[asterisk-users] Can not hear called party

[Singer Wang]

Hi,

I'm about to leave for work, but I will send you a email from work.
We're having similar problems with our setup, and we're been working on
it for two weeks now. I would like to share notes with you to see if we
can find anything.

thanks,
Singer Wang

On Wed, 2006-12-06 at 14:10 +0100, Thomas RULMONT wrote:
> Hello,
> 
> We have a problem on a recent asterisk install with Polycom 30x
> phones;
> 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.
> 
> The configuration is the following:
> - Asterisk 1.2.9.1
> - Zaptel 1.2.6
> - Libpri 1.2.3
> - chan_misdn 0.3.1-rc17
> - 6 Polycom IP300 (firmware 1.6.3.0067)
> - 1 Polycom IP301 (same firmware)
> - 2 Grandstream HT487 1.0.7.11 ATA
> - 1 2*BRI ISDN card from Beronet working in P2P TE mode
> - Standard Supermicro server P4 3.0GHz 1GB RAM RAID 1
> 
> The problem only occurs with Polycom phones, since we add the 301
> (never used) on the network.
> 
> SIP caracteristics:
> * no reinvite (occurs also with reinvite)
> * ulaw
> * No NAT, dynamic addresses
> * 240 seconds between registrations
> 
> RTP on ports 2220 to 31000
> 
> 
> After sniffing packets while not working, it seems packets are
> correctly sent on the negotiated ports, example:
> INVITE from asterisk to polycom with RTP 15126
> OK from polycom to asterisk with RTP 2236
> ACK from asterisk to the polycom
> Bi-directional UDP Flow between 2236 <--> 15126 until BYE (with
> asterisk in the path as reinvite is no set)
> 
> Does anyone have any idea one the cause of this strange situation ?
> 
> Thomas.
> 
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