SIP Trunks - SME

Hello,
We are in stage of configuring SIP Trunk between 2 CUCM clusters (cluster A & cluster B ) via SME.
Can any body help how to configure incoming & outgoing calls on SME using SIP ? 
For Example:
For a call from user A (cluster A) to user B (cluster B)
User A from Cluster A will use Route Pattern for outgoing calls, what parameters need to be configure in SME to answer call coming from cluster A ?
Then for SME outgoing call towards cluster B, do we need to configure RP ? In cluster B what parameters need to be configure to answer call coming from SME ?
I am trying to understand the call leg step by step from one CUCM cluster to other through SME.
We have 3 Digits of extension in Cluster A & we have 4 Digits of extension in Cluster B.
Full numbers of ip phone in each cluster is 8 Digit & we want 8 digit calling pattern to call from cluster A to B & vice versa.
Thank in advance.
Regards,
Conie

Got it.
We have only one SME server at time & we are planning to use Inter-cluster trunk between clusters as backup incase if SME stop working for any malfunctioning reason.
For inter-cluster trunks we will have :
1- Inter-Cluster trunk in Cluster A point to Cluster B ?
2- Inter-Cluster trunk in Cluster B point to Cluster A ?
3- In cluster A RL, set inter-cluster trunk for cluster B as secondary ?
4- In cluster B RL, set inter-cluster trunk for cluster A as secondary ?
should i use same calling approach as we discussed above via Route Patterns only ?
As we have CUCM 8.6 in both clusters, we are planning to use H.323 Inter-Cluster (Non Gatekeeper) Trunk as backup.
And PSTN lines as backup for H.323 Inter-cluster.
Do you have any comments regarding our design ? are we on right track with this failover approach ?
Regards,
Conie
Thank you to being answering my questions appreciated your answers helped me alot.

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

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

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

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

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

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

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

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

  • 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?

  • Route pattern to SIP trunk problem

    Hello, I have a 2801 router that has been configured with CME and a working SIP connection to my local ISP.
    Tested with calls via CME so I know for sure that the SIP config and dial plan is fine on this gateway.
    Next I wanted to try out CUCM so I set up a CUCM 8.6 box that is connected to the 2801 router to use as it's SIP gateway.
    The only change I made to the gateway router config was to alter the "ip option 150" address so that the phones go to CUCM for their configs etc (which they do with no problems).
    Then I set up a SIP trunk in CUCM along with a route pattern which is to use the SIP trunk within the Gateway/Route list option.
    But when I make a call that matches this route pattern all I get is the intermittent beep message from the phone. I cannot route calls succesfully through it.
    I have checked network connectivity and all is fine. The IP address I specfied in CUCM for the SIP trunk is simply one of the interfaces on the 2801 router and it is definitley reachable.
    I also activated "debug ccsip all" on the 2801 gateway router but nothing appears. So it seems like the calls are not even reaching the 2801 gateway ?
    Is the problem possibly a conflit between CME on the gateway router and my CUCM ?
    Do I need to disable CME somehow on the gateway first ?  Or am I not doing something correct in the CUCM config ?
    Thank you kindly for any suggestions.
    ps. I have attached a couple of screenshots of my config.

