Re: [asterisk-users] Re: Zaptel install - Fedora Core 5
Tzafrir Cohen wrote: On Tue, Aug 22, 2006 at 08:42:36AM -0500, Rich Adamson wrote: Tomislav Parčina wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I did yum update last week and here is my current kernel: I had no problem at all with zaptel. I am only using TDM400P though, in case that matters. Hi Anto! The thing is that I can't rely on yum update for asterisk installation. I would like something that will work like this: I install FC5 from CD/DVD, install RPM's that I need from my ftp server or from CD, install zaptel, libpri, asterisk... So, I need to download rpm's that will allow me to install zaptel/libpri/asterisk without using yum update (I need to make all installations the same). Why bother with the rpm's? Because you have some other programs on your system other than Asterisk. And because you want a reproducable build. Guess that depends a lot on personal objectives, styles, and whether asterisk code has been modified locally. Once the reproducable build is operational and one has to maintain the code, reproducable builds sort of go out the window (eg, customer/system A has a problem, but not customer B through Z). Using the Branch SVN checkout approach always provides the most up to date code as opposed to replicating buggy stuff via rpms. ___ --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] Compilation
Which version of Web-MeetMe did you download? The process up to 2.0.1 is, well, annoying. Copy app_cbmysql.c to ./asterisk/apps and modify the Makefile to include the application. The project is now hosted on SourceForge and has a much improved build process, but I have not built a release tarball yet. Dan From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Khaled ChehabSent: Tuesday, August 22, 2006 4:01 AMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Cc: [EMAIL PROTECTED]Subject: [asterisk-users] Compilation Dear I am installing Web-MeetMe ,one of the requirements is app_cbmysql.c I have it but ,how can I compile it . Regards *No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates.This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium.If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person.Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects.* ___ --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] Realtime Extensions -- Comments?
-Original Message- From: Matthew Crocker [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 22, 2006 10:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime Extensions -- Comments? Add a boolean field to the table then create a view based on the value of that field CREATE TABLE `extensions_table_data` ( `id` int(11) NOT NULL auto_increment, 'isActive' boolean NOT NULL default 'True', `context` varchar(20) NOT NULL default '', `exten` varchar(20) NOT NULL default '', `priority` tinyint(4) NOT NULL default '0', `app` varchar(20) NOT NULL default '', `appdata` varchar(128) NOT NULL default '', PRIMARY KEY (`context`,`exten`,`priority`), KEY `id` (`id`) ) TYPE=MyISAM; Uhm... what abouts comments? What if I wanted to temporarily deactivate a couple of extensions? Without a comment flag, I'd have to completely remove those entries from the extensions table! That's not very friendly is it... Is there a better way? Then Create view extensions_table as SELECT id,context,exten,priority,app,appdata from extensions_table_data where IsActive = True; Then you can just flip the IsActive bit on/off to remove extensions That's the workaround I've implemented. ___ --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] Unable to match on CallerID in an include block
What version of asterisk ? Julian Steve Hanselman wrote: Is there any reason why I can’t use the xxx/callerid format in an include section? It doesn’t seem to work, but if I paste the lines into the main section where I include the block it does? E.g. this doesn’t work [telewest] Include = spamblock [spamblock] _X./12345,s,macro(spamcall) Whereas this does: [telewest] _X./12345,s,macro(spamcall) Any ideas? Steve The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendataco.uk ___ --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] Realtime Extensions -- Comments?
note http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions comment from Philipp Dunkel. On 22 Aug 2006, at 17:13, Douglas Garstang wrote: -Original Message- From: Jeremy McNamara [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 22, 2006 9:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime Extensions -- Comments? Douglas Garstang wrote: Uhm... what abouts comments? What if I wanted to temporarily deactivate a couple of extensions? Without a comment flag, I'd have to completely remove those entries from the extensions table! That's not very friendly is it... Is there a better way? Yes, DON'T USE REALTIME! I wish it was that easy. We started looking at realtime again, because the option of building the config files with a script querying the database became daunting. It doesn't matter where you turn in Asterisk, there's gotcha's. For example, you can't put the hint stuff into realtime, and there's no inherint way to comment extensions. 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] Realtime Extensions -- Comments?
He meant he added another column in the database table... In the asterisk source everything database related for the default database stuff is explicitly named... Like INSERT INTO blah(col1,col2,col3) Values(foo,foo,foo) - Original Message - From: Douglas Garstang [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, August 22, 2006 12:14 PM Subject: RE: [asterisk-users] Realtime Extensions -- Comments? -Original Message- From: Jason Parker [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 22, 2006 9:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime Extensions -- Comments? - Douglas Garstang [EMAIL PROTECTED] wrote: The unofficial docs on the voip wiki for the realtime extensions table structure is: CREATE TABLE `extensions_table` ( `id` int(11) NOT NULL auto_increment, `context` varchar(20) NOT NULL default '', `exten` varchar(20) NOT NULL default '', `priority` tinyint(4) NOT NULL default '0', `app` varchar(20) NOT NULL default '', `appdata` varchar(128) NOT NULL default '', PRIMARY KEY (`context`,`exten`,`priority`), KEY `id` (`id`) ) TYPE=MyISAM; Uhm... what abouts comments? What if I wanted to temporarily deactivate a couple of extensions? Without a comment flag, I'd have to completely remove those entries from the extensions table! That's not very friendly is it... Is there a better way? 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 I cheated, and just added a comments field to the table. Asterisk only reads fields by name, so extra columns don't hurt at all. How did an extra field that Asterisk doesn't know anything about, change it's behaviour? 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 No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.405 / Virus Database: 268.11.5/425 - Release Date: 8/22/2006 ___ --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] SLA.conf
Thanks Kevin, This is very exciting news. Let me know what if anything I can do to help test this. I have a Grandstream GXP2000 and Aastra 9133i phone. What sort's of phones and or features are required to support this? The Aastra supports Broadworks SLA and standard SIP BLF but not alternate device state BLF (yet!) such as metermaid call parking. -Original Message- From: Kevin P. Fleming [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 22, 2006 7:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SLA.conf - shadowym [EMAIL PROTECTED] wrote: I found this indication that Shared Line Appearance is possibly in SVN. Is it or is this just an indication that it is up and coming? http://bugs.digium.com/view.php?id=7701 There is an initial implementation of SLA in SVN trunk right now, but it is buggy and needs some redesign. We hope to get that work done before Asterisk 1.4 is released. -- Kevin P. Fleming Senior Software Engineer Digium, 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] Realtime Extensions -- Comments?
On Tue, 2006-08-22 at 10:14 -0600, Douglas Garstang wrote: -Original Message- From: Jason Parker [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 22, 2006 9:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime Extensions -- Comments? - Douglas Garstang [EMAIL PROTECTED] wrote: The unofficial docs on the voip wiki for the realtime extensions table structure is: CREATE TABLE `extensions_table` ( `id` int(11) NOT NULL auto_increment, `context` varchar(20) NOT NULL default '', `exten` varchar(20) NOT NULL default '', `priority` tinyint(4) NOT NULL default '0', `app` varchar(20) NOT NULL default '', `appdata` varchar(128) NOT NULL default '', PRIMARY KEY (`context`,`exten`,`priority`), KEY `id` (`id`) ) TYPE=MyISAM; Uhm... what abouts comments? What if I wanted to temporarily deactivate a couple of extensions? Without a comment flag, I'd have to completely remove those entries from the extensions table! That's not very friendly is it... Is there a better way? 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 I cheated, and just added a comments field to the table. Asterisk only reads fields by name, so extra columns don't hurt at all. How did an extra field that Asterisk doesn't know anything about, change it's behaviour? Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- I think he meant a comments field, to describe the extension. Asterisk doesn't care about extra fields in the db, but it won't use them to it's benefit unless it's programmed in. -- 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
RE: [asterisk-users] Unable to match on CallerID in an include block
Hi Julian, Ah, a very good point, I put that in my first cut but had completely forgotten in this one! 1.2.10 Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: 22 August 2006 17:30 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Unable to match on CallerID in an include block What version of asterisk ? Julian Steve Hanselman wrote: Is there any reason why I can't use the xxx/callerid format in an include section? It doesn't seem to work, but if I paste the lines into the main section where I include the block it does? E.g. this doesn't work [telewest] Include = spamblock [spamblock] _X./12345,s,macro(spamcall) Whereas this does: [telewest] _X./12345,s,macro(spamcall) Any ideas? Steve The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendataco.uk ___ --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 The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.uk ___ --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] Realtime Extensions -- Comments?
- Douglas Garstang [EMAIL PROTECTED] wrote: -Original Message- From: Jason Parker [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 22, 2006 9:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime Extensions -- Comments? - Douglas Garstang [EMAIL PROTECTED] wrote: The unofficial docs on the voip wiki for the realtime extensions table structure is: CREATE TABLE `extensions_table` ( `id` int(11) NOT NULL auto_increment, `context` varchar(20) NOT NULL default '', `exten` varchar(20) NOT NULL default '', `priority` tinyint(4) NOT NULL default '0', `app` varchar(20) NOT NULL default '', `appdata` varchar(128) NOT NULL default '', PRIMARY KEY (`context`,`exten`,`priority`), KEY `id` (`id`) ) TYPE=MyISAM; Uhm... what abouts comments? What if I wanted to temporarily deactivate a couple of extensions? Without a comment flag, I'd have to completely remove those entries from the extensions table! That's not very friendly is it... Is there a better way? 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 I cheated, and just added a comments field to the table. Asterisk only reads fields by name, so extra columns don't hurt at all. How did an extra field that Asterisk doesn't know anything about, change it's behaviour? Doug. I guess I misunderstood what you meant by comments. It appeared to be two unrelated questions. -- Jason Parker Digium ___ --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] Linksys SPA-941 Message Waiting Indicator
On 8/20/06, Rich Adamson [EMAIL PROTECTED] wrote: voiplist wrote: Any ideas? Would be nice to use this feature as we have with other Sipura products in the past. Sounds like a config problem with asterisk. The 941/942's here worked just fine right out of the box. 941 message waiting is working here as well w/ asterisk 1.2.9 w/ no special config. Cliff -- === Cliff Brake http://bec-systems.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
RE: [asterisk-users] Apache for FastAGI
Personally I will build multi-thread server to work as AGI server. Apache is an option if you know how to implement programs on web server, but know how to build your server. You can hardcode the HTTP header into a global variable in your dialplan, and send the header to Apache first, then follow your dynamic agi request. [EMAIL PROTECTED] 08/22/06 7:59 AM I'm not sure how one would build a HTTP header on the client side, given that all you have to work with is a single line entry in extensions.conf. -Original Message- From: Tielin Xu [mailto:[EMAIL PROTECTED] Sent: Friday, August 18, 2006 12:41 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Apache for FastAGI It is an valid option, but you have to build a HTTP header in your request to your web server, which CGI programs or Java servlets on web server could interpret your request from Asterisk. Tielin [EMAIL PROTECTED] 08/18/06 11:28 AM Here's an idea... Rather than writing your own multi-thread socket server for use with FastAGI, has anyone tried to use an Apache web server instead? After all, it does all that for you. I just gave it a shot, but Asterisk tries to send all the agi params to the web server, which it doesn't like it... [Fri Aug 18 12:25:28 2006] [error] [client xxx.yyy.141.162] Invalid URI in request agi_network: yes 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 ___ --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 and spandsp
Hi, I'm trying to get my asterisk recieve faxes via rxfax and spandsp. It creates the connection, recieves some info about the sender and dies. Result is not valid tif file of size 330 or 334 B. Debug messages are shown only sometimes and it ends with something like FLOW Fast carrier training failed FLOW Fast carrier down. Libtiff 3.7.1, asterisk 1.2.9.1, spandsp 0.0.2pre21 installed manually on gentoo. Has anyone made asterisk to recieve faxes this way? (or any other, I'm slowly running out of ideas) I'd be grateful for any reply Jan Fousek ___ --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 and spandsp
Hi, I'm trying to get my asterisk recieve faxes via rxfax and spandsp. It creates the connection, recieves some info about the sender and dies. Result is not valid tif file of size 330 or 334 B. Debug messages are shown only sometimes and it ends with something like FLOW Fast carrier training failed FLOW Fast carrier down. Libtiff 3.7.1, asterisk 1.2.9.1, spandsp 0.0.2pre21 installed manually on gentoo. Has anyone made asterisk to recieve faxes this way? (or any other, I'm slowly running out of ideas) I'd be grateful for any reply Jan Fousek ___ --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] Unable to match on CallerID in an include block
I suspect that your dialplan is more than you show ;)
It works just fine for me with svn trunk
[from-sip]
include = common
[common]
exten = 1234,1,NoOp(Hmm ${CALLERID(num)})
exten = 1234/7708,1,NoOp(Here)
If I dial 1234 from my 7708 extension, I get the NoOp(Here)
If I dial 1234 from my 7701 extension, I get the NoOp(Hmm 7701)
Julian.
