[asterisk-users] Application Layer Gateway for SIP protocol

Hello everybody, I want to know if Asterisk can provide any solution to perform NAT traversal for SIP protocol which means that it implements the functions of an ALG (Application Layer Gateway). Thanks. ___ -- Bandwidth and Colocation Provide

Re: [asterisk-users] canreinvite question

In the sip.conf [2001] ... Canreinvite=yes [2002] ... Canreinvite=no Cordialement, BERGANZ François http://www.acropolistelecom.net  Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. -Message d'origine- De : asterisk-users-boun...@lists.digium.com [mailto:asterisk-us

[asterisk-users] Conference with an AGI inside Queue for password change

Hi, I have a typical call center with queues and agents added via AddQueueMember. One of my requirement is to implement a forgot password function. If a caller does not remember the password, he calls up an unauthenticated line and the agent manually authenticates him. Then the caller should have

Re: [asterisk-users] Increase DTMF Tone Duration

Andres wrote: > We are running 1.4.22 and have been experiencing problems with certain > IVRs and DTMF Tone duration. We would like to be able to increase DTMF > Tone duration by 50 to 100ms over what the user is pressing on his > phone. We have a PRI test circuit and an analyer in between to

Re: [asterisk-users] Authorize & Microsoft SQL

There is some code somewhere on the Asterisk/Linux box getting the SQL data, be it a program, script or batch file. There is something initiating the T-SQL code... SELECT * FROM supportcases WHERE id = 123456789 This code comes from the client, not the server. The Asterisk box will hav

Re: [asterisk-users] Authorize & Microsoft SQL

On Thu, 18 Dec 2008, Gregory Malsack wrote: > Steve, my friends setup does not utilize perl/php code. His > communication is directly between asterisk and mysql, there is no middle > man. This is what I was hoping for with ms sql. But it doesn't sound > like that will be the case. If all you w

Re: [asterisk-users] Authorize & Microsoft SQL

Steve, my friends setup does not utilize perl/php code. His communication is directly between asterisk and mysql, there is no middle man. This is what I was hoping for with ms sql. But it doesn't sound like that will be the case. Thanks for everything! Greg -Original Message- From: aste

Re: [asterisk-users] Authorize & Microsoft SQL

This is exactly what you need. Get your friends perl/php script and the SQL code will be near identical, or at least you will have no problem changing it yourself even if you don't know SQL. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lis

Re: [asterisk-users] Authorize & Microsoft SQL

The solution you mentioned is really no different. The only difference are the following: 1. Different database drivers. Someone just emailed them, see the email after my response in this thread. 2. Slightly different queries, but will be very easy to recode. For Example mySQL pus

Re: [asterisk-users] Authorize & Microsoft SQL

This much I already know. This information is easily found through a simple google search. What I'm looking for is if anyone knows what a dialplan would look like that would perform an ODBC query to an ODBC database. I've seen minuet documentation on ODBCget, which is what I'm thinking will do t

Re: [asterisk-users] Authorize & Microsoft SQL

Thanks for the reply Steve. I think you may have given me the idea I need. Here’s what I was really going for though. A friend of mine did this same thing as what I am looking to do, however he does this with a mysql database. Here’s basically what he does, 1. Reads a string from the c

Re: [asterisk-users] Authorize & Microsoft SQL

All you need is odbc and freetds. Then it will integrate very smoothly. Fred Posner f...@teamforrest.com Direct: +1 (503) 914-0999 -Original Message- From: "Steve Wofford" Date: Thu, 18 Dec 2008 19:46:36 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asteri

Re: [asterisk-users] Authorize & Microsoft SQL

There is nothing that ties asterisk and MS SQL together. MS SQL is just a database and many Windows base PBX use MS SQL to store CDR amongst many other things. What you do w/ the SQL data is up to you and can do anything really. Can you provide some more information on how this data is go get t

