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Does Freeswitch complies with SIP Connect technical recommendation?
by Yehavi Bourvine 22 Mar '11

22 Mar '11

Hello, We published an RFP for our SIP connection to the PSTN. One of the suppliers asked whether Freeswitch complies with the SIP CONNECT recomendation. I have to answer them soon, and don't have enough time to read and evaluate the entire document at present. Does Freeswitch fullfills all the recomendations there? Thanks, __Yehavi:

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More then one number for registered user
by Boris Kovalenko 22 Mar '11

22 Mar '11

Hello! Is this possible to have one registered user and more then one associated number with it (without tricks with dialplan)? For example: <user id="XXX" number-alias="YYY,ZZZ....."> -- Regards, Boris

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I need to register all of my DIDs with a single SIP trunk..
by Randy Andrade 22 Mar '11

22 Mar '11

Greetings, I'm hoping someone can help me out here. I'm relatively new to FS, but not to general telephony and I have a small amount of SIP experience. My FS installation is, so far, a fairly straightforward PBX setup with a single SIP trunk with 21 DIDs to the ITSP that I work at. I have everything working as far as inbound and outbound calls, except that in order for me to use a DID as the outbound caller id, that DID needs to be explicitly registered with the ITSP, and not just the pilot number for the SIP trunk. When I try to create a second gateway *.xml file to register a second DID, I get an error "Registration Failed with status Operation has no matching challenge [904]" on the FS console. As I mentioned, I work at this ITSP and I program similar SIP trunks into Adtran TA9xx boxes for SIP over T1 deliveries. On the Adtrans, I'm able to define a single SIP trunk, and have multiple register statements within that trunk, one for each DID. I'm wondering if there's a way to do something similar in FS. I'm running a very recent (I update from the GIT tree about twice a week) version of FS on a Debian Linux OS, and, I have several local extensions working and able to call in and out; I just want to be able to outpulse various DIDs as my caller-id without getting calls rejected. Here's a copy of my gateway .xml file, with relevant information changed: <include> <gateway name="gatewayname">  <param name="username" value="1234567890"/>      <param name="from-domain" value="gatewayname"/>  <param name="password" value="XXXXXXXX"/>    <param name="proxy" value="proxyname"/>      <param name="register" value="true"/>      <param name="caller-id-in-from" value="true"/>    <param name="ping" value="30"/> <param name="extension-in-contact" value="true"/> </gateway> </include> Again, I don't know if I would do it here (I would assume so) but I'm wondering if there's some way (maybe multiple "extension" statements? I dunno..) to register multiple DIDs in the same gateway / SIP trunk. Thanks!

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Diversion Header
by Gustavo Espeche 22 Mar '11

22 Mar '11

Hi all we are having a problem with a customer that send the call with header Diversion, and our Freeswitch reject the call because don't understand the header. Some one know how can fix this problem? Best Regards. Gustavo Espeche www.easyipcall.com

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can't make mod_shout on latest git
by Neil Patel 22 Mar '11

22 Mar '11

>From latest git, I'm getting this error on make all: making all mod_shout Creating mod_shout.la... quiet_libtool: link: cannot find the library `/usr/lib/gcc/i486-linux-gnu/4.3.3/../../../../lib//libvorbis.la' or unhandled argument `/usr/lib/gcc/i486-linux-gnu/4.3.3/../../../../lib// libvorbis.la' make[5]: *** [mod_shout.la] Error 1 make[4]: *** [all] Error 1 make[3]: *** [mod_shout-all] Error 1 make[2]: *** [all-recursive] Error 1 make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 Looks like there is a path specification error/typo (I have libvorbis.la in /usr/lib/)? Not sure how to fix, but would be great if someone could. Best, Neil

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No ringback/audio suddenly
by Mark Campbell-Smith 22 Mar '11

22 Mar '11

Hi! I have an issue with no ringback or audio that has just started in the last few weeks. No changes in FS, which has been rock solid for a long time. I think my voip provider Pennytel has upgraded or changed their servers. Anyway, the difference in signalling is as follows: This does not work: INVITE sip:number@sip.pennytel.com SIP/2.0 SIP/2.0 100 Trying SIP/2.0 401 Unauthorized ACK sip:number@sip.pennytel.com SIP/2.0 INVITE sip:number@sip.pennytel.com SIP/2.0 SIP/2.0 100 Trying SIP/2.0 183 Session Progress SIP/2.0 180 Ringing SIP/2.0 200 OK ACK sip:202.85.243.105:5061 SIP/2.0 BYE sip:202.85.243.105:5061 SIP/2.0 SIP/2.0 200 OK This works INVITE sip:number@sip.pennytel.com SIP/2.0 SIP/2.0 100 Trying SIP/2.0 401 Unauthorized ACK sip:number@sip.pennytel.com SIP/2.0 INVITE sip:number@sip.pennytel.com SIP/2.0 SIP/2.0 100 Trying SIP/2.0 183 Session Progress SIP/2.0 200 OK ACK sip:202.85.243.105:5061 SIP/2.0 And I see after the Ringing signal in the case that does not work (there is no ringing signal in the case that does work): 2011-03-21 20:28:25.677621 [DEBUG] sofia.c:4641 Channel sofia/external/0393011079 skipping state [proceeding][180] received after a RINGING Can anyone point me to a reason for the difference in signalling and what I can do to get audio 100% of the time? THanks

