- FreeSWITCH-users - FreeSWITCH Mailing Lists

unable to bridge sofia call to jabbin
by Pekka K. Kurki 29 Dec '06

29 Dec '06

trzing to bridge to jabbin brings me just the cmpp message 405 not-allowed... why? 2006-12-29 16:06:23 [NOTICE] switch_core.c:2683 switch_core_standard_on_execute() Execute bridge(dingaling/jingle/pekka(a)salo.intellectics.com/Psi) SEND[<presence type='subscribe' to='pekka(a)salo.intellectics.com/Psi' from='pekka-free(a)salo.intellectics.com/talk'><status>Call Me!</status></presence>] SEND[<presence type='subscribed' to='pekka(a)salo.intellectics.com/Psi' from='pekka-free(a)salo.intellectics.com/talk'><status>Call Me!</status></presence>] SEND[<presence type='probe' from='pekka-free(a)salo.intellectics.com/talk' to='pekka(a)salo.intellectics.com/Psi'/>] RECV[<presence to="pekka-free(a)salo.intellectics.com/talk" from="pekka(a)salo.intellectics.com/Psi" type="error"><error code="405" type="cancel"><not-allowed xmlns="urn:ietf:params:xml:ns:xmpp-stanzas"/></error></presence>] RECV[<presence from="pekka(a)salo.intellectics.com/Psi" to="pekka-free(a)salo.intellectics.com/talk"> <priority>5</priority> <c xmlns="http://jabber.org/protocol/caps" node="http://www.google.com/xmpp/client/caps" ver="0.10" ext="voice-v1"/> </presence>] SEND[<presence type='subscribe' to='pekka(a)salo.intellectics.com/Psi' from='pekka-free(a)salo.intellectics.com/talk'><status>Call Me!</status></presence>] RECV[<presence to="pekka-free(a)salo.intellectics.com/talk" from="pekka(a)salo.intellectics.com/Psi" type="error"><error code="405" type="cancel"><not-allowed xmlns="urn:ietf:params:xml:ns:xmpp-stanzas"/></error></presence>] 2006-12-29 16:00:35 [DEBUG] mod_dingaling.c:1568 channel_outgoing_channel() Unknown Recipient! 2006-12-29 16:00:35 [DEBUG] mod_dingaling.c:590 terminate_session() Terminate called from line 1569 state=CS_NEW 2006-12-29 16:00:35 [NOTICE] switch_core.c:3232 switch_core_session_destroy() Close Channel N/A 2006-12-29 16:00:35 [ERR] switch_ivr.c:2491 switch_ivr_originate() Cannot Create Outgoing Channel! 2006-12-29 16:00:35 [DEBUG] switch_ivr.c:2832 switch_ivr_originate() Originate Resulted in Error Cause: 1 [UNALLOCATED] 2006-12-29 16:00:35 [INFO] mod_bridgecall.c:69 audio_bridge_function() Originate Failed. Cause: UNALLOCATED 2006-12-29 16:00:35 [NOTICE] mod_bridgecall.c:72 audio_bridge_function() Hangup sofia/sip/1001(a)85.126.75.149 [CS_EXECUTE] [UNALLOCATED] 2006-12-29 16:00:35 [INFO] switch_channel.c:969 switch_channel_perform_hangup() Kill sofia/sip/1001(a)85.126.75.149 [1] 2006-12-29 16:00:35 [DEBUG] switch_core.c:2908 switch_core_session_run() (sofia/sip/1001(a)85.126.75.149) State HANGUP 2006-12-29 16:00:35 [DEBUG] mod_sofia.c:1244 sofia_on_hangup() Channel sofia/sip/1001(a)85.126.75.149 hanging up, cause: UNALLOCATED 2006-12-29 16:00:35 [DEBUG] mod_sofia.c:1264 sofia_on_hangup() Responding to INVITE with: 404 2006-12-29 16:00:35 [DEBUG] switch_core.c:2612 switch_core_standard_on_hangup() Standard HANGUP sofia/sip/1001(a)85.126.75.149, cause: UNALLOCATED 2006-12-29 16:00:35 [DEBUG] switch_core.c:3352 switch_core_session_thread() Session 3 (sofia/sip/1001(a)85.126.75.149) Locked, Waiting on external entities 2006-12-29 16:00:35 [INFO] switch_core.c:3359 switch_core_session_thread() Session 3 (sofia/sip/1001(a)85.126.75.149) Ended 2006-12-29 16:00:35 [NOTICE] switch_core.c:3232 switch_core_session_destroy() Close Channel sofia/sip/1001(a)85.126.75.149 send 530 bytes to udp/[85.126.75.150]:5060 at 15:00:35.476725: -- (Mr.) Pekka K. Kurki Intellectics GesmbH http://www.intellectics.com Helenenstrasse 44, A-2500 Baden, Austria Tel. +43 2256 815811 Mobile +43 699 12532539 Fax +43 2256 815819 (Tel. US: 360-2267729 UK: 0870 4787064) sip:pekka.kurki@sip.intellectics.com h323:432256815811@sip.intellectics.com email: Pekka.Kurki(a)intellectics.com www: http://pekka.kurki.at

