[asterisk-users] hints and server-side DND (do not disturb)
Hi, Currently I'm using hints to determine SIP presence. As I understand it, a SIP extension can be labeled as busy, ringing, etc, based on a channel status. So a channel MUST be present. If it isn't then the extension is considered to be available. If my statement is correct then is there a way to set the extesnion as busy even if there's no channel associated with this extension? eg. when an extension sets server-side DND (Do Not Disturb), it actually sets a boolean value in astdb. Whenever asterisk tries to route a call to this extension, it first checks this value. Obviously, there's no way I can use hints in this scenario, or is there? Is it possible to somehow create a dummy channel whenever an extension sets server-side DND (custom context) and delete it whenever it unsets it? Thanks, Vieri -- _ -- 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] hints and server-side DND (do not disturb)
יעע -Original Message- From: Vieri rentor...@yahoo.com Sender: asterisk-users-boun...@lists.digium.com Date: Tue, 17 Apr 2012 23:27:10 To: asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] hints and server-side DND (do not disturb) Hi, Currently I'm using hints to determine SIP presence. As I understand it, a SIP extension can be labeled as busy, ringing, etc, based on a channel status. So a channel MUST be present. If it isn't then the extension is considered to be available. If my statement is correct then is there a way to set the extesnion as busy even if there's no channel associated with this extension? eg. when an extension sets server-side DND (Do Not Disturb), it actually sets a boolean value in astdb. Whenever asterisk tries to route a call to this extension, it first checks this value. Obviously, there's no way I can use hints in this scenario, or is there? Is it possible to somehow create a dummy channel whenever an extension sets server-side DND (custom context) and delete it whenever it unsets it? Thanks, Vieri -- _ -- 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] g729 freezes 1.8
Hi folks, I was recently installing g729 support as usual (register, cp codec_g729 to modules) and found out that the resulting asterisk instances freezes whenever g729 commands are executed in the command line. The registration process went OK and I can see the module being loaded by asterisk: [Apr 18 09:54:29] NOTICE[23775] codec_g729a.c: G.729A transcoding module version 1.8.0_3.1.5, Copyright (C) 1999-2009 Digium, Inc. [Apr 18 09:54:29] NOTICE[23775] codec_g729a.c: This module is supplied under a commercial license granted by Digium, Inc. [Apr 18 09:54:29] NOTICE[23775] codec_g729a.c: Please see the full license text supplied by the accompanying [Apr 18 09:54:29] NOTICE[23775] codec_g729a.c: register utility, or ask for a copy from Digium. [Apr 18 09:54:29] NOTICE[23775] codec_g729a.c: This product includes software developed by the OpenSSL Project [Apr 18 09:54:29] NOTICE[23775] codec_g729a.c: for use in the OpenSSL Toolkit. (http://www.openssl.org/) [Apr 18 09:54:29] NOTICE[23775] codec_g729a.c: Copyright (C) 1998-2006 The OpenSSL Project [Apr 18 09:54:29] VERBOSE[23775] manager.c: == Manager registered action G729LicenseStatus [Apr 18 09:54:29] VERBOSE[23775] manager.c: == Manager registered action G729LicenseList [Apr 18 09:54:29] VERBOSE[23775] codec_g729a.c: == Host-ID: 7e:f8:73:46:bf:58:23:8f:82:10:6d:e5:59:6f:90:04:a8:30:74:fe [Apr 18 09:54:29] VERBOSE[23775] codec_g729a.c: == Found license 'G729-ZNSEND59XW4H' providing 25 channels [Apr 18 09:54:29] VERBOSE[23775] codec_g729a.c: == Found total of 25 G.729 licenses After I can, sometimes, execute g729 CLI commands: *CLI g729 show hostid Host-ID: 7e:f8:73:46:bf:58:23:8f:82:10:6d:e5:59:6f:90:04:a8:30:74:fe But no g729 seems to be loaded: *CLI core show translation Translation times between formats (in microseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 siren7 siren14 slin16 g719 speex16 testlaw g723 - - - -- - - - - - - - - - - - - - - gsm - - 2 2- 2 1 2 - - 4001 - 2 - - 4003 - - 2 ulaw - 2 - 2- 2 1 2 - - 4001 - 2 - - 4003 - - 2 alaw - 2 2 -- 2 1 2 - - 4001 - 2 - - 4003 - - 2 g726aal2 - - - -- - - - - - - - - - - - - - - adpcm - 2 2 2- - 1 2 - - 4001 - 2 - - 4003 - - 2 slin - 1 1 1- 1 - 1 - - 4000 - 1 - - 4002 - - 1 lpc10 - 2 2 2- 2 1 - - - 4001 - 2 - - 4003 - - 2 g729 - - - -- - - - - - - - - - - - - - - speex - - - -- - - - - - - - - - - - - - - ilbc - 2 2 2- 2 1 2 - - - - 2 - - 4003 - - 2 g726 - - - -- - - - - - - - - - - - - - - g722 - 2 2 2- 2 1 2 - - 4001 - - - - 4001 - - 2 siren7 - - - -- - - - - - - - - - - - - - - siren14 - - - -- - - - - - - - - - - - - - - slin16 - 3 3 3- 3 2 3 - - 4002 - 1 - - - - - 3 g719 - - - -- - - - - - - - - - - - - - - speex16 - - - -- - - - - - - - - - - - - - - testlaw - 2 2 2- 2 1 2 - - 4001 - 2 - - 4003 - - - Whenever I try to execute g729 show licencese, the CLI freezes and, in a few, asterisk itself freeezes: *CLI g729 show licenses *CLI core show translation *CLI core show translation I've tried both 1.8.8.1 and 1.8.11.0 asterisk versions, and generic and barcelona g729 binaries from http://downloads.digium.com/pub/telephony/codec_g729/asterisk-1.8.0/x86-64/ Whenever I delete the codec_g729a.so module everything runs smoothly.
