Re: [asterisk-users] Can I call myself on the same machine
On Tue, 23 Mar 2010, ayodele abejide wrote: From: cur...@telecomabmex.com To: asterisk-users@lists.digium.com Date: Mon, 22 Mar 2010 17:54:27 -0600 Subject: Re: [asterisk-users] Can I call myself on the same machine On Mon, 2010-03-22 at 23:40 +, ayodele abejide wrote: I am a newbie to asterisk, I have a complete installation of asterisk running on my ubuntu machine and I have x-lite installed also, I would like to know if I can call myself on the same machine, because whenever I try to call myself I get an engaged tone. My configuration file settigns are as below: If you plan to use a softphone on the same machine then you need to tell it to use a different port to listen for SIP. Asterisk and your softphone are both trying to listen to port 5060, that is why it always fails. I tried port 5061 for my softphone, but the same problem occurs Your soft-phone needs to be able to handle 2 concurrent calls (ie. have call waiting enabled) Not sure if the free version of x-lite allows this. Gordon -- _ -- 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] chan_ss7 issue
Dear all, Do you have come acrross with this issue. My ss7 link get fluctuating. It use chan_ss7 version 1.0.95-beta. I have 8 E1s running on a DL380 server. This enable to have calls from sip to ss7 and vice versa. However ss7 links are not stable. linkset siuc, link l1, schannel 1, sls 0, NOT_ALIGNED, rx: 1, tx: 2/4, sentseq/lastack: 127/127, total 4034145216, 4031118560 linkset siuc, link l5, schannel 1, sls 1, INSERVICE, rx: 5, tx: 3/3, sentseq/lastack: 95/95, total 4030833616, 4028245568 ^[[a[r...@localhost ~]# asterisk -rx ss7 link status linkset siuc, link l1, schannel 1, sls 0, NOT_ALIGNED, rx: 2, tx: 4/4, sentseq/lastack: 127/127, total 4034149872, 4031123216 linkset siuc, link l5, schannel 1, sls 1, INSERVICE, rx: 2, tx: 3/3, sentseq/lastack: 100/100, total 4030838272, 4028250224 ^[[a[r...@localhost ~]# asterisk -rx ss7 link status linkset siuc, link l1, schannel 1, sls 0, NOT_ALIGNED, rx: 1, tx: 3/4, sentseq/lastack: 127/127, total 4034154480, 4031127824 linkset siuc, link l5, schannel 1, sls 1, DOWN, rx: 5, tx: 4/4, sentseq/lastack: 100/101, total 4030842880, 4028254832 ^[[a[r...@localhost ~]# asterisk -rx ss7 link status linkset siuc, link l1, schannel 1, sls 0, NOT_ALIGNED, rx: 3, tx: 1/4, sentseq/lastack: 127/127, total 4034159456, 4031132800 linkset siuc, link l5, schannel 1, sls 1, DOWN, rx: 0, tx: 0/4, sentseq/lastack: 100/101, total 4030847840, 4028259792 ^[[a[r...@localhost ~]# asterisk -rx ss7 link status linkset siuc, link l1, schannel 1, sls 0, NOT_ALIGNED, rx: 4, tx: 4/4, sentseq/lastack: 127/127, total 4034164432, 4031137776 linkset siuc, link l5, schannel 1, sls 1, NOT_ALIGNED, rx: 2, tx: 4/4, sentseq/lastack: 127/127, total 4030852816, 4028264768 ^[[a[r...@localhost ~]# asterisk -rx ss7 link status linkset siuc, link l1, schannel 1, sls 0, NOT_ALIGNED, rx: 5, tx: 3/4, sentseq/lastack: 127/127, total 4034169312, 4031142640 linkset siuc, link l5, schannel 1, sls 1, PROVING, rx: 4, tx: 2/4, sentseq/lastack: 127/127, total 4030857696, 4028269632 [r...@localhost ~]# asterisk -rx ss7 link status And I get a log as [Mar 23 14:18:40] NOTICE[30389] mtp.c: Empty Zaptel output buffer detected, outgoing packets may have been lost on link 'l1'. [Mar 23 14:18:40] NOTICE[30389] mtp.c: Full Zaptel input buffer detected, incoming packets may have been lost on link 'l1' (count=64. [Mar 23 14:18:40] NOTICE[30389] mtp.c: Full Zaptel input buffer detected, incoming packets may have been lost on link 'l5' (count=64. [Mar 23 14:18:40] NOTICE[30389] mtp.c: Empty Zaptel output buffer detected, outgoing packets may have been lost on link 'l5'. [Mar 23 14:18:40] NOTICE[30389] mtp.c: Excessive poll delay 12718! [Mar 23 14:18:40] NOTICE[30389] mtp.c: Full Zaptel input buffer detected, incoming packets may have been lost on link 'l1' (count=64. [Mar 23 14:18:40] NOTICE[30389] mtp.c: Full Zaptel input buffer detected, incoming packets may have been lost on link 'l5' (count=64. [Mar 23 14:18:40] NOTICE[30389] mtp.c: Empty Zaptel output buffer detected, outgoing packets may have been lost on link 'l1'. [Mar 23 14:18:40] NOTICE[30389] mtp.c: Empty Zaptel output buffer detected, outgoing packets may have been lost on link 'l5'. [r...@localhost ~]# Can anybody help me on this. It will be great help. Kind Rgds Daminda -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which folder for sounds?
