Calling Party No Display in SIP Trunk

Hi All,
We are integrating the CUCM & CME via SIP trunk where we facing the problem in Caller ID.
In CUCM & CME we have 4 digit extension. It will transformed to 8 digit Enterprise number while calling from/to cucm & cme. Once the call answered it displays the 4 digit extension of called party instead of 8 digit enterprise no.
Hope it's updating the Phone caller Id based on "P-Asserted Identity" in SIP messages . My requirements is it should display 8 digit no & Name.
Flow :   Phone --> CUCM --> SIP Trunk --> CME --> Phone.
Any Suggestions !!!

Hi All,
I just tried with SIP Profiles on the dial peer to remove the Remote-party ID.
When call initiated from CME to CUCM , "Remote Party Id"  is removed in Invite. When CUCM reply with 180 ringing it contains "Remote Party Id" but sip profile not having any effect.
This to avoid the displaying CUCM internal number series in CME instead of DID.
Can you advice how it can be achieve ?
Configuration
voice class sip-profiles 10
request ANY sip-header Remote-Party_ID remove
response ANY sip-header Remote-Party-ID remove
dial-peer voice 100 voip
destination-pattern 91T
session protocol sipv2
session target ipv4:<ip address>
voice-class sip profiles 10
Call Flow
SCCP Phone -- CME --- SIP -- CUCM --- SCCP Phone.
Thanks in Advance

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    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##
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    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
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    access-list 103 permit udp 10.1.10.0 0.0.0.3 any eq 2000
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    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
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    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
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     member local-plus
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     member national
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     member local
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     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
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     direct-inward-dial
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     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.

  • Route SIP REFER to SIP Trunk based on DN

    Cisco UCM 9 is connected to a third-party PBX over SIP Trunk. Third-party PBX sends a SIP REFER message to Cisco UCM to call a DN on the third-party PBX. Cisco UCM responds with SIP 404 Not Found as it does not recognize the DN of the third-party PBX.
    How do I configure Cisco Unified Communication Manager 9 to route this call back out over the SIP Trunk to the third-party PBX based on the DN (Not IP)?
    Cisco UCM contains a route pattern 53xxx to route to SIP_Trunk_3rdParty.
    Third-party PBX contains a SIP Proxy and Call Server. The call should route to the SIP Proxy IP. The SIP REFER contains "Refer-To" 53xxx@ThirdPartyCallServerIP
    I added a SIP Route Pattern on CUCM to route calls for ThirdPartyCallServerIP to SIP_Trunk_3rdParty. This works in routing the call to ThirdPartyCallServerIP, however I need the call to route to 53xxx@ThirdPartySIPproxyIP for it to be successful.
    Direct calls from CUCM to ThirdParty PBX 53XXX@ThirdPartySIPproxyIP are successful. SIP REFER coming into CUCM to request CUCM to call ThirdParty fail.
    Any ideas on what configuration on CUCM I could try to get CUCM to route the call to thrid-party based on the SIP REFER?

    Thanks for the reply Vivek.
    Partitions:
         -  ThirdPartyPBX
         -  CiscoEndpoints
    Calling Search Space: "ThirdParty_Cisco" contain both of the above partitions.
    Route Pattern 531XX and 80965 are assigned to Route Partition "ThirdPartyPBX"
    Cisco UCM Main site phones are in CSS "ThirdParty_Cisco" and DN is in Route Partition "CiscoEndpoints". DN is in CSS "ThirdParty_Cisco".
    Trunk "SIP_Trunk_3rdParty"  - Inbound and Outbound Calls are in CSS "ThirdParty_Cisco".
    Trunk SIP information has "Rerouting CSS", "Out-of-Dialog Refer CSS", and Subscribe CSS as "ThirdParty_Cisco".
    Cisco continues to respond to with SIP 404 not found. CUCM does not seem to match the SIP refer to the CSS or Route partition with with 531XX route pattern.
    The SIP Refer is coming from DN 80965 over the SIP Trunk from the Third-party PBX.
    Perhaps I'm missing something in my CSS config?
    Any other method for CUCM to match SIP Refer to a Route Pattern?

  • + Globalization and SIP Trunks

    I'm trying to add the +E.164 number to Missed Call Field.  On a SIP Trunk, I know I don't have support for ISDN TON (Subscriber, National, International), so when I push the Calling Party Number to a Transformation how do I ensure I tag Local, National, and International Calling Party Number correctly?  Configuration examples for those who have done this would be appreciated.

