Re: [asterisk-users] SIP provider registration attempts
--- On Tue, 2/23/10, Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote: Look at qualify= for sip.conf, and consider to extend your diaplan for a better routing decision with a snippet like this Actually, I noticed that setting qualify= alone solves my issue. I apparently don't require extra dialplan logic because if the peer is unreachable (according to qualify state) then I guess that Asterisk's Dial() immediately fails. Thanks, Vieri -- _ -- 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] Running safe_asterisk
Tilghman Lesher wrote: On Tuesday 23 February 2010 05:27:55 Per Jessen wrote: To be honest I don't remember any more, I just know my queueing doesn't work unless I reload. I think it's a timing issue at startup - that app_queue gets loaded too early or something. ah, here is my question about the same, but back in 2007: http://lists.digium.com/pipermail/asterisk-users/2007-May/188072.html You need to load the chan_local.so channel before pbx_config.so loads, so that your Local channels have the right devicestate. Adding 'preload = chan_local.so', followed by 'preload = pbx_config.so', to your /etc/asterisk/modules.conf should be sufficient. Thanks Tilghman - that works! I also added chan_sip.so. /Per Jessen, Zürich -- _ -- 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] directrtp with SIP + H.323
24 feb 2010 kl. 01.22 skrev Kristian Kielhofner: On Tue, Feb 23, 2010 at 8:22 AM, Michelle Dupuis supp...@ocg.ca wrote: We're creating a SIP gateway for a client that will take one leg of a call in via SIP, and out the other side via H.323. To minimize load on the gateway, we would like to have the RTP stream bypass the gatewayy altogether (directrtp/reinvite). Is this possible with these to protocols? Thanks Yate claims it can do this: http://yate.null.ro/pmwiki/index.php?n=Main.H323ToSIPSignallingProxy There are two ways - either by reinvites, which according to Kevin won't work with H323, or by doing it right in the call setup. If we did that, we would stumble into the same problem as we have with this function in SIP - which goes all back to the media negotiation framework (see http://www.voip-forum.com/asterisk/2010-02/asterisk-18-lts-wishlist-1-media-negiotiation-framework/ ). Asterisk currently just communicates an answered call as answered over the bridge without any attributes. This is the reason why the code has been marked experimental for many releases and no one has solved it. In order for this to work, you either need exactly the same codec attributes or a way to handle the ANSWER control frame (like John Martin did in the videocaps branch). The hooks are all there if you want to experiment with this in the H.323 channel. It's certainly possible. But it is not a function I would support generally (which is why the directrtp call setup function remains experimental). /O -- _ -- 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] Denying call transfer to certain extensions
Hi!
What I want is, if a call coming from a trunk 100 rings, and if the
caller wants to be transfered to 101, the transfer is denied. In other
words, 101 can't get transfered calls.
WHat about using featuresmap to replace the usual transfer application
with code that tests to see the origin of the cal ind if it is from the
100 do something else, otherwise transfer as expected.
Or look at the channel variables
${BLINDTRANSFER},
${GOTO_ON_BLINDXFR},
${TRANSFER_CONTEXT} and
${FORWARD_CONTEXT} as well as
${TRANSFERSTATUS}
as described in doc/channelvariables.txt.
Philipp
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Re: [asterisk-users] SIP provider registration attempts
Hi! Look at qualify= for sip.conf, and consider to extend your diaplan for a better routing decision with a snippet like this Actually, I noticed that setting qualify= alone solves my issue. I apparently don't require extra dialplan logic because if the peer is unreachable (according to qualify state) then I guess that Asterisk's Dial() immediately fails. That's right, but a) you might want to make a routing decision already before starting Dial() for a smoother handling of calls, and b) the extra code helps to differentiate the case when qualify says all is fine, yet the peer still cannot be reached for whatever reason. Asterisk will translate the SIP error code into a HANGUPCAUSE, and with that translation you loose a lot of information. Philipp -- _ -- 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] Looping over AstDB
Hello list, anybody has handy an example of how to loop over an ASTDB family by getting all the keys in the dialplan? Like I have the AstDB set as: /test/102 : 205 /test/106 : 203 /test/113 : 209 I would like to get (in any order) the 102, 106 and 113 as members of the family test. TIA, l. -- Loway - home of QueueMetrics - http://queuemetrics.com -- _ -- 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] Wrong MOH
Hello, I´m running Asterisk 1.6.1.11 . Ive got 2 classes in my musiconhold.conf: [general] [default] mode=files directory=/var/lib/asterisk/moh [signal] mode=files directory=/var/lib/asterisk/moh_signal I use this for 2 different queues and it works fine. When I call an IAX User directly and he puts me on hold I hear the [signal] sound files and not the [default] ones, but why? Any pointers please ? Thanks - Oliver -- Schrÿder Assistance und Consulting GmbH Lohdiecksweg 6 59457 Werl Fon 02922-8037-490 Fax 02922-8037-540 -- Sitz der Gesellschaft: Werl HRB 6747 Amtsgericht Arnsberg Geschÿftsfÿhrer: Lutz Schrÿder Steuernummer: 5343/5708/1703 -- -- _ -- 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] Manager Logged off
Hi People, I don't know if my problem should be reported in this forum, but maybe somebody knows about it. I'm using the tool .NET WebService Studio to test the web service which is working with asterisk by AMI. It is working fine, the dialplan is executed correctly... the problem is when the web service is consumed by my program (Genexus). I've been checking the log and the differences when I use .NET Webservice Studio and when I use my program... The difference I found is that the Manager is logged off and that is the reason why the dialplan is not executed fine in the second case. [Feb 24 08:08:20] DEBUG[1212] app_meetme.c: Cmdline: 7|k|1 [Feb 24 08:08:20] VERBOSE[1212] logger.c: == Manager 'asteriskWS' logged off from 67.63.42.120 The previous log shows that DEBUG line which is not shown in the log when I use the .NET Webservice Studio. Anybody knows why the Manager is logged off? Thanks, _ ¿Aún no sabes qué móvil eres? ¡Descúbrelo aquí! http://www.quemovileres.com/-- _ -- 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] Multiple instances of Asterisk on the same host...
