Re: [asterisk-users] AGI doesn't execute PHP5 script
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Stefan-Michael. Guenther (in-put GbR) wrote: Hi, Am Dienstag, 8. August 2006 13:41 schrieb Matt Riddell (NZ): Stefan-Michael. Guenther (in-put GbR) wrote: Hi, I'm trying to start a PHP5 script via the AGI Interface. The asterisk version is Asterisk 1.2.5-BRIstuffed-0.3.0-PRE-1k and I followed the instructions on http://www.voip-info.org/tiki-print.php?page=Asterisk+AGI+php The problem is, as you can see from the output in the CLI, that Asterisk claims that it executes the script, but nothing happens. It doesn't create the file /tmp/asterisk and it doesn't send an email. When I execute the script manually on the command line, it is executes without an error, the file is there and the email, too. ^^^ Try running it from the command line and see what happens I guess you meant the test.php script, right? Executing php5 scripts on the command line isn't a problem at all, only when they are started through AGI. This particular script also? Are you using AGI DEBUG in console? - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFE2XwPS6d5vy0jeVcRAlCtAJoDPCIIgX4XxVELnoQYEmvc1l+oLwCcDioH 7AyLQZcKjqrJMBxqNiM4qI8= =tgJm -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI doesn't execute PHP5 script
Hi, Am Mittwoch, 9. August 2006 08:09 schrieb Matt Riddell (NZ): The problem is, as you can see from the output in the CLI, that Asterisk claims that it executes the script, but nothing happens. It doesn't create the file /tmp/asterisk and it doesn't send an email. When I execute the script manually on the command line, it is executes without an error, the file is there and the email, too. ^^^ Try running it from the command line and see what happens I guess you meant the test.php script, right? Executing php5 scripts on the command line isn't a problem at all, only when they are started through AGI. This particular script also? Are you using AGI DEBUG in console? yes, I can execute test.php on the command line and it runs as expected. Wenn I call it via AGI nothing happens. Yes, the first mail contained the output of the script with agi debug, set verbose 10, set debug 10 set before. Here it is again: asterisk*CLI dial [EMAIL PROTECTED] -- Executing Answer(OSS/dsp, ) in new stack Console call has been answered -- Executing AGI(OSS/dsp, test.php) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/test.php AGI Tx agi_request: test.php AGI Tx agi_channel: OSS/dsp AGI Tx agi_language: en AGI Tx agi_type: Console AGI Tx agi_uniqueid: asterisk-6958-1155024459.47 AGI Tx agi_callerid: unknown AGI Tx agi_calleridname: unknown AGI Tx agi_callingpres: 0 AGI Tx agi_callingani2: 0 AGI Tx agi_callington: 0 AGI Tx agi_callingtns: 0 AGI Tx agi_dnid: unknown AGI Tx agi_rdnis: unknown AGI Tx agi_context: guenther AGI Tx agi_extension: 111 AGI Tx agi_priority: 2 AGI Tx agi_enhanced: 0.0 AGI Tx agi_accountcode: AGI Tx LI -- AGI Script test.php completed, returning 0 -- Executing Hangup(OSS/dsp, ) in new stack Hangup on console And it doesn't make a difference whether I use the dial command or a sip phone to call extension 111 in context [guenther]. Strange, isn't it? Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice over IP - Lösungen ___ --Bandwidth 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] realtime+mysql
I'm attempting to setup asterisk running real-time with mysql. Right now I
can get asterisk to start and run but a show dialplan shows basically
nothing other than parking extensions. I'm watching the full log also for
debug messages and I can see that asterisk is connecting to mysql with out a
issue but for some reason it does not seam to read the extensions table.
Below is a few of my configs..
#My slimmed down modules.conf
[EMAIL PROTECTED] asterisk]# cat /etc/asterisk/modules.conf
[modules]
autoload=no
load = res_config_mysql.so
load = res_crypto.so
load = res_features.so
load = chan_features.so
load = chan_iax2.so
load = pbx_realtime.so
load = app_realtime.so
#My extconfig.conf
[EMAIL PROTECTED] asterisk]# cat /etc/asterisk/extconfig.conf
[settings]
extensions = mysql,asterisk,extensions
#mysqldump of my asterisk database
DROP TABLE IF EXISTS `extensions`;
CREATE TABLE `extensions` (
`context` varchar(20) NOT NULL default 'default',
`extension` varchar(20) NOT NULL default '',
`priority` int(2) NOT NULL default '1',
`application` varchar(20) NOT NULL default '',
`args` varchar(50) default NULL,
`descr` text,
`flags` int(1) NOT NULL default '0',
PRIMARY KEY (`context`,`extension`,`priority`)
) ENGINE=MyISAM DEFAULT CHARSET=latin1;
--
-- Dumping data for table `extensions`
--
INSERT INTO `extensions` VALUES
('sortcalls','1949265snip',1,'Wait','10','Wait(10)',0)
INSERT INTO `extensions` VALUES
('sortcalls','_1949265snip',1,'Wait','10','Wait(10)',0);
My iax.conf is setup to forward calls to sortcalls context... this is what i
get when a call comes in..
Aug 8 22:47:58 NOTICE[10074]: chan_iax2.c:7303 socket_read: Rejected
connect attempt from 64.61.93.87, request '1949265snip@sortcalls' does not
exist
Aug 8 22:47:58 NOTICE[10074]: chan_iax2.c:7303 socket_read: Rejected
connect attempt from 64.61.93.90, request '1949265snip@sortcalls' does not
exist
Anybody know whats going wrong here?
--
~Shaun
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Re: [asterisk-users] realtime+mysql
If i am not mistaken, you need to have another IAX user tables to store all the iax users. HEre is some example http://www.voip-info.org/wiki/view/Asterisk+RealTime+IAX
good luck!On 8/9/06, Shaun [EMAIL PROTECTED] wrote:
I'm attempting to setup asterisk running real-time with mysql.Right now Ican get asterisk to start and run but a show dialplan shows basicallynothing other than parking extensions.I'm watching the full log also for
debug messages and I can see that asterisk is connecting to mysql with out aissue but for some reason it does not seam to read the extensions table.Below is a few of my configs..#My slimmed down modules.conf
[EMAIL PROTECTED] asterisk]# cat /etc/asterisk/modules.conf[modules]autoload=noload = res_config_mysql.soload = res_crypto.soload = res_features.soload = chan_features.soload = chan_iax2.so
load = pbx_realtime.soload = app_realtime.so#My extconfig.conf[EMAIL PROTECTED] asterisk]# cat /etc/asterisk/extconfig.conf[settings]extensions = mysql,asterisk,extensions#mysqldump of my asterisk database
DROP TABLE IF EXISTS `extensions`;CREATE TABLE `extensions` (`context` varchar(20) NOT NULL default 'default',`extension` varchar(20) NOT NULL default '',`priority` int(2) NOT NULL default '1',
`application` varchar(20) NOT NULL default '',`args` varchar(50) default NULL,`descr` text,`flags` int(1) NOT NULL default '0',PRIMARY KEY(`context`,`extension`,`priority`)) ENGINE=MyISAM DEFAULT CHARSET=latin1;
Dumping data for table `extensions`--INSERT INTO `extensions` VALUES('sortcalls','1949265snip',1,'Wait','10','Wait(10)',0)INSERT INTO `extensions` VALUES('sortcalls','_1949265snip',1,'Wait','10','Wait(10)',0);
My iax.conf is setup to forward calls to sortcalls context... this is what iget when a call comes in..Aug8 22:47:58 NOTICE[10074]: chan_iax2.c:7303 socket_read: Rejectedconnect attempt from
64.61.93.87, request '1949265snip@sortcalls' does notexistAug8 22:47:58 NOTICE[10074]: chan_iax2.c:7303 socket_read: Rejectedconnect attempt from 64.61.93.90, request '1949265snip@sortcalls' does not
existAnybody know whats going wrong here?--~Shaun___--Bandwidth and Colocation provided by
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[asterisk-users] problem with queues
Hi all Does any one experience the scenario in which an agent sits behind any trunk and serves the queue. Bcz I have tried and the queue acts very veared as when it tries the agent extension across the trunk it starts music onhold and when the agent is busy/not responding then it stops music on hold again when it starts hunting the agents across a trunk again starts the MOH and again stops the MOH when it can't find the agent online. To make thing worse if the MOH is dynamic then the user hears different MOH tones after a period of 1 or two seconds, and if u have enabled voice prompts in the queue then the result is even more bad. Best Regards Mohammad Zeeshan Latif smime.p7s Description: S/MIME cryptographic signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FW: problem with queues
Hi all Does any one experience the scenario in which an agent sits behind any trunk and serves the queue. Bcz I have tried and the queue acts very veared as when it tries the agent extension across the trunk it starts music onhold and when the agent is busy/not responding then it stops music on hold again when it starts hunting the agents across a trunk again starts the MOH and again stops the MOH when it can't find the agent online. To make thing worse if the MOH is dynamic then the user hears different MOH tones after a period of 1 or two seconds, and if u have enabled voice prompts in the queue then the result is even more bad. Best Regards Mohammad Zeeshan Latif ___ --Bandwidth 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] G729
I registered 5 g729 codec and the result was , I cant use these channels because all channels are not available even I have no call on the system 5/0 encoders/decoders of 5 licensed channels are currently in use Please help * 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
[asterisk-users] UK mobile reject codes
If I make a call to a mobile phone from an ISDN30 PRI line and the call is A) not answered (but no voicemail) or B) the call is rejected, is there any list anywhere of the different codes returned to the ISDN (hangupcause) For example, if I call an o2 mobile, and reject is pressed then I get a hangupcode 16. However, if I call a virgin mobile, and reject is pressed, a hangupcode of 31 is returned. Anyone know why this may be the case ? Or does anyone have a list for the mobile operators ? Many 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
[asterisk-users] Re: realtime+mysql
IAX is being read from the flat config like it
normally is. I can verify this because asterisk registers with my
provider.
-- ~Shaun
"Sharon Lim" [EMAIL PROTECTED] wrote
in message news:[EMAIL PROTECTED]...If
i am not mistaken, you need to have another IAX user tables to store all the
iax users. HEre is some example http://www.voip-info.org/wiki/view/Asterisk+RealTime+IAX
good luck!
On 8/9/06, Shaun
[EMAIL PROTECTED]
wrote:
I'm
attempting to setup asterisk running real-time with mysql.Right
now Ican get asterisk to start and run but a show dialplan shows
basicallynothing other than parking extensions.I'm watching
the full log also for debug messages and I can see that asterisk is
connecting to mysql with out aissue but for some reason it does not seam
to read the extensions table.Below is a few of my configs..#My
slimmed down modules.conf [EMAIL PROTECTED] asterisk]# cat
/etc/asterisk/modules.conf[modules]autoload=noload =
res_config_mysql.soload = res_crypto.soload =
res_features.soload = chan_features.soload = chan_iax2.so
load = pbx_realtime.soload = app_realtime.so#My
extconfig.conf[EMAIL PROTECTED] asterisk]# cat
/etc/asterisk/extconfig.conf[settings]extensions =
mysql,asterisk,extensions#mysqldump of my asterisk database
DROP TABLE IF EXISTS `extensions`;CREATE TABLE `extensions`
(`context` varchar(20) NOT NULL default
'default',`extension` varchar(20) NOT NULL default
'',`priority` int(2) NOT NULL default '1',
`application` varchar(20) NOT NULL default
'',`args` varchar(50) default NULL,`descr`
text,`flags` int(1) NOT NULL default
'0',PRIMARY
KEY(`context`,`extension`,`priority`)) ENGINE=MyISAM DEFAULT
CHARSET=latin1; Dumping data for table
`extensions`--INSERT INTO `extensions`
VALUES('sortcalls','1949265snip',1,'Wait','10','Wait(10)',0)INSERT
INTO `extensions`
VALUES('sortcalls','_1949265snip',1,'Wait','10','Wait(10)',0);
My iax.conf is setup to forward calls to sortcalls context...
this is what iget when a call comes in..Aug8 22:47:58
NOTICE[10074]: chan_iax2.c:7303 socket_read: Rejectedconnect attempt
from 64.61.93.87, request
'1949265snip@sortcalls' does notexistAug8
22:47:58 NOTICE[10074]: chan_iax2.c:7303 socket_read: Rejectedconnect
attempt from 64.61.93.90, request
'1949265snip@sortcalls' does not existAnybody
know whats going wrong
here?--~Shaun___--Bandwidth
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Re: [asterisk-users] Re: realtime+mysql
Sorry if i am wrong. Did you add something in extensions.conf to identify your context ? Something like this http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions
BTW, how come your extensions got snip? I taught extension is a number that you dial?On 8/9/06, Shaun
[EMAIL PROTECTED] wrote:
IAX is being read from the flat config like it
normally is. I can verify this because asterisk registers with my
provider.