On Mon, Mar 22, 2010 at 09:38:26PM -0400, sean darcy wrote: 1.6.2: -- Executing [...@incoming-pstn-line:4] VoiceMail(DAHDI/4-1, 1...@default,u) in new stack -- DAHDI/4-1 Playing '/var/spool/asterisk/voicemail/default/100/unavail.gsm' (language 'en') [Mar 22 17:15:46] WARNING[31145]: file.c:650 ast_openstream_full: File vm-intro does not exist in any format [Mar 22 17:15:46] WARNING[31145]: file.c:953 ast_streamfile: Unable to open vm-intro (format 0x4 (ulaw)): No such file or directory But: locate vm-intro /var/lib/asterisk/sounds/en/vm-intro.gsm /var/lib/asterisk/sounds/en/vm-intro.ulaw /var/lib/asterisk/sounds/en/vm-intro.wav head -12 /etc/asterisk/asterisk.conf [directories](!) ; remove the (!) to enable this As long as it is not enabled, the compile-time defaults are used. astetcdir = /etc/asterisk astmoddir = /usr/lib64/asterisk/modules astvarlibdir = /var/lib/asterisk astdbdir = /var/lib/asterisk astkeydir = /var/lib/asterisk astdatadir = /var/lib/asterisk astagidir = /var/lib/asterisk/agi-bin astspooldir = /var/spool/asterisk astrundir = /var/run/asterisk astlogdir = /var/log/asterisk So in which folder are these sounds supposed to be? Unless you specifid the full path explicitly, the sound files are looked for in the following pathes (in the following order). Suppose you wanted the sound file called: 'somewhere/soundname' DATADIR/sounds/LANG_FULL/somewhere/soundname DATADIR/sounds/LANG/somewhere/soundname DATADIR/sounds/somewhere/soundname DATADIR/sounds/DEFAULTLANG/soundname DATADIR defaults to /var/lib/asterisk (unless you use the Debian/Ubuntu packages...) and can be set in asterisk.conf otherwise. LANG_FULL is the complete value of LANGUAGE, if it is set. LANG is the value of LANGUAGE sliced after the first '_'. For instance, if LANG_FULL was 'en_US_Whatever', LANG will be 'en'. DEFAULTLANG defaults to 'en' and can be set to a default value in asterisk.conf. Normally people don't change it. Hmm... 'core show settings' does not show datadir. Should it? -- 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 -- 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-ss7]Chan_ss7 issue
Dear all, Do you have come acrross with this issue. My ss7 link get fluctuating. It use chan_ss7 version 1.0.95-beta. I have 8 E1s running on a DL380 server with Digium E1 cards ( 4 port cards). This enable to have calls from sip to ss7 and vice versa. However ss7 links are not stable. linkset siuc, link l1, schannel 1, sls 0, NOT_ALIGNED, rx: 1, tx: 2/4, sentseq/lastack: 127/127, total 4034145216, 4031118560 linkset siuc, link l5, schannel 1, sls 1, INSERVICE, rx: 5, tx: 3/3, sentseq/lastack: 95/95, total 4030833616, 4028245568 ^[[a[r...@localhost ~]# asterisk -rx ss7 link status linkset siuc, link l1, schannel 1, sls 0, NOT_ALIGNED, rx: 2, tx: 4/4, sentseq/lastack: 127/127, total 4034149872, 4031123216 linkset siuc, link l5, schannel 1, sls 1, INSERVICE, rx: 2, tx: 3/3, sentseq/lastack: 100/100, total 4030838272, 4028250224 ^[[a[r...@localhost ~]# asterisk -rx ss7 link status linkset siuc, link l1, schannel 1, sls 0, NOT_ALIGNED, rx: 1, tx: 3/4, sentseq/lastack: 127/127, total 4034154480, 4031127824 linkset siuc, link l5, schannel 1, sls 1, DOWN, rx: 5, tx: 4/4, sentseq/lastack: 100/101, total 4030842880, 4028254832 ^[[a[r...@localhost ~]# asterisk -rx ss7 link status linkset siuc, link l1, schannel 1, sls 0, NOT_ALIGNED, rx: 3, tx: 1/4, sentseq/lastack: 127/127, total 4034159456, 4031132800 linkset siuc, link l5, schannel 1, sls 1, DOWN, rx: 0, tx: 0/4, sentseq/lastack: 100/101, total 4030847840, 4028259792 ^[[a[r...@localhost ~]# asterisk -rx ss7 link status linkset siuc, link l1, schannel 1, sls 0, NOT_ALIGNED, rx: 4, tx: 4/4, sentseq/lastack: 127/127, total 4034164432, 4031137776 linkset siuc, link l5, schannel 1, sls 1, NOT_ALIGNED, rx: 2, tx: 4/4, sentseq/lastack: 127/127, total 4030852816, 4028264768 ^[[a[r...@localhost ~]# asterisk -rx ss7 link status linkset siuc, link l1, schannel 1, sls 0, NOT_ALIGNED, rx: 5, tx: 3/4, sentseq/lastack: 127/127, total 4034169312, 4031142640 linkset siuc, link l5, schannel 1, sls 1, PROVING, rx: 4, tx: 2/4, sentseq/lastack: 127/127, total 4030857696, 4028269632 [r...@localhost ~]# asterisk -rx ss7 link status And I get a log as [Mar 23 14:18:40] NOTICE[30389] mtp.c: Empty Zaptel output buffer detected, outgoing packets may have been lost on link 'l1'. [Mar 23 14:18:40] NOTICE[30389] mtp.c: Full Zaptel input buffer detected, incoming packets may have been lost on link 'l1' (count=64. [Mar 23 14:18:40] NOTICE[30389] mtp.c: Full Zaptel input buffer detected, incoming packets may have been lost on link 'l5' (count=64. [Mar 23 14:18:40] NOTICE[30389] mtp.c: Empty Zaptel output buffer detected, outgoing packets may have been lost on link 'l5'. [Mar 23 14:18:40] NOTICE[30389] mtp.c: Excessive poll delay 12718! [Mar 23 14:18:40] NOTICE[30389] mtp.c: Full Zaptel input buffer detected, incoming packets may have been lost on link 'l1' (count=64. [Mar 23 14:18:40] NOTICE[30389] mtp.c: Full Zaptel input buffer detected, incoming packets may have been lost on link 'l5' (count=64. [Mar 23 14:18:40] NOTICE[30389] mtp.c: Empty Zaptel output buffer detected, outgoing packets may have been lost on link 'l1'. [Mar 23 14:18:40] NOTICE[30389] mtp.c: Empty Zaptel output buffer detected, outgoing packets may have been lost on link 'l5'. [r...@localhost ~]# Can anybody help me on this. It will be great help. Kind Rgds Daminda -- _ -- 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] Better SIP security please! Was: (no subject)
21 mar 2010 kl. 18.22 skrev Philipp von Klitzing: Hi Olle! The work I started during Christmas - Named ACL's - is a starting point that other developers can use to develop all kind of schemes. http://www.voip-forum.com/asterisk/2010-01/manageable-access-control-lists -asterisk-nacls/ Very interesting. Doesn't look like this has any chance to secure 1.4 installations though, I am afraid. The code was written both for trunk and 1.4. It won't be included in 1.4 release though, right. /O -- _ -- 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] SIP codec negotiation / manipulation