    Hello Sam,
    After a lot of research on my personal Lab, I was able to achieve Globalization in SIP Trunks. This is how I did:
    I created two partitions, CALLING-XFORM-DP-PT, and CALLING-XFORM-TRK-PT.
    One is for the Device Pool, and the other one is for the SIP Trunk. Create the CSSs and add your partitions > CALLING-XFORM-DP-CSS with CALLING-XFORM-DP-PT inside and CALLING-XFORM-DP-CSS with CALLING-XFORM-TRK-PT inside.
    Now create two Device Pools, one for the Trunk and one for your Phones.
    PHONES-DP and TRUNK-DP were the Device Pools I created. I am trying to be as specific as possible so everybody understands, so bear with me. :)
    Now, put the CSSs accordingly. Go to PHONES-DP "Device Mobility Related Information" field and put CALLING-XFORM-DP-CSS there. Save and reset.
    On TRUNK-DP you are not require to do anything, of course there is an "unless", but let's stick with your main concern, otherwise this post will be bigger than it will be.
    Go to the Trunk itself and add the CALLING-XFORM-TRK-CSS to the "Incoming Calling Party Settings" field. Uncheck the "Use Device Pool CSS" and leave "Strip Digits" and "Prefix" fields in its defaults. Save and Reset.
    Now, create your Calling Party Transformation Patterns. I will explain to you using the Brazilian national standard.
    For Local Calls, here in Sao Paulo, we receive 11 which is the area code and 8 or 9 digits after that for local number. So, I created the Calling Party Xform mask as follows:
    1 - 11.9XXXXXXXX > discard digit instruction: Predot > Prefix +5511
    2 - 11.[2-9]XXXXXXX > discard digit instruction: Predot > Prefix +5511
    I didn't change the TON or the Plan because it doesn't matter. SIP doesn't recognize these fields.
    Now, same thing for national calls. We receive the area code and 8 or 9 digits. Since we already have specific Patterns for Local calls, it is ok to have a more generic pattern for other area codes
    1 - [1-9][1-9]9XXXXXXXX > No Discard Digit Instruction > Prefix +55
    For international calls, our SP always sends 00 in front of the number. Knowing that, it was easy to simply add the following pattern.
    1 - 00.! > discard digit instruction: Predot > Prefix +
    Now, you create the Calling Party Transformation Patterns for Phone Device Pools. Achieving Localization as well.
    Local Numbers
    1 - \+5511.9XXXXXXXX > discard digit instruction: Predot > No need to prefix anything
    2 - \+5511.[2-9]XXXXXXX > discard digit instruction: Predot > No need to prefix anything
    National Numbers
    3 - \+55.[1-9][1-9]9XXXXXXXX > discard digit instruction: Predot > Prefix 0
    4 - \+55.[1-9][1-9][2-9]XXXXXXX > discard digit instruction: Predot > Prefix 0
    (in Brazil we used to have the numbers delivered to us as 0 + area code + number. That Prefix is just to achieve localization for us)
    There is no need to do anything for International Numbers at this level, because if the number is different from +55, there is nothing to be done, just receive the +E164 format.
    After all that you will be able to even do Plus Dialing. If you need any help setting it up, please let me know.
    Don't forget that if your provider sends you the digits in a way that you can't do much on CUCM, try to identify the numbers by adding a translation pattern in your Trunk (if it is Cisco, of course) before the numbers enter on CUCM. Then use the Calling Party Transformation Patterns to strip off any identification you gave for the inbound numbers.
    I hope this helps.
    Please rate useful posts.

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    Hi,
    No as you said you have already created the Trunk between the Sonus and Lync you can use the same trunk. but in the Sonus you have to configure the outbound routes from which Trunk you have to send the calls for the new Trunk provider or the Level3.
    check this
    https://support.sonus.net/display/UXDOC41/SIP+Trunking+Between+SIP+Border+Elements
    Whenever you see a helpful reply, click on Vote As Helpful & click on Mark As Answer if a post answers your question.

  • 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"

  • Third Party Phone over SIP Trunk with CUCM 9.x

    Hi all,
    I have a problem where my Third Party SIP phones wont go over the SIP trunk configured in my CUCM 9.x cluster. My Cisco phones work fine and goes out the trunk. I have noticed a distinct difference in wireshark with the invite packets from Third Party SIP phones and the Cisco ones.
    I have configured the SIP trunk in CUCM with the following route pattern (60.!#)and configured it with associated group and list. Heres the differense between the invite packets from Cisco and Third Party phones.
    Cisco Phone: INVITE sip.60xxxx%23@ipadress
    Third Party SIP Phone:  INVITE sip:[email protected]
    It seems the Cisco phones gets some extra configured the Third Party ones dont...
    Thanks in advance for any help.
    //Per

