September 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

IAX Origination error
by Meftah Tayeb 17 Sep '09

17 Sep '09

hello, i have a problem while originating a call using mod_iax: originate IAX/guest(a)pbx.digium.com/s 1000 freeswitch will crach and exit automatikaly anthm updated the mod_iax but same problem see the trace in the Pastebin <http://pastebin:freeswitch@pastebin.freeswitch.org/1040> (http://pastebin:freeswitch@pastebin.freeswitch.org/1040) this is the asterisk definition of this extension: exten => 500,n,Dial(IAX2/guest@pbx.digium.com/s@default) ; Call the Asterisk demo any help? thanks __________ Information from ESET NOD32 Antivirus, version of virus signature database 4435 (20090917) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com __________ Information from ESET NOD32 Antivirus, version of virus signature database 4435 (20090917) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com __________ Information from ESET NOD32 Antivirus, version of virus signature database 4435 (20090917) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com

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Please Help: TDM Hardware For Testing
by Michael Collins 17 Sep '09

17 Sep '09

Hello all! We could use your help with something. We've had several volunteers willing to assist with testing and debugging various scenarios with TDM circuits. However, there is a need to get TDM hardware into the hands of those volunteers. If you have old TDM hardware, be it analog or digital, that is sitting around collecting dust and you'd like to donate it (or loan it) to the cause then please email me off list. Thanks! -MC

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mod_conference performance
by Роберт Тверитнер 17 Sep '09

17 Sep '09

Hi guys! I've tested FreeSWITCH conference module performance trying to figure out maximum number of simultaneous calls my FS box can serve. It took all 100% of CPU with only 50 calls (in average depending on conference rate) and "leaking stream handle" messages started appearing. The environment I was testing in: OS - Windows Server 2007 SP1 64 Bit CPU - Dual-core AMD Opteron 1216 HE 2.4 GHz RAM - 2 GB FreeSwitch version 1.0.4 (14460) I've written a test program that used to originate calls once in 5 seconds from the other box. These calls were routed to particular conference room I was testing. I had a number of rooms with different rate (8000-32000) and interval (20,30) settings and with perpetual-sound turned on steraming music continiously. I've switched off all unnecessary modules, but left logging on in order to trace what was happening later. Client test softphone used respective speex codec according to conference room rate. This is a dialplan I used: <extension name="test_conference"> <condition field="destination_number" expression="^(800020)$" break="on-true"> <action application="conference" data="$1@default20"/> </condition> <condition field="destination_number" expression="^(800030)$" break="on-true"> <action application="conference" data="$1@default30"/> </condition> <condition field="destination_number" expression="^(1600020)$" break="on-true"> <action application="conference" data="$1@wideband20"/> </condition> <condition field="destination_number" expression="^(1600030)$" break="on-true"> <action application="conference" data="$1@wideband30"/> </condition> <condition field="destination_number" expression="^(3200020)$" break="on-true"> <action application="conference" data="$1@ultrawideband20"/> </condition> <condition field="destination_number" expression="^(3200030)$" break="on-true"> <action application="conference" data="$1@ultrawideband30"/> </condition> </extension> My questions are: Do you know any way I can increase my FS conference capacity? What do I have to tune in FS or in my box? Best regards, Robert.

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Re: [Freeswitch-users] 480 Temporarily Unavailable/503 Max Sessions (was 502 Bad Gateway: Destination out of order error)
by email lists 17 Sep '09

