Re: [asterisk-users] State of Asterisk+Virtualization+Timing
On Sun, Nov 06, 2011 at 03:50:21PM +, Gordon Henderson wrote:
On Tue, 1 Nov 2011, Nic Colledge wrote:
Have you thought about using LXC rather than OpenVZ.
+1
There are a few references to allowing guest access to timing
hardware online.
Simples. Load up the dahdi modules in the host and all the containers
see it.
Be sure to also create /dev/dahdi/{ctl,timer,pseudo,channel} for the
container, as they're likely not be allowed to create device files.
Here is a link to the nice little blog entry I used a reference when doing this
in testing... http://www.whmcr.com/2011/06/18/dahdi-in-lxc/
(Not my blog).
Nic.
--
_
-- 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] State of Asterisk+Virtualization+Timing
Have you thought about using LXC rather than OpenVZ. There are a few references to allowing guest access to timing hardware online. I've only been playing with it recently and haven't used it in production yet but plan to soon. As for general thoughts about virtualising asterisk, I tried it in the past (about a year ago) on KVM and VMWare and it didn't work too well for me. Regardless of whether you are using LXC / OpenVZ / KVM / Whatever, you should be careful not to have too much other stuff running on the box. If asterisk has to wait to get CPU time you will really notice it, this isn't a problem with other applications like say a webserver. Nic. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson Sent: 01 November 2011 17:08 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] State of Asterisk+Virtualization+Timing Greetings- I'm about to dive into the process of virtualizing some of my Asterisk (primarily 1.4.x) infrastructure. In the past, when looking at virt solutions, the primary issue preventing me from moving was the lack of proper timing. We do not need it for MeetMe but rather for IAX2 trunking. I'd like to use either OpenVZ or KVM, but each seem to have independent issues that need to be addressed: OpenVZ - Better resource usage, lower overhead. Primary issue is how to grant access to host node timing source (physical device, or dahdi_dummy in /dev/dahdi/) to the containerized Asterisk process. KVM - Higher overhead, easier installation, 'true virtualization'. Primary issue is not timing per se, but KVM scheduling. Timing source, while present from dahdi_dummy natively may still not get proper scheduling by KVM process. This could also affect general call quality (even non IAX2 trunked voice), DTMF, etc. I have to believe there are others running virtualized Asterisk installations with some degree of success on OpenVZ or KVM. Care to share your thoughts? --Tim -- _ -- 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] security: SIP header spoofing CHANNEL(recvip)?
I was wondering if these could be spoofed recently when reading the docs.
Have you tried peerip rather than recvip?
Does that give the same result?
Nic.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alejandro Recarey
Sent: 25 August 2011 11:34
To: Asterisk Users Mailing List
Subject: [asterisk-users] security: SIP header spoofing CHANNEL(recvip)?
I am currently suffering various SIP attacks. I am using the following
extension to record the caller's IP address:
exten = h,n,set(CDR(srcip)=${CHANNEL(recvip)})
However, in recent attacks, this IP address is not correct, and I
believe that they are spoofing it. I am using asterisk 1.6.2.15.
Does the CHANNEL(recvip) variable record IP show in the SIP header
instead of the real, UDP source IP? If the CHANNEL(recvip) variable
records the IP address set in the SIP header, and not the real IP
address, how can I obtain the REAL IP address of the caller?
--
_
-- 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] is res_timing_timerfd module stable in 1.8?
New Text at Bottom: --- hi: my current system is 1.6.2. I have dahdi hardware card. I must disable res_timing_timerfd module or sometimes phone calls would become silent suddenly. I don't know the situation in 1.8. I heard that timing is still a problem in 1.8. should I keep using res_timing_dahdi or I can use res_timing_timerfd to get some benefit if I upgrade to 1.8? thank a lot for information!! Regards, tbskyd -- Hi, There are a few issues on Mantis that people think are related to timing and timerfd. I'm not sure what the benefits of timerfd over dahdi are when you have the dahdi hardware (maybe someone else could comment). I'm currently testing 1.8.4 with dahdi timing (no hardware) to see if it solves some of the problems we have been having. Its working so far (touch wood) but its only three days and about 500 calls in. I would say if you have the dahdi hardware timer then use it. If you have been having a timing related issue with 1.6 and/or 1.8 have a look on issues.asterisk.org and see if anyone else has reported similar problems that you can add a bit more info to, if not make your own report. Regards, Nic. -- _ -- 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 crashes on high IO load
