Forwarding with SIP Trunking and Retaining CID

We are using SIP trunking to get to the PSTN and CUCM 7.1.  The SIP provider only permits calls originating from their own DIDs.
So how can we allow our users to forward all calls to say their cell phone.  AND when a call comes in we want them to be able to see the original caller ID. Is it possible?  What is the mojo?  Thanks!

Hi,
for the SIP trunks there are no limits from the system. Check this out: https://supportforums.cisco.com/message/3795863#3795863
If you have different voice codecs for your phones and the trunk you need DSP ressources for transcoding.
As I know the UC520 have a PVDM2-64 with 4 DSP chips. You can use the DSP calculator from cisco to find out how many DSPs you need. But keep in mind that conferencing and transcoding can't share a DSP processor.
For example 1 DSP for conferencing and 3 for transcoding.
best regards
Christian

Similar Messages

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  • PROBLEM WITH FORWARDING ALL - SIP TRUNK

    Hello,
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    Hi,
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  • Problems between an UC520 and Asterisk with sip trunk

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    David.

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  • NexVortex SIP trunk and UC500 default timeout settings?

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    Hi Steven,
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    voice class codec 1
     codec preference 1 g711alaw
    voice register global
     mode cme
     source-address 10.1.1.1 port 5060
     load 9971 sip9971.9-2-2
     load 9951 sip9951.9-2-2
     load 8961 sip8961.9-2-2
     timezone 23
    voice source-group CCA_SIP_SOURCE_GROUP_CUE_CME
     access-list 2
     translation-profile incoming SIP_Incoming
    voice source-group CCA_SIP_SOURCE_GROUP_EXTERNAL
     access-list 3
    voice translation-rule 9
     rule 1 /0041449475090/ /90/
     rule 2 /0041449475091/ /91/
     rule 3 /0041449475092/ /92/
     rule 4 /0041449475093/ /93/
     rule 5 /0041449475094/ /94/
     rule 6 /0041449475095/ /95/
     rule 7 /0041449475096/ /96/
     rule 8 /0041449475097/ /97/
     rule 9 /0041449475098/ /98/
     rule 10 /0041449475099/ /99/
    voice translation-rule 410
     rule 1 /^0\(.*\)/ /\1/
     rule 15 /^..$/ /0041449475090/
    voice translation-rule 411
     rule 1 /^0\(.*\)/ /ABCD0\1/
    voice translation-rule 412
     rule 1 /^ABCD\(.*\)/ /\1/
    voice translation-rule 422
     rule 15 /^ABCD\(.*\)/ /\1/
    voice translation-rule 1000
     rule 1 /.*/ //
    voice translation-rule 1111
     rule 1 /^9\([1-9]\)$/ /004144947509\1/
     rule 15 /^..$/ /0041449475090/
    voice translation-rule 1112
     rule 1 /^0/ //
    voice translation-rule 2000
     rule 1 /0041449475098/ /98/
    voice translation-rule 2001
     rule 1 /0041449475097/ /97/
    voice translation-rule 2002
     rule 1 /^6/ //
    voice translation-rule 2222
    voice translation-profile AA_Profile
     translate called 2001
    voice translation-profile CALLER_ID_TRANSLATION_PROFILE
     translate calling 1111
    voice translation-profile CallBlocking
     translate called 2222
    voice translation-profile OUTGOING_TRANSLATION_PROFILE
     translate called 1112
    voice translation-profile PSTN_CallForwarding
     translate redirect-target 410
     translate redirect-called 410
    voice translation-profile PSTN_Outgoing
     translate calling 1111
     translate called 1112
     translate redirect-target 410
     translate redirect-called 410
    voice translation-profile SIP_Called_9
     translate calling 3265
     translate called 9
    voice translation-profile SIP_Incoming
     translate called 411
    voice translation-profile SIP_Passthrough
     translate called 412
    voice translation-profile SIP_Passthrough_CallBlocking
     translate called 422
    voice translation-profile VM_Profile
     translate called 2000
    voice translation-profile XFER_TO_VM_PROFILE
     translate redirect-called 2002
    voice translation-profile nondialable
     translate called 1000
    voice-card 0
     dspfarm
