UC540 SIP trunk "allowed IP addresses" limitation?

Testing out a UC540 and I'm using Callcentric.  Working great, but the feature under the SIP trunk configuring/Advanced that allows you to set the IP addresses of machines allowed to connect to the UC540 only supports a maximum of 100 entries.  Callcentric wants 204.11.192.0/24 or 204.11.192.0-204.11.192-255 unblocked.
The UC540 doesn't seem to support inputting a range in any notation I'm aware of, maybe I'm missing something?
I contacted Callcentric and asked if they could be more specific, LOL, but they said they change things on their end from time to time.
Anyone else figure out a way around this?

Okay, follow-up to the question.
How bad an idea is it to leave this security feature disabled?
I have a bit of experience in this area, I use an Asterisk (FreePBX distro) for my work phone system, and Callcentric is my SIP trunker.  Before I limited incoming connections to my FreePBX server to the Callcentric range above, there would be occasions (maybe once every 2-3 weeks) where someone (probably up to no good) found my open port(s) and all my phones would start ringing simultaneously.  It lasted for about 20-30 seconds, and then would stop.
My assumption was that someone was looking for security vulnerabilities so they could use my Asterisk box to place their LD (maybe even overseas) calls.  They never got anywhere, though, as the Asterisk was fully patched, and my passwords for my handsets were strong.
I wonder if the Cisco UC540 will suffer the same sort of result, or worse, or maybe even not so much.
Obviously I can simply select another trunker.  The thing I like about Callcentric are the rates and also the dashboard that allows me to make a lot of adjustments (like forwarding calls to a cell phone number, and the ability to configure faxing reception, too).  So if anyone has a suggestion for a supported SIP trunker that supports the same types of features, I'd be open to that, as well.
BTW, I did ask Callcentric one more time, with a link to this topic, whether they can restrict that range to better than 0-255, but they declined.  I see their point, just wish there was another simple way around this.

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     isdn switch-type basic-net3
     isdn point-to-point-setup
     isdn incoming-voice voice
     isdn sending-complete
     isdn static-tei 0
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     shutdown
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     isdn point-to-point-setup
     isdn incoming-voice voice
     isdn sending-complete
     isdn static-tei 0
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     ssid cisco-data
     ssid cisco-voice
     speed basic-1.0 basic-2.0 basic-5.5 6.0 9.0 basic-11.0 12.0 18.0 24.0 36.0 48.0 54.0
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     antenna receive right
     antenna transmit right
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     bridge-group 100 subscriber-loop-control
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    ip nat inside source list 1 interface FastEthernet0/0 overload
    ip route 0.0.0.0 0.0.0.0 192.168.10.1
    ip route 10.1.10.1 255.255.255.255 Integrated-Service-Engine0/0
    access-list 1 remark SDM_ACL Category=2
    access-list 1 permit 10.1.1.0 0.0.0.255
    access-list 1 permit 192.168.10.0 0.0.0.255
    access-list 1 permit 10.1.10.0 0.0.0.3
    access-list 2 remark CCA_SIP_SOURCE_GROUP_ACL_INTERNAL
    access-list 2 remark SDM_ACL Category=1
    access-list 2 permit 192.168.10.2
    access-list 2 permit 10.1.10.0 0.0.0.3
    access-list 2 permit 192.168.10.0 0.0.0.255
    access-list 2 permit 10.1.1.0 0.0.0.255
    access-list 3 remark CCA_SIP_SOURCE_GROUP_ACL_EXTERNAL
    access-list 3 remark SDM_ACL Category=1
    access-list 3 permit 212.147.47.216
    access-list 3 deny   any
    access-list 100 remark auto generated by SDM firewall configuration
    access-list 100 remark SDM_ACL Category=1
    access-list 100 deny   ip 192.168.10.0 0.0.0.255 any
    access-list 100 deny   ip host 255.255.255.255 any
    access-list 100 deny   ip 127.0.0.0 0.255.255.255 any
    access-list 100 permit ip any any
    access-list 101 remark auto generated by SDM firewall configuration##NO_ACES_8##
    access-list 101 remark SDM_ACL Category=1
    access-list 101 permit tcp 10.1.1.0 0.0.0.255 eq 2000 any
    access-list 101 permit udp 10.1.1.0 0.0.0.255 eq 2000 any
    access-list 101 deny   ip 10.1.1.0 0.0.0.255 any
    access-list 101 deny   ip 192.168.10.0 0.0.0.255 any
    access-list 101 deny   ip 192.168.1.0 0.0.0.255 any
    access-list 101 deny   ip host 255.255.255.255 any
    access-list 101 deny   ip 127.0.0.0 0.255.255.255 any
    access-list 101 permit ip any any
    access-list 102 remark auto generated by SDM firewall configuration##NO_ACES_6##
    access-list 102 remark SDM_ACL Category=1
    access-list 102 deny   ip 10.1.10.0 0.0.0.3 any
    access-list 102 deny   ip 10.1.1.0 0.0.0.255 any
    access-list 102 deny   ip 192.168.1.0 0.0.0.255 any
    access-list 102 deny   ip host 255.255.255.255 any
    access-list 102 deny   ip 127.0.0.0 0.255.255.255 any
    access-list 102 permit ip any any
    access-list 103 remark auto generated by SDM firewall configuration##NO_ACES_8##
    access-list 103 remark SDM_ACL Category=1
    access-list 103 permit tcp 10.1.10.0 0.0.0.3 any eq 2000
    access-list 103 permit udp 10.1.10.0 0.0.0.3 any eq 2000
    access-list 103 deny   ip 10.1.10.0 0.0.0.3 any
    access-list 103 deny   ip 192.168.10.0 0.0.0.255 any
    access-list 103 deny   ip 192.168.1.0 0.0.0.255 any
    access-list 103 deny   ip host 255.255.255.255 any
    access-list 103 deny   ip 127.0.0.0 0.255.255.255 any
    access-list 103 permit ip any any
    access-list 104 remark auto generated by SDM firewall configuration##NO_ACES_14##
    access-list 104 remark SDM_ACL Category=1
    access-list 104 deny   ip 10.1.10.0 0.0.0.3 any
    access-list 104 deny   ip 10.1.1.0 0.0.0.255 any
    access-list 104 permit ip any any
    access-list 104 permit udp host 8.8.8.8 eq domain any
    access-list 104 permit icmp any any echo-reply
    access-list 104 permit icmp any any time-exceeded
    access-list 104 permit icmp any any unreachable
    access-list 104 deny   ip 10.0.0.0 0.255.255.255 any
    access-list 104 deny   ip 172.16.0.0 0.15.255.255 any
    access-list 104 deny   ip 192.168.0.0 0.0.255.255 any
    access-list 104 deny   ip 127.0.0.0 0.255.255.255 any
    access-list 104 deny   ip host 255.255.255.255 any
    access-list 104 deny   ip host 0.0.0.0 any
    access-list 104 deny   ip any any
    control-plane
    bridge 1 route ip
    bridge 100 route ip
    voice-port 0/0/0
     cptone CH
     station-id name FAX
     station-id number 99
     caller-id enable
    voice-port 0/0/1
     cptone CH
     shutdown
     caller-id enable
    voice-port 0/0/2
     cptone CH
     shutdown
     caller-id enable
    voice-port 0/0/3
     cptone CH
     shutdown
     caller-id enable
    voice-port 0/1/0
     compand-type a-law
     cptone CH
     bearer-cap Speech
    voice-port 0/1/1
     compand-type a-law
     cptone CH
     bearer-cap Speech
    voice-port 0/4/0
     auto-cut-through
     signal immediate
     input gain auto-control -15
     description Music On Hold Port
    sccp local Loopback0
    sccp ccm 10.1.1.1 identifier 1 version 4.0
    sccp
    sccp ccm group 1
     associate ccm 1 priority 1
     associate profile 2 register mtpa4934c6ee4e0
    dspfarm profile 2 transcode
     description CCA transcoding for SIP Trunk VTX
     codec g711ulaw
     codec g711alaw
     codec g729ar8
     codec g729abr8
     maximum sessions 10
     associate application SCCP
    dial-peer cor custom
     name internal
     name local
     name local-plus
     name international
     name national
     name national-plus
     name emergency
     name toll-free
    dial-peer cor list call-internal
     member internal
    dial-peer cor list call-local
     member local
    dial-peer cor list call-local-plus
     member local-plus
    dial-peer cor list call-national
     member national
    dial-peer cor list call-national-plus
     member national-plus
    dial-peer cor list call-international
     member international
    dial-peer cor list call-emergency
     member emergency
    dial-peer cor list call-toll-free
     member toll-free
    dial-peer cor list user-internal
     member internal
     member emergency
    dial-peer cor list user-local
     member internal
     member local
     member emergency
     member toll-free
    dial-peer cor list user-local-plus
     member internal
     member local
     member local-plus
     member emergency
     member toll-free
    dial-peer cor list user-national
     member internal
     member local
     member local-plus
     member national
     member emergency
     member toll-free
    dial-peer cor list user-national-plus
     member internal
     member local
     member local-plus
     member national
     member national-plus
     member emergency
     member toll-free
    dial-peer cor list user-international
     member internal
     member local
     member local-plus
     member international
     member national
     member national-plus
     member emergency
     member toll-free
    dial-peer voice 1 pots
     destination-pattern 99
     port 0/0/0
     no sip-register
    dial-peer voice 2 pots
     port 0/0/1
     no sip-register
    dial-peer voice 3 pots
     port 0/0/2
     no sip-register
    dial-peer voice 4 pots
     port 0/0/3
     no sip-register
    dial-peer voice 5 pots
     description ** MOH Port **
     destination-pattern ABC
     port 0/4/0
     no sip-register
    dial-peer voice 6 pots
     description tcatch all dial peer for BRI/PRIv
     translation-profile incoming nondialable
     incoming called-number .%
     direct-inward-dial
    dial-peer voice 50 pots
     description ** incoming dial peer **
     incoming called-number ^AAAA$
     direct-inward-dial
     port 0/1/0
    dial-peer voice 51 pots
     description ** incoming dial peer **
     incoming called-number ^AAAA$
     direct-inward-dial
     port 0/1/1
    dial-peer voice 2000 voip
     description ** cue voicemail pilot number **
     translation-profile outgoing XFER_TO_VM_PROFILE
     destination-pattern 98
     b2bua
     session protocol sipv2
     session target ipv4:10.1.10.1
     voice-class sip outbound-proxy ipv4:10.1.10.1
     dtmf-relay rtp-nte
     codec g711ulaw
     no vad
    dial-peer voice 2001 voip
     description ** cue auto attendant number **
     translation-profile outgoing PSTN_CallForwarding
     destination-pattern 97
     b2bua
     session protocol sipv2
     session target ipv4:10.1.10.1
     voice-class sip outbound-proxy ipv4:10.1.10.1
     dtmf-relay rtp-nte
     codec g711ulaw
     no vad
    dial-peer voice 2012 voip
     description ** cue prompt manager number **
     translation-profile outgoing PSTN_CallForwarding
     destination-pattern 96
     b2bua
     session protocol sipv2
     session target ipv4:10.1.10.1
     voice-class sip outbound-proxy ipv4:10.1.10.1
     dtmf-relay rtp-nte
     codec g711ulaw
     no vad
    dial-peer voice 1000 voip
     permission term
     description ** Incoming call from SIP trunk (VTX) **
     session protocol sipv2
     session target sip-server
     incoming called-number .%
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     fax rate 14400
     fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1001 voip
     corlist outgoing call-local
     description ** star code to SIP trunk (VTX) **
     destination-pattern *..
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     fax rate 14400
     fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1003 voip
     description ** Passthrough Inbound Calls for PSTN from CUE **
     translation-profile incoming SIP_Passthrough
     b2bua
     session protocol sipv2
     session target ipv4:10.1.10.1
     incoming called-number ABCDT
     dtmf-relay rtp-nte
     codec g711ulaw
     no vad
    dial-peer voice 1005 voip
     description ** Passthrough Inbound Calls for MWI from CUE **
     b2bua
     session protocol sipv2
     session target ipv4:10.1.10.1
     incoming called-number A80T
     dtmf-relay rtp-nte
     codec g711ulaw
     no vad
    dial-peer voice 1009 voip
     description ** Passthrough Inbound Calls for Internal Extensions from CUE **
     b2bua
     session protocol sipv2
     session target ipv4:10.1.10.1
     incoming called-number ^..$
     dtmf-relay rtp-nte
     codec g711ulaw
     no vad
    dial-peer voice 1033 voip
     corlist outgoing call-local
     description **CCA*Switzerland*Short Code Services**
     translation-profile outgoing PSTN_Outgoing
     preference 1
     destination-pattern 0187
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1042 voip
     corlist outgoing call-emergency
     description **CCA*Switzerland*Ambulance / Poisioning**
     translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
     preference 1
     destination-pattern 0014[45]
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1041 voip
     corlist outgoing call-emergency
     description **CCA*Switzerland*REGA Air Rescue**
     translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
     preference 1
     destination-pattern 00333333333
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1025 voip
     corlist outgoing call-national
     description **CCA*Switzerland*National Destination Numbers**
     translation-profile outgoing PSTN_Outgoing
     preference 1
     destination-pattern 00[789]1.......
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1020 voip
     corlist outgoing call-national
     description **CCA*Switzerland*Regional Announcement VM**
     translation-profile outgoing PSTN_Outgoing
     preference 1
     destination-pattern 01600
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1040 voip
     corlist outgoing call-emergency
     description **CCA*Switzerland*REGA Air Rescue**
     translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
     preference 1
     destination-pattern 000333333333
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1043 voip
     corlist outgoing call-emergency
     description **CCA*Switzerland*Ambulance / Poisioning**
     translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
     preference 1
     destination-pattern 014[45]
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1035 voip
     corlist outgoing call-national
     description **CCA*Switzerland*Mobile Numbers**
     translation-profile outgoing PSTN_Outgoing
     preference 1
     destination-pattern 007[46789].......
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1024 voip
     corlist outgoing call-national-plus
     description **CCA*Switzerland*Personal Numbering**
     translation-profile outgoing PSTN_Outgoing
     preference 1
     destination-pattern 00878......
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1029 voip
     corlist outgoing call-national
     description **CCA*Switzerland*Voicemail Access**
     translation-profile outgoing PSTN_Outgoing
     preference 1
     destination-pattern 00860.........
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1036 voip
     corlist outgoing call-national
     description **CCA*Switzerland*VPN Access**
     translation-profile outgoing PSTN_Outgoing
     preference 1
     destination-pattern 00869.............
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1027 voip
     corlist outgoing call-national-plus
     description **CCA*Switzerland*Premium Rate (Business)**
     translation-profile outgoing PSTN_Outgoing
     preference 1
     destination-pattern 00900......
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1026 voip
     corlist outgoing call-national
     description **CCA*Switzerland*Test Numbers**
     translation-profile outgoing PSTN_Outgoing
     preference 1
     destination-pattern 00868T
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1034 voip
     corlist outgoing call-national-plus
     description **CCA*Switzerland*Shared Cost numbers**
     translation-profile outgoing PSTN_Outgoing
     preference 1
     destination-pattern 0084[0248]......
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1038 voip
     corlist outgoing call-emergency
     description **CCA*Switzerland*Emergency**
     translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
     preference 1
     destination-pattern 0011[278]
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1037 voip
     corlist outgoing call-toll-free
     description **CCA*Switzerland*Toll Free Numbers**
     translation-profile outgoing PSTN_Outgoing
     preference 1
     destination-pattern 00800......
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1039 voip
     corlist outgoing call-emergency
     description **CCA*Switzerland*Emergency**
     translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
     preference 1
     destination-pattern 011[278]
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1032 voip
     corlist outgoing call-national
     description **CCA*Switzerland*National Destination Numbers**
     translation-profile outgoing PSTN_Outgoing
     preference 1
     destination-pattern 00[23456]........
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1023 voip
     corlist outgoing call-international
     description **CCA*Switzerland*International Calls**
     translation-profile outgoing PSTN_Outgoing
     preference 1
     destination-pattern 000T
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1031 voip
     description **CCA*Switzerland*Premium Rate (Social)**
     translation-profile outgoing PSTN_Outgoing
     preference 1
     destination-pattern 0090[16]......
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1030 voip
     corlist outgoing call-national
     description **CCA*Switzerland*Short Code**
     translation-profile outgoing PSTN_Outgoing
     preference 1
     destination-pattern 014[0357]
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1045 voip
     corlist outgoing call-emergency
     description **CCA*Switzerland*REGA/Glaciers Air Rescue**
     translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
     preference 1
     destination-pattern 0141[45]
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1028 voip
     corlist outgoing call-national-plus
     description **CCA*Switzerland*Directory Enquiries**
     translation-profile outgoing PSTN_Outgoing
     preference 1
     destination-pattern 018[15].
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1021 voip
     corlist outgoing call-national
     description **CCA*Switzerland*Short Code**
     translation-profile outgoing PSTN_Outgoing
     preference 1
     destination-pattern 011[45].
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1022 voip
     corlist outgoing call-national
     description **CCA*Switzerland*Short Code Services**
     translation-profile outgoing PSTN_Outgoing
     preference 1
     destination-pattern 01[67].
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1044 voip
     corlist outgoing call-emergency
     description **CCA*Switzerland*REGA/Glaciers Air Rescue**
     translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
     preference 1
     destination-pattern 00141[45]
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 2002 voip
     description ** cue voicemail PSTN number **
     translation-profile outgoing VM_Profile
     destination-pattern xxx$
     b2bua
     session protocol sipv2
     session target ipv4:10.1.10.1
     voice-class sip outbound-proxy ipv4:10.1.10.1
     dtmf-relay rtp-nte
     codec g711ulaw
     no vad
    dial-peer voice 2003 voip
     description ** cue auto attendant PSTN number **
     translation-profile outgoing AA_Profile
     destination-pattern xxx$
     b2bua
     session protocol sipv2
     session target ipv4:10.1.10.1
     voice-class sip outbound-proxy ipv4:10.1.10.1
     dtmf-relay rtp-nte
     codec g711ulaw
     no vad
    dial-peer voice 1110 pots
     preference 9
     destination-pattern xxx
     port 0/0/0
     no sip-register
    dial-peer voice 3006 voip
     description SIP
     translation-profile incoming SIP_Called_9
     session protocol sipv2
     session target sip-server
     incoming called-number xxx.
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    no dial-peer outbound status-check pots
    sip-ua
     keepalive target dns:site1.365873.trk.ipvoip.ch
     authentication username xxx password 7 xxx
     no remote-party-id
     retry invite 2
     retry register 10
     timers connect 100
     timers keepalive active 100
     registrar dns:site1.365873.trk.ipvoip.ch expires 3600
     sip-server dns:site1.365873.trk.ipvoip.ch
     host-registrar
    telephony-service
     sdspfarm units 5
     sdspfarm transcode sessions 10
     sdspfarm tag 2 mtpa4934c6ee4e0
     video
     fxo hook-flash
     max-ephones 40
     max-dn 300
     ip source-address 10.1.1.1 port 2000
     auto assign 1 to 1 type bri
     calling-number initiator
     service phone videoCapability 1
     service phone ehookenable 1
     service phone ehookEnable 1
     service dnis overlay
     service dnis dir-lookup
     service dss
     timeouts interdigit 5
     system message SwissT.Net
     url services http://10.1.10.1/voiceview/common/login.do
     url authentication http://10.1.10.1/voiceview/authentication/authenticate.do
     cnf-file location flash:
     cnf-file perphone
     user-locale U4 load CME-locale-de_DE-German-8.1.2.2.tar
     network-locale U4
     load 521G-524G cp524g-8-1-17
     load 525G spa525g-7-5-4
     load 501G spa50x-30x-7-5-2b
     load 502G spa50x-30x-7-5-2b
     load 504G spa50x-30x-7-5-2b
     load 508G spa50x-30x-7-5-2b
     load 509G spa50x-30x-7-5-2b
     load 525G2 spa525g-7-5-4
     load 301 spa50x-30x-7-5-2b
     load 303 spa50x-30x-7-5-2b
     time-zone 23
     time-format 24
     date-format dd-mm-yy
     keepalive 30 auxiliary 4
     voicemail 98
     max-conferences 8 gain -6
     call-forward pattern .T
     call-forward system redirecting-expanded
     hunt-group logout HLog
     moh flash:/media/music-on-hold.au
     multicast moh 239.10.16.16 port 2000
     web admin system name cisco secret 5 xxx
     dn-webedit
     time-webedit
     transfer-system full-consult dss
     transfer-pattern .T
     transfer-pattern 0.T
     transfer-pattern 6.. blind
     secondary-dialtone 0
     night-service day Sun 17:00 09:00
     night-service day Mon 17:00 09:00
     night-service day Tue 17:00 09:00
     night-service day Wed 17:00 09:00
     night-service day Thu 17:00 09:00
     night-service day Fri 17:00 09:00
     night-service day Sat 17:00 09:00
     fac standard
     create cnf-files version-stamp Jan 01 2002 00:00:00
    ephone-template  1
     url services 1 http://10.1.10.1/voiceview/common/login.do VoiceviewExpress
     service phone webAccess 0
     softkeys remote-in-use  Newcall
     softkeys idle  Redial Pickup Mobility Newcall Cfwdall Gpickup Dnd Login
     softkeys seized  Cfwdall Endcall Redial Pickup Gpickup Callback
     softkeys connected  Hold Endcall Trnsfer Mobility TrnsfVM Confrn Acct Park
     button-layout 7931 2
    ephone-template  15
     url services 1 http://10.1.10.1/voiceview/common/login.do VoiceviewExpress
     softkeys remote-in-use  Newcall
     softkeys idle  Redial Newcall Mobility Cfwdall Pickup Gpickup Dnd Login
     softkeys seized  Cfwdall Endcall Redial Pickup Gpickup Callback
     softkeys connected  Hold Endcall Mobility Trnsfer TrnsfVM Confrn Acct Park
     button-layout 7931 2
    ephone-template  16
     url services 1 http://10.1.10.1/voiceview/common/login.do VoiceviewExpress
     softkeys remote-in-use  Newcall
     softkeys idle  Redial Newcall Mobility Cfwdall Pickup Gpickup Dnd Login
     softkeys seized  Cfwdall Endcall Redial Pickup Gpickup Callback
     softkeys connected  Hold Endcall Mobility Trnsfer TrnsfVM Confrn Acct Park
    ephone-template  17
     url services 1 http://10.1.10.1/voiceview/common/login.do VoiceviewExpress
     softkeys remote-in-use  CBarge Newcall
     softkeys idle  Redial Newcall Mobility Cfwdall Pickup Gpickup Dnd Login
     softkeys seized  Cfwdall Endcall Redial Pickup Gpickup Callback
     softkeys connected  Hold Endcall Mobility Trnsfer TrnsfVM Confrn Acct Park
    ephone-template  18
     url services 1 http://10.1.10.1/voiceview/common/login.do VoiceviewExpress
     softkeys remote-in-use  CBarge Newcall
     softkeys idle  Redial Newcall Mobility Cfwdall Pickup Gpickup Dnd Login
     softkeys seized  Cfwdall Endcall Redial Pickup Gpickup Callback
     softkeys connected  Hold Endcall Mobility Trnsfer TrnsfVM Confrn Acct Park
     button-layout 7931 2
    ephone-dn  9
     number BCD no-reg primary
     description MoH
     moh out-call ABC
    ephone-dn  292
     number xxx
     description SIP Main Number registration
     preference 10
    ephone-dn  293  dual-line
     number 90 secondary xxx no-reg both
     label Zentrale
     description 90
     name Zentrale
     call-forward busy 98
     call-forward noan 98 timeout 20
    ephone-dn  294  dual-line
     number 94 secondary xxx no-reg both
     label LL
     description Lehrling Lehrnende
     name Lehrling Lehrnende
     mobility
     snr xxx delay 1 timeout 30 cfwd-noan 98
     snr ring-stop
     call-forward busy 98
     call-forward noan 98 timeout 20
    ephone-dn  295  dual-line
     number 93 secondary xxx no-reg both
     label CM
     description
     name
     snr xxx delay 1 timeout 30 cfwd-noan 98
     snr ring-stop
     call-forward busy 98
     call-forward noan 98 timeout 10
    ephone-dn  296  dual-line
     number 92 secondary xxx no-reg both
     label EE
     description
     name
     mobility
     call-forward busy 98
     call-forward noan 98 timeout 20
    ephone-dn  297  dual-line
     number 91 secondary xxx no-reg both
     label RS
     description
     name
     mobility
     snr xxx delay 1 timeout 30 cfwd-noan 98
     snr ring-stop
     call-forward busy 98
     call-forward noan 98 timeout 10
    ephone-dn  298
     number 6.. no-reg primary
     description ***CCA XFER TO VM EXTENSION***
     call-forward all 98
    ephone-dn  299
     number A801.. no-reg primary
     mwi off
    ephone-dn  300
     number A800.. no-reg primary
     mwi on
    ephone  1
     device-security-mode none
     mac-address A44C.11A0.B648
     ephone-template 1
     max-calls-per-button 2
     username "xxx" password xxx
     type 525G2
     button  1:296 2:293 3m297 4m295
     button  5m294
    ephone  2
     device-security-mode none
     mac-address A44C.11A0.B566
     ephone-template 1
     max-calls-per-button 2
     username "xxx" password xxx
     type 525G2
     button  1:297 2:293 3m296 4m295
     button  5m294
    ephone  3
     device-security-mode none
     mac-address A44C.11A0.B5C4
     ephone-template 1
     max-calls-per-button 2
     username "xxx" password xxx
     type 525G2
     button  1:295 2:293 3m297 4m296
     button  5m294
    ephone  4
     device-security-mode none
     mac-address A44C.11A0.B67A
     ephone-template 1
     max-calls-per-button 2
     username "xxx" password xxx
     type 525G2
     button  1:294 2:293 3m297 4m296
     button  5m295
    alias exec cca_voice_mode PBX
    alias exec cca_vm_notification schedule from_time=00 to_time=24
    alias exec clid-ALL_BRI ;1:0-4;1:0-9;1:0-9;1:1-9
    alias exec clid-SIP ;1:1-9;1:1-9;1:1-9
    banner login ^CCisco Configuration Assistant. Version: 3.2 (3). Fri Jul 04 13:18:33 CEST 2014^C
    line con 0
     no modem enable
    line aux 0
    line 2
     no activation-character
     no exec
     transport preferred none
     transport input all
    line vty 0 4
     transport preferred none
     transport input all
    line vty 5 100
     transport preferred none
     transport input all
    ntp master
    ntp server 91.240.0.5 prefer
    en

