Calls drop after 15 min

[martinbuffleo]

              I have the following set up:

               SIP Trunk                                 SIP Trunk

PSTN--------------------------------CME-------------------------------------------------Lync

                                                 |                                                           |

                                                 |                                                   --------------

                                                 |                                                   |              |

                                             Cisco 7940                      Lync Soft client         Dial in Conference

Calls from PSTN to CME are fine

Calls from CME to Lync are fine.

calls from Lync to PSTN hang up at the 15 min mark.

I have the following configured.

allow-connections sip to sip

no supplementary-service sip moved-temporarily

no supplementary-service sip refer

no supplementary-service sip handle-replaces

sip

  midcall-signaling passthru

does any one have any ideas?

[Suresh Subramanian]

Please provide the complete configuration.

[Suresh Subramanian]

Also debug voice ccapi inout & debug ccsip message for a test call with calling, called numbers n timestamp

[Saurabh Agnihotri]

Hi Martin,

It looks like a "session expires timer" issue. But as Suresh has mentioned debugs would be needed to understand what is causing it.

Regards,

Saurabh

[Jorge Armijo]

Please share the "sh run" and "sh version" from the CME and capture the debugs during the failure:

-debug voice ccapi inout

-debug ccsip messages

Use the following commands to capture the loggs in the best possible way:

https://supportforums.cisco.com/docs/DOC-16310

Then, share the debugs and we will check them.

--
Jorge Armijo

Please remember to rate helpful responses and identify helpful or correct answers.

[martinbuffleo]

                   Please see sh run attached.

About to capture requested debugs

[martinbuffleo]

                   Please find attached the requested debug

[Jagpreet Barmi]

Hi,

I checked the debugs and the running config.

Here is my analysis:

* CME receives a Re-Invite after 15 minutes from the provider

* After receiving the Re-invite from PSTN, CME forwards the INVITE to lync server without DTMF capability (a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15)

Due to this, the Lync responds back with "SIP/2.0 488 Invalid SDP: Gateway ParseSdpOffer Error: No DTMF support on Gateway side."

On further analysis, the reason why CME sends the re-Invite without DTMF because the re-Invite received from the service provider had  "a=silenceSupp:off - - - -" which kind of forces CME to think that this is fax call (passthrough) and not doesn't send DTMF capabilities to Lync Server.

Possible solutions:

1. Under voice service voip we have the following configuration which can be removed.

sip

  midcall-signaling passthru

2. Under voice service voip, use the following command

fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none

HTH,

Jagpreet Singh Barmi

[Ayodeji Okanlawon]

+5 Jagpreet..

Excellent find. Two questions come to mind

1. Why will the asterisk guys use this as a session refresher..Looks like this is what they are doing

2. Why is CCME not hanlding this correctly. This is a valid parameter as speficifed in RFC 3108

Looking online, looks like this has caused issues and its very common with asterisk.

Would be interesting to know what your PBX provider says if you contact them and ask why they are sending you a silenceSUPP parameter in the middle of a call

Please rate all useful posts

"opportunity is a haughty goddess who waste no time with those who are unprepared"

[martinbuffleo]

Thanks for the replies.

I have forwarded this discussion to my SIP provider.

[martinbuffleo]

I applied the above which didn't appear to make a difference.

Then I set my dial peers to all use "dtmf-relay rtp-nte"

and I have a call that has been up for 21 min

Thanks for the help.

Now just need to work out why I can't join a conference, even though Lync is accepting my meetin number.

[jj.arellano]

i had both configurations so, i only deleted sip.... and it works!!!!
thanks Jagpreet.
Jesus Arellano.