Re: [asterisk-users] How to park calls on a specific extension
At 08:40 PM 11/29/2006, you wrote: Either write what you want, or learn to use what we have and hope that SLA when it appears is better. Parking is not the best solution, I think that's the problem with the Asterisk community right now. Anytime something is suggested, the response is either write it yourself or deal with what is there. Not my intention to make you feel that way, and I'm sorry I didn't see it as someone attempting to make a suggestion, but instead saw it as someone complaining about something that didn't work the way they wanted. Asterisk is far from perfect, but even with that it enabled me to do things I never imagined being able to do with my home phone and all because I couldn't find a decently priced 4 line cordless phone. I have the same struggle with my wife who has little humor for those occasional misplaced commas that stop her from calling mom till I find and fix it, but smiles at our phone bill that dropped by abut 60%. Ira ___ --Bandwidth 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] Siemens Gigaset C450 IP vs S450 IP
On Wed, 2006-11-29 at 22:57 -0500, Cory Andrews wrote: Andrew - I have been told they have no plans to introduce US distribution or availability on these products in the foreseeable future. I was told this by one of the channel managers from Siemens. I received some eval units of some of the Siemens SIP products from a reseller in EU, and they are quite nice, but they were all kitted for 220. Couldn't you connect the base unit into the same socket as the cooker? ;) I just wish we could use Aastra 480i CT units over here in Europe, same kind of incompatibility problem. (Wrong frequencies) -- Dave Cotton [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] Cut function on semicolon separator
Hi,
I have the most stupid problem in my dialplan.
I need to do something as trivial as splitting a string, with a semicolon as
separator.
I was thinking the 'CUT' function would be perfect for this.
But the problem is the semicolon. In the dialplan it is always understood as
a separator for parameters.
What I have tried so far:
[macro-eva-on-sip]
exten = s,1,NoOp(${CALLERID(name)})
exten = s,n,NoOp(${CALLERID(num)})
exten = s,n,Set(v=${CALLERID(num)})
exten = s,n,Set(sep=;)
exten = s,n,NoOp(${CUT(v,sep,1)})
exten = s,n,Dial(SIP/evavox/${MACRO_EXTEN})
exten = s,n,Hangup()
I'm convinced there's a very simple solution to this, but I don't see it.
Anybody?!
Grtz,
Koen
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Re: [asterisk-users] What's up with the Manager Interface?!?!
- Original Message - From: Douglas Garstang [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, November 30, 2006 12:16 AM Subject: RE: [asterisk-users] What's up with the Manager Interface?!?! -Original Message- From: Steve Edwards [mailto:[EMAIL PROTECTED] Sent: Wednesday, November 29, 2006 2:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] What's up with the Manager Interface?!?! On Wed, 29 Nov 2006, Douglas Garstang wrote: G. Here's another example... Action: Command Command: sip show peer 2944093 Response: Follows Privilege: Command * Name : 2944093 Secret : Set MD5Secret: Not set Context : 180o_CallStart Subscr.Cont. : 180o_WatchBLF Why the HELL is there an asterisk before 'Name'? Now I have to strip the bloody thing out! And why is there TWO empty lines before it? Good grief! Doug. Would it be a better use of your time to fix the offending modules rather than kludge your code to handle the inconsistencies? Is AMI spec'd or would that be the first step? Steve, No... I'm not a C programmer. A standard interface would be a first step. :) Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- Doug, Instead of getting upset try fixing it. What happend to the cheery Doug that I knew from a month ago ? Run outa Jack ? I can try to get you a bottle :) ___ --Bandwidth 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] What's up with the Manager Interface?!?!
- Original Message - From: Douglas Garstang [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, November 30, 2006 12:19 AM Subject: RE: [asterisk-users] What's up with the Manager Interface?!?! -Original Message- From: James Texter [mailto:[EMAIL PROTECTED] Sent: Wednesday, November 29, 2006 3:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What's up with the Manager Interface?!?! Doug, Your issue isn't with the manager. It's with the CLI output you are trying to hijack via manager :D If you run sip show peer 2944093 in the CLI, you'll see a blank line, followed by a line that is * Name. It appears what you really want is a manager Action to show a sip peer, in which case I would recommend adding a new manager command that returns a string which is much more machine readable. Remember, CLI output is designed to be human readable. James. Ok... that sounds like an objective distinction. Maybe it's just the output that I get as a result of: Action: Command Command: foo that's causing problems. eg: Action: Command Command: sip show subscriptions I don't know why every CLI command doesn't have a corresponding action. I won't be adding any new manager commands, as I am not a C programmer. Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- Do what I do when I used to get desperate. Throw money at a developer (or what I ended up doing is getting a partner that is a C programmer). ___ --Bandwidth 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] Cut function on semicolon separator
Hi Koen,
Try:
exten = s,n,NoOp(CUT(${v},${sep},1))
Cheers
Koen Van Impe wrote:
Hi,
I have the most stupid problem in my dialplan.
I need to do something as trivial as splitting a string, with a
semicolon as separator.
I was thinking the 'CUT' function would be perfect for this.
But the problem is the semicolon. In the dialplan it is always
understood as a separator for parameters.
What I have tried so far:
[macro-eva-on-sip]
exten = s,1,NoOp(${CALLERID(name)})
exten = s,n,NoOp(${CALLERID(num)})
exten = s,n,Set(v=${CALLERID(num)})
exten = s,n,Set(sep=;)
exten = s,n,NoOp(${CUT(v,sep,1)})
exten = s,n,Dial(SIP/evavox/${MACRO_EXTEN})
exten = s,n,Hangup()
I'm convinced there's a very simple solution to this, but I don't see it.
Anybody?!
Grtz,
Koen
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[asterisk-users] Distinctive ring
Hi list!
I need help with distinctive ring on Cisco 7940 phone. I'm using Asterisk 1.2.5
(I know, I should upgrade) and in dial plan I have:
exten = _64X,n,Set(_ALERT_INFO=Chirp2)
exten = _64X,n,Dial(SIP/${EXTEN},30,wWtT)
On Cisco in Settings = Ring type I have Chirp1 and Chirp2. By default
phone is ringing sound Chirp1. For internal calls I'm using dial plan I have
sent you above. Problem is that Cisco doesn't ring with Chirp2, but with
slightly different Chirp1 (instead of ring, pause, ring he sounds ring ring
pause).
Is there any way that my Cisco 7940, thru dial plan, can ring Chirp2 instead of
Chirp1?
Thank you for your time!