    Hello, thanks for helping.
    I activated "debug voice ccapi inout" as well as "debug ccsip all" on the gateway but nothing showed up.
    Therefore I deduce the call is not even making it to across the SIP trunk into the gateway router ?
    As I am a newbie trying this out for the first time, it is guranteed to be something really simple.
    I have included my running config from the gateway router below..
    One addition I made was to add an incoming dial peer. That is "dial peer 5,  description CUCM SIP trunk".
    I set it up with a destination patter 2... to match my phone config on CUCM which have numbering in the 2000 range.
    Sorry, I got RTMT up and running but could not get any meaningful results from it. I need to learn up on that.
    I did however run a 'dialed number analysis' from CUCM direct and have attached the result. It seems the dialled number "99" is matching the route pattern OK.
    So why is it not then moving down the SIP trunk to my gateway and getting picked up by the incoming dial peer ?
    Thanks if you guys can offer any more help.
    version 15.1
    service timestamps debug datetime msec
    service timestamps log datetime msec
    no service password-encryption
    hostname Router
    boot-start-marker
    boot system flash:c2801-ipvoicek9-mz.151-2.T0a.bin
    boot-end-marker
    no aaa new-model
    clock timezone nzst 13 0
    dot11 syslog
    ip source-route
    ip dhcp pool DATA_SCOPE
       network 192.168.200.0 255.255.255.0
       default-router 192.168.200.1
       dns-server 8.8.8.8
    ip dhcp pool VOICE_SCOPE
       network 192.168.100.0 255.255.255.0
       default-router 192.168.100.1
       option 150 ip 192.168.2.115
    ip dhcp pool MGMT_SCOPE
       network 192.168.1.0 255.255.255.0
       default-router 192.168.1.99
    ip cef
    ip name-server 4.2.2.2
    no ipv6 cef
    multilink bundle-name authenticated
    voice class codec 1
    codec preference 1 g711alaw
    codec preference 2 g729r8
    codec preference 3 g711ulaw
    codec preference 4 ilbc
    voice translation-rule 1
    rule 1 /^9/ //
    voice translation-profile Strip9ToGetOut
    translate called 1
    voice-card 0
    crypto pki token default removal timeout 0
    crypto pki trustpoint TP-self-signed-2995340181
    enrollment selfsigned
    subject-name cn=IOS-Self-Signed-Certificate-2995340181
    revocation-check none
    crypto pki certificate chain TP-self-signed-2995340181
    certificate self-signed 01
      3082023E 308201A7 A0030201 02020101 300D0609 2A864886 F70D0101 04050030
      31312F30 2D060355 04031326 494F532D 53656C66 2D536967 6E65642D 43657274
      69666963 6174652D 32393935 33343031 3831301E 170D3733 30363034 31393534
      32305A17 0D323030 31303130 30303030 305A3031 312F302D 06035504 03132649
      4F532D53 656C662D 5369676E 65642D43 65727469 66696361 74652D32 39393533
      34303138 3130819F 300D0609 2A864886 F70D0101 01050003 818D0030 81890281
      8100C34D C8ECBB53 E01373A3 2E286B78 2D23042B 1C8588B1 A7861899 BA1C6860
      AE1D7868 2A59E3BC 54D0A457 8FFDE27F C09104E5 C7A429F3 74CD9DA8 4A980366
      675CC27C CDB94838 821CC05F 2C0AC2BC D882C132 6CAA1FA6 6DA740E4 562428B1
      12B741F1 A50C9246 4CC35EDA DEE1D038 3883BB35 A91ABF8B 483E4160 F5FA4B5A
      9A570203 010001A3 66306430 0F060355 1D130101 FF040530 030101FF 30110603
      551D1104 0A300882 06526F75 74657230 1F060355 1D230418 30168014 72119640
      F3396E1F E4168086 D31D8619 0D8337FF 301D0603 551D0E04 16041472 119640F3
      396E1FE4 168086D3 1D86190D 8337FF30 0D06092A 864886F7 0D010104 05000381
      81003B5A 29DE3A1E C5AB6092 E8D90650 C80752FC 0AAC93FD C5DE3D69 071B08FA
      D4013232 81CA07E7 15F90190 6A3AD6A0 1D05F0F2 13479568 888332A5 F81E2681
      7DA44095 4D11CFB7 CA79579A 8D95DE54 7B00173C E2C50573 A310C8C9 1487FEFC
      CE35B66E 9EF94CFA 8D6D6DCD ADC78132 2709F198 6DF2F0FA D80CC088 D0C4C7D1 080B
          quit
    license udi pid CISCO2801 sn FTX0947W07M
    username xxx privilege 15 password 0 xxx
    interface FastEthernet0/0
    ip address 192.168.3.50 255.255.255.0
    duplex auto
    speed auto
    interface FastEthernet0/1
    no ip address
    duplex auto
    speed auto
    interface FastEthernet0/1.2
    encapsulation dot1Q 2
    ip address 192.168.2.1 255.255.255.0
    interface FastEthernet0/1.99
    encapsulation dot1Q 99
    ip address 192.168.1.99 255.255.255.0
    interface FastEthernet0/1.100
    description voice_VLAN
    encapsulation dot1Q 100
    ip address 192.168.100.1 255.255.255.0
    interface FastEthernet0/1.200
    description data_VLAN
    encapsulation dot1Q 200
    ip address 192.168.200.1 255.255.255.0
    ip forward-protocol nd
    ip http server
    ip http authentication local
    ip http secure-server
    ip route 0.0.0.0 0.0.0.0 192.168.3.1
    logging esm config
    tftp-server flash:/phone/7940-7960/P00307020200.bin alias P00307020200.bin
    tftp-server flash:/phone/7940-7960/P00307020200.loads alias P00307020200.loads
    tftp-server flash:/phone/7940-7960/P00307020200.sb2 alias P00307020200.sb2
    tftp-server flash:/phone/7940-7960/P00307020200.sbn alias P00307020200.sbn
    control-plane
    mgcp fax t38 ecm
    dial-peer voice 1 voip
    description local_7_Digit_Calling
    translation-profile outgoing Strip9ToGetOut
    destination-pattern 9[2-9]......
    session protocol sipv2
    session target ipv4:203.184.16.2
    voice-class codec 1 
    dial-peer voice 2 voip
    description international_calling
    translation-profile outgoing Strip9ToGetOut
    destination-pattern 900T
    session protocol sipv2
    session target ipv4:203.184.16.2
    voice-class codec 1 
    dial-peer voice 3 voip
    description national_calling
    translation-profile outgoing Strip9ToGetOut
    destination-pattern 90[34679].......
    session protocol sipv2
    session target ipv4:203.184.16.2
    voice-class codec 1 
    dial-peer voice 4 voip
    translation-profile outgoing Strip9ToGetOut
    destination-pattern 90[34679].......
    dial-peer voice 5 voip
    description CUCM SIP trunk
    destination-pattern 2...
    session protocol sipv2
    session target ipv4:192.168.2.115
    voice-class codec 1 
    sip-ua
    authentication username xxxxxxxxxx password xxxxxxxx
    060
    telephony-service
    max-ephones 10
    max-dn 20
    ip source-address 192.168.1.99 port 2000
    load 7960-7940 P00307020200
    max-conferences 4 gain -6
    transfer-system full-consult
    create cnf-files version-stamp Jan 01 2002 00:00:00
    ephone-dn  1  dual-line
    number 1000
    name Lydia Francis
    ephone-dn  2  dual-line
    number 1001
    name Leah Francis
    ephone-dn  3  dual-line
    number 1002
    n
    ephone-dn  4  dual-line
    number 1003
    ephone  1
    mac-address C80A.A970.01DE
    type CIPC
    button  2:2
    ephone  2
    mac-address 000C.3070.8705
    button  1:1 2:15
    ephone  3
    mac-address 000C.8546.5954
    button  1:3 2:15
    line con 0
    logging synchronous
    line aux 0
    line vty 0 4
    privilege level 15
    login local
    transport input telnet ssh
    scheduler allocate 20000 1000
    ntp server 195.43.74.123
    end