    Hi Patrick
    I am working on this one as well. I have a UC560 with SIP Trunk provider Les.NET.
    It was working fine until a few weeks ago when something changed on the provider end and broke it. My hunch it is something to do with the SIP REFER.
    http://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unified-communications-manager-express/91535-cme-sip-trunking-config.html
    Here is an excerpt from the above page:
    Call Transfer
    When a call comes in on an SIP trunk to an SCCP Phone or CUE AutoAttendant (AA) and is transferred, the CME by default will send a SIP REFER message to the SP proxy. Most SP Proxy Servers do not support the REFER method. This needs to be configured in order to force the CME to hairpin the call:
    Router(config)#voice service voip
    Router(conf-voi-serv)#no supplementary-service sip refer
    Figure 3 shows the behavior of the CME system with the REFER method disabled.

  • Unable to place call on calls on hold - SIP Trunk from CUCM to CUBE and from CUBE to ISTP

    Hi Cisco Community,
    I have a SIP Trunk setup between the CUCM and CUBE and another SIP Dial Peers from the CUBE to the ITSP. All incoming/outgoing calls, DTMF-Relay works well except one thing which is the ability to hold the call.
    On the SIP Trunk from the CUCM to the CUBE, I did not select “MTP” because when I do so, I am forced to select my preferred MTP codec which when selected G.729/G.729a, all my outgoing calls goes out using G.729r8. This codec works well for most of the calls until the ITSP replies back with G.729br8. When this condition occur, my call simply fails (this is very intermittent and only some random numbers).
    That said, I have some issues with DTMF Relay when I select MTP on the SIP Trunk. DTMF Relay only works if the call is G.729r8 all the way from the CUCM to the ITSP. If the ITSP replies back with G.729br8, the call might established but will simply be “voice-only”.
    The current setup is no MTP is selected and everything is working perfectly. I am happy with that until I place a call on hold, which when I do so, the call immediately terminate. Could you please help me understand why?
    I have all media resources configured such as G.729r8 MTP, G.729br8 MTP, G.711u MTP, Transcoding with all codecs, etc. All MRG and MRGL are configured on all devices and SIP Trunks.
    Below is an example of a call that is connected with the current setup:
    Note:
    IP: 10.18.81.2 (CUBE)
    IP: 10.18.81.11 (CUCM SUB)
    IP: 10.111.111.254 (ITSP SBC)
    PM-HO-VG-01#
    PM-HO-VG-01#
    Nov 30 11:44:29.938: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received: 
    INVITE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72063a5aba5d
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>
    Date: Sun, 30 Nov 2014 11:44:29 GMT
    Call-ID: [email protected]
    Supported: timer,resource-priority,replaces
    Min-SE:  1800
    User-Agent: Cisco-CUCM9.1
    Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
    CSeq: 101 INVITE
    Expires: 180
    Allow-Events: presence, kpml
    Supported: X-cisco-srtp-fallback,X-cisco-original-called
    Call-Info: <sip:10.18.81.11:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
    Cisco-Guid: 1020645888-0000065536-0000124117-0189862410
    Session-Expires:  1800
    Contact: <sip:[email protected]:5060>
    Max-Forwards: 70
    Content-Length: 0
    Nov 30 11:44:29.942: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72063a5aba5d
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>
    Date: Sun, 30 Nov 2014 11:44:29 GMT
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Allow-Events: telephone-event
    Server: Cisco-SIPGateway/IOS-15.2.4.M5
    Content-Length: 0
    Nov 30 11:44:29.946: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    INVITE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>
    Date: Sun, 30 Nov 2014 11:44:29 GMT
    Call-ID: [email protected]
    Supported: 100rel,timer,resource-priority,replaces,sdp-anat
    Min-SE:  1800
    Cisco-Guid: 1020645888-0000065536-0000124117-0189862410
    User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    CSeq: 101 INVITE
    Timestamp: 1417347869
    Contact: <sip:[email protected]:5060>
    Call-Info: <sip:10.18.81.2:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
    Expires: 180
    Allow-Events: telephone-event
    Max-Forwards: 69
    Session-Expires:  1800
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 301
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 7676 6958 IN IP4 10.18.81.2
    s=SIP Call
    c=I
    PM-HO-VG-01#N IP4 10.18.81.2
    t=0 0
    m=audio 22256 RTP/AVP 18 0 8 101
    c=IN IP4 10.18.81.2
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=yes
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    Nov 30 11:44:29.950: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Received: 
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Timestamp: 1417347869
    Nov 30 11:44:30.658: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Received: 
    SIP/2.0 180 Session Progress
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>;tag=71913148-1417348035284
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Timestamp: 1417347869
    Supported: 
    Contact: <sip:[email protected]:5060;transport=udp>
    Session: Media
    Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
    X-BroadWorks-Correlation-Info: bbf94839-a234-4237-95e6-a7037322f0f4
    Content-Type: application/sdp
    Content-Length: 355
    v=0
    o=BroadWorks 316169737 1 IN IP4 10.111.111.254
    s=-
    c=IN IP4 10.111.111.254
    t=0 0
    m=audio 20074 RTP/AVP 18 101 100
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=rtpmap:100 X-NSE/8000
    a=fmtp:100 200-202
    a=X-sqn:0
    a=X-cap: 1 audio RTP/AVP 100
    a=X-cpar: a=rtpmap:100 X-NSE/8000
    a=X-cpar: a=fmtp:100 200-202
    a=X-cap: 2 image udptl t38
    Nov 30 11:44:30.662: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    SIP/2.0 183 Session Progress
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72063a5aba5d
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>;tag=3C365010-1E42
    Date: Sun, 30 Nov 2014 11:44:29 GMT
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    Allow-Events: telephone-event
    Contact: <sip:[email protected]:5060>
    Supported: sdp-anat
    Server: Cisco-SIPGateway/IOS-15.2.4.M5
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 289
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 7965 2747 IN IP4 10.18.81.2
    s=SIP Call
    c=IN IP4 10.18.81.2
    t=0 0
    m=audio 22350 RTP/AVP 18 101 19
    c=IN IP4 10.18.81.2
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=yes
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=rtpmap:19 CN/8000
    a=ptime:20
    Nov 30 11:44:31.226: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Received: 
    SIP/2.0 180 Session Progress
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>;tag=71913148-1417348035284
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Timestamp: 1417347869
    Supported: 
    Contact: <sip:[email protected]:5060;transport=udp>
    Session: Media
    Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
    X-BroadWorks-Correlation-Info: bbf9
    PM-HO-VG-01#4839-a234-4237-95e6-a7037322f0f4
    Content-Type: application/sdp
    Content-Length: 355
    v=0
    o=BroadWorks 316169737 1 IN IP4 10.111.111.254
    s=-
    c=IN IP4 10.111.111.254
    t=0 0
    m=audio 20074 RTP/AVP 18 101 100
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=rtpmap:100 X-NSE/8000
    a=fmtp:100 200-202
    a=X-sqn:0
    a=X-cap: 1 audio RTP/AVP 100
    a=X-cpar: a=rtpmap:100 X-NSE/8000
    a=X-cpar: a=fmtp:100 200-202
    a=X-cap: 2 image udptl t38
    Nov 30 11:44:31.630: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Received: 
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>;tag=71913148-1417348035284
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Timestamp: 1417347869
    Supported: 
    Contact: <sip:[email protected]:5060;transport=udp>
    Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
    Accept: application/media_control+xml,application/sdp,application/xml
    X-BroadWorks-Correlation-Info: bbf94839-a234-4237-95e6-a7037322f0f4
    Content-Type: application/sdp
    Content-Length: 355
    v=0
    o=BroadWorks 316169737 1 IN IP4 10.111.111.254
    s=-
    c=IN IP4 10.111.111.254
    t=0 0
    m=audio 20074 RTP/AVP 18 101 100
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=rtpmap:100 X-NSE/8000
    a=fmtp:100 200-202
    a=X-sqn:0
    a=X-cap: 1 audio RTP/AVP 100
    a=X-cpar: a=rtpmap:100 X-NSE/8000
    a=X-cpar: a=fmtp:100 200-202
    a=X-cap: 2 image udptl t38
    Nov 30 11:44:31.630: //64511/3CD5D2000001/SIP/Call/sipSPICallInfo: 
    The Call Setup Information is:
    Call Control Block (CCB) : 0x0x3D7B1458
    State of The Call        : STATE_ACTIVE
    TCP Sockets Used         : NO
    Calling Number           : 27218091323
    Called Number            : 0862000000
    Source IP Address (Sig  ): 10.18.81.2
    Destn SIP Req Addr:Port  : 10.111.111.254:5060
    Destn SIP Resp Addr:Port : 10.111.111.254:5060
    Destination Name         : 10.111.111.254
    Nov 30 11:44:31.630: //64511/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo: 
    Number of Media Streams: 1
    Media Stream             : 1
    Negotiated Codec         : g729br8
    Negotiated Codec Bytes   : 20
    Nego. Codec payload      : 18 (tx), 18 (rx)
    Negotiated Dtmf-relay    : 6
    Dtmf-relay Payload       : 101 (tx), 101 (rx)
    Source IP Address (Media): 10.18.81.2
    Source IP Port    (Media): 22256
    Destn  IP Address (Media): 10.111.111.254
    Destn  IP Port    (Media): 20074
    Orig Destn IP Address:Port (Media): [ - ]:0
    Nov 30 11:44:31.630: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    ACK sip:[email protected]:5060;transport=udp SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC81D00
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>;tag=71913148-1417348035284
    Date: Sun, 30 Nov 2014 11:44:29 GMT
    Call-ID: [email protected]
    Max-Forwards: 70
    CSeq: 101 ACK
    Allow-Events: telephone-event
    Content-Length: 0
    Nov 30 11:44:31.634: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72063a5aba5d
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>;tag=3C365010-1E42
    Date: Sun, 30 Nov 2014 11:44:29 GMT
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    Allow-Events: telephone-event
    Contact: <sip:[email protected]:5060>
    Supported: replaces
    Supported: sdp-anat
    Server: Cisco-SIPGateway/IOS-15.2.4.M5
    Supported: timer
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 289
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 7965 2747 IN IP4 10.18.81.2
    s=SIP Call
    c=IN IP4 10.18.81.2
    t=0 0
    m=audio 22350 RTP/AVP 18 101 19
    c=IN IP4 10.18.81.2
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=yes
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=rtpmap:19 CN/8000
    a=ptime:20
    Nov 30 11:44:31.726: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received: 
    ACK sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72075e3a02c1
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>;tag=3C365010-1E42
    Date: Sun, 30 Nov 2014 11:44:29 GMT
    Call-ID: [email protected]
    Max-Forwards: 70
    CSeq: 101 ACK
    Allow-Events: presence, kpml
    Content-Type: application/sdp
    Content-Length: 236
    v=0
    o=CiscoSystemsCCM-SIP 9082578 1 IN IP4 10.18.81.11
    s=SIP Call
    c=IN IP4 10.18.80.40
    b=TIAS:8000
    b=AS:8
    t=0 0
    m=audio 21928 RTP/AVP 18 101
    a=rtpmap:18 G729/8000
    a=ptime:20
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    Nov 30 11:44:31.730: //64510/3CD5D2000001/SIP/Call/sipSPICallInfo: 
    The Call Setup Information is:
    Call Control Block (CCB) : 0x0x3D816D70
    State of The Call        : STATE_ACTIVE
    TCP Sockets Used         : NO
    Calling Number           : 0218091323
    Called Number            : 0862000000
    Source IP Address (Sig  ): 10.18.81.2
    Destn SIP Req Addr:Port  : 10.18.81.11:5060
    Destn SIP Resp Addr:Port : 10.18.81.11:5060
    Destination Name         : 10.18.81.11
    Nov 30 11:44:31.730: //64510/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo: 
    Number of Media Streams: 1
    Media Stream             : 1
    Negotiated Codec         : g729br8
    Negotiated Codec Bytes   : 20
    Nego. Codec payload      : 18 (tx), 18 (rx)
    Negotiated Dtmf-relay    : 6
    Dtmf-relay Payload       : 101 (tx), 101 (rx)
    Source IP Address (Media): 10.18.81.2
    Source IP Port    (Media): 22350
    Destn  IP Address (Media): 10.18.80.40
    Destn  IP Port    (Media): 21928
    Orig Destn IP Address:Port (Media): [ - ]:0
    Nov 30 11:44:31.730: //64510/3CD5D2000001/SIP/Call/sipSPICallInfo: 
    The Call Setup Information is:
    Call Control Block (CCB) : 0x0x3D816D70
    State of The Call        : STATE_ACTIVE
    TCP Sockets Used         : NO
    Calling Number           : 0218091323
    Called Number            : 0862000000
    Source IP Address (Sig  ): 10.18.81.2
    Destn SIP Req Addr:Port  : 10.18.81.11:5060
    Destn SIP Resp Addr:Port : 10.18.81.11:5060
    Destination Name         : 10.18.81.11
    PM-HO-VG-01#
    Nov 30 11:44:31.730: //64510/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo: 
    Number of Media Streams: 1
    Media Stream             : 1
    Negotiated Codec         : g729br8
    Negotiated Codec Bytes   : 20
    Nego. Codec payload      : 18 (tx), 18 (rx)
    Negotiated Dtmf-relay    : 6
    Dtmf-relay Payload       : 101 (tx), 101 (rx)
    Source IP Address (Media): 10.18.81.2
    Source IP Port    (Media): 22350
    Destn  IP Address (Media): 10.18.80.40
    Destn  IP Port    (Media): 21928
    Orig Destn IP Address:Port (Media): [ - ]:0
    PM-HO-VG-01#sh sip
    PM-HO-VG-01#sh sip-ua call
    PM-HO-VG-01#sh sip-ua calls 
    Total SIP call legs:2, User Agent Client:1, User Agent Server:1
    SIP UAC CALL INFO
    Call 1
    SIP Call ID                : [email protected]
       State of the call       : STATE_ACTIVE (7)
       Substate of the call    : SUBSTATE_NONE (0)
       Calling Number          : 27218091323
       Called Number           : 0862000000
       Bit Flags               : 0xC04018 0x10000100 0x0
       CC Call ID              : 64511
       Source IP Address (Sig ): 10.18.81.2
       Destn SIP Req Addr:Port : [10.111.111.254]:5060
       Destn SIP Resp Addr:Port: [10.111.111.254]:5060
       Destination Name        : 10.111.111.254
       Number of Media Streams : 1
       Number of Active Streams: 1
       RTP Fork Object         : 0x0
       Media Mode              : flow-through
       Media Stream 1
         State of the stream      : STREAM_ACTIVE
         Stream Call ID           : 64511
         Stream Type              : voice+dtmf (0)
         Stream Media Addr Type   : 1
         Negotiated Codec         : g729br8 (20 bytes)
         Codec Payload Type       : 18 
         Negotiated Dtmf-relay    : rtp-nte
         Dtmf-relay Payload Type  : 101
         QoS ID                   : -1
         Local QoS Strength       : BestEffort
         Negotiated QoS Strength  : BestEffort
         Negotiated QoS Direction : None
         Local QoS Status         : None
         Media Source IP Addr:Port: [10.18.81.2]:22256
         Media Dest IP Addr:Port  : [10.111.111.254]:20074
    Options-Ping    ENABLED:NO    ACTIVE:NO
       Number of SIP User Agent Client(UAC) calls: 1
    SIP UAS CALL INFO
    Call 1
    SIP Call ID                : [email protected]
       State of the call       : STATE_ACTIVE (7)
       Substate of the call    : SUBSTATE_NONE (0)
       Calling Number          : 0218091323
       Called Number           : 0862000000
       Bit Flags               : 0xC0401E 0x10000100 0x80004