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PA and ALSA
by Frank Rosengart 21 Mar '11

21 Mar '11

Hi, I' trying to use FS as a softphone, because it seems to be the only SIP UA, with up-to-date and full CELT support (thanks for that, great work!) But I'm really struggling with my audio config. I'm using Ubuntu 10.10, x86 with Gnome -> Pulseaudio. 1. mod_alsa simply segfaults immediately after receiving a call 'application="bridge" data="alsa"' or 'alsa call 123' from the console 2. I am running FS on two different computers. One, which is upgraded from earlier times, portaudio works ok (not perfect, but I think that are Alsa related issues). Another computer, with a fresh 10.10 install, mod_portaudio can not find any audio devices. mod_portaudio.so links libasound, so I assume that Alsa is included. aplay -L lists available devices: pulse Playback/recording through the PulseAudio sound server front:CARD=NVidia,DEV=0 HDA NVidia, ALC889A Analog Front speakers ... Any ideas what is missing here? 3. Has the portaudio lib, which included with FS, the PA 'dmix patch' applied? http://www.portaudio.com/trac/ticket/31 4. Has anyone started to build mod_pulseaudio ? I know, there are latency issues, but we are talking about VoIP with codec with frame sizes of 10ms and more.. Portaudio is really not the first choice for sound APIs on Linux anymore... 5. Same config, using CELT on Windows, I get noise glitches every second when calling an FS server, answering with the echo app. They are very regular, so it's maybe a sample rate issue somewhere in the path... Thanks for any hints. Frank

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Re: [Freeswitch-users] mod_fsv, record and playback options (William Kendi ...)
by Joshua Shaffner 21 Mar '11