17 Sep '09

Anthony, I increased the ulimit parameters as suggested. I then ran into a few other Freeswitch errors that required me to edit switch.conf.xml to increase the max sessions, and max sessions per second parameters (to 100000, and 100, respectively). Now, after a few minutes of running calls at 20 calls per second with a 3 min. duration, I start seeing an influx of "480 Temporarily Unavailable" SIP error messages. Here is a snip of the log: ------------------------------------------- 2009-09-16 19:35:19.18798 [DEBUG] switch_core_session.c:932 Send signal sofia/external/2015692700(a)72.245.84.150 [BREAK] 2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:434 (sofia/internal/9726074300@72.245.84.24:5060) State HANGUP 2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:398 (sofia/external/2015692700(a)72.245.84.150) Running State Change CS_REPORTING 2009-09-16 19:35:19.18798 [DEBUG] mod_sofia.c:338 Channel sofia/internal/9726074300@72.245.84.24:5060 hanging up, cause: NORMAL_CLEARING 2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:612 (sofia/external/2015692700(a)72.245.84.150) State REPORTING 2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:53 sofia/external/2015692700(a)72.245.84.150 Standard REPORTING, cause: NO_ANSWER 2009-09-16 19:35:19.18798 [DEBUG] mod_sofia.c:417 Responding to INVITE with: 480 2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:612 (sofia/external/2015692700(a)72.245.84.150) State REPORTING going to sleep 2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:411 (sofia/external/2015692700(a)72.245.84.150) State Change CS_REPORTING -> CS_DESTROY 2009-09-16 19:35:19.18798 [DEBUG] switch_core_session.c:1068 Session 3395 (sofia/external/2015692700(a)72.245.84.150) Locked, Waiting on external entities 2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:46 sofia/internal/9726074300@72.245.84.24:5060 Standard HANGUP, cause: NORMAL_CLEARING 2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:434 (sofia/internal/9726074300@72.245.84.24:5060) State HANGUP going to sleep 2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:434 (sofia/external/2015692915(a)72.245.84.150) State HANGUP 2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:46 sofia/external/2015692915(a)72.245.84.150 Standard HANGUP, cause: NO_ANSWER 2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:434 (sofia/external/2015692915(a)72.245.84.150) State HANGUP going to sleep 2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:476 (sofia/internal/9726074300@72.245.84.24:5060) State Change CS_HANGUP -> CS_REPORTING 2009-09-16 19:35:19.18798 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/9726074300@72.245.84.24:5060 [BREAK] 2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/9726074300@72.245.84.24:5060) Running State Change CS_REPORTING 2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:612 (sofia/internal/9726074300@72.245.84.24:5060) State REPORTING 2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:53 sofia/internal/9726074300@72.245.84.24:5060 Standard REPORTING, cause: NORMAL_CLEARING 2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:612 (sofia/internal/9726074300@72.245.84.24:5060) State REPORTING going to sleep 2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:411 (sofia/internal/9726074300@72.245.84.24:5060) State Change CS_REPORTING -> CS_DESTROY 2009-09-16 19:35:19.18798 [DEBUG] switch_core_session.c:1068 Session 3392 (sofia/internal/9726074300@72.245.84.24:5060) Locked, Waiting on external entities 2009-09-16 19:35:19.18798 [NOTICE] switch_core_session.c:1086 Session 3392 (sofia/internal/9726074300@72.245.84.24:5060) Ended 2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:434 (sofia/external/2015692915(a)72.245.84.150) State HANGUP 2009-09-16 19:35:19.18798 [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/9726074300@72.245.84.24:5060 [CS_DESTROY] 2009-09-16 19:35:19.18798 [DEBUG] mod_sofia.c:338 Channel sofia/external/2015692915(a)72.245.84.150 hanging up, cause: NO_ANSWER 2009-09-16 19:35:19.18798 [DEBUG] mod_sofia.c:406 Sending CANCEL to sofia/external/2015692915(a)72.245.84.150 2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:46 sofia/external/2015692915(a)72.245.84.150 Standard HANGUP, cause: NO_ANSWER 2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:434 (sofia/external/2015692915(a)72.245.84.150) State HANGUP going to sleep 2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:564 (sofia/internal/9726074300@72.245.84.24:5060) State DESTROY 2009-09-16 19:35:19.18798 [DEBUG] mod_sofia.c:255 sofia/internal/9726074300@72.245.84.24:5060 SOFIA DESTROY 2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:60 sofia/internal/9726074300@72.245.84.24:5060 Standard DESTROY 2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:564 (sofia/internal/9726074300@72.245.84.24:5060) State DESTROY going to sleep 2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:434 (sofia/internal/2126822269@72.245.84.24:5060) State HANGUP 2009-09-16 19:35:19.18798 [DEBUG] mod_sofia.c:338 Channel sofia/internal/2126822269@72.245.84.24:5060 hanging up, cause: NORMAL_CLEARING 2009-09-16 19:35:19.18798 [DEBUG] mod_sofia.c:417 Responding to INVITE with: 480 ------------------------------------------- Is there some other limitation on freeswitch I may be hitting? I tried throttling down to 10 calls per second (3 min. duration) and here are some snippets from tailing the log file: ------------------------------------------- 2009-09-16 20:01:47.422656 [DEBUG] switch_ivr_originate.c:2138 Originate Resulted in Error Cause: 102 [RECOVERY_ON_TIMER_EXPIRE] 2009-09-16 20:01:47.496066 [DEBUG] switch_ivr_originate.c:2138 Originate Resulted in Error Cause: 102 [RECOVERY_ON_TIMER_EXPIRE] 2009-09-16 20:01:47.623316 [DEBUG] switch_ivr_originate.c:2138 Originate Resulted in Error Cause: 102 [RECOVERY_ON_TIMER_EXPIRE] 2009-09-16 20:02:59.777664 [CRIT] mod_sofia.c:2601 Error Creating Session 2009-09-16 20:02:59.777664 [CRIT] mod_sofia.c:2601 Error Creating Session 2009-09-16 20:02:59.777664 [CRIT] mod_sofia.c:2601 Error Creating Session 2009-09-16 20:02:59.788134 [CRIT] mod_sofia.c:2601 Error Creating Session nta: sent 503 Maximum Calls In Progress for INVITE (1) nta: sent 503 Maximum Calls In Progress for INVITE (1) nta: sent 503 Maximum Calls In Progress for INVITE (1) nta: sent 503 Maximum Calls In Progress for INVITE (1) nta: sent 502 Bad Gateway for INVITE (1) nta: sent 502 Bad Gateway for INVITE (1) nta: sent 502 Bad Gateway for INVITE (1) nta: sent 503 Maximum Calls In Progress for INVITE (1) 2009-09-16 20:02:59.788134 [CRIT] mod_sofia.c:2601 Error Creating Session 2009-09-16 20:02:59.788134 [CRIT] mod_sofia.c:2601 Error Creating Session 2009-09-16 20:02:59.788134 [CRIT] mod_sofia.c:2601 Error Creating Session nta: sent 502 Bad Gateway for INVITE (1) nta: sent 502 Bad Gateway for INVITE (1) nta: sent 502 Bad Gateway for INVITE (1) nta: sent 502 Bad Gateway for INVITE (1) 2009-09-16 20:02:59.861110 [CRIT] switch_time.c:500 Over Session Rate of 100! nta: sent 480 Temporarily Unavailable for INVITE (1) nta: sent 480 