December 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

How to change Hang UP Cause and send it to the A-Leg
by Josue Diaz Cruz 13 Dec '11

13 Dec '11

I am trying to send a different hangup cause to the caller party. In the Dial plan i am using this public xml file (the asterisks "*" in the code are digits ommited): <extension name="lcr_******"> <condition field="network_addr" expression="***.***.***.***"/> <condition field="destination_number" expression="^5***#(\d+)$"> <action application="set" data="absolute_codec_string=G729"/> <action application="set" data="continue_on_fail=yes"/> <action application="set" data="sip_ignore_remote_cause=true"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="sofia/gateway/co*****/5***#$1"/> <action application="hangup" data="NORMAL_CIRCUIT_CONGESTION"/> </condition> </extension> when a call is made, the dialplan shows me that all is ok This is the Freeswitch server A who play the normal circuit congestion dialplan: 2011-12-13 12:59:37.390188 [NOTICE] switch_channel.c:907 New Channel sofia/external/***********(a)109.***.28.*** [5a591069-4d76-45c9-93ea-f38e85d17aa2] 2011-12-13 12:59:37.390188 [INFO] mod_dialplan_xml.c:336 Processing *******<********>->5*******#21********** in context public 2011-12-13 12:59:37.390188 [NOTICE] switch_channel.c:907 New Channel sofia/external/5*******#21********** [11fc5e4e-045f-4ce9-81ec-5049c248cb34] 2011-12-13 12:59:37.414191 [NOTICE] sofia.c:5872 Hangup sofia/external/5*******#21********** [CS_CONSUME_MEDIA] [CALL_REJECTED] 2011-12-13 12:59:37.426196 [INFO] mod_dptools.c:2696 Originate Failed. Cause: CALL_REJECTED 2011-12-13 12:59:37.426196 [NOTICE] mod_dptools.c:916 Hangup sofia/external/***********(a)109.***.28.*** [CS_EXECUTE] [NORMAL_CIRCUIT_CONGESTION] 2011-12-13 12:59:37.426196 [NOTICE] switch_core_session.c:1353 Session 304414 (sofia/external/***********(a)109.***.28.***) Ended 2011-12-13 12:59:37.426196 [NOTICE] switch_core_session.c:1355 Close Channel sofia/external/***********(a)109.***.28.*** [CS_DESTROY] 2011-12-13 12:59:37.426196 [NOTICE] switch_core_session.c:1353 Session 304415 (sofia/external/5*******#21**********) Ended 2011-12-13 12:59:37.426196 [NOTICE] switch_core_session.c:1355 Close Channel sofia/external/5*******#21********** [CS_DESTROY] And this is the other freeswitch i am using for testing pourposes to check if the cause code is the right one: 2011-12-13 16:05:13.556630 [NOTICE] switch_channel.c:779 New Channel sofia/external/***********(a)109.***.28.*** [8a188730-f366-4a49-964e-e4e66fae973c] 2011-12-13 16:05:13.560645 [INFO] mod_dialplan_xml.c:331 Processing 1000 <1000>->500321261452652 in context public 2011-12-13 16:05:13.560645 [NOTICE] switch_channel.c:779 New Channel sofia/external/5*******#21********** [e03f896f-f935-42f9-b83f-49a30de216e7] 2011-12-13 16:05:13.600636 [NOTICE] sofia.c:5051 Hangup sofia/external/5*******#21********** [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] 2011-12-13 16:05:13.604647 [INFO] mod_dptools.c:2393 Originate Failed. Cause: NORMAL_TEMPORARY_FAILURE 2011-12-13 16:05:13.604647 [NOTICE] mod_dptools.c:2456 Hangup sofia/external/***********(a)109.***.28.*** [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] 2011-12-13 16:05:13.604647 [NOTICE] switch_core_session.c:1228 Session 108345 (sofia/external/***********(a)109.***.28.***) Ended 2011-12-13 16:05:13.604647 [NOTICE] switch_core_session.c:1230 Close Channel sofia/external/***********(a)109.***.28.*** [CS_DESTROY] 2011-12-13 16:05:13.604647 [NOTICE] switch_core_session.c:1228 Session 108346 (sofia/external/5*******#21**********) Ended 2011-12-13 16:05:13.604647 [NOTICE] switch_core_session.c:1230 Close Channel sofia/external/5*******#21********** [CS_DESTROY] For me to receive a 34 or a 41 it is not a problem, but for my client they can manage 34 but not 41 error to redirect traffic to other provider. Can someone help me with this issue? I beg you pardon for my english Josue Diaz Cruz Departamento Tecnico y Soporte <mailto:jdiaz@coinfru.com> jdiaz(a)coinfru.com

