Re: [asterisk-users] Asterisk routine maintenance activities

Kurian Thayil wrote: > On Wed, 2009-04-22 at 15:24 +1000, Lee, John (Sydney) wrote: > >> Daily Asterisk restart >> > > Do you think its mandatory in production env? Daily? No. However, after implementing a weekly restart of Asterisk, I've found the instance of lockups and CPU utilisa

[asterisk-users] CDR feature not working properly for "failed call attempt"

Hi Asterisk Developers/users, I am facing a problem while using the cdr feature of asterisk(version asterisk1.4.24.1). Whenever I make a call using a “*.call” file and it gets failed , it don't produce the CDR for that channel as it falls into “OutgoingSpoolFailed” channel As there is no such chann

[asterisk-users] SendImage on SIP channel

Hi, I am using Asterisk 1.6.0.7 and I have a SIP channel configured. I connected a video phone. The sip.conf also has videosupport=yes in the global section. When I use the SendImage application in dialplan, the SENDIMAGERESULT value is set to "NOSUPPORT". The phone shows video is enabled and

[asterisk-users] mISDN DTMF endless tone

Hello all, I have set up a server with an OpenVox 1BRI card using mISDN driver and asterisk 1.4.22. All things works properly except when dtmf tones are sent from PSTN and to PSTN. The cases are different: - from PSTN, when the end user press a digit I got that right via mISDN (seen with ISDN ast

[asterisk-users] E1 not synchronized

Hello, We're using OpenVox D410 to connect Asterisk 1.4.21.2 box to E1 line from local telco operator. Once in a while we experienced lines always busy. I'm not sure but reported as sometime on making outgoing calls only, sometime both outgoing and incoming. Rebooting (not just restarting asteris

[asterisk-users] Should you use UserEvents for monitoring calls ?

Hi, I need to monitor call activity from a custom application software. The goal is to display things like who is on call or not, who has forwarded his call to his voicemail, etc ... When using manager's login command with Event parameter set to on, I'm getting tens of events I don't care about b

Re: [asterisk-users] CDR feature not working properly for "failed call attempt"

As far as I know, just reinstall 1.4.20, and your problem goes away. CS From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vikas Sent: Wednesday, April 22, 2009 4:12 AM To: asterisk-users@lists.digium.com; asterisk-...@lists.digium.co

Re: [asterisk-users] Faxing and TIFF files

D Tucny wrote: > 2009/4/22 Michael > > > I use GPL Ghostscript 8.6.2 to produce the TIFF files for faxing. > > Does anyone know of a way, either while producing the file, or > after, to tell > how many pages have been produced? (without manually v

Re: [asterisk-users] [asterisk-dev] How to get to 10.000 open calls

# moving to -users as this belongs there. It is a nice idea to run several Asterisk processes simultenously, it will defineately help with multithreading. However I would suggest trying less instances - that would perhaps give greater benefit, as Asterisk has it's own threading. For example 8 inst

[asterisk-users] module load chan_dahdi.so gives several WARNING-messages

I have 2 questions about the following output on the Asterisk CLI : asterisk*CLI> reload chan_dahdi.so The 'reload' command is deprecated and will be removed in a future release. Please use 'module reload' instead. [Apr 22 15:02:33] NOTICE[3634]: loader.c:580 ast_module_reload: The module 'chan_d

Re: [asterisk-users] Queue() Ignore Hangup Request

Solution: http://bugs.digium.com/view.php?id=12655&nbn=10 > -Original Message- > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > boun...@lists.digium.com] On Behalf Of Lee, John (Sydney) > Sent: Wednesday, 22 April 2009 3:56 PM > To: Asterisk Users Mailing List - No

Re: [asterisk-users] How to get to 10.000 open calls

Venefax schrieb (auf asterisk-dev): > It should work > exactly like in OpenSIP. Then why not actually use OpenSIPS? Philipp Kempgen -- AMOOCON 2009, May 4-5, Rostock / Germany -> http://www.amoocon.de Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de AMOOMA GmbH - Bach

Re: [asterisk-users] Queue() Ignore Hangup Request

"Lee, John (Sydney)" writes: > Solution: http://bugs.digium.com/view.php?id=12655&nbn=10 I'm probably being stupid here... "Closing this as setting the configuration option to no should produce the same effect as removing the code that produced working results." So you have to add a configurat

Re: [asterisk-users] How to get to 10.000 open calls

On 22 Apr 2009, at 14:31, Philipp Kempgen wrote: > Venefax schrieb (auf asterisk-dev): >> It should work >> exactly like in OpenSIP. > Then why not actually use OpenSIPS? Touche ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com --

Re: [asterisk-users] Looking for good IAX ATA

> > In places where SIP won't work for some reason, I register the phone to > > asterisk on my laptop which then converts the SIP channel to IAX. > > How did you do this? Were you using Wi-Fi to talk to the laptop (which was > using Wi-Fi to talk to the world?) Yes, that's how I do it. > A nat