       CC Call ID              : 64510
       Source IP Address (Sig ): 10.18.81.2
       Destn SIP Req Addr:Port : [10.18.81.11]:5060
       Destn SIP Resp Addr:Port: [10.18.81.11]:5060
       Destination Name        : 10.18.81.11
       Number of Media Streams : 1
       Number of Active Streams: 1
       RTP Fork Object         : 0x0
       Media Mode              : flow-through
       Media Stream 1
         State of the stream      : STREAM_ACTIVE
         Stream Call ID           : 64510
         Stream Type              : voice+dtmf (1)
         Stream Media Addr Type   : 1
         Negotiated Codec         : g729br8 (20 bytes)
         Codec Payload Type       : 18 
         Negotiated Dtmf-relay    : rtp-nte
         Dtmf-relay Payload Type  : 101
         QoS ID                   : -1
         Local QoS Strength       : BestEffort
         Negotiated QoS Strength  : BestEffort
         Negotiated QoS Direction : None
         Local QoS Status         : None
         Media Source IP Addr:Port: [10.18.81.2]:22350
         Media Dest IP Addr:Port  : [10.18.80.40]:21928
    Options-Ping    ENABLED:NO    ACTIVE:NO
       Number of SIP User Agent Server(UAS) calls: 1
    PM-HO-VG-01#
    PM-HO-VG-01#
    PM-HO-VG-01#
    As you can see, the call is connected and everything is working perfectly. When I press the hold button, here is what I get:
    NOTE: I have # debug ccsip messages and #debug ccsip calls (running)
    PM-HO-VG-01#
    PM-HO-VG-01#
    Nov 30 11:44:49.210: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received: 
    INVITE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720852ab8b92
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>;tag=3C365010-1E42
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    Supported: timer,resource-priority,replaces
    Min-SE:  1800
    User-Agent: Cisco-CUCM9.1
    Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
    CSeq: 102 INVITE
    Max-Forwards: 70
    Expires: 180
    Allow-Events: presence
    Supported: X-cisco-srtp-fallback
    Supported: Geolocation
    Contact: <sip:[email protected]:5060>
    Content-Type: application/sdp
    Content-Length: 244
    v=0
    o=CiscoSystemsCCM-SIP 9082578 2 IN IP4 10.18.81.11
    s=SIP Call
    c=IN IP4 0.0.0.0
    b=TIAS:8000
    b=AS:8
    t=0 0
    m=audio 21928 RTP/AVP 18 101
    a=rtpmap:18 G729/8000
    a=ptime:20
    a=inactive
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    Nov 30 11:44:49.218: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720852ab8b92
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>;tag=3C365010-1E42
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    CSeq: 102 INVITE
    Allow-Events: telephone-event
    Server: Cisco-SIPGateway/IOS-15.2.4.M5
    Content-Length: 0
    Nov 30 11:44:49.218: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    INVITE sip:[email protected]:5060;transport=udp SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC9241
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>;tag=71913148-1417348035284
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    Supported: 100rel,timer,resource-priority,replaces,sdp-anat
    Min-SE:  1800
    Cisco-Guid: 1020645888-0000065536-0000124117-0189862410
    User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    CSeq: 102 INVITE
    Max-Forwards: 70
    Timestamp: 1417347889
    Contact: <sip:[email protected]:5060>
    Call-Info: <sip:10.18.81.2:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
    Expires: 180
    Allow-Events: telephone-event
    Content-Type: application/sdp
    Content-Length: 271
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 7676 6959 IN IP4 10.18.81.2
    s=SIP Call
    c=IN IP4 0.0.0.0
    t=0 0
    m=audio 22256 RTP/AVP 18 101
    c=IN IP4 0.0.0.0
    a=inactive
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=yes
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=ptime:20
    Nov 30 11:44:49.278: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Received: 
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC9241
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>;tag=71913148-1417348035284
    Call-ID: [email protected]
    CSeq: 102 INVITE
    Timestamp: 1417347889
    Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
    Supported: 
    Accept: application/media_control+xml,application/sdp,application/xml
    Contact: <sip:[email protected]:5060;transport=udp>
    X-BroadWorks-Correlation-Info: bbf94839-a234-4237-95e6-a7037322f0f4
    Content-Type: application/sdp
    Content-Length: 360
    v=0
    o=BroadWorks 316169737 2 IN IP4 10.111.111.254
    s=-
    c=IN IP4 0.0.0.0
    t=0 0
    m=audio 20074 RTP/AVP 18 101 100
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=rtpmap:100 X-NSE/8000
    a=fmtp:100 200-202
    a=X-sqn:0
    a=X-cap: 1 audio RTP/AVP 100
    a=X-cpar: a=rtpmap:100 X-NSE/8000
    a=X-cpar: a=fmtp:100 200-202
    a=X-cap: 2 image udptl t38
    a=inactive
    Nov 30 11:44:49.278: //64511/3CD5D2000001/SIP/Call/sipSPICallInfo: 
    The Call Setup Information is:
    Call Control Block (CCB) : 0x0x3D7B1458
    State of The Call        : STATE_ACTIVE
    TCP Sockets Used         : NO
    Calling Number           : 27218091323
    Called Number            : 0862000000
    Source IP Address (Sig  ): 10.18.81.2
    Destn SIP Req Addr:Port  : 10.111.111.254:5060
    Destn SIP Resp Addr:Port : 10.111.111.254:5060
    Destination Name         : 10.111.111.254
    Nov 30 11:44:49.278: //64511/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo: 
    Number of Media Streams: 1
    Media Stream             : 1
    Negotiated Codec         : g729br8
    Negotiated Codec Bytes   : 20
    Nego. Codec payload      : 18 (tx), 18 (rx)
    Negotiated Dtmf-relay    : 6
    Dtmf-relay Payload       : 101 (tx), 101 (rx)
    Source IP Address (Media): 10.18.81.2
    Source IP Port    (Media): 22256
    Destn  IP Address (Media): 0.0.0.0
    Destn  IP Port    (Media): 20074
    Orig Destn IP Address:Port (Media): [ - ]:0
    Nov 30 11:44:49.282: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    ACK sip:[email protected]:5060;transport=udp SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECA2633
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>;tag=71913148-1417348035284
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    Max-Forwards: 70
    CSeq: 102 ACK
    Allow-Events: telephone-event
    Content-Length: 0
    Nov 30 11:44:49.282: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720852ab8b92
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>;tag=3C365010-1E42
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    CSeq: 102 INVITE
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    Allow-Events: telephone-event
    Contact: <sip:[email protected]:5060>
    Supported: replaces
    Supported: sdp-anat
    Server: Cisco-SIPGateway/IOS-15.2.4.M5
    Supported: timer
    Content-Type: application/sdp
    Content-Length: 271
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 7965 2748 IN IP4 10.18.81.2
    s=SIP Call
    c=IN IP4 0.0.0.0
    t=0 0
    m=audio 22350 RTP/AVP 18 101
    c=IN IP4 0.0.0.0
    a=inactive
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=yes
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=ptime:20
    Nov 30 11:44:49.282: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received: 
    ACK sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72094953dfea
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>;tag=3C365010-1E42
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    Max-Forwards: 70
    CSeq: 102 ACK
    Allow-Events: presence
    Content-Length: 0
    Nov 30 11:44:49.290: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received: 
    INVITE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720a6918040f
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>;tag=3C365010-1E42
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    Supported: timer,resource-priority,replaces
    Min-SE:  1800
    User-Agent: Cisco-CUCM9.1
    Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
    CSeq: 103 INVITE
    Max-Forwards: 70
    Expires: 180
    Allow-Events: presence
    Supported: X-cisco-srtp-fallback
    Supported: Geolocation
    Contact: <sip:[email protected]:5060>
    Content-Length: 0
    Nov 30 11:44:49.294: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720a6918040f
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>;tag=3C365010-1E42
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    CSeq: 103 INVITE
    Allow-Events: telephone-event
    Server: Cisco-SIPGateway/IOS-15.2.4.M5
    Content-Length: 0
    Nov 30 11:44:49.294: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    INVITE sip:[email protected]:5060;transport=udp SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECB16F3
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>;tag=71913148-1417348035284
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    Supported: timer,resource-priority,replaces,sdp-anat
    Min-SE:  1800
    Cisco-Guid: 1020645888-0000065536-0000124117-0189862410
    User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    CSeq: 103 INVITE
    Max-Forwards: 70
    Timestamp: 1417347889
    Contact: <sip:[email protected]:5060>
    Expires: 180
    Allow-Events: telephone-event
    Content-Length: 0
    Nov 30 11:44:49.338: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Received: 
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECB16F3
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>;tag=71913148-1417348035284
    Call-ID: [email protected]
    CSeq: 103 INVITE
    Timestamp: 1417347889
    Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
    Supported: 
    Accept: application/media_control+xml,application/sdp,application/xml
    Contact: <sip:[email protected]:5060;transport=udp>
    Content-Type: application/sdp
    Content-Length: 306
    v=0
    o=BroadWorks 316169737 3 IN IP4 10.111.111.254
    s=-
    c=IN IP4 10.111.111.254
    t=0 0
    m=audio 20074 RTP/AVP 18 101
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=X-sqn:0
    a=X-cap: 1 audio RTP/AVP 100
    a=X-cpar: a=rtpmap:100 X-NSE/8000
    a=X-cpar: a=fmtp:100 200-202
    a=X-cap: 2 image udptl t38
    Nov 30 11:44:49.342: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    SIP/2.0 2
    PM-HO-VG-01#00 OK
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720a6918040f
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>;tag=3C365010-1E42
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    CSeq: 103 INVITE
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    Allow-Events: telephone-event
    Contact: <sip:[email protected]:5060>
    Supported: replaces
    Supported: sdp-anat
    Server: Cisco-SIPGateway/IOS-15.2.4.M5
    Supported: timer
    Content-Type: application/sdp
    Content-Length: 289
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 7965 2749 IN IP4 10.18.81.2
    s=SIP Call
    c=IN IP4 10.18.81.2
    t=0 0
    m=audio 22350 RTP/AVP 18 101 19
    c=IN IP4 10.18.81.2
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=yes
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=rtpmap:19 CN/8000
    a=ptime:20
    Nov 30 11:44:49.350: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received: 
    ACK sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720b594cd517
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>;tag=3C365010-1E42
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    Max-Forwards: 70
    CSeq: 103 ACK
    Allow-Events: presence
    Content-Type: application/sdp
    Content-Length: 213
    v=0
    o=CiscoSystemsCCM-SIP 9082578 3 IN IP4 10.18.81.11
    s=SIP Call
    c=IN IP4 10.18.81.10
    t=0 0
    m=audio 4000 RTP/AVP 18
    a=X-cisco-media:umoh
    a=rtpmap:18 G729/8000
    a=ptime:20
    a=fmtp:18 annexb=no
    a=sendonly
    Nov 30 11:44:49.354: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    BYE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECC55
    From: <sip:[email protected]>;tag=3C365010-1E42
    To: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
    Max-Forwards: 70
    Timestamp: 1417347889
    CSeq: 101 BYE
    Reason: Q.850;cause=86
    P-RTP-Stat: PS=874,OS=17480,PR=872,OR=17440,PL=0,JI=0,LA=0,DU=17
    Content-Length: 0
    Nov 30 11:44:49.354: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    BYE sip:[email protected]:5060;transport=udp SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECD1ECD
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>;tag=71913148-1417348035284
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
    Max-Forwards: 70
    Timestamp: 1417347889
    CSeq: 104 BYE
    Reason: Q.850;cause=65
    P-RTP-Stat: PS=872,OS=17440,PR=952,OR=19040,PL=0,JI=0,LA=0,DU=17
    Content-Length: 0
    Nov 30 11:44:49.374: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Received: 
    SIP/2.0 200 Race Condition
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECD1ECD
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>;tag=71913148-1417348035284
    Call-ID: [email protected]
    Timestamp: 1417347889
    CSeq: 104 BYE
    Content-Length: 0
    Nov 30 11:44:49.374: //64511/3CD5D2000001/SIP/Call/sipSPICallInfo: 
    The Call Setup Information is:
    Call Control Block (CCB) : 0x0x3D7B1458
    State of The Call        : STATE_DEAD
    TCP Sockets Used         : NO
    Calling Number           : 27218091323
    Called Number            : 0862000000
    Source IP Address (Sig  ): 10.18.81.2
    Destn SIP Req Addr:Port  : 10.111.111.254:5060
    Destn SIP Resp Addr:Port : 10.111.111.254:5060
    Destination Name         : 10.111.111.254
    Nov 30 11:44:49.374: //64511/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo: 
    Number of Media Streams: 1
    Media Stream             : 1
    Negotiated Codec         : g729br8
    Negotiated Codec Bytes   : 20
    Nego. Codec payload      : 18 (tx), 18 (rx)
    Negotiated Dtmf-relay    : 6
    Dtmf-relay Payload       : 101 (tx), 101 (rx)
    Source IP Address (Media): 10.18.81.2
    Source IP Port    (Media): 22256
    Destn  IP Address (Media): 10.111.111.254
    Destn  IP Port    (Media): 20074
    Orig Destn IP Address:Port (Media): [ - ]:0
    Nov 30 11:44:49.374: //64511/3CD5D2000001/SIP/Call/sipSPICallInfo: 
    Disconnect Cause (CC)    : 65
    Disconnect Cause (SIP)   : 200
    Nov 30 11:44:49.406: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Received: 
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECC55
    From: <sip:[email protected]>;tag=3C365010-1E42
    To: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    CSeq: 101 BYE
    Content-Length: 0
    Nov 30 11:44:49.406: //64510/3CD5D2000001/SIP/Call/sipSPICallInfo: 
    The Call Setup Information is:
    Call Control Block (CCB) : 0x0x3D816D70
    State of The Call        : STATE_DEAD
    TCP Sockets Used         : NO
    Calling Number           : 0218091323
    Called Number            : 0862000000
    Source IP Address (Sig  ): 10.18.81.2
    Destn SIP Req Addr:Port  : 10.18.81.11:5060
    Destn SIP Resp Addr:Port : 10.18.81.11:5060
    Destination Name         : 10.18.81.11
    Nov 30 11:44:49.406: //64510/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo: 
    Number of Media Streams: 1
    Media Stream             : 1
    Negotiated Codec         : g729br8
    Negotiated Codec Bytes   : 20
    Nego. Codec payload      : 18 (tx), 18 (rx)
    Negotiated Dtmf-relay    : 6
    Dtmf-relay Payload       : 101 (tx), 101 (rx)
    Source IP Address (Media): 10.18.81.2
    Source IP Port    (Media): 22350
    Destn  IP Address (Media): 0.0.0.0
    Destn  IP Port    (Media): 21928
    Orig Destn IP Address:Port (Media): [ - ]:0
    Nov 30 11:44:49.406: //64510/3CD5D2000001/SIP/Call/sipSPICallInfo: 
    Disconnect Cause (CC)    : 86
    Disconnect Cause (SIP)   : 200
    PM-HO-VG-01#

