[asterisk-users] Call recovery feature
Hello, what about: This feature means you can restart Asterisk after a failure (or asterisk restart itself with safe_asterisk), and keep existing calls up with only a few seconds of audio dropped. That would be a feature! there is a other pbx that has this feature... Anyone else would like to see that feature? just want to start some brainstorming, Kristijan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which SpanDSP version to play with Asterisk 10 and T.38/T.30 gatewaying ?
I use the latest spandsp source from the freeswitch git. There you have also a changelog documenting the differences. Steve Underwood commit here the latest changes in spandsp source. http://fisheye.freeswitch.org/changelog/freeswitch.git/libs/spandsp Kristijan 2012/1/11 Olivier oza_4...@yahoo.fr: Hi, Maybe I missed it while checking it, but which spandsp version is recommended to play with Asterisk 10 and T.38/T.30 gatewaying ? I can see both spandsp-0.0.6pre17.tgz and spandsp-0.0.6pre18.tgz here (http://www.soft-switch.org/downloads/spandsp/) but I couldn't find a changelog documenting differences between them. So I prefer to double check ask for recommendations. Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.7 and ReceiveFAX
remove the c argument Kristijan 2011/10/7 Administrator TOOTAI ad...@tootai.net: Hi, I setup my first stock 1.8.7 asterisk (Ubuntu LTS 10.04 packages taken from deb http://packages.asterisk.org/deb lucid main) including dahdi from this same repository. No FFA involved. On incoming calls (only SIP, no telephony card), fax detection is working but reception failed with -- Executing [fax@from-TOOTAiAudio:19] ReceiveFAX(SIP/tootaiAUDIO-0564, /tmp/1317991071.1614.tiff,c) in new stack [Oct 7 14:37:52] WARNING[6961]: res_fax.c:1651 receivefax_exec: ReceiveFAX does not support polling == Spawn extension (from-TOOTAiAudio, fax, 19) exited non-zero on 'SIP/tootaiAUDIO-0564' What can be the problem? Thanks for any hint. -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] mISDN and 1.8
Gergo, why do you want to use mISDN? Use Dahdi. Or do you want to use a very exotic isdn card which is only supported by mISDN? tell us more. My long time experience with mISDN_v1 is, v1 has major echo and fax problems. because the audio signals are transported very unsynchronic because of the kernel driver architecture which was made for data transmission, and originally not for voice and fax applications. mISDN_v2 with chan_lcr has vital improvements for voice. especially together with OSLEC as echo canceller. but still has some issues with fax transmission. And since Dahdi supports the important BRI card based on HFC-8S/4S colonge chip and via http://code.google.com/p/zaphfc/ also the cheap HFC-S cards. No reason anymore to use mISDN. Kristijan 2011/9/26 Gergo Csibra csi...@gmail.com: Monday, September 26, 2011, 7:20:10 PM, Kevin wrote: On 09/26/2011 11:35 AM, Gergo Csibra wrote: Hi, are there anybody, who using the chan_misdn included with Asterisk v1.8? If yes what mISDN version used v1 or v2? Yes, I can read on many pages for mISDN2 I need to use chan_lcr, but this informations are 2-3 years old, and I can't imagine asterisk v1.8 chan_misdn works only with linux kernelv2.6.24 which is quite old. chan_misdn only supports mISDN version 1. This means I need to hack a v2.6.24 kernel remove the mISDNv2 and install mISDN v1? -- Best regards, Gergo mailto:csi...@gmail.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DMTF via rfc2833 and SIP-INFO simultaneously
the provider/carrier changed his setting how to submit DTMF to our asterisk. It was set to send SIP-INFO and rfc2833 to send only rfc2833 Kristijan 2011/9/13 virendra bhati virbh...@gmail.com: Hi , What was the solution of that problem ? Did provider change the setting at there end or else ? On Tue, Sep 13, 2011 at 7:37 PM, Kristijan Vrban vrban.l...@googlemail.com wrote: hello Virendra, thx for your response. but after i made clear to the carrier that i want the dmtf only via rfc2833 and not via rfc2833 and SIP-INFO simultaneously, the problem is fixed. Kristijan 2011/9/13 virendra bhati virbh...@gmail.com: Hi 1st check that how many manager is connected into the server. 1 or more then you can say that 2 DTMF is capture by asterisk for same events. manager show connected Username IP Address root 127.0.0.1 it should be one only. I face the same case then I found that more then 1 manager was working into the server. On Fri, Aug 26, 2011 at 3:11 PM, Kristijan Vrban vrban.l...@googlemail.com wrote: Hello, i have a carrier that send DMTF via rfc2833 and SIP-INFO simultaneously. That has the effect, that asterisk read every dtmf twice. and yes, it's mainly the carriers mistake. but is there a configure option, that asterisk accept only one DMTF method for inbound dtmf? Kristijan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Thanks and regards Virendra Bhati +91-9172341457 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Thanks and regards Virendra Bhati +91-9172341457 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DMTF via rfc2833 and SIP-INFO simultaneously
