Re: [asterisk-users] Hangup extensions via CLI?
On Wed, 11 Feb 2009 12:34:12 +0100, Lenz Emilitri wrote:
This is a bit of trickery, but could not resist :)
This will kill a channel that is connected to SIP/201
asterisk -rx soft hangup $(asterisk -rx 'show channels' | grep SIP/201
| awk '{ print $1 '} )
what if there're also channels sip/201, sip/2011, sip/2012, sip/2013 et
al ?
--
Regards, /\_/\ All dogs go to heaven.
din...@alphaque.com(0 0) http://www.openmalaysiablog.com/
+==oOO--(_)--OOo==+
| for a in past present future; do|
| for b in clients employers associates relatives neighbours pets; do |
| echo The opinions here in no way reflect the opinions of my $a $b. |
| done; done |
+=+
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Set caller ID to anonymous
On Wed, 14 Jan 2009 16:09:02 +0100, philipp-chemn...@gmx.de wrote: setting the caller ID works perfect. Detecting if a caller is or isn't registered is the problem. I'm using sip. wouldnt ChanIsAvail() or regexten/regcontext settings in sip.conf assist in this ? -- Regards, /\_/\ All dogs go to heaven. din...@alphaque.com(0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Set caller ID to anonymous
On Wed, 14 Jan 2009 09:24:05 +0100, philipp-chemn...@gmx.de wrote: Hi guys, I am trying to set the caller ID to 'Anonymous anonymous' if the caller is not registered to the asterisk server. But I can't find a solution. put registered users in one context which dials out, and unregistered users in another which sets the callerid and then dials out. -- Regards, /\_/\ All dogs go to heaven. din...@alphaque.com(0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip clients for smart phones?
On Fri, 3 Oct 2008 12:00:16 -0800, Babcock, Michael Alex wrote: is it frig or fring? On Oct 3, 2008, at 11:49 AM, Tariq .. wrote: try using Frig.. it's a great client with an SIP client.. i tried it on IPhone and on my N82 Nokia phone.. it works great on GPRS and Wi- Fi... and i DO use it with my two Asterisk Servers.. regards only problem with fring is that it makes a connection to fring's servers, and then from there to your asterisk server. this results in a round-trip via the US for most of us here in malaysia. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bizarre international call problem.
On Fri, 26 Sep 2008 15:04:30 -0400 (EDT), Ken D'Ambrosio wrote: So I'm confused: any ideas on how this worked when the PBX was hooked straight to the PSTN? Is there some SS7 signal or something that says, This is an international call, when the number has no 011 preface? I'd hate to have to revert, but I will if need be... *sigh* the provider may be tagging it on. have you checked pridialplan, or prilocaldialplan settings and playing around with that in zapata.conf ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Astricon people please post the announcement
On Thu, 25 Sep 2008 18:00:00 +0100, Tim Panton wrote: It's essentially a channel driver. Licensed per channel in the same way that the g729 codec is. which would mean that us freebsd folks are going to be left out. oh well. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Do You Know What the Problem With CDMA is?
On Thu, 25 Sep 2008 12:21:41 -0400, Jason Aarons \(US\) wrote: A lot of places you still can't get GSM in the US.it has improved...but GSM 3G coverage is lacking compared to EVDO/CDMA. which isn't usually a problem as all 3G phones i've seen also use GSM, and the phones switch to GSM when 3G coverage isn't available. You start to explain about GSM and their eyes open wide as they realize they need a unlocked GSM phone from a electronics shop and SIM chip from actually, if you're using a gsm/3G phone, and your carrier has a roaming agreement with a malaysian carrier (there're 3 big ones and 1 small one, by the way), then it shouldnt matter. of course, they'll sock roaming charges on you. some company named Digi sold in 7-Eleven and some scratch off cards for refills using SMS. that's just one of the three, and its a prepaid gsm card you're referring to. you could've also picked up a celcom or a maxis prepaid card, or not worry about that and just roam with a gsm phone. In reality my roaming fees for Intl are too high, I'll get a pre-paid in-country phone before I get phone bill for Intl roaming. My data connection syncs email all day long. i hear the vodaphone 3G service hits you a fixed monthly fee for use anywhere in the world for a data/3G connection. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk GSM Gateway Project
On Sun, 22 Jun 2008 08:45:10 -0500, Michael Graves wrote: I have a small Portech GSM gateway. It works well. It's GSMSIP which seems to me a better solution than FXO/FXS type interfaces. They make gateways up to 32 port for E-1 interconnect. what did they cost, michael ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about SS7
On Wed, 14 May 2008 17:06:54 -0400, Alexander Lopez wrote: SS7 helps carriers maximize the use of the circuits that interconnect them with others. Instead of using a channel and having it open for 30 seconds as the call is setup, user gets signaling (busy, ringing, not in service), and call is torn down. It can get the result in a split second with out using any of my channels, all out of band and digital rather than analog, (see 2600 signaling) simplistically, ss7 is like sip which sets up the call, and the circuit itself is the rtp streams which are then built when the call is connected. likewise, you can have the sip exchange go through one path/route and rtp through another. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Disable transfer on all calls
On Tue, 22 Apr 2008 11:54:41 +0100, Grey Man wrote: The best option is to put a SIP Proxy in front of your Asterisk sever and block REFER requests. or just comment out the block in chan_sip.c which handles the refers. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] callers in queue passed to agents who accept only one call at a time
On Fri, 28 Mar 2008 06:33:42 -0700 (PDT), Vieri wrote: However, I can't use ringinuse=no in queues.conf because I'm running 1.2.27 (or is there a backport/patch?). iirc, there is a patch to backport ringinuse to 1.2.x. it's on mantis somewhere. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandstream BLF and Call-limit
On Fri, 28 Mar 2008 12:09:58 -0500, Peder @ NetworkOblivion wrote: Any idea? If I remove call-limit on the sip.conf entries, it all goes back to working fine. I tried 2, 9 and 99 on the call-limit and they all have the same issues. I can't imagine why call-limit causes hints to stop updating correctly. on the phone being monitored (sipA in this case), do an 'sip debug peer' and see the different notify messages sent to sipA. this would provide an indication, at the very least to developers if something in asterisk is broken. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] help with no audio
On Tue, 01 Apr 2008 13:32:28 -0400, Jared Smith wrote: On Tue, 2008-04-01 at 13:24 -0400, Jerry Geis wrote: I call into the dialplan and try to play demo-congrats and I hear nothing. Firewall is disabled. Everything is on the 192.168.1.X network for this simple configuration. The tftp server is giving the polycom phone the config files. Any ideas why I dont hear audio? Do you happen to have an unconfigured T1 card in your machine? That's the most common problem I see for people when they get no audio at all coming out of Asterisk. we've seen sites where just configuring the T1/E1 card alone is not enough, we'd need to plug the card with a loopback cable or connect it to a live E1 for rtp to work. any clues why this is the case ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] txfax not working with spandsp