    Hi Manish,
    Again, excellent feedback. Much appreciated.
    I will try the commands suggested above and see if I can get DTMF to work correctly while interworking H.323 and SIP.
    But my ultimate goal is to have SIP all way from the CUCM to the CUBE and from the CUBE to the ITSP.
    If I enable SIP Early Offer with MTP on the CUCM going to the CUBE, all SIP Invite sent from the CUCM to the CUBE uses G.729r8 as the codec and once the call is established using G.729r8 should the ITSP reply with that codec, the call succeed and I am able to see an active MTP session using G.729 when I issue the command # show sccp connections.
    One thing that I saw is that my ITSP love so much sending G.729br8 most of the times, so even if using SIP EO with MTP on the SIP Trunk to the CUBE, when I sent my INVITE out from the CUCM to the CUBE using G.729r8, specially on call center numbers such as 0800 numbers, you will see that the call established but the codec being negotiated is G.729br8 which is voice only (missing DTMF).
    I will be doing some intensive test again later on this week and will send the logs. 
    Here is my question to both of you:
    Which is the best way of having a proper SIP to SIP setup all the way that will not pose any problem?
    Do I have to enable Early Offer on the SIP Profile used by the CUBE SIP Trunk or should I use the normal Standard SIP Profile? Do I need to enable MTP on my CUBE SIP Trunk or not?
    From the CUBE point of view, I have a voice class codec that support G.729r8 or G.729br8 and the DTMF Relay method supported by the ITSP is RFC 2833.
    I will send more logs for each scenario. I think that we are getting close to the resolution of this problem.
    Thanks again for your support fellows.

