sip door entry system to sip call fails

[barry florence]

call manager 10.5.1.10000-7

(2N Helios IP ) door system(2042)-> cucm->DX650(3513)

Time stamp:16:23:27

Hi

    when call initiated from door system to user (DX650), it rings, once answered on the phone it disconnects the call and gives fast busy tone instead of connecting the call.  Same calls work in jabber and also in sccp phones.  I have attached the call manager trace, please help on this.

Thanks

Barry

[barry florence]

Thanks for your reply and time.  This issue is haunting me, however I have given them the workaround by giving them 8945.  I will  wait for your reply and will also see if i can find anything in the weekend.

Thanks again.

[Ayodeji Okanlawon]

Barry,

The 8945 works with both audio and video? Can you send me the logs for call with the 8945?

[barry florence]

Hi Ayodeji,

called number: 3513

       Sorry I have incorrectly said previously that the working phone is 8945(sccp), however i just realise the working phone is 8945 sip phone.  I have tested on this phone (8945-sip), the audio and video both has worked as said before, I have attached the logs as requested.

Thanks

[barry florence]

Hi Ayodeji,

     Did you had time to look into the traces,   I am trying to look as well  the previously attached trace however i am not good at sip traces, i can see that there are four legs on SDP negotiation

1.  Door Entry to CUCM (Early Offfer)

2.  phone(8945 and dx650) to cucm 200ok (Delayed offer)

3. cucm to phone  (delayed offer ack)

4. cucm to Door entry (Early offer ack)

after looking at the working phone  and nonworking sip traces, i can see there is difference in above call leg 2,  the rest of the call legs are more less same.  for example, the difference is the working one recieve only, the non-working one is sendrev only and video codec parameters are different, other than it hard for me to understand.  Could you please shed some lights.

Thanks.

---------------------

2.   Delay Offer 200ok

DX650 to cucm

04609131.002 |16:23:30.080 |AppInfo  |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.1.4.49 on port 41226 index 536 with 2522 bytes:
[938085,NET]
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.1.3.5:5060;branch=z9hG4bK56122278d148
From: <sip:2042@10.1.3.5>;tag=302167~23001403-1e76-4ea5-8dd4-52d6f82b946c-27747696
To: <sip:3513@10.1.3.5>;tag=5cfc665d4a0d088022cc6b0e-33f1d21d
Call-ID: aa0caf00-4b6197ff-298c-503010a@10.1.3.5
Date: Wed, 14 Jan 2015 16:23:29 GMT
CSeq: 101 INVITE
Server: Cisco-CP-DX650/10.1.2
Contact: <sip:0bed8813-62a5-6dcf-6e3d-90cc088a245d@10.1.4.49:41226;transport=tcp>;video
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
Remote-Party-ID: "Navjot Mary" <sip:3513@10.1.3.5>;party=called;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-7.0.0,X-cisco-xsi-8.5.1
Allow-Events: kpml,dialog
Recv-Info: conference
Recv-Info: x-cisco-conference
Content-Length: 1451
Content-Type: application/sdp
Content-Disposition: session;handling=optional
v=0
o=Cisco-SIPUA 1129 0 IN IP4 10.1.4.49
s=SIP Call
t=0 0
m=audio 18126 RTP/AVP 0 8 18 102 116 101
c=IN IP4 10.1.4.49
a=trafficclass:conversational.audio.avconf.aq:admitted
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:102 L16/16000
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
m=video 27814 RTP/AVP 100 112 126 97
c=IN IP4 10.1.4.49
b=TIAS:4000000
a=trafficclass:conversational.video.avconf.aq:admitted
a=rtpmap:100 H264/90000
a=fmtp:100 profile-level-id=640028;packetization-mode=1;level-asymmetry-allowed=1;max-fps=6000;max-rcmd-nalu-size=256000
a=imageattr:* recv [x=1024,y=600,q=0.60] [x=1280,y=720,q=0.50]
a=rtpmap:112 H264/90000
a=fmtp:112 profile-level-id=4D0028;packetization-mode=1;level-asymmetry-allowed=1;max-fps=6000;max-rcmd-nalu-size=256000
a=imageattr:* recv [x=1024,y=600,q=0.60] [x=1280,y=720,q=0.50]
a=rtpmap:126 H264/90000
a=fmtp:126 profile-level-id=428028;packetization-mode=1;level-asymmetry-allowed=1;max-fps=6000;max-rcmd-nalu-size=256000
a=imageattr:* recv [x=1024,y=600,q=0.60] [x=1280,y=720,q=0.50]
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=428028;packetization-mode=0;level-asymmetry-allowed=1;max-fps=6000;max-rcmd-nalu-size=256000
a=imageattr:* recv [x=1024,y=600,q=0.60] [x=1280,y=720,q=0.50]
a=rtcp-fb:* nack pli
a=rtcp-fb:* ccm fir
a=rtcp-fb:* ccm tmmbr
a=sendrecv

