In article <0IGZ005UMQCSFOE0@osl1sminn1.broadpark.no>,
Bjorn Ove Kristiansen <bok2@online.no> wrote:>
> I've set up an Asterisk box with four local SIP users. The Asterisk box
uses
> a SIP provider for placing external calls and receiving incoming calls as
> well. In other words, there's no PSTN or ISDN lines attached to the
box.
> Codec in use is G.711 alaw.
>
> I've set up a queue with music-on-hold etc. Sound files are the
standard moh
> mp3s. When I call the queue from local sip clients everything works fine, I
> hear the music and all seems well. However, if I call from a landline to
the
> asterisk box through the SIP-provider, I can only hear the music whenever I
> am talking to myself, making some noise etc.
The SIP implementation in Asterisk relies on the timing of the incoming
stream to time its outgoing stream. If the incoming stream stops, so does
the outgoing. You need to get your provider to disable Silence Suppression
on your SIP link. I would guess they don't have many Asterisk users, or
they would have encountered this issue already.
> I figure it has something to do with silence suppression, but wherever I
> search I only find information on how to disable this on SIP clients. Since
> this is a problem that arises somewhere between the SIP provider and the
> asterisk box, I am lost. Is there any way to get around this in the
asterisk
> configuration, or is it simply that the SIP provider itself is refusing to
> transmit comfort noice?
Almost. It is probably that they ARE transmitting comfort noise, which
is not sent continuously, but just at the beginning of a period of silence.
It is a packet that tells the other end "give the user comfort noise from
now until you get more audio from me". To work with Asterisk, the provider
needs to send full audio packets all the time instead.
Cheers
Tony
--
Tony Mountifield
Work: tony@softins.co.uk - http://www.softins.co.uk
Play: tony@mountifield.org - http://tony.mountifield.org