- FreeSWITCH-users - FreeSWITCH Mailing Lists

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by martin.hoole＠emailn.de 25 Sep '14

25 Sep '14

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Registration Failed with status DNS Error [503[
by Steven Schoch 25 Sep '14

25 Sep '14

This is probably a newbie problem, but I've been knocking my head over it for a few hours this morning, so I thought I'd ask for some help. I was running FreeSWITCH 1.2.7 on CentOS 5.8 (as a VM), and it was working fine. I upgraded to the latest FreeSWITCH 1.5.14b on CentOS 6.5. I still need to configure SELinux, so I turned that off for now. I also was suspecting the firewall, so I turned that off as well. I copied the configuration over, and the voicemail recordings, etc. Now we can receive calls, but we can't place external calls. We're using FlowRoute, and I'm getting this: 2014-09-24 13:37:42.695288 [NOTICE] sofia_reg.c:427 Registering flowroute 2014-09-24 13:37:42.695288 [ERR] sofia_reg.c:2292 flowroute Registration Failed with status DNS Error [503]. failure #2 2014-09-24 13:37:43.715285 [WARNING] sofia_reg.c:482 flowroute Failed Registration [503], setting retry to 30 seconds. The DNS seems to resolve sip.flowroute.com just fine, when using ping. And incoming calls are working! Where should I look next? By the way, I can call the FreeSWITCH conference line, so it can DNS lookup conference.freeswitch.org just fine. -- Steve

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connecting freeswitch to service provider
by akhil garg 25 Sep '14

25 Sep '14

Hi, --------------------- -------------------- | Freeswitch 1 | |Freeswitch 2 | ------------------------------------> | |----------------------------------------------------------->| |<------------------------------------------ (user 100 registered to FS1) | | | | (user 200 registered to FS2) -------------------- -------------------- Freeswitch 1: ------------------ Created a gateway ( with User 2222) in external sip_profile which registers to Freeswitch2 User 100 registers to Freeswitch1 in external sip_profile. Public dialplan is as below <extension name=gw_To_fs2"> <condition field="destination_number" expression="^200$"> <action application="set" data="dialed_extension=$1"/> <action application="set" data="dialed_domain=IP of FS2"/> <action application="bridge" data="sofia/gateway/sip_provider_1/${dialed_extension}"/> </condition> </extension> Freeswitch 2: ------------------ User 200 registers to Freeswitch2 in external sip_profile. Public dialplan is as below <extension name="client_1"> <condition field="destination_number" expression="^200$"> <action application="set" data="dialed_domain=IP of FS2"/> <action application="set" data="dialed_extension=$1"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="user/${dialed_extension}@ ${dialed_domain}"/> </condition> </extension> Query: --------- I am able to call 200 from user 100 using gateway (configured on Freeswitch1 and registered on Freeswitch2) How can I call User 100 from from user 200 without confuring a gateway on Freeswitch2? -- regards, akhil

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slow?
by Stanislav Sinyagin 25 Sep '14

25 Sep '14

all FreeSWITCH file and Git repositories are extremely slow to download from. Is it some temporary network problem? git clone from stash took about an hour, with 25kB/sec.

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FS-6849: mod_python error everytime session is invoked for the first time right after a freeswitch start
by Nuno Reis 25 Sep '14

