Console Dialing in Freeswitch

Baskar

28 Nov 2008 28 Nov '08

7:38 a.m.

*Hi,* *It is possible to dial outbound through console dialing. Yes means me How?* *Another question whether there is any api command for console dialing.* * Thank you with regards, N.Baskar *

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Michael Collins

28 Nov 28 Nov

7:55 a.m.

Why, yes there is! You want the "originate" command: http://wiki.freeswitch.org/wiki/Mod_commands#originate I'm not sure I understand your API command question, however you can definitely execute originate commands via the event socket: http://wiki.freeswitch.org/wiki/Event_Socket Could you give us a little more detail on what you are trying to accomplish? We might be able to help you figure out an elegant solution. -MC On Thu, Nov 27, 2008 at 11:38 PM, Baskar <yudha2008@gmail.com> wrote:

...

*Hi,*

*It is possible to dial outbound through console dialing. Yes means me How?*

*Another question whether there is any api command for console dialing.*

* Thank you with regards, N.Baskar *

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Baskar

1 Dec 1 Dec

10:54 a.m.

Hi, * **It is possible to dial outbound through console dialing. Yes means me How ?** Without using the softphone how can i dial outbound from freeswitch console itself. * * I want to Know without using any softphone for calling. It is possible in asterisk. we can dial from console itself. * * So i want to know it is possible in freeswitch.* Warm Regards, N.Baskar

Giovanni Maruzzelli

11:15 a.m.

Hello Baskar, in FS it is possible to call from console using the endpoint mod_portaudio. Please have a look at http://wiki.freeswitch.org/wiki/Freeswitch_softphone , it is *NOT REAL SOFTPHONE* it is FS used *LIKE* a softphone. Exactly as in Asterisk with chan_alsa or chan_oss. Sincerely, Giovanni Maruzzelli ========================================= Contact person : Mr Giovanni Maruzzelli Company : celliax Website: www.celliax.org Address : via Pierlombardo 9, 20135 Milano Country/Territory : Italy Business Email: gmaruzz at celliax dot org Phone : 39-347-2665618 Fax : 39-02-87390039 On Mon, Dec 1, 2008 at 11:54 AM, Baskar <yudha2008@gmail.com> wrote:

...

Hi,

It is possible to dial outbound through console dialing. Yes means me How ?

Without using the softphone how can i dial outbound from freeswitch console itself.

I want to Know without using any softphone for calling.

It is possible in asterisk. we can dial from console itself.

So i want to know it is possible in freeswitch.

Warm Regards, N.Baskar

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Baskar

2 Dec 2 Dec

6:24 a.m.

*Hi Giovanni Maruzzelli*, To list the available devices i have given this command *pa devlist* *output:* freeswitch@hp30094686650.optimus.co.in> pa devlist 2008-12-02 11:27:34 [CONSOLE] switch_console.c:255 switch_console_process() Unknown Command: pa But when i check in my system *hwconf *there is auido drives *class: AUDIO bus: PCI detached: 0 driver: snd-intel8x0 desc: "Intel Corporation 82801EB/ER (ICH5/ICH5R) AC'97 Audio Controller" vendorId: 8086 deviceId: 24d5 subVendorId: 8086 subDeviceId: 0c4a pciType: 1 pcidom: 0 pcibus: 0 pcidev: 1f pcifn: 5* How to resolve the problem. Can u correct me where i am wrong.Can u just describe what is the error also. Thanks for reply *-- Warm Regards, N.Baskar *

Giovanni Maruzzelli

6:50 a.m.

Hi Baskar, you have to compile and enable the module mod_portaudio. Please edit the modules.conf in the main directory of the FS sources, and remove the "#" before mod_portaudio. Also, after compilation and installation ("make install"), in the directory /usr/local/freeswitch/conf/autoload/ edit the file modules.conf.xml so to enable the portaudio module and edit the portaudio.conf.xml to reflect your setup. Sincerely, Giovanni Maruzzelli ========================================= Cell : 39-347-2665618 Fax : 39-02-87390039 On Tue, Dec 2, 2008 at 7:24 AM, Baskar <yudha2008@gmail.com> wrote:

...

