SIP Trunk Questions

Integration with 3rd Party Presence service

2013-05-10T11:32:22Z

Is it possible to integrate CUCM with 3rd party presence server (not CUPS)? For example I wanted to integrate CUCM with IBM presence service or Oracle Communication Presence using SIP trunking. Do you see any major challenges in doing so? The required use cases are as follows: - CUCM should be able to publish presence of SIP clients - CUCM should be able to subscribe for buddy list both internal and external to CUCM - CUCM notfiies presence update upong receiving NOTIFY from Presence server - Presence server maintains registered buddylist. CUCM provides buddy list through register

RE: Cisco Call Recording Interface random error Reason: Q.850;cause=47

2012-11-26T21:40:15Z

This kind of error is usually seen when a transcoder, MTP or similar media-path helper resource is either not configured or has run out of available instances. Ensuring that DTMF signaling type and codec usage is unified and configured correctly may help reduce the need for such resources.

Cisco Call Recording Interface random error Reason: Q.850;cause=47

2012-11-26T20:51:38Z

Hello, One of our customers is having problem with Cisco Call Recording interface. Most of the calls getting recorded fine except in case of consulation transfer. In case of consulation transfer some calls (to the same extension) get recorded, other calls getting error Reason: Q.850;cause=47. Any suggestions, ideas? Thanks bad call Wireshark Message Header Via: SIP/2.0/UDP 10.32.60.12:5060;branch=z9hG4bK152da74fee0f0 From: "Rob Trefz Agent" ;tag=ae0b7800-6bc8-4225-972e-582b03635c76-44193253 To: ;tag=002f1860 Date: Mon, 19 Nov 2012 15:03:11 GMT Call-ID: [url=mailto:3acde700-aa14a2f-c1cb-c3c200a@10.32.60.12]3acde700-aa14a2f-c1cb-c3c200a@10.32.60.12[/url] User-Agent: Cisco-CUCM8.0 Max-Forwards: 70 P-Preferred-Identity: "Rob Trefz Agent" CSeq: 102 BYE Reason: Q.850;cause=47 Content-Length: 0 Good Call Wireshark Message Header Via: SIP/2.0/UDP 10.32.60.12:5060;branch=z9hG4bK152b64154c8fe From: "Rob Trefz Agent" ;tag=ae0b7800-6bc8-4225-972e-582b03635c76-44193217 To: ;tag=302d2e62 Date: Mon, 19 Nov 2012 14:56:44 GMT Call-ID: [url=mailto:54226380-aa148ac-c1c1-c3c200a@10.32.60.12]54226380-aa148ac-c1c1-c3c200a@10.32.60.12[/url] User-Agent: Cisco-CUCM8.0 Max-Forwards: 70 P-Preferred-Identity: "Rob Trefz Agent" CSeq: 102 BYE Content-Length: 0

How to monitor - alert SIP trunk state in UCM 8.6 and later

2012-10-16T21:00:41Z

Enabled SIP profile, Options Ping on the trunk. It works fine but I can't find this in RTMT to report on.. Anybody got this working? Thanks Kees

RE: Cisco CME and NET VX900 SIP trunk

2012-09-28T20:22:17Z

[quote=AAD VAN DER ZANDEN]Has anyone once achieved thsi? We are trying to connect the CME SIP trunk ( 12.4.22T advipservices ) on a 2811 to a VX900 We can call from SIP phone registered on VX to phone registered on cisco CME but not back so it looks like a dial-peer issue though I can't think of anything . There is no dial tone on SIP phone at CME ( SPA922 ) Any suggestions? Here is our config: voice service voip allow-connections sip to sip sip registrar server expires max 600 min 60 ! ! voice class codec 1 codec preference 1 g729br8 codec preference 2 g729r8 codec preference 3 g711alaw ! voice register global mode cme source-address xx.xx.xx.xx port 5060 max-dn 8 max-pool 8 authenticate register dialplan-pattern 2 690.... extension-length 4 tftp-path flash: create profile sync 0007272453516527 ! voice register dn 1 number 6902222 allow watch ! voice register pool 1 id mac [mac-address of client] type 7970 dtmf-relay sip-notify voice-class codec 1 username xxxx password xxxxx ! dial-peer voice 1 voip description ** SIP TRUNK ** destination-pattern 686.... voice-class codec 1 voice-class sip dtmf-relay force rtp-nte session protocol sipv2 session target ipv4:[IP of remote VX SIP server] dtmf-relay rtp-nte no vad ! telephony-service em logout 0:0 0:0 0:0 max-ephones 10 max-dn 20 ip source-address xx.xx.xx.xx port 2000 load 7960-7940 P00308000500 dialplan-pattern 1 690.... extension-length 4 max-conferences 8 gain -6 transfer-system full-consult create cnf-files version-stamp Jan 01 2002 00:00:00 ! [/quote] I had a problem like this also. My problem was the VX900. The CME IP source address was a loopback (192.168.20.1) and the CME router was the default gateway (192.168.1.1) of the network the VX was on. I could call from a PBX switch through the VX to the CME, but I couldn't call back. The problem was the VX was seeing the default gateway as the inbound calling party not the loopback address. To fix this I made the SIP inbound the default gaetway and everything worked fine. I guess you could also just make the CME IP source address the default gateway. I hope this helps someone, because this is the only place I could find anything about making a VX and a CME communicate. If you go to the CLI of the VX Enable->Trace all (Trace none, to turn off) and then call you can see what the issue is.