Re: [asterisk-users] g729 freezes 1.8
On Wednesday 18 April 2012, samuel wrote: Hi folks, I was recently installing g729 support as usual (register, cp codec_g729 to modules) and found out that the resulting asterisk instances freezes whenever g729 commands are executed in the command line. . stuff deleted . Are you sure your g729 module, your Asterisk and your kernel are of the same bittedness? You cannot load 32-bit modules into an application which was compiled as 64- bit. This is not a problem if you built everything yourself from Source Code; but if anything was supplied pre-compiled and binary-only, you need to compile your Asterisk to match it. (And next time, insist on the Source Code; after all, you're paying money for it. Your right to know trumps other people's rights to keep secrets from you.) -- AJS Answers come *after* questions. -- _ -- 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] device state of a realtime queue member
On Tue, 2012-04-17 at 11:53 -0400, Matt Hamilton wrote: I'm trying to find if a realtime queue member is paused or not from the dialplan. For a paused, not in use phone, DEVICE_STATE returns not in use only. Is there a function that will tell if the phone is paused or not (other than querying the database directly)? Thanks, Matt Hi You could use queue show or queue show queue name asterisk console commands 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
Re: [asterisk-users] g729 freezes 1.8
I'm pretty sure it's not a problem of 32-64 bits: Asterisk 1.8.11.0 built by root on a x86_64 running Linux on 2012-04-18 07:45:43 UTC and I downladed the binaries from http://downloads.digium.com/pub/telephony/codec_g729/asterisk-1.8.0/x86-64/ And asterisk loads the module, as you can see in the log files I sent. So it doesn't look like a problem with 32-64 bits Thanks for the answer, Samuel. On 18 April 2012 10:33, A J Stiles asterisk_l...@earthshod.co.uk wrote: On Wednesday 18 April 2012, samuel wrote: Hi folks, I was recently installing g729 support as usual (register, cp codec_g729 to modules) and found out that the resulting asterisk instances freezes whenever g729 commands are executed in the command line. . stuff deleted . Are you sure your g729 module, your Asterisk and your kernel are of the same bittedness? You cannot load 32-bit modules into an application which was compiled as 64- bit. This is not a problem if you built everything yourself from Source Code; but if anything was supplied pre-compiled and binary-only, you need to compile your Asterisk to match it. (And next time, insist on the Source Code; after all, you're paying money for it. Your right to know trumps other people's rights to keep secrets from you.) -- AJS Answers come *after* questions. -- _ -- 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] g729 freezes 1.8
Just to confirm, I try to load a 32 bits g729 binary and asterisk doesn't load it and complain about it: [Apr 18 12:38:50] WARNING[2033] loader.c: Error loading module 'codec_g729a.so': /usr/lib/asterisk/modules/codec_g729a.so: wrong ELF class: ELFCLASS32 [Apr 18 12:38:50] WARNING[2033] loader.c: Module 'codec_g729a.so' could not be l oaded. On 18 April 2012 12:36, samuel sam...@gmail.com wrote: I'm pretty sure it's not a problem of 32-64 bits: Asterisk 1.8.11.0 built by root on a x86_64 running Linux on 2012-04-18 07:45:43 UTC and I downladed the binaries from http://downloads.digium.com/pub/telephony/codec_g729/asterisk-1.8.0/x86-64/ And asterisk loads the module, as you can see in the log files I sent. So it doesn't look like a problem with 32-64 bits Thanks for the answer, Samuel. On 18 April 2012 10:33, A J Stiles asterisk_l...@earthshod.co.uk wrote: On Wednesday 18 April 2012, samuel wrote: Hi folks, I was recently installing g729 support as usual (register, cp codec_g729 to modules) and found out that the resulting asterisk instances freezes whenever g729 commands are executed in the command line. . stuff deleted . Are you sure your g729 module, your Asterisk and your kernel are of the same bittedness? You cannot load 32-bit modules into an application which was compiled as 64- bit. This is not a problem if you built everything yourself from Source Code; but if anything was supplied pre-compiled and binary-only, you need to compile your Asterisk to match it. (And next time, insist on the Source Code; after all, you're paying money for it. Your right to know trumps other people's rights to keep secrets from you.) -- AJS Answers come *after* questions. -- _ -- 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] FXO - GSM Gateway Problem
Hi,
I have a problem where calling out of asterisk when the call is answered
dahdi hangs up immediately.
For example: Sip Client A calls external number. Route: SIP - FXO - GSM
Gateway -External Landline.
However when that external landline answers the call dahdi hangs up
immediately .
Going the other way is fine (External Landline - GSM Gateway - FXO -
SIP).
I've tried multiple different internet searches and can't seem to find any
information on this problem.
Below are my config files.
Sip.conf
[office-phone](!)
type=friend
context=sipofficephone
host=dynamic
nat=yes
#secret=
dtmfmode=auto
disallow=all
;allow=ulaw
allow=alaw
allow=GSM
[lewisphone](office-phone);lewis mobile
secret=
Chan_dahdi.conf
[channels]
signalling=fxs_ks
context=pstnincomming
group=0
channel = 1
Extensions.conf
[sipofficephone]
exten = _X.,1,Verbose(2,Call from VoIP network to ${EXTEN})
same = n,Dial(DAHDI/1/${EXTEN})
same = n,Hangup()
[pstnincomming]Diamon
exten = s,1,Answer()
same = n,Dial(SIP/lewisphone)
same = n,Hangup()
Asterisk CLI Output (Verbose 3)
My comments bold.
== Using SIP RTP CoS mark 5
-- Executing [@sipofficephone:1] Verbose(SIP/lewisphone-000a,
2,Call from VoIP network to ) in new stack
== Call from VoIP network to
-- Executing [@sipofficephone:2] Dial(SIP/lewisphone-000a,
DAHDI/1/) in new stack
-- Called DAHDI/1/
-- DAHDI/1-1 answered SIP/lewisphone-000a GSM Gateway Answering Call
then Sending it out.