On Tue, 23 Feb 2010, Tzafrir Cohen wrote: On Tue, Feb 23, 2010 at 12:19:31PM +, Gordon Henderson wrote: On Tue, 23 Feb 2010, Tzafrir Cohen wrote: But then again, lxc uses much of the work on containers done also by and for OpenVZ. Sort of like the VMWare/Xen/KVM story all over again, with lxc playing the role of KVM. And LXC got into the kernel before the others - what that means is anyones guess - probably because it was sponsored/written by IBM? KVM was not sponsored by any big-name company (RedHat, IBM and the others had their bets on Xen, at the time). Indeed, so it just goes to show.. LXC is growing on me - I like that I can see the underlying 'container' root directory from the host - makes it trivial to distribute config files, updates, etc. Also takes up minimal disk space as each container can be hard links to a 'base' container with the exception of a file files in /var/ that get written to. Update the base container and all the others update automatically... Asterisk seems to run OK too, as does dahdi_dummy.. Just ran up 4 containers and 4 dahdi_tests, one in each container and they're all reporting the same (99.997, etc.) (And this is on my test box with 256MB of RAM in it) Will place some test calls through it later on. I reckon I can get about 8 contaners going on this box... A crude plan is to get container 0 to call c1, then to call c2, then ... and c7 does echo() and then point sipp at it and see what happens - if it can survive a number of calls on that hardware, (1.8Hz Celeron) I'll be more than happy on a proper server... One thing that does fail is asterisk -p - well, it doesn't fail, just prints Unable to set high priority, so that's possibly something that the container isn't allowing - more research required. Gordon -- _ -- 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] voip host in israel
hi, do u know a good, inexpensive hosting company in israel, that will host voip? i want to have my asterisk server here, in the u.s., to hook up to a voip host in israel. most traffic would be to israel. would prefer one base rate to any landline location in israel. and, something reasonable for cell phones. thanks in advance. g. Date: Tue, 23 Feb 2010 12:55:37 +0200 From: tzafrir.co...@xorcom.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Multiple instances of Asterisk on the same host... On Mon, Feb 22, 2010 at 11:23:29PM +, Gordon Henderson wrote: On Mon, 22 Feb 2010, Roderick A. Anderson wrote: Gordon Henderson wrote: Interesting thread recently about virtual servers... I'm thinking of doing something similar - right now looking at Containers (lxc) rather than proper virtualisation though, however it got me thinking of a poor mans virtualisation solution... This would assume you have a real server to start with and full root access... I was thinking of simply running multiple asterisks on the same box, each with their own /etc/asterisk config directory (in e.g. /home/v1/etc/asterisk, /home/v2/etc/asterisk and so on - obviously give them unique /home/v1/spool/asterisk/ , etc. directories too, but for the most part things like /var/lib/asterisk/sounds and modules can be shared. (exception being astdb!) It just means a custom /etc/asterisk/asterisk.conf file for each instance and asterisk being started with the correct config file - /home/v1/etc/asterisk.conf, etc. So giving each asterisk it's own IP address (eth0:1, eth0:2, etc.) and changing the bindaddr parameter in each one to suit multiple IP addresses bound to the 'host' would seem to be the way to do it - each asterisk can still use ztdummy/dhadidummy for timing if required (or does it stop multiple asterisks opening it?) Anyone done this or contemplated doing it? I have heard of a company, name completely escapes me right now, that appears to use Linux-Vserver. I am trying to find the time to move my business system to a Linux-Vserver from a Micro-Linux Asterisk Server and the only issue I'm aware of is DAHDI/ZAPTEL might have to be run in the host instead of the guests. Then some permissions set so the guests can access it DAHDI. My aim is to actually use LXC as it has kernel level support (as of 2.6.29) and will be supported by most distros soon if not already. Linux-Vserver appears to be depreciated by at least Debian, probably Ubuntu too, but I've no idea about the world of Red Hat/Fedora/Centos, etc.. I tried OpenVZ, but it seems to have even poorer support, and no updated for some time either. Actually: Linux-VServer is deprecated much in favour of OpenVZ. The OpenVZ developers have been much more willing to work with the upstream kernel maintainers. But then again, lxc uses much of the work on containers done also by and for OpenVZ. Sort of like the VMWare/Xen/KVM story all over again, with lxc playing the role of KVM. -- 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 _ Hotmail: Trusted email with Microsoft’s powerful SPAM protection. http://clk.atdmt.com/GBL/go/201469226/direct/01/-- _ -- 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] subject: 1.4 vs 1.6
Hi Guys We are using asterisk 1.4 on all of our platforms for a while now. Some of our partners recommended to use asterisk 1.6 in order to improve overall stability and performance. Can someone please let me know if you have a such experience? Also, do you have any other negative or positive comments on 1.6 Very much thanking you for your help!!! Juan _ Hotmail: Powerful Free email with security by Microsoft. https://signup.live.com/signup.aspx?id=60969-- _ -- 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] IAX devices not registering after upgrade to
On Tue, Feb 23, 2010 at 09:28:23PM +0200, Rudi Oosthuizen wrote: Hi All, We have encountering issue that IAX enable voice gateways not registering with asterisk after upgrade from asterisk 1.4.18.1 - 1.4.29 Before that IAX works very well. If any one have similar issue and solution for that let me know. Check for ERROR[] chan_iax2.c: Call rejected, CallToken Support required. If add requirecalltoken=no to Iax trunk. http://downloads.asterisk.org/pub/security/AST-2009-006.html http://downloads.asterisk.org/pub/security/IAX2-security.html -- 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
Re: [asterisk-users] IAX devices not registering after upgrade to
Tzafrir Cohen wrote: http://downloads.asterisk.org/pub/security/AST-2009-006.html http://downloads.asterisk.org/pub/security/IAX2-security.html And more importantly, the UPGRADE files included in the source code that the OP downloaded pointed to all of this stuff. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- 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] subject: 1.4 vs 1.6
On Wed, Feb 24, 2010 at 9:20 AM, Juan Sandro juan.san...@hotmail.com wrote: Hi Guys We are using asterisk 1.4 on all of our platforms for a while now. Some of our partners recommended to use asterisk 1.6 in order to improve overall stability and performance. Can someone please let me know if you have a such experience? Also, do you have any other negative or positive comments on 1.6 If it isn't broke, don't 'fix' it. There are benefits to 1.6, like dramatically enhanced SIP support, much faster dialplan processing, easier faxing, changes to dialplan syntax, and lots of other features. I would say the improvement of going to 1.6 is only if you are trying to expect more from the same gear, or want the new features. If you're not actually having problems, don't change anything. -- _ -- 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] subject: 1.4 vs 1.6
Well.. we do from time to time have SIP attacks, Core dumps and lately very weird issues with Cisco phone becoming unreachable. Anyone had issues with Cisco 7940 where by ALL of the phones will for 30-90 seconds become unreachable? All phones are on T1 MPLS network using Cisco 26xx routers.. Juan Date: Wed, 24 Feb 2010 09:56:50 -0500 From: dbackeb...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] subject: 1.4 vs 1.6 On Wed, Feb 24, 2010 at 9:20 AM, Juan Sandro juan.san...@hotmail.com wrote: Hi Guys We are using asterisk 1.4 on all of our platforms for a while now. Some of our partners recommended to use asterisk 1.6 in order to improve overall stability and performance. Can someone please let me know if you have a such experience? Also, do you have any other negative or positive comments on 1.6 If it isn't broke, don't 'fix' it. There are benefits to 1.6, like dramatically enhanced SIP support, much faster dialplan processing, easier faxing, changes to dialplan syntax, and lots of other features. I would say the improvement of going to 1.6 is only if you are trying to expect more from the same gear, or want the new features. If you're not actually having problems, don't change anything. -- _ -- 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 _ Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. https://signup.live.com/signup.aspx?id=60969-- _ -- 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] subject: 1.4 vs 1.6