RE: BYE with error code QOS 65

2012-09-27T12:50:52Z

Below is the error message i got it for QOS 65, can you explain what it means? /////////////////////////// UCM Logs for error QOS 65 (attched the full logs) ////////////////////////////////////////////// (1,68,3807)/ci=23234797/ccbId=27359/scbId=0/getXCiscoViPRFallbackIDAndDTMFKey: Device type 8, Pstn Fallback is not enabled|1,100,57,1.41671^172.16.14.167^* 16:34:51.332 |//SIP/SIPCdpc(1,68,3807)/ci=23234797/ccbId=27359/scbId=0/appendReasonHdr: appendReasonHdr - Invalid Disconnect Cause(cause=65), No Reason Header Appended|1,100,57,1.41671^172.16.14.167^* 16:34:51.332 |//SIP/SIPCdpc(1,68,3807)/ci=23234797/ccbId=27359/scbId=0/addTransparencyInfo: Transparency info is NULL. Not attaching anything.|1,100,57,1.41671^172.16.14.167^* 16:34:51.332 |//SIP/SIPCdpc(1,68,3807)/ci=23234797/ccbId=27359/scbId=0/handleSIPConnectInd: releasing and changing state to disconnectSent|

BYE with error code QOS 65

2012-09-27T12:45:34Z

Hi, Instaed of ACK with SDP message, i am getting BYE with error code QOS 65 for my 200OK response. Can you help me to solve this issue? Below are the transations(attched uCM log file) /////////////////////////////////////////////// Message from UCM to recorder ////////////////////////////////////////////////////////////////////// INVITE sip:6565@172.16.14.167:5060 SIP/2.0 Via: SIP/2.0/TCP 192.168.10.185:5060;branch=z9hG4bK54af35a841d1;received=192.168.10.185 From: ;tag=27627~4cd274ee-1ce3-47c7-8736-342fd5b194ca-23235053 To: Date: Thu, 27 Sep 2012 11:41:44 GMT Call-ID: [url=mailto:4e7f5180-6413b78-203f-b90aa8c0@192.168.10.185]4e7f5180-6413b78-203f-b90aa8c0@192.168.10.185[/url] Supported: timer,resource-priority,replaces Min-SE: 1800 User-Agent: Cisco-CUCM8.5 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 Supported: Geolocation Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=500" Cisco-Guid: 1316966784-0000065536-0000003283-3104483520 Session-Expires: 1800 P-Preferred-Identity: Remote-Party-ID: ;party=calling;screen=no;privacy=off Contact: ;isFocus Max-Forwards: 70 Content-Length: 0 ///////////////////////////////////// Message to UCM from Recorder //////////////////////////////////////////// SIP/2.0 183 Session Progress Via: SIP/2.0/TCP 192.168.10.185:5060;branch=z9hG4bK54b256023c51;received=192.168.10.185 From: ;tag=27630~4cd274ee-1ce3-47c7-8736-342fd5b194ca-23235056 To: ;tag=108a49b5b8b835440c71cca3 Call-ID: [url=mailto:4e7f5180-6413b78-2041-b90aa8c0@192.168.10.185]4e7f5180-6413b78-2041-b90aa8c0@192.168.10.185[/url] CSeq: 101 INVITE Allow: INVITE,ACK,CANCEL,BYE,MESSAGE Content-Type: application/sdp Content-Length: 145 v=0 o=MyRecoder 12036987 1 IN IP4 172.16.14.167 s=SIP Call m=audio 21242 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=ptime:20 SIP/2.0 200 OK Via: SIP/2.0/TCP 192.168.10.185:5060;branch=z9hG4bK54b256023c51;received=192.168.10.185 From: ;tag=27630~4cd274ee-1ce3-47c7-8736-342fd5b194ca-23235056 To: ;tag=108a49b5b8b835440c71cca3 Call-ID: [url=mailto:4e7f5180-6413b78-2041-b90aa8c0@192.168.10.185]4e7f5180-6413b78-2041-b90aa8c0@192.168.10.185[/url] CSeq: 101 INVITE Allow: INVITE,ACK,CANCEL,BYE,MESSAGE Content-Type: application/sdp Content-Length: 145 v=0 o=MyRecoder 12036987 1 IN IP4 172.16.14.167 s=SIP Call m=audio 21242 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=ptime:20 ////////////////// Message from UCM to recorder /////////////////////////////////////////////// BYE sip:6565@172.16.14.167:5060 SIP/2.0 Via: SIP/2.0/TCP 192.168.10.185:5060;branch=z9hG4bK54b43ca6cc43;received=192.168.10.185 From: ;tag=27630~4cd274ee-1ce3-47c7-8736-342fd5b194ca-23235056 To: ;tag=108a49b5b8b835440c71cca3 Date: Thu, 27 Sep 2012 11:41:44 GMT Call-ID: [url=mailto:4e7f5180-6413b78-2041-b90aa8c0@192.168.10.185]4e7f5180-6413b78-2041-b90aa8c0@192.168.10.185[/url] User-Agent: Cisco-CUCM8.5 Max-Forwards: 70 P-Preferred-Identity: CSeq: 102 BYE Reason: Q.850;cause=65 Content-Length: 0