hello Virendra, thx for your response. but after i made clear to the carrier that i want the dmtf only via rfc2833 and not via rfc2833 and SIP-INFO simultaneously, the problem is fixed. Kristijan 2011/9/13 virendra bhati virbh...@gmail.com: Hi 1st check that how many manager is connected into the server. 1 or more then you can say that 2 DTMF is capture by asterisk for same events. manager show connected Username IP Address root 127.0.0.1 it should be one only. I face the same case then I found that more then 1 manager was working into the server. On Fri, Aug 26, 2011 at 3:11 PM, Kristijan Vrban vrban.l...@googlemail.com wrote: Hello, i have a carrier that send DMTF via rfc2833 and SIP-INFO simultaneously. That has the effect, that asterisk read every dtmf twice. and yes, it's mainly the carriers mistake. but is there a configure option, that asterisk accept only one DMTF method for inbound dtmf? Kristijan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Thanks and regards Virendra Bhati +91-9172341457 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DMTF via rfc2833 and SIP-INFO simultaneously
Hello, i have a carrier that send DMTF via rfc2833 and SIP-INFO simultaneously. That has the effect, that asterisk read every dtmf twice. and yes, it's mainly the carriers mistake. but is there a configure option, that asterisk accept only one DMTF method for inbound dtmf? Kristijan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk spontaneous reboot
use gdb (The GNU Project Debugger) to take a look into the core dump gdb asterisk core.sip.pbx.tld-2011-08-26T08:07:35+0200 Kristijan 2011/8/26 Jonas Kellens jonas.kell...@telenet.be: Hello, Today I restart the MySQL-DB (/sbin/service mysqld restart) and I could no longer connect to asterisk (/usr/sbin/asterisk -r) for a few seconds. There is now a core dump present in /tmp : -rw--- 1 root root 88M Aug 26 08:07 core.sip.pbx.tld-2011-08-26T08:07:35+0200 How can I get usefull information about what went wrong ? Because a spontaneous reboot of Asterisk has never happened before when just restarting the MySQL-deamon. Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue Group not forwaring calls to agents
did you find al solution for this issues? i fight with the same problem. kristijan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AST_DEVICE_UNAVAILABLE vs. AST_DEVICE_UNKNOWN for new loaded realtime peers
Hello is use realtime sip-peers in 1.8, and have the problem, that when a peer is loaded from database, the devstate is AST_DEVICE_UNAVAILABLE and the the peers can not be called from the queue. because the app_queue only calls agens in state AST_DEVICE_NOT_INUSE or AST_DEVICE_UNKNOWN. My question: is this behavior configurable, or is s scource code change necessary? My opinion is, that when a sip peer is loaded from database, and it is not a dynamic host, the the devstate should be AST_DEVICE_UNKNOWN, because in this moment it is UNKNOWN, and the app_queue whould try to call the peer. Kristijan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Latest DAHDI/libpri/Asterisk 1.8 1x BRI port HFC based ISDN card?
http://code.google.com/p/zaphfc/ 2011/5/23 Patrick Lists asterisk-l...@puzzled.xs4all.nl: Hi, I would appreciate some advice on the following: how does one use a single BRI port HFC chipset based ISDN cards with the latest DAHDI, libpri and Asterisk 1.8? For Asterisk 1.4 I would first install mISDN mISDNuser and then build Asterisk 1.4. Now I noticed that Digium's HFC chipset based B410P is supported by the latest DAHDI libpri but reading the source of the wcb4xxp driver it only seems to support HFC chipset based cards with _2_ or more BRI ports. What do I use for a _single_ BRI port card with latest DAHDI, libpri and Asterisk 1.8? Thanks and regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good by asterisk 1.4? Please not.