Wednesday, February 24, 2010, 3:56:50 PM, David wrote: On Wed, Feb 24, 2010 at 9:20 AM, Juan Sandro juan.san...@hotmail.com wrote: Hi Guys We are using asterisk 1.4 on all of our platforms for a while now. Some of our partners recommended to use asterisk 1.6 in order to improve overall stability and performance. Can someone please let me know if you have a such experience? Also, do you have any other negative or positive comments on 1.6 If it isn't broke, don't 'fix' it. There are benefits to 1.6, like dramatically enhanced SIP support, much faster dialplan processing, easier faxing, changes to dialplan syntax, and lots of other features. I would say the improvement of going to 1.6 is only if you are trying to expect more from the same gear, or want the new features. If you're not actually having problems, don't change anything. Yes, and check this page: http://www.asterisk.org/asterisk-versions as you can see, the 1.4 version is LTS, and the 1.6 isn't, but the upcoming 1.8 will be LTS too. So don't change to 1.6 :) -- Best regards, Gergomailto:csi...@gmail.com -- _ -- 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] Problems in Asterisk Real Time (Urgent help )
Hello, Asterisk Real time database worked on astersik 1.6.2.0 but now i am working on Asterisk to latest version which is 1.6.2.2 ,there is a a warning [Feb 24 16:26:14] WARNING[4053]: config.c:2025 find_engine: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available [Feb 24 16:26:14] NOTICE[4053]: chan_sip.c:21500 handle_request_register: Registration from '555sip:5...@192.168.50.109 sip%3a...@192.168.50.109' failed for '192.168.50.105' - No matching peer found is there a problem in version compatability? if anyone knows anything ,help me please. -- Ahmed Magdy Mahmoud -- _ -- 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] Problems in Asterisk Real Time (Urgent help )
-BEGIN PGP SIGNED MESSAGE- Hash: SHA512 Hi It might seem that you have installed 1.6.2.2 over 1.6.2.0, but not updated the correct modules. If you - from the CLI - do a mode show like sql. What is your result? - - Tommy ahmed magdy skrev: Hello, Asterisk Real time database worked on astersik 1.6.2.0 but now i am working on Asterisk to latest version which is 1.6.2.2 ,there is a a warning [Feb 24 16:26:14] WARNING[4053]: config.c:2025 find_engine: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available [Feb 24 16:26:14] NOTICE[4053]: chan_sip.c:21500 handle_request_register: Registration from '555sip:5...@192.168.50.109 mailto:sip%3a...@192.168.50.109' failed for '192.168.50.105' - No matching peer found is there a problem in version compatability? if anyone knows anything ,help me please. -- Ahmed Magdy Mahmoud -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEAREKAAYFAkuFSfIACgkQ573V05EH/pZ/UACeJM3gj0TA6ckbrKB8mvY1A76+ NKEAn3wP9+ba51Tr1Gvq87M/z4U8MSiD =ns4Z -END PGP SIGNATURE- -- _ -- 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] Re-INVITE on BYE
Hi gurus, In need of a little help here. I¹m trying to do the Asterisk media release by using canreinvite=yes. But I found weird behaviour when comes to BYE. Below are my current setup: Client A is registered to Opensips Client B is registered to Asterisk A Opensips Asterisk B On hangup below are the SIP flow which I¹ve notice from the Asterisk server itself: 1. Opensips forward the BYE to Asterisk 2. Asterisk response with 200 OK 3. Asterisk send INVITE to B 4. B response with 200 OK with SDP 5. Asterisk reply with ACK 6. Asterisk send BYE to B 7. B response with 200 OK Shouldn¹t Asterisk forward the BYE to B instead of issuing a re-INVITE then BYE? P.s: I¹ve also attached the traces. Regards, Lawrence sip:+60121110...@211.24.134.120 SIP/2.0 Record-Route: sip:211.24.134.121;lr;ftag=a0a48613 Via: SIP/2.0/UDP 211.24.134.121;branch=z9hG4bKe46a.7367b334.0 Via: SIP/2.0/UDP 211.24.134.122:5062;received=211.24.134.122;branch=z9hG4bK-d8754z-917805ee170c25a5-1---d8754z-;rport=5062 Max-Forwards: 69 Contact: sip:6013...@211.24.134.122:5062;transport=UDP To: sip:60121110...@211.24.134.121;transport=UDP;tag=as4483b7b3 From: 601sip:6013...@211.24.134.121;transport=UDP;tag=a0a48613 Call-ID: OWFkNDhlOTBkMDg2NDkxNzEwZTc1M2JlYjNmYmI5Yzk. CSeq: 2 BYE User-Agent: Zoiper rev.5528 Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/UDP 211.24.134.121;branch=z9hG4bKe46a.7367b334.0;received=211.24.134.121 Via: SIP/2.0/UDP 211.24.134.122:5062;received=211.24.134.122;branch=z9hG4bK-d8754z-917805ee170c25a5-1---d8754z-;rport=5062 Record-Route: sip:211.24.134.121;lr;ftag=a0a48613 From: 601sip:6013...@211.24.134.121;transport=UDP;tag=a0a48613 To: sip:60121110...@211.24.134.121;transport=UDP;tag=as4483b7b3 Call-ID: OWFkNDhlOTBkMDg2NDkxNzEwZTc1M2JlYjNmYmI5Yzk. CSeq: 2 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 INVITE sip:lawre...@211.24.134.123:1187;rinstance=f878778da25aff09;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 211.24.134.120:5060;branch=z9hG4bK23e035f3;rport From: 601 sip:6013...@211.24.134.120;tag=as307ae54b To: sip:lawre...@211.24.134.123:1187;rinstance=f878778da25aff09;transport=UDP;tag=38208e1d Contact: sip:6013...@211.24.134.120 Call-ID: 4f71da4e74e598224436fda203d36...@211.24.134.120 CSeq: 104 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 244 v=0 o=root 11515 11517 IN IP4 211.24.134.120 s=session c=IN IP4 211.24.134.120 t=0 0 m=audio 10036 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv SIP/2.0 200 OK Via: SIP/2.0/UDP 211.24.134.123:1187;branch=z9hG4bK23e035f3;rport=5060 Contact: sip:lawre...@211.24.134.123:1187;rinstance=f878778da25aff09;transport=UDP To: sip:lawre...@211.24.134.123:1187;rinstance=f878778da25aff09;transport=UDP;tag=38208e1d From: 601sip:6013...@211.24.134.120:5060;tag=as307ae54b Call-ID: 4f71da4e74e598224436fda203d36...@211.24.134.120 CSeq: 104 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp User-Agent: Zoiper rev.5324 Content-Length: 254 v=0 o=Zoiper_user 0 0 IN IP4 211.24.134.123 s=Zoiper_user c=IN IP4 211.24.134.123 t=0 0 m=audio 8000 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv ACK sip:lawre...@211.24.134.123:1187;rinstance=f878778da25aff09;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 211.24.134.120:5060;branch=z9hG4bK57b4af45;rport From: 601 sip:6013...@211.24.134.120;tag=as307ae54b To: sip:lawre...@211.24.134.123:1187;rinstance=f878778da25aff09;transport=UDP;tag=38208e1d Contact: sip:6013...@211.24.134.120 Call-ID: 4f71da4e74e598224436fda203d36...@211.24.134.120 CSeq: 104 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 BYE sip:lawre...@211.24.134.123:1187;rinstance=f878778da25aff09;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 211.24.134.120:5060;branch=z9hG4bK5756d79d;rport From: 601 sip:6013...@211.24.134.120;tag=as307ae54b To: sip:lawre...@211.24.134.123:1187;rinstance=f878778da25aff09;transport=UDP;tag=38208e1d Call-ID: 4f71da4e74e598224436fda203d36...@211.24.134.120 CSeq: 105 BYE User-Agent: Asterisk PBX Max-Forwards: 70 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/UDP 211.24.134.123:1187;branch=z9hG4bK5756d79d;rport=5060 Contact: sip:lawre...@211.24.134.123:1187;rinstance=f878778da25aff09;transport=UDP To: sip:lawre...@211.24.134.123:1187;rinstance=f878778da25aff09;transport=UDP;tag=38208e1d From: 601sip:6013...@211.24.134.120:5060;tag=as307ae54b Call-ID: 4f71da4e74e598224436fda203d36...@211.24.134.120 CSeq: 105 BYE
Re: [asterisk-users] Problems in Asterisk Real Time (Urgent help )
Have you check if MySql is already running? Have you check HD space? regards. 2010/2/24 ahmed magdy amagdy.ibra...@gmail.com Hello, Asterisk Real time database worked on astersik 1.6.2.0 but now i am working on Asterisk to latest version which is 1.6.2.2 ,there is a a warning [Feb 24 16:26:14] WARNING[4053]: config.c:2025 find_engine: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available [Feb 24 16:26:14] NOTICE[4053]: chan_sip.c:21500 handle_request_register: Registration from '555sip:5...@192.168.50.109 sip%3a...@192.168.50.109' failed for '192.168.50.105' - No matching peer found is there a problem in version compatability? if anyone knows anything ,help me please. -- Ahmed Magdy Mahmoud -- _ -- 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 -- Juan. -- _ -- 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] subject: 1.4 vs 1.6
Gergo Csibra escribió: Wednesday, February 24, 2010, 3:56:50 PM, David wrote: On Wed, Feb 24, 2010 at 9:20 AM, Juan Sandro juan.san...@hotmail.com wrote: Hi Guys We are using asterisk 1.4 on all of our platforms for a while now. Some of our partners recommended to use asterisk 1.6 in order to improve overall stability and performance. Can someone please let me know if you have a such experience? Also, do you have any other negative or positive comments on 1.6 If it isn't broke, don't 'fix' it. There are benefits to 1.6, like dramatically enhanced SIP support, much faster dialplan processing, easier faxing, changes to dialplan syntax, and lots of other features. I would say the improvement of going to 1.6 is only if you are trying to expect more from the same gear, or want the new features. If you're not actually having problems, don't change anything. Yes, and check this page: http://www.asterisk.org/asterisk-versions as you can see, the 1.4 version is LTS, and the 1.6 isn't, but the upcoming 1.8 will be LTS too. So don't change to 1.6 :) That sounds reasonable, but as I have seen through several years following the asterisk project, when 1.8.0 will be released it will be far less stable than the more used and mature 1.6.0.X, for example. I would prefer to do a middle step for upgrading, that would be 1.4.X - 1.6.0.X - 1.8.X when it becomes really stable. Asterisk history has shown us that a newly released branch, no matter if it's LTS on the new release schema, will need time and a large user base that adopts it to report bugs and help stabilize it. I would not underestimate the actual 1.6.X branches. Just IMHO, any opinions welcome. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center -- _ -- 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] Re-INVITE on BYE