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)270248
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
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Re: [asterisk-users] mISDN
Hi for all I've a problem. I'm trying to detect the progress of an invalid call. For example, if I phone to a busy number (or invalid number), my misdn always detect ring. Have you got any suggestion? 2006/11/29, Patrick [EMAIL PROTECTED]: On Wed, 2006-11-29 at 16:38 +0100, Timothy Parez wrote: I get the following with debug on: P[ 3] I IND :SETUP oad:497978546 dad:50556010 pid:10 state:none P[ 3] -- channel:1 mode:TE cause:16 ocause:16 rad: cad: P[ 3] -- info_dad: onumplan:2 dnumplan:2 rnumplan: cpnnumplan:0 P[ 3] -- Bearer: Speech P[ 3] -- Codec: Alaw P[ 0] -- * NEW CHANNEL dad:50556010 oad:497978546 P[ 3] -- CTON: Unknown P[ 3] EXPORT_PID: pid:10 P[ 3] -- PRES: Restricted (0) P[ 3] -- SCREEN: Unscreened (0) Nov 29 16:39:40 WARNING[7221]: chan_misdn.c:4735 chan_misdn_log: Extension can never match, so disconnecting P[ 3] I SEND:RELEASE oad:0497978546 dad:050556010 pid:10 P[ 3] -- bc_state:BCHAN_CLEANED P[ 3] -- channel:1 mode:TE cause:16 ocause:1 rad: cad: P[ 3] -- info_dad: onumplan:2 dnumplan:2 rnumplan: cpnnumplan:0 P[ 3] I IND :RELEASE_COMPLETE oad: dad: pid:10 state:EXTCANTMATCH P[ 3] -- channel:0 mode:TE cause:16 ocause:16 rad: cad: P[ 3] -- info_dad: onumplan:0 dnumplan:0 rnumplan:0 cpnnumplan:0 P[ 3] hangup_chan P[ 3] - hangup P[ 3] * IND : HANGUPpid:10 ctx:inisdn dad:050556010 oad:0497978546 Afaik the dad:050556010 is the destination number and oad:0497978546 is the origination number. I think you need to change your dialplan by adding a 0 in front of your 5055 entries. 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
[asterisk-users] Digium TE405P dtmf issue
Hi Group, I have an asterisk running as media gateway with a Digium TE405P 2nd Gen rev 2 with echo cancellation. It is interconnected to a telco carrier via ISDN Pri. The voice quality is clear except that sometimes a hear a beep sound that occure around 5 to 10 secs in the middle of the conversation. When I check the logs in the asterisk, I found this. Nov 30 00:48:38 DEBUG[27705] chan_zap.c: Exception on 17, channel 5 Nov 30 00:48:38 DEBUG[27705] chan_zap.c: Got event Event 131121(131121) on channel 5 (index 0) Nov 30 00:48:38 DEBUG[27705] chan_zap.c: DTMF Down '1' Nov 30 00:48:38 DEBUG[27705] chan_zap.c: Exception on 17, channel 5 Nov 30 00:48:38 DEBUG[27705] chan_zap.c: Got event Event 262193(262193) on channel 5 (index 0) Nov 30 00:48:38 DEBUG[27705] chan_zap.c: Detected digit '1' Nov 30 00:48:38 DEBUG[27705] chan_zap.c: Exception on 17, channel 5 Nov 30 00:48:38 DEBUG[27705] chan_zap.c: Got event Event 131121(131121) on channel 5 (index 0) Nov 30 00:48:38 DEBUG[27705] chan_zap.c: DTMF Down '1' Nov 30 00:48:38 DEBUG[27705] chan_zap.c: Exception on 17, channel 5 Nov 30 00:48:38 DEBUG[27705] chan_zap.c: Got event Event 262193(262193) on channel 5 (index 0) Nov 30 00:48:38 DEBUG[27705] chan_zap.c: Detected digit '1' Can please someone help me on how can I omit this problem. I am currently running asterisk 1.2.13. Thanks in advance... Leonimar Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail beta. http://new.mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Trouble with regexten
Can anyone help with the use of regexten? (* 1.4.3) I've got Asterisk creating extensions for my SIP phones using regexten but I can't seem to figure out how to make use of them once they're registered. Here's my dialplan for from-sip (the SIP's default context): asterisk*CLI dialplan show from-sip [ Context 'from-sip' created by 'pbx_config' ] '98766' =1. Dial(Sip/Tim) [pbx_config] 2. Hangup() [pbx_config] Include ='sip_autoreg'[pbx_config] Include ='widgets'[pbx_config] -= 1 extension (2 priorities) in 1 context. =- asterisk*CLI and here's sip_autoreg (the regexten context): asterisk*CLI dialplan show sip_autoreg [ Context 'sip_autoreg' created by 'pbx_config' ] '114' = 2. Dial(Sip/Tim) [pbx_config] 3. Hangup() [pbx_config] [ Context 'sip_autoreg' created by 'SIP' ] '112' = 1. Noop(Russell) [SIP] '113' = 1. Noop(Richard) [SIP] '114' = 1. Noop(Tim) [SIP] -= 4 extensions (5 priorities) in 2 contexts. =- asterisk*CLI Dialing 98766 from Sip/Russell rings Sip/Tim as expected. Dialing 114 gives Not Found :-( I'm very confused any ideas why this doesn't work? -- Regards, Russell | Russell Brown | MAIL: [EMAIL PROTECTED] PHONE: 01780 471800 | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.uk | ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 200+ analog phones connected to FXS modules
I am trying to find out the best way to replace one of our hardware PBXs. It currently has 200+ analog phones connected to it. The idea is to take advantage of the already installed phone cables (big building) so I'm trying to avoid the use of ethernet adapters (if possible). However, I'm realizing that it's an expensive setup and will definitely require two or more cooperating Asterisk servers (cluster) mainly due to PCI slot availability. I am aware of the TDM2400P card. One could put 6 FXS uqad-modules and would serve 24 analog phones. However, I would need at least 9 of these PCI cards which could be placed in 2 or 3 servers. Is there another way of doing this (hopefully cheaper and more convenient)? Thank you for your suggestions. Vieri Yahoo! Music Unlimited Access over 1 million songs. http://music.yahoo.com/unlimited ___ --Bandwidth 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] 200+ analog phones connected to FXS modules
Take a look at Channel Banks On 11/30/06, Vieri [EMAIL PROTECTED] wrote: I am trying to find out the best way to replace one of our hardware PBXs. It currently has 200+ analog phones connected to it. The idea is to take advantage of the already installed phone cables (big building) so I'm trying to avoid the use of ethernet adapters (if possible). However, I'm realizing that it's an expensive setup and will definitely require two or more cooperating Asterisk servers (cluster) mainly due to PCI slot availability. I am aware of the TDM2400P card. One could put 6 FXS uqad-modules and would serve 24 analog phones. However, I would need at least 9 of these PCI cards which could be placed in 2 or 3 servers. Is there another way of doing this (hopefully cheaper and more convenient)? Thank you for your suggestions. Vieri ___ --Bandwidth 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] 200+ analog phones connected to FXS modules
On nov/30/2006, Vieri wrote: Is there another way of doing this (hopefully cheaper and more convenient)? VoIP Gateways with 48 FXS ports. -- Paco Brufal[EMAIL PROTECTED] ServiTux Servicios Informáticos S.L. Tel. 966 160 600 / Fax. 966 160 601 ___ --Bandwidth 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] Cut function on semicolon separator
Peter,
Thanks for your reply!
It didn't work though.
There's actually already a problem setting the semicolon as value for the
'sep' variable.
*The functions:*
exten = s,n,Set(sep=';')
exten = s,n,NoOp(${CUT(v,${sep},1)})
*The output:*
-- Executing Set(SIP/1649-09ca84f0, sep=) in new stack
-- Executing NoOp(SIP/1649-09ca84f0, 1649;phonecontext=Exp_Net) in new
stack
fyi, v is a variable holding 1649;phonecontext=Exp_Net
So the question is now: how can I set a variable to hold a semicolon as
variable.
And can I then use this variable as separator in the Cut function?
On 11/30/06, Peter Lindquist [EMAIL PROTECTED] wrote:
Hi Koen,
Try:
exten = s,n,NoOp(CUT(${v},${sep},1))
Cheers
Koen Van Impe wrote:
Hi,
I have the most stupid problem in my dialplan.
I need to do something as trivial as splitting a string, with a semicolon
as separator.
I was thinking the 'CUT' function would be perfect for this.
But the problem is the semicolon. In the dialplan it is always understood
as a separator for parameters.
What I have tried so far:
[macro-eva-on-sip]
exten = s,1,NoOp(${CALLERID(name)})
exten = s,n,NoOp(${CALLERID(num)})
exten = s,n,Set(v=${CALLERID(num)})
exten = s,n,Set(sep=;)
exten = s,n,NoOp(${CUT(v,sep,1)})
exten = s,n,Dial(SIP/evavox/${MACRO_EXTEN})
exten = s,n,Hangup()
I'm convinced there's a very simple solution to this, but I don't see it.
Anybody?!
Grtz,
Koen
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RE: [asterisk-users] 200+ analog phones connected to FXS modules
I think It would be cheaper to use dedicated VoIP PSTN Gateways, such as audiocodes or similar (audiocodes is actually a bad example, as their not that cheap). But dedicated ATA hardware with 24 or more ports. Jon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vieri Sent: 30. november 2006 10:15 To: asterisk-users@lists.digium.com Subject: [asterisk-users] 200+ analog phones connected to FXS modules I am trying to find out the best way to replace one of our hardware PBXs. It currently has 200+ analog phones connected to it. The idea is to take advantage of the already installed phone cables (big building) so I'm trying to avoid the use of ethernet adapters (if possible). However, I'm realizing that it's an expensive setup and will definitely require two or more cooperating Asterisk servers (cluster) mainly due to PCI slot availability. I am aware of the TDM2400P card. One could put 6 FXS uqad-modules and would serve 24 analog phones. However, I would need at least 9 of these PCI cards which could be placed in 2 or 3 servers. Is there another way of doing this (hopefully cheaper and more convenient)? Thank you for your suggestions. Vieri Yahoo! Music Unlimited Access over 1 million songs. http://music.yahoo.com/unlimited ___ --Bandwidth 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] AGI PHP Issues (AGI script runs but phone hangs up too quickly)
Sorry to re-post this but I'm sure it's something simple that someone has
found before.
To summarise:
Dial plan answers the phone
AGI script executes
AGI debug in console show phonetics ABC - However no audio at all on the
phone and this step is less than 1 second.
Dial plan Busy
Phone hangs up.
Total time less than a second.
This is such a simple AGI script do I need the PHPAGI Library - this seems
like a sledgehammer to crack a peanut.
Thanks again.
Original post:
I am attempting my first go at a simple AGI application using PHP (Getting
Asterisk to SAY PHONETIC ABC). I have dabbled with PHP but I am by no means
a professional standard developer.
