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<font face="Verdana">Hello,<br>
<br>
We have a problem on a recent asterisk install with Polycom 30x phones;<br>
Sometimes (can not reproduce or find the logic of the problem after one
week one analysis), the called party (even incoming or outgoing call)
can not hear the calling party, as other flow works (caller hears
called). This occurs between 5 and 10% of the time.<br>
<br>
The configuration is the following:<br>
- Asterisk 1.2.9.1<br>
- Zaptel 1.2.6<br>
- Libpri 1.2.3<br>
- chan_misdn 0.3.1-rc17<br>
- 6 Polycom IP300 (firmware 1.6.3.0067)<br>
- 1 Polycom IP301 (same firmware)<br>
- 2 Grandstream HT487 1.0.7.11 ATA<br>
- 1 2*BRI ISDN card from Beronet working in P2P TE mode<br>
- Standard Supermicro server P4 3.0GHz 1GB RAM RAID 1<br>
<br>
The problem only occurs with Polycom phones, since we add the 301
(never used) on the network.<br>
<br>
SIP caracteristics:<br>
* no reinvite (occurs also with reinvite)<br>
* ulaw<br>
* No NAT, dynamic addresses<br>
* 240 seconds between registrations<br>
<br>
RTP on ports 2220 to 31000<br>
<br>
<br>
After sniffing packets while not working, it seems packets are
correctly sent on the negotiated ports, example:<br>
INVITE from asterisk to polycom with RTP 15126<br>
OK from polycom to asterisk with RTP 2236<br>
ACK from asterisk to the polycom<br>
Bi-directional UDP Flow between 2236 <--> 15126 until BYE (with
asterisk in the path as reinvite is no set)<br>
<br>
Does anyone have any idea one the cause of this strange situation ?<br>
<br>
Thomas.<br>
<br>
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