August 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

EXCHANGE_ROUTING_ERROR after ~70 transfer to dialplan extebsion
by Kozak Vladimir 03 Aug '11

03 Aug '11

Hi all. FreeSWITCH Version 1.0.head (git-9ff8f53 2011-05-03 12-13-52 -0400) I have problem with transfer channel to FS extension. I send from external system to FS command as "SendMsg " + uniqueId + "\n" + "call-command: execute\n" + "execute-app-name: transfer\n" + "execute-app-arg: " + destination + "\n\n"; after ~70 correct transfer actions FS send BYE to my phone and send HANGUP_EVENT with cause EXCHANGE_ROUTING_ERROR to my system. (FS log attached) from FS loggs: 2011-08-09 04:48:38.525293 [DEBUG] mod_sofia.c:457 Channel sofia/internal/1000(a)vkozak.starpoundtech.net hanging up, cause: EXCHANGE_ROUTING_ERROR 2011-08-09 04:48:38.525293 [DEBUG] mod_sofia.c:500 Sending BYE to sofia/internal/1000(a)vkozak.starpoundtech.net what for does limitation of transfer operations exist? how can I avoid this issue? is't possible to increase number of hops for transfer command?

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Help setting up SIP reg
by lloyd thomas 03 Aug '11

03 Aug '11

*Hi I need a little help setting up a SIP registration for a provider that does not use auth.* *All I have is info below.* ** * * SBC/Proxy IP: 80.40.150.150:5060 Authentication: Trusted IP – 88.221.85.33 Assigned DDI: 01869******, 01869****** DTMF Method: RFC2833 Status: Live No. of trunks: 2x Session Timer: 1800 Profile*:* Generic (35060) Apparently the following is used for * [vibe] type = friend host = 80.40.150.150

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FreeSWITCH Conference Call Today
by Michael Collins 03 Aug '11

03 Aug '11

Hello all! Today's agenda is: http://wiki.freeswitch.org/wiki/FS_weekly_2011_08_03 We have a few things to discuss plus a sneak peek at a new SIP logging project that will be presented at ClueCon. (We'll have the author on a future conf call to talk to us.) -Michael

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Grandstreams and TLS
by Joshua Foshee 03 Aug '11

03 Aug '11

Has anyone got the GXV3175 or GXV3140 to work with TLS? I have tried TLSV1 and SSL23 on the TLS but neither one of them will get to authenticate correctly. If you have can you post the Freeswitch TLS config and the phone settings that you got to work? Thanks, Josh

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Cancel sent on outgoing call completion
by Chris Cureau 03 Aug '11

03 Aug '11

I'm hoping that someone might be able to figure this one out...I've tried everything that I know of! On an outbound call, I can see the buildup and the connection to the gateway. The calked party begins to ring very briefly, and then it looks like a CANCEL is being sent. I'm pretty sure it isn't isolated to a phone since all my phones are doing it now. This is using current git as of last night. http://pastebin.freeswitch.org/16953 Any hints appreciated! -- Chris

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Calls not established : See the error details
by Dhairya Vora 03 Aug '11