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Freeswitch Dialplan question (bridging two outgoing calls)
by Saurabh Aggarwal 29 Dec '06

29 Dec '06

How do you configure an extension in the dialplan such that when you dial it, it makes two outgoing calls, and bridges them together (not really a conference, since I don't want to be in the call). Is it possible? If not, how would i create such a conference? I saw the example (in the sample config) to create a conference and dial out to 1 user, but how do I dial out 2 users? Thanks. _________________________________________________________________ Fixing up the home? Live Search can help. http://imagine-windowslive.com/search/kits/default.aspx?kit=improve&locale=…

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freeswitch: src/switch_core.c:1360: switch_core_session_strdup: Assertion `session != ((void *)0)' failed.
by Pekka K. Kurki 25 Dec '06

25 Dec '06

any time i try to bridge sofia channels i get: freeswitch: src/switch_core.c:1360: switch_core_session_strdup: Assertion `session != ((void *)0)' failed. Aborted (FC5, svn version of last week) br pekka -- (Mr.) Pekka K. Kurki Intellectics GesmbH http://www.intellectics.com Helenenstrasse 44, A-2500 Baden, Austria Tel. +43 2256 815811 Mobile +43 699 12532539 Fax +43 2256 815819 (Tel. US: 360-2267729 UK: 0870 4787064) sip:pekka.kurki@sip.intellectics.com h323:432256815811@sip.intellectics.com email: Pekka.Kurki(a)intellectics.com www: http://pekka.kurki.at

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Happy Holidays.....
by Brian West 25 Dec '06

25 Dec '06

Dear Fellow FreeSWICHers, That time of year is approaching where I am trying to get things lined up for Cluecon '07. I'm doing a call for speakers for the '07 season. We are shooting for the last week in June and working hard to finalize the location (which is in Chicago as it is every year). Now if you wish to speak at Cluecon let me know... if you have an idea for a topic you would like to see or someone you think should be at Cluecon then please send me a followup email and we can discuss possible topics. Cluecon is open to all open source telephony and its a very developer centered conference. If you need me to sell the idea to your boss please let me know I will make the call. ;) Thanks, Brian West

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Re: [Freeswitch-users] freeswitch windows popup error msg while using sip dialing 888
by michael mendel 24 Dec '06

24 Dec '06

Hello > > I get this error message while using XLITE dialing 888 using FS compiled > for windows. > > *Debug Assertion Failed* > *Program:c:/Freeswitch* > *File:dbgheap.c* > *Line:692* > *Experssion:_CrtIsValidHeapPointer(PUserData)* > > Regards, > Michael Mendel > > >

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Freeswitch registration voipdiscount/sipdiscount
by Pekka K. Kurki 21 Dec '06

21 Dec '06

has anybody set up registration to voipdiscount/sipdiscount? I had no luck so far... br, pekka -- (Mr.) Pekka K. Kurki Intellectics GesmbH http://www.intellectics.com Helenenstrasse 44, A-2500 Baden, Austria Tel. +43 2256 815811 Mobile +43 699 12532539 Fax +43 2256 815819 (Tel. US: 360-2267729 UK: 0870 4787064) sip:pekka.kurki@sip.intellectics.com h323:432256815811@sip.intellectics.com email: Pekka.Kurki(a)intellectics.com www: http://pekka.kurki.at

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(no subject)
by michael mendel 20 Dec '06

20 Dec '06

Hello I have an FS running on XP I have two friends that are registered to my FS. they use X-LITE on XP also they have successed to dial to 888 and also to test 1234, but when they try to dial one friend to the other and vice versa it's not ringing . I have put my freeswitch.xml at http://pastebin.freeswitch.org/187, and my friends use the dial Number(it is also there user name): 110 and 114. Regards, Michael Mendel

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Re: [Freeswitch-users] (no subject)
by Anthony Minessale 20 Dec '06