@digium 1. What happened with the 1.4 patches that still wait on issues.asterisk.org? e.g. issue #19108 2. What happened with bugfix patches for 1.4 made by the community. Will those be ignored now? (e.g. i have one more a memleak fix for 1.4 in preparation, that i can publish earliest after 2011-04-21) Kristijan 2011/4/15 Julian Lyndon-Smith aster...@dotr.com: 1.4 svn has a nasty bug in it at the moment. Would love to see that fixed ;) https://issues.asterisk.org/view.php?id=18951 Julian On 15 April 2011 14:22, Satish Patel satish...@hotmail.com wrote: You know we don't have choise. I had remembered when we shifted 1.2 to first release of 1.4 and we had many issue. Same thing right now I'm dealing with 1.8 things take time to stabilized. Good luck!! -- Sent from my iPhone On Apr 15, 2011, at 8:33 AM, Kristijan Vrban vrban.l...@googlemail.com wrote: Security only fixes: 2011-04-21 So in six days, no more bugfix patches will committed into 1.4-branch :( Is a prolongation possible? Because 1.4 is so reliable now. It would be a great loss. And no, 1.8 is not (yet) a replacement. Kristijan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Follow Ode To Politics by HB Tasker at http://twitter.com/HBTasker -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Good by asterisk 1.4? Please not.
Security only fixes: 2011-04-21 So in six days, no more bugfix patches will committed into 1.4-branch :( Is a prolongation possible? Because 1.4 is so reliable now. It would be a great loss. And no, 1.8 is not (yet) a replacement. Kristijan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ReceiveFAX issue.
also the Answer is redundant in logger.conf console = warning,error,notice,debug,fax 2011/1/24 David Backeberg dbackeb...@gmail.com: On Mon, Jan 24, 2011 at 2:53 PM, Bryant Zimmerman brya...@zktech.com wrote: I am testing out inbound faxing using res_fax and res_fax_spandsp.so My system answers the call but then sets there on the ReseiveFax line then comes back with an error that it exceeded the maximum retries. How would I go about debugging this? Below is my very simple dialplan code I am using, and the fax show version gives the following as well. Record the call with Monitor() or MixMonitor(). Listen to the call. See if you can figure out anything obvious. Why are you doing a Wait(2)? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which version to use: 1.4 or 1.6 or 1.8
there is no reason not to use 1.8 when you start a new installation. 1.8 is the new five years long term support version Kristijan 2010/12/15 bilal ghayyad bilmar...@yahoo.com: Hi All; I need to know which version of asterisk to use, if to be 1.4 or 1.6 or 1.8? For example, when to decide that I have to go for 1.6 or I have to go for 1.8? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] app_voicemail: 3 for advanced options does not have an effect _while_ the vm-message is played
3 for advanced options does not have an effect _while_ the vm-message is played. all other options like 7 delete or 6 next message are working. after the vm-message is played, it's working. the question: is this intentional? or a bug? the available documentation does not describe this case. Kristijan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-1.8.0-beta4 - compile error
if you dont need asterisk as a fax maschine, just disable res_fax_spandsp with make menuconfig. if you want fax support, first remove all old spandsp lib/header, and install the latest spandsp on you system from: http://www.soft-switch.org/downloads/snapshots/spandsp/?C=M;O=D then again ./configure make kristijan 2010/8/24 Václav Strachoň vaclav.strac...@gmail.com: Hi, I tried to compile asterisk-1.8.0-beta4 but after ./configure make I've got following error: [CC] res_fax.c - res_fax.o [LD] res_fax.o - res_fax.so [CC] res_fax_spandsp.c - res_fax_spandsp.o res_fax_spandsp.c:117: error: field âfax_stateâ has incomplete type res_fax_spandsp.c:118: error: field ât38_stateâ has incomplete type res_fax_spandsp.c: In function âspandsp_fax_startâ: res_fax_spandsp.c:580: error: dereferencing pointer to incomplete type res_fax_spandsp.c: In function âspandsp_fax_cli_show_sessionâ: res_fax_spandsp.c:682: error: dereferencing pointer to incomplete type res_fax_spandsp.c:682: error: dereferencing pointer to incomplete type make[1]: *** [res_fax_spandsp.o] Error 1 make: *** [res] Error 2 Can somebody help me? asterisk-1.8.0-beta4 Linux 2.6.18-164.15.1.el5 #1 SMP CentOS release 5.2 (Final) Thanks, Vaclav Strachon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 beta3 - Unable to stop/start/restart deamon