My script seems to execute ok, and I can see asterisk playing the sounds but
my phone goes from ringing to busy, and I don't hear the phontics.
Below are the relevant bits from my PHP, Console, and extensions.conf.
I would be most grateful if someone could show me the way.
Thanks in advance:
Chris
Asterisk ver: 1.2.10
PHP:
#!/usr/local/php/bin/php -q
?php
$stdin = fopen('php://stdin', 'r');
$stdout = fopen('php://stdout', 'w');
$stdlog = fopen('/var/log/asterisk/my_agi.log', 'w');
while (!feof($stdin)) {
$temp = fgets($stdin);
$temp = str_replace(\n,,$temp);
$s = explode(:,$temp);
$agivar[$s[0]] = trim($s[1]);
if (($temp == ) || ($temp == \n)) {
break;
}
}
fputs($stdout,SAY PHONETIC \abc\ \#\ \n);
fflush($stdout);
$msg = fgets($stdin,1024);
fputs($stdlog,$msg . \n);
?
Extensions.conf:
exten = 4343,1,Answer
exten = 4343,2,AGI(example.php)
exten = 4343,3,Busy
AGI Debug:
AGI Rx SAY PHONETIC abc #
-- Playing 'phonetic/a_p' (language 'en')
-- Playing 'phonetic/b_p' (language 'en')
-- Playing 'phonetic/c_p' (language 'en')
-- AGI Script example.php completed, returning 0
-- Executing Busy(SIP/4321-081b9498, ) in new stack
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[asterisk-users] Re: Cisco 7940 Firmware 8.2
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Greetings, I am cutting my teeth with SIP phones and my first issue is getting a Cisco 7940 to Authenticate with my VoIP provider (BBTelsys). I did read some notes on the vo-ip website about 7.5 being the better firmware version. Has anyone had trouble with 8.2 and SIP registering? Should I just downgrade to 7.5 and give it a go? I think SIP uses UDP 5060 correct? The phone is behind a firewall(NAT) I figure this might be an issue as well. Thoughts? Thank you for your response. I'm using 7.4 firmware. I didn't noticed any problems. I'm not familiar that and further firmware brings anything that will make me change firmware. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cut function on semicolon separator
Try using set without ' or . I mean:
exten = s,n,Set(sep=;)
And next step try using CUT with and without ${..}.
exten = s,n,Noop(${CUT(v,sep,1)})
or
exten = s,n,Noop(${CUT(v,${sep},1)})
First parameter is using variable without surrounding ${..}.
bye,
a
On 11/30/06, Koen Van Impe [EMAIL PROTECTED] wrote:
Peter,
Thanks for your reply!
It didn't work though.
There's actually already a problem setting the semicolon as value for the
'sep' variable.
The functions:
exten = s,n,Set(sep=';')
exten = s,n,NoOp(${CUT(v,${sep},1)})
The output:
-- Executing Set(SIP/1649-09ca84f0, sep=) in new stack
-- Executing NoOp(SIP/1649-09ca84f0, 1649;phonecontext=Exp_Net) in new
stack
fyi, v is a variable holding 1649;phonecontext=Exp_Net
So the question is now: how can I set a variable to hold a semicolon as
variable.
And can I then use this variable as separator in the Cut function?
On 11/30/06, Peter Lindquist [EMAIL PROTECTED] wrote:
Hi Koen,
Try:
exten = s,n,NoOp(CUT(${v},${sep},1))
Cheers
Koen Van Impe wrote:
Hi,
I have the most stupid problem in my dialplan.
I need to do something as trivial as splitting a string, with a semicolon
as separator.
I was thinking the 'CUT' function would be perfect for this.
But the problem is the semicolon. In the dialplan it is always understood
as a separator for parameters.
What I have tried so far:
[macro-eva-on-sip]
exten = s,1,NoOp(${CALLERID(name)})
exten = s,n,NoOp(${CALLERID(num)})
exten = s,n,Set(v=${CALLERID(num)})
exten = s,n,Set(sep=;)
exten = s,n,NoOp(${CUT(v,sep,1)})
exten = s,n,Dial(SIP/evavox/${MACRO_EXTEN})
exten = s,n,Hangup()
I'm convinced there's a very simple solution to this, but I don't see it.
Anybody?!
Grtz,
Koen
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Re: [asterisk-users] How to park calls on a specific extension
On Thu, Nov 30, 2006 at 12:03:24AM -0600, Lacy Moore - Aspendora wrote: The question is what is the best interface? On our old system, we put the caller on hold, went to another phone, pressed pickup and then entered the extension where the call is on hold. I never liked that, especially if I was at an extension that wasn't mine. By the time, I got to where I needed to be, or someone called me and told me to pick that call, I would forget what extension. The same thing, I believe, will happen with the current park method. I don't know what would help with that, maybe better vitamins to prevent memory loss? :-) I don't know. Maybe a receptionist console that could tell who is on park, their phone number and caller id info along with who put them on park? If you integrate with the voice mail, so that you can pull a user's audio name for an extension, the pickup extension can say Do you want to pick up the call put on hold by 'Lacy Moore' or 'Joe Smith' or 'waiting room' or 'extension 242' Hopefully little need for memory. I'm wondering if maybe we are looking at having to have different ways of doing it. Being able to transfer the call to a line button, and being able to press that line button to pick up the call, and having the status shown, may be the better solution for small companies. Problem there is only some phones have line buttons, and when they have them they are scarce and there's many things you might like to do with them, and dedicating them to this would be low on my list. Dedicating one speed dial to a pickup call command that picks up the solo call or reads you the names/numbers of the calls on hold, or puts them on your screen if you have a screen -- that makes more sense, and it does well on every phone. Then if you want to have line buttons which read hints based on the number of calls held. I'm going to show my ignorance here. Since the phone displays the number we dialed,or the incoming caller information on the screen (we're talking those with displays), is there anyway to have it so that when the call is parked, it also shows the parking spot the caller is parked on? Kind of like hold does now? I know nothing about the SIP protocol, so I don't know if this is possible or not. Yes, some phones can receive text messages back from the server. Not all of them. But if you have a system where parking is just pressing hold, then all you need to know at worst is the name or extension of the phone you're on, and that's usualy already on the phone screen or even written on in pen! ___ --Bandwidth 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] Cut function on semicolon separator
_The functions:_
exten = s,n,Set(sep=';')
exten = s,n,NoOp(${CUT(v,${sep},1)})
Have you tried to put a '\' in front of the ';': Set(sep='\;')?
Peter
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Re: [asterisk-users] AgentCallbackLogin deprecated?
On Tue, 28 Nov 2006 17:57:04 -0600 Octavio Ruiz (Ta^3) [EMAIL PROTECTED] wrote: Is there an isolated example somewhere of how to use existing dialplan logic and dynamic queue membership to simulate the current behaviour? http://svn.digium.com/view/asterisk/trunk/doc/queues-with-callback-members.txt Thanks for that - didn't realise the mainline docs contained such useful and comprehensive information these days! Why? Seems that reinventing the well was the agentcallbacklogin implementation, when it could be happend in dialplan logic. Cool, in conjunction with the one-line patches at http://bugs.digium.com/view.php?id=7736 I think I have the ACD functionality I need without bothering with chan_agent :) Cheers, Gavin. ___ --Bandwidth 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] codec error message
Hi all, I get the following message in the CLI after enabling video function. I have searched about the codec 126 but nothing found. Anybody can tell me how to fix the problem? Nov 30 15:54:27 NOTICE[16508]: rtp.c:576 ast_rtp_read: Unknown RTP codec 126 received Nov 30 15:54:27 NOTICE[16508]: rtp.c:576 ast_rtp_read: Unknown RTP codec 126 received ___ --Bandwidth 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] Cut function on semicolon separator
All,
The last Peter got it right! :-)
The final solution:
exten = s,n,Set(sep='\;')
exten = s,n,NoOp(${CUT(v,${sep},1)})
Thanks for you input and have a very nice day!
Koen
On 11/30/06, Peter Boehm [EMAIL PROTECTED] wrote:
_The functions:_
exten = s,n,Set(sep=';')
exten = s,n,NoOp(${CUT(v,${sep},1)})
Have you tried to put a '\' in front of the ';': Set(sep='\;')?
