- FreeSWITCH-users - FreeSWITCH Mailing Lists

Javascript: record ringing of session
by Birgit Arkesteijn 25 Nov '08

25 Nov '08

Hi, In Javascript, I do the following: var mSession = new Session("{ignore_early_media=true,originate_timeout=8}" + endpoint_url); mSession.execute("record_session", recordfile); This works very well, however, the client would like the rings to be recorded as well. I've tried using various channel variables, but 'new Session' only returns after the call is answered. I tried calling apiExecute("record", <file_path>); before 'new Session', but that returns an error. Is is possible to record the rings (in javascript)? And if so, how would I do that? Thanks, Birgit -- -- Birgit Arkesteijn, birgit(a)westhawk.co.uk, -- Westhawk Ltd, Albion Wharf, 19 Albion Street, Manchester M1 5LN, UK -- Company no: 1769350 -- Registered Office: -- 15 London Road, Stockton Heath, Warrington WA4 6SJ. UK. -- tel.: +44 (0)161 237 0660 -- <URL: http://www.westhawk.co.uk>

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Big problems with disconnects
by Dennis 25 Nov '08

25 Nov '08

hi, i have big problems with disconnects, when bridging and unbridging calls. because i had random diconnects when testing fs from one softphone to the other, i set up a little dialplan (socket outbound), to do some hardcore testing. after the inbound is answered, i do an originate to an outbound. when the outbound answers, i bridge the inbound and outbound. then i do uuid_transfer, 5 seconds later a uuid_bride, 5 seconds later a uuid_transfer and so on and so on. latest after 5 minutes of switching, the connection seems to hang for about 29 seconds and then ends. this ALWAYS happens after the debug message: switch_core_state_machine.c:187 switch_core_standard_on_park() sofia/internal/98765 Standard PARK see http://pastebin.freeswitch.org/6281 at line 82. i repeated that for at least 30 times - always with the same result. what could be the problem? thanks

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WANPIPE and zaptel with FreeSWITCH
by Evgeniy Zolotov 25 Nov '08

25 Nov '08

Hello all ! FreeSWITCH works good with Sangoma A 10X cards in Linux ( it was checked by us ). For this cards FreeSWITCH uses TDM API interface, which was written specially for FreeSWITCH into few last releases of WANPIPE. In this case for timeslots such names are created: /dev/wptdm_NN - ( N - is a digit), in the openzap.conf this spans called [wanpipeN]. We know that Asterisk works with WANPIPE over zaptel interface. In this case for timeslots such names are created: /dev/zap/N - ( N - is a digit), in the openzap.conf this spans called [zapN]. Does anybody try to use FreeSWITCH with such interface, when spans in the openzap.conf called [zap], but not [wanpipeN]?

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Question about user authentication on 301 redirect
by Juan Backson 25 Nov '08

25 Nov '08

Hi, I have openser set up as sip registrar, location, and redirect server. When a INVITE is received in openser, it would send back a 301 to redirect the user agent to one of the freeswitch servers. The question I have is that what mechanism can be used within freewitch to tell whether the incoming request is redirected from openser( user agents that have already)? In that case, does it mean freeswitch would have to authenticate the user agent again? Thanks for your suggestion. JB

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Jack and portaudio support in freeswitch
by Ashutosh 25 Nov '08

25 Nov '08

Hello, are there any plans to make freeswitch compatible with jack and puredata, like russel has done for asterisk ( http://www.russellbryant.net/blog/2008/01/13/jack-interfaces-for-asterisk/) . Such a setup will enable us to manipulate or play with rtp in realtime like (http://www.lobstertech.com/code/voicechanger/) . Since both jack and puredata have a posix c++ interface, it shudnt be difficult. cheers!! ashutosh

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I am a new user
by Faisal Maqsoodi 25 Nov '08

25 Nov '08

I ve just installed freeswitch on my system. What should i try on it now to take a start? I ve gone through the theory of pbx and all that. faisal

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mod_perl core dump
by regs＠kinetix.gr 25 Nov '08

25 Nov '08

I have a perl script (for dialplan generation) that works fine. When I try to use the DBI module I get a segmentation fault. My OS is Linux CentOS 5.2 and I am using freeswitch-1.0.1. If I can recall correctly, some other guy had the same problem a few months ago but I cannot find the mailing list entry. Has anyone faced this problem besides me?