On Fri, 21 Dec 2007 08:50:28 -0500, David Boyd wrote: On Fri, 2007-12-21 at 18:41 +0800, Dinesh Nair wrote: the attached log with verbose=6 and debug=6 refers. we've got a sangoma A104 (no hwec) with PRI ports 1 3 loopbacked to each other. we're trying to have txfax send out on one of those pri ports with rxfax listening on the other side, hence having asterisk send a fax to itself. we however have bad, and i mean really bad (10%) success rates. we're currently using asterisk 1.2.24 with spandsp 0.0.4-20071214 (snapshot of 14/12/07) and we keep getting Fax send not successful - result (25) No response after sending a page. errors. ECM is turned on in both app_txfax.c and app_rxfax.c. from what we gather just reading the code, time T4 expires in txfax because apparently rxfax is not sending a response back out, and thus after the maximum message retries (3) txfax just drops the call, leading rxfax to say that the call was dropped prematurely. does anyone know what's going on here, and if there is a version of spandsp which could work in this scenario ? timing issues on the PRI you mean ? none that we can see clearly. anything specific we need to investigate ? zttest returns 100% all of the time. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] txfax not working with spandsp
the attached log with verbose=6 and debug=6 refers. we've got a sangoma A104 (no hwec) with PRI ports 1 3 loopbacked to each other. we're trying to have txfax send out on one of those pri ports with rxfax listening on the other side, hence having asterisk send a fax to itself. we however have bad, and i mean really bad (10%) success rates. we're currently using asterisk 1.2.24 with spandsp 0.0.4-20071214 (snapshot of 14/12/07) and we keep getting Fax send not successful - result (25) No response after sending a page. errors. ECM is turned on in both app_txfax.c and app_rxfax.c. from what we gather just reading the code, time T4 expires in txfax because apparently rxfax is not sending a response back out, and thus after the maximum message retries (3) txfax just drops the call, leading rxfax to say that the call was dropped prematurely. does anyone know what's going on here, and if there is a version of spandsp which could work in this scenario ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ Dec 21 18:32:05 VERBOSE[205] logger.c: -- Attempting call on Zap/g1/1002 for [EMAIL PROTECTED]:1 (Retry 1) Dec 21 18:32:05 DEBUG[205] chan_zap.c: Using channel 1 Dec 21 18:32:05 VERBOSE[205] logger.c: -- Requested transfer capability: 0x00 - SPEECH Dec 21 18:32:05 DEBUG[205] devicestate.c: Changing state for Zap/1 - state 2 (In use) Dec 21 18:32:05 DEBUG[205] devicestate.c: Changing state for Zap/1 - state 2 (In use) Dec 21 18:32:05 DEBUG[205] app_queue.c: Device 'Zap/1' changed to state '2' (In use) but we don't care because they're not a member of any queue. Dec 21 18:32:05 DEBUG[205] app_queue.c: Device 'Zap/1' changed to state '2' (In use) but we don't care because they're not a member of any queue. Dec 21 18:32:05 VERBOSE[205] logger.c: -- Accepting call from '' to '1002' on channel 0/1, span 2 Dec 21 18:32:05 DEBUG[205] chan_zap.c: No echo cancellation requested Dec 21 18:32:05 DEBUG[205] devicestate.c: Changing state for Zap/32 - state 2 (In use) Dec 21 18:32:05 DEBUG[205] pbx.c: Launching 'Answer' Dec 21 18:32:05 VERBOSE[205] logger.c: -- Executing Answer(Zap/32-1, ) in new stack Dec 21 18:32:05 DEBUG[205] pbx.c: Launching 'Wait' Dec 21 18:32:05 VERBOSE[205] logger.c: -- Executing Wait(Zap/32-1, ) in new stack Dec 21 18:32:05 DEBUG[205] pbx.c: Launching 'Set' Dec 21 18:32:05 VERBOSE[205] logger.c: -- Executing Set(Zap/32-1, FAXFILE=/tmp/FAX-1198233125.1.tiff) in new stack Dec 21 18:32:05 DEBUG[205] pbx.c: Function result is '1002' Dec 21 18:32:05 DEBUG[205] pbx.c: Launching 'Set' Dec 21 18:32:05 VERBOSE[205] logger.c: -- Executing Set(Zap/32-1, NEWFAXFILE=/var/spool/asterisk/fax/FAX-1002--11982331251198233125.1.tiff) in new stack Dec 21 18:32:05 DEBUG[205] pbx.c: Launching 'RxFAX' Dec 21 18:32:05 VERBOSE[205] logger.c: -- Executing RxFAX(Zap/32-1, /tmp/FAX-1198233125.1.tiff|debug) in new stack Dec 21 18:32:05 DEBUG[205] app_queue.c: Device 'Zap/32' changed to state '2' (In use) but we don't care because they're not a member of any queue. Dec 21 18:32:05 DEBUG[205] devicestate.c: Changing state for Zap/32 - state 2 (In use) Dec 21 18:32:05 DEBUG[205] channel.c: Set channel Zap/32-1 to read format slin Dec 21 18:32:05 DEBUG[205] channel.c: Set channel Zap/32-1 to write format slin Dec 21 18:32:05 DEBUG[205] app_queue.c: Device 'Zap/32' changed to state '2' (In use) but we don't care because they're not a member of any queue. Dec 21 18:32:05 DEBUG[205] chan_zap.c: Queuing frame from PRI_EVENT_PROCEEDING on channel 0/1 span 1 Dec 21 18:32:05 DEBUG[205] chan_zap.c: No echo cancellation requested Dec 21 18:32:05 VERBOSE[205] logger.c: Channel Zap/1-1 was answered. Dec 21 18:32:05 DEBUG[205] pbx.c: Launching 'Answer' Dec 21 18:32:05 VERBOSE[205] logger.c: -- Executing Answer(Zap/1-1, ) in new stack Dec 21 18:32:05 DEBUG[205] pbx.c: Launching 'Set' Dec 21 18:32:05 VERBOSE[205] logger.c: -- Executing Set(Zap/1-1, CDR(userfield)=FAX-1) in new stack Dec 21 18:32:05 DEBUG[205] pbx.c: Launching 'TxFAX' Dec 21 18:32:05 VERBOSE[205] logger.c: -- Executing TxFAX(Zap/1-1, /var/spool/asterisk/outgoing_fax/page.1.1.tiff|caller|debug) in new stack Dec 21 18:32:05 DEBUG[205] channel.c: Set channel Zap/1-1 to read format slin Dec 21 18:32:05 DEBUG[205] channel.c: Set channel Zap/1-1 to write format slin Dec 21 18:32:05 DEBUG[205] devicestate.c: Changing state for Zap/1 - state 2 (In use) Dec 21 18:32:05
Re: [asterisk-users] Fax on asterisk
On Tue, 4 Dec 2007 20:45:08 -0500 (EST), Alex Balashov wrote: This sounds like the app_rxfax module has a dependency on some other module which implements T.30 handling, and that this module is either not loaded, or that its symbol table is not being shared in the monolithic core. there's actually a mismatch in your spandsp library, i believe you're using an older one. download 0.0.4pre15, which seems to work well these days. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Cisco calling Name
On Sat, 1 Dec 2007 00:42:43 -0500, John Bittner wrote: Anyone see an issue on asterisk 1.2 that it will not accept the invite from a Cisco gateway. If I turn off voice service voip signaling are you sure you've got ulaw enabled for that peer in sip.conf ? and the invite trace shows that the cisco is not sending any cname. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind a PIX firewall?
On Tue, 27 Nov 2007 09:40:56 -0500, Matt wrote: This is a dual NAT situation. PIX on Asterisk side, and Netgear on phone side. HOWEVER.The Asterisk box has it's own IP but it is being tunneled through the PIX.I guess the PIX must be messing something up? could you post a 'sip debug peer ' of the call ? depending on your setup, you may need to set externip in sip.conf to the external ip addy of the pix firewall, so the addresses placed in the SIP packets are correct. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Opinions on Release Numbering
On Wed, 10 Oct 2007 12:54:42 -0500, Russell Bryant wrote: I proposed calling the release candidates 1.6.3-rc1, 1.6.3-rc2, etc. Another proposal has been using 1.5 to indicate that it is a release candidate. For example, 1.5.3, 1.5.3.1, 1.5.3.2, etc., would be the release candidates for the upcoming 1.6.3 release. the former is more obvious than the latter. i kind of like asterisk's release numbering mechanism where the even numbered dot releases are stable/production while the odd numbered ones are for development. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] online active call watching
On Mon, 10 Sep 2007 13:43:46 -0800, Mojo with Horan Company, LLC wrote: Though still in the proof-of-concept stage, my project AstSee from http://www.astsee.com/ might be fun to play with if you're using linux/XWindows. There are screenshots there. that may be so, but without source, there's no way we can test it on freebsd. i'll stick with fop for the timebeing, thank you. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax Throughput