    Thanks for the answer
    Yeah i have DNS configured and i have the trunk pointed to a domain destination SRV record and like i said it works fine when calling from a Cisco phone. I tried changing the domain to an ip address but same result. I also changed the Plycom phone from being registered towards the domain of CUCM to an IP adress of CUCM and then the SIP INVITE messages in wireshark began to look kinda the same expet for the "%23" section but it still dont work.
    When i look at the Real Time Data in RTMT the orig and final called from the cisco phone has stripped the 60 and forwared the rest of the number towards the correct domain for the SIP trunk.
    When looking at the data from the Polycom phone the orig and final called data still contains the 60 prefix part and the called device name field is empty.  The termination Cause Code is that the number requested is Unallocated/Unassigned..
    In other words something is missing to get CUCM to strip 60 from the Polycom phones dialed number and send it towards the SIP trunk like it does when the Cisco phones call it.
    Unfortunatley i dont have the meens to attach the trace...
    Thanks again for any help/advice
    With regards, Per.

  • 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

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

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

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

  • Outbound Call Failure - SIP Trunk

    All phones are unable to dial a single target number on the PSTN.  The symptom is that it rings once and goes fast busy.
    The call flow is:
    Phone >>> CUCM >>> CUBE >>> Verizon SIP Trunk >>> PSTN >>> Target Number
    As seen in the CUBE debug ccsip messages, the CUBE receives a "SIP/2.0 480 Temporarily unavailable" message.  debug ccsip messages, dial-peer and voice class information follows:
    Received:
    INVITE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/TCP 192.168.106.11:5060;branch=z9hG4bK6c4bb65f140c8
    From: "" <sip:[email protected]>;tag=4037968~a812c08a-f7f0-43a7-a92c-e5ac2a38867c-65745615
    To: <sip:[email protected]>
    Date: Wed, 18 Dec 2013 21:48:27 GMT
    Call-ID: [email protected]
    Supported: timer,resource-priority,replaces
    Min-SE:  1800
    User-Agent: Cisco-CUCM8.6
    Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
    CSeq: 101 INVITE
    Expires: 180
    Allow-Events: presence, kpml
    Supported: X-cisco-srtp-fallback,X-cisco-original-called
    Call-Info: <sip:192.168.106.11:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
    Cisco-Guid: 0520523008-0000065536-0000067523-0191539392
    Session-Expires:  1800
    P-Asserted-Identity: "" <sip:[email protected]>
    Remote-Party-ID: "" <sip:[email protected]>;party=calling;screen=yes;privacy=off
    Contact: <sip:[email protected]:5060;transport=tcp>
    Max-Forwards: 70
    Content-Type: application/sdp
    Content-Length: 390
    v=0
    o=CiscoSystemsCCM-SIP 4037968 1 IN IP4 192.168.106.11
    s=SIP Call
    c=IN IP4 10.139.64.171
    b=TIAS:64000
    b=AS:64
    t=0 0
    m=audio 30688 RTP/AVP 0 8 116 18 101
    a=rtpmap:0 PCMU/8000
    a=ptime:20
    a=rtpmap:8 PCMA/8000
    a=ptime:20
    a=rtpmap:116 iLBC/8000
    a=ptime:20
    a=maxptime:60
    a=fmtp:116 mode=20
    a=rtpmap:18 G729/8000
    a=ptime:20
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
    Sent:
    INVITE sip:[email protected]:5073 SIP/2.0
    Via: SIP/2.0/UDP 10.139.64.52:5060;branch=z9hG4bK636B313D7
    Remote-Party-ID: "" <sip:[email protected]>;party=calling;screen=yes;privacy=off
    From: "" <sip:[email protected]>;tag=78FC5414-198D
    To: <sip:[email protected]>
    Date: Wed, 18 Dec 2013 21:40:10 GMT
    Call-ID: [email protected]
    Supported: 100rel,timer,resource-priority,replaces,sdp-anat
    Min-SE:  1800
    Cisco-Guid: 0520523008-0000065536-0000067523-0191539392
    User-Agent: Cisco-SIPGateway/IOS-12.x
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    CSeq: 101 INVITE
    Timestamp: 1387402810
    Contact: <sip:[email protected]:5060>
    Expires: 180
    Allow-Events: telephone-event
    Max-Forwards: 69
    Session-Expires:  1800
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 348
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 4778 3356 IN IP4 10.139.64.52
    s=SIP Call
    c=IN IP4 10.139.64.52
    t=0 0
    m=audio 23372 RTP/AVP 0 8 116 18 101
    c=IN IP4 10.139.64.52
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:116 iLBC/8000
    a=fmtp:116
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=yes
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
    079120: Dec 18 2013 16:40:10.008: //314738/1F068D000001/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 100 Trying
    Via: SIP/2.0/TCP 192.168.106.11:5060;branch=z9hG4bK6c4bb65f140c8
    From: "" <sip:[email protected]>;tag=4037968~a812c08a-f7f0-43a7-a92c-e5ac2a38867c-65745615
    To: <sip:[email protected]>
    Date: Wed, 18 Dec 2013 21:40:09 GMT
