* You are viewing Posts Tagged ‘codec’

t38modem v2, which version or patch of asterisk?

Hi I’m developing for my company an Asterisk+t38modem+Hylafax solution.
I’m struggling in asterisk< ->t38modemv2 comunication. I’ve tried lot of
asterisk version but noone seem to function well. In asterisk 1.8.8.8
for example i can receive fax but I’m not able to send faxes due to a
“SIP/2.0 488 Not acceptable here” in asterisk 1.6 and 1.4 T38 codec
seems to be unsupported.
Asterisk 1.4.20 reports:
channel.c:3153 ast_channel_make_compatible: No path to translate from
SIP/T38modem-0-084e8dd8(256) to SIP/audiocodes_mp114-084d7858(8)
Is there an Asterisk version that could go with t38modem?
thank a lot.
I hope that someone answer becouse I’m very not so far to be unenploied…

no audio using g729A for Cisco AS5300 sip peer

Hi,

We need help in enabling g729a codec for our SIP peer that’s using Cisco
AS5300.
Our codec is purchased from Digium.

We are able to dial out the numbers and answer the call, but there’s no
audio. This is when only g729a is allowed.

We noticed when they also allow ulaw codec on their side, the codec used
falls back to ulaw and the problem is gone.

Set Call Codec in extension.conf

Hi All,
I am trying to set call codec at extension.conf but it doesnt work … its like my command doesnt change anything

exten=6500,1,Answer
exten=6500,2,Playback(welcome)
exten=6500,3,SIPAddHeader(email:${SIP_HEADER(email)})
exten=6500,4,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H:%M:%S_%A)}_${SIP_HEADER(email)}.wav,b)
exten=6500,5,set(SIP_CODEC=gsm) — this is not changed ….
exten=6500,6,Queue(${EXTEN})

can any body help me with that?

Codec warnings after upgrade to 1.8

I’m getting various codec related warnings after upgrading to 1.8. Did I miss something in the UPGRADE file? Does Asterisk no longer transcode 8-)?

WARNING[11123]: channel.c:4909 ast_write: Codec mismatch on channel DAHDI/i1/12124221200-74 setting write format to g722 from ulaw native formats 0×4 (ulaw)

And

WARNING[11120]: channel.c:4909 ast_write: Codec mismatch on channel SIP/interglobe-sip-000001e6 setting write format to g722 from ulaw native formats 0×4 (ulaw)

File Convert

Hi users,

I have Asterisk 1.6.2.20 in Ubuntu 10.04. I am trying to convert a gsm file
to G729 using file convert, but I am facing error as follows,

file convert /tmp/welcome.gsm /tmp/welcome.g729
Failed to convert /tmp/welcome.gsm to /tmp/welcome.g729!
Command ‘file convert /tmp/welcome.gsm /tmp/welcome.g729′ failed.
[Dec 20 17:24:18] WARNING[2221]: translate.c:256 ast_translator_build_path:
No translator path from g723 to alaw
[Dec 20 17:24:18] WARNING[2221]: file.c:184 ast_writestream: Unable to
translate to format g729, source format gsm

Even though I have the module format_g729.so. Do I need to have licensed
G729 codec for this? or codec_g729.so?

Kindly let me know how to convert the file.

Regards