[asterisk-users] Chan_sip Entry

[Kutman.DK at forces.gc.ca]

Hello,
 
Yes, I also believe that this is some sort of codec issue.  Here is my sip.conf file:
 
[201]<?xml:namespace prefix = o ns = "urn:schemas-microsoft-com:office:office" />

type=friend

;secret=201

record_out=Adhoc

record_in=Adhoc

qualify=no

port=5060

nat=no

host=dynamic

dtmfmode=rfc2833

dial=SIP/201

context=from-internal

canreinvite=no

callerid=device <201>

 

[202]

type=friend

;secret=202

record_out=Adhoc

record_in=Adhoc

qualify=no

port=5060

nat=no

host=dynamic

dtmfmode=rfc2833

dial=SIP/202

context=from-internal

canreinvite=no

callerid=device <202>
 
Note: The "secret" is commented out so that there is no authentication when registering with the Jain-Sip phones.
 
Thanks,
 
 

-----Original Message-----
From: Gerald A [mailto:geraldablists at gmail.com]
Sent: Tuesday, September 11, 2007 5:12 PM
To: Kutman DK at ADM(Mat) DAEPM(R&CS)@Ottawa-Hull
Subject: Re: [asterisk-users] Chan_sip Entry


Hi again,


On 9/11/07, Kutman.DK at forces.gc.ca < Kutman.DK at forces.gc.ca  <mailto:Kutman.DK at forces.gc.ca> > wrote: 


I am trying to get to Jain Sip softphones to call one another via an Asterisk server.  When I call from phone 1 to phone 2 there is audio transmission both ways, but when I call from phone 2 to phone 1 I don't get audio transmission and reception both ways.  When I look at the asterisk log file it has an entry which says: 
"Oooh, format changed to 2".


Usually this is a codec selection problem. Are both Jain's the same version?

Maybe posting your sip.conf for the phones might help.

Thanks, 
Gerald.
 

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