December 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

OT: Spa2102 and call transfer
by Jonas Gauffin 09 Dec '09

09 Dec '09

Hello, I can't get call transfer to work with a SPA2102 adapter. I don't think it has something to do with FS, but I'm hoping someone here could help me. I do not get a new line in the phone (by pressing the R button), all DTMF tones are sent as audio to the other connected phone. Anyone got it working? Thanks, Jonas

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Re: [Freeswitch-users] CLIP on FXS channels with mod_openzap
by François Legal 09 Dec '09

09 Dec '09

Sure, I'll try that. I'm just building freeswitch-snapshot that I downloaded from files.freeswitch.org I also experience, when bridging a call from an FXS to FXO the call is cut after a random time (this does not appear when bridging SIP to FXO). Might this upgrade fix this problem also ? François On Mon, 30 Nov 2009 10:48:26 -0600, Anthony Minessale wrote: can you test svn trunk or latest pre release of 1.0.5 On Mon, Nov 30, 2009 at 9:36 AM, François Legal wrote: Hello, I'm using Freeswitch with a Sangoma A400 card, and I'm having CLIP problems on the FXS ports. When I ring on FXS ports, the connected phone does not display callerid/callerid-name. I tried turning the stuff of in openzap.conf.xml () but it did not help. As a side note, turning this on on the FXO ports drops the callerid information on incoming calls. Running freeswitch 1.0.4 on linux 2.6.27. François _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org [2] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [3] UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [4] http://www.freeswitch.org [5] -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ [6] ClueCon http://www.cluecon.com/ [7] Twitter: http://twitter.com/FreeSWITCH_wire [8] AIM: anthm MSN:anthony_minessale@hotmail.com [9] GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com [10] IRC: irc.freenode.net [11] #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org [12] iax:guest@conference.freeswitch.org/888 [13] googletalk:conf+888@conference.freeswitch.org [14] pstn:213-799-1400 Links: ------ [1] mailto:devel@thom.fr.eu.org [2] mailto:FreeSWITCH-users@lists.freeswitch.org [3] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [4] http://lists.freeswitch.org/mailman/options/freeswitch-users [5] http://www.freeswitch.org [6] http://www.freeswitch.org/ [7] http://www.cluecon.com/ [8] http://twitter.com/FreeSWITCH_wire [9] mailto:MSN%3Aanthony_minessale@hotmail.com [10] mailto:PAYPAL%3Aanthony.minessale@gmail.com [11] http://irc.freenode.net [12] mailto:sip%3A888@conference.freeswitch.org [13] http://iax:guest@conference.freeswitch.org/888 [14] mailto:googletalk%3Aconf%2B888@conference.freeswitch.org

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[OpenZAP] Does Dahdi (ex-Zaptel) require Asterisk?
by Fred-145 09 Dec '09

09 Dec '09

Hello I'd like to install OpenZAP so I can use a TDM card with Freeswitch, but I'm getting a software error althought the TDM card seems detected (lspci -v OK). Dahdi was successfully compiled from source code. Is it OK to just install Dahdi 2.2.0 without Asterisk before going ahead with OpenZAP? The reason I ask, is that in another forum, someone mentionned "/etc/asterisk/chan_dahdi.conf". Here's the output from dahdi_cfg -vvv: DAHDI Tools Version - 2.2.0 DAHDI Version: 2.2.0.2 Echo Canceller(s): Configuration ====================== Channel map: Channel 01: FXS Kewlstart (Default) (Echo Canceler: none) (Slaves: 01) 1 channels to configure. DAHDI_CHANCONFIG failed on channel 1: No such device or address (6) Here's ls -l /dev/dahdi/: total 0 crw-rw---- 1 root root 196, 254 Dec 8 13:38 channel crw-rw---- 1 root root 196, 0 Dec 8 13:38 ctl crw-rw---- 1 root root 196, 255 Dec 8 13:38 pseudo crw-rw---- 1 root root 196, 253 Dec 8 13:38 timer Has someone succesfully installed Dahdi without Asterisk? Thank you. -- View this message in context: http://old.nabble.com/-OpenZAP--Does-Dahdi-%28ex-Zaptel%29-require-Asterisk… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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continue_on_fail
by Peter P GMX 09 Dec '09

