- FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Recording to a Shoucast Stream
by Anthony Minessale 31 Oct '07

31 Oct '07

like this: shout://user:pass@host.com/mount.mp3 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: David Revill <davidrevill(a)datarun.co.uk> To: freeswitch-users(a)lists.freeswitch.org Sent: Tuesday, October 30, 2007 9:26:02 AM Subject: [Freeswitch-users] Recording to a Shoucast Stream Can any one help with an example of how to record to a Shoutcast stream ? I have mod_shout installed and can playback from Shoutcast but just need a simple example of the application call. Thanks __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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Von
by Neil Weaver 31 Oct '07

31 Oct '07

Hello everyone, Are there any freeswitch folks here at Von in Boston? If so, lets get together!

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Re: [Freeswitch-users] Compile error, switch_swig.c
by tuhl＠ix.netcom.com 30 Oct '07

30 Oct '07

Thanks! The 2-line change fixed the compile error. Tom At 10:38 AM 10/30/2007, Trixter aka Bret McDanel wrote: >On 10/30/07, <mailto:tuhl@ix.netcom.com>tuhl(a)ix.netcom.com ><<mailto:tuhl@ix.netcom.com>tuhl(a)ix.netcom.com> wrote: >Hi, I'm in the process of updating freeswitch on a Linux server and >found this compile error. The list of commands I used to get the >source and compile, is at the end of the email. > >... > > > >there is a patch which fixes this issue. Its part of my patches for >gcc 4.3 compatibility (just coincidence that its in there as I >enabled every module to try to make gcc 4.3 happy). If you want to >test it, and this file is only one or two lines if you wanna just >pull that out, feel free. > > ><http://jira.freeswitch.org/browse/MODAPP-44>http://jira.freeswitch.org/browse/MODAPP-44 > > >-- >Trixter <http://www.0xdecafbad.com>http://www.0xdecafbad.com Bret McDanel >Belfast +44 28 9099 6461 US +1 516 687 5200 ><http://www.trxtel.com>http://www.trxtel.com the phone company that pays you! >_______________________________________________ >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 =============== tuhl(a)ix.netcom.com

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Recording to a Shoucast Stream
by David Revill 30 Oct '07

30 Oct '07

Can any one help with an example of how to record to a Shoutcast stream ? I have mod_shout installed and can playback from Shoutcast but just need a simple example of the application call. Thanks

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Re: [Freeswitch-users] Audio probs with newer (and may be older)FreeSwitch versions under WinXP
by Anthony Minessale 30 Oct '07

30 Oct '07

if you do report it to jira, please include a pacap of the sip and rtp traffic as well as the entire console log of freeswitch with the env TPORT_LOG=1 set before you start. 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: "Sluschny, Thomas" <Thomas.Sluschny(a)siemens.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Friday, October 26, 2007 6:50:15 AM Subject: Re: [Freeswitch-users] Audio probs with newer (and may be older)FreeSwitch versions under WinXP DIV { MARGIN:0px;} I use FreeSwitch as a SIP phone, that means i also use portaudio device, and this under WinXP. Scenario is that i use command: pa call xxx with xxx configured as bridge to sofia/profile/xxx(a)device.ip Called devices: e. g. siemens optipoint 400 standard sip, optipoint 410 advance S, proprietary linux box with osip. Its also the same if one device is calling freeswitch (pa answer to offhook incoming call). Everything is live audio, no file playing. Audio codec is G.711u with 20ms (packetsize 160). I will test with lastest svn on weekend at home, if still exists i'll open a jira entry... Thomas Von: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] Im Auftrag von Anthony Minessale Gesendet: Donnerstag, 25. Oktober 2007 21:15 An: freeswitch-users(a)lists.freeswitch.org Betreff: Re: [Freeswitch-users] Audio probs with newer (and may be older)FreeSwitch versions under WinXP Can you describe the entire setup? What OS is FS running on, windows I assume in this case? is this a sip call? If so what device/phone is making the call? is this a call to an extension playing a file or a bridge to another channel? please describe in full detail all information you have about how you are setup and what you are trying to do when it doesn't work, including how you have configured everything. 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: "Sluschny, Thomas" <Thomas.Sluschny(a)siemens.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Thursday, October 25, 2007 9:18:41 AM Subject: [Freeswitch-users] Audio probs with newer (and may be older) FreeSwitch versions under WinXP Hi all, i have little problems with audio quality in newer FreeSwitch versions: At most i use a very old FreeSwitch version (03/2007 or so). Here i see typically all 60 ms three audio packets at once. This is not really nice for jitter and delay, but most equipment can cope with it. BUT: In newer FreeSwitch-Version (last tested snapshot from 22.10.2007) Audio sounds really bad for me! Now i see in Wireshark the same behavior like before (3 packets all 60ms), but now the Timestamp for all 3 packets is the same! And no equipment can cope with it (test some boxes and SipPhones). So my question: When it has been changed, or better: where is the code i can play with to fix this (at least for me). I tried my Work PC and a Laptop under WinXP, both the same. Anyone else has this problem? best regards, Thomas Sluschny __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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Audio probs with newer (and may be older) FreeSwitch versions under WinXP
by Sluschny, Thomas 27 Oct '07