Are you using IAX? There are some problems causing crashes for us related to laggyness on IAX channels with 1.8 versions. There are a bunch of problems with IAX related to https://issues.asterisk.org/view.php?id=17521 Nic. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Maximilian Grobecker Sent: 04 April 2011 16:24 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk crashes on high IO load Hello Thorsten, the system has 4 GB RAM and about 2,5 GB free so swap space is not used or exhausted. Maybe the high load is not cause of this crashes but it's the only thing the crashes can be reproduced with. Thank you! Maximilian Grobecker Am 04.04.2011 16:03, schrieb Thorsten Göllner: Take a look with top at your system when high io load is seen. Maybe the machine is running out of ram and starts swapping? Am 04.04.2011 15:04, schrieb Maximilian Grobecker: Hi! I'm writing to this list because I've got a very confusing issue with our Asterisk 1.8.3.2 installation. On high IO load on the hard drives Asterisk becomes instable and crashes after a few minutes. I tried to reproduce this by running bonnie++ on the hardware while making calls. The calls didn't get disturbed (no noises or crackles) but after about five minutes Asterisk suddenly crashed without any further error messages. Are you experiencing the same problem? I'm really confused now why Asterisk crashes... Thank you! Maximilian Grobecker -- _ -- 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
Re: [asterisk-users] 1.8 realtime - segfault
Hi, This may be related to an issue I added to the bug tracker. Problems around using Local Channels across realtime / non-realtime contexts in 1.8. Nic. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Naomi Rosenberg Sent: 21 March 2011 16:22 To: asterisk-users@lists.digium.com Subject: [asterisk-users] 1.8 realtime - segfault Hi, I have installed Asterisk 1.8 and am using realtime dialplan contexts from a mysql table. Asterisk keeps segfaulting. When I trace the thread ids associated with the segfaults in the full log, all they have in common is netsock2.c: == Using SIP RTP CoS mark 5 which is probably a red herring since it appears so often in threads that do not segfault. When compiling, I ticked all the addons that mentioned mysql, as well as the DON'T OPTIMIZE flag, but other than that I left it to the defaults. I would really like some ideas about what might be causing this. If you need any more information please let me know what information you need and I will try and help you to help me! Thank you, Naomi -- _ -- 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] Realtime and Local Channel Crash Problem 1.8.3-rc2
Hi, I have been having a problem with asterisk crashing when using local channels and realtime on asterisk 1.8.3-rc2. The example given here is I think the easiest way to reproduce this problem. In extensions.conf I have: [internal] switch = Realtime/extensions/p exten = 301,1,Answer() exten = 301,2,Dial(Local/501@internal) exten = 301,3,Hangup() exten = 501,1,Answer() exten = 501,2,Playback(demo-echotest) exten = 501,3,Echo() exten = 501,4,Hangup() Where dialling 301 puts you through to 501 and you hear the echo test message fine. However if I move 501 to the realtime database extensions table and remove it from extensions.conf asterisk hangs on the local channel dial, then completely dies a few minutes later (console stops responding to commands) with killall -9 asterisk being the only way to stop it. In both cases I can dial 501 directly with no problem. The last message on the console (with verbose 10) -- Executing [300@internal:2] Dial(SIP/1014-0001, Local/501@internal) Everything works fine with the exact same setup and asterisk 1.8.1.2. Thanks, Nic. -- _ -- 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] Realtime and Local Channel Crash Problem 1.8.3-rc2
Text below.. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonathan Thurman Sent: 15 February 2011 19:39 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime and Local Channel Crash Problem 1.8.3-rc2 On Tue, Feb 15, 2011 at 10:15 AM, Nic Colledge n...@njcolledge.net wrote: I have been having a problem with asterisk crashing when using local channels and realtime on asterisk 1.8.3-rc2. Nic, I can reproduce this using the latest SVN for the 1.8 branch. I don't get the console locking, but SIP definitely deadlocks every time. If you want to open a ticket, I'll upload the bt/threads/locks info that I have. -Jonathan Guys, It's after a couple of attempts of core show channels that the console goes bonkers. I'm reporting the issue on issues.asterisk.org now. Will post the number here once I have uploaded my back traces. Thanks. Nic. -- _ -- 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] Realtime and Local Channel Crash Problem 1.8.3-rc2
Text below.. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonathan Thurman Sent: 15 February 2011 19:39 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime and Local Channel Crash Problem 1.8.3-rc2 On Tue, Feb 15, 2011 at 10:15 AM, Nic Colledge n...@njcolledge.net wrote: I have been having a problem with asterisk crashing when using local channels and realtime on asterisk 1.8.3-rc2. Nic, I can reproduce this using the latest SVN for the 1.8 branch. I don't get the console locking, but SIP definitely deadlocks every time. If you want to open a ticket, I'll upload the bt/threads/locks info that I have. -Jonathan Guys, It's after a couple of attempts of core show channels that the console goes bonkers. I'm reporting the issue on issues.asterisk.org now. Will post the number here once I have uploaded my back traces. Thanks. Nic. Further to my last issue# is 0018818. Thanks, Nic. -- _ -- 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] Saving the monitor file on new file always using Monitor(wav, Record1, m)
Try using ${UNIQUEID} to get the unique id of the current call. That or
something like CDR(uniqueid). Forget which off the top of my head.