  • Callcentric SIP Trunk (ITSP -- 2811 CUBE -- CUCM 8.6

    I have a SIP trunk from call centric that goes into my lab gear - they appear to be a good sip service due to cost but I'm having some trouble getting calls to route correctly. The call flow is Callcentric.com ITSP (SIP) --> 2811 (acting as cube) -->SIP Trunk --> CUCM 8.6. Phones are registered to CUCM.
    I have the sip trunk registered and calls come in to the router (I see them in ccsip message/call debugs) The 2811 running  15.1(4)M7). Callcentric sends the username of the customer in the sip Invite instead of the called number, the called number is in the TO field. I have several DID’s from Callcentric (18452055544, 18452055545, 18452055546) for my lab. There are a few configs on here for CME where the customer number (17772253754) is simply translated to their phone DN - which is fine if you only have 1 DN with callcentric but more than 1 and thats not feasible since every inbound did will be matched to that 17772253754 translation/phone dn.
    I’m using the a guide from http://tblog.cisco.be/2011/02/17/cube-conditional-sip-profiles/ using the Copy function as described http://www.cisco.com/c/en/us/products/collateral/ios-nx-os-software/ios-software-release-15-1-3-t/product_bulletin_c25-635704.html
    I haven’t been able to find anything where they actually explain all the header fields so Its mostly trial and error.. so far mostly error.  I think I’m close.. but who knows. Any assistance would be greatly appreciated
    voice class sip-profiles 1
    request INVITE peer-header sip TO copy ".sip:(.*)@." u01
    request INVITE sip-header SIP-Req-URI modify ".*@(.*)" "INVITE sip:\u01@\1"
    CUCM (single/pub)- 192.168.1.200
    2811 acting as cube - 192.168.1.203
    Calling Number - 18165297500
    Called Number - 18452055544
    vrtr1#show  sip register status
    Line                             peer       expires(sec) registered P-Associ-URI
    ================================ ========== ============ ========== ============
    17772253754                      -1         20           yes
    vrtr1#
    The Call Setup Information is:
    Call Control Block (CCB) : 0x49646C28
    State of The Call        : STATE_DEAD
    TCP Sockets Used         : NO
    Calling Number           : 18165297500
    Called Number            : 17772253754 (my customer number not called number)
    Source IP Address (Sig  ): 192.168.1.203 (my 2811 router)
    Destn SIP Req Addr:Port  : 204.11.192.159:5080
    Destn SIP Resp Addr:Port : 204.11.192.159:5080
    Destination Name         : 204.11.192.159
    Feb 14 11:20:53.303: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    INVITE sip:[email protected]:5060 SIP/2.0
    v: SIP/2.0/UDP 204.11.192.159:5080;branch=z9hG4bK-805ff2443b18502ff96181045b62dd74
    f: <sip:[email protected]>;tag=3601387252-874282
    t: <sip:[email protected]>
    i: [email protected]
    CSeq: 1 INVITE
    Max-Forwards: 8
    m: <sip:[email protected]:5080;transport=udp>
    Supported: timer
    c: application/sdp
    l: 350
    v=0
    o=NexTone-MSW 2147483647 2147483647 IN IP4 204.11.192.159
    s=sip call
    c=IN IP4 204.11.192.159
    t=0 0
    m=audio 61094 RTP/AVP 18 0 8 101
    a=fmtp:18 annexb=no
    a=fmtp:101 0-15
    a=rtpmap:101 telephone-event/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:18 G729/8000
    a=ptime:20
    a=sendrecv
    a=silenceSupp:off - - - -
    a=setup:actpass
    Feb 14 11:20:53.327: //936/310B294680AD/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 204.11.192.159:5080;branch=z9hG4bK-805ff2443b18502ff96181045b62dd74
    From: <sip:[email protected]>;tag=3601387252-874282
    To: <sip:[email protected]>
    Date: Fri, 14 Feb 2014 17:20:53 GMT
    Call-ID: [email protected]
    CSeq: 1 INVITE
    Allow-Events: telephone-event
    Server: Cisco-SIPGateway/IOS-12.x
    Content-Length: 0
    Feb 14 11:20:53.327: //936/310B294680AD/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 404 Not Found
    Via: SIP/2.0/UDP 204.11.192.159:5080;branch=z9hG4bK-805ff2443b18502ff96181045b62dd74
    From: <sip:[email protected]>;tag=3601387252-874282
    To: <sip:[email protected]>;tag=35399D8-63
    Date: Fri, 14 Feb 2014 17:20:53 GMT
    Call-ID: [email protected]
    CSeq: 1 INVITE
    Allow-Events: telephone-event
    Server: Cisco-SIPGateway/IOS-12.x
    Reason: Q.850;cause=1
    Content-Length: 0
    Feb 14 11:20:53.419: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    ACK sip:[email protected]:5060 SIP/2.0
    v: SIP/2.0/UDP 204.11.192.159:5080;branch=z9hG4bK-805ff2443b18502ff96181045b62dd74
    f: <sip:[email protected]>;tag=3601387252-874282
    t: <sip:[email protected]>;tag=35399D8-63
    i: [email protected]
    CSeq: 1 ACK
    Max-Forwards: 10
    l: 0
    u all
    Feb 14 11:20:57.067: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    ACK sip:18452055544;cic=0288;rn=6465471001;[email protected]:5070 SIP/2.0
    v: SIP/2.0/UDP 204.11.192.159:5080;branch=z9hG4bK-6bceae47efe9f53b4234698a32ac8beb
    f: <sip:[email protected]>;tag=3601387252-874282
    t: <sip:[email protected]>;tag=35399D8-63
    i: [email protected]
    CSeq: 1 ACK
    Max-Forwards: 8
    l: 0
    ************************** Running Config **************************
    sh run
    vrtr1#sh running-config
    Building configuration...
    Current configuration : 4189 bytes
    ! Last configuration change at 00:34:03 CST Fri Feb 14 2014
    ! NVRAM config last updated at 20:26:58 CST Thu Feb 13 2014
    ! NVRAM config last updated at 20:26:58 CST Thu Feb 13 2014
    version 15.1
    service timestamps debug datetime msec localtime
    service timestamps log datetime msec localtime
    no service password-encryption
    hostname vrtr1
    boot-start-marker
    boot system flash:
    boot system flash flash:c2800nm-ipvoicek9-mz.151-4.M7.bin
    boot-end-marker
    card type t1 0 0
    logging buffered 4096 notifications
    enable password cisco
    no aaa new-model
    memory-size iomem 5
    clock timezone CST -6 0
    clock summer-time CST recurring
    no network-clock-participate wic 0
    dot11 syslog
    ip source-route
    ip cef
    ip name-server 192.168.1.9
    no ipv6 cef
    multilink bundle-name authenticated
    voice service voip
    ip address trusted list
      ipv4 192.168.1.0 255.255.255.0
      ipv4 204.11.192.0 255.255.255.0
    allow-connections h323 to h323
    allow-connections h323 to sip
    allow-connections sip to h323
    allow-connections sip to sip
    supplementary-service h450.12
    no supplementary-service sip moved-temporarily
    no supplementary-service sip refer
    sip
      bind control source-interface FastEthernet0/0
      bind media source-interface FastEthernet0/0
      registrar server expires max 1800 min 1800
      localhost dns:callcentric.com
      outbound-proxy dns:callcentric.com
    voice class codec 1
    codec preference 1 g729r8
    codec preference 2 g711ulaw
    voice class sip-profiles 1
    request INVITE peer-header sip TO copy ".sip:(.*)@." u01
    request INVITE sip-header SIP-Req-URI modify ".*@(.*)" "INVITE sip:\u01@\1"
    voice-card 0
    crypto pki token default removal timeout 0
    license udi pid CISCO2811 sn FTX1133A4QR
    controller T1 0/0/0
    cablelength long 0db
    interface FastEthernet0/0
    description ** LAN **
    ip address 192.168.1.203 255.255.255.0
    duplex auto
    speed auto
    h323-gateway voip interface
    h323-gateway voip bind srcaddr 192.168.1.203
    interface FastEthernet0/1
    no ip address
    shutdown
    duplex auto
    speed auto
    ip forward-protocol nd
    no ip http server
    no ip http secure-server
    ip route 0.0.0.0 0.0.0.0 192.168.1.1
    snmp mib persist circuit
    control-plane
    voice-port 0/1/0
    voice-port 0/1/1
    voice-port 0/1/2
    voice-port 0/1/3
    ccm-manager mgcp
    no ccm-manager fax protocol cisco
    ccm-manager music-on-hold
    ccm-manager config server 192.168.1.200 
    ccm-manager config
    mgcp
    mgcp call-agent 192.168.1.200 2427 service-type mgcp version 0.1
    mgcp dtmf-relay voip codec all mode out-of-band
    mgcp rtp unreachable timeout 1000 action notify
    mgcp modem passthrough voip mode nse
    mgcp package-capability rtp-package
    mgcp package-capability sst-package
    mgcp package-capability pre-package
    no mgcp package-capability res-package
    no mgcp package-capability fxr-package
    no mgcp timer receive-rtcp
    mgcp sdp simple
    mgcp fax t38 inhibit
    mgcp rtp payload-type g726r16 static
    mgcp bind control source-interface FastEthernet0/0
    mgcp bind media source-interface FastEthernet0/0
    mgcp profile default
    dial-peer voice 999100 pots
    service mgcpapp
    port 0/1/0
    dial-peer voice 999101 pots
    service mgcpapp
    port 0/1/1
    dial-peer voice 999102 pots
    service mgcpapp
    port 0/1/2
    dial-peer voice 999103 pots
    service mgcpapp
    port 0/1/3
    dial-peer voice 999010 pots
    service mgcpapp
    port 0/1/0
    dial-peer voice 6 voip
    description ## INBOUND DID to CUCM ##
    session protocol sipv2
    session target ipv4:192.168.1.200
    incoming called-number 17772253754
    voice-class sip profiles 1
    dtmf-relay h245-alphanumeric
    no vad
    dial-peer voice 7 voip
    description ## INBOUND DID to CUCM ##
    session protocol sipv2
    session target ipv4:192.168.1.200
    incoming called-number 1845205554[4-5]
    voice-class sip profiles 1
    dtmf-relay h245-alphanumeric
    no vad
    sip-ua
    credentials username 17772253754 password 7 106C1B49111F17194D realm callcentric.com
    authentication username 17772253754 password 7 08035E1E1D11000553 realm callcentric.com
    no remote-party-id
    retry invite 2
    retry register 10
    timers connect 100
    mwi-server dns:callcentric.com expires 3600 port 5060 transport udp
    registrar dns:callcentric.com expires 3600
    sip-server dns:callcentric.com
    host-registrar
    line con 0
    line aux 0
    line vty 0 4
    password cisco
    login
    transport input all
    scheduler allocate 20000 1000
    ntp server 199.102.46.72
    ntp server 23.227.162.123 prefer
    end
    exit

