Re: [asterisk-users] asterisk config file online editor
With some mods it surely did the trick
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mojo with Horan
Company, LLC
Sent: miércoles, 20 de febrero de 2008 01:49 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk config file online editor
No problem, hope it gets you where you need to be :)
Moj
Anton Krall wrote:
This is a good start, thx Moj
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mojo with
Horan Company, LLC
Sent: martes, 19 de febrero de 2008 01:35 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk config file online editor
Like 15 lines of php and html?
?php
$fn = /etc/asterisk/extensions.conf;
if ($_REQUEST['action'] == write $_REQUEST['contents'] != )
{
rename($fn, $fn...date(U));
$fp = fopen($fn, wt);
fwrite($fp, $_REQUEST['contents']);
fclose($fp);
}
?
form
h1?=$fn?/h1
textarea name=contents?php include $fn ?/textarea
input type=hidden name=action value=write
input type=submit value=Save File input type=reset value=Reset
/form
Security holes galore! clean it up a bit :) And check on permissions
issues, that your httpd can write to the file.
Moj
Anton Krall wrote:
Guys, Im looking for a good text file editor for asterisk config files
that can be embedded on a web page for online editing (on an
interface),
any recommendations?
Anton Krall
Direccion General
Intruder Consulting
A Division of IntruderEnterprises S.A. de C.V.
www.Intruder.com.mx
www.IntruderStore.com.mx
Tel. 3872-2200 ext. 201
Tel. 01-800-INTRUDER (01-800-468-7833)
Email: [EMAIL PROTECTED]
Como lo estoy haciendo? Contacte a mi Director:
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[asterisk-users] asterisk config file online editor
Guys, Im looking for a good text file editor for asterisk config files that can be embedded on a web page for online editing (on an interface), any recommendations? Anton Krall Direccion General Intruder Consulting A Division of IntruderEnterprises S.A. de C.V. www.Intruder.com.mx www.IntruderStore.com.mx Tel. 3872-2200 ext. 201 Tel. 01-800-INTRUDER (01-800-468-7833) Email: [EMAIL PROTECTED] Como lo estoy haciendo? Contacte a mi Director: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk config file online editor
This is a good start, thx Moj
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mojo with
Horan Company, LLC
Sent: martes, 19 de febrero de 2008 01:35 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk config file online editor
Like 15 lines of php and html?
?php
$fn = /etc/asterisk/extensions.conf;
if ($_REQUEST['action'] == write $_REQUEST['contents'] != )
{
rename($fn, $fn...date(U));
$fp = fopen($fn, wt);
fwrite($fp, $_REQUEST['contents']);
fclose($fp);
}
?
form
h1?=$fn?/h1
textarea name=contents?php include $fn ?/textarea
input type=hidden name=action value=write
input type=submit value=Save File input type=reset value=Reset
/form
Security holes galore! clean it up a bit :) And check on permissions
issues, that your httpd can write to the file.
Moj
Anton Krall wrote:
Guys, Im looking for a good text file editor for asterisk config files
that can be embedded on a web page for online editing (on an
interface),
any recommendations?
Anton Krall
Direccion General
Intruder Consulting
A Division of IntruderEnterprises S.A. de C.V.
www.Intruder.com.mx
www.IntruderStore.com.mx
Tel. 3872-2200 ext. 201
Tel. 01-800-INTRUDER (01-800-468-7833)
Email: [EMAIL PROTECTED]
Como lo estoy haciendo? Contacte a mi Director:
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Re: [asterisk-users] app_voicemail for spanish
Hi Bilal You can take a look inside app_voicemail.c to check if you language is already defined... if it isn't then you will need to do some copy and paste to enable yours.. at least that's what I saw but then again, after testing, seems Spanish is fully implemented on 1.2.x I was just missing some speech files. Hope this helps AK -Original Message- From: bilal ghayyad [mailto:[EMAIL PROTECTED] Sent: miércoles, 16 de enero de 2008 08:07 a.m. To: asterisk-users@lists.digium.com Cc: Anton Krall Subject: Re: app_voicemail for spanish Hi AK; I would like to ask a question: where is the problem if u record the prompted messages in ur voice and as u need? Does not work? Also, if that the situation: how can I determine the needed voicemail language? For example I need ARABIC language, so what should I do to have arabic prompts? Regards Bilal - Im looking at app_voicemail (remember, this is on 1.2.x) and there seems to be some syntax changes for Spanish but doesn't seem to have all that's required... Ill file a bug report on mantis. AK -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Joakimsen Sent: lunes, 14 de enero de 2008 05:58 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] app_voicemail for spanish The language support is supposed to be there I know I've played with it and there are at least SOME grammatical changes (don't recall which right now) But if further language support is needed you should file a bugreport. On Jan 14, 2008 5:04 PM, Anton Krall [EMAIL PROTECTED] wrote: Guys, anybody has a 1.2.x compatible app_voicemail patched for Spanish prompts that can handle for example, instead of saying trabajo mensjes would say mensajes de trabajo o mensajes trabajo (inverse)? Also can handle singular and plural (mensaje vs. mensajes)? Anton Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_voicemail for spanish
Will do AK -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Joakimsen Sent: lunes, 14 de enero de 2008 11:48 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] app_voicemail for spanish No features are being added for 1.2 so I'd check to see if 1.4 has the changes you need before filing a bugreport. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] app_voicemail for spanish
Guys, anybody has a 1.2.x compatible app_voicemail patched for Spanish prompts that can handle for example, instead of saying trabajo mensjes would say mensajes de trabajo o mensajes trabajo (inverse)? Also can handle singular and plural (mensaje vs. mensajes)? Anton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_voicemail for spanish
Im looking at app_voicemail (remember, this is on 1.2.x) and there seems to be some syntax changes for Spanish but doesn't seem to have all that's required... Ill file a bug report on mantis. AK -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Joakimsen Sent: lunes, 14 de enero de 2008 05:58 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] app_voicemail for spanish The language support is supposed to be there I know I've played with it and there are at least SOME grammatical changes (don't recall which right now) But if further language support is needed you should file a bugreport. On Jan 14, 2008 5:04 PM, Anton Krall [EMAIL PROTECTED] wrote: Guys, anybody has a 1.2.x compatible app_voicemail patched for Spanish prompts that can handle for example, instead of saying trabajo mensjes would say mensajes de trabajo o mensajes trabajo (inverse)? Also can handle singular and plural (mensaje vs. mensajes)? Anton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] bug in 1.2.24
Thank you for the example Isaac. I did as you mentioned and now it seems to
be working perfectly.
Saludos
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Isaac Xiao
Sent: jueves, 13 de septiembre de 2007 10:33 p.m.
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] bug in 1.2.24
Here is our dial plan. You need to avoid double recording as well when
you transfer the call to other extension.
exten = 7141,3,Set(CALLFILENAME=q${EXTEN}-${TIMESTAMP}-${UNIQUEID})
exten = 7141,4,Set(__FROM-EXT-QUEUES=ext-queues)
exten = 7141,5,MixMonitor(${CALLFILENAME}.gsm|b)
exten = 7141,6,Playback(custom/None)
exten = 7141,7,Queue(7141|t|||7200)
Here is the CLI log.
-- Executing Playback(Zap/9-1, monitoring) in new stack
-- Playing 'monitoring' (language 'md')
-- Executing Playback(Zap/9-1, press-1-to-msg) in new stack
-- Playing 'press-1-to-msg' (language 'md')
-- Executing Goto(Zap/9-1, ext-queues|7141|1) in new stack
-- Goto (ext-queues,7141,1)
-- Executing NoOp(Zap/9-1, do not answer call before entering
queue) in new stack
-- Executing SetCIDName(Zap/9-1, CN) in new stack
-- Executing Set(Zap/9-1,
CALLFILENAME=q7141-20070914-132445-1189740177.10324) in new stack
-- Executing Set(Zap/9-1, __FROM-EXT-QUEUES=ext-queues) in new
stack
-- Executing MixMonitor(Zap/9-1,
q7141-20070914-132445-1189740177.10324.gsm|b) in new stack
-- Executing Playback(Zap/9-1, custom/None) in new stack
-- Executing Queue(Zap/9-1, 7141|t|||7200) in new stack
So Yes. As long as Zap/9-1 channel (customer's channel) not hangs up, it
will be always recorded.
Isaac
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Re: [asterisk-users] bug in 1.2.24
Thank Isaac, Ill try it this way.. Im currently using this before entering
the queue so calls from the queue are recorded:
exten =
s,n,SetVar(MONITOR_FILENAME=/var/spool/asterisk/${TIMESTAMP}-${UNIQUEID}-${C
ALLERIDNUM}-Queue-Ventas)
exten = s,n,SetVar(TRANSFER_CONTEXT=internalphones)
So I could just run mixmonitor instead of those lines and thats it? Queue
call will be recorded and everything that happens afterwards if it is
transferred?
Saludos
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Isaac Xiao
Sent: martes, 11 de septiembre de 2007 06:24 p.m.
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] bug in 1.2.24
It is not a bug. attended Transfer is using Local channel, if you have a
look the debug log from CLI, you will see why it fails. To solve this
problem, enable recording before the calls go into the queue.
Exten = ,1,MixMonitor(...)
Exten = ,2,Goto(ext-queue, , 1)
This will ensure you to record the customer/caller's channel instead of
exten's channel. So no matter where you transfer the call and as long as
the caller not hangup the call, it will be always recorded.
By the way, 1.2.24 stable, we got problem with 1.2.21. 1.2.17 seems
stable.
Good luck,
Isaac Xiao
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[asterisk-users] bug in 1.2.24
GUys.. I dont know if this is a known bug or not but I just tested and replicated this one over and over again. It involves call transfer from calls that entered the pbx via a queue.. say a call comes in and its thrown in a queue, somebody answers the call but then wants to transfer the call to somebody else outside the queue, of course... the bug comes in here.. Im using mixmonitor to record calls and when this scenario happens, the recording of the call coming in is OK, the call when in the queue and taking to the agent is OK, but then, when the agent transfers the call using attended transfer, mixmonitor stops recording... this doesn't happen if the call is transfer using BLIND transfer, just when using ATTENDED. Anybody seen this? Any bug fix or patch for 1.2.24 for this? Thx guys ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_conference
Mostly I want to try something new, always testing new stuff.. Ive read some interesting stuff about app_conference. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Moises Silva Sent: domingo, 02 de septiembre de 2007 11:15 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] app_conference Well, if you have control over incoming codecs, yeah sure I recommend it. However, because of the iLBC problem I never solved ( choppy sound ), if you don't have control over codecs joining the conference, may be meet me is still better fo you. Why do you want to move away from meetme? On 9/1/07, Anton Krall [EMAIL PROTECTED] wrote: Hi Moises. So, would you recommend app_conference over meetme? Knowing what you know about it? Saludos -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Moises Silva Sent: jueves, 30 de agosto de 2007 09:06 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] app_conference Anton, I used app_conference last year, debugged some problems with voice frames of 240 samples and made some fixes to the code. This is the result: http://www.moythreads.com/app-conference-ast-1.2.12.1-nov-6-2006.tar.bz2 I reported the problem to iaxclient-devel mailing list, as noted here: http://osdir.com/ml/telephony.pbx.asterisk.iaxclient.devel/2006-11/msg00016. html But never got response, not sure if is still under development. Right now iLBC voice frames will not work, sound will be choppy, I have not had the time/skills to fix it. Moy On 8/30/07, Anton Krall [EMAIL PROTECTED] wrote: Is app_conference designed only for 1.4? I tried compiling against 1.2.24 and but get a no such file while looking for autoconf.h which is a file only used in 1.4... anybody running app_conference on 1.2? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Within C++, there is a much smaller and cleaner language struggling to get out. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Within C++, there is a much smaller and cleaner language struggling to get out. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_conference
Hi Moises. So, would you recommend app_conference over meetme? Knowing what you know about it? Saludos -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Moises Silva Sent: jueves, 30 de agosto de 2007 09:06 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] app_conference Anton, I used app_conference last year, debugged some problems with voice frames of 240 samples and made some fixes to the code. This is the result: http://www.moythreads.com/app-conference-ast-1.2.12.1-nov-6-2006.tar.bz2 I reported the problem to iaxclient-devel mailing list, as noted here: http://osdir.com/ml/telephony.pbx.asterisk.iaxclient.devel/2006-11/msg00016. html But never got response, not sure if is still under development. Right now iLBC voice frames will not work, sound will be choppy, I have not had the time/skills to fix it. Moy On 8/30/07, Anton Krall [EMAIL PROTECTED] wrote: Is app_conference designed only for 1.4? I tried compiling against 1.2.24 and but get a no such file while looking for autoconf.h which is a file only used in 1.4... anybody running app_conference on 1.2? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Within C++, there is a much smaller and cleaner language struggling to get out. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] app_conference
Is app_conference designed only for 1.4? I tried compiling against 1.2.24 and but get a no such file while looking for autoconf.h which is a file only used in 1.4... anybody running app_conference on 1.2? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] app_conference and asterisk 1.2.24
Is app_conference designed only for 1.4? I tried compiling against 1.2.24 and but get a no such file while looking for autoconf.h which is a file only used in 1.4... anybody running app_conference on 1.2? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: click to call
Steven Have you been able to custommized the interface for babar's iax solution? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Sent: Lunes, 04 de Junio de 2007 07:17 a.m. To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: click to call I am using the free http://www.babarnazmi.citril.com/forum/viewtopic.php?t=7sid=fd8047cffb13074 969d3418064f4eb31 It is working as you described. It appears to be working well. -- -- Steven http://www.glimasoutheast.org Anton Krall [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] The idea is to put some kind of embedded app on the website so customers with mics can just click an icon or image and connect to our sales people or customer support staff... So far for what I've seen, there is some misconception of the terms.. click to dial can mean if you see a number on a webpage, click on it and your softphone will dial it.. but can also mean click on the image and it will connect you to the sales people, for example. I'm looking for the latter. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mail-lists Sent: Jueves, 31 de Mayo de 2007 10:18 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] click to call Anton Krall wrote: I have been looking around for examples or code on making a click to call application for web sites... has anybody had any luck on this topic? Is there any open source code out ther that could do this? What we have done in the past is created url's like this : sip:4044565941. Xlite will register itself as the sip handler on your system. If you want a generic click to call (ability to call numbers on any given website) check out moziax ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] click to call
Hi Gordon So, mexuar solution was that java softphone that you talked about? Any other small softphone type solution around, something on the same lines of what you described, something that the user could download but could be preconfigured or passed parameters to so they user wont have to mess with settings. Regards AK -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gordon Henderson Sent: Sábado, 02 de Junio de 2007 03:09 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] click to call On Fri, 1 Jun 2007, Anton Krall wrote: So Guys, no go on this topic? I trialled a click-to-dial application recently. It generated a lot of controversy on the list (search the archives) because various people said it couldn't be done/wouldn't work, etc. Then there were whinges about the commercial nature of the application (it's licensed, not free, and details were being posted to the -users list) and so on. Personally, I didn't see why as the creators of the code were simply replying to questions asked by list members, however... (That's probably why you've not gotten many replies ;-) So the thing I trialled was a button on a web page which downlaoded a soft-phone program written in Java to your browser. The soft-phone uses the IAX protocol to connect to an asterisk server, then depending on the javascript that you write to encapsulate the button on the web page, you have the ability to specify username password (to authenticate back to the asterisk server) and number to dial - the number you dial could even be entered via more javascript on the webpage, and the asterisk server at the back-end can then do what it needs to do with the number - dial an extension in a closed system, or even initiate a dial-out to the PSTN, if the server as such a connection and the connection is authorised. The end-user pushing the button doesn't need to see any of this at all - it can all be embedded in the javascript behind the button. You can specify callerId too, or dial different numbers, so the person answering the call could use this information to know what web page you are on for example. You can even embed it into an email signature with a different number then you could tell if they are calling you in reply to an email, and so on. (And much as I hate big HTML based email signatures, if done correctly this could be quite effective - and it doesn't need to download the Java - about 120KB until you click on the button) (They have a demonstration client which works with the Tesco VoIP service - you enter your Tesco username/password, then get a phone application with buttons, etc. The Tesco VoIP system unusually uses IAX rather than SIP as their transport mechanism!) I tried the application on a WinXP box, Linux box and Mac, and as long as the sound system was setup to work with the headset microphone, it just worked - At last, Java doing what it was supposed to be doing, working correctly cross platform! Some of the whinges to the list were that a soft-phone couldn't possibly be written in Java as Java was too heavyweight - well, this is the latter part of the first decade of the new millennium and Java has come a long way since it was first released, and they couldn't be further from the truth - in use on my 2GHz Linux box, it was using about 2-3% CPU, and at 120KB to download, is no worse than your average mid-resolution camera image these days. If this is what you're after, then go to http://www.mexuar.com/products_connect.shtml They were happy to give me a time-limited trial of the software, which I used, and found worked really well. You will need to write some html and javascript to encapsulate it into your own web page, but that's not hard to do and examples are provided. Now all I need is some clients to sell it to ;-) Gordon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Jueves, 31 de Mayo de 2007 10:58 a.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] click to call The idea is to put some kind of embedded app on the website so customers with mics can just click an icon or image and connect to our sales people or customer support staff... So far for what I've seen, there is some misconception of the terms.. click to dial can mean if you see a number on a webpage, click on it and your softphone will dial it.. but can also mean click on the image and it will connect you to the sales people, for example. I'm looking for the latter. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mail-lists Sent: Jueves, 31 de Mayo de 2007 10:18 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] click to call Anton Krall wrote: I have been looking around for examples
RE: [asterisk-users] click to call
True, maybe I didnt make myself clear on that point, what i meant was, Im not looking for an app that would let people click a sip: URL type to make a call using their already installed softphone but rather allow any user that visits our website to click on something and either open a web softphone or download a small one thats preconfigured and allow them to call a predefined extension on our asterisk server. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Sábado, 02 de Junio de 2007 08:23 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] click to call So far for what I've seen, there is some misconception of the terms.. click to dial can mean if you see a number on a webpage, click on it and your softphone will dial it.. but can also mean click on the image and it will connect you to the sales people, for example. I think the misconception is on your part. No matter what, the client will have to run some sort of softphone application. Whether it is implemented in Java, an exe, ActiveX, or some other 3rd party app. There is no magic image that makes phone calls. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Friday, June 01, 2007 10:53 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] click to call So Guys, no go on this topic? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Jueves, 31 de Mayo de 2007 10:58 a.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] click to call The idea is to put some kind of embedded app on the website so customers with mics can just click an icon or image and connect to our sales people or customer support staff... So far for what I've seen, there is some misconception of the terms.. click to dial can mean if you see a number on a webpage, click on it and your softphone will dial it.. but can also mean click on the image and it will connect you to the sales people, for example. I'm looking for the latter. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mail-lists Sent: Jueves, 31 de Mayo de 2007 10:18 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] click to call Anton Krall wrote: I have been looking around for examples or code on making a click to call application for web sites... has anybody had any luck on this topic? Is there any open source code out ther that could do this? What we have done in the past is created url's like this : sip:4044565941. Xlite will register itself as the sip handler on your system. If you want a generic click to call (ability to call numbers on any given website) check out moziax - ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Anton Krall.vcf Description: Binary data ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] click to call
Thank you for the explanation Dean, you are right on the money and could be more precise. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Sábado, 02 de Junio de 2007 04:34 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] click to call Joseph, This issue is people get confused; Click-to-Call - the ability to enter a number on a web page (or 2 numbers in the case of apps like JaJah) and have a centralized server deliver a two legged outbound call resulting in a 2 (or more) party conference call. This is inbuilt in Asterisk and can be found/implemented very easily by referring to the voip-info site for Dynamically generated call files. Click-to-Talk is different. It is the ability to using a browser to visit a web site, by clicking on a hyperlinked image or initiating the call in some other way your browser downloads either a java applet in the case of Mexuar, JiaxClient, Barbizan and a few other java solutions or an Active-X client in the case of Estara. These applets are basically installing a softphone onto the browser that is configured to dial a particular extension eg throught to sales or technical support. Its frustrating people confusing the terms but hopefully over time people will understand the differences. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Joseph Bajin Sent: Saturday, 2 June 2007 12:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] click to call You shouldn't need a softphone to do Click to Call.. The idea is pretty simple, and maybe I am missing something since I am haven't worked with Asterisk enough, but basically you start off by making the call to the Initial Party, Park the Call, Call the Other Party and then Connect them together.. Seems pretty simple and easy enough to do. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] click to call