Nic.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad
Sent: 01 January 2011 17:43
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Saving the monitor file on new file always using
Monitor(wav, Record1, m)
Dear List;
For each call (in specific case), I need to do a record and save in a spearated
file, so I am thinking the best thing is to save based on the time.
Monitor(wav,Record1,m)
So, how can I make the file name to be based on the current time (which is
changed always, or based on the some unique paramter (related to the call it
self).
Any advise?
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
Re: [asterisk-users] Asterisk 1.8 IAX Registration
Paul, Thanks, I'll try this patch later tonight. Nic. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger Sent: 28 October 2010 03:27 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.8 IAX Registration On Tue, Oct 26, 2010 at 8:26 PM, Paul Belanger paul.belan...@polybeacon.com wrote: I'm going to try and look at this during Astricon :) Ok, just uploaded a new patch on https://issues.asterisk.org/view.php?id=18202 Let me know if it worked. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) | Blog: http://blog.polybeacon.com | Twitter: http://twitter.com/pabelanger -- _ -- 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 IAX Registration
Paul, Further to my last, I think I found another small related issue with IAX which is generating the following error: [Oct 24 14:42:12] ERROR[15589]: netsock2.c:94 ast_sockaddr_stringify_fmt: getnameinfo(): ai_family not supported To reproduce this issue, setup a phone in iax.conf or your realtime table, goto Zoiper press register in the Preferences / IAX Account. The phone will register correctly when your patch is applied. Then press unregister in Zoiper. On my realtime peers the error then shows up on the console, but for my static iax.conf peers it does not. If you then do a iax2 show peers on the console, the error is displayed. Notice that the value for Host in the command output is a empty string (example below). Name/UsernameHost Mask Port Status 111 (D) 255.255.255.255 0 Unmonitored Initially (for static iax.conf peers) before registration this value is null and does not cause the error to be displayed on a iax2 show peers command (example below). So I'm guessing that somewhere on un-registration this is set to when it should be set to null. Name/UsernameHost Mask Port Status 111 (null) (D) 255.255.255.255 0 Unmonitored Thanks, Nic. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nic Colledge Sent: 24 October 2010 14:31 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.8 IAX Registration Paul, I applied your patch to 1.8.0 and I'm happy to report it has fixed the problem I was experiencing. Thanks again. Nic. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger Sent: 23 October 2010 22:06 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.8 IAX Registration On Sat, Oct 23, 2010 at 3:03 PM, Paul Belanger paul.belan...@polybeacon.com wrote: Okay, just reproduced your issue and looking at the code now. :) Ok, think I fixed it. You can either apply this patch to 1.8.0, or svn update the branch I'm working on. Feedback is welcome. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) | Blog: http://blog.polybeacon.com | Twitter: http://twitter.com/pabelanger -- _ -- 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 IAX Registration
Paul, I made a debug log of the register and unregister process for a single Zoiper client using IAX and have emailed it direct to you. The error shows in the file as: [Oct 24 19:07:32] ERROR[1403] netsock2.c: getnameinfo(): ai_family not supported Thanks, Nic. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger Sent: 24 October 2010 18:40 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.8 IAX Registration On Sun, Oct 24, 2010 at 10:06 AM, Nic Colledge n...@njcolledge.net wrote: Further to my last, I think I found another small related issue with IAX which is generating the following error: Do you mind collecting a debug log [1]? Having some issues reproducing this. [1] http://svn.asterisk.org/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) | Blog: http://blog.polybeacon.com | Twitter: http://twitter.com/pabelanger -- _ -- 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] Asterisk 1.8 IAX Registration