[asterisk-users] Authorize & Microsoft SQL

Hello Everyone, I have an installation where the client has a Microsoft SQL database that holds all of their case information. They would like the asterisk system to require users to enter a valid case number when making an outgoing call. I’m seeing some documentation regarding people using

[asterisk-users] Increase DTMF Tone Duration

Hi, We are running 1.4.22 and have been experiencing problems with certain IVRs and DTMF Tone duration. We would like to be able to increase DTMF Tone duration by 50 to 100ms over what the user is pressing on his phone. We have a PRI test circuit and an analyer in between to measure tone dur

Re: [asterisk-users] Message 0841984

What's scary is if the account used was a legitimate user and was hijacked, and either spoofed or the computer has a trojan that gives them control. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing

[asterisk-users] dahdi-linux 2.1.0.3 and dahdi-tools 2.1.0.2 released

The Asterisk development team has released dahdi-linux 2.1.0.3 and dahdi-tools 2.1.0.2. These releases are available for immediate download from http://downloads.digium.com. dahdi-linux was updated in order to fix several issues when compiling on kernels prior to 2.6.18 as well as a condition i

[asterisk-users] RESOLVED - Re: Zaptel / TDM400P card stopped working (in Fedora 8 system)

Langdon Stevenson wrote: > Hi > > I have a Dell PE2300 with a Digium TDM400P line card in it (with one > module to handle an inbound phone line). This is running on a Fedora 8 > system with Asterisk 1.4.21.2-1.fc8 > > This system has been working nicely for about 12 months. After a recent >

Re: [asterisk-users] Asterisk 1.4.22 Queues problems (Fifo or not ?)

Atis Lezdins a écrit : > You could enable "core set verbose 3" and "core set debug 1", and then > post corresponding log when you see this happens. > Ok so, what you are saying is that this shouldn't happens ? (in normal conditions the queue should be a fifo) Anyway, i had the idea this was ca

Re: [asterisk-users] Message 0841984

Well a good hanging might bemore satisfying, anyway. Details... ;) On Thu, 18 Dec 2008, David Gibbons wrote: > Last I checked, Lynch mobs don't shoot people. > > > I wonder if there would be interest in organizing a bounty for a lynching > mob, that would track down these !...@#$# silly excuse

Re: [asterisk-users] Message 0841984

Last I checked, Lynch mobs don't shoot people. I wonder if there would be interest in organizing a bounty for a lynching mob, that would track down these !...@#$# silly excuses for human beings and shoot them. If we all chipped in a few dollars I bet we could hire someone. --Dave

Re: [asterisk-users] Message 0841984

On Thu, 18 Dec 2008, John Todd wrote: >>> Dear asterisk-us...@lists.digium.com! >>> Lovers package at discount price! >>> Discount price store: ID 406858 >>> http://tba.dojmoquj.cn?faz >>> Pfizer is a licensee of the TRUSTe Privacy Program. >>> ? 2001-2008 Pfizer Inc. All rights reserved. >>> >

Re: [asterisk-users] Message 0841984

On Dec 18, 2008, at 7:55 AM, Jeff LaCoursiere wrote: > > Surely only list members should be allowed to post unmoderated? > > On Thu, 18 Dec 2008, ad...@viagra.com wrote: > >> Dear asterisk-us...@lists.digium.com! >> Lovers package at discount price! >> Discount price store: ID 406858 >> http://tb

Re: [asterisk-users] Asterisk 1.4.22 Queues problems (Fifo or not ?)

On Thu, Dec 18, 2008 at 9:44 PM, Benoit wrote: > Atis Lezdins a écrit : >> On Thu, Dec 18, 2008 at 8:50 PM, Darrin Henshaw wrote: >> >>> I believe you are correct Atis. >>> >>> Philipp within your queue setup do you have any announcements? If so read >>> the posting on >>> queues.conf(http://ww

Re: [asterisk-users] Asterisk 1.4.22 Queues problems (Fifo or not ?)