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Allow-Events: telephone-event
    Server: Cisco-SIPGateway/IOS-12.x
    Content-Length: 0
    +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
    079121: Dec 18 2013 16:40:10.080: //314739/1F068D000001/SIP/Msg/ccsipDisplayMsg:
    Received:
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 10.139.64.52:5060;branch=z9hG4bK636B313D7
    From: "" <sip:[email protected]>;tag=78FC5414-198D
    To: <sip:[email protected]>
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Timestamp: 1387402810
    +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
    079122: Dec 18 2013 16:40:11.176: //314739/1F068D000001/SIP/Msg/ccsipDisplayMsg:
    Received:
    SIP/2.0 180 Ringing
    Via: SIP/2.0/UDP 10.139.64.52:5060;branch=z9hG4bK636B313D7
    From: "" <sip:[email protected]>;tag=78FC5414-198D
    To: <sip:[email protected]>;tag=182903799-1387403308449
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Timestamp: 1387402810
    Supported:
    Contact: <sip:[email protected]:5073;transport=udp>
    Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
    Content-Length: 0
    +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
    Sent:
    SIP/2.0 180 Ringing
    Via: SIP/2.0/TCP 192.168.106.11:5060;branch=z9hG4bK6c4bb65f140c8
    From: "" <sip:[email protected]>;tag=4037968~a812c08a-f7f0-43a7-a92c-e5ac2a38867c-65745615
    To: <sip:[email protected]>;tag=78FC58A8-1B6B
    Date: Wed, 18 Dec 2013 21:40:09 GMT
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    Allow-Events: telephone-event
    Remote-Party-ID: <sip:[email protected]>;party=called;screen=no;privacy=off
    Contact: <sip:[email protected]:5060;transport=tcp>
    Server: Cisco-SIPGateway/IOS-12.x
    Content-Length: 0
    +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
    079128: Dec 18 2013 16:40:12.384: //314739/1F068D000001/SIP/Msg/ccsipDisplayMsg:
    Received:
    SIP/2.0 480 Temporarily unavailable
    Via: SIP/2.0/UDP 10.139.64.52:5060;branch=z9hG4bK636B313D7
    From: "" <sip:[email protected]>;tag=78FC5414-198D
    To: <sip:[email protected]>;tag=182903799-1387403308449
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Timestamp: 1387402810
    Content-Length: 0
    +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
    Sent:
    SIP/2.0 480 Temporarily Not Available
    Via: SIP/2.0/TCP 192.168.106.11:5060;branch=z9hG4bK6c4bb65f140c8
    From: "" <sip:[email protected]>;tag=4037968~a812c08a-f7f0-43a7-a92c-e5ac2a38867c-65745615
    To: <sip:[email protected]>;tag=78FC58A8-1B6B
    Date: Wed, 18 Dec 2013 21:40:09 GMT
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Allow-Events: telephone-event
    Server: Cisco-SIPGateway/IOS-12.x
    Reason: Q.850;cause=18
    Content-Length: 0
    +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
    079146: Dec 18 2013 16:40:12.388: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    ACK sip:[email protected]:5073 SIP/2.0
    Via: SIP/2.0/UDP 10.139.64.52:5060;branch=z9hG4bK636B313D7
    From: "" <sip:[email protected]>;tag=78FC5414-198D
    To: <sip:[email protected]>;tag=182903799-1387403308449
    Date: Wed, 18 Dec 2013 21:40:10 GMT
    Call-ID: [email protected]
    Max-Forwards: 70
    CSeq: 101 ACK
    Allow-Events: telephone-event
    Content-Length: 0
    +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
    079147: Dec 18 2013 16:40:12.404: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    ACK sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/TCP 192.168.106.11:5060;branch=z9hG4bK6c4bb65f140c8
    From: "" <sip:[email protected]>;tag=4037968~a812c08a-f7f0-43a7-a92c-e5ac2a38867c-65745615
    To: <sip:[email protected]>;tag=78FC58A8-1B6B
    Date: Wed, 18 Dec 2013 21:48:27 GMT
    Call-ID: [email protected]
    Max-Forwards: 70
    CSeq: 101 ACK
    Allow-Events: presence, kpml
    Content-Length: 0
    +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
    dial-peer voice 9100 voip
    description inboubd dial-peer for outgoing calls from CUCM (11D)
    preference 1
    session protocol sipv2
    incoming called-number ^1..........$
    voice-class codec 10
    dtmf-relay rtp-nte digit-drop
    ip qos dscp cs5 media
    ip qos dscp cs3 signaling
    no vad 
    outbound DP
    dial-peer voice 8100 voip
    description outbound dial-peer for outgoing calls to Verizon (11D)
    destination-pattern ^1..........$
    session protocol sipv2
    session target sip-server
    voice-class codec 10
    voice-class sip dtmf-relay force rtp-nte
    voice-class sip early-offer forced
    dtmf-relay rtp-nte digit-drop
    ip qos dscp cs5 media
    ip qos dscp cs3 signaling
    no vad
    no supplementary-service sip moved-temporarily
    no supplementary-service sip refer
    +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
    voice class codec 1
    codec preference 1 g729r8
    codec preference 2 g711ulaw
    voice class codec 10
    codec preference 1 transparent
    voice class codec 2
    codec preference 1 g711ulaw
    codec preference 2 g722-64