use the init script from debian http://svn.debian.org/viewsvn/pkg-voip/asterisk/trunk/debian/asterisk.init?revision=8502view=markup the one from the asterisk source seems to be broken, if have the same issue kristijan 2010/8/16 unsero...@aol.com: I am using Debian Lenny, not RedHat. -Original Message- From: Faisal Hanif fai...@vopium.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Mon, Aug 16, 2010 11:33 am Subject: Re: [asterisk-users] Asterisk 1.8 beta3 - Unable to stop/start/restart deamon did you copied rc.redhat.asterisk script from contrib/init.d/ forlder to /etc/init.d/ folder? Regards, Faisal Hanif On 8/16/2010 2:28 PM, unsero...@aol.com wrote: No ideas? Sorry but I'm new to Linux and I am wondering why I can't stop or start the deamon the way it works with 1.6.1.20. I installed Asterisk 1.8 with all defaults set. Maybe something is missing in any conf file? Make sure it starts without the daemon. Try asterisk -cvvv. Does it start then? sean -- Yes, without the daemon it starts and i don't see any errors. It also starts automatically after a system boot. But I am wondering why I can't stop|start|restart using /etc/init.d/asterisk start|stop|restart like in 1.6? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] res_fax_digium and T.38 error correction
Hello, i just had some fax abortions because of some packet loss. so i startet to examine in the pcap recording from the res_fax_digium, if the T.38 EC mode redundancy was really used. So i watched into it, and compared it with a t.38 pcap from spandsp (same asterisk setup, but with app_fax) and i see differences in t38.error_recovery (error-recovery: secondary-ifp-packets) With spandsp here are three items, and with res_fax_digium zero items. (t38.secondary_ifp_packets) I this the the t.38 error correction? I ask this questions, because the fax for asterisk admin manual, there are no information about the T.38 error correction, and if i better use Redundancy or FEC. Kristijan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Internal timing bad for Fax?
Hello, i just made the reproducible watching: I send a Fax from asterisk (trunk) with spandsp (latest snapshot) via T.38 - Audiocodes Mediant 2000 (FW 5.60.43.5) - PSTN Fax With Internal timing Enabled, the Fax break after the first quarter from the first page is transfered. With Internal timing Disabled, the fax is transferred flawless. Both test with pthread timing module on a QEMU Virtual maschine So, is internal timing bad for Fax? Kristijan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Unavailable issue with SIP realtime and app_queue (*-1.4)
Hello, when is use SIP realtime and i (re)start asterisk, then all SIP user are Unavailable, and never go to Not in use because the phones are registered on opensips. And res_config_mysql does load the user only, when the SIP does a call (or get called) an then chan_sip give app_queue the information, that this SIP user is available. Is there a possibly to force app_queue to give the call to chan_sip, even if app_queue thinks the SIP user is Unavailable? Kristijan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Extensions Reload | Asterisk Freezes ? 1.4
I can confirm this issue. And my setup is similar to yours. kristijan 2010/4/19 Positively Optimistic positivelyoptimis...@gmail.com: Good day.. We have what I consider to be a large dialplan (-= 1501 extensions (2559 priorities) in 99 contexts. =-) If we have more than 10 or so channels up (all SIP, no TDM) and issue the extensions reload command.. quite often, asterisk will completely freeze up... requiring us to either kill and restart the process or restart the box... I should probably also share that when watching the log files, at the time the extensions reload command executes... there are no exceptions and/or issues reported while parsing the dialplan... Has anyone in the community experienced this.. and/or have any suggestions ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1.4 chan_sip use internal IP for dialog-info+xml SUBSCRIBE, why?