-- Hanging up on 'DAHDI/1-1' Dest answering call to which DAHDI hangs up
-- Hungup 'DAHDI/1-1'
== Spawn extension (sipofficephone, , 2) exited non-zero on
'SIP/lewisphone-000a'
Best Regards
Lewis
digitalselect-e
www.Digital-Select.com http://www.digital-select.com/
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Re: [asterisk-users] Asterisk 1.8.11.0 Debian Squeeze packages with T.38 gateway, queue hints and fixed RFC4235 (notifycid=yes)
Hi, Il 18/04/2012 00:39, Kevin P. Fleming ha scritto: You guys know that it works in Asterisk 10, but you say you can't use Asterisk 10 for some reason that I don't understand. 1) No Debian packages for v10. If you have to maintain lots of servers, installing from sources is a big burden. Compile, install and forget isn't the way I work: if I have to apply a fix or close a security hole I can easily push the patches to my build server which will recompile all the branches I maintain, then every server will automatically upgrade with cron jobs. 2) A new whole of problems when upgrading production machines from a working 1.8.x to v10. That will mean parsing configs manually, find the problems and fixing them. 3) Third parties utilities/hardware/modules. I'm still waiting for a fix for my Sangoma BRI card which did broke when upgrading... You need a compatible version of third parties components to use recent versions of asterisk/dahdi/whatever and upgrading third parties components does always mean problems. 4) Isn't v10 supposed to be beta/non-production/non-long-term-support?[1] If we want to honor what Digium says we should use 1.8 for production servers when reliability is important. Backporting a single unstable feature is much better than the whole thing. 5) What was the purpose of the t38gateway-1.8 branch? Why did it existed at all if not to allow users to use t38 gw in production servers? I even read about the possibility to backport t38 gw to 1.8 as a plugin, but it seems it isn't a requested feature (which is strange because I know peoples who stopped using asterisk because of the lack of t38 gw). I really don't want to do polemics: I always used pstn for the faxes until now and I will keep using it. No problem. Cheers, Niccolò [1]https://wiki.asterisk.org/wiki/display/AST/Proposed+changes+to+Asterisk+release+and+support+cycles -- _ -- 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] g729 freezes 1.8
On Wednesday 18 April 2012, samuel wrote: On 18 April 2012 10:33, A J Stiles asterisk_l...@earthshod.co.uk wrote: Are you sure your g729 module, your Asterisk and your kernel are of the same bittedness? I'm pretty sure it's not a problem of 32-64 bits: Asterisk 1.8.11.0 built by root on a x86_64 running Linux on 2012-04-18 07:45:43 UTC and I downladed the binaries from http://downloads.digium.com/pub/telephony/codec_g729/asterisk-1.8.0/x86-64/ And asterisk loads the module, as you can see in the log files I sent. So it doesn't look like a problem with 32-64 bits Ah, well. It's always worth a shot, though. It could still be a missing library; run `ldd` on the .so file(s), and make sure all needed libraries are installed. -- AJS Answers come *after* questions. -- _ -- 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 1.8.10 getaddrinfo
your destination address is not correct, on CLI, check what is actually being passed in Dial application Regards, Zohair Raza On Wed, Apr 18, 2012 at 2:04 AM, motty.cruz motty.c...@gmail.com wrote: Hello All, I'm gettint this error, started recently when I upgraded to 1.8.10 from 1.8.4. [Apr 17 08:03:52] ERROR[9099]: netsock2.c:263 ast_sockaddr_resolve: getaddrinfo(external out, (null), ...): Name or service not known [Apr 17 08:03:52] WARNING[9099]: chan_sip.c:26503 sip_request_call: Unable to find IP address for host externalout. We will not use this remote IP address Does anybody have an idea how to fix error above? Thanks, motty -- _ -- 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] FXO - GSM Gateway Problem
Hi,
It can be codec negotiation error or else plese try to print hangupcause
sent from telco
On Wed, Apr 18, 2012 at 4:27 PM, Tech t...@digital-select.com wrote:
Hi,
** **
I have a problem where calling out of asterisk when the call is answered
dahdi hangs up immediately.
For example: Sip Client A calls external number. Route: SIP - FXO - GSM
Gateway -External Landline.
However when that external landline answers the call dahdi hangs up
immediately .
** **
Going the other way is fine (External Landline - GSM Gateway - FXO -
SIP).
** **
I've tried multiple different internet searches and can't seem to find any
information on this problem.
** **
Below are my config files.
** **
*Sip.conf*
[office-phone](!)
type=friend
context=sipofficephone
host=dynamic
nat=yes
#secret=
dtmfmode=auto
disallow=all
;allow=ulaw
allow=alaw
allow=GSM
** **
[lewisphone](office-phone);lewis mobile
secret=
** **
*Chan_dahdi.conf*
[channels]
signalling=fxs_ks
context=pstnincomming
group=0
channel = 1
** **
** **
*Extensions.conf*
[sipofficephone]
exten = _X.,1,Verbose(2,Call from VoIP network to ${EXTEN})
same = n,Dial(DAHDI/1/${EXTEN})
same = n,Hangup()
** **
[pstnincomming]Diamon
exten = s,1,Answer()
same = n,Dial(SIP/lewisphone)
same = n,Hangup()
** **
** **
*Asterisk CLI Output (Verbose 3)*
My comments bold.
** **
== Using SIP RTP CoS mark 5
-- Executing [@sipofficephone:1]
Verbose(SIP/lewisphone-000a, 2,Call from VoIP network to ) in
new stack
== Call from VoIP network to
-- Executing [@sipofficephone:2] Dial(SIP/lewisphone-000a,
DAHDI/1/) in new stack
-- Called DAHDI/1/
-- DAHDI/1-1 answered SIP/lewisphone-000a *GSM Gateway Answering
Call then Sending it out.*
-- Hanging up on 'DAHDI/1-1' *Dest answering call to which DAHDI
hangs up*
-- Hungup 'DAHDI/1-1'
== Spawn extension (sipofficephone, , 2) exited non-zero on
'SIP/lewisphone-000a'
** **
** **
** **
Best Regards
*
*
Lewis
[image: digitalselect-e]
www.Digital-Select.com http://www.digital-select.com/
*
*
** **
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Re: [asterisk-users] Asterisk 1.8.11.0 Debian Squeeze packages with T.38 gateway, queue hints and fixed RFC4235 (notifycid=yes)
On 04/18/2012 06:08 AM, Niccolò Belli wrote: Hi, Il 18/04/2012 00:39, Kevin P. Fleming ha scritto: You guys know that it works in Asterisk 10, but you say you can't use Asterisk 10 for some reason that I don't understand. 1) No Debian packages for v10. If you have to maintain lots of servers, installing from sources is a big burden. Compile, install and forget isn't the way I work: if I have to apply a fix or close a security hole I can easily push the patches to my build server which will recompile all the branches I maintain, then every server will automatically upgrade with cron jobs. This is a valid point, and we'll get this corrected. Our package repository should have packages for Asterisk 10, but it doesn't. 2) A new whole of problems when upgrading production machines from a working 1.8.x to v10. That will mean parsing configs manually, find the problems and fixing them. I haven't seen any rash of problems with config files when users upgrade from 1.8 to 10; in fact, we've changed development policies specifically in order to avoid breaking existing working configurations during upgrades, except when they are unavoidable. 3) Third parties utilities/hardware/modules. I'm still waiting for a fix for my Sangoma BRI card which did broke when upgrading... You need a compatible version of third parties components to use recent versions of asterisk/dahdi/whatever and upgrading third parties components does always mean problems. Do you expect Debian-style packages to include these third-party components in Asterisk? If you are talking about DAHDI specifically, moving to Asterisk 10 does not change DAHDI requirements at all. 4) Isn't v10 supposed to be beta/non-production/non-long-term-support?[1] If we want to honor what Digium says we should use 1.8 for production servers when reliability is important. Backporting a single unstable feature is much better than the whole thing. Asterisk 10 is not 'beta' or 'non-production', I have no idea where you are getting such an idea. Yes, it is a 'standard', not 'long term support' release, but it is still fully supported and intended for production use (it is not a 'developer' release). If you want Digium to be able to support your installation, especially for a long term, adding in a series of complex patches that significantly change behavior will not lead to a supportable system; if you report an issue against your patched version of Asterisk, the first response will be to replicate the problem without the patches in place, which defeats the purpose of using a 'supported' release. 