25 Sep '14

Using gdb I've figured out that in mod_python.c:281: ... result = PyEval_CallObjectWithKeywords(function, arg, (PyObject *) NULL); Py_DECREF(function); ... variable result is always null when a python script trying to use the session run the first time right after freeswitch startup. That will make it enter the else logic in line 291 in the same file ending with a crash. I'll still have to drill more to find out why this is happening now. Anyone has a clue on what's going on? BR, -- *Nuno Miguel Reis* | *Unified Communication** Systems* M. +351 913907481 | nreis(a)wavecom.pt WAVECOM-Soluções Rádio, S.A. Cacia Park | Rua do Progresso, Lote 15 3800-639 AVEIRO | Portugal T. +351 309 700 225 | F. +351 234 919 191 *GPS <http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a…> | www.wavecom.pt <http://www.wavecom.pt/>** <http://www.wavecom.pt/>* [image: Description: Description: WavecomSignature] <http://www.wavecom.pt/pt/wavecom/premios.php> [image: Publicity] <http://www.wavecom.pt/pt/mail_eventos.php> On Tue, Sep 23, 2014 at 3:53 AM, Nuno Reis <nreis(a)wavecom.pt> wrote: > This is the output after calling the script right after start: > > 2014-09-23 03:49:45.801129 [NOTICE] mod_python.c:212 Invoking py module: > teste > 2014-09-23 03:49:45.801129 [NOTICE] switch_cpp.cpp:1328 Testing123... > 2014-09-23 03:49:45.801129 [NOTICE] switch_cpp.cpp:1328 Dumping current > CDR INFO > 2014-09-23 03:49:45.801129 [ERR] mod_python.c:293 Error calling python > script > 2014-09-23 03:49:45.801129 [ERR] mod_python.c:164 Python Error by calling > script "teste": <type 'exceptions.AttributeError'> > Message: 'PySwigObject' object has no attribute 'getXMLCDR' > Exception: None > > Traceback (most recent call last) > File: "/opt/freeswitch/scripts/teste.py", line 12, in handler > > All the following script calls work just fine. > > *Note:* If you want to test the code snippet i've added before don't > forget to add: > > import freeswitch > > Looking forward to hear from you. > > Best Regards, > > -- > > *Nuno Miguel Reis* | *Unified Communication** Systems* > M. +351 913907481 | nreis(a)wavecom.pt > WAVECOM-Soluções Rádio, S.A. > Cacia Park | Rua do Progresso, Lote 15 > 3800-639 AVEIRO | Portugal > T. +351 309 700 225 | F. +351 234 919 191 > *GPS > <http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a…> > | www.wavecom.pt <http://www.wavecom.pt/>** <http://www.wavecom.pt/>* > > [image: Description: Description: WavecomSignature] > <http://www.wavecom.pt/pt/wavecom/premios.php> > > [image: Publicity] <http://www.wavecom.pt/pt/mail_eventos.php> > > > > On Tue, Sep 23, 2014 at 3:44 AM, Nuno Reis <nreis(a)wavecom.pt> wrote: > >> Hi guys. >> >> I've reported this before and i can tell that it is still happening in >> the latest v1.4. >> >> How reproduce: >> >> >> 1. just create a dialplan that calls a python script >> 2. dial that dialplan entry >> >> >> Example code (works with anything that invokes *session*): >> >> #!/usr/bin/env python >> # -*- coding: utf-8 -*- >> >> from __future__ import absolute_import >> # http://docs.python.org/whatsnew/pep-328.html >> >> def handler(session, args): >> freeswitch.consoleLog("notice", "Testing123...") >> freeswitch.consoleLog("notice", "Dumping current CDR INFO") >> freeswitch.consoleLog("info", str(session.getXMLCDR())) >> freeswitch.consoleLog("notice", "The END!") >> >> >> mod_pyhton crashes everytime the script is called right after a >> freeswitch start (the very first time session is invoked inside a >> mod_python python script). >> After the first time (error), everything starts to work just fine. >> This doesn't happen in e6b8b3e4f2deac73547bc9b9a2cb3e1d3147d101 from >> v1.2.stable >> >> Should i open a jira on this? >> >> -- >> >> *Nuno Miguel Reis* | *Unified Communication** Systems* >> M. +351 913907481 | nreis(a)wavecom.pt >> WAVECOM-Soluções Rádio, S.A. >> Cacia Park | Rua do Progresso, Lote 15 >> 3800-639 AVEIRO | Portugal >> T. +351 309 700 225 | F. +351 234 919 191 >> *GPS >> <http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a…> >> | www.wavecom.pt <http://www.wavecom.pt/>** <http://www.wavecom.pt/>* >> >> [image: Description: Description: WavecomSignature] >> <http://www.wavecom.pt/pt/wavecom/premios.php> >> >> [image: Publicity] <http://www.wavecom.pt/pt/mail_eventos.php> >> >> >> >

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mod_dingaling + GoogleHangout + TLS not working
by Luis Daniel Lucio Quiroz 25 Sep '14

25 Sep '14

Hello In my kopete, i can only log into google hangout if i use these parameters: server: talk.google.com port: 5223 (ssl) sasl: plain other change wont work. It seems that mod_dingaling only supports TLS and not SSL. I have tryed server: talk.google.com port: 5222 (default one) tls: on sasl: plain and tls: off sasl: md5 and non of them works. Do you have any advice?