Steve Hanselman wrote:
Hi Julian,
Ah, a very good point, I put that in my first cut but had completely
forgotten in this one!
1.2.10
Steve
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Julian
Lyndon-Smith
Sent: 22 August 2006 17:30
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Unable to match on CallerID in an include
block
What version of asterisk ?
Julian
Steve Hanselman wrote:
Is there any reason why I can't use the xxx/callerid format in an
include section?
It doesn't seem to work, but if I paste the lines into the main
section
where I include the block it does?
E.g. this doesn't work
[telewest]
Include = spamblock
[spamblock]
_X./12345,s,macro(spamcall)
Whereas this does:
[telewest]
_X./12345,s,macro(spamcall)
Any ideas?
Steve
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RE: [asterisk-users] Realtime Extensions -- Comments?
Yes, but that doesn't stop Asterisk from still treating that row as a valid extension. -Original Message- From: Don [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 22, 2006 10:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime Extensions -- Comments? He meant he added another column in the database table... In the asterisk source everything database related for the default database stuff is explicitly named... Like INSERT INTO blah(col1,col2,col3) Values(foo,foo,foo) - Original Message - From: Douglas Garstang [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, August 22, 2006 12:14 PM Subject: RE: [asterisk-users] Realtime Extensions -- Comments? -Original Message- From: Jason Parker [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 22, 2006 9:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime Extensions -- Comments? - Douglas Garstang [EMAIL PROTECTED] wrote: The unofficial docs on the voip wiki for the realtime extensions table structure is: CREATE TABLE `extensions_table` ( `id` int(11) NOT NULL auto_increment, `context` varchar(20) NOT NULL default '', `exten` varchar(20) NOT NULL default '', `priority` tinyint(4) NOT NULL default '0', `app` varchar(20) NOT NULL default '', `appdata` varchar(128) NOT NULL default '', PRIMARY KEY (`context`,`exten`,`priority`), KEY `id` (`id`) ) TYPE=MyISAM; Uhm... what abouts comments? What if I wanted to temporarily deactivate a couple of extensions? Without a comment flag, I'd have to completely remove those entries from the extensions table! That's not very friendly is it... Is there a better way? 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 I cheated, and just added a comments field to the table. Asterisk only reads fields by name, so extra columns don't hurt at all. How did an extra field that Asterisk doesn't know anything about, change it's behaviour? 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 -- -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.405 / Virus Database: 268.11.5/425 - Release Date: 8/22/2006 ___ --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] LOUD MP3 Hold Music
How do you lower the volume of MP3 hold music? Dennis Clark DENPRO WRK 207.618.1998 CEL 443.415.0527 FAX 1.888.811.8809 [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
Re: [asterisk-users] LOUD MP3 Hold Music
I have the opposite problem. I can hardly hear the hold music at all.On 8/22/06, Dennis P. Clark [EMAIL PROTECTED] wrote:How do you lower the volume of MP3 hold music?Dennis Clark DENPROWRK 207.618.1998CEL 443.415.0527FAX 1.888.811.8809[EMAIL PROTECTED]___--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] LOUD MP3 Hold Music
David Freeman wrote: I have the opposite problem. I can hardly hear the hold music at all. On 8/22/06, *Dennis P. Clark* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: How do you lower the volume of MP3 hold music? I'm certainly not an expert on MOH, but I don't believe there are any volume control knobs to be tweaked in asterisk itself. You might want to take a look at the configs/musiconhold.conf.sample file as there were some parameters that impacted high/med/quiet modes, but I'm thinking they only applied to the old mpg123 music app. Could easily be wrong. Best guess... probably have to use something like sox to change the volume of the mp3 (or whatever) file itself. ___ --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] How can I implement Music on Call Transfer?
Hi friends,I did music on hold. How can we implement music on call transfer? I am unable to find any tutorial about setting up music on call transfer, when call is transferring from one extension to another extension. So, I have two doubts. They are:1) How can I put music on call transfer (Not in music on hold)?2) Music on hold and Music on transfer are the same?Looking forward to your response. Thank you.Regards,Chandra. Get your email and more, right on the new Yahoo.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
[asterisk-users] CTI
Hello, Have someone implemented * like as a CTI platform with IVR, VoiceMail, Fax to tiff files, etc etc using Digium/Sangoma Dual T1/E1 interface cards? Does it work ok? Is the audio quality good when all the ports are in use? Is there any issue regarding on faxes on digital trunks? How do you suggest implementing this kind of solution? Best Regards, Fernando ___ --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] Trunk with multiple IPs?
How do I enter a trunk with multiple IPs. xyz voip provider has 4 IPs and I want to allow incoming from any of them: 1.1.1.1, 1.1.1.2, 1.1.1.3 and 1.1.1.4 Do I put 4 separate host= lines, do I put a single host=line that is comma separated or do I have to set up 4 separate incoming trunks? TIA, Warren ___ --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] Re: Zap and SendDTMF??
I was able to get it working with a meetme.
The D and M options seemed to lock the audio channel for too long.
This is what worked:
exten = 5481,1,NoOp(${TIMESTAMP} paging Group 1 Office Page)
exten = 5481,2,DIAL(Zap/g2/5110,,G(ext-local-custom^5485^1))
exten = 5482,1,NoOp(${TIMESTAMP} paging Group 2 Shop Page)
exten = 5482,2,DIAL(Zap/g2/5110,,G(ext-local-custom^5486^1))
exten = 5484,1,NoOp(${TIMESTAMP} paging Group 4 All Page)
exten = 5484,2,DIAL(Zap/g2/5110,,G(ext-local-custom^5487^1))
exten = 5485,1,MeetMe(5488|1Axd)
exten = 5485,2,SIPDtmfMode(inband)
exten = 5485,3,Wait(1)
exten = 5485,4,SendDTMF(1)
exten = 5485,5,MeetMe(5488|1xd)
exten = 5486,1,MeetMe(5488|1Axd)
exten = 5486,2,SIPDtmfMode(inband)
exten = 5486,3,Wait(1)
exten = 5486,4,SendDTMF(2)
exten = 5486,5,MeetMe(5488|1xd)
exten = 5487,1,MeetMe(5488|1Axd)
exten = 5487,2,SIPDtmfMode(inband)
exten = 5487,3,Wait(1)
exten = 5487,4,SendDTMF(4)
exten = 5487,5,MeetMe(5488|1xd)
--
--
Steven
http://www.glimasoutheast.org
Steven [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
I have tried it with exten = 5481,2,DIAL(IAX2/5480,,D(w1)) and it also does
not work.
I have since moved it to an analog extension on a legacy PBX.
I have tried:
exten = 5481,3,DIAL(Zap/g2/5110,,D(1))
and a macro with SendDTMF.
It works fine if I dial 5110, then enter the number of the zone I wish to
page.
If I dial 5481 with is intended to dial zone 1 automatically, I get a 3-4
second delay before I can speak or it gets cut off.
Note: I am referring to 3-4 seconds after the DTMF digit 1 is sent. I
understand that it should be muted during the D option.
--
--
Steven
http://www.glimasoutheast.org
Alexander Lopez [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
Try This
exten = 5481,1,NoOp(${TIMESTAMP} paging Group 1 Office Page)
exten = 5481,2,DIAL(IAX2/5480/w1||)
SNIP
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RE: [asterisk-users] LOUD MP3 Hold Music
If MP3s are too loud then their should be an internal function that modifies all MP3s in the folder to be at one consistent volume (normalization). Anybody know of a way to do this? Here is my musiconhold.conf just in case I missed something. [classes] Random = quietmp3:/var/lib/asterisk/mohmp3 Dennis Clark DENPRO WRK 207.618.1998 CEL 443.415.0527 FAX 1.888.811.8809 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Tuesday, August 22, 2006 2:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] LOUD MP3 Hold Music David Freeman wrote: I have the opposite problem. I can hardly hear the hold music at all. On 8/22/06, *Dennis P. Clark* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: How do you lower the volume of MP3 hold music? I'm certainly not an expert on MOH, but I don't believe there are any volume control knobs to be tweaked in asterisk itself. You might want to take a look at the configs/musiconhold.conf.sample file as there were some parameters that impacted high/med/quiet modes, but I'm thinking they only applied to the old mpg123 music app. Could easily be wrong. Best guess... probably have to use something like sox to change the volume of the mp3 (or whatever) file itself. ___ --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: Zaptel install - Fedora Core 5
On Tue, Aug 22, 2006 at 11:16:38AM -0500, Rich Adamson wrote: Tzafrir Cohen wrote: On Tue, Aug 22, 2006 at 08:42:36AM -0500, Rich Adamson wrote: Tomislav Parčina wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I did yum update last week and here is my current kernel: I had no problem at all with zaptel. I am only using TDM400P though, in case that matters. Hi Anto! The thing is that I can't rely on yum update for asterisk installation. I would like something that will work like this: I install FC5 from CD/DVD, install RPM's that I need from my ftp server or from CD, install zaptel, libpri, asterisk... So, I need to download rpm's that will allow me to install zaptel/libpri/asterisk without using yum update (I need to make all installations the same). Why bother with the rpm's? Because you have some other programs on your system other than Asterisk. And because you want a reproducable build. Guess that depends a lot on personal objectives, styles, and whether asterisk code has been modified locally. Once the reproducable build is operational and one has to maintain the code, reproducable builds sort of go out the window (eg, customer/system A has a problem, but not customer B through Z). The flip side to this is that only A will get the fix, and not B to Z. This is why also we try to make sure things will be customizable at run-time. However, if you need to provide the same build to both system A and system B, there's double work. Using the Branch SVN checkout approach always provides the most up to date code as opposed to replicating buggy stuff via rpms. What's the problem of building an rpm vs. latest 1.2 / trunk? The only issue is the existing patches. I posted a few monthes ago how to use an 'external' to get the debian/ directory from the pkg-voip repo nd build a package directly from svn using svn-buildpackage. Pick your branch/tag. You may have a set of local changes you wish to maintain. The way of rpm/deb is to maintain them as a set of patches. One possible alternaitve is to try to use an Svk repository with the Digium repo as an aside upstream. It would still be a bit tricky. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [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
Re: [asterisk-users] Realtime Extensions -- Comments?