5) What was the purpose of the t38gateway-1.8 branch? Why did it existed at all if not to allow users to use t38 gw in production servers? I even read about the possibility to backport t38 gw to 1.8 as a plugin, but it seems it isn't a requested feature (which is strange because I know peoples who stopped using asterisk because of the lack of t38 gw). You'd have to ask the community developer who created the branch what his intentions were with it; it's not an 'official' release of Asterisk, and at this point it isn't supported by anyone. The T.38 gateway code was significantly reworked to get it merged into trunk (which became Asterisk 10), because the 1.8 version had a lot of serious issues. That code is most definitely *not* ready for production, especially given how difficult T.38 interoperability is in general. T.38 gateway support isn't available as a 'plugin' for older releases because those releases don't have the necessary APIs and functionality needed to make it work. Adding those into an older release would risk destabilizing that release, and would dramatically increase the testing and support burden. I really don't want to do polemics: I always used pstn for the faxes until now and I will keep using it. No problem. If you feel that having a discussion about what makes sense for users to do and not to do is 'polemics', then fine, you can do whatever you like. Just please stop trying to assign blame or fault to people because this old, unsupported branch doesn't do what you want, especially when there is a current, fully supported release that will do what you want. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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] g729 freezes 1.8
On 04/18/2012 06:13 AM, A J Stiles wrote: On Wednesday 18 April 2012, samuel wrote: On 18 April 2012 10:33, A J Stilesasterisk_l...@earthshod.co.uk wrote: Are you sure your g729 module, your Asterisk and your kernel are of the same bittedness? I'm pretty sure it's not a problem of 32-64 bits: Asterisk 1.8.11.0 built by root on a x86_64 running Linux on 2012-04-18 07:45:43 UTC and I downladed the binaries from http://downloads.digium.com/pub/telephony/codec_g729/asterisk-1.8.0/x86-64/ And asterisk loads the module, as you can see in the log files I sent. So it doesn't look like a problem with 32-64 bits Ah, well. It's always worth a shot, though. It could still be a missing library; run `ldd` on the .so file(s), and make sure all needed libraries are installed. The simplest route to solving this problem is to contact Digium's support department; this is a Digium commercial product and you are entitled to technical support. The simple answer to your question is no, there are no known incompatibilities between Asterisk 1.8 and Digium's G.729 codec modules (if there were, we'd fix them). -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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] asterisk 1.4.39 and dahdi 2.6
Dear Warren; Yes, first thing I do is the make all and make install for dahdi, then I do ./configure and make and make install for asterisk. But I do not find the chan_dahdi under the /usr/lib/asterisk/modules. WHY? If I used asterisk 1.8, then I do not have any problem. What I am missing? Do I have to do dahdi_gencof before starting the compilation and installation of asterisk 1.4? Regards Bilal - Hi All; Is it normal if I used asterisk 1.4 and dahdi, then I will not find chan_dahdi under /usr/lib/asterisk/modules? And I will not be able to type dahdi commands (dahdi restart for example) in the asterisk CLI? Actually what I found only the following: app_dahdibarge.so app_dahdiras.so app_dahdiscan.so codec_dahdi.so So, it is available only with asterisk 1.8? Well, does this mean it is preferred to use zaptel with asterisk 1.4? Did you compile asterisk with DAHDI support? i.e Did you install DAHDI, then run ./configure on Asterisk Source and then install? Or did you install asterisk first, then DAHDI? I've successfully used DAHDI with Asterisk 1.4, so there must be some issue. Please give us information about how you installed everything. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com http://www.selbytech.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
[asterisk-users] Far end nat traversal not working
We use a obfuscation software to encrypt/mangle both SIP/RTP which sits before asterisk. What happens is sometimes we don't get any voice. after some rtp set debug we found out that when received ip of the rtp stream is router's public ip, everything works cleanly. But sometimes we get the private ip's of the client as received address in rtp stream which results in no voice. it seems asterisk because of some unknown reason failed to traverse nat for the media stream. How asterisk manages nat is not known to me. But common SIP nat traversal methods dictate that first it modifies SDP to put its address as the destination address for both side. Then it waits for a rtp packet (symmetric rtp) to know to what port it should send media. I'm not understanding at what stage it fails. Because of wrong IP is shown i'm suspecting that its because of not writing SDP correctly. For what reason it happens still unknown to us. Any pointer to how it should be debugged? What reason behind this strange behavior is still unknown to us. Thanks in advance. -- -aft -- _ -- 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] FXO - GSM Gateway Problem
Thanks Dhaval for taking the time to look at my question.
I have tried to print the hangup cause however as you can see below it
doesn't show that section of the dialplan.
I have ammended below the CLI and extensions.conf with the changes I made.
ASTERISK CLI
== Using SIP RTP CoS mark 5
-- Executing [01493857917@sipofficephone:1]
Verbose(SIP/lewisphone-000d, 2,Call from VoIP network to
01493857917) in new stack
== Call from VoIP network to 01493857917
-- Executing [01493857917@sipofficephone:2]
Dial(SIP/lewisphone-000d, DAHDI/1/01493857917) in new stack
-- Called DAHDI/1/01493857917
-- DAHDI/1-1 answered SIP/lewisphone-000d
-- Hanging up on 'DAHDI/1-1'
-- Hungup 'DAHDI/1-1'
== Spawn extension (sipofficephone, 01493857917, 2) exited non-zero on
'SIP/lewisphone-000d'
extensions.conf
[sipofficephone]
exten = _X.,1,Verbose(2,Call from VoIP network to ${EXTEN})
same = n,Dial(DAHDI/1/${EXTEN})
same = n,Verbose(2, Hangup Cause ${HANGUPCAUSE})
same = n,Hangup()
[pstnincomming]
exten = s,1,Answer()
same = n,Dial(SIP/lewisphone)
same = n,Hangup()
Best Regards
Lewis
digitalselect-e
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL
INDRODIYA
Sent: 18 April 2012 13:18
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FXO - GSM Gateway Problem
Hi,
It can be codec negotiation error or else plese try to print hangupcause
sent from telco
On Wed, Apr 18, 2012 at 4:27 PM, Tech t...@digital-select.com wrote:
Hi,
I have a problem where calling out of asterisk when the call is answered
dahdi hangs up immediately.
For example: Sip Client A calls external number. Route: SIP - FXO - GSM
Gateway -External Landline.
However when that external landline answers the call dahdi hangs up
immediately .
Going the other way is fine (External Landline - GSM Gateway - FXO -
SIP).
I've tried multiple different internet searches and can't seem to find any
information on this problem.
Below are my config files.
Sip.conf
[office-phone](!)
type=friend
context=sipofficephone
host=dynamic
nat=yes
#secret=
dtmfmode=auto
disallow=all
;allow=ulaw
allow=alaw
allow=GSM
[lewisphone](office-phone);lewis mobile
secret=
Chan_dahdi.conf
[channels]
signalling=fxs_ks
context=pstnincomming
group=0
channel = 1
Extensions.conf
[sipofficephone]
exten = _X.,1,Verbose(2,Call from VoIP network to ${EXTEN})
same = n,Dial(DAHDI/1/${EXTEN})
same = n,Hangup()
[pstnincomming]Diamon
exten = s,1,Answer()
same = n,Dial(SIP/lewisphone)
same = n,Hangup()
Asterisk CLI Output (Verbose 3)
My comments bold.