  • Configuring Level3 SIP trunk with Lync 2013

    Hi, I ran into some issues trying to configure SIP trunk from Level 3 and I was hoping someone here can help. We have our mediation server collocated with FE and SIP traffic goes from public IP, port 5060 via NAT, to local IP on FE, port 5060.
    Level 3 provided us with one signaling IP and two RTP IPs.
    I tried multiple trunk configuration settings and I can see that when I'm placing a call from Lync to an outside number I'm getting INVITE from Level 3 signaling IP, the session is established, phone rings, but there is no audio on either side. There's also
    a METHOD NOT ALLOWED message coming from them, which doesn't tell me much about what's happening.
    If I call to a Level 3 DID (assigned to my Lync user account) there's also INVITE from their side, but later I receive a CANCEL from them due to idle session. The phone never rings.
    Questions:
    1) Does anyone have Level 3 SIP trunks configured and can share their Get-TrunkConfiguration settings? What settings should I have for encryption, refer, sessionTimer / RTCP, and others? Level 3 refuses to provide any additional information besides IPs.
    2) Do I understand this correctly that when configuring PSTN gateways in topology, one of the RTP IPs should be entered in the  "alternate media IP" field? We have SIP trunks from another provider (which work fine), and they only use one IP
    for everything, so I don't have any experience configuring separate SIP and media IPs with Lync.
    Thanks, and let me know if I should provide additional info.