Hi, Have just been testing asterisk 1.8.0, 1.8.0-rc5 and a trunk version from about half an hour ago. IAX Friends (Zoiper Softphones) don't stay registered for more than a few seconds they start out with status unknown and quickly become unreachable, I am using realtime with postgresql, with tables and configuration that have worked fine for IAX in 1.6 and a trunk release from a few months ago. I have not tested any other IAX clients other than Zoiper Softphones. Operating system is Ubuntu 9.10, with the default kernel recompiled for AMD CPUs and a couple of other small changes. I get the following errors on the console after a registration followed by unregistration of the zoiper client: [Oct 23 15:22:50] ERROR[695]: chan_iax2.c:11917 iax2_poke_peer: Bad address cast to IPv4 [Oct 23 15:22:50] ERROR[695]: chan_iax2.c:1742 iax2_getpeername: Bad address cast to IPv4 [Oct 23 15:22:50] ERROR[695]: chan_iax2.c:8770 update_registry: Bad address cast to IPv4 [Oct 23 15:22:53] ERROR[702]: chan_iax2.c:1742 iax2_getpeername: Bad address cast to IPv4 [Oct 23 15:22:53] ERROR[703]: chan_iax2.c:1742 iax2_getpeername: Bad address cast to IPv4 [Oct 23 15:22:53] ERROR[694]: chan_iax2.c:1742 iax2_getpeername: Bad address cast to IPv4 [Oct 23 15:22:53] NOTICE[702]: chan_iax2.c:8618 reg_source_db: IAX/Registry astdb host:port invalid - '192.168.1.111:4569' [Oct 23 15:22:53] ERROR[693]: netsock2.c:94 ast_sockaddr_stringify_fmt: getnameinfo(): ai_family not supported [Oct 23 15:22:53] NOTICE[703]: chan_iax2.c:8618 reg_source_db: IAX/Registry astdb host:port invalid - '192.168.1.111:4569' [Oct 23 15:22:53] NOTICE[694]: chan_iax2.c:8618 reg_source_db: IAX/Registry astdb host:port invalid - '192.168.1.111:4569' [Oct 23 15:22:53] ERROR[695]: netsock2.c:94 ast_sockaddr_stringify_fmt: getnameinfo(): ai_family not supported [Oct 23 15:22:53] ERROR[695]: chan_iax2.c:2305 peercnt_modify: Bad address cast to IPv4 Is this a configuration issue or something in asterisk? Thanks in advnace. Nic Colledge -- _ -- 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 IAX Registration
Sorry forgot to add this into my initial email. The same happens with phones configured in iax.conf and the Realtime database table. [Oct 23 16:49:52] ERROR[1220]: chan_iax2.c:8770 update_registry: Bad address cast to IPv4 etc. Nic. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: 23 October 2010 16:07 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.8 IAX Registration What happens without using Realtime ? -- _ -- 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 IAX Registration
Paul, Thanks for your reply, I saw that issue on the tracker when I was trying to find a solution earlier but it looked completely different so I didn't give it much thought. This happens on my install with both Realtime phones and those configured in iax.conf I have just tried the branch you suggested and the problem remains. It's worse with qualify=yes but still happens (albeit less frequently) with qualify=no. My iax.conf general section: [general] bindport=4569 bindaddr=0.0.0.0 srvlookup=yes jitterbuffer=no forcejitterbuffer=no context=default rtcachefriends=yes rtautoclear=yes A typicall iax.conf user: [111] type=friend auth=md5 host=dynamic context=internal mailbox=111 accountcode=IAX/111 disallow=all allow=ulaw allow=alaw allow=g729 requirecalltoken=auto qualify=yes secret=secret111 Thanks again, Nic. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger Sent: 23 October 2010 17:58 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.8 IAX Registration On Sat, Oct 23, 2010 at 11:53 AM, Nic Colledge n...@njcolledge.net wrote: Sorry forgot to add this into my initial email. The same happens with phones configured in iax.conf and the Realtime database table. https://issues.asterisk.org/view.php?id=18183 I was able to reproduce a problem with realtime IAX2 yesterday, do you mind trying the branch listed on the issue tracker. Also, please post a copy of your iax2.conf file that will reproduce the issue (mask any passwords). -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) | Blog: http://blog.polybeacon.com | Twitter: http://twitter.com/pabelanger -- _ -- 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] CEL ODBC problem in 1.8.0
Hi,
I have been experimenting with CEL in a trunk version of asterisk for some time
and have upgraded my test machine to 1.8.0 today.
Made a few calls and it looks like the eventtype field is missing in the CEL
insert query when using ODBC. I see the following errors on the console:
[Oct 22 21:46:09] WARNING[952]: res_odbc.c:634 ast_odbc_prepare_and_execute:
SQL Execute returned an error -1: 23502: ERROR: null value in column
eventtype violates not-null constraint;
Error while executing the query (101)
[Oct 22 21:46:09] WARNING[952]: res_odbc.c:646 ast_odbc_prepare_and_execute:
SQL Execute error -1! Attempting a reconnect...
[Oct 22 21:46:09] WARNING[952]: res_odbc.c:742 ast_odbc_sanity_check:
Connection is down attempting to reconnect...
[Oct 22 21:46:09] NOTICE[952]: res_odbc.c:1472 odbc_obj_connect: Connecting
PostgreSQL-asterisk
[Oct 22 21:46:09] NOTICE[952]: res_odbc.c:1502 odbc_obj_connect: res_odbc:
Connected to PostgreSQL-asterisk [asterisk-connector]
[Oct 22 21:46:09] WARNING[952]: cel_odbc.c:733 odbc_log: Insert failed on
'PostgreSQL-asterisk:celnew'. CEL failed: INSERT INTO celnew
(uniqueid,linkedid,eventtime,userdeftype,cid_name,cid_num,cid_ani,cid_rdnis,cid_dnid,exten,context,channame,appname,appdata,accountcode,peeraccount,userfield,peer,amaflags)
VALUES ('co-1287780365.0','co-1287780365.0',{ ts '2010-10-22 21:46:06'
},'','135','441X','441X','','136','136','internal','SIP/135-','','','SIP/135','SIP/135','','',3)
I have copied my old config files and am using the postgresql database which
worked fine with the old (trunk) version from ages ago.