== Using SIP RTP CoS mark 5
-- Executing [@sipofficephone:1] Verbose(SIP/lewisphone-000a,
2,Call from VoIP network to ) in new stack
== Call from VoIP network to
-- Executing [@sipofficephone:2] Dial(SIP/lewisphone-000a,
DAHDI/1/) in new stack
-- Called DAHDI/1/
-- DAHDI/1-1 answered SIP/lewisphone-000a GSM Gateway Answering Call
then Sending it out.
-- Hanging up on 'DAHDI/1-1' Dest answering call to which DAHDI hangs up
-- Hungup 'DAHDI/1-1'
== Spawn extension (sipofficephone, , 2) exited non-zero on
'SIP/lewisphone-000a'
Best Regards
Lewis
digitalselect-e
www.Digital-Select.com http://www.digital-select.com/
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[asterisk-users] Realtime asterisk 10.3.0
Hi there,
I setup realtime asterisk 10.3.0 with backend mysql server. everything seems to
work fine except when I tried to enable the extensions for dialplan to be
obtained from mysql, I got an empty dialplan. I am not sure why this happening
(i don't know it's a bug or misconfiguration on my behalf)? here is some of
configuration may help to assist to solve this problem. thank you for your help
in advance.
extconfig.conf file:
[settings]
sipusers = odbc,aster,sip_buddies
sippeers = odbc,aster,sip_buddies
extensions = odbc,aster,extensions
extensions.conf file:
[internal]
exten = Realtime/internal@extensions
mysql table for extensions:
++--+---+--+--+--+
| id | context | exten | priority | app | appdata |
++--+---+--+--+--+
| 1 | internal | 1235 | 1 | Dial | SIP/${EXTEN} |
| 2 | internal | 1234 | 1 | Dial | SIP/${EXTEN} |
++--+---+--+--+--+
localhost*CLI dialplan show
[ Context 'app_queue_gosub_virtual_context' created by 'app_queue' ]
's' = 1. NoOp() [app_queue]
[ Context 'app_dial_gosub_virtual_context' created by 'app_dial' ]
's' = 1. NoOp() [app_dial]
[ Context 'parkedcalls' created by 'features' ]
'700' = 1. Park() [features]
[ Context 'internal' created by 'pbx_config' ]
-= 3 extensions (3 priorities) in 4 contexts. =---
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[asterisk-users] Delete Session timer ?
Hi can i don't sent into the SIP invite the Session Timer ? on asterisk 1.6 Best regards Olivier -- _ -- 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] Delete Session timer ?
On Wed, Apr 18, 2012 at 05:42:18PM +0200, Olivier CALVANO wrote: Hi can i don't sent into the SIP invite the Session Timer ? on asterisk 1.6 Have you tried 'session-timers=refuse' ? -- Barry -- _ -- 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] Delete Session timer ?
If I interpret the original question correctly, this will make his call drop. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Barry Miller Sent: Wednesday, April 18, 2012 10:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Delete Session timer ? On Wed, Apr 18, 2012 at 05:42:18PM +0200, Olivier CALVANO wrote: Hi can i don't sent into the SIP invite the Session Timer ? on asterisk 1.6 Have you tried 'session-timers=refuse' ? -- Barry -- _ -- 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] BUSY vs. CONGESTION
If you are dialing out on an in-use line, you should get the Congested
message. Of course you could put the CHANAVAIL command ahead of dial to
avoid this.
Question two - Goto is a straight jump, gotoif jumps on a condition.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph
Sent: Friday, April 13, 2012 7:34 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] BUSY vs. CONGESTION
I have two lines, fax voice.
I usually call out on fax line (to have voice line available)
I need to set the dial line based on dial-status. When I try to call out on
fax line and it is receiving a fax will I get a BUSY or CONGESTION signal?
What is the difference in dial plan condition: goto and gotoif
exten = 1,1,Dial(SIP/7780${EXTEN}@pstn-9998,60,tr)
exten = 1,2,Goto(1-${DIALSTATUS},1)
exten = 1-BUSY,1,Dial(SIP/9780${EXTEN}@pstn-,60,tr)
exten = 1-CONGESTION,1,Dial(SIP/9780${EXTEN}@pstn-,60,tr)
vs.
exten = 1,1,Dial(SIP/7780${EXTEN}@pstn-9998,60,tr)
exten = 1,n,GotoIf($[${DIALSTATUS}=BUSY]?line2)
exten = 1,n,GotoIf($[${DIALSTATUS}=CONGESTION]?line2)
exten = 1-n(line2),1,Dial(SIP/9780${EXTEN}@pstn-,60,tr)
--
Joseph
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Re: [asterisk-users] BUSY vs. CONGESTION
No, if you are dialing to a TN which is in use you get a BUSY, except on FXO
signaled ports which are always considered ANSSWERED when the PBX finishes
dialing.
If you are trying to dial out via a LINE which is in use, you would likely get
a CONTESTION.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Wednesday, April 18, 2012 12:12 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] BUSY vs. CONGESTION
If you are dialing out on an in-use line, you should get the Congested message.
Of course you could put the CHANAVAIL command ahead of dial to avoid this.
Question two - Goto is a straight jump, gotoif jumps on a condition.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph
Sent: Friday, April 13, 2012 7:34 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] BUSY vs. CONGESTION
I have two lines, fax voice.
I usually call out on fax line (to have voice line available)
I need to set the dial line based on dial-status. When I try to call out on
fax line and it is receiving a fax will I get a BUSY or CONGESTION signal?
What is the difference in dial plan condition: goto and gotoif
exten = 1,1,Dial(SIP/7780${EXTEN}@pstn-9998,60,tr)
exten = 1,2,Goto(1-${DIALSTATUS},1)
exten = 1-BUSY,1,Dial(SIP/9780${EXTEN}@pstn-,60,tr)
exten = 1-CONGESTION,1,Dial(SIP/9780${EXTEN}@pstn-,60,tr)
vs.
exten = 1,1,Dial(SIP/7780${EXTEN}@pstn-9998,60,tr)
exten = 1,n,GotoIf($[${DIALSTATUS}=BUSY]?line2)
exten = 1,n,GotoIf($[${DIALSTATUS}=CONGESTION]?line2)
exten = 1-n(line2),1,Dial(SIP/9780${EXTEN}@pstn-,60,tr)
--
Joseph
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Re: [asterisk-users] hints and server-side DND (do not disturb)
On Wed, Apr 18, 2012 at 1:27 AM, Vieri rentor...@yahoo.com wrote:
Hi,
Currently I'm using hints to determine SIP presence. As I understand it, a
SIP extension can be labeled as busy, ringing, etc, based on a channel
status. So a channel MUST be present. If it isn't then the extension is
considered to be available.
