Sir,
I have install Asterisk 1.4.22 on Centos 5 everthing is fine except one problem i have install 2 Audiocode mp 124 fxs when user of one audicode call the user of 2nd audiocde user there is a voice delay but when user of same audicode call another user of same audiocode then there is no problem.
Both audiocode working fine on Asterisk 1.4.0 olny isuue with Asterisk 1.4.0 when transfer a call than one way audio.
Please help
Rajeev.
Sir,
I have update Asterisk 1.4.22 to Asterisk 1.6.0.1 but same problem actully its not a audio delay its a audio loss during conversation.
Please help
Rajeev.
OS?
Firewall Details and network configuration?
Console/Log messages at time of loss?
Does SIP go down at the time of audio loss, if so SIP traces?
Is the failure one way or two ways?
Internally or externally bridged?
Etc.
Sir,
I have tried on Fedora Core 5 as well as on Centos 5
no message on console
sip is not down because one side audio working fine
alll firewell disable
failure only one side
internal as well as external
Rajeev.
It would be advisable to turn up the verbosity to the point where you can tell that SIP has gone down from the dialplan events, although I agree that is is probably still up.
Re-reading, I’m not sure that you have SIP at all. Is Asterisk acting as a SIP switch or are both parties Zap/dahdi ones?
On the other hand, I would have expected you to challenge the firewall question if there is no VoIP.
I’m not sure if you understood the question about internal versus external bridging. If you get this problem with external bridging the fault is not with Asterisk, unless it has reconfigured the bridging at the exact time of failure.
If you understood this to mean inside or outside of NAT, NAT has similar effects to firewalls in that it may infer connections which can time out.
Hi
Sounds like a reinvite issue.
set canreinvite=no to the sip.conf for peers on the audiocodes and see what happens
Ian
Sir,
Actually its not a nat issue becuase i have using 5 audiocodes and rest off 4 working vary well except one audiocode.
When i have use Asterisk 1.4.0 than all 5 audiocode working fine but there is a problem in Asterisk 1.4.0 when i have transfer call than there is one side audio that’s why i have upgrade asterisk.
If you suggest what i have to do to resolve audio problem in Asterisk 1.4.0 than i have revert back to Asterisk 1.4.0.
When i have use canreinvite=no then same problem.
Please help.
Rajeev.
Please help.
Rajeev.