22 mar 2010 kl. 14.54 skrev Kevin Sandy: On 3/21/2010 4:05 AM, Olle E. Johansson wrote: 17 mar 2010 kl. 16.37 skrev Kevin Sandy: We're having an odd issue with codec negotiation from one of our SIP providers. Here's the basic situation. We receive an invite from them advertising support for G711, G729, and G723. In our response, we send back that we support G711 and G729. In about half the cases, this results in no problems, with audio being encoded with G711. The other half of the time, they send us a second invite requesting G729. However, they proceed to send us a G711 encoded audio stream... They have somewhat acknowledged the problem, but their advice is for us to only accept a single codec in our 200 OK. We don't want to disable either; we have customers using G729, so we'd like to avoid transcoding when possible, but we also do some T38 faxing, which I believe requires G711 to start off. My first thought was to selectively force the codec on inbound calls - if it is for a voice number, use 729, otherwise 711. However, I can't find any way of doing this within Asterisk. (We do have an OpenSIPS server sitting between us and the provider, and I could use OpenSIPS features to do this; however, right now the OpenSIPS server is fairly dumb - it's only proxying traffic between us and the provider and knows nothing about our specific DIDs.) A couple more details in case anyone has seen a similar issue. The provider is Broadvox, and this issue only seems to manifest on calls coming to them via Skype. They claim to not have any direct link with Skype, but it seems odd that the problem would be specific to Skype callers if the call is coming to Broadvox as a standard PSTN call. Is there any way to do this? Am I totally missing something and making a stupid mistake, or making the issue more complicated than it needs to be? The problem here is that you have a proxy in between, so Asterisk can't have separate peer configurations, since all the SIP messages are from the same IP and thus the same peer. I have a branch that implements peer matching in this specific configuration, which means that you can have different codec configurations for different partners even though there's a proxy in front of Asterisk. https://origsvn.digium.com/svn/asterisk/team/oej/pinetree-1.4 Please try this branch and give feedback. There should be some docs in sip.conf for the new matchrule setting. /O I'd be interested in trying this out - but the site doesn't seem to be responding. :) Sorry, gave you the developer URL. Too quick copy and paste... Here's a correct one: http://svn.digium.com/svn/asterisk/team/oej/pinetree-1.4 /O -- _ -- 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] Integrate a CPE with Asterisk in MGCP
[020202020202] context=mgcp host=dynamic canreinvite=no dtmfmode=rfc2833 nat=yes threewaycalling=yes transfer=yes ; transfer requires threewaycalling=yes. Use FLASH to transfer callwaiting=yes ; this might be a cause of trouble for ip10s cancallforward=yes line=aaln/1 Order of variables is important. Try this configuration: [020202020202] nat=yes canreinvite=no context=mgcp callerid=020202020202 host=dynamic canreinvite=no dtmfmode=rfc2833 threewaycalling=yes transfer=yes; transfer requires threewaycalling=yes. Use FLASH to transfer callwaiting=yes ; this might be a cause of trouble for ip10s cancallforward=yes ;wcardep = * ; This option is required by some devices line= aaln/1 line= virtual/nat-timeout -- _ -- 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] Install dahdi on Xen virtual console
Hi, We are trying compile dahdi on amazon vertual instance. When we are compiling dahdi we receieve following error. You do not appear to have the sources for the 2.6.21.7-2.fc8xen kernel installed. We are helpless on getting this 2.6.21.7-2 sources. Please help to get this compile. -- Thanks Regards, Vidura Senadeera, Sri Lanka. msn/yahoo/skype Ids - vidurased -- _ -- 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] Install dahdi on Xen virtual console
On Tue, Mar 23, 2010 at 06:27:40PM +0530, Vidura Senadeera wrote: Hi, We are trying compile dahdi on amazon vertual instance. When we are compiling dahdi we receieve following error. You do not appear to have the sources for the 2.6.21.7-2.fc8xen kernel installed. We are helpless on getting this 2.6.21.7-2 sources. Please help to get this compile. Look under http://archive.kernel.org/fedora-archive/releases/8/Fedora/ -- 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 -- 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] Install dahdi on Xen virtual console
Hi there , are you using any king of Iax trunk or Duguim interface on this VM? Because if is just for sip you dont need dahdi you can compile asterisk and work on it. Daniel Abreu On 23 Mar 2010, at 12:57 PM, Vidura Senadeera wrote: helpless on getting this 2.6.21.7-2 sources. Please help to get this co -- _ -- 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] Install dahdi on Xen virtual console
- Daniel Leite de Abreu dlab...@gmail.com escreveu: Hi there , are you using any king of Iax trunk or Duguim interface on this VM? Because if is just for sip you dont need dahdi you can compile asterisk and work on it. He will need DAHDI if he plans on using MeetMe(). Also, internal timing is only available on 1.6+. So if he plans on running 1.4, he will need DAHDI in order to playback audio files. -- _ -- 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] Install dahdi on Xen virtual console
While working with Rhino hardware I was told by their technical support that no virtual machine software gives access to the PCI bus, so using zaptel or dahdi is not an option over the virtual machines. Although somebody has said otherwise on this list but make sure you actually have access to the dahdi hardware from within your virutal machines before trying to compile these drivers. If it is possible to do so, it would help me too. -- Zeeshan A Zakaria On 2010-03-23 9:20 AM, Daniel Leite de Abreu dlab...@gmail.com wrote: Hi there , are you using any king of Iax trunk or Duguim interface on this VM? Because if is just for sip you dont need dahdi you can compile asterisk and work on it. Daniel Abreu On 23 Mar 2010, at 12:57 PM, Vidura Senadeera wrote: helpless on getting this 2.6.21.7-2 sources... -- _ -- Bandwidth and Colocatio... -- _ -- 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] Using asterisk as avaya definity recordingserver
On Mon, Mar 22, 2010 at 7:56 PM, Rafael Prado Rocchi pr...@practis.com.brwrote: Hi, it's not that simple. It requires deep modification on asterisk and dahdi sources to work the way you want. Why? I must confess I still don't quite understand what he wants, from what I've read the legacy pbx will place a secondary call via ISDN ( did he mean PRI? ) therefore Asterisk will just Record(), what is it that is not so simple about that? -- Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Install dahdi on Xen virtual console
For that ztdummy does the work fine without worrying about enabling the actual hardware. On 2010-03-23 9:49 AM, VinÃcius Fontes vinic...@canall.com.br wrote: - Daniel Leite de Abreu dlab...@gmail.com escreveu: Hi there , are you using any king of Iax trunk or Duguim interface on this VM? Because if ... He will need DAHDI if he plans on using MeetMe(). Also, internal timing is only available on 1.6+. So if he plans on running 1.4, he will need DAHDI in order to playback audio files. -- _ -- Bandwidth and Colocati... -- _ -- 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 crash - segmentation fault