Im not sure, but there is a commented column that could have 0(not commented) or 1(commented) as values. Is this right? P.S.: I got it from voip--info.org on the realtime Static page... D e n i s G a l v ã o iSolve - Solve Is Our Business Av. Candido de Abreu, 526 1610A CEP: 80530-000 - Curitiba - PR +55 41 3252-2977 r 101 http://www.isolve.com.br On 22 de ago de 2006, at 11:20, Douglas Garstang wrote: The unofficial docs on the voip wiki for the realtime extensions table structure is: CREATE TABLE `extensions_table` ( `id` int(11) NOT NULL auto_increment, `context` varchar(20) NOT NULL default '', `exten` varchar(20) NOT NULL default '', `priority` tinyint(4) NOT NULL default '0', `app` varchar(20) NOT NULL default '', `appdata` varchar(128) NOT NULL default '', PRIMARY KEY (`context`,`exten`,`priority`), KEY `id` (`id`) ) TYPE=MyISAM; Uhm... what abouts comments? What if I wanted to temporarily deactivate a couple of extensions? Without a comment flag, I'd have to completely remove those entries from the extensions table! That's not very friendly is it... Is there a better way? 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] Dialplan or matching
Hey David,
Yes, it can, you just have to play around with the logic and what you
are comparing and when you can do the comparison.
Try something like this:
exten = _18XXNXX,1, NoOP()
exten = _18XXNXX,n,gotoif(${EXTEN}:2:2 = 00 | ${EXTEN}:2:2 =
66 | ${EXTEN}:2:2 = 77 | ${EXTEN}:2:2 = 88)?TRUE:FALSE
exten = _18XXNXX,n(TRUE),Dial()
exten = _18XXNXX,n(FALSE), HangUp()
I'm sure you can take it from there. You can remove the first line with
the NoOP but I normally feel it is good to give an instruction cycle to
Asterisk (and any program) when jumping to another extension (or
function), is it needed, no, but you never know.
Kevin
David Cook wrote:
Maybe I'm daft, but can asterisk to 'or' logic in dialplan matches
sort of like the SPA's can?
Tollfree numbers for example. I can have a line for each combination:
exten = _1800NXX, Dial,
exten = _1866NXX, Dial,
exten = _1877NXX, Dial,
exten = _1888NXX, Dial,
But I want to do is something like this:
exten = _18[0678][0678]NXX, Dial, .
Or to prevent the logic error which albeit small, the above would create:
exten = _18[00,66,77,88:2]NXX, Dial, ..
(representing that the next 2 chars must equal any of '00'.'66','77'
or '88'
Is there any syntax that allows this??
dbc.
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Re: SV: [Asterisk-Users] Nokia E61
I know I am responding to an old post but dont think you would want to change the title of your site from osCommerce to your name ? - Original Message - From: Sam Tam [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Wednesday, July 19, 2006 7:05 AM Subject: RE: SV: [Asterisk-Users] Nokia E61 WE have found this type of phone work better than E61 http://cyber-telecom.net/shop/product_info.php/cPath/21/products_id/31 Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Fredrik Emil Jensen Sent: Tuesday, July 18, 2006 4:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: SV: [Asterisk-Users] Nokia E61 Yes. It works at the same time. The problem is the NAT, you will be able to dial in to the phone as long as the NAT table knows where to redirect the packet too, but when the firewall/router loses its table (usually it will timeout after xx sec/min) you will only be able to dial outgoing call. But I see that the phone support TCP, has anyone tried it with SIP TCP through NAT? Or what about running it through the VPN software that is also on the phone? Or what about the http://sofia-sip.sourceforge.net/ has anyone tried that to see if it works with NAT. The phone it's the best SIP / WIFI phone that I have tried, easy to choose which connection (GMS/InternetPhone) you want to dial through and very good sound. For your guys that are planning on using this through hot-spot etc, you can use example use Birdstep Roaming client (http://www.smartroaming.com), and I can also see that iPass is also creating/created a client http://www.ipass.com/pressroom/pressroom_releases.html?rid=201 based on the GRIC stander. This client will log you on the hot-spot automatic. So its only one problem now it's the NAT issue, I guess you can tunnel this traffic, but that's its another client, and more latency! /Fredrik Jensen -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Devraj Mukherjee Sent: 6. juli 2006 04:55 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: SV: [Asterisk-Users] Nokia E61 Does the GSM and Wi-Fi phone feature work at the same time? :) Thanks for your time On 7/5/06, Amund Nygaard [EMAIL PROTECTED] wrote: Hello I done some more testing, i have no problems connection behind natted networks. It even connected with 3G, but as you can imagine G711 is not very suited for that :P BR Amund Nygaard -Opprinnelig melding- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne av Antonio Rabena Sendt: 5. juli 2006 10:26 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: SV: [Asterisk-Users] Nokia E61 Hi, I had no issues connecting/calling to my asterisk box (on public ip), even my phone is behind a hotspot. Its just that i need to use G711 codec. At 03:34 PM 7/5/2006, you wrote: Hello Has anyone tried a Nokia E6x phone when it is natted? Like behind a hotspot or similar? BR Amund Nygaard -Opprinnelig melding- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne av Devraj Mukherjee Sendt: 4. juli 2006 12:49 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [Asterisk-Users] Nokia E61 Thanks guys. How about the quality of the call etc? Are you happy with the phone, do you recommend them? ___ --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 ___ --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] PRI Ethernet Bridge
Hi Everyone, I am looking into implementing a PRI with our asterisk system. Does anyone recommend a certain PRI to T1 ethernet bridge like red-phone? Are their any installation hiccups I should look out for? Thanks Julian ___ --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] Apache for FastAGI
Assuming you use Perl for AGI scripting, which you should be doing anyways ;-) *cough* You made a typo... you really meant to say 'python'. :P flame Python is to Perl what Pascal was to C. A nice toy ;-) /flame ___ --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] Anybody using Eicon SoftIP with Asterisk
I am trying to configure Eicon's SIP version of SoftIP to work with Asterisk. The documentation is poor and my general knowledge of SIP communications is limited. I am getting a circuits busy message when trying to call the IP address of the server with SoftIP. If anybody has gotten this to work please respond. Thank you. Tim ___ --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] SpanDSP Error
I am running Asterisk 1.2.7 and SpanDSP0.0.2 with Kernel version 2.6.17-1.2141_FC4. I am getting an error after compiling the SpanDSP and putting the .c files in the correct place and then patching the make file. All goes well but then when asterisk is opened, I get this message: [app_rxfax.so]Aug 22 16:55:22 WARNING[13995]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_rxfax.so: undefined symbol: t30_completion_code_to_str Aug 22 16:55:22 WARNING[13995]: loader.c:554 load_modules: Loading module app_rxfax.so failed! Can anyone help me? Thank you, Christian Jensen ___ --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] LOUD MP3 Hold Music
I'm using a Windows software call mp3gain. It can normalize directory. David -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Dennis P. Clark Envoyé : 22 août 2006 15:35 À : [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Objet : RE: [asterisk-users] LOUD MP3 Hold Music If MP3s are too loud then their should be an internal function that modifies all MP3s in the folder to be at one consistent volume (normalization). Anybody know of a way to do this? Here is my musiconhold.conf just in case I missed something. [classes] Random = quietmp3:/var/lib/asterisk/mohmp3 Dennis Clark DENPRO WRK 207.618.1998 CEL 443.415.0527 FAX 1.888.811.8809 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Tuesday, August 22, 2006 2:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] LOUD MP3 Hold Music David Freeman wrote: I have the opposite problem. I can hardly hear the hold music at all. On 8/22/06, *Dennis P. Clark* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: How do you lower the volume of MP3 hold music? I'm certainly not an expert on MOH, but I don't believe there are any volume control knobs to be tweaked in asterisk itself. You might want to take a look at the configs/musiconhold.conf.sample file as there were some parameters that impacted high/med/quiet modes, but I'm thinking they only applied to the old mpg123 music app. Could easily be wrong. Best guess... probably have to use something like sox to change the volume of the mp3 (or whatever) file itself. ___ --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] Polycom 501 vs 601 provisioning
Most of the time, the sample provided by Polycom is the same than the other version. So you dont need to use the new xml. If they add something, its written in the release (pdf) so you just need to add what they say. David De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Douglas Garstang Envoyé: 22 août 2006 11:05 À: Asterisk Users Mailing List - Non-Commercial Discussion Objet: RE: [asterisk-users] Polycom 501 vs 601 provisioning Yes, the config files are the same across a single version of the SIP application software for 301/501/601. If you upgrade however, you'll need to use the new xml files supplied with that version. Polycom needs to get a clue and either make the files backwards compatible or provide a conversion utility. Doug. -Original Message- From: Mike [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 22, 2006 8:44 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Polycom 501 vs 601 provisioning Hi, I have a bunch of Polycom 501 provisioned centrally, and its all working fine. I am toying with the idea of adding one 601, but I will only do so if the provisioning acceptes the same files (i.e. sip.cfg can be common to both 501 and 601s) so as not to burden me any further just for one phone. Is this the case? Mike ___ --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 and spandsp
list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I cheated, and just added a comments field to the table. Asterisk only reads fields by name, so extra columns don't hurt at all. How did an extra field that Asterisk doesn't know anything about, change it's behaviour? Doug. -- Message: 3 Date: Tue, 22 Aug 2006 11:16:38 -0500 From: Rich Adamson [EMAIL PROTECTED] Subject: Re: [asterisk-users] Re: Zaptel install - Fedora Core 5 To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=UTF-8; format=flowed Tzafrir Cohen wrote: On Tue, Aug 22, 2006 at 08:42:36AM -0500, Rich Adamson wrote: Tomislav Par�ina wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I did yum update last week and here is my current kernel: I had no problem at all with zaptel. I am only using TDM400P though, in case that matters. Hi Anto! The thing is that I can't rely on yum update for asterisk installation. I would like something that will work like this: I install FC5 from CD/DVD, install RPM's that I need from my ftp server or from CD, install zaptel, libpri, asterisk... So, I need to download rpm's that will allow me to install zaptel/libpri/asterisk without using yum update (I need to make all installations the same). Why bother with the rpm's? Because you have some other programs on your system other than Asterisk. And because you want a reproducable build. Guess that depends a lot on personal objectives, styles, and whether asterisk code has been modified locally. Once the reproducable build is operational and one has to maintain the code, reproducable builds sort of go out the window (eg, customer/system A has a problem, but not customer B through Z). Using the Branch SVN checkout approach always provides the most up to date code as opposed to replicating buggy stuff via rpms. -- Message: 4 Date: Tue, 22 Aug 2006 09:19:14 -0700 From: Dan Austin [EMAIL PROTECTED] Subject: RE: [asterisk-users] Compilation To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Which version of Web-MeetMe did you download? The process up to 2.0.1 is, well, annoying. Copy app_cbmysql.c to ./asterisk/apps and modify the Makefile to include the application. The project is now hosted on SourceForge and has a much improved build process, but I have not built a release tarball yet. Dan From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Khaled Chehab Sent: Tuesday, August 22, 2006 4:01 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Cc: [EMAIL PROTECTED] Subject: [asterisk-users] Compilation Dear I am installing Web-MeetMe ,one of the requirements is app_cbmysql.c I have it but ,how can I compile it . Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060822/245f726d/attachment-0001.htm -- Message: 5 Date: Tue, 22 Aug 2006 10:19:55 -0600 From: Douglas Garstang [EMAIL PROTECTED] Subject: RE: [asterisk-users] Realtime Extensions -- Comments? To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 -Original Message- From: Matthew Crocker [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 22, 2006 10:03 AM To: Asterisk Users
[asterisk-users] Asterisk, two eth and two providers
Hi, I'm thinking on setting up an asterisk server with two providers. One will let us to make international calls and provide to us a TollFree number. The other will provide local numbers (i'm from Argentina). The problem is that the local number provider requires a dedicated connection and the asterisk server is behind a NAT. There's no problem with the first provider, i just forward the ports and modify the required firewall rules and it's done. The problem comes with the second provider. They gave to me an IP, GW, nmask, etc because it's an static IP. No problem with that.. i configured eth1 with that information.. and works great. The problem comes when setting up asterisk because i just can set one externip in sip.conf file. Also, the packets are beign forwarded to the server (i checked this with TCPDump) but asterisk doesn't give an answer at all. INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Does asterisk support this kind of setup? i mean two providers (one behind NAT and the other with a dedicated connection) with thifferent eth controllers. If you have questions or doubts about what i said. Feel free to ask i'm really looking forward to solve this problem. Best regards, Santiago ___ --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] Re: Asterisk IAXmodem HylaFax?