Peter
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RE: [asterisk-users] 200+ analog phones connected to FXS modules
You could put at least two Rhino quad t1 cards and that would give you 8 times 24 ports and I heard of one with those cards plus a dual t1 card which is 240 extensions on one server. this would take up 3 pci slots. on Thursday 11/30/2006 Jon Schøpzinsky([EMAIL PROTECTED]) wrote I think It would be cheaper to use dedicated VoIP PSTN Gateways, such as audiocodes or similar (audiocodes is actually a bad example, as their not that cheap). But dedicated ATA hardware with 24 or more ports. Jon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vieri Sent: 30. november 2006 10:15 To: asterisk-users@lists.digium.com Subject: [asterisk-users] 200+ analog phones connected to FXS modules I am trying to find out the best way to replace one of our hardware PBXs. It currently has 200+ analog phones connected to it. The idea is to take advantage of the already installed phone cables (big building) so I'm trying to avoid the use of ethernet adapters (if possible). However, I'm realizing that it's an expensive setup and will definitely require two or more cooperating Asterisk servers (cluster) mainly due to PCI slot availability. I am aware of the TDM2400P card. One could put 6 FXS uqad-modules and would serve 24 analog phones. However, I would need at least 9 of these PCI cards which could be placed in 2 or 3 servers. Is there another way of doing this (hopefully cheaper and more convenient)? Thank you for your suggestions. Vieri Yahoo! Music Unlimited Access over 1 million songs. http://music.yahoo.com/unlimited ___ --Bandwidth 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [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] 200+ analog phones connected to FXS modules
I would just guess that the PCI bus would be pretty busy, with 3 T1 cards. Couldn't that be a problem? Jon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John covici Sent: 30. november 2006 12:07 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] 200+ analog phones connected to FXS modules You could put at least two Rhino quad t1 cards and that would give you 8 times 24 ports and I heard of one with those cards plus a dual t1 card which is 240 extensions on one server. this would take up 3 pci slots. on Thursday 11/30/2006 Jon Schøpzinsky([EMAIL PROTECTED]) wrote I think It would be cheaper to use dedicated VoIP PSTN Gateways, such as audiocodes or similar (audiocodes is actually a bad example, as their not that cheap). But dedicated ATA hardware with 24 or more ports. Jon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vieri Sent: 30. november 2006 10:15 To: asterisk-users@lists.digium.com Subject: [asterisk-users] 200+ analog phones connected to FXS modules I am trying to find out the best way to replace one of our hardware PBXs. It currently has 200+ analog phones connected to it. The idea is to take advantage of the already installed phone cables (big building) so I'm trying to avoid the use of ethernet adapters (if possible). However, I'm realizing that it's an expensive setup and will definitely require two or more cooperating Asterisk servers (cluster) mainly due to PCI slot availability. I am aware of the TDM2400P card. One could put 6 FXS uqad-modules and would serve 24 analog phones. However, I would need at least 9 of these PCI cards which could be placed in 2 or 3 servers. Is there another way of doing this (hopefully cheaper and more convenient)? Thank you for your suggestions. Vieri Yahoo! Music Unlimited Access over 1 million songs. http://music.yahoo.com/unlimited ___ --Bandwidth 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 200+ analog phones connected to FXS modules
You could go for 2 quad pri cards + channel banks or for TDMoE or usb channel banks. The last option would be the cheaper and more scalable one imho www.spidermux.org www.xorcom.com Joachim John covici wrote: You could put at least two Rhino quad t1 cards and that would give you 8 times 24 ports and I heard of one with those cards plus a dual t1 card which is 240 extensions on one server. this would take up 3 pci slots. on Thursday 11/30/2006 Jon Schøpzinsky([EMAIL PROTECTED]) wrote I think It would be cheaper to use dedicated VoIP PSTN Gateways, such as audiocodes or similar (audiocodes is actually a bad example, as their not that cheap). But dedicated ATA hardware with 24 or more ports. Jon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vieri Sent: 30. november 2006 10:15 To: asterisk-users@lists.digium.com Subject: [asterisk-users] 200+ analog phones connected to FXS modules I am trying to find out the best way to replace one of our hardware PBXs. It currently has 200+ analog phones connected to it. The idea is to take advantage of the already installed phone cables (big building) so I'm trying to avoid the use of ethernet adapters (if possible). However, I'm realizing that it's an expensive setup and will definitely require two or more cooperating Asterisk servers (cluster) mainly due to PCI slot availability. I am aware of the TDM2400P card. One could put 6 FXS uqad-modules and would serve 24 analog phones. However, I would need at least 9 of these PCI cards which could be placed in 2 or 3 servers. Is there another way of doing this (hopefully cheaper and more convenient)? Thank you for your suggestions. Vieri Yahoo! Music Unlimited Access over 1 million songs. http://music.yahoo.com/unlimited ___ --Bandwidth 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] 200+ analog phones connected to FXS modules
rhino tells me no, they have a computer you can buy on which they have tested such things. I don't have this myself, however. on Thursday 11/30/2006 Jon Schøpzinsky([EMAIL PROTECTED]) wrote I would just guess that the PCI bus would be pretty busy, with 3 T1 cards. Couldn't that be a problem? Jon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John covici Sent: 30. november 2006 12:07 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] 200+ analog phones connected to FXS modules You could put at least two Rhino quad t1 cards and that would give you 8 times 24 ports and I heard of one with those cards plus a dual t1 card which is 240 extensions on one server. this would take up 3 pci slots. on Thursday 11/30/2006 Jon Schøpzinsky([EMAIL PROTECTED]) wrote I think It would be cheaper to use dedicated VoIP PSTN Gateways, such as audiocodes or similar (audiocodes is actually a bad example, as their not that cheap). But dedicated ATA hardware with 24 or more ports. Jon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vieri Sent: 30. november 2006 10:15 To: asterisk-users@lists.digium.com Subject: [asterisk-users] 200+ analog phones connected to FXS modules I am trying to find out the best way to replace one of our hardware PBXs. It currently has 200+ analog phones connected to it. The idea is to take advantage of the already installed phone cables (big building) so I'm trying to avoid the use of ethernet adapters (if possible). However, I'm realizing that it's an expensive setup and will definitely require two or more cooperating Asterisk servers (cluster) mainly due to PCI slot availability. I am aware of the TDM2400P card. One could put 6 FXS uqad-modules and would serve 24 analog phones. However, I would need at least 9 of these PCI cards which could be placed in 2 or 3 servers. Is there another way of doing this (hopefully cheaper and more convenient)? Thank you for your suggestions. Vieri Yahoo! Music Unlimited Access over 1 million songs. http://music.yahoo.com/unlimited ___ --Bandwidth 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [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] Setting RTP ports for Asterisk?