My Asterisk 1.2.40 process crashes regularly in the is_zero_or_null function at: return (*vp-u.s == 0 || (to_integer (vp) vp-u.i == 0)); My gdb trace is at: http://pastebin.com/raw.php?i=hmhzZxye Other examples here: http://lists.digium.com/pipermail/asterisk-users/2010-March/245927.html Can anyone please help? -- _ -- 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 make upgrades with Asterisk
Thanks Zeeshan, In fact,i have RealTime configured and working... What i want is to make an upgrade of libpri and wanpipe at least, asterisk and zaptel will be like i have now... Do you think that recompile/upgrade this softwares version will produce a problem? what steps should i do? Is it necessary to recompile asterisk if i make an upgrade of libpri? this recompilation will affect the realtime or the well bahavior of the server? Working with an old version of asterisk like 1.4.21.2 with the newest version of libpri is recommended or not? Thanks in advance for all your advices! Message: 11 Date: Mon, 22 Mar 2010 23:15:00 -0400 From: Zeeshan Zakaria zisha...@gmail.com Subject: Re: [asterisk-users] How to make upgrades with Asterisk To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: 5ad99e891003222015o1727265cr4860d4e8a0b96...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 If it is a production server, you should not do the upgrade on it. Setup a new server with upgraded software, migrate all the data, test it and make sure it works fine. There are things like CDR and voicemail which are constantly being updated, meaning just before the final migration, you should copy them to the new server. I have done some migrations and have found clonezilla to be a wonderful tool for this purpose. You create a second server on any computer, and once it is ready, clone it on a USB stick or CD, or on another computer via SSH, then clone the production server for backup purposes on a medium of your choice, and finally restore this new server image on to the production server. If something goes wrong, you'll be able to restore the server back to its functional state from the backup cloned image. When I migrated my own production server from 1.2 to 1.4, I did the rehearsal many times, and very carefully drafted the whole migation plan. This also included asking all the users to copy and delete their voicemails before the day of migration. It took me about two weeks in planning and making sure every single setting will be migrated, before I was comfotable to do the migration, which took hardly an hour, and went just perfectly smooth. Personally I am of the opinion that if it is not really necessary, don't upgrade it. Will 1.6 give you something which you don't have in 1.4? It'll have its own issues and learning curve. I tried it once and it was only a pain for my setup, specially with real-time architecture, and a few other things which I can't remember now. -- Zeeshan A Zakaria -- _ -- 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] distribuited ACD on many asterisk nodes
Dear All i'm planning to develop for a customer a particular implementation of Asterisk. The aim of the project is to share different users between different Asterisk inbound call center . I'm planning to have a sync for some of the QueueMemberStatus informations between all the nodes, then a particular (external) ACD algorithm will decide to transfer a parked call to the final user. I want to trigger an action when an event of type QueueMemberStatus is detected on the manager socket, and then propagate this information to the other Asterisk nodes using some XMPP features or something else. This architecture allows to share users between different call center without having a complete replication of all the nodes (each node can decide how much resources give to the cluster of call center). So each node can have its own configuration and requires only a manager access to share users information and thansfer call. Do you know if there is something similar somewhere ? Maybe Asterisk has already some magic sauce to do that ? ;-) Thanks to all -- /*/ nik600 http://www.kumbe.it -- _ -- 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 make upgrades with Asterisk
Hi, Do clone the existing server in any case. I am not sure about libpri but zaptel I have recompiled various times without recompiling asterisk, so I think it should be ok with libpri too. Wanpipe doesn't interact with asterisk, but with zaptel only so it should not be a problem to update it. None of these upgrades will affect realtime. However switching to asterisk 1.6 should be done only on a separate machine with extreme care and detail, as I mentioned earlier. If you really want to take risk on your production machine, setup an 8 hour window at night and inform users of the maintenance schedule. Then do it as follows: 1. Clone the existing machine. But learn conezilla first, it is very easy, check it on check my instructions on www.ilovetovoip.com. 2. Update libpri and wanpipe on your server and test that it hasn't broken anything. 3. If you are not satisfied with the update, restore the server to its original state from the cloned image. 4. If everything works fine, leave it as it is. In a day or two you'll get update from your users if there will be any issues, based on which you can decide whether to roll back or not. -- Zeeshan A Zakaria On 2010-03-23 10:56 AM, Danny Dias ing.diasda...@gmail.com wrote: Thanks Zeeshan, In fact,i have RealTime configured and working... What i want is to make an upgrade of libpri and wanpipe at least, asterisk and zaptel will be like i have now... Do you think that recompile/upgrade this softwares version will produce a problem? what steps should i do? Is it necessary to recompile asterisk if i make an upgrade of libpri? this recompilation will affect the realtime or the well bahavior of the server? Working with an old version of asterisk like 1.4.21.2 with the newest version of libpri is recommended or not? Thanks in advance for all your advices! Message: 11 Date: Mon, 22 Mar 2010 23:15:00 -0400 From: Zeeshan Zakaria zisha...@gmail.com Subject: Re: [asterisk-users] How to make upgrades with Asterisk To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: 5ad99e891003222015o1727265cr4860d4e8a0b96...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 If it is a production server, you should not do the upgrade on it. Setup a new server with up... -- _ -- 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] Minimalize jitter in VoIP calls
Hello list, what can I do to minimalize the jitter in SIP-calls at server level ? If at local network level, there is a VoIP-router and their is a physical network dedicated to IP-phones, but there is still jitter. When using a Hosted Asterisk server, which settings on the Asterisk-server can minimalize the jitter between the VoIP-router and the Asterisk-server on the public internet ?? Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Classic NO AUDIO problem - DD-WRT and NAT forwarding - HELP PLEASE!