Lawrence Na (my-...@vyke) wrote: On hangup below are the SIP flow which I’ve notice from the Asterisk server itself: 1. Opensips forward the BYE to Asterisk 2. Asterisk response with 200 OK 3. Asterisk send INVITE to B 4. B response with 200 OK with SDP 5. Asterisk reply with ACK 6. Asterisk send BYE to B 7. B response with 200 OK Shouldn’t Asterisk forward the BYE to B instead of issuing a re-INVITE then BYE? Asterisk does not 'forward' messages or requests, since it is not a proxy. In this case, it is redirecting B's media back to itself in case the dialplan contains any steps to be done with B's channel before it is destroyed. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- 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] identify the costumer
Hi People, I work in a company that are using asterisk as pbx. I need a way to identify what client my employees are calling. For example: - For each call that an employee of my company make to a customer, must identify the client name in the CDR table. - Is there a way of my employee enter a code to identify the client and then enter a phone number to make the call? I would like to identify the customer's name in the table CDR. Thanks, Douglas -- _ -- 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] identify the costumer
Douglas Pasqua wrote: Hi People, I work in a company that are using asterisk as pbx. I need a way to identify what client my employees are calling. For example: - For each call that an employee of my company make to a customer, must identify the client name in the CDR table. - Is there a way of my employee enter a code to identify the client and then enter a phone number to make the call? I would like to identify the customer's name in the table CDR. throw the information into a database and lookup the numbers before displaying the output with some other app. once they are in the db great but to build that up you could use any of the reverse lookup services and then just ask whoever made the calls to fill in whatever else is missing until you are up to a high hit rate of matches in your database. Have a procedure to enter new numbers for new clients. if you use something like sugar, just use the data in there to do your lookups since that is likely where the staff are looking up the numbers to make the calls in the first place. Thanks, Douglas -- _ -- 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] identify the costumer
-BEGIN PGP SIGNED MESSAGE- Hash: SHA512 Douglas Pasqua skrev: I need a way to identify what client my employees are calling. For example: - For each call that an employee of my company make to a customer, must identify the client name in the CDR table. - Is there a way of my employee enter a code to identify the client and then enter a phone number to make the call? I would like to identify the customer's name in the table CDR. The easiest way of doing this is probably to have a separate database table with the customer names and phone numbers, and create a view that has the phone number as a key for both tables. This of course, depends on you having CDRs stored in a database, as well as up to date customer records. - - Tommy -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEAREKAAYFAkuFWAIACgkQ573V05EH/pYkvACeIVdlokIMk3Mnve7virfqRUsY 8FQAmQFHdoZXHKPX2J2QfKj0p+BcI0Dg =PZvI -END PGP SIGNATURE- -- _ -- 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] identify the costumer
Why not set your clients up as extensions so your employee's call them
with an extension code instead of dialing a number? For example
Exten = 1001,1,Dial(DAHDI/g1/18005551212)
Exten = 1002,1,Dial(DAHDI/g1/18005551213)
Exten = 1003,1,Dial(DAHDI/g1/18005551214)
Or more efficiently
Exten = _1xxx,1,Set(custid=${DB(${EXTEN})})
Exten = _1xxx,n,Dial(DAHDI/g1/${custid})
Solution 2 could be modified to use MYSQL, I just use ASTDB bc I don't care
for MYSQL.
In users.conf, set up the client companies as users like
[1001]
Username=abc widgets
[1002]
Username=att
[1003]
Username=IBM
Just a thought
Danny Nicholas
--
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From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Douglas Pasqua
Sent: Wednesday, February 24, 2010 10:36 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] identify the costumer
Hi People,
I work in a company that are using asterisk as pbx.
I need a way to identify what client my employees are calling. For example:
- For each call that an employee of my company make to a customer, must
identify the client name in the CDR table.
- Is there a way of my employee enter a code to identify the client and then
enter a phone number to make the call? I would like to identify the
customer's name in the table
CDR.
Thanks,
Douglas
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Re: [asterisk-users] identify the costumer
Op 24-02-10 17:35, Douglas Pasqua schreef: Hi People, I work in a company that are using asterisk as pbx. I need a way to identify what client my employees are calling. For example: - For each call that an employee of my company make to a customer, must identify the client name in the CDR table. - Is there a way of my employee enter a code to identify the client and then enter a phone number to make the call? I would like to identify the customer's name in the table CDR. SNOM phones have a CMC soft function key. When enabled, this allows you to enter a customer ID during a call. This is sent in a SIP info message to asterisk, and subsequently copied to the CDR userfield. Ron Thanks, Douglas -- NeoNova BV innovatieve internetoplossingen http://www.neonova.nl Science Park 140 1098 XG Amsterdam info: 020-5611300 servicedesk: 020-5611302 fax: 020-5611301 KvK Amsterdam 34151241 Op dit bericht is de volgende disclaimer van toepassing: http://www.neonova.nl/maildisclaimer -- _ -- 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] Virtual machine timing (KVM)
FWIW, we recently moved a 1.4.29 Asterisk system onto a VMWare guest machine and with 40+ call legs (20+ calls), it isn't even breaking a sweat. We have had no complaints from users nor have we noticed any degradation in voice quality, be it live, voicemail or conference bridge (with six participants). The underlying hardware is an HP ProLiant DL360 G5 (Xeon 5160 3gz, 2 cores) with 20gb of memory and the VMWare version is ESXi 4. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sean Brady Sent: Friday, February 19, 2010 12:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Virtual machine timing (KVM) To get MeetMe working properly, I know some sort of timing device provided by the zaptel package is required (even if it means the zt_dummy). But, on a virtual machine I know that the Linux timing won't work as expected. Is it possible to then dedicate a physical device like a USB port or something to the virtual machine to use for the timing interrupts? The 2.2.1 version of DAHDI using DAHDI dummy seems to be working adequately in a Xen environment on CentOS for me, although I haven't been using MeetMe. Have you run into issues with it specifically? Which version of DAHDI are you using? If there are some issues that you have found I would like to know... Thanks, Sean -- _ -- 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
[asterisk-users] AMD: HANGUP
*Code:*
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing Playback(Local/91441425477...@default-b9f2,1,
sip-silence) in new stack
-- Playing 'sip-silence' (language 'en')
-- Executing AGI(Local/91441425477...@default-b9f2,1, agi://
127.0.0.1:4577/call_log) in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing AMD(Local/91441425477...@default-b9f2,1,
2000|2000|1000|5000|120|50|4|256) in new stack
-- AMD: Local/91441425477...@default-b9f2,1 00 (null) (Fmt: 64)
-- AMD: initialSilence [2000] greeting [2000] afterGreetingSilence
[1000] totalAnalysisTime [5000] minimumWordLength [120] betweenWordsSilence
[50] maximumNumberOfWords [4] silenceThreshold [256]
== Spawn extension (default, 91441425477375, 2) exited non-zero on
'Local/91441425477...@default-1e22,2'
-- Executing DeadAGI(Local/91441425477...@default-1e22,2, agi://
127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-16-ANSWER-35-15)
in new stack
-- AGI Script agi://
127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-16-ANSWER-35-15completed,
returning 0
-- AMD: HANGUP
-- Executing DeadAGI(Local/91441425477...@default-1e22,1, agi://
127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-0---) in
new stack
-- AGI Script agi://
127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-0---completed,
returning 0
== Spawn extension (default, 91441425477388, 2) exited non-zero on
'Local/91441425477...@default-86e4,2'
-- Executing DeadAGI(Local/91441425477...@default-86e4,2, agi://
127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-16-ANSWER-41-15)
in new stack
-- AGI Script agi://
127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-16-ANSWER-41-15completed,
returning 0
-- AMD: HANGUP
-- Executing DeadAGI(Local/91441425477...@default-86e4,1, agi://
127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-0---) in
new stack
-- AGI Script agi://
127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-0---completed,
returning 0
vici*CLI
My agent are NOT getting calls.
