Thanks Milena, There were some issues with the service provider for G729. Changing to G711 solved the issue. -Raja. On Fri, Nov 5, 2010 at 8:16 PM, Milena <testeador01@gmail.com> wrote:
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@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@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@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@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@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@192.168.3.175) Ended 2010-11-04 13:13:28.634279 [NOTICE] switch_core_session.c:1184 Close Channel sofia/internal/1000@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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