     dsp services dspfarm
    fax interface-type fax-mail
    license udi pid UC540W-BRI-K9 sn FGL163220SL
    archive
     log config
      logging enable
      logging size 600
      hidekeys
    username admin privilege 15 secret xxx
    username xxx password 0 ""
    username xxx password 0 ""
    ip tftp source-interface Loopback0
    bridge irb
    interface Loopback0
     description $FW_INSIDE$
     ip address 10.1.10.2 255.255.255.252
     ip access-group 101 in
     ip nat inside
     ip virtual-reassembly in
    interface FastEthernet0/0
     description $FW_OUTSIDE$
     no ip address
     ip inspect SDM_LOW out
     ip virtual-reassembly in
     ip verify unicast reverse-path
     load-interval 30
     shutdown
     duplex auto
     speed auto
    interface Integrated-Service-Engine0/0
     description cue is initialized with default IMAP group
     ip unnumbered Loopback0
     ip nat inside
     ip virtual-reassembly in
     service-module ip address 10.1.10.1 255.255.255.252
     service-module ip default-gateway 10.1.10.2
    interface FastEthernet0/1/0
     no ip address
     macro description cisco-desktop
     spanning-tree portfast
    interface FastEthernet0/1/1
     switchport voice vlan 100
     no ip address
     macro description cisco-phone
     spanning-tree portfast
    interface FastEthernet0/1/2
     switchport voice vlan 100
     no ip address
     macro description cisco-phone
     spanning-tree portfast
    interface FastEthernet0/1/3
     switchport voice vlan 100
     no ip address
     macro description cisco-phone
     spanning-tree portfast
    interface FastEthernet0/1/4
     switchport voice vlan 100
     no ip address
     macro description cisco-phone
     spanning-tree portfast
    interface FastEthernet0/1/5
     switchport voice vlan 100
     no ip address
     macro description cisco-phone
     spanning-tree portfast
    interface FastEthernet0/1/6
     switchport voice vlan 100
     no ip address
     macro description cisco-phone
     spanning-tree portfast
    interface FastEthernet0/1/7
     switchport voice vlan 100
     no ip address
     macro description cisco-phone
     spanning-tree portfast
    interface FastEthernet0/1/8
     no ip address
     macro description cisco-desktop
     spanning-tree portfast
    interface BRI0/1/0
     no ip address
     isdn switch-type basic-net3
     isdn point-to-point-setup
     isdn incoming-voice voice
     isdn sending-complete
     isdn static-tei 0
    interface BRI0/1/1
     no ip address
     shutdown
     isdn switch-type basic-net3
     isdn point-to-point-setup
     isdn incoming-voice voice
     isdn sending-complete
     isdn static-tei 0
    interface Dot11Radio0/5/0
     no ip address
     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
     station-role root
     antenna receive right
     antenna transmit right
    interface Dot11Radio0/5/0.1
     encapsulation dot1Q 1 native
     bridge-group 1
     bridge-group 1 subscriber-loop-control
     bridge-group 1 spanning-disabled
     bridge-group 1 block-unknown-source
     no bridge-group 1 source-learning
     no bridge-group 1 unicast-flooding
    interface Dot11Radio0/5/0.100
     encapsulation dot1Q 100
     bridge-group 100
     bridge-group 100 subscriber-loop-control
     bridge-group 100 spanning-disabled
     bridge-group 100 block-unknown-source
     no bridge-group 100 source-learning
     no bridge-group 100 unicast-flooding
    interface Vlan1
     no ip address
     bridge-group 1
     bridge-group 1 spanning-disabled
    interface Vlan100
     no ip address
     bridge-group 100
     bridge-group 100 spanning-disabled
    interface BVI1
     description $FW_INSIDE$
     ip address 192.168.10.2 255.255.255.0
     ip access-group 102 in
     ip nat inside
     ip virtual-reassembly in
    interface BVI100
     description $FW_INSIDE$
     ip address 10.1.1.1 255.255.255.0
     ip access-group 103 in
     ip nat inside
     ip virtual-reassembly in
    ip forward-protocol nd
    ip http server
    ip http authentication local
    ip http secure-server
    ip http path flash:/gui
    ip dns server
    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.