On Wed, 27 Jun 2007 09:08:21 -0500, Matthew Fredrickson wrote: You fixed your clocking then. That was what I was thinking of. Make sure that your Dialogic card is also pulling timing from the Digium card as well. What version of zaptel drivers are you running? on a related issue, using asterisk 1.2.21 and spandsp 0.0.4 as well as the relevant rxfax and txfax, a loopback fax over an E1 PRI always goes thru at 9600bps. is there a way to increase this, or is it due to the line itself ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe + IAX2 + Asterisk 1.2.18 = calls dropped
On Thu, 26 Apr 2007 12:39:32 -0400, Dave Miller wrote: Dave Miller wrote on 4/26/07 11:46 AM: We upgraded our asterisk server to 1.2.18 last night to pick up the security update. Since then, any calls coming in on IAX2 links get dropped if they try to enter a MeetMe conference room. The log shows this: Apr 26 08:33:16 NOTICE[27362]: chan_iax2.c:3167 iax2_read: I should never be called! Hanging up. I've temporarily worked around it by switching our inbound provider to use SIP instead of IAX, but that's not an ideal solution. Quick turnaround on the bug tracker, bug is resolved fixed already :) http://bugs.digium.com/view.php?id=9600 guess that'll be fixed in the next release. is there a patch for this against 1.2.18 ? it would sure help those who're tracking the release tarballs instead of having to svn and compile it. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP failover between Sip Providers
On Wed, 18 Apr 2007 09:04:22 +0200, Knud Müller wrote: I think it can be done by using the dialplan and the database to store the statistical information but maybe there is an easier way that integrates better with asterisk!? i dont think you'd even need a database with statistics. just have all calls sent to provider A with an automatic failover to provider B if the call can't be completed through A. you'd need to go look at the DIALSTATUS variable for that. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] Nokia E60 firmware update break SIP
On Mon, 16 Apr 2007 20:14:40 -0700, Martin Joseph wrote: The phone no longer registers with asterisk, although it displays the little icon as though it has, and it doesn't even seem to try to pass calls to asterisk... So, I would avoid 3.06330904 20-11-06 RM-49 i've got an E61 running the same firmware revision and it works fine and dandy with asterisk 1.2.17. one thing you may want to do is to delete all your SIP profiles in the phone and reconfigure it from scratch. upgrading firmware from 2.x to 3.x broke something which wasnt forward compatible. we had similar issues, but deleting all profiles and reconfiguring from scratch fixed it. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TM Malaysia E1 PRI signaling
On Tue, 17 Apr 2007 20:55:44 -0400, Jason Aarons \(US\) wrote: Anyone configured a E1 PRI in Kuala Lampur, Malaysia with TM Malaysia? What signaling did they provide, framing, formatting? we have many times for our customers. E1 EuroISDN with CCS, HDB3, CRC4. works great out of the box. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] injecting audio announcements into sip channel
On Mon, 16 Apr 2007 10:48:40 +0100, Mark Reardon wrote: 1) But how do I inject them into the SIP channel. 2) How do I time the injection so that the correct message is played at the correct time. take a look at the L() option to Dial(). -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] To use asterisk or proprietary hardware, that is the question
On 02/25/07 22:16 Doug Lytle said the following: zaptel-base.c. This prevents the Sangoma Setup script from patching zaptel. The fix (Found by Googling) was to rename every instance of ok, the sangoma scripts on freebsd do not patch the zaptel-bsd source in any way, so this shouldn't affect those on *BSD. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] To use asterisk or proprietary hardware, that is the question
On 02/25/07 22:16 Doug Lytle said the following: My experience from yesterday shows that zaptel.c has been renamed to zaptel-base.c. This prevents the Sangoma Setup script from patching zaptel. The fix (Found by Googling) was to rename every instance of ok, the sangoma scripts on freebsd do not patch the zaptel-bsd source in any way, so this shouldn't affect those on *BSD. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] To use asterisk or proprietary hardware, that is the question
On 02/25/07 06:26 Darrick Hartman said the following: Kristian is working with Sangoma to get wanpipe supported once again in Asterisk. is there a reason why wanpipe stopped working with asterisk ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP 406 error - cause?
On 02/22/07 06:04 Michelle Dupuis said the following: I'm working on calls coming in to an asterisk box as H.323, and going out as SIP to a remote device (a VoiceMaster). The remote device is refusing the calls with SIP error 406 (Not Acceptable). I have attached the SIP debug output below. It looks like codecs overlaps - can anyone see why the call is being refused? 406s are usually returned because there're no common codecs for the call. check the codecs available on the voicemaster. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Timeout in IAX vs SIP
On 02/01/07 02:15 Olle E Johansson said the following: both channels should act the same unless there's a configuration that's giving wrong information to chan_sip, like you having a username= or defaultip= setting. how does a username= entry in sip.conf affect dialling behaviour when the phone is not registered ? by default as a matter of practice, we have username=something for our peers, though they may be on dynamic IP addresses and register with asterisk. is what we're doing a Bad Thing(tm) ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transfer on RTP timeout?
On 01/28/07 18:52 Florian Overkamp said the following: Nokia seems to have done something like this in their E-series (E60 etc) with Avaya and Cisco. Anyone have a lowdown on the technical stuff there ? i think that's a FMC (fixed mobile convergence) client which both avaya and cisco wrote for the E series platform. my stock E61 doesn't have such a client, though it has the SIP 2.0 symbian client. as for the original poster, what you can probably do is to trap the hangup, and perhaps modify app_dial.c to set the hangup cause in DIALSTATUS for RTP timeouts, then take appropriate redialling action as part of the h extension. do note that this is off the cuff, and i'm not sure how difficult it'd be to do this. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Goto not jumping to current context
in a simple dialplan like follows:
[firstcontext]
include = secondcontext
include = thirdcontext
include = fourthcontext
[fourthcontext]
_03X.,1,Goto(${EXTEN:2},1)
_X.,1,DoSomething()
_X.,2,Hangup()
the Goto() for exten _03X. seems to start the search for the jump within
firstcontext, thus possibly matching an exten in secondcontext or
thirdcontext first before hitting the matchall in fourthcontext. obviously,
a simple fix would be to change it to Goto(fourthcontext,${EXTEN:2},1).
however, i dont remember Goto working this way. shouldn't a Goto search
within the current context first when the context parameter is ommitted ?
it's asterisk 1.2.14 in FreeBSD 6.1 though.
--
Regards, /\_/\ All dogs go to heaven.
[EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/
+==oOO--(_)--OOo==+
| for a in past present future; do|
| for b in clients employers associates relatives neighbours pets; do |
| echo The opinions here in no way reflect the opinions of my $a $b. |
| done; done |
+=+
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How big a pipe can IAX2 go?
On 01/05/07 06:18 Zoa said the following: It used to be a problem to have very big iax2 trunks (e.g. 100 channels). anyone remember why this was so, and if a bug was opened on this for 1.2 ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] one way rtp stream (Sent alwax to 127.0.0.1)
On 12/29/06 06:04 Hans-Jürgen Brand said the following: Found problem xlite client Version 3 Build 34025 send wrong rtp port to asterisk. But I don't know how to change this at xlite have you tried nat=yes in sip.conf for the peer ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How accurate is show translation?
