08-25-2017, 07:47 AM
Hello,
I've some troubles with a T41P (the only Yealink phone one my LAN network). When I answer to an external call, the caller can hear me (T41P-> external : audio ok) but I can't hear him (external -> T41P : no audio).
If I answer the call with an other SIP phone (which is ringing at the same time that T41P) there is no issue.
The SIP server and the phones are on the same network (i.e. no nat and no routing).
The strange thing is that in the PCAP traces every thing looks good and furthermore the RTP voice stream from the external caller is actually received by the T41P !
(I was able to play the both stream - T41P->external and external->T41P - with Wireshark).
After some tests, it looks like that the issue happens if the RTP stream is first established by the SIP server and secondly by the T41P. In the opposite case (RTP first established by the T41P) every thing works.
When I get this issue it is possible to get the audio working by putting the call on hold and resuming it : after the call has been resumed, the caller can be heard.
If I call an external number there is no issue (audio works fine in the two ways - in this case, RTP stream is first established by the T41P).
For information :
SIP server : Asterisk 14 (IP 192.168.101.27)
T41P : version 36.81.0.110 (IP 192.168.101.30)
in attachment : the PCAP file [attachment=4557] and the .bin configuration file [attachment=4558] (I can't attach the allconfig.tgz file :/)
Someone have some ideas ?
Let me know if you want more informations.
Best regards,
Jonathan
I've some troubles with a T41P (the only Yealink phone one my LAN network). When I answer to an external call, the caller can hear me (T41P-> external : audio ok) but I can't hear him (external -> T41P : no audio).
If I answer the call with an other SIP phone (which is ringing at the same time that T41P) there is no issue.
The SIP server and the phones are on the same network (i.e. no nat and no routing).
The strange thing is that in the PCAP traces every thing looks good and furthermore the RTP voice stream from the external caller is actually received by the T41P !
(I was able to play the both stream - T41P->external and external->T41P - with Wireshark).
After some tests, it looks like that the issue happens if the RTP stream is first established by the SIP server and secondly by the T41P. In the opposite case (RTP first established by the T41P) every thing works.
When I get this issue it is possible to get the audio working by putting the call on hold and resuming it : after the call has been resumed, the caller can be heard.
If I call an external number there is no issue (audio works fine in the two ways - in this case, RTP stream is first established by the T41P).
For information :
SIP server : Asterisk 14 (IP 192.168.101.27)
T41P : version 36.81.0.110 (IP 192.168.101.30)
in attachment : the PCAP file [attachment=4557] and the .bin configuration file [attachment=4558] (I can't attach the allconfig.tgz file :/)
Someone have some ideas ?
Let me know if you want more informations.
Best regards,
Jonathan