-- AMD: HANGUP ??
Is that an Issue ?
How to solve it ?
I have below entry for 8369 :
*Code:*
; VICIDIAL_auto_dialer transfer script AMD with Load Balanced:
exten = 8369,1,Playback(sip-silence)
exten = 8369,2,AGI(agi://127.0.0.1:4577/call_log)
exten = 8369,3,AMD(2000|2000|1000|5000|120|50|4|256)
exten = 8369,4,AGI(VD_amd.agi,${EXTEN})
exten = 8369,5,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-LB)
exten = 8369,6,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-LB)
exten = 8369,7,Hangup
Amd.conf has :
*Code:*
; initial_silence: Maximum silence duration before the greeting. If exceeded
then MACHINE.
; greeting: Maximum length of a greeting. If exceeded then MACHINE.
; after_greeting_silence: Silence after detecting a greeting. If exceeded
then HUMAN
; total_analysis_time: Maximum time allowed for the algorithm to decide on a
HUMAN or PERSON
; min_word_length: Minimum duration of Voice to considered as a word
; between_words_silence: Minimum duration of silence after a word to
considere the audio what follows as a new word
; maximum_number_of_words: Maximum number of words in the greeting. If
exceeded then MACHINE
[AnsweringMachineDetector]
initial_silence= 3500
greeting = 1500
after_greeting_silence = 300
total_analysis_time= 5000
min_word_length= 120
between_words_silence = 50
maximum_number_of_words= 5
silence_threshold = 256
--
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Re: [asterisk-users] subject: 1.4 vs 1.6
On Wednesday 24 February 2010 10:16:25 Miguel Molina wrote: Gergo Csibra escribió: Wednesday, February 24, 2010, 3:56:50 PM, David wrote: On Wed, Feb 24, 2010 at 9:20 AM, Juan Sandro juan.san...@hotmail.com wrote: Hi Guys We are using asterisk 1.4 on all of our platforms for a while now. Some of our partners recommended to use asterisk 1.6 in order to improve overall stability and performance. Can someone please let me know if you have a such experience? Also, do you have any other negative or positive comments on 1.6 If it isn't broke, don't 'fix' it. There are benefits to 1.6, like dramatically enhanced SIP support, much faster dialplan processing, easier faxing, changes to dialplan syntax, and lots of other features. I would say the improvement of going to 1.6 is only if you are trying to expect more from the same gear, or want the new features. If you're not actually having problems, don't change anything. Yes, and check this page: http://www.asterisk.org/asterisk-versions as you can see, the 1.4 version is LTS, and the 1.6 isn't, but the upcoming 1.8 will be LTS too. So don't change to 1.6 :) That sounds reasonable, but as I have seen through several years following the asterisk project, when 1.8.0 will be released it will be far less stable than the more used and mature 1.6.0.X, for example. I would prefer to do a middle step for upgrading, that would be 1.4.X - 1.6.0.X - 1.8.X when it becomes really stable. Asterisk history has shown us that a newly released branch, no matter if it's LTS on the new release schema, will need time and a large user base that adopts it to report bugs and help stabilize it. I would not underestimate the actual 1.6.X branches. Additionally, it's worth noting that the dates above are meant to be the EARLIEST dates that development, security fixes, etc. will end. It is quite possible that we will elect to extend some of them. The whole idea is to give companies advance notice of at least six months before we stop supporting a release. The end is coming; but it might be delayed. :-) -- Tilghman -- _ -- 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] AMD: HANGUP
It looks like your channel has been hungup during the AMD application,
not that the AMD application is hanging up the call. The source is your
friend (http://www.asterisk.org/doxygen/asterisk1.4/app__amd_8c.html):
00205 /* If we fail to read in a frame, that means they hung up */
00206 if (!(f = ast_read
http://www.asterisk.org/doxygen/asterisk1.4/channel_8c.html#7ef6737309dc9e8b6c4a7cb4800638b1(chan)))
{
00207 if (option_verbose
http://www.asterisk.org/doxygen/asterisk1.4/group__main__options.html#ga294d0efa6a89c1a3d162787cac4fff5
2)
00208 ast_verbose
http://www.asterisk.org/doxygen/asterisk1.4/logger_8c.html#81d26348827b996085d4cb6be3e2c348(VERBOSE_PREFIX_3
http://www.asterisk.org/doxygen/asterisk1.4/logger_8h.html#24b0f46e22f4ea3226fa082e955dd4ef
AMD: HANGUP\n);
00209 if (option_debug
http://www.asterisk.org/doxygen/asterisk1.4/group__main__options.html#g40f8fb2e731031d99f732f515cec680f)
00210 ast_log
http://www.asterisk.org/doxygen/asterisk1.4/logger_8c.html#93dd824dff97fe84941d6d71b7a3710e(LOG_DEBUG
http://www.asterisk.org/doxygen/asterisk1.4/logger_8h.html#6ff63e8955665c4a58b1598f2b07c51a,
Got hangup\n);
00211 strcpy(amdStatus, HANGUP);
00212 break;
00213 }
So basically check that the channel is not being hungup during
application execution.
Regards,
--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center
David @ULC escribió:
*Code:*
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing Playback(Local/91441425477...@default-b9f2,1,
sip-silence) in new stack
-- Playing 'sip-silence' (language 'en')
-- Executing AGI(Local/91441425477...@default-b9f2,1,
agi://127.0.0.1:4577/call_log http://127.0.0.1:4577/call_log) in
new stack
-- AGI Script agi://127.0.0.1:4577/call_log
http://127.0.0.1:4577/call_log completed, returning 0
-- Executing AMD(Local/91441425477...@default-b9f2,1,
2000|2000|1000|5000|120|50|4|256) in new stack
-- AMD: Local/91441425477...@default-b9f2,1 00 (null)
(Fmt: 64)
-- AMD: initialSilence [2000] greeting [2000] afterGreetingSilence
[1000] totalAnalysisTime [5000] minimumWordLength [120]
betweenWordsSilence [50] maximumNumberOfWords [4] silenceThreshold [256]
== Spawn extension (default, 91441425477375, 2) exited non-zero on
'Local/91441425477...@default-1e22,2'
-- Executing DeadAGI(Local/91441425477...@default-1e22,2,
agi://127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-16-ANSWER-35-15
http://127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-16-ANSWER-35-15)
in new stack
-- AGI Script
agi://127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-16-ANSWER-35-15
http://127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-16-ANSWER-35-15
completed, returning 0
-- AMD: HANGUP
-- Executing DeadAGI(Local/91441425477...@default-1e22,1,
agi://127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-0---
http://127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-0---)
in new stack
-- AGI Script
agi://127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-0---
http://127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-0---
completed, returning 0
== Spawn extension (default, 91441425477388, 2) exited non-zero on
'Local/91441425477...@default-86e4,2'
-- Executing DeadAGI(Local/91441425477...@default-86e4,2,
agi://127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-16-ANSWER-41-15
http://127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-16-ANSWER-41-15)
in new stack
-- AGI Script
agi://127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-16-ANSWER-41-15
http://127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-16-ANSWER-41-15
completed, returning 0
-- AMD: HANGUP
-- Executing DeadAGI(Local/91441425477...@default-86e4,1,
agi://127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-0---
http://127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-0---)
in new stack
-- AGI Script
agi://127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-0---
http://127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-0---
completed, returning 0
vici*CLI
My agent are NOT getting calls.
