March 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

test
by John Skopis (Lists) 06 Mar '08

06 Mar '08

test

3 2

Dialplan apps questions
by Michael Collins 05 Mar '08

05 Mar '08

Could someone define what these apps do? deflect intercept The wiki doesn't have these defined but we have sample extensions that use these dp apps. I will update the wiki with the definitions of these apps. Thanks, MC

3 5

Re: [Freeswitch-users] freeswitch in vmware
by Anthony Minessale 05 Mar '08

05 Mar '08

make current does a svn update cleans the tree and rebuilds. Are you not using a standard checkout? If you are behind a web proxy you need to do the initial checkout to port 81 http://svn.freeswitch.org:81/svn/freeswitch/trunk you may want to start this from scratch. 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: Cavalera Claudio Luigi <Claudio.Cavalera(a)italtel.it> To: freeswitch-users(a)lists.freeswitch.org Sent: Wednesday, March 5, 2008 3:25:49 AM Subject: Re: [Freeswitch-users] freeswitch in vmware Hello Anthony, it fails on the first command: make current Fetching external item into 'libs/openzap' svn: PROPFIND request failed on '/svn/openzap/trunk' svn: PROPFIND of '/svn/openzap/trunk': 403 Forbidden (http://svn.openzap.org:81) make: *** [current] Error 1 It's strange, I think my web proxy is involved in this. Ciao Claudio ________________________________ From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Tuesday, March 04, 2008 8:07 PM To: freeswitch-users(a)lists.freeswitch.org Subject: Re: [Freeswitch-users] freeswitch in vmware can you test this for me? first "make current" to update and rebuild the code. then test it again, if it still is happening, edit src/include/switch_private.h look at the top for /* Define if you have clock_gettime() */ #define HAVE_CLOCK_GETTIME 1 /* Define if you have clock_nanosleep() */ #define HAVE_CLOCK_NANOSLEEP 1 and comment them /* Define if you have clock_gettime() */ //#define HAVE_CLOCK_GETTIME 1 /* Define if you have clock_nanosleep() */ //#define HAVE_CLOCK_NANOSLEEP 1 then close it and from the main src dir execute "make current" again once it's done retest if the problem persists, edit the file again and restore it by removing the comments and add this extra line anywhere at the top #define SWITCH_USE_CLOCK_FUNCS 1 and again "make sure" and retest please report the results and we can see if there is something we can do. 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: Brian Young ARA/CFD <byoung(a)ara.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Tuesday, March 4, 2008 11:08:02 AM Subject: Re: [Freeswitch-users] freeswitch in vmware I am also a new user. Running under WinXP, and have experienced very similar results: choppy music for default 9999, but clear for other recordings such as default 3000 (conf-alone.wav) and conference controls (mute, unmute, etc). -Brian Young <brian.young(a)ara.com> -----Original Message----- From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Cavalera Claudio Luigi Sent: Tuesday, March 04, 2008 11:28 AM To: freeswitch-users(a)lists.freeswitch.org Subject: Re: [Freeswitch-users] freeswitch in vmware I've solved the timing issues in fs vmware, I mean that those error messages are now history. However the problem with moh dialing 9999 still remains. I've tried from my pc and another one. Music is "choppy". I've also tried a 1000 to 1001 sip call using zoiper and it sounds crystal clear. Strange. Is there any default iax user in the system? Best Regards, Claudio > -----Original Message----- > From: freeswitch-users-bounces(a)lists.freeswitch.org > [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of > Michael Jerris > Sent: Tuesday, March 04, 2008 3:13 PM > To: freeswitch-users(a)lists.freeswitch.org > Subject: Re: [Freeswitch-users] freeswitch in vmware > > This definately looks like a broken clock in the vm, but executing the > event late by 65 seconds is a lot. did you pause your vm? > This could > be related to clock issues in slackware as well, I know people who run > fine in vmware so it is certainly doable. Why not just run freeswitch > on your native os instead of in vmware? > > Mike > > > On Mar 4, 2008, at 6:49 AM, Cavalera Claudio Luigi wrote: > > > Maybe this debug would also be useful: > > 2008-03-04 12:44:31 [WARNING] switch_scheduler.c:114 > > task_thread_loop() > > Task was executed late by 65 seconds 1 heartbeat (core) > > 2008-03-04 12:46:47 [CRIT] mod_local_stream.c:208 > read_stream_thread() > > Leaking stream handle! [switch_ivr_play_file() > > src/switch_ivr_play_say.c:738] > > > >> -----Original Message----- > >> From: Cavalera Claudio Luigi > >> Sent: Tuesday, March 04, 2008 12:48 PM > >> To: 'freeswitch-users(a)lists.freeswitch.org' > >> Subject: freeswitch in vmware > >> > >> 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. > > > -------------------------------------------------------------- > --------------------------------------- > > > > > > _______________________________________________ > > 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/freesw > itch-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/freesw > itch-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 ________________________________ Never miss a thing. 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3 3