09 Dec '09

I have a Problem with continue_on_fail. I have setup a hunt group <action application="set" data="continue_on_fail=NO_ANSWER,USER_BUSY"/> <action application="bridge" data="sofia/external/219@10.11.12.243,sofia/external/223@10.11.12.234,sofia/external/236@10.11.12.188,sofia/external/101@10.11.12.245"/> <action application="bridge" data= (dialstring for fallback user ) I want the fallback user to be called whenever none of the previously called 3 gateway numbers picks up or if they are all busy. Therefore continue_on_fail=NO_ANSWER,USER_BUSY The fallback user is called, however if any of the previously called gateways picks up and then hangs up, the fallback user is called afterwards. Means: The fallback user is always called. I had expected, that continue_on_fail=NO_ANSWER,USER_BUSY would not fire the next bridge if it gets a NORMAL_CLEARING. Am I thinking wrongly about this? I have added <action application="set" data="hangup_after_bridge=true"/> and this works, but I would like to specify more in detail the conditions when to follow the next hunt group entry. Best regards Peter

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Re: [Freeswitch-users] FXO PCI card with Pfsense Freeswitch.
by Ryanny Lin 09 Dec '09

09 Dec '09

hi I don't know this driver for freebsd works or not, maybe you can check it out and try it. Here is the Digium's SVN repository: http://svn.digium.com/svn/dahdi/ It looks like driver for freebsd is under testing. Or use zaptel driver ... Change to root, then cd /usr/ports/misc/zaptel make install && make clean uh..., maybe PfSense doesn't have /usr/ports ...you can just install zaptel. On Tue, Dec 8, 2009 at 3:25 PM, Orien Love <orien at tx.rr.com <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>> wrote: > > >* I am looking for a 4 port FXO card to use with my PfSense installation > *>* of freeswitch. does anybody know if the > *>* Asterisk Trixbox 4 FXO Voip Digium TDM410 will work on pfsense? > *>* or could somebody recommend one that would. > *>* > *>* Thank You > *>* Orien > *>* > *>* _______________________________________________ > *>* FreeSWITCH-users mailing list > *>* FreeSWITCH-users at lists.freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> > *>* http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > *>* UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > *>* http://www.freeswitch.org > *>** > > -- Sincerely regards, Wen-Jen

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Re: [Freeswitch-users] Lua and database access to core_db
by Jon Bruel 09 Dec '09

09 Dec '09

I got the combination Lua with direct access to the core Sqlite database to work. Hurray, maybe I'm not as stupid as A.M II hints... The problem was that Lua did not "like": require "luasql.sqlite" env = luasql.sqlite() con = assert(env:connect("/usr/local/freeswitch/db/core.db")) After changing it to require "luasql.sqlite3" env = luasql.sqlite3() con = assert(env:connect("/usr/local/freeswitch/db/core.db")) And seeing that there was a symlink in one of the right directories called with the name: sqlite3.so, it worked. Changing the core db into a MySQL via ODBC caused some problems even after it seemed to work. For instance, console help caused an error with an error description indicating that a SQL SELECT query including the reserved word key has been fired. It this problem likely to be solved if I used another version of the MySQL? Jon Brüel Consiglia Telecommunications DK-2960 Rungsted Kyst Tel: +45 45 16 1000 Mob: +45 26 15 30 60 CVR: 27047882

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FXO PCI card with Pfsense Freeswitch.
by Orien Love 08 Dec '09

08 Dec '09

I am looking for a 4 port FXO card to use with my PfSense installation of freeswitch. does anybody know if the Asterisk Trixbox 4 FXO Voip Digium TDM410 will work on pfsense? or could somebody recommend one that would. Thank You Orien

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Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP
by Jerry Richards 08 Dec '09

08 Dec '09

Here is the Pastebin Link: http://pastebin.freeswitch.org/11432 Thanks, Jerry _____ From: Jerry Richards [mailto:jerry.richards@teotech.com] Sent: Tuesday, December 08, 2009 12:35 PM To: 'Michael Jerris'; 'freeswitch-users(a)lists.freeswitch.org' Subject: RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP Anthony and Michael, I downloaded the latest trunk, rebuilt it, and re-ran the test with the logs that Anthony told me to turn on. I put the results up in the PasteBin. Best Regards, Jerry _____ From: Jerry Richards [mailto:jerry.richards@teotech.com] Sent: Monday, December 07, 2009 10:49 AM To: 'Michael Jerris'; 'freeswitch-users(a)lists.freeswitch.org' Subject: RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP When I got the latest trunk the make gets an error. Should I perhaps disable the mod_amr? making all mod_amr make[5]: *** No rule to make target '/mod_amr.c', needed by 'mod_amr.so'. Stop The method I used to get the latest trunk follows: svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch Best Regards, Jerry _____ From: Jerry Richards [mailto:jerry.richards@teotech.com] Sent: Monday, December 07, 2009 7:44 AM To: 'Michael Jerris'; 'freeswitch-users(a)lists.freeswitch.org' Subject: RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP I am changing the 3pcc setting because one of my gateways sends INVITEs without SDP. I will try to update to the latest trunk today and capture traces as Anthony described. If I can't do it today, it might be at the end of the week. Best Regards, Jerry _____ From: Michael Jerris [mailto:mike@jerris.com] Sent: Saturday, December 05, 2009 7:30 PM To: Jerry Richards Subject: Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP Jerry- Any update on this? Mike On Dec 4, 2009, at 3:59 PM, Anthony Minessale wrote: Why are you changing the 3pcc setting, is this an invite with no sdp? you need to take a trace from FS. 1) update to latest trunk first so line number match up. 2) issue these commands sofia profile internal siptrace on console loglevel debug save the output and put it on pastebin http://pastebin.freeswitch.org <http://pastebin.freeswitch.org/> On Fri, Dec 4, 2009 at 2:47 PM, Jerry Richards <jerry.richards(a)teotech.com> wrote: I have Mediant 1000 gateway, and for some reason, when I make an outbound call, FS enters the CS_CONSUME_MEDIA state and never connects the call. A Wireshark trace shows that FS is replying to the gateway's inbound RTP packets with ICMP DESTINATION UNREACHABLE. But the gateway is sending RTP packets to the same port that FS specified in the outbound INVITE. It appears in the log that FS is discarding the 200 OK from the gateway. I disabled the Firewall and SELinux on the Freeswitch machine. I tried changing <enable-3pcc> to "true" and also "proxy", but it has no effect. Anyone know what could be the issue? I posted the Freeswitch log in the pastebin. Best Regards, Jerry _______________________________________________ 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 <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 <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net/> #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@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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Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP
by Jerry Richards 08 Dec '09