  • Issue with instant ringback when using sip trunk to SP

    Hi all,
    We use CUCM 8.0.2.
    We have a SIP trunk to a SP connected via one of our Cisco 2911 routers configured as a CUBE.
    Cisco IOS Software, C2900 Software (C2900-UNIVERSALK9-M), Version 15.0(1)M3, RELEASE SOFTWARE (fc2)
    c2900-universalk9-mz.SPA.150-1.M3.bin
    Cisco CISCO2911/K9 (revision 1.0)
    Technology Package License Information for Module:'c2900'
    Technology Technology-package
                      Current       Type
    ipbase        ipbasek9      Permanent
    security      securityk9    Permanent
    uc              uck9            Permanent
    data           None            None
    We also have several ISDN lines that run out via various Cisco routers configured as H323 gateways.
    We use 7945 and CIPC for our phones.
    We're having an issue with calls going via the SIP trunk where we hear ringing instantly after dialling - but before the actual device at the other end starts ringing (considerable difference).
    Using the SIP trunk: If I make a call to my mobile phone - I hear ringing instantly - about 3 rings before my mobile phone actually starts ringing - undesireable.
    Using H323 gateway: If I make a call to my mobile phone - I hear silence for a bit - then ringing when the mobile starts ringing - desired.
    Using SIP trunk: If I make a call to a landline that is ready - it rings instantly for at least 1 ring - before the actual phone I'm calling starts ringing - undesireable.
    Using H323 gateway: There is a momentary pause before hearing ringing on my phone and the phone I dialled - desired.
    Using SIP trunk: If I make a call to a landline that is off-hook (with no call-waiting/etc.) - it rings once and then returns the busy signal (the worst issue) - undesireable.
    Using H323 gateway: There is a momentary pause before hearing busy signal - desired.
    Phone to phone internally (same network): Operates as expected (instantly rings locally and on the phone I'm calling). Between phones that utilise the SIP trunk and phones that utilise the H323 gateways within the same network - communication is instant and as expected.
    Any ideas why this happens and how to stop it?
    I want it to not ring until the situation is known and that it can provide the appropriate feedback (ringing/busy/etc.).
    Some possibly relevant config (note that there is a known bug with this IOS that meant I had to declare the codec in each dial-peer as the voice class would not work):
    voice service voip
    address-hiding
    mode border-element
    allow-connections sip to sip
    sip
      bind control source-interface GigabitEthernet0/0
      bind media source-interface GigabitEthernet0/0
      header-passing error-passthru
      early-offer forced
      midcall-signaling passthru
    interface GigabitEthernet0/0
    ip address x.x.x.x 255.255.255.252
    ip access-group acl.SIP-IN in
    no ip redirects
    no ip unreachables
    ip verify unicast reverse-path
    ip virtual-reassembly
    duplex full
    speed 100
    no cdp enable
    gateway
    timer receive-rtp 1200
    sip-ua
    connection-reuse
    gatekeeper
    shutdown
    dial-peer voice 1 voip
    description *** INBOUND CALLS FROM CARRIER ***
    translation-profile incoming SIPTRUNK-INCOMING
    session protocol sipv2
    incoming called-number #blah blah#
    dtmf-relay rtp-nte
    codec g711alaw
    ip qos dscp cs5 media
    no vad
    dial-peer voice 61 voip
    description **** WA, SA AND NT NUMBERS ****
    destination-pattern 0[8]........
    session protocol sipv2
    session target ipv4:<MY SP's SIP SERVER>
    incoming called-number 0[8]........
    dtmf-relay rtp-nte
    codec g711alaw
    ip qos dscp cs5 media
    no vad
    dial-peer voice 81 voip
    description **** MOBILE NUMBERS ****
    destination-pattern 0[4]........
    session protocol sipv2
    session target ipv4:<MY SP's SIP SERVER>
    incoming called-number 0[4]........
    dtmf-relay rtp-nte
    codec g711alaw
    ip qos dscp cs5 media
    no vad
    dial-peer voice 500 voip
    description *** INBOUND SIP TRUNK TO CUCM PUB ***
    translation-profile outgoing SIPTRUNK-CALLING-ADD-0
    preference 1
    destination-pattern 5[12]..
    session protocol sipv2
    session target ipv4:<OUR CUCM PUBLISHER IP>
    dtmf-relay rtp-nte
    codec g711alaw
    ip qos dscp cs5 media
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    Cheers,
    Brett

    I ended up resolving this issue as follows:
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    Now, I get the expected delay before hearing ringback.
    Solved!