-- ~Shaun
Sharon Lim [EMAIL PROTECTED] wrote
in message news:[EMAIL PROTECTED]
...If
i am not mistaken, you need to have another IAX user tables to store all the
iax users. HEre is some example http://www.voip-info.org/wiki/view/Asterisk+RealTime+IAX
good luck!
On 8/9/06, Shaun
[EMAIL PROTECTED]
wrote:
I'm
attempting to setup asterisk running real-time with mysql.Right
now Ican get asterisk to start and run but a show dialplan shows
basicallynothing other than parking extensions.I'm watching
the full log also for debug messages and I can see that asterisk is
connecting to mysql with out aissue but for some reason it does not seam
to read the extensions table.Below is a few of my configs..#My
slimmed down modules.conf [EMAIL PROTECTED] asterisk]# cat
/etc/asterisk/modules.conf[modules]autoload=noload =
res_config_mysql.soload = res_crypto.soload =
res_features.soload = chan_features.soload = chan_iax2.so
load = pbx_realtime.soload = app_realtime.so#My
extconfig.conf[EMAIL PROTECTED] asterisk]# cat
/etc/asterisk/extconfig.conf[settings]extensions =
mysql,asterisk,extensions#mysqldump of my asterisk database
DROP TABLE IF EXISTS `extensions`;CREATE TABLE `extensions`
(`context` varchar(20) NOT NULL default
'default',`extension` varchar(20) NOT NULL default
'',`priority` int(2) NOT NULL default '1',
`application` varchar(20) NOT NULL default
'',`args` varchar(50) default NULL,`descr`
text,`flags` int(1) NOT NULL default
'0',PRIMARY
KEY(`context`,`extension`,`priority`)) ENGINE=MyISAM DEFAULT
CHARSET=latin1; Dumping data for table
`extensions`--INSERT INTO `extensions`
VALUES('sortcalls','1949265snip',1,'Wait','10','Wait(10)',0)INSERT
INTO `extensions`
VALUES('sortcalls','_1949265snip',1,'Wait','10','Wait(10)',0);
My iax.conf is setup to forward calls to sortcalls context...
this is what iget when a call comes in..Aug8 22:47:58
NOTICE[10074]: chan_iax2.c:7303 socket_read: Rejectedconnect attempt
from 64.61.93.87, request
'1949265snip@sortcalls' does notexistAug8
22:47:58 NOTICE[10074]: chan_iax2.c:7303 socket_read: Rejectedconnect
attempt from 64.61.93.90, request
'1949265snip@sortcalls' does not existAnybody
know whats going wrong
here?--~Shaun___--Bandwidth
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Re: [asterisk-users] cmd DIAL - Who picked up the call?
No, i cannot open a bug on this, cause i dont have a PRI that uses zap. so if there were any questions, you had to answer them. But there is a similar bug, using mISDN. http://bugs.digium.com/view.php?id=7435 and a solution for ME, dont know if it will help you: http://fuhrmannek.de/projects/asterisk/download/res_features-misdn-bugfix.diff good luck KAI Koopmann, Jan-Peter schrieb: On Tuesday, August 01, 2006 9:36 AM Kai Ober wrote: when you park a call (asterisk feature defautl keys: #700 ...) at your isdn phone and you forgot to catch the call on another phone, the phone from where you parked the call, should ring after 45 seconds (default) does this work for you? (which asterisk version dou you have?) 1.2.9.1 bristuffed and no it does not seem to work. It seems to mixup src and dst channel: == Parked Zap/4-1 on 701. Will timeout back to extension [from_internalisdn] s, 1 in 300 seconds The call came from another extension and another context. Therefore the callback will fail (and _does_ fail)... Will you file a bug report and give me the bug number? ___ --Bandwidth 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] AGI doesn't execute PHP5 script [SOLVED]
Hi, I have solved it (but don't understand yet, why it works)!!! SuSE 10.1 uses different configuration files for the cli version and the cgi version of PHP5. When I modify the first line in the script from #! /usr/bin/php5 to #! /usr/bin/php5 -c /etc/php5/cli/ the php scripts gets executed! I guess I have to compare the parameters in the two configuration files to get the important detail. AGI Tx agi_accountcode: AGI Tx LI -- AGI Script test.php completed, returning 0 -- Executing Hangup(OSS/dsp, ) in new stack Hangup on console And it doesn't make a difference whether I use the dial command or a sip phone to call extension 111 in context [guenther]. Strange, isn't it? Yeah. That response is usually when things are not happening properly. Matt, the ouput hasn't changed, although ist script is executed properly. Why do you thing that this output shows a failure? Also are you using php -q? No, this isn't required for php5 (taken from README.CLI): * CLI is started up in quiet mode by default. (-q switch kept for compatibility) Thanks for your help and suggestions, Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice over IP - Lösungen ___ --Bandwidth 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] Two card NT-TE mode
I want to connect two ISDN bri card directly. Is it necessary to use a cross-cable? I use a fritz card in TE mode and a Atlantis Card (zaphfc) in NT mode. Thank's Matteo ___ --Bandwidth 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 Siemens Legacy PBX
James Arscott [EMAIL PROTECTED] writes: Hi Small progress, though combining the suggest below, enabling overlapdial and a few other things I have got the following : When you hit 9 on the simenes, you hear a dial tone. As soon as you hit another number to start dialling it complains with some generic error on the siemens handset. What I see from asterisk at the same time [...] -- Starting simple switch on 'Zap/62-1' -- Accepting overlap call from '697000' to 'unspecified' on channel 0/31, span 2 Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 1/0x1) (Originator) Message type: INFORMATION (123) [70 02 81 39]LI Called Number (len= 4) [ Ext: 1 TON: Unknown Number Type (0) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '9' ] -- Processing IE 112 (cs0, Called Party Number) Hmm? That's not overlap dialing. You get the complete called number in one single Message. -- Processing IE 112 (cs0, Called Party Number) NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Overlap Receiving, peerstate Overlap sending Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 1/0x1) (Terminator) Message type: RELEASE COMPLETE (90) [08 02 81 81]LI Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ] ... and Asterisk's answer is: Unallocated number. It seems your Siemens PBX doesn't do it right. We had some issues with other PBXs and Asterisk when Asterisk was the NET-side. Try to reverse the roles, so that Siemens is NET and Asterisk CPE. That helped here with an Alcatel. cu, Wolfgang ___ --Bandwidth 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-dev] Questions regarding g.729 and g.711 in Asterisk
Olle E Johansson wrote: 7 aug 2006 kl. 14.37 skrev Rich Adamson: You have two choices to correct the behavior. One, change the asterisk definitions so as to show a preference (disallow=all, allow=g729,ulaw), or, two, change the sip phone's definition to prefer g729 as its first choice. Or use the SIP_CODEC variable in the dialplan to set a prefered codec for the call. Wow, what a great function, is there an IAX2_CODEC cariable too? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ever donate Software to Digium? If you did your a fool.
With respect, I don't think you understand the dynamics of growing a business. If we are all to benefit from the continued development of Asterisk then it is in our own best interests for Digium to succeed, because their success is for our benefit. Your posting is unfortunate as it disregards the considerable effort, cost and time put into Asterisk by Mark and Digium. By the way, I have no relation with Digium other than to derive a considerable benefit from open source software developed by Mark/Digium and a lot of other programmers, for which I am extremely grateful. Digium is not being given a whole load of money - the investors will want a slice of the company and the future profits. That's how VC funding works. Randall H. wrote: If you gave software to Digium then you helped Mark become very rich. http://abcnews.go.com/Technology/wireStory?id=2290152 ___ --Bandwidth 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] Two card NT-TE mode
On Wed, Aug 09, 2006 at 12:05:30PM +0200, support_list wrote: I want to connect two ISDN bri card directly. Is it necessary to use a cross-cable? No. A standard (non-crossed) ethernet cable will work, as long as it has all 8 wires. -- 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] Ever donate Software to Digium? If you did youra fool.
If you gave software to Digium then you helped Mark become very rich. What's wrong with making Mark a rich man? He has come up with a great new product and I'm sure he has risked a lot to get it to you. Asterisk is free so he owes you nothing. How about you take your jealousy elsewhere or maybe put your energy into doing something worthwhile. Regards Mark Brooker T: 02 4959 8670 M: 0415 846 865 F: 02 4950 5609 E: [EMAIL PROTECTED] W: http://www.mbit.com.au -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of George Gardiner Sent: Wednesday, 9 August 2006 8:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Ever donate Software to Digium? If you did youra fool. With respect, I don't think you understand the dynamics of growing a business. If we are all to benefit from the continued development of Asterisk then it is in our own best interests for Digium to succeed, because their success is for our benefit. Your posting is unfortunate as it disregards the considerable effort, cost and time put into Asterisk by Mark and Digium. By the way, I have no relation with Digium other than to derive a considerable benefit from open source software developed by Mark/Digium and a lot of other programmers, for which I am extremely grateful. Digium is not being given a whole load of money - the investors will want a slice of the company and the future profits. That's how VC funding works. Randall H. wrote: If you gave software to Digium then you helped Mark become very rich. http://abcnews.go.com/Technology/wireStory?id=2290152 ___ --Bandwidth 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] Re: Sangoma A200D and DTMF Detection
We experienced this problem with a Sangoma A104D card. With echo cancel turned on, the card was not detecting incoming DTMF digits to our IVR properly. However, when we added the line relaxdtmf=yes to zapata.conf, the problem went away. If the relaxdtmf setting is not curing the problem for you, I would suggest sending an email to [EMAIL PROTECTED]. Rana Dutt Softel Solutions rdutt at softelinc dot 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] AGI doesn't execute PHP5 script [SOLVED]
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Stefan-Michael. Guenther (in-put GbR) wrote: Yeah. That response is usually when things are not happening properly. Matt, the ouput hasn't changed, although ist script is executed properly. Why do you thing that this output shows a failure? :) Usually my scripts have hundreds of lines of debug statements, so when I see nothing come back, I'm always a little concerned! :) - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFE2cYRS6d5vy0jeVcRAt8WAJ9W15a0rslNlGx8YDXl13dwbFMI0gCfWX8c 58YUsSa6/3Bpr/yyd/VYcps= =r7Pv -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Ever donate Software to Digium? If you didyoura fool.
Not only that but Asterisk and Digium has enabled ALL of us to market and produce and support a product for businesses that would have no other alternative but to spend even more money on the big boys or get smaller less featured phone systems without the benefits of VOIP. We all succeed in this scenario and the resources Digium has put into this product has helped us just as much (if not more) than it has helped them. You only see this type of jealousy from people who haven't made an impact, open or closed source. I see people on the lists complaining about having to pay $10 for a g729 codec or that some of the digital interface cards are a lot of money - that's such a small thing to complain about when you are getting Asterisk for $0 and it's enabled you to make money! Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of MBIT Technologies Sent: Wednesday, August 09, 2006 7:20 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Ever donate Software to Digium? If you didyoura fool. If you gave software to Digium then you helped Mark become very rich. What's wrong with making Mark a rich man? He has come up with a great new product and I'm sure he has risked a lot to get it to you. Asterisk is free so he owes you nothing. How about you take your jealousy elsewhere or maybe put your energy into doing something worthwhile. Regards Mark Brooker T: 02 4959 8670 M: 0415 846 865 F: 02 4950 5609 E: [EMAIL PROTECTED] W: http://www.mbit.com.au -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of George Gardiner Sent: Wednesday, 9 August 2006 8:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Ever donate Software to Digium? If you did youra fool. With respect, I don't think you understand the dynamics of growing a business. If we are all to benefit from the continued development of Asterisk then it is in our own best interests for Digium to succeed, because their success is for our benefit. Your posting is unfortunate as it disregards the considerable effort, cost and time put into Asterisk by Mark and Digium. By the way, I have no relation with Digium other than to derive a considerable benefit from open source software developed by Mark/Digium and a lot of other programmers, for which I am extremely grateful. Digium is not being given a whole load of money - the investors will want a slice of the company and the future profits. That's how VC funding works. Randall H. wrote: If you gave software to Digium then you helped Mark become very rich. http://abcnews.go.com/Technology/wireStory?id=2290152 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ever donate Software to Digium? If you did your a fool.