If you're looking for alternatives to Zetafax why not look at AsterFax (http://asterfax.sourceforge.net)? Your clients can use their existing e-mail client to send faxes. Warrick ___ --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] Missing Extension
Is there a way to have asterisk send an email when a extension disappears or is disconnected? Roger Workman Business Development Upperclassman/Universal Holdings LLC Voice: 304.324.3800 Fax: 304.324.3801 ICQ: 4447584 FWD Network: 56505 Website: http://www.upperclassman.net Billing Questions: billing @upperclassman.net Rental Questions: rentals @upperclassman.net Maintenance: help @upperclassman.net This e-mail and any of its attachments may contain sensitive information, which is privileged, confidential, or subject to copyright belonging to RW Management Inc, Universal Holdings LLC or Upperclassman LLC. This e-mail is intended solely for the use of the individual or entity to which it is addressed. If you are not the intended recipient of this e-mail, you are hereby notified that any dissemination, distribution, copying, or action taken in relation to the contents of and attachments to this e-mail is strictly prohibited and may be unlawful. If you have received this e-mail in error, please notify the sender immediately and permanently delete the original and any copy of or printout of this e-mail. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Warrick Zedi Sent: Tuesday, August 22, 2006 6:29 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: Asterisk IAXmodem HylaFax? If you're looking for alternatives to Zetafax why not look at AsterFax (http://asterfax.sourceforge.net)? Your clients can use their existing e-mail client to send faxes. Warrick ___ --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] Strange SIP response
Hi, I am getting the following message on the CLI: -- Got SIP response 480 Temporarily Unavailable back from 192.168.1.60 -- SIP/EXT23-d910 is circuit-busy and the call hangs up. The peer is correctly registered and I'm not getting unavailable messages. I really need help with this error. -- MENSAJE ENVIADO CON WMAIL 1.01 UNIVERSIDAD DEL CAUCA ___ --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] No CLID from PSTN using X100P FXO Card
Greetings!
Asterisk 1.2.10, Zaptel 1.2.7
For AGES have been using a box setup with 1.0.9 on it and an clone X100P
FXO card. Right now, it's only set up to listen to the PSTN line and
grab the CLID and shove that to an AGI so it can IM me who is calling.
However, reinstalled the box from ground up and installed 1.2.10 and now
CLID isn't working at all. The PSTN line is still transmitting it, as
I've plugged in my Uniden cordless with CLID and it shows up fine on
there, but getting absolutely nothing inside the ${CALLERIDNUM} and
${CALLERIDNAME} variables.
Thoughts? I'm pretty sure I have everything set up correctly, but I'll
post the relevant sections of my conf files in case I don't.
Many thanks,
Nathan
*** zapata.conf ***
[trunkgroups]
[channels]
context=pstn
signalling=fxs_ks
usecallerid=yes
channel = 1
*** zaptel.conf ***
fxsks = 1
loadzone = us
defaultzone = us
*** extensions.conf ***
[pstn]
exten = s,1,Wait(1)
exten =
s,2,agi,hoodahek_dbhandle|${CALLERIDNAME}|${CALLERIDNUM}|${UNIQUEID}
exten = s,3,agi,hoodahek_notify|${CALLERIDNAME}|${CALLERIDNUM}|landline
exten = s,4,Wait(30)
exten = s,5,Dial(SIP/3015841533,20);
exten = s,6,VoiceMail,u1
exten = s,7,Hangup
--
-
Nathan E. Pralle
www.nathanpralle.com
-
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RE: [asterisk-users] Strange SIP response
Have you run SIP DEBUG PEER 192.168.1.60? It may help...tcpdump is also one of my personal favorites -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Diego Andres Asenjo G. Sent: Tuesday, August 22, 2006 6:50 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Strange SIP response Hi, I am getting the following message on the CLI: -- Got SIP response 480 Temporarily Unavailable back from 192.168.1.60 -- SIP/EXT23-d910 is circuit-busy and the call hangs up. The peer is correctly registered and I'm not getting unavailable messages. I really need help with this error. -- MENSAJE ENVIADO CON WMAIL 1.01 UNIVERSIDAD DEL CAUCA ___ --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] PRI and Asterisk
Hi Everyone Any opinions on this? Thanks Julian From: [EMAIL PROTECTED]To: asterisk-users@lists.digium.comDate: Tue, 18 Jul 2006 01:29:57 +Subject: [asterisk-users] PRI and Asterisk Hi All,I am planning to order a PRI and would like to know your opinions on a devices like the Redfone redbridge. Basically any PRI to Asterisk interface that has worked well for you.Thanks,Julian ___ --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] Dialplan or matching
Thanks Kevin! That's what is great about these forums. I never thought
of using gotoif() inside ... one of those Doh! moments.
I included your concept in my standard [dial-ld] context with
${EXTEN}:1:3=800, etc. rather than by 2's, (so it doesn't overlap with
8XX area codes) and select my local loop as the first pick.
dbc.
Kevin Smith wrote:
Hey David,
Yes, it can, you just have to play around with the logic and what you
are comparing and when you can do the comparison.
Try something like this:
exten = _18XXNXX,1, NoOP()
exten = _18XXNXX,n,gotoif(${EXTEN}:2:2 = 00 | ${EXTEN}:2:2
= 66 | ${EXTEN}:2:2 = 77 | ${EXTEN}:2:2 = 88)?TRUE:FALSE
exten = _18XXNXX,n(TRUE),Dial()
exten = _18XXNXX,n(FALSE), HangUp()
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Re: [asterisk-users] SpanDSP Error
Christian Jensen wrote: I am running Asterisk 1.2.7 and SpanDSP0.0.2 with Kernel version 2.6.17-1.2141_FC4. I am getting an error after compiling the SpanDSP and putting the .c files in the correct place and then patching the make file. All goes well but then when asterisk is opened, I get this message: [app_rxfax.so]Aug 22 16:55:22 WARNING[13995]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_rxfax.so: undefined symbol: t30_completion_code_to_str Aug 22 16:55:22 WARNING[13995]: loader.c:554 load_modules: Loading module app_rxfax.so failed! I guess the answer to your problem will be the same as every other time the same question has been asked here, on IRC, in private e-mails to me, etc. 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
[asterisk-users] Hint extension issue - bug?
