No sound is playing


Please excuse my bad english :wink:

I installed Asterisk and all works good. I can register the Softphone on Asterisk and can dial to the server and from the server to the phone.

But i tried to do the “test-call” where “Hello World” is playing. The Softphone dial normally and in the console is no error. There the file is playing. But i can´t hear the sound on my telephone…

I´m using a dedicated webserver with Debian 6.0 minimal. Asterisk is the newest version (10.6.1).

What can I try to do the sound working? Thanks for your help :smile:

Please explain the reference to a web server.

Missing audio tends to be the result of firewall or NAT misconfiguration, especially if you are running demo dialplans, which should answer in all the right places.

The reference to the web server was because most of the users here work with VM´s or something else.

Your tip with NAT was very helpfull, now i can hear the “Hello World”-sound, but if i call another telephone, i got the connection but no sound…

My sip.conf looks like:




Edit: I try to “hold” the connection with x-lite and after i disabled “hold”, I could speak and my partner heard me. But I´m still not hearing him. After he pressed “hold” I heard him but he didn´t hear me :smiley: Any solutions?

I didn’t mean nat settings in Asterisk, I meant in the router. nat=yes is intended for phones which are inside NAT when Asterisk is on a public address, and the phones don’t have good NAT support.

I would suggest starting with hardware SIP phones, or operating with all public or all private addresses.

For security you need to turn off allowguest. The use of type=peer is also advisable.

Hello friends.
I decided to ask here in case not to start a new thread.

I have some phones/extensions with alaw and g729 allowed codecs.
They work fine with both variants of codecs.

But the only thing I’ve noticed , is that .gsm audio files can not be played by Asterisk, when the phones are configured to use g729 codec.

Is it normal?
Does it mean that we must have audi ofiles of different format, depending on the codec we use in the sip-phone ?

You need to have a chain of codecs in Asterisk between the input and output formats. Most codecs are selected during the make menuconfig step of a source install (and all codecs for which the pre-requesites are met are, I think, installed). G.729 is different, because it is encumbered by patents. You need to download the codec binary, from Digium, and buy enough licences.

As long as you have the G.729 format module, which knows nothing of the patented technology, Asterisk can connect calls through in G.729 if both sides speak it. However, voice announcements and music on hold will need to be pre-transcoded, tones will not work at all (many tones are generated in the phone and will work), and features that impicilty require transcoding like MixMonitor and answering machine detection will not work.

david55, thank you one more time for a complete information.
And a queston appeared - is there some list of recommended audio formats for using with different codecs?
For example: alaw - .wav

.wav is slin, .WAV is gsm There is a CLI command that gives estimated costs of converting between codecs. Generally slin converts accurately to all 8kHz codecs. Conversion to G.711 and ADPCM (I forget the G. number) is cheap from slin. The high compression codecs are expensive to convert to from any other format. Converting from them to G.711 will produce a result no worse than the original. Conversions between high compression codecs are likely to produce something worse than either.

You are allowed to have multiple file formats, and Asterisk will choose its idea of the best match.