If my statement is correct then is there a way to set the extesnion as
busy even if there's no channel associated with this extension?
eg. when an extension sets server-side DND (Do Not Disturb), it actually
sets a boolean value in astdb. Whenever asterisk tries to route a call to
this extension, it first checks this value. Obviously, there's no way I can
use hints in this scenario, or is there? Is it possible to somehow create a
dummy channel whenever an extension sets server-side DND (custom
context) and delete it whenever it unsets it?
I've done something similar using night-mode type logic. All calls
coming into the system first do a check against the db to see if night-mode
is enabled or not. If it is, route calls to voicemail, if it's not, route
calls normally. You can also use custom hints to set busy lamps on
appropriate phones. The receptionist can then hit the monitored button on
her phone to turn on or turn off night-mode. Here's some snippets from
existing dialplan...
[mainmenu]
; Main IVR
exten = s,1,Verbose(Inbound call to main number - checking if night mode
or normal)
exten = s,n,Set(NIGHTMODE=${DB(nightmode/enable)})
exten = s,n,GotoIf($[${NIGHTMODE} = 1]?nightmode)
exten = s,n,Verbose(Normal mode - Proceeding Normally)
exten = s,n,...
exten = s,n,...
exten = s,n,...
exten = s,n(nightmode),Verbose(Night mode - going straight to voicemail)
exten = s,n,Voicemail(@default,su)
exten = s,n,Hangup()
[internal]
; Night Mode
exten = *280,1,Answer()
exten = *280,n,GotoIf($[${DB(nightmode/enable)} = 1]?disable:enable)
exten = *280,n(enable),Verbose(Enabling night mode)
exten = *280,n,Set(DB(nightmode/enable)=1)
exten = *280,n,Set(DEVICE_STATE(Custom:lamp)=BUSY)
exten = *280,n,Playback(enabled)
exten = *280,n,Hangup()
exten = *280,n(disable),Verbose(Disabling night mode)
exten = *280,n,Set(DB(nightmode/enable)=0)
exten = *280,n,Set(DEVICE_STATE(Custom:lamp)=NOT_INUSE)
exten = *280,n,Playback(disabled)
exten = *280,n,Hangup()
--
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com http://www.selbytech.com
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Re: [asterisk-users] Asterisk 1.8.11.0 Debian Squeeze packages with T.38 gateway, queue hints and fixed RFC4235 (notifycid=yes)
Kevin P. Fleming wrote: This is a valid point, and we'll get this corrected. Our package repository should have packages for Asterisk 10, but it doesn't. How likely is it that a Centos 6 repo might be setup at the same time? -- _ -- 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 1.8.11.0 Debian Squeeze packages with T.38 gateway, queue hints and fixed RFC4235 (notifycid=yes)
On 04/18/2012 11:23 AM, Dan Austin wrote: Kevin P. Fleming wrote: This is a valid point, and we'll get this corrected. Our package repository should have packages for Asterisk 10, but it doesn't. How likely is it that a Centos 6 repo might be setup at the same time? It's on our list, but since the RPMs are primarily designed to support AsteriskNOW, and AsteriskNOW is still built on CentOS 5, it's not a high priority. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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] device state of a realtime queue member
Thanks Ishfaq, I need something from within the dialplan though. From: i...@pack-net.co.uk To: asterisk-users@lists.digium.com Date: Wed, 18 Apr 2012 10:06:38 +0100 Subject: Re: [asterisk-users] device state of a realtime queue member On Tue, 2012-04-17 at 11:53 -0400, Matt Hamilton wrote: I'm trying to find if a realtime queue member is paused or not from the dialplan. For a paused, not in use phone, DEVICE_STATE returns not in use only. Is there a function that will tell if the phone is paused or not (other than querying the database directly)? Thanks, Matt Hi You could use queue show or queue show queue name asterisk console commands 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] Asterisk 1.8.11.0 Debian Squeeze packages with T.38 gateway, queue hints and fixed RFC4235 (notifycid=yes)
Il 18/04/2012 14:50, Kevin P. Fleming ha scritto: Do you expect Debian-style packages to include these third-party components in Asterisk? If you are talking about DAHDI specifically, moving to Asterisk 10 does not change DAHDI requirements at all. No, I just pointed out that upgrading to a new asterisk version (ie 1.6 - 1.8) can lead to regressions when using third parties components. For example two years ago there was a bug with sangoma cards and asterisk 1.8 and now there is another one with dahdi 2.6. If you feel that having a discussion about what makes sense for users to do and not to do is 'polemics', then fine, you can do whatever you like. Just please stop trying to assign blame or fault to people because this old, unsupported branch doesn't do what you want, especially when there is a current, fully supported release that will do what you want. I think you misunderstood: I just wanted to point out that *I do not blame anyone*, I was just speaking about the reasons because of I prefer to not upgrade to v10. About asterisk 10, it seems I misunderstood the new release cycle, my fault. Niccolò -- _ -- 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 1.8.11.0 Debian Squeeze packages with T.38 gateway, queue hints and fixed RFC4235 (notifycid=yes)
Il 18/04/2012 14:50, Kevin P. Fleming ha scritto: we'll get this corrected That's an awesome news indeed. Niccolò -- _ -- 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] Delete Session timer ?
On Wed, Apr 18, 2012 at 10:59:21AM -0500, Danny Nicholas wrote: If I interpret the original question correctly, this will make his call drop. I imagine this would only happen if the remote end _requires_ RFC 4028 timers, which I don't think is very common. I was thinking he would refuse session timers for just those peers that had problems with them, like one of my ITSPs. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Barry Miller Sent: Wednesday, April 18, 2012 10:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Delete Session timer ? On Wed, Apr 18, 2012 at 05:42:18PM +0200, Olivier CALVANO wrote: Hi can i don't sent into the SIP invite the Session Timer ? on asterisk 1.6 Have you tried 'session-timers=refuse' ? -- Barry -- _ -- 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] device state of a realtime queue member
You can use system() to do this from the dialplan From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Hamilton Sent: Wednesday, April 18, 2012 11:42 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] device state of a realtime queue member Thanks Ishfaq, I need something from within the dialplan though. From: i...@pack-net.co.uk To: asterisk-users@lists.digium.com Date: Wed, 18 Apr 2012 10:06:38 +0100 Subject: Re: [asterisk-users] device state of a realtime queue member On Tue, 2012-04-17 at 11:53 -0400, Matt Hamilton wrote: I'm trying to find if a realtime queue member is paused or not from the dialplan. For a paused, not in use phone, DEVICE_STATE returns not in use only. Is there a function that will tell if the phone is paused or not (other than querying the database directly)? Thanks, Matt Hi You could use queue show or queue show queue name asterisk console commands 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] FXO - GSM Gateway Problem
Hi
I have had issues with wiring for incoming calls causing what looks like a
hangup when answered but in those cases the call stays up and asterisk thinks
its a new call. Have seen it on Avaya too
If it is wiring can you test a different incoming line?
Cheers duncan
On 19/04/2012, at 1:54 AM, Tech t...@digital-select.com wrote:
Thanks Dhaval for taking the time to look at my question.