EVERYONE PLEASE DON'T FEED THE TROLL!! That post was done only for the sake of generating responses, and we do no one any favors by taking the bait. thx. B. ___ --Bandwidth 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] Phone Newbie Questions
First let me just say that I am a total newbie when it comes to phones but I have several years of Linux and it experience. I have been tasked with offering a competing solution to our current phone providers based on asterisk... To show my complete ignorance I am going to try and describe out setup, or as much as I know about it. We have a PRI line with a single number that rolls over to several lines. We also have a handful of analogue fax lines (are these part of the PRI...I dunno?). There are also a few 800 numbers...(not sure if that fact really matters to our phone system or not). From my bit of research I am pretty sure asterisk does support PRI but please correct me if I am wrong 'cause none of this matters if it doesn't... First is asterisk really capable of supporting a 100+ user base of phones? I understand (or at least think I do) that I need one of the Digium Digital TDM Cards and that is where the PRI connection gets pluged into...how then do I connect the asterisk system to the phone network that exists in the office? (Told you I was clueless) Any guidance would be appreciated or even links to an introduction on phone systems would be great (pictures would help too ;) ). Brian ___ --Bandwidth 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] Using a DB for Configurations
Thanks David But what I was more looking for was storing the configuation file eg extensions.conf as a database file in MY SQL and then have asterisk load the table from MYSQL as opposed the text file extensions.conf ? Is there any benefit in this ? Thanks all Barry ___ --Bandwidth 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] Ever donate Software to Digium? If you did your afool.
Lol, your definition of rich and mine are obviously very different. $13m I'd call that working capital. He hasn't sold the company, he hasn't walk away to retire in Anguilla. Personally I'm very thankful for the work put in by Digium, it allows me to run a pabx in my home office with functionality I wouldn't otherwise be able to get from Cisco etc. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Randall H. Sent: Wednesday, 9 August 2006 1:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Ever donate Software to Digium? If you did your afool. If you gave software to Digium then you helped Mark become very rich. http://abcnews.go.com/Technology/wireStory?id=2290152 ___ --Bandwidth 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] Handling inbound and outbound calls passed from a proxy
you must add option insecure=very|yes|no in sip.conf, see http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conffor more info by default incoming calls goes into default context have you checked if registration has occuredin sipproxy?check debug messages in asterisk console 2006/8/9, kjcsb [EMAIL PROTECTED]: I need to handle the following scenarios:1. UA1 -- SIP Proxy -- Asterisk2. UA2 -- SIP Proxy -- Asterisk -- PSTN gateway (SIP) I have configured a trunk to register with the SIP proxy:trunk1register=user1:[EMAIL PROTECTED]/DID1UA1 calls [EMAIL PROTECTED] and the call is recognised as being to DID1. I set up an inbound route for DID1 and route the call as appropriate. That dealswith scenario 1.I then tried to configure another trunk to handle scenario 2:trunk2context=from-internalhost=SIP.Proxy type=peerregister=user2:[EMAIL PROTECTED]A call to PSTN1 from the UA is passed to the SIP proxy which recognises itas PSTN call. The SIP proxy updates the From details and passes the call to Asterisk which (I presume) puts the call into the from-internal context anddials the outbound route appropriately.However that setup messes up scenario 1 which now gives a 404 back to UA1. Ipresume Asterisk is not differentiating between a call made to user1 from UA1 and a call made to PSTN1 from user2. It's just seeing a call fromSIP.Proxy and putting it into the from-internal context.Could anyone advise how I would set up Asterisk to cope with both thesescenarios? I could setup DID2 but I don't know how to pass the call onto the PSTN gateway. I am using AMP/FreePBX but if someone could advise the generalprinciples I would appreciate it.ThanksCameron___--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] Two card NT-TE mode
Tzafrir Cohen wrote: On Wed, Aug 09, 2006 at 12:05:30PM +0200, support_list wrote: I want to connect two ISDN bri card directly. Is it necessary to use a cross-cable? No. A standard (non-crossed) ethernet cable will work, as long as it has all 8 wires. I disagree. If the pin layout of the ISDN cards is in TE mode on both cards, then you do need a crossover cable. Usually the cards are in TE pin layout, except some cards have onboard jumpers for changing the pin-layout to NT mode (like junghanns quadbri). I guess in your case you need a BRI crossover cable. attached is a picture of an ISDN BRI crossover cable (from Diva Server Adapters Installation Guide) regards klaus ___ --Bandwidth 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] Ever donate Software to Digium? If you did your afool.
I'll feed the Troll!!! Mark deserves this, he has given us, all of us, a way to make and/or save money. Kudos to him and the staff at Digium. I, for one feel Mark owes me nothing but I still feel like I owe him and the project much of my uncompleted work. Way to GO!! Mark. May the extra cash available allow you to grow Digium into what ONLY you can dream Another Asterisk 'fool', Alex -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Randall H. Sent: Wednesday, August 09, 2006 1:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Ever donate Software to Digium? If you did your afool. If you gave software to Digium then you helped Mark become very rich. http://abcnews.go.com/Technology/wireStory?id=2290152 ___ --Bandwidth 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] Phone Newbie Questions
Brian,What you need are some sources of good information to get you started. Based on what you wrote there is a lot to cover - impossible in an email.Buy and read the book from this link - http://www.oreilly.com/catalog/asterisk/the pdf version can be found here but having one to read is much better.http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 With a background in Linux, once you learn a bit more setting up asterisk will be easy. And yes Asterisk can handle a 100+ users with relatively standard hardware.Colin On 8/9/06, Brian Becker [EMAIL PROTECTED] wrote: First let me just say that I am a total newbie when it comes to phonesbut I have several years of Linux and it experience.I have beentasked with offering a competing solution to our current phoneproviders based on asterisk... To show my complete ignorance I am going to try and describe outsetup, or as much as I know about it.We have a PRI line with asingle number that rolls over to several lines.We also have ahandful of analogue fax lines (are these part of the PRI...I dunno?). There are also a few 800 numbers...(not sure if that fact reallymatters to our phone system or not).From my bit of research I ampretty sure asterisk does support PRI but please correct me if I amwrong 'cause none of this matters if it doesn't... First is asterisk really capable of supporting a 100+ user base of phones?I understand (or at least think I do) that I need one of the DigiumDigital TDM Cards and that is where the PRI connection gets pluged into...how then do I connect the asterisk system to the phone networkthat exists in the office? (Told you I was clueless)Any guidance would be appreciated or even links to an introduction onphone systems would be great (pictures would help too ;) ). Brian___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sipura SPA-3000 vs Sangoma A200
Hi there, I'm looking to set up a home-office PBX/Asterisk lab using a VIA EPIA motherboard as an always on, low powered solution. I am trying to find out the differences between a solution using an external ATA (like the Sipura SPA-3000) or an internal PCI card (like the Sangoma A200 with 2 FXO 2 FXS ports). Can someone tell me what the major functionality and limitations are when using the two devices with Asterisk? I am looking for things like configuration, power draw, sending faxes, etc. So far I have these: Sipura SPA-3000 - External - can be located remotely from Asterisk server, can be used with an embedded solution (NSLU2) or a server with no PCI slot (smaller footprint and lower power draw). The problem with this is that there are more power blocks/adaptors needed. Draws about 5W (according to the statistics). Ports - 1xFXO and 1xFXS Cost - 80 UK pounds Features not in Sangoma A200: Sangoma A200 (A20101) Internal PCI - half or full hight. Ports - 2xFXO and 2xFXO, power consumption ??? cost: 200 UK pounds Features not in Sipura SPA-3000: Thanks in advance! ___ --Bandwidth 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] Jabber: Difference between client and component
Is there any difference between having asterisk as a jabber client or jabber component ? Does anyone know what settings need to be set (!) in order to connect as a component to a wildfire server ? 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] Sipura SPA-3000 vs Sangoma A200
Stephen G wrote: Hi there, I'm looking to set up a home-office PBX/Asterisk lab using a VIA EPIA motherboard as an always on, low powered solution. I am trying to find out the differences between a solution using an external ATA (like the Sipura SPA-3000) or an internal PCI card (like the Sangoma A200 with 2 FXO 2 FXS ports). Can someone tell me what the major functionality and limitations are when using the two devices with Asterisk? I am looking for things like configuration, power draw, sending faxes, etc. So far I have these: Sipura SPA-3000 - External - can be located remotely from Asterisk server, can be used with an embedded solution (NSLU2) or a server with no PCI slot (smaller footprint and lower power draw). The problem with this is that there are more power blocks/adaptors needed. Draws about 5W (according to the statistics). Ports - 1xFXO and 1xFXS Cost - 80 UK pounds Features not in Sangoma A200: Sangoma A200 (A20101) Internal PCI - half or full hight. Ports - 2xFXO and 2xFXO, power consumption ??? cost: 200 UK pounds Features not in Sipura SPA-3000: One of the biggest differences is the spa3k is rather limited in terms of echo cancellation. If your pstn line is outside the limits of the spa3k's echo canceller, you'll have less then acceptable audio quality. Unfortunately, there isn't any nice way to identify your pstn line characters (etc) without trying it. Also, a few users complain about voice interpreted as dtmf signaling under some circumstances. Having spent two years with the spa3k in multiple environments, I'd suggest your alternative choice of the A200 card is a better one. Power consumption has nothing to do with your choices really. ___ --Bandwidth 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] Sipura SPA-3000 vs Sangoma A200
Rich, Thanks for the quick reply and your advice. My main goal is to build a small, energy efficient, always on server that will be able to run Asterisk and connect up to the PSTN, with FAX ability. The PCI card makes the solution cleaner, but it is harder to find small cases/motherboards that support PCI other than a custom Mini-ITX solution. I agree that the power consumption should not be much different between the Sipura and Sangoma A200, although I need to have a large enough power supply and an extra 4-pin computer power cable to power any FXS ports according to: http://wiki.sangoma.com/sangoma-hardware#A200. - Original Message From: Rich Adamson [EMAIL PROTECTED] Having spent two years with the spa3k in multiple environments, I'd suggest your alternative choice of the A200 card is a better one. Power consumption has nothing to do with your choices really. ___ --Bandwidth 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 Configuration
Hi All I am new member to asterisk mailing list. I have complied the asterisk and it is running fine. I have configured two extensions in extensions.conf exten = 228,1,Dial exten = 234,1,Dial and configured the xlite soft phone. when I am calling from 234 to 228 it is unable to establish the call. I am able to here all automated playback IVR. ex.500, 600 can any one help to configure the inbound / outbound calls and how to add sip users. -Linga Reddy ___ --Bandwidth 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 Configuration
Hi: First at all: You SIP phones are right register on sip.conf file? Cris From: R.Linga Reddy [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk Configuration Date: Wed, 09 Aug 2006 19:40:50 +0530 Hi All I am new member to asterisk mailing list. I have complied the asterisk and it is running fine. I have configured two extensions in extensions.conf exten = 228,1,Dial exten = 234,1,Dial and configured the xlite soft phone. when I am calling from 234 to 228 it is unable to establish the call. I am able to here all automated playback IVR. ex.500, 600 can any one help to configure the inbound / outbound calls and how to add sip users. -Linga Reddy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Dont just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/ ___ --Bandwidth 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 Configuration
You need to tell asterisk what to dial. Check the dial command syntax and probably the sip.conf file.On 8/9/06, R.Linga Reddy [EMAIL PROTECTED] wrote:HiAllI am new member to asterisk mailing list. I have complied the asterisk and it is running fine.I have configuredtwo extensions in extensions.confexten = 228,1,Dialexten = 234,1,Dialand configured the xlite soft phone. when I am calling from 234 to 228 it is unable to establish the call.I am able to here all automated playback IVR. ex.500, 600can any one help to configure the inbound / outbound calls and how toadd sip users.-Linga Reddy ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- BruceNortex Networks ___ --Bandwidth 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] Phone Newbie Questions
Yes it supports PRIOn 8/9/06, Colin MacMillan [EMAIL PROTECTED] wrote: Brian,What you need are some sources of good information to get you started. Based on what you wrote there is a lot to cover - impossible in an email.Buy and read the book from this link - http://www.oreilly.com/catalog/asterisk/the pdf version can be found here but having one to read is much better. http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 With a background in Linux, once you learn a bit more setting up asterisk will be easy. And yes Asterisk can handle a 100+ users with relatively standard hardware.Colin On 8/9/06, Brian Becker [EMAIL PROTECTED] wrote: First let me just say that I am a total newbie when it comes to phonesbut I have several years of Linux and it experience.I have beentasked with offering a competing solution to our current phoneproviders based on asterisk... To show my complete ignorance I am going to try and describe outsetup, or as much as I know about it.We have a PRI line with asingle number that rolls over to several lines.We also have ahandful of analogue fax lines (are these part of the PRI...I dunno?). There are also a few 800 numbers...(not sure if that fact reallymatters to our phone system or not).From my bit of research I ampretty sure asterisk does support PRI but please correct me if I amwrong 'cause none of this matters if it doesn't... First is asterisk really capable of supporting a 100+ user base of phones?I understand (or at least think I do) that I need one of the DigiumDigital TDM Cards and that is where the PRI connection gets pluged into...how then do I connect the asterisk system to the phone networkthat exists in the office? (Told you I was clueless)Any guidance would be appreciated or even links to an introduction onphone systems would be great (pictures would help too ;) ). Brian___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephony www.baldwintechsolutions.comPhone: 518-631-2855 x205Fax: 518-631-2856 ___ --Bandwidth 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] Sipura SPA-3000 vs Sangoma A200
Stephen, In my experience setting up an office PBX, I started with several SPA-3000's and eventually decided to go with an A200d. There were two reasons for changing to the A200: 1. Hardware echo canceller. Despite all the configuration settings I tried, there was always a faint echo with the SPA-3000 that the users kept mentioning.The A200's echo canceller killed the echo in conversations - saving both the user's and my sanity. The only echo now is when people are using speakerphones with the speaker volume cranked up (SNOM 320's). 2. I had several instances with the SPA-3000s where they dropped calls or bridged two local people into the same PSTN line. This could have partially been the way I was trying to use it - relying on the 3000 to reject a request for an outgoing call before trying the next one. Any erratic behaviour seemed to coincide with it re-registering to the asterisk server. I was able to reproduce dropped calls on 4 different units with different FW versions by setting the register time to 1s, and hammering it with multiple outgoing calls. Since I went with the A200, I didn't follow this up or investigate further. It seems that YMMV with the SPA-3000 - my opinion is that they work great for home use, but I wouldn't use them for a business application. Most fax transmissions worked well for both of them when going directly from the FXO to the FXS ports, but not 100%. We eventually plugged the fax directly into it's own dedicated line - taking the fax line out of the phone system. Configuration of the 3000's looks daunting at first, but isn't too bad after you've done it a couple times. I had some issues getting the echo canceller on the A200 working when it was first installed, but after I got it working (their tech support responded quickly) I re-installed the entire system twice from scratch and couldn't reproduce the issue. I don't know the power consumption of the A200. If you are sizing up equipment, I wouldn't rule out the digium cards. I did some limited testing with a Digium TDM 400P - I don't recall any issues.I think you need to go to a TDM2400P to get the HWEC. Perhaps a SBC with a PCI Slot would suit your purpose if you decide to go with an internal card? Regards, Dana Harding - Original Message - From: Stephen G [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, August 09, 2006 7:20 AM Subject: [asterisk-users] Sipura SPA-3000 vs Sangoma A200 Hi there, I'm looking to set up a home-office PBX/Asterisk lab using a VIA EPIA motherboard as an always on, low powered solution. I am trying to find out the differences between a solution using an external ATA (like the Sipura SPA-3000) or an internal PCI card (like the Sangoma A200 with 2 FXO 2 FXS ports). Can someone tell me what the major functionality and limitations are when using the two devices with Asterisk? I am looking for things like configuration, power draw, sending faxes, etc. So far I have these: Sipura SPA-3000 - External - can be located remotely from Asterisk server, can be used with an embedded solution (NSLU2) or a server with no PCI slot (smaller footprint and lower power draw). The problem with this is that there are more power blocks/adaptors needed. Draws about 5W (according to the statistics). Ports - 1xFXO and 1xFXS Cost - 80 UK pounds Features not in Sangoma A200: Sangoma A200 (A20101) Internal PCI - half or full hight. Ports - 2xFXO and 2xFXO, power consumption ??? cost: 200 UK pounds Features not in Sipura SPA-3000: Thanks in advance! ___ --Bandwidth 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] Ever donate Software to Digium? If you did your a fool.