Hi Everyone, I have tried to set the box to DMZ and also tried to port forward 5060 TCP/UDP and 1/2 UDP to the server IP but it's no use and there is a no audio issue. I am pretty certain it's a NAT issue as the sip call establishes. I also made a succesful IAX2 call through IAX trunking and zap lines on this server but sip doesn't work. I register to an extension but even dialing *97 for voicemail wont' give me any audio. Picture posted here shows my DD-WRT NAT setting: *http://tinypic.com/r/21cuqlu/5* Any input will be much appreciated. This is running latest PBXinaFLASH (which has FreePBX) and I tried using externip=x.x.x.x/255.255.248.0 in /etc/asterisk/sip_nat.conf but it was of no use. Thanks, Bruce -- _ -- 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 crash - segmentation fault
--- On Tue, 3/23/10, Vieri rentor...@yahoo.com wrote: My Asterisk 1.2.40 process crashes regularly in the is_zero_or_null function at: return (*vp-u.s == 0 || (to_integer (vp) vp-u.i == 0)); My gdb trace is at: http://pastebin.com/raw.php?i=hmhzZxye Other examples here: http://lists.digium.com/pipermail/asterisk-users/2010-March/245927.html Can anyone please help? And my Asterisk log shows the following right before the crash: Mar 23 12:32:37 VERBOSE[9054] logger.c: -- Executing ExecIf(SIP/4070-09464648, 0|Set|REALCALLERIDNUM=4070) in new stac k Mar 23 12:32:37 DEBUG[9054] app_macro.c: Executed application: ExecIf Mar 23 12:32:37 DEBUG[9054] pbx.c: Expression result is '0' Mar 23 12:32:37 DEBUG[9054] pbx.c: Function result is '4070' Mar 23 12:32:37 DEBUG[9054] pbx.c: Expression result is '0' Mar 23 12:32:37 DEBUG[9054] pbx.c: Expression result is '1' Mar 23 12:32:37 DEBUG[9054] pbx.c: Expression result is '0' Mar 23 12:32:37 WARNING[9054] ast_expr2.y: Conversion of 0 to integer under/overflowed! What does this mean? -- _ -- 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] In Berlin this week? Kamailio/Asterisk community dinner on Thursday
Friends, Daniel and I are running a Kamailio SIP Masterclass this week in Berlin. When travelling around like this, we often invite the community to come and meet us in a nice restaurant. We offer good company and fun discussions about Kamailio, SIP-router.org and Asterisk - but the drinks and food are on you. At least yours :-) Berlin is the city where Sip Express Router was born. Many SER/SIP-router and Kamailio developers live here, so we suspect that you'll find a good set of core developers joining us. Hint: Buying a beer for a developer is generally considered a good thing. Buying too many will affect the commits the next day... The bad code submissions can be reverted easily, so don't worry about it. We'll just have to handle the situation... - Where? The Lemke Brauhaus, Luisenplatz1, 10585 Berlin (close to Schloss Charlottenburg). - Time? 19.00 Berlin time - URL: http://www.brauhaus-lemke.com/index.php?area=4 Please send me a not off-list if you think you can participate, so that we can get a properly sized table. If you want to take a chance, just show up. Either way, you're welcome! This is also a good way to prepare for the VoipAthon - the 24 hour Voip Users Group session. Don't miss that! http://voipathon.org/ The next Asterisk SIP Masterclass will be hosted by Telespeak in the UK. Check their web site for information! I suspect we can find beer or someting compatible in that area too :-) Regards, /Olle -- _ -- 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] Strange Meetme disconnects
Running * version 1.6.1.17. My meetme conferences automagically disconnect users approximately 5-15 seconds after the user is connected. This occurs regardless of whether music on hold is active or not. [Mar 23 11:34:36] -- Executing Macro(SIP/SDN_TMCKEE-00e9, confroom,1808) [Mar 23 11:34:36] -- Executing [...@macro-confroom:1] Answer(SIP/SDN_TMCKEE-00e9, ) in new stack [Mar 23 11:34:36] -- Executing [...@macro-confroom:2] Wait(SIP/SDN_TMCKEE-00e9, 1) in new stack [Mar 23 11:34:37] -- Executing [...@macro-confroom:3] MeetMe(SIP/SDN_TMCKEE-00e9, 1808,PpcMs) in new stack [Mar 23 11:34:37] == Parsing '/etc/asterisk/meetme.conf': [Mar 23 11:34:37] == Found [Mar 23 11:34:37] -- Created MeetMe conference 1013 for conference '1808' [Mar 23 11:34:37] -- SIP/SDN_TMCKEE-00e9 Playing 'conf-onlyperson.ulaw' (language 'en') [Mar 23 11:34:41] -- Started music on hold, class 'default', on SIP/SDN_TMCKEE-00e9 [Mar 23 11:34:41] -- Stopped music on hold on SIP/SDN_TMCKEE-00e9 [Mar 23 11:34:42] -- Started music on hold, class 'default', on SIP/SDN_TMCKEE-00e9 [Mar 23 11:34:46] -- Executing [...@macro-queue:8] Queue(SIP/CLTPBX-00e8, HDSK-QUEUE,tck,,,300) in new stack [Mar 23 11:34:46] -- Started music on hold, class 'default', on SIP/CLTPBX-00e8 [Mar 23 11:35:00] -- Hungup 'DAHDI/pseudo-1141253725' [Mar 23 11:35:00] == Spawn extension (macro-confroom, s, 3) exited non-zero on 'SIP/SDN_TMCKEE-00e9' in macro 'confroom' [Mar 23 11:35:00] == Spawn extension (sdn-dialout, 1808, 1) exited non-zero on 'SIP/SDN_TMCKEE-00e9' [Mar 23 11:35:00] -- Stopped music on hold on SIP/SDN_TMCKEE-00e9 Does anyone have any clues? I'm using the DAHDI Dummy timing source on this machine. pbx01*CLI dahdi show status Description Alarms IRQbpviol CRC4 Fra Codi Options LBO DAHDI_DUMMY/1 (source: HRtimer) 1UNCONFI 0 0 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1) Tim McKee -- _ -- 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] Sip module and dns