Hi, I'm using the hint extension to monitoring the status of some extensions. If the extension is defined as a friend, the monitoring doesn't work any more. It only work if I define it as a peer. Is that right ? I mean, I supposed that an extension defined as a friend should have all the functionality of user and peer types. Is this documented somewhere? How can I know the status of an extension of type friend? I hope someone could bring me some light about this issue. Thanks in advance. Lucas Alvarez ___ --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] Strange SIP response
Rushowr wrote: Have you run SIP DEBUG PEER 192.168.1.60? It may help...tcpdump is also one of my personal favorites Yes, I have used it. The lines are extracted from a sip debug on the CLI. I'm going to paste more lines: Sip read: SIP/2.0 480 Temporarily Unavailable To: sip:[EMAIL PROTECTED]:6198;tag=e4331437 From: 24307022sip:[EMAIL PROTECTED];tag=as288765a2 Via: SIP/2.0/UDP 172.16.1.3:5060;branch=z9hG4bK73129cd1;received=172.16.1.3 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Content-Length: 0 7 headers, 0 lines -- Got SIP response 480 Temporarily Unavailable back from 192.168.1.50 Transmitting: ACK sip:[EMAIL PROTECTED]:6198 SIP/2.0 Via: SIP/2.0/UDP 172.16.1.3:5060;branch=z9hG4bK73129cd1 From: 24307022 sip:[EMAIL PROTECTED];tag=as288765a2 To: sip:[EMAIL PROTECTED]:6198;tag=e4331437 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 192.168.1.50:6198 -- SIP/EXT25-a454 is circuit-busy == Everyone is busy/congested at this time I have not detected packet losses even. Thanks for your response. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Diego Andres Asenjo G. Sent: Tuesday, August 22, 2006 6:50 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Strange SIP response Hi, I am getting the following message on the CLI: -- Got SIP response 480 Temporarily Unavailable back from 192.168.1.60 -- SIP/EXT23-d910 is circuit-busy and the call hangs up. The peer is correctly registered and I'm not getting unavailable messages. I really need help with this error. -- MENSAJE ENVIADO CON WMAIL 1.01 UNIVERSIDAD DEL CAUCA ___ --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] Prompts recording for Asterisk
Hi, this question may sound a little dumb, but I need opinion of who are already using asterisk. questions are : 1) which format is best suited for asterisk (.gsm, .wav etc, also what sampling rate and bit size) 2) What are the best sources (cost effective) to get prompts recorded. thanks in advance. Nitin ___ --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] Setting the contact header on outbound INVITE
Is there anyway to set the Contact header on outbound INVITEs such as there is for the REGISTER? I would also like to be able to set the Contact header on responses. Thanks, Michael This email may contain confidential information. If you are not the intended recipient, please advise by return email and delete immediately without reading or forwarding to others. -- Cbeyond ___ --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] Problems with Authorization and INVITE
Hello, I'm trying to do click-to-dial using a script from * http://www.azxws.com/asterisk/ It is a perl script that talks to the Asterisk Manager. I have my asterisk box setup and registering with the provider but when I execute the script, I can see from my ngrep dumps that asterisk is sending out the invite and getting a 401 unauthorized message. I would then expect Asterisk to resend the INVITE with the required info but it is only sending an ACK message. My context contains the username and secret. Any thought as to why the second INVITE is not being sent? Regards Ken ___ --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] Speech Recognition Apps
Im thinking of taking another run at www.Tellme.com to set up an open access Pay-As-You-Go SIP gateway for their Speech Recognition services. I tried to do this about a year ago http://www.voip-info.org/wiki/view/Tellme and whilst the initial enthusiasm was good they ended up more or less building the same idea but partnering with Skype. Now that its a year later on hopefully the timing is right and as I have another application that I could build out if this was available want to take another run. In order to build up momentum it would be great to hear from anyone, actually building speech recognition apps with sphinx or who has an existing Asterisk application that could be ported across if this gateway was made available, the more people we get on board the more likely this is to happening. If anyone has anything else to ad please reply or email directly for more confidential matters. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +1-917-207-3420 Mb +61-2-9016-5642 (Sydney in-dial). ___ --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] Non-zaptel hardware based timing sources
What would the recommended timing source be other than zaptel cards for Asterisk svn trunk on Linux 2.6.x? I have a server that was running with 2 te410p's. I've converted to using SIP so I don't need the cards anymore -- other than to provide a timing source for our applications -- mostly meetme conferencing. What have you had good experiences with? ztdummy? zaprtc? ??? Would I be better off leaving one of the t1 cards in? What about an xp101? Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --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] PRI and Asterisk
Never tested Redfone box. Digium and Sangoma cards works fine for me. Jorge Mendoza Julian Varanini wrote: Hi Everyone Any opinions on this? Thanks Julian From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Tue, 18 Jul 2006 01:29:57 + Subject: [asterisk-users] PRI and Asterisk Hi All, I am planning to order a PRI and would like to know your opinions on a devices like the Redfone redbridge. Basically any PRI to Asterisk interface that has worked well for you. Thanks, Julian ___ --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] Simple CDR parser to print to webpage
Hello -I'm searching for a simple php or perl script to parse Asterisk's CDR csv into a formatted webpage - anyone have any suggestions?-- --Christopher T Aloi-- ___ --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] problem with asterisk billing time...
Hi Im using asterisk 1.0.9 with Linux Ubuntu 5.10 and I have a problem with the time in the cdr-csv. The time registered for the calls is 5 hours earlier than he actual time, it seems respond to the COT and not to the UCT that represents the actual time #date -u Tue Aug 22 19:52:16 UCT 2006 Correct time #date Tue Aug 22 14:52:18 COT 2006 Wrong time appears in the cdr records of asterisk Does anybody know how to fix this? Thanks and good luck with * Jose Manuel Cortes ___ --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] Hint extension issue - bug?
Are you having this problem with the trunk? -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Lucas Alvarez Envoyé : 22 août 2006 18:23 À : Asterisk Developers Mailing List; Asterisk Users Mailing List - Non-Commercial Discussion Objet : [asterisk-users] Hint extension issue - bug? Hi, I'm using the hint extension to monitoring the status of some extensions. If the extension is defined as a friend, the monitoring doesn't work any more. It only work if I define it as a peer. Is that right ? I mean, I supposed that an extension defined as a friend should have all the functionality of user and peer types. Is this documented somewhere? How can I know the status of an extension of type friend? I hope someone could bring me some light about this issue. Thanks in advance. Lucas Alvarez ___ --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] problem with asterisk billing time...
JOSE MANUEL CORTES DAVID wrote: Does anybody know how to fix this? You can start by using a recent version of Asterisk. Current development work is heading toward v1.4. v1.0 is stupid old. Then ensure your Timezone is set properly. Jeremy McNamara ___ --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 1.2.11, Asterisk-Addons 1.2.4 and Zaptel 1.2.8 Released
The Asterisk Development Team is pleased to announce new releases of three of our projects: Asterisk 1.2.11 includes a number of bug fixes, along with an update to the chan_misdn driver for mISDN devices, including Digium's new B410P quad BRI interface card. Asterisk-Addons 1.2.4 includes quite a number of bug fixes and performance improvements in the H.323 channel driver maintained by Objective Systems (chan_ooh323). Zaptel 1.2.8 include a small number of bug fixes, an update to properly licensed header files for the included Octasic API kit and a Makefile target to make it easier for users to build the mISDN kernel and userspace drivers for the B410P. As always, the release files are available on the Digium FTP servers at http://ftp.digium.com, in both tarball and patch file form. All of the release files have been signed with our GPG keys and the signature files are available in the same directories as the release files. Thanks for using and supporting Asterisk! ___ --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] How to set externip in sip.conf automatically?
I need to give Asterisk access to my external IP address to prevent
the NAT problem where caller cannot hear the callee's voice.
According to Asterisk - The Future of Telephony page 92 Environment
Variables:
Environment variables are a way of accessing Unix environment
variables from within Asterisk. They are referenced in the form of
${ENV{var}}
where var is the Unix environment variable you wish to reference.
My external IP is placed each night in a file call /etc/myip and placed
in the $MYIP variable by /etc/bashrc when an shell is loaded.
So I have /etc/myip refreshed each night in a cron job
and when a shell is opened /etc/bashrc does:
export MYIP=`cat /etc/myip`
To access the variable in sip.conf I have tried:
externip=${ENV(EXTERNIP)}
and
${ENV($EXTERNIP)}
but neither seems to work.
Is this the correct syntax? Did I misinterpret the book?
I say neither seems to work because When I hard code
externip=69.91.84.176
there are no NAT problems but when I try to access the $MYIP variable
either of the ways above NAT prevents me hearing the callee's voice.
I have tried but not found a way to directly access the contents of MYIP
to the console using the CLI. Is there a way to see or set _any_ Linux
enviromnent variable using the CLI? More generally, how do I access the
Linux shell from the CLI?
The problem with simply using
externip=69.91.94.176
is that number is subject to change and I don't know an easy way to
automatically write the value into sip.conf programatically.
I could have just said how do I do this but wanted to show that I've
done my homework.
Thanks for any help.
Larry
--
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Linux
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[asterisk-users] Working Sipura 3000 or Linksys 3102 configuration?
Hello I'm having a problem with the Linksys 3102: With incoming PSTN calls, I can hear the caller through the X-Ten softphone, but he can't hear me. The problem is worse with Sjphone and the GrandStream 100 hardphone, as I get no sound in either direction. FWIW... - the SIP client, the PBX and the Linksys are all connected to a switch, with no firewall anywhere - the only way I can get the Linksys to notify the PBX of an incoming PSTN call is using the following settings: * PSTN Line PSTN-To-VoIP Gateway Setup PSTN Ring Thru Line 1 = yes * User 1 Call Forward Settings Cfwd All Dest = fxo (where fxo is the account also used in PSTN Line Subscriber Information to register with the PBX) Dial plans in either Line 1 or PSTN Line don't make it. Could someone upload his configuration of the Linksys (File Save as file) so I can compare with what I have? Since both ends use G711u as their default codec and there's no firewall between them, I suspect I'm totally wrong when it comes to configuring the Linksys as a simple SIP gateway (no use for the FXS port at this point). Possibly some routing issue. Thank you. -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.405 / Virus Database: 268.11.4/424 - Release Date: 21/08/2006 ___ --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] Strange SIP response
Diego Andres Asenjo G. wrote: Hi, I am getting the following message on the CLI: -- Got SIP response 480 Temporarily Unavailable back from 192.168.1.60 -- SIP/EXT23-d910 is circuit-busy and the call hangs up. The peer is correctly registered and I'm not getting unavailable messages. I really need help with this error. Check the sip device config and make sure Do Not Disturb (DND), Call Forwarding, etc, have not be set. ___ --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 IP430 won't finish boot
On 8/21/06, Noah Miller [EMAIL PROTECTED] wrote: Hi Doug - One thing you could try that may or may not help: a different FTP server. I've been using ProFTPd (particularly because you can configure it to use the Polycom default username and password, so you don't have to manually type in those settings on each phone). If that doesn't do it, go for Polycom support. It sounds like you're doing everything according to their book, and it isn't working. If so, they should be the ones to fix it. - Noah ___ Why do you think the problem may be with the FTP server? I've been running vsftpd on several different systems, all with Polycom's. 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
Re: [asterisk-users] Polycom IP430 won't finish boot
DM wrote: Why do you think the problem may be with the FTP server? I've been running vsftpd on several different systems, all with Polycom's. There were reports that the Polycoms preferred some FTP servers over others, but I also use vsftpd (using the default PlcmsSpIp username/password combo) quite successfully on my five provisioning servers. cYa, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: 1 3000 SQUIZ (77849) Fitzroy, VIC T: +61 (0) 3 9235 5400 3065 F: +61 (0) 3 9235 5444 W: http://www.squiz.net/ . Open Source - Own it - Squiz.net ./ ___ --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] Strange SIP response
I had the same problem. The problem was another sip extensions whit the same ip. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Rich Adamson Enviado el: Martes, 22 de Agosto de 2006 11:21 p.m. Para: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] Strange SIP response Diego Andres Asenjo G. wrote: Hi, I am getting the following message on the CLI: -- Got SIP response 480 Temporarily Unavailable back from 192.168.1.60 -- SIP/EXT23-d910 is circuit-busy and the call hangs up. The peer is correctly registered and I'm not getting unavailable messages. I really need help with this error. Check the sip device config and make sure Do Not Disturb (DND), Call Forwarding, etc, have not be set. ___ --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] Calls over VPN
Is anybody making calls over VPN? If so what is the penalty as encryption is involved. I was planning to use VPN to register Sipura units to my local asterisk this way I don't have to deal with NAT issues. -- #Joseph ___ --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] Calls over VPN
We did a setup of 70 sites connected back to a central Asterisk box, and it worked very well over an MPLS VPN. regards, PaulH AsteriskIT www.asteriskit.com.au On Tue, 2006-08-22 at 20:43 -0600, Joseph wrote: Is anybody making calls over VPN? If so what is the penalty as encryption is involved. I was planning to use VPN to register Sipura units to my local asterisk this way I don't have to deal with NAT issues. ___ --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] Calls over VPN
We did a setup of 70 sites connected back to a central Asterisk box, and it worked very well over an MPLS VPN. regards, PaulH AsteriskIT www.asteriskit.com.au The best part about VPN is that it makes it harder for the ISPs to track and mess with. ;-) -- Henry J. Cobb http://www.io.com/~hcobb/ ___ --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] Calls over VPN
MPLS is a VPN, but it doesn't use encryption in most cases.-brandonOn 8/22/06, Paul Hales [EMAIL PROTECTED] wrote:We did a setup of 70 sites connected back to a central Asterisk box, and it worked very well over an MPLS VPN.regards,PaulHAsteriskITwww.asteriskit.com.auOn Tue, 2006-08-22 at 20:43 -0600, Joseph wrote: Is anybody making calls over VPN?If so what is the penalty as encryption is involved. I was planning to use VPN to register Sipura units to my local asterisk this way I don't have to deal with NAT issues.___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Brandon GalbraithEmail: [EMAIL PROTECTED] AIM: brandong00Voice: 630.400.6992A true pirate starts drinking before the sun hits the yard-arm. Ya. --thelost ___ --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] Calls over VPN
Funnily enough, in this case the ISP supported our choice of running Asterisk... PaulH AsteriskIT www.asteriskit.com.au On Tue, 2006-08-22 at 22:04 -0500, Henry J. Cobb wrote: We did a setup of 70 sites connected back to a central Asterisk box, and it worked very well over an MPLS VPN. regards, PaulH AsteriskIT www.asteriskit.com.au The best part about VPN is that it makes it harder for the ISPs to track and mess with. ;-) ___ --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] problems with wevbmail
I can login on the web http://myasterisk.com/cgi-bin/vmail.cgi without problems but i cant see the messages on any folder. Thanks, Sergio. ___ --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] How to set externip in sip.conf automatically?
