March 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Voice Mail - Core dump
by Volker Wend 07 Mar '12

07 Mar '12

Hi, I try to use the voice mail modul and get always a core dump (first usage). Any Ideas how to proceed ? Best regards, Volker Wend 2012-03-06 14:45:53.016696 [NOTICE] mod_dptools.c:1135 Channel [sofia/sipinterface_1/0173123456@172.25.156.1:5060] has been answered 2012-03-06 14:46:09.856705 [NOTICE] sofia.c:628 Hangup sofia/sipinterface_1/01736024861@172.25.156.1:5060 [CS_EXECUTE] [NORMAL_CLEARING] Segmentation fault (core dumped) /bin/cat: Schreibfehler: Datenübergabe unterbrochen (broken pipe) 2012-03-06 14:46:09.956696 [NOTICE] switch_core_session.c:1398 Session 9 (sofia/sipinterface_1/01736024861@172.25.156.1:5060) Ended 2012-03-06 14:46:09.956696 [NOTICE] switch_core_session.c:1400 Close Channel sofia/sipinterface_1/01736024861@172.25.156.1:5060 [CS_DESTROY] [cid:hospital.jpg]<http://www.immanuel.de/einrichtungen/hospital-dienstleistung-beratung-gmbh> Volker Wend Projektleiter Informationstechnologie, Medizintechnik und Prozessmanagement hospital Dienstleistung + Beratung GmbH Am kleinen Wannsee 5 . 14109 Berlin Tel. +49 (33638) 83261 . Fax +49 (30) 80505 137 Mobil +49 (173) 6024861 . v.wend(a)hospital-berlin.de<mailto:v.wend@hospital-berlin.de> hospital Dienstleistung + Beratung GmbH Am Kleinen Wannsee 5A . 14109 Berlin Amtsgericht Charlottenburg HRB 73271 Geschäftsführer: Udo Schmidt Steuernummer 27/604/01051 Eine Gesellschaft der Immanuel Diakonie www.immanuel.de DEM LEBEN ZULIEBE. Der Umwelt zuliebe: Müssen Sie diese Mail wirklich ausdrucken?

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web application to call to fs
by Wagner 06 Mar '12

06 Mar '12

hello , is there any Web application that i could use to let a user call to my ivr through my website? thanks

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chat API UTF-8
by John Rose 06 Mar '12

06 Mar '12

When using chat API for SIP MESSAGE and sending UTF-8 should the text be URL encoded? John

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Why doesn't continue_on_fail continue?
by Harry Coin 06 Mar '12

06 Mar '12

This is a 'why doesn't continue_on_fail' continue question. session:setVariable("continue_on_fail", "true"); session:setVariable("ignore_early_media", "true"); session:setVariable("hangup_after_bridge", "true"); # Try some local registered extensions session:execute("bridge", "[leg_timeout=28]loopback/202/default/XML"); # Ok, this works, we get here only when nobody picks up. # Use the dial plan leading to a bridge sofia/internal/xxx@voip2pstn1|sofia/internal/xxx@voip2pstn session:execute("bridge", "[leg_timeout=30][accountcode=4499090]sofia/internal/5551212@${domain}"); # If fails for any reason (all outgoing voip->pstn lines are busy or if voip2pstn device is dead/busy) continue here. # Problem is -- We never get here if the above fails. # The reason is 'NORMAL_TEMPORARY_FAILURE' but it just hangs up the calling extension. session:execute("bridge", "[leg_timeout=136]loopback/*99008"); #should leave voicemail, never gets called. What am I doing wrong? Harry

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Re: [Freeswitch-users] Choopy one-way noise (FreeTDM)
by Christian Benke 06 Mar '12