So Guys, no go on this topic? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Jueves, 31 de Mayo de 2007 10:58 a.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] click to call The idea is to put some kind of embedded app on the website so customers with mics can just click an icon or image and connect to our sales people or customer support staff... So far for what I've seen, there is some misconception of the terms.. click to dial can mean if you see a number on a webpage, click on it and your softphone will dial it.. but can also mean click on the image and it will connect you to the sales people, for example. I'm looking for the latter. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mail-lists Sent: Jueves, 31 de Mayo de 2007 10:18 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] click to call Anton Krall wrote: I have been looking around for examples or code on making a click to call application for web sites... has anybody had any luck on this topic? Is there any open source code out ther that could do this? What we have done in the past is created url's like this : sip:4044565941. Xlite will register itself as the sip handler on your system. If you want a generic click to call (ability to call numbers on any given website) check out moziax - Anton Krall.vcf Description: Binary data ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] click to call
I have been looking around for examples or code on making a click to call application for web sites... has anybody had any luck on this topic? Is there any open source code out ther that could do this? Regards AK ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] click to call
The idea is to put some kind of embedded app on the website so customers with mics can just click an icon or image and connect to our sales people or customer support staff... So far for what I've seen, there is some misconception of the terms.. click to dial can mean if you see a number on a webpage, click on it and your softphone will dial it.. but can also mean click on the image and it will connect you to the sales people, for example. I'm looking for the latter. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mail-lists Sent: Jueves, 31 de Mayo de 2007 10:18 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] click to call Anton Krall wrote: I have been looking around for examples or code on making a click to call application for web sites... has anybody had any luck on this topic? Is there any open source code out ther that could do this? What we have done in the past is created url's like this : sip:4044565941. Xlite will register itself as the sip handler on your system. If you want a generic click to call (ability to call numbers on any given website) check out moziax ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Anton Krall.vcf Description: Binary data ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk 1.4 and r2mfc or unicall
Hi Guys.. I want to see what the R2mfc community has been up to. Anybody moved to 1.4? what have you done regarding unicall? Any updates or are you stuck with 1.2.X too? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] windows mobile 5 softphone for square screen devices
Well Guys.. I just bought the X-PDA one and indeed it has 240x240 support, costs around 29 USD and very well worth it, works great on an iPAQ 6945 via wireless and using my BT headset all sound goes to the headset and not the speaker, which is great and solves the eternal problem of having to listen to your call thru the speaker (not the phone speaker but the hands free one) AK |-Original Message- |From: [EMAIL PROTECTED] [mailto:asterisk-users- |[EMAIL PROTECTED] On Behalf Of Timothy Parez |Sent: Wednesday, January 17, 2007 9:40 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [asterisk-users] windows mobile 5 softphone for square screen devices | |Hi, | |I've been looking for a good SIP application for Windows Mobile for ages. |I found speaQ, but it has the same problem as any other softphone for |Windows Mobile. | |You see, it uses the speaker to output the conversation instead of the |phone speaker, |you know the one that is used when you make a normal phone call with |your WM Mobile PDA/Smartphone. |At first I was asking myself if every SIP client developer out there is |down right stupid but |in the end I found out this is actually Microsoft blocking access to |that phone speaker. |The claim that allowing the developers to access it would allow for |invasion of privacy (like recording phone calls). |So unless someone can work around this, softphones for WM will remain |quite useless. | |Timothy. | |Anton Krall wrote: | Guys, anybody has seen or is using some kind of softphone on any square | screen device with WM5? Ive tried sjlabs one and xten for pocket pc and they | do work on Wm5 but they are designed for standard screens, anybody using | anything on square ones? | | | | ___ | --Bandwidth and Colocation provided by Easynews.com -- | | asterisk-users mailing list | To UNSUBSCRIBE or update options visit: |http://lists.digium.com/mailman/listinfo/asterisk-users | | | |___ |--Bandwidth and Colocation provided by Easynews.com -- | |asterisk-users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] windows mobile 5 softphone for square screen devices
Cant remember the url but I googled it. Xten also without luck.. the main problem is the 240x240 screen... |-Original Message- |From: [EMAIL PROTECTED] [mailto:asterisk-users- |[EMAIL PROTECTED] On Behalf Of mitcheloc |Sent: Wednesday, January 17, 2007 1:48 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [asterisk-users] windows mobile 5 softphone for square screen devices | |I've been trying the SJPhone with no luck. Where did you download the |Xten version from? | |On 1/16/07, Anton Krall [EMAIL PROTECTED] wrote: | Guys, anybody has seen or is using some kind of softphone on any square | screen device with WM5? Ive tried sjlabs one and xten for pocket pc and they | do work on Wm5 but they are designed for standard screens, anybody using | anything on square ones? | | | | ___ | --Bandwidth and Colocation provided by Easynews.com -- | | asterisk-users mailing list | To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | | |-- | |Mitchel Constantin |Snap - A desktop user interface for Asterisk |www.snapanumber.com |___ |--Bandwidth and Colocation provided by Easynews.com -- | |asterisk-users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] windows mobile 5 softphone for square screen devices
Guys, anybody has seen or is using some kind of softphone on any square screen device with WM5? Ive tried sjlabs one and xten for pocket pc and they do work on Wm5 but they are designed for standard screens, anybody using anything on square ones? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] no unicall on 1.4
Hahahahaha |-Original Message- |From: [EMAIL PROTECTED] [mailto:asterisk-users- |[EMAIL PROTECTED] On Behalf Of Matt Riddell (NZ) |Sent: Sunday, January 07, 2007 3:13 PM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [asterisk-users] no unicall on 1.4 | |-BEGIN PGP SIGNED MESSAGE- |Hash: SHA1 | |Eric ManxPower Wieling wrote: | Anton Krall wrote: | This is exactly one of the things that Steve and I discussed a bit ago... | when did asterisk turn from an open source project with very good | developers | into a business that only focuses in $$$? | | I imagine that happened around the time they sold their soul to the | venture capitalists. 8-) | | Oddly, I download and install Asterisk for free all the time. | |Oh, you must be using the warez version then. | |31337 then aren't you! | |:D | |- -- |Cheers, | |Matt Riddell |___ | |http://www.sineapps.com/news.php (Daily Asterisk News - html) |http://wap.sineapps.com (Daily Asterisk News for your cellphone) |http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) |-BEGIN PGP SIGNATURE- |Version: GnuPG v1.4.2 (MingW32) |Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org | |iD8DBQFFoWJbS6d5vy0jeVcRArCpAJ9nJUq1NHzN/X8DrCMe7yB8LtNXkwCcCRfj |2KojUWrXmmJ/x55GMwvYZoI= |=tUpw |-END PGP SIGNATURE- |___ |--Bandwidth and Colocation provided by Easynews.com -- | |asterisk-users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] no unicall on 1.4
I agree with your last argument, exactly how it should be, Digium's free asterisk version has no warranties, as it should be, if you want support, you have to buy ... that's the way it works and should work. My complain is regarding the issue that all related asterisk projects like unicall were too spread around and usually no updated info is available, you can find bits and pieces of info around about new releases (like in voip-info) but like what happened with unicall and 1.4, nobody knew Steve was droping support for it ... hell, even somebody just saw the news and asked about it... that's what sometimes worries me, the owner of the project has the social responsibility of updating the news and keeping the community in touch with whats new, etc. for example SugarCRM, you can find everything on their page, news, etc. in the case of Digium and Asterisk, information is scattered around and sometimes its hard to find news about certain stuff, like unicall and this case. Anyway, we are drifting from the initial point which was to hope and support further development of R2MFC on the asterisk community so I propose a bounty to get Steve (sheesh, that's sounded like a hit bounty :)) I meant, a bounty to convince Steve to help the community that still uses R2 what do you say? |-Original Message- |From: [EMAIL PROTECTED] [mailto:asterisk-users- |[EMAIL PROTECTED] On Behalf Of Moises Silva |Sent: Friday, January 05, 2007 10:58 PM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [asterisk-users] no unicall on 1.4 | |On 1/5/07, Anton Krall [EMAIL PROTECTED] wrote: | I think you are misunderstanding several points here Moises. |May be | | I do give Digium a break like you said, that's why you have options |I dont understand this. How is related that you give Digium a break, |with the fact |that I have the option of use sangoma, Digium or clone hardware? | | But from the programmers perspective, if some open source software gets too | controlled by a company, then the synergy of open source is kind of broken, |MySQL is open source, but it is mainly ( only ) developed by the |company itself, so please dont contribute to the widely missuse of the |open source concept. Open Source does not mean free, Open Source |does not mean anyone is welcomed to contribute code. | | all projects have rules and what I meant is that Digium right now is too | controlling and that's why some programmers have decided to leave asterisk |Once again, the rules are not the main problem ( important factor |though ), the main problem are the technical differences. | | and pursue other projects in which they fell more comfortable and can | provide code in an easier manner. |Good for them!, more options for everyone. | | What I really have never liked is cases like what happened to unicall, there | was no news or announcements, simply , in 1.4 it's not there anymore :) |Anton, honestly, this is the part that rings into my head. Digium |NEVER supported formally the development of Unicall, Unicall was NEVER |part of the formal Asterisk release or addons, so it was NEVER there!. |Unicall was only Steves project that happens to work with Asterisk. If |I write a channel driver, as long as I dont sign and fax a disclaimer |to Digium and is accepted, Digium does not have ANY responsibility |about my code, in this case Steve's code. Even AFTER this, please |execute this: | |asterisk -vvvr |show warranty | |and if you have the free edition as I suppose you will see a BIG | |NO WARRANTY | |If you want more support BUY ( if you havent done already ) Asterisk |bussiness/enterprise edition. | |Kind Regards and Good Look! | |Moises | |-- |Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; |___ |--Bandwidth and Colocation provided by Easynews.com -- | |asterisk-users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] no unicall on 1.4
Couldnt agree with you more Lee. I think its very difficult for a software company to be able to stay focused on developing the software while been profitable, thats why many companies turn to consulting services (Sun), other develop hardware (IBM and OS/2 :)) Digium has been doing a great job with asterisk but like you said, access to the code without hassle was a plus for developers, now, well, they are probably going to turn to some other open source project like openpbx and probably Digium will end up hiring their own programmers and in time, asterisk could stop been an open source project and become a commercial only software. One thing thats interesting is how Digium turned to hardware like the TDM cards for getting money but as of now, Sangoma for example offers a better product (to me at least due to my experience with Digium hardware, timing sources, HW compatibility, etc.) so let me ask the awkward question: what is Digium doing 100% right? HW? No, software, used to, but maybe not now, so? Dont get me wrong, I love asterisk and will stick with 1.2 until something comes along (openpbx goes stable, etc.) but come on guys at Digium, focus... open source software was about making something by the community for the community, not getting XXX million USD in VC while losing your best programmers, the community and the R2MFC market :) |-Original Message- |From: [EMAIL PROTECTED] [mailto:asterisk-users- |[EMAIL PROTECTED] On Behalf Of Lee Howard |Sent: Thursday, January 04, 2007 10:14 PM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [asterisk-users] no unicall on 1.4 | |Anton Krall wrote: | |This is exactly one of the things that Steve and I discussed a bit ago... |when did asterisk turn from an open source project with very good developers |into a business that only focuses in $$$? | | | |Well, I think that there can be no doubt that there still are some very |good developers working on Asterisk, but yes, I do understand what |you're saying, and I think that we're not the only ones that have |noticed it. In particular I've noticed how the disclaimer requirement is |a sore spot, and as well how impossibly difficult it is for Digium |competitors to get their patches applied to the code base: | |http://bugs.digium.com/view.php?id=7742 | |Thats why openpbx was born I guess | | | |In part, yes. I think that some of these things are like lead weights to |the Asterisk development process - I think that Steve Underwood |appreciated the unfettered CVS commit access to the OpenPBX repository. |That's a once-in-a-lifetime opportunity that Asterisk may have been able |to have, itself, possibly. I see Steve's participation in OpenPBX as a |big selling point (i.e. real T.38 gatewaying and actual spandsp |integration). However, there's a lot of momentum behind Asterisk, and |that's compensated somewhat for its lead weights up until now, and |OpenPBX can't seem to get a public release out. | |At Cluecon last year in Chicago anthm told the conference how it was his |belief that it would be better to start from scratch than to fix up all |of the problems with Asterisk like OpenPBX is attempting - and thus we |have FreeSWITCH. | |So there are lots of possibilities out there, and I can only think that |the lead weights in the Asterisk development process will eventually |lead to more issues than with chan_unicall. | |For example, samba is still free, and people are making a profit from it by |giving out consulting services for deploying samba.. that is a good working |scenario asterisk used to be the same can you spell greedy :)? | | | |Well, when you sell consulting services for deploying Samba your |business focus is still on the software. If they were selling |Samba-related hardware or were heavily involved in selling Samba-related |things like books and tee-shirts, etc., instead of actually working the |software itself... well, then I think you'd see the same kinds of |problems that you're frustrated with now. It's all too easy for that |business activity to become a conflict of interest when it's not |directly related to the user-experienced software itself. | |Lee. | |___ |--Bandwidth and Colocation provided by Easynews.com -- | |asterisk-users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users BEGIN:VCARD VERSION:2.1 X-MS-SIGNATURE:YES N;LANGUAGE=en-us:Krall;Anton FN:Anton Krall ORG:Intruder Consulting TITLE:A Division of IntruderEnterprises S.A. de C.V. TEL;WORK;VOICE:+52 (55) 5781-5112 x 201 TEL;WORK;VOICE:+52 (55) 5985-2430 x 201 X-MS-OL-DEFAULT-POSTAL-ADDRESS:0 URL;WORK:http://www.intruder.com.mx EMAIL;PREF;INTERNET:[EMAIL PROTECTED] PHOTO;TYPE=JPEG;ENCODING=BASE64: /9j/4AAQSkZJRgABAQEAYABgAAD/2wBDAAYEBQYFBAYGBQYHBwYIChAKCgkJChQODwwQFxQY GBcUFhYaHSUfGhsjHBYWICwgIyYnKSopGR8tMC0oMCUoKSj/2wBDAQcHBwoIChMKChMoGhYa
RE: [asterisk-users] no unicall on 1.4 (was: OnHook Call Announcement...)
Hi Josue, as of today at least, Steve Underwood is focusing his efforts into making unicall be the basis for openpbx so will not be devoting more time into unicall and asterisk. This could change maybe but thats what he told me a few days ago. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Josué Conti Sent: Friday, January 05, 2007 3:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] no unicall on 1.4 (was: OnHook Call Announcement...) Hi All,as good? Steve Underwood will not work more with channel Unicall for the Asterisk? It will be discontinued? Best Regards Josué 2007/1/4, Moises Silva [EMAIL PROTECTED]: 1.2, Zap and Unicall work fine 1.4 Only Zap working, Unicall is broken On 1/4/07, Erick Perez [EMAIL PROTECTED] wrote: Question: So for people using E1 with R2 or PRI as signaling, what are my options in asterisk 1.4 and 1.2? On 1/4/07, Anton Krall [EMAIL PROTECTED] wrote: Well Moises, if you do, please drop me a line and I will gladly test it. I was mentioning digium because AFAIK, the guys at digium are in touch with the programmers and contributors so I thought maybe they would have an insight on whats going to happen with unicall on 1.4, I mean, somebody at the source should know right? Many people still use unicall so I thought somebody would pick up the ball, maybe that's going to be you hopefuly. Let me know how it goes. |-Original Message- |From: [EMAIL PROTECTED] [mailto:asterisk-users- |bounces@ lists.digium.com] On Behalf Of Moises Silva |Sent: Wednesday, January 03, 2007 5:22 PM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [asterisk-users] no unicall on 1.4 (was: OnHook Call Announcement...) | |On 1/3/07, Anton Krall [EMAIL PROTECTED] wrote: | And probably wont be as Steve Underwood explained to me that he is now supporting |openpbx and has stopped support for unicall on asterisk 1.4 | | Can anybody at digium confirm? Is unicall going to be left out of 1.4? | |This has nothing to do with Digium, it has to do with anybody wanting |to code the version for 1.4, AFAIK Steve never worked for Digium and |Digium never distributed Unicall driver. | |Porting Unicall to 1.4 is in my TODO since 1 month ago, may be this |month I will have the time to give a look at the code and try to make |it work on 1.4, if somebody else cant do it before. | |Regards. | |-- |Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; |___ |--Bandwidth and Colocation provided by Easynews.com -- | |asterisk-users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] no unicall on 1.4
I think you are misunderstanding several points here Moises. I do give Digium a break like you said, thats why you have options, you can use digium cards or sangoma cards, it's up to you, I use digium cards from time to time because I like to support digium in what they are doing. But from the programmers perspective, if some open source software gets too controlled by a company, then the synergy of open source is kind of broken, all projects have rules and what I meant is that Digium right now is too controlling and thats why some programmers have decided to leave asterisk and pursue other projects in which they fell more comfortable and can provide code in an easier manner. What I really have never liked is cases like what happened to unicall, there was no news or announcements, simply , in 1.4 it's not there anymore :) and you had to find out for yourself so, in one hand we have a lot of code control and on the other a lack of communication from programmers, to digium (which controls the code) to us... this also applies for documentation and changes, I know there is a change file but to be honest, sometimes it's not clear enough and you have to go into the code and take a look at what was changed... but this I can live with, after all, like you said, it is open source.. And I just want to say this again, I (and probably some others) am willing to pay some $$ to Steve Underwood if he would consider porting unicall to 1.4 Steve, please come back! :) |-Original Message- |From: [EMAIL PROTECTED] [mailto:asterisk-users- |[EMAIL PROTECTED] On Behalf Of Moises Silva |Sent: Friday, January 05, 2007 9:41 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [asterisk-users] no unicall on 1.4 | |On 1/5/07, Anton Krall [EMAIL PROTECTED] wrote: | when did asterisk turn from an open source project with very good developers | nto a business that only focuses in $$$? |They are not mutually exclusive. | | That's why openpbx was born I guess |I dont think so. I think is more because of technical disagreements. | | For example, samba is still free, and people are making a profit from it by |When does Asterisk stoped being free, sorry but I missed something? :) | | In any case, I (and maybe some other folks) would definitely pay some $$ to | Steve is he would consider supporting unicall for 1.4... I've always | believed that if you make money with something, why not give some to the | good programmers that made it happen... |Of course. What many buissiness people dont get is that programmers |have to eat ;) |And companies like Intruder and Office Connect are selling services |based on software created by people for free, the least think those |companies can do is support the programmers with money, the same thing |you are getting, right? | | | probably Digium will end up hiring their own programmers... |There are several programmers that work for Digium, but that does not mean |the project stops being open source ( http://en.wikipedia.org/wiki/Open_source ) | | what is Digium doing 100% right? HW? No, software, used to, but maybe not now, |so? |Sangoma has a much bigger background in electronics, give Digium a |break, they keep working hard on software, and I dont think they are |going to quit, soon or later the software and hardware will get |better. | | come on guys at Digium, focus... | open source software was about making something by the community for the | community, not getting XXX million USD in VC while losing your best | programmers, the community and the R2MFC market :) |Actually you need to read a little more about the open source term, |and dont use it loosely. | |I think critics are good, eventually will push Digium to do better |software, but Digium is still a company that needs to make money: |money != evil. | |At the end, is open source/freesoftware, if you dont like it, nobody |is stopping you from change it. | |Kind Regards | |-- |Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; |___ |--Bandwidth and Colocation provided by Easynews.com -- | |asterisk-users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users BEGIN:VCARD VERSION:2.1 X-MS-SIGNATURE:YES N;LANGUAGE=en-us:Krall;Anton FN:Anton Krall ORG:Intruder Consulting TITLE:A Division of IntruderEnterprises S.A. de C.V. TEL;WORK;VOICE:+52 (55) 5781-5112 x 201 TEL;WORK;VOICE:+52 (55) 5985-2430 x 201 X-MS-OL-DEFAULT-POSTAL-ADDRESS:0 URL;WORK:http://www.intruder.com.mx EMAIL;PREF;INTERNET:[EMAIL PROTECTED] PHOTO;TYPE=JPEG;ENCODING=BASE64: /9j/4AAQSkZJRgABAQEAYABgAAD/2wBDAAYEBQYFBAYGBQYHBwYIChAKCgkJChQODwwQFxQY GBcUFhYaHSUfGhsjHBYWICwgIyYnKSopGR8tMC0oMCUoKSj/2wBDAQcHBwoIChMKChMoGhYa KCgoKCgoKCgoKCgoKCgoKCgoKCgoKCgoKCgoKCgoKCgoKCgoKCgoKCgoKCgoKCgoKCj/wAAR CAAXAEgDASIAAhEBAxEB/8QAHwAAAQUBAQEBAQEAAAECAwQFBgcICQoL/8QAtRAA
RE: [asterisk-users] no unicall on 1.4 (was: OnHook Call Announcement...)