    I created the new voice class and mapped it to the outgoing dial-peer 8100. The call was then successful. 
    See new voice class:
    #sh run | be voice class codec 11
    voice class codec 11
    codec preference 1 g711ulaw
    codec preference 2 g711alaw
    codec preference 3 g729r8
    See revised dial-peer 8100:
    dial-peer voice 8100 voip
    description outbound dial-peer for outgoing calls to Verizon (11D)
    destination-pattern ^1..........$
    session protocol sipv2
    session target sip-server
    voice-class codec 11
    voice-class sip dtmf-relay force rtp-nte
    voice-class sip early-offer forced
    dtmf-relay rtp-nte digit-drop
    ip qos dscp cs5 media
    ip qos dscp cs3 signaling
    no vad
    no supplementary-service sip moved-temporarily
    no supplementary-service sip refer
    My only remaining question is why did the CUBE invite NOT include the m line for g729r8? 
    ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
    See the ccapi inout snippet showing the hit with dial-peer 8100:
    ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
    080927: Dec 19 2013 15:27:57.810: //316459/32C4F8800001/CCAPI/ccCallSetupRequest:
       Destination=, Calling IE Present=TRUE, Mode=0,
       Outgoing Dial-peer=8100, Params=0x2B912E08, Progress Indication=NULL(0)
    +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
    See the debug ccsip messages output showing original CUCM invite received by CUBE with 5 a line references:
    +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
    080907: Dec 19 2013 15:27:57.806: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    INVITE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/TCP 192.168.106.11:5060;branch=z9hG4bK6d715c9c6ad1
    From: "XXXXXXXXXX" ;tag=4077346~a812c08a-f7f0-43a7-a92c-e5ac2a38867c-65761788
    To:
    Date: Thu, 19 Dec 2013 20:36:14 GMT
    Call-ID: [email protected]
    Supported: timer,resource-priority,replaces
    Min-SE:  1800
    User-Agent: Cisco-CUCM8.6
    Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
    CSeq: 101 INVITE
    Expires: 180
    Allow-Events: presence, kpml
    Supported: X-cisco-srtp-fallback,X-cisco-original-called
    Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=500"
    Cisco-Guid: 0851769472-0000065536-0000068412-0191539392
    Session-Expires:  1800
    P-Asserted-Identity: "XXXXXXXXXX"
    Remote-Party-ID: "XXXXXXX" ;party=calling;screen=yes;privacy=off
    Contact:
    Max-Forwards: 70
    Content-Type: application/sdp
    Content-Length: 464
    v=0
    o=CiscoSystemsCCM-SIP 4077346 1 IN IP4 192.168.106.11
    s=SIP Call
    c=IN IP4 10.139.64.52
    b=TIAS:64000
    b=AS:64
    t=0 0
    m=audio 26738 RTP/AVP 0 8 116 116 18 101
    a=rtpmap:0 PCMU/8000
    a=ptime:20
    a=rtpmap:8 PCMA/8000
    a=ptime:20
    a=rtpmap:116 iLBC/8000
    a=ptime:20
    a=maxptime:60
    a=fmtp:116 mode=20
    a=rtpmap:116 iLBC/8000
    a=ptime:30
    a=maxptime:60
    a=fmtp:116 mode=30
    a=rtpmap:18 G729/8000
    a=ptime:20
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
    See ccsip messages output showing CUBE sending invite to Verizon:
    +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
    Sent:
    INVITE sip:[email protected]:5073 SIP/2.0
    Via: SIP/2.0/UDP 10.139.64.52:5060;branch=z9hG4bK63F9C611
    Remote-Party-ID: "David Callahan" ;party=calling;screen=yes;privacy=off
    From: "David Callahan" ;tag=7DE0957C-1CAB
    To:
    Date: Thu, 19 Dec 2013 20:27:57 GMT
    Call-ID: [email protected]
    Supported: 100rel,timer,resource-priority,replaces,sdp-anat
    Min-SE:  1800
    Cisco-Guid: 0851769472-0000065536-0000068412-0191539392
    User-Agent: Cisco-SIPGateway/IOS-12.x
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    CSeq: 101 INVITE
    Timestamp: 1387484877
    Contact:
    Expires: 180
    Allow-Events: telephone-event
    Max-Forwards: 69
    Session-Expires:  1800
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 259
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 6966 4178 IN IP4 10.139.64.52
    s=SIP Call
    c=IN IP4 10.139.64.52
    t=0 0
    m=audio 32502 RTP/AVP 0 8 101
    c=IN IP4 10.139.64.52
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15