Atis Lezdins a écrit : > On Thu, Dec 18, 2008 at 8:50 PM, Darrin Henshaw wrote: > >> I believe you are correct Atis. >> >> Philipp within your queue setup do you have any announcements? If so read >> the posting on >> queues.conf(http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf)

[asterisk-users] [Fwd: Asterisk client for ekiga.net NAT problem]

I am experiencing a "606 not Acceptable" error trying to set up an Asterisk server as an ekiga.net client. My server is behind a firewall with NAT routing. I have googled this problem and read about Asterisk feeding its local ip address to ekiga.net. That seems to be my problem. I tried putti

Re: [asterisk-users] Dial timeout with SIP - how to set timeout for INVITE ACK

On Dec 18, 2008, at 7:22 AM, Julien Chavanton wrote: > I have a concern with Dial command, I want to enable a secondary > route with a remote partner, if the first route fails then we use > the second one : > > > Solution1: it will try both (there will be 2 simultanious actives > calls ring

Re: [asterisk-users] Asterisk 1.4.22 Queues problems (Fifo or not ?)

On Thu, Dec 18, 2008 at 8:50 PM, Darrin Henshaw wrote: > I believe you are correct Atis. > > Philipp within your queue setup do you have any announcements? If so read the > posting on > queues.conf(http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf), > announcements will have an eff

Re: [asterisk-users] Asterisk 1.4.22 Queues problems (Fifo or not ?)

Philipp Kempgen a écrit : > Benoit schrieb: > > >> I'm having a question with asterisk queue system, is it a fifo or a lifo >> or random ? >> > > Depends on the strategy. > http://www.voip-info.org/wiki-Asterisk+call+queues > >Philipp Kempgen > ? The strategy is for call distribution

Re: [asterisk-users] Dial timeout with SIP - how to set timeout for INVITE ACK

It's a valid concern, but be prepared for people to tell you that this should be done with the qualify parameter to determine if a host is up and running. Not the most ideal way to handle it, I'll agree. But the SIP proxy functionality of Asterisk is limited (as it's not intended to be a SIP proxy)

Re: [asterisk-users] Asterisk 1.4.22 Queues problems (Fifo or not ?)

Atis Lezdins a écrit : > Calls are distributed in Priority+FIFO. Do you set ${QUEUE_PRIO} > before sending call to queue? Perhaps you're forgetting it in some > part of dialplan. > Thanks for the hint, could be usefull, however i'm not using it anywhere right now __

Re: [asterisk-users] Asterisk 1.4.22 Queues problems (Fifo or not ?)

I believe you are correct Atis. Philipp within your queue setup do you have any announcements? If so read the posting on queues.conf(http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf), announcements will have an effect on the order that calls are picked up. Cheers, Darrin Hensha

Re: [asterisk-users] Latest AstManProxy

2008/12/18 Freddi Hansen > > Hi, > > > > I unsuccessfully tried to download AstManProxy, clicking over download > > button in http://github.com/davetroy/astmanproxy/tree/master . > > It failed with "XML error". > > I guess I have to insert here not to get caught by top or bottom post > filters. >

[asterisk-users] canreinvite question

Is it possible to allow reinvites to/from specific devices? For example; exten 2001 and 2002 can reinvite to each other, but not 2003 and 2004 exten 2003 and 2004 can reinvite to each other, but not 2001 and 2002 Can that be done? Devices 2001 & 2002 are behind one firewall, and 2003 & 2004 ar

Re: [asterisk-users] Asterisk 1.4.22 Queues problems (Fifo or not ?)