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mod_say_pt sound file list
by Pete 13 Dec '11

13 Dec '11

Hi, Could anybody please provide a list of sound files that mod_say_pt requires ? Regards, Pete

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Error in my_thread_global_end(): nn threads didn't exit
by Anthony Cosgrove 13 Dec '11

13 Dec '11

Hey folks, So I'm using Lua to process fax calls but I've noticed that after the call has ended and things have settled down I get the following message on the console: Error in my_thread_global_end(): nn threads didn't exit From the testing I've been doing it seems to happen when I use the ODBC functions of freeswitch.Dbh() -- I've posted my debug output on pastebin but it didn't reveal much if anything. Log output @ http://pastebin.freeswitch.org/17995 Lua code being used @ http://pastebin.freeswitch.org/17997 Has anyone else gotten this message with their ODBC functions? Threads not exiting = menos buenos Thanks, Anthony (zorprime)

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Please help me
by Farooq Hussain 12 Dec '11

12 Dec '11

hello everyone, I am using FXO Card with freeTDM. Now I can make calls from FXO and I want to receive a call from my cell phone but We are not getting where we went we wrong and sip error code destination out of order. <include> <extension name="public_did"> <condition field="destination_number" expression="^(+92321402191)$"> <action application="set" data="domain_name=$${domain}"/> <action application="say" data="en number iterated ${ani}"/> </condition> </extension> </include> -- Thanks Farooq Hussain

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spandsp/src/dtmf.c: 'duration' isn't a membership of 'dtmf_rx_state_t'?
by mazilo 12 Dec '11

12 Dec '11

In recent commit (70c1c03c93f172ea18e33350897d58e3a82028d0), I noticed SPANDSP has undergone some heavy overhaul. Of particularly interesting to me is the src/dtmf.c file (which also got patched). ATM, I just couldn't locate a reference within the scope of spandsp source on the 'duration' membership in 'dtmf_rx_state_t' as shown in the following excerpt of src/dtmf.c patch: Anyone? ----- FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/spandsp-src-dtmf-c-duration-i… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Invalid SAY Interface [en]
by gautam 12 Dec '11

12 Dec '11

Hello, The voicmail and IVR prompts in my system are not saying out the numbers. The rest of the playback is happening correctly. In the log I'm getting the following error for the digits: >>[ERR] switch_ivr_play_say.c:287 Invalid SAY Interface [en]! The prompt that I here is something like this: "The person at extension is not available" and "Press to play the recording, press to listen to the recording. " The files are there in the digits folder, and the voicemail is getting saved correctly. Please help me fix this. Gautam -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Invalid-SAY-Interface-en-tp70… Sent from the freeswitch-users mailing list archive at Nabble.com.

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[INCOMPATIBLE_DESTINATION] Call failed - Fritzbox/Freeswitch - help required
by Sherif Omran 12 Dec '11