George Gardiner wrote: Digium is not being given a whole load of money - the investors will want a slice of the company and the future profits. That's how VC funding works. More like selling your soul to the Devil, actually. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --Bandwidth 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] Two card NT-TE mode
Yes, I have read this part of the Diva Server Adapters Installation Guide and I think that it is necessary use a cross cable. Matteo Klaus Darilion wrote: Tzafrir Cohen wrote: On Wed, Aug 09, 2006 at 12:05:30PM +0200, support_list wrote: I want to connect two ISDN bri card directly. Is it necessary to use a cross-cable? No. A standard (non-crossed) ethernet cable will work, as long as it has all 8 wires. I disagree. If the pin layout of the ISDN cards is in TE mode on both cards, then you do need a crossover cable. Usually the cards are in TE pin layout, except some cards have onboard jumpers for changing the pin-layout to NT mode (like junghanns quadbri). I guess in your case you need a BRI crossover cable. attached is a picture of an ISDN BRI crossover cable (from Diva Server Adapters Installation Guide) regards klaus ___ --Bandwidth 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] Prague PTT?
Is anyone familiar with the Telco in Prague? We have an issue with the connection that will be made from the Telco demark when we do an IPT installation next week. -jason - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ --Bandwidth 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] ESCAUX releases net.PBX Free Edition
* ESCAUX releases net.PBX Free Edition, a free and Open Source version of its original ESCAUX net.PBX product. * ESCAUX net.PBX is a turnkey Asterisk solution designed for the SME and Corporate customer. We made Asterisk easy. Expercience for yourself and download the ESCAUX net.PBX Free Edition today. The software is ready for download at: http://www.escaux.com/netpbx ESCAUX net.PBX runs directly from CD, requires no installation and does not overwrite your harddisk. Your configuration changes are centrally stored on our servers and become instantly available at every system reboot. Simply place our Smart Live CD in your PC, follow the Configuration Wizard and show-off to friends and colleagues with your Web controlled, Asterisk based, Business IP PBX. The net.PBX Free Edition is a full featured IP PBX system suitable for Business use. You can install and run the Free Edition on your own servers and benefit from a close to zero cost IP PBX system for your company. On Demand commercial support packages are available to assist you with installation or configuration but you can also refer to our net.PBX Community Forum. The ESCAUX Free Edition is an Open Source project hosted at SourceForge. At any moment in time you can decide to migrate towards a Commercial version of our product and benefit from our unmatched Service Level Agreement, Guaranteed response and Repair times. Best regards, the ESCAUX development team ___ --Bandwidth 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] Deployment for less than 10 phones
Im doing some research about how to deploy asterisk in small offices. So far I have seen the soekris implementation with astlinux and it sounds good. Please share your comments/ideas for the following configuration: Note: Pure PBX only, no routing/firewall functions needed. Small Office #1 Up to 10 analog phones (FXS) Up to 3 or 4 PSTN lines (FXO) Asterisk providing standard pbx features and voicemail.(no call center stuff) Codec is G711 Small Office #2 Up to 10 Voip Phones (sip) with g711 Up to 3 or 4 incoming SIP lines via Ethernet from the VOIP provider Asterisk providing standard pbx features and voicemail. No PSTN connectivity (or maybe just one emergency port???) The idea to use G711 is to minimize transcoding and to maintain the costs to a bare minimum. Either using a standard PC or a soekris board (Epygi Quadro is too expensive and I dont need the routing functions). Usually is accepted that using G711 on each leg, it needs 30MHZ per voice channel so a 300MHZ computer will give me the 10 calls I need while keeping the CPU transcoding to a minimum. Soekris boards/case cannot fit a TDM400 card unless that has changed recently, Any ideas if sangoma cards fit? Also, the net4801-60 soekris board has a 266MHZ cpu so i will only get about 8 calls. However I need some light here8 calls FXS to ZAP? SIP to SIP? Suggestions for small form factor cases are welcomed. Thanks for all your comments. Thanks for our comments. -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] em wink, TE110P, * answers too soon
Hi, I've been googling all over the place and have read the relevant articles in the Digium knowledge base. I have tried all the suggestions I found in the K.B. Spent some time on the asterisk irc, tweaking some parameters as people thereon thought would be helpful, but to no avail. I am trying to set up * on an em wink trunk currently attached to an Avaya Merlin Magix system. The provider of the T1 is McLeodUSA; our location is St Paul MN USA. I am in the process of getting more specific timing information from their tech support, but it takes days. I can call into the * PBX from my cell phone just fine. I can call between the two grandstream phones I bought for testing just fine. Here's the problem. When a call comes into *, * attempts to route it to an extension prematurely. For example, if the DTMF digits coming from upstream are '538', * tries to send the call to extn '53'. I still receive the '8', but too late. Here's a snip from /var/log/asterisk/messages where the incoming DID digits are '535': Aug 7 22:30:00 DEBUG[31492] chan_zap.c: Monitor doohicky got event Ring/Answered on channel 1 Aug 7 22:30:00 DEBUG[31478] devicestate.c: Changing state for Zap/1 - state 2 (In use) Aug 7 22:30:00 VERBOSE[31493] logger.c: Asterisk Ready. -- Starting simple switch on 'Zap/1-1' Aug 7 22:30:00 DEBUG[31494] app_queue.c: Device 'Zap/1' changed to state '2' (In use) but we don't care because they're not a member of any queue. Aug 7 22:30:01 DEBUG[31493] chan_zap.c: DTMF digit: 5 on Zap/1-1 Aug 7 22:30:01 DEBUG[31493] chan_zap.c: DTMF digit: 3 on Zap/1-1 Aug 7 22:30:01 DEBUG[31493] chan_zap.c: Enabled echo cancellation on channel 1 Aug 7 22:30:01 VERBOSE[31493] logger.c: == Unknown extension '53' in context 'demo' requested Aug 7 22:30:04 DEBUG[31493] channel.c: Set channel Zap/1-1 to write format gsm Aug 7 22:30:04 DEBUG[31493] channel.c: Scheduling timer at 160 sample intervals Aug 7 22:30:04 VERBOSE[31493] logger.c: -- Playing 'ss-noservice' (language 'en') Aug 7 22:30:04 DEBUG[31493] chan_zap.c: DTMF digit: 5 on Zap/1-1 Aug 7 22:30:07 DEBUG[31493] chan_zap.c: Exception on 20, channel 1 Aug 7 22:30:07 DEBUG[31493] chan_zap.c: Got event On hook(1) on channel 1 (index 0) Aug 7 22:30:07 DEBUG[31493] chan_zap.c: disabled echo cancellation on channel 1 Aug 7 22:30:07 DEBUG[31493] channel.c: Scheduling timer at 0 sample intervals Aug 7 22:30:07 DEBUG[31493] channel.c: Hanging up channel 'Zap/1-1' Aug 7 22:30:07 DEBUG[31493] chan_zap.c: zt_hangup(Zap/1-1) Aug 7 22:30:07 DEBUG[31493] chan_zap.c: Hangup: channel: 1 index = 0, normal = 20, callwait = -1, thirdcall = -1 Aug 7 22:30:07 DEBUG[31493] chan_zap.c: disabled echo cancellation on channel 1 Aug 7 22:30:07 DEBUG[31493] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/1-1 Aug 7 22:30:07 DEBUG[31493] chan_zap.c: Updated conferencing on 1, with 0 conference users Aug 7 22:30:07 VERBOSE[31493] logger.c: -- Hungup 'Zap/1-1' Aug 7 22:30:07 DEBUG[31478] devicestate.c: Changing state for Zap/1 - state 0 (Unknown) Aug 7 22:30:07 DEBUG[31495] app_queue.c: Device 'Zap/1' changed to state '0' (Unknown) but we don't care because they're not a member of any queue. Here are some settings from /etc/asterisk/zapata.conf: [trunkgroups] [channels] wink=300 rxwink=300 start=3000 context=default switchtype=national toneduration=100 usecallerid=no cidsignalling=dtmf hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=no rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no callprogress=no switchtype = national context = demo signalling = em_w group = 1 channel = 1-20 It has occurred to me that I could just set immediate=yes, read the incoming DTMF digits into a variable, and route to the appropriate extension. That seems more fragile to me since we could someday (when I'm not here) start getting more than 3 digits (caller id, for example). Plus I'd like to make it work the way it's *supposed* to. Any help/suggestions are appreciated! Cheers, -- Steve Linabery B94B C3C7 8A27 FF09 3C9D E992 5A20 2492 D5F5 EE51 This electronic message transmission contains information from the sender's organization that may be proprietary, confidential and/or privileged. The information is intended only for the use of the individual(s) or entity named above. If you are not the intended recipient, be aware that any disclosure, copying or distribution or use of the contents of this information is prohibited. If you have received this electronic transmission in error, please notify the sender immediately by replying to the address listed in the From: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
[asterisk-users] Tri-Link Technologies?
Anyone ever hear of a company...that isn't around anymore named Tri-Link Technologies? Apparently they had some system called a Vortex system...I have a few of the voip desk phones here and am trying to find some info on them to see if it is possible to reuse them for anything. Actually a descent looking phone...but can't really find any good info on the phones theirself... They were just called "Vortex PC Phone" Thanks, Don ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Autoreply: [asterisk-users] Deployment for less than 10 phones
Attualmente non sono in sede. Per richieste urgenti contattare lo 800 919299 o inviare una mail a [EMAIL PROTECTED] oppure a [EMAIL PROTECTED] Cordiali Saluti Giuseppe Parlato Area Network mailto:[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
Autoreply: [asterisk-users] em wink, TE110P, * answers too soon
Attualmente non sono in sede. Per richieste urgenti contattare lo 800 919299 o inviare una mail a [EMAIL PROTECTED] oppure a [EMAIL PROTECTED] Cordiali Saluti Giuseppe Parlato Area Network mailto:[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
[asterisk-users] How to, astcc and virtuemart
Hi, i'm trying to setup virtuemart with astcc (which is already working ok), i've seen messages from JP Carballo and he has done that, i would like to have a little help please. thanks. Julio Caceres _ Visita MSN Latino Entretenimiento: ¡música, cine, chismes, TV y más...! http://latino.msn.com/entretenimiento/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Autoreply: [asterisk-users] Tri-Link Technologies?