I have tried to print the hangup cause however as you can see below it
doesn't show that section of the dialplan.
I have ammended below the CLI and extensions.conf with the changes I made.
ASTERISK CLI
== Using SIP RTP CoS mark 5
-- Executing [01493857917@sipofficephone:1]
Verbose(SIP/lewisphone-000d, 2,Call from VoIP network to 01493857917)
in new stack
== Call from VoIP network to 01493857917
-- Executing [01493857917@sipofficephone:2]
Dial(SIP/lewisphone-000d, DAHDI/1/01493857917) in new stack
-- Called DAHDI/1/01493857917
-- DAHDI/1-1 answered SIP/lewisphone-000d
-- Hanging up on 'DAHDI/1-1'
-- Hungup 'DAHDI/1-1'
== Spawn extension (sipofficephone, 01493857917, 2) exited non-zero on
'SIP/lewisphone-000d'
extensions.conf
[sipofficephone]
exten = _X.,1,Verbose(2,Call from VoIP network to ${EXTEN})
same = n,Dial(DAHDI/1/${EXTEN})
same = n,Verbose(2, Hangup Cause ${HANGUPCAUSE})
same = n,Hangup()
[pstnincomming]
exten = s,1,Answer()
same = n,Dial(SIP/lewisphone)
same = n,Hangup()
Best Regards
Lewis
image001.gif
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL INDRODIYA
Sent: 18 April 2012 13:18
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FXO - GSM Gateway Problem
Hi,
It can be codec negotiation error or else plese try to print hangupcause sent
from telco
On Wed, Apr 18, 2012 at 4:27 PM, Tech t...@digital-select.com wrote:
Hi,
I have a problem where calling out of asterisk when the call is answered
dahdi hangs up immediately.
For example: Sip Client A calls external number. Route: SIP - FXO - GSM
Gateway -External Landline.
However when that external landline answers the call dahdi hangs up
immediately .
Going the other way is fine (External Landline - GSM Gateway - FXO - SIP).
I've tried multiple different internet searches and can't seem to find any
information on this problem.
Below are my config files.
Sip.conf
[office-phone](!)
type=friend
context=sipofficephone
host=dynamic
nat=yes
#secret=
dtmfmode=auto
disallow=all
;allow=ulaw
allow=alaw
allow=GSM
[lewisphone](office-phone);lewis mobile
secret=
Chan_dahdi.conf
[channels]
signalling=fxs_ks
context=pstnincomming
group=0
channel = 1
Extensions.conf
[sipofficephone]
exten = _X.,1,Verbose(2,Call from VoIP network to ${EXTEN})
same = n,Dial(DAHDI/1/${EXTEN})
same = n,Hangup()
[pstnincomming]Diamon
exten = s,1,Answer()
same = n,Dial(SIP/lewisphone)
same = n,Hangup()
Asterisk CLI Output (Verbose 3)
My comments bold.
== Using SIP RTP CoS mark 5
-- Executing [@sipofficephone:1] Verbose(SIP/lewisphone-000a,
2,Call from VoIP network to ) in new stack
== Call from VoIP network to
-- Executing [@sipofficephone:2] Dial(SIP/lewisphone-000a,
DAHDI/1/) in new stack
-- Called DAHDI/1/
-- DAHDI/1-1 answered SIP/lewisphone-000a GSM Gateway Answering Call
then Sending it out.
-- Hanging up on 'DAHDI/1-1' Dest answering call to which DAHDI hangs up
-- Hungup 'DAHDI/1-1'
== Spawn extension (sipofficephone, , 2) exited non-zero on
'SIP/lewisphone-000a'
Best Regards
Lewis
image001.gif
www.Digital-Select.com
--
Re: [asterisk-users] device state of a realtime queue member
You can use system() to do this from the dialplan I'll give that a try. Seems like there is no dialplan function for that yet. I guess querying the database via func_odbc is another option. Thanks. From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Wed, 18 Apr 2012 12:10:55 -0500 Subject: Re: [asterisk-users] device state of a realtime queue member You can use system() to do this from the dialplan From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Hamilton Sent: Wednesday, April 18, 2012 11:42 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] device state of a realtime queue member Thanks Ishfaq, I need something from within the dialplan though. From: i...@pack-net.co.uk To: asterisk-users@lists.digium.com Date: Wed, 18 Apr 2012 10:06:38 +0100 Subject: Re: [asterisk-users] device state of a realtime queue member On Tue, 2012-04-17 at 11:53 -0400, Matt Hamilton wrote: I'm trying to find if a realtime queue member is paused or not from the dialplan. For a paused, not in use phone, DEVICE_STATE returns not in use only. Is there a function that will tell if the phone is paused or not (other than querying the database directly)? Thanks, Matt Hi You could use queue show or queue show queue name asterisk console commands 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 -- _ -- 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] Delete Session timer ?
yes i have put this option, but asterisk sent in the Header that he support the Session Timers, the sip server of the operator sent a session timer too and asterisk ignor it. my objectifs is asterisk don't sent the session timer Le 18 avril 2012 17:56, Barry Miller asterisk-us...@notanet.net a écrit : On Wed, Apr 18, 2012 at 05:42:18PM +0200, Olivier CALVANO wrote: Hi can i don't sent into the SIP invite the Session Timer ? on asterisk 1.6 Have you tried 'session-timers=refuse' ? -- Barry -- _ -- 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] Delete Session timer ?
Which version of Asterisk are you using? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier CALVANO Sent: Wednesday, April 18, 2012 2:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Delete Session timer ? yes i have put this option, but asterisk sent in the Header that he support the Session Timers, the sip server of the operator sent a session timer too and asterisk ignor it. my objectifs is asterisk don't sent the session timer Le 18 avril 2012 17:56, Barry Miller asterisk-us...@notanet.net a écrit : On Wed, Apr 18, 2012 at 05:42:18PM +0200, Olivier CALVANO wrote: Hi can i don't sent into the SIP invite the Session Timer ? on asterisk 1.6 Have you tried 'session-timers=refuse' ? -- Barry -- _ -- 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] Custom Application recording problem
Hi Dale,
Thanks for the correction gosub() worked. There was a problem with pressing
option 3 so I removed extension 4.
