[asterisk-dev] SIP able two way to put on hold: which one is buggy in Asterisk 11?
gregory.esnaud at niji.fr
Wed Jul 10 01:58:35 CDT 2013
I came to you about a bug, or misconfiguration maybe, on our infra.
We have Cisco's hardphone (7940 & 7941) dispatched on several sites, with an asterisk (v11.3) on a central site.
Phone A receive a PSTN call (PSTN call through a Cisco 2921), then put it on hold:
If the parameter <rfc2543Hold> is set to "false" on the phone, the phone send a=sendonly with its own ip address in connection information. Here is our issue: sometimes it works, and sometimes the call is lost (the pstn call leg never hearing the music and the hardphone cannot resume the call it thinks to have put on hold).
But if the parameter is set to "True", the phone send a=inactive with 0.0.0.0 as connection information. And then it works at each time, never the call legs are lost....
Is there a best "on hold" recommendation with asterisk? Or the Cisco's SIP layer buggy?
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