On 12/23/06 09:51 Leo Ann Boon said the following: I would love to hear how others are using the results from show translation in system dimensioning. So far, I feel that dimensioning an Asterisk box is still mostly guesstimation :). Currently, I'm using the 30MHz per call rule to dimension. on a Pentium D 2.80Ghz, we've sustained 300 simultaneous IAX2 calls terminating in a dialplan loop that answers the call, waits 2 seconds, plays demo-instruct and loops again. a cursory examination revealed that a large portion of the CPU was used to handle NIC interrupts. occasionally we got a chan_iax2.so error which said, Maximum trunk data space exceeded to... this seems to be controlled by the MAX_TRUNKDATA constant in chan_iax2.c which is set to 40ms of SLIN for 200 calls. it'd be nice to know what this constant is for and what would the implications of increasing it be. [cc'ed to -dev as well] -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Page() Function Timeout
On 11/16/06 06:06 David Gagnon said the following: Which version are you using? There was a problem in 1.2.12.1 with the page application. Update to 1.2.13. what was the problem ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.12.1 dies after asterisk -rx and never comes up again
On 10/11/06 21:15 Joseph said the following: I quits on my as well, when I try to make a second call. There is a bug report on it: http://bugs.digium.com/view.php?id=7972 this seems like a configuration error within FreePBX and isnt really a bug in asterisk. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 488 Not acceptable here sent by Asterisk - SIPdebug follows
On 09/20/06 15:06 Dinesh Nair said the following: On 09/19/06 16:59 Steve Langstaff said the following: I wonder whether you are experiencing the following bug (since the SIP INVITE has a multipart SDP body): http://bugs.digium.com/view.php?id=7124nbn=4 thanks for the link, however, on 18th may 2006, kpfleming's note says, This should be fixed in both 1.2 branch and trunk, and i'm using 1.2.12.1 which was just released this week. looking thru the current chan_sip.c code, it does seem like kevin's modified patch has been committed into the branch i'm using, so this isnt the problem. [am cc'ing reply into -dev because a bug report was opened on this at http://bugs.digium.com/view.php?id=8010 with a patch provided] i've managed to track this down to a loop which terminated prematurely in find_sdp() in chan_sip.c. this bug would have prevented proper handling of multipart/mixed content types due to the loop which searches for the end of the block ending prematurely and setting req-sdp_start req-sdp_end. i've provided patches for trunk and 1.2.x in the bug entry, as i think this should also be committed to 1.2.x. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 488 Not acceptable here sent by Asterisk - SIPdebug follows
On 09/19/06 16:59 Steve Langstaff said the following: I wonder whether you are experiencing the following bug (since the SIP INVITE has a multipart SDP body): http://bugs.digium.com/view.php?id=7124nbn=4 thanks for the link, however, on 18th may 2006, kpfleming's note says, This should be fixed in both 1.2 branch and trunk, and i'm using 1.2.12.1 which was just released this week. looking thru the current chan_sip.c code, it does seem like kevin's modified patch has been committed into the branch i'm using, so this isnt the problem. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 488 Not acceptable here sent by Asterisk - SIP debug follows
the situation Asterisk -- SIP --- SIPGW --- SIP Phone SIP Phone is trying to call asterisk dialplan: exten = 0224577501,1,Answer() exten = 0224577501,2,Playback(demo-instruct) exten = 0224577501,3,Hangup() however, asterisk 1.2.12.1 (on FreeBSD 6.1) sends back a 488 Not acceptable here with a CLI message of WARNING[162]: chan_sip.c:3529 process_sdp: Insufficient information for SDP (m = '', c = '') it seems to be dropping out in process_sdp() because it can't find the m= or the c=. this is a little odd, so am wondering if this has triggered some edge case in find_sdp(), get_sdp() or get_sdp_iterate(). i've been poring thru the code (as the box is remote, and i cant duplicate it locally), but can't find exactly where in chan_sip.c its borking. any advice would be much appreciated. the SIP debug is attached below: (10.14.32.179 is the SIPGW, 10.14.32.164 is Asterisk) begin sip debug -- SIP read from 10.14.32.179:5060: INVITE sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.14.32.179:5060 Via: SIP/2.0/UDP 10.14.32.189:5060 Record-Route: sip:10.14.32.179:5060 Supported: replaces User-Agent: SIP201 (lp201_sip0423.bin) Contact: sip:[EMAIL PROTECTED]:5060 From: sip:[EMAIL PROTECTED]:5060 ;tag=a0e20bd-13c4-132e6-4aed8ab-2ea8 To: sip:[EMAIL PROTECTED]:5060;user=phone Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE History-Info: sip:[EMAIL PROTECTED]:5060;index 1 Content-Type: multipart/mixed;boundary=unique-boundary Content-Length: 474 --unique-boundary Content-Type: application/sdp v=0 o=SIP201 12367 0 IN IP4 10.14.32.189 s=SIP201 Session i=Audio Session c=IN IP4 10.14.32.189 t=0 0 m=audio 16384 RTP/AVP 4 18 0 8 18 a=rtpmap:4 G723/8000/1 a=rtpmap:18 G729/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:18 G729/8000/1 --unique-boundary Content-Type: application/isup;version=Indonesia Content-Transfer-Encoding: binary --- (14 headers 21 lines)--- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 10.14.32.179 : 5060 (non-NAT) Found peer 'RISTI' Sep 19 09:38:53 WARNING[162]: chan_sip.c:3529 process_sdp: Insufficient information for SDP (m = '', c = '') Transmitting (no NAT) to 10.14.32.179:5060: SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 10.14.32.179:5060;received=10.14.32.179 Via: SIP/2.0/UDP 10.14.32.189:5060 From: sip:[EMAIL PROTECTED]:5060 ;tag=a0e20bd-13c4-132e6-4aed8ab-2ea8 To: sip:[EMAIL PROTECTED]:5060;user=phone;tag=as5a7aa73d Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: QubeTalk ECS Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- Destroying call '[EMAIL PROTECTED]' suria*CLI -- SIP read from 10.14.32.179:5060: ACK sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.14.32.179:5060 Via: SIP/2.0/UDP 10.14.32.189:5060 Record-Route: sip:10.14.32.179:5060 Contact: sip:[EMAIL PROTECTED]:5060 User-Agent: SIP201 (lp201_sip0423.bin) From: sip:[EMAIL PROTECTED]:5060 ;tag=a0e20bd-13c4-132e6-4aed8ab-2ea8 To: sip:[EMAIL PROTECTED]:5060;user=phone ;tag=as5a7aa73d Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK Content-Length:0 --- (11 headers 0 lines)--- Destroying call '[EMAIL PROTECTED]' end sip debug -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Makefile.moddir_rules: No such file or directory
On 09/13/06 07:22 Ronald Wiplinger said the following: I need h.264 and tried therefore svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk asterisk supports h.264 in passthru mode. we've tested this with eyebeam video SIP clients without problems. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] modifying the INVITE headers
On 09/11/06 18:36 Paco Brufal said the following: Hello, Here in Spain there is a VoIP provider (Telefonica) that only works if when you make an outgoing call, the SIP headers are like this: INVITE sip:phonenumber@telefonica.net SIP/2.0 But Asterisk is sending this: INVITE sip:phonenumber@sbc.ngn.rima-tde.net SIP/2.0 because sbc.ngn.rima-tde.net is the register host. My config is this: try adding fromdomain=telefonica.net in the config for that peer. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Max Time
On 08/27/06 13:23 Rushowr said the following: Set(TIMEOUT(absolute)=seconds) Change seconds to the number of seconds you want to allow a call to last alternatively, look at the L() option to Dial. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Nobody is responding. Why? (Implement music on transfer)
On 08/26/06 23:52 Crazy Boy said the following: Hi friends, I did music on hold. How can we implement music on call transfer? I am unable to find any tutorial about setting up music on call transfer, i'm not exactly sure what you're intending to do, but MoH is already active and played for attended transfers. blind transfers will relay the call indication tones. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Nokia E60/61/70 and SIP
On 08/24/06 09:02 El Flynn said the following: Just wondering -- has anyone used the SIP phone feature on the Nokia E60/61/70 phones? We're trying to see if this would be an OK phone to get for the company, particularly since we're already running Asterisk. SIP works well with asterisk, with some caveats: 1. you need qualify set as the wifi radio on the phone sucks big oranges 2. the phone routinely loses IP connectivity, leading to reg failures 3. when two simultaneous calls, GSM and SIP, come in the phone hangs more often than not 4. be prepared to reboot constantly for simple config changes on the phone. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] linuxdevices.com: Trolltech woos developers with open Linux phone Who'll be the first with * on a mobile?