12 Dec '11

Hello I am trying to configure Freeswitch, so that 2 telephones each one is connected to a fritzbox and at different locations can speak. The call rings but when the callee lifts up, call drops. I would appreciate if any body can help. Below is a log of the call Thanks in advance regards, Sherif 2011-12-12 07:38:32.098496 [DEBUG] switch_core_session.c:1164 Session 45 (sofia/sipinterface_1/sip:2004@78.138.90.31) Locked, Waiting on external entities 2011-12-12 07:38:32.101287 [DEBUG] switch_core_codec.c:146 sofia/sipinterface_2/2005(a)sip.server.com Restore previous codec PCMA:8. 2011-12-12 07:38:32.101287 [DEBUG] switch_ivr_originate.c:3228 Originate Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION] 2011-12-12 07:38:32.101287 [ERR] switch_ivr_originate.c:2430 Cannot create outgoing channel of type [user] cause: [INCOMPATIBLE_DESTINATION] 2011-12-12 07:38:32.101287 [DEBUG] switch_ivr_originate.c:3228 Originate Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION] 2011-12-12 07:38:32.101287 [INFO] mod_dptools.c:2355 Originate Failed. Cause: INCOMPATIBLE_DESTINATION 2011-12-12 07:38:32.101287 [NOTICE] mod_dptools.c:2418 Hangup sofia/sipinterface_2/2005(a)sip.server.com [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] 2011-12-12 07:38:32.101287 [DEBUG] switch_channel.c:2102 Send signal sofia/sipinterface_2/2005(a)sip.server.com [KILL] 2011-12-12 07:38:32.101287 [DEBUG] switch_core_session.c:1021 Send signal sofia/sipinterface_2/2005(a)sip.server.com [BREAK] 2011-12-12 07:38:32.101287 [DEBUG] switch_core_state_machine.c:348 (sofia/sipinterface_2/2005(a)sip.server.com) State EXECUTE going to sleep 2011-12-12 07:38:32.101287 [DEBUG] switch_core_state_machine.c:314 (sofia/sipinterface_2/2005(a)sip.server.com) Running State Change CS_HANGUP 2011-12-12 07:38:32.101287 [DEBUG] switch_core_state_machine.c:499 (sofia/sipinterface_2/2005(a)sip.server.com) State HANGUP 2011-12-12 07:38:32.101287 [DEBUG] mod_sofia.c:408 sofia/sipinterface_2/2005(a)sip.server.com Overriding SIP cause 488 with 488 from the other leg 2011-12-12 07:38:32.101287 [DEBUG] mod_sofia.c:414 Channel sofia/sipinterface_2/2005(a)sip.server.com hanging up, cause: INCOMPATIBLE_DESTINATION 2011-12-12 07:38:32.101287 [NOTICE] switch_core_session.c:1182 Session 45 (sofia/sipinterface_1/sip:2004@78.138.90.31) Ended 2011-12-12 07:38:32.101287 [NOTICE] switch_core_session.c:1184 Close Channel sofia/sipinterface_1/sip:2004@78.138.90.31 [CS_DESTROY] 2011-12-12 07:38:32.101287 [DEBUG] switch_core_state_machine.c:428 (sofia/sipinterface_1/sip:2004@78.138.90.31) Running State Change CS_DESTROY 2011-12-12 07:38:32.101287 [DEBUG] switch_core_state_machine.c:439 (sofia/sipinterface_1/sip:2004@78.138.90.31) State DESTROY 2011-12-12 07:38:32.101287 [DEBUG] mod_sofia.c:341 sofia/sipinterface_1/sip:2004@78.138.90.31 SOFIA DESTROY 2011-12-12 07:38:32.101287 [DEBUG] switch_core_state_machine.c:60 sofia/sipinterface_1/sip:2004@78.138.90.31 Standard DESTROY 2011-12-12 07:38:32.101287 [DEBUG] switch_core_state_machine.c:439 (sofia/sipinterface_1/sip:2004@78.138.90.31) State DESTROY going to sleep 2011-12-12 07:38:32.107730 [DEBUG] mod_sofia.c:476 Responding to INVITE with: 488 2011-12-12 07:38:32.107730 [DEBUG] switch_core_state_machine.c:46 sofia/sipinterface_2/2005(a)sip.server.com Standard HANGUP, cause: INCOMPATIBLE_DESTINATION 2011-12-12 07:38:32.107730 [DEBUG] switch_core_state_machine.c:499 (sofia/sipinterface_2/2005(a)sip.server.com) State HANGUP going to sleep 2011-12-12 07:38:32.107730 [DEBUG] switch_core_state_machine.c:333 (sofia/sipinterface_2/2005(a)sip.server.com) State Change CS_HANGUP -> CS_REPORTING 2011-12-12 07:38:32.107730 [DEBUG] switch_core_session.c:1021 Send signal sofia/sipinterface_2/2005(a)sip.server.com [BREAK] 2011-12-12 07:38:32.107730 [DEBUG] switch_core_state_machine.c:314 (sofia/sipinterface_2/2005(a)sip.server.com) Running State Change CS_REPORTING 2011-12-12 07:38:32.107730 [DEBUG] switch_core_state_machine.c:590 (sofia/sipinterface_2/2005(a)sip.server.com) State REPORTING 2011-12-12 07:38:32.107730 [DEBUG] switch_core_state_machine.c:53 sofia/sipinterface_2/2005(a)sip.server.com Standard REPORTING, cause: INCOMPATIBLE_DESTINATION 2011-12-12 07:38:32.107730 [DEBUG] switch_core_state_machine.c:590 (sofia/sipinterface_2/2005(a)sip.server.com) State REPORTING going to sleep 2011-12-12 07:38:32.107730 [DEBUG] switch_core_state_machine.c:327 (sofia/sipinterface_2/2005(a)sip.server.com) State Change CS_REPORTING -> CS_DESTROY 2011-12-12 07:38:32.107730 [DEBUG] switch_core_session.c:1021 Send signal sofia/sipinterface_2/2005(a)sip.server.com [BREAK] 2011-12-12 07:38:32.107730 [DEBUG] switch_core_session.c:1164 Session 44 (sofia/sipinterface_2/2005(a)sip.server.com) Locked, Waiting on external entities 2011-12-12 07:38:32.107730 [NOTICE] switch_core_session.c:1182 Session 44 (sofia/sipinterface_2/2005(a)sip.server.com) Ended 2011-12-12 07:38:32.107730 [NOTICE] switch_core_session.c:1184 Close Channel sofia/sipinterface_2/2005(a)sip.server.com [CS_DESTROY] 2011-12-12 07:38:32.107730 [DEBUG] switch_core_state_machine.c:428 (sofia/sipinterface_2/2005(a)sip.server.com) Running State Change CS_DESTROY 2011-12-12 07:38:32.107730 [DEBUG] switch_core_state_machine.c:439 (sofia/sipinterface_2/2005(a)sip.server.com) State DESTROY 2011-12-12 07:38:32.107730 [DEBUG] mod_sofia.c:341 sofia/sipinterface_2/2005(a)sip.server.com SOFIA DESTROY 2011-12-12 07:38:32.107730 [DEBUG] switch_rtp.c:584 [ zrtp engine]: STOP STREAM ID=10 mode=CLEAR state=NOZRTP. 2011-12-12 07:38:32.107730 [DEBUG] switch_rtp.c:584 [ zrtp]: Stream ID=0 UNKNOWN switching <NONE> ---> <NONE>. 2011-12-12 07:38:32.107730 [DEBUG] switch_rtp.c:584 [ zrtp engine]: STOP STREAM ID=0 mode=UNKNOWN state=NONE. 2011-12-12 07:38:32.107730 [DEBUG] switch_rtp.c:584 [ zrtp engine]: STOP STREAM ID=0 mode=UNKNOWN state=NONE. 2011-12-12 07:38:32.107730 [DEBUG] switch_core_state_machine.c:60 sofia/sipinterface_2/2005(a)sip.server.com Standard DESTROY 2011-12-12 07:38:32.107730 [DEBUG] switch_core_state_machine.c:439 (sofia/sipinterface_2/2005(a)sip.server.com) State DESTROY going to sleep