Asterisk 1.4.29 BLF-SUBSCRIBE go to internal IP (ngrep output): U 2010/03/02 11:34:06.013515 212.78.xxx.xxx:2048 - 62.134.xxx.xxx:5060 SUBSCRIBE sip:1...@62.134.xxx.xxx SIP/2.0..Via: SIP/2.0/UDP 212.78.xxx.xxx:2048;branch=z9hG4bK-d28tfohos0vh;rport..From: sip:k922002...@62.134.xxx.xxx;tag=vyx8c0trgx..To: sip:1...@62.134.xxx.xxx;tag=as13e7cb7c..Call-ID: 3c2768d8487f-rbzdwjzdbgcs..CSeq: 1163 SUBSCRIBE..Contact: sip:k922002...@192.168.55.31:2048;reg-id=1..max-forwards: 70. .event: dialog..user-agent: snom320/8.2.25..expires: 60..Accept: application/dialog-info+xml..Content-Length: 0 U 2010/03/02 11:34:06.053870 192.168.4.109:5060 - 192.168.55.31:2048 SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP 212.78.xxx.xxx:2048;branch=z9hG4bK-d28tfohos0vh;rport;received=212.78.xxx.xxx..From: sip:k922002...@62.134.xxx.xxx;tag=vyx8c0 trgx..To: sip:1...@62.134.xxx.xxx;tag=as13e7cb7c..Call-ID: 3c2768d8487f-rbzdwjzdbgcs..CSeq: 1163 SUBSCRIBE..User-Agent: asterisk 1.4.29..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO..Supported: replaces..WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=5a8a1268..Content-Length: 0 But VM-SUBSCRIBE go to external IP (ngrep output): U 2010/03/02 11:33:46.362857 212.78.xxx.xxx:2048 - 62.134.xxx.xxx:5060 SUBSCRIBE sip:aster...@62.134.xxx.xxx SIP/2.0..Via: SIP/2.0/UDP 212.78.xxx.xxx:2048;branch=z9hG4bK-lubj3r12xmcy;rport..From: sip:k922002...@62.134.xxx.xxx;tag=bov99lxeez ..To: sip:voicem...@62.134.xxx.xxx;tag=as586c72f7..Call-ID: 3c2670215123-ymlw0ru3an2r..CSeq: 851 SUBSCRIBE..Contact: sip:k922002...@192.168.55.31:2048;reg-id=1..max-f orwards: 70..event: message-summary..user-agent: snom320/8.2.25..expires: 60..Accept: application/simple-message-summary..Content-Length: 0 U 2010/03/02 11:33:46.363003 62.134.xxx.xxx:5060 - 212.78.xxx.xxx:2048 SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP 212.78.xxx.xxx:2048;branch=z9hG4bK-lubj3r12xmcy;rport;received=212.78.xxx.xxx..From: sip:k922002...@62.134.xxx.xxx;tag=bov99l xeez..To: sip:voicem...@62.134.xxx.xxx;tag=as586c72f7..Call-ID: 3c2670215123-ymlw0ru3an2r..CSeq: 851 SUBSCRIBE..User-Agent: asterisk 1.4.29..Allow: INVITE, ACK, CANCEL, OPT IONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO..Supported: replaces..WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=66b3b8ff..Content-Length: 0 Why? Look's like a bug for me? sip show peer K922002626: * Name : K922002626 Realtime peer: Yes, cached Secret : Set MD5Secret: Not set Context : K9220 Subscr.Cont. : Not set Language : de AMA flags: Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : Mailbox : 3...@k9220 VM Extension : voicemail LastMsgsSent : 0/0 Call limit : 5 Dynamic : Yes Callerid : MaxCallBR: 384 kbps Expire : 232 Insecure : no Nat : No ACL : No T38 pt UDPTL : No CanReinvite : Yes PromiscRedir : No User=Phone : No Video Support: No Trust RPID : No Send RPID: No Subscriptions: Yes Overlap dial : No DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr-IP : 212.78.xxx.xxx Port 2048 Defaddr-IP : 0.0.0.0 Port 2048 Def. Username: K922002626 SIP Options : (none) Codecs : 0x80e (gsm|ulaw|alaw|g726) Codec Order : (alaw:20,ulaw:20,g726:20,gsm:20) Auto-Framing: No Status : OK (47 ms) Useragent: snom320/8.2.25 Reg. Contact : sip:k922002...@212.78.xxx.xxx:2048 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T.38 with reinvite
