September 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Opus codec issue
by jonathan augenstine 26 Sep '14

26 Sep '14

I have been working with WebRTC and utilizing the opus codec. The sound quality on calls is great. I have been encountering some file issues however and want to get some input on how to best proceed. When the call is has both ends in Opus, or if Freeswitch is transcoding the call has no issues. The only issues I encounter are the following. 1. fs_encode fails to create a valid opus file When fs_encode executes, you do see mod_opus load successfully. However, the file that is created cannot be read by opus.org utilities, specifically opusinfo. The command line I use is: fs_encode -l mod_opus infile.wav outfile.opus 2. Streaming a file that is encoded in opus format fails If I encode a file using opus.org tools, opusenc, I have a valid opus file and I am able to open it with other audio applications. If I call into freeswitch and try to stream out that file, there is no error reported in the console logs, but the audio heard is garbled. Basically the mod_opus appears to work but reading, writing files appears to have an issue. My questions. 1. Is there anything I might have configured incorrectly? 2. What information can I grab to help troubleshoot this issue? Jonathan

4 10

FreeSWITCH Friday FreeForAll Reminder!
by Ken Rice 26 Sep '14

26 Sep '14

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888@conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon

1 0

is there any way to apply mod_shout "volume" setting on a per stream basis?
by Nathan Neulinger 26 Sep '14

26 Sep '14

i.e. in the param section of a local_stream? It doesn't look like anything in place that would allow this - is it something that might reasonably be possible to add, or would it be involved? -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

1 0

Re: [Freeswitch-users] Internal Sip Profile public or default context?
by Matthew Shealy 26 Sep '14

26 Sep '14

My Freeswitch is hosted on an external vps. Would I want to put the Internal profile behind the firewall and not allow connections to it. Or since I will be using Bria to make and receive my calls would I open it up and connect my registered users that way? - Matthew ---------- Forwarded message ---------- > From: Michael Collins <msc(a)freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Mon, 22 Sep 2014 14:03:10 -0700 > Subject: Re: [Freeswitch-users] Internal Sip Profile public or default > context? > This would be the default context to use for any unauthenticated calls. In > other words, that context parameter is telling the Internal SIP profile > which dialplan context to use when an unauth'd call comes in. Typically > this would be inbound SIP provider calls, such as calls in to a DID number. > You definitely want unauth'd calls to be handled by your > public/untrusted/paranoid dialplan context. To paraphrase Brian West: > "paranoid by default" is a good policy. If the user is authenticated then > the user_context value is used. > > -MC > > On Sat, Sep 20, 2014 at 6:21 PM, Matthew Shealy <shealyw2(a)gmail.com> > wrote: > >> Hi guys, >> >> In the documentation it says: >> >> "Authenticated users of this profile are normally set (via the variable >> user_context) to use the dialplan defined in the default context." >> >> >> https://confluence.freeswitch.org/display/FREESWITCH/Configuring+FreeSWITCH >> >> So why after a fresh compile is the $conf/sip_profiles/internal.xml >> Profile set to <param name="context" value="public"/> >> >> >> Thanks, >> Matthew >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >

1 0

(no subject)
by martin.hoole＠emailn.de 25 Sep '14

25 Sep '14

1 0

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

3 11

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

2 1

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.

4 4

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> >> >> >> >

1 0

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?

1 0