On 08/16/06 23:35 Robert Michel said the following: I think the BCM chip is for the GSM stuff, for GUI and applications the XScale chip - so for running asterisk, the XScale will be the processor. why would you want to run asterisk on the phone ? ideally, it should be running a softphone and connecting back over WiFi or 3G (HSPDA ??) to an asterisk installation. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending SIP 183 Session Progressing
On 08/17/06 14:56 Olle E Johansson said the following: Don't do it within chan_sip, do it within the dialplan by using playback with the no answer option before you dial out... yes, that will force early media and cause sip_write() to force send a 183. thanx, this should work. i'll test it out and report back. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sending SIP 183 Session Progressing
i'm not sure if this is a -users or a -dev question, but am sending it here anyways. discussion could move to -dev if chan_sip.c code needs to be amended/explained. first up, all this on asterisk 1.2.10 on freebsd 6.1. here's the beef: from a particular sip softphone we're playing with, we notice that calls to another SIP phone (same LAN) result in the /lack/ of a ringing tone on the softphone. however, calls from the same softphone to a PSTN/Mobile number (through a TE405P) result in proper behaviour on the softphone with a ringing tone. an ethereal trace of both types of calls results in only one difference. for calls to the PSTN/Mobile thru libpri/chan_zap, asterisk returns a SIP 183 Session Progress[1] packet in between the 100 Trying and 180 Ringing, while for calls from the softphone to another SIP phone it's 100 Trying followed immediately by 180 Ringing. so my question is, is the softphone behaving correctly in not playing a ringing tone to the user without the 183 packet inspite of the 180 Ringing packet being received ? alternatively, since we aren't able to change the softphone, will i break anything big if i force asterisk to send the 183 packet immediately after sending the 100 Trying packet in sip_indicate() ? alternatively, in reading the RFCs, i came across RFC3398 which speficies mappings between ISDN Cause Codes and SIP responses. has this mapping been implemented in asterisk at the moment, either in 1.2 or the upcoming 1.4 ? [1] the 183 Session Progress packet is triggered by the receipt of a PRI PROGRESS indicator from libpri, which gets translated to a AST_CONTROL_PROGRESS and thence a 183 Session Progress to SIP. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending SIP 183 Session Progressing
On 08/15/06 23:30 Michael J. Tubby B.Sc (Hons) G8TIC said the following: I suspect your problem is with the softphone implementation... definitely, the SIP spec iianm says that UACs should play a ringing tone when the 180 is received. Occasionally calls which go from 100 - 180 without going via the 183 result in the Cisco ringing and combined rining genrated by the telephone exchange which is weird but ok. the supplementary question then is, since i can't change the softphone would i break anything if i forced the sending of the 183 packet anyways from within chan_sip ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoipNow 1.2.0 Beta
On 07/29/06 02:49 Miles Scruggs said the following: http://forum.4psa.com/showthread.php?t=455 Take it for a ride around the block and tell them what you think. As powerful as the config files, and command line interface is, there is is there anywhere we can take a look at screenshots without having to download the entire package ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue announcement issues
On 07/27/06 03:28 Phil Jordan said the following: Jul 26 20:05:22 DEBUG[16371] channel.c: Scheduling timer at 160 sample intervals Jul 26 20:05:22 DEBUG[16371] channel.c: Avoiding initial deadlock for 'IAX2/phil-5' Jul 26 20:05:22 VERBOSE[16371] logger.c: -- Called IAX2/phil Jul 26 20:05:22 DEBUG[16371] channel.c: Generator got voice, switching to phase locked mode Jul 26 20:05:22 DEBUG[16371] channel.c: Scheduling timer at 0 sample intervals Jul 26 20:05:22 VERBOSE[16371] logger.c: -- Call accepted by 82.11.45.110 (format gsm) Jul 26 20:05:22 VERBOSE[16371] logger.c: -- Format for call is (gsm) Jul 26 20:05:22 VERBOSE[16371] logger.c: -- IAX2/phil-5 is ringing Jul 26 20:05:56 DEBUG[16371] chan_sip.c: Stopping retransmission on it does seem that IAX2/phil is still logged in as an agent of the queue, thus the caller is delivered to that agent and no hold times, position or periodic announcements are made. what does 'show queue hasbean' and 'show agents' say ? this may be the case because you have persistentagents=yes in queues.conf. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue announcement issues
On 07/26/06 14:58 Phil Jordan said the following: Before I get round to posting my configs for critique, is this a BSD port issue? I see stuff around on the net re the BSD port, to the no, it isn't a BSD port issue. many people run asterisk from ports with ACDs without any problems. in your situation, you'd probably need to provide more information (CLI verbose output, for starters) before someone can give you a more accurate solution. effect that there are some issues with Asterisk applications which are related to timers. What exactly is meant by that please? Is that what the zaptel-bsd drivers have the ztdummy timer and they're in ports and subversion. look for zaptel-bsd in the wiki. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: SV: [Asterisk-Users] Nokia E61
On 07/18/06 04:03 Fredrik Emil Jensen said the following: the packet too, but when the firewall/router loses its table (usually it will timeout after xx sec/min) you will only be able to dial outgoing can't you use qualify to get the nat device to keep the mapping ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] B2BUA Webbased and Click 2 dial apps
Hi Marnus, That is a good idea, I didnt think of thatJ thanks Dinesh. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marnus van Niekerk Sent: Thursday, July 06, 2006 4:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] B2BUA Webbased and Click 2 dial apps Also have a look at .call files. You web app can just create a .call file and then move it to the right location and asterisk will place the call No manager interface needed. Marnus van Niekerk Opportunity is missed by most people because it isdressed in overalls and looks like work.Thomas Alva Edison - Inventor of 1093 patents,including the light bulb, phonogram and motion pictures. Dinesh wrote: Hello, I have a requirement of bridging 2 sip connections via asterisk, which has to be web based. A person has to go to a webpage and enter his from sip uri(say sip1) and enter another sip uri(say sip2). Upon pressing the connect button, the webpage needs to send say a dial sip1 uri and dial dip uri 2 and bridge the call? Do I need any special sip api for this? Any ideas will be niceJ. Does this webpage has to be on asterisk server running on the machine? Or can it be passed as a string to the server from the webserver? Regards, Dinesh Birlasekaran Network Engineer, ComIT, Institute of Molecular and Cell Biology 61 Biopolis Drive, Singapore 138673 HP : 92962676 DID : 65869804 Fax : 67791117 Email : [EMAIL PROTECTED] WWW: www.imcb.a-star.edu.sg ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] B2BUA Webbased and Click 2 dial apps
I am not for the billing part, as its sip based, and its educational calls only. I mean between sip.edu community and my educational institute. So practically any sip uri should be able to be dialed from the website. I dunno I am just asking the ideas for the group. Regards, Dinesh. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marco Mouta Sent: Friday, July 07, 2006 5:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] B2BUA Webbased and Click 2 dial apps It would be hard to bill all this calls, if you are using dialout call files instead of Asterisk Manager API no ? How would you colect the call duraction of both call legs? Thks, Marco Mouta On 7/6/06, Marnus van Niekerk [EMAIL PROTECTED] wrote: Also have a look at .call files. You web app can just create a .call file and then move it to the right location and asterisk will place the call No manager interface needed. Marnus van Niekerk Opportunity is missed by most people because it is dressed in overalls and looks like work. Thomas Alva Edison - Inventor of 1093 patents, including the light bulb, phonogram and motion pictures. Dinesh wrote: Hello, I have a requirement of bridging 2 sip connections via asterisk, which has to be web based. A person has to go to a webpage and enter his from sip uri(say sip1) and enter another sip uri(say sip2). Upon pressing the connect button, the webpage needs to send say a dial sip1 uri and dial dip uri 2 and bridge the call? Do I need any special sip api for this? Any ideas will be niceJ. Does this webpage has to be on asterisk server running on the machine? Or can it be passed as a string to the server from the webserver? Regards, Dinesh Birlasekaran Network Engineer, ComIT, Institute of Molecular and Cell Biology 61 Biopolis Drive, Singapore 138673 HP : 92962676 DID : 65869804 Fax : 67791117 Email : [EMAIL PROTECTED] WWW: www.imcb.a-star.edu.sg ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Com os melhores cumprimentos, Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] B2BUA Webbased and Click 2 dial apps
Hello, I have a requirement of bridging 2 sip connections via asterisk, which has to be web based. A person has to go to a webpage and enter his from sip uri(say sip1) and enter another sip uri(say sip2). Upon pressing the connect button, the webpage needs to send say a dial sip1 uri and dial dip uri 2 and bridge the call? Do I need any special sip api for this? Any ideas will be niceJ. Does this webpage has to be on asterisk server running on the machine? Or can it be passed as a string to the server from the webserver? Regards, Dinesh Birlasekaran Network Engineer, ComIT, Institute of Molecular and Cell Biology 61 Biopolis Drive, Singapore 138673 HP : 92962676 DID : 65869804 Fax : 67791117 Email : [EMAIL PROTECTED] WWW: www.imcb.a-star.edu.sg ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WebPhone
On 07/04/06 00:16 Jean-Denis Girard said the following: It should be working. What happens exactly: is this an installation problem, or what ? Can you try running Firefox from an xterm, there should be some messages, eg. it dies with FATAL ERROR: No connection to network client in a popup window. the Debug window shows a whole bunch of Socket not alive messages. the commandline start of -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WebPhone
On 07/03/06 12:51 Tzafrir Cohen said the following: Web pages, evenwith javascript, are still very limited. For instance, they cannot establish UDP communication on their own with other places. An arbitrary TCP connection is also not so trivial. presently yes, however this will soon change as browsers open up more of their API as they evolve to become containers for applications written in javascript. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WebPhone
On 07/03/06 15:41 Jean-Denis Girard said the following: MozPhone no longer depends on any external libraries (libiaxclient is statically compiled in, and jslib is now included). So install is very simple, like any other firefox extension. It is correct that newer cant seem to get it to work on Mozilla/5.0 (X11; U; FreeBSD i386; en-US; rv:1.8) Gecko/20051228 Firefox/1.5. any chance this is on the radar ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WebPhone
On 06/29/06 04:41 Forrest Beck said the following: Here is a firefox plugin that connects to asterisk via IAX protocol. http://moziax.mozdev.org/ works only on windows, right ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WebPhone
On 06/29/06 05:17 Tzafrir Cohen said the following: But it's not a web phone by any means. Writing a soft phone in HTML and javascript is practically impossible. with the amount of interest in AJAX, DHTML and the much hyped Web 2.0, this may soon be a possibility as the browsers open up more of their container API. Google Desktop, anyone ? :) -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Work required - modify Asterisk + SEMS
On 06/29/06 01:18 Jeremy McNamara said the following: why not setup a listen only meetme for the 'listeners' and talk only for the 'talker'? isnt the Page() application used for stuff like this ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Gizmo and Asterisk analysis
On 06/25/06 19:01 Roy Sigurd Karlsbakk said the following: seem to pass all SIP and RTP traffic through their own servers... See http://karlsbakk.net/asterisk/gizmo-project.php for details interesting. but isnt Gizmo an open source client ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI Issue - Calls being rejected with unacceptable channel
On 06/23/06 01:22 Andy Brezinsky said the following: Protocol Discriminator: Q.931 (8) len=47 Call Ref: len= 2 (reference 15996/0x3E7C) (Originator) Message type: SETUP (5) Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 48764/0xBE7C) (Terminator) Message type: CALL PROCEEDING (2) Protocol Discriminator: Q.931 (8) len=14 Call Ref: len= 2 (reference 48764/0xBE7C) (Terminator) Message type: CONNECT (7) i may be way offbase with this, but on our PRI calls, we usually have asterisk sending an ALERTING between the CALL PROCEEDING and CONNECT. this seems borne out by... Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 15996/0x3E7C) (Originator) Message type: RELEASE (77) Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2 (reference 15996/0x3E7C) (Originator) Message type: STATUS (125) [08 03 83 e5 07] Cause (len= 5) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Transit network (3) Ext: 1 Cause: Message not compatible with call state (101), class = Protocol Error (6) ] ...Message not compatible with call state STATUS returned by the other side. you may want to experiment with a Wait(2) before the Answer(). -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hitting * in a queue call hangs up?