When I use the PGSQL native driver instead of ODBC with the same postgresql
table it works fine.
Is this a bug?
P.s. great work on 1.8.0 by the way, thanks to all the developers, testers and
everyone involved.
Thanks,
Nic Colledge
n...@njcolledge.netmailto:n...@njcolledge.net
--
_
-- 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] How to tell if there is a transfer from CDR?
Hi, I use CEL or Call Event Logging in 1.8 to get a more concise picture of what happened in a call. We use it for a bunch of stuff including billing attended and unattended transfers differently. If you are thinking of upgrading, it's worth a try. Nic. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F Sent: 05 September 2010 03:54 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to tell if there is a transfer from CDR? Last time I analyzed this (I believe back in 1.2) there was no way of telling. However a blind transfered call would generate 2 CDR recoreds: 1. For the part of the call with the transferrer and transfered. 2. For the part of the call with the transferee and transfered. The call duration for the 2nd record would include the time of the 1st record as well. So if part one took 20 seconds and part 2 40 seconds, then the 2nd record would have 60 seconds as billable. The only workaround was to check the BLINDTRANSFER var and reset cdr if it was populated. Please members of this list, I would love to hear more input as I'm sure this has changed. Also I would not be surprised that I'm wrong in my analysis as more than 4 years has passed since and I might have forgotten. TIA On Fri, Sep 3, 2010 at 5:06 PM, Carlos Chavez cur...@telecomabmex.com wrote: Is there any way to know if a call was transferred from reading the CDR? Any relation in fields like UNIQUEID? Something that can be scripted to make a special report? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -- _ -- 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
Re: [asterisk-users] How to tell if there is a transfer from CDR?
Bryant, We have been using a pre-1.8 trunk version of asterisk that has been pretty stable for us. We have a fairly small user base currently and decided to take the risk with a trunk version after some testing basically because of the availability of CEL as it lets us do a bunch of things we couldn't do with 1.6 CDR. I have briefly tested the beta3 of 1.8 and it seemed ok but were holding off for the release version (with a if it ain't broke don't fix it mentality). In our environment and my limited experience 1.8 is shaping up to be a great release (Great work guys, thanks to everyone working on asterisk!) I recommend you fire it up somewhere to test and see if you still have the issues, CEL can be pretty verbose and confusing at first but it does give you a lot more information about what happened during a call and when. On the other hand you may see it working and decide it's not for you. As for DTMF we only have a few IVR Menu style interfaces that don't currently see much use so I can't really give a definitive answer, but we have not had any problems with it. Worst case scenario, you test it and find some problems. I can't speak for the developers, but while it's in beta, now's the time to find them! Regards, Nic. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant Zimmerman Sent: 05 September 2010 21:51 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to tell if there is a transfer from CDR? Nic How stable is 1.8 really? It sounds like you are running it in production is this the case? CDR Transfer issues and rfc2833 DTMF issues are hitting us hard with 1.6.2.x. We want to move as soon as 1.8 is stable enough. Thanks Bryant From: Nic Colledge n...@njcolledge.net Hi, I use CEL or Call Event Logging in 1.8 to get a more concise picture of what happened in a call. We use it for a bunch of stuff including billing attended and unattended transfers differently. If you are thinking of upgrading, it's worth a try. Nic. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F Sent: 05 September 2010 03:54 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to tell if there is a transfer from CDR? Last time I analyzed this (I believe back in 1.2) there was no way of telling. However a blind transfered call would generate 2 CDR recoreds: 1. For the part of the call with the transferrer and transfered. 2. For the part of the call with the transferee and transfered. The call duration for the 2nd record would include the time of the 1st record as well. So if part one took 20 seconds and part 2 40 seconds, then the 2nd record would have 60 seconds as billable. The only workaround was to check the BLINDTRANSFER var and reset cdr if it was populated. Please members of this list, I would love to hear more input as I'm sure this has changed. Also I would not be surprised that I'm wrong in my analysis as more than 4 years has passed since and I might have forgotten. TIA On Fri, Sep 3, 2010 at 5:06 PM, Carlos Chavez cur...@telecomabmex.com wrote: Is there any way to know if a call was transferred from reading the CDR? Any relation in fields like UNIQUEID? Something that can be scripted to make a special report? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -- _ -- 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 -- _ -- 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
[asterisk-users] AppDial in CEL Data