RE: SIP trunk - CUCM Rejection of SIP messages

2012-09-21T09:55:29Z

i`m sorry but could you show me a your "200 OK" messages? i have a problem that cm does not send to me a 101 ACK messages i don`t know what is the problem of my sip messages

SIP trunk does NOT response against 200 OK messages

2012-09-21T09:52:33Z

hello, i`m trying to get a rtp stream i got a INVITE message like below from cm =============== from CM ======================= INVITE sip:2999@172.17.17.15:5060 SIP/2.0 Via: SIP/2.0/TCP 172.17.17.11:5060;branch=z9hG4bK1b066218d23 From: ;tag=536~916fa243-9f70-4a7b-9599-bb6044937543-20942073 To: Date: Fri, 21 Sep 2012 07:23:49 GMT Call-ID: 48333600-5c11605-1b1-b1111ac@172.17.17.11 Supported: timer,resource-priority,replaces Min-SE: 1800 User-Agent: Cisco-CUCM8.6 Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY CSeq: 101 INVITE Expires: 180 Allow-Events: presence, kpml Supported: X-cisco-srtp-fallback Supported: Geolocation Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=500" Cisco-Guid: 1211315712-0000065536-0000000433-0185668012 Session-Expires: 1800 P-Preferred-Identity: Remote-Party-ID: ;party=calling;screen=no;privacy=off Contact: ;isFocus Max-Forwards: 70 Content-Length: 0 ============================================================================================== so, i generate a 200 OK message and send ============== sent messages ============================================ SIP/2.0 200 OK Via: SIP/2.0/TCP 172.17.17.11:5060;branch=z9hG4bK1b066218d23 From: ;tag=536~916fa243-9f70-4a7b-9599-bb6044937543-20942073 To: Call-ID: 48333600-5c11605-1b1-b1111ac@172.17.17.11 User-Agent: Cisco-CUCM8.6 Allow: INVITE,OPTIONS,INFO,BYE,CANCEL,ACK,PRACK,UPDATE,REFER,SUBSCRIBE,NOTIFY,PUBLISH CSeq: 101 INVITE Contact: ;isFocus Content-Type: application/sdp Content-Length: 154 v=0 o=PhoneUP_Recorder 5000 1 IN IP4 172.17.17.15 s=SIP Call c=IN IP4 172.17.17.15 t=0 0 m=audio 5000 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 =============================================================================== but, any response message does not come please, help me what is wrong....

Transfer to Gateway

2012-09-01T14:29:47Z

Is there any way to make it so that when phones regiestered to CUCM transfer calls from a call coming in on a gateway use REFER instead of INVITE/UPDATE? I am pretty sure at this point that this is impossible but would be great if there was a way.