Attualmente non sono in sede. Per richieste urgenti contattare lo 800 919299 o inviare una mail a [EMAIL PROTECTED] oppure a [EMAIL PROTECTED] Cordiali Saluti Giuseppe Parlato Area Network mailto:[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
[asterisk-users] Snom MWI
Anyone having issues with the message waiting indicator and retrieve button on SNOM 320's and 360's. [EMAIL PROTECTED] ast]# asterisk -rx show version Asterisk 1.2.10 built by root @ myhost on a i686 running Linux on 2006-07-24 23:42:12 UTC Verbosity is at least 10 Some users get calls and when they hit the retrieve button, nothing comes through, yet for others it does. For yet others, they press retrieve, the MWI light goes off and they have to hit retrieve again to get to voicemail. Also... I have a sidecar on a SNOM 360, I have extensions programmed on it, yet the only lines that flash are the lines where the phones are offline. Any thoughts on this? On the sidecar SNOM, its running the latest firmware. ___ --Bandwidth 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: Autoreply: [asterisk-users] Tri-Link Technologies?
Dude I am English - Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, August 09, 2006 12:56 PM Subject: Autoreply: [asterisk-users] Tri-Link Technologies? Attualmente non sono in sede. Per richieste urgenti contattare lo 800 919299 o inviare una mail a [EMAIL PROTECTED] oppure a [EMAIL PROTECTED] Cordiali Saluti Giuseppe Parlato Area Network mailto:[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 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.405 / Virus Database: 268.10.8/414 - Release Date: 8/9/2006 ___ --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
Autoreply: [asterisk-users] How to, astcc and virtuemart
Attualmente non sono in sede. Per richieste urgenti contattare lo 800 919299 o inviare una mail a [EMAIL PROTECTED] oppure a [EMAIL PROTECTED] Cordiali Saluti Giuseppe Parlato Area Network mailto:[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
Autoreply: [asterisk-users] Snom MWI
Attualmente non sono in sede. Per richieste urgenti contattare lo 800 919299 o inviare una mail a [EMAIL PROTECTED] oppure a [EMAIL PROTECTED] Cordiali Saluti Giuseppe Parlato Area Network mailto:[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
Autoreply: Re: Autoreply: [asterisk-users] Tri-Link Technologies?
Attualmente non sono in sede. Per richieste urgenti contattare lo 800 919299 o inviare una mail a [EMAIL PROTECTED] oppure a [EMAIL PROTECTED] Cordiali Saluti Giuseppe Parlato Area Network mailto:[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] Snom MWI
Anyone having issues with the message waiting indicator and retrieve button on SNOM 320's and 360's. When the MWI light is not lit on a 360, and the user hits the voicemail button, the Snom phone dials the extension 'unknown', 'default' or 'asterisk'. If you don't have an unknown etc extension in your dialplan your get Forbidden. To cover this, you need these lines in a context that is accessible to all the phones: exten = default,1,VoicemailMain() exten = asterisk,1,VoicemailMain() exten = unknown,1,VoicemailMain() exten = Unknown,1,VoicemailMain() Also in the Snom setup web page you have to map whatever keystroke you use for voicemail in the line setup. For example, in Line 1, under the definition for Voicemail, you put in *98 or whatever you normally use to access voicemail *without* a voicemail key (*98 is standard in North America and is I believe the default for a Trixbox-style distribution) Also... I have a sidecar on a SNOM 360, I have extensions programmed on it, yet the only lines that flash are the lines where the phones are offline. Ensure the key type for your like mapping is set to type: Destination and the extension number of the phone is set in the Number field. Once you submit the form on the setup webpage, the phone should expand the entry to something like: sip:[EMAIL PROTECTED];user=phone Also you have to make sure your hints in the dialplan are setup correctly. Hints for the sidecar are a bit tricky; I have found that it works best to have the 'hint' priority directly underneath the definition for the extension: exten = ,1,Dial(SIP/,25) exten = ,2,Voicemail() exten = ,hint,SIP/ hth ___ --Bandwidth 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: Autoreply: [asterisk-users] Tri-Link Technologies?
It's a horrible, horrible autonotice that this person is unavailable. Expect to see lots of these.To contribute to the topic, I also can't find much on this phone ;)Dave On 8/9/06, Don [EMAIL PROTECTED] wrote: Dude I am English- Original Message -From: [EMAIL PROTECTED]To: asterisk-users@lists.digium.com Sent: Wednesday, August 09, 2006 12:56 PMSubject: Autoreply: [asterisk-users] Tri-Link Technologies?Attualmente non sono in sede. Perrichieste urgenti contattare lo 800919299 o inviare una mail a [EMAIL PROTECTED] oppure a [EMAIL PROTECTED].Cordiali SalutiGiuseppe ParlatoArea Networkmailto:[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--No virus found in this incoming message.Checked by AVG Free Edition. Version: 7.1.405 / Virus Database: 268.10.8/414 - Release Date: 8/9/2006___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Integrating Asterisk with a Panasonic D500 using MFC/R2
I am trying to get my Asterisk server to talk to a Panasonic D500 PBX using an E1 connection. The card for the Panasonic uses MFC/R2 and I have installed Unicall. Calls from the Asterisk server to the Panasonic go through without a hitch and I can call any extension I want. The problem is that I cannot get any calls from the Panasonic. I have the following log from a call: Aug 9 00:27:09 WARNING[14641] chan_unicall.c: Unicall/1 event Detected Aug 9 00:27:10 WARNING[14641] chan_unicall.c: MFC/R2 UniCall/1 - 3 on [2/ 2/Seize ack /Seize ack] Aug 9 00:27:10 WARNING[14641] chan_unicall.c: MFC/R2 UniCall/1 1 on - [2/ 2/Seize ack /Seize ack] Aug 9 00:27:10 WARNING[14641] chan_unicall.c: MFC/R2 UniCall/1 - 3 off [2/ 2/Group A /DNIS request ] Aug 9 00:27:10 WARNING[14641] chan_unicall.c: MFC/R2 UniCall/1 1 off - [2/ 2/Group A /DNIS request ] Aug 9 00:27:11 WARNING[14641] chan_unicall.c: MFC/R2 UniCall/1 - 3 on [2/ 2/Group A /DNIS request ] Aug 9 00:27:11 WARNING[14641] chan_unicall.c: MFC/R2 UniCall/1 1 on - [2/ 2/Group A /DNIS request ] Aug 9 00:27:11 WARNING[14641] chan_unicall.c: MFC/R2 UniCall/1 - 3 off [2/ 2/Group A /DNIS request ] Aug 9 00:27:11 WARNING[14641] chan_unicall.c: MFC/R2 UniCall/1 1 off - [2/ 2/Group A /DNIS request ] Aug 9 00:27:11 WARNING[14641] chan_unicall.c: MFC/R2 UniCall/1 - 3 on [2/ 2/Group A /DNIS request ] Aug 9 00:27:11 WARNING[14641] chan_unicall.c: MFC/R2 UniCall/1 1 on - [2/ 2/Group A /DNIS request ] Aug 9 00:27:12 WARNING[14641] chan_unicall.c: MFC/R2 UniCall/1 - 3 off [2/ 2/Group A /DNIS request ] Aug 9 00:27:12 WARNING[14641] chan_unicall.c: MFC/R2 UniCall/1 1 off - [2/ 2/Group A /DNIS request ] Aug 9 00:27:12 WARNING[14641] chan_unicall.c: MFC/R2 UniCall/1 - 3 on [2/ 2/Group A /DNIS request ] Aug 9 00:27:12 WARNING[14641] chan_unicall.c: MFC/R2 UniCall/1 6 on - [2/ 2/Group A /DNIS request ] Aug 9 00:27:12 WARNING[14641] chan_unicall.c: MFC/R2 UniCall/1 - 3 off [2/ 2/Group C /Category req ] Aug 9 00:27:12 WARNING[14641] chan_unicall.c: MFC/R2 UniCall/1 6 off - [2/ 2/Group C /Category req ] Aug 9 00:27:13 WARNING[14641] chan_unicall.c: MFC/R2 UniCall/1 - 6 on [2/ 2/Group C /Category req ] Aug 9 00:27:13 WARNING[14641] chan_unicall.c: MFC/R2 UniCall/1 1 on - [2/ 2/Group C /Category req ] Aug 9 00:27:13 WARNING[14641] chan_unicall.c: MFC/R2 UniCall/1 - 6 off [2/ 2/Group C /ANI request ] Aug 9 00:27:13 WARNING[14641] chan_unicall.c: MFC/R2 UniCall/1 1 off - [2/ 2/Group C /ANI request ] Aug 9 00:27:13 WARNING[14641] chan_unicall.c: MFC/R2 UniCall/1 - B on [2/ 2/Group C /ANI request ] Aug 9 00:27:13 WARNING[14641] chan_unicall.c: MFC/R2 UniCall/1 R2 prot. err. [2/ 2/Group C /ANI request ] cause 32772 - Unexpected MF6 signal Aug 9 00:27:13 WARNING[14641] chan_unicall.c: MFC/R2 UniCall/1 1001 - [1/ 1/Idle /Idle ] Aug 9 00:27:13 WARNING[14641] chan_unicall.c: Unicall/1 event Protocol failure According to the Panasonic tech person the problem is that Asterisk insists on getting the Group C request for a callerid. I tried to disable callerid in unicall.conf but the result was always the same. I have mx,10,4 so after I get the first four digits I get the protocol error. I know many people have integrated with PRI, but the card on the panasonic does not support that. -- Carlos Chavez Prats Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 signature.asc Description: This is a digitally signed message part ___ --Bandwidth 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: Autoreply: [asterisk-users] Tri-Link Technologies?
It says on the bottom for use only with vortex system...but...I was just hoping to possibly find a way to use it for something else since the company is long gone and no way to contact them. - Original Message - From: David Freeman To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, August 09, 2006 1:37 PM Subject: Re: Autoreply: [asterisk-users] Tri-Link Technologies? It's a horrible, horrible autonotice that this person is unavailable. Expect to see lots of these.To contribute to the topic, I also can't find much on this phone ;)Dave On 8/9/06, Don [EMAIL PROTECTED] wrote: Dude I am English- Original Message -From: [EMAIL PROTECTED]To: asterisk-users@lists.digium.com Sent: Wednesday, August 09, 2006 12:56 PMSubject: Autoreply: [asterisk-users] Tri-Link Technologies?Attualmente non sono in sede. Perrichieste urgenti contattare lo 800919299 o inviare una mail a [EMAIL PROTECTED] oppure a [EMAIL PROTECTED].Cordiali SalutiGiuseppe ParlatoArea Networkmailto:[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--No virus found in this incoming message.Checked by AVG Free Edition. Version: 7.1.405 / Virus Database: 268.10.8/414 - Release Date: 8/9/2006___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message.Checked by AVG Free Edition.Version: 7.1.405 / Virus Database: 268.10.8/414 - Release Date: 8/9/2006 ___ --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] RE: Ever donate Software to Digium? If you did your afool.