Hi, I am using CEL to more accurate billing information with some success. However there is an ambiguity in the CEL data when multiple destinations are specified in the DIAL command. For example, if I have Dial(SIP/outboundA/100SIP/outboundA/101SIP/outboundB/200SIP/outboundB/201) this is reflected in the dial command data that shows up in CEL. The problem is in some situations it is difficult to tell which one of these destinations answered the call because the CEL_Answer event does not store the destination number anywhere. It would be nice if the appdata of the CEL_Answer event were the part of the dial command which was used to create that channel so say SIP/outboundA/101 rather than (Outgoing Line). I am currently assuming the order in which the channels were created corresponds to the order in which the destinations appear in the dial command to find the answered destination. This works fine most of the time, it only fails when one of more of the outgoing channels could not be created. Would it be possible to change this? Thanks, Nic -- _ -- 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] Full transfer details on inbound calls
Hi, This may be no use to you if you are using 1.4 but Call Event Logging (or CEL) that is currently in trunk should provide an easier way to do this. All events associated with a call e.g. Answer, Hangup, Bridge start, Transfer etc. are logged to the usual back-ends. We use postgresql via ODBC. Nic. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik Sent: 13 April 2010 14:50 To: Asterisk Subject: [asterisk-users] Full transfer details on inbound calls Hi We're using asterisk 1.4.17 using RealTime and my boss has decided that we should keep a track of the full history of incoming calls i.e. who and when they were transferred to. The asterisk CDR only holds the initial answering channel for any call and not any further transfers that may have happened. The idea we are toying with is getting the time and the originating channel from the cdr, and then searching the full asterisk logs for the channel identifier string. Obviously we would have to have the verbose output going to a file and make sure that the verbosity in the console is always at least 5. I've done enough testing to see that is is possible i...@trinity:/var/log/asterisk$ grep 'Apr 13' full | grep SIP/xxx.xxx.xxx.xxx-082090e8 | grep answered [Apr 13 13:31:11] VERBOSE[17120] logger.c: -- SIP/811-08214f50 answered SIP/xxx.xxx.xxx.xxx-082090e8 [Apr 13 13:31:31] VERBOSE[17120] logger.c: -- SIP/808-08212f08 answered SIP/xxx.xxx.xxx.xxx-082090e8 The above output shows that the originating channel was answered by sip extension 811 and then by 808 20 seconds later. I am also considering parsing the full log into a mysql database and doing the searching in there. My question is is this a good way to go about what I'm trying to achieve or is there a simpler/less process intensive method that I'm missing. Thanks in advance Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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] rtcachefriends qualify
Hi, I think so, maybe someone can help clarify this for me also. I have: rtcachefriends=yes rtautoclear=yes in sip.conf and was under the impression that this caches the settings from the database until a user unregisters. When they unregister the data is removed from the cache (rtautoclear). For me this was a nice compromise. This is from memory but I’m pretty sure I got this from the documentation online, if someone can confirm what I’m saying that would be sweet. Thanks. Nic. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jonas kellens Sent: 01 March 2010 14:06 To: Asterisk Mailing Subject: [asterisk-users] rtcachefriends qualify [Mar 1 14:54:07] WARNING[15290]: chan_sip.c:17669 build_peer: Qualify is incompatible with dynamic uncached realtime. Please either turn rtcachefriends on or turn qualify off on peer 'gerrie' Am I correct that when I turn on rtcachefriends in sip.conf, database-changes in my MySQL-DB will not be reflected untill a reload ?? Am I correct that when I turn off qualify in my realtime sip-database, I could be confronted with NAT-problems for SIP-peers that are behind a NAT-router ? Is this the choice I need to take ? Greetingz, Jonas -- _ -- 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] init.d error when installing trunk
Hi, The last few times I have installed trunk versions of asterisk on Ubuntu I have seen this error after doing a make config for asterisk. install: cannot stat `contrib/init.d/etc_default_asterisk': No such file or directory The init.d links then fail to work properly (e.g. /etc/init.d/asterisk restart) after installation. Most recently I installed asterisk (SVN-trunk-r248269) on Ubuntu Server 9.10. Have I missed something in the install process somewhere? Thanks, Nic. -- _ -- 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 Unregisteres IAX Friend Randomly
Bump! And some more information (see below for initial problem): This problem is intermittent, but you don't have to wait long for it to happen. Also, sometimes when the reregister happens (and the client has been wrongly unregistered) asterisk sends the correct response to the client indicating this has been a success when the database is not updated with a new regseconds time. Any idea as to what I've done wrong / what's going on? Thanks in advance. Nic. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nic Colledge Sent: 11 December 2009 15:17 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk Unregisteres IAX Friend Randomly Hi, I've been having a strange problem recently where real-time asterisk will unregister a IAX friend at random times when the registration should not have expired. I have a Zoiper soft phone client (on windows) connecting to asterisk over a LAN (no firewalls). The default reregister time of 60 seconds is used, but the asterisk server unregisters the client (sets regseconds to 0 in the database) after a seemingly random time after registration say, 15 seconds (which is 45 seconds before registration expiry) I set iax2 debug on, and set the core debug level to 5 so I could see the IAX2 control frames on the console. I use pgAdmin-III to watch the value of regseconds in the database change from a registered value to 0. When the unregister happens, there are no frames sent to / received from the client, and nothing else on the asterisk console. It just seems like asterisk decided to unregister the client for no reason. At this point placing a call to the client will fail, until the client reregisters (at the correct time) 45 seconds later. I have seen this happen on both 1.6.2-rc8 and a trunk asterisk from yesterday (on different machines). I'm currently installing 1.6.0 to test that as well. This only seems to happen with real-time asterisk. (I'm using Postgres for the backend database and the pgsql driver in extconfig.conf) Any ideas what's going on here? Is this a known issue? Thanks in advance. Regards, Dr. Nic Colledge ___ -- 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 Unregisteres IAX Friend Randomly
