- FreeSWITCH-users - FreeSWITCH Mailing Lists

Update to MODENDP-272
by John Platts 02 Dec '09

02 Dec '09

I have uploaded the dialplan and JavaScript files used to process calls to MODENDP-272. I have even done a make current to revision 15755, and the blind transfer is still failing. _________________________________________________________________ Windows 7: Unclutter your desktop. Learn more. http://www.microsoft.com/windows/windows-7/videos-tours.aspx?h=7sec&slideid…

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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 42, Issue 12
by sharad 02 Dec '09

02 Dec '09

Hello We also faced the similar issue. Actually it is caused bacause hold on music files are missing. either you save all the music files or configure your dialplan accordingly. Sharad , Coral Telecom, India ----- Original Message ----- From: <freeswitch-users-request(a)lists.freeswitch.org> To: <freeswitch-users(a)lists.freeswitch.org> Sent: Wednesday, December 02, 2009 3:24 PM Subject: FreeSWITCH-users Digest, Vol 42, Issue 12 > 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: OSP Interop w/ Trans Nexus (Max Clark) > 2. Re: CLIP on FXS channels with mod_openzap (Fran?ois Legal) > 3. Bridging/Connecting Freeswitch servers (Otis) > 4. Re: Transfer Problem (Jonathan Barou) > -------------------------------------------------------------------------------- > _______________________________________________ > 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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freeswitch changes c= to FS's IP with bypass_media=true
by Juan Backson 02 Dec '09

02 Dec '09

Hi, I found that with bypass_media=true, freeswitch would change c= to FS's own IP. I think this is a misconfiguration. Does anyone know what config could have caused that? thanks, jb

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OSP Interop w/ Trans Nexus
by Gregory Boehnlein 02 Dec '09

02 Dec '09

Anyone know if Freeswitch can work w/ OSP and use something like Transnexus for CDR/Rating?

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Re: [Freeswitch-users] Choppy sound with PCMU
by erandr-junk＠usa.net 02 Dec '09

02 Dec '09

Thanks. I tried that... Just forcing SPA to 20ms didn't change anything. Just installing SVN trunk didn't fix it either, but setting that option afterwards surely did the trick. One thing I've noticed while staring at the console is that it *looks like* that w/o the new setting the stuttering happens when FS either re-registers itself with the provider or one of the SPA's port re-registers with FS. ------ Original Message ------ Received: Tue, 01 Dec 2009 05:33:26 PM EST From: Anthony Minessale <anthony.minessale(a)gmail.com> To: freeswitch-users(a)lists.freeswitch.org Subject: Re: [Freeswitch-users] Choppy sound with PCMU > linksys has had a bug for eons that can be fixed by setting the ptime (or > rtp packet size in their terms) > in it's firmware to .20 instead of .30 > > Asterisk does not use async RTP like we do so it's never a problem > you can disable the timer by setting the channel var rtp_timer_name=none or > sofia param rtp-timer-name to none in the sofia profile. > > You should also test this on latest SVN trunk or wait for pre8 > > > > On Tue, Dec 1, 2009 at 3:52 PM, eaf <erandr-junk(a)usa.net> wrote: > > > > > I should also add, after browsing through some topics here, that my SIP > > provider sends 172-byte RTP frames, which is in accordance with ptime:20 > > that it gives to FreeSWITCH. > > > > > > eaf wrote: > > > > > > Hi, > > > > > > I'm trying to migrate from Asterisk to FreeSWITCH (really like the way > > how > > > it can be programmed), but ran into one issue with sound quality that I > > > just cannot workaround by myself. I would describe the sound problem as > > > being "choppy". From time to time small portions of the other party's > > > voice are dropped, so the voice kind of stutters. This is not too bad, > > but > > > is really noticeable, happens in every call and I don't experience the > > > same with Asterisk running on the same box. I attached two files: > > > freeswitch.wav and asterisk.mp3 to illustrate my point. > > > > > > Issue completely goes away, if I set inbound-proxy-media to true. > > > > > > The way how I test is to connect SPA-2000 via 10mbps LAN to the box > > > directly exposed to internet, and then dial a toll-free via FutureNine (a > > > SIP provider). > > > > > > The codec in use is PCMU. Can't really try PCMA or anything else with > > this > > > provider. Only PCMU. Tried to match ptime of provider (30) with ptime of > > > the SPA, didn't get any improvement. Tried turning off recording, no > > > change either. > > > > > > What puzzles me is that even with greedy codec negotiations and with PCMU > > > on both sides of FreeSWITCH, it's still saying that > > > TRANSCODING_NECESSARY. I'm attaching relevant portion of freeswitch.log > > to > > > illustrate. > > > > > > The box isn't particularly fast: Linux (Debian 4), CPU - AMD Geode LX800 > > > with 997 bogomips. 256MB RAM. Only one call in progress, so I hope that > > > it's not a performance issue. > > > > > > http://old.nabble.com/file/p26594250/freeswitch.wav freeswitch.wav > > > > > > http://old.nabble.com/file/p26594250/asterisk.mp3 asterisk.mp3 > > > > > > http://old.nabble.com/file/p26594250/freeswitch.log freeswitch.log > > > > > > Tried both 1.0.4 and 1.0.5pre5. Same results. > > > > > > What should I do next? Calls are consistently bad with FreeSWITCH, and > > > consistently show no glitches with Asterisk. > > > > > > > > > > -- > > View this message in context: > > http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26599565.html > > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > > > > _______________________________________________ > > 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 <MSN%3Aanthony_minessale(a)hotmail.com> > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale(a)gmail.com> > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org <sip%3A888(a)conference.freeswitch.org> > iax:guest@conference.freeswitch.org/888 > googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888(a)conference.freeswitch.org> > pstn:213-799-1400 > > _______________________________________________ > 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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Ring Forever
by Esben Stien 02 Dec '09