On Thu, Dec 18, 2008 at 8:39 PM, Philipp Kempgen wrote: > Benoit schrieb: > >> I'm having a question with asterisk queue system, is it a fifo or a lifo >> or random ? > > Depends on the strategy. > http://www.voip-info.org/wiki-Asterisk+call+queues > Strategy affects which agent will be next to g

Re: [asterisk-users] Problems with ztdummy

2008/12/18 Stephen Brown Jr > I'm having trouble with ztdummy and I can't seem to figure it out. I > am running Zaptel 1.4.12.1 under Debian 4.0 with latest updates > applied and I have compiled Zaptel from source along with a new kernel > from Debian sources to include 1khz timer support. Do y

Re: [asterisk-users] Dial timeout with SIP - how to set timeout for INVITE ACK

"You want to know if the remote address/proxy is up and running before you bother trying to wait on it for very long. Is this right?" , yes this would be a good start ? - But the IP could be up and the SIP service down, we need a signaling timeout, I beleive a good way in term of responsabili

Re: [asterisk-users] Asterisk 1.4.22 Queues problems (Fifo or not ?)

Benoit schrieb: > I'm having a question with asterisk queue system, is it a fifo or a lifo > or random ? Depends on the strategy. http://www.voip-info.org/wiki-Asterisk+call+queues Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr.

Re: [asterisk-users] Problems with ztdummy

> I think I had a board once where I had to specify ACPI=no to the kernel I believe that has taken care of it, it's working now :) I took out all of the ACPI support in the kernel, I don't need it anyhow as this box will be online 24/7 (like any server I suppose) > Any chance you could check this

Re: [asterisk-users] Latest AstManProxy

> Hi, > > I unsuccessfully tried to download AstManProxy, clicking over download > button in http://github.com/davetroy/astmanproxy/tree/master . > It failed with "XML error". I guess I have to insert here not to get caught by top or bottom post filters. You might want to use the version at:

Re: [asterisk-users] Asterisk 1.4.22 Queues problems (Fifo or not ?)

On Thu, Dec 18, 2008 at 8:21 PM, Benoit wrote: > I'm having a question with asterisk queue system, is it a fifo or a lifo > or random ? > > Sometimes when we have people waiting in the queue and new agents are > connected to handle the load the first call that is handled is not the > one which > i

Re: [asterisk-users] ael vs conf

On Thu, 2008-12-18 at 15:36 +0200, Tzafrir Cohen wrote: > On Thu, Dec 18, 2008 at 10:48:03AM -0200, David fire wrote: > > hi > > what i should use? ael or conf??? > > lua ? > > > thanks > > David > > Silly (and untested) tip of the day: > > Instead of using include, use templates: > > If

[asterisk-users] Asterisk 1.4.22 Queues problems (Fifo or not ?)

Hi, I'm having a question with asterisk queue system, is it a fifo or a lifo or random ? Sometimes when we have people waiting in the queue and new agents are connected to handle the load the first call that is handled is not the one which is already waiting for 4min, but the new one which has j

Re: [asterisk-users] DAHDI install dont need download of echo cancel

Jerry Geis schrieb: >> Jerry Geis schrieb: >> >/ Is there a way to install DAHDI and NOT download the echo canceler files? >> />/ I dont have firewall access and its failing. >> />/ I dont need the files as there is no card installed. >> />/ >> />/ How do I get past this? >> / >> If I remember cor

Re: [asterisk-users] DAHDI install dont need download of echo cancel

Mr. James W. Laferriere wrote: > Hello Tzafir , > > On Thu, 18 Dec 2008, Tzafrir Cohen wrote: >> On Thu, Dec 18, 2008 at 12:48:59PM -0500, Matt Watson wrote: >>> after you have configured zaptel manually the first time, copy the >>> menuselect.makeopts file that is generated in the root dire

Re: [asterisk-users] Dial timeout with SIP - how to set timeout for INVITE ACK

> > > *From:* asterisk-users-boun...@lists.digium.com on behalf of Philipp > Kempgen > *Sent:* Thu 18/12/2008 4:17 PM > *To:* Asterisk Users > *Subject:* Re: [asterisk-users] Dial timeout with SIP - how to set > timeout for I

Re: [asterisk-users] DAHDI install dont need download of echo cancel

Hello Tzafir , On Thu, 18 Dec 2008, Tzafrir Cohen wrote: > On Thu, Dec 18, 2008 at 12:48:59PM -0500, Matt Watson wrote: >> after you have configured zaptel manually the first time, copy the >> menuselect.makeopts file that is generated in the root directory of the >> zaptel source to a fil