Hi Giovanni Maruzzelli,

To list the available devices i have given this command pa devlist output: freeswitch@hp30094686650.optimus.co.in> pa devlist 2008-12-02 11:27:34 [CONSOLE] switch_console.c:255 switch_console_process() Unknown Command: pa

But when i check in my system hwconf there is auido drives

class: AUDIO bus: PCI detached: 0 driver: snd-intel8x0 desc: "Intel Corporation 82801EB/ER (ICH5/ICH5R) AC'97 Audio Controller" vendorId: 8086 deviceId: 24d5 subVendorId: 8086 subDeviceId: 0c4a pciType: 1 pcidom: 0 pcibus: 0 pcidev: 1f pcifn: 5

How to resolve the problem. Can u correct me where i am wrong.Can u just describe what is the error also.

Thanks for reply -- Warm Regards, N.Baskar

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Baskar

7:27 a.m.

*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@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@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@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@hp30094686650 bin]# * *Thanks for the reply. Correct me were i am wrong.* *Warm Regards, N.Baskar *

Michael S Collins

7:35 a.m.

Does the core dump always happen in this call scenario? If so, can you get a back trace? Put it on pastebin. That will hopefully help narrow down the issue. -MC Sent from my iPhone On Dec 1, 2008, at 11:27 PM, Baskar <yudha2008@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@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@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@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@hp30094686650 bin]#

Thanks for the reply. Correct me were i am wrong.

Warm Regards, N.Baskar

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Giovanni Maruzzelli

7:39 a.m.

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@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@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@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@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@hp30094686650 bin]#

Thanks for the reply. Correct me were i am wrong.

Warm Regards, N.Baskar

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Baskar

10:57 a.m.

Hi, *After starting the freeswitch I try to dial a extension from console* *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?* * I have pasted the console events in pastebin in this path: **http://fr.pastebin.ca/1273382 ** What is the error? Can any one correct me where I am wrong and try to resolve the problem. I want to know why Fs Aborted what should be done to recover from Aborted.* -- Warm Regards, N.Baskar

Michael Jerris

11:02 a.m.

This appears to be a somewhat older version of svn trunk. Please re- test with current svn trunk Thanks Mike On Dec 2, 2008, at 5:57 AM, Baskar wrote:

...

Hi,

After starting the freeswitch I try to dial a extension from console 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?

I have pasted the console events in pastebin in this path:

http://fr.pastebin.ca/1273382

What is the error? Can any one correct me where I am wrong and try to resolve the problem.

I want to know why Fs Aborted what should be done to recover from Aborted.

Baskar

11:07 a.m.

*Hi, This is the svn version i have installed before a month FreeSWITCH Version 1.0.trunk (10130M) * -- *Warm Regards, N.Baskar*

Anthony Minessale

4:09 p.m.

from the source tree of FS please type "make current" when it completes, retest the call. On Tue, Dec 2, 2008 at 5:07 AM, Baskar <yudha2008@gmail.com> wrote:

...

*Hi,

This is the svn version i have installed before a month

FreeSWITCH Version 1.0.trunk (10130M) * -- *Warm Regards, N.Baskar*

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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400

Baskar

3 Dec 3 Dec

3:32 p.m.