I have tried this with windows firewall both on and off - same problem. Thanks, Nic. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis Sent: 14 December 2009 14:53 To: 'Asterisk Users List' Subject: Re: [asterisk-users] Asterisk Unregisteres IAX Friend Randomly Are you sure this isn't a Windows zeroconfig problem? If Win drops the connection while * is talking to your client, the registration could drop too.. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nic Colledge Sent: Monday, December 14, 2009 9:33 AM To: Asterisk Users List Subject: Re: [asterisk-users] Asterisk Unregisteres IAX Friend Randomly Bump! And some more information (see below for initial problem): This problem is intermittent, but you don't have to wait long for it to happen. Also, sometimes when the reregister happens (and the client has been wrongly unregistered) asterisk sends the correct response to the client indicating this has been a success when the database is not updated with a new regseconds time. Any idea as to what I've done wrong / what's going on? Thanks in advance. Nic. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nic Colledge Sent: 11 December 2009 15:17 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk Unregisteres IAX Friend Randomly Hi, I've been having a strange problem recently where real-time asterisk will unregister a IAX friend at random times when the registration should not have expired. I have a Zoiper soft phone client (on windows) connecting to asterisk over a LAN (no firewalls). The default reregister time of 60 seconds is used, but the asterisk server unregisters the client (sets regseconds to 0 in the database) after a seemingly random time after registration say, 15 seconds (which is 45 seconds before registration expiry) I set iax2 debug on, and set the core debug level to 5 so I could see the IAX2 control frames on the console. I use pgAdmin-III to watch the value of regseconds in the database change from a registered value to 0. When the unregister happens, there are no frames sent to / received from the client, and nothing else on the asterisk console. It just seems like asterisk decided to unregister the client for no reason. At this point placing a call to the client will fail, until the client reregisters (at the correct time) 45 seconds later. I have seen this happen on both 1.6.2-rc8 and a trunk asterisk from yesterday (on different machines). I'm currently installing 1.6.0 to test that as well. This only seems to happen with real-time asterisk. (I'm using Postgres for the backend database and the pgsql driver in extconfig.conf) Any ideas what's going on here? Is this a known issue? Thanks in advance. Regards, Dr. Nic Colledge ___ -- 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 Unregisteres IAX Friend Randomly
Again, more info: Since I added rtcachefriends=yes this problem went away, but I don't really want the friends to be cached, because I want changed to be applied ASAP. Does anyone else have experience of the peers being unregistered before their time with rtcachefriends=no? Nic. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nic Colledge Sent: 14 December 2009 15:34 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Unregisteres IAX Friend Randomly I have tried this with windows firewall both on and off - same problem. Thanks, Nic. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis Sent: 14 December 2009 14:53 To: 'Asterisk Users List' Subject: Re: [asterisk-users] Asterisk Unregisteres IAX Friend Randomly Are you sure this isn't a Windows zeroconfig problem? If Win drops the connection while * is talking to your client, the registration could drop too.. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nic Colledge Sent: Monday, December 14, 2009 9:33 AM To: Asterisk Users List Subject: Re: [asterisk-users] Asterisk Unregisteres IAX Friend Randomly Bump! And some more information (see below for initial problem): This problem is intermittent, but you don't have to wait long for it to happen. Also, sometimes when the reregister happens (and the client has been wrongly unregistered) asterisk sends the correct response to the client indicating this has been a success when the database is not updated with a new regseconds time. Any idea as to what I've done wrong / what's going on? Thanks in advance. Nic. