I don’t know what trixbox does for pre-config. You can try iax2 debug, you can try a differen client to make sure it is not realted (try a sip client) and then do a rtp debug to see if the media stream is flowing, etc.
Surely there has to be something going on here I am missing…
What would keep this thing from playing sound successfully to the end user? It literally neve advances past the attempt to play sound…
hmm - I did not look that closely (sorry). How did you load in that sound file. Check the log - it seems like a format that it can’t handle and the log may say something about it falling over and failing. Also check it’s permissions and the permissions of the sound directory.
It won’t even play sound for the basic things like “play our extnsion” which are system recordings that come in the install.
did you check the log and permissions of the sound directory? (that is usually the cause if it can’t play any sounds)
I made sure the sounds directory has read all.
still no sound. No sound from anything in fact.
The TIme Is, Your Extention Is, my IVR… MOH test, Nothing.
What the deuce? Which log shoudl I check?
Thanks!
Same problem for me too.
tried to remove the drivers for te110p, zaptel.conf and zapata.conf but leaving the card inside: no sound
removed the card: ok sound.
I would like to know if this could be an asterisk problem or specifically freepbx problem or O.S. Problem.
I have OpenSuse 10.2 on a very old Intel PII 450 MHz
I have a similar problem here.
I can’t play audio files to a SIP channel and I can’t hear any sound in the SIP end of a call from PSTN to SIP, for example. Using two different SIP clients, the diagnose is that the ingoing codec (asterisk -> SIP client) is not set, even though both the SIP clients and Asterisk have ulaw and alaw available. I tried to restrict the codec to ulaw in Asterisk without any success.
Also, the same audio files can be played on a Zap channel and I can do PSTN to H323 calls with no issues.
I checked the permissions on the audio files and everything seems fine.
I am running Asterisk 1.2.23 compiled from source on a Fedora Core 7, using FreePBX 2.3.1.
SIP calls are going through an openSER registrar + mediaproxy, but even without it (that’s to say with the SIP client registered to Asterisk), there is this one-way audio problem. All SIP phones are on the same IP space as the PBX so there is no problem of NAT. I installed FreePBX on top of an existing Asterisk installation and I didn’t have this problem before the FreePBX installation.
I am using an Intel Celeron 2.53GHz with a Digium TE110P.
Here is an example of channel where I can’t play a file:
localhost*CLI> sip show channel [email protected]
* SIP CallI>
Direction: Outgoing
Call-ID: [email protected]
Our Codec Capability: 4
Non-Codec Capability: 1
Their Codec Capability: 4
Joint Codec Capability: 4
Format ulaw
Theoretical Address: 192.168.1.120:5060
Received Address: 192.168.1.120:5060
NAT Support: RFC3581
Audio IP: 192.168.1.120 (local)
Our Tag: as5b62021a
Their Tag: 7c7d177dbcea
SIP User agent:
Username: alice
Peername: openSER
Original uri: sip: [email protected]
Need Destroy: 0
Last Message: Tx: ACK
Promiscuous Redir: Yes
Route: sip:192.168.1.120;lr=on;ftag=as5b62021a
DTMF Mode: rfc2833
SIP Options: (none)
Did the other persons who mentioned the problem solve it? Is there anything new on that issue? Seems related to the presence of the Digium card but I had no problem before installing FreePBX so there might be something that I can change at that level.
Thanks,
//Max
The only calls affected by this problem are SIP-to-Zap calls (and Zap-to-SIP) calls. The Asterisk-to-SIP codec seems not to be set whereas the SIP-to-Asterisk codec is set to ulaw. I checked in the SIP messages, and if I am not mistaken, it seems that both codecs were negociated and set.
If anybody has an idea, thank you for sharing.
//Max
Check to ensure asterisk is trying to play the IVR files back from the correct directory. On my default install, IVR files were uploaded to /var/lib/asterisk/sounds/custom while asterisk is trying to play sounds back from /usr/share/asterisk/sounds/custom.
The following fixed it for me.
cd /usr/share/asterisk/sounds; ln -s /var/lib/asterisk/sounds/custom
I had no sound even when placing normal calls with Asterisk so I don’t think rights over a few recorded sounds would help in my case. I have now made a fresh install and I will have a look deeper into FreePBX later, that’s to say once I will have a setup without OpenSER and MediaProxy…
Just asking. I had the same issue with the same card if you have a station module on the card but no power. You can either plug in the card or remove the station module and it should work.
sir
please send me all the setup of voip server on fedora core 7
i have installed all the packages of voip server and the
show me a some errores
(does not connect the database)
and i also installed the mysql server
sir kindly send me a setup of voip on fedora core 7 starting to end.
sir please…help me
thanks
qasimkasi,
This is considered hijacking a thread and is frowned upon. Please keep this request to the already existing post you have going. Doing this is considered rude to the original author and if it is kept up people will NOT answer your posts. also posting here limits your responses to those that know how to solve no audio problems and/or one way audio problems.
Please be considerate of others.