01-29-2014 10:16 PM - edited 03-18-2019 11:19 AM
Hi,
Yesterday I configured my third party sip phone which is yealink in this case on cucm and successfully registered it with cucm, despite of registration i have some calling issue in this phone. I am able to make outbound calls from this phone to any other phone however issue is related to inbound calls.I tried calling its DN from anywhere but call disconnect after sometime. Also didnt get any proper sip session trace in RTMT. Kindly suggest some step to sortout this issue.
Thanks
01-29-2014 10:41 PM
Does the call actually get set up with audio both ways? How much time does it take before the call is disconnected? Can you capture detailed callmanager traces for a test call
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a0080094e89.shtml
HTH
Manish
01-30-2014 02:26 AM
Dear Manish,
Call normally dicsonnected after 30-40 sec with termination code 102 in session trace. PFB SDI trace with 5030 is Thirdparty sip phone and 5033 is c7945. Looking forward for your suggestion.
CallingPartyNumber=5033
|DialingPartition=
|DialingPattern=5030
|FullyQualifiedCalledPartyNumber=5030
|DialingPatternRegularExpression=(5030)
|DialingWhere=
|PatternType=Enterprise
|PotentialMatches=NoPotentialMatchesExist
|DialingSdlProcessId=(0,0,0)
|PretransformDigitString=5030
|PretransformTagsList=SUBSCRIBER
|PretransformPositionalMatchList=5030
|CollectedDigits=5030
|UnconsumedDigits=
|TagsList=SUBSCRIBER
|PositionalMatchList=5030
|VoiceMailbox=
|VoiceMailCallingSearchSpace=PT-LHR-LOCAL:PT-Local:Unityvmpt:PT-F6-Local:PT-ISL-LOCAL:PT-KHI-LOCAL:PT_Operator_LHR:PT_Operator_KHI:PT_Operator_ISL
|VoiceMailPilotNumber=7103
|RouteBlockFlag=RouteThisPattern
|RouteBlockCause=0
|AlertingName=Syed Ahmer
|UnicodeDisplayName=Syed Ahmer
|DisplayNameLocale=1
|OverlapSendingFlagEnabled=0
12:17:38.028 |//SIP/SIPUdp/wait_SdlSPISignal: Outgoing SIP UDP message to 172.16.200.21:[5062]:
[23928282,NET]
INVITE sip:5030@172.16.200.21:5062 SIP/2.0
Via: SIP/2.0/UDP 10.100.200.11:5060;branch=z9hG4bK1ca0cc6e317649
From: "Syed Ahmer" <5033>;tag=8787406~039e2a80-8561-4586-8954-d01ed2aa12c8-2462119185033>
To: <5030>5030>
Date: Thu, 30 Jan 2014 07:17:38 GMT
Call-ID: 84e24a80-2e91fc72-1a6940-bc8640a@10.100.200.11
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
Send-Info: conference, x-cisco-conference
Alert-Info: <>>
Contact: <5033>5033>
Remote-Party-ID: "Syed Ahmer" <5033>;party=calling;screen=yes;privacy=off5033>
Max-Forwards: 70
Content-Length: 0
|14,100,50,1.14103336^10.163.14.4^SEP00230432C828
12:17:38.028 |EnvProcessUdpPort - EnvProcessUdpHandler::fireSignal() varId = 0|14,100,50,1.14103336^10.163.14.4^SEP00230432C828
12:17:38.028 |EnvProcessUdpHandler::fireSignal - SEND: index = 0, handler = 0xaf299320|*^*^*
12:17:38.028 |EnvProcessUdpPort::fireSignal - SEND, destination = 172.16.200.21:5062|*^*^*
12:17:38.028 |EnvProcessUdpPort - EnvProcessUdpHandler::send(buff, 850, 172.16.200.21:5062)|*^*^*
01-30-2014 02:42 AM
Hi Syed,
This only shows the Invite message without SDP, are there no other messages if you filter through
Call-ID: 84e24a80-2e91fc72-1a6940-bc8640a@10.100.200.11
Manish
01-30-2014 03:00 AM
Dear Manish,
I didnt find any other sip messages except register and invite in SDI trace.
09-15-2017 12:30 AM
I am facing the similar issue on Avaya phone registered on cisco call manager 9.X.
both phone avaya and cisco registered to same call manager and DN are in same partiton .
issue Avaya can call cisco phone with both way audio normal/good call .
however cisco try to reach avay DN in same partion call get disconnected with after some time i.e. 30 to 40 second
and no traces even.
Regards
Lalit Arora
09-19-2017 10:58 PM
Finally a Solution to this issue.
Noticed that the Alert-Info Header is causing the problem, part of the INVITE message being sent out to AVAYA Phone (Third Party SIP device in my case).
Alert-Info: <file://Bellcore-dr1/>
1. Configured a SIP Normalization Script to remove the Alert-Info Header from the INVITE message.
2. Applied the Normalization script to a SIP Profile.
3. Applied the SIP Profile to the Avaya Phone.
4. Test calls were successful both-sided with proper 2-way audio communication.
09-19-2017 11:00 PM - edited 09-19-2017 11:02 PM
Thank you Vishal
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