Larry, am I missing something but you seem to be putting the externip
into the MYIP variable but reading some EXTERNIP variable through
$ENV{}. Shouldn't you be doing something like externip=${ENV{MYIP}}?
The other issue is also the use of curly brackets as opposed to
paranthesis. The snip from the manual seems to use curly brackets but
you're using paranthesis in your example above.
Just silly things to watch out for :D
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[asterisk-users] Multiple site multi server setup
Hi List, I want to do a layout like this: Corporate-Asterisk01 Site A-Asterisk02Site B-Asterisk03 I will have phones register to each server at each location, and also want to store the users voicemail there. Now here is my question. Can I setup the phones (Polycom I was hoping) to register to the server its connected to, and the main server, so in the event the server it is normally connected is down it could still make interoffice calls and make calls via the PSTN from the main server. Would this be done with RegOp() function or something else? Also then all the servers would have to be connected via NFS due to the fact of the voicemails wanting to be stored on another machine while its primary is down, or is this not even possible? Any help would be great! Thanks Ron ___ --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] Calls over VPN
I was thinking of using openVPN -- #Joseph On Tue, 2006-08-22 at 22:06 -0500, Brandon Galbraith wrote: MPLS is a VPN, but it doesn't use encryption in most cases. -brandon On 8/22/06, Paul Hales [EMAIL PROTECTED] wrote: We did a setup of 70 sites connected back to a central Asterisk box, and it worked very well over an MPLS VPN. ___ --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] Prompts recording for Asterisk
Nitin, I'm sure others have better advice but there's no best format per se. Whatever makes asterisk and more importantly the CPU work less in playing those prompts is probably best. from what I understand (*) picks up the best suited format based on the capabilities of the channel and endpoint. If you have endpoints that connect using different codecs, you'd want to have the prompts in all of those formats on your machine and (*) will pick up the relevant ones thus avoiding transcoding. You can find all information on this page: http://www.voip-info.org/wiki-Asterisk+sound+files. What I'm going to do is have the prompts be recorded in a wav (44khz) and then downsample them to 8kHz 16 bits windows wav. Then use the Asterisk 'convert' utility to convert all prompts to all diff formats I expect people to use. Hope this helps ___ --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] How to set externip in sip.conf automatically?
RR wrote:
Larry, am I missing something but you seem to be putting the externip
into the MYIP variable but reading some EXTERNIP variable through
$ENV{}. Shouldn't you be doing something like externip=${ENV{MYIP}}?
The other issue is also the use of curly brackets as opposed to
paranthesis. The snip from the manual seems to use curly brackets but
you're using paranthesis in your example above.
Just silly things to watch out for :D
That was an very good catch RR.
I had _thought_ the curly brackets were correct.
So I went through both ways (with and without the $ in $MYIP)
making sure there were two pairs of curlies and the result was the same.
Calling a POTS phone on my desk from the SIP phone, cannot hear myself
talk into the POTS phone while listening on the SIP phone.
I'd really feel better with a method of examining the externip variable
in CLI but that apparently can't happen. However, each time I just use
externip=xx.xx.xx.xx the call works fine.
--
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Linux
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Re: [asterisk-users] Missing Extension
Can you explain what you mean by disappears? or by disconnected? On 8/22/06, Roger Workman [EMAIL PROTECTED] wrote: Is there a way to have asterisk send an email when a extension disappears or is disconnected? Roger Workman Business Development Upperclassman/Universal Holdings LLC Voice: 304.324.3800 Fax: 304.324.3801 ICQ: 4447584 FWD Network: 56505 Website: http://www.upperclassman.net Billing Questions: billing @upperclassman.net Rental Questions: rentals @upperclassman.net Maintenance: help @upperclassman.net This e-mail and any of its attachments may contain sensitive information, which is privileged, confidential, or subject to copyright belonging to RW Management Inc, Universal Holdings LLC or Upperclassman LLC. This e-mail is intended solely for the use of the individual or entity to which it is addressed. If you are not the intended recipient of this e-mail, you are hereby notified that any dissemination, distribution, copying, or action taken in relation to the contents of and attachments to this e-mail is strictly prohibited and may be unlawful. If you have received this e-mail in error, please notify the sender immediately and permanently delete the original and any copy of or printout of this e-mail. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Warrick Zedi Sent: Tuesday, August 22, 2006 6:29 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: Asterisk IAXmodem HylaFax? If you're looking for alternatives to Zetafax why not look at AsterFax (http://asterfax.sourceforge.net)? Your clients can use their existing e-mail client to send faxes. Warrick ___ --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] How to set externip in sip.conf automatically?
Mate, I'm beginning to think that it can't be done. As in, maybe
you're not allowed to put anything into externip other a valid IP
address and the $ENV{} variable doesn't really work there. You might
want to decipher your externip by registering your server with a
dynamic dns service and then lookup your IP through an nslookup
periodically. Then do some sort of a check and if the address has
dynamically changed, then rewrite your sip.conf file and do a CLI 'sip
reload' or 'restart when convenient'. Not sure why your IP address
should change that frequently anyway, so the approach I mentioned
should cover you. Maybe there are better suggestions out there.
BTW, In the newer versions, maybe it's in 1.4 only, you can use the
keyword 'externhost' where you specify the FQDN of the server, and it
will then lookup your external ip all on its own. You can then use
externrefresh to tell (*) to look it up every so often. Not sure what
the status of this feature is in current 1.2.x releases
Good luck
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[asterisk-users] How can I implement Music on Call Transfer?
Hi friends,I did music on hold. How can we implement music on call transfer? I am unable to find any tutorial about setting up music on call transfer, when call is transferring from one extension to another extension. So, I have two doubts. They are:1) How can I put music on call transfer (Not in music on hold)?2) Music on hold and Music on transfer are the same?Looking forward to your response. Thank you.Regards,Chandra. Do you Yahoo!? Get on board. You're invited to try the new Yahoo! Mail.___ --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] Variable to show caller id for a current call?
Wait a minutewhy are you putting 227 into the CALLERID function? You
should read this:
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+callerid
The (number) portion is the argument to CALLERID telling it what to give
you, not what to insert/write
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Warren (mailing lists)
Sent: Monday, August 21, 2006 3:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Variable to show caller id for a
current call?
But how do you get that with GetVar? I am trying to do this
through the API. I tried:
Action: GetVar
Variable CALLERID(227)
and I tries:
Action: GetVar
Variable ${CALLERID(227)}
Neither returned anything.
How can I do this? Alternately... Is there a way to have a
program fired off when an extension rings that will have the
caller id passed to it as part of the call?
W
Rushowr wrote:
${CALLERID(number)}
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Warren (mailing lists)
Sent: Monday, August 21, 2006 1:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Variable to show caller id for a
current call?
Is there a variable that can be gotten with GetVar to show the
callerid of the current incoming call in progress at a sip
extension?
For instance, a caller from 516-922-9463 calls extension 234.
I would like to be able to be able to get back the
516-922-9463 if I pass 234.
Also, can this be done while the extension is ringing?
TIA,
Warren
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Re: [asterisk-users] Asterisk and spandsp
On Tue, Aug 22, 2006 at 03:27:57PM +0200, Jan Fousek wrote: Hi, I'm trying to get my asterisk recieve faxes via rxfax and spandsp. It creates the connection, recieves some info about the sender and dies. Result is not valid tif file of size 330 or 334 B. Debug messages are shown only sometimes and it ends with something like FLOW Fast carrier training failed FLOW Fast carrier down. Libtiff 3.7.1, asterisk 1.2.9.1, spandsp 0.0.2pre21 installed manually on gentoo. Has anyone made asterisk to recieve faxes this way? How are you conected to your carrier? Have you tried a newer version? Try the latest 0.0.2 (0.0.2pre26). -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [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
Re: [asterisk-users] Re: Asterisk IAXmodem HylaFax?
On Wed, Aug 23, 2006 at 08:28:51AM +1000, Warrick Zedi wrote: If you're looking for alternatives to Zetafax why not look at AsterFax (http://asterfax.sourceforge.net)? Your clients can use their existing e-mail client to send faxes. AsterFax is a rather comlicated setup. Hylafax+IAXModem looks like something that does less wheel reinventings. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [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
Re: [asterisk-users] Re: Asterisk IAXmodem HylaFax?