Hi, *I have newly installed freeswitch in another machine. **After starting the freeswitch I try to dial a extension from console but when i call extension 1002 from freeswitch console, call is connected to extension 1002, but FS is aborted but call is established in1002.* *When i dial from console the call get connected and freeswitch is cut.* *OUtput:* *FreeSWITCH Version 1.0.trunk (10567) Started. Crash Protection [Disabled] Max Sessions[1000] Session Rate[30] SQL [Enabled] 2008-12-03 21:02:21 [CONSOLE] mod_local_stream.c:142 read_stream_thread() Can't open directory: /usr/local/freeswitch/sounds/music/16000* * freeswitch@hp30094686650.optimus.co.in> pa devlist* *API CALL [pa(devlist)] output: 0;/dev/dsp;16;4 1;Intel ICH5: Intel ICH5 (hw:0,0);2;6 2;Intel ICH5: Intel ICH5 - MIC ADC (hw:0,1);2;0 3;Intel ICH5: Intel ICH5 - MIC2 ADC (hw:0,2);2;0 4;Intel ICH5: Intel ICH5 - ADC2 (hw:0,3);2;0 5;Intel ICH5: Intel ICH5 - IEC958 (hw:0,4);0;2 6;front;0;6 7;surround40;0;4 8;surround41;0;128 9;surround50;0;128 10;surround51;0;6 11;iec958;0;2 12;spdif;0;2 13;default;128;128 14;dmix;0;2* *freeswitch@hp30094686650.optimus.co.in> pa call 1002* *2008-12-03 21:06:11 [NOTICE] switch_channel.c:564 switch_channel_set_name() New Channel portaudio/1002 [fae97d5b-3480-410e-af0a-192d00710537] freeswitch@hp30094686650.optimus.co.in> 2008-12-03 21:06:12 [NOTICE] mod_portaudio.c:1586 place_call() Channel [portaudio/1002] has been answered API CALL [pa(call 1002)] output: SUCCESS:1:fae97d5b-3480-410e-af0a-192d00710537 2008-12-03 21:06:12 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing FreeSWITCH->1002 in context default 2008-12-03 21:06:12 [WARNING] switch_ivr.c:1840 switch_ivr_set_user() can't find user [default@] 2008-12-03 21:06:12 [INFO] mod_dptools.c:872 info_function() CHANNEL_DATA: Channel-State: [CS_EXECUTE] Channel-State-Number: [4] Channel-Name: [portaudio/1002] Unique-ID: [fae97d5b-3480-410e-af0a-192d00710537] 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: [fae97d5b-3480-410e-af0a-192d00710537] Caller-Source: [mod_portaudio] Caller-Context: [default] Caller-Channel-Name: [portaudio/1002] Caller-Profile-Index: [1] Caller-Profile-Created-Time: [1228318571584600] Caller-Channel-Created-Time: [1228318571584600] Caller-Channel-Answered-Time: [1228318572164620] 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] 2008-12-03 21:06:12 [INFO] mod_dptools.c:858 log_function() Answer-State []n 2008-12-03 21:06:12 [INFO] switch_ivr_async.c:1577 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 1 execute_extension::dx XML features 2008-12-03 21:06:12 [INFO] switch_ivr_async.c:1577 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 2 record_session::/usr/local/freeswitch/recordings/0000000000.2008-12-03-21-06-12.wav 2008-12-03 21:06:12 [INFO] switch_ivr_async.c:1577 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 3 execute_extension::cf XML features 2008-12-03 21:06:12 [NOTICE] switch_channel.c:564 switch_channel_set_name() New Channel sofia/internal/sip:1002@172.20.179.201:37046;rinstance=e6259d34a17a130a [e9a905cd-dc7c-49b1-b3f7-1cd52c1129d1]* *freeswitch: src/switch_core_io.c:202: switch_core_session_read_frame: Assertion `(*frame)->codec != ((void *)0)' failed. Aborted (core dumped) [root@hp30094686650 bin]# * * After installing current svn trunk also i get the same error.I cant able to recover the failure .Correct me were i am wrong. Thanks Regards, N.Baskar *

Michael S Collins

4:04 p.m.

Baskar, Which operating system are you running? I would like to try and duplicate symptoms on one of my boxes, all of which run CentOS 5.x -MC Sent from my iPhone On Dec 3, 2008, at 7:32 AM, Baskar <yudha2008@gmail.com> wrote:

...

Hi,

I have newly installed freeswitch in another machine.

After starting the freeswitch I try to dial a extension from console but when i call extension 1002 from freeswitch console, call is connected to extension 1002, but FS is aborted but call is established in1002.

When i dial from console the call get connected and freeswitch is cut.

OUtput:

FreeSWITCH Version 1.0.trunk (10567) Started. Crash Protection [Disabled] Max Sessions[1000] Session Rate[30] SQL [Enabled] 2008-12-03 21:02:21 [CONSOLE] mod_local_stream.c:142 read_stream_thread() Can't open directory: /usr/local/freeswitch/ sounds/music/16000

freeswitch@hp30094686650.optimus.co.in> pa devlist API CALL [pa(devlist)] output: 0;/dev/dsp;16;4 1;Intel ICH5: Intel ICH5 (hw:0,0);2;6 2;Intel ICH5: Intel ICH5 - MIC ADC (hw:0,1);2;0 3;Intel ICH5: Intel ICH5 - MIC2 ADC (hw:0,2);2;0 4;Intel ICH5: Intel ICH5 - ADC2 (hw:0,3);2;0 5;Intel ICH5: Intel ICH5 - IEC958 (hw:0,4);0;2 6;front;0;6 7;surround40;0;4 8;surround41;0;128 9;surround50;0;128 10;surround51;0;6 11;iec958;0;2 12;spdif;0;2 13;default;128;128 14;dmix;0;2

