Shared-line CME

Hi everyone,

 

I am trying to set up a shared-line between SCCP and SIP Phones.

 

ephone-dn 1 octo
number 4001
shared-line sip

 

ephone 1
 mac-address 001F.CAE8.335C
 ephone-template
 type 7962
 button 1:1

voice register dn 1
 number 4002 

voice register dn 2
 number  4001
 huntstop channel 1
 shared-line

voice register pool 1
 id mac F029.2959.500D
 type 9971
 number 1 dn 1
 number 2 dn 2
 no call-waiting
 dtmf-relay sip-kpml rtp-nte
 codec g711u
 no vad
 camera
 video 

 

Both phones ring every time but when I answer from SIP Phone the call is drop right away. If I answer from SCCP Phone the call goes ok I can Hold and Resume from SIP Phone.

What can I do to answer directy from the SIP Phone?

Comments

  • Hi,

     

      It looks like you have properly created a shared line on a SIP and SCCP phone at the same time.  I would be interested in what your CME is saying when you pick up the line on the 9971 SIP phone. Can you do a 'debug ccsip messages" and 'debug ccsip error' on the CME and show the ouput from the time the 9971 registers with the CME up till you initiate the call and then pick up the line?

    Thanks,

     

     

  • Hi,

     

      I have been looking through the logs for the last day and trying to understand the exact error messages and call flow.  I should have asked you for the the show ccapi in out debug as well to make it a little easier but I can see half of the conversation between the skinny phone and the Site C CME by seeing the SIP messages Site C CME is sending back to HQ (the first SIP/2.0 180 Ringing response it sends to HQ) 

    I will spend some more time in the morning. I think the issue stems from when they added support for shared lines on SIP and SCCP phones in CME. I will check my notes but I know for our internal version running 8.6 on IOS 15.1(4) it does not support this feature. I hopefully will figure out why the issue is happening by reading through mor eof the RFC 3216 and the Cisco docs and be able to help. 

    For me, this is also a learning experience as I try better understand the limitations of our own phone system.

     

     

  • Swap your voice register dn 1 and 2 so that the ephone-dn of the shared line is the same number as the voice register dn. Also there is a known issue with a specific hardware revision of the 29xx router.  I forget if it's for the sip or the sccp phone but one of them I would add the max calls command in the dn. 

     

    Good luck 

  • Also what codec does the sccp phone when you double press ? On the successful call?

  • Hi guys,

    Thanks for trying to help me lol.

    It is something related to codec. I can call from PSTN and answer from SIP Phone. the issue is when I call from HQ or SB to SC through CUBE.

    HQ to SC I am using g729.

     

    In my cube I have the dial-peer

     

    Dial-peer voice 4001 voip

    Destination-pattern ^4...$

    Incoming called-number ^4...$

    Session protocol sip2

    Session target ipv4:142.102.66.254

    Dtmf-relay sip-kpml rtp-nte

    Voice-class sip g729 annex

    No vad

     

    In my SC I have the dial-peer

    Dial-peer voice 2300 voip

    Destination-pattern [23]...$

    Incoming called-number [23]...$

    Session protocol sip2

    Session target ipv4:142.102.65.254

    Dtmf-relay sip-kpml rtp-nte

    Codec g729

    No vad

     

    Is there anyway where can I debug mismatch codec?

     

    Thanks guys.

  • It appears you have 729 and the 711 on the phone. How is your transcoded?


    Sent from my iPhone. Any typos are the result of its intelligence and not mine. 

    On Oct 22, 2016, at 5:01 AM, ottoravasco <[email protected]> wrote:

    Hi guys,

    Thanks for trying to help me lol.

    It is something related to codec. I can call from PSTN and answer from SIP Phone. the issue is when I call from HQ or SB to SC through CUBE.

    HQ to SC I am using g729.

     

    In my cube I have the dial-peer

     

    Dial-peer voice 4001 voip

    Destination-pattern ^4...$

    Incoming called-number ^4...$

    Session protocol sip2

    Session target ipv4:142.102.66.254

    Dtmf-relay sip-kpml rtp-nte

    Voice-class sip g729 annex

    No vad

     

    In my SC I have the dial-peer

    Dial-peer voice 2300 voip

    Destination-pattern [23]...$

    Incoming called-number [23]...$

    Session protocol sip2

    Session target ipv4:142.102.65.254

    Dtmf-relay sip-kpml rtp-nte

    Codec g729

    No vad

     

    Is there anyway where I can debug miscodec?

     

    Thanks guys.




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  • Is your transcoder working?  Debug VoIP dialpeer inout might help. I stand behind my previous comment about maxcalls and the dn number matching. 

  • I tried to change the DN to make ephone-dn and voice register dn the same TAG. It's on "bug" of the version 15.4M1 if I remember correctly. You do not need the same TAG to the shared-line work.

     

    Regarding to max-calls I add it and nothing change lol.

    The codec, normally used through CUBE is G729.

     

     

     

  • This is the link of debug voip dialpeer inout

    https://drive.google.com/file/d/0B5nZEnspRlZ7MkJXc256eWFvakU/view?usp=sharing

     

    It does not show me anything wrong.

     

    Which debug command do you guy use to check if there is any mismatch codec?

  • What's your inbound dial-peer on SC?  What's your transcoder config?  What codec is negotiated when the sccp phone answers? 

  • Worth noting that we ran into a SCCP/SIP shared line issue at a customer site.  Couldn't get consistent ring on both devices.  I had one of my guys work with TAC and they never got it working.   TAC called it a bug, but that may have been taking the easy way out. 
    I only looked at it for about 15 minutes, but all the signaling looked to be doing everything it should.  

    Take it FWIW

    Sent from my iPhone

    On Oct 23, 2016, at 10:41 AM, ottoravasco <[email protected]> wrote:

    This is the link of debug voip dialpeer inout

    https://drive.google.com/file/d/0B5nZEnspRlZ7MkJXc256eWFvakU/view?usp=sharing

     

    It does not show me anything wrong.

     

    Which debug command do you guy use to check if there is any mismatch codec?




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