Well Moises, if you do, please drop me a line and I will gladly test it. I was mentioning digium because AFAIK, the guys at digium are in touch with the programmers and contributors so I thought maybe they would have an insight on whats going to happen with unicall on 1.4, I mean, somebody at the source should know right? Many people still use unicall so I thought somebody would pick up the ball, maybe thats going to be you hopefuly. Let me know how it goes. |-Original Message- |From: [EMAIL PROTECTED] [mailto:asterisk-users- |[EMAIL PROTECTED] On Behalf Of Moises Silva |Sent: Wednesday, January 03, 2007 5:22 PM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [asterisk-users] no unicall on 1.4 (was: OnHook Call Announcement...) | |On 1/3/07, Anton Krall [EMAIL PROTECTED] wrote: | And probably wont be as Steve Underwood explained to me that he is now supporting |openpbx and has stopped support for unicall on asterisk 1.4 | | Can anybody at digium confirm? Is unicall going to be left out of 1.4? | |This has nothing to do with Digium, it has to do with anybody wanting |to code the version for 1.4, AFAIK Steve never worked for Digium and |Digium never distributed Unicall driver. | |Porting Unicall to 1.4 is in my TODO since 1 month ago, may be this |month I will have the time to give a look at the code and try to make |it work on 1.4, if somebody else cant do it before. | |Regards. | |-- |Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; |___ |--Bandwidth and Colocation provided by Easynews.com -- | |asterisk-users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] no unicall on 1.4
This is exactly one of the things that Steve and I discussed a bit ago... when did asterisk turn from an open source project with very good developers into a business that only focuses in $$$? Thats why openpbx was born I guess For example, samba is still free, and people are making a profit from it by giving out consulting services for deploying samba.. that is a good working scenario asterisk used to be the same can you spell greedy :)? In any case, I (and maybe some other folks) would definitely pay some $$ to Steve is he would consider supporting unicall for 1.4... I've always believed that if you make money with something, why not give some to the good programmers that made it happen... Just my $0.02 |-Original Message- |From: [EMAIL PROTECTED] [mailto:asterisk-users- |[EMAIL PROTECTED] On Behalf Of Lee Howard |Sent: Thursday, January 04, 2007 7:48 PM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [asterisk-users] no unicall on 1.4 | |Barzilai Spinak wrote: | | 2) Why is it that Digium never gave a damn about E1/MFC/R2... | | |My guess would be because they don't have any E1/MFC/R2 hardware |products to market. | |Lee. |___ |--Bandwidth and Colocation provided by Easynews.com -- | |asterisk-users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users BEGIN:VCARD VERSION:2.1 X-MS-SIGNATURE:YES N;LANGUAGE=en-us:Krall;Anton FN:Anton Krall ORG:Intruder Consulting TITLE:A Division of IntruderEnterprises S.A. de C.V. TEL;WORK;VOICE:+52 (55) 5781-5112 x 201 TEL;WORK;VOICE:+52 (55) 5985-2430 x 201 X-MS-OL-DEFAULT-POSTAL-ADDRESS:0 URL;WORK:http://www.intruder.com.mx EMAIL;PREF;INTERNET:[EMAIL PROTECTED] PHOTO;TYPE=JPEG;ENCODING=BASE64: /9j/4AAQSkZJRgABAQEAYABgAAD/2wBDAAYEBQYFBAYGBQYHBwYIChAKCgkJChQODwwQFxQY GBcUFhYaHSUfGhsjHBYWICwgIyYnKSopGR8tMC0oMCUoKSj/2wBDAQcHBwoIChMKChMoGhYa KCgoKCgoKCgoKCgoKCgoKCgoKCgoKCgoKCgoKCgoKCgoKCgoKCgoKCgoKCgoKCgoKCj/wAAR CAAXAEgDASIAAhEBAxEB/8QAHwAAAQUBAQEBAQEAAAECAwQFBgcICQoL/8QAtRAA AgEDAwIEAwUFBAQAAAF9AQIDAAQRBRIhMUEGE1FhByJxFDKBkaEII0KxwRVS0fAkM2JyggkK FhcYGRolJicoKSo0NTY3ODk6Q0RFRkdISUpTVFVWV1hZWmNkZWZnaGlqc3R1dnd4eXqDhIWG h4iJipKTlJWWl5iZmqKjpKWmp6ipqrKztLW2t7i5usLDxMXGx8jJytLT1NXW19jZ2uHi4+Tl 5ufo6erx8vP09fb3+Pn6/8QAHwEAAwEBAQEBAQEBAQECAwQFBgcICQoL/8QAtREA AgECBAQDBAcFBAQAAQJ3AAECAxEEBSExBhJBUQdhcRMiMoEIFEKRobHBCSMzUvAVYnLRChYk NOEl8RcYGRomJygpKjU2Nzg5OkNERUZHSElKU1RVVldYWVpjZGVmZ2hpanN0dXZ3eHl6goOE hYaHiImKkpOUlZaXmJmaoqOkpaanqKmqsrO0tba3uLm6wsPExcbHyMnK0tPU1dbX2Nna4uPk 5ebn6Onq8vP09fb3+Pn6/9oADAMBAAIRAxEAPwDI8HeCb3xFHJclhb2MY+adumfQDua62b4Q iBGluNWEcOPlZreQZJ/4DXS/DAXn/CBxS6fArXscc5gbPOflGfrgmun8NNfr4dt5vESs2oG2 lRNxy7ZZcYHryfwzXo4nGVouTi0knY+ew+CoyhHmV21e58/XvhaWz1XWLGZpC+n27TkxoGBw QOckYHPXn6VFdeDdftbV7i4050iRFkbMiblRsYYrnIXkc4xXWeJNRFp408Xosc9217YyW6mF d20llO5vQDFUdQ8TxT6pr2pi0vBaalpqafCzLwHEcanJzjGUbpXoxqVGk/66HnzpUU2n3f6/ 8D7zO1D4f67bao9lbQJdukEdw7xyKFRXAPJJ4wTjJ9M9Kxl0DVHu7a1Wzcz3MJuIUBHzxgMS w56YVj+FdnrevxXMesW8FjqaX2oaba27RPDjyzDsBPXJUhOuO9LpPijRoDo+p3UOp/bNP017 ARxxqYmyroH3k5/j6Y696FUqct2v6t/mEqNFysnZf8H07amJ4Z8CarrU1o0sRtbK4jeVZmZS 2xVJ3CMsGKkjGcY5rNs/CutXmmrfW1i727Kzod6hnVfvMqE7mA55ANdvYa7psOraXq9/b6ol /YW/9lNBEiGFpVRkBD7hjg5K4696bpnjOC30XSrqSC9huNNtjZqYrOFkkYbtpEzAsn3uQAf1 pOpVvov61/4BSo0LJN/1p/wdNzg9Q0DU9P0221C9tWhtLkBoXZl+cEZBAznGO+KKveM9Qa/b Rw1vcQfZtOhtsTLt3Fc5ZfbNFdEG2rs46qjGVo7Gh4Z8e3uhaetpHbW9wiZ8syru2Z64BrRn +KmqPaslva2ttcFCnnxIFYA9en86KKyeGpSlzOKuaRxdaK5VLQ4qHVLiOaeRisrTDD+YM8g5 B+oIpG1KdtPSzITy0xhsfMQCzAfTLMfxoorblRjzPuWItevY74XeY2l2svK8EM5c5APPzMeO nY8VX/tGb+zjZbY/LJ+9t+bGc4z6Zooo5UHPLuOOq3JnMx2eYbn7Xnb0fOfyqP8AtCf7IbbK +Ue2OfvBv5iiijlQcz7ljXNUGqSwuIBDsVt2GzuZmLM34k0UUU0klZClJyd2f//Z X-MS-OL-DESIGN;CHARSET=utf-8:card xmlns=http://schemas.microsoft.com/office/outlook/12/electronicbusinesscards; ver=1.0 layout=top bgcolor=ffimg xmlns= align=tright area=25 use=photo/fld xmlns= prop=name align=left dir=ltr style=b color=00 size=10/fld xmlns= prop=org align=left dir=ltr color=00 size=8/fld xmlns= prop=title align=left dir=ltr color=00 size=8/fld xmlns= prop=blank size=8/fld xmlns= prop=telwork align=left dir=ltr color=00 size=8label align=left color=626262Work/label/fldfld xmlns= prop=telwork2 align=left dir=ltr color=00 size=8label align=left color=626262Work/label/fldfld xmlns= prop=email align=left dir=ltr color=00 size=8label align=left color=626262Email amp; MSN: /label/fldfld xmlns= prop=webwork align=left dir=ltr color=00 size=8label align=left color=626262Web: /label/fldfld xmlns= prop=blank size=8/fld xmlns= prop=blank size=8/fld xmlns= prop=blank size=8/fld xmlns= prop=blank size=8/fld xmlns= prop=blank size=8/fld xmlns= prop=blank size=8/fld xmlns= prop=blank size=8/fld xmlns= prop=blank size=8//card REV:20061218T091156Z END:VCARD ___ --Bandwidth and Colocation
[asterisk-users] no unicall on 1.4 (was: OnHook Call Announcement...)
And probably wont be as Steve Underwood explained to me that he is now supporting openpbx and has stopped support for unicall on asterisk 1.4 Can anybody at digium confirm? Is unicall going to be left out of 1.4? |-Original Message- |From: [EMAIL PROTECTED] [mailto:asterisk-users- |[EMAIL PROTECTED] On Behalf Of Carlos Chavez |Sent: Tuesday, January 02, 2007 6:02 PM |To: Asterisk |Subject: [asterisk-users] OnHook Call Announcement... | | I have a customer that is asking for a feature called On Hook Call |Announcement. The way he explains it is that when someone is on another call you can |sort of break in into their conversation but only the local person hears you and not the |external caller. | | Basically he wants to use this function so he can call anyone in the company |even if they are already on a call (he is the big boss). I saw that there is a feature |coming in 1.4 called Whisper paging that may do something like this but I need to know |if it is possible to do it in 1.2 because there is still no support for Unicall on 1.4 | |-- |Telecomunicaciones Abiertas de Mexico S.A. de C.V. |Carlos Chvez Prats |Director de Tecnologa |+52-55-91169161 ext 2001 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] 1.4 and unicall
No update on unicall and 1.4? |-Original Message- |From: [EMAIL PROTECTED] [mailto:asterisk-users- |[EMAIL PROTECTED] On Behalf Of Anton Krall |Sent: Tuesday, December 26, 2006 6:15 AM |To: asterisk-users@lists.digium.com |Subject: [asterisk-users] 1.4 and unicall | |Guys, anybody knows if 1.4 has support for unicall or if/which version of |unicall will compile on it? | | |___ |--Bandwidth and Colocation provided by Easynews.com -- | |asterisk-users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] 1.4 and unicall
I hope so, he is the only guy working on mfcr2 right now. I have unicall working on 1.2 perfectly but if there will be no unicall support for 1.4, that would be a show stopper unless we use a mfcr2 converter... anybody knows any? Something that can convert mfcr2 to pri? |-Original Message- |From: [EMAIL PROTECTED] [mailto:asterisk-users- |[EMAIL PROTECTED] On Behalf Of Barzilai Spinak |Sent: Thursday, December 28, 2006 8:26 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [asterisk-users] 1.4 and unicall | |I asked the same a while ago, without any kind of conclusive answer. |But you have to consider that these are special dates |I just spent all night studying/modifying mfcr2.c to my needs but |I've never looked at the unicall code or the asterisk channel API. |With respect to MFC/R2, and according to what it saw, it seems fairly |complete on the incoming part of the protocol, but the outgoing logic is |kind of crude. |I wonder if Steve Underwood is still actively working on it. | |BarZ | |Anton Krall wrote: | No update on unicall and 1.4? | | |-Original Message- | |From: [EMAIL PROTECTED] [mailto:asterisk-users- | |[EMAIL PROTECTED] On Behalf Of Anton Krall | |Sent: Tuesday, December 26, 2006 6:15 AM | |To: asterisk-users@lists.digium.com | |Subject: [asterisk-users] 1.4 and unicall | | | |Guys, anybody knows if 1.4 has support for unicall or if/which version of | |unicall will compile on it? | | | | | |___ | |--Bandwidth and Colocation provided by Easynews.com -- | | | |asterisk-users mailing list | |To UNSUBSCRIBE or update options visit: | | http://lists.digium.com/mailman/listinfo/asterisk-users | | | | ___ | --Bandwidth and Colocation provided by Easynews.com -- | | asterisk-users mailing list | To UNSUBSCRIBE or update options visit: |http://lists.digium.com/mailman/listinfo/asterisk-users | | |___ |--Bandwidth and Colocation provided by Easynews.com -- | |asterisk-users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users BEGIN:VCARD VERSION:2.1 X-MS-SIGNATURE:YES N;LANGUAGE=en-us:Krall;Anton FN:Anton Krall ORG:Intruder Consulting TITLE:A Division of IntruderEnterprises S.A. de C.V. TEL;WORK;VOICE:+52 (55) 5781-5112 x 201 TEL;WORK;VOICE:+52 (55) 5985-2430 x 201 X-MS-OL-DEFAULT-POSTAL-ADDRESS:0 URL;WORK:http://www.intruder.com.mx EMAIL;PREF;INTERNET:[EMAIL PROTECTED] PHOTO;TYPE=JPEG;ENCODING=BASE64: /9j/4AAQSkZJRgABAQEAYABgAAD/2wBDAAYEBQYFBAYGBQYHBwYIChAKCgkJChQODwwQFxQY GBcUFhYaHSUfGhsjHBYWICwgIyYnKSopGR8tMC0oMCUoKSj/2wBDAQcHBwoIChMKChMoGhYa KCgoKCgoKCgoKCgoKCgoKCgoKCgoKCgoKCgoKCgoKCgoKCgoKCgoKCgoKCgoKCgoKCj/wAAR CAAXAEgDASIAAhEBAxEB/8QAHwAAAQUBAQEBAQEAAAECAwQFBgcICQoL/8QAtRAA AgEDAwIEAwUFBAQAAAF9AQIDAAQRBRIhMUEGE1FhByJxFDKBkaEII0KxwRVS0fAkM2JyggkK FhcYGRolJicoKSo0NTY3ODk6Q0RFRkdISUpTVFVWV1hZWmNkZWZnaGlqc3R1dnd4eXqDhIWG h4iJipKTlJWWl5iZmqKjpKWmp6ipqrKztLW2t7i5usLDxMXGx8jJytLT1NXW19jZ2uHi4+Tl 5ufo6erx8vP09fb3+Pn6/8QAHwEAAwEBAQEBAQEBAQECAwQFBgcICQoL/8QAtREA AgECBAQDBAcFBAQAAQJ3AAECAxEEBSExBhJBUQdhcRMiMoEIFEKRobHBCSMzUvAVYnLRChYk NOEl8RcYGRomJygpKjU2Nzg5OkNERUZHSElKU1RVVldYWVpjZGVmZ2hpanN0dXZ3eHl6goOE hYaHiImKkpOUlZaXmJmaoqOkpaanqKmqsrO0tba3uLm6wsPExcbHyMnK0tPU1dbX2Nna4uPk 5ebn6Onq8vP09fb3+Pn6/9oADAMBAAIRAxEAPwDI8HeCb3xFHJclhb2MY+adumfQDua62b4Q iBGluNWEcOPlZreQZJ/4DXS/DAXn/CBxS6fArXscc5gbPOflGfrgmun8NNfr4dt5vESs2oG2 lRNxy7ZZcYHryfwzXo4nGVouTi0knY+ew+CoyhHmV21e58/XvhaWz1XWLGZpC+n27TkxoGBw QOckYHPXn6VFdeDdftbV7i4050iRFkbMiblRsYYrnIXkc4xXWeJNRFp408Xosc9217YyW6mF d20llO5vQDFUdQ8TxT6pr2pi0vBaalpqafCzLwHEcanJzjGUbpXoxqVGk/66HnzpUU2n3f6/ 8D7zO1D4f67bao9lbQJdukEdw7xyKFRXAPJJ4wTjJ9M9Kxl0DVHu7a1Wzcz3MJuIUBHzxgMS w56YVj+FdnrevxXMesW8FjqaX2oaba27RPDjyzDsBPXJUhOuO9LpPijRoDo+p3UOp/bNP017 ARxxqYmyroH3k5/j6Y696FUqct2v6t/mEqNFysnZf8H07amJ4Z8CarrU1o0sRtbK4jeVZmZS 2xVJ3CMsGKkjGcY5rNs/CutXmmrfW1i727Kzod6hnVfvMqE7mA55ANdvYa7psOraXq9/b6ol /YW/9lNBEiGFpVRkBD7hjg5K4696bpnjOC30XSrqSC9huNNtjZqYrOFkkYbtpEzAsn3uQAf1 pOpVvov61/4BSo0LJN/1p/wdNzg9Q0DU9P0221C9tWhtLkBoXZl+cEZBAznGO+KKveM9Qa/b Rw1vcQfZtOhtsTLt3Fc5ZfbNFdEG2rs46qjGVo7Gh4Z8e3uhaetpHbW9wiZ8syru2Z64BrRn +KmqPaslva2ttcFCnnxIFYA9en86KKyeGpSlzOKuaRxdaK5VLQ4qHVLiOaeRisrTDD+YM8g5 B+oIpG1KdtPSzITy0xhsfMQCzAfTLMfxoorblRjzPuWItevY74XeY2l2svK8EM5c5APPzMeO nY8VX/tGb+zjZbY/LJ+9t+bGc4z6Zooo5UHPLuOOq3JnMx2eYbn7Xnb0fOfyqP8AtCf7IbbK +Ue2OfvBv5iiijlQcz7ljXNUGqSwuIBDsVt2GzuZmLM34k0UUU0klZClJyd2f//Z X-MS-OL-DESIGN;CHARSET=utf-8:card xmlns=http://schemas.microsoft.com/office/outlook/12/electronicbusinesscards; ver=1.0 layout=top bgcolor=ffimg xmlns= align=tright area=25 use=photo/fld xmlns= prop=name align=left dir=ltr style=b color=00 size=10/fld xmlns= prop=org align=left dir=ltr color=00 size=8/fld xmlns= prop=title align=left dir=ltr color=00 size=8/fld xmlns= prop=blank
[asterisk-users] 1.4 and unicall
Guys, anybody knows if 1.4 has support for unicall or if/which version of unicall will compile on it? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] agi+cepstral driving me nuts
Too bad Cepstral hasnt still made a decent Spanish voice, the ones they
have still sound too computer like, not like the English ones they have
which sound great!
|-Original Message-
|From: [EMAIL PROTECTED] [mailto:asterisk-users-
|[EMAIL PROTECTED] On Behalf Of Julian J. M.
|Sent: Tuesday, December 26, 2006 6:26 PM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [asterisk-users] agi+cepstral driving me nuts
|
|Why don't you try app_swift?
|http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Swift
|
|This one even compiles on 1.4, and has buffering, meaning that it
|doesn't have to wait for the tts to generate the complete output.
|
|http://www.loopfree.net/app_swift/
|
|exten = s,1,AGI(getinfo.php)
|exten = s,2,Swift( ${RESULT_INFORMATION} )
|
|Julián J. M.
|
|On 12/26/06, blackwater dev [EMAIL PROTECTED] wrote:
| I just got cepstal working fine in the dial plan using code like:
|
| exten = 511,5,AGI(cepstral.pl|Welcome to my house finder. At the beep
| enter your zip code.)
|
|
| The php script it calls is based on the nerdvittles weather one so it
calls
| a webpage which prints to the screen, the nerdvittles code uses system to
| generate the .wav file then has the dial plan call it via:
|
| //php script
| $retcode2 = system (flite -f $tmptext -o $tmpwave) ;
|
| //extensions
| exten = 411,9,NoOp(Wave file: ${TMPWAVE})
| exten = 411,10,Playback(${TMPWAVE})
|
|
| Since I am using capstral, I simply changed the line to below which works
| fine from the command line but when calling, I never hear it, it just
hangs
| up. Is it timing out? Is there a better way to do this? How can I
return
| just a string of Text to read so I don't have to create the .wav file
then
| play it?
|
| $retcode2 = system (swift -n Diane -m text -f $tmptext -o $tmpwave) ;
|
|
| Thanks!
|
|___
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|
|asterisk-users mailing list
|To UNSUBSCRIBE or update options visit:
| http://lists.digium.com/mailman/listinfo/asterisk-users
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RE: [Asterisk-Users] asterisk shutdown
Guys.
I was able to put asterisk on debug for a while and another shutdown took
place, here are the logs... I was wondering.. I notice that we are using a
lot of manager actions and can it be that if you place a lot of manager
actions in a short period of time, that it makes asterisk do a shutdown?
Maybe a bug?