Re: [asterisk-users] Dial timeout with SIP - how to set timeout for INVITE ACK

>So why not use 30 and let Asterisk take care of the SIP details/ >timeouts? Asterisk will wait the until it receive "answer" or timeout I need to timeout a SIP call on SIP INVITE ACK, in ISDN for exmaple this is translated to PROCEEDING Meaning "I have received the call, now I will look what

Re: [asterisk-users] DAHDI install dont need download of echo cancel

On Thu, Dec 18, 2008 at 12:48:59PM -0500, Matt Watson wrote: > after you have configured zaptel manually the first time, copy the > menuselect.makeopts file that is generated in the root directory of the > zaptel source to a file /etc/zaptel.makeopts. > > presumably this is available for people th

Re: [asterisk-users] DAHDI install dont need download of echo cancel

after you have configured zaptel manually the first time, copy the menuselect.makeopts file that is generated in the root directory of the zaptel source to a file /etc/zaptel.makeopts. presumably this is available for people that have moved on to DAHDI as well, and I would guess it should be /etc/

Re: [asterisk-users] queue question

2008/12/18 David fire > you can use the h extencion, when a call is hangup this extencions is > executed > (doble posting in action) > > 2008/12/18 Giedrius Augys > >> Hello, >> >>Is it possible, that after the call was established between client and >> agent and one of the them hangups the

[asterisk-users] Ghost in the Channel-Banks

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I've been struggling with an ongoing problem the last month. Here is the layout of the wiring: T1 from ISP > DiTech Echo Cancel device > Voice Channel-Bank (same) T1 from ISP > (same) DiTech Echo Cancel device > asterisk1 server zap card > fax channe

Re: [asterisk-users] Problems with ztdummy

On Thu, Dec 18, 2008 at 10:49:03AM -0500, Stephen Brown Jr wrote: > I'm having trouble with ztdummy and I can't seem to figure it out. I > am running Zaptel 1.4.12.1 under Debian 4.0 with latest updates > applied and I have compiled Zaptel from source along with a new kernel > from Debian sources t

Re: [asterisk-users] DAHDI install dont need download of echo cancel

> > Jerry Geis schrieb: > >/ Is there a way to install DAHDI and NOT download the echo canceler files? > />/ I dont have firewall access and its failing. > />/ I dont need the files as there is no card installed. > />/ > />/ How do I get past this? > / > If I remember correctly you can un-check th

Re: [asterisk-users] killall -9 (was: Re: (no subject))

Thanks for asking, This is caused by kill -9 and after starting by both ways... I cannot start DNS resolving of trunks. Regards 2008/12/18 Philipp Kempgen > Leonja Cerebro schrieb: > > > I have problem after killall -9 asterisk > > and asterisk -f > > Can you narrow down the problem a bit? > Is

Re: [asterisk-users] DAHDI install dont need download of echo cancel

Jerry Geis schrieb: > Is there a way to install DAHDI and NOT download the echo canceler files? > I dont have firewall access and its failing. > I dont need the files as there is no card installed. > > How do I get past this? If I remember correctly you can un-check them in `make menuselect`.

Re: [asterisk-users] Dial timeout with SIP - how to set timeout for INVITE ACK

Julien Chavanton schrieb: > I have a concern with Dial command, I want to enable a secondary route with a > remote partner, if the first route fails then we use the second one : > Solution1: it will try both (there will be 2 simultanious actives calls > ringing) this is not clean when calling an

Re: [asterisk-users] Problems with ztdummy

Stephen Brown Jr wrote: > rtc: lost some interrupts at 1024Hz BIOS settings? I've seen some rtc issues with a Red Hat derivative where the motherboard was bad, others I've seen BIOS settings affect the real time clocks. I think I had a board once where I had to specify ACPI=no to the kernel,