On 06/20/06 18:20 Matt said the following: It seems 1.2.9.1 does not correct this behavior... can I correct it somehow? matt, i believe i've already sent this to the list. the bug at http://bugs.digium.com/view.php?id=6897 has the fix for 1.2.x as agent-endcall.patch. apply that, and hitting '*' during a queue call wont hang up the call. to hangup the call you'd then need to use whatever was defined for disconnect in features.conf. also not that endcall=no in agents.conf. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 Vs SIP cpu load
On 06/13/06 22:49 Colin Anderson said the following: Although this may have changed in the newer 1.2.X series of Asterisk, I believe that Asterisk does not support SMP from the perspective of isnt asterisk multithreaded ? on a proper OS thread implementation, threads can migrate across CPUs, can't they ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Attended transfer and queue
On 06/12/06 20:21 Matt said the following: AHHH! We use the Xfer button on our Aastra 9133is to do transfers for some reason (see another post I just made) when I hit * queue calls disconnect. take a look at http://bugs.digium.com/view.php?id=6897 which solves this problem. also, since this has been committed to 1.2 and trunk, i would think that 1.2.9.1 would also have this patch applied. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Attended transfer and queue
On 06/12/06 20:42 Dinesh Nair said the following: i would think that 1.2.9.1 would also have this patch applied. not it doesnt. my patch was only committed for trunk, though mantis does have the patch that works on 1.2.x as well. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Attended transfer and queue
On 06/12/06 21:11 Matt said the following: What version of Asterisk are you running, that you are able to dial *2 and the * isn't hanging up like it is for me? because i wrote and applied the patch ? :) -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Config Revision Control
On 06/03/06 22:10 Kevin P. Fleming said the following: - Michiel van Baak [EMAIL PROTECTED] wrote: Then the svn automerge thingie Kevin wrote for the asterisk svn tree is automerging changes to the 'common' tree to all the server trees. unrelated to asterisk obviously, but is there somewhere i can download the svn automerge patch of kevin's ? i'd love to have automerge running on our internal svn servers here. :) -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P not recognised on FreeBSD system
On 05/19/06 16:30 Chris Hastie said the following: I've just received an OEM Wildcard X100P FXO card. Installing into my FreeBSD 5.4-RELEASE box it doesn't appear to be recognised at all. Since have you downloaded, compiled and installed the zaptel-bsd drivers ? if you haven't, instructions for getting them are at http://www.voip-info.org/wiki-FreeBSD+zaptel for the X100P, you'd need to kldload zaptel.ko and wcfxo.ko, though be warned that the wcfxo.ko driver has not had much development in yonks. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P not recognised on FreeBSD system
On 05/19/06 18:57 Chris Hastie said the following: Yes, I have these. The modules load, but ztcfg complains ZT_CHANCONFIG failed on channel 1: No such device or address (6) and as I said, it doesn't appear that the card has been recognised by the kernel. could you try the X100P in anther system to rule out issues with the Via board you're using ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Ringing indication not working as expected
On 05/18/06 18:45 Sebastian Kayser said the following: So although the Zap interface is used for both types of external calls (snom - POST, snom - PSTN) the ringing indication to my snoms fails for calls to the PSTN. we've got the following: E1 PRI --- Asterisk ---+--- FXS Gateway --- Analog Phones | +--- FXS Gateway --- Analog Phones | +--- FXS Gateway --- Analog Phones | +--- FXS Gateway --- Analog Phones we see the same problem for /some/ of the phones on a single fxs gateway, but not for the /other/ phones on the /same/ gateway. i'd always thought that this may have been caused by a lost SIP 180 Ringing packet between asterisk and the fxs gateway, but perhaps now i may look inside asterisk itself to try to see if it's causing it somewhere. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple Registers
On 05/17/06 04:00 Noah Miller said the following: only one registration. You can register from multiple devices, but only the one that has most recently registered will receive calls. Put another way, when the second device registers it will unregister the first device. exactly as you've put it for incoming calls. however, in practice, both devices will be able to make outgoing calls. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATXFER
On 05/11/06 19:46 Josué Conti said the following: functions informs to transfer and the transference is ok. However, if the agent tries to effect an attended transference the ATXFER, knocks down the call. All the agents of this queue are with canreinvite=no in i'm guessing that the feature code for ATXFER in features.conf begins with a '*'. this '*' is trapped instead by chan_agent, which is a hardcoded value within chan_agent to hang up the call. that's the same symptoms you're seeing. i've submitted a patch for this, which has been committed to trunk, so the next release of asterisk should have an endcall parameter in agents.conf which allows you to turn off this feature. however, if you're using 1.2.x, and you need it now, you can apply the 1.2 related patch from http://bugs.digium.com/view.php?id=6897 apply agent-endcall.patch for 1.2.x. as noted above, this patch has already been committed to trunk. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Queue Transfer
On 05/02/06 20:50 Josué Conti said the following: To activate the transferences of calls in asterisk, I effected: SIP.CONF in sip of the agent I qualified canreinvite=no, so that asterisk monitors this transference. EXTENSIONS.CONF I qualified the parameters tT in the command Dial FEATURES.CONF I qualified [ featuremap ] to blindxfer = # ; to atxfer = * 7 did you use the t and T options to Queue() in the dialplan ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Queue Transfer
On 04/29/06 20:15 Josué Conti said the following: Dinesh the agents they receive a call and this call will have to be transferred, them uses only functions hold and trnsf in device i'm not sure how the polycom's hold and trnsf buttons are mapped, but using blindxfer and atxfer dtmf keypresses marks the agent unused upon a transfer. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Queue Transfer
On 04/29/06 10:06 Josué Conti said the following: is that if the agent transfers the call, for another user and this user takes care of the call, the status of the agent in the show agents is of that it the same continues speaking (talking to zap) with circuit how are you performing the transfer ? are they blind/attended transfers using the keystrokes in features.conf ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Pinouts for T1/E1 crossover cable WAS RE: [Asterisk-Users] what cable to connect a legacy PBX to a TE410P ?
On 04/25/06 05:58 Sangoma Techdesk said the following: At Sangoma we do quite a lot of back-to back T1 and E1 connections. T1 is not a very fussy connection, as the baud rate is only about 750 kbps. In our experience, for error free communications you can use the following rules of thumb: Up to 50 ft: Flat patch cable Up to 500 ft: Ordinary twisted telephone cable Cat 5 may be overkill unless you are going hundreds of feet. we've faced weird intermittent problems and we suspect it's related to electrical interference caused by power cables et al in the server rack. we've seen this with both sangoma and digium cards when attempting to connect asterisk boxes to carrier E1s provided by the local operator. the cables used are normal cat5 UTP cables. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] queues and the '*' key
On 04/21/06 05:35 Sean Kennedy said the following: I have a vague memory of reading about this somewhere, but searched @ the wiki AND through google aren't turning up anything useful. take a look at http://bugs.digium.com/view.php?id=6897 there's a patch there for 1.2 with another for trunk which has been committed already. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Background music in call
On 04/16/06 10:51 C F said the following: use feauters.conf and the application map section. i may be wrong, but that's not the same as background music during a call. iianm, using playback() or background() in features.conf turns off the call audio and plays the selected file. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Segfault on Inbound call?