[Jul 4 12:59:59] VERBOSE[8953]: [Jul 4 12:59:59] == Parsing
'/etc/asterisk/manager.conf': [Jul 4 12:59:59] VERBOSE[8953]: [Jul 4
12:59:59] == Parsing
'/etc/asterisk/manager.conf': [Jul 4 12:59:59] Found
[Jul 4 12:59:59] DEBUG[8953]: 0.0.0.0/0.0.0.0/0.0.0.0 appended to acl for
peer
[Jul 4 12:59:59] DEBUG[8953]: 127.0.0.1/255.255.255.255/255.255.255.255
appended to acl for peer
[Jul 4 12:59:59] DEBUG[8953]: 10.0.0.0/255.0.0.0/255.0.0.0 appended to acl
for peer
[Jul 4 12:59:59] DEBUG[8953]: # Testing 127.0.0.1 with 0.0.0.0
[Jul 4 12:59:59] DEBUG[8953]: # Testing 127.0.0.1 with 127.0.0.1
[Jul 4 12:59:59] DEBUG[8953]: # Testing 127.0.0.1 with 10.0.0.0
[Jul 4 12:59:59] DEBUG[8953]: Manager received command 'Command'
[Jul 4 12:59:59] DEBUG[8953]: Manager received command 'Logoff'
[Jul 4 12:59:59] DEBUG[8954]: Manager received command 'Login'
[Jul 4 12:59:59] VERBOSE[8954]: [Jul 4 12:59:59] == Parsing
'/etc/asterisk/manager.conf': [Jul 4 12:59:59] VERBOSE[8954]: [Jul 4
12:59:59] == Parsing
'/etc/asterisk/manager.conf': [Jul 4 12:59:59] Found
[Jul 4 12:59:59] DEBUG[8954]: 0.0.0.0/0.0.0.0/0.0.0.0 appended to acl for
peer
[Jul 4 12:59:59] DEBUG[8954]: 127.0.0.1/255.255.255.255/255.255.255.255
appended to acl for peer
[Jul 4 12:59:59] DEBUG[8954]: 10.0.0.0/255.0.0.0/255.0.0.0 appended to acl
for peer
[Jul 4 12:59:59] DEBUG[8954]: # Testing 127.0.0.1 with 0.0.0.0
[Jul 4 12:59:59] DEBUG[8954]: # Testing 127.0.0.1 with 127.0.0.1
[Jul 4 12:59:59] DEBUG[8954]: # Testing 127.0.0.1 with 10.0.0.0
[Jul 4 12:59:59] DEBUG[8954]: Manager received command 'Command'
[Jul 4 12:59:59] DEBUG[8954]: Manager received command 'Logoff'
[Jul 4 12:59:59] DEBUG[8955]: Manager received command 'Login'
[Jul 4 12:59:59] VERBOSE[8955]: [Jul 4 12:59:59] == Parsing
'/etc/asterisk/manager.conf': [Jul 4 12:59:59] VERBOSE[8955]: [Jul 4
12:59:59] == Parsing
'/etc/asterisk/manager.conf': [Jul 4 12:59:59] Found
[Jul 4 12:59:59] DEBUG[8955]: 0.0.0.0/0.0.0.0/0.0.0.0 appended to acl for
peer
[Jul 4 12:59:59] DEBUG[8955]: 127.0.0.1/255.255.255.255/255.255.255.255
appended to acl for peer
[Jul 4 12:59:59] DEBUG[8955]: 10.0.0.0/255.0.0.0/255.0.0.0 appended to acl
for peer
[Jul 4 12:59:59] DEBUG[8955]: # Testing 127.0.0.1 with 0.0.0.0
[Jul 4 12:59:59] DEBUG[8955]: # Testing 127.0.0.1 with 127.0.0.1
[Jul 4 12:59:59] DEBUG[8955]: # Testing 127.0.0.1 with 10.0.0.0
[Jul 4 12:59:59] DEBUG[8955]: Manager received command 'Command'
[Jul 4 12:59:59] DEBUG[8955]: Manager received command 'Logoff'
[Jul 4 13:00:02] DEBUG[8663]: update_call_counter(114) - decrement call
limit counter
[Jul 4 13:00:02] DEBUG[8663]: Acked pending invite 102
[Jul 4 13:00:02] DEBUG[8663]: Stopping retransmission on
'[EMAIL PROTECTED]' of Request 102: Match Found
[Jul 4 13:00:02] DEBUG[8663]: Exiting with DIALSTATUS=CANCEL.
[Jul 4 13:00:02] VERBOSE[8663]: [Jul 4 13:00:02] == Spawn extension
(internalphones, 114, 7) exited non-zero on 'SIP/101-f035'
[Jul 4 13:00:02] VERBOSE[8669]: [Jul 4 13:00:02] == End MixMonitor
Recording SIP/101-f035
[Jul 4 13:00:02] VERBOSE[8669]: [Jul 4 13:00:02] == Executing [chmod 777
/digrec/20060704-125935-1152035975.864-from-101-to-114.wav]
[Jul 4 13:00:02] DEBUG[8663]: Spy Ãd removed from channel SIP/101-f035
[Jul 4 13:00:02] DEBUG[8663]: cdr_mysql: inserting a CDR record.
[Jul 4 13:00:02] DEBUG[8663]: cdr_mysql: SQL command as follows: INSERT
INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duratio
n,billsec,disposition,amaflags,accountcode,uniqueid,userfield) VALUES
('2006-07-04 12:59:35','\Recepcion\ 101','101','114','internalphones',
'SIP/101-f03
5','SIP/114-b115','Dial','SIP/114|120|rtwTW',27,0,'NO
ANSWER',3,'Recepcion','1152035975.864','[Recepcion - Miguel Angel Sanz] ')
[Jul 4 13:00:02] DEBUG[8663]: update_call_counter(101) - decrement call
limit counter
[Jul 4 13:00:02] DEBUG[8319]: update_call_counter(101) - decrement call
limit counter
[Jul 4 13:00:02] DEBUG[8319]: Acked pending invite 102
[Jul 4 13:00:02] DEBUG[8319]: Stopping retransmission on
'[EMAIL PROTECTED]' of Request 102: Match Found
[Jul 4 13:00:02] DEBUG[8305]: Didn't get a frame from channel: SIP/101-d287
[Jul 4 13:00:02] DEBUG[8305]: Bridge stops bridging channels UniCall/4-1
and SIP/101-d287
[Jul 4 13:00:02] DEBUG[8305]: update_call_counter(101) - decrement call
limit counter
[Jul 4 13:00:02] DEBUG[8227]: Didn't get a frame from channel: SIP/101-a2b9
[Jul 4 13:00:02] DEBUG[8227]: Bridge stops bridging channels Zap/35-1 and
SIP/101-a2b9
[Jul 4 13:00:02] DEBUG[8227]: update_call_counter(101) - decrement call
limit counter
[Jul 4 13:00:02]
RE: [Asterisk-Users] asterisk shutdown
Well guys. Another day of shutdowns. [Jul 3 07:00:08] VERBOSE[3086]: [Jul 3 07:00:08] Beginning asterisk shutdown [Jul 3 07:02:23] VERBOSE[2657]: [Jul 3 07:02:23] Beginning asterisk shutdown [Jul 3 10:59:01] VERBOSE[3083]: [Jul 3 10:59:01] Beginning asterisk shutdown [Jul 3 11:29:02] VERBOSE[8720]: [Jul 3 11:29:02] Beginning asterisk shutdown [Jul 3 11:38:01] VERBOSE[16667]: [Jul 3 11:38:01] Beginning asterisk shutdown [Jul 3 15:26:40] VERBOSE[2733]: [Jul 3 15:26:40] Beginning asterisk shutdown [Jul 3 15:42:01] VERBOSE[3159]: [Jul 3 15:42:01] Beginning asterisk shutdown No apparent reason on logs... Mmhh... How can I enable more detailed debugging? I can run it for 1 day or so even if it hits performace a bit.. Just to be sure what the source of the problem is. |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Anton Krall |Sent: Friday, June 30, 2006 9:11 PM |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: RE: [Asterisk-Users] asterisk shutdown | |No log entries yet that might show whats happening and you are |correct, I cant run under strace as it would hit performance quite bad. | |:( I will continue to look into the logs and hope something |will show up so I can post further.. If anybody else |experiencing this can come up with some log entry, please share. | ||-Original Message- ||From: [EMAIL PROTECTED] ||[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir ||Cohen ||Sent: Friday, June 30, 2006 2:13 AM ||To: asterisk-users@lists.digium.com ||Subject: Re: [Asterisk-Users] asterisk shutdown || ||On Thu, Jun 29, 2006 at 10:54:58PM -0500, Anton Krall wrote: || So, no answers? Nobody knowd why this might be happening? ||Nobody else || experiencing this? || ||Is this a reproducable issue? Have you turned on verbosity and debug ||and log them (e.g. the full log)? || ||If still no messages and this is reproducable, consider running ||asteriskunder strace (-f). Though I figure that this could be a major ||performance hit. || ||-- ||Tzafrir Cohen sip:[EMAIL PROTECTED] ||icq#16849755 iax:[EMAIL PROTECTED] ||+972-50-7952406 ||[EMAIL PROTECTED] http://www.xorcom.com ||___ ||--Bandwidth and Colocation provided by Easynews.com -- || ||Asterisk-Users mailing list ||To UNSUBSCRIBE or update options visit: || http://lists.digium.com/mailman/listinfo/asterisk-users || | |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk shutdown
No log entries yet that might show whats happening and you are correct, I cant run under strace as it would hit performance quite bad. :( I will continue to look into the logs and hope something will show up so I can post further.. If anybody else experiencing this can come up with some log entry, please share. |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Tzafrir Cohen |Sent: Friday, June 30, 2006 2:13 AM |To: asterisk-users@lists.digium.com |Subject: Re: [Asterisk-Users] asterisk shutdown | |On Thu, Jun 29, 2006 at 10:54:58PM -0500, Anton Krall wrote: | So, no answers? Nobody knowd why this might be happening? |Nobody else | experiencing this? | |Is this a reproducable issue? Have you turned on verbosity and |debug and log them (e.g. the full log)? | |If still no messages and this is reproducable, consider |running asteriskunder strace (-f). Though I figure that this |could be a major performance hit. | |-- |Tzafrir Cohen sip:[EMAIL PROTECTED] |icq#16849755 iax:[EMAIL PROTECTED] |+972-50-7952406 |[EMAIL PROTECTED] http://www.xorcom.com |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk shutdown
So, no answers? Nobody knowd why this might be happening? Nobody else experiencing this? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Anton Krall |Sent: Wednesday, June 28, 2006 7:03 PM |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: RE: [Asterisk-Users] asterisk shutdown | |Same version, same problem... | ||-Original Message- ||From: [EMAIL PROTECTED] ||[mailto:[EMAIL PROTECTED] On Behalf Of Doug ||Lytle ||Sent: Wednesday, June 28, 2006 11:42 AM ||To: Asterisk Users Mailing List - Non-Commercial Discussion ||Subject: Re: [Asterisk-Users] asterisk shutdown || ||Anton Krall wrote: || Guys. || || Ive seen on my asterisk messages log that asterisk has ||shutdown itself || about || 12 times in 5 days... The logs show nothing but: || || ||What version? || ||I'm running 1.2.9.1 and saw one of my Asterisk process, this morning, ||just shut down for no apparent reason. I didn't have a |console at the ||time. The logs don't show anything. || ||Doug || ||-- || ||Ben Franklin quote: || ||Those who would give up Essential Liberty to purchase a little ||Temporary Safety, deserve neither Liberty nor Safety. || || ||___ ||--Bandwidth and Colocation provided by Easynews.com -- || ||Asterisk-Users mailing list ||To UNSUBSCRIBE or update options visit: || http://lists.digium.com/mailman/listinfo/asterisk-users || || | |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk shutdown
Guys. Ive seen on my asterisk messages log that asterisk has shutdown itself about 12 times in 5 days... The logs show nothing but: [Jun 28 09:40:02] WARNING[3172]: Unicall/4 event Drop call [Jun 28 09:40:02] WARNING[3172]: Unicall/4 event Release call [Jun 28 09:40:02] VERBOSE[3172]: [Jun 28 09:40:02] -- Unicall/4 released [Jun 28 09:40:02] VERBOSE[3084]: [Jun 28 09:40:02] Asterisk cleanly ending (15). [Jun 28 09:40:03] VERBOSE[28320]: [Jun 28 09:40:03] Asterisk Event Logger Started /var/log/asterisk/event_log [Jun 28 09:41:01] VERBOSE[28368]: [Jun 28 09:41:01] Beginning asterisk shutdown [Jun 28 09:41:01] VERBOSE[28368]: [Jun 28 09:41:01] Executing last minute cleanups [Jun 28 09:41:01] VERBOSE[28368]: [Jun 28 09:41:01] Asterisk cleanly ending (15). [Jun 28 09:41:01] VERBOSE[28457]: [Jun 28 09:41:01] Asterisk Event Logger Started /var/log/asterisk/event_log [Jun 28 09:41:01] VERBOSE[28457]: [Jun 28 09:41:01] == Parsing '/etc/asterisk/dnsmgr.conf': [Jun 28 09:41:01] VERBOSE[28457]: [Jun 28 09:41:01] == Parsin g '/etc/asterisk/dnsmgr.conf': [Jun 28 09:41:01] Found [Jun 28 09:41:01] NOTICE[28457]: Managed DNS entries will be refreshed every 1200 seconds. [Jun 28 09:41:01] VERBOSE[28457]: [Jun 28 09:41:01] Asterisk Dynamic Loader loading preload modules: [Jun 28 09:41:01] VERBOSE[28457]: [Jun 28 09:41:01] == Parsing '/etc/asterisk/modules.conf': [Jun 28 09:41:01] VERBOSE[28457]: [Jun 28 09:41:01] == Parsi ng '/etc/asterisk/modules.conf': [Jun 28 09:41:01] Found [Jun 28 09:41:01] VERBOSE[28457]: [Jun 28 09:41:01] == Manager registered action Ping [Jun 28 09:41:01] VERBOSE[28457]: [Jun 28 09:41:01] == Manager registered action Events [Jun 28 09:41:01] VERBOSE[28457]: [Jun 28 09:41:01] == Manager registered action Logoff [Jun 28 09:41:01] VERBOSE[28457]: [Jun 28 09:41:01] == Manager registered action Hangup [Jun 28 09:41:01] VERBOSE[28457]: [Jun 28 09:41:01] == Manager registered action Status [Jun 28 09:41:01] VERBOSE[28457]: [Jun 28 09:41:01] == Manager registered action Setvar [Jun 28 09:41:01] VERBOSE[28457]: [Jun 28 09:41:01] == Manager registered action Getvar As you can see, there are no noticeable errors or anything so.. Anybody has seen this before? Is there any way to make asterisk more verbose? Im running it as -cg Any hints? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk shutdown
Same version, same problem... |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Doug Lytle |Sent: Wednesday, June 28, 2006 11:42 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] asterisk shutdown | |Anton Krall wrote: | Guys. | | Ive seen on my asterisk messages log that asterisk has |shutdown itself | about | 12 times in 5 days... The logs show nothing but: | | |What version? | |I'm running 1.2.9.1 and saw one of my Asterisk process, this |morning, just shut down for no apparent reason. I didn't have |a console at the time. The logs don't show anything. | |Doug | |-- | |Ben Franklin quote: | |Those who would give up Essential Liberty to purchase a |little Temporary Safety, deserve neither Liberty nor Safety. | | |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk shutdown
Im my case, the box is closed down so I dont think its an intruder issue... Im puzzled... From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of William PiperSent: Wednesday, June 28, 2006 4:41 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] asterisk shutdown On 6/28/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Wed, Jun 28, 2006 at 04:39:29PM -0400, William Piper wrote: The same thing happened to me... I had to get a linux expert to take care of it for me. I believe the files were either "libpam" or "libss". They were telling asterisk to shutdown. I believe they deleted the files it that fixed it.Is it following a glibc upgrade or something? I'm not sure what happened only that the answer had to do with those files. Hell, I don't even know what the files do, all I know is that the files were likea trojan and were masked as the name of a legitimate file. The affect was the same as you are having though. Asterisk is not linked with pam at all. libnss is the glibc name serviceswitch and has dynamically loading code. Tighten down your firewall.Could you be more specific? This problem happened more than once to us. I believe someone founda weekness in our system found a way to access the server. We setup tight iptables and changed passwords and it hasn't happened again. I don't even know if this is the same thing, but it took about 7 hours of a linux guru to figure it out. I figure it couldn't hurt telling you about it. bp ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] sangoma unicall m2rfc
Steve. Im also getting a lot of these: Jun 20 10:34:58] WARNING[16786]: chan_unicall.c:2644 handle_uc_event: Unicall/1 event Dialing [Jun 20 10:35:01] WARNING[16786]: chan_unicall.c:2644 handle_uc_event: Unicall/1 event Far end disconnected [Jun 20 10:35:01] WARNING[16786]: chan_unicall.c:2930 handle_uc_event: CRN 32818 - far disconnected cause=Switching equipment congestion [42] |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Steve Underwood |Sent: Monday, June 19, 2006 7:15 PM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] sangoma unicall m2rfc | |Anton Krall wrote: | |Uys, Steve Underwood | |I just got a Sangoma A101 card and Im using unicall 0.0.3.pre9 for |R2MFC, I get the far and local end unblocked but as soon as I try to |make a call I get dialing and then protocol failure.. | |Do you guys know if there are any issues with sangoma and unicall? |Anybody has an a101 card working with unicall and r2mfc? | |Are you out there Steve? :) | | | |Lots of people are using Sangoma cards successfully with Unicall. | |Regards, |Steve |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sangoma unicall m2rfc
Uys, Steve Underwood I just got a Sangoma A101 card and Im using unicall 0.0.3.pre9 for R2MFC, I get the far and local end unblocked but as soon as I try to make a call I get dialing and then protocol failure.. Do you guys know if there are any issues with sangoma and unicall? Anybody has an a101 card working with unicall and r2mfc? Are you out there Steve? :) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] sangoma unicall m2rfc
Are you around Steve? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Anton Krall |Sent: Monday, June 19, 2006 11:58 AM |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: [Asterisk-Users] sangoma unicall m2rfc | |Uys, Steve Underwood | |I just got a Sangoma A101 card and Im using unicall 0.0.3.pre9 |for R2MFC, I get the far and local end unblocked but as soon |as I try to make a call I get dialing and then protocol failure.. | |Do you guys know if there are any issues with sangoma and |unicall? Anybody has an a101 card working with unicall and r2mfc? | |Are you out there Steve? :) | | |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] sangoma unicall m2rfc
Any particular tips I should consider? Its very weird why I get protocol failure with sangoma and using the same config with digium cards it works ok. BTW how can I make unicall more verbose on asterisk logs? For example, Im getting a lot of dropped calls (not due to load or anything) so I want to know more about whats happening with unicall during the call, can this be done? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Steve Underwood |Sent: Monday, June 19, 2006 7:15 PM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] sangoma unicall m2rfc | |Anton Krall wrote: | |Uys, Steve Underwood | |I just got a Sangoma A101 card and Im using unicall 0.0.3.pre9 for |R2MFC, I get the far and local end unblocked but as soon as I try to |make a call I get dialing and then protocol failure.. | |Do you guys know if there are any issues with sangoma and unicall? |Anybody has an a101 card working with unicall and r2mfc? | |Are you out there Steve? :) | | | |Lots of people are using Sangoma cards successfully with Unicall. | |Regards, |Steve |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] sangoma unicall m2rfc
Moises, please contact me offlist, I have a couple of questions regarding your doc. Thx |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Moises Silva |Sent: Monday, June 19, 2006 9:10 PM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] sangoma unicall m2rfc | |Anton, Please read this document I wrote, is in spanish: | |http://phpmexic.u33.0web-hosting.com/wordpress/misc/mfcr2-aster |isk-unicall.pdf | |There you will find some ideas about how to debug problems with mfcr2. |In my blog I have some other files that you can download. |Check http://moy.ivsol.net/ | |Regards | |On 6/19/06, Anton Krall [EMAIL PROTECTED] wrote: | Any particular tips I should consider? Its very weird why I get | protocol failure with sangoma and using the same config with digium | cards it works ok. | | BTW how can I make unicall more verbose on asterisk logs? |For example, | Im getting a lot of dropped calls (not due to load or |anything) so I | want to know more about whats happening with unicall during |the call, | can this be done? | | |-Original Message- | |From: [EMAIL PROTECTED] | |[mailto:[EMAIL PROTECTED] On Behalf Of Steve | |Underwood | |Sent: Monday, June 19, 2006 7:15 PM | |To: Asterisk Users Mailing List - Non-Commercial Discussion | |Subject: Re: [Asterisk-Users] sangoma unicall m2rfc | | | |Anton Krall wrote: | | | |Uys, Steve Underwood | | | |I just got a Sangoma A101 card and Im using unicall 0.0.3.pre9 for | |R2MFC, I get the far and local end unblocked but as soon |as I try to | |make a call I get dialing and then protocol failure.. | | | |Do you guys know if there are any issues with sangoma and unicall? | |Anybody has an a101 card working with unicall and r2mfc? | | | |Are you out there Steve? :) | | | | | | | |Lots of people are using Sangoma cards successfully with Unicall. | | | |Regards, | |Steve | |___ | |--Bandwidth and Colocation provided by Easynews.com -- | | | |Asterisk-Users mailing list | |To UNSUBSCRIBE or update options visit: | | http://lists.digium.com/mailman/listinfo/asterisk-users | | | | | | ___ | --Bandwidth and Colocation provided by Easynews.com -- | | Asterisk-Users mailing list | To UNSUBSCRIBE or update options visit: |http://lists.digium.com/mailman/listinfo/asterisk-users | | | |-- |Su nombre es GNU/Linux, no solamente Linux, mas info en |http://www.gnu.org; |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Changing RO vars like SRC
Guys, is there a way to set CDR vards like SRC, I tried using set but asterisk complains they are RO vars. What Im trying to do is a small way to let users make calls from someone elses extension but auth using a password and seitch credential to their own so the call appears on CDR as made from their extension and not the one they are actually using. Is there a way to do this and somebody has done this before? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Roaming Users
Guys. I have a couple of agis that when trying to dial a local call, LD, etc. ask the user for a password and then checks against a DB to see if they can call or not. My newi dea here is to allow users to roam between extensions, for example, user 1 can go to users 2 phone and when ask for the password, enter his own and depending on privs, be able to call or not. Problem that I have here is CDR. How can I drop the current CDR record (which has users 2 extension inside) and create a new one with users 1 info on it as if the call was been made from users 1 from and not users 2? Anybody done something like this? Thx! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Roaming Users