[asterisk-users] Zaptel tone debug

Hi all! i have a question about a problem There exists some way of enabling debug for tones of zaptel? Of such way of being able to obtain the parameter busypattern for a specific provider. I have the idea that can be enabled from the source code of asterisk Thanks -- Ing Francisco Roq

Re: [asterisk-users] SendImage on SIP channel

This may be a "left-field" .02, but it seems that you would need the JPEG codec up on the channel to send an image. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: Wednesday, April 22, 2009

Re: [asterisk-users] SendImage on SIP channel

On Wednesday 22 April 2009 03:47:05 Sai P. Varanasi wrote: > Hi, > I am using Asterisk 1.6.0.7 and I have a SIP channel configured. I > connected a video phone. The sip.conf also has videosupport=yes in the > global section. > When I use the SendImage application in dialplan, the SENDIMAGERESULT

Re: [asterisk-users] Asterisk process ended

Hi All, So, there’s no solution without core dump to debug the crash ? Should I start my Asterisk process with –g option and waiting for another crash to get the core dump ? Regards, A.L De : Adrien Lemoine [mailto:alemo...@legos.fr] Envoyé : mardi 21 avril 2009 18:46 À : 'aste

Re: [asterisk-users] module load chan_dahdi.so gives several WARNING-messages

jonas kellens wrote: > > I have 2 questions about the following output on the Asterisk CLI : > Jonas Please do not hijack threads. Please start a new message. There is a good chance that many people didn't read your message because it's in reply to another, unrelated one. Barry

[asterisk-users] Conference problem

Hello all,   I have some issues with the MeetMe application.   The working topology is as follows. The Asterisk (1.4.22-rc5) is connected through SIP trunk to a Call Manager (6.1.2) which is connected to a Cisco Voice Gateway. The Gateway is connected to PSTN through a PRI. The calls are forward

Re: [asterisk-users] Peer 'iaxfax' is now UNREACHABLE! Time: 3

Lee Howard wrote: > Marco wrote: > >> I've IAXModem and asterisk Asterisk 1.4.24 running on the same machine. >> They are linked together through localhost. I've turned qualify on for the >> iax peer. Periodically I've this message: >> >> [Apr 20 23:47:46] NOTICE[4641]: chan_iax2.c:9049 __iax2_p

Re: [asterisk-users] Asterisk process ended

On Wed, 22 Apr 2009, Adrien Lemoine wrote: So, there?s no solution without core dump to debug the crash ? There is no solution without information. Without a core dump or error logs, its time to break out the "Magic 8 Ball." Asterisk (v1.2.7.1) runs on RedHat AS 4 without ?g option. As

Re: [asterisk-users] Asterisk process ended

Y, we are you running asterisk 1.2.x ? On Wed, Apr 22, 2009 at 11:39 AM, Steve Edwards wrote: > I have a customer stuck (coincidentally) on 1.2.7.1 because of some custom > hacks to meetme. They have to restart every 3 or 4 months due to a memory Steve, why can't you do your customer a favor and

Re: [asterisk-users] Asterisk process ended

Adrien Lemoine escribió: Hi All, So, there's no solution without core dump to debug the crash ? Should I start my Asterisk process with --g option and waiting for another crash to get the core dump ? You're right, because as you saw nothing weird in your logs you would be blind w

Re: [asterisk-users] Conference problem

run a "sip debug" and check whether it's asterisk disconnecting the calls (usually a SIP BYE message) or whether Asterisk is getting the disconnect from your Cisco GW Martin On Wed, Apr 22, 2009 at 10:56 AM, Cristi Iconaru wrote: > Hello all, > > I have some issues with the MeetMe application. >

Re: [asterisk-users] Asterisk process ended

On Wed, 22 Apr 2009, Martin wrote: > On Wed, Apr 22, 2009 at 11:39 AM, Steve Edwards > wrote: >> I have a customer stuck (coincidentally) on 1.2.7.1 because of some >> custom hacks to meetme. They have to restart every 3 or 4 months due to >> a memory > Steve, why can't you do your customer a

[asterisk-users] Upgrading from 1.4.21.2 to 1.6.0.5 breaks sql queries with backslashes?