    Hi,                                                              
    On Lync topology PSTN gateways interface, please check if you enter gateway listening port 5060 and enable TCP option.
    Please also check if you enable refer support on Lync Server Control Panel, if you enable it please uncheck it.
    You can compare the trunk configuration for Level 3 in the part “Sample Trunk Configuration for Level 3” in the link below with yours’, it is for Lync server 2010 but similar for Lync server 2013:
    http://blogs.technet.com/b/nexthop/archive/2013/04/10/configuring-lync-2010-server-to-work-with-level-3-sip-trunking-services.aspx
    Best Regards,
    Eason Huang
    Eason Huang
    TechNet Community Support

  • Best way to implement SIP Options Pings on a SIP Trunk

    I wanted to see if anyone had suggestions on the best way to configure SIP Options Pings.
    Typically I would configure them per dial-peer.  However, I really want to do it per destination IP address.  I do not want a SIP Options ping for every single dial-peer being sent out every X seconds.  
     Example:
    In my case I have 4 SIP trunks in the same CUBE.  Each pointing to a different destination IP.  There are 6 dial-peers per SIP trunk.  I really do not want 24 option pings going out every X seconds.  I guess I've never actually did a debug to see how many pings are going out at a time but I am assuming it sends one for each dial-peer or does it?
    If I am correct in my assumption, is there way to only send one ping per destination IP and if that single IP goes unresponsive then all 6 dial-peers go down?

    Hello,
    The OOD option ping is sent per dial-peer to the destination.
    Restrictions
    •The Cisco Unified Border Element OOD Options ping feature can only be configured at the VoIP Dial-peer level.
    •All dial peers start in an active (not busied out) state on a router boot or reboot.
    •If a dial-peer has both an outbound proxy and a session target configured, the OOD options ping is sent to the outbound proxy address first.
    •Though multiple dial-peers may point to the same SIP server IP address, an independent OOD options ping is sent for each dial-peer.
    •If a SIP server is configured as a DNS hostname, OOD Options pings are sent to all the returned addresses until a response is received.
    •Configuration for Cisco Unified Border Element OOD and TDM Gateway OOD are different, but can co-exist.
    //Suresh
    Please rate all the useful posts.

  • SIP trunking monitoring and usage report

    Hello all,
    Recently the IP telephony system has migrated to SIP trunking from PRI.   SIP trunking is running on CUBE 2900 serials.
    Could anyone advise the following?    much appreciate it.
    .  SIP trunking monitoring :  
       If MIB needed,   any brief steps to configure it on monitored device and monitoring system?
    . SIP trunking usage report :  
       Any software/application could generate the usage report?   and show the cocurrent calls?
    Thanks again,
    master002

    Thanks Nadeem for the reply,  looks Variphy Insight is the option?
    I have installed RTMT and was able to get the real time data instead of the usage report.  I also have CUOM -- operation manager, I haven't customized it to monitor SIP trunking yet.
    As my previous post mentioned,  any one could share the idea or experience on SIP trunking monitoring and usage report?  
    Thanks again,

  • Redirect SIP Trunk calls to FXO port

    Hi,
    This is the scenario. There are 3 branches, two of them are Cisco Call Manager Express and one of them is Elastix-based.
    So, as the image explains, the three branches have SIP trunks fully operational. The branches are in different cities, so the numbers structure changes. In city A it begins with 2, in B begins with 3 and in C begins with 4. Every POTS number is a 7 digit number (2XXXXXX, 3XXXXXX, 4XXXXXX). And every user, in every branch, have a 4 digit number beginning with the city code (2XXX, 3XXX, 4XXX).
    But, every time city A wants to make a call to a POTS number in city B, it goes across the A´s FXO line. So it charges a inter-city cost to the call.
    The client wants that every time a city A user wants to call a POTS number in city B, goes over the SIP trunk to city B and use the FXO on the city B call manager.
    I have made a pattern for city A. So, everytime the user dials 3XXXXXX, it does not use the city A´s FXO, but it goes to the branch in city B.
    What do I have to do now in branch B´s Call Manager Express to redirect that call to a local FXO?
    Thanks in advanced!
    Regards
    PS. There is  a diagram of the topology. Want to do what the red line is doing.