03 Aug '11

I don't understand why two users (registered directly to freeswitch) are unable to connect each other? When a registered user 1001 calls registered user 1002, call fails giving this error ************************************************************************************************************************************************************ freeswitch(a)localhost.localdomain> 2011-08-03 10:07:01.941089 [WARNING] sofia_reg.c:1337 SIP auth challenge (INVITE) on sofia profile 'internal' for [1002(a)172.16.10.211] from ip 172.16.10.248 2011-08-03 10:07:01.941089 [NOTICE] switch_channel.c:897 New Channel sofia/internal/1000(a)172.16.10.211 [0bf31c06-8379-4831-88d7-a7df3c4ecd49] 2011-08-03 10:07:01.941089 [INFO] mod_dialplan_xml.c:336 Processing 1000 <1000>->1002 in context default 2011-08-03 10:07:01.941089 [INFO] switch_core_session.c:1281 sofia/internal/ 1000(a)172.16.10.211 setting session heartbeat to 60 second(s). 2011-08-03 10:07:01.941089 [ERR] mod_event_socket.c:457 Socket Error! 2011-08-03 10:07:01.941089 [NOTICE] switch_core_state_machine.c:189 sofia/internal/1000(a)172.16.10.211 has executed the last dialplan instruction, hanging up. 2011-08-03 10:07:01.941089 [NOTICE] switch_core_state_machine.c:191 Hangup sofia/internal/1000(a)172.16.10.211 [CS_EXECUTE] [NORMAL_CLEARING] 2011-08-03 10:07:01.961094 [NOTICE] switch_core_session.c:1347 Session 25 (sofia/internal/1000(a)172.16.10.211) Ended 2011-08-03 10:07:01.961094 [NOTICE] switch_core_session.c:1349 Close Channel sofia/internal/1000(a)172.16.10.211 [CS_DESTROY] ************************************************************************************************************************************************************ When I make an outbound call to my mobile, calls are not going through the custom gateway ************************************************************************************************************************************************************ freeswitch(a)localhost.localdomain> 2011-08-03 10:03:43.608234 [WARNING] sofia_reg.c:1337 SIP auth challenge (INVITE) on sofia profile 'internal' for [00919876543210(a)172.16.10.211] from ip 172.16.10.213 2011-08-03 10:03:43.728232 [NOTICE] switch_channel.c:897 New Channel sofia/internal/1002(a)172.16.10.211 [334906be-9cb0-44f2-9660-f9377c2ecfba] 2011-08-03 10:03:43.728232 [INFO] mod_dialplan_xml.c:336 Processing 1002 <1002>->00919876543210 in context default 2011-08-03 10:03:43.728232 [INFO] switch_core_session.c:1281 sofia/internal/ 1002(a)172.16.10.211 setting session heartbeat to 60 second(s). 2011-08-03 10:03:43.728232 [ERR] mod_event_socket.c:457 Socket Error! 2011-08-03 10:03:43.728232 [NOTICE] switch_core_state_machine.c:189 sofia/internal/1002(a)172.16.10.211 has executed the last dialplan instruction, hanging up. 2011-08-03 10:03:43.728232 [NOTICE] switch_core_state_machine.c:191 Hangup sofia/internal/1002(a)172.16.10.211 [CS_EXECUTE] [NORMAL_CLEARING] 2011-08-03 10:03:43.728232 [NOTICE] switch_core_session.c:1347 Session 24 (sofia/internal/1002(a)172.16.10.211) Ended 2011-08-03 10:03:43.728232 [NOTICE] switch_core_session.c:1349 Close Channel sofia/internal/1002(a)172.16.10.211 [CS_DESTROY] ************************************************************************************************************************************************************

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FS Status Command Description
by Leonardo P. Bidinoto 03 Aug '11

03 Aug '11

Hello Dan, I know this might be a noobie question, but ....... *fs_cli -x'status'* UP 0 years, 1 day, 1 hour, 2 minutes, 45 seconds, 167 milliseconds, 852 microseconds FreeSWITCH is ready 1377 session(s) since startup 153 session(s) 0/30 ==> *what means this zero before the "/30"? sometimes is changing to 1, like this ( 153 session(s) 1/30 )* 1000 session(s) max min idle cpu 0.00/76.00 -- Leonardo Pires Bidinoto Voice Technology www.voicetechnology.com.br

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Commercial support
by Chris Cureau 03 Aug '11

03 Aug '11

What's the best way to get paid support for freeswitch either on a per incident or contract basis?

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(no subject)
by Sam 03 Aug '11

03 Aug '11

I am writing a perl script to bridge a call and need to be able to pass back any early media to the caller so I have set "ignore_early_media=false". My problem is that when I am getting early media and then the call finishes, the app seems to be calling the bridge app again to make a new call, even though I have no loops programmed into the script. I have set "hangup_after_bridge=true" but that does not help. Any help would be greatly appreciated. ###################################My code############################################# #!/usr/bin/perl -w our $session; my $ringback_tone = "%(2000,4000,440,480)"; #US RINGBACK TONE if ($session->ready ()) { #set bridge settings $session->execute("set", "ringback=$ringback_tone"); #set the ringback tone type $session->execute("set", "instant_ringback=true"); #set to ring instantly $session->execute("set", "ignore_early_media=false"); #set to NOT ignore early media $session->execute("set", "call_timeout=20"); #only works if "ignore_early_media=true" $session->execute("set", "bridge_answer_timeout=20"); $session->execute("set", "progress_timeout=15"); $session->execute("set", "continue_on_fail=false"); $session->execute("set", "hangup_after_bridge=true"); $session->execute("set", "bridge_pre_execute_bleg_app=info"); $session->execute("bridge", "sofia/gateway/carrier1/5214498052059"); $session->hangup(); }

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Limit number of voicemails
by Sascha Daniels 03 Aug '11

03 Aug '11

Hi together. For a small appliance I need to limit the number of voicemails for each user. I know that I can limit the length. That is the first step. Is there a way to disable the message recording, when the maximum number of voicemails is reached? Unfortunately I didn't find anything in the documentation. Kind regards Sascha -- AMOOMA GmbH - Bachstr. 124 - 56566 Neuwied --> http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister Montabaur B14998 Bücher: http://das-asterisk-buch.de - http://ruby-auf-schienen.de

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