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mod_opal segmentation fault error
by Rogelio Perez 09 Sep '09

09 Sep '09

Hi guys, My FS setup was working smoothly with mod_opal enabled until I had to rebuild everything from scratch. Now I have compiled everything following the same procedure (I even have a script for that) and mod_opal stopped working. The SVN commands are: svn co https://opalvoip.svn.sourceforge.net/svnroot/opalvoip/ptlib/ trunk ptlib svn co https://opalvoip.svn.sourceforge.net/svnroot/opalvoip/opal/branches/v3_6 opal ...and the compilation commands follow the documentation isntructions and there are no output errors. I start FS with mod_opal disabled and then when I run "load mod_opal"I get the error: Segmentation fault (core dumped). The log output shows nothing, and I see there are core.xxxxx files on the FS directory but I dont know how to read them. Any ideas? Thanks, Rogelio

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Mod fax best practices
by Rob Forman 09 Sep '09

09 Sep '09

Hi all, I built a mod_fax setup which is working well. Hats off to the authors for the module. I receive faxes regularly without issue. I can also send faxes, but seem to have a higher failure rate, especially with faxes over 10 pages, usually with error 21: No response after sending a page. I know fax-over-IP can be dicey sometimes (don't know if T.38 would help but see that mod_fax doesn't support yet anyway), but here's my question: Are there any best practices when using mod_fax? Codecs to use or avoid, jitter settings, OS tuning, etc? Things that you guys have learned through live use. Sorry if its a newbie question but I've read through the documentation and wiki but haven't seen much in this. About the system: Dedicated hosted server, Debian 5, Freeswitch 1.0.4. Nothing else running on the system. Any thoughts or lessons learned would be greatly appreciated. Cheers, Rob Forman

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Call Forwarding Question
by Nikolai Geordzhev 09 Sep '09

09 Sep '09

Hi, I`m trying to implement Call Forwarding in my FS setup. I set a user variable managing the type of forwarding (busy,no answer,unconditional) and the destination the phone is forwarded to: <user id='102'> <variables> <variable name='fwd_type' value='unconditional'/> <variable name='fwd_unconditional_number' value='103"/> </variables> </user> Then in my dialplan I check if the called user has ${fwd_type} set and if so I set the destination_number to ${fwd_unconditional_number}. After this operation I bridge the call to the new destination_number. The issue in this scenario is that I get a cdr in which A (101) calls C(103) and I need to have two cdrs, one from A(101) to B(102) and one from B(102) to C(103). Does anyone know how can this be implemented. 10x, NG

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[ERR] mod_sofia.c:2645 Invalid Gateway
by Jerry Richards 09 Sep '09

09 Sep '09

I have phones registered internally and can call among them. However, when I dial "711" from an internal phone, freeswitch replies with "484 Address Incomplete" with reason "INVALID_NUMBER_FORMAT". At the server console, I see the following error: [ERR] mod_sofia.c:2645 Invalid Gateway Does anyone know why I get this error? Is there something more I must do to add the gateway below? I already added the following to the usr/local/freesitch/conf/dialplan/default.xml: <extension name="Testing - Mediant 1000"> <condition field="destination_number" expression="^(711)$"> <action application="bridge" data="sofia/gateway/mediant1000/$1"/> </condition> </extension> I already created a usr/local/freeswitch/conf/sip_profiles/external/mediant1000.xml file: <include> <gateway name="192.168.72.253"> <param name="username" value="TEOGateWay"/> <param name="password" value="tI0W*Cj@b"/> <param name="register" value="false"/> </gateway> </include> Best Regards, Jerry

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stability problems
by Christian Löschenkohl 09 Sep '09

09 Sep '09

hello we have regular (every 4-6 days) stability problems with freeswitch when the problme occurs - no registers are done bythe server (olny 1 ack of the initial register) - no more calls are working - the calls are all ending with a timeout (cdr caues ORIGINATOR_CANCEL) - only a restart of the whole server cures the problem the server doesn't crash or segfault my first try was to enable the crash-protection flag, but with no difference the server is restartet every night and the last stand still was after about 15h uptime the system is an sun fire 2400 with debian 64 bit system how could i offer you more information to solve this big problem br -- Ing. Christian Löschenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl(a)xpirio.com

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auto_hunt=true vs execute_extenstion
by Max Ivanov 09 Sep '09

09 Sep '09

Hi all! Is there any difference between auto_hunt=True and execute_extenstion?

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mod_fax not working
by Mathieu Parent 09 Sep '09

09 Sep '09

Hi, I just started testing Freeswitch, this works well. This is powerfull and easy to administrate. I just have one problem and one request. The request is BOUNTY-5: IMAP integration of voicemail (I will try to propose a patch if this is within my competencies) The problem is with mod_fax not receiving any fax. Relevant extension is: <extension name="fax2mail"> <condition field="destination_number" expression="^(9998)$"> <action application="set" data="fax_verbose=true"/> <action application="set" data="fax_disable_v17=false"/> <action application="set" data="fax_use_ecm=true"/> <action application="answer" /> <action application="playback" data="silence_stream://2000"/> <action application="rxfax" data="/tmp/${uuid}.rxfax.tiff"/> <action application="hangup"/> </condition> </extension> Test case is a 1-page fax coming from a Minolta printer connected to a Cisco VG248 (via RJ11). The VG248 port is registred on a Cisco Call Manager communicating with FS via a SIP trunk. Log is attached: - first try from 11:04:05 to 11:04:31 - 2nd try from 11:04:57 to 11:05:22 - 3rd try from 11:05:48 to 11:06:08 Do you need more information? Regards Mathieu Parent

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Using mod_managed to create full FreeSWITCH modules
by Josh Rivers 09 Sep '09

09 Sep '09

The wiki says: mod_managed exposes nearly the entire FreeSWITCH C API (courtesy of SWIG). This allows creation of not just API functions and call apps, but any type of module that FreeSWITCH supports (codecs, endpoints, etc.). The types are in the FreeSWITCH.Native namespace. FreeSWITCH.Native. The FreeSWITCH.Native.freeswitch type contains static members to access all the functions. Does anybody have a starting point they can share for a non-API/APP managed module. I'd like to build something that runs in the SWITCH_MODULE_SHUTDOWN_FUNCTION/SWITCH_MODULE_SHUTDOWN_FUNCTION/SWITCH_MODULE_SHUTDOWN_FUNCTION lifecycle. How can this be done? Thanks! Josh

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nibblebill zero balance
by Gabriel Kuri 09 Sep '09

09 Sep '09

We've been testing mod_nibblebill, it's a great module, cheers to the author! A couple questions regarding nibblebill: 1) We noticed that when an account is below the minimum balance and a call is attempted with that account, FS begins to connect the B leg for the call but then cancels the INVITE. Why is FS sending the INVITE for the B leg only to CANCEL it immediately, if the account is below the minimum balance? I would think it would just send the 404 immediately back to the user without sending any INVITEs to establish the B leg? 2) Any progress on implementing minimum billing increments (ie 60/60) ? Cheers, Gabe

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ESL: DTMF event is not coming
by Nagalenoj 09 Sep '09

09 Sep '09

Dear friends, I am using freeswitch-1.0.4. When I execute the sample script(/libs/esl/perl/server2.pl), it is not receiving the DTMF events. When I execute the same program in freeswitch-1.0.3, it's receiving the event. Do I miss something to configure/upgrade.? -- View this message in context: http://www.nabble.com/ESL%3A-DTMF-event-is-not-coming-tp25326328p25326328.h… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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