[Freeswitch-users] Console Dialing in Freeswitch

[Giovanni Maruzzelli]

Baskar,

that is bizarre.

Seems there is a problem with mod_sofia, the module that manages SIP
connection to the SIP client at 1002 extension.

Maybe someone else on the list can be of more help.

Sincerely,

Giovanni Maruzzelli

=========================================
Cell : 39-347-2665618
Fax : 39-02-87390039



On Tue, Dec 2, 2008 at 8:27 AM, Baskar <yudha2008 at gmail.com> wrote:
> Hi,
>
> I have updated all the above events you told .It's working fine but when i
> call extension 1002 from freeswitch console, call is connected to extension
> 1002, but FS is aborted but call is established in1002. what shall i do.
> what was the error.
>
> Full freeswitch get cut.
>
> output:
> freeswitch at hp30094686650.optimus.co.in> pa call 1002
> 2008-12-02 12:54:05 [NOTICE] switch_channel.c:553 switch_channel_set_name()
> New Channel portaudio/1002 [20b1163a-29c7-4369-bdb5-27398dc1a263]
> 2008-12-02 12:54:07 [NOTICE] mod_portaudio.c:1555 place_call() Channel
> [portaudio/1002] has been answered
> API CALL [pa(call 1002)] output:
> SUCCESS:1:20b1163a-29c7-4369-bdb5-27398dc1a263
>
> 2008-12-02 12:54:07 [INFO] mod_dialplan_xml.c:232 dialplan_hunt() Processing
> FreeSWITCH->1002 in context default
> 2008-12-02 12:54:07 [WARNING] switch_ivr.c:1805 switch_ivr_set_user() can't
> find user [default@]
> freeswitch at hp30094686650.optimus.co.in> 2008-12-02 12:54:07 [INFO]
> mod_dptools.c:902 info_function() CHANNEL_DATA:
> Channel-State: [CS_EXECUTE]
> Channel-State-Number: [4]
> Channel-Name: [portaudio/1002]
> Unique-ID: [20b1163a-29c7-4369-bdb5-27398dc1a263]
> Call-Direction: [inbound]
> Answer-State: [answered]
> Channel-Read-Codec-Name: [L16]
> Channel-Read-Codec-Rate: [8000]
> Channel-Write-Codec-Name: [L16]
> Channel-Write-Codec-Rate: [8000]
> Caller-Dialplan: [XML]
> Caller-Caller-ID-Name: [FreeSWITCH]
> Caller-Caller-ID-Number: [0000000000]
> Caller-Network-Addr: [172.20.176.32]
> Caller-Destination-Number: [1002]
> Caller-Unique-ID: [20b1163a-29c7-4369-bdb5-27398dc1a263]
> Caller-Source: [mod_portaudio]
> Caller-Context: [default]
> Caller-Channel-Name: [portaudio/1002]
> Caller-Profile-Index: [1]
> Caller-Profile-Created-Time: [1228202645898038]
> Caller-Channel-Created-Time: [1228202645898038]
> Caller-Channel-Answered-Time: [1228202647630133]
> Caller-Channel-Progress-Time: [0]
> Caller-Channel-Progress-Media-Time: [0]
> Caller-Channel-Hangup-Time: [0]
> Caller-Channel-Transfer-Time: [0]
> Caller-Screen-Bit: [true]
> Caller-Privacy-Hide-Name: [false]
> Caller-Privacy-Hide-Number: [false]
> variable_channel_name: [portaudio/1002]
> variable_endpoint_disposition: [ANSWER]
> variable_read_codec: [L16]
> variable_read_rate: [8000]
> variable_write_codec: [L16]
> variable_write_rate: [8000]
> variable_use_profile: [nat]
> variable_dialed_ext: [1002]
> variable_current_application: [info]
>
>
> 2008-12-02 12:54:07 [INFO] mod_dptools.c:888 log_function() Answer-State []n
> 2008-12-02 12:54:07 [INFO] switch_ivr_async.c:1536
> switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 1 execute_extension::dx XML
> features
> 2008-12-02 12:54:07 [INFO] switch_ivr_async.c:1536
> switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 2
> record_session::/usr/local/freeswitch/recordings/0000000000.2008-12-02-12-54-07.wav
> 2008-12-02 12:54:07 [INFO] switch_ivr_async.c:1536
> switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 3 execute_extension::cf XML
> features
> 2008-12-02 12:54:07 [NOTICE] switch_channel.c:553 switch_channel_set_name()
> New Channel
> sofia/internal/1002 at 172.20.179.201:23878;rinstance=de482996ac747c8d
> [f7f80a05-be75-414b-bcea-4e5a34c3351e]
> freeswitch: src/switch_core_io.c:179: switch_core_session_read_frame:
> Assertion `(*frame)->codec != ((void *)0)' failed.
> Aborted (core dumped)
> [root at hp30094686650 bin]#
>
> Thanks for the reply. Correct me were i am wrong.
>
> Warm Regards,
> N.Baskar
>
>
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