    In this situation I would do an answer-address based on ANI so you are specifically identifying your site A and then just piggy back off the local FXO out.
    So assuming you are sending just 4 digits over the SIP for each site:
    Dial-peer voice X voip
    answer-address "blah"
    protocol sipv2
    ...(whatever else you need to configure in these dots)
    At this point your CME at site B will take the call see that it is destined for a POTs line and it should send it out whatever local dial-peer you have setup for that site when they dial out to the PSTN locally.
    EDIT:
    Then again, you probably already have a general incoming dial-peer, the above design would just be specific for your site A and isn't really needed.

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

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

    you can use Session Trace in RTMT to check who is disconnecting the call and why.

  • Why do we need MTP in the SIP trunk for CVP warm transfers

    Hi All,
    Why do we need to enable MTP in SIP trunk between CUCM and CVP for CVP based trasnfers???
    Thanks in advance!!
    Regards,
    Thammaya Gupta K.

    I saw also in the CDR logs that the IP Phone media transport going to CUBE is in G711.And as well in the wireshark capture of the IP communicator that the CUCM invoke to use the g711 codec but as per ITSP logs they are now in the g729.
    @ Jamie If I un-tick the MTP point required in SIP trunk will make the call leg from IP Phone to CUBE g729 (w/o hw resource), I have also tried to use g729 preferred originating codec, but still the IP Phone is using g711.
    I have seen a documentation states:
    " To configure G.729 codecs for use with a SIP trunk, you must use a hardware MTP or transcoder that supports the G.729 codec." - I read this on the CUCM help page under configuring SIP trunk setting.
    Our ultimate goal is to use g729 without using HW MTP/ transcoder.
    IP Phone ->CUCM SIP Trunk ->CUBE-> ITSP

  • Cca xml export for SIP Trunk service providers, request or schema.

    Hi
    Please can I have the complete xml schema for SIP trunks so I can create a proper template?
    Exporting from CCA 3.2.2 gives me an incomplete and sometimes inaccurate xml file
    example: toll fraud protection  - I would like to add this in to the template - how
    SIP registration: If I set this to 120 on CCA and export - the exported template remains as 3600
    thanks
    Adam

    Hi Adam,
    Are you saying when you try to export your SIP trunk settings via CCA that your settings are incorrect in the xml file?  If so, I recommend opening a case to troubleshoot the issue as it sounds like a bug to me.

  • BE6000 and SIP trunk

    Hello!
    Now we use Cisco IP telephony based on Cisco CME. Cisco CME is installed on 2821 router with IOS C2800NM-ADVENTERPRISEK9-M, Version 12.4(24)T7. A SIP trunk is configured between CME router and telephony provider.
    We plan to upgrade our telephone system and begin using Cisco CBE6000. I study the documentation and in "Cisco Business Edition Unified Communications and Collaboration Solutions for Small and Midsized Companies", I find that for SIP trunk to works in CBE6000 a Cisco Integrated Services Router with Cisco Unified Border Element is required. Or other quote from this document "Public switched telephone network (PSTN) connections using SIP is available through any Cisco Integrated Services Router voice gateway".
    In this regard there is a question: what is the functions of this router when I make a trunk between CBE6000 and telephony provider SIP gateway and whether it is possible to use existing Cisco 2821 router as such router.
    Thanks.

    Hi Ayodeji,
    I too have a somewhat similar query.
    We have UC560 running currently at our company, the UC560 is connected to an ISDN modem from the network provider. We are now upgrading to BE6000. To connect to the ISDN E1/T1 line, we've ordered Cisco 2921 with a E1/T1 VWIC card (VWIC3-2MFT-T1/E1) which will connect to BE6000 and internal LAN. I just wanted to know what would I need to configure on 2921?
    Do I need to install CUBE or just configure it to connect to ISDN line?
    Many thanks for any help.
    regards,

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