-- AMD: HANGUP ??
Is that an Issue ?
How to solve it ?
I have below entry for 8369 :
*Code:*
; VICIDIAL_auto_dialer transfer script AMD with Load Balanced:
exten = 8369,1,Playback(sip-silence)
exten = 8369,2,AGI(agi://127.0.0.1:4577/call_log
http://127.0.0.1:4577/call_log)
exten = 8369,3,AMD(2000|2000|1000|5000|120|50|4|256)
exten = 8369,4,AGI(VD_amd.agi,${EXTEN})
exten = 8369,5,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-LB)
exten = 8369,6,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-LB)
exten = 8369,7,Hangup
Amd.conf has :
*Code:*
Re: [asterisk-users] Which H.323 to use in Ast 1.6
I have always used ooh323 between Avaya and Asterisk. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis Sent: Tuesday, February 23, 2010 2:24 PM To: 'Asterisk Users List' Subject: [asterisk-users] Which H.323 to use in Ast 1.6 We're doing a project that requires H.323 to an Avaya. Does anyone have experience to share on which H.323 driver to use in asterisk 1.6? Is the diference between h323 and ooh323 still worth the extra effort? (We've only installed h323 under 1.4) If you have setup/config experience with this setup in Asterisk 1.6 please share! Thanks, MD -- _ -- 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] Encrypted calls between mobile gsm and isdn (asterisk)
Hi All, are you aware of any solution which can encrypt calls between a mobile gsm and isdn (asterisk) ? Thanks for your attention, have a nice day. Mike -- _ -- 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] Question
Ok so a while back I found an example for having a number dial multiple
numbers and then whoever answers and confirms gets the call. (don't
recall who the example was from, but thank you!)
But Now today I've been playing with TTS and STT and came across the
BackgroundDetect command. Now If I use this allow it works fine. But
when I try and use it with this it never actually detects me talking -
or if it does it doesn't connect the caller so that the Wait time
expires and it goes on.
So my question is how can I make this work to where you can talk and it
will connect you to the caller or press 1. Not now where you just press
1. Which a lot of the time I can't get my phone out of my pocket,
unlocked, and press 1 before it is sent to VM
[default]
exten =
_XX,1,Monitor(wav,/var/store/calls/PersonalLine-${STRFTIME($
{EPOCH},,%Y%m%d-%H%M%S)}-${CALLERID(num)}-${EXTEN},mb)
exten = _XX,2,dial(${bellbu}/${EXTEN:4},40,rM(screen)) ;
without r it seems to pass a second or two of audio first
exten = _XX,4,Hangup ; You can also substitute this with a
Voicemail destination or other alternative destination
[macro-screen]
;exten = s,1,Wait(1)
;exten = s,n,Background(/var/lib/asterisk/sounds/press1) ; substitute a
different playback file if you need to
;exten = s,n,WaitExten(5) ; the value is the Wait time before we assume
the call is not accepted
;exten = 1,1,NoOp(Caller accepted) ; Do not set MACRO_RESULT to
anything to connect the caller
;exten = i,1,Set(MACRO_RESULT=CONTINUE)
;exten = t,1,Set(MACRO_RESULT=CONTINUE)
exten = s,1,Wait(1)
exten = s,n,BackgroundDetect(/var/lib/asterisk/sounds/press1)
exten = s,n,WaitExten(10) ; the value is the Wait time before we assume
the call is not accepted
exten = 1,1,NoOp(Caller accepted) ; Do not set MACRO_RESULT to anything
to connect the caller
exten = i,1,Set(MACRO_RESULT=CONTINUE)
exten = t,1,Set(MACRO_RESULT=CONTINUE)
exten = talk,1,NoOp(Caller accepted)
[Inbound]
exten = 4095551212,1,NoOP()
exten = 4095551212,n,Dial(LOCAL/111222LOCAL/222333,40)
exten = 4095551212,n,Voicemail(1...@default)
James Shigley
--
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[asterisk-users] BYE message not relayed to caller
Hello, I have a setup that includes a cellphone a proxy running Kamailio and rtpproxy and a SIP server (VoipSwitch/Asterisk). Call flow works well while using Asterisk, however when VoipSwitch is used i find that the BYE message from VoipSwitch has an RURI = acco...@voipswitch, so the proxy ends up repeatedly sending BYE messages to VoipSwitch instead of sending them to the Cellphone, causing the Cellphone to never hangup. However when using Asterisk the BYE message is forwarded to the cellphone and both endpoints of the call hangup. I show below the SIP message flow while using VoipSwitch. Cell Phone Kamailio VoipSwitch | | | |INVITE| | |-| | |100 Trying| | |-| | | |INVITE| | |-| | |100 trying| | |-| | |183SessionProg| | |-| |183SessionProg| | |-| | | |200 OK| |200 OK|-| |-| | | ACK | | |-| | | | ACK | | |-| | | BYE | | |-|- BYE,ruri=acco...@voipswitch | | BYE | | |-| | | BYE | | |-| Is this issue caused by the SIP server or some other element along the SIP message flow ? Does anybody know the difference in SIP message handling between VoipSwitch and Asterisk or can anybody point me to an online resource ? -- Thanks and Regards, Vikram Ragukumar. -- _ -- 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] Which H.323 to use in Ast 1.6
Could you share your config for the Asterisk and Avaya side too? Thanks -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jamie A. Stapleton Sent: Wednesday, February 24, 2010 3:37 PM To: Asterisk Users List Subject: Re: [asterisk-users] Which H.323 to use in Ast 1.6 I have always used ooh323 between Avaya and Asterisk. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis Sent: Tuesday, February 23, 2010 2:24 PM To: 'Asterisk Users List' Subject: [asterisk-users] Which H.323 to use in Ast 1.6 We're doing a project that requires H.323 to an Avaya. Does anyone have experience to share on which H.323 driver to use in asterisk 1.6? Is the diference between h323 and ooh323 still worth the extra effort? (We've only installed h323 under 1.4) If you have setup/config experience with this setup in Asterisk 1.6 please share! Thanks, MD -- _ -- 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] AMD: HANGUP
I changed my VOIP, and now things are ok.
But didnt understand, how can VOIP can affect it ?
On Wed, Feb 24, 2010 at 11:53 PM, David @ULC ucoms2...@gmail.com wrote:
*Code:*
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing Playback(Local/91441425477...@default-b9f2,1,
sip-silence) in new stack
-- Playing 'sip-silence' (language 'en')
-- Executing AGI(Local/91441425477...@default-b9f2,1, agi://
127.0.0.1:4577/call_log) in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing AMD(Local/91441425477...@default-b9f2,1,
2000|2000|1000|5000|120|50|4|256) in new stack
-- AMD: Local/91441425477...@default-b9f2,1 00 (null) (Fmt:
64)
-- AMD: initialSilence [2000] greeting [2000] afterGreetingSilence
[1000] totalAnalysisTime [5000] minimumWordLength [120] betweenWordsSilence
[50] maximumNumberOfWords [4] silenceThreshold [256]
== Spawn extension (default, 91441425477375, 2) exited non-zero on
'Local/91441425477...@default-1e22,2'
-- Executing DeadAGI(Local/91441425477...@default-1e22,2, agi://
127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-16-ANSWER-35-15)
in new stack
-- AGI Script agi://
127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-16-ANSWER-35-15completed,
returning 0
-- AMD: HANGUP
-- Executing DeadAGI(Local/91441425477...@default-1e22,1, agi://
127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-0---)
in new stack
-- AGI Script agi://
127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-0---completed,
returning 0
== Spawn extension (default, 91441425477388, 2) exited non-zero on
'Local/91441425477...@default-86e4,2'
-- Executing DeadAGI(Local/91441425477...@default-86e4,2, agi://
127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-16-ANSWER-41-15)
in new stack
-- AGI Script agi://
127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-16-ANSWER-41-15completed,
returning 0
-- AMD: HANGUP
-- Executing DeadAGI(Local/91441425477...@default-86e4,1, agi://
127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-0---)
in new stack
-- AGI Script agi://
127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-0---completed,
returning 0
vici*CLI
My agent are NOT getting calls.
