March 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Overzealous 'nixevent'?
by freeswitch＠dalethatcher.com 14 Mar '08

14 Mar '08

I've been looking at improving the performance of an event socket app and have started trying to use 'nixevent' to restrict the traffic. However I've run into an issue where CUSTOM events go missing if I nix events. Environment: Freeswitch rev: 7886 Platform: Linux 2.6.22-14-generic #1 SMP (ubuntu) To reproduce I have this in my public.xml dialplan: <extension name="conference"> <condition field="destination_number" expression="^conference$"> <action application="answer"/> <action application="conference" data="conf"/> </condition> </extension> I set up socket A with the commands: auth ClueCon event plain all Socket B is set up with: auth ClueCon event plain all nixevent HEARTBEAT I then use a softphone to dial the 'conference' extension in the dialplan. On socket A I get the CUSTOM conference events, on socket B I don't get HEARTBEAT but I also get no CUSTOM conference events. I've also tried explicit enabling via: event plain CUSTOM conference::maintenence but still get no events (the spelling of maintenance is from the wiki). Can anyone else replicate or spot something I'm missing? thanks, - Dale Thatcher

3 2

Re: [Freeswitch-users] Incoming call audio problem
by Josip Djuricic 14 Mar '08

14 Mar '08

Ok, seems like I can't include any file larger then 100kb, so I can send the trace to people that want to check it out. As i said I still get no audio, but I'm rid of sip 415. From the trace you can see that there is a RTP from softphone trough FS, all the way to Cisco AS5300, but it doesn't get out on PSTN. Also FS get's no RTP from AS5300. This is all happening when I call from PSTN side into Voip network. When the call is coming from voip phone everything works well. Thank you all Djuricic wrote: > Hi there, > > I am including the new trace, this time with sip and rtp. > > I hope that somebody can see something I don't, cause there is still > no audio. > > This time I am doing the trace behind a NAT with a x-lite softphone > because I am at home. > > Josip Djuricic wrote: >> I meant GTD ;) >> >> Anyway, the traces now looks alot better, no SIP 415 anymore, but >> there is still no voice, and rtp seems to be going only from the >> phone to the FS >> >> Josip Djuricic wrote: >>> I can disable GDP but still no audio, I'm checkin the traces now >>> >>> Brian West wrote: >>>> Oh this bug has already been sent to MikeJ and Anthm. This is one >>>> strange invite. anyway its being worked on. You might want to see >>>> what in your config causes the addition of the application/gtd >>>> type. I suspect 'supplementary-service pass-through' >>>> >>>> /b >>>> >>>> >>>> INVITE sip:38514001148@89.18.32.19 >>>> <mailto:38514001148@89.18.32.19>:5080 SIP/2.0 Via: SIP/2.0/UDP >>>> 10.0.0.19:5060;x-route-tag="tgrp:Iskra" From: >>>> <sip:38513391900@10.0.0.19 >>>> <mailto:38513391900@10.0.0.19>>;tag=2D49486C-1855 To: >>>> <sip:38514001148@89.18.32.19 <mailto:38514001148@89.18.32.19>> >>>> Date: Thu, 13 Mar 2008 14:39:39 GMT Call-ID: >>>> 253738A2-F04211DC-81719B71-D67790FB(a)10.0.0.19 >>>> <mailto:253738A2-F04211DC-81719B71-D67790FB@10.0.0.19> Supported: >>>> timer,100rel Min-SE: 1800 Cisco-Guid: >>>> 624255889-4030861788-2171509617-3598160123 User-Agent: >>>> Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, >>>> PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO CSeq: 101 INVITE >>>> Max-Forwards: 6 Remote-Party-ID: <sip:38513391900@10.0.0.19 >>>> <mailto:38513391900@10.0.0.19>>;party=calling;screen=yes;privacy=off >>>> Timestamp: 1205419179 Contact: <sip:38513391900@10.0.0.19 >>>> <mailto:38513391900@10.0.0.19>:5060> Expires: 180 Allow-Events: >>>> telephone-event MIME-Version: 1.0 Content-Type: >>>> multipart/mixed;boundary=uniqueBoundary Content-Length: 523 >>>> --uniqueBoundary Content-Type: application/sdp v=0 >>>> o=CiscoSystemsSIP-GW-UserAgent 5947 1401 IN IP4 10.0.0.19 s=SIP >>>> Call c=IN IP4 10.0.0.19 t=0 0 m=audio 16850 RTP/AVP 8 c=IN IP4 >>>> 10.0.0.19 a=rtpmap:8 PCMA/8000 a=ptime:20 --uniqueBoundary >>>> Content-Type: application/gtd Content-Disposition: >>>> signal;handling=optional IAM, PRN,isdn*,,NET5*, USI,rate,c,3,c,1 >>>> USI,lay1,alaw TMR,02 CPN,04,,1,14001148 CGN,04,,1,y,4,13391900 >>>> CPC,09 FCI,,,,,,,y, GCI,25356391f04211dc816e9b71d67790fb >>>> --uniqueBoundary-- >>>> On Mar 13, 2008, at 9:51 AM, Josip Djuricic wrote: >>>> >>>>> supplementary-service pass-through >>>> >>>> ------------------------------------------------------------------------ >>>> >>>> _______________________________________________ >>>> 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 >> >

