I upgraded from RC2 last night, but have a major call quality issue. Heres our setup: 1 FXS and 1 FXO card. Incoming/Outgoing calls via IAX trunking from our provider. G729 running between us and the VoIP provider. Two handsets, one BudgetTone 102 and a Cisco 7940G running the 7.2 SIP firmware. Both these phones are using ULAW to the server, and we have plenty of G729 licenses on the server. Now the BudgetTone does not have any problems at all. The Cisco however has severe breakup on the incoming audio from the VoIP provider. The caller can hear me fine. In some cases the call is okay for 10seconds or so then slowly gets worse, but most of the time I just cant make out what they are saying. The link we are running this on is a 1500/256 ADSL line, we are not running much else on this link so it does not appear to be bandwidth related. I tried changing codecs that the phone uses, in particular to G729 so there is no conversion taking place, but it does not make any difference at all. I just dont understand why it only affects the Cisco. I just switched it back to RC2 and it works fine again. This phone does not have any problems at all when taking/making calls over the Zaptel interface. Any ideas?
steve@daviesfam.org
2004-Sep-30 00:24 UTC
[Asterisk-Users] Asterisk 1.00 Call quality problem
On Thu, 30 Sep 2004, Nick Cobley wrote:> 1 FXS and 1 FXO card. > Incoming/Outgoing calls via IAX trunking from our provider. G729 > running between us and the VoIP provider. > Two handsets, one BudgetTone 102 and a Cisco 7940G running the 7.2 > SIP firmware. > Both these phones are using ULAW to the server, and we have plenty of > G729 licenses on the server. > > Now the BudgetTone does not have any problems at all. The Cisco > however has severe breakup on the incoming audio from the VoIP > provider. The caller can hear me fine. In some cases the call is okay > for 10seconds or so then slowly gets worse, but most of the time I > just cant make out what they are saying.Make sure you aren't trying to use trunking and the iax jitter buffer together. As of now they don't work properly together. So turn off one or the other for the link to your VOIP provider. As at 1.0 the jitter buffer can be enabled and disabled per peer/user. See iax.conf.sample. Steve