Guys. I have a couple of agis that when trying to dial a local call, LD, etc. ask the user for a password and then checks against a DB to see if they can call or not. My newi dea here is to allow users to roam between extensions, for example, user 1 can go to users 2 phone and when ask for the password, enter his own and depending on privs, be able to call or not. Problem that I have here is CDR. How can I drop the current CDR record (which has users 2 extension inside) and create a new one with users 1 info on it as if the call was been made from users 1 from and not users 2? Anybody done something like this? Thx! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Unicall Protocol Failure
Muchas gracias Felix, voy a probar a ver que tal jala. Tu tuviste ese miusmo problema? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martinez FelixSent: Thursday, June 01, 2006 9:28 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Unicall Protocol Failure Cambiando un timer que existe en el archivo mfcr2.c La variable DEFAULT_T1 tiene el valor 5000, incrementalo a 2, compilas, instalas y listo mas o menos en la linea de codigo 102 actual #define DEFAULT_T1 5000 despues #define DEFAULT_T1 2 Espero te sirva. On 5/30/06, Anton Krall [EMAIL PROTECTED] wrote: Steve Underwood:Steve, why do some numbers give protocol errors? Ive noticed here in Mexicothat certain numbers when dialed return protocol failure and a busy tone.Any idea why this happens and why with only certain phone numbers? ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Volume configuration on Polycom Soundpoint501phone
Here ya go: For configuring the speakerphone volume after a phone restart (default volume) you need to set in sip.cfg: voice.gain.rx.digital.chassis="X" X been the default volume -15, -9 ... 0... 9...15 in 3 intervals For configuring the ring volume after a phone restart (default volume) you need to set in sip.cfg: voice.gain.rx.digital.ringer="X" X been the default volume -30, -27 ... 0 in -3 intervals For configuring the handset volume after a phone restart (default volume) you need to set in sip.cfg: voice.gain.rx.digital.handset="X" X been the default volume -30, -27 ... 0 in -3 intervals Hope this helps guys From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]Sent: Thursday, May 25, 2006 9:47 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] Volume configuration on Polycom Soundpoint501phone Could not find your post for 4 months ago. -- Original message ------ From: "Anton Krall" [EMAIL PROTECTED] Yes, check a post that I made about 4 months ago, I posted the cofig for setting the speaker, handset and ring volumes .. |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Jerry Jones |Sent: Thursday, May 04, 2006 3:15 PM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Volume configuration on Polycom |Soundpoint 501phone | |Edit your config files to enable persistance | |Will remain across multiple calls, but not reboots | | |On May 4, 2006, at 2:51 PM, Jim Freeze wrote: | | We are using the polycom 501 phones, and are having some challenges | with the volume setting. When a phone call comes in, the |user ups the | volume for the handset, but they have to repeat that for every call. | | Currently, the volume level seems to reset itself at about 60%. | Is there a way for the user to change their default volume level? | | Thanks | | -- | Jim Freeze | ___ | --Bandwidth and Colocation provided by Easynews.com -- | | Asterisk-Users mailing list | To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Aster isk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Unicall Protocol Failure
Pues muchas gracias por el tip. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martinez FelixSent: Friday, June 02, 2006 8:35 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Unicall Protocol Failure si, al principio, de ahora en adelante en todas la instalaciones qe hago codificamos ese parametro On 6/2/06, Anton Krall [EMAIL PROTECTED] wrote: Muchas gracias Felix, voy a probar a ver que tal jala. Tu tuviste ese miusmo problema? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Martinez FelixSent: Thursday, June 01, 2006 9:28 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Unicall Protocol Failure Cambiando un timer que existe en el archivo mfcr2.c La variable DEFAULT_T1 tiene el valor 5000, incrementalo a 2, compilas, instalas y listo mas o menos en la linea de codigo 102 actual #define DEFAULT_T1 5000 despues #define DEFAULT_T1 2 Espero te sirva. On 5/30/06, Anton Krall [EMAIL PROTECTED] wrote: Steve Underwood:Steve, why do some numbers give protocol errors? Ive noticed here in Mexicothat certain numbers when dialed return protocol failure and a busy tone.Any idea why this happens and why with only certain phone numbers? ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unicall Protocol Failure
Steve Underwood: Steve, why do some numbers give protocol errors? Ive noticed here in Mexico that certain numbers when dialed return protocol failure and a busy tone. Any idea why this happens and why with only certain phone numbers? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dumping queue_log to MySQL
Im using the fifo approach.. working great so far! From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin SavoySent: Friday, May 05, 2006 8:57 AMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: [Asterisk-Users] Dumping queue_log to MySQL Anyone have a working solution for this? I played with the demo that came with QueueMetrics to see how they were doing it and it was working for a bit but now somehow every night it stopped. Perl and Tail are still running on the server but the information is not dumping to the MySQL database. I dont get any error messages anywhere telling me why it stops. As far as tail and perl are concerned everything is fine. We will be using this for a call center and need more reliability. Anyone got one working? Thanks _ Kevin Savoy Business Unit Telecom Analyst 2218 4th Ave W Williston, ND 58801 Ph: 701-774-4023 Fax: 701-774-2901 http://www.novo1.com Novo 1 is a service mark of Novo 1, Inc ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Volume configuration on Polycom Soundpoint 501phone
Yes, check a post that I made about 4 months ago, I posted the cofig for setting the speaker, handset and ring volumes .. |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Jerry Jones |Sent: Thursday, May 04, 2006 3:15 PM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Volume configuration on Polycom |Soundpoint 501phone | |Edit your config files to enable persistance | |Will remain across multiple calls, but not reboots | | |On May 4, 2006, at 2:51 PM, Jim Freeze wrote: | | We are using the polycom 501 phones, and are having some challenges | with the volume setting. When a phone call comes in, the |user ups the | volume for the handset, but they have to repeat that for every call. | | Currently, the volume level seems to reset itself at about 60%. | Is there a way for the user to change their default volume level? | | Thanks | | -- | Jim Freeze | ___ | --Bandwidth and Colocation provided by Easynews.com -- | | Asterisk-Users mailing list | To UNSUBSCRIBE or update options visit: |http://lists.digium.com/mailman/listinfo/asterisk-users | |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: SV: [Asterisk-Users] Polycom 501 - Disable DND feature?
Do you know if you can disable soft keys like the blind xfer key that shows on the screen? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Derek Listmail Acct |Sent: Thursday, May 04, 2006 6:52 PM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: SV: [Asterisk-Users] Polycom 501 - Disable DND feature? | |You can disable the DND button completly. I think that will |get you what you want. | |I don't have a 500/501 handy to find out which button it is, |but you can check in Menu - Status - Diagnostics - Test |Hardware - Keypad Diagnostics. | |It's button 9 on my 600 and this disabled it: | |keys key.scrolling.timeout=1 key.IP_600.9.function.prim=Null/ | |--Derek | | | | Well, yes and no. I tested that before and it causes a silent ring | instead of a call rejection. I actually want to disable the entire | feature. So the phone always rings unless you're actually on |the phone. | | Thanks for the reply though! | | Regards, | Jan | | | | Från: [EMAIL PROTECTED] | [mailto:[EMAIL PROTECTED] För Jerry Jones | Skickat: den 4 maj 2006 15:00 | Till: Asterisk Users Mailing List - Non-Commercial Discussion | Ämne: Re: [Asterisk-Users] Polycom 501 - Disable DND feature? | | | Attribute Values Default Interpretation call.rejectBusyOnDnd |0, 1 1 If | set to 1, reject all incoming calls with the reason busy if | do-not-disturb is enabled. | | Have not used, but looks like it may ignore the key if this is 0 | | Let us know... | | | On May 4, 2006, at 2:22 AM, [EMAIL PROTECTED] | [EMAIL PROTECTED] | wrote: | | | Hi, | | Is it possible to disable the DND feature on a Polycom 501? | | Regards, | Jan | ___ | --Bandwidth and Colocation provided by Easynews.com -- | | Asterisk-Users mailing list | To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | | ___ | --Bandwidth and Colocation provided by Easynews.com -- | | Asterisk-Users mailing list | To UNSUBSCRIBE or update options visit: |http://lists.digium.com/mailman/listinfo/asterisk-users | | |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] I am looking for a webphone on MY SITE
Anyway to check out Corraleta? :) |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Dean Collins |Sent: Wednesday, April 26, 2006 1:12 PM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: RE: [Asterisk-Users] I am looking for a webphone on MY SITE | |Ahh it's about time you released to market anyway. | | | | -Original Message- | From: [EMAIL PROTECTED] |[mailto:asterisk-users- | [EMAIL PROTECTED] On Behalf Of Tim Panton | Sent: Wednesday, 26 April 2006 1:05 PM | To: Asterisk Users Mailing List - Non-Commercial Discussion | Subject: Re: [Asterisk-Users] I am looking for a webphone on MY SITE | | | On 26 Apr 2006, at 16:24, Dean Collins wrote: | | Do a google on Mexuar. | | That's my cover blown :-) | | Tim Panton | [EMAIL PROTECTED] | | | | ___ | --Bandwidth and Colocation provided by Easynews.com -- | | Asterisk-Users mailing list | To UNSUBSCRIBE or update options visit: |http://lists.digium.com/mailman/listinfo/asterisk-users | | |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Background asynchronous AGI
Deadagi? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Tony Mountifield |Sent: Tuesday, April 25, 2006 5:17 AM |To: asterisk-users@lists.digium.com |Subject: [Asterisk-Users] Background asynchronous AGI | |I have been writing a lot of AGI programs in C with good success. |I would like somehow to have an AGI program continue in the |background while the pbx execution returns to the dialplan and |continues. Is this possible? I was thinking that perhaps I |could fork or create another thread within the AGI prog. | |The reason I want to do so is in order to monitor external |information (e.g. credit limit and realtime cost of the |current call) and then perhaps hang up the call, transfer it |or play an announcement to it. | |I'm aware I could do this with a separate control program |using the Manager API, but I like the idea of it being done |per-call on demand using AGI if possible. | |Can anyone suggest any ideas or better techniques? | |Thanks in advance! |Tony |-- |Tony Mountifield |Work: [EMAIL PROTECTED] - http://www.softins.co.uk |Play: [EMAIL PROTECTED] - http://tony.mountifield.org |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Unicall MFRC2 Problems with BrT.
Ah! Ok, that's the file I have, so its ok :) Thx Moises |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Moises Silva |Sent: Monday, April 24, 2006 9:12 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Unicall MFRC2 Problems with BrT. | |sorry for my english, I did not explain myself correctly. I |mean I downloaded the file Today, never meant to say that |the file was uploaded Today. I know the file is recent |enough because i looked for a change in mfcr2.c source that I |know was put there recently. | |Regards | |On 4/22/06, Anton Krall [EMAIL PROTECTED] wrote: | Are you sure its from today? | | The file has dates | | libmfcr2-0.0.3.tar.gz 30-Mar-2006 09:06 346K | | Also inside th tar the changelog has nothing inside and the |news file | has nothing too. | | How did you see it was from today? | | | |-Original Message- | |From: [EMAIL PROTECTED] | |[mailto:[EMAIL PROTECTED] On Behalf |Of Moises | |Silva | |Sent: Saturday, April 22, 2006 9:21 AM | |To: Asterisk Users Mailing List - Non-Commercial Discussion | |Subject: Re: [Asterisk-Users] Unicall MFRC2 Problems with BrT. | | | |hum, the last time i downloaded something every file has different | |dates. However, im looking at a new version that i have downloaded | |today: | | | |http://www.soft-switch.org/downloads/unicall/unicall-0.0.3pre9/ | libmfcr2-0.0.3.tar.gz | | | |And checking the source it seems that tar is the most |recent version. | |I check the version looking in the C code for a fix i know must be | |there, in mfcr2.c line 2780, after the generation tone it |must OR the | |signal with 0x80. | | | |Let me tell you that I have not tested that version. I have |a custom | |version that i fixed (because it gave me the same error you |have) and | |I sent the fix to Steve Underwood, but he told me that my |fix was not | |error proof, and that may fail (I have | |1 month now in a production server with no problems tough), so he | |made a similar fix, and told me that was more reliable. The link I | |just gave you is for the TAR with Steve Underwood fix. | | | |I guess you already contacted me off-list to quote you for my | |consultory. If you still have problems let me know and i |may be able | |to help you through SSH. | | | |Best Regards | | | |On 4/21/06, Anton Krall [EMAIL PROTECTED] wrote: | | Moises, how can I find out which version Im running, on | |Steves ftp all | | say | | 0.0.3 or the date also says the same date. | | | | | | |-Original Message- | | |From: [EMAIL PROTECTED] | | |[mailto:[EMAIL PROTECTED] On Behalf | |Of Moises | | |Silva | | |Sent: Friday, April 21, 2006 9:43 AM | | |To: Asterisk Users Mailing List - Non-Commercial Discussion | | |Subject: Re: [Asterisk-Users] Unicall MFRC2 Problems with BrT. | | | | | |A couple of weeks ago, libmfcr2 has a small error in the tone | | |signaling for the call setup, that was fixed 2 weeks ago or so, | | |please, wich version of libmfcr2 are you using? if you dont | |know try | | |upgrading to the latest version. Im pretty much sure |that you have | | |the very same problem we had. | | | | | |Regards | | | | | |On 4/21/06, Jefferson Carvalho |[EMAIL PROTECTED] wrote: | | | Hello All, | | | | | | I'm facing problems with Unicall on this scenario : | | | | | | CentOS 4.3 - Running on x86_64 | | | Asterisk 1.2.7.1 | | | Zaptel 1.2.5 | | | | | | When running zttool , shows all Spans OK. | | | | | | But I can't receive and make calls. | | | | | | I tried to change many parameters and still doesn't work. | | | | | | Any clues ? | | | | | | * unicall.conf | | | | | | [channels] | | | | | | language=br | | | | | | context=incoming-pstn | | | usecallerid=yes | | | hidecallerid=no | | | immediate=no | | | callwaitingcallerid=yes | | | threewaycalling=yes | | | transfer=yes | | | cancellforward=yes | | | callreturn=yes | | | echocancel=yes | | | echocancelwhenbridged=yes | | | | | | rxgain=0.0 | | | txgain=0.0 | | | faxdetect=both | | | loglevel=255 | | | protocolclass=mfcr2 | | | protocolvariant=br,20,4 | | | protocolend=cpe | | | group=1 | | | callgroup=1 | | | | | | channel = 1-15 | | | channel = 17-31 | | | channel = 32-46 | | | channel = 48-62 | | | channel = 63-77 | | | channel = 94-108 | | | channel = 110-124 | | | | | | * zaptel.conf * | | | | | | loadzone=br | | | defaultzone=br | | | | | | | | | span=1,1,0,cas,hdb3 | | | cas=1-15:1101 | | | cas=17-31:1101 | | | | | | span=2,0,0,cas,hdb3 | | | cas=32-46:1101 | | | cas=48-62:1101 | | | | | | | | | span=3,0,0,cas,hdb3 | | | cas=63-77:1101 | | | cas=79-93:1101 | | | | | | span=4,0,0,cas,hdb3 | | | cas=94-108:1101 | | | cas=110-124:1101 | | | | | | | | | | | | * lor error * | | | | | | -- Executing Dial(SIP/1000-1de2, | | | Unicall/g1/40020022|40|Ttr) in new stack Apr 20 19:13:57 | | | WARNING[30676]: chan_unicall.c:627 | | | unicall_report: MFC/R2 | | | UniCall/1 Call control(1) | | | Apr 20 19:13
RE: [Asterisk-Users] Unicall MFRC2 Problems with BrT.
Are you sure its from today? The file has dates libmfcr2-0.0.3.tar.gz 30-Mar-2006 09:06 346K Also inside th tar the changelog has nothing inside and the news file has nothing too. How did you see it was from today? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Moises Silva |Sent: Saturday, April 22, 2006 9:21 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Unicall MFRC2 Problems with BrT. | |hum, the last time i downloaded something every file has |different dates. However, im looking at a new version that i |have downloaded |today: | |http://www.soft-switch.org/downloads/unicall/unicall-0.0.3pre9/ libmfcr2-0.0.3.tar.gz | |And checking the source it seems that tar is the most recent version. |I check the version looking in the C code for a fix i know |must be there, in mfcr2.c line 2780, after the generation tone |it must OR the signal with 0x80. | |Let me tell you that I have not tested that version. I have a |custom version that i fixed (because it gave me the same error |you have) and I sent the fix to Steve Underwood, but he told |me that my fix was not error proof, and that may fail (I have |1 month now in a production server with no problems tough), so |he made a similar fix, and told me that was more reliable. The |link I just gave you is for the TAR with Steve Underwood fix. | |I guess you already contacted me off-list to quote you for my |consultory. If you still have problems let me know and i may |be able to help you through SSH. | |Best Regards | |On 4/21/06, Anton Krall [EMAIL PROTECTED] wrote: | Moises, how can I find out which version Im running, on |Steves ftp all | say | 0.0.3 or the date also says the same date. | | | |-Original Message- | |From: [EMAIL PROTECTED] | |[mailto:[EMAIL PROTECTED] On Behalf |Of Moises | |Silva | |Sent: Friday, April 21, 2006 9:43 AM | |To: Asterisk Users Mailing List - Non-Commercial Discussion | |Subject: Re: [Asterisk-Users] Unicall MFRC2 Problems with BrT. | | | |A couple of weeks ago, libmfcr2 has a small error in the tone | |signaling for the call setup, that was fixed 2 weeks ago or so, | |please, wich version of libmfcr2 are you using? if you dont |know try | |upgrading to the latest version. Im pretty much sure that you have | |the very same problem we had. | | | |Regards | | | |On 4/21/06, Jefferson Carvalho [EMAIL PROTECTED] wrote: | | Hello All, | | | | I'm facing problems with Unicall on this scenario : | | | | CentOS 4.3 - Running on x86_64 | | Asterisk 1.2.7.1 | | Zaptel 1.2.5 | | | | When running zttool , shows all Spans OK. | | | | But I can't receive and make calls. | | | | I tried to change many parameters and still doesn't work. | | | | Any clues ? | | | | * unicall.conf | | | | [channels] | | | | language=br | | | | context=incoming-pstn | | usecallerid=yes | | hidecallerid=no | | immediate=no | | callwaitingcallerid=yes | | threewaycalling=yes | | transfer=yes | | cancellforward=yes | | callreturn=yes | | echocancel=yes | | echocancelwhenbridged=yes | | | | rxgain=0.0 | | txgain=0.0 | | faxdetect=both | | loglevel=255 | | protocolclass=mfcr2 | | protocolvariant=br,20,4 | | protocolend=cpe | | group=1 | | callgroup=1 | | | | channel = 1-15 | | channel = 17-31 | | channel = 32-46 | | channel = 48-62 | | channel = 63-77 | | channel = 94-108 | | channel = 110-124 | | | | * zaptel.conf * | | | | loadzone=br | | defaultzone=br | | | | | | span=1,1,0,cas,hdb3 | | cas=1-15:1101 | | cas=17-31:1101 | | | | span=2,0,0,cas,hdb3 | | cas=32-46:1101 | | cas=48-62:1101 | | | | | | span=3,0,0,cas,hdb3 | | cas=63-77:1101 | | cas=79-93:1101 | | | | span=4,0,0,cas,hdb3 | | cas=94-108:1101 | | cas=110-124:1101 | | | | | | | | * lor error * | | | | -- Executing Dial(SIP/1000-1de2, | | Unicall/g1/40020022|40|Ttr) in new stack Apr 20 19:13:57 | | WARNING[30676]: chan_unicall.c:627 | | unicall_report: MFC/R2 | | UniCall/1 Call control(1) | | Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:627 unicall_report: | | MFC/R2 | | UniCall/1 Make call | | Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:627 unicall_report: | | MFC/R2 | | UniCall/1 Making a new call with CRN 32769 Apr 20 19:13:57 | | WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2 | | UniCall/1 0001 - [1/ 1/Idle /Idle ] | | -- Called g1/40020022 | | Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:2644 |handle_uc_event: | | Unicall/1 event Dialing | | Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:627 | |unicall_report: MFC/R2 | | UniCall/1 - [1/ 40/Seize /Idle ] | | Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:627 | |unicall_report: MFC/R2 | | UniCall/1 4 on - [2/ 40/Group I /Idle ] | | Apr 20 19:14:02 WARNING[30676]: chan_unicall.c:627 | |unicall_report: MFC/R2 | | UniCall/1 R2 prot. err. [2/ 40/Group I /DNIS | | ] cause | | 32769 - T1 timed out
RE: [Asterisk-Users] Unicall MFRC2 Problems with BrT.