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DTMF
by Baskar 25 Nov '08

25 Nov '08

Hi, In barging if we want to pass the DTMF signals. For example in barging - 2 to speak with the uuid - 1 to speak with the other half - 3 to engage a three way - 0 to restore eavesdrop. - * to next channel. I want pass these DTMF signals through event socket api uuid_send_dtmf <uuid> <dtmf_data> i did not know what is dtfm_data if dtmf_data is these DTMF signals values api uuid_send_dtmf b3df12bf-35c6-4a02-abaa-770c0d7bf358 *2* *output:* * Content-Type: api/response Content-Length: 14 -ERR no reply* I want to know what is dtmf_data? Any one correct me to solve my problem. -- Warm Regards, N.Baskar

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mod curl user directory from web service
by Antti Kokko 24 Nov '08

24 Nov '08

Hello, Thanks in advance. I have a problem with mod curl user directory. I have now configured system to fetch user directory from a web service. It works fine and the users get to the FS and users can register with a SIP client. I`m assuming the system is using default dialplans, sip-profiles etc. because I have just directory -section in xml_curl.conf. Anyway after registering if two clients wants to start a session I get immediate hangup by FS. Here is an example of my user directory: <?xml version="1.0" encoding="UTF-8" standalone="no"?> <document type="freeswitch/xml"> <section name="directory" description="User Directory"> <domain name="192.168.0.197"> <user id="duck" mailbox="duck"> <params> <param name="password" value="1234" /> <param name="vm-password" value="1234" /> </params> <variables> <variable name="accountcode" value="duck" /> <variable name="user-context" value="default" /> <variable name="effective_caller_id_name" value="duck" /> <variable name="effective_caller_id_number" value="duck" /> </variables> </user> </domain> </section> </document> And here is an example log I get from FS: 2008-11-24 14:08:48 [NOTICE] switch_channel.c:553 switch_channel_set_name() New Channel sofia/internal/duck(a)192.168.0.197[36959c5c-a2b0-f045-aac0-051b80911fb1] 2008-11-24 14:08:48 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing testuser->duck2 in context public 2008-11-24 14:08:48 [NOTICE] switch_core_state_machine.c:160 switch_core_standard_on_execute() Hangup sofia/internal/duck2(a)192.168.0.197[CS_EXECUTE] [NORMAL_CLEARING] nta: sent 480 Temporarily Unavailable for INVITE (13404) nua(03C30980): removing session usage nua(03C30980): call state changed: received -> terminated 2008-11-24 14:08:48 [NOTICE] switch_core_session.c:878 switch_core_session_thread() Session 6 (sofia/internal/duck(a)192.168.0.197) Ended 2008-11-24 14:08:48 [NOTICE] switch_core_session.c:880 switch_core_session_thread() Close Channel sofia/internal/duck(a)192.168.0.197[CS_HANGUP]

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Originating to local extensions and mod_loopback
by Martin 24 Nov '08

24 Nov '08

Hi all, I've been looking for a way to originate a call out to a local extension without specifying the domain. The originate is performed by a separate program via event socket and hence has no access to the ${domain} variable. In previous releases I'm sure you were able to simply do: originate sofia/internal/1000. In more up-to-date releases, only originate user/1000@domain seems to do the trick. I thought I'd found the answer in mod_loopback, originating to loopback/1000 works but this causes more problems than it solves in that the jitter in the audio of the resultant call is unacceptable on most of the machines I've tested on. - Centos 5.2 on at least a P4 with > 1 gig of RAM. I'm testing primarily on svn trunk 10248 - I've tried building newer versions with no luck. If there's no reliable way of doing this, I'll find a way to synchronise the domain variable with my program via external means, thought I'd ask first in case I'm missing something simple. Cheers, Martin -- Martin Cole

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