[Asterisk-Dev] chan_sip goes "deaf" - SIP monitor thread stops

[s***@daviesfam.org]

Hi,

I'm looking for any help or suggestions as to how to track down this
issue, which sure looks like a bug to me:

We have a site where chan_sip will suddenly go deaf. No untoward messages
in the logs, except that it receives nothing (no Sip read: messages), nor
do any retransmits take place.

This is often triggered when the operator blind-transfers a call into a
queue. The queue rings 14 SIP phones in parallel. (This does usually
work properly).

In the case which I examined closely, the first half or so of the phones
Asterisk sends INVITE and gets back trying/ringing. For the other half
Asterisk says its sending the INVITE, but never gets anything back and
neither retransmits the INVITEs.

No further SIP packets are seen until Asterisk is restarted. No "Sip
read:" logs.

[Matt Riddell]

Post by s***@daviesfam.org
We have a site where chan_sip will suddenly go deaf. No untoward messages
in the logs, except that it receives nothing (no Sip read: messages), nor
do any retransmits take place.
This is often triggered when the operator blind-transfers a call into a
queue. The queue rings 14 SIP phones in parallel. (This does usually
work properly).

Sounds like a problem I had at a customer site earlier this week. I
fixed it by setting incominglimit=1 in the sip.conf for each of the phones.

It seemed to be caused by a second call ringing a SIP line that already
had a call in progress.