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

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    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.

  • 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

  • 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.
    Our call flow is as 
    IP Phones => CUCM --->SIPTRUNK--->CUBE=>SIPTRUNK=>SP
    I have attached the config for CUBE and debug ccsip messages output for both incoming and outgoing calls.
    Please if some help in sorting out this issue, Thanks in Advance
    Tasneem

    Inbound call>>>>
    The reason you are experiencing this is that your CUBE is requesting PRACK and your provider is not responding to it..
    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"

  • 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.

  • Video only enabled when call is initiated from one direction across SIP Trunk

    wonder If anyone can shed some light on this.
    I have an issue between two cucm clusters, tied together with a SIP trunk. 
    If we dial from Australia to the US there is two way video and audio.  If the US calls Australia, there is only audio.   I have run a test call from the US through VLT and have found the following SDP's  (see below). When The US make a video enabled call to australia the message "Video is not available, Remote party has video off" on the US phone screen.
    Both clusters have the SIP trunk set up with the same codec settings and video bandwidth between reqions and locations.  the SIP trunk is configured pretty much stock standard and identical at both ends, yet the SDP seem to want to negotiate different Video Parameters  (again see SDP's below). CUCM in australia is 10.61.2.82.
    what other settings can I check to get video to work when calls get initiated from either direction,...................
    both phones are SIP 8941's, again audio is no problem in both directions.
    =======this is from the phone in Australia to the CUCM in australia phone IP 10.61.4.112======================================
    45870304.002 |09:02:07.941 |AppInfo  |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.61.4.112 on port 34271 index 53563 with 2089 bytes:
    [344530309,NET]
    SIP/2.0 200 OK
    Via: SIP/2.0/TCP 10.61.2.82:5060;branch=z9hG4bKe0103892bbb75
    From: "Anonymous" <sip:[email protected]>;tag=109791678~1b5af941-cea2-4a00-a0bd-15a532224d7d-59374526
    To: <sip:[email protected]>;tag=5057a887bfdd550c0d321a20-7f843426
    Call-ID: [email protected]
    Date: Wed, 29 Apr 2015 23:02:07 GMT
    CSeq: 101 INVITE
    Server: Cisco-CP8941/9.4.2
    Contact: <sip:[email protected]:34271;transport=tcp>;video
    Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
    Remote-Party-ID: "Dennis Mink - 33935" <sip:[email protected]>;party=called;id-type=subscriber;privacy=off;screen=yes
    Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-7.0.0,X-cisco-xsi-8.5.1
    Allow-Events: kpml,dialog
    Recv-Info: conference
    Recv-Info: x-cisco-conference
    Content-Length: 966
    Content-Type: application/sdp
    Content-Disposition: session;handling=optional
    v=0
    o=Cisco-SIPUA 28123 0 IN IP4 10.61.4.112
    s=SIP Call
    t=0 0
    m=audio 16736 RTP/AVP 0 8 18 102 9 116 101
    c=IN IP4 10.61.4.112
    a=trafficclass:conversational.audio.avconf.aq:admitted
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:102 L16/16000
    a=rtpmap:9 G722/8000
    a=rtpmap:116 iLBC/8000
    a=fmtp:116 mode=20
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=sendrecv
    m=video 16738 RTP/AVP 126 97
    c=IN IP4 10.61.4.112
    b=TIAS:2000000 
    a=trafficclass:conversational.video.avconf.aq:admitted   <----this is missing from US SDP
    a=rtpmap:126 H264/90000
    a=fmtp:126 profile-level-id=428014;packetization-mode=1;level-asymmetry-allowed=1;max-mbps=36000;max-fs=1200;max-rcmd-nalu-size=1300
    a=imageattr:126 send * recv [x=640,y=480]
    a=rtpmap:97 H264/90000
    a=fmtp:97 profile-level-id=428014;packetization-mode=0;level-asymmetry-allowed=1;max-mbps=36000;max-fs=1200
    a=imageattr:97 send * recv [x=640,y=480]
    a=rtcp-fb:* ccm tmmbr
    a=sendrecv
    ============below is coming from the US (phone IP is 10.1.109.81)================
    04/30/2015 09:02:08.169 Send 10.61.4.112 SIP ACK bfa99a00-541162ed-71da57-52023d0a NotAvail
    SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.61.4.112 on port 34271 index 53563 
    [344530326,NET]
    ACK sip:[email protected]:34271;transport=tcp SIP/2.0
    Via: SIP/2.0/TCP 10.61.2.82:5060;branch=z9hG4bKe010481f320b08
    From: "Anonymous" <sip:[email protected]>;tag=109791678~1b5af941-cea2-4a00-a0bd-15a532224d7d-59374526
    To: <sip:[email protected]>;tag=5057a887bfdd550c0d321a20-7f843426
    Date: Wed, 29 Apr 2015 23:02:05 GMT
    Call-ID: [email protected]
    User-Agent: Cisco-CUCM10.0
    Max-Forwards: 70
    CSeq: 101 ACK
    Allow-Events: presence
    Content-Type: application/sdp
    Content-Length: 456
    SDP Message
    ====================================================
    v=0
    o=CiscoSystemsCCM-SIP 109791678 1 IN IP4 10.61.2.82
    s=SIP Call
    c=IN IP4 10.1.109.81
    b=TIAS:8000
    b=AS:8
    t=0 0
    m=audio 16412 RTP/AVP 18 101
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=trafficclass:conversational.audio.aq:admitted   <---what does this do here, and how?
    m=video 0 RTP/SAVP 31 34 96 97      <-----------port 0. why?
    a=rtpmap:31 H261/90000
    a=rtpmap:34 H263/90000
    a=rtpmap:96 H263-1998/90000
    a=rtpmap:97 H264/90000
    a=content:main
    a=inactive