  • SIP Trunk not accepting inbound calls

    I have a CME setup using Engin as a SIP provider
    I am able to dial out with no issue, however my inbound calls do not work, they divert to the Engin voicemail
    My SIP registration is OK and the number is configured as the primary DN on one of my phones
    Router#sh sip-ua register status
    Line                             peer       expires(sec) registered P-Associ-URI
    ================================ ========== ============ ========== ============
    038682XXXX                       -1         1124         yes       
    101                              20001      45           no        
    102                              20003      18           no        
    103                              20005      45           no        
    104                              20006      45           no        
    I do see the call come in if I debug the dial peer, but it only seems to match an outgoing dp
    I am seeing a couple of disconnect cause codes that I cant seem to find any relavent information on in the CCSIP debugs
    Router#
    Sep  8 18:15:15.009: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SESSION_REFRESH_REQ
    Sep  8 18:15:15.009: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: dir:2, method:102, resp_code:0, container:4F947560
    Sep  8 18:15:15.009: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLPrintTDContainer: Peer-Event: E_STSL_PASS_ST_PARAMS, SE Value:1800, SE Refresher:uas, Min-SE Value:1800, flags:2001
    Sep  8 18:15:15.017: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SESSION_REFRESH_RESP
    Sep  8 18:15:15.017: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: dir:1, method:102, resp_code:100, container:4F947DF8
    Sep  8 18:15:15.025: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_DISCONNECT
    Sep  8 18:15:15.025: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SESSION_REFRESH_RESP
    Sep  8 18:15:15.025: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: dir:1, method:102, resp_code:403, container:4F947B38
    Sep  8 18:15:15.061: //156/83D9548A803F/SIP/Call/sipSPICallInfo:
    The Call Setup Information is:
    Call Control Block (CCB) : 0x4C39C570
    State of The Call        : STATE_DEAD
    TCP Sockets Used         : NO
    Calling Number           : 0417XXXXXX
    Called Number            : 038682XXXX
    Source IP Address (Sig  ): 211.30.48.136
    Destn SIP Req Addr:Port  : 203.161.164.69:5060
    Destn SIP Resp Addr:Port : 203.161.164.69:5060
    Destination Name         : 203.161.164.69
    Sep  8 18:15:15.061: //156/83D9548A803F/SIP/Call/sipSPIMediaCallInfo:
    Number of Media Streams: 1
    Media Stream             : 1
    Negotiated Codec         : g711ulaw
    Negotiated Codec Bytes   : 160
    Nego. Codec payload      : 0 (tx), 0 (rx)
    Negotiated Dtmf-relay    : 0
    Dtmf-relay Payload       : 0 (tx), 0 (rx)
    Source IP Address (Media): 211.30.48.136
    Source IP Port    (Media): 17768
    Destn  IP Address (Media): 203.161.164.69
    Destn  IP Port    (Media): 18314
    Orig Destn IP Address:Port (Media): [ - ]:0
    Sep  8 18:15:15.061: //156/83D9548A803F/SIP/Call/sipSPICallInfo:
    Disconnect Cause (CC)    : 21
    Disconnect Cause (SIP)   : 403
    Any Ideas
    Doug