On 04/14/06 20:05 Matt said the following: When it patched the zaptel source... if I have usecallerid=yes on then it crashes... if I turn usecallerid=no then it is fine. we've tested the sangoma A101, A102 and A104 cards with usercallerid=yes, and it hasn't crashed. this is on FreeBSD though, and the sangoma driver installation did not patch the zaptel-bsd drivers at all. ymmv on linux. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [asterisk-dev] Announcing Astmanproxy 1.20
On 04/08/06 11:26 [EMAIL PROTECTED] said the following: I'm pleased to announce the release of Astmanproxy 1.20, the fast, flexible proxy server for Asterisk's Manager Interface. Astmanproxy we've just started using astmanproxy, and i'll soon be submitting a couple of patches which addresses the following: 1. Building astmanproxy on FreeBSD 2. having astmanproxy reconnect if asterisk dies and restarts who should i submit patches to ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [asterisk-dev] Announcing Astmanproxy 1.20
On 04/08/06 11:26 [EMAIL PROTECTED] said the following: I'm pleased to announce the release of Astmanproxy 1.20, the fast, flexible proxy server for Asterisk's Manager Interface. Astmanproxy we've just started using astmanproxy, and i'll soon be submitting a couple of patches which addresses the following: 1. Building astmanproxy on FreeBSD 2. having astmanproxy reconnect if asterisk dies and restarts who should i submit patches to ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk-ooh323, asterisk 1.2.6 and netmeeting
On 04/06/06 05:36 Avi Miller said the following: If I dialled from a SIP phone on Asterisk 1 to the Phone on the Avaya, it worked fine. If I dialled from a phone on the Avaya, the SIP phone would ring, but the call would drop as soon as it was answered because of codec negotiation failure. absolutely the same symptoms. my architecture is as follows: OHPHONE Asterisk SIP Client calls from the SIP client to OHPHONE work fine with audio et al passed both ways. calls from OH PHONE to the SIP client dont. just after the SIP client answers, the call dies. i tried your suggestion of removing all disallow and allow lines in ooh323.conf, but with that, even calls from SIP to H323 (which were working) stop working. it does lend credence to the theory that it's a codec nego issue though. the debug and verbose output of a failed H323 to SIP call is below (6262 is the SIP exten and 6996 is the OHPHONE H.323): Apr 6 13:59:37 VERBOSE[201] logger.c: -- Executing Dial(OOH323/192.168.1.169-0361, SIP/6262|40|owWtT) in new stack Apr 6 13:59:37 DEBUG[201] chan_sip.c: Setting NAT on RTP to 0 Apr 6 13:59:37 DEBUG[201] chan_sip.c: Setting NAT on VRTP to 0 Apr 6 13:59:37 DEBUG[201] acl.c: # Testing 192.168.1.164 with 192.168.1.0 Apr 6 13:59:37 DEBUG[201] chan_sip.c: Outgoing Call for 6262 Apr 6 13:59:37 VERBOSE[201] logger.c: -- Called 6262 Apr 6 13:59:37 DEBUG[201] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found Apr 6 13:59:37 VERBOSE[201] logger.c: -- SIP/6262-960b is ringing Apr 6 13:59:37 DEBUG[201] channel.c: Driver for channel 'OOH323/192.168.1.169-0361' does not support indication 3, emulating it Apr 6 13:59:37 DEBUG[201] channel.c: Prodding channel 'OOH323/192.168.1.169-0361' Apr 6 13:59:37 DEBUG[201] channel.c: Scheduling timer at 160 sample intervals Apr 6 13:59:37 DEBUG[201] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]' Apr 6 13:59:37 DEBUG[201] acl.c: # Testing 192.168.1.151 with 192.168.1.0 Apr 6 13:59:37 DEBUG[201] chan_sip.c: SIP message could not be handled, bad request: [EMAIL PROTECTED] Apr 6 13:59:38 DEBUG[201] chan_sip.c: Acked pending invite 102 Apr 6 13:59:38 DEBUG[201] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Apr 6 13:59:38 DEBUG[201] chan_sip.c: build_route: Contact hop: sip:[EMAIL PROTECTED]:5060 Apr 6 13:59:38 VERBOSE[201] logger.c: -- SIP/6262-960b answered OOH323/192.168.1.169-0361 Apr 6 13:59:38 WARNING[201] src/chan_h323.c: Don't know how to indicate condition -1 on ooh323c_7 Apr 6 13:59:38 DEBUG[201] channel.c: Scheduling timer at 0 sample intervals Apr 6 13:59:38 VERBOSE[201] logger.c: -- Attempting native bridge of OOH323/192.168.1.169-0361 and SIP/6262-960b Apr 6 13:59:38 DEBUG[201] channel.c: Didn't get a frame from channel: OOH323/192.168.1.169-0361 Apr 6 13:59:38 DEBUG[201] channel.c: Bridge stops bridging channels OOH323/192.168.1.169-0361 and SIP/6262-960b Apr 6 13:59:38 DEBUG[201] chan_sip.c: update_call_counter(6262) - decrement call limit counter Apr 6 13:59:38 DEBUG[201] app_dial.c: Exiting with DIALSTATUS=ANSWER. Apr 6 13:59:38 VERBOSE[201] logger.c: == Spawn extension (macro-stdexten, s-DIAL, 1) exited non-zero on 'OOH323/192.168.1.169-0361' in macro 'stdexten' Apr 6 13:59:38 VERBOSE[201] logger.c: == Spawn extension (macro-stdexten, s-DIAL, 1) exited non-zero on 'OOH323/192.168.1.169-0361' Apr 6 13:59:39 DEBUG[201] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 103: Match Found Apr 6 13:59:40 DEBUG[201] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]' note that channel.c says it didnt get a frame from OHPHONE and that it subsequent stops bridging the channels. now to go figure out why this is so. any pointers would be appreciated. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Questions on call recording and conference.
