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[yudha2008 at gmail.com]

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

[msc at freeswitch.org]

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 (yudha2008@gmail.com)> wrote:

[yudha2008 at gmail.com]

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

[gmaruzz at celliax.org]

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:

[yudha2008 at gmail.com]

Hi Giovanni Maruzzelli,
To list the available devices i have given this command pa devlistoutput:freeswitch@hp30094686650.optimus.co.in (freeswitch@hp30094686650.optimus.co.in)> pa devlist 2008-12-02 11:27:34 [CONSOLE] switch_console.c:255 switch_console_process() Unknown Command: paBut when i check in my system hwconf there is auido drives class: AUDIObus: PCI detached: 0driver: snd-intel8x0desc: "Intel Corporation 82801EB/ER (ICH5/ICH5R) AC'97 Audio Controller" vendorId: 8086deviceId: 24d5subVendorId: 8086 subDeviceId: 0c4apciType: 1pcidom: 0 pcibus: 0pcidev: 1fpcifn: 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

[gmaruzz at celliax.org]

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:

[yudha2008 at gmail.com]

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 (freeswitch@hp30094686650.optimus.co.in)> pa call 10022008-12-02 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 answeredAPI CALL [pa(call 1002)] output: SUCCESS:1:20b1163a-29c7-4369-bdb5-27398dc1a2632008-12-02 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 (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 features2008-12-02 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 features2008-12-02 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

[msc at freeswitch.org]

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 (yudha2008@gmail.com)> wrote:

[gmaruzz at celliax.org]

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:

[yudha2008 at gmail.com]

Hi,

<!-- /* Style Definitions */ p.MsoNormal, li.MsoNormal, div.MsoNormal {mso-style-parent:""; margin:0in; margin-bottom:.0001pt; mso-pagination:widow-orphan; font-size:12.0pt; font-family:"Times New Roman"; mso-fareast-font-family:"Times New Roman";} @page Section1 {size:8.5in 11.0in; margin:1.0in 1.25in 1.0in 1.25in; mso-header-margin:.5in; mso-footer-margin:.5in; mso-paper-source:0;} div.Section1 {page:Section1;} --> 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

[mike at jerris.com]

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:

[mike at jerris.com]

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:

[yudha2008 at gmail.com]

Hi,

This is the svn version i have installed before a month

FreeSWITCH Version 1.0.trunk (10130M)

--
Warm Regards,
N.Baskar

[anthony.minessale at g...]

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 (yudha2008@gmail.com)> wrote:

[yudha2008 at gmail.com]

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 (freeswitch@hp30094686650.optimus.co.in)> pa devlist
API CALL [pa(devlist)] output: 0;/dev/dsp;16;41;Intel ICH5: Intel ICH5 (hw:0,0);2;6 2;Intel ICH5: Intel ICH5 - MIC ADC (hw:0,1);2;03;Intel ICH5: Intel ICH5 - MIC2 ADC (hw:0,2);2;0 4;Intel ICH5: Intel ICH5 - ADC2 (hw:0,3);2;05;Intel ICH5: Intel ICH5 - IEC958 (hw:0,4);0;2 6;front;0;67;surround40;0;48;surround41;0;128 9;surround50;0;12810;surround51;0;611;iec958;0;2 12;spdif;0;213;default;128;12814;dmix;0;2
freeswitch@hp30094686650.optimus.co.in (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 (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 features2008-12-03 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 features2008-12-03 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