Hi , I had some problems in the past with sip trunks, asterisk-users Digest, Vol 68, Issue 4, message 6, and had a reply (message 9) saying that It could be a dns issue. Well today I had a problem again with sip module and it really seams a dns issue. I have an asterisk, version 1.4.26.1, that has 4 bri access and two sip trunks. I'm having internet access problems and when this happens and if one of the trunks tries to reregister its panic time!!! All the sip peers goes unreachable, trunks and phones, and the sip module freezes, sip reload takes many many time to act. My solution is to remove the sip trunks from the configuration. But why this happens? Why If there is no dns resolution of the trunks sip module freezes ? This is more strange because if by some reason the internet is down but still exists dns cache all is ok. (of course sip trunks unreachable) This is supposed to be like this? There is no dns tunning for asterisk+sip? To avoid this I'm starting to think putting bind in the asterisk server and publishing there the zones of the sip trunks. (Or instead of names start using the ip's) Any comments? Regards, Luis Silva -- _ -- 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] G.711a or G.711u ???
Dear all, I have an Asterisk SIP server in a LAN environment and I want your opinion in order to decide the use of an audio codec: What audio codec is better, G.711a or G.711u ??? Which suites to my LAN voip calls ??? Thank you !!! Alejandro -- _ -- 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] G.711a or G.711u ???
On 24/03/10 8:41 AM, Alejandro Cabrera Obed wrote: Dear all, I have an Asterisk SIP server in a LAN environment and I want your opinion in order to decide the use of an audio codec: What audio codec is better, G.711a or G.711u ??? Which suites to my LAN voip calls ??? It basically comes down to where the system is being used and what codecs you're using upstream. G.711a is aLaw and G.711u is uLaw. uLaw is predominantly used in the USA. aLaw is used in most of the rest of the world (although I think Japan might use uLaw). If you're using an ISDN card then it will be talking aLaw or uLaw depending on where you are. The idea is to avoid transcoding - i.e. converting between one format and another. So, if you're using a VoIP provider instead of ISDN, it will depend on what they're using. If your VoIP provider is outside of the US and accepts aLaw, then that's likely what you want to use (bear in mind that they might still use an upstream provider who uses G.729 etc). Easiest option is to just choose aLaw or uLaw based on your country. -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- 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] G.711a or G.711u ???
Doesn't really matter unless you want to use zaptel lines on one end and VoIP lines on the other end, and want to avoid mismatch between your telco and your IP phones codecs. Transcoding between aLaw and uLaw in my experience can degrade voice quality. If you are using zaptel, and your telco supports aLaw, then use aLaw otherwise use uLaw. uLaw is North American standard and aLaw almost rest of the world. -- Zeeshan A Zakaria On 2010-03-23 3:48 PM, Alejandro Cabrera Obed aco1...@gmail.com wrote: Dear all, I have an Asterisk SIP server in a LAN environment and I want your opinion in order to decide the use of an audio codec: What audio codec is better, G.711a or G.711u ??? Which suites to my LAN voip calls ??? Thank you !!! Alejandro -- _ -- 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] G.711a or G.711u ???
2010/3/23 Alejandro Cabrera Obed aco1...@gmail.com: Dear all, I have an Asterisk SIP server in a LAN environment and I want your opinion in order to decide the use of an audio codec: What audio codec is better, G.711a or G.711u ??? Which suites to my LAN voip calls ??? That depends most on the codec used in your country IF you terminate calls to ISDN. In th USA (and some other countries) you would use G711u and in Europe etc. you would use G711a. Chris -- _ -- 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] G.711a or G.711u ???