good question. i never investigated this issue more exact. Any other T.38 more knowing here if this is possibly anyway? Kristijan 2010/2/12 Deepesh D deep.d2...@gmail.com: Hello, Is it possible to use asterisk in T.38 pass through mode with reinvite? My fax calls are getting disconnected if canreinvite=yes. It works only if I make canreinvite=no. Normal calls work in both cases. Thanks -- _ -- 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 -- _ -- 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] ReceiveFAX and SendFAX questions
it's because after ReceiveFAX (i use asterisk 1.6.2.1 with
spandsp-0.0.6pre17), it jumps to the hangup exten. So this is my
dialplan for ReceiveFAX:
[fax-in]
exten = s,1,ReceiveFAX(/tmp/fax-${CDR(uniqueid)}.tif)
exten = s,n,Hangup()
exten = h,1,NoOp(### FAXSTATUS: ${FAXSTATUS})
exten = h,n,NoOp(###FAXERROR: ${FAXERROR})
exten = h,n,NoOp(### FAXMODE: ${FAXMODE})
exten = h,n,NoOp(###FAXPAGES: ${FAXPAGES})
exten = h,n,NoOp(### FAXBITRATE: ${FAXBITRATE})
exten = h,n,NoOp(### FAXRESOLUTION: ${FAXRESOLUTION})
exten = h,n,NoOp(### REMOTESTATIONID: ${REMOTESTATIONID})
exten = h,n,System(/usr/bin/tiff2pdf /tmp/fax-${CDR(uniqueid)}.tif -o
/tmp/fax-${CDR(uniqueid)}.pdf)
exten = h,n,System(/home/kristijan.vrban/test/fax_perl/send_email.pl
${CUSTOMER} ${EFAX} /tmp/fax-${CDR(uniqueid)}.pdf ${CALLERID(num):1}
${FAXPAGES})
and the exten = 101,1,Answer() and exten = 101,2,Wait(3) are
redundant in you dialplan.
Kristijan
2010/1/24 Magnus Benngård magnu...@inputinterior.se:
Morning,
Have some questions regarding receiving and sending faxes...
1:st example:
exten = 101,1,Answer()
exten = 101,2,Wait(3)
exten = 101,3,ReceiveFAX(/var/spool/asterisk/tmp/fax.tiff)
exten = 101,4,System(tiff2pdf -p A4 /var/spool/asterisk/tmp/fax.tiff
/var/spool/asterisk/tmp/fax.pdf)
exten = 101,5,System(mutt -s 'New FAX for you sir' -a
/var/spool/asterisk/tmp/fax.pdf magnu...@inputinterior.se /dev/null)
I do receive the fax, the fax got converted to a pdf but 101,5 never get
executed, when i look in cli, last line is 101,4... can any1 se why?
2:nd example:
exten = 101,1,Answer()
exten = 101,2,Wait(3)
exten = 101,3,ReceiveFAX(/var/spool/asterisk/tmp/fax.tiff)
exten = 101,4,System(fax.sh)
cat /usr/bin/fax.sh
tiff2pdf -p A4 /var/spool/asterisk/tmp/fax.tiff
/var/spool/asterisk/tmp/fax.pdf
mutt -s 'New FAX for you sir' -a /var/spool/asterisk/tmp/fax.pdf
bo...@inputinterior.se /dev/null
That works, i receive the fax as an attachment, but as I asked before why is
not example 1 working?