On 03/31/06 08:24 Wai Wu said the following: In Asterisk, what happens to the files when both legs of the call hangs up? Is there a way to create a conference room on the flight? i.e. without pre-defining the conference ID in meetme.conf. look at the 'd' option to MeetMe. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: queue issue
On 04/06/06 19:17 Tomislav Parèina said the following: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... on a related note, we notice that if we've set atxfer = *1 in features.conf and blindxfer=#1, then attended transfers dont work. somehow, the Queue app captures the '*' and hangs up the call. is this the behaviour others have observed ? obviously, since we've used *2 for auto monitor, that doesnt work as well. Yes, this is well known (problem?). I have solved it by editing features.conf file. i've opened a bug and provided a fix for this at http://bugs.digium.com/view.php?id=6897 on investigation into the source, it wasnt the queue app but rather chan_agent which was doing this. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pickup() h323
On 04/06/06 04:41 Dan Austin said the following: Chan_ooh323 just worked. The code is, to a infrequent programmer, easy to read, extend and fix bugs. ok, i'm not getting into a my H323 is better than yours argument, but we've been struggling to get OOH323 working with OHPHONE. symptoms are that calls from SIP -- OhPhone work fine, but when OhPhone calls SIP, the call is hungup the moment the SIP phone answers. any clues why ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk-ooh323, asterisk 1.2.6 and netmeeting
On 04/05/06 13:52 Dinesh Nair said the following: On 04/05/06 13:17 Avi Miller said the following: I had a similar problem connecting Asterisk to an Avaya IP403 via OOH323: In the end, I removed all the disallow=all and allow=codec lines in Asterisk. This seems to have allowed the two systems to overcome the codec negotiation problems they were having and proceed with actual audio transfer. :) we'll try with this, but further testing reveals that the H.323 negotiation over port 1720 happens fine, with H.245 then being done over another TCP port tuple. we didnt see the RTP port session being created/negotiated. i'm assuming from the asterisk-ooh323 docs that it uses asterisk's builtin RTP mechanism, and this should be over UDP. there were no UDP packets being exchanged at all. we will try your suggestion however. more tests reveal that with ohphone, calls from SIP-ohphone work fine with audio passed both ways. however when ohphone calls a SIP device, the call is hungup when the SIP device answers. obviously, SIP-IAX and SIP-SIP calls work fine, so there's nothing wrong with the SIP device per se. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk-ooh323, asterisk 1.2.6 and netmeeting
has anyone managed to get these three beasties to work together ? we're using ooh323 from asterisk-addons-1.2.2, asterisk 1.2.6 and microsoft netmeeting default from windows xp. the symptoms are that calls from a SIP client to NetMeeting rings on NetMeeting, but upon answering the call in NetMeeting, no audio is passed between the two. eventually, the call times out and hangs up. on a reverse call from NetMeeting to the SIP client, the SIP client rings and when it's answered, the same thing happens, i.e. no audio is passed. the same happens when netmeeting calls an IVR-related app like Directory, SayDigits et al. my ooh323.conf file is attached. also, here's the asterisk console output with ooh323 debug on: NetMeeting H323 to SIP --- onNewCallCreated ooh323c_7 +++ onNewCallCreated ooh323c_7 --- ooh323_onReceivedSetup ooh323c_7 --- find_user +++ find_user Adding capabilities to call(incoming, ooh323c_7) --- configure_local_rtp +++ configure_local_rtp +++ ooh323_onReceivedSetup - Determined context default, extension 6384 --- onAlerting ooh323c_7 --- find_call +++ find_call +++ onAlerting ooh323c_7 -- Executing Dial(OOH323/mms mms-fa6a, SIP/6384|40|owWtT) in new stack -- Called 6384 -- SIP/6384-d9f2 is ringing - ooh323_indicate 3 on call ooh323c_7 ooh323_indicate 3 on ooh323c_7 -- SIP/6384-d9f2 answered OOH323/mms mms-fa6a - ooh323_indicate -1 on call ooh323c_7 Apr 4 18:16:34 WARNING[3021]: src/chan_h323.c:952 ooh323_indicate: Don't know how to indicate condition -1 on ooh323c_7 ooh323_indicate -1 on ooh323c_7 --- ooh323_answer +++ ooh323_answer -- Attempting native bridge of OOH323/mms mms-fa6a and SIP/6384-d9f2 --- onCallEstablished ooh323c_7 --- find_call +++ find_call +++ onCallEstablished ooh323c_7 --- onCallCleared ooh323c_7 --- find_call +++ find_call == Spawn extension (macro-stdexten, s-DIAL, 1) exited non-zero on 'OOH323/mms mms-fa6a' in macro 'stdexten' == Spawn extension (macro-stdexten, s-DIAL, 1) exited non-zero on 'OOH323/mms mms-fa6a' --- ooh323_hangup hanging mms mms +++ ooh323_hangup --- ooh323_destroy Destroying mms mms +++ ooh323_destroy SIP to H323 NetMeeting -- Executing Dial(SIP/6384-b575, OOH323/6985|40|owWtT) in new stack --- ooh323_request - data 6985 format 0x8 (alaw) --- find_peer +++ find_peer +++ ooh323_request --- ooh323_call- 6985 +++ ooh323_call -- Called 6985 --- onNewCallCreated ooh323c_o_3 --- find_call +++ find_call setting callid number 6384 Outgoing call 6985(ooh323c_o_3) - Codec prefs - (ulaw) Adding capabilities to call(outgoing, ooh323c_o_3) Adding g711 ulaw capability to call(outgoing, ooh323c_o_3) --- configure_local_rtp +++ configure_local_rtp +++ onNewCallCreated ooh323c_o_3 --- onAlerting ooh323c_o_3 --- find_call +++ find_call +++ onAlerting ooh323c_o_3 -- OOH323/6985-e521 is ringing --- onCallEstablished ooh323c_o_3 --- find_call +++ find_call +++ onCallEstablished ooh323c_o_3 -- OOH323/6985-e521 answered SIP/6384-b575 -- Attempting native bridge of SIP/6384-b575 and OOH323/6985-e521 --- ooh323_hangup hanging 6985 +++ ooh323_hangup == Spawn extension (macro-stdexten, s-DIAL, 1) exited non-zero on 'SIP/6384-b575' in macro 'stdexten' == Spawn extension (macro-stdexten, s-DIAL, 1) exited non-zero on 'SIP/6384-b575' --- onCallCleared ooh323c_o_3 --- find_call +++ find_call +++ onCallCleared --- ooh323_destroy Destroying 6985 +++ ooh323_destroy -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ [general] h323id=QubeTalkECS callerid=QubeTalkECS gatekeeper=DISABLE ; DISCOVER or IP addy logfile=/var/spool/asterisk/log/h323_log gateway=no ; or yes faststart=yes h245tunneling=yes port=1720 bindaddr=0.0.0.0 context=default [6970] ip=192.168.1.160
Re: [Asterisk-Users] asterisk-ooh323, asterisk 1.2.6 and netmeeting
On 04/05/06 13:17 Avi Miller said the following: I had a similar problem connecting Asterisk to an Avaya IP403 via OOH323: In the end, I removed all the disallow=all and allow=codec lines in Asterisk. This seems to have allowed the two systems to overcome the codec negotiation problems they were having and proceed with actual audio transfer. :) we'll try with this, but further testing reveals that the H.323 negotiation over port 1720 happens fine, with H.245 then being done over another TCP port tuple. we didnt see the RTP port session being created/negotiated. i'm assuming from the asterisk-ooh323 docs that it uses asterisk's builtin RTP mechanism, and this should be over UDP. there were no UDP packets being exchanged at all. we will try your suggestion however. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] queue issue
On 04/05/06 21:37 Dov Bigio said the following: - The agent transferred the call to an user (not a queue), by dialing the atxtransfer (1) key defined in features.conf on a related note, we notice that if we've set atxfer = *1 in features.conf and blindxfer=#1, then attended transfers dont work. somehow, the Queue app captures the '*' and hangs up the call. is this the behaviour others have observed ? obviously, since we've used *2 for auto monitor, that doesnt work as well. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk, QSIG and Tenovis PBX?
On 03/31/06 23:29 Jim Houser said the following: Looking at the TE100P I don't see it listed Q.SIG as supported. We have it working great as PRI. Am I wrong about the Q.SIG support? Q.SIG and the like are supported from libpri. we got it working with a TE410P, but i'm sure getting to work with the single span cards shouldnt be much different. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: H323 problems
On 04/04/06 19:20 Tomislav Parèina said the following: Ooh323 channel driver from asterisk-addons-1.2.1 has same problem have you managed to get this working ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Benchmarking an Asterisk Server with 14k users
On 03/31/06 00:28 Stefan-Michael. Guenther (in-put GbR) said the following: To make it clear: We don't want to compare the three system against each other. The asterisk server is running on a completely different hardware. We what are the hardware and OS specs for the asterisk server ? this will form the crux of what you're testing. 7,000 simultaneous calls seems high for a single server to handle, you may need to build a cluster of asterisk servers to handle this. signate has claimed 5,000 simultaneous calls on their asterisk based product. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk, QSIG and Tenovis PBX?
On 03/31/06 19:49 Wolfgang Zweimueller said the following: My conclusion with Q.SIG: do not use it at this implementation level. YMMV. i'll beg to differ. we've used Q.SIG successfully with an Ericsson MD110 for a customer in thailand. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RTP frame size location?
On 03/29/06 13:06 Andres said the following: It works perfectly with other values we have tested of 40 and 60. We currently use 60 on all our servers. It cuts down on bandwidth for a G279 call to about 15Kbps. with 60ms packets, is a packet loss or two noticable ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to create [new_context] in extensions.conf?
On 03/24/06 07:39 Larry Alkoff said the following: That's how I _thought_ it worked but extens in such a created [context_name] are not seen or used by Asterisk to dial out. There is something missing. have you included the new context in the context where your phones are set to ? include = new_context -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Remote dialtone
On 03/22/06 21:55 Jason Bachman said the following:
Karlos,
Sounds like you want ignorepat = 2 (or 3) in the context that holds the
dial patterns. This will continue the dialtone after you dial 2 or 3 in
your dialplan.
IE:
[system-2]
ignorepat = 3
exten = _3XX,s,1,Dial(IAX2/system-2/${EXTEN})
ignorepat wont work for SIP or IAX2 phones however since they send the
entire called number as a single SIP/IAX2 packet.
--
Regards, /\_/\ All dogs go to heaven.
[EMAIL PROTECTED](0 0)http://www.alphaque.com/
+==oOO--(_)--OOo==+
| for a in past present future; do|
| for b in clients employers associates relatives neighbours pets; do |
| echo The opinions here in no way reflect the opinions of my $a $b. |
| done; done |
+=+
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