Tzafrir, When last did you look at AsterFax? What do you believe is required to set it up? In what way are there wheel reinventings in either HylaFax or AsterFax? Tzafrir Cohen wrote: On Wed, Aug 23, 2006 at 08:28:51AM +1000, Warrick Zedi wrote: If you're looking for alternatives to Zetafax why not look at AsterFax (http://asterfax.sourceforge.net)? Your clients can use their existing e-mail client to send faxes. AsterFax is a rather comlicated setup. Hylafax+IAXModem looks like something that does less wheel reinventings. ___ --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] No retry after DNS failure
Today I had a brief power outage which caused the Asterisk server and DSL modem to reboot. The Asterisk server came up before the internet connection was working, so it failed when try to look up some of the hosts for my outbound voip providers in sip.conf. Asterisk never recovered from that, i.e. it never retried so those providers were unavailable. The only provider that was still available was one that I had entered the IP address for, rather than the host name. Have any of you run into this issue, and if so, how have you solved it? It seems that since Asterisk periodically tries to reregister it should also retry the DNS lookup at the same time, rather than never trying again if the lookup fails. This would indicate that Asterisk would also fail if the voip provider changed the IP address of its server because Asterisk would never see the new IP address. Here are some workarounds I thought of, but none of them are particularly good: 1) Get a UPS so my machines won't reboot when the power fails. This actually might not solve anything, because I'm connected to a remote DSLAM in my neighborhood that I believe does not have backup power, so it won't work when the power is out. But perhaps Asterisk is more robust after it has booted (I'll have to test this). 2) Change all host names in sip.conf to IP addresses. This is kind of ugly and also will break when a voip provider changes their IP address. There is a reason for DNS! 3) Have a cron job send asterisk periodic sip reload commands. 4) Delay the start of asterisk until the internet connection has come up. This could cause me to be without any phones if there is any delay or failure in bringing up the network (I also have zap channels and PSTN lines). 5) A hybrid of ideas 3 and 4 above: Have a startup script that waits for the internet connection to come up, and then sends a sip reload command to Asterisk. Any other ideas? John ___ --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 601 Issues
Updating Asterisk is worth a go - I know of someone else who contacted us with a distorted music on hold problem, and an Asterisk updated fixed it. PaulH AsteriskIT www.asteriskit.com.au On Mon, 2006-08-21 at 16:14 +0800, Nathan Alberti wrote: On 20/08/2006, at 8:38 PM, Paul Hales wrote: Does anything pop up on the Asterisk screen? Does music on hold work fine? PaulH On Fri, 2006-08-18 at 13:13 +0800, Nathan Alberti wrote: Nothing strange on the asterisk console... just stopped and started hold on channel. If I repeatedly take a call on and off hold sometimes it will work, other times it they will hear distorted hold music, other times they will hear silence, the same thing happens with voice. Driving me nuts :) I have tried; New firmware on the Polycom New IOS on the router 2 x New Switches Next is Asterisk version. Nathan. ___ --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 retry after DNS failure
Today I had a brief power outage which caused the Asterisk server and DSL modem to reboot. The Asterisk server came up before the internet connection was working, so it failed when try to look up some of the hosts for my outbound voip providers in sip.conf. Asterisk never recovered from that, i.e. it never retried so those providers were unavailable. The only provider that was still available was one that I had entered the IP address for, rather than the host name. snip Are you sure that it was Asterisk? Did you try an nslookup after the network came up while Asterisk wasn't working? How long did you leave it before taking matters into your own hands? DNS will do negative caching as well as positive. If sip.sometel.com failed to resolve (because it doesn't exist or because your network wasn't up), your caching dns server or resolver or both may remember this as a 'negative cache' entry, so that it remembers that it doesn't exist. If your asterisk server is responsible for the network connection, then the ifup/pon script should take care of flushing the dns cache for you for exactly this reason. If you have another server or router that is responsible for the connection then the server running asterisk will have no idea that its negative cache is no longer valid. hth James ___ --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 retry after DNS failure
I have the similar problem but with IAX.I have two servers. First is primary with dynamic IP and open 4569 port. Second is behind firewall.If first server is being disconnected for some time, the second server cannot reconnect. I have to either restart asterisk or stop it, wait for some time and start it again. The same situation is happening when primary server changes IP address.On 8/22/06, John Marvin [EMAIL PROTECTED] wrote:Today I had a brief power outage which caused the Asterisk server and DSL modem to reboot. The Asterisk server came up before the internetconnection was working, so it failed when try to look up some of thehosts for my outbound voip providers in sip.conf.Asterisk never recovered from that, i.e. it never retried so thoseproviders were unavailable. The only provider that was still availablewas one that I had entered the IP address for, rather than the host name.Have any of you run into this issue, and if so, how have you solved it? It seems that since Asterisk periodically tries to reregister it shouldalso retry the DNS lookup at the same time, rather than never tryingagain if the lookup fails. This would indicate that Asterisk would also fail if the voip provider changed the IP address of its server becauseAsterisk would never see the new IP address.Here are some workarounds I thought of, but none of them areparticularly good:1) Get a UPS so my machines won't reboot when the power fails. This actually might not solve anything, because I'm connected to a remoteDSLAM in my neighborhood that I believe does not have backup power, soit won't work when the power is out. But perhaps Asterisk is more robust after it has booted (I'll have to test this).2) Change all host names in sip.conf to IP addresses. This is kind ofugly and also will break when a voip provider changes their IP address.There is a reason for DNS! 3) Have a cron job send asterisk periodic sip reload commands.4) Delay the start of asterisk until the internet connection has comeup. This could cause me to be without any phones if there is any delay or failure in bringing up the network (I also have zap channels and PSTNlines).5) A hybrid of ideas 3 and 4 above: Have a startup script that waits forthe internet connection to come up, and then sends a sip reload command to Asterisk.Any other ideas?John___--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 -- Best regards,Michael Strelnikov ___ --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] 1.2.10 and 1.2.9.1
Hello good people, I'm sure this has been brought up previously but I basically wanted to wait to resurrect this topic till 1.2.10 has been out for a little while, like a cpl of mths. Now I think it has and I just wanted to request for peope who've chosen to upgrade their systems to 1.2.10 to provide their opinions (whomsoever chooses to provide one) about its stability and/or bug fixes as opposed to 1.2.9.1. I'd read a lot of mails about people having upgraded to 1.2.9.1. only to realise that they were better off with 1.2.7 or 1.2.6. Has this been the case with 1.2.10 or is this definately a more stable release specifically with regards to voicemail w/realtime and MeetMe. Thanks in advance to all who respond :) ___ --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 retry after DNS failure
James Harper wrote: Are you sure that it was Asterisk? Did you try an nslookup after the network came up while Asterisk wasn't working? How long did you leave it before taking matters into your own hands? Fairly sure. I didn't realize there was a problem until about 6 hours after the power outage. Local calls go out via the PSTN, so I didn't notice the problem until I tried to make a long distance call. It took a few minutes to diagnose the problem. It was when I did a sip show peers and noticed it was missing my outbound voip providers that I realized there was a problem. A sip reload immediately brought them back. DNS will do negative caching as well as positive. If sip.sometel.com failed to resolve (because it doesn't exist or because your network wasn't up), your caching dns server or resolver or both may remember this as a 'negative cache' entry, so that it remembers that it doesn't exist. I'm fairly sure there is a different error response between name doesn't exist as opposed to a failure to connect to an authoratative name server in the chain. For example if the name servers responsible for .com said that sometel.com didn't exist, or the nameservers for sometel.com said that sip.sometel.com didn't exist then that should be negatively cached. But if the name servers for .com said that the IP address for the name servers for sometel.com are xx.xx.xx.xx and then your dns server can't connect to xx.xx.xx.xx then a different error response is returned, and I don't believe that is or should be negatively cached. If your asterisk server is responsible for the network connection, then the ifup/pon script should take care of flushing the dns cache for you for exactly this reason. If you have another server or router that is responsible for the connection then the server running asterisk will have no idea that its negative cache is no longer valid. Well, you may have hit upon a reason why others may not see this problem. The asterisk server is not responsible for the connection, an external DSL modem/router is responsible. However, I run a local DNS server, so it is available to Asterisk. Perhaps Asterisk is more robust when it can't contact the DNS server, as opposed to it being able to contact the DNS server and the DNS server responds with an error. Perhaps Asterisk doesn't differentiate how it behaves depending on the type of error. John ___ --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] How to modify incoming DNIS?
Hello everyone,
I have a question that seems simple, but I am stumped. Basically, I have
several incoming SIP trunk from gateways connected to physical E1 lines
around the world. Everything works, but the DNIS that comes in are non
standard and sometimes conflict. It varies not only by country, but also
by carrier within a country and even by switch within a carrier. I want
the complete E.164 number, but I rarely get this.
What I would like to do is modify the incoming number as soon as it
enters Asterisk. What happens now is that DNIS 12345678 from SIP trunk A
maps to the same extension as 12345678 from SIP trunk B, even though one
number is actually from Europe and the other from Asia. This causes
many, many conflicts. I want to remap the incoming DNIS to 4412345678
and 6012345678.
Unfortunately, the simple solution doesn't seem to work. The following
fails
(CALLERID(num) is updated correctly, but EXTEN never gets changed.)
[from-sip-trunk-A]
exten = _1234,1,Set(EXTEN=44${EXTEN})
exten = _1234,2,Set(CALLERID(num)=44${CALLERID(num)})
exten = _1234,3,Goto(s,1)
exten = s,1,Goto(incoming-call,${EXTEN},1)
[from-sip-trunk-B]
exten = _1234,1,Set(EXTEN=60${EXTEN})
exten = _1234,2,Set(CALLERID(num)=60${CALLERID(num)})
exten = _1234,2,Goto(s,1)
exten = s,1,Goto(incoming-call,${EXTEN},1)
No error messages are generated, but the EXTEN variable is never
updated! Right after changing it, I print it out with a NoOP and the
value is not updated.
The incoming-call context makes extensive use of macros and other
applications which assume EXTEN will be set correctly. How can I force
EXTEN to be set to the value I want instead of the value that arrives on
the trunk?
Please help. This sounds so simple but I am now very frustrated. The
idea of rewriting every application in the system because EXTEN can't be
changed is very unappealing. I'm already 4 days late getting this up and
running.
Thank you for any assistance,
Chris
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[asterisk-users] Snom360 with 6.2.2 firmware
Hi all, I'm using a Snom360 with bristuffed asterisk and iwant to known if is possibile realizesomthing of this: I receive an incoming call andthen answeredI want to transfer it to a cell phone (or another pubblic number), so press "transfer" on the phone, call the number and only if the called party is avaible i want to transfer the call. Infact with the transfer key, when i send the number, i lost the state of call, and i do not known if the called party was avaible or no. Is there a way to realize this ? Thanks very much in advance Giordano ___ --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] lbProxy
Hello Ive been trying to use the lbProxy SIP load balancing proxy. After I actually got it compiled, using the CMSOFAZ.COM version, I began experimenting. I quickly ran into a problem. Heres my setup: wan--|lan Phone - lbProxy - Asterisk lbProxy sends all of the sip packets to Asterisk, but when asterisk responds, it chooses the port from which the request came, instead of using the port number in the Via statement. This is a good thing normally, due to NAT, but even if I set nat=no or nat=never, it still responds to this port instead of the port in the Via Statement. Has anybody gotten this to work, and can explain how. Im about 1 day away from writing my own load balancing sip proxy, but would love if I could use lbProxy instead :) Jon -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.405 / Virus Database: 268.11.4/424 - Release Date: 21-08-2006 ___ --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] H323 can not register to remote openh323gk?