SendFAX question:
exten = 101,1,Answer()
exten = 101,2,Wait(3)
exten = 101,3,ReceiveFAX(/var/spool/asterisk/tmp/fax.tiff)
exten = 101,4,Some magical way to setup the channel to: SIP/033211101
exten = 101,5,SendFAX(/var/spool/asterisk/tmp/fax.tiff)
033211101 is an ATA (SPA2102) registered to *.
I wonder if it is possible to do something like my example or not?
Any suggestions?
I was looking at:
http://www.evilspurv.net/blog/2010/01/sending-pdfs-as-fax-with-asterisk/
I could do something like that but i would prefer to have all in the
dialplan without need for an external program.
/Magnus
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Re: [asterisk-users] Faxing: Anyone have a compiled executable?
On a well set up system you should be able to send or receive those pages all day. If you can't, you probably have timing issues in your Asterisk setup. This is a uncleared question. What does timing issue exactly mean? 1) Enable internal timing and use one of the res_timing_*.so (with asterisk =1.6.2) 2) Does it mean, use only one of the res_timing_*.so and no internal timing ? 3) Or something completely different? Kristijan -- _ -- 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] Faxing: Anyone have a compiled executable?
yeah, but what about internal_timing = yes in asterisk.conf yes or no for faxing? Or is this option irrelevant for app_fax/spandsp ? 2010/1/8 William Stillwell (Lists) william.stillwell-li...@ablebody.net: In version prior to 1.6, timing is very critical for faxing, and the use of a timing source improves fax sending/receiving., and if no timing source was used, then you would use zt_dummy, but I am not sure how reliable that is or was.. And from what I am reading, v1.6 is far better with faxing, and I would assume the res_timing_*.so is an improved version of the later zt_dummy timing source. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kristijan Vrban Sent: Friday, January 08, 2010 4:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Faxing: Anyone have a compiled executable? On a well set up system you should be able to send or receive those pages all day. If you can't, you probably have timing issues in your Asterisk setup. This is a uncleared question. What does timing issue exactly mean? 1) Enable internal timing and use one of the res_timing_*.so (with asterisk =1.6.2) 2) Does it mean, use only one of the res_timing_*.so and no internal timing ? 3) Or something completely different? Kristijan -- _ -- 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 -- _ -- 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 -- _ -- 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 with gdb
super quick asterisk in gdb howto: compile asterisk with DONT_OPTIMIZE (in make menuconfig - Compiler flags) gdb asterisk run -cvv wait for the crash bt bt full and now make the patch :) Kristijan 2009/12/24 Tzafrir Cohen tzafrir.co...@xorcom.com On Thu, Dec 24, 2009 at 12:13:55PM +0530, Goyal, Amit wrote: Hi All, Can some help me with how to run Asterisk with gdb. What specifically do you want to do? What do you want to check? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- 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 ___ -- 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] spandsp version
magnus, simple answer: just use the latest version available. and if something is not working inside the t.30/t.38 protocol, try the latest spanpshot: http://www.soft-switch.org/downloads/snapshots/spandsp/?C=M;O=Dand if something i still not working, give a good description how to reproduce the problem. Kristijan ___ -- 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] Asterisk as Outbound Proxy ?
Hello, short question: is there a possibility to use asterisk as an outbound proxy? iam open for any suggestions, use asterisk trunk, dirty patches, ugly workarounds, everything. What is want to build is: SIP Phone - via TLS/SRTP - Asterisk as outbound proxy - via UDP/RTP - VoIP-Provider So Asterisk should just forward any incoming SIP messages (INVITE, REGISTER) to the VoIP-Provider and do SIP TLS- SIP UDP and SRTP - RTP translation (via *1.6.2 and the SRTP patch) Kristijan ___ -- 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] Digium Echo cancellation.