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Problems with local phones not registering
by Ted Schober 12 Dec '11

12 Dec '11

I have freeswitch running with the internal and dialplan .xml files near virgin. (I added PASSWORD.xml as suggested in the Wiki One account on each of two of my Grandstream GXP-2000 phones on the eth0 local net of 192.168.61.X register just fine to default extensions of the dialplan. They get their addresses from a home routher by DHCP. I have been unable to get any other accounts on these phones, or any additional phones to register. The second phone registers on port 5062 (the default for account 2) for some reason. It will not register on 5060 even if I turn off account 2. The registered phones call and receive calls from each other and the provider gateway. I have checked the phone options, and they are identical except for the phone IP, the phone extension number. The FS server does have three NIC cards - one on another segment, and one that is disconnected. Any ideas where to look for the problem? Thanks!

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Calls failing using Second Life Viewer and Parcels
by Jayden 12 Dec '11

12 Dec '11

Hi everyone, I am new to Freeswitch and am a first time poster here so any help would be much appreciated. I have tried to search for similar problems but have had no success. I have recently started using Freeswitch 1.0.head (2011-11-30) on Windows because I had trouble compiling the source version obtained from git. I am using Opensim version 0.7.2 and Second Life Viewer 3.2.1 and I am using parcels to restrict communication to certain areas. The problem I am having is that when I move through parcels too quickly the call will fail to reconnect to the Freeswitch service and at this point I have not been able to find a way to reconnect without logging out and back in to the second life client. I realise this is not purely due to Freeswitch, as when using Hippo Viewer (0.6.3) I seem to be unable to "break" the connection in this manner, however Hippo Viewer is not an option because it does not allow the viewing of mesh objects. If anyone can suggest a way that I might be able to get around this by either changing some Freeswitch configurations or Second Life settings that would be great. Otherwise, if there are any ways to simply reconnect to the service without logging in and out of Second Life that would still be of great help, as the project I am working on will be used as a training simulation and logging in and out is not likely to be an acceptable solution by the client. If you would like to help and would like to see any console outputs just let me know and I will try to provide them. Thanks in advance, Jayden -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Calls-failing-using-Second-Li… Sent from the freeswitch-users mailing list archive at Nabble.com.

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conference xxxx record .mp3 file is shorter than actual time.
by Stephen Dame 11 Dec '11

11 Dec '11

Quick question.. Recording 1hr 55min conference is only 1:49 long. No voice data appears to be lost. Not sure if it's a sampling issue, or silence not getting recorded in .mp3.. I have tried a few times and looks like it's about 10% shorter and tried with 16,000 and 48,000 conferences I thought it might have been energy setting, set to 0. I'm also recording out to shout: and can see data 100% of time transmitting in media player. Is there a setting in freeswitch, or lame to ensure exact timing on recorded file. I'm setting recording from fs_cli with conference xxxxx record path/file.mp3 Not sure how to set a different rate in fs_cli for recording. Thanks in advance. Regards, Stephen

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