Martin Glazer wrote:
I didn't receive any response, but I'm still struggling with this
issue - The problem is intermittent, making it difficult to
troubleshoot. At least I've now managed to capture a tcpdump of a
problem call.
I am seeing ICMP port unreachable messages from the server to the
phone when the one way audio occurs.
This is my current theory, but I'm not sure it makes sense.
Somewhere during a communication with the server, either the server or
phone misses a SIP message and so one of them still believes it is on
a call.
Next outbound call made by the phone, when the phone sends the RTP
stream, it is being sent to the old UDP port on the server from the
previous conversation. The server rejects this with an ICMP port
unavailable as it is expecting RTP from the phone on another port.
This results in one way audio - the external person being called
cannot hear the callee.
When an inbound call is made, the server initiates the RTP ports with
the phone and two way conversation is heard.
Does this make any sense? I have a dump file of the conversation if
anyone is interested?
Is anyone else seeing UDP issues with Asterisk 1.2.23 or even 1.2.21?
Any thoughts on trying to determine what is causing this?
For network tracking use wireshark and pcapsipdump (from sourceforge.)
You can use pcapsipdump to capture all packets related to a call and use
wireshark to view the resulting per-call file. You can also use
wireshark alone to see all traffic which may be useful to you as well
and you can use it to extract the audio component, mix it and/or play
each side and see if there actually is audio.
And don't forget simpler things like:
1. Is the problem analog? Could it be the handset cord or the handset
mic or speaker that's not working? Mute on the phone?
2. Check for updated firmware. Check the release notes and see if this
is a known problem.
3. Check the Ethernet. Could this be a half-duplex/full duplex problem
or a faulty cable.
Good luck.
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