-- AMD: HANGUP ??
Is that an Issue ?
How to solve it ?
I have below entry for 8369 :
*Code:*
; VICIDIAL_auto_dialer transfer script AMD with Load Balanced:
exten = 8369,1,Playback(sip-silence)
exten = 8369,2,AGI(agi://127.0.0.1:4577/call_log)
exten = 8369,3,AMD(2000|2000|1000|5000|120|50|4|256)
exten = 8369,4,AGI(VD_amd.agi,${EXTEN})
exten = 8369,5,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-LB)
exten = 8369,6,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-LB)
exten = 8369,7,Hangup
Amd.conf has :
*Code:*
; initial_silence: Maximum silence duration before the greeting. If
exceeded then MACHINE.
; greeting: Maximum length of a greeting. If exceeded then MACHINE.
; after_greeting_silence: Silence after detecting a greeting. If exceeded
then HUMAN
; total_analysis_time: Maximum time allowed for the algorithm to decide on
a HUMAN or PERSON
; min_word_length: Minimum duration of Voice to considered as a word
; between_words_silence: Minimum duration of silence after a word to
considere the audio what follows as a new word
; maximum_number_of_words: Maximum number of words in the greeting. If
exceeded then MACHINE
[AnsweringMachineDetector]
initial_silence= 3500
greeting = 1500
after_greeting_silence = 300
total_analysis_time= 5000
min_word_length= 120
between_words_silence = 50
maximum_number_of_words= 5
silence_threshold = 256
--
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[asterisk-users] audio glitches in conference
I'm having a problem with conferences both meetme and app_conference, though I've done most of the testing with meetme. Essentially, I get little gaps in the audio - usually fewer than a dozen or so samples, though it does vary. They seem to occur at random, but I usually get one ever few seconds, on average. They also seem to delay some buffer somewhere, so that if I start recording (via eagi) after the conference has been established for half an hour or so, the stream received by the eagi script delayed by about 10 seconds. First, the preliminaries: I'm on a debian lenny system, using the 1:1.4.21.2~dfsg-3 asterisk package. This is a dedicated server - was running xen, but I've shut down all the domU's to test if they were interfering, so now there's no sharing going on. I've been testing with a simple eagi script to grab the audio from the conference: #!/bin/sh cat /dev/fd/3 /tmp/audio.raw I've been testing it using the following dialplan extensions: [test] exten = testeagi,1,Answer exten = testeagi,n,Wait(3) exten = testeagi,n,EAGI(testeagi.sh) exten = testmeet,1,Answer exten = testmeet,n,MeetMe(testconf,1qd) exten = testsound,1,Answer exten = testsound,n,Playback(testbeep-asterisk) (testbeep is just 30 seconds of sine wave) I've been trying things like this: originate Local/testso...@test extension teste...@test The recorded audio plays back fine - no glitches. (an example is at http://www.vecotourism.org/audio17.wav) originate Local/teste...@test extension testm...@test originate Local/testso...@test extension testm...@test This does have the glitches. (an example is at http://www.vecotourism.org/audio18.wav) What could be causing this? And is there anything else I could be doing to debug it? Thanks. -- Jon-o Addleman - http://www.redowl.ca -- _ -- 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] Problems with Linksys IP Phone SPA 942
Hi people, I'm having problems of connection with a Linksys SPA IP PHONE 942 when I use the WAN port, most of the time when I try to connect to the network or restart the IP Phone I can't get internet connectivity . I tried using both static IP and DHCP, but the problem is the same. Some have had similar problem with this brand of IP Phone? Thanks for the help and attention. Hugs Veja quais são os assuntos do momento no Yahoo! +Buscados http://br.maisbuscados.yahoo.com-- _ -- 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] Problems with Linksys IP Phone SPA 942
How did you connect your phone to the network ? Please describe your connection. Jimmy -Original Message-From: renato...@yahoo.com.brSent: Wed, 24 Feb 2010 15:51:35 -0800 (PST)To: asterisk-users@lists.digium.comSubject: [asterisk-users] Problems with Linksys IP Phone SPA 942 Hi people,I'm having problems of connection with a Linksys SPA IP PHONE 942 when I use the WAN port, most of the time when I try to connect to the network or restart the IP Phone I can't get internet connectivity . I tried using both static IP and DHCP, but the problem is the same. Some have had similar problem with this brand of IP Phone?Thanks for the help and attention.Hugs Veja quais são os assuntos do momento no Yahoo! + Buscados: Top 10 - Celebridades - Música - Esportes Free 3D Marine Aquarium Screensaver Watch dolphins, sharks orcas on your desktop! Check it out at www.inbox.com/marineaquarium -- _ -- 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] audio glitches in conference
Jonathan- I'm having a problem with conferences both meetme and app_conference, though I've done most of the testing with meetme. Essentially, I get little gaps in the audio - usually fewer than a dozen or so samples, though it does vary. They seem to occur at random, but I usually get one ever few seconds, on average. They also seem to delay some buffer somewhere, so that if I start recording (via eagi) after the conference has been established for half an hour or so, the stream received by the eagi script delayed by about 10 seconds. How did you measure the gaps? Using signal or speech analysis software to display the recording? If you measure number of samples between the gaps, does it correspond to multiples of RTP packet payload length (for example, for 8 kHz G711 multiples of 80 samples between gaps) ? -Jeff First, the preliminaries: I'm on a debian lenny system, using the 1:1.4.21.2~dfsg-3 asterisk package. This is a dedicated server - was running xen, but I've shut down all the domU's to test if they were interfering, so now there's no sharing going on. I've been testing with a simple eagi script to grab the audio from the conference: #!/bin/sh cat /dev/fd/3 /tmp/audio.raw I've been testing it using the following dialplan extensions: [test] exten = testeagi,1,Answer exten = testeagi,n,Wait(3) exten = testeagi,n,EAGI(testeagi.sh) exten = testmeet,1,Answer exten = testmeet,n,MeetMe(testconf,1qd) exten = testsound,1,Answer exten = testsound,n,Playback(testbeep-asterisk) (testbeep is just 30 seconds of sine wave) I've been trying things like this: originate Local/testso...@test extension teste...@test The recorded audio plays back fine - no glitches. (an example is at http://www.vecotourism.org/audio17.wav) originate Local/teste...@test extension testm...@test originate Local/testso...@test extension testm...@test This does have the glitches. (an example is at http://www.vecotourism.org/audio18.wav) What could be causing this? And is there anything else I could be doing to debug it? Thanks. -- Jon-o Addleman - http://www.redowl.ca -- _ -- 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] OT: Problems with Linksys IP Phone SPA 942
Sent: Wednesday, February 24, 2010 5:52 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Problems with Linksys IP Phone SPA 942 Hi people, I'm having problems of connection with a Linksys SPA IP PHONE 942 when I use the WAN port, most of the time when I try to connect to the network or restart the IP Phone I can't get internet connectivity . I tried using both static IP and DHCP, but the problem is the same. Some have had similar problem with this brand of IP Phone? Thanks for the help and attention. Hugs Veja quais são os assuntos do momento no Yahoo! + Buscados: Top 10 - Celebridades - Música - Esportes --- First, have you tried a known working ethernet cable? Also, if you have VLAN enabled on the phone, it will send out a DHCP request, but then ignore the reply unless the reply has VLAN in it. If I where you, I would sniff the packets using www.wireshark.org and a true hub (or switch capable of mirroring ports). If you have it set to DHCP and you don't see it send out any DHCP requests, I would call Linksys and see what they say about troubleshooting it. This is the filter to use in WireShark to only see Linksys device packets: eth.addr[0:3] == 00-0e-08 Optionally you could add bootp || sip to the end make Wireshark only show DHCP and SIP activity. One of the companies I work for pre-provisions Linksys phones and blind drop ships them for ITSPs. We also handle the ITSPs returns. Anyway, we have booted, pre-provisioned, shipped, and handled returns for almost 10,000 SPA942s in the past year and had 47 that were actually bad and Linksys replaced them all. Maybe 10 of those where what I attributed as the WAN port was broken, the other 37 where mainly hung firmware and won't power problems. About once a week though we do get a phone back where the end user says it won't connect to the network and the ITSP just replaced it without hassling the customer to troubleshoot. We boot it behind our DHCP server, and it works fine. Then we reset it to factory defaults, re-provision, and re-ship it and normally don't get it back a second time. Sometimes it looks like the end user has mucked around in the VLAN settings or it is set to static IP and we have to change these before DHCP actually works but I can always get it back. I'd be interested in hearing what you find out as I have always been curious about why those once a weeks don't work at the end user but have no way of finding it out. I'm sure Linksys would like to know as well. One thing made be feel dumb the other day. We boot these phones 10 at a time and a new guy had accidentally plugged in one of them using the LAN port. This caused all sorts of problems in the network for some reason. After chasing my tail in the DHCP server and power cycling the switch and this and that for 2 hours, I found that one, changed it to the WAN, and am still too mad at myself to actually investigate why that broke the whole system. Regards, Shanon Swafford Cell: 972.989.3242 Email: sha...@dfwavc.com http://www.dfwavc.com/ http://www.ntxinternet.com/ http://www.ntxinternet.com/shanon_swafford.html -- _ -- 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] Do i need install Dahdi or libpri ?