    Hi Dennis,
    On US phone SDP media attribute is inactive.
    a=rtpmap:97 H264/90000
    a=content:main
    a=inactive
    Are you sure that audio works ? Can you please share all the SIP messages of both the scenarios.
    Thanks
    Manish

  • UC560 SIP Trunk, IOS Jabber, MWI

    Has anyone successfully been able to configure complete functionality of these on the UC560?  Making IOS Jabber work, breaks sip trunk configuration and the MWI.  I do currently have a case opened with TAC, but they have yet to find the solution.  Currently the sip trunk is working as is the MWI, the IOS Jabber phone is registered, but can not make or receive phone calls. 

    Good morning
    Hi John, thanks for using our forum, my name is Johnnatan and I am part of the Small business Support community. You can post your question in "Small Business Voice and Conferencing>SBCS - UC500" so you can have more feedback on your case, more users will see it there. You can move your post using  the actions panel on the right.
    Greetings,
    Johnnatan Rodriguez Miranda.
    Cisco Network Support Engineer.

  • SIP Trunk to 2811

    Hi,
    We have Cisco 2811 with CCME and also act as H.323 gateway for branch VOIP traffic. Cisco IP phones are registered in the router and there's a PRI E1 connection to PSTN cloud from the router. One Service provider offers SIP trunk.
    Q1. Is it possible to configure SIP trunk in 2811 and enable calls between H.323, SIP & PSTN?
    Q2. If possible, what are the pre-requisities ?
              * IOS
              * Additional Resources
    Q3. Highly appriciate if someone can provide  sample configuration with simple dial peers to enable voice routing between SIP, H.323 & PSTN cloud?
    Thank you
    Lakmal

    Thanks for the prompt support.
    Existing setup: refer attached diagram & router configs
    As explained earlier Location-1 is connected with PSTN PRI e1 and 2 mbps data link.
    Call flow:
    Location 01:
    E1 Number series: 038-42911XX
    Local Extensions: 33XX
    PSTN Access Code: 9 followed by 7 digits for local and 10 digits for national calls
    Location 01 can directly dial location02 by using 4-digit extension (62[345]x)
    Location 02:
    Local Extensions: 62[345]X
    PSTN Access C ode: 9 followed by 7 digits for local and 10 digits for national calls. These calls will connect to PSTN cloud through Location 01.
    Location 02 can directly dial location-1 by using 4-digit extension (33xx)
    In the proposed setup, the SIP trunk colored in red is introduced.  Could you please help me to configure the SIP trunk in the 2811 router and route calls through it? For example assume we need to route PSTN calls through SIP trunk using access code 8 which is currently doing by access code 9. Also need to terminate incoming calls from SIP trunk to 3300 extension. Please check whether the current IOS is capable?
    If you require any further details, please let me know.
    Thank you

  • Calls from Sip Trunk to UC540 and then to CUE returned ** Service Unavailable**