3 6

Re: [Freeswitch-users] 404 error
by Josip Djuricic 13 Mar '08

13 Mar '08

Dear Kurt, could you please send your dialplans, it seems like a routing problem. Thank you Josip

1 0

(no subject)
by Ben Levy 13 Mar '08

13 Mar '08

Is there any SS7 to sip functionality built into this project? Specifically I'm looking for a H324M to Sip gateway and don't want to use Asterisk. Thanks in advance for and answers you might have. Regards, Ben Levy President APEX Voice Communications, Inc THE APPLICATION SERVER COMPANY www.apexvoice.com Email: blevy(a)apexvoice.com Tel: +1.818.379.8400 x 103 Visit us at the following events... VON Mexico <http://www.von.com/2008/mexico/web/> Von Spring February 26 - 28, 2008 March 17 - 20, 2008 Mexico City, Mexico San Jose, CA, USA Booth 932 Booth 133 The information transmitted is intended only for the person(s) or entity to which it is addressed and may contain confidential and/or legally privileged material. Delivery of this message to any person other than the intended recipient(s) is not intended in any way to waive privilege or confidentiality. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by entities other than the intended recipient is prohibited. If you receive this in error, please contact the sender and delete the material from any computer.

2 1

Incoming call audio problem
by Josip Djuricic 13 Mar '08

13 Mar '08

Hi there, I was already writing about this problem, and I still didn't managed to fix it. I thought it was a NAT problem, but now without NAT the problem still remains. The error is like this, when I call from pstn, the call goes trough Cisco AS5300, then goes to FS, then to it's final destination - Ip Phone. Now you can answer the phone, but you can't hear anything on any side. And FS returns lots of SIP 415 (unsupported media type) to Cisco AS5300. When you call from Voip phone to the same pstn number it all works well, so I'm starting to be confused more and more. (and in this case there is no 415 messages). I believe cisco is sending something that FS can't understand, in the INVITE's but that's just a wild guess cause I ain't sure what can FS exactly understand very well. Whoever want's I can send him a tcpdump, it's about 7mb tarred, so I can't send it to the list. Thank you all, Josip

5 14

Re: [Freeswitch-users] freeswitch in vmware UPDATE
by Anthony Minessale 12 Mar '08

12 Mar '08

I spoke too soon, I mean try it now =D revision 7857! Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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 iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: John Skopis (Lists) <jlists(a)skopis.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Tuesday, March 11, 2008 7:35:00 AM Subject: Re: [Freeswitch-users] freeswitch in vmware UPDATE Anthony Minessale wrote: > can everyone who has any audio problems please > retest with after doing "make current" or a fresh checkout. > [snip] using Revision: 7855 no dice...same as before. I could try contacting vmware but I doubt we would get very far _______________________________________________ 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 ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs

2 1

woomera specific purpose (or maybe opal)
by Josip Djuricic 11 Mar '08

11 Mar '08

Hi there, I come to you again with very specific question. Is there a possibility to make woomera or opal work with freeswitch, cause I need it for specific purpose only. I don't need the call to work at all, what I would need would be if a call comes to woomera channel, just drop the call and generate specific release code. It is required for specific billing purpose in combination with mvts. Thank you very much. btw. about that other problem I had with incoming calls rtp not working, it was a nat problem. Josip

1 1

freeswitch in vmware
by Cavalera Claudio Luigi 11 Mar '08

11 Mar '08

Hello guys, at the moment the only linux I've got handy is a slackware 12.0 under vmware server 1.04 I've installed freeswitch to start a sort of comparison with asterisk. I've registered a SIP 1000 user and dialed 9999 (I could not wait :-). The config is default. The music is choppy, I mean silence - music - silence within a second! Sometimes it sounds good for ten seconds or so. Could this be due to clock skew problems of virtual machines? 008-03-04 12:43:24 [CRIT] switch_time.c:321 softtimer_runtime() Forward Clock Skew Detected! Best Regards, Claudio Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. -----------------------------------------------------------------------------------------------------

6 25

Re: [Freeswitch-users] IAX2 DTMF -> SIP RTP not reliable
by Anthony Minessale 11 Mar '08

11 Mar '08

nonetheless, can you please see my bugnote in jira. ===== can you edit /usr/local/freeswitch/conf/autoload_configs/iax.conf.xml and set the debug param to 1. Then reproduce the case where it's not working. We can see if the mod_iax is even getting the digits. ===== also be sure to sync to the current release by entering "make current" from the build root just to ensure your problem still exists in the latest revision. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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 iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Birgit Arkesteijn <birgit(a)westhawk.co.uk> To: freeswitch-users(a)lists.freeswitch.org Sent: Tuesday, March 11, 2008 9:36:22 AM Subject: Re: [Freeswitch-users] IAX2 DTMF -> SIP RTP not reliable Hi Anthony, I'm fairly sure that it's not an IAX problem. IAX2 messages are properly acked, sound (between IAX2 and SIP) is ok, etc, etc. The problem is (only) with DTMF, as far as I can tell. It's more likely to be in core or sofia_sip? Thanks, Birgit On 11/03/08 13:20, Anthony Minessale wrote: > Just to warn you we do not use mod_iax all that much. It was used as a > tool to test the core of FreeSWITCH in the early stages of development > and we just kinda left it hanging around since it does at least work. > We had to add a bunch of code to the stock iaxclient lib to make it > threadsafe etc because the lib is designed to handle 1 or 2 calls and we > are using it in a b2bua type env where hundreds of concurrent calls may > be possible. That said, since we more or less forked the code to avoid > generating support issues for the other soft-phones that use iaxclient, > you may want to check if there were any bugs in the real version of the > library that match your issue. > > http://sourceforge.net/projects/iaxclient > > > We've had little to no demand for IAX2 so we have not focused very much > on it. We do have strategies for creating a more robust and perhaps even > an "improved" implementation of the IAX2 protocol. Alas, it's pretty > much purely an academic exercise because there is little reason to > implement it. > > > > > > > > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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 > iax:guest@conference.freeswitch.org/888 > googletalk:conf+888@conference.freeswitch.org > pstn:213-799-1400 > > > ----- Original Message ---- > From: Birgit Arkesteijn <birgit(a)westhawk.co.uk> > To: Freeswitch-users(a)lists.freeswitch.org > Sent: Monday, March 10, 2008 10:52:40 AM > Subject: [Freeswitch-users] IAX2 DTMF -> SIP RTP not reliable > > Hi, > > We're running FreeSWITCH svn Trunk 7697 on SuSE Linux 9.3 (x86-64). > > I'm experiencing problems with DTMF: > The IAX2 DTMF frames aren't always transferred into RTP packets. > Sometimes none of the frames are translated, sometimes only the first > one is omitted. All the IAX2 DTMF packets are ack-ed. > > Our setup: > PhoneFromHere.com <http://PhoneFromHere.com> applet (IAX2 protocol) > -> FreeSWITCH (ns2.westhawk.co.uk <http://ns2.westhawk.co.uk>) > -> SIP to SystemX (sofia/default/123(a)66.212.134.192) > > > I searched on jira.freeswitch.org <http://jira.freeswitch.org> for a bug > that could explain this, but > failed, so I submitted this as a bug, see > http://jira.freeswitch.org/browse/FSCORE-109 > > Has anyone else experienced the same or a similar problem? > > Thanks, Birgit > -- -- Birgit Arkesteijn, birgit(a)westhawk.co.uk, -- Westhawk Ltd, Albion Wharf, 19 Albion Street, Manchester M1 5LN, UK -- Company no: 1769350 -- Registered Office: -- 15 London Road, Stockton Heath, Warrington WA4 6SJ. UK. -- tel.: +44 (0)161 237 0660 -- <URL: http://www.westhawk.co.uk> _______________________________________________ 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 ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs

1 0

Re: [Freeswitch-users] IAX2 DTMF -> SIP RTP not reliable
by Anthony Minessale 11 Mar '08

11 Mar '08

Just to warn you we do not use mod_iax all that much. It was used as a tool to test the core of FreeSWITCH in the early stages of development and we just kinda left it hanging around since it does at least work. We had to add a bunch of code to the stock iaxclient lib to make it threadsafe etc because the lib is designed to handle 1 or 2 calls and we are using it in a b2bua type env where hundreds of concurrent calls may be possible. That said, since we more or less forked the code to avoid generating support issues for the other soft-phones that use iaxclient, you may want to check if there were any bugs in the real version of the library that match your issue. http://sourceforge.net/projects/iaxclient We've had little to no demand for IAX2 so we have not focused very much on it. We do have strategies for creating a more robust and perhaps even an "improved" implementation of the IAX2 protocol. Alas, it's pretty much purely an academic exercise because there is little reason to implement it. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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 iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Birgit Arkesteijn <birgit(a)westhawk.co.uk> To: Freeswitch-users(a)lists.freeswitch.org Sent: Monday, March 10, 2008 10:52:40 AM Subject: [Freeswitch-users] IAX2 DTMF -> SIP RTP not reliable Hi, We're running FreeSWITCH svn Trunk 7697 on SuSE Linux 9.3 (x86-64). I'm experiencing problems with DTMF: The IAX2 DTMF frames aren't always transferred into RTP packets. Sometimes none of the frames are translated, sometimes only the first one is omitted. All the IAX2 DTMF packets are ack-ed. Our setup: PhoneFromHere.com applet (IAX2 protocol) -> FreeSWITCH (ns2.westhawk.co.uk) -> SIP to SystemX (sofia/default/123(a)66.212.134.192) I searched on jira.freeswitch.org for a bug that could explain this, but failed, so I submitted this as a bug, see http://jira.freeswitch.org/browse/FSCORE-109 Has anyone else experienced the same or a similar problem? Thanks, Birgit -- -- Birgit Arkesteijn, birgit(a)westhawk.co.uk, -- Westhawk Ltd, Albion Wharf, 19 Albion Street, Manchester M1 5LN, UK -- Company no: 1769350 -- Registered Office: -- 15 London Road, Stockton Heath, Warrington WA4 6SJ. UK. -- tel.: +44 (0)161 237 0660 -- <URL: http://www.westhawk.co.uk> _______________________________________________ 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 ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

2 1