When I call into my Asterisk box via my VoIP line (using gsm codec) and then
try to make an outgoing DISA call over PSTN I get the following:

[Nov 1 15:12:54] WARNING[17694]: chan_dahdi.c:8930 dahdi_write: Cannot
handle frames in gsm format
[Nov 1 15:12:54] WARNING[17694]: app_dial.c:1401 wait_for_answer: Unable to
forward voice or dtmf

Obviously, it looks like asterisk is not converting the gsm frames to
whatever it needs to send over the PSTN. I never had this problem with the
1.6.x series but it started as soon as I upgraded to 1.8.0 and dahdi-2.4.0.
My Asterisk machine has a TDM-410 card installed for the interface to the
PSTN.

Any ideas?

What am I doing wrong…to get no responses at all

Thx

From: href=”mailto:asterisk-users-bounces@lists.digium.com”>asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Ujjval Karihaloo
Sent: Thursday, October 07, 2010 7:35 PM
To: Asterisk Users Mailing List – Non-Commercial Discussion
Subject: [asterisk-users] REINVITE with Auth Credentials has different SDP Codec

Hi I have a call from Service Provider (SP) to Asterisk to User

User sends a T38 REINVITE

Asterisk passes that to SP

SP challenges the INVITE

Asterisk sends INVITE with credentials but sends G711ulaw in the SDP instead of T38 udptl…

Obviously Fax fails..

Any ideas on how I can maintain the T38 SDP when SP challenges the mid-Call T38 REINVITE?

Hi I have a call from Service Provider (SP) to Asterisk to User

User sends a T38 REINVITE

Asterisk passes that to SP

SP challenges the INVITE

Asterisk sends INVITE with credentials but sends G711ulaw in the SDP instead of T38 udptl…

Obviously Fax fails..

Any ideas on how I can maintain the T38 SDP when SP challenges the mid-Call T38 REINVITE?

Hi,

I am facing some voice drop in inbound, outbound, and IVR. But while
checking the process of the CPU and memory utilization is very less.

Mem: 21304K used, 36500K free, 0K shrd, 1896K buff, 13228K cached

The voice drop is in systemic. I am not too sure what to check… all the
configuration and codec are set to proper…

Anything to do with tos in sip.conf or something else… I am using Asterisk
1.2 version.

Hi list,

I’m trying to test an IVR system with recorded prompts and would
like to be able to call 1234 and have the codec be gsm, 2234 slin, 3234
ulaw, etc. I know I can set up 3 users where #1 is gsm, #2 is ulaw and #3
is slin; Need it the other way so I can do DAHDI–> IAX testing.

Any ideas? Google wasn’t really helpful on this one.

Danny Nicholas