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Leg stuck in a conference
by Oleg Stolyar 24 Sep '14

24 Sep '14

Guys, I wonder if anyone else ran into this situation: Very rarely I have a conference participant stuck in a conference. When I try to run kick command on that conference, it returns "OK kicked <member ID>" but nothing happens. In the logs there is only this: db1790fe-fa75-42a6-9054-75c600b9539c 2014-09-23 15:43:07.066947 [DEBUG] mod_conference.c:5619 Send signal sofia/<profile>/anonymous(a)anonymous.invalid [BREAK] As an additional piece of information if I use uuid_kill on the leg, it does disconnect the leg and ends the conference. Any suggestions?

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is there a group_confirm_timeout setting?
by Phillip Jones 24 Sep '14

24 Sep '14

Hi there, I have question about group_confirm and I am hoping someone can help: I am bridging a call and prompting the called party to press 1 to answer. Works great: <action application="answer"/> <action application="set" data="group_confirm_file=phrase:ppn_to_accept_call_from"/> <action application="set" data="group_confirm_key=1"/> <action application="bridge" data="{leg_timeout=10}user2/1001 <action application="say" data="en number pronounced 54321"/> You will notice that leg_timeout is 10 (set low to test). So that gives the caller just a couple of seconds to answer. So I add: * <action application="set" data="group_confirm_cancel_timeout=true"/>* This cancels that leg_timeout - the call now will stay up until the called party presses 1 or either party hangs up. If a call goes to a cell phone voicemail - this can result in a long message of repeated "press 1 to accept call". And I cannot move the calling party on to the next dialplan action. I tried originate_timeout and call_timeout and all seemed to be cancelled by group_confirm_cancel_timeout, as you might expect. Is there a way to set a timeout in that group_confirm stage? For example a *group_confirm_timeout *setting. That would set the seconds the called party has to confirm? Thanks in anticipation. Phil

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Re: [Freeswitch-users] session:playAndGetDigits() play two files When digits are fetched
by Snabel Kabiya 24 Sep '14

24 Sep '14

Hi, I'm calling the method from lua script local digits = session:playAndGetDigits(minLength, maxLength, maxAttempts, 5000, termChar, menuPrompt, noMatchPrompt, grammarRegex, "digits", 1500); Now when i tried using file_string: menuPrompt_part1=/usr/audio/phrase1.wav menuPrompt_part2=/usr/audio/phrase2.wav local digits = session:playAndGetDigits(minLength, maxLength, maxAttempts, 5000, termChar, file_string://menuPrompt_part1!menuPrompt_part2, noMatchPrompt, grammarRegex, "digits", 1500); i got the following error: *2014-09-24 07:58:25.677222 [ERR] mod_lua.cpp:203 /usr/local/freeswitch/scripts/....: '<name>' expected near '/'* what did i miss? Thanks, Snabel ---------- Forwarded message ---------- > From: Michael Collins <msc(a)freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Thu, 18 Sep 2014 10:08:38 -0700 > Subject: Re: [Freeswitch-users] session:playAndGetDigits() play two files > When digits are fetched > I think this might be what you're looking for: > https://wiki.freeswitch.org/wiki/Mod_file_string > > -MC > > On Thu, Sep 18, 2014 at 5:56 AM, Snabel Kabiya <snabel(a)lexifone.com> > wrote: > >> Hi, >> >> I'm using this method >> https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits >> I've this parameter >> >> - file = Sound file to play while digits are fetched >> >> i wanted to ask if i can play two files? if yes what's the syntax? >> >> Thanks, >> Snabel >> >>

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Dialplan Tools pickup - Bug/Memory Leak
by Lloyd Aloysius 24 Sep '14

24 Sep '14

Hello Switch crashing all the time when using the Pickup not immediately, but after couple times. Back trace show the pickup but do know where this is happening. https://jira.freeswitch.org/browse/FS-6851 Any help is appreciated. Thank you LLoyd

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