fwiw, they both 'sound' the same, so if your question was specifically about not pest-terminated calls it still doesn't matter. :-) For that you could use something like g.722 provided your endpoints support it. In that case it really DOES make a big difference and the bandwidth is the same. -Karl - Original Message - From: Alejandro Cabrera Obed aco1...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, March 23, 2010 2:41 PM Subject: [asterisk-users] G.711a or G.711u ??? Dear all, I have an Asterisk SIP server in a LAN environment and I want your opinion in order to decide the use of an audio codec: What audio codec is better, G.711a or G.711u ??? Which suites to my LAN voip calls ??? Thank you !!! Alejandro -- _ -- 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 crash - segmentation fault
On Tuesday 23 March 2010 11:53:03 Vieri wrote: --- On Tue, 3/23/10, Vieri rentor...@yahoo.com wrote: My Asterisk 1.2.40 process crashes regularly in the is_zero_or_null function at: return (*vp-u.s == 0 || (to_integer (vp) vp-u.i == 0)); My gdb trace is at: http://pastebin.com/raw.php?i=hmhzZxye Other examples here: http://lists.digium.com/pipermail/asterisk-users/2010-March/245927.html Can anyone please help? And my Asterisk log shows the following right before the crash: Mar 23 12:32:37 VERBOSE[9054] logger.c: -- Executing ExecIf(SIP/4070-09464648, 0|Set|REALCALLERIDNUM=4070) in new stac k Mar 23 12:32:37 DEBUG[9054] app_macro.c: Executed application: ExecIf Mar 23 12:32:37 DEBUG[9054] pbx.c: Expression result is '0' Mar 23 12:32:37 DEBUG[9054] pbx.c: Function result is '4070' Mar 23 12:32:37 DEBUG[9054] pbx.c: Expression result is '0' Mar 23 12:32:37 DEBUG[9054] pbx.c: Expression result is '1' Mar 23 12:32:37 DEBUG[9054] pbx.c: Expression result is '0' Mar 23 12:32:37 WARNING[9054] ast_expr2.y: Conversion of 0 to integer under/overflowed! What does this mean? It's quite clearly a bug, but given that 1.2 is in security maintenance mode, it's not a bug that will ever be fixed in an official release of Asterisk. Your best bet is to bite the bullet and upgrade to 1.4. -- Tilghman -- _ -- 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] G.711a or G.711u ???
On Tue, Mar 23, 2010 at 1:41 PM, Alejandro Cabrera Obed aco1...@gmail.com wrote: Dear all, I have an Asterisk SIP server in a LAN environment and I want your opinion in order to decide the use of an audio codec: What audio codec is better, G.711a or G.711u ??? Which suites to my LAN voip calls ??? As others have stated, use whatever matches your country. In terms of the network, the traffic is exactly the same for both a-law and u-law. Both are 8khz sampled 8 bits per sample PCM (64kbps). The only difference between the two is the companding (http://en.wikipedia.org/wiki/Companding) algorithm that's used for the audio emphasis. To the network, the packets are 100% identical. Andrew Thank you !!! Alejandro -- _ -- 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] permit/deny in sip.conf iax.conf
Does anyone know the rationale behind why deny/permit values can not be specified in 'general' setting of sip.conf iax.conf In other words, if I want to deny everyone, then allow selectively permit specific hosts or subnets, I can't do so without first deny'ing all in EVERY user/peer definition. Too verbose. Naturally I can accomplish this using templates, but it DOES seem a bit odd since [general] is in essence ALREADY a 'template' of sorts for all parameters not otherwise specified. Is there an architectural reason for this or do I misunderstand a concept somewhere? -K -- _ -- 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] permit/deny in sip.conf iax.conf
Karl Fife wrote: Naturally I can accomplish this using templates, but it DOES seem a bit odd since [general] is in essence ALREADY a 'template' of sorts for all parameters not otherwise specified. You should use templates; the [general] section never should been an implicit template, but it has been that way forever so we can't change it. The [general] section *should* have only been for settings that apply to the SIP channel driver as a whole, and *not* for providing defaults to entities configured for the driver. Unfortunately, it has both purposes. -- 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 -- 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] permit/deny in sip.conf iax.conf
Karl Fife wrote: Naturally I can accomplish this using templates, but it DOES seem a bit odd since [general] is in essence ALREADY a 'template' of sorts for all parameters not otherwise specified. On Tue, 23 Mar 2010, Kevin P. Fleming wrote: You should use templates; the [general] section never should been an implicit template, but it has been that way forever so we can't change it. The [general] section *should* have only been for settings that apply to the SIP channel driver as a whole, and *not* for providing defaults to entities configured for the driver. Unfortunately, it has both purposes. It may not be as intended, but from a user standpoint, it seems logical and convenient to establish policy in [general] and make exceptions in the entities as needed. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] permit/deny in sip.conf iax.conf
Steve Edwards wrote: It may not be as intended, but from a user standpoint, it seems logical and convenient to establish policy in [general] and make exceptions in the entities as needed. Right... for when you have one policy. When you have two policies, each that apply to a dozen or more entries in the config file, then it really doesn't help, it harms. Templates solve that problem completely, because each policy can be its own (named!) template, and they can be combined. Since templates are also very easy to use for the single policy case, they are a better solution to teach people (and they're also easier to implement in the configuration code of the module). In other modules created since chan_sip, we've intentionally avoided this problem, and you'll note that in nearly every other module, the [general] section is exactly that; general settings for the module, and not defaults. -- 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 -- 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] permit/deny in sip.conf iax.conf
I see. - Original Message - From: Steve Edwards asterisk@sedwards.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, March 23, 2010 4:06 PM Subject: Re: [asterisk-users] permit/deny in sip.conf iax.conf Karl Fife wrote: Naturally I can accomplish this using templates, but it DOES seem a bit odd since [general] is in essence ALREADY a 'template' of sorts for all parameters not otherwise specified. On Tue, 23 Mar 2010, Kevin P. Fleming wrote: You should use templates; the [general] section never should been an implicit template, but it has been that way forever so we can't change it. The [general] section *should* have only been for settings that apply to the SIP channel driver as a whole, and *not* for providing defaults to entities configured for the driver. Unfortunately, it has both purposes. It may not be as intended, but from a user standpoint, it seems logical and convenient to establish policy in [general] and make exceptions in the entities as needed. That makes sense. Thanks for the clarification. I'll call my global template [colonel] so there's no confusion :-) -K -- _ -- 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] permit/deny in sip.conf iax.conf
Steve Edwards wrote: It may not be as intended, but from a user standpoint, it seems logical and convenient to establish policy in [general] and make exceptions in the entities as needed. On Tue, 23 Mar 2010, Kevin P. Fleming wrote: Right... for when you have one policy. When you have two policies, each that apply to a dozen or more entries in the config file, then it really doesn't help, it harms. Templates solve that problem completely, because each policy can be its own (named!) template, and they can be combined. Since templates are also very easy to use for the single policy case, they are a better solution to teach people (and they're also easier to implement in the configuration code of the module). In other modules created since chan_sip, we've intentionally avoided this problem, and you'll note that in nearly every other module, the [general] section is exactly that; general settings for the module, and not defaults. OK. You win :) -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Safe_asterisk doesn't exists???