Vincent Delporte wrote: Hello When I make calls from home to the PSTN by going through the Net - Asterisk - the Net - VoIP provider - PSTN, I get no sound either way. I assume it's because I must tell Asterisk to use fixed ranges of UDP ports and map ports accordingly on the NAT firewall under which it is located on the LAN at work. Here's the schema: home NAT Internet NAT Asterisk NAT Internet VoIP provide PSTN callee I took care of the NAT at home by using fixed ports in X-Lite + used STUN, so I guess the problem is located on the Asterisk side. 1. What are the settings (in sip.conf?) to tell Asterisk to use specific ports for RTP? 2. With this kind of setup, does Asterisk stay in the loop to forward RTP packets, or do X-Lite at home and the VoIP provider send RTP to each other directly? 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 try rtp.conf :-D 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
[asterisk-users] Server Compatibility questions... IBM and Dell
Does anyone on list have experience with Digium hardware in the following servers: Dell poweredge SC440 IBM xSeries x226 Have just had major hassles getting TE205P ISDN cards going in these boxes. No joy so far. Anyone managed to do it yet? Thanks. Mark ___ --Bandwidth 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] Monitoring awareness
Hi Steve, Ok Playback could be used here, indeed. But if you are using automonitor - by default activated by (*1) - I think there is no way how to implement this. Am I right? Thanks, Ondrej Steve Totaro wrote: [EMAIL PROTECTED] wrote: Hello, I'm discovering asterisk, it seem to be a great soft. I have seen a fonction to record calls that's a great fontion but there is something disturbing me. When the record start, except if the recorder prevent the other part, he is not aware of the recording... I dont find a way from the feature.conf how to play a sound when a monitor start to record :/ Either play a file with a beep or a verbal message that this call may be recorded for such and such reason. This can be done easily in the dialplan by calling playback or background prior to monitor. Depending on local laws, you may be OK if just one party on the call knows it is being recorded. Other states have different laws. I have no idea how the law works when one caller is in one state with one set of laws and the other caller is in a different state with different laws. Thanks, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The information contained in this e-mail and in any attachments is confidential and is designated solely for the attention of the intended recipient(s). If you are not an intended recipient, you must not use, disclose, copy, distribute or retain this e-mail or any part thereof. If you have received this e-mail in error, please notify the sender by return e-mail and delete all copies of this e-mail from your computer system(s). Please direct any additional queries to: [EMAIL PROTECTED] 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
Re: [asterisk-users] Voicemail, SQL ODBC
RR wrote: Norbert, mate, I don't know why you're having so much problems. Do you wanna post your extconfig.conf here? just to humour us? I have it running with MSSQLServer a more complicated prospect than mySQL which has a dedicated driver for it, and it still works. RR, mate, I don't think that I have so many problems. 1.) I asked a simple question: Is it (still not) possible to connect Asterisk directly (= without ODBC) to mySQL for the purpose of storing voicemail data? Now, some posts later I've got a simple answer: No! 2.) It's not exactly clear to me why my extconfig.conf should humour you 3.) You're telling me (and everybody else here) that you have *it* running with MSSQL. But you're neither telling what *it* exactly is or does nor *how* you made it running. Maybe you want your extconfig.conf post here? ___ --Bandwidth 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 connection to a PBX
Probably find you have less hassle ditching the proprietary PBX's altogether and just use the * boxes at each end of an IAX trunk. Probably be a cheaper solution in the long run. On 11/30/06, asterisk-robert [EMAIL PROTECTED] wrote: Inital setup for testing will be 2-4 channels in order to prove the concept. When successful we may include some PBX systems that do not have available T-1 slots. On Wed, 29 Nov 2006 19:18:50 -0800, Tom Lynn [EMAIL PROTECTED] wrote: How many channels do you require? I'd favor T1 for a few reasons. Higher port density means fewer cards per system, which will mean fewer interrupts. T1s won't require you to tune analog levels. Echo probability will be lower. On 11/29/06, asterisk-robert [EMAIL PROTECTED] wrote: We are thinking of setting up an Asterisk system to route calls between 2 of our factories. Our idea is to connect an Asterisk box to each PBX and then use SIP(or IAX) to truck between the 2 systems on our internal network. I would be interested in any ideas regarding the connection points: 1. Is using Asterisk a good solution? 2. Is using a T-1 card the best way to connect the PBX and Asterisk? 3. If analog is used for the connection is it better for Asterisk to use FXO or FXS cards? Any ideas are appreciated. Robert ___ --Bandwidth 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 -- regards, Mark P. Edwards ___ --Bandwidth 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] Voicemail, SQL ODBC
Norbert Zawodsky wrote: RR wrote: Norbert, mate, I don't know why you're having so much problems. Do you wanna post your extconfig.conf here? just to humour us? I have it running with MSSQLServer a more complicated prospect than mySQL which has a dedicated driver for it, and it still works. RR, mate, I don't think that I have so many problems. 1.) I asked a simple question: Is it (still not) possible to connect Asterisk directly (= without ODBC) to mySQL for the purpose of storing voicemail data? Now, some posts later I've got a simple answer: No! 2.) It's not exactly clear to me why my extconfig.conf should humour you 3.) You're telling me (and everybody else here) that you have *it* running with MSSQL. But you're neither telling what *it* exactly is or does nor *how* you made it running. Maybe you want your extconfig.conf post here? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users how about just some simple RTFM? http://www.voip-info.org/wiki/view/Asterisk+RealTime+Voicemail http://www.voip-info.org/wiki/view/Asterisk+Voicemail+ODBC+storage 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: [asterisk-users] Polycom 601 Second Incoming Call
you can change the configs to have multiple beeps, and adjust the timing of them, but when we tried the problem then is the beep is not added to the incoming audio, but replaces it, so you lose the far end speaking, went back to default. On Nov 29, 2006, at 3:34 PM, Dovid B wrote: Hi List, I have a Polycom 601 that when the user is on the phone they only hear one beep and the CID of the second incoming call is not shown. Is there a way to have the CID show up for the second call ? And a way to configure the phone to beep more often if there is another call coming in. The problem is that if the receptionist is on the phone and looking up something on the PC she some times dosent realize that a new call is coming in. Thanks. Dovid ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom 601 Second Incoming Call
On Wed, Nov 29, 2006 at 11:34:41PM +0200, Dovid B said: I have a Polycom 601 that when the user is on the phone they only hear one beep and the CID of the second incoming call is not shown. Is there a way to have the CID show up for the second call ? And a way to configure the phone to beep more often if there is another call coming in. The problem is that if the receptionist is on the phone and looking up something on the PC she some times dosent realize that a new call is coming in. Thanks. I can't offer any help here, but just a ditto to your question. Nothing seems obvious to me that would change this behavior in the XML. The problem is annoying enough that I was thinking of writing a little desktop applet that would popup with this info, but the phone should do this by default. ___ --Bandwidth 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] AgentCallbackLogin deprecated?
On Tue, 28 Nov 2006 17:57:04 -0600 Octavio Ruiz (Ta^3) [EMAIL PROTECTED] wrote: Why? Seems that reinventing the well was the agentcallbacklogin implementation, when it could be happend in dialplan logic. Hm, now that I have examined this in more depth, I still seem to be missing one vital piece of the puzzle. The queues-with-callback-members.txt tutorial assumes that one agent (as a specific human being) is always reachable at a specific phone. This is not the case, and why I investigated chan_agent in the first instance. Our agents sit at any phone and log in, so their ACD groups follow them. This is what I really meant about re-inventing the wheel, since with AgentCallbackLogin removed, surely I'll have to maintain my own database tables of which agent is available at which extension? I'm hoping I've just overlooked something really obvious :) Cheers, Gavin, ___ --Bandwidth 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] Server Compatibility questions... IBM and Dell
I've got a Dell SC440 running just fine with a Digium TDM-400 card in it. It's running CentOS-64bit. Mark Edwards wrote: Does anyone on list have experience with Digium hardware in the following servers: Dell poweredge SC440 IBM xSeries x226 Have just had major hassles getting TE205P ISDN cards going in these boxes. No joy so far. Anyone managed to do it yet? Thanks. Mark ___ --Bandwidth 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] SIP transfer from agent fails
I have seen a couple of posts related to this, but no workaround. Setup; Asterisk 1.2.13 with Polycom IP501 phones Caller is sent to the queue with the t option Agent is logged in via AgentCallbackLogin on an extension that is in a context that includes exclusively agent extensions. Agent is set up with ackcall=yes (# to answer) Call comes in, agent takes the call, attempts to transfer to another extension using a SIP transfer on the Polycom phone. Call drops when completing the transfer. The caller goes on hold as they should, the second call is dialed and answers successfully, but the completion of the transfer fails and the call is dropped. There is a internal server error 500 logged on the console from the phone of the agent the originally answered the call. This used to work in earlier versions, quit working some time around 1.2.10 I think. ___ --Bandwidth 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] Server Compatibility questions... IBM and Dell