Event
by Jonas Gauffin 05 Mar '08

05 Mar '08

Isn't the channel state incorrect in the event below? No calls have been made. proto: sip login: sip%3Amod_sofia%40192.168.0.58%3A5070 from: jonas%40192.168.0.58 status: Registered rpid: unknown event_type: presence alt_event_type: dialog event_count: 0 unique-id: sip channel-state: CS_HANGUP answer-state: resubscribe Event-Name: PRESENCE_IN Core-UUID: b93eae74-8e4e-5946-bd96-15e2661acb92 FreeSWITCH-Hostname: jg-hempc FreeSWITCH-IPv4: 192.168.0.58 FreeSWITCH-IPv6: 127.0.0.1 Event-Date-Local: 2008-03-05%2016%3A56%3A54 Event-Date-GMT: Wed,%2005%20Mar%202008%2015%3A56%3A54%20GMT Event-Date-timestamp: 1204732614031250 Event-Calling-File: sofia_presence.c Event-Calling-Function: sofia_presence_resub_callback Event-Calling-Line-Number: 555

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Re: [Freeswitch-users] Does dialplan. bridge application can detect call hangup earlier?
by Anthony Minessale 05 Mar '08

05 Mar '08

What is it that you are calling in that example? Is it a pstn gateway or something? It sounds like the gateway you are calling always answers the call before it tries to dial the far end destination and does not have proper supervision handling. The correct method would be that the gateway places the far end call without answering it and passes the indications across as they are encountered by the far end. You may want to send your question to whoever is in charge of that gateway as I am fairly certain we do basic call scenarios correctly since we have had people interop with several major carriers. 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: Steve Xu <516888(a)gmail.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Tuesday, March 4, 2008 6:01:56 PM Subject: [Freeswitch-users] Does dialplan. bridge application can detect call hangup earlier? Hi all, Here is another question. I try to use dialplan <action application="bridge" data="sofia/default/999917728693819(a)66.55.44.33"/> to bridge two SIP endpoints. Here are the strange pattern I found during call handling. 1.The called party still keep ringing for a long time period even I have hanged up the calling party already. I have to pick up the called party to stop the ring sometimes, however the call record shows the call has been answered. I think FS will detect the calling party status and drop the call if calling party hung up. It does not need to waiting for the status of called party changes. 2.The calling party does not hear anything ( ie. a busy tone) when the called party hang up the call during the normal conversation, the call keep connecting there until the calling party hang up, then FS start realize the call ended. Is there some configuration setting can make FS to detect the call status earlier and make the call handling properly? or may be I used the wrong action application (bridge)to handle the calls? Thanks again, Steven Xu -----Inline Attachment Follows----- _______________________________________________ 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

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Re: [Freeswitch-users] Does freeswitch can only pass through for g729 and g723?
by Anthony Minessale 05 Mar '08

05 Mar '08

Yes you can build and install both these codecs in passthru mode which will only allow bridging the calls and not any transcoding. You can also try the new proxy_media mode or bypass_media mode which does transparent proxy of the calls. http://wiki.freeswitch.org/wiki/Bypass_Media http://wiki.freeswitch.org/wiki/Proxy_Media And finally you will soon be able to purchace g729 licenses for use with FreeSWITCH so you can do media trans coding as well. 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: dingo <dingo800(a)gmail.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Tuesday, March 4, 2008 9:21:45 PM Subject: [Freeswitch-users] Does freeswitch can only pass through for g729 and g723? Hi all, Does freeswitch can only pass through for g729 and g723? I have a phone support only g729 or g723. What can I do to have the great feature of freeswitch with my phone? Any ideas? -----Inline Attachment Follows----- _______________________________________________ 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 ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping

1 0

Cant start freeswitch
by Henk Oegema 05 Mar '08

05 Mar '08

After a server restart it was ok :)

1 0

Cant start freeswitch
by Henk Oegema 05 Mar '08

05 Mar '08

I just compiled Freeswitch on a Kubuntu 7.10 system without any errors. :) However, when I want to start Freeswith (in /usr/local/freeswitch/bin): $ sudo ./freeswitch I get: Cannot lock pid file /usr/local/freeswitch/log/freeswitch.pid. :( (I also tested Freeswitch on a Debian system, and there it's ok)

3 2

Does freeswitch can only pass through for g729 and g723?
by dingo 05 Mar '08

05 Mar '08

Hi all, Does freeswitch can only pass through for g729 and g723? I have a phone support only g729 or g723. What can I do to have the great feature of freeswitch with my phone? Any ideas?

1 0

Does dialplan. bridge application can detect call hangup earlier?
by Steve Xu 05 Mar '08

05 Mar '08

Hi all, Here is another question. I try to use dialplan <action application="bridge" data="sofia/default/999917728693819(a)66.55.44.33"/> to bridge two SIP endpoints. Here are the strange pattern I found during call handling. 1.The called party still keep ringing for a long time period even I have hanged up the calling party already. I have to pick up the called party to stop the ring sometimes, however the call record shows the call has been answered. I think FS will detect the calling party status and drop the call if calling party hung up. It does not need to waiting for the status of called party changes. 2.The calling party does not hear anything ( ie. a busy tone) when the called party hang up the call during the normal conversation, the call keep connecting there until the calling party hang up, then FS start realize the call ended. Is there some configuration setting can make FS to detect the call status earlier and make the call handling properly? or may be I used the wrong action application (bridge)to handle the calls? Thanks again, Steven Xu

1 0