I really don't understand the complaint. Fonality gets a $5 mil. investment for building its own system on top of Asterisk -- no complaint. But Mark Co. can't get VC for their own business / enterprise / support architecture? Everything that we-all is contributing is part of the open source Asterisk system, which Mark dreamed up and which continues to be an unbelievably exciting piece of software. One day soon I hope programmers who work for me will contribute to the code base. The fact that Digium just got $13.7 mil. to keep doing what they're doing merely ensures that we who use the open-source base will all be benefiting from a stable and growing team of professionals pushing the project forward. Congrats, Digium. I for one am delighted, and look forward to joining the fools. Yaakov Menken ___ --Bandwidth 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] High Availability
Hey guys, I'm currently investigating solutions about High Availability solution, I've found out about this webpage on voip-info.org: http://www.voip-info.org/wiki/view/Asterisk+High+Availability+Solutions But that's cool for the voice and stuff. But what about the recording. If I don't want to put too much load on the main servers that will process the calls, is there a way that we can setup 2 extra servers, that will handle the call recording ? I'm guessing that call recording might generate a lot of work and i/o on the disk. But not sure, since I've never done that before and not even sure if its a good idea to transmit that over the network ? Anyway, if anyone has any other link about High Availability Solutions, that would be great! Thanks, ___ --Bandwidth 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] DTMF codes in feature.conf not comming through
I'm running Asterisk 1.2.7.1 using entirely SIP connections, but I have a problem with DTMF signaling. In the features.conf, I have set up sequences using * and # followed by a single digit for transfers etc. But when I then press '*' or '#' during a call, only each other is passed on. All other DTMF signals are working great. Is there a way to guarantee that single '*' and '#' are passed on (respecting a featuredigittimeout )? And is there a way do make a call NOT using the featuremap and therefore grapping the DTMF tones? On a call-to-call basis or for a specific SIP client? Regards, Henrik My feature map: [general] parkext = 700 ; What ext. to dial to park parkpos = 701-720 ; What extensions to park calls on context = parkedcalls ; Which context parked calls are in parkingtime = 20 ; Number of seconds a call can be parked for ; (default is 45 seconds) transferdigittimeout = 8 ; Number of seconds to wait between digits when transfering a call featuredigittimeout = 500 ; Max time (ms) between digits for ; feature activation. Default is 500 courtesytone = beep ; Sound file to play to the parked caller ; when someone dials a parked call adsipark = yes ; if you want ADSI parking announcements pickupexten = *8; Configure the pickup extension. Default is *8 xfersound = beep ; to indicate an attended transfer is complete xferfailsound = beeperr; to indicate a failed transfer [featuremap] blindxfer = #1; Blind transfer disconnect = *0 ; Disconnect automon = *3 ; One Touch Record atxfer = *7 ; Attended transfer ___ --Bandwidth 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] BriStuff | HFC-S | Progress() | Early B3 on incoming calls from PSTN
Hi folks, I'm currently trying to get some early audio, i.e. audio without a connection, to the caller to give some cost-free info while the alerting phase. The WIKI's info on the Progress() application says, that just Progress() before e.g. a Background(soundfile|n) should work but it doesn't. In fact, the call times out (The person you've called is temporarily not available) if there is no ringing indication for longer than 30 secs or so (long sound). As soon as the Dial application is called, the caller gets a normal ringing indication. Dial(...|m) causes the caller to hear nothing and the call to timeout. So, I guess, in addition to Progress(), * has to tell the PSTN that the call is proceeding AND that there is audio, i.e. inband information, available and it should be passed to the caller. I guess, the focus should be on the progress indication as the call timeout shows, that the PSTN isn't satisfied with just what Progress() sends. Does anyone got this Early-B3 working with BriStuff and HFC-S? Regards, Stefan -- (o_ Stefan Gofferje| SCLT, MCP, CCSA //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler Koch - the original point and click interface Asterisk 1.2.9.1-BRIstuffed-0.3.0-PRE-1r | chan_sccp ___ --Bandwidth 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] HFC-S Cards in the UK
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Having spent some time chasing up the importers today, it appears that there is no real shortage of the cards yet. However they are seen a legacy card now with people switching to broadband. MRi claim to have 100-200 cards available, and their distributors are only ordering when they get orders. Dynamode seem to have dropped the PCI ISDN card. No idea on Billion, we don't usually use their distributors. Sitecom are moving office at the moment and are not even answering the phone! So while there HFC cards out there, it seems that they are going to get harder to find. - -- Ron Wellsted [EMAIL PROTECTED] http://www.wellsted.org.uk N 52.567623, W 2.137621 Linux Counter No. 202120 FWD:519961 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2.2 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iQEVAwUBRNoum0tP/KMNOfRbAQIghgf7Br3VITjABhoF2ogSMAr+Nx+1GU3eE3op ioPifC+uDGHyxH+q1gYYneS+/yRl1ILWYSkcuIZl1HGhbMh12cbzbSTCHeDN+l+u 6yZaflZxYWaZE/286PY0NvfHwc8O/fW4Hnzu4LzFoxdOr/JN7nHgb0P/28/16nMI W1tsGnCEzvy+tD05CWrwViA4xZY7+6Qt1Ry0aZC4Epl3pIkTxUdGWyDhYelBz9SP acm4nVShd5LacwQZ/HRmO2+fpvSI8coe2WXGdLXjjaMJU+HuA7UFBco9YlnxcOcK jyWx8XZLWzho45Qv2oTZR6X+IpzfbZ0yPYHTBrZmfOJ+ZjEfS+knEw== =ycXc -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] can Digium FXS channels support been half mile to 1 mile length away from phone?
Hi all, I need to setup 6 phones about 3/4 of a mile from the main box, (can't do it with VoIP yet because of networking issues), does anyone knows if the boards can resist such a length for FXS ports. Right now there is a Dialogic MSI160 working fine. The actual length in a straight line is about half a mile, it does have a single cable point to point, but can't be sure where the cable goes since it has to pass from one building to another, so it might end up been quite a bit more distance. If the answer is they can't handle such a distance does anyone knows if there is an alternative, (line booster or whatever similar ) thanks ___ --Bandwidth 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] Integrating Asterisk with a Panasonic D500 using MFC/R2
I had a similar problem with a Siemens, most probably you are specifying the wrong number of expected ANI digits. Try with mx,0,4 as protocolvariant, that will tell Unicall to expect 0 ANI digits, but of course, in Asterisk environment you wont be able to get callerid. Play around incrementing the ANI expected digits in protocolvariant. Regards On 8/9/06, Carlos Chavez [EMAIL PROTECTED] wrote: I am trying to get my Asterisk server to talk to a Panasonic D500 PBX using an E1 connection. The card for the Panasonic uses MFC/R2 and I have installed Unicall. Calls from the Asterisk server to the Panasonic go through without a hitch and I can call any extension I want. The problem is that I cannot get any calls from the Panasonic. I have the following log from a call: Aug 9 00:27:09 WARNING[14641] chan_unicall.c: Unicall/1 event Detected Aug 9 00:27:10 WARNING[14641] chan_unicall.c: MFC/R2 UniCall/1 - 3 on [2/ 2/Seize ack /Seize ack] Aug 9 00:27:10 WARNING[14641] chan_unicall.c: MFC/R2 UniCall/1 1 on - [2/ 2/Seize ack /Seize ack] Aug 9 00:27:10 WARNING[14641] chan_unicall.c: MFC/R2 UniCall/1 - 3 off [2/ 2/Group A /DNIS request ] Aug 9 00:27:10 WARNING[14641] chan_unicall.c: MFC/R2 UniCall/1 1 off - [2/ 2/Group A /DNIS request ] Aug 9 00:27:11 WARNING[14641] chan_unicall.c: MFC/R2 UniCall/1 - 3 on [2/ 2/Group A /DNIS request ] Aug 9 00:27:11 WARNING[14641] chan_unicall.c: MFC/R2 UniCall/1 1 on - [2/ 2/Group A /DNIS request ] Aug 9 00:27:11 WARNING[14641] chan_unicall.c: MFC/R2 UniCall/1 - 3 off [2/ 2/Group A /DNIS request ] Aug 9 00:27:11 WARNING[14641] chan_unicall.c: MFC/R2 UniCall/1 1 off - [2/ 2/Group A /DNIS request ] Aug 9 00:27:11 WARNING[14641] chan_unicall.c: MFC/R2 UniCall/1 - 3 on [2/ 2/Group A /DNIS request ] Aug 9 00:27:11 WARNING[14641] chan_unicall.c: MFC/R2 UniCall/1 1 on - [2/ 2/Group A /DNIS request ] Aug 9 00:27:12 WARNING[14641] chan_unicall.c: MFC/R2 UniCall/1 - 3 off [2/ 2/Group A /DNIS request ] Aug 9 00:27:12 WARNING[14641] chan_unicall.c: MFC/R2 UniCall/1 1 off - [2/ 2/Group A /DNIS request ] Aug 9 00:27:12 WARNING[14641] chan_unicall.c: MFC/R2 UniCall/1 - 3 on [2/ 2/Group A /DNIS request ] Aug 9 00:27:12 WARNING[14641] chan_unicall.c: MFC/R2 UniCall/1 6 on - [2/ 2/Group A /DNIS request ] Aug 9 00:27:12 WARNING[14641] chan_unicall.c: MFC/R2 UniCall/1 - 3 off [2/ 2/Group C /Category req ] Aug 9 00:27:12 WARNING[14641] chan_unicall.c: MFC/R2 UniCall/1 6 off - [2/ 2/Group C /Category req ] Aug 9 00:27:13 WARNING[14641] chan_unicall.c: MFC/R2 UniCall/1 - 6 on [2/ 2/Group C /Category req ] Aug 9 00:27:13 WARNING[14641] chan_unicall.c: MFC/R2 UniCall/1 1 on - [2/ 2/Group C /Category req ] Aug 9 00:27:13 WARNING[14641] chan_unicall.c: MFC/R2 UniCall/1 - 6 off [2/ 2/Group C /ANI request ] Aug 9 00:27:13 WARNING[14641] chan_unicall.c: MFC/R2 UniCall/1 1 off - [2/ 2/Group C /ANI request ] Aug 9 00:27:13 WARNING[14641] chan_unicall.c: MFC/R2 UniCall/1 - B on [2/ 2/Group C /ANI request ] Aug 9 00:27:13 WARNING[14641] chan_unicall.c: MFC/R2 UniCall/1 R2 prot. err. [2/ 2/Group C /ANI request ] cause 32772 - Unexpected MF6 signal Aug 9 00:27:13 WARNING[14641] chan_unicall.c: MFC/R2 UniCall/1 1001 - [1/ 1/Idle /Idle ] Aug 9 00:27:13 WARNING[14641] chan_unicall.c: Unicall/1 event Protocol failure According to the Panasonic tech person the problem is that Asterisk insists on getting the Group C request for a callerid. I tried to disable callerid in unicall.conf but the result was always the same. I have mx,10,4 so after I get the first four digits I get the protocol error. I know many people have integrated with PRI, but the card on the panasonic does not support that. -- Carlos Chavez Prats Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Sipura SPA-3000 vs Sangoma A200
I echo (pun intended) Rich's response. The Spa3k is ~ok~ but echo has always been a problem for my home office. The A200D works flawlessly. I'm looking to set up a home-office PBX/Asterisk lab using a VIA EPIA motherboard as an always on, low powered solution. I have seen an A200D in a soekris 4801 (http://www.soekris.com) box running astlinux. I say saw, because it was at a show and the box wasn't plugged in. It was Jim VanMeggelen - one of the authors of the O'Reilly Asterisk book. You might want to drop him a line. The Sangoma has a 4-pin molex for power supply connection to augment the PCI bus when you need to generate ring voltage for FXS ports. The soekris (by default) won't give you that so either you put FXS external or you figure out how to get +5/+12 VDC to the Sangoma. Actually, you may want to check with Sangoma ... maybe you only need 5 or 12 but they just match the molex to be compliant with all PC hardware. I am trying to find out the differences between a solution using an external ATA (like the Sipura SPA-3000) or an internal PCI card (like the Sangoma A200 with 2 FXO 2 FXS ports). The nice thing about the SPA3K is that upon registration failure or power failure the FXO FXS ports get hardwired together so you get a power safe environment. The nice thing about the Sangoma is that it supports ring contexts by distinctive ring. I believe this is also called Ident-a-call in many places. For a home office this is great. I have a second number that rings my primary line with a different ring pattern for ~ 4.00/mth. rather than the expense of a second line. I program that ring pattern into zapata.conf and push those calls directly to Zap/4 (my fax) and other calls to Zap/3 (my house), etc dbc. ___ --Bandwidth 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] Ignoring the # key on a call
I'm using Asterisk 1.2.7.1, and if I press the # key when I'm on a call, I get a prompt to transfer the call. This, of course, interferes with any IVR system I'm using, as many systems will ask me to enter a number, then press the # key. Is there any way I can get Asterisk to ignore this key when I'm on a call? Patrick ___ --Bandwidth 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] wildcard always busy
Hi guys,I am fighting to get a Wildcard TE405P working but it always start and put all channels in use. 14 TE4/0/1/14 Clear (In use) 15 TE4/0/1/15 Clear (In use) 16 TE4/0/1/16 HDLCFCS (In use) 17 TE4/0/1/17 Clear (In use)I've tried to downgrade zaptel and asterisk but it didn't solve the problem.Here is my zaptel.conf:span=1,0,1,ccs,hdb3,crc4,yellow #span=2,0,1,ccs,hdb3,crc4,yellow#span=3,0,1,ccs,hdb3,crc4,yellow#span=4,0,1,ccs,hdb3,crc4,yellowbchan=1-15,17-31dchan=16#bchan=32-46,48-62#dchan=47#bchan=63-77,79-93#dchan=78#bchan=94-108,110-124 #dchan=109loadzone = usloadzone = brdefaultzone = brAnd here my zapata.conf:[trunkgroups][channels]context=defaultswitchtype=nationalpridialplan=national rxwink=300 ; Atlas seems to use long (250ms) winksusecallerid=yeshidecallerid=nocallwaiting=yesusecallingpres=yescallwaitingcallerid=yesthreewaycalling=yestransfer=yescanpark=yes cancallforward=yescallreturn=yesechocancel=yesechocancelwhenbridged=yesechotraining=yesechotraining=400rxgain=0.0txgain=0.0group=1callgroup=1pickupgroup=1immediate=no ;group=1signalling=pri_cpechannel = 1-15,17-31;channel = 32-46,48-62,63-77,79-93;channel = 94-108,110-124I always got this error:chan_zap.c:8324 pri_dchannel: PRI got event: No more alarm (5) on Primary D-channel of span 1 And this when I try to make a call:chan_zap.c:2298 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel anyway!Asterisk is unable to make and receive calls from E1. Is there something wrong in the configuration? How can I put it to work?-- Ralph LiebessohnICQ: 74835911Skype: liebessohn ___ --Bandwidth 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] Integrating Asterisk with a Panasonic D500 using MFC/R2
Hi Carlos,I had the same problem and spent a lot of time studying the MFC/R2 protocol but the problem is in the libmfcr2 package version!!Try using the packages in: http://www.soft-switch.org/downloads/unicall/unicall-0.0.3pre7And not in pre9.Both pre7 and pre9 have libmfcr2-0.0.3.tar.gz with the same name.. but in different sizes. They are different.After that everything started do work !!Good Luck I am trying to get my Asterisk server to talk to a Panasonic D500 PBXusing an E1 connection. The card for the Panasonic uses MFC/R2 and Ihave installed Unicall. Calls from the Asterisk server to the Panasonic go through without a hitch and I can call any extension I want. Theproblem is that I cannot get any calls from the Panasonic. I have thefollowing log from a call: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] can Digium FXS channels support been half mile to 1 mile length away from phone?