Thanks Joe. Although youre card isn't quite the same as the one I am trying to use you've given me a possible idea to play around with - to try and get the 64 bit stuff going and see if that has some sort of positive effect... Still out there looking for someone with a 205, 207, 405 or 407 in there though... many thanks, Mark. On 12/1/06, Joe Dennick [EMAIL PROTECTED] wrote: I've got a Dell SC440 running just fine with a Digium TDM-400 card in it. It's running CentOS-64bit. Mark Edwards wrote: Does anyone on list have experience with Digium hardware in the following servers: Dell poweredge SC440 IBM xSeries x226 Have just had major hassles getting TE205P ISDN cards going in these boxes. No joy so far. Anyone managed to do it yet? Thanks. Mark ___ --Bandwidth 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 -- regards, Mark P. Edwards ___ --Bandwidth 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] extension launch into AGI
I've got a simple set up with 1 fxo port and 1 fxs port in a Digium card connected to a POTS line and a phone set (physical extension). I've got all incoming calls launching directly into an AGI script. I'd like to do the same for the physical extension. In other words, when picking up the hand set, the AGI is launched without dialing any digits. check http://www.voip-info.org/wiki-Asterisk+config+zapata.conf keyword is : immediate ___ --Bandwidth 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] Voicemail, SQL ODBC
RR, mate, I don't think that I have so many problems. 1.) I asked a simple question: Is it (still not) possible to connect Asterisk directly (= without ODBC) to mySQL for the purpose of storing voicemail data? Now, some posts later I've got a simple answer: No! Oh, haha sorry about that, I read these emails just to take a break from my regular work and filter them by keyword voicemail as that's all I use (*) for (and conference). Maybe I don't know enough about DB Connectivity by I thought the MySQL driver I was mentioning earlier is a direct connector to MySQL and doesn't need ODBC. ODBC I thought was for applications to talk to DBs for which there's no specific driver. So if instead of using unixodbc you compile with res_mysql (which you have for your CDRs) and then configure your res_mysql.conf with the DB info + in your extconfig.conf say something like voicemail = mysql,DSN,vm table it should work. But what do I know. Maybe someone can confirm this. 2.) It's not exactly clear to me why my extconfig.conf should humour you 1) it's just a phrase (i.e. humour me) and 2) Wanted to see if you're configuring your extconfig.conf properly, along the lines of what I said above 3.) You're telling me (and everybody else here) that you have *it* running with MSSQL. But you're neither telling what *it* exactly is or does nor *how* you made it running. Maybe you want your extconfig.conf post here? Umm *it* is the whatever the subject of the email and discussion is(?) and how I got it running is by what Derek just said :P. Ihave to use unixODBC, FreeTDS to get it to work with MSSQL server and store the voicemails in a DB. ___ --Bandwidth 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] Trouble with regexten
Creating a context in your extensions.conf with the same name as your regcontext will cause all kinds of oddness to happen, among them this. What you need to do is have a differently-named context in extensions.conf with your 2-n priorities and include sip_autoreg in that. Regards, - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Russell Brown Sent: Thursday, November 30, 2006 4:14 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Trouble with regexten Can anyone help with the use of regexten? (* 1.4.3) I've got Asterisk creating extensions for my SIP phones using regexten but I can't seem to figure out how to make use of them once they're registered. Here's my dialplan for from-sip (the SIP's default context): asterisk*CLI dialplan show from-sip [ Context 'from-sip' created by 'pbx_config' ] '98766' =1. Dial(Sip/Tim) [pbx_config] 2. Hangup() [pbx_config] Include ='sip_autoreg'[pbx_config] Include ='widgets'[pbx_config] -= 1 extension (2 priorities) in 1 context. =- asterisk*CLI and here's sip_autoreg (the regexten context): asterisk*CLI dialplan show sip_autoreg [ Context 'sip_autoreg' created by 'pbx_config' ] '114' = 2. Dial(Sip/Tim) [pbx_config] 3. Hangup() [pbx_config] [ Context 'sip_autoreg' created by 'SIP' ] '112' = 1. Noop(Russell) [SIP] '113' = 1. Noop(Richard) [SIP] '114' = 1. Noop(Tim) [SIP] -= 4 extensions (5 priorities) in 2 contexts. =- asterisk*CLI Dialing 98766 from Sip/Russell rings Sip/Tim as expected. Dialing 114 gives Not Found :-( I'm very confused any ideas why this doesn't work? -- Regards, Russell | Russell Brown | MAIL: [EMAIL PROTECTED] PHONE: 01780 471800 | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.uk | ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Trouble using 2 IAX2 DiDs provided by different ITSPs
Asterisk 1.2.7 Redhat 9 I have DiDs from two different ITSP both set up as IAX2. Each one works when it's the only one in my iax.conf, but when I have them both defined in iax.conf at the same time, only one will work. My iax.conf is provided below. Any ideas how to fix? I'd like to use both DiDs! Thanks, H My iax.conf is below. When I dial the DiD provided by ITSP_B, the other ITSP seems to reject it. For example when I call the ITSP_B DiD, I get the following error message: Nov 29 21:50:17 NOTICE[23106]: chan_iax2.c:7203 socket_read: Host IP failed to authenticate as ITSP_A iax.conf == [general] register = my UserID:my password@ITSP A Server #1 domain register = my UserID:my password@ITSP A Server #2 domain register = my UserID:my password@ITSP B #1 domain notransfer=yes bindport=4569 bindaddr=0.0.0.0 bandwidth=low disallow=all allow=ulaw allow=g729 jitterbuffer=yes forcejitterbuffer=no tos=lowdelay autokill=yes [ITSP_B] context=incoming-iax type=friend qualify=2000 host=ITSP B #1 domain user=my UserID username=my UserID auth=md5 secret=my password disallow=all allow=ulaw ; ; *** ITSP_A Inbound *** [ITSP_A] context=incoming-iax type=user auth=md5 username=my UserID secret=my password disallow=all allow=ulaw ; ; *** ITSP_A Outbound *** [ITSP_A-Out] type=peer host=ITSP A Server #1 domain auth=md5 username=my UserID secret=my password disallow=all qualify=yes allow=ulaw ; [ITSP_A-Out2] type=peer host=ITSP A Server #2 domain auth=md5 username=my UserID secret=my password disallow=all qualify=yes allow=ulaw ___ --Bandwidth 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] Loosing IAX connection between offices
Setup: Office A: router: Linksys WRT54GS running SVEASOFT Alchemy-pre7a v3.37.6.8sv Asterisk: v.1.2.4 static IP Office B: router: Linksys WRT54GL running Linksys firmware v4.30.2 Asterisk: v.1.2.7.1 dynamic IP (using dyndns name) Office A is set up with refresh dns and cron job for iax2 reload every 5 minutes. It rarely looses connection to Office B. Surprisingly, Office B is the one loosing connection with Office A. I'm surprised because Office A is the one with the static IP address. When I do a IAX2 Show Peers, the connection will show as UNKNOWN or UNAVAILABLE. After loosing connection, the only way I can get it to reestablish is to reboot the * box. IAX2 reload doesn't solve it. I haven't been able to establish if it loosing the connection at a specific duration. Though, it seems to be random. iax.conf of Office B: [general] bindport = 4569 bindaddr = 0.0.0.0 disallow=all allow=ulaw allow=alaw allow=gsm mailboxdetail=yes notransfer=yes ;(- just added yesterday) [officeb-user] type=user secret=secret host=static ip address context=from-internal [officea] username=officea-user type=peer secret=secret qualify=4000 host=static ip address context=from-internal Any ideas on why Office B is loosing connection to Office A? or how to re-establish connection without rebooting? ___ --Bandwidth 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] Loosing IAX connection between offices
The linksys firmware on the WRT54G's on hardware versions 5 and above are notorious for layer 2 problems. Can you swap out that router? DM wrote: Setup: Office A: router: Linksys WRT54GS running SVEASOFT Alchemy-pre7a v3.37.6.8sv Asterisk: v.1.2.4 static IP Office B: router: Linksys WRT54GL running Linksys firmware v4.30.2 Asterisk: v.1.2.7.1 dynamic IP (using dyndns name) Office A is set up with refresh dns and cron job for iax2 reload every 5 minutes. It rarely looses connection to Office B. Surprisingly, Office B is the one loosing connection with Office A. I'm surprised because Office A is the one with the static IP address. When I do a IAX2 Show Peers, the connection will show as UNKNOWN or UNAVAILABLE. After loosing connection, the only way I can get it to reestablish is to reboot the * box. IAX2 reload doesn't solve it. I haven't been able to establish if it loosing the connection at a specific duration. Though, it seems to be random. iax.conf of Office B: [general] bindport = 4569 bindaddr = 0.0.0.0 disallow=all allow=ulaw allow=alaw allow=gsm mailboxdetail=yes notransfer=yes ;(- just added yesterday) [officeb-user] type=user secret=secret host=static ip address context=from-internal [officea] username=officea-user type=peer secret=secret qualify=4000 host=static ip address context=from-internal Any ideas on why Office B is loosing connection to Office A? or how to re-establish connection without rebooting? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jason The place where you made your stand never mattered, only that you were there... and still on your feet ___ --Bandwidth 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] IP call to extensions off my server
I have an asterisk server with TDM2402 card that has about 10 extensions on it. Both video phones and just audio phones. Normal calls coming in are received on the TDM lines and routed to an extension. If someone wants to call me based on my servers IP address and reach an extension on my server how is that done??? Jerry ___ --Bandwidth 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] Call recording with Asterisk BE
Hi Ed -
With Asterisk BE I am trying to record calls coming to a queue,. I am getting
the call to record, however the file name that the file saves to, is not the
correct one.
In my extensions.conf, I have the following entry to set the file name.
exten= 0072,4,Set(AGENTFILENAME=${CALLERID(number)}-${TIMESTAMP}-${EXTEN:4})
exten= 0072,5,Monitor(wav,${AGENTFILENAME}),m
exten= 0072,6,Queue(NOC)
I have also tried
exten= 0072,4,Set(AGENTFILENAME=${CALLERID(number)}-${TIMESTAMP}-${EXTEN:4})
exten= 0072,5,Monitor(wav,${AGENTFILENAME},m)
but this is what I am getting in the file name.
agent-1656-1164843488-241-in.wav
In the Asterisk console the name appears correctly however.
First off, use the second syntax for Monitor() - i.e. with the 'm'
inside the parentheses. Second, the ${EXTEN:4} effectively wipes out
your extension number since it is only four digits, so it's not really
necessary (unless you're also using it with longer extensions).
Beyond that I can tell you that this looks to be an issue with
Asterisk BE. I just tested and the same syntax works correctly with
Asterisk 1.2.X. It would be a good idea to bring this up with Digium
support.