    Thank you for the reply. I've updated the dial-peers as sugested. I'm now seeing an invite go out to my CUCM however the call fails with a 403 (forbidden) which appears to come from the ITSP (Callcentric). I've included a new set of ccsip message debugs and the dial-peers as adjusted. Please let me know what you think.
    dial-peer voice 6 voip
    description ## INBOUND CALL from ITSP ##
    session protocol sipv2
    session target sip-server
    incoming called-number 17772253754
    voice-class sip profiles 1
    dtmf-relay rtp-nte
    no vad
    dial-peer voice 100 voip
    description ## INBOUND DID to CUCM ##
    destination-pattern 17772253754
    session protocol sipv2
    session target ipv4:192.168.1.200
    voice-class sip profiles 1
    dtmf-relay rtp-nte
    no vad
    dial-peer voice 7 voip
    description ## INBOUND DID to CUCM ##
    session protocol sipv2
    session target ipv4:192.168.1.200
    incoming called-number 1845205554[4-5]
    voice-class sip profiles 1
    dtmf-relay rtp-nte
    no vad
    Feb 15 10:18:11.424: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    INVITE sip:[email protected]:5060 SIP/2.0
    v: SIP/2.0/UDP 204.11.192.164:5080;branch=z9hG4bK-d78945b444598bc22c8509d069f4789d
    f: ;tag=3601469891-655
    t: [email protected]>
    i: [email protected]
    CSeq: 1 INVITE
    Max-Forwards: 8
    m:
    Supported: timer
    c: application/sdp
    l: 350
    v=0
    o=NexTone-MSW 2147483647 2147483647 IN IP4 204.11.192.164
    s=sip call
    c=IN IP4 204.11.192.164
    t=0 0
    m=audio 61782 RTP/AVP 18 0 8 101
    a=fmtp:18 annexb=no
    a=fmtp:101 0-15
    a=rtpmap:101 telephone-event/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:18 G729/8000
    a=ptime:20
    a=sendrecv
    a=silenceSupp:off - - - -
    a=setup:actpass
    Feb 15 10:18:11.456: //2419/9933162D820E/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 204.11.192.164:5080;branch=z9hG4bK-d78945b444598bc22c8509d069f4789d
    From: ;tag=3601469891-655
    To: [email protected]>
    Date: Sat, 15 Feb 2014 16:18:11 GMT
    Call-ID: [email protected]
    CSeq: 1 INVITE
    Allow-Events: telephone-event
    Server: Cisco-SIPGateway/IOS-12.x
    Content-Length: 0
    Feb 15 10:18:11.460: //2420/9933162D820E/SIP/Msg/ccsipDisplayMsg:
    Sent:
    INVITE sip:@192.168.1.200:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK9A91F35
    From: [email protected]>;tag=8408644-12C8
    To:
    Date: Sat, 15 Feb 2014 16:18:11 GMT
    Call-ID: [email protected]
    Supported: 100rel,timer,resource-priority,replaces,sdp-anat
    Min-SE:  1800
    Cisco-Guid: 2570262061-2509443555-2182021079-2501285341
    User-Agent: Cisco-SIPGateway/IOS-12.x
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    CSeq: 101 INVITE
    Timestamp: 1392481091
    Contact:
    Expires: 180
    Allow-Events: telephone-event
    Max-Forwards: 7
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 273
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 2786 1511 IN IP4 192.168.1.203
    s=SIP Call
    c=IN IP4 192.168.1.203
    t=0 0
    m=audio 18168 RTP/AVP 18 101
    c=IN IP4 192.168.1.203
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=ptime:20
    Feb 15 10:18:11.552: //2420/9933162D820E/SIP/Msg/ccsipDisplayMsg:
    Received:
    SIP/2.0 407 Proxy Authentication Required
    v: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK9A91F35;rport=57100;received=24.123.98.94
    f: [email protected]>;tag=8408644-12C8
    t:
    i: [email protected]
    CSeq: 101 INVITE
    Proxy-Authenticate: Digest realm="callcentric.com", domain="sip:callcentric.com", nonce="8ae6b7b1cea74cf401e8a26fd3c7371b", opaque="", stale=TRUE, algorithm=MD5
    l: 0
    Feb 15 10:18:11.560: //2420/9933162D820E/SIP/Msg/ccsipDisplayMsg:
    Sent:
    ACK sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK9A91F35
    From: [email protected]>;tag=8408644-12C8
    To:
    Date: Sat, 15 Feb 2014 16:18:11 GMT
    Call-ID: [email protected]
    Max-Forwards: 70
    CSeq: 101 ACK
    Allow-Events: telephone-event
    Content-Length: 0
    Feb 15 10:18:11.560: //2420/9933162D820E/SIP/Msg/ccsipDisplayMsg:
    Sent:
    INVITE sip:@192.168.1.200:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK9AA1BA3
    From: [email protected]>;tag=8408644-12C8
    To:
    Date: Sat, 15 Feb 2014 16:18:11 GMT
    Call-ID: [email protected]
    Supported: 100rel,timer,resource-priority,replaces,sdp-anat
    Min-SE:  1800
    Cisco-Guid: 2570262061-2509443555-2182021079-2501285341
    User-Agent: Cisco-SIPGateway/IOS-12.x
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    CSeq: 102 INVITE
    Timestamp: 1392481091
    Contact:
    Expires: 180
    Allow-Events: telephone-event
    Proxy-Authorization: Digest username="17772253754",realm="callcentric.com",uri="sip:[email protected]:5060",response="a381f10fbbfbd255b444569fef0dddfe",nonce="8ae6b7b1cea74cf401e8a26fd3c7371b",opaque="",algorithm=MD5
    Max-Forwards: 7
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 273
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 2786 1511 IN IP4 192.168.1.203
    s=SIP Call
    c=IN IP4 192.168.1.203
    t=0 0
    m=audio 18168 RTP/AVP 18 101
    c=IN IP4 192.168.1.203
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=ptime:20
    Feb 15 10:18:11.648: //2420/9933162D820E/SIP/Msg/ccsipDisplayMsg:
    Received:
    SIP/2.0 403 Incorrect Authentication
    v: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK9AA1BA3;rport=57100;received=24.123.98.94
    f: [email protected]>;tag=8408644-12C8
    t:
    i: [email protected]
    CSeq: 102 INVITE
    l: 0
    Feb 15 10:18:11.660: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    ACK sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK9AA1BA3
    From: [email protected]>;tag=8408644-12C8
    To:
    Date: Sat, 15 Feb 2014 16:18:11 GMT
    Call-ID: [email protected]
    Max-Forwards: 70
    CSeq: 102 ACK
    Allow-Events: telephone-event
    Content-Length: 0
    Feb 15 10:18:11.660: //2419/9933162D820E/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 403 Forbidden
    Via: SIP/2.0/UDP 204.11.192.164:5080;branch=z9hG4bK-d78945b444598bc22c8509d069f4789d
    From: ;tag=3601469891-655
    To: [email protected]>;tag=8408714-B60
    Date: Sat, 15 Feb 2014 16:18:11 GMT
    Call-ID: [email protected]
    CSeq: 1 INVITE
    Allow-Events: telephone-event
    Server: Cisco-SIPGateway/IOS-12.x
    Reason: Q.850;cause=57
    Content-Length: 0
    Feb 15 10:18:11.752: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    apsc-vrtr1#ACK sip:[email protected]:5060 SIP/2.0
    v: SIP/2.0/UDP 204.11.192.164:5080;branch=z9hG4bK-d78945b444598bc22c8509d069f4789d
    f: ;tag=3601469891-655
    t: [email protected]>;tag=8408714-B60
    i: [email protected]
    CSeq: 1 ACK
    Max-Forwards: 10
    l: 0
    vrtr1#u al
    Feb 15 10:18:14.776: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    ACK sip:18452055544;cic=0288;rn=6465471001;[email protected]:5070 SIP/2.0
    v: SIP/2.0/UDP 204.11.192.164:5080;branch=z9hG4bK-e437c2c5cac5f1a6e147c1cd7c98aad7
    f: ;tag=3601469891-655
    t: [email protected]>;tag=8408714-B60
    i: [email protected]
    CSeq: 1 ACK
    Max-Forwards: 8
    l: 0

  • DTMF tones from CUCUM 9 thru H323 GW out SIP trunk not working

      This is the setup.  Currently in lab environment for a client, but needs to go into production
    IP Phone -> CUCM 9 -> H323 GW -> SIP Trunk -> Proprietary device -> Analog phone
    Calls complete both ways with no issues.  Proprietary devices only uses G711ulaw, so I have configured a xcoder on the H323 GW to transcode to G729 across the WAN link (between the CUCM cluster and the H323 GW).
    Pressing keys/sending DTMF tones from the IP phone are not heard in the analog phone
    Running a debug voice ccpai inout at the H323 gateway shows me that the DTMF tones are being received the GW and are being sent along.  See below:
    Seaport#
    Seaport#
    Seaport#! Pressing digit "9" on VoIP phone
    Seaport#
    Seaport#
    Seaport#
    Seaport#
    *Nov  5 15:41:57.637: //1181/00B99D4F0500/CCAPI/cc_api_call_digit_begin:
       Consume mask is not set. Relaying Digit 9 to dstCallId 0x49E
    *Nov  5 15:41:57.637: //1181/00B99D4F0500/CCAPI/cc_relay_digit_begin_for_3way_conference:
       Check DTMF relay digit begin for 3way conf
    *Nov  5 15:41:57.713: //1181/00B99D4F0500/CCAPI/cc_api_call_digit_end:
       Consume mask is not set. Relaying Digit 9 to dstCallId 0x49E
    *Nov  5 15:41:57.713: //1181/00B99D4F0500/CCAPI/cc_relay_digit_end_for_3way_conference:
       Check DTMF relay digit end for 3way conf
    Seaport#
    Seaport#! Pressing digit "9" on VoIP phone                " on VoIP phone                 5" on VoIP phone              
    Seaport#
    Seaport#
    Seaport#
    *Nov  5 15:42:14.913: //1181/00B99D4F0500/CCAPI/cc_api_call_digit_begin:
       Consume mask is not set. Relaying Digit 5 to dstCallId 0x49E
    *Nov  5 15:42:14.913: //1181/00B99D4F0500/CCAPI/cc_relay_digit_begin_for_3way_conference:
       Check DTMF relay digit begin for 3way conf
    *Nov  5 15:42:14.989: //1181/00B99D4F0500/CCAPI/cc_api_call_digit_end:
       Consume mask is not set. Relaying Digit 5 to dstCallId 0x49E
    *Nov  5 15:42:14.989: //1181/00B99D4F0500/CCAPI/cc_relay_digit_end_for_3way_conference:
       Check DTMF relay digit end for 3way conf
    Seaport#
    Seaport#! Pressing digit "       5" on VoIP phone              
    Seaport#
    Seaport#
    Seaport#
    *Nov  5 15:42:14.913: //1181/00B99D4F0500/CCAPI/cc_api_call_digit_begin:
       Consume mask is not set. Relaying Digit 5 to dstCallId 0x49E
    *Nov  5 15:42:14.913: //1181/00B99D4F0500/CCAPI/cc_relay_digit_begin_for_3way_conference:
       Check DTMF relay digit begin for 3way conf
    *Nov  5 15:42:14.989: //1181/00B99D4F0500/CCAPI/cc_api_call_digit_end:
       Consume mask is not set. Relaying Digit 5 to dstCallId 0x49E
    *Nov  5 15:42:14.989: //1181/00B99D4F0500/CCAPI/cc_relay_digit_end_for_3way_conference:
       Check DTMF relay digit end for 3way conf
    Seaport#
    Seaport#
    However, debug ccsip does not give me any indications that the DTMF tone is being sent out the SIP trunk.  Debug ccsip all attached.
    Relevant portions of the H323 configuration are below
    voice service voip
    no ip address trusted authenticate
    allow-connections h323 to h323
    allow-connections h323 to sip
    allow-connections sip to h323
    allow-connections sip to sip
    fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
    sip
      bind control source-interface Loopback0
      bind media source-interface Loopback0
    voice class codec 1
    codec preference 1 g711ulaw
    codec preference 2 g729r8
    codec preference 3 g729br8
    interface Loopback0
    ip address 172.16.88.254 255.255.255.255
    ip pim sparse-dense-mode
    h323-gateway voip interface
    h323-gateway voip bind srcaddr 172.16.88.254
    interface GigabitEthernet0/1
    ip address 192.168.200.254 255.255.255.0
    duplex auto
    speed auto
    interface Loopback0
    ip address 172.16.88.254 255.255.255.255
    ip pim sparse-dense-mode
    h323-gateway voip interface
    h323-gateway voip bind srcaddr 172.16.88.254
    interface GigabitEthernet0/1                                   <- interface to proprietary device
    ip address 192.168.200.254 255.255.255.0
    duplex auto
    speed auto
    interface GigabitEthernet0/2                                  <-interface to Local LAN supporting IP Phones
    ip address 10.10.10.254 255.255.255.0
    duplex auto
    speed auto
    sccp local GigabitEthernet0/2
    sccp ccm 10.10.10.254 identifier 1 priority 1 version 3.1
    sccp ccm group 1
    bind interface GigabitEthernet0/2
    associate ccm 1 priority 1
    associate profile 10 register xcoder_1
    dspfarm profile 10 transcode 
    codec g711ulaw
    codec g711alaw
    codec g729ar8
    codec g729abr8
    maximum sessions 10
    associate application SCCP
    dial-peer voice 2 voip
              description Default Incoming Dial Peer
    incoming called-number .
    voice-class codec 1 
    dtmf-relay h245-alphanumeric h245-signal rtp-nte
    dial-peer voice 6 voip
    destination-pattern 90052..                      <- DN of analog phone
    session protocol sipv2
    session target ipv4:192.168.200.1            <- IP of proprietary device
    codec g711ulaw
    no vad
    sip-ua
    registrar ipv4:172.16.88.254 expires 3600
    no transport tcp
    telephony-service
    sdspfarm units 4
    sdspfarm transcode sessions 2
    sdspfarm tag 1 xcoder_1
    I also ran the debug voip rtp session named-event all but nothing was displayed when I pressed the digits on the IP Phone.
    Jeff

    Please configure "dtmf-relay rtp-nte" command under SIP dial-peers.
    Jorge Armijo
    Please remember to rate helpful responses and identify helpful or correct answers.