08 Dec '09

Anthony and Michael, I downloaded the latest trunk, rebuilt it, and re-ran the test with the logs that Anthony told me to turn on. I put the results up in the PasteBin. Best Regards, Jerry _____ From: Jerry Richards [mailto:jerry.richards@teotech.com] Sent: Monday, December 07, 2009 10:49 AM To: 'Michael Jerris'; 'freeswitch-users(a)lists.freeswitch.org' Subject: RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP When I got the latest trunk the make gets an error. Should I perhaps disable the mod_amr? making all mod_amr make[5]: *** No rule to make target '/mod_amr.c', needed by 'mod_amr.so'. Stop The method I used to get the latest trunk follows: svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch Best Regards, Jerry _____ From: Jerry Richards [mailto:jerry.richards@teotech.com] Sent: Monday, December 07, 2009 7:44 AM To: 'Michael Jerris'; 'freeswitch-users(a)lists.freeswitch.org' Subject: RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP I am changing the 3pcc setting because one of my gateways sends INVITEs without SDP. I will try to update to the latest trunk today and capture traces as Anthony described. If I can't do it today, it might be at the end of the week. Best Regards, Jerry _____ From: Michael Jerris [mailto:mike@jerris.com] Sent: Saturday, December 05, 2009 7:30 PM To: Jerry Richards Subject: Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP Jerry- Any update on this? Mike On Dec 4, 2009, at 3:59 PM, Anthony Minessale wrote: Why are you changing the 3pcc setting, is this an invite with no sdp? you need to take a trace from FS. 1) update to latest trunk first so line number match up. 2) issue these commands sofia profile internal siptrace on console loglevel debug save the output and put it on pastebin http://pastebin.freeswitch.org <http://pastebin.freeswitch.org/> On Fri, Dec 4, 2009 at 2:47 PM, Jerry Richards <jerry.richards(a)teotech.com> wrote: I have Mediant 1000 gateway, and for some reason, when I make an outbound call, FS enters the CS_CONSUME_MEDIA state and never connects the call. A Wireshark trace shows that FS is replying to the gateway's inbound RTP packets with ICMP DESTINATION UNREACHABLE. But the gateway is sending RTP packets to the same port that FS specified in the outbound INVITE. It appears in the log that FS is discarding the 200 OK from the gateway. I disabled the Firewall and SELinux on the Freeswitch machine. I tried changing <enable-3pcc> to "true" and also "proxy", but it has no effect. Anyone know what could be the issue? I posted the Freeswitch log in the pastebin. Best Regards, Jerry _______________________________________________ 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 <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 <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net/> #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@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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Re: [Freeswitch-users] Choppy sound with PCMU
by erandr-junk＠usa.net 08 Dec '09

08 Dec '09

Wow... Thinking about this timer setting and about how it converted send()/recv() from non-blocking to blocking, I straced freeswitch when it was supposed to be idle. It never pauses! It keeps going in and out of select() every millisecond! Why?? ------ Original Message ------ Received: Tue, 01 Dec 2009 08:31:46 PM EST From: erandr-junk(a)usa.net To: <freeswitch-users(a)lists.freeswitch.org> Subject: Re: [Freeswitch-users] Choppy sound with PCMU > 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 > > > > > > > _______________________________________________ > 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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