- Noah
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Re: [asterisk-users] Monitoring awareness
I think you are right or i didn't find how to to it without using a conference. And even with conference didn't find a smart way to make it. Ondrej Valousek a écrit : Hi Steve, Ok Playback could be used here, indeed. But if you are using automonitor - by default activated by (*1) - I think there is no way how to implement this. Am I right? Thanks, Ondrej Steve Totaro wrote: [EMAIL PROTECTED] wrote: Hello, I'm discovering asterisk, it seem to be a great soft. I have seen a fonction to record calls that's a great fontion but there is something disturbing me. When the record start, except if the recorder prevent the other part, he is not aware of the recording... I dont find a way from the feature.conf how to play a sound when a monitor start to record :/ Either play a file with a beep or a verbal message that this call may be recorded for such and such reason. This can be done easily in the dialplan by calling playback or background prior to monitor. Depending on local laws, you may be OK if just one party on the call knows it is being recorded. Other states have different laws. I have no idea how the law works when one caller is in one state with one set of laws and the other caller is in a different state with different laws. Thanks, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The information contained in this e-mail and in any attachments is confidential and is designated solely for the attention of the intended recipient(s). If you are not an intended recipient, you must not use, disclose, copy, distribute or retain this e-mail or any part thereof. If you have received this e-mail in error, please notify the sender by return e-mail and delete all copies of this e-mail from your computer system(s). Please direct any additional queries to: [EMAIL PROTECTED] 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 ___ --Bandwidth 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] zombie SIP channels after CURL cnam lookup
Can anyone suggest a reason why these channels might end up zombies?
The process is;
Call comes in via SIP into a context that appends the caller ID name as
follows;
[cnam-lookup]
exten =
_[2-9]X,1,set(CALLERID(name)=${CURL(http://cnam.provider.com/?co
mpanyId=555password=passwordnumber=${CALLERID(num)})})
exten = _[2-9]X,2,goto(subscriber-numbers|${EXTEN}|1)
the call is then sent to the context where the extension is defined.
This works well with high volume, but there are occasionally zombie
channels as a result, can not track down the cause;
Channel Location State Application(Data)
SIP/1.1.251.9-b6700 [EMAIL PROTECTED] Ring(None)
SIP/1.1.251.9-b6a0d [EMAIL PROTECTED] Ring(None)
SIP/1.1.251.9-b6ad0 [EMAIL PROTECTED] Ring(None)
SIP/1.1.251.9-b7dcf [EMAIL PROTECTED] Ring(None)
SIP/1.1.251.9-b675f [EMAIL PROTECTED] Ring(None)
The channels listed above have appeared in show channels for 2 days now.
I assume it was either because the CURL response was not returned, since
we are still in he context cnam-lookup and the next step is a goto.
Is there a way to set an absolute timeout for the set command and
continue in the dialplan if that timeout is exceeded, without impacting
timeouts further down the line?
The cnam response should come within 200ms.
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Re: [asterisk-users] extension launch into AGI
Time Bandit wrote: I've got a simple set up with 1 fxo port and 1 fxs port in a Digium card connected to a POTS line and a phone set (physical extension). I've got all incoming calls launching directly into an AGI script. I'd like to do the same for the physical extension. In other words, when picking up the hand set, the AGI is launched without dialing any digits. check http://www.voip-info.org/wiki-Asterisk+config+zapata.conf keyword is : immediate Perfect. 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
RE: [asterisk-users] IP call to extensions off my server
That is a huge question, but the short answer is; They sent you s SIP invite to the [EMAIL PROTECTED] including whatever credentials are required to authenticate them based on how you have them defined in your sip.conf. You could allow anonymous, but be careful that the context it comes into does not allow the call out on your TDM line and get ready for VoIP spam!. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis Sent: Thursday, November 30, 2006 8:29 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] IP call to extensions off my server I have an asterisk server with TDM2402 card that has about 10 extensions on it. Both video phones and just audio phones. Normal calls coming in are received on the TDM lines and routed to an extension. If someone wants to call me based on my servers IP address and reach an extension on my server how is that done??? Jerry ___ --Bandwidth 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] 200+ analog phones connected to FXS modules
Or you could use a couple of these boxes: http://www.xorcom.com/astribank/features-32.html On 11/30/06, Vieri [EMAIL PROTECTED] wrote: I am trying to find out the best way to replace one of our hardware PBXs. It currently has 200+ analog phones connected to it. The idea is to take advantage of the already installed phone cables (big building) so I'm trying to avoid the use of ethernet adapters (if possible). However, I'm realizing that it's an expensive setup and will definitely require two or more cooperating Asterisk servers (cluster) mainly due to PCI slot availability. I am aware of the TDM2400P card. One could put 6 FXS uqad-modules and would serve 24 analog phones. However, I would need at least 9 of these PCI cards which could be placed in 2 or 3 servers. Is there another way of doing this (hopefully cheaper and more convenient)? Thank you for your suggestions. Vieri Yahoo! Music Unlimited Access over 1 million songs. http://music.yahoo.com/unlimited ___ --Bandwidth 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] meetme monitoring
Hello, I've a monitoring problem with app_meetme, I'd like to record a zap channel, which goes to a meetme conference Monitor doesn't record the voice of another members in the conference. Thanks any help Tamas ___ --Bandwidth 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] Billing Software
We are looking for an offline billing solution. We have a couple of particular requirements: 1) Since it's offline, we need to be able to import the CDR. 2) A way to support account credits based on referrals. Meaning, that if a member refers a new account, that member would get a free month of service, or similar type credits. 3) Generate invoices in either HTML or PDF format so they can be printed or emailed to the actual customers. Does anyone know of a package that supports this? Would prefer open source. Thanks, Daniel ___ --Bandwidth 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] Modprobe Zaptel
Hi Better late than ever, I guess, On Mon, Nov 27, 2006 at 10:18:56PM +, Julian Varanini wrote: Hi all For some dumb reason I decided to upgrade from Mandriva 2006 to 2007, thinking I could install asterisk all over again. Anyway I did install asterisk, zaptel and libpri. After install I ran modprobe zaptel which said zaptel not found. Thanks to help on this mailing list I had a fix to this problem and edited the Makefile located in /usr/src/linux/ to read -6mdv (instead of -6mdvcustom) which matched the uname -r. As an alternative: make KSRC=/path/to/kernel/src However in most systems that source tree would be pointed from /lib/modules/`uname -r`/build , and this is wher eZaptel looks by default . Then I installed zaptel again and the drivers were still installed in /lib/modules/2.6.17-6mdvcustom and not /lib/modules/2.6.17-6mdv. This may suggest that the tree agaist which you built is not configured to work with your configuration. After many reinstalls and reboots I could not find why they were still moved to that location, so I just moved them from -6mdvcustom to -6mdv and modprobe zaptel did not display any errors. However I needed to run modprobe wcte11xp for it to actually load the driver, when I did not need to do this in 2006 Does anyone know why this is? I can get it all to start up at boot using rc.local, but when I installed zaptel on Mandriva 2006 it loaded at boot on its own. In a properly-set system, hotplug would load the drivers. Which hardware do you have? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth 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: MeetMe announcements and SIP channels
In article [EMAIL PROTECTED], Mike [EMAIL PROTECTED] wrote: Just curious if anyone knows of any hacks to enable announce entry/exit in MeetMe conferences with SIP (as opposed to ZAP) channels since the |i option will not work with SIP. The |i option does indeed work with SIP. You do have to have the zaptel driver loaded, and either ztdummy or a card driver, AND asterisk must load chan_zap.so Meetme won't work at all without zaptel, but if you want entry/exit announcements and/or recording to work, you also need chan_zap.so If you have done the above and it still doesn't work, what behaviour and log messages do you observe? Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: AGI PHP Issues (AGI script runs but phone hangs up too quickly)
In article [EMAIL PROTECTED], Chris Blunt [EMAIL PROTECTED] wrote: Sorry to re-post this but I'm sure it's something simple that someone has found before. To summarise: Dial plan answers the phone AGI script executes AGI debug in console show phonetics ABC - However no audio at all on the phone and this step is less than 1 second. Dial plan Busy Phone hangs up. Perhaps Asterisk is playing the phonetics in the background (like Background instead of like Playback)? Try putting a Wait(5) after the AGI (or even a sleep(5) within it), to see. General question for anyone that knows: Is there a way in the Dialplan or or in AGI to wait for Background messages to finish playing? Perhaps doing a Playback of a file containing a very short length of silence? Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trouble with regexten
When using autoreg, is there any way to extract the userid somehow? IE:
SIP.com
regcontext=registrations
[123]
regexten=2125551212
extensions.conf
[phones]
include = registrations
exten = _212NXX,2,Dial(SIP/${VARIABLE}))
exten = _212NXX,3,VoiceMail(u${EXTEN})
Honestly I dont see the point of autoreg unless this can be done...