20 Dec '06

The syntax for calling a registered sip user is sofia/<profile_name>/<user_id>%<domain_name> you have: sofia/test/110%84.108.22.188:5060 This is fist off invalid because :5060 is not valid here. secondly, your output shows your phone using this register line: REGISTER sip:mikemendel.no-ip.org SIP/2.0 this means the domain you are using is "no-ip.org" not "84.108.22.188" if you actually want it to work you need to call sofia/test/110%no-ip.org because the domains have to match. you also could rename your sofia profile from test to no-ip.org Then you could simply call /sofia/no-ip.org/110 Whatever you put as the domain in your sip phone must be addressed in the config. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com JABBER:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: michael mendel <mikemendel(a)gmail.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Wednesday, December 20, 2006 4:10:32 PM Subject: Re: [Freeswitch-users] (no subject) On 12/20/06, Michael Jerris <mike(a)jerris.com> wrote: Can you please post the entire output of the failed call, everything from when the call comes in, through the error message you posted below. Hello Michael The Error apears at line 943 of http://pastebin.freeswitch.org/191 Regards, Michael Mendel From: michael mendel [mailto: mikemendel(a)gmail.com] Sent: Wednesday, December 20, 2006 2:47 PM To: freeswitch-users(a)lists.freeswitch.org Subject: Re: [Freeswitch-users] (no subject) Hello Mike Thank you for your advice but I have already tried that issue , I still don't get any Ringing, Sorry. http://pastebin.freeswitch.org/190 than I get after putting % instead of @ 2006-12-20 21:30:28 [ERR] switch_ivr.c:2430 switch_ivr_originate() Cannot Create Outgoing Channel! Regards, Michael Mendel _______________________________________________ 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 __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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Re: [Freeswitch-users] OpenVXML
by Anthony Minessale 20 Dec '06

20 Dec '06

That's good to know cos when I first encuntered vxml and got to the point where I found out I had to buy all the editors I said "oh well I guess I'll have to wait for someone to make a free editor before I try this" But if there is a freely available development tool for it, it makes the idea of making a vxml parser more interesting. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com JABBER:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Tyler <tyler(a)openmethods.com> To: freeswitch-users(a)lists.freeswitch.org. Cc: Tim Barnes <tim(a)openmethods.com> Sent: Wednesday, December 20, 2006 1:25:09 PM Subject: [Freeswitch-users] OpenVXML Hello Everyone.... If you are building VoiceXML applications, you should check out openVoiceXML, the only open source visual development tool for VoiceXM OpenVXML is an open source GUI-Based IVR application development tool. The tool consists of an application designer (Visio-like) and a runtime that dynamically generates VoiceXML. OpenVXML is currently licensed under the Eclipse Public License and can be found under the VTP project on Eclipse.org.I www.eclipse.org/vtp There is a very helpful mailing list, documentation, and flash tutorials to assist you in getting started with the tool (links below). (Mailing List) http://dev.eclipse.org/newslists/news.eclipse.technology.voicetools/maillis… (Documentation) http://www.eclipse.org/vtp/openvxml/Getting_Started.php (Flash Tutorials) http://www.eclipse.org/vtp/openvxml-announce/content/html/openvxml_demo.htm Listed below are 5 good reasons to check out OpenVXML 1) The openVXML media system allows developers to build branded media libraries with minimal effort that are simple to maintain. The development tool also supports inheritance between brands, which allows developers to do very little work when building multiple applications that are functionally similar, but require different caller experiences, such as when a retail application needs needs to support multiple lines of business (business and cosumer). 2) The framework provides built-in modules that support database reads and web service invocations without requiring custom coding. 3) Developers can extend the openVXML framework with custom modules to perform specific back-end integration methods like message bus, complex VUI tasks, or anything else possible in the Java language. 4)The framework contains an abstraction layer that allows it to be customized for any possible voice platform; it even has the ability to run on multiple VXML platforms concurrently, running against the same application. The runtime server and designer have been used with the Genesys Voice Portal (GVP, Voxeo's Prophecy, VoxPilot, BeVocal and OpenVXi. 5) The openVXML Studio exports Java web applications that run on any Java Servlets 2.3 compatible server, allowing users to deploy on anything from single Tomcat instances to massive clusters of WebLogic servers. Because the designer is Eclipse based, you can run it on Windows, MacOS, and most flavors of Linux. Feedback is greatly appreciated. Please let m know what you think of the tool. Thanx. Tyler VanWinkle _______________________________________________ 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 __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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Re: [Freeswitch-users] C SAY API
by Anthony Minessale 20 Dec '06