[asterisk-users] killall -9 (was: Re: (no subject))

Leonja Cerebro schrieb: > I have problem after killall -9 asterisk > and asterisk -f Can you narrow down the problem a bit? Is that caused by the kill -9 (instead of a normal stop)? Or by asterisk -f (instead of /etc/init.d/asterisk start)? > Asterisk stops to send to DNS resolving of trunks

Re: [asterisk-users] Message 0841984

Surely only list members should be allowed to post unmoderated? On Thu, 18 Dec 2008, ad...@viagra.com wrote: > Dear asterisk-us...@lists.digium.com! > Lovers package at discount price! > Discount price store: ID 406858 > http://tba.dojmoquj.cn?faz > Pfizer is a licensee of the TRUSTe Privacy Pro

Re: [asterisk-users] Idle threads

Hi, I noticed something bad happening on our systems lately. We have lots of asterisk threads running, but most of them are completely idle - strace doesn't show anything happening there. The only thread doing work seems to do everything. I see it sending mysql queries, writing logs, sending both S

Re: [asterisk-users] Message 0841984

Dear asterisk-us...@lists.digium.com! Lovers package at discount price! Discount price store: ID 406858 http://tba.dojmoquj.cn?faz Pfizer is a licensee of the TRUSTe Privacy Program. © 2001-2008 Pfizer Inc. All rights reserved. ___ -- Bandwidth and Colo

[asterisk-users] Problems with ztdummy

I'm having trouble with ztdummy and I can't seem to figure it out. I am running Zaptel 1.4.12.1 under Debian 4.0 with latest updates applied and I have compiled Zaptel from source along with a new kernel from Debian sources to include 1khz timer support. The modules build fine, yet when I load the

[asterisk-users] DAHDI install dont need download of echo cancel

Is there a way to install DAHDI and NOT download the echo canceler files? I dont have firewall access and its failing. I dont need the files as there is no card installed. How do I get past this? Jerry ___ -- Bandwidth and Colocation Provided by http://

Re: [asterisk-users] zaptel-Error

Is there somebody who knows what to do? -Ursprüngliche Nachricht- Von: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von fidibus83 Gesendet: Donnerstag, 18. Dezember 2008 11:02 An: 'Asterisk Users Mailing List - Non-Commercial Discussio

[asterisk-users] Dial timeout with SIP - how to set timeout for INVITE ACK

I have a concern with Dial command, I want to enable a secondary route with a remote partner, if the first route fails then we use the second one : Solution1: it will try both (there will be 2 simultanious actives calls ringing) this is not clean when calling an endusers exten => _X.,1,Dia

[asterisk-users] speelcheeK? (was Re: top posting again [was: Re: CDR Design])

Drew Gibson wrote: > David fire wrote: > >> you are soamming my mail box whit this useless discution >> the solution is doble posting (top and bottom) >> > Is it the spelling that is difficult or the typing? > speel cheeK 2 heard. C? Kind of reminds me of that movie... Johnny Dangerous

[asterisk-users] stream a file on a channel using AMI

Hi using AMI, is it possile to stream a file on a specific channel? Thanks to all in advance. -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSU

[asterisk-users] Asterisk 1.4.23-rc3 Released

The Asterisk.org development team has created the third release candidate for Asterisk 1.4.23. 1.4.23-rc3 is available for immediate download from http://downloads.digium.com/. This release candidate contains multiple fixes since 1.4.23-rc2 including issues with transfers and DTMF. For a full list

Re: [asterisk-users] top posting again [was: Re: CDR Design]

David fire wrote: > you are soamming my mail box whit this useless discution > the solution is doble posting (top and bottom) > > 2008/12/17 Andrew Kohlsmith (lists) > > > On December 17, 2008 05:03:00 pm Eric "ManxPower" Wieling wrote: > > To me top posting is l

