[Asterisk-Users] Calls between SIP and IAX

[Bohuslav Coufal]

Thank You for answer.

 

As I try, the problem occurs when the call come to IAX channel in unknow
format of codec. When the calls come in IAX channel with correct codec
format (ulaw in my case) calls are O.K.

 

Is it possible to set generally, that i'm using in all devices ulaw
format (calls from H.323 trunk doesn't set it correct).

 

Thanks,

 

Bob.

 

________________________________

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Moises
Silva
Sent: Saturday, October 01, 2005 7:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Calls between SIP and IAX

 

asterisk console output and details about config files and networking
are welcome, and i think, desirable.

best regards

On 10/1/05, Bohuslav Coufal <bcoufal at onyx.cz> wrote:

Hi all,

I have a trouble when I try to configure asterisk to make calls between
IAX and SIP. IAX I'm using to connect between asterisks a on SIP I have
phones. The calls come from higher asterisk to my on IAX, SIP phone is 
ringing and when I hang up then dial command ends and connection is
loss.

When I'll make connection between asterisks on SIP then all work fine.

Does anybody has any suggestions?

Bob.

P.S . - I'm using asterisk 1.0.9 on FC3.

_______________________________________________
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users at lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit: 
   http://lists.digium.com/mailman/listinfo/asterisk-users




-- 
"Su nombre es GNU/Linux, no solamente Linux, mas info en
http://www.gnu.org" 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051001/82deeb71/attachment.htm