[Freeswitch-users] Dialplan with Intercall causes Error.

Milena testeador01 at gmail.com
Fri Nov 5 07:46:48 PDT 2010


Hello

You're probably trying to bridge a channel using g729 with one using another
codec, which requires transcoding and you don't have g729 module with
licenses therefore being unable to bridge it.

To be sure of what is going on, check your logs on log level debug, because
the ones you posted are not very informative. Most likely you will need to
change your softphone's codec or the other party's to also admit a different
one than g729, or set both parties to g729 so it won't require transcoding.

If you need more help, please use pastebin to paste logs:
pastebin.freeswitch.org and make sure your logs are on debug level. Then
post the link in this list.

-milena

On Thu, Nov 4, 2010 at 12:29 PM, Raja Rokkam <rokkamraja at gmail.com> wrote:

> Hi,
>
>    I get the below error when I try contacting another SIP service provider
> using a freeswitch dialplan:
>
> 2010-11-04 13:13:28.409290 [NOTICE] switch_channel.c:669 New Channel
> sofia/internal/1000 at 192.168.3.175 [afcd49a9-351c-4439-8f81-a4f4e45b5c99]
> 2010-11-04 13:13:28.415386 [INFO] mod_dialplan_xml.c:418 Processing
> 1000->5099 in context default
> 2010-11-04 13:13:28.418291 [NOTICE] switch_channel.c:669 New Channel
> sofia/external/+18662087191 [61b94ef6-bd16-4b27-9866-09e5694b28c2]
> 2010-11-04 13:13:28.547282 [INFO] sofia.c:662 Update Callee ID to
> "+18662087191" <+18662087191>
> 2010-11-04 13:13:28.547282 [WARNING] sofia_glue.c:3290 Hello,
> I see you have a Sonus!
> FYI, Sonus cannot follow the RFC on the proper way to send DTMF.
> Sadly, my creator had to spend several hours figuring this out so I thought
> you'd like to know that!
> Don't worry, DTMF will work but you may want to ask them to fix it......
> 2010-11-04 13:13:28.553283 [NOTICE] sofia.c:4733 Channel
> [sofia/external/+18662087191] has been answered
> 2010-11-04 13:13:28.565282 [NOTICE] switch_ivr_originate.c:3103 Channel
> [sofia/internal/1000 at 192.168.3.175] has been answered
> 2010-11-04 13:13:28.610279 [ERR] mod_g729.c:145 This codec is only usable
> in passthrough mode!
> 2010-11-04 13:13:28.610279 [ERR] switch_core_io.c:726 Codec G.729 decoder
> error!
> 2010-11-04 13:13:28.610279 [NOTICE] switch_ivr_bridge.c:637 Hangup
> sofia/external/+18662087191 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
> 2010-11-04 13:13:28.631289 [NOTICE] switch_core_state_machine.c:185
> sofia/internal/1000 at 192.168.3.175 has executed the last dialplan
> instruction, hanging up.
> 2010-11-04 13:13:28.631289 [NOTICE] switch_core_state_machine.c:187 Hangup
> sofia/internal/1000 at 192.168.3.175 [CS_EXECUTE] [NORMAL_CLEARING]
> 2010-11-04 13:13:28.631289 [NOTICE] switch_core_session.c:1182 Session 2
> (sofia/external/+18662087191) Ended
> 2010-11-04 13:13:28.631289 [NOTICE] switch_core_session.c:1184 Close
> Channel sofia/external/+18662087191 [CS_DESTROY]
> 2010-11-04 13:13:28.634279 [NOTICE] switch_core_session.c:1182 Session 1
> (sofia/internal/1000 at 192.168.3.175) Ended
> 2010-11-04 13:13:28.634279 [NOTICE] switch_core_session.c:1184 Close
> Channel sofia/internal/1000 at 192.168.3.175 [CS_DESTROY]
>
>
> Can you please let me know what could be the reason? I am using the latest
> Freeswitch server, and PJSIP for the softphone library.
>
>
>
> Thanks,
> Raja.
>
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