[asterisk-users] Call routing in voicemail

Dear All, When one configures a standalone asterisk voicemail and attach to a legacy PBX, and the PBX transfers a busy, no-response or switchedoff extension to Asterisk. What would be the source and destination IDs that would get to asyerisk voicemail? Thanks, ... ___

Re: [asterisk-users] ael vs conf

??? 2008/12/18 Tzafrir Cohen > On Thu, Dec 18, 2008 at 10:48:03AM -0200, David fire wrote: > > hi > > what i should use? ael or conf??? > > lua ? > > > thanks > > David > > Silly (and untested) tip of the day: > > Instead of using include, use templates: > > If you have: > > > [a] >

Re: [asterisk-users] user entry as variables

Use the AGI script to collect the digits instead of doing it in your dialplan. j On Wed, 17 Dec 2008, Michael wrote: > I want to take series of user entered (via phone keypad) options/numeric entry > fields and use these with an AGI script. I have looked through voip-info and > I can't find an

Re: [asterisk-users] How to tell when a issue actually gets in a released version

Philipp Kempgen wrote: > It (svnbot) says: >> U branches/1.4/main/dial.c >> >> >> r104841 | mmichelson | 2008-02-27 15:45:47 -0600 (Wed, 27 Feb 2008) | 17 >> lines > > Which means it has been commited to the 1.4 branch at r

[asterisk-users] Best FXS SIGNALING and ECHO CANCELLATION Method in Middle East countries (SYRIA)

Hello .. Any one Know what is the best FXS signaling and echo cancellation in middle east countries (SYRIA is my country). thanks in advances _ Send e-mail faster without improving your typing skills. http://windowslive.com/Explore

Re: [asterisk-users] ael vs conf

On Thu, Dec 18, 2008 at 10:48:03AM -0200, David fire wrote: > hi > what i should use? ael or conf??? lua ? > thanks > David Silly (and untested) tip of the day: Instead of using include, use templates: If you have: [a] exten => _12X [b] include => a exten => _1. This won't work as pl

Re: [asterisk-users] ael vs conf

David fire wrote: > hi > what i should use? ael or conf??? I think it's personal choice at this point, I recently switched everything to ael, it's easier to read and follow the flow and less typing of redundant stuff. I have found that ael finds more of my fat finger mistakes when it compiles i

Re: [asterisk-users] Latest AstManProxy [SOLVED]

2008/12/18 Philipp Kempgen > Olivier schrieb: > > > I unsuccessfully tried to download AstManProxy, clicking over download > > button in http://github.com/davetroy/astmanproxy/tree/master . > > It failed with "XML error". > > Try again. It works. You're right : now it works ! I can't explain wh

[asterisk-users] Qualify = UNKNOWN

Hi, I have about 200 SIP phones with qualify=yes in my Realtime sip db. Some phones show "UNKNOWN" as status when I do a "sip show peers" (as opposed to, say, "50ms"). They are also counted as Offline (at the bottom of the show sip peers result) Those phones work perfectly with no perceptible

[asterisk-users] ael vs conf

hi what i should use? ael or conf??? thanks David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-user

Re: [asterisk-users] queue question

you can use the h extencion, when a call is hangup this extencions is executed (doble posting in action) 2008/12/18 Giedrius Augys > Hello, > >Is it possible, that after the call was established between client and > agent and one of the them hangups the call , the cmd queue executes cmd > (g

[asterisk-users] queue question

Hello, Is it possible, that after the call was established between client and agent and one of the them hangups the call , the cmd queue executes cmd (gosub, macro) or something... Thanks for advance -- Pagarbiai / Best Regards, Giedrius Augys ___

[asterisk-users] (no subject)

Hello, I have problem after killall -9 asterisk and asterisk -f Asterisk stops to send to DNS resolving of trunks Regards -- We never did too much talking anyway So don't think twice, it's all right ___ -- Bandwidth and Colocation Provided by http://ww

Re: [asterisk-users] starting call recording using AMI or other stuff

Hello, I have problem after killall -9 asterisk and asterisk -f Asterisk stops to send to DNS resolving of trunks Regards -- We never did too much talking anyway So don't think twice, it's all right ___ -- Bandwidth and Colocation Provided by http://www

Re: [asterisk-users] Latest AstManProxy

Olivier schrieb: > I unsuccessfully tried to download AstManProxy, clicking over download > button in http://github.com/davetroy/astmanproxy/tree/master . > It failed with "XML error". Try again. It works. Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.