20 Dec '06

We don't have something like that at the moment but it would not be very difficult to create. We do have a mod_perl which lets you do native api calls with perl but that is not complete yet. Your best bet for ivr right now is to look at mod_spidermonkey which is the javascript IVR enginge that has a lot of control over the call and is fairly easy to pick up. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com JABBER:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: David Villasmil Govea <david.villasmil(a)gmail.com> To: freeswitch-users(a)lists.freeswitch.org Cc: freeswitch-dev(a)lists.freeswitch.org Sent: Wednesday, December 20, 2006 4:33:23 AM Subject: Re: [Freeswitch-users] C SAY API I'm not asking about asterisk, you got me wrong. What I am asking is wether there is a module for interacting with freeswitch in perl. Just LIKE asterisk.. nmot asking about asterisk at all... not using it anymore. David On 12/20/06, Kannaiyan Natesan <gbpnkans(a)gmail.com> wrote: This is the first time replying to you. This is not asterisk mailing list. I think we need moderation for this list to avoid unwanted emails. Check it out in the asterisk mailing list. Regards, Kannaiyan On 12/20/06, David Villasmil Govea <david.villasmil(a)gmail.com> wrote: > Hello All, > > this is my third time asking the same question: Is there anything > analog to asterisk AGI, with perl? > > Thank you > > David > > > On 12/19/06, Nikolay Kolev <me(a)nikolay.com> wrote: > > This is all good, but I'm afraid that similarly to your Asterisk product, > > it's probably not going to be open source and free (as in "free beer"). I > > will be more than happy if I assumed wrongly. > > > > Nikolay Kolev > > -----Original Message----- > > From: tech(a)i6net.com [mailto:tech@i6net.com] > > Sent: Tuesday, December 19, 2006 2:10 AM > > To: freeswitch-users(a)lists.freeswitch.org ; Brian West > > Cc: freeswitch-users(a)lists.freeswitch.org > > Subject: Re: [Freeswitch-users] C SAY API > > > > Hello, > > > > i6net is trying to port the VXIasterisk to FreeSwitch (project named > > VXIfreeswitch). > > The module is called mod_vxml (Brian confirm me the availability for this > > namming module). > > We have made the first connection between our VoiceXML browser and the > > module. > > We need help to : > > - have a configuration file (XML). > > - have a CLI admin commands to get some execution stats. > > > > > > Tech > > > > Selon Brian West <brian.west(a)mac.com>: > > > > > 1. Get OpenVXI to compile .... > > > 2. Actually get it to work. > > > 3. Must lean towards Open Source. > > > > > > I have yet to get it working on even linux following the actual build > > > instructions. If anyone wants to do the research to get this to > > > compile on linux then we can work on Mac and Windows. > > > > > > > > > /b > > > > > > On Dec 18, 2006, at 10:40 PM, Kannaiyan Natesan wrote: > > > > > > > Voicexml will be the best bet. > > > > > > > > Any cons towards using this? > > > > > > > > Kannaiyan > > > > > > > > On 12/18/06, Victor Gamov <vitspec(a)gmail.com> wrote: > > > >> > > > >> On 12/16/06, Anthony Minessale < anthmct(a)yahoo.com > wrote: > > > >>> > > > >>> > > > >>> > > > >>> > > > >>> Hi, > > > >>> > > > >>> I'm working on a muiti-lingual say api for ivr programmers to be > > > >>> able > > > >>> to generate audio for saying numbers, counting items or > > > >>> expressing amounts > > > >> of things such as currency. > > > >> > > > >> [skip] > > > >> > > > >> Hi Anthony! > > > >> > > > >> I think that it's not bad idea to have VoiceXML module in > > > >> FreeSWITCH. Using > > > >> this module anyone can easely write any IVR application (voicemail > > > >> and > > > >> other). If so, logic for multi-lingual api and other complex > > > >> things may be > > > >> carry out from FreeSWITCH. > > > >> > > > >> Some words about multi-lingual support. > > > >> Some times ago I write VoiceXML applications speaked in English, > > > >> Russian and > > > >> some other langs. The basic idea was liked to your say() function and > > > >> Kannaiyan's sequencer: we use module which parses input digits and > > > >> interpret > > > >> it as number, money, date, etc and predefined VoiceXML files for each > > > >> language to properly construct phrases like "You have 100 > > > >> messages" (EN) or > > > >> "In your voicebox 100 messages" (RU). It works fine for all langs. > > > >> > > > >> -- > > > >> CU, > > > >> Victor Gamov > > > >> _______________________________________________ > > > >> 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 > > > > > > > > > _______________________________________________ > > > 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 > > > > > > _______________________________________________ > > 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 > > > > > > -- > DVG > > -- > Imagination is more important than knowledge > Albert Einstein > _______________________________________________ > 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 -- DVG -- Imagination is more important than knowledge Albert Einstein _______________________________________________ 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 __________________________________________________ Do You Yahoo!? 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