  • Cucm , Cube via Sip and Sip Trunk to ISP , Outgoing calls not working

    Hi
    We have issue with the outgoing calls to sip trunk
    Below is the config and the debugs
    It will be great if you give your thoughts since we have stuck here
    My thoughts are:
    i see that for unknown reason the called number is going with 4 digits instead of 8 digits
    i dont see any sip message comming from ISP
    Maybe the call not going there ? to isp trunk? From the trace the call hit the correct dialpeer 888 but i see 4 digits as a called number , but i dodnt understant the reason to translated in 4 digits the called number.Not apply a translation rule for that
    confused!!!
    Calling Numbner:22324086
    Called Number: 23823690
    CUCM:192.168.1.241 and 242
    CUBE:192.168.1.10
    voice service voip
    ip address trusted list
      ipv4 0.0.0.0 0.0.0.0
    dtmf-interworking rtp-nte
    allow-connections h323 to h323
    allow-connections h323 to sip
    allow-connections sip to h323
    allow-connections sip to sip
    supplementary-service h450.12
    no supplementary-service h225-notify cid-update
    no supplementary-service sip moved-temporarily
    no supplementary-service sip refer
    fax protocol none
    no fax-relay sg3-to-g3
    h323
    sip
      registrar server
      localhost dns:bbtb.cyta.com.cy
      outbound-proxy dns:sbg.bbtb.cyta.com.cy
      no update-callerid
      early-offer forced
    voice class codec 2
    codec preference 1 g711alaw
    codec preference 2 g711ulaw
    codec preference 3 g729br8
    codec preference 4 g729r8
    voice translation-rule 1
    rule 1 /.*\(....\)/ /\1/
    voice translation-rule 3
    rule 1 /^9/ //
    voice translation-rule 4
    rule 1 /\+/ /900/
    rule 2 /^\(9\)\(.......$\)/ /99\2/
    rule 3 /^\(2\)\(.......$\)/ /92\2/
    rule 4 /^0/ /90/
    rule 5 /^1/ /9001/
    rule 6 /^3/ /9003/
    rule 7 /^4/ /9004/
    rule 8 /^5/ /9005/
    rule 9 /^6/ /9006/
    rule 10 /^7/ /9007/
    rule 11 /^8/ /9008/
    rule 12 /^9/ /9009/
    rule 13 /^2/ /9002/
    voice translation-rule 5
    rule 1 // /2232/
    rule 2 /^9/ //
    voice translation-profile SIP_Incoming
    translate calling 4
    translate called 1
    voice translation-profile SIP_Outgoing
    translate calling 5
    translate called 3
    interface FastEthernet0/0
    ip address 192.168.1.10 255.255.255.0
    duplex auto
    speed auto
    interface FastEthernet0/1
    description **SIP TRUNK WITH CYTA**
    ip address 10.249.13.130 255.255.255.252
    duplex auto
    speed auto
    interface FastEthernet0/0
    ip address 192.168.1.10 255.255.255.0
    duplex auto
    speed auto
    interface FastEthernet0/1
    description **SIP TRUNK WITH CYTA**
    ip address 10.249.13.130 255.255.255.252
    duplex auto
    speed auto
    dial-peer voice 889 voip
    description **SIP Trunk to CUCM**
    destination-pattern 4086
    session protocol sipv2
    session target ipv4:192.168.1.242:5060
    voice-class codec 2 
    voice-class sip dtmf-relay force rtp-nte
    no voice-class sip outbound-proxy  
    voice-class sip bind control source-interface FastEthernet0/0
    voice-class sip bind media source-interface FastEthernet0/0
    dtmf-relay sip-notify
    no vad
    dial-peer voice 890 voip
    description **SIP Trunk to CUCM2**
    destination-pattern 4086
    session protocol sipv2
    session target ipv4:192.168.1.241:5060
    voice-class codec 2 
    voice-class sip dtmf-relay force rtp-nte
    no voice-class sip outbound-proxy  
    voice-class sip bind control source-interface FastEthernet0/0
    voice-class sip bind media source-interface FastEthernet0/0
    dtmf-relay sip-notify
    no vad
    dial-peer voice 888 voip
    description **SIP Trunk to CYTA OUTGOING**
    translation-profile incoming SIP_Incoming
    translation-profile outgoing SIP_Outgoing
    destination-pattern 9T
    session protocol sipv2
    session target sip-server
    incoming called-number .
    voice-class codec 2 
    voice-class sip dtmf-relay force rtp-nte
    dtmf-relay rtp-nte
    no vad
    voice service voip
    ip address trusted list
      ipv4 0.0.0.0 0.0.0.0
    dtmf-interworking rtp-nte
    allow-connections h323 to h323
    allow-connections h323 to sip
    allow-connections sip to h323
    allow-connections sip to sip
    supplementary-service h450.12
    no supplementary-service h225-notify cid-update
    no supplementary-service sip moved-temporarily
    no supplementary-service sip refer
    fax protocol none
    no fax-relay sg3-to-g3
    h323
    sip
      registrar server
      localhost dns:bbtb.cyta.com.cy
      outbound-proxy dns:sbg.bbtb.cyta.com.cy
      no update-callerid
      early-offer forced
    voice class codec 2
    codec preference 1 g711alaw
    codec preference 2 g711ulaw
    codec preference 3 g729br8
    codec preference 4 g729r8
    voice translation-rule 1
    rule 1 /.*\(....\)/ /\1/
    voice translation-rule 3
    rule 1 /^9/ //
    voice translation-rule 4
    rule 1 /\+/ /900/
    rule 2 /^\(9\)\(.......$\)/ /99\2/
    rule 3 /^\(2\)\(.......$\)/ /92\2/
    rule 4 /^0/ /90/
    rule 5 /^1/ /9001/
    rule 6 /^3/ /9003/
    rule 7 /^4/ /9004/
    rule 8 /^5/ /9005/
    rule 9 /^6/ /9006/
    rule 10 /^7/ /9007/
    rule 11 /^8/ /9008/
    rule 12 /^9/ /9009/
    rule 13 /^2/ /9002/
    voice translation-rule 5
    rule 1 // /2232/
    rule 2 /^9/ //
    voice translation-profile SIP_Incoming
    translate calling 4
    translate called 1
    voice translation-profile SIP_Outgoing
    translate calling 5
    translate called 3
    dial-peer voice 889 voip
    description **SIP Trunk to CUCM**
    destination-pattern 4086
    session protocol sipv2
    session target ipv4:192.168.1.242:5060
    voice-class codec 2 
    voice-class sip dtmf-relay force rtp-nte
    no voice-class sip outbound-proxy  
    voice-class sip bind control source-interface FastEthernet0/0
    voice-class sip bind media source-interface FastEthernet0/0
    dtmf-relay sip-notify
    no vad
    dial-peer voice 890 voip
    description **SIP Trunk to CUCM2**
    destination-pattern 4086
    session protocol sipv2
    session target ipv4:192.168.1.241:5060
    voice-class codec 2 
    voice-class sip dtmf-relay force rtp-nte
    no voice-class sip outbound-proxy  
    voice-class sip bind control source-interface FastEthernet0/0
    voice-class sip bind media source-interface FastEthernet0/0
    dtmf-relay sip-notify
    no vad
    dial-peer voice 888 voip
    description **SIP Trunk to CYTA OUTGOING**
    translation-profile incoming SIP_Incoming
    translation-profile outgoing SIP_Outgoing
    destination-pattern 9T
    session protocol sipv2
    session target sip-server
    incoming called-number .
    voice-class codec 2 
    voice-class sip dtmf-relay force rtp-nte
    dtmf-relay rtp-nte
    no vad

    Hi Aok
    I change the default value for IPVMS from g711ulaw to g711alaw but the results remained the same
    Also i have  restarted the IPVMS
    SIP-GW#
    SIP-GW#
    *Mar  5 14:19:57.854: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    INVITE sip:[email protected]:5060;transport=tcp SIP/2.0
    Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK79816bbd4196
    From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
    To: ;tag=125E62C-1354
    Date: Tue, 05 Mar 2013 13:52:31 GMT
    Call-ID: [email protected]
    Supported: timer,resource-priority,replaces
    Min-SE:  1800
    User-Agent: Cisco-CUCM9.0
    Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
    CSeq: 102 INVITE
    Max-Forwards: 70
    Expires: 180
    Allow-Events: presence
    Supported: X-cisco-srtp-fallback
    Supported: Geolocation
    Session-Expires:  1800;refresher=uac
    P-Asserted-Identity:
    Remote-Party-ID: ;party=calling;screen=yes;privacy=off
    Contact:
    Content-Type: application/sdp
    Content-Length: 244
    v=0
    o=CiscoSystemsCCM-SIP 38874 2 IN IP4 192.168.1.241
    s=SIP Call
    c=IN IP4 0.0.0.0
    b=TIAS:64000
    b=AS:64
    t=0 0
    m=audio 24784 RTP/AVP 8 101
    a=rtpmap:8 PCMA/8000
    a=ptime:20
    a=inactive
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    *Mar  5 14:19:57.878: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
    Sent:
    INVITE sip:[email protected]:5060;transport=udp SIP/2.0
    Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK355253C
    From: [email protected]>;tag=125E594-5C7
    To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
    Date: Tue, 05 Mar 2013 14:19:57 GMT
    Call-ID: [email protected]
    Route:
    Supported: 100rel,timer,resource-priority,replaces,sdp-anat
    Min-SE:  1800
    Cisco-Guid: 3698592896-0000065536-0000000107-4043417792
    User-Agent: Cisco-SIPGateway/IOS-12.x
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    CSeq: 102 INVITE
    Max-Forwards: 70
    Timestamp: 1362493197
    Contact:
    Expires: 60
    Allow-Events: telephone-event
    Content-Type: application/sdp
    Content-Length: 262
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 6506 3807 IN IP4 10.249.13.130
    s=SIP Call
    c=IN IP4 10.249.13.130
    t=0 0
    m=audio 19234 RTP/AVP 8 101
    c=IN IP4 10.249.13.130
    a=inactive
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=ptime:20
    *Mar  5 14:19:57.878: //716/DC740C800000/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 100 Trying
    Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK79816bbd4196
    From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
    To: ;tag=125E62C-1354
    Date: Tue, 05 Mar 2013 14:19:57 GMT
    Call-ID: [email protected]
    CSeq: 102 INVITE
    Allow-Events: telephone-event
    Server: Cisco-SIPGateway/IOS-12.x
    Content-Length: 0
    *Mar  5 14:19:57.926: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
    Received:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK355253C
    To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
    From: [email protected]>;tag=125E594-5C7
    Call-ID: [email protected]
    CSeq: 102 INVITE
    Contact:
    Require: timer
    Session-Expires: 1800;refresher=uac
    Content-Type: application/sdp
    Content-Length: 213
    Allow: ACK, BYE, CANCEL, INVITE, OPTIONS, PRACK, REFER, NOTIFY, UPDATE, INFO
    Accept: application/media_control+xml
    Accept: application/sdp
    Accept: application/x-broadworks-call-center+xml
    v=0
    o=BroadWorks 96335268 2 IN IP4 10.224.42.164
    s=-
    c=IN IP4 10.224.42.72
    t=0 0
    m=audio 54932 RTP/AVP 8 101
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=ptime:20
    a=inactive
    *Mar  5 14:19:57.942: //716/DC740C800000/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 200 OK
    Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK79816bbd4196
    From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
    To: ;tag=125E62C-1354
    Date: Tue, 05 Mar 2013 14:19:57 GMT
    Call-ID: [email protected]
    CSeq: 102 INVITE
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    Allow-Events: telephone-event
    Contact:
    Supported: replaces
    Supported: sdp-anat
    Server: Cisco-SIPGateway/IOS-12.x
    Session-Expires:  1800;refresher=uac
    Require: timer
    Supported: timer
    Content-Type: application/sdp
    Content-Length: 259
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 9410 5774 IN IP4 192.168.1.10
    s=SIP Call
    c=IN IP4 192.168.1.10
    t=0 0
    m=audio 19314 RTP/AVP 8 101
    c=IN IP4 192.168.1.10
    a=inactive
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=ptime:20
    *Mar  5 14:19:57.946: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
    Sent:
    ACK sip:[email protected]:5060;transport=udp SIP/2.0
    Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK3562A4
    From: [email protected]>;tag=125E594-5C7
    To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
    Date: Tue, 05 Mar 2013 14:19:57 GMT
    Call-ID: [email protected]
    Route:
    Max-Forwards: 70
    CSeq: 102 ACK
    Allow-Events: telephone-event
    Content-Length: 0
    *Mar  5 14:19:57.946: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    ACK sip:[email protected]:5060;transport=tcp SIP/2.0
    Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK798246ab3597
    From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
    To: ;tag=125E62C-1354
    Date: Tue, 05 Mar 2013 13:52:31 GMT
    Call-ID: [email protected]
    Max-Forwards: 70
    CSeq: 102 ACK
    Allow-Events: presence
    Content-Length: 0
    *Mar  5 14:19:58.146: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    INVITE sip:[email protected]:5060;transport=tcp SIP/2.0
    Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK7983739137ab
    From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
    To: ;tag=125E62C-1354
    Date: Tue, 05 Mar 2013 13:52:31 GMT
    Call-ID: [email protected]
    Supported: timer,resource-priority,replaces
    Min-SE:  1800
    User-Agent: Cisco-CUCM9.0
    Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
    CSeq: 103 INVITE
    Max-Forwards: 70
    Expires: 180
    Allow-Events: presence
    Supported: X-cisco-srtp-fallback
    Supported: Geolocation
    Session-Expires:  1800;refresher=uac
    P-Asserted-Identity:
    Remote-Party-ID: ;party=calling;screen=yes;privacy=off
    Contact:
    Content-Length: 0
    *Mar  5 14:19:58.158: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
    Sent:
    INVITE sip:[email protected]:5060;transport=udp SIP/2.0
    Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK3571933
    From: [email protected]>;tag=125E594-5C7
    To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
    Date: Tue, 05 Mar 2013 14:19:58 GMT
    Call-ID: [email protected]
    Route:
    Supported: timer,resource-priority,replaces,sdp-anat
    Min-SE:  1800
    Cisco-Guid: 3698592896-0000065536-0000000107-4043417792
    User-Agent: Cisco-SIPGateway/IOS-12.x
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    CSeq: 103 INVITE
    Max-Forwards: 70
    Timestamp: 1362493198
    Contact:
    Expires: 60
    Allow-Events: telephone-event
    Content-Length: 0
    *Mar  5 14:19:58.158: //716/DC740C800000/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 100 Trying
    Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK7983739137ab
    From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
    To: ;tag=125E62C-1354
    Date: Tue, 05 Mar 2013 14:19:58 GMT
    Call-ID: [email protected]
    CSeq: 103 INVITE
    Allow-Events: telephone-event
    Server: Cisco-SIPGateway/IOS-12.x
    Content-Length: 0
    *Mar  5 14:19:58.218: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
    Received:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK3571933
    To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
    From: [email protected]>;tag=125E594-5C7
    Call-ID: [email protected]
    CSeq: 103 INVITE
    Contact:
    Require: timer
    Session-Expires: 1800;refresher=uac
    Content-Type: application/sdp
    Content-Length: 216
    Allow: ACK, BYE, CANCEL, INVITE, OPTIONS, PRACK, REFER, NOTIFY, UPDATE, INFO
    Accept: application/media_control+xml
    Accept: application/sdp
    Accept: application/x-broadworks-call-center+xml
    v=0
    o=BroadWorks 96335268 3 IN IP4 10.224.42.164
    s=-
    c=IN IP4 10.224.42.72
    t=0 0
    m=audio 54932 RTP/AVP 8 18 96 99
    a=rtpmap:96 AMR/8000
    a=rtpmap:99 telephone-event/8000
    a=fmtp:99 0-15
    a=ptime:20
    a=sendrecv
    *Mar  5 14:19:58.234: //716/DC740C800000/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 200 OK
    Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK7983739137ab
    From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
    To: ;tag=125E62C-1354
    Date: Tue, 05 Mar 2013 14:19:58 GMT
    Call-ID: [email protected]
    CSeq: 103 INVITE
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    Allow-Events: telephone-event
    Contact:
    Supported: replaces
    Supported: sdp-anat
    Server: Cisco-SIPGateway/IOS-12.x
    Session-Expires:  1800;refresher=uac
    Require: timer
    Supported: timer
    Content-Type: application/sdp
    Content-Length: 283
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 9410 5775 IN IP4 192.168.1.10
    s=SIP Call
    c=IN IP4 192.168.1.10
    t=0 0
    m=audio 19314 RTP/AVP 8 18 101
    c=IN IP4 192.168.1.10
    a=rtpmap:8 PCMA/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=yes
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    *Mar  5 14:19:58.242: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    ACK sip:[email protected]:5060;transport=tcp SIP/2.0
    Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK7985648033f2
    From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
    To: ;tag=125E62C-1354
    Date: Tue, 05 Mar 2013 13:52:31 GMT
    Call-ID: [email protected]
    Max-Forwards: 70
    CSeq: 103 ACK
    Allow-Events: presence
    Content-Type: application/sdp
    Content-Length: 192
    v=0
    o=CiscoSystemsCCM-SIP 38874 3 IN IP4 192.168.1.241
    s=SIP Call
    c=IN IP4 192.168.1.241
    t=0 0
    m=audio 4000 RTP/AVP 8
    a=X-cisco-media:umoh
    a=rtpmap:8 PCMA/8000
    a=ptime:20
    a=sendonly
    *Mar  5 14:19:58.262: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
    Sent:
    ACK sip:[email protected]:5060;transport=udp SIP/2.0
    Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK358582
    From: [email protected]>;tag=125E594-5C7
    To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
    Date: Tue, 05 Mar 2013 14:19:58 GMT
    Call-ID: [email protected]
    Route:
    Max-Forwards: 70
    CSeq: 103 ACK
    Allow-Events: telephone-event
    Content-Type: application/sdp
    Content-Length: 259
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 6506 3808 IN IP4 10.249.13.130
    s=SIP Call
    c=IN IP4 10.249.13.130
    t=0 0
    m=audio 19234 RTP/AVP 8 99
    c=IN IP4 10.249.13.130
    a=sendonly
    a=rtpmap:8 PCMA/8000
    a=rtpmap:99 telephone-event/8000
    a=fmtp:99 0-15
    a=ptime:20
    SIP-GW#
    SIP-GW#sh voip rtp connections
    VoIP RTP active connections :
    No. CallId     dstCallId  LocalRTP RmtRTP     LocalIP                                RemoteIP
    1     716        717        19314    4000     192.168.1.10                           192.168.1.241
    2     717        716        19234    54932    10.249.13.130                          10.224.42.72
    Found 2 active RTP connections