Yep, there are analogue line boosters for your requirement. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Manrique Feoli Sent: Wednesday, 9 August 2006 3:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] can Digium FXS channels support been half mile to 1 mile length away from phone? Hi all, I need to setup 6 phones about 3/4 of a mile from the main box, (can't do it with VoIP yet because of networking issues), does anyone knows if the boards can resist such a length for FXS ports. Right now there is a Dialogic MSI160 working fine. The actual length in a straight line is about half a mile, it does have a single cable point to point, but can't be sure where the cable goes since it has to pass from one building to another, so it might end up been quite a bit more distance. If the answer is they can't handle such a distance does anyone knows if there is an alternative, (line booster or whatever similar ) thanks ___ --Bandwidth 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: Autoreply: [asterisk-users] Snom MWI
[EMAIL PROTECTED] wrote: Attualmente non sono in sede. Per richieste urgenti contattare lo 800 919299 o inviare una mail a [EMAIL PROTECTED] oppure a [EMAIL PROTECTED] Cordiali Saluti Giuseppe Parlato Area Network mailto:[EMAIL PROTECTED] PLONK! Regards, Austin. signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Autoreply: [asterisk-users] Snom MWI
Austin Denyer wrote: PLONK! Regards, Austin. and you sent this to a public list? you're a fucking idiot. -A ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] BriStuff | HFC-S | Progress() | Early B3 on incoming calls from PSTN
Hi, On Wed, 2006-08-09 at 20:50 +0200, Stefan Gofferje wrote: Hi folks, I'm currently trying to get some early audio, i.e. audio without a connection, to the caller to give some cost-free info while the alerting phase. Many (if not all) telco does not allow sending inband audio to the caller if not connected. So your caller will always get the ringtone, that's generated by the telco. Or they won't be able to sell toll free numbers :) Some telcos allow sending inband audio for few seconds on the PRI, but depends on the agreements you made. matteo. ___ --Bandwidth 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: Autoreply: [asterisk-users] Snom MWI
Austin Denyer wrote: [EMAIL PROTECTED] wrote: Attualmente non sono in sede. Per richieste urgenti contattare lo 800 919299 o inviare una mail a [EMAIL PROTECTED] oppure a [EMAIL PROTECTED] Cordiali Saluti Giuseppe Parlato Area Network mailto:[EMAIL PROTECTED] PLONK! Regards, Austin. I'd like to apologize profusely to anyone offended by my previous post. It was not intended to go to Asterisk-Users but to Austin in private. I will graciously accept any and all flames for my unacceptably poor behavior. Andrew ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] can Digium FXS channels support been half mile to 1 mile length away from phone?
Manrique Feoli wrote: Hi all, I need to setup 6 phones about 3/4 of a mile from the main box, (can't do it with VoIP yet because of networking issues), does anyone knows if the boards can resist such a length for FXS ports. Right now there is a Dialogic MSI160 working fine. The actual length in a straight line is about half a mile, it does have a single cable point to point, but can't be sure where the cable goes since it has to pass from one building to another, so it might end up been quite a bit more distance. If the answer is they can't handle such a distance does anyone knows if there is an alternative, (line booster or whatever similar ) With the exception of the ring generator on the digium card, the fxs port specs are basically the same as telephone company specs. Telco lines in rural areas often times exceed six miles. So, yes the digium card will handle your half mile just fine. The ring generator on the card is much smaller then those used in the telco, and is likely limited to maybe two or three ringers at the distant location. Don't plan on attaching multiple phones to the fxs port at your remote location. (Same is basically true with the sangoma card.) ___ --Bandwidth 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: Re: [asterisk-users] Force peer source ip
On 7/25/06, Leo Ann Boon [EMAIL PROTECTED] wrote: What is your net mask? 255.255.255.0? You can try in sip.conf: externip=212.xxx.xxx.xxx localnet=192.168.0.0/255.255.0.0 A bit late answer, but I haven't got around to test it until now. No, this doesn't work. Maybe it's my Asterisk (1.0.x)? -- regards, Robin ___ --Bandwidth 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] Ignoring the # key on a call
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 patrick wrote: I'm using Asterisk 1.2.7.1, and if I press the # key when I'm on a call, I get a prompt to transfer the call. This, of course, interferes with any IVR system I'm using, as many systems will ask me to enter a number, then press the # key. Is there any way I can get Asterisk to ignore this key when I'm on a call? Have a look inside features.conf, where you can change the hash transfer (or pound) to ## or whatever you'd like. - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFE2j5gS6d5vy0jeVcRAsORAJwKfICCPAfBK5UQtPnd+un44wHFZgCfbZJh GEByq1FUGXXl6xw+CG3NLMo= =wrcV -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ignoring the # key on a call
You can set the transfer feature to be whatever key press you want. If you want to go that route, you can set it up so that # doesn't initiate a transfer, and Asterisk will just pass the # as DTMF to the other side of the call. My transfer is *#. Use features.conf to set this up.AlexOn 8/9/06, patrick [EMAIL PROTECTED] wrote: I'm using Asterisk 1.2.7.1, and if I press the # key when I'm on a call, I get a prompt to transfer the call. This, of course, interfereswith any IVR system I'm using, as many systems will ask me to enter anumber, then press the # key. Is there any way I can get Asterisk to ignore this key when I'm on a call?Patrick___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar[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: Autoreply: [asterisk-users] Snom MWI
Andrew D Kirch wrote: Austin Denyer wrote: [EMAIL PROTECTED] wrote: Attualmente non sono in sede. Per richieste urgenti contattare lo 800 919299 o inviare una mail a [EMAIL PROTECTED] oppure a [EMAIL PROTECTED] Cordiali Saluti Giuseppe Parlato Area Network mailto:[EMAIL PROTECTED] PLONK! Regards, Austin. I'd like to apologize profusely to anyone offended by my previous post. It was not intended to go to Asterisk-Users but to Austin in private. I will graciously accept any and all flames for my unacceptably poor behavior. That's your only Oooopps for the day. ;) ___ --Bandwidth 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: Ever donate Software to Digium? If you did your afool.
On 8/9/06, Yaakov Menken [EMAIL PROTECTED] wrote: I really don't understand the complaint. Fonality gets a $5 mil. investment for building its own system on top of Asterisk -- no complaint. But Mark Co. can't get VC for their own business / enterprise / support architecture? Everything that we-all is contributing is part of the open source Asterisk system, which Mark dreamed up and which continues to be an unbelievably exciting piece of software. One day soon I hope programmers who work for me will contribute to the code base. The fact that Digium just got $13.7 mil. to keep doing what they're doing merely ensures that we who use the open-source base will all be benefiting from a stable and growing team of professionals pushing the project forward. Congrats, Digium. I for one am delighted, and look forward to joining the fools. The original post must be the biggest troll in history. I think all of us are thankful for having Asterisk. Infact, I'm glad that Digium is getting good $$ out of it (apart from all the hardware sales) since many of us can't afford to pay Mark the sort of money he deserves for his time spent writing Asterisk. I know Asterisk has earned me a few bucks from the various installations I've done, so I'm happy to hear this bit of news. Of course, Asterisk is a community-supported project so my thanks also goes to all the people that have made it what it is today. Maybe one day soon I will be able to contribute to it (apart from a few Wiki additions I've done). Cheers, Gonzalo ___ --Bandwidth 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: Ever donate Software to Digium? If you didyour afool.
There is no doubt Asterisk is a nice peiece of software...and it does a nice job on our prepaid/postpaid apps... But it will need to evolve even more quickly to keep up with what freeswitch will have within the next few months...unless they plan to stay mainly under the idea of relatively small business use. Thread issues are virtually non existant in freeswitch. And Digium has fallen short with their hardware where Sangoma has picked up the ball and ran with it. I am not trashing Digium by the way...I just see a lot of changes they will need to make for the future if they want to stay on top of that which they have basically invented. Sangoma for instance...their support seems phenominal...Digium's on the other hand...I have talked to people they that seem to know less about asterisk than I do...and that just shouldn't be the case. It happens everywhere though...applications, games, automobiles, etc...Take an example of the gaming industry verant basically put MMORPG games on the map...but they didn't evolve and Everquest and Everquest 2 fell to the wayside of newer more creative ideas. It's hard to stay on top in the computer world. - Original Message - From: Gonzalo Servat [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, August 09, 2006 4:26 PM Subject: [asterisk-users] Re: Ever donate Software to Digium? If you didyour afool. On 8/9/06, Yaakov Menken [EMAIL PROTECTED] wrote: I really don't understand the complaint. Fonality gets a $5 mil. investment for building its own system on top of Asterisk -- no complaint. But Mark Co. can't get VC for their own business / enterprise / support architecture? Everything that we-all is contributing is part of the open source Asterisk system, which Mark dreamed up and which continues to be an unbelievably exciting piece of software. One day soon I hope programmers who work for me will contribute to the code base. The fact that Digium just got $13.7 mil. to keep doing what they're doing merely ensures that we who use the open-source base will all be benefiting from a stable and growing team of professionals pushing the project forward. Congrats, Digium. I for one am delighted, and look forward to joining the fools. The original post must be the biggest troll in history. I think all of us are thankful for having Asterisk. Infact, I'm glad that Digium is getting good $$ out of it (apart from all the hardware sales) since many of us can't afford to pay Mark the sort of money he deserves for his time spent writing Asterisk. I know Asterisk has earned me a few bucks from the various installations I've done, so I'm happy to hear this bit of news. Of course, Asterisk is a community-supported project so my thanks also goes to all the people that have made it what it is today. Maybe one day soon I will be able to contribute to it (apart from a few Wiki additions I've done). Cheers, Gonzalo ___ --Bandwidth 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: 268.10.8/414 - Release Date: 8/9/2006 ___ --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] Mac Address Authentication Methods
Hello all ! I am looking for a way to use an ata or ipphone mac address as a part of the sip registration. I've tried using arp tables, but it only works on local networks not thru the internet.. I know that vonage uses some kind of mac auth, but I think that this is an special feature of their equipments ain't it? Have anyone accomplished this? Any ideas would be very very appreciated Thanks Daniel Botelho P. Moraes Email/Msn: [EMAIL PROTECTED] Tel.:+55-11-8183-8896 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] can Digium FXS channels support been half mile to 1 mile length away from phone?