Moises, how can I find out which version Im running, on Steves ftp all say 0.0.3 or the date also says the same date. |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Moises Silva |Sent: Friday, April 21, 2006 9:43 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Unicall MFRC2 Problems with BrT. | |A couple of weeks ago, libmfcr2 has a small error in the tone |signaling for the call setup, that was fixed 2 weeks ago or |so, please, wich version of libmfcr2 are you using? if you |dont know try upgrading to the latest version. Im pretty much |sure that you have the very same problem we had. | |Regards | |On 4/21/06, Jefferson Carvalho [EMAIL PROTECTED] wrote: | Hello All, | | I'm facing problems with Unicall on this scenario : | | CentOS 4.3 - Running on x86_64 | Asterisk 1.2.7.1 | Zaptel 1.2.5 | | When running zttool , shows all Spans OK. | | But I can't receive and make calls. | | I tried to change many parameters and still doesn't work. | | Any clues ? | | * unicall.conf | | [channels] | | language=br | | context=incoming-pstn | usecallerid=yes | hidecallerid=no | immediate=no | callwaitingcallerid=yes | threewaycalling=yes | transfer=yes | cancellforward=yes | callreturn=yes | echocancel=yes | echocancelwhenbridged=yes | | rxgain=0.0 | txgain=0.0 | faxdetect=both | loglevel=255 | protocolclass=mfcr2 | protocolvariant=br,20,4 | protocolend=cpe | group=1 | callgroup=1 | | channel = 1-15 | channel = 17-31 | channel = 32-46 | channel = 48-62 | channel = 63-77 | channel = 94-108 | channel = 110-124 | | * zaptel.conf * | | loadzone=br | defaultzone=br | | | span=1,1,0,cas,hdb3 | cas=1-15:1101 | cas=17-31:1101 | | span=2,0,0,cas,hdb3 | cas=32-46:1101 | cas=48-62:1101 | | | span=3,0,0,cas,hdb3 | cas=63-77:1101 | cas=79-93:1101 | | span=4,0,0,cas,hdb3 | cas=94-108:1101 | cas=110-124:1101 | | | | * lor error * | | -- Executing Dial(SIP/1000-1de2, Unicall/g1/40020022|40|Ttr) | in new stack Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:627 | unicall_report: MFC/R2 | UniCall/1 Call control(1) | Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:627 unicall_report: | MFC/R2 | UniCall/1 Make call | Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:627 unicall_report: | MFC/R2 | UniCall/1 Making a new call with CRN 32769 Apr 20 19:13:57 | WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2 | UniCall/1 0001 - [1/ 1/Idle /Idle ] | -- Called g1/40020022 | Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:2644 handle_uc_event: | Unicall/1 event Dialing | Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:627 |unicall_report: MFC/R2 | UniCall/1 - [1/ 40/Seize /Idle ] | Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:627 |unicall_report: MFC/R2 | UniCall/1 4 on - [2/ 40/Group I /Idle ] | Apr 20 19:14:02 WARNING[30676]: chan_unicall.c:627 |unicall_report: MFC/R2 | UniCall/1 R2 prot. err. [2/ 40/Group I /DNIS | ] cause | 32769 - T1 timed out | Apr 20 19:14:02 WARNING[30676]: chan_unicall.c:627 |unicall_report: MFC/R2 | UniCall/1 4 off - [1/ 1/Idle /Idle ] | Apr 20 19:14:02 WARNING[30676]: chan_unicall.c:627 |unicall_report: MFC/R2 | UniCall/1 1001 - [1/ 1/Idle /Idle ] | Apr 20 19:14:02 WARNING[30676]: chan_unicall.c:2644 handle_uc_event: | Unicall/1 event Protocol failure | -- Unicall/1 protocol error. Cause 32769 Apr 20 19:14:02 | WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2 | UniCall/1 Channel echo cancel | Apr 20 19:14:03 WARNING[30676]: chan_unicall.c:627 unicall_report: | MFC/R2 | UniCall/1 Channel gains | Apr 20 19:14:03 WARNING[30676]: chan_unicall.c:627 unicall_report: | MFC/R2 | UniCall/1 Channel switching | -- Hungup 'UniCall/1-1' | == Everyone is busy/congested at this time (1:0/0/1) | == Auto fallthrough, channel 'SIP/1000-1de2' status is |'CHANUNAVAIL' | Apr 20 19:14:03 WARNING[30664]: chan_unicall.c:627 |unicall_report: MFC/R2 | UniCall/1 - 1011 [1/ 1/Idle /Idle ] | Apr 20 19:14:03 WARNING[30664]: chan_unicall.c:627 |unicall_report: MFC/R2 | UniCall/1 1001 - [1/ 1/Idle /Idle ] | | Jefferson Carvalho | | | | | ___ | --Bandwidth and Colocation provided by Easynews.com -- | | Asterisk-Users mailing list | To UNSUBSCRIBE or update options visit: |http://lists.digium.com/mailman/listinfo/asterisk-users | | | |-- |Su nombre es GNU/Linux, no solamente Linux, mas info en |http://www.gnu.org; |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ --Bandwidth and Colocation provided by
RE: [Asterisk-Users] Digium cards, so disappointing !
Can you send the output of zttest ? Whats your average and what kind of hardware are you using? That will give people pointers of what to use/expect. |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Boris Bakchiev |Sent: Thursday, April 20, 2006 8:52 PM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: RE: [Asterisk-Users] Digium cards, so disappointing ! | |Our production asterisk server has TE411P and we route close |to 50-70K of calls per month through its ports. |We have NEVER EVER had any issues with faxing (close to |3k/month) with faxes connected on one of the spans of the card. | |Moreover, we have had quite a success receiving the faxes with |iaxmodem+hylafax thanks to Lee Howard that we're now gradually |switching |the fax machines to iaxmodem+hylafax combo. | |Faxes are sensitive to timing and configuration settings of |your asterisk. |Once your system is tuned to perfection you should have no |problems faxing at all despite the official stance from Digium. | | | issues). Then we switched to a TE411P for the hardware echo | cancellation. Now we want to receive fax ( 20/day) on it and guess | what ? Since April 2006 (again a few months after we bought |our brand | new card), officially, fax communications is not supported with | Digium cards ( |http://www.voip-info.org/wiki-Asterisk+fax |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Faxing and PCI (was Re: [Asterisk-Users] Digium cards,
Hi Steve! I tried the gain tweaking on the E1 just to check but can you tell me if that thing I was talking about dropping calls when gains are +2 -2 for example is the right behavior? I think my main problem is latency on the te10p cards. I have a te110p card and 2 tdm04b and te110p's is missing irqs (frame slips) according to zttest, zttolol and when trying to check the interrupts, I see Im getting 997 or 990 every 10 secs. Also, zttest shows an average of 99.90 not the needed 99.98 or better. |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Steve Underwood |Sent: Tuesday, April 18, 2006 8:09 PM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: Faxing and PCI (was Re: [Asterisk-Users] Digium cards, | |Hi Anton, | |If you need to tweak gains for FAXing, something is badly |wrong. The modems operate over a wide range of signal levels. | |Steve | |Anton Krall wrote: | |Do you know if you can tweak gains if using unicall? I tried it once |and if you move the gains on zaptel using a te110p with |unicall on E1, |when gains are +2 or -1, calls do not complete, forget even about |faxing :) | | | ||-Original Message- ||From: [EMAIL PROTECTED] ||[mailto:[EMAIL PROTECTED] On Behalf Of Doug ||Lytle ||Sent: Tuesday, April 18, 2006 10:18 AM ||To: Asterisk Users Mailing List - Non-Commercial Discussion ||Subject: Re: Faxing and PCI (was Re: [Asterisk-Users] Digium cards, || ||Steve Underwood wrote: || Doug Lytle wrote: || || Nicholas Kathmann wrote: || || || If you need to tweak gains something is seriously wrong. || ||The 2 fax machines that I was having problem with were failing to ||train at 9600bps, they would then try at 7200 and finally train at ||4800. ||Around 15 pages into the fax they would fail with a, Failed |to detect ||high speed-data carrier and disconnect. Increasing the |rxgain to 3.0 ||and they now train at 9600bps and faxes complete. || ||This PRI is connected to our Definity G3. || ||Doug | | | |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Outgoing voice distortion with Unicall
If the voice distortion sounds like clack clack clack las if you had a fan right next to you (remember when you talk directly to a fan in front of you, the other side gets your voice like in intervals), if thats the case, exactly, your frame size should be 20ms, sipura and some other atas come by default with 30 packet sizes, after changing to 20, all worked fine. Hope that helps. |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Stepan Hradsky |Sent: Wednesday, April 19, 2006 1:38 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Outgoing voice distortion with Unicall | |Hi, | |I had similar problem and problem was in SIP ATA device (we |use Sipura 2100). They was set from factory to send 30ms voice |frame, when we change frame to 20ms everything work perfectly. | |Stepan | |Carlos Chavez napsal(a): | I am having a strange problem with [EMAIL PROTECTED] 2.7 (Asterisk |1.2.5) with a | TE210P card and Unicall. I have compiled everything and |Unicall seems | to be working well. The only problem we are having is that the | outgoing voice is a bit distorted. When someone from the |inside calls | (all phones are connected to ATA using SIP) they can hear the other | person perfectly, but the remote hears them distorted. I |have checked | and rechecked the configurations and everything seems fine. |I really | do not know if this is a software problem, a hardware problem or a | line quality problem. Anyone has any idea how to pinpoint |the source of the problem? | | | |-- | -- | | ___ | --Bandwidth and Colocation provided by Easynews.com -- | | Asterisk-Users mailing list | To UNSUBSCRIBE or update options visit: |http://lists.digium.com/mailman/listinfo/asterisk-users | | |-- |Oddeleni spravy site a pece o zakazniky |ha-vel internet s.r.o. |internet x voice x family x cafe . rEvoluce | |Svabinskeho 9 |702 00 Moravská Ostrava | |tel./fax: +420 552 305 306 | |email: [EMAIL PROTECTED] |www: http://www.ha-vel.cz | |Oddeleni pece o zakazniky: +420 552 305 345 Dohledove centrum: |+420 552 305 321 | |Neodstranujte prosim zadnou cast tohoto e-mailu pri pripadne |dalsi komunikaci k tomuto tematu. |Please do not remove any parts of this e-mail message in |further communication about this issue. | |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] polycom blind transfer button
Guys, this is a weird question but has anybody disabled the blind button that appears on polycoms or know if you can disable the use of blind transfers on polycoms to make any transfer attended? Thx! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Faxing and PCI (was Re: [Asterisk-Users] Digium cards,
Do you know if you can tweak gains if using unicall? I tried it once and if you move the gains on zaptel using a te110p with unicall on E1, when gains are +2 or -1, calls do not complete, forget even about faxing :) |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Doug Lytle |Sent: Tuesday, April 18, 2006 10:18 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: Faxing and PCI (was Re: [Asterisk-Users] Digium cards, | |Steve Underwood wrote: | Doug Lytle wrote: | | Nicholas Kathmann wrote: | | | If you need to tweak gains something is seriously wrong. | |The 2 fax machines that I was having problem with were failing |to train at 9600bps, they would then try at 7200 and finally |train at 4800. |Around 15 pages into the fax they would fail with a, Failed |to detect high speed-data carrier and disconnect. Increasing |the rxgain to 3.0 and they now train at 9600bps and faxes complete. | |This PRI is connected to our Definity G3. | |Doug | |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Performance: Xeon or Opteron?
That's great news! Seen sangoma is beating digium :) |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of yusuf |Sent: Tuesday, April 18, 2006 10:57 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Performance: Xeon or Opteron? | |I have used many sangoma cards, and have not had *any* irq issues | |Anton Krall wrote: | Has anybody used the sangoma fxo cards with asterisk? Anybody using | multiple cards? Problems with irq and such (same as with |digium ones)? | | | | |-Original Message- | |From: [EMAIL PROTECTED] | |[mailto:[EMAIL PROTECTED] On Behalf Of John | |Novack | |Sent: Wednesday, April 12, 2006 10:29 AM | |To: [EMAIL PROTECTED] | |Cc: Asterisk Users Mailing List - Non-Commercial Discussion | |Subject: Re: [Asterisk-Users] Performance: Xeon or Opteron? | | | | | | | |Rich Adamson wrote: | | | | | | While talking with one of the sangoma folks very recently, he was | | rather emphatic the pci bus was designed to share | |interrupts. I was | | a little concerned as a test server had the wanpipe driver | |sharing an | | interrupt with libata and uhc1_hcd. His comment was |that's the way | | its suppose to work, sharing interrupts as needed. I've not had | | any recognizable issues with the A200D card at all, and faxing | |via a A200D | | fxs port to a A200D fxo (pstn) port functions 100% reliably. | | | | What that would suggest is the TDM400 pci firmware |(whether on card | | logic or whatever) is the source of at least part of the | |TDM400 shared | | interrupt issue. I don't have any digium T1/E1 cards |laying around, | | but if memory serves correctly, the T1/E1 cards do not |use the same | | pci controller chip. That would suggest the T1/E1 cards are | |less of an | | issue then with the TDM400 card. | | | |That's good to know, but considering the response from |Digium on the | |TDM400 ( try another motherboard) when there didn't seem to even be | |an int. sharing issue, the card just couldn't be seen at all , and | |the support I received from Sangoma on a recent FXS issue that was | |resolved within a few days, I would tend to go with Sangoma for the | |T1 card, if and when I have the need. | | | |John Novack | | | |___ | |--Bandwidth and Colocation provided by Easynews.com -- | | | |Asterisk-Users mailing list | |To UNSUBSCRIBE or update options visit: | | http://lists.digium.com/mailman/listinfo/asterisk-users | | | | | | ___ | --Bandwidth and Colocation provided by Easynews.com -- | | Asterisk-Users mailing list | To UNSUBSCRIBE or update options visit: |http://lists.digium.com/mailman/listinfo/asterisk-users | | | |-- |thanks, |yusuf |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] polycom blind transfer button
I cant seem to find such an option on the xml config files so far :( |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Jonathan k. Creasy |Sent: Tuesday, April 18, 2006 3:56 PM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: RE: [Asterisk-Users] polycom blind transfer button | |I could be wrong but off the top of my head I think that it is |in the features section of the config file. | |-Jonathan | |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Anton Krall |Sent: Tuesday, April 18, 2006 4:47 PM |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: [Asterisk-Users] polycom blind transfer button | |Guys, this is a weird question but has anybody disabled the |blind button that appears on polycoms or know if you can |disable the use of blind transfers on polycoms to make any |transfer attended? | |Thx! | |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] te110p and interrupts
I don't know if this only works with multiple cpus but I have HT enabled and it shows cpu0 and cpu1 .. I tried the first part of this email and still the kernel boots and shows 2 cpus.. Will this only work with 2 real cpus? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Begumisa Gerald M |Sent: Monday, April 17, 2006 5:13 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] te110p and interrupts | |Hi Paul, | |Thanks for the message! | | On Sun, 16 Apr 2006, Paul Hewlett wrote: | [...] | I am curious.. | | Have you tried disabling CPU1 by setting isolcpus=1 on |the kernel | command line ? | | This will make the kernel ignore the second CPU - you |can then run | asterisk on it by using the taskset command (from schedutils) | | taskset 0x0001 asterisk -p | | and asterisk wlll run on a CPU all on its own. I was about to try | this and wondered if you might give it a try and report back. | |I haven't done this yet. Once we have physical access to the |machine, I'll make sure we try this out and see what |difference it makes. | | |Cheers! |Gerald. |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] te110p and interrupts
What you say makes a lot of sense... Well.. I disabled HT and Im still having irqmisses :( I stil have to try the latency idea.. Hope that helps.. |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Paul Hewlett |Sent: Monday, April 17, 2006 12:19 PM |To: asterisk-users@lists.digium.com |Subject: Re: [Asterisk-Users] te110p and interrupts | |On Monday 17 April 2006 12:39, Anton Krall wrote: | I don't know if this only works with multiple cpus but I have HT | enabled and it shows cpu0 and cpu1 .. I tried the first part of this | email and still the kernel boots and shows 2 cpus.. Will this only | work with 2 real cpus? | | I believe so. Hyperthreading is not really SMP. | | Additionally I think that HT should not be used if asterisk |is doing a lot of transcoding - whilst HT gives u 2 'pipes' |there is still only one FPU and heavy transcoding will simply |bottleneck at the FPU(NB SSE/MMX count as FP since they use |the FP registers.). If u turn HT on, a 1% penalty is incurred |fir the mutexes required so heavy use of the FPU will actually |show a decrease in speed compared to no HT. In theory at least. :-) | |Paul Hewlett | | |-Original Message- | |From: [EMAIL PROTECTED] | |[mailto:[EMAIL PROTECTED] On Behalf Of | |Begumisa Gerald M | |Sent: Monday, April 17, 2006 5:13 AM | |To: Asterisk Users Mailing List - Non-Commercial Discussion | |Subject: Re: [Asterisk-Users] te110p and interrupts | | | |Hi Paul, | | | |Thanks for the message! | | | | On Sun, 16 Apr 2006, Paul Hewlett wrote: | | [...] | | | | I am curious.. | | | | Have you tried disabling CPU1 by setting isolcpus=1 on | | | |the kernel | | | | command line ? | | | | This will make the kernel ignore the second CPU - you | | | |can then run | | | | asterisk on it by using the taskset command (from schedutils) | | | | taskset 0x0001 asterisk -p | | | | and asterisk wlll run on a CPU all on its own. I was |about to try | | this and wondered if you might give it a try and report back. | | | |I haven't done this yet. Once we have physical access to |the machine, | |I'll make sure we try this out and see what difference it makes. | | | | | |Cheers! | |Gerald. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Performance: Xeon or Opteron?
My main concerns would be, can you have multiple cards like this on a system, for example, I now have a te110p and 2 tdm04b and Im getting irqmisses on the te110p (according to zttool and zttest) which makes fax receiving on the te110p almost impossible.. Plus, voice is getting frame slips. I was hoping sangoma cards could be more enterprise friendly. |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |John Novack |Sent: Thursday, April 13, 2006 4:57 PM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Performance: Xeon or Opteron? | |Have a 2FXO 2FXS card working now. |More forgiving of the PCI bus. The software installation is a |little mean, with the outlined procedure making assumptions |about the installers knowledge and familiarity with Sangoma |products, and in some places it doesn't really discriminate |between their T1 and A200 cards. |I found one defect in their FXS driver, which they have now fixed. | |Overall seems to be a good product, slightly more affordable |and less of a problem child than the Digium/TigerJet TDM400 | |John Novack | |Anton Krall wrote: | |Has anybody used the sangoma fxo cards with asterisk? Anybody using |multiple cards? Problems with irq and such (same as with digium ones)? | | | ||-Original Message- ||From: [EMAIL PROTECTED] ||[mailto:[EMAIL PROTECTED] On Behalf Of John ||Novack ||Sent: Wednesday, April 12, 2006 10:29 AM ||To: [EMAIL PROTECTED] ||Cc: Asterisk Users Mailing List - Non-Commercial Discussion ||Subject: Re: [Asterisk-Users] Performance: Xeon or Opteron? || || || ||Rich Adamson wrote: || || || While talking with one of the sangoma folks very recently, he was || rather emphatic the pci bus was designed to share ||interrupts. I was || a little concerned as a test server had the wanpipe driver ||sharing an || interrupt with libata and uhc1_hcd. His comment was |that's the way || its suppose to work, sharing interrupts as needed. I've |not had any || recognizable issues with the A200D card at all, and faxing ||via a A200D || fxs port to a A200D fxo (pstn) port functions 100% reliably. || || What that would suggest is the TDM400 pci firmware |(whether on card || logic or whatever) is the source of at least part of the ||TDM400 shared || interrupt issue. I don't have any digium T1/E1 cards |laying around, || but if memory serves correctly, the T1/E1 cards do not use |the same || pci controller chip. That would suggest the T1/E1 cards are ||less of an || issue then with the TDM400 card. || ||That's good to know, but considering the response from Digium on the ||TDM400 ( try another motherboard) when there didn't seem to |even be an ||int. sharing issue, the card just couldn't be seen at all , and the ||support I received from Sangoma on a recent FXS issue that was ||resolved within a few days, I would tend to go with Sangoma |for the T1 ||card, if and when I have the need. || ||John Novack || ||___ ||--Bandwidth and Colocation provided by Easynews.com -- || ||Asterisk-Users mailing list ||To UNSUBSCRIBE or update options visit: || http://lists.digium.com/mailman/listinfo/asterisk-users || || | |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | | | |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Digium cards, so disappointing !