    Hi Tapan,
    Firstly the topology is as follows
    ISP/VOIP provider - Internet - Cable modem - 2800 CME router - IP Phone
    The VM is provided by the ISP
    debug ccsip messages
    Sep  9 10:21:41.488: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    INVITE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 203.161.164.69:5060;branch=z9hG4bKv0hlbo2030i1c85sm7b0.1
    From: "0417XXXXXX"[email protected];user=phone>;tag=SD1uttd01-353510938-1315527701453-
    To: "Doug Goding"[email protected]>
    Call-ID: SD1uttd01-c27fdbf3f1de4ad1f0f2e1342b210494-au418e3
    CSeq: 633854439 INVITE
    Contact:
    Supported: 100rel,timer
    Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
    Accept: multipart/mixed,application/media_control+xml,application/sdp
    Min-SE: 60
    Session-Expires: 1800;refresher=uas
    Max-Forwards: 9
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 317
    v=0
    o=BroadWorks 18275729 1 IN IP4 203.161.164.69
    s=-
    c=IN IP4 203.161.164.69
    t=0 0
    m=audio 18128 RTP/AVP 18 8 0 101
    c=IN IP4 203.161.164.69
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=yes
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=bsoft: 1 image udptl t38
    Sep  9 10:21:41.508: //4375/867C64EB8067/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 203.161.164.69:5060;branch=z9hG4bKv0hlbo2030i1c85sm7b0.1
    From: "0417XXXXXX"[email protected];user=phone>;tag=SD1uttd01-353510938-1315527701453-
    To: "Doug Goding"[email protected]>
    Date: Fri, 09 Sep 2011 00:21:41 GMT
    Call-ID: SD1uttd01-c27fdbf3f1de4ad1f0f2e1342b210494-au418e3
    CSeq: 633854439 INVITE
    Allow-Events: telephone-event
    Server: Cisco-SIPGateway/IOS-12.x
    Content-Length: 0
    Sep  9 10:21:41.516: //4375/867C64EB8067/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 403 Forbidden
    Via: SIP/2.0/UDP 203.161.164.69:5060;branch=z9hG4bKv0hlbo2030i1c85sm7b0.1
    From: "0417XXXXXX"[email protected];user=phone>;tag=SD1uttd01-353510938-1315527701453-
    To: "Doug Goding"[email protected]>;tag=39373A4-586
    Date: Fri, 09 Sep 2011 00:21:41 GMT
    Call-ID: SD1uttd01-c27fdbf3f1de4ad1f0f2e1342b210494-au418e3
    CSeq: 633854439 INVITE
    Allow-Events: telephone-event
    Server: Cisco-SIPGateway/IOS-12.x
    Reason: Q.850;cause=21
    Content-Length: 0
    Sep  9 10:21:41.544: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    ACK sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 203.161.164.69:5060;branch=z9hG4bKv0hlbo2030i1c85sm7b0.1
    CSeq: 633854439 ACK
    From: "0417XXXXXX"[email protected];user=phone>;tag=SD1uttd01-353510938-1315527701453-
    To: "Doug Goding"[email protected]>;tag=39373A4-586
    Call-ID: SD1uttd01-c27fdbf3f1de4ad1f0f2e1342b210494-au418e3
    Max-Forwards: 9
    Content-Length: 0
    Voice Config
    Router#
    voice service voip
    allow-connections h323 to sip
    allow-connections sip to h323
    allow-connections sip to sip
    sip
      registrar server expires max 3600 min 3600
      localhost dns:mel.byo.engin.com.au
      no call service stop
    voice class codec 1
    codec preference 1 g711ulaw
    voice translation-rule 10
    rule 1 /^0/ //
    voice translation-rule 11
    rule 1 /^.*/ /0386821234/
    voice translation-profile PSTN_Outgoing
    translate calling 11
    voice-card 0
    dsp services dspfarm
    mgcp profile default
    sccp local Vlan100
    sccp ccm 10.1.100.1 identifier 1 version 7.0
    sccp
    sccp ccm group 1
    bind interface Vlan100
    associate ccm 1 priority 1
    associate profile 1 register confdsp
    dspfarm profile 1 conference 
    codec g711ulaw
    codec g711alaw
    codec g729ar8
    codec g729abr8
    codec g729r8
    codec g729br8
    maximum sessions 4
    associate application SCCP
    dial-peer voice 99 voip
    translation-profile outgoing PSTN_Outgoing
    destination-pattern .T
    session protocol sipv2
    session target sip-server
    dtmf-relay rtp-nte
    codec g711ulaw
    no vad
    dial-peer voice 100 voip
    session protocol sipv2
    session target dns:mel.byo.engin.com.au
    incoming called-number 0386821234
    dtmf-relay sip-notify
    codec g711ulaw
    no vad
    dial-peer voice 110 voip
    description ** Incoming call from SIP trunk (Generic SIP Trunk Provider) **
    session protocol sipv2
    session target dns:mel.byo.engin.com.au
    incoming called-number .%
