Re: Incoming calls issue in Third Party SIP Phone

sahmerzia · ‎01-29-2014

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

Manish Gogna · ‎01-29-2014

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

sahmerzia · ‎01-30-2014

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-d01ed2aa12c8-246211918

To: <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>

Remote-Party-ID: "Syed Ahmer" <5033>;party=calling;screen=yes;privacy=off

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)|*^*^*

Manish Gogna · ‎01-30-2014

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

sahmerzia · ‎01-30-2014

Dear Manish,

I didnt find any other sip messages except register and invite in SDI trace.

LalitArora89 · ‎09-15-2017

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

Vishal Gupta. · ‎09-19-2017

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.

LalitArora89 · ‎09-19-2017

Thank you Vishal