hallo Dhaval, to save money, you can use oslec as software EC in dahdi, then you dont need hardware echo cancellation. AFAIK oslec now in dahdi trunk and in the linux kernel. if you have problems with oslec, you can try the oslec mailing list: https://lists.sourceforge.net/lists/listinfo/freetel-oslec Kristijan Vrban 2009/8/27 DHAVAL INDRODIYA dhaval.it01...@gmail.com hi all, any one know, about echo cancellation with digium card, is it actually needed or it okay if we dont purchase because it increase price which half of new card, regards Dhaval ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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-user] Which policy for ISDN BRI support in NT/PtMP ?
hello, i made a experimental patch for libpri to have NT/PTMP mode, answers please on asterisk-dev at: http://lists.digium.com/pipermail/asterisk-dev/2009-May/038455.html Kristijan 2009/5/14 Kristijan Vrban vrban.l...@googlemail.com good news, i just made my isdn device ring! ok, after it ring, any timout then hangup up the chan, but a ringing from chan_dahdi via bri_net_ptmp - isdn_device was possible. to made this happen i made some very crude hacks inside libpri, but i hope the next days i can offer a patch that offer some basic nt_ptmp functionality. Stay tuned :) Kristijan 2009/5/12, Tzafrir Cohen tzafrir.co...@xorcom.com: On Tue, May 12, 2009 at 08:05:49AM +0200, Olivier wrote: 2009/5/12 Kristijan Vrban vrban.l...@googlemail.com For those also need NT over PtMP, i started a initial patch for it. Very limited at the moment, only one incoming call to chan_dahdi from one device is possible. But i was pleasantly surprised that NT-ptmp is working anyway Get the patch here: http://bugs.digium.com/view.php?id=15048 Or rather: works in one direction: calls from a phone to the NT work. Calls in the other way don't make it. I believe it's much better than nothing, though, and I'm testing this patch in the new debs I have. That is great news !!! How best can we contribute to make this happen ? Test this. Report how it (mis)behaves. And maybe try to trace why calls from the NT side don't get through. Will the output most probably be a new libpri 1.4.X (or 1.6.X) or will it also include a new Asterisk version ? For starters there will likely be some changes required in libpri . There is no libpri 1.6.x and not likely to be one in the near future. The trunk of libpri is branches/1.4 . -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- 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 ___ -- 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-user] Which policy for ISDN BRI support in NT/PtMP ?
good news, i just made my isdn device ring! ok, after it ring, any timout then hangup up the chan, but a ringing from chan_dahdi via bri_net_ptmp - isdn_device was possible. to made this happen i made some very crude hacks inside libpri, but i hope the next days i can offer a patch that offer some basic nt_ptmp functionality. Stay tuned :) Kristijan 2009/5/12, Tzafrir Cohen tzafrir.co...@xorcom.com: On Tue, May 12, 2009 at 08:05:49AM +0200, Olivier wrote: 2009/5/12 Kristijan Vrban vrban.l...@googlemail.com For those also need NT over PtMP, i started a initial patch for it. Very limited at the moment, only one incoming call to chan_dahdi from one device is possible. But i was pleasantly surprised that NT-ptmp is working anyway Get the patch here: http://bugs.digium.com/view.php?id=15048 Or rather: works in one direction: calls from a phone to the NT work. Calls in the other way don't make it. I believe it's much better than nothing, though, and I'm testing this patch in the new debs I have. That is great news !!! How best can we contribute to make this happen ? Test this. Report how it (mis)behaves. And maybe try to trace why calls from the NT side don't get through. Will the output most probably be a new libpri 1.4.X (or 1.6.X) or will it also include a new Asterisk version ? For starters there will likely be some changes required in libpri . There is no libpri 1.6.x and not likely to be one in the near future. The trunk of libpri is branches/1.4 . -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- 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 ___ -- 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-user] Which policy for ISDN BRI support in NT/PtMP ?
For those also need NT over PtMP, i started a initial patch for it. Very limited at the moment, only one incoming call to chan_dahdi from one device is possible. But i was pleasantly surprised that NT-ptmp is working anyway Get the patch here: http://bugs.digium.com/view.php?id=15048 Kristijan ___ -- 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