    Hi to all
    i have something strange here and i need your assistance
    Call Flow:
    Sip trunk-->UC540--> CUE
    When calls coming to UC540 from outside and then going to cue then we send back service unavailable.I made a translation and i sent directly the incoming calls to CUE
    The same behavior is also if i send the calls to dummy number and then from there set forward all to voice mail.
    Incoming voicemail is working fine
    Incoming calls to phones also ok
    Uc540: 8.6
    CUE: 8.6.5
    A number: 99999999
    B number: 22777777
    Voice Mail Number:111
    Attached is the trace
    i see that we hit the correct dial peers .
    I have enable only trancoder since MTP is not register ( don't know why , but i don't think also that is necessary..
    voice service voip
     ip address trusted list
      ipv4 172.16.80.0 255.255.255.0
      ipv4 172.16.81.0 255.255.255.0
     allow-connections sip to sip
     supplementary-service h450.12
     no supplementary-service sip moved-temporarily
     no supplementary-service sip refer
     supplementary-service media-renegotiate
     sip
      no update-callerid
    dial-peer voice 1000 voip
     description **SIP TRUNK**
     translation-profile incoming SIP-INCOMING
     translation-profile outgoing SIP-OUTGOING
     destination-pattern 9T
     modem passthrough nse codec g711alaw
     session protocol sipv2
     session target sip-server
     incoming called-number .T
     voice-class codec 2  
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     fax-relay ecm disable
     no fax-relay sg3-to-g3
     fax rate 9600
     fax protocol pass-through g711alaw
     no vad
    dial-peer voice 2001 voip
     description ** cue voicemail pilot number **
     destination-pattern 111
     b2bua
     session protocol sipv2
     session target ipv4:10.1.10.1
     incoming called-number 111
     no voice-class sip outbound-proxy   
     dtmf-relay sip-notify
     codec g711ulaw
     no vad
    Regards
    chrysostomos

    Hi
    Interface                  IP-Address      OK? Method Status                Protocol
    FastEthernet0/0            unassigned      YES NVRAM  up                    up
    FastEthernet0/0.10         192.168.0.10    YES DHCP   up                    up   ----> For internet
    FastEthernet0/0.20         10.151.5.130    YES NVRAM  up                    up  ------> For sip trunk
    In0/0                      10.1.10.2       YES unset  up                    up    --------> default gw for cue
    Vlan1                      unassigned      YES unset  up                    up
    Vlan100                    unassigned      YES unset  up                    up
    Vlan200                    unassigned      YES unset  up                    down
    Vlan300                    unassigned      YES unset  up                    down
    NVI0                       10.1.10.2       YES unset  up                    up
    BVI1                       192.168.20.1    YES NVRAM  up                    up
    BVI100                     10.1.1.1        YES NVRAM  up                    up   ---------> ip for cme
    Loopback0                  10.1.10.2       YES NVRAM  up                    up   ------> default gw for cue
    dial-peer voice 2001 voip
     description ** cue voicemail pilot number **
     destination-pattern 111
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     no voice-class sip outbound-proxy
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     voice-class sip bind media source-interface BVI100
     dtmf-relay sip-notify
     codec g711ulaw
     no vad
    interface FastEthernet0/0.10
     description **FOR INTERNET**
     encapsulation dot1Q 10
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     ip access-group 105 in
     ip nat outside
     ip inspect SDM_LOW out
     ip virtual-reassembly in
    interface FastEthernet0/0.20
     description **FOR SIP TRUNK WITH ISP**
     encapsulation dot1Q 20
     ip address 10.151.5.130 255.255.255.240
    ip route 10.1.10.1 255.255.255.255 Integrated-Service-Engine0/0
    ping 10.1.10.1 source bvi100
    Type escape sequence to abort.
    Sending 5, 100-byte ICMP Echos to 10.1.10.1, timeout is 2 seconds:
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    Anything to advice?

  • 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
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     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
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     duplex auto
     speed auto
    interface Serial0/3/0
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     shutdown
     clock rate 2000000
    interface Serial0/3/1
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     clock rate 250000
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     shutdown
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     network 10.0.30.0 0.0.0.255
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    no ip http secure-server
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    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
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     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
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     codec g711ulaw
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     max-dn 48
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     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
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     transport preferred none
     transport input all
     transport output lat pad telnet rlogin lapb-ta mop udptn v120 ssh
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     speed 115200
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    scheduler allocate 20000 1000
    end
    Branch_SIP#show debug
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    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#

  • PROBLEM WITH FORWARDING ALL - SIP TRUNK

    Hello,
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    I have this scenario: PSTN - SIP GW - CUCM6.1 - SIP TRUNK - CUM8.6
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    Content-Length: 0
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    Hi,
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    From: <sip:[email protected]>;tag=e1d37fe6-cb7b-46e7-a868-6fe81d6bb391-40468009
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    Content-Length: 0
    User-Agent: Cisco-CUCM6.1
    To: <sip:[email protected]>
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    Expires: 180
    Call-ID: [email protected]
    Via: SIP/2.0/TCP 145.245.235.201:5060;branch=z9hG4bK3863af2fe80489
    CSeq: 101 INVITE
    Session-Expires:  1800
    Max-Forwards: 67
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    To: <sip:[email protected]>;tag=DA70FE78-21AE
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Reason: Q.850;cause=100
    Content-Length: 0
    The problem is because in the redirect header, the comma "," is not a valid parameter so in the alerting name I removed the comma after the surname (i.e. "Elodie, Mary" to "Elodie Mary" and now is working.
    We will change the alerting name by the moment and I will also investigate if there is a parameter to just not divert this name because is not needed this info in forwarded calls.
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