I must agree with you. I too buy Digium cards because I want to support the development of asterisk. Asterisk is a great product but digum cards are a pain, they say they don't support faxing but a lot of people that are implementing asterisk demand or need faxin as a day to day service on their PBX's. Sad to see that faxing is nearly impossible on digium cards. To me is like saying here you have a great car but.. It cannot handle a car stereo :( |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Tony ROBIN |Sent: Thursday, April 13, 2006 7:01 PM |To: asterisk-users@lists.digium.com |Subject: [Asterisk-Users] Digium cards, so disappointing ! | | |I am so fed up with Digium cards. My company first owned a |TE410P, I installed it in a Dell server and enjoyed its |instability (we bought it months before Digium warned about |the incompatibility issues). Then we switched to a TE411P for |the hardware echo cancellation. Now we want to receive fax ( |20/day) on it and guess what ? Since April 2006 (again a few |months after we bought our brand new card), officially, fax |communications is not supported with Digium cards ( |http://www.voip-info.org/wiki-Asterisk+fax ). |Of course, I should have guessed that it is far too much to |ask to a $2495 card ! Is the fax extension in Asterisk just |there to push us to the competing products ? | |We hesitated to buy another Digium card after the problems |with TE410P, but I told myself it was nice to support Asterisk |by buying some Digium cards. Now Digium make us regret our |buys and a disappointed customer is a lost customer forever... |Too sad... |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Digium cards, so disappointing !
Problem is, how to make sure you system WILL have 100% on zttest before buying the cards.. You need to have stability, compatibility and certainty that what you buy is going to work :( Anybody had similar problems or success stories with sangoma cards? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Lee Howard |Sent: Thursday, April 13, 2006 7:22 PM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Digium cards, so disappointing ! | |Tony ROBIN wrote: | |Now we want to receive fax ( 20/day) on it and guess what ? Since |April 2006 (again a few months after we bought our brand new card), |officially, fax communications is not supported with Digium cards ( |http://www.voip-info.org/wiki-Asterisk+fax ). |Of course, I should have guessed that it is far too much to ask to a |$2495 card ! Is the fax extension in Asterisk just there to push us |to the competing products ? | | |If your zttest has good results (mostly 100%, nothing less |than 99.98%) then you should be able to receive faxes (I'd |suggest iaxmodem+HylaFAX) despite Digium's disclaimer. | |I do not excuse Digium, however, from sidelining fax the way |that they have. | |Lee. | |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Digium cards, so disappointing !
Aaron, have you tried using 1 te110p and 2 tdm04b on the same server? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Aaron Daniel |Sent: Thursday, April 13, 2006 7:19 PM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Digium cards, so disappointing ! | |*shrugs* Ya win some ya lose some. We've spent about 10 grand |plus on Digium cards and have been pretty satisfied with ours |:) Faxes have been working great for over 6 months and the |cards work wonderfully in our Dell servers. They just need |more documentation on the different configuration options you |can pass on load... I think the only problems we've really had |are configuration related, or bad hardware on our part, oh, |and a server room fry that took out more than just the |Asterisk servers :-P | |Aaron | |On Fri, 14 Apr 2006, Tony ROBIN wrote: | | | I am so fed up with Digium cards. My company first owned a TE410P, | I installed it in a Dell server and enjoyed its instability (we | bought it months before Digium warned about the incompatibility | issues). Then we switched to a TE411P for the hardware echo | cancellation. Now we want to receive fax ( 20/day) on it and | guess what ? Since April 2006 (again a few months after we bought | our brand new card), officially, fax communications is not | supported with Digium cards ( |http://www.voip-info.org/wiki-Asterisk+fax ). | Of course, I should have guessed that it is far too much to ask | to a $2495 card ! Is the fax extension in Asterisk just there | to push us to the competing products ? | | We hesitated to buy another Digium card after the problems with | TE410P, but I told myself it was nice to support Asterisk by | buying some Digium cards. Now Digium make us regret our buys and | a disappointed customer is a lost customer forever... Too sad... | ___ | --Bandwidth and Colocation provided by Easynews.com -- | | Asterisk-Users mailing list | To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | |-- |Aaron Daniel |Computer Systems Technician |Sam Houston State University |[EMAIL PROTECTED] |(936) 294-4198 |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Digium cards, so disappointing !
What do you mean Matt? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Matt Riddell (IT) |Sent: Friday, April 14, 2006 7:53 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Digium cards, so disappointing ! | |Anton Krall wrote: | I must agree with you. I too buy Digium cards because I want to | support the development of asterisk. Asterisk is a great product but | digum cards are a pain, they say they don't support faxing but a lot | of people that are implementing asterisk demand or need |faxin as a | day to day service on their PBX's. | | Sad to see that faxing is nearly impossible on digium cards. |To me is | like saying here you have a great car but.. It cannot handle a car | stereo :( | |Is this not possibly also related to the patenting issues on |the email to fax gateways? | |-- |Cheers, | |Matt Riddell |___ | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Performance: Xeon or Opteron?
Well certainly sangoma cards are worth looking into now that they also offer analog port cards, anybody knows if there is a reseller in Mexico? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Rich Adamson |Sent: Friday, April 14, 2006 8:09 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Performance: Xeon or Opteron? | |I'd have to guess that combination of cards with almost any |mobo would be considered an overloaded system. If you replaced |the two TDM04b cards with an A200D or TDM2400 card, most of |those irqmisses (etc) would probably go away; but that's a |somewhat educated guess on my part. | |Factually, the sangoma cards integrate with the pci bus in a |much more stable/usable way then does the digium TDM card (and |I believe the te110 if it uses the TigerJet pci chip). | | |Anton Krall wrote: | My main concerns would be, can you have multiple cards like |this on a | system, for example, I now have a te110p and 2 tdm04b and Im getting | irqmisses on the te110p (according to zttool and zttest) which makes | fax receiving on the te110p almost impossible.. Plus, voice |is getting | frame slips. | | I was hoping sangoma cards could be more enterprise friendly. | | |-Original Message- | |From: [EMAIL PROTECTED] | |[mailto:[EMAIL PROTECTED] On Behalf Of John | |Novack | |Sent: Thursday, April 13, 2006 4:57 PM | |To: Asterisk Users Mailing List - Non-Commercial Discussion | |Subject: Re: [Asterisk-Users] Performance: Xeon or Opteron? | | | |Have a 2FXO 2FXS card working now. | |More forgiving of the PCI bus. The software installation is |a little | |mean, with the outlined procedure making assumptions about the | |installers knowledge and familiarity with Sangoma products, and in | |some places it doesn't really discriminate between their T1 |and A200 | |cards. | |I found one defect in their FXS driver, which they have now fixed. | | | |Overall seems to be a good product, slightly more |affordable and less | |of a problem child than the Digium/TigerJet TDM400 | | | |John Novack | | | |Anton Krall wrote: | | | |Has anybody used the sangoma fxo cards with asterisk? |Anybody using | |multiple cards? Problems with irq and such (same as with |digium ones)? | | | | | | | ||-Original Message- | || | ||Rich Adamson wrote: | || | || | || While talking with one of the sangoma folks very |recently, he was | || rather emphatic the pci bus was designed to share | ||interrupts. I was | || a little concerned as a test server had the wanpipe driver | ||sharing an | || interrupt with libata and uhc1_hcd. His comment was | |that's the way | || its suppose to work, sharing interrupts as needed. I've | |not had any | || recognizable issues with the A200D card at all, and faxing | ||via a A200D | || fxs port to a A200D fxo (pstn) port functions 100% reliably. | || | || What that would suggest is the TDM400 pci firmware | |(whether on card | || logic or whatever) is the source of at least part of the | ||TDM400 shared | || interrupt issue. I don't have any digium T1/E1 cards | |laying around, | || but if memory serves correctly, the T1/E1 cards do not use | |the same | || pci controller chip. That would suggest the T1/E1 cards are | ||less of an | || issue then with the TDM400 card. | || | ||That's good to know, but considering the response from Digium on | ||the TDM400 ( try another motherboard) when there didn't seem to | |even be an | ||int. sharing issue, the card just couldn't be seen at all |, and the | ||support I received from Sangoma on a recent FXS issue that was | ||resolved within a few days, I would tend to go with Sangoma | |for the T1 | ||card, if and when I have the need. | || | ||John Novack | |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Digium cards, so disappointing !
Im my case, faxes are coming in thru an E1 (using unicall) and into iaxmodem/hylafax.. So it should pretty much work.. Except the te110p is missing frames so faxes are getting desynced. |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Rich Adamson |Sent: Friday, April 14, 2006 8:19 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Digium cards, so disappointing ! | |I believe the TDM2400 has the capability of doing on-card |fxo-fxs data flows (without hitting the pci bus), but that |function has not yet been implemented. Its basically |required to support faxes in an analog environment. When it |is implemented, that card should work. The TDM400 card will |not work in 99% of the deployments. | |Faxing via T1 cards is known to work in a fairly large number |of deployments, but its likely to be highly dependent on |exactly where the fax machine is located relative to *. Eg, |incoming pstn fax via a T1 that is expected to be switched to |a sip ata adapter has lots of technical and specific |infrastructure dependencies that have to be addressed by the |implementor / engineer. The plug-n-play approach will have a |very high failure rate. | | |Anton Krall wrote: | I must agree with you. I too buy Digium cards because I want to | support the development of asterisk. Asterisk is a great product but | digum cards are a pain, they say they don't support faxing but a lot | of people that are implementing asterisk demand or need |faxin as a | day to day service on their PBX's. | | Sad to see that faxing is nearly impossible on digium cards. |To me is | like saying here you have a great car but.. It cannot handle a car | stereo :( | | | |-Original Message- | |From: [EMAIL PROTECTED] | |[mailto:[EMAIL PROTECTED] On Behalf Of Tony | |ROBIN | |Sent: Thursday, April 13, 2006 7:01 PM | |To: asterisk-users@lists.digium.com | |Subject: [Asterisk-Users] Digium cards, so disappointing ! | | | | | |I am so fed up with Digium cards. My company first owned a |TE410P, I | |installed it in a Dell server and enjoyed its instability (we | |bought it months before Digium warned about the incompatibility | |issues). Then we switched to a TE411P for the hardware echo | |cancellation. Now we want to receive fax ( | |20/day) on it and guess what ? Since April 2006 (again a few months | |after we bought our brand new card), officially, fax |communications | |is not supported with Digium cards ( | |http://www.voip-info.org/wiki-Asterisk+fax ). | |Of course, I should have guessed that it is far too much to |ask to a | |$2495 card ! Is the fax extension in Asterisk just there |to push us | |to the competing products ? | | | |We hesitated to buy another Digium card after the problems with | |TE410P, but I told myself it was nice to support Asterisk by buying | |some Digium cards. Now Digium make us regret our buys and a | |disappointed customer is a lost customer forever... | |Too sad... | |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Digium cards, so disappointing !
My main concern is that many people want to benefit from integrated faxing like voicemail, for example, sending faxes via pdfto their email. That's one of the pluses of having an asterisk/voip system. |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Rich Adamson |Sent: Friday, April 14, 2006 8:37 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Digium cards, so disappointing ! | |Anton Krall wrote: | Problem is, how to make sure you system WILL have 100% on zttest | before buying the cards.. You need to have stability, compatibility | and certainty that what you buy is going to work :( | | Anybody had similar problems or success stories with sangoma cards? | |Running zttest on my box with both a TDM04b and A200D |installed indicates and average of 99.96% for both. Not sure |how accurate that might be as the A200D card appears as a 24 |channel interface in terms of /dev/zap even though only four |ports are equipped. | |The TDM04b won't support faxes on this box under any |circumstances and I've played around with about every possible |pci latency, etc, change that folks have suggested in the last |two years. | |Based on my heavily invested testing to date (which includes |about two years of doing this), the only usable fax support |thus far comes from using the A200D card with the fax machine |directly connected to a fxs port on that card, and an fxo |(pstn) port on the exact same card. Those fax tests have been |100% solid using a cheap/older Brother fax machine. | | | |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Unicall and Fax
Carlos, have you tested your te110p (or any T1/E1 card) to see if you are missing irq, frame slips? Ive tested a te110p with unicall (depends which version are you using) and I am able to receive a few raxes after a very long time, Im getting frame slips wich makes it very hard but Im trying to figure out the problem with digum and the advice of the guys here on the list. Hope that helps. |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Carlos Chavez |Sent: Friday, April 14, 2006 10:24 AM |To: Asterisk |Subject: [Asterisk-Users] Unicall and Fax | | Has anyone been able to send a fax through a Unicall |channel? I am unable to send or receive faxes using either |rxfax or a fax machine connected to an ATA. Can someone point |me in the right direction? | |-- |Carlos Chavez |Director de Tecnología |Telecomunicaciones Abiertas de México S.A. de C.V. |Tel: +52-55-91169161 Ext 2001 | |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Performance: Xeon or Opteron?
Has anybody used the sangoma fxo cards with asterisk? Anybody using multiple cards? Problems with irq and such (same as with digium ones)? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |John Novack |Sent: Wednesday, April 12, 2006 10:29 AM |To: [EMAIL PROTECTED] |Cc: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Performance: Xeon or Opteron? | | | |Rich Adamson wrote: | | | While talking with one of the sangoma folks very recently, he was | rather emphatic the pci bus was designed to share |interrupts. I was | a little concerned as a test server had the wanpipe driver |sharing an | interrupt with libata and uhc1_hcd. His comment was that's the way | its suppose to work, sharing interrupts as needed. I've not had any | recognizable issues with the A200D card at all, and faxing |via a A200D | fxs port to a A200D fxo (pstn) port functions 100% reliably. | | What that would suggest is the TDM400 pci firmware (whether on card | logic or whatever) is the source of at least part of the |TDM400 shared | interrupt issue. I don't have any digium T1/E1 cards laying around, | but if memory serves correctly, the T1/E1 cards do not use the same | pci controller chip. That would suggest the T1/E1 cards are |less of an | issue then with the TDM400 card. | |That's good to know, but considering the response from Digium |on the TDM400 ( try another motherboard) when there didn't |seem to even be an int. sharing issue, the card just couldn't |be seen at all , and the support I received from Sangoma on a |recent FXS issue that was resolved within a few days, I would |tend to go with Sangoma for the T1 card, if and when I have the need. | |John Novack | |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] te110p and interrupts
Hi Andrew... Thank you very much for the info. I didn't recompile the kernel, Im using a generic 2.6 kernel but its worth taking a look at what you said.. Where can I find (which file) the Hz the kernel was precompiled to? Also, Im running 1 te110p and 2 tdm cards, probably I'll disable 1 card later but I will need at least 1 tdm and the te110p for my E1. So you suggest not disabling any apic/acpi stuff then.. How about HT? should I disable that on the bios? Im using supermicro servers and eventhough I get voice calls without problems, Im getting a few IRQ Misses from time to time which makes faxing on the E1 very difficult (the original problem). What do you think? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Andrew Kohlsmith |Sent: Tuesday, April 11, 2006 7:37 AM |To: asterisk-users@lists.digium.com |Subject: Re: [Asterisk-Users] te110p and interrupts | |On Monday 10 April 2006 19:20, Mojo with Horan Company, LLC wrote: | Try booting with apic off, I think it's noapic kernel |option. Notice | this is APIC and not ACPI, which you referred to. Then get your | boards on different REAL irqs. | |Please do not open your mouth to spout nonsense if you do not |know what you're talking about. | |APIC interrupts are far more real than emulated XT-PIC |interrupts. If the IO-APIC can put each device on its own |high interrupt it means that that INT# signal on the PCI |slot *is* on a totally separate, physical IRQ line which is |routed into that APIC. When you disable native APIC mode you |force it into compatibility mode, where it essentially |performs a logical OR on the real, separate IRQ lines and |gives you a single, edge-triggered i8259-style low IRQ. | |(In reality it's not a logical OR since the XT (i8259-style) |interrupts are edge-triggered, not level triggered, but that's |neither here nor there for this particular discussion.) | |Again, if the IO-APIC is reporting that the card is on its own |IRQ, it really, truly, honestly *IS* on its own IRQ. The |reason that it is suggested to disable the IO-APIC is that on |many low-end systems, the IO-APIC is plain old broken and |causing other issues. I don't think I've run across a system |board in the last year or two with that issue, though. It's |always been on older P3 and early P4 systems. | |Anton, your problem is very likely simple interrupt load. You |have three Digium cards in there, and they're all generating |their own 1000Hz interrupt. |If you did the newbie thing and compiled your kernel with a HZ |value of '1000' because you felt it would be better, you have |that overhead as well. | |Your system is very likely just having trouble coping with so |many interrupts. |My personal opinion is that you should sell the three Digium |cards and buy a single dualspan card and a cheap channel bank. | Your interrupt load will drop by 2/3 and your system will be |FAR happier. | |So check the kernel HZ value first; I set *all* my Asterisk |systems to the old style HZ of 100; there is simply no need |for anything more on a server, |*especially* if you've already got hardware providing a real |1000Hz interrupt instead of ztdummy trying to emulate such a thing. | |-A. |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] te110p and interrupts
Zttool shows no irqmisses on the te110p card? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Kenneth Lussier |Sent: Tuesday, April 11, 2006 7:05 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] te110p and interrupts | |On Mon, 2006-04-10 at 19:25 -0500, Eric ManxPower Wieling wrote: | Anton Krall wrote: | I will try that and see what happens... | This server is a supermicro one.. Anybody else had issues |like this | on supermicro? Any hints on how to resolv them? | | If I remember correctly, supermicro bios does let you |assign irq to | certain pci ports right? Will that help? | | Also, is there a way/is it recommended to increase the latency for | these devices? | | We tried to use a supermicro board once. We could not make it work | with Asterisk and a Digium TE card if there was any usage of the IDE | controller. We made the system a new file server and went with an | Intel branded board. | |I'm running a Supermicro system with a te110p and a tdm40b. I |have HT turned off in the bios and I have acpi=off noapic |added as boot options in grub. The cards work just fine. | |FYI, |Kenny | | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] te110p and interrupts
Guys. I have an issue with a te110p card and also some tdm04b cards on the same system: Zttest returns this for the tdm04b cards: [EMAIL PROTECTED] ~]# /usr/src/zaptel-1.2.4/zttest 38 -v Opened pseudo zap interface, measuring accuracy... 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8223 sample intervals 99.621582% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8400 sample intervals 97.460938% 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8438 sample intervals 96.997070% --- Results after 15 passes --- Best: 100.00 -- Worst: 96.997070 -- Average: 99.601237 And for the te110p: [EMAIL PROTECTED] ~]# /usr/src/zaptel-1.2.4/zttest 1 -v Opened pseudo zap interface, measuring accuracy... 8192 samples in 8239 sample intervals 99.426270% 8192 samples in 8200 sample intervals 99.902344% 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8384 sample intervals 97.656250% 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8193 sample intervals 99.987793% --- Results after 20 passes --- Best: 100.00 -- Worst: 97.656250 -- Average: 99.843140 Im getting problems receiving faxes which can be caused by this. As you can see, the cards are not sharing IRQs as far as I cant tell CPU0 CPU1 0: 17697848 17714488IO-APIC-edge timer 7: 2 2IO-APIC-edge parport0 8: 1 0IO-APIC-edge rtc 9: 0 1 IO-APIC-level acpi 14: 103455 112606IO-APIC-edge ide0 15: 0 0IO-APIC-edge libata 169: 0 0 IO-APIC-level uhci_hcd:usb2 185:2793868 12 IO-APIC-level eth0 201: 0 0 IO-APIC-level ehci_hcd:usb1 209: 0 0 IO-APIC-level uhci_hcd:usb3 217: 19672684 15531101 IO-APIC-level wcte11xp 225: 17824822 17371238 IO-APIC-level wctdm 233: 15323598 19896224 IO-APIC-level wctdm So, can anybody point me in the right direction in order to debug the problem with zttest and quality of sound? How can I change prioritiess for the cards or maybe adjust or see if they get their right number of interrupts, etc? Thx for the help guys! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] queue_log timestamp?