  • Best Practice to Integrate CER with RedSky E911 Anywhere via SIP Trunk

    We are trying to integrate CER 9 with RedSky for V911 using a SIP trunk and need assistance with best practice and configuration. There is very little documentation regarding "best practice" for routing these calls to RedSky. This trunk will be handling the majority of our geographically dispersed company's 911  calls.
    My question is: should we use an IPsec tunnel for this? The only reference I found was this: http://www.cisco.com/c/en/us/solutions/collateral/enterprise-networks/virtual-office/deployment_guide_c07-636876.htmlm which recommends an IPsec tunnel for the SIP trunk to Intrado. I would think there are issues with an unsecure SIP trunk for 911 calls. Looking for advice or specifics on how to configure this. Does the SIP trunk require a CUBE or is a CUBE only required for the IPsec tunnel?
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    you can use Session Trace in RTMT to check who is disconnecting the call and why.

  • SIP trunk incoming and outgoing calls issue

    Hi Everyone,
    We recently installad a SIP trunk and terminated on CUBE and CUCM but we have issues on incoming and outgoing calls, When someone dial in from outside he keeps listening the dailing ring even after we pick up the phone and at the end the callers time exipres and call gets disconnected.
    For Dailing out, the dialed number rings and caller hear the dailing ring as well but if someone pick the phone it apprears that call is connected but no audio in it, dead air.
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    Tasneem

    Inbound call>>>>
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    Here we have your cube sending 180 ringing with "Require 100rel"..This was sent several times and your ITSP didnt respond probably because they do not support 100rel...(It is Huwaei after all, they do what they like)
    Sent:
    SIP/2.0 180 Ringing
    Via: SIP/2.0/UDP 10.205.20.50:5060;branch=z9hG4bKkpw18wp35hfw2c23h51ww6a8aT19871
    From: ;tag=sbc080552fph4hp-CC-25
    To: ;tag=256F3440-12C
    Date: Thu, 16 Jan 2014 13:31:34 GMT
    Call-ID: isbc6818c4kfhaa4ca1k5f8k1awsh52f1ccw@SoftX3000
    CSeq: 1 INVITE
    Require: 100rel
    RSeq: 2507
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    Allow-Events: telephone-event
    Remote-Party-ID: "TEST STC" ;party=called;screen=yes;privacy=off
    Contact:
    Record-Route:
    Server: Cisco-SIPGateway/IOS-15.2.4.M2
    Content-Length: 0
    AFter the CUBE didnt get any response, it then replied with Gateway Timeout...
    Jan 16 13:31:54.550: //31880/5A4406E48184/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 504 Gateway Timeout
    Via: SIP/2.0/UDP 10.205.20.50:5060;branch=z9hG4bKkpw18wp35hfw2c23h51ww6a8aT19871
    From: ;tag=sbc080552fph4hp-CC-25
    To: ;tag=256F3440-12C
    Call-ID: isbc6818c4kfhaa4ca1k5f8k1awsh52f1ccw@SoftX3000
    CSeq: 1 INVITE
    Reason: Q.850;cause=102
    Content-Length: 0
    I suggest you disable this parameter..and test again
    voice service voip
    sip
    rel1xx disable
    Please rate all useful posts
    "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"