Temporarily Unavailable for INVITE (1) nta: sent 480 Temporarily Unavailable for INVITE (1) nta: sent 480 Temporarily Unavailable for INVITE (1) nta: sent 480 Temporarily Unavailable for INVITE (1) nta: sent 480 Temporarily Unavailable for INVITE (1) ------------------------------------------- I'm going to try increasing the "session rate", but I'm wondering if there are other limitations I'm running into. Any thoughts/suggestions? Vladimir ________________________________________ From: freeswitch-users-bounces(a)lists.freeswitch.org [freeswitch-users-bounces(a)lists.freeswitch.org] On Behalf Of Anthony Minessale [anthony.minessale(a)gmail.com] Sent: Tuesday, September 15, 2009 6:32 PM To: freeswitch-users(a)lists.freeswitch.org Subject: Re: [Freeswitch-users] 502 Bad Gateway: Destination out of order error It's probably from this 2009-09-15 14:34:55.838365 [ERR] sofia_glue.c:2503 AUDIO RTP REPORTS ERROR: [Socket Error!] 2009-09-15 14:34:55.838365 [NOTICE] sofia_glue.c:2504 Hangup sofia/external/411(a)192.168.0.150<mailto:411@192.168.0.150> [CS_CONSUME_MEDIA][DESTINATION_OUT_OF_ORDER ] 2009-09-15 14:34:55.838365 [ERR] sofia.c:3796 RTP Error! your machine failed to produce a socket when requested. My blind guess, you are on a 32 bit machine and you do not have the ulimits set for enough file descriptors etc.. ulimit -c unlimited ulimit -d unlimited ulimit -f unlimited ulimit -i unlimited ulimit -n 999999 ulimit -q unlimited ulimit -u unlimited ulimit -v unlimited ulimit -x unlimited ulimit -s 244 ulimit -l unlimited On Tue, Sep 15, 2009 at 5:22 PM, email lists <email.list.subscriber(a)gmail.com<mailto:email.list.subscriber@gmail.co m>> wrote: Hello All, Wondering if anyone has experienced this issue before. I've attached a snip of the log file where the error occurs and could use some leads on this. What's interesting is that the call appears to complete as normal, and a radius stop message even gets generated, though the duration is ~1 second. <snip> h323-disconnect-time = "h323-disconnect-time=14:35:01.000 UTC Tue Sep 15 2009" h323-connect-time = "h323-connect-time=14:34:59.000 UTC Tue Sep 15 2009" </snip> While there are a lot of pieces involved, the call scenario is pretty basic (no transfers, no holds, etc.), just a few redirects that Freeswitch appears to be able to handle without issue. Attached is a dumb'd down call ladder. I tried different rates at which I generate the calls, but it didn't seem to correlate to the amount of errors I am seeing. Sending a total of 100 calls, with a call duration of 10 seconds: @10 calls per second = 14 "502" errors. @5 calls per second = 4 "502" errors. @4 calls per second = NO ERRORS (1st run) @4 calls per second = 39 "502" errors. Please let me know if any additional information is needed. Thanks in advance for all help. Vladimir _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org<mailto:FreeSWITCH-users@lists.fr eeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use rs http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com<mailto:MSN%3Aanthony_minessale@hotma il.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<mailto:PAYPAL%3Aanthon y.minessale(a)gmail.com> IRC: irc.freenode.net<http://irc.freenode.net> #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org<mailto:sip%3A888@conference.freeswit ch.org> iax:guest@conference.freeswitch.org/888<http://iax:guest@conference.fr eeswitch.org/888> googletalk:conf+888@conference.freeswitch.org<mailto:googletalk%3Aconf %2B888(a)conference.freeswitch.org> pstn:213-799-1400