21 Mar '11

William, I can not find jira ticket. Got a link? I would like to play with your modification. Anthm, any reason in particular why one should go with mod_mp4? Can it play back video to sip video caller? On Mon, Feb 21, 2011 at 5:21 PM, <freeswitch-users-request(a)lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: Using 16 KHz sounds (Michael Collins) > 2. Re: FS with Microsoft Forefront TMG (curriegrad2004) > 3. Re: FS with Microsoft Forefront TMG (Malay Thakershi) > 4. Re: mod_fsv, record and playback options (William Kendi ...) > > > ---------- Forwarded message ---------- > From: Michael Collins <msc(a)freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Date: Mon, 21 Feb 2011 12:35:54 -0800 > Subject: Re: [Freeswitch-users] Using 16 KHz sounds > It depends on why there is choppy audio. My guess is that going to 16k won't help. You should update to latest git and re-test, preferably on a system that is not in production. See if you can narrow down the conditions under which the audio is not good. Does it happen when the system is under load? Does it happen on every call, or only on certain calls? Things like that. > -MC > > On Sun, Feb 20, 2011 at 12:51 PM, Malay Thakershi <mthakershi(a)gmail.com> wrote: >> >> Hello, >> I use Cepstral in my mod_managed FS application. I mainly use Session.Speak and PlayAndGetDigits in my code to play WAV / audio text. >> When I started using FS and got a stable program running, I used Cepstral Allison 8 KHz voice. But frequently I get choppy type of sound. Earlier it was acceptable but now some callers seem to have difficulty understanding the call audio. >> Would it help if I get 16 KHz sounds / Cepstral license? What are changes I would need to make? >> Thank you for any help. >> Malay Thakershi >> _______________________________________________ >> 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 >> > > > > ---------- Forwarded message ---------- > From: curriegrad2004 <curriegrad2004(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Date: Mon, 21 Feb 2011 13:28:35 -0800 > Subject: Re: [Freeswitch-users] FS with Microsoft Forefront TMG > Publish ports 5060 and the range of rtp ports you specified in the > switch.conf.xml file. That's how I did it in ISA 2006 > > On Mon, Feb 21, 2011 at 12:22 PM, Malay Thakershi <mthakershi(a)gmail.com> wrote: >> Hello, >> Anyone has tried using FS successfully behind Microsoft Forefront Threat >> Management Gateway? >> Thank you for any direction. >> Malay >> _______________________________________________ >> 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 >> >> > > > > > ---------- Forwarded message ---------- > From: Malay Thakershi <mthakershi(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Date: Mon, 21 Feb 2011 16:03:15 -0600 > Subject: Re: [Freeswitch-users] FS with Microsoft Forefront TMG > Hello, >  >  >  > <param name="rtp-enable-zrtp" value="true"/> > I never had to change anything in this file. So I think my FS configuration working on default settings. > So therefore, you mean to say if I just follow normal procedure, it should work, right? > Malay > > > On Mon, Feb 21, 2011 at 3:28 PM, curriegrad2004 <curriegrad2004(a)gmail.com> wrote: >> >> Publish ports 5060 and the range of rtp ports you specified in the >> switch.conf.xml file. That's how I did it in ISA 2006 >> >> On Mon, Feb 21, 2011 at 12:22 PM, Malay Thakershi <mthakershi(a)gmail.com> wrote: >> > Hello, >> > Anyone has tried using FS successfully behind Microsoft Forefront Threat >> > Management Gateway? >> > Thank you for any direction. >> > Malay >> > _______________________________________________ >> > 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 > > > > ---------- Forwarded message ---------- > From: "William Kendi ..." <william.nishio(a)voicetechnology.com.br> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Date: Mon, 21 Feb 2011 19:21:26 -0300 > Subject: Re: [Freeswitch-users] mod_fsv, record and playback options > I did it! > > After some tears and sweats, I finally managed to create a working FSV demuxer module for the FFMPEG project! > > With this module, files in the FSV format now can be converted to any other format through the FFMPEG project! > > To install: > > 1). Put the "fsvdec.c" file in the "libavformat" directory. > 2). Insert the line "REGISTER_DEMUXER (FSV, fsv);" in the file "allformats.c" also in the "libavformat" directory. > 3). Insert the line "OBJS-$(CONFIG_FSV_DEMUXER) += fsvdec.o" in the file "Makefile" also in the "libavformat" directory. > 4). Build the FFMPEG project using "make install". > > Now the FSV format seems to be more usable and I am trying to figure how to use the FreeSWITCH JIRA. > > 2011/1/21 Anthony Minessale <anthony.minessale(a)gmail.com> >> >> Sure, send it to Jira and we'll get it in. >> Though, I'm surprised you would not want to use the mod_mp4 now that >> it exists =D the FSV was sort if a hack I made up on a whim. >> >> >> >> On Fri, Jan 21, 2011 at 5:17 PM, William Suffill >> <william.suffill(a)gmail.com> wrote: >> > Best to add the patches/details into Jira [http://jira.freeswitch.org] so it >> > can be tracked and reviewed for being added to the source tree. >> > >> > >> > _______________________________________________ >> > 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 >> > >> > >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale@hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888@conference.freeswitch.org >> googletalk:conf+888@conference.freeswitch.org >> pstn:+19193869900 >> >> _______________________________________________ >> 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 > >

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unable to insert channel variable in new call leg w/in Lua script
by elijah 21 Mar '11

21 Mar '11

In a Lua script I instantiate a new call leg and (attempt to) insert a new channel variable: local session = freeswitch.Session("user/1001(a)stuff.com") if session:hangupCause() == "SUCCESS" then session:execute("export", "my_variable=" .. myVariable) session:transfer("ext_in_mycontext", "XML", "mycontext") However, I am not able to retrieve my custom channel variable later. This call leg is established correctly but there is no point in this call leg's life where I can seem to find my variable. Is there another method by which I should try to insert a channel variable in a newly created call leg?

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audio dropouts with celt
by Frank Rosengart 21 Mar '11

21 Mar '11

Hi, I'm connecting two FS'es with local audio (PA) each, via SIP using the CELT codec. I experience occasional audio dropouts, maybe one in five seconds. But not regulary. Sometimes it's fine for a minute. The recorded waveform shows there is just silence for some milliseconds. Sometimes the audio is garbled for a portion of a second. This happens even when the computers are on the same LAN, but it's getting worse when using public internet. All systems are running FS from GIT, with CELT 0.10.0. I have done the following debugging: - recording the session: recorded on both sides, local audio is ok, remote side has dropouts. pa loopback has no issues -> no soundcard/driver issue - running wireshark, using the RTP analyzer: on PCMA,G722 there is no packet loss, no notable jitter (<20ms). The RTP analyzer does not show any jitter info on CELT payload packets. I'm not sure if it can see lost packets. -> I'm quite sure there is no packet loss. How timing critical is CELT? Any ideas how I can go on with debugging? Or is this better posted to the CELT developers? Is anyone running FS+CELT with dropout-free audio? Thanks! Frank

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