  • 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#

  • Design Question Sip Trunk

    Hi,
    I have a requirement to move from a h323 environment to a SIP environment. I am looking for best practises especially around security. I have 2 CUCM servers (8.5) located in separate cities for redundancy. I also have 2 voice gateways which at the moment are h323 to the PSTN, each located at different cities.
    My  requirements are:
    1. Creat a sip trunk to the provider instaed of using PRI.
    2. If the Wan link fails on one gateway to provider, the alternate router in the other location should be able to receive the setup messages and if a user logs on via extension mobility, should be able to answer the call.
    Are there any simplified design docos about for this? I am hesitant to create a SIP trunk straight to the provider for security, so thinking of terminating the call on the voice routers with CUBE. I'm pretty sure this is run of the mill and would appreciate some input.
    Cheers!
    Pieter

    +5 to Chris..Always use CUBE...
    Here are more ideas..
    1. Create two sip trunks, first one to Cube 1, second to CUBE 2
        on your sip trunk set DTMF as no preference
    2. Assign the trunks to CUCM group with your two servers in it
    3. Configure route groups with Circular algorithm distribution (this way you have  load balancing on your two cubes)
    4. Configure dial-peers on your CUBE gateway to point in preferential order to your cucm servers
    5. Use dtmf-relay rtp-nte on your dial-peer (ensure you have sip as the protocol on your dial-peers)
    6. configure your codec selection properly on your dial-peers
    7. configure your region settings properly between your sip trunk and phones.
    8. Evaluate if you need xcoders, provide one if you do and ensure you set your region correctly between your xcoder and CUBE
    9. Is there voicemail involved? Ensure you set your region settings between cube and voicemail correctly, otherwise calls to voicemail may invoke xcoder (from experience)
    10. Provision adequate bandwidth for your calls. Once you move to sip, you loose the luxury of e1 channels. You are solely relying on Bandwidth. Ensure you have adequate bandwidth for your concurrent calls
    11. Provision QoS for your calls
    12. Is there fax involved? You need to think carefully on this one. What fax method do you want to use. T.38/pass through. Does your provider support T.38?
    13. Have a thorough test plan. Test call transfers, call on hold, call forward etc
    Just a few pointers..Careful planning and implementation is required for a successful SIP implementation

  • LYNC 2010 with Multiple Sites and Pools

    We have LYNC 2010 Enterprise Pool in North America with 5 Front End servers, 2 Edge Servers. No Director servers deployed. We want to enable Dial-In Conferencing feature for all users across the Globe (to avoid 3<sup>rd</sup> party conferencing
    cost…) and proposing 2 additional pools ( one in EMEA and another in APAC regions). I am looking for best practices to deploy multiple pools on multiple central sites.
    Do we need to deploy Director Servers? (Microsoft recommend Director, however is this mandatory for multiple pools?)
    How can we handle Simple URLs and DNS entries in multiple Edge pools (edge pool in each frontend pool)?
    Do we need to publish different ISA firewall rules for simple URLs and mobility?
    As we are not planning to replace PBX now, and only looking to provide dial-in Conferencing Bridge with SIP trunking, what is the best practice to provide site resilience in case one central site is down?
    Thanks in advance for your help…
    Tek-Nerd

    Hello.
    1. This is a recommendation and it's not mandatory even with multiples pools. The only point if you don't deploy director is that for the first connection, all sig wil go through one pool (the one behind the srv and dns record) and after this pool redirect
    the user to the good one (and also do the authentification) .Also director is made to avoid this type of load on one pool (connection from other pool and from edge authentification), but still not mandatory.
    2.Simple URL point only on one entry (or pool of RP) somewhere in the world (after there are some unofficial workaround) but don't forget that behind that there is the external web service URL (for each pool) and also an edge association to one pool. And
    the user is really using thoses URL and svc (edge associated to the pool as the ext web url)
    For the external web svc & edge association, you can create a local edge pool closer to your internal pool so for those user the media will be closer and don't go trhough the world. (exemple you add one emea pool, the media that come from internet goes
    through the emea edge and emea RP handling Lync external web services). But the simple urls still point to your main edge/RP pool.
    3. Like point before, not mandatory, recommendation if you don't want your media going around the world and coming back. Be carefull the association is still fixed (depending of the user's pool parameter), so if the user travel still using original edge/rp
    settings.
    4. You can create multiple trunk with lync so have backup in case of one is no more working. After this is hard to say what is the best, this is depending of your infrastructure.
    David

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