freeswitch@hp30094686650.optimus.co.in> pa call 1002 2008-12-03 21:06:11 [NOTICE] switch_channel.c:564 switch_channel_set_name() New Channel portaudio/1002 [fae97d5b-3480-410e-af0a-192d00710537] freeswitch@hp30094686650.optimus.co.in> 2008-12-03 21:06:12 [NOTICE] mod_portaudio.c:1586 place_call() Channel [portaudio/1002] has been answered API CALL [pa(call 1002)] output: SUCCESS:1:fae97d5b-3480-410e-af0a-192d00710537

2008-12-03 21:06:12 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing FreeSWITCH->1002 in context default 2008-12-03 21:06:12 [WARNING] switch_ivr.c:1840 switch_ivr_set_user() can't find user [default@] 2008-12-03 21:06:12 [INFO] mod_dptools.c:872 info_function() CHANNEL_DATA: Channel-State: [CS_EXECUTE] Channel-State-Number: [4] Channel-Name: [portaudio/1002] Unique-ID: [fae97d5b-3480-410e-af0a-192d00710537] 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: [fae97d5b-3480-410e-af0a-192d00710537] Caller-Source: [mod_portaudio] Caller-Context: [default] Caller-Channel-Name: [portaudio/1002] Caller-Profile-Index: [1] Caller-Profile-Created-Time: [1228318571584600] Caller-Channel-Created-Time: [1228318571584600] Caller-Channel-Answered-Time: [1228318572164620] 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]

2008-12-03 21:06:12 [INFO] mod_dptools.c:858 log_function() Answer- State []n 2008-12-03 21:06:12 [INFO] switch_ivr_async.c:1577 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 1 execute_extension::dx XML features 2008-12-03 21:06:12 [INFO] switch_ivr_async.c:1577 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 2 record_session::/ usr/local/freeswitch/recordings/0000000000.2008-12-03-21-06-12.wav 2008-12-03 21:06:12 [INFO] switch_ivr_async.c:1577 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 3 execute_extension::cf XML features 2008-12-03 21:06:12 [NOTICE] switch_channel.c:564 switch_channel_set_name() New Channel sofia/internal/sip:1002@172.20.179.201 :37046;rinstance=e6259d34a17a130a [e9a905cd-dc7c-49b1- b3f7-1cd52c1129d1] freeswitch: src/switch_core_io.c:202: switch_core_session_read_frame: Assertion `(*frame)->codec != ((void *)0)' failed. Aborted (core dumped) [root@hp30094686650 bin]#

After installing current svn trunk also i get the same error.I cant able to recover the failure .Correct me were i am wrong.

Thanks Regards, N.Baskar

_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

Anthony Minessale

5:24 p.m.

please clean all the core.* files reproduce the problem which will generate a core.xyz file (xyz is some number) run the command. gdb /usr/local/freeswitch/bin/freeswitch core.xzy when it loads issue the command bt and send me the output. -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400

Baskar

11 Dec 11 Dec

8:51 a.m.