Hi, all. I've been searching google, bug reports and forums and have looked in all the asterisk-users list archives back to 2003 but haven't seen an answer to this, so thought I'd post here. The problem seems to be that Asterisk 1.6.0.5 is sending backslashes (needed to escape commas and so fo

Re: [asterisk-users] Asterisk process ended

Thanks for your answer Steve. So if I understand correctly, the best solution is to update Asterisk rather than strive to find the bug? Regards, A.L Le Mer 22 avril 2009 19:05, Martin a écrit : > Y, we are you running asterisk 1.2.x ? > > > On Wed, Apr 22, 2009 at 11:39 AM, Steve Edwards > wr

Re: [asterisk-users] Asterisk process ended

The bug was probably found and corrected a couple years ago so why waste time trying to chase it down again? On Wed, 2009-04-22 at 20:59 +0200, alemo...@legos.fr wrote: > Thanks for your answer Steve. > > So if I understand correctly, the best solution is to update Asterisk > rather than

[asterisk-users] how to know the channel from the iax phone side?

hi is there any way to know the channel from the phone side? (an iax phone) Thanks David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http:

Re: [asterisk-users] Asterisk process ended

Hi Miguel, Thank you for your detailed reply. I will discuss these options with my colleagues and consider updating or production of logs in a future crash. Thank you all for the clarification. Regards, A.L On Wed, April 22, 2009 7:02 pm, Miguel Molina wrote: > Adrien Lemoine escribió: > >> >

Re: [asterisk-users] Upgrading from 1.4.21.2 to 1.6.0.5 breaks sql queries with backslashes?

On Wednesday 22 April 2009 13:34:29 Kristina Harris wrote: > Hi, all. I've been searching google, bug reports and forums and have > looked in all the asterisk-users list archives back to 2003 but haven't > seen an answer to this, so thought I'd post here. > > The problem seems to be that Asterisk 1

[asterisk-users] random hangups: how to debug?

I've a TDM400 with dahdi 2.1.0.4, asterisk 1.6.1-rc5. Asterisk is randomly hanging up calls coming over the pstn. Often it happens right as the call is answered: -- Starting simple switch on 'DAHDI/4-1' [Apr 22 17:09:38] NOTICE[20123]: chan_dahdi.c:7505 ss_thread: Got event 18 (Ring Begin)

Re: [asterisk-users] Looking for good IAX ATA

On Wed, 22 Apr 2009, Yahya Mohammad wrote: [snip] > > Btw, there are other options such as Fring, which I believe uses a > proprietary protocol from the cellphone to Fring servers, which is then > converted to SIP, Skype, Yahoo etc. > > I live in the middle east, and the state run ISPs block SI

Re: [asterisk-users] Upgrading from 1.4.21.2 to 1.6.0.5 breaks sql queries with backslashes?

On Wed, 22 Apr 2009, Tilghman Lesher wrote: > Correct. You no longer have to escape anything in the MYSQL command, at > all. This was done as a one-time flag-day event, for the upgrade from > 1.4 to 1.6. It makes future dialplans much easier to write, albeit with > a one-time conversion. Thanks

Re: [asterisk-users] Remote BLF / hint on IAX2 trunk

On 17/04/2009 5:44 p.m., Marco Sambo wrote: > So thanks, but in Asterisk 1.4.24 is not present in any way?? > Any mystique solution?? You could use func_devstate (which is available in the next version of 1.4 or currently available as a backport) and some shared database tricks. -- Kind Regards

Re: [asterisk-users] Simultaneous Calls at a time

On 17/04/2009 6:15 a.m., Gordon Henderson wrote: > On Thu, 16 Apr 2009, David @ULC wrote: > >> Even I thought so thats why I tried with 4 VOIP provider and things didn't >> change. :-( > > What phones? > > I've seen this with Snom 370 phones and ISDN (which I don't think is > related) I put it down

Re: [asterisk-users] Jabber and Presence

On 18/04/2009 2:28 a.m., Gavin Henry wrote: > Hi all, > > What other open source tools are people using for this? I was looking > at Openfire and their asterisk plugin. > > Is it easy to roll your own with res_jabber.so ?? I used openfire in the past, but have now changed over to using ejabberd.

[asterisk-users] Step-by-Step Asterisk and Cisco 1760 Help

I am up to post 5 on my step-by-step but I hit a bit of a snag and so far my searches have failed me, I hope someone can help. (By the way, I added an asterisk index for quick navigation on the blog http://qvlweb.blogspot.com/2009/04/asterisk-pbx-install-index.html.) Here is the snag and I am

[asterisk-users] voice quality

Hi all, I wonder who has the same voice quality problem as what we have. Below is our configuration. Company --- asterisk 1.4.22 (g729) --- CISCO --- T1 --- customer Sometimes, customers told me that they heard our voice not very clear, like a call from far far away. I heard the recording is ok

Re: [asterisk-users] asterisk 420 Bad Response

21 apr 2009 kl. 11.46 skrev Khaled W. Chehab: > Dears, > > When my GW send a call to asterisk v 1.4.24 , > Asterisk send Status: 420 bad extension (unsupported) > Why? Any modifications should be done one sip.conf > regards > > Your gateway is propably requiring a SIP extension Asterisk d

Re: [asterisk-users] voice quality

On Thu, 23 Apr 2009, Rilawich Ango wrote: > Hi all, > I wonder who has the same voice quality problem as what we have. > Below is our configuration. > Company --- asterisk 1.4.22 (g729) --- CISCO --- T1 --- customer > > Sometimes, customers told me that they heard our voice not very clear, > like