Hello my friends, I'm very worry about a problem i'm having...my asterisk got freez some times, every 5 or 6 days with NO trace in /var/log/asterisk/messages What i want to know is if safe_asterisk has something to be with this? This is what i have on my server: [r...@mypbx ~]# ps -A | grep asterisk 9118 ?00:01:30 asterisk [r...@dreampbx ~]# ps aux | grep asterisk root 9118 0.1 0.3 29668 12520 ? Sl Mar22 1:30 /usr/sbin/asterisk -f -vvvg -c root 12096 0.0 0.0 4140 640 pts/1S+ 18:40 0:00 grep asterisk I have another asterisk servers working and the commands above always shows safe _asterisk as a process... This safe_asterisk could be the cause of my problems? how does it works? how can i activate it? Thanks in advance for your valuable help! DD -- _ -- 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] Safe_asterisk doesn't exists???
On Tue, 23 Mar 2010, Danny Dias wrote: This safe_asterisk could be the cause of my problems? how does it works? how can i activate it? safe_asterisk is a script that usually lives at /usr/sbin/safe_asterisk. The script runs in the background. If it detects that Asterisk died, it can send you an email before restarting Asterisk. safe_asterisk is not the problem, but it can be useful as a band-aid until you find the real problem. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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 1.6.1.12 with Grandstream HT502 T38 Fax
Hi All, I'm in the lab with Asterisk 1.6.1.12 and several ATA's testing T38. I hit a snag with the Grandstream HT502. It only seems to nail up a session at 9600bps. The Grandstream GXW4104 nails up consistently at 14400bps. I'm using the same equipment in the same configuration, just switching out the ATA. I have the latest firmware on each unit. Any ideas on what could cause this? The configuration is pretty simple so I don't think I'm missing anything there. I'm guessing there is a built in speed limit on the HT502? Thanks. JR -- _ -- 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] AMD reporting NOTSURE most of the time
I am running Asterisk and using Answer machine detection with call files on a virtual Vcloud server running Centos 5.3 and LAMP. I am finding that AMD is only detecting HUMAN or MACHINE for about 30% of the calls (I sent over 50,000 outbound calls last week, and 70% said NOTSURE). I have a suspicion that the problem may be due to the timing source on virtual server when its under load delivering lots of asterisk calls, since the AMDSTATUS always reports things such as:- AMDSTATUS:NOTSURE-AMDCAUSE:TOOLONG-5500 My AMD.conf settings are all set to default:- [general] initial_silence = 2500 ; Maximum silence duration before the greeting. ; If exceeded then MACHINE. greeting = 1500 ; Maximum length of a greeting. If exceeded then MACHINE. after_greeting_silence = 800; Silence after detecting a greeting. ; If exceeded then HUMAN total_analysis_time = 5000 ; Maximum time allowed for the algorithm to decide ; on a HUMAN or MACHINE min_word_length = 100 ; Minimum duration of Voice to considered as a word between_words_silence = 50 ; Minimum duration of silence after a word to consider ; the audio what follows as a new word maximum_number_of_words = 5 ; Maximum number of words in the greeting. ; If exceeded then MACHINE silence_threshold = 256 Just wondering if any of you AMD users have any ideas as to what I should check. When I view this on the console I see that it jumps to too long almost immediately:- AMD: initialSilence [2500] greeting [1500] afterGreetingSilence [800] totalAnalysisTime [5000] minimumWordLength [100] betweenWordsSilence [50] maximumNumberOfWords [5] silenceThreshold [256] maximumWordLength [5000] -- AMD: Channel [SIP/faktortel-1385]. Changed state to STATE_IN_SILENCE -- AMD: Channel [SIP/faktortel-1385]. Too long... -- AMD: Channel [SIP/faktortel-1385]. Too long... Thanks Steve -- _ -- 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] Mobile phone shut down, but Queue() Ring as usual
hi, all i use Queue() to call a Mobile phone, there is only one mobile phone in the queue. even if the mobile phone shut down, Queue() is ring in the cli verbose as mobile phone is normally working. what i want to see is if the mobile phone is shut down. queue() will end immediately to tell on one in the queue. is there any method to do this ? -- Best regards, Sucan -- _ -- 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] pstn calls not picked up
I have a PSTN line coming into FXO port 4 on a TDM400P. Incoming calls are not being picked up. I don't find anything unusual in asterisk log. I am clueless where I should look. I also find zapata-additional.conf empty. The trouble started when the system was accidentally shut down and rebooted. Any help ? How do I diagnose if the TDM400P is not fried ? Thanks, -braman -- _ -- 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] Mobile phone shut down, but Queue() Ring as usual
Try the same as in http://lists.digium.com/pipermail/asterisk-users/2010-March/246316.html just make sure to add this in the [channels] context ;) Hope it helps. Alyed 2010/3/23 Zhang Shukun bit...@gmail.com hi, all i use Queue() to call a Mobile phone, there is only one mobile phone in the queue. even if the mobile phone shut down, Queue() is ring in the cli verbose as mobile phone is normally working. what i want to see is if the mobile phone is shut down. queue() will end immediately to tell on one in the queue. is there any method to do this ? -- Best regards, Sucan -- _ -- 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] pstn calls not picked up
Hello, Please Confirm if the dahdi/Zaptel service is running . check your channels status. On Wed, Mar 24, 2010 at 9:29 AM, Balu Raman brama...@gmail.com wrote: I have a PSTN line coming into FXO port 4 on a TDM400P. Incoming calls are not being picked up. I don't find anything unusual in asterisk log. I am clueless where I should look. I also find zapata-additional.conf empty. The trouble started when the system was accidentally shut down and rebooted. Any help ? How do I diagnose if the TDM400P is not fried ? Thanks, -braman -- _ -- 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 -- Best Regards Shakeel Abbas -- _ -- 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