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nic Colledge Sent: 11 December 2009 15:17 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk Unregisteres IAX Friend Randomly Hi, I've been having a strange problem recently where real-time asterisk will unregister a IAX friend at random times when the registration should not have expired. I have a Zoiper soft phone client (on windows) connecting to asterisk over a LAN (no firewalls). The default reregister time of 60 seconds is used, but the asterisk server unregisters the client (sets regseconds to 0 in the database) after a seemingly random time after registration say, 15 seconds (which is 45 seconds before registration expiry) I set iax2 debug on, and set the core debug level to 5 so I could see the IAX2 control frames on the console. I use pgAdmin-III to watch the value of regseconds in the database change from a registered value to 0. When the unregister happens, there are no frames sent to / received from the client, and nothing else on the asterisk console. It just seems like asterisk decided to unregister the client for no reason. At this point placing a call to the client will fail, until the client reregisters (at the correct time) 45 seconds later. I have seen this happen on both 1.6.2-rc8 and a trunk asterisk from yesterday (on different machines). I'm currently installing 1.6.0 to test that as well. This only seems to happen with real-time asterisk. (I'm using Postgres for the backend database and the pgsql driver in extconfig.conf) Any ideas what's going on here? Is this a known issue? Thanks in advance. Regards, Dr. Nic Colledge ___ -- 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 Unregisteres IAX Friend Randomly
Hi, I've been having a strange problem recently where real-time asterisk will unregister a IAX friend at random times when the registration should not have expired. I have a Zoiper soft phone client (on windows) connecting to asterisk over a LAN (no firewalls). The default reregister time of 60 seconds is used, but the asterisk server unregisters the client (sets regseconds to 0 in the database) after a seemingly random time after registration say, 15 seconds (which is 45 seconds before registration expiry) I set iax2 debug on, and set the core debug level to 5 so I could see the IAX2 control frames on the console. I use pgAdmin-III to watch the value of regseconds in the database change from a registered value to 0. When the unregister happens, there are no frames sent to / received from the client, and nothing else on the asterisk console. It just seems like asterisk decided to unregister the client for no reason. At this point placing a call to the client will fail, until the client reregisters (at the correct time) 45 seconds later. I have seen this happen on both 1.6.2-rc8 and a trunk asterisk from yesterday (on different machines). I'm currently installing 1.6.0 to test that as well. This only seems to happen with real-time asterisk. (I'm using Postgres for the backend database and the pgsql driver in extconfig.conf) Any ideas what's going on here? Is this a known issue? Thanks in advance. Regards, Dr. Nic Colledge ___ -- 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] Channel Variable
Hi
I have been using the CHANNEL variable as a way of checking if a user is
allowed to make outgoing calls, and what their source caller ID should be
(these values are in a database).
This works all of the time with SIP and most of the time with IAX, however
sometimes with IAX the channel variable seems to be wrong.
I have been using Zoiper as my IAX client and Asterisk 1.6.2.0-rc6.
For the sake of debugging I have Verbose(1,Outgoing Call Handler
${CUT(CHANNEL,-,1)}) in the (internal - not default) dial plan.
Most of the time the channel variable is IAX2/10007 which is the desired
behaviour (with 10007 being the IAX username) but some of the time
IAX2/192.168.1.111:4569 is shown instead.
I would like to know why this is happening and if there is anything that can be
done to make it show the IAX2/10007 form every time?
I realise that I could use ${CDR(accountcode)} instead, and as it happens this
returns the correct account code value in both cases. However, I wanted to be
able to do this on a per-channel basis and multiple channels currently share a
common accountcode.
Any ideas what's going on here, is there something obvious I'm missing?
Thanks in advance.
Regards,
Nic
___
-- 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] GotoIfTime problem - possible bug
Hi, I'm currently doing some testing against asterisk 1.6.1.7-rc2 (keep meaning to upgrade) and am having a problem with the GotoIfTime dial plan function. The asterisk book says that day of week field can include the ampersand () to combine multiple days / day ranges but this gives me an error. For example monwed gives the error (in the asterisk console): [Nov 23 18:04:27] WARNING[11387]: pbx.c:6249 get_range: Invalid day of week 'monwed', assuming none Does anyone else have experience of this problem? Are there any patches / newer versions to get around this? Thanks in advance. Regards, Dr. Nic Colledge ___ -- 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] GotoIfTime problem - possible bug
Thanks very much, I'll fire up 1.6.2 and see how I go. Nic. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: 23 November 2009 18:20 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] GotoIfTime problem - possible bug On Monday 23 November 2009 12:11:02 pm Nic Colledge wrote: I'm currently doing some testing against asterisk 1.6.1.7-rc2 (keep meaning to upgrade) and am having a problem with the GotoIfTime dial plan function. The asterisk book says that day of week field can include the ampersand () to combine multiple days / day ranges but this gives me an error. For example monwed gives the error (in the asterisk console): [Nov 23 18:04:27] WARNING[11387]: pbx.c:6249 get_range: Invalid day of week 'monwed', assuming none Does anyone else have experience of this problem? Are there any patches / newer versions to get around this? That was an error on my part, when I helped review the book prior to publication. (I was incidentally thinking of the arguments to the CUT() function.) However, given that it was a good idea, it has been implemented in the forthcoming 1.6.2 release, currently in release candidate status. -- 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 ___ -- 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