06 Mar '12

On 2 March 2012 14:13, Saugort Dario Garcia Tovar <dgarcia(a)anew.com.ve> wrote: > After change tdm board have you notice difference, the issue persist? No more audio-quality issues after replacing the A500 with a OpenVOX B100p. I forgot to mention we where able to reproduce this issue in our lab, after upgrading wanpipe the problem appeared to be gone - but we didn't test throughly and when we did the same at the clients machine, it didn't have the same effect. Will try to reproduce the problem in the lab again. Regards, Christian

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Calls drop after 1:48 (kinda)
by BloodyIron 06 Mar '12

06 Mar '12

Hi Folks, Okay so this one is a bit tricky to reproduce. One of our extensions will have their calls dropped preicsely after 1 minutes 48 seconds ( 1:48 ) of call time. Another extension on the same segment of the network does not do the same thing. This extension does this for every single call type, be it external or internal calls. Now you may think, well it may be a busted phone. We reset the phone to factory defaults and saw no improvement. Furthermore we are seeing it elsewhere in our freeswitch installation, as in other extensions on other sites are seeing the same issue. These extensions are behind a NAT, however the freeswitch server is publically facing (as in public IP), with a passive firewall between it and the world (no routing, no NAT, etc). Just to be clear, we are also using fusionpbx to control the installation, as typing everything into an xml is not very efficient (but we're not scared of working with xml files either). Right now, we're gonna try adding " " to our sofia.conf.xml file to address it, beyond this we are unsure what to do. Can anyone speak on this matter? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Calls-drop-after-1-48-kinda-t… Sent from the freeswitch-users mailing list archive at Nabble.com.

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MWI does not working anymore?
by Yehavi Bourvine 06 Mar '12

06 Mar '12

Hello, I've noticed a problem started in the last two weeks, but I don't see anyone else reporting it... MWI status is not updated anymore. It is only updated at the initial boot of the phone. Do others that use GIT from the last two weeks notice it? Thanks, __Yehavi:

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EC2 - One-way Voice / NAT Issue
by Sara Higfler 06 Mar '12

06 Mar '12

Hi, I'm sure everyone is tired of NAT related questions, but I am really struggling here :( I have freeswitch configured on an EC2 instance, with an elastic IP address. I've followed all of the NAT configuration instructions on the EC2 wiki (http://wiki.freeswitch.org/wiki/Amazon_ec2) and have researched widely on other possible solutions to no avail. Most scenarios work fine, but I'm having issue with one particular call path (see scenario 4 below). Firewall is correctly configured on the EC2 machine. Scenario 1: x-lite to x-lite - Both clients running behind firewall/NAT - Voice path in both directions Scenario 2: x-lite to PSTN - x-lite running behind firewall/NAT - PSTN is connected via SIP-trunk from Freeswitch to VoIPon ITSP - Voice path in both directions Scenario 3: PSTN to PSTN - Both endpoints connected via SIP-trunk (VoIPon ITSP) via Freeswitch - Voice path in both directions Scenario 4: PSTN to x-lite - Exact reverse of scenario 2 - Voice path one way (PSTN to x-lite) Is there something obvious that would prevent this specific scenario from working? I've dumped a SIP trace for the fourth scenario onto pastebin... http://pastebin.com/dg1XqxsJ I'd really appreciate some assistance and guidance! Kind regards, Sara

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switch_r_sdp or switch_l_sdp modify the media ip or port?
by hjz51 06 Mar '12

06 Mar '12

Hi , Who can tell me how to use switch_r_sdp or switch_l_sdp modify the media ip or port? Now I am want to redirect media to a transcoding card. But I don't know how to rewrite the rtp ip and port point to the transcoding card. Please can you give me the way to do it. Thanks very much!

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Flipping CID from ....
by Miha Zoubek 06 Mar '12

06 Mar '12

Hi, Flipping CID from "018108500" <018108500> to "Outbound Call" <38651357952>, FS does this after attended trafers happens. For b leg in my outbound call I can see there written Outbound Call not number. Is is possible to change this? I was looking at variable but did not noticed that caller-id is set to Outbound Call. Thanks! Miha

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