    dtmf-relay rtp-nte
    no vad
    dial-peer voice 90 voip
    description Melbourne 03 Numbers
    translation-profile outgoing PSTN_Outgoing
    destination-pattern [89].......
    session protocol sipv2
    session target sip-server
    dtmf-relay rtp-nte
    codec g711ulaw
    no vad
    dial-peer voice 91 voip
    description National Numbers
    translation-profile outgoing PSTN_Outgoing
    destination-pattern 0[278]........
    session protocol sipv2
    session target sip-server
    dtmf-relay rtp-nte
    codec g711ulaw
    no vad
    dial-peer voice 92 voip
    description Vic/Tas 03 numbers
    translation-profile outgoing PSTN_Outgoing
    destination-pattern [56].......
    session protocol sipv2
    session target sip-server
    dtmf-relay rtp-nte
    codec g711ulaw
    no vad
    dial-peer voice 93 voip
    description Mobile numbers
    translation-profile outgoing PSTN_Outgoing
    destination-pattern 04........
    session protocol sipv2
    session target sip-server
    dtmf-relay rtp-nte
    codec g711ulaw
    no vad
    dial-peer voice 94 voip
    description 13XXXX numbers
    translation-profile outgoing PSTN_Outgoing
    destination-pattern 13[1-9]...
    session protocol sipv2
    session target sip-server
    dtmf-relay rtp-nte
    codec g711ulaw
    no vad
    dial-peer voice 96 voip
    description 1300/1800 numbers
    translation-profile outgoing PSTN_Outgoing
    destination-pattern 1[38]00......
    session protocol sipv2
    session target sip-server
    dtmf-relay rtp-nte
    codec g711ulaw
    no vad
    dial-peer voice 98 voip
    description Emergency 000
    translation-profile outgoing PSTN_Outgoing
    destination-pattern 000
    session protocol sipv2
    session target sip-server
    dtmf-relay rtp-nte
    codec g711ulaw
    no vad
    sip-ua
    credentials username 0386821234 password 7 XXXX realm voice.mibroadband.com.au
    authentication username 0386821234 password 7 XXXX
    nat symmetric role active
    nat symmetric check-media-src
    no remote-party-id
    retry invite 2
    retry register 10
    timers connect 100
    mwi-server dns:mel.byo.engin.com.au expires 3600 port 5060 transport udp unsolicited
    registrar dns:mel.byo.engin.com.au expires 3600
    sip-server dns:mel.byo.engin.com.au
    connection-reuse
    telephony-service
    sdspfarm conference mute-on #1 mute-off #2
    sdspfarm units 2
    sdspfarm tag 1 confdsp
    conference hardware
    max-ephones 42
    max-dn 144
    ip source-address 10.1.100.1 port 2000
    calling-number initiator
    service phone videoCapability 1
    service phone displayOnDuration 00:01
    service phone displayOnTime 08:30
    service phone displayOffTime 17:30
    service phone displayIdleTimeout 00:01
    service phone displayOnWhenIncomingCall 1
    system message Cisco CME
    load 7941 SCCP41.8-4-2S
    load 7942 SCCP42.8-4-2S
    load 7945 SCCP45.8-4-2S
    load 7961 SCCP41.8-4-2S
    load 7962 SCCP42.8-4-2S
    load 7965 SCCP45.8-4-2S
    load ata ATA030204SCCP090202A
    time-zone 48
    date-format dd-mm-yy
    voicemail 90125200
    mwi relay
    max-conferences 8 gain -6
    call-forward pattern .T
    call-forward system redirecting-expanded
    moh music-on-hold.au
    web admin system name cisco secret 5 $1$d8/H$glhLiCCWXmFSUp6BtwGho0
    dn-webedit
    time-webedit
    transfer-system full-consult
    transfer-pattern 0.T
    create cnf-files version-stamp 7960 Jul 06 2011 10:32:45
    ephone-dn  1  dual-line
    number 038682XXXX
    label 101
    name 7965
    mwi sip
    ephone-dn  2  dual-line
    number 102
    label 102
    name 7941
    ephone-dn  3  dual-line
    number 103
    label 103
    name 7920
    ephone  1
    device-security-mode none
    video
    mac-address 0023.5EB8.6E4E
    type 7965
    button  1:2 2:1
    ephone  3
    device-security-mode none
    mac-address 0019.0633.A933
    max-calls-per-button 2
    type 7920
    button  1:3
    ephone  10
    device-security-mode none
    mac-address 0019.E7B7.BAB3
    max-calls-per-button 2
    type ata
    button  1:1

  • SIP Trunk Question - Outbound Calls Fail

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    Hi
    909 is what I meant to dial as that is the help desk for the telco. 
    I tried mobile numbers as well getting the same error codes. And international numbers. 
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