Hi,all: in /etc/asterisk/h323.conf I setting gatekeeper=192.168.0.19 secret=3001 and on server 192.168.0.19 I running a openh323gk and add a user 3001 and password is 3001 too, but when I booting asterisk, I got messages : Error registering with gatekeeper "192.168.0.19".Aug 22 15:58:22 ERROR[2590]: chan_h323.c:2373 load_module: Gatekeeper registration failed. I don't know why? tengulre ___ --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] GSM gateway and FXO ATA
Hi list!
I'm trying to connect Analog GSM gateway (2N Ateus) with Asterisk 1.2.5 over
Grandstream HT488 ATA. I have plugged gateway to FXO port of ATA. Incoming
calls work the way I call GSM number and then I get DISA to call inside
company. Outgoing call work well when I call VoIP number of ATA which calls
gateway and then I dial number I wish to call over gateway. As I said, it works
fine.
Now I would like to dial ATA_number+number_I_wish_to_call so that I don't have
to dial twice when I'm trying to establish outgoing call from company thru
gateway.
I have tried this but it doesn't work well.
; to dial outside thru GSM gateway
exten = _456.,1,Dial(SIP/${EXTEN},30,tTD(${EXTEN:3}))
exten = _456.,n,Hangup
This is what I see on CLI:
-- Executing Dial(SIP/577-104c, SIP/4560989970434|30|tTD(248)) in new
stack
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Hangup(SIP/577-104c, ) in new stack
== Spawn extension (sip, 4560989970434, 2) exited non-zero on 'SIP/577-104c'
Why asterisk thinks that gateway is busy when it's not?
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
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Re: [asterisk-users] Re: what is the real use of AEL?
In the above, Jean-Michel puts it right on the table: of what
possible use is AEL? Why am I bothering to waste my time with it? It's a
valid question! It deserves some discussion!
First of all I'd like to thank for all the good answers and valid points
people have made to this question.
snip/
Sorry for the diversion. My answer to Jean-Michel's straightforward
question goes along some different lines than rushowr. I never really
cared how fast/efficient the extension engine was-- it's obvious I'm
not writing stuff for thousands of concurrent users like rushowr. But in
the majority of cases, it's the apps that are run from AEL that take up
all the execution time. As long as AEL execution time is pretty minimal
between priorities, it's probably going to be OK. (users of dialplans
for intensively loaded sites may HIGHLY disagree!)
I admit it, I haven't done the test. Using FastAGI is just enough speed
for me :-)
My first reason for getting excited about AEL, enough so, to rewrite the
parser to make it more user-friendly, and add a few bells and whistles,
was that it provided an opportunity to code dialplans with higher
level constructs than gotos. Truly, AEL is to extension.conf format,
as programming languages are to assembler.
While I can see that it is a nice thing to have, in my opinion it makes
configuration files look less and less like configuration files and
more like a programming language. And I don't know, I doubt that mixing
these two things look very good.
Now to be honest, I'll probably be using AEL (assuming it's there to
stay) for complicated dialplan constructs, but #includes are going to be
good to avoid having a conf file that looks like a mix of apples and
potatoes :)
But, in this, Jean-Michel
is right to ask: we already have several possible programming languages,
perl, C, java, PHP, ruby, and so on. The one advantage AEL holds over
all of them, is that its structure parallels the data model used in
Asterisk. That data model is composed mainly of contexts, extensions,
and priorities. Because AEL follows that structure, it is easier to
write dialplans than it would be in other languages. Most AGI scripts
don't have to deal with anything above the priority level... and if
you do want to generate an entire dialplan from the innards of a perl
script, I doubt it would easier to read and understand than an AEL
script, nor do I think it would be anywhere near as concise.
That's where I beg to differ. How is the code snippet more concise /
readable than, say:
#!/usr/bin/perl
use FictionalAPI; # imports NoOp, Verbose...
sub loop {
my $iterations = 100;
my $time1 = time;
NoOp('hello') for (1..$iterations);
my $time2 = time;
my $diff = $time2 - $time1;
my $prisecs = 4 * $iterations / $diff;;
Verbose(The time diff is $diff seconds);
Verbose(Which means the priorities/sec = $prisecs);
SayNumber($prisecs);
}
That assumes you have FictionalAPI, which is why I highlighted the need
to clean, well defined API and good IPC communication between Asterisk
and external systems / program.
The next reason I spent time on it was code quality. There is no lint
yet for extensions.conf. We've seen little things like misspellings
of exten = into extem = silently drop that priority, which may
take months to spot and fix. Not that a thorough linter couldn't be
written, but I did add tons of checks to the AEL parser, to spot common
errors at compile time, rather than find them at run-time in a
production dial-plan.
Now that's awesome. The OCamel language has a pretty crazy type infering
compiler (i.e. the compiler infers the type of variables from their use
in the code rather than a declaration) you might want to take a look at.
This is as good as type checking goes IMHO :)
Haven't we all taken a course on software
quality, or read some articles at least? How much do errors cost
if found at compile time, as opposed to the cost of finding them at
run time? The earlier the phase at which errors are found and
eliminated, the cheaper the error. Right? and so, AEL is a tool for
you to reduce your costs of generating dial plans. I might add here,
that other languages also do similar checks at parse time; but some of
the checks that AEL will do, are specific to the underlying data model.
You won't necessarily get those kind of checks out of perl or php.
Valid point. However, Perl has a nice fix: the Test::More suite and
things like Mock::Object, which let you write pretty comprehensive test
suites to do some kind of code quality (any decent CPAN module has a
test suite...). That being said, you are right, type checking is
important and it saves time. I wish Perl had Ocamel parser / compiler
features :)
[BTW, haven't you ever stopped, after you have finished writing a
dialplan, just as you are about to put it in production on a live
asterisk server with tons of
Re: [asterisk-users] GSM gateway and FXO ATA
What you need is something like:
exten = _456.,1,Dial(SIP/[EMAIL PROTECTED],30,tTD(${EXTEN:3}))
regards,
PaulH
AsteriskIT
www.asteriskit.com.au
On Tue, 2006-08-22 at 10:59 +0200, Tomislav Parčina wrote:
Hi list!
I'm trying to connect Analog GSM gateway (2N Ateus) with Asterisk 1.2.5 over
Grandstream HT488 ATA. I have plugged gateway to FXO port of ATA. Incoming
calls work the way I call GSM number and then I get DISA to call inside
company. Outgoing call work well when I call VoIP number of ATA which calls
gateway and then I dial number I wish to call over gateway. As I said, it
works fine.
Now I would like to dial ATA_number+number_I_wish_to_call so that I don't
have to dial twice when I'm trying to establish outgoing call from company
thru gateway.
I have tried this but it doesn't work well.
; to dial outside thru GSM gateway
exten = _456.,1,Dial(SIP/${EXTEN},30,tTD(${EXTEN:3}))
exten = _456.,n,Hangup
This is what I see on CLI:
-- Executing Dial(SIP/577-104c, SIP/4560989970434|30|tTD(248)) in new
stack
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Hangup(SIP/577-104c, ) in new stack
== Spawn extension (sip, 4560989970434, 2) exited non-zero on 'SIP/577-104c'
Why asterisk thinks that gateway is busy when it's not?
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
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Re: [asterisk-users] GSM gateway and FXO ATA
Assuming the 456 is the ATA number and the outside number is always 10
digits.
exten = _456.,1,Dial(SIP/${EXTEN:0:3}/${EXTEN:-10},tT)
but then it might as well be
exten = _456.,1,Dial(SIP/456/${EXTEN:-10},tT)
; to dial outside thru GSM gateway
exten = _456.,1,Dial(SIP/${EXTEN},30,tTD(${EXTEN:3}))
exten = _456.,n,Hangup
M
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Re: [asterisk-users] 1.2.10 and 1.2.9.1
RR wrote: I'd read a lot of mails about people having upgraded to 1.2.9.1. only to realise that they were better off with 1.2.7 or 1.2.6. Has this been the case with 1.2.10 or is this definately a more stable release I've had good results with 1.2.10. But, I'm only using queues lightly. Current uptime is 2 weeks and I haven't had a crash yet. 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
Re: [asterisk-users] No retry after DNS failure
John Marvin wrote: Today I had a brief power outage which caused the Asterisk server and DSL modem to reboot. The Asterisk server came up before the internet connection was working, so it failed when try to look up some of the hosts for my outbound voip providers in sip.conf. Asterisk never recovered from that, i.e. it never retried so those providers were unavailable. The only provider that was still available was one that I had entered the IP address for, rather than the host name. Have any of you run into this issue, and if so, how have you solved it? It seems that since Asterisk periodically tries to reregister it should also retry the DNS lookup at the same time, rather than never trying again if the lookup fails. This would indicate that Asterisk would also fail if the voip provider changed the IP address of its server because Asterisk would never see the new IP address. Here are some workarounds I thought of, but none of them are particularly good: 1) Get a UPS so my machines won't reboot when the power fails. This actually might not solve anything, because I'm connected to a remote DSLAM in my neighborhood that I believe does not have backup power, so it won't work when the power is out. But perhaps Asterisk is more robust after it has booted (I'll have to test this). 2) Change all host names in sip.conf to IP addresses. This is kind of ugly and also will break when a voip provider changes their IP address. There is a reason for DNS! 3) Have a cron job send asterisk periodic sip reload commands. 4) Delay the start of asterisk until the internet connection has come up. This could cause me to be without any phones if there is any delay or failure in bringing up the network (I also have zap channels and PSTN lines). 5) A hybrid of ideas 3 and 4 above: Have a startup script that waits for the internet connection to come up, and then sends a sip reload command to Asterisk. Any other ideas? If memory serves correctly, most of the above has been raised as issues in the past and the suggested work around has been to run a dns caching server on the asterisk box. FWIW, I always use IP addresses instead of dns names. But, I don't have to deal with dynamic ip changes of any device either. ___ --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 forum - forum.globalvoicenet.com
There use to be forum on this site http://forum.globalvoicenet.com/ and you could read mails from this list on Forum-like fashion. Does anybody know what happened? Have they moved somewhere else? Thank you for info. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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 retry after DNS failure
Rich Adamson wrote: If memory serves correctly, most of the above has been raised as issues in the past and the suggested work around has been to run a dns caching server on the asterisk box. That's exactly what I am doing, unless you mean that the dns caching server caches results over a reboot. FWIW, I always use IP addresses instead of dns names. But, I don't have to deal with dynamic ip changes of any device either. I don't have to deal with any dynamic IP changes either, other than the possibility of one of my voip providers changing the IP address of their servers, which is probably a very rare occurence. John ___ --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] Re: Zaptel install - Fedora Core 5
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I did yum update last week and here is my current kernel: I had no problem at all with zaptel. I am only using TDM400P though, in case that matters. Hi Anto! The thing is that I can't rely on yum update for asterisk installation. I would like something that will work like this: I install FC5 from CD/DVD, install RPM's that I need from my ftp server or from CD, install zaptel, libpri, asterisk... So, I need to download rpm's that will allow me to install zaptel/libpri/asterisk without using yum update (I need to make all installations the same). -- Tomislav a Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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