*Hi, when in dial from console inbound is working fine when i dial outbound it is not working in console dialing. * FreeSWITCH Version 1.0.trunk (10567) Started. Crash Protection [Disabled] Max Sessions[1000] Session Rate[30] SQL [Enabled] 2008-12-11 14:17:03 [CONSOLE] mod_local_stream.c:142 read_stream_thread() Can't open directory: /usr/local/freeswitch/sounds/music/16000 *freeswitch@localhost> pa devlist* API CALL [pa(devlist)] output: 0;/dev/dsp;16;4 1;Intel ICH5: Intel ICH5 (hw:0,0);2;6 2;Intel ICH5: Intel ICH5 - MIC ADC (hw:0,1);2;0 3;Intel ICH5: Intel ICH5 - MIC2 ADC (hw:0,2);2;0 4;Intel ICH5: Intel ICH5 - ADC2 (hw:0,3);2;0 5;Intel ICH5: Intel ICH5 - IEC958 (hw:0,4);0;2 6;front;0;6 7;surround40;0;4 8;surround41;0;128 9;surround50;0;128 10;surround51;0;6 11;iec958;0;2 12;spdif;0;2 13;default;128;128 14;dmix;0;2 * After that i dial 3 in softphone Output:* freeswitch@localhost> 2008-12-11 14:17:16 [NOTICE] switch_channel.c:564 switch_channel_set_name() New Channel sofia/internal/1003@172.20.177.117[d49468e5-be90-40f6-8ffe-58c56651d87a] 2008-12-11 14:17:16 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing FreeSwitch->3 in context default 2008-12-11 14:17:16 [NOTICE] switch_channel.c:564 switch_channel_set_name() New Channel portaudio/1003 [bfc0023c-0725-40a8-a187-574bdab40c40] 2008-12-11 14:17:17 [NOTICE] mod_portaudio.c:235 channel_on_init() Ring-Ready portaudio/1003! *pa answer* 2008-12-11 14:17:24 [NOTICE] mod_portaudio.c:1404 answer_call() Channel [portaudio/1003] has been answered API CALL [pa(answer)] output: Answered 1 channels. freeswitch@localhost> 2008-12-11 14:17:24 [NOTICE] switch_ivr_originate.c:1509 switch_ivr_originate() Channel [sofia/internal/ 1003@172.20.177.117] has been answered pa hangup 2008-12-11 14:17:32 [NOTICE] mod_portaudio.c:1365 hangup_call() Hangup portaudio/1003 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] API CALL [pa(hangup)] output: OK freeswitch@localhost> 2008-12-11 14:17:32 [NOTICE] switch_core_state_machine.c:168 switch_core_standard_on_execute() Hangup sofia/internal/1003@172.20.177.117 [CS_EXECUTE] [NORMAL_CLEARING] 2008-12-11 14:17:32 [INFO] mod_cdr_csv.c:207 my_on_hangup() CHANNEL_DATA: Channel-State: [CS_HANGUP] Channel-State-Number: [10] Channel-Name: [sofia/internal/1003@172.20.177.117] Unique-ID: [d49468e5-be90-40f6-8ffe-58c56651d87a] Call-Direction: [inbound] Answer-State: [answered] Caller-Username: [1003] Caller-Dialplan: [XML] Caller-Caller-ID-Name: [FreeSwitch] Caller-Caller-ID-Number: [1003] Caller-Network-Addr: [172.20.177.201] Caller-Destination-Number: [3] Caller-Unique-ID: [d49468e5-be90-40f6-8ffe-58c56651d87a] Caller-Source: [mod_sofia] Caller-Context: [default] Caller-Channel-Name: [sofia/internal/1003@172.20.177.117] Caller-Profile-Index: [1] Caller-Profile-Created-Time: [1228985236919311] Caller-Channel-Created-Time: [1228985236919311] Caller-Channel-Answered-Time: [1228985244720686] Caller-Channel-Progress-Time: [1228985237539254] Caller-Channel-Progress-Media-Time: [0] Caller-Channel-Hangup-Time: [1228985252428470] Caller-Channel-Transfer-Time: [0] Caller-Screen-Bit: [true] Caller-Privacy-Hide-Name: [false] Caller-Privacy-Hide-Number: [false] Other-Leg-Username: [1003] Other-Leg-Dialplan: [XML] Other-Leg-Caller-ID-Name: [Extension 1003] Other-Leg-Caller-ID-Number: [1003] Other-Leg-Network-Addr: [172.20.177.201] Other-Leg-Unique-ID: [bfc0023c-0725-40a8-a187-574bdab40c40] Other-Leg-Source: [mod_sofia] Other-Leg-Context: [default] Other-Leg-Channel-Name: [portaudio/1003] Other-Leg-Screen-Bit: [true] Other-Leg-Privacy-Hide-Name: [false] Other-Leg-Privacy-Hide-Number: [false] 2008-12-11 14:17:32 [NOTICE] switch_core_session.c:956 switch_core_session_thread() Session 1 (sofia/internal/1003@172.20.177.117) Ended 2008-12-11 14:17:32 [NOTICE] switch_core_session.c:958 switch_core_session_thread() Close Channel sofia/internal/ 1003@172.20.177.117 [CS_HANGUP] 2008-12-11 14:17:32 [NOTICE] switch_core_session.c:956 switch_core_session_thread() Session 2 (portaudio/1003) Ended 2008-12-11 14:17:32 [NOTICE] switch_core_session.c:958 switch_core_session_thread() Close Channel portaudio/1003 [CS_HANGUP] * Then i tried it for outbound Output:* *pa call 1003* 2008-12-11 14:17:39 [NOTICE] switch_channel.c:564 switch_channel_set_name() New Channel portaudio/1003 [14de48f4-40cb-42bb-8f43-084d4df7ec89] 2008-12-11 14:17:39 [NOTICE] mod_portaudio.c:1586 place_call() Channel [portaudio/1003] has been answered API CALL [pa(call 1003)] output: SUCCESS:2:14de48f4-40cb-42bb-8f43-084d4df7ec89 freeswitch@localhost> 2008-12-11 14:17:39 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing FreeSWITCH->1003 in context default 2008-12-11 14:17:39 [ERR] mod_sofia.c:2102 sofia_outgoing_channel() Invalid Gateway 2008-12-11 14:17:39 [NOTICE] mod_sofia.c:2301 sofia_outgoing_channel() Close Channel N/A [CS_NEW] 2008-12-11 14:17:39 [ERR] switch_ivr_originate.c:1063 switch_ivr_originate() Cannot create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT] 2008-12-11 14:17:39 [INFO] mod_dptools.c:1868 audio_bridge_function() Originate Failed. Cause: INVALID_NUMBER_FORMAT 2008-12-11 14:17:39 [NOTICE] mod_dptools.c:1895 audio_bridge_function() Hangup portaudio/1003 [CS_EXECUTE] [INVALID_NUMBER_FORMAT] 2008-12-11 14:17:39 [INFO] mod_cdr_csv.c:207 my_on_hangup() CHANNEL_DATA: Channel-State: [CS_HANGUP] Channel-State-Number: [10] Channel-Name: [portaudio/1003] Unique-ID: [14de48f4-40cb-42bb-8f43-084d4df7ec89] Call-Direction: [inbound] Answer-State: [answered] Caller-Dialplan: [XML] Caller-Caller-ID-Name: [FreeSWITCH] Caller-Caller-ID-Number: [0000000000] Caller-Network-Addr: [172.20.177.117] Caller-Destination-Number: [1003] Caller-Unique-ID: [14de48f4-40cb-42bb-8f43-084d4df7ec89] Caller-Source: [mod_portaudio] Caller-Context: [default] Caller-Channel-Name: [portaudio/1003] Caller-Profile-Index: [1] Caller-Profile-Created-Time: [1228985259280611] Caller-Channel-Created-Time: [1228985259280611] Caller-Channel-Answered-Time: [1228985259508649] Caller-Channel-Progress-Time: [0] Caller-Channel-Progress-Media-Time: [0] Caller-Channel-Hangup-Time: [1228985259512660] Caller-Channel-Transfer-Time: [0] Caller-Screen-Bit: [true] Caller-Privacy-Hide-Name: [false] Caller-Privacy-Hide-Number: [false] 2008-12-11 14:17:39 [NOTICE] switch_core_session.c:956 switch_core_session_thread() Session 3 (portaudio/1003) Ended 2008-12-11 14:17:39 [NOTICE] switch_core_session.c:958 switch_core_session_thread() Close Channel portaudio/1003 [CS_HANGUP] *Correct me were in am wrong . i have done all the updates and i install freeswitch newly . I am using Centos 5.2 I also attached the default.xml in this mail. Correct me were in am wrong. -- Thanks, N.Baskar*

Michael Jerris

11:32 a.m.

Check your dialplan, you have a gateway name that is not configured properly. On Dec 11, 2008, at 3:51 AM, Baskar wrote:

...

2008-12-11 14:17:39 [ERR] mod_sofia.c:2102 sofia_outgoing_channel() Invalid Gateway

Michael Jerris

2 Dec 2 Dec

10:49 a.m.

What revision of freeswitch is this? Can you please test this with svn trunk? Mike On Dec 2, 2008, at 2:27 AM, Baskar 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@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@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@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@hp30094686650 bin]#

Thanks for the reply. Correct me were i am wrong.

Warm Regards, N.Baskar

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Anthony Minessale
Baskar
Giovanni Maruzzelli
Michael Collins
Michael Jerris
Michael S Collins