  • Lync 2013 with SIP trunk with panasonic kx-tde200

    Hi
    My company has installed a panasonic ip-pbx kx-tde for multiline with 100 number range for telephone service.
    Now my company is going to replace multiline by sip trunk . It will still work with Panasonic pbx box just need to reprogramme to be able to connected to the sip proxy which is managed by internet service provider.
    For this scenario , would Lync 2013 voice work if I just add PSTN gateway which is the ip of panasonic pbx address to the frontend in topology ? Or I may need a mediation server as a must requirement  to make lync voice work?
    Thanks
    WenFei

    Media bypass allow a call to basically skip the mediation server once it's established and go directly from gateway (in this case the PBX) and the endpoint (the telephone handset or Lync client) More information here: http://technet.microsoft.com/en-us/library/gg398719.aspx 
    By having this (if your PBX supports it) you reduce the load on the mediation servers. Before you go too far down the road also make sure that your PBX supports SIP trunks that are SIP over TCP (as Lync doesn't work with SIP over UDP)
    Sort of, the easiest way is to add the .com as an additional SIP domain in Topology builder, you will need to create DNS records for it (both internal and external) and you will need to reissue the certs with additional SANs to support the second domain.
    YOu will also need to update all the users to use the new suffix of xxx.com. So it's not a small task.
    If this helped you please click "Vote As Helpful" if it answered your question please click "Mark As Answer" | Blog
    www.lynced.com.au | Twitter
    @imlynced

  • SIP phone registering on SIP trunk

    Hi,
    i have a UC 500 connected to our phone provider using a SIP trunk.
    All the phones are SPA508 G
    All is working fine !
    Then, some days ago i added a SIP phone (extention 350) on the UC500, that also worked fine, and then after some minutes all our incoming/outgoing calls were blocked.
    I called my provider that told me that our IP was banned because they have seen to much registration attempt from a bad user that was "350"
    I can confirm with a "sh sip-ua register status" command that i had two sip registration : my SIP trunk and the SIP phone
    Then it seems that the UC 500 is trying to register the SIP phone on the SIP trunk ?
    What am i doing wrong ?
    Is there a command to avoid that ?
    Bellow is how the SIP phone and the SIP trunk are configured
    Many thanks for your help, i was unable to find anything about that, but i guess somebody already had this problem !
    The SIP phone -------------------------------------------------------------------------
    voice service voip
     allow-connections h323 to h323
     allow-connections h323 to sip
     allow-connections sip to h323
     allow-connections sip to sip
     supplementary-service h450.12
     fax protocol none
     modem passthrough nse codec g711ulaw
     sip
      registrar server expires max 3600 min 120
      no update-callerid
    voice class codec 1
     codec preference 1 g711ulaw
     codec preference 2 g729r8
    voice register global
     mode cme
     source-address 10.1.1.1 port 5060
     max-dn 20
     max-pool 20
     load 9971 sip9971.9-2-2
     load 9951 sip9951.9-2-2
     load 8961 sip8961.9-2-2
     load 7971 term71.default
     authenticate register
     authenticate realm xxxxxx.com
     timezone 13
     hold-alert
     mwi stutter
     mwi reg-e164
     create profile sync 0636240803635305
    voice register dn  1
     number 350
     name Conference
     label Conference
    voice register pool  1
     id mac 1234.1234.1234
     number 1 dn 1
     username 350 password 1234
     codec g711ulaw
    The SIP trunk ----------------------------------------------------------------------
    sip-ua
     credentials username user1234 password 1234 realm sipgw9.provider.com
     authentication username user1234 password 1234 no remote-party-id
     retry invite 2
     retry register 10
     timers connect 100
     registrar dns:sipgw9.provider.com expires 3600
     sip-server dns:sipgw9.provider.com

    I'm still searching on the forum, and maybe i found somthing related to my problem, not sure... any advice ?
    Disable outbound proxy on voice register global as by default it will use the outbound proxy configured on the system which would not make sense
    voice register global
      no outbound-proxy
    found there : https://supportforums.cisco.com/discussion/10760741/uc500-sip-server-and-sip-trunk

  • Confused by basic SIP Trunk configuration.

    I've went through a few basic SIP trunk configurations and Youtube videos the last couple days but can't figure out what I'm doing wrong.
    I've set up H323 and MGCP no problem, but I can't figure out the SIP trunk set up. I'm guessing there are some concepts I'm not understanding yet.
    I've got a CUCM lab set up. A 2851 PSTN Simulator, 2851 H323 Gateway at the Main site with a 9.0 CUCM setup in that site and a Branch site that I'm trying to set up as a SIP trunk to connect two phones.
    CUCM is on the 192.168.5.x/24 subnet. 172.16.0.x/24 is the subnet connecting the serial(internet) cable between the two gateways in which I'm trying to establish the trunk between.
    The Branch phones are still registering with the CUCM at the main site. The Route Pattern is looking to the Branch Route List which has the SIP Trunk listed. I'm just getting a fast busy when trying to place a call from the branch site to the main site.
    The most frustrating thing I'm not understanding, is that the debug ccsip and call debugs on my SIP Branch gateway shows absolutely nothing.  I've tried registering the branch phones with the SIP Trunk, but stopped when I figured that shouldn't be necessary.
    If someone can make some sense of this, I'd truly appreciate it!

    Hello Aditya and thanks for the consideration!
    I do have a direct IP connection, but I want to set up a SIP trunk and use it just to know how to do it before I do it in production. 
    I did end up deleting the phones from CUCM so they can register with the 2851 CME that I'm setting up as a SIP trunk. So it is registering there, and I set the allow connections and bind sip commands.
    I am now getting Debugs and calls from the SIP Trunk router going to CUCM, but the error message is No Codec, and I Get the fast busy after the call rings on the CUCM Main Site side. So looks like the negotiation is failing. Here is my CLI for the SIP Trunk now after the changes have been made and phones registered to the SIP Branch site as well as the Debug when I tried to place a call to extension "5000":
    Note: I did try to change the codecs in the dial-peers to g729r8 instead of 711 and same fast busy after answering.
    ==============================================
    Branch_SIP#show run
    Building configuration...
    Current configuration : 3529 bytes
    ! Last configuration change at 03:15:11 UTC Thu Apr 2 2015
    version 15.1
    service timestamps debug datetime msec
    service timestamps log datetime msec
    no service password-encryption
    hostname Branch_SIP
    boot-start-marker
    boot-end-marker
    ! card type command needed for slot/vwic-slot 0/2
    enable secret 5 $1$hOXF$gvfmWW1ZIQE0mAMVg.u1c/
    no aaa new-model
    dot11 syslog
    ip source-route
    ip cef
    ip dhcp excluded-address 10.0.10.1 10.0.10.10
    ip dhcp excluded-address 10.0.30.1 10.0.30.10
    ip dhcp pool Data
     network 10.0.10.0 255.255.255.0
     default-router 10.0.10.254
     option 150 ip 192.168.5.250
     dns-server 192.168.5.200
    ip dhcp pool Voice
     network 10.0.30.0 255.255.255.0
     default-router 10.0.30.254
     dns-server 192.168.5.200
     option 150 ip 172.16.0.1
    ip dhcp pool data
     option 150 ip 172.16.0.2
    no ipv6 cef
    multilink bundle-name authenticated
    voice service voip
     allow-connections sip to sip
     sip
      bind media source-interface Loopback1
    voice-card 0
    crypto pki token default removal timeout 0
    license udi pid CISCO2851 sn FTX1031A2FM
    redundancy
    interface Loopback1
     ip address 2.2.2.2 255.255.255.255
    interface GigabitEthernet0/0
     no ip address
     duplex auto
     speed auto
    interface GigabitEthernet0/0.10
     encapsulation dot1Q 10
     ip address 10.0.10.254 255.255.255.0
    interface GigabitEthernet0/0.30
     encapsulation dot1Q 30
     ip address 10.0.30.254 255.255.255.0
    interface GigabitEthernet0/1
     no ip address
     shutdown
     duplex auto
     speed auto
    interface Serial0/3/0
     no ip address
     shutdown
     clock rate 2000000
    interface Serial0/3/1
     ip address 172.16.0.1 255.255.255.0
     clock rate 250000
    interface Internal-Service-Module0/0
     no ip address
     shutdown
     !Application: CUE Running on AIM2
     hold-queue 512 out
    router eigrp 1
     network 0.0.0.0
     network 2.2.2.2 0.0.0.0
     network 10.0.0.0
     network 10.0.10.0 0.0.0.255
     network 10.0.30.0 0.0.0.255
     network 172.16.0.0
    ip forward-protocol nd
    no ip http server
    no ip http secure-server
    ip route 0.0.0.0 0.0.0.0 172.16.0.2
    tftp-server flash:term45.default.loads
    tftp-server flash:jar45sccp.8-5-3TH1-6.sbn
    tftp-server flash:cnu45.8-5-3TH1-6.sbn
    tftp-server flash:apps45.8-5-3TH1-6.sbn
    tftp-server flash:dsp45.8-5-3TH1-6.sbn
    tftp-server flash:cvm45sccp.8-5-3TH1-6.sbn
    control-plane
    voice-port 0/0/0
    voice-port 0/0/1
    mgcp profile default
    dial-peer voice 1 voip
     description **Incoming Call from SIP Trunk**
     session protocol sipv2
     session target sip-server
     codec g711ulaw
    dial-peer voice 2 voip
     description **Outgoing Call to SIP Trunk**
     destination-pattern 5...
     session protocol sipv2
     session target sip-server
     codec g711ulaw
    sip-ua
     sip-server ipv4:192.168.5.250
    telephony-service
     codec g711ulaw
     max-ephones 24
     max-dn 48
     ip source-address 172.16.0.1 port 2000
     system message SIP Branch Site
     cnf-file location flash:
     load 7960-7940 P00308010200.bin
     max-conferences 8 gain -6
     transfer-system full-consult
    ephone-dn  1
     number 4008
    ephone-dn  2
     number 4005
    ephone  1
     device-security-mode none
     mac-address 001D.A21A.2065
     button  1:1
    line con 0
     exec-timeout 0 0
    line aux 0
    line 194
     no activation-character
     no exec
     transport preferred none
     transport input all
     transport output lat pad telnet rlogin lapb-ta mop udptn v120 ssh
     stopbits 1
     speed 115200
    line vty 0 4
     password cisco
     login
     transport input all
    line vty 5 15
     password cisco
     login
     transport input all
    scheduler allocate 20000 1000
    end
    Branch_SIP#show debug
    TFTP:
      TFTP Event debugging is on
    CCSIP SPI: SIP Call Statistics tracing is enabled       (filter is OFF)
    Branch_SIP#
    *Apr  2 03:20:32.351: //25/0C496935804A/SIP/Call/sipSPICallInfo:
    The Call Setup Information is:
    Call Control Block (CCB) : 0x4B6C5C28
    State of The Call        : STATE_DEAD
    TCP Sockets Used         : NO
    Calling Number           : 4008
    Called Number            : 5005
    Source IP Address (Sig  ): 172.16.0.1
    Destn SIP Req Addr:Port  : 192.168.5.250:5060
    Destn SIP Resp Addr:Port : 192.168.5.250:5060
    Destination Name         : 192.168.5.250
    *Apr  2 03:20:32.351: //25/0C496935804A/SIP/Call/sipSPIMediaCallInfo:
    Number of Media Streams: 1
    Media Stream             : 1
    Negotiated Codec         : No Codec
    Negotiated Codec Bytes   : 0
    Nego. Codec payload      : 255 (tx), 255 (rx)
    Negotiated Dtmf-relay    : 0
    Dtmf-relay Payload       : 0 (tx), 0 (rx)
    Source IP Address (Media): 2.2.2.2
    Source IP Port    (Media): 19472
    Destn  IP Address (Media):  -
    Destn  IP Port    (Media): 0
    Orig Destn IP Address:Port (Media): [ - ]:0
    *Apr  2 03:20:32.351: //25/0C496935804A/SIP/Call/sipSPICallInfo:
    Disconnect Cause (CC)    : 63
    Disconnect Cause (SIP)   : 503
    Branch_SIP#

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