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session record does not for very short calls
by Frank ＠ Impact 17 Sep '09

17 Sep '09

FreeSWITCH Version 1.0.trunk (12790M) I have this in my DP <action application="set" data="RECORD_ANSWER_REQ=true"/> <action application="set" data="RECORD_STEREO=true"/> <action application="record_session" data="/mnt/rd/file.wav"/> works fine as long as the call is long enough. But if the call is only, say, 3-4 seconds long (or something very short like that), then the wav file is never created with the audio in it. Is there a work around for this?

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How to process s extension in FS
by Ahmed Munir 17 Sep '09

17 Sep '09

Hi, How can I process s extension in FS? Is there other way around of doing it? Kindly advice me. -- Regards, Ahmed Munir

3 2

FreeSWITCH and OpenBTS integration
by Alberto Escudero 17 Sep '09

17 Sep '09

I am happy to let you know that FreeSWITCH route calls from OpenBTS, the open base station based on the Universal Software Radio USRP. Yes! Calls from a standard handset to a GSM base station connected to FreeSWITCH If you want to read more about the idea check: http://openbts.sourceforge.net/ http://www.it46.se/entry/380 (our effort to deploy the technology in a developing region) I have put a few notes for others to give it a try available here: https://sourceforge.net/apps/trac/openbts/wiki/OpenBTS/SettingUpFreeSWITCH Let me know what is the best place in FreeSWITCH wiki to add and keep updated this information -- Stopping junk mailers is good for the environment

5 7

DB table sip_dialogs is always empty
by kokoska.rokoska 17 Sep '09

17 Sep '09

I'm sorry to resend this post, but even after few hours I can't see it in the mailing-list... Thanks. Best regards, kokoska.rokoska kokoska rokoska napsal(a): > Hello, > > I have setted-up odbc-dsn on all my FreeSWITCH sofia profiles and based > on logs FS connected to the dsns correctly. But during the calls I can't > see any rows in sip_dialogs table. When I run "show channels" from > console it works fine. > All other tables are "populated and maintained" as I expect :-) > > I'm running FreeSWITCH on 64bit Centos 5.3 with Postgresql 8.1 DB... > > Could you, please, point what I'm missing, or where I'm wrong? Thank you > very much! > > > Best regards, > > kokoska.rokoska >

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faxrx error 13 Unexpected message received
by Travis Stutsman 17 Sep '09

17 Sep '09

In my attempts to receive a fax from a PSTN fax machine, the transaction fails with error code 13 "Unexpected message received". Verbose logging is on for mod_fax. Here is an exerpt: ################################################# 2009-09-15 10:41:39.215123 [DEBUG] mod_fax.c:137 FLOW T.30 Rx: CFR with final frame tag 2009-09-15 10:41:39.215123 [DEBUG] mod_fax.c:137 FLOW T.30 Rx: ff 13 84 2009-09-15 10:41:39.215123 [DEBUG] mod_fax.c:137 FLOW T.30 In state 12 2009-09-15 10:41:39.215123 [DEBUG] mod_fax.c:137 FLOW T.30 Unexpected CFR received in state 12 ...... ======================================================================== ====== 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:180 Fax processing not successful - result (13) Unexpected message received. 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:185 Remote station id: ********** 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:186 Local station id: SpanDSP Fax Ident 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:187 Pages transferred: 0 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:189 Total fax pages: 0 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:190 Image resolution: 8031x3850 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:191 Transfer Rate: 14400 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:193 ECM status on 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:194 remote country: 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:195 remote vendor: 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:196 remote model: 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:198 ======================================================================== ====== ################################################# I have successfully received a fax from faxzero.com on this installation, as have I successfully sent a fax from freeswitch to the PSTN fax machine in question. I've been digging, but there doesn't seem to be a whole lot of information on faxing. I guess I could use a bit of direction. Any input is much appreciated. Thanks! -- Travis

3 5

Limit_Hash
by Matt Riddell 17 Sep '09

17 Sep '09

Hi, I've moved this discussion to users as it seems my query is moving in that direction :) So, upon looking at limit_hash, it appears to do what I need to do. My question then becomes, how do I set a hash for an originated call? It seems that limit_hash is an application rather than a channel variable, and so far I've been doing most things without touching the dialplan. So, say I want to originate 9 calls, 3 from 3 customers. I would like to mark the calls with my_customer_group_1 through 3, and then use the limit_hash_usage command to verify the count of channels in each group. I therefore have a few questions: 1. Can I mark a call in the originate statement? 2. How do I use the limit_hash_usage command? The wiki states: You can verify the usage of any resource with the limit_hash_usage api call. limit_hash_usage <realm> <id> Is realm the same as a SIP realm? Is id the hash that I have used to mark the call with? Just making sure :) -- Cheers, Matt Riddell Director

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