Thanks Rich, I was expecting that. I was worried because on other type of cards if you set the phone too far you'll burn the port of the card, mainly because of the lack of capacity to keep such a long line up. Now when you say 2 or three ringers, you mean 2 or three ring events or phones ringing at once or what did you mean? I need 6 phones now, and not expecting more than 2 ringing at once. (hopefully) cheers Manrique Rich Adamson escribió: Manrique Feoli wrote: Hi all, I need to setup 6 phones about 3/4 of a mile from the main box, (can't do it with VoIP yet because of networking issues), does anyone knows if the boards can resist such a length for FXS ports. Right now there is a Dialogic MSI160 working fine. The actual length in a straight line is about half a mile, it does have a single cable point to point, but can't be sure where the cable goes since it has to pass from one building to another, so it might end up been quite a bit more distance. If the answer is they can't handle such a distance does anyone knows if there is an alternative, (line booster or whatever similar ) With the exception of the ring generator on the digium card, the fxs port specs are basically the same as telephone company specs. Telco lines in rural areas often times exceed six miles. So, yes the digium card will handle your half mile just fine. The ring generator on the card is much smaller then those used in the telco, and is likely limited to maybe two or three ringers at the distant location. Don't plan on attaching multiple phones to the fxs port at your remote location. (Same is basically true with the sangoma card.) ___ --Bandwidth 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] Jitterbuffer on SIP
Thank You Patrick,After some minor problems in some file paths I had success compiling.The only problem was the codec_g726 witch does an illegal call and Asterisk doesn´t come up. But I only use g729 or g723 so I´ve deleted it from the modules directory and asterisk came up. I´m going to test it now. Thanks again!!On Tue, 2006-08-08 at 10:45 -0300, Thierry Querette wrote: Hi, Is that a way to patch a running asterisk 1.2.9.1 instalation with the experimental SIP Jitterbuffer support ?Yes, see http://www.asterisk-backports.orghttp://asterisk-backports.org/downloads/ast_jb-1.2.9.1+rtp-keep-jb+fax +g726.patchThe jb seems to work fine on my setup (with low usage).Regards,Patrick ___ --Bandwidth 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] HFC-S Cards in the UK
Ron Wellsted [EMAIL PROTECTED] writes: So while there HFC cards out there, it seems that they are going to get harder to find. We got a few of these from Conrad. They are in Germany and I am not sure if this one is the same as ours. But at EUR 24.95 per card you cannot loose to much. http://www.conrad.de/goto.php?artikel=955078 hth, Wolfgang ___ --Bandwidth 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: Snom MWI
CA == Colin Anderson [EMAIL PROTECTED] writes: Anyone having issues with the message waiting indicator and retrieve button on SNOM 320's and 360's. CA When the MWI light is not lit on a 360, and the user hits the CA voicemail button, the Snom phone dials the extension 'unknown', CA 'default' or 'asterisk'. Actually it calls whatever extension you have defined as the voice mail extension when provisioning it -- user_mailbox1 is the one for line 1. When the MWI light is lit, it instead dials whatever URL it was sent by asterisk in the message. /Benny ___ --Bandwidth 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] Jitterbuffer on SIP
Hi, same by me, the patch affects app_rxfax, app_txfax and the G.726 codec from the spandsp. However, it doesn't link it to the libspandsp properly, asterisk complains: undefined symbol: g726_encode. I added to modules.conf the line noload = codec_g726.so and asterisk comes up again. Thanks for a link to patch. Jan Fousek __ Od: [EMAIL PROTECTED] Komu: asterisk-users@lists.digium.com Datum: 09.08.2006 23:26 Předmět: [asterisk-users] Jitterbuffer on SIP Thank You Patrick, After some minor problems in some file paths I had success compiling. The only problem was the codec_g726 witch does an illegal call and Asterisk doesn´t come up. But I only use g729 or g723 so I´ve deleted it from the modules directory and asterisk came up. I´m going to test it now. Thanks again!! On Tue, 2006-08-08 at 10:45 -0300, Thierry Querette wrote: Hi, Is that a way to patch a running asterisk 1.2.9.1 instalation with the experimental SIP Jitterbuffer support ? Yes, see http://www.asterisk-backports.org http://asterisk-backports.org/downloads/ast_jb-1.2.9.1+rtp-keep-jb+fax +g726.patch The jb seems to work fine on my setup (with low usage). Regards, Patrick ___ --Bandwidth 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] Mac Address Authentication Methods
Daniel Botelho P. Moraes wrote: Hello all ! I am looking for a way to use an ata or ipphone mac address as a part of the sip registration. I've tried using arp tables, but it only works on local networks not thru the internet.. I know that vonage uses some kind of mac auth, but I think that this is an special feature of their equipments ain't it? Have anyone accomplished this? Any ideas would be very very appreciated The vonage stuff uses the mac address string because it is guaranteed to be unique, and that string is available for scripts within (at least some of) the ata's to use. Vonage does not have access to the mac address from a networking perspective, which is what you've already noted. If you wanted to come up with some sort of mass-produced ata style box that could register itself with servers and use a unique password, what would you use? (Probably the mac address combined with an additional string.) If you look at the sipura devices (and other devices with sipura software), you'll find the mac address is readily available and preloaded into a script variable for you to use. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] can Digium FXS channels support been half mile to 1 mile length away from phone?
Don't forget to do lightening and surge protection. On 8/9/06, Manrique Feoli [EMAIL PROTECTED] wrote: Thanks Rich, I was expecting that. I was worried because on other type of cards if you set the phone too far you'll burn the port of the card, mainly because of the lack of capacity to keep such a long line up. Now when you say 2 or three ringers, you mean 2 or three ring events or phones ringing at once or what did you mean? I need 6 phones now, and not expecting more than 2 ringing at once. (hopefully) cheers Manrique Rich Adamson escribió: Manrique Feoli wrote: Hi all, I need to setup 6 phones about 3/4 of a mile from the main box, (can't do it with VoIP yet because of networking issues), does anyone knows if the boards can resist such a length for FXS ports. Right now there is a Dialogic MSI160 working fine. The actual length in a straight line is about half a mile, it does have a single cable point to point, but can't be sure where the cable goes since it has to pass from one building to another, so it might end up been quite a bit more distance. If the answer is they can't handle such a distance does anyone knows if there is an alternative, (line booster or whatever similar ) With the exception of the ring generator on the digium card, the fxs port specs are basically the same as telephone company specs. Telco lines in rural areas often times exceed six miles. So, yes the digium card will handle your half mile just fine. The ring generator on the card is much smaller then those used in the telco, and is likely limited to maybe two or three ringers at the distant location. Don't plan on attaching multiple phones to the fxs port at your remote location. (Same is basically true with the sangoma card.) ___ --Bandwidth 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] Ever donate Software to Digium? If you did your a fool.
You stinking slefish pig, you use asterisk for free and this is what you say in return. How does it go (Cleaning out my closet): http://www.loglar.com/song.php?id=10620 look at the end of verse 3 thats for you. On 8/9/06, Randall H. [EMAIL PROTECTED] wrote: If you gave software to Digium then you helped Mark become very rich. http://abcnews.go.com/Technology/wireStory?id=2290152 ___ --Bandwidth 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: Sangoma A200D and DTMF Detection
Thanks Rana, We got a prompt response from Sangoma and gave them SSH access to troubleshoot the issue. Modifying relaxdtmf on its own did not help but if used in combination with rxgain it made all the difference. The system is working pretty good now with these two modifications: relaxdtmf=yes rxgain=6.0 Rana Dutt wrote: We experienced this problem with a Sangoma A104D card. With echo cancel turned on, the card was not detecting incoming DTMF digits to our IVR properly. However, when we added the line relaxdtmf=yes to zapata.conf, the problem went away. If the relaxdtmf setting is not curing the problem for you, I would suggest sending an email to [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]. Rana Dutt Softel Solutions rdutt at softelinc dot 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 -- Andres ___ --Bandwidth 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 many digits are collected
When Background or Playback is used in a dial plan how many digits are collected and what variable are they returned in? I'm trying to do a simple auto attendant and having very little luck -- One day at a time, one second if that's what it takes ___ --Bandwidth 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 many digits are collected
When you use playback no digits are collected. When you use backround
the digits go to an available extension in that context, for example:
[ivrcontext]
exten = s,1,Background(testfile)
exten = _X,1,Noop(user pressed ${EXTEN})
this will gotot Noop for any single digit that is pressed, you will be
able to see in the CLI what they pressed.
app_read allows you to have it in a variable.
On 8/9/06, Bruce Ferrell [EMAIL PROTECTED] wrote:
When Background or Playback is used in a dial plan how many digits are
collected and what variable are they returned in?
I'm trying to do a simple auto attendant and having very little luck
--
One day at a time, one second if that's what it takes
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Re: [asterisk-users] Ever donate Software to Digium? If you did your a fool.
On 8/9/06, Randall H. [EMAIL PROTECTED] wrote: If you gave software to Digium then you helped Mark become very rich. http://abcnews.go.com/Technology/wireStory?id=2290152 ___ --Bandwidth and Colocation provided by Easynews.com -- Congrats to Mark & Co @ Digium! It's well deserved recognition of all that you have achieved...and the even greater possibilities that lay ahead. OTOH, now that there's VC funding the game has changed...potentially alot. There may be a whole new set of expectations, and people to answer to in ways that haven't been the case to this point. However hard everyone's worked to get here...now the real work starts. But now there's a greater depth of means to tap in getting it done. Michael ___ --Bandwidth 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] Ever donate Software to Digium? If you did your a fool.
Now they can buy Sangoma with cash and still have a couple million left over :) (remember it's a Canadian company USD$1 = CAN$0.89) http://tinyurl.com/p4epp Company cash value of Sangoma USD$10.3 million. MATT--- On 8/9/06, Michael Graves [EMAIL PROTECTED] wrote: On 8/9/06, Randall H. [EMAIL PROTECTED] wrote: If you gave software to Digium then you helped Mark become very rich. http://abcnews.go.com/Technology/wireStory?id=2290152 ___ --Bandwidth and Colocation provided by Easynews.com -- Congrats to Mark Co @ Digium! It's well deserved recognition of all that you have achieved...and the even greater possibilities that lay ahead. OTOH, now that there's VC funding the game has changed...potentially alot. There may be a whole new set of expectations, and people to answer to in ways that haven't been the case to this point. However hard everyone's worked to get here...now the real work starts. But now there's a greater depth of means to tap in getting it done. Michael ___ --Bandwidth 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] Warning - Voiplink.com doesn't deliver - stuck in a hole
Is anyone else having problems with them? Order placed online 13 days ago. voiplink.com charged my cc for the product 11 days ago. They can't seem to ship Linksys spa-942 that they claim to have in stock. Order is still pending on their web site. Calls to them confirm no shipment but also get no results. Anyone know of a good supplier for the spa-942? ___ --Bandwidth 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] Callback and Asterisks
Good morning, all,I am in immediate need of configuring an Asterix to act as wake up call system.I need:1. user calls in and at the prompt enters his room number.2. Asterisks then checks DB to ensure that that room number exists3. Asterisk then prompts user to enter time4. user enters time he wants a wake up call5. Asterisk calls back at that time and plays a messageCan this be done with Asterisks?Vic ___ --Bandwidth 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] Warning - Voiplink.com doesn't deliver - stuck in a hole
I have had the same experience with a Grandstream order from them - 7 days and no product. They even told me it was shipping Monday, but couldn't produce a tracking number on Tuesday. Pretty lame. On 8/9/06, Tom [EMAIL PROTECTED] wrote: Is anyone else having problems with them? Order placed online 13 days ago. voiplink.com charged my cc for the product 11 days ago. They can't seem to ship Linksys spa-942 that they claim to have in stock. Order is still pending on their web site. Calls to them confirm no shipment but also get no results. Anyone know of a good supplier for the spa-942? ___ --Bandwidth 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] Callback and Asterisks
Vic wrote: I am in immediate need of configuring an Asterix to act as wake up call system. Amazing: http://www.google.com/search?q=asterisk+wake+up+call+site%3Avoip-info.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP trunks: order or type
I have two trunks to the same machine (x.x.x.2), one is type=friend, other is type=peer. Asterisk seems to choose which trunk to use by the order by which they are set out in sip.conf. When a incoming call comes into Asterisk, it always uses the last trunk. My understanding was that a peer trunk can't receive incoming calls. Does Asterisk ignore the type when dealing with incoming calls from the same host/machine ? I want all incoming calls to use the back-trunk only. When I change the order around from what it looks like below it works perfectly. I've been told that order of things appearing in sip.conf should not matter. --Shaun sip.conf: [back-trunk] type= friend username= 8880006111 secret = vv host= x.x.x.2 dtmfmode= rfc2833 nat = no canreinvite = no insecure= port,invite qualify = no disallow= all allow = ulaw allow = alaw allow = g729 context = shared-back-trunk-incoming [back-trunk-ulaw] type= peer username= 8880006113 secret = vv host= x.x.x.2 dtmfmode= rfc2833 nat = no canreinvite = no insecure= port,invite qualify = no disallow= all allow = ulaw context = shared-back-trunk-ulaw-incoming Asterisk CLI: Aug 10 12:17:15 DEBUG[21756]: chan_sip.c:7242 check_user_full: Setting NAT on RTP to 0 Aug 10 12:17:15 DEBUG[21756]: chan_sip.c:10497 handle_request_invite: Checking SIP call limits for device 8880006113 Aug 10 12:17:15 DEBUG[21756]: chan_sip.c:1401 __sip_ack: Stopping retransmission on '[EMAIL PROTECTED]' of Response 1: Match Found ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
