March 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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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help
by Joshua Shaffner 21 Mar '11

21 Mar '11

On Mon, Mar 21, 2011 at 4:10 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. Mod_fifo outbound strategy enterprise (Marc de Corny) > 2. Does Freeswitch complies with SIP Connect technical > recommendation? (Yehavi Bourvine) > 3. unable to insert channel variable in new call leg w/in Lua > script (elijah) > 4. Re: New FreeSWITCH Module: mod_ladspa (Frank Carmickle) > 5. Delayed start of tone_detect (Stephen Wilde) > 6. Re: New FreeSWITCH Module: mod_ladspa (Michael Collins) > 7. Re: Understanding break="never" in condition-tag (Michael Collins) > > > ---------- Forwarded message ---------- > From: Marc de Corny <marcdecorny(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Date: Mon, 21 Mar 2011 17:59:16 +0000 > Subject: [Freeswitch-users] Mod_fifo outbound strategy enterprise > Hi all, > I am getting the hang of the fifo now and it is proving very stable which is what we all need. > Got one question however regarding mod_fifo and the outbound strategies. > From looking at the outbput of the fifo list commands I can see that that parameter is always set to "ringall". which is fine, but I can also see many stats on that number of calls and the last call that the members have taken. Is there also a way of sending the calls to the longest idle or something of that nature? Is that done by setting the outbound_strategy to "enterprise" or is there another way to achieve that. > Is there also a command to change that value without editing the XML as I am trying to make everything dynamic. > any input is very much appreciated. > thanks > Marc > > ---------- Forwarded message ---------- > From: Yehavi Bourvine <yehavi.bourvine(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Date: Mon, 21 Mar 2011 20:46:20 +0200 > Subject: [Freeswitch-users] Does Freeswitch complies with SIP Connect technical recommendation? > 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: > > ---------- Forwarded message ---------- > From: elijah <elijah(a)crankenstein.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Date: Mon, 21 Mar 2011 11:50:44 -0700 > Subject: [Freeswitch-users] unable to insert channel variable in new call leg w/in Lua script > 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? > > > > ---------- Forwarded message ---------- > From: Frank Carmickle <frank(a)carmickle.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Date: Mon, 21 Mar 2011 15:27:39 -0400 > Subject: Re: [Freeswitch-users] New FreeSWITCH Module: mod_ladspa > Hey Michael > This is cool. Has there been any thought to making freeswitch a jack client. That IMHO would be very very useful. That would allow sip, or any other protocol that fs supports, to be a way to get audio in to professional sound production software like ardour, audacity or ecasound. > Thanks > --FC > On Mar 18, 2011, at 6:56 PM, Michael Collins wrote: > > Hello all! > The FreeSWITCH developers have added a cool new module: mod_ladspa. Check out the story: > http://www.freeswitch.org/node/313 > Have fun! > -Michael > _______________________________________________ > 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: Stephen Wilde <wstephen80(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Date: Mon, 21 Mar 2011 20:33:28 +0100 > Subject: [Freeswitch-users] Delayed start of tone_detect > I start the tone_detect on answer using "execute_on_answer" as: > <action application="export" data="nolocal:execute_on_answer=tone_detect MYTONE 680 ro +5000 set MYTONE=1" /> > but I have the necessity to start the tone_detect for example 500ms after answer: there is a way to do that in dialplan? > > ---------- Forwarded message ---------- > From: Michael Collins <msc(a)freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Date: Mon, 21 Mar 2011 12:39:18 -0700 > Subject: Re: [Freeswitch-users] New FreeSWITCH Module: mod_ladspa > Not that I know of, but "patches welcome"!! :) > -MC > > On Mon, Mar 21, 2011 at 12:27 PM, Frank Carmickle <frank(a)carmickle.com> wrote: >> >> Hey Michael >> This is cool. Has there been any thought to making freeswitch a jack client. That IMHO would be very very useful. That would allow sip, or any other protocol that fs supports, to be a way to get audio in to professional sound production software like ardour, audacity or ecasound. >> Thanks >> --FC >> On Mar 18, 2011, at 6:56 PM, Michael Collins wrote: >> >> Hello all! >> The FreeSWITCH developers have added a cool new module: mod_ladspa. Check out the story: >> http://www.freeswitch.org/node/313 >> Have fun! >> -Michael >> _______________________________________________ >> 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: Michael Collins <msc(a)freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Date: Mon, 21 Mar 2011 13:10:26 -0700 > Subject: Re: [Freeswitch-users] Understanding break="never" in condition-tag > >> >> I can't/didn't contribute much to this discussion, but I do follow it. My idea and (little) testing experience supports the explanation of Dimitry. >> Either way, it 'proofs' there is confusion about the working of the 'break' in condition tags. >> If I can find the time, I will try to do some tests / build an example myself. Thank you both for the answers and insights so far. >> > > Okay, for those of you who have the bridge book, please go to page 167-168. Darren did a really good job of explaining this. For those of you who don't have the book, SHAME! :) > The break flag determines what happens on processing conditions within a single extension. Let's say that chan var ${freeswitch} has the value "rocks" and ${foo} has the value "bar". > <extension name="break_flag_is_not_confusing"> > <condition "${freeswitch}" expression="^lame$"> >  > </condition> >  > <condition "${foo}" expression="bar"> > ... > </condition> > </extension> > Look at the above snipped. The first condition evaluates to false, which means all processing for this particular extension stops. The parse "breaks" out of this extension and moves on to the next one. Now look at this extension with the the break-flag set to "never"... > <extension name="break_flag_is_not_confusing"> > <condition "${freeswitch}" expression="^lame$" break="never"> >  > </condition> >  > <condition "${foo}" expression="bar"> > ... > </condition> > </extension> > See the difference? With the break flag set to "never" then it does not matter whether the condition is evaluated as true or false - the extension's next condition gets evaluated. Why do you need the break="never"? Simple: when no "break" is specified there is an implied break="on-false" for the condition. In other words, by default, if one condition inside of an extension evaluates to false then all processing for that extension stops. This has the effect of allowing you to "stack" conditions to create a logical AND. > So, in summary, the break flag will control how the parser behaves when evaluating conditions. Normally when a single condition fails, the whole extension is skipped, however when you do break="never" on a condition then it does not matter if that particular condition is true or false - extension processing will continue. > Hope this helps. If it doesn't then read chapters 5 and 8 of the book. Chapter 5 (me) is a gentle intro into the dialplan. Chapter 8 (Darren) talks more extensively about the advanced concepts of dialplan processing. > -MC > > _______________________________________________ > 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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pass variable from dialplan to bridge
by dunchan 21 Mar '11

21 Mar '11

Hi! I have a simple question, how can i pass a varible from dialplan to the bridge? I want to fill the 'from-user' field depends on sip user settings. I've tryed the following in dialplan: ... some conditions ... <action application="set" data="from-user=${outbound_caller_id_number}"/> <action application="bridge" data="sofia/gateway/upc/$1"/> in a gateway xml i tryed: <gateway name="upc"> <param name="username" value="..."/> <param name="password" value="..."/> <param name="proxy" value="..."/> <param name="register" value="false"/> <param name="from-user" value="${from-user}"/> </gateway> Above config ha no effect :( any suggestions? thx, Viktor

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Recommendations for multi-channel provider of Singapore DID's
by Chris Fowler 21 Mar '11

21 Mar '11

Hi, We use InPhonex for International DID's. Unfortunately in Singapore they are unable to provide DID's with more than 2 Channels. Can anyone recommend a provider that can provide DID's with more than two channels in Singapore (we need 4 - 20 Channel DID's)? Thanks, Chris.

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