02 Dec '09

I'd like to set up an extension that would just ring forever. When a person calls this extension, it would ring until the end of times. I've tried several ways to do this, without luck, and I don't find any information on this on the wiki. Any pointers how to do this?. -- Esben Stien is b0ef@e s a http://www. s t n m irc://irc. b - i . e/%23contact sip:b0ef@ e e jid:b0ef@ n n

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FreeSWITCH Survey: What Environment Do You Normally Use?
by Michael Collins 02 Dec '09

02 Dec '09

Hi folks, I'm doing a little survey to get an idea of what everyone prefers to use for their operating environment, like 32 vs. 64 bit, Linux vs. Windows, etc. Please log in to the main page and check out this node: http://www.freeswitch.org/node/206 Select the environment that you use the most or prefer to use. In a week or so I will send out the final tally. I'm sure the only question is who will come in second place after 64-bit CentOS/Red Hat. :) -Michael

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Re: [Freeswitch-users] Choppy sound with PCMU
by eaf 01 Dec '09

01 Dec '09

I should also add, after browsing through some topics here, that my SIP provider sends 172-byte RTP frames, which is in accordance with ptime:20 that it gives to FreeSWITCH. eaf wrote: > > Hi, > > I'm trying to migrate from Asterisk to FreeSWITCH (really like the way how > it can be programmed), but ran into one issue with sound quality that I > just cannot workaround by myself. I would describe the sound problem as > being "choppy". From time to time small portions of the other party's > voice are dropped, so the voice kind of stutters. This is not too bad, but > is really noticeable, happens in every call and I don't experience the > same with Asterisk running on the same box. I attached two files: > freeswitch.wav and asterisk.mp3 to illustrate my point. > > Issue completely goes away, if I set inbound-proxy-media to true. > > The way how I test is to connect SPA-2000 via 10mbps LAN to the box > directly exposed to internet, and then dial a toll-free via FutureNine (a > SIP provider). > > The codec in use is PCMU. Can't really try PCMA or anything else with this > provider. Only PCMU. Tried to match ptime of provider (30) with ptime of > the SPA, didn't get any improvement. Tried turning off recording, no > change either. > > What puzzles me is that even with greedy codec negotiations and with PCMU > on both sides of FreeSWITCH, it's still saying that > TRANSCODING_NECESSARY. I'm attaching relevant portion of freeswitch.log to > illustrate. > > The box isn't particularly fast: Linux (Debian 4), CPU - AMD Geode LX800 > with 997 bogomips. 256MB RAM. Only one call in progress, so I hope that > it's not a performance issue. > > http://old.nabble.com/file/p26594250/freeswitch.wav freeswitch.wav > > http://old.nabble.com/file/p26594250/asterisk.mp3 asterisk.mp3 > > http://old.nabble.com/file/p26594250/freeswitch.log freeswitch.log > > Tried both 1.0.4 and 1.0.5pre5. Same results. > > What should I do next? Calls are consistently bad with FreeSWITCH, and > consistently show no glitches with Asterisk. > > -- View this message in context: http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26599565.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Choppy sound with PCMU
by eaf 01 Dec '09

01 Dec '09

Hi, I'm trying to migrate from Asterisk to FreeSWITCH (really like the way how it can be programmed), but ran into one issue with sound quality that I just cannot workaround by myself. I would describe the sound problem as being "choppy". From time to time small portions of the other party's voice are dropped, so the voice kind of stutters. This is not too bad, but is really noticeable, happens in every call and I don't experience the same with Asterisk running on the same box. I attached two files: freeswitch.wav and asterisk.mp3 to illustrate my point. Issue completely goes away, if I set inbound-proxy-media to true. The way how I test is to connect SPA-2000 via 10mbps LAN to the box directly exposed to internet, and then dial a toll-free via FutureNine (a SIP provider). The codec in use is PCMU. Can't really try PCMA or anything else with this provider. Only PCMU. Tried to match ptime of provider (30) with ptime of the SPA, didn't get any improvement. Tried turning off recording, no change either. What puzzles me is that even with greedy codec negotiations and with PCMU on both sides of FreeSWITCH, it's still saying that TRANSCODING_NECESSARY. I'm attaching relevant portion of freeswitch.log to illustrate. The box isn't particularly fast: Linux (Debian 4), CPU - AMD Geode LX800 with 997 bogomips. 256MB RAM. Only one call in progress, so I hope that it's not a performance issue. http://old.nabble.com/file/p26594250/freeswitch.wav freeswitch.wav http://old.nabble.com/file/p26594250/asterisk.mp3 asterisk.mp3 http://old.nabble.com/file/p26594250/freeswitch.log freeswitch.log Tried both 1.0.4 and 1.0.5pre5. Same results. What should I do next? Calls are consistently bad with FreeSWITCH, and consistently show no glitches with Asterisk. -- View this message in context: http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26594250.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Cpu peak to 100% Openzap E1 R2
by Dome Charoenyost 01 Dec '09

01 Dec '09

Dear All, I got problem about openzap. I plan to use 4E1 R2 (Phone EQ card). I use zaptel driver from http://e400p.phoniceq.com/driver/ (1.4.12.1) and freeswitch 1.0.5 pre 7 My Server Intel(R) Core(TM)2 Quad CPU Q6600 @ 2.40GHz and 2 GB memmory. when i start freeswitch if enable echo cencel in zt.conf CPU peak to 100% when i disable echo cancel FS can start but when first call incominng got 100% again. Someone can help me? BG Dome

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