May 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] text to speech IVRs and MOH
by pete＠privateconnect.com 22 May '09

22 May '09

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Retrieving voicemail
by Larry Marshall 22 May '09

22 May '09

In the http://wiki.freeswitch.org/wiki/Getting_Started_Guide under Basic Calling, it states "Call your own extension number to login to your voicemail box". When I dial my extension (1000) from an internal softphone at the same extension, it just asks me to leave message. It does not give me the opportunity to hear my messages. What am I missing. I am using the default configuration. Thanks, Lars

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Best way to determine if a bridge was successful in Lua
by Matthew Fong 22 May '09

22 May '09

I'm using a lua script to control an IVR, and would like to know how I can tell if a session:execute("bridge","sofia/gateway/blahblah"); was successful or not it seems the response from session:execute is nil regardless if the bridge was successful or not whats the best way? Thanks --matt

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Re: [Freeswitch-users] Segmentation fault with xmlrpc shutdown?
by Lon Baker 21 May '09

21 May '09

I updates to the latest trunk this morning and still get the same segmentation issue on 32 bit Ubuntu server 9. The exact command I am issuing is fsctl shutdown asap. Lon

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Conference Statistics
by Jon DiVita 21 May '09

21 May '09

Hello, all. Being new to FS, I was curious if there are any logs/cdrs which could be generated to gather statistics about a conference call? I'm mainly looking for call duration and user count. So far, my CDR's only have individual user CDR's, but nothing for a conference bridge. Thanks!

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Documentation error?
by Larry Marshall 21 May '09

21 May '09

On the page http://wiki.freeswitch.org/wiki/Configuring_SIP under Configuration, it speaks about the vars.xml file. Specifically it states, "In this file, there is only one parameter that you need to specify. That parameter is $${sip_profile}." I can't find the variable, nor can I grep for its assignment in conf. Am I missing something? Lars

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Segmentation fault with xmlrpc shutdown?
by Lon Baker 21 May '09

21 May '09

When I issue a fsctl shutdown via xmlrpc I get a segmentation fault on Ubuntu server 9. It a 32 bit version with all packages fully up to date running freeswitch 1.0.3 Any ideas? Lon

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JavaScript session Transfer
by Baskar 20 May '09

20 May '09

*Hi, I have an issue in transfer the call through JavaScript session. In JavaScript Session i have dialed 2 numbers One is mobile number and another one is extension number I want both the call to transfer in the conference room using JavaScript session Whether transfer is possible in JavaScript session . If possible means how? Can any one assist me to solve above process. Thanks in advance. -- Warm Regards, N.Baskar *

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openzap and progress detection
by Francois Delawarde 20 May '09

20 May '09

Hello, I'm in Spain with an analog TDM400 Clone from OpenVox (with 1xFXO +1xFXS), and am trying first to make the FXO work with Openzap and Freeswitch (using dahdi 2.2.0-rc4). Openzap perfectly detects and loads the spans, but I'm currently enable to dial out with the FXO module, it doesn't dial anything and times-out after 30 seconds. I believe it has to do with some tone detection and therefore have a few questions: - When I plug in the line, dahdi sends an event (event 17) that is ignored by OZ. Can we enable some type of battery check in OZ before dialing out, or is there some variable to monitor battery (oz dump doesn't show the battery status)? - Does OZ use the polarity switch events sent by dahdi (in kewlstart mode) for answer and hangup detection? - Apparently, OZ does tone progress detection by frequency, but here in Spain, most tones use the same 425Hz frequency with different on-off timing. Is it possible to detect those? - As some PBX in Spain transfer calls by first hanging up and picking up on another phone, can we enable/disable parts of polarity switch and/or tone progress detection (ex: (hangup)/(answer)onpolarityswitch in Asterisk) - Some lines here are connected to very old FXS from operators that have low sound quality and can take a few seconds to give a dial tone when picking up. Is it possible to introduce a delay before sending DTMF digits when dialing? Is it possible to "relax" DTMF detection, and tweak DTMF settings (make them a bit longer, with a longer pause for the other side to detect)? My "ideal" case to make it work in every case around here would be to: - have OZ fail to dial if battery is not present (and be able to fetch battery status somehow) - disable tone progress (sometimes call ends up on some local PBX that answers and provides US tones which are different) - be able to have an initial pause before dialing with DTMF digits - use polarity switch to detect remote answer, but not hangup (for transfer issues) Is the above possible? Thanks in advance, François.

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Core Dump with signal 6, Aborted
by Mariusz Kołodziejczyk WP 20 May '09

20 May '09

Hello I have a problem with core dumped with signal 6 Aborted in gdb looks: Core was generated by `/usr/local/freeswitch//bin/freeswitch'. Program terminated with signal 6, Aborted. .... .... .... #0 0xb7f4f410 in __kernel_vsyscall () #1 0xb7c11085 in raise () from /lib/tls/i686/cmov/libc.so.6 #2 0xb7c12a01 in abort () from /lib/tls/i686/cmov/libc.so.6 #3 0xb7c49b7c in ?? () from /lib/tls/i686/cmov/libc.so.6 #4 0xb7cd3138 in __fortify_fail () from /lib/tls/i686/cmov/libc.so.6 #5 0xb7cd30f0 in __stack_chk_fail () from /lib/tls/i686/cmov/libc.so.6 #6 0xb7ee6f64 in __stack_chk_fail_local () from /usr/local/freeswitch/lib/libfreeswitch.so.1 #7 0xb7e345d8 in record_callback (bug=0xb01c0488, user_data=0xa73892b8, type=63780) at src/switch_ivr_async.c:455 #8 0x0000f844 in ?? () Backtrace stopped: previous frame inner to this frame (corrupt stack?) FreeSwitch version: FreeSWITCH Version 1.0.3 (exported) Linux: Linux pbx 2.6.24-23-server #1 SMP Mon Jan 26 00:55:21 UTC 2009 i686 GNU/Linux I'm using ISDN PRI via digium card and sip gateways This problem occurs when i'm using only voip for outbound calls. By default inbound calls alway comes from ISDN PRI. Thanks

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