27 Oct '07

Hi all, i have little problems with audio quality in newer FreeSwitch versions: At most i use a very old FreeSwitch version (03/2007 or so). Here i see typically all 60 ms three audio packets at once. This is not really nice for jitter and delay, but most equipment can cope with it. BUT: In newer FreeSwitch-Version (last tested snapshot from 22.10.2007) Audio sounds really bad for me! Now i see in Wireshark the same behavior like before (3 packets all 60ms), but now the Timestamp for all 3 packets is the same! And no equipment can cope with it (test some boxes and SipPhones). So my question: When it has been changed, or better: where is the code i can play with to fix this (at least for me). I tried my Work PC and a Laptop under WinXP, both the same. Anyone else has this problem? best regards, Thomas Sluschny

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Re: [Freeswitch-users] Call problems
by Anthony Minessale 26 Oct '07

26 Oct '07

please retest with latest trunk as a few issues were fixed with transfers today. Be aware that the transfers work best with bridged calls. It sounds like that is what you are doing so it should be ok but there are a few things to consider when calling an extension that is just "playback" for example. Say you call an extension that has playback foo.wav: Technically the file is playing on the inbound leg and this is considered a 1 legged call. so if you blind transfer this extension there is nothing to transfer cos there is only 1 leg so in this case we will not allow it. if you attended transfer this extension, we will transfer the new channel you called to the same extension as the one you were on and hangup the original so the playback will start over for the called party. if you want to call and ivr or playfile extension and then transfer it you have to make a hairpin call back into the system. so instead of making 1000 lead to playback foo.wav make it lead to sofia/default/playback-foo(a)same.box then you have 2 legs and it's possible to transfer the playback. 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: Jonas Gauffin <jonas.gauffin(a)gmail.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Thursday, October 25, 2007 6:55:45 AM Subject: Re: [Freeswitch-users] Call problems Ok. I've created a jira entry. On 10/25/07, Michael Jerris <mike(a)jerris.com> wrote: > We would need a full trace from freeswitch including the TPORT_LOG to > determine that. > > > On 10/25/07, Jonas Gauffin < jonas.gauffin(a)gmail.com> wrote: > > > > Hello > > > > I have three phones, all registered with FreeSwitch. > > > > I did the following: > > a) Called from phone A to phone B > > b) B picked up > > c) A put B on hold > > c) A called phone C > > d) C picked up > > e) Pressed transfer on A's phone > > > > Both B and C got disconnected instead of being connected to each other. > > > > I did another variant: > > A calls B, B calls C, B presses transfer. > > A things that he is still connected, B and C gets disconnected. > > > > Are these problems in my phones, or are they FS problems? > > > > //Jonas > > > > _______________________________________________ > > 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 __________________________________________________ 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] Audio probs with newer (and may be older) FreeSwitch versions under WinXP
by Anthony Minessale 26 Oct '07

26 Oct '07

Can you describe the entire setup? What OS is FS running on, windows I assume in this case? is this a sip call? If so what device/phone is making the call? is this a call to an extension playing a file or a bridge to another channel? please describe in full detail all information you have about how you are setup and what you are trying to do when it doesn't work, including how you have configured everything. 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: "Sluschny, Thomas" <Thomas.Sluschny(a)siemens.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Thursday, October 25, 2007 9:18:41 AM Subject: [Freeswitch-users] Audio probs with newer (and may be older) FreeSwitch versions under WinXP Hi all, i have little problems with audio quality in newer FreeSwitch versions: At most i use a very old FreeSwitch version (03/2007 or so). Here i see typically all 60 ms three audio packets at once. This is not really nice for jitter and delay, but most equipment can cope with it. BUT: In newer FreeSwitch-Version (last tested snapshot from 22.10.2007) Audio sounds really bad for me! Now i see in Wireshark the same behavior like before (3 packets all 60ms), but now the Timestamp for all 3 packets is the same! And no equipment can cope with it (test some boxes and SipPhones). So my question: When it has been changed, or better: where is the code i can play with to fix this (at least for me). I tried my Work PC and a Laptop under WinXP, both the same. Anyone else has this problem? best regards, Thomas Sluschny __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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IAX2 <--> H323
by AFShin Abdollahi 26 Oct '07

26 Oct '07

Hi all, I'm looking for a Solution for IAX2 <--> H323 and SIP <---> H323. Can I rely on FreeSwitch to do these for me? Our Switch work with H323 only, so I want to find a way to translate other protocols to our switch. If it does, Any sample configurations, any documents, guide, is really appreciated. Regards, AFShin

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Presence doesn't work on windows
by Hassan Firouzbakht 26 Oct '07

26 Oct '07

Hi, It seems Presence doesn't work on windows or at least there is a special configuration that I am not aware about that. I appreciate if somebody tell me what is wrong in here. Here is the procedures: 1) Sent SIP Subscribe , Request event is = Presence.winfo (Watcher info) 2) Received 202 - Accept. 3) Waited for SIP Notify message. 4) Nothing happened Again 1) Sent SIP Subscribe to add a buddy into my contact list, Request event is = presence. 2) Received 202 -Accept. 3) Waited for SIP Notify message, 4) Nothing happened In another scenario 1) Sent SIP Publish with my status in pidf+xml format 2) Received 202 -Accept. 4) Nothing happened Thanks a lot, Hassan _________________________________________________________________ R U Ready for Windows Live Messenger Beta 8.5? Try it today! http://entertainment.sympatico.msn.ca/WindowsLiveMessenger

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