Unexpected X.509 subject name for TLS SIP Trunk

2012-08-17T15:55:26Z

Hello, I am trying to configure a TLS connection to a SIP trunk for secure recording. I have generated a test certificate and uploaded it to the CUCM and added its subject name to a SIP Trunk Security Profile and assigned that profile to the SIP Trunk I am using setting the SRTP Allowed and "Whenusing both sRTP and TLS" for the secure traffic option. When I try to record a call the CUCM sends me an INVITE over a TLS connection (looking at Wireshark) but then after the 200OK etc. it sends a BYE straight away. Lpooking at the logs using RTMT I can see: SIPHandler(1,100,71,1) |SIPTcp(1,100,63,1) |1,100,17,70.3^*^* |[T:N-H:0,N:0,L:0,V:0,Z:0,D:0] connIdx= 74 --remoteIP=192.0.0.57 --remotePort = 5061 --X509SubjectName /CN=My Recording/ST=Someplace/C=UK/O=My Recorders Ltd --Cipher AES128-SHA --SubjectAltname = then: TLS InvalidX509NameInCertificate Error (reason 2), Rcvd=Red, Expected=O=My Recorders Ltd,C=UK,ST=Someplace,CN=My Recording then the CUCM rejects the call as the TLS connection is unsecure. The subject name is the same as the subject name in the CUCM Security->Certificates list When I extract the subject name from the certificate in OpenSSL I get: Subject: CN=My Recording, ST=Someplace, C=UK, O=My Recorders Ltd I don't understand why the certificate name is being displayed differently at different places in the logs. Why does the CUCM not like the subject name of the certificate when all parties are using the same self-signed test certificate?

Ringback for CUBE

2012-07-19T22:46:57Z

i am trying to have CUBE to generate ringback tones for inbound SIP calls on behalf of our SIPserver that is sitting behind CUBE and it does not generate Ringback and since CUBE does not natively generate tones, i wonder if someone can help me with an IVR TCL script to make CUBE play ringback tone twards ITSP as soon as it receives SIP 180 w/SDP message from the internal SIPserver Thanks in advance.

Autosvar: New Message from Abdul Rasheed in Cisco Unified Communications Ma

2012-07-16T19:45:38Z

Hi, I am on vacation and will be back Aug 6. For urgent matters call my office at +46108787000. Kind Regards, //Dan-Anders Höök

RE: SIP-trunk redundancy

2012-07-16T19:37:23Z

Hi Giggesh, we have the same issue with the fail over configuration. In our case the fail over will happen only after 1 minute. We are using UDP for SIP trunk outgoing traffic. Shall we reduce the Retry count for SIP INVITE for making the failover very fast? Thanks and Regards Abdul Rasheed

RE: SIP trunk - CUCM Rejection of SIP messages

2012-02-28T22:51:45Z

thanks Abdul Rasheed for your reply, now it is working, and the rtp streams came to my recorder pc. but my problem is that i do not know how to save these rtp streams as a file [wav or mp3]., so if you know away to do please tell me to study it. thanks again for your help

RE: SIP trunk - CUCM Rejection of SIP messages

2012-02-28T17:42:12Z

Dear Ahmed, You have missed out a few messages during your SIP negotitations. You need to go through the SIP trunk messaging guide available in the CDN website to implement those messages. regards Rasheed

RE: SIP trunk - CUCM Rejection of SIP messages

2012-02-24T13:43:47Z

Hello Abdul Rasheed; i have the same bye message with the same reason, but my case is 1- SIP Trunk send me the invitations 2- my sip server response with 200 ok 3- SIP Trunk sent to my ACK 101 then he sent to me bye message and he cotinue in sending it. i read about that my 200 ok response for the invitation should have the SDP and codecs, and if these codecs are not compatable with cucm you will get that bye message. my case is 1- i created SIP Trunk and give it my pc ip. 2- i run my sip application on my pc . 3- i got two invitations for the call on ip phone with automatic recording 4- i sent 200 ok for these two invitations 5- cucm sent me ack 101 6- cucm continue sending me bye message . please if you ses that i missed any thing here, like should i configure any thing in network or in my sip SAILFIN server or in CUCM configurations, tell me please, Thanks,

RE: SIP trunk - CUCM Rejection of SIP messages

2012-02-19T19:52:03Z

Hello Everybody , For the recording using BIB and SIP trunk, the rejection of INVITE messages with a BYE of reason "Bearer capability not implemented" what does this stands for ? THis is not happening always, but at sometimes only. Please let me know if anybody has got the same error and the possible reason . it will be a great help for me Regards Rasheed

RE: SIP trunk - CUCM Rejection of SIP messages

2012-02-09T09:48:19Z

Hello, can anybody tell me some reasons for the above error. Why does a SIP message get a Q.850 cause =65 (bearer capability not implemented) error? Regards Rasheed

RE: CUCM sends INVITE after 900 sec

2012-01-21T00:20:48Z

cause=86 looks to be Network Timeout, so I agree the SIP session refresh mechanim mentioned above is likely the root cause. I guess you can either modify the recorder side to interop with the way UCM does this refresh, or modify the timer so that in practice it never occurs (as above.)