Source Filename: SDL001_100_000731.txt

-------------------------------------------

2. 200ok
8945 phone to call manager

07254147.001 |17:00:46.729 |AppInfo  |//SIP/SIPUdp/wait_SdlSPISignal: Outgoing SIP UDP message to 10.1.4.93:[5060]:
[1504223,NET]
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.4.93:5060;rport;branch=z9hG4bK2813
From: "Front Door" <sip:2042@10.1.3.5>;tag=26462
To: <sip:3513@10.1.3.5>;tag=490081~23001403-1e76-4ea5-8dd4-52d6f82b946c-27754391
Date: Mon, 19 Jan 2015 17:00:44 GMT
Call-ID: 14072
CSeq: 20 INVITE
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
Allow-Events: presence
Supported: replaces
Server: Cisco-CUCM10.5
Send-Info: conference, x-cisco-conference
Remote-Party-ID: "Navjot Mary" <sip:3513@10.1.3.5>;party=called;screen=yes;privacy=off
Contact: <sip:3513@10.1.3.5:5060>;video
Content-Type: application/sdp
Content-Length: 449
v=0
o=CiscoSystemsCCM-SIP 490081 1 IN IP4 10.1.3.5
s=SIP Call
c=IN IP4 10.1.4.29
b=TIAS:768000
b=AS:768
t=0 0
m=audio 17098 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=video 17100 RTP/AVP 126
b=TIAS:704000
a=rtpmap:126 H264/90000
a=fmtp:126 profile-level-id=428014;packetization-mode=1;max-mbps=36000;max-fs=1200;max-rcmd-nalu-size=1300;level-asymmetry-allowed=1
a=content:main
a=recvonly

Source Filename: traces.txt

---------------

[Ayodeji Okanlawon]

Barry,

I have looked at the traces again and they are exactly the same. The only difference is the port number that is used for video and audio calls are different on the 8945 and DX650. This is expected since these ports are dynamic ports.

I have attached the two traces and the relevant 200 OK and ACK. Its the same.

What I would suggest is to either downgrade the firmware on the DX650 or Upgrade it. Downgrade it if its running the latest firmware or upgrade it if its not. Your last option is to OPEN a TAC case or contact helios and confirm that this product will work with DX650

To understand SIP traces have a look at this document I wrote. It will help you.

https://supportforums.cisco.com/document/113271/understanding-sip-traces

[barry florence]

Hi Ayodeji,

      Thanks for your reply.   I will try to work on the firmware by next week and update you, if not as you suggested will go for the tac.  I will always look into your articles whenever i have issues with sip traces or dsp, infact most of my sip trace reading knowledge i got it from there.  you are doing a great job by sharing your knowledge and inspiration to me, God bless you.  Thanks for the help.

Barry

[SwathyEdayath]

Hi Barry, could you resolve this problem ? do u remember ..? I am going through same issue now..

Regards

Swathy