Re: [asterisk-users] Alcatel OXE + Asterisk as external IVR

2008/12/18 Artifex Maximus > On Wed, Dec 17, 2008 at 2:16 PM, Olivier wrote: > > 2008/12/17 Artifex Maximus > >> On Wed, Dec 17, 2008 at 11:52 AM, Olivier wrote: > >> > 2008/12/17 Artifex Maximus > >> > If you don't expect to get more than 15 (or 12) calls at a time, I > don't > >> > see > >>

[asterisk-users] Latest AstManProxy

Hi, I unsuccessfully tried to download AstManProxy, clicking over download button in http://github.com/davetroy/astmanproxy/tree/master . It failed with "XML error". How can you download AstManProxy ? Has the project moved to somewhere else ? Have its features been deprecated and replaced by some

Re: [asterisk-users] Alcatel OXE + Asterisk as external IVR

On Wed, Dec 17, 2008 at 2:16 PM, Olivier wrote: > 2008/12/17 Artifex Maximus >> On Wed, Dec 17, 2008 at 11:52 AM, Olivier wrote: >> > 2008/12/17 Artifex Maximus >> > If you don't expect to get more than 15 (or 12) calls at a time, I don't >> > see >> > any real downside to use option 2. >> Ofte

Re: [asterisk-users] Asterisk AGX addons compile issues

2008/12/18 Michael > Has anyone seen this before, and know what is happening? > > u...@host:~/asterisk/agx-ast-addons# ./build.sh > -- Configuring done > -- Generating done > -- Build files have been written to: /root/asterisk/agx-ast-addons > [ 11%] Building C object CMakeFiles/app_devstate.dir/

Re: [asterisk-users] Voice Packets latency

Dear Sir, Appending to what I said in my previous email, please find below an rtp packets received on my asterisk server and sent to the carrier...Please let me know if length packet is OK Sent RTP packet to Carrier_IP:17380 (type 18, seq 045348, ts 3706128072, len 20) Got RTP packet fr

Re: [asterisk-users] zaptel-Error

But when I do yum list zaptel: Zaptel.i386 1.4.2.1-1.fc7 installed -Ursprüngliche Nachricht- Von: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von fidibus83 Gesendet: Donnerstag, 18. Dezember 2008 10:59 An: 'Asterisk Users Mailing Lis

Re: [asterisk-users] zaptel-Error

Zaptel version is 1.4.11 -Ursprüngliche Nachricht- Von: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Tzafrir Cohen Gesendet: Donnerstag, 18. Dezember 2008 10:54 An: asterisk-users@lists.digium.com Betreff: Re: [asterisk-users] zapt

Re: [asterisk-users] zaptel-Error

On Thu, Dec 18, 2008 at 10:08:42AM +0100, fidibus83 wrote: > > I have a Linux-Server with a Digium Wildcard TE110P. I install and configure > zaptel. But I have an error when I execute ztcfg –vv: > > 31 channels configured. > ioctl(ZT_LOADZONE) failed: Invalid argument > Notice: Configuration fi

[asterisk-users] zaptel-Error

Hello, my English isn’t very good but I try to explain my problem. Hopefully you can understand me. :-) I have a Linux-Server with a Digium Wildcard TE110P. I install and configure zaptel. But I have an error when I execute ztcfg –vv: 31 channels configured. ioctl(ZT_LOADZONE) failed: Invalid

[asterisk-users] Voice Packets latency

Dear All, I would like to ask please if there is any buffer for RTP packets sent through asterisk and if there is a way to change the value in order to increase it Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asteri