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[Freeswitch-users] Console Dialing in Freeswitch

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msc at freeswitch.org
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PostPosted: Wed Dec 03, 2008 11:09 am    Post subject: [Freeswitch-users] Console Dialing in Freeswitch Reply with quote

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 (yudha2008@gmail.com)> wrote:



Quote:
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


[url=mailto:freeswitch@hp30094686650.optimus.co.in]freeswitch@hp30094686650.optimus.co.in (freeswitch@hp30094686650.optimus.co.in)[/url]> 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
[url=mailto:freeswitch@hp30094686650.optimus.co.in]freeswitch@hp30094686650.optimus.co.in (freeswitch@hp30094686650.optimus.co.in)[/url]> 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] [url=mailto:freeswitch@hp30094686650.optimus.co.in]freeswitch@hp30094686650.optimus.co.in (freeswitch@hp30094686650.optimus.co.in)[/url]> 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: [1002@172.20.179.201 (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


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anthony.minessale at g...
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PostPosted: Wed Dec 03, 2008 12:26 pm    Post subject: [Freeswitch-users] Console Dialing in Freeswitch Reply with quote

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 ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400
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yudha2008 at gmail.com
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PostPosted: Thu Dec 11, 2008 4:01 am    Post subject: [Freeswitch-users] Console Dialing in Freeswitch Reply with quote

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 (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 (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 (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 (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 (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 (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 (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
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mike at jerris.com
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PostPosted: Thu Dec 11, 2008 6:34 am    Post subject: [Freeswitch-users] Console Dialing in Freeswitch Reply with quote

Check your dialplan, you have a gateway name that is not configured
properly.


On Dec 11, 2008, at 3:51 AM, Baskar wrote:

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


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