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Snom D712 hangs up when call is put on hold

Hello.


We're having an issue with Snom D712 phones hanging up instead of putting calls on hold.


We have connected the snom phone to an Asterisk 10.0 PBX and when a call is made from the PBX to the Snom phone and the user presses Hold on the Snom, the call is terminated instead of being put on hold. This is done using the F2 soft-key that displays 'Hold' in the display. The same happens if the Line-keys are used to put the call on hold.


If the Snom initiates the call to the PBX, it seems to work as intended and the call is correctly put on hold.


I've attached a log file for the broken case, and it seems the Snom phone sends a 'BYE' even though no errors are listed in the log.


The addresses in the log file are 172.16.0.1 (our PBX) and 172.16.182.1 (the Snom phone).


I've also attached PCAP files of the SIP session for two cases. One working where the Snom phone initiated the call to the PBX and one broken where the PBX initiated the call.


Any suggestions on how we can fix this issue?

pcap
pcap
txt

Best Answer

Hi Knut,


I have a couple ideas for you to try.


In the broken scenario, in the Hold request (reINVITE) I see the following SDP:


v=0
o=root 2051144610 2051144612 IN IP4 172.16.182.1
s=call
c=IN IP4 172.16.182.1
t=0 0
m=audio 63872 RTP/AVP 9 0 8 3 99 112 18 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:Z936pBv+fhN5ORHL52eduCgyasoR5reVf9Xulv4L
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:99 G726-32/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendonly
m=video 0 R


To which the server replies:


v=0
o=root 1585620859 1585620860 IN IP4 172.16.0.1
s=Asterisk PBX v10.12.3.ics.9
c=IN IP4 172.16.0.1
t=0 0
m=audio 24028 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=recvonly


The reply is missing the "m=video" line, for this reason I would first recommend to try enabling this setting: http://wiki.snom.com/Settings/allow_mismatched_sdp_answers


If that does not work, then maybe you could also try to disable SRTP: http://wiki.snom.com/Settings/user_srtp


I hope this helps.


Thanks

Catalina



Answer

Hi Knut,


I have a couple ideas for you to try.


In the broken scenario, in the Hold request (reINVITE) I see the following SDP:


v=0
o=root 2051144610 2051144612 IN IP4 172.16.182.1
s=call
c=IN IP4 172.16.182.1
t=0 0
m=audio 63872 RTP/AVP 9 0 8 3 99 112 18 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:Z936pBv+fhN5ORHL52eduCgyasoR5reVf9Xulv4L
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:99 G726-32/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendonly
m=video 0 R


To which the server replies:


v=0
o=root 1585620859 1585620860 IN IP4 172.16.0.1
s=Asterisk PBX v10.12.3.ics.9
c=IN IP4 172.16.0.1
t=0 0
m=audio 24028 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=recvonly


The reply is missing the "m=video" line, for this reason I would first recommend to try enabling this setting: http://wiki.snom.com/Settings/allow_mismatched_sdp_answers


If that does not work, then maybe you could also try to disable SRTP: http://wiki.snom.com/Settings/user_srtp


I hope this helps.


Thanks

Catalina



1 person likes this

Hello.


Thank you for your response. The allow_mismatched_sdp_answers option fixed the issue.


SRTP was already disabled on the active identity, so I find it a bit strange that it would try to set up an encrypted connection.


Thanks for your help!

You are welcome, I am glad the problem is solved.


It does not look like SRTP is disabled for the identity. I see SRTP being set up as well in the initial INVITE in trace sip-dump-working.pcap

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