Below is the final [sub-timo]
[sub-timo]
exten = s,1,Set(RecordingType=${ARG1})
exten = s,n,Set(TIMEOUT(digit)=2) ; Set Digit Timeout to 5
seconds
exten = s,n,Set(TIMEOUT(response)=2) ; Set Response Timeout to 10
seconds
exten = s,n,Answer
exten = s,n,NoOp(${CALLERID(num)})
exten = s,n,Set(number=${CALLERID(num)})
exten = s,n,NoOp(${number})
exten = s,n(recordmsg),Background(recmsg1) ;Please say yo message after
the beep and end with a hash
exten =
s,n,Record(/var/www/html/timo/crystalrecords/${RecordingType}/${number}.wav)
exten =
s,n(playmsg),Playback(/var/www/html/timo/crystalrecords/${RecordingType}/${n
umber})
exten = s,n(askuser),Background(ackrec) ;Press 1 to replay or 2 to
re-record, 3 to save
exten = s,11,WaitExten(5)
exten = 1,1,Goto(s,playmsg)
exten = 2,1,Goto(s,recordmsg) ; re-record message
exten = 3,1,AGI(${RecordingType}.php)
exten = s,1,Background(invalidentry)
exten = s,n,Goto(s,askuser)
exten = t,1,Playback(thankyoubye)
exten = t,n,Return
Inorder for the system to recognize invalid selections, I also changed
exten = i,1,Background(invalidentry)
exten = i,n,Goto(s,askuser)
To
exten = s,1,Background(invalidentry)
exten = s,n,Goto(s,askuser)
Thank you very much for the help.
Kind Regards
Billy
On 4/17/12 11:11 PM, Dale Noll dn...@wi.rr.com wrote:
Billy,
I really should have had my coffee before answering you previous
message. My head was in the wrong place (not saying where) and I sent
you down the wrong path.
Macro() is not the answer because of the WaitExten(). When WaitExten is
used in a Macro(), it does not match within the macro, it matches an
extension within the context where the macro was called. This is what
is causing your errors.
What you really should do is use gosub(), not macro().
Here is the recording routine
[sub-timo]
exten = s,1,Set(RecordingType=${ARG1})
exten = s,n,Set(TIMEOUT(digit)=2) ; Set Digit Timeout to 5
seconds
exten = s,n,Set(TIMEOUT(response)=2) ; Set Response Timeout to
10 seconds
exten = s,n,Answer
exten = s,n,NoOp(${CALLERID(num)})
exten = s,n,Set(number=${CALLERID(num)})
exten = s,n,NoOp(${number})
exten = s,n(recordmsg),Background(recmsg1) ;Please say yo message
after the beep and end with a hash
exten =
s,n,Record(/var/www/html/timo/crystalrecords/${RecordingType}/${number}.gsm)
exten =
s,n(playmsg),Playback(/var/www/html/timo/crystalrecords/${RecordingType}/${num
ber})
exten = s,n(askuser),Background(ackrec) ;Press 1 to replay or 2 to
re-record, 3 to save
exten = s,11,WaitExten(5)
exten = 1,1,Goto(s,playmsg)
exten = 2,1,Goto(s,recordmsg) ; re-record message
exten = 3,1,Goto(4,1)
exten = 4,1,AGI($RecordingType}.php)
exten = 4,n,Return()
exten = i,1,Background(invalidentry)
exten = i,n,Goto(s,askuser)
exten = t,1,Playback(thankyoubye)
exten = t,n,Return
I know big change there eh? Note: I did make some changes to extension
4, but that was fix syntax error, not because of the change from macro
to gosub.
The difference is really how you call it.
exten = 3552,1,Gosub(sub-timo,s,1(contentdb))
exten = 3552,n,Hangup()
Also note. I have not tested this code. I have something similar in
place, but not your specific code.
Oh. You should be able to remove the 'include = timo' from the
[from-internal-custom] context.
Dale
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Re: [asterisk-users] Custom Application recording problem
On Thu, 19 Apr 2012, Billy Kaye wrote:
Below is the final [sub-timo]
[sub-timo]
exten = s,1,Set(RecordingType=${ARG1})
[snip]
exten = s,1,Background(invalidentry)
Doesn't this cause an error to be logged when you reload the dialplan?
Does 'dialplan show s@sub-timo' show what you expect?
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] Delete Session timer ?
On Wed, Apr 18, 2012 at 08:52:02PM +0200, Olivier CALVANO wrote: Le 18 avril 2012 17:56, Barry Miller asterisk-us...@notanet.net a ?crit : On Wed, Apr 18, 2012 at 05:42:18PM +0200, Olivier CALVANO wrote: Hi can i don't sent into the SIP invite the Session Timer ? on asterisk 1.6 Have you tried 'session-timers=refuse' ? yes i have put this option, but asterisk sent in the Header that he support the Session Timers, the sip server of the operator sent a session timer too and asterisk ignor it. my objectifs is asterisk don't sent the session timer I'm not sure about 1.6, but on 1.8 and 10, my INVITEs normally say Supported: replaces, timer *unless* the peer definition has 'session-timers=refuse'. Then I see Supported: replaces and CLI 'sip show channel ' on an outgoing call to this peer shows SIP Options:(none) Session-Timer: Inactive Isn't this what you're looking for? -- Barry -- _ -- 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] FXO - GSM Gateway Problem
snip I have a problem where calling out of asterisk when the call is answered dahdi hangs up immediately. /snip I'd make sure both answeronpolarityswitch and hanguponpolarityswitch are either commented out or set to no. from chan_dahdi.conf; ; Use a polarity reversal to mark when a outgoing call is answered by the ; remote party. ; ;answeronpolarityswitch=yes ; ; In some countries, a polarity reversal is used to signal the disconnect of a ; phone line. If the hanguponpolarityswitch option is selected, the call will ; be considered hung up on a polarity reversal. ; ;hanguponpolarityswitch=yes ; Alec Davis -- _ -- 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 1.4.39 and dahdi 2.6
Dear Warren; Yes I am compiling and installing dahdi first and then I start by asterisk 1.4.39 but I do not find chan_dahdi under /usr/lib/asterisk/modules, but if I used asterisk 1.8, it is working fine. From the other side: I tried asterisk 1.4.44 and same thing (I am not able to see the chan_dahdi) !! By the way, I am using ubuntu. Which asterisk 1.4 version that you tried it with dahdi and you were able to find the chan_dahdi? Really I tried too many attempts and until now I am not able to find a solution ! What I am missing? Regards Bilal - Hi All; Is it normal if I used asterisk 1.4 and dahdi, then I will not find chan_dahdi under /usr/lib/asterisk/modules? And I will not be able to type dahdi commands (dahdi restart for example) in the asterisk CLI? Actually what I found only the following: app_dahdibarge.so app_dahdiras.so app_dahdiscan.so codec_dahdi.so So, it is available only with asterisk 1.8? Well, does this mean it is preferred to use zaptel with asterisk 1.4? Did you compile asterisk with DAHDI support? i.e Did you install DAHDI, then run ./configure on Asterisk Source and then install? Or did you install asterisk first, then DAHDI? I've successfully used DAHDI with Asterisk 1.4, so there must be some issue. Please give us information about how you installed everything. -- Thanks, --Warren Selby, dCAP -- _ -- 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] upgrading from asterisk 1.4 to 1.6
Hi I have installed asterisk 1.4 and asterisk-gui 2.0, the problem is that it cannot upload the .gsm which i record through voice menu prompt, it gives error uploading is supported in asterisk 1.6 or higher. Can anyone help me? thanks -- _ -- 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