If you do php, try this: ? $format = '%d/%m/%Y %H:%M:%S'; $strf = strftime($format,$argv[1]); echo $strf\n; ? Copy the code into program.php So run it as: php -q progra.php 232313123.2 23232323 been the timestmap you want translated into real date Hope this helps. |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of Johann |Sent: Monday, April 10, 2006 4:53 PM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] queue_log timestamp? | |It's a unixtime stamp. It's the number of seconds since the |epoch(Jan 1, 1970). | | |[EMAIL PROTECTED] wrote: | Hi, | | How do I read (make sense of) the timestamp in the queue_log? I'm | probably just slow but I don't understand it. | | Thanks! | | Regards, | Jan | ___ | --Bandwidth and Colocation provided by Easynews.com -- | | Asterisk-Users mailing list | To UNSUBSCRIBE or update options visit: |http://lists.digium.com/mailman/listinfo/asterisk-users | | |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] te110p and interrupts
This system has acpi enabled. This is how the cards looks like: 217: 22201132 17764597 IO-APIC-level wcte11xp225: 20253291 19707036 IO-APIC-level wctdm233: 17510466 22474179 IO-APIC-level wctdm but if you run lspci .vb to see how the system sees the cards: 02:02.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device 795e:0001 Flags: bus master, medium devsel, latency 32, IRQ 5 I/O ports at 2000 Memory at dd20 (32-bit, non-prefetchable) Capabilities: [40] Power Management version 2 02:08.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b119:0001 Flags: bus master, medium devsel, latency 32, IRQ 5 I/O ports at 2800 Memory at dd201000 (32-bit, non-prefetchable) Capabilities: [40] Power Management version 2 03:02.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b119:0001 Flags: bus master, medium devsel, latency 32, IRQ 5 I/O ports at 3000 Memory at dd40 (32-bit, non-prefetchable) Capabilities: [40] Power Management version 2 I dont get it, the system sees everything on IRQ5 eventhough the kernel sees everything in diff. IRQ: 02:02.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface02:08.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface03:02.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Also, latency for the cards is 32... i enabled the watchdog on zttool and I do see Missed IRQ/interrupts on zttooll for the te110p card... how can I fix this? Ive read some stuff around but so far.. nothing makes sense to me... :( From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Infobox PeruSent: Monday, April 10, 2006 5:47 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] te110p and interrupts use lspci -vb for detecting interrupt conflicts.. On 4/10/06, Anton Krall [EMAIL PROTECTED] wrote: Guys. I have an issue with a te110p card and also some tdm04b cards on the same system:Zttest returns this for the tdm04b cards:[EMAIL PROTECTED] ~]# /usr/src/zaptel-1.2.4/zttest 38 -vOpened pseudo zap interface, measuring accuracy...8192 samples in 8191 sample intervals 99.987793%8192 samples in 8192 sample intervals 100.00%8192 samples in 8192 sample intervals 100.00%8192 samples in 8223 sample intervals 99.621582%8192 samples in 8192 sample intervals 100.00% 8192 samples in 8191 sample intervals 99.987793%8192 samples in 8192 sample intervals 100.00%8192 samples in 8192 sample intervals 100.00%8192 samples in 8191 sample intervals 99.987793%8192 samples in 8400 sample intervals 97.460938%8192 samples in 8191 sample intervals 99.987793%8192 samples in 8192 sample intervals 100.00%8192 samples in 8192 sample intervals 100.00%8192 samples in 8191 sample intervals 99.987793%8192 samples in 8438 sample intervals 96.997070%--- Results after 15 passes ---Best: 100.00 -- Worst: 96.997070 -- Average: 99.601237And for the te110p:[EMAIL PROTECTED] ~]# /usr/src/zaptel-1.2.4/zttest 1 -v Opened pseudo zap interface, measuring accuracy...8192 samples in 8239 sample intervals 99.426270%8192 samples in 8200 sample intervals 99.902344%8192 samples in 8191 sample intervals 99.987793%8192 samples in 8191 sample intervals 99.987793%8192 samples in 8192 sample intervals 100.00%8192 samples in 8191 sample intervals 99.987793%8192 samples in 8384 sample intervals 97.656250%8192 samples in 8191 sample intervals 99.987793%8192 samples in 8191 sample intervals 99.987793%8192 samples in 8191 sample intervals 99.987793%8192 samples in 8191 sample intervals 99.987793%8192 samples in 8192 sample intervals 100.00%8192 samples in 8191 sample intervals 99.987793%8192 samples in 8192 sample intervals 100.00%8192 samples in 8192 sample intervals 100.00%8192 samples in 8192 sample intervals 100.00%8192 samples in 8192 sample intervals 100.00% 8192 samples in 8191 sample intervals 99.987793%8192 samples in 8192 sample intervals 100.00%8192 samples in 8193 sample intervals 99.987793%--- Results after 20 passes ---Best: 100.00 -- Worst: 97.656250 -- Average: 99.843140Im getting problems receiving faxes which can be caused by this.As you can see, the cards are not sharing IRQs as far as I cant tell CPU0 CPU10: 17697848 17714488IO-APIC-edgetimer 7:22IO-APIC-edgeparport08:10IO-APIC-edgertc9:01 IO-APIC-levelacpi14: 103455 112606IO-APIC-edgeide015:00IO-APIC-edgelibata169:00 IO-APIC-leveluhci_hcd:usb2185:2793868 12 IO-APIC-leveleth0201:00 IO-APIC-levelehci_hcd:usb1209:00 IO-APIC-leveluhci_hcd:usb3217
RE: [Asterisk-Users] te110p and interrupts
I will try that and see what happens... This server is a supermicro one.. Anybody else had issues like this on supermicro? Any hints on how to resolv them? If I remember correctly, supermicro bios does let you assign irq to certain pci ports right? Will that help? Also, is there a way/is it recommended to increase the latency for these devices? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Mojo with Horan Company, LLC |Sent: Monday, April 10, 2006 6:20 PM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] te110p and interrupts | |Try booting with apic off, I think it's noapic kernel option. |Notice this is APIC and not ACPI, which you referred to. |Then get your boards on different REAL irqs. | |Moj | |Anton Krall wrote: | This system has acpi enabled. This is how the cards looks like: | | 217: 22201132 17764597 IO-APIC-level wcte11xp | 225: 20253291 19707036 IO-APIC-level wctdm | 233: 17510466 22474179 IO-APIC-level wctdm | | but if you run lspci .vb to see how the system sees the cards: | | 02:02.0 Network controller: Tiger Jet Network Inc. Tiger3XX |Modem/ISDN | interface | Subsystem: Unknown device 795e:0001 | Flags: bus master, medium devsel, latency 32, IRQ 5 | I/O ports at 2000 | Memory at dd20 (32-bit, non-prefetchable) | Capabilities: [40] Power Management version 2 |02:08.0 Network | controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface | Subsystem: Unknown device b119:0001 | Flags: bus master, medium devsel, latency 32, IRQ 5 | I/O ports at 2800 | Memory at dd201000 (32-bit, non-prefetchable) | Capabilities: [40] Power Management version 2 | | 03:02.0 Network controller: Tiger Jet Network Inc. Tiger3XX |Modem/ISDN | interface | Subsystem: Unknown device b119:0001 | Flags: bus master, medium devsel, latency 32, IRQ 5 | I/O ports at 3000 | Memory at dd40 (32-bit, non-prefetchable) | Capabilities: [40] Power Management version 2 | | I dont get it, the system sees everything on IRQ5 eventhough the | kernel sees everything in diff. IRQ: | | 02:02.0 Network controller: Tiger Jet Network Inc. Tiger3XX |Modem/ISDN | interface 02:08.0 Network controller: Tiger Jet Network Inc. |Tiger3XX | Modem/ISDN interface 03:02.0 Network controller: Tiger Jet Network | Inc. Tiger3XX Modem/ISDN interface Also, latency for the cards is | 32... | | i enabled the watchdog on zttool and I do see Missed |IRQ/interrupts on | zttooll for the te110p card... how can I fix this? | | Ive read some stuff around but so far.. nothing makes sense to me... | :( | | | |--- |- | *From:* [EMAIL PROTECTED] | [mailto:[EMAIL PROTECTED] *On Behalf Of | *Infobox Peru | *Sent:* Monday, April 10, 2006 5:47 PM | *To:* Asterisk Users Mailing List - Non-Commercial Discussion | *Subject:* Re: [Asterisk-Users] te110p and interrupts | | use lspci -vb for detecting interrupt conflicts.. | | On 4/10/06, *Anton Krall* [EMAIL PROTECTED] | mailto:[EMAIL PROTECTED] wrote: | | Guys. I have an issue with a te110p card and also some tdm04b | cards on the | same system: | | Zttest returns this for the tdm04b cards: | | [EMAIL PROTECTED] ~]# /usr/src/zaptel-1.2.4/zttest 38 -v | Opened pseudo zap interface, measuring accuracy... | | 8192 samples in 8191 sample intervals 99.987793% | 8192 samples in 8192 sample intervals 100.00% | 8192 samples in 8192 sample intervals 100.00% | 8192 samples in 8223 sample intervals 99.621582% | 8192 samples in 8192 sample intervals 100.00% | 8192 samples in 8191 sample intervals 99.987793% | 8192 samples in 8192 sample intervals 100.00% | 8192 samples in 8192 sample intervals 100.00% | 8192 samples in 8191 sample intervals 99.987793% | 8192 samples in 8400 sample intervals 97.460938% | 8192 samples in 8191 sample intervals 99.987793% | 8192 samples in 8192 sample intervals 100.00% | 8192 samples in 8192 sample intervals 100.00% | 8192 samples in 8191 sample intervals 99.987793% | 8192 samples in 8438 sample intervals 96.997070% | --- Results after 15 passes --- | Best: 100.00 -- Worst: 96.997070 -- Average: 99.601237 | | And for the te110p: | | [EMAIL PROTECTED] ~]# /usr/src/zaptel-1.2.4/zttest 1 -v | Opened pseudo zap interface, measuring accuracy... | | 8192 samples in 8239 sample intervals 99.426270% | 8192 samples in 8200 sample intervals 99.902344% | 8192 samples in 8191 sample intervals 99.987793% | 8192 samples in 8191 sample intervals 99.987793
RE: [Asterisk-Users] te110p and interrupts
This is a Xeon with HT... I remember I disabled HT on supermicros bios but then again, maybe somebody enabled it again.. Thats why its showing 2 cpus? Ill disable it too.. Thx for the hint.. BTW why did you commented the timer line? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Boris Bakchiev |Sent: Monday, April 10, 2006 6:26 PM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: RE: [Asterisk-Users] te110p and interrupts | |Is this dual CPU/Core or just P4 with HT enabled? |If it is P4, I would recommend to disable HT. | |Try changing PCI slots for one of the cards (if you have spare |PCI slots). | | CPU0 CPU1 | 0: 17697848 17714488IO-APIC-edge timer |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] te110p and interrupts
Damn :) any others? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Eric ManxPower Wieling |Sent: Monday, April 10, 2006 7:25 PM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] te110p and interrupts | |Anton Krall wrote: | I will try that and see what happens... | This server is a supermicro one.. Anybody else had issues |like this on | supermicro? Any hints on how to resolv them? | | If I remember correctly, supermicro bios does let you assign irq to | certain pci ports right? Will that help? | | Also, is there a way/is it recommended to increase the latency for | these devices? | |We tried to use a supermicro board once. We could not make it |work with Asterisk and a Digium TE card if there was any usage |of the IDE controller. We made the system a new file server |and went with an Intel branded board. |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] queueue recording and what to do next
Thank you very much for the tip Matt. I was wondering, besides recording the queues, I also use mixmonitor on my dialplans for some extensions, does mixmonitor also use sommix to mix the call legs are is mixmonitor mixing realtime using inernal asterisk functions? I say this because I just to make sure that by replacing sommix I wont brake anything else. |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of Matt Roth |Sent: Thursday, April 06, 2006 12:31 PM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] queueue recording and what to do next | |Anton Krall wrote: | |Guys, if you define recording on queues.conf and also define a |monitor_filename var on your dialplna, you can record a queue |call but, |isthere a way to do something with the file after the call |ends? I need |to move the file to some other place but I cant find where to |define a |command to run after a queue call finishes. | |Any hints? | |Anton, | |In queues.conf set: | |monitor-join=yes | |for all queues that you are recording. This will cause soxmix |to be executed at the end of the call in order to join the leg |files into a single recording. | |Then backup the soxmix binary and replace it with a script |that does whatever you want with the leg files. Your script |will be passed 3 arguments (see show application monitor at |the CLI for more details): | |1) The -in leg filename. |2) The -out leg filename. |3) A target mixed filename. | |Odds are that you'll only care about the first two arguments. |Here is the script that I use to move the leg files from a RAM |disk over an NFS mount to a remote machine which handles |mixing and archiving the recordings: | |[EMAIL PROTECTED] ~]# cat /usr/bin/soxmix |#!/bin/bash | |/bin/nice -n 19 mv --target-directory=/digrec-nfs/ $1 $2 if [ |$? -ne 0 ]; then |echo Failed to mv '$3' /var/log/asterisk/mvdr_log |exit 1 |fi | |exit 0 | |As you can see, I'm using the third argument to log any failed moves. |So far there haven't been any. | |There is another option besides replacing soxmix with a custom |program. |You can use the dialplan variables MONITOR_EXEC and |MONITOR_EXEC_ARGS to tell Monitor() to use another program |to mix the leg files. I have found this method to be |unreliable. Roughly 1% of the time, Monitor() was not calling |the program defined by MONITOR_EXEC. Replacing soxmix has |worked for me 100% of the time (we handle 10,000 - 13,000 |recordings a day), so I recommend it as the preferred solution. | |Matthew Roth |InterMedia Marketing Solutions |Software Engineer and Systems Developer |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Asterisk in production as a fax server, anyone?
Im running hylafax, iaxmodem and spandsp with asterisk on unicall with mfcr2 e1 in Mexico and Im having a very hard time getiing faxes to work.. I hear click on the lines and they are E1 so.. I have no clue whats happening but could it be frame slips? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |MBIT Technologies |Sent: Tuesday, April 04, 2006 6:19 PM |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: RE: [Asterisk-Users] Re: Asterisk in production as a |fax server,anyone? | |I'm not sure if this has been brought up before but I don't |think Asterisk is the package you would want to use if you are |going to implement a fax production server. A more mature |package in this area is Hylafax. I think you will find it will |be far more reliable than Asterisk in the case of a fax |production server. Heres a link | |http://www.hylafax.org | | |Regards | | |Mark Brooker |T: 02 4959 8670 |M: 0415 846 865 |F: 02 9882 0947 |E: [EMAIL PROTECTED] |W: http://www.mbit.com.au | | |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Paulo Scardine |Sent: Wednesday, 5 April 2006 9:09 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Re: Asterisk in production as a |fax server,anyone? | |Don Pobanz escreveu: | | Frame slips are NOT motherboard related! | |I had problems with some combinations of motherboards, memory |sizes and linux kernel versions. | |There are timing problems that also causes frame slips, like |buffer overruns or underruns, but these are software related. | |-- |Paulo | |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] queueue recording and what to do next
Guys, if you define recording on queues.conf and also define a monitor_filename var on your dialplna, you can record a queue call but, isthere a way to do something with the file after the call ends? I need to move the file to some other place but I cant find where to define a command to run after a queue call finishes. Any hints? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Can't recieve Fax: No carrier detected - Asterisk + iaxmodem + Hylafaxv --- sorry.wrong log.
It really makes that much diff. using slinear vs. ulaw? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Colin Anderson |Sent: Tuesday, April 04, 2006 11:26 AM |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: RE: [Asterisk-Users] Can't recieve Fax: No carrier |detected - Asterisk + iaxmodem + Hylafaxv --- sorry.wrong log. | | -- Format for call is ulaw | |Try the slin codec, I didn't have good results until I used slin. |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] queueue recording and what to do next
Thx Michel, I didn't think about that. |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Michiel van Baak |Sent: Tuesday, April 04, 2006 3:42 PM |To: asterisk-users@lists.digium.com |Subject: Re: [Asterisk-Users] queueue recording and what to do next | |On 14:36, Tue 04 Apr 06, Anton Krall wrote: | Guys, if you define recording on queues.conf and also define a | monitor_filename var on your dialplna, you can record a queue call | but, isthere a way to do something with the file after the |call ends? | I need to move the file to some other place but I cant find where to | define a command to run after a queue call finishes. | | Any hints? | |You can use the exten = h,1,deadagi() to process it. |At least that's how we do it with faxes. | |exten = h,1,deadagi(processfax.php) ;put the fax in db and |generate pdf on filesys | |Good luck |-- |Michiel van Baak |[EMAIL PROTECTED] |http://michiel.vanbaak.info |GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D | |Why is it drug addicts and computer afficionados are both |called users? | |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] polycom overlap dialing?
What do you mean by overlap dialing? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of C F |Sent: Monday, April 03, 2006 3:26 PM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] polycom overlap dialing? | |Use the impossible match setting in sip.cfg (or maybe it's in |phone.cfg). In any case polycom documentation covers it all. | |On 4/3/06, Jerry Jones [EMAIL PROTECTED] wrote: | I had emaile poly support a couple months ago and they |replied not yet. | | | On Apr 2, 2006, at 7:14 AM, Noah Miller wrote: | | Is there any way to get a polycom 601 to do overlap dialing? | | I can't find anything on the subject, which confirms my |initial hunch: | I really doubt it. You could probably work something up in | asterisk, though. | | - Noah | ___ | --Bandwidth and Colocation provided by Easynews.com -- | | Asterisk-Users mailing list | To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ | --Bandwidth and Colocation provided by Easynews.com -- | | Asterisk-Users mailing list | To UNSUBSCRIBE or update options visit: |http://lists.digium.com/mailman/listinfo/asterisk-users | |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] applicationmap
Guys. Ive been plyaing around with applicationmap in features.conf Im sing it to playback a sound to the caller but here is a problem, how does the callee know when the sound has finished playing if he cant hear it at all so I was wondering, has anybody played around with this? Is there a setting in applicationmap that has both the caller and callee? Something like mamamia1 = #8,both,Background,/var/lib/asterisk/sounds/acube/mamamia/mamamia ? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Meetme admin
This may sound complicated but how about adding another extenion in extensions.conf where you define meetme with the admin flags and make another extension for normal users without the flags. Plus you can use apps like authenticate or maybe even mysql statements to auth the admin in the extension before making him enter meetme. Just my $.2 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jordan NovakSent: Monday, April 03, 2006 3:42 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Meetme admin I have found meetmeadmin, sounds good and all but does anyone have a code snippet or idea on how to do this. I want to allow one person to be an admin, mute, kick, join other users. How do I differentiate the admin from the rest and then allow him to kick join and so forth without interrupting the conference or have the others hear digits being dialed. Jordan Novak Communications Technician ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] applicationmap
Yep, done all that and it is working, my problem here is that when using an app like playback, sound is just hear as per your definition, either callee or caller, but I need both :) Hope they integrate that soon. |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of C F |Sent: Monday, April 03, 2006 4:07 PM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] applicationmap | |both has been a suggestion on the bug tracker, but I havn't seen it |being implemented. Right now the only way I know of getting |applicationmap to work is like this: |1. Create it in features.conf |2. use Set(DYNAMIC_FEATURES) in the dialplan to make the |applicationmap available for the caller/callee as defined in |features.conf. this has to be done before you use app_dial |3. Make sure asterisk is in the mediapath when using dial, sometimes |this means adding t or T to the dial command. |Please note, there is no security whatsoever to disallow the other |party from activating an application map. | |On 4/3/06, Anton Krall [EMAIL PROTECTED] wrote: | Guys. | | Ive been plyaing around with applicationmap in features.conf | | Im sing it to playback a sound to the caller but here is a |problem, how does | the callee know when the sound has finished playing if he |cant hear it at | all so I was wondering, has anybody played around with this? | | Is there a setting in applicationmap that has both the |caller and callee? | Something like | | mamamia1 = | #8,both,Background,/var/lib/asterisk/sounds/acube/mamamia/mamamia ? | | ___ | --Bandwidth and Colocation provided by Easynews.com -- | | Asterisk-Users mailing list | To UNSUBSCRIBE or update options visit: |http://lists.digium.com/mailman/listinfo/asterisk-users | |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
