I have the following set up:
SIP Trunk SIP Trunk
| | |
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
does any one have any ideas?
Also debug voice ccapi inout & debug ccsip message for a test call with calling, called numbers n timestamp
It looks like a "session expires timer" issue. But as Suresh has mentioned debugs would be needed to understand what is causing it.
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:
Then, share the debugs and we will check them.
Please remember to rate helpful responses and identify helpful or correct answers.
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.
1. Under voice service voip we have the following configuration which can be removed.
2. Under voice service voip, use the following command
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
Jagpreet Singh Barmi
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
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"opportunity is a haughty goddess who waste no time with those who are unprepared"
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.