hello,all
there is a AudioCodes Mediant 2000 out there. i want to realise ip to
PSTN and PSTN to ip connection.
after some configuration on AudioCodes Mediant 2000, PSTN to ip
connecttion works.
next ,i want to dial from asterisk to PSTN now. i have see the sample
in the extensions.conf relevent to PSTN as follow:
; If you are freely delivering calls to the PSTN, list them here
;
;exten = _1256428,1,Dial(DAHDI/G2/${EXTEN:7}) ; Expose all of 256-428
;exten = _1256325,1,Dial(DAHDI/G2/${EXTEN:7}) ; Ditto for 256-325
but above shows something about DAHDI card.
my question is:
a, Do i need install DAHDI or libpri in my system?
b, how to write in dialplan to realise connection to PSTN.
That'sPSTN-AudioCodes Mediant
2000---IP(asterisk)-AudioCodes Mediant 2000
--PSTN
?
Thanks very much!
--
Best regards,
Sucan
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Re: [asterisk-users] OT: Problems with Linksys IP Phone SPA 942
On Wednesday 24 February 2010 19:23:01 Shanon Swafford wrote: One thing made be feel dumb the other day. We boot these phones 10 at a time and a new guy had accidentally plugged in one of them using the LAN port. This caused all sorts of problems in the network for some reason. After chasing my tail in the DHCP server and power cycling the switch and this and that for 2 hours, I found that one, changed it to the WAN, and am still too mad at myself to actually investigate why that broke the whole system. DHCP is designed in such a way that you can legitimately have multiple DHCP servers on the same network. The first DHCP server which replies and meets the DHCP client's requirements will be the server to which the client registers. If the Linksys DHCP server is faster (or if you have several switches and it replies to some hosts faster), then those hosts will likely use the Linksys as their DHCP server. You could technically avoid this situation by provisioning some DHCP option that the Linksys does not and making all of your DHCP clients require that option, but that takes quite a bit away from the zeroconf usage of DHCP. Or you could set up a rule on your managed switch such that broadcasts to UDP port 67 only hit the switch port on which your intended DHCP server is located. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- 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] Re-INVITE on BYE
Hi Kevin, Thx for your kind response. Is there any options/steps that I could trigger to skip from redirecting the media back to Asterisk? Regards, Lawrence -- _ -- 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] audio glitches in conference
I'm having a problem with conferences both meetme and app_conference, though I've done most of the testing with meetme. What version of DAHDI are you running? -- _ -- 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] Re-INVITE on BYE
Lawrence Na (my-...@vyke) wrote: Thx for your kind response. Is there any options/steps that I could trigger to skip from redirecting the media back to Asterisk? If your mail client allows, please *reply* to messages in a thread, rather than starting a new thread with the same subject. This way people who find that thread in the list archives can see all the messages in the thread. Thanks. To answer your question, though, no, there is no method available in Asterisk today to modify this behavior. Are you just curious, or do think it is actually causing a problem? -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- 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 n-way DTMF detection
Hello, I have setup the n-way conferencing with Asterisk and it's working when I use with my budgetone 100 phone but it doesn't work for any of the voip software or other ATA that I have. When I turned the debug on, I see that the correct keys (*0) were entered but asterisk doesn't detect the signal to trigger the features event. I have set a test extension to get the input dtmf key and say the digit out. They are getting correctly on the IVR but when using n-way conferencing, it's not taking it. Here is the output of testing DTMF with IVR. v103*CLI v103*CLI -- Executing [...@from-internal:1] Read(SIP/-b6807538, digito||10) in new stack -- Accepting a maximum of 10 digits. * DTMF-relay event received: 8 * DTMF-relay event received: 5 * DTMF-relay event received: 2 -- User entered '852' -- Executing [...@from-internal:2] SayDigits(SIP/-b6807538, 852) in new stack -- SIP/-b6807538 Playing 'digits/8' (language 'en') -- SIP/-b6807538 Playing 'digits/5' (language 'en') -- SIP/-b6807538 Playing 'digits/2' (language 'en') -- Executing [...@from-internal:3] Hangup(SIP/-b6807538, ) in new stack == Spawn extension (from-internal, 88, 3) exited non-zero on 'SIP/-b6807538' -- Executing [...@from-internal:1] Macro(SIP/-b6807538, hangupcall) in new stack -- Executing [...@macro-hangupcall:1] ResetCDR(SIP/-b6807538, vw) in new stack -- Executing [...@macro-hangupcall:2] NoCDR(SIP/-b6807538, ) in new stack -- Executing [...@macro-hangupcall:3] GotoIf(SIP/-b6807538, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,6) -- Executing [...@macro-hangupcall:6] GotoIf(SIP/-b6807538, 1?skipblkvm) in new stack -- Goto (macro-hangupcall,s,9) -- Executing [...@macro-hangupcall:9] GotoIf(SIP/-b6807538, 1?theend) in new stack -- Goto (macro-hangupcall,s,11) -- Executing [...@macro-hangupcall:11] Hangup(SIP/-b6807538, ) in new stack == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/-b6807538' in macro 'hangupcall' == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/-b6807538' v103*CLI bash-3.1# anh here is the console log of the Asterisk when pressing the key during callerA is on the phone with CallerB. v103*CLI v103*CLI * DTMF-relay event received: * * DTMF-relay event received: 0 v103*CLI Wondering that if anyone know what could be wrong here. My asterisk version is Asterisk 1.4.20. -Tri -- _ -- 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] curl and ssl certificate
Hello, Is it possible use asterisk curl function with ssl sertificate? Thanks -- Best Regards, Giedrius -- _ -- 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