On 11/30/06, Watkins, Bradley [EMAIL PROTECTED] wrote:
Creating a context in your extensions.conf with the same name as your
regcontext will cause all kinds of oddness to happen, among them this.
What you need to do is have a differently-named context in
extensions.conf with your 2-n priorities and include sip_autoreg in
that.
Regards,
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Russell Brown
Sent: Thursday, November 30, 2006 4:14 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Trouble with regexten
Can anyone help with the use of regexten? (* 1.4.3)
I've got Asterisk creating extensions for my SIP phones using
regexten but I can't seem to figure out how to make use of
them once they're registered.
Here's my dialplan for from-sip (the SIP's default context):
asterisk*CLI dialplan show from-sip
[ Context 'from-sip' created by 'pbx_config' ]
'98766' =1. Dial(Sip/Tim) [pbx_config]
2. Hangup() [pbx_config]
Include ='sip_autoreg'[pbx_config]
Include ='widgets'[pbx_config]
-= 1 extension (2 priorities) in 1 context. =-
asterisk*CLI
and here's sip_autoreg (the regexten context):
asterisk*CLI dialplan show sip_autoreg
[ Context 'sip_autoreg' created by 'pbx_config' ]
'114' = 2. Dial(Sip/Tim) [pbx_config]
3. Hangup() [pbx_config]
[ Context 'sip_autoreg' created by 'SIP' ]
'112' = 1. Noop(Russell) [SIP]
'113' = 1. Noop(Richard) [SIP]
'114' = 1. Noop(Tim) [SIP]
-= 4 extensions (5 priorities) in 2 contexts. =-
asterisk*CLI
Dialing 98766 from Sip/Russell rings Sip/Tim as expected.
Dialing 114 gives Not Found :-(
I'm very confused any ideas why this doesn't work?
--
Regards,
Russell
| Russell Brown | MAIL: [EMAIL PROTECTED] PHONE: 01780 471800 |
| Lady Lodge Systems | WWW Work: http://www.lls.com |
| Peterborough, England | WWW Play: http://www.ruffle.me.uk |
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Re: [asterisk-users] Loosing IAX connection between offices
I assume you are referring to the 54GL in Office B? What about replacing the firmware with SVEASOFT or DDWRT? Would this fix it? On 11/30/06, jason [EMAIL PROTECTED] wrote: The linksys firmware on the WRT54G's on hardware versions 5 and above are notorious for layer 2 problems. Can you swap out that router? DM wrote: Setup: Office A: router: Linksys WRT54GS running SVEASOFT Alchemy-pre7a v3.37.6.8sv Asterisk: v.1.2.4 static IP Office B: router: Linksys WRT54GL running Linksys firmware v4.30.2 Asterisk: v.1.2.7.1 dynamic IP (using dyndns name) ___ --Bandwidth 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] zaptel compilation problems with linux 2.6.19
Hello! I have problems compiling zaptel (tried 1.2.11, 1.2.10 and 1.4.0-beta2 -- all give the same error) with 2.6.19 kernel CC [M] /home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/card_fxo.o In file included from /home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/xpd.h:26, from /home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/card_fxo.c:28: /home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/xdefs.h:93: error: conflicting types for 'bool' include/linux/types.h:36: error: previous declaration of 'bool' was here In file included from /home/roman/install/asterisk/zaptel-1.4.0-beta2/zconfig.h:9, from /home/roman/install/asterisk/zaptel-1.4.0-beta2/zaptel.h:34, from /home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/xproto.h:29, from /home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/xpd.h:27, from /home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/card_fxo.c:28: include/linux/config.h:10:3: warning: no newline at end of file make[3]: *** [/home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/card_fxo.o] Error 1 make[2]: *** [/home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp] Error 2 make[1]: *** [_module_/home/roman/install/asterisk/zaptel-1.4.0-beta2] Error 2 make[1]: Leaving directory `/home/roman/install/kernel/linux-2.6.19' make: *** [linux26] Error 2 seems that commenting out typedef int bool; in xpp/xdefs.h on line 93 works that out, but don't know if it's completely right thing to do Roman ___ --Bandwidth 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: Spandsp rxfax txtax fails no errors
I'm compiling from downloded source:
http://soft-switch.org/downloads/spandsp/spandsp-0.0.3pre26.tgz
and
http://soft-switch.org/downloads/snapshots/spandsp/test-apps-asterisk-1.4/*
on a slackware 11 system with asterisk 1.4 beta3
Note in the message below I've added information about another step
taken in debugging the problem by modifying app_txfax.c and recompiling.
It finally gave me a log entry and exited. The console noted that it
exited with a non zero value.
Thanks
Thanks for the response!!!
I enabled debuging in the menuselect configuration for compiling
asterisk 1.4 beta3. In logging.conf enabled debug loggin to the
/var/log/asterisk/debug file and to the console. Restarted (not just
reload) asterisk and there is plenty of general debugging info in the
debug log file. I also am calling the fax apps with debug argument as
follows
exten = fax,n,rxfax(${FAXFILE}|debug)
and
exten = fax,n,rxfax(${FAXFILE}|debug)
Looking at the code in app_rxfax.c and app_txfax.c there should be
plenty of information in the debug log on failure or success. However
I haven't found any debug log information that should be generated. It
is like it just does a return 0 at the beginning of the application.
I found some documentation on the system() call that says that the
dial plan will jump to n+101 priority if the return value is not 0.
So I setup the dial plan:
[outgoingfax]
exten = out_fax,1,Wait(2)
exten = out_fax,2,txfax(${TXFAX_NAME}|caller|debug)
exten = out_fax,3,system(echo sent fax file ${TXFAX_NAME}
/tmp/fax.log )
exten = out_fax,4,Hangup
exten = out_fax,103,system(echo failed fax file ${TXFAX_NAME}
/tmp/fax.log )
exten = h,1,Hangup()
No /tmp/fax.log file created at all.
asterisk -rdddv
-- Executing [EMAIL PROTECTED]:1] Wait(SIP/inettrunk-081e8100,
2) in new stack
-- Executing [EMAIL PROTECTED]:2] TxFAX(SIP/inettrunk-081e8100,
/tmp/test.tif) in new stack
[Nov 29 13:26:13] DEBUG[28613]: pbx_spool.c:391 scan_service: Delaying
retry since we're currently running
'[EMAIL PROTECTED]@ol/asterisk/outgoing/fax.call'
[Nov 29 13:26:24] DEBUG[28613]: pbx_spool.c:391 scan_service: Delaying
retry since we're currently running
'/var/spool/asterisk/outgoing/fax.call'
[Nov 29 13:26:35] DEBUG[28613]: pbx_spool.c:391 scan_service: Delaying
retry since we're currently running 'h�,[EMAIL PROTECTED]/asterisk/outgoing/fax.call'
From this it looks like it just gets stuck in the TxFAX app.
I've modified app_txfax.c slightly to see if the app can run and return
with the following code near the beginning in txfax_exec
code:
uint8_t __buf[sizeof(uint16_t)*MAX_BLOCK_SIZE + 2*AST_FRIENDLY_OFFSET];
uint8_t *buf = __buf + AST_FRIENDLY_OFFSET;
ast_log(LOG_WARNING, Made it in and going out. Giving up.\n);
return -1;
if (chan == NULL)
code end
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[asterisk-users] T1's in St. Lucia
Does anyone on this list know of a reputable T1/PRI provider in St. Lucia? If so, what monthly costs am I looking at? I do know that Cable and Wireless are the biggest Telco. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with ZapRAS and asterisk
Hi, I am trying to use Asterisk cmd ZapRAS (http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ZapRAS), I pathed the ppp daemon ftp://ftp.digium.com/pub/zaptel/misc/, but when I try to use it, I obtain the following error: Connected to Asterisk 1.2.4 currently running on TSU-R1 (pid = 7242) Verbosity was 0 and is now 44 -- Accepting call from '123456789' to '9022' on channel 0/1, span 1 -- Executing Answer(Zap/1-1, ) in new stack -- Executing ZapRAS(Zap/1-1, debug|64000|noauth|netmask|255.255.255.0|192.168.10.2:192.168.10.1) in new stack -- Starting RAS on Zap/1-1 Nov 30 17:02:03 WARNING[7293]: app_zapras.c:172 run_ras: wait4 returned -1: No child processes -- RAS on Zap/1-1 terminated with status 0 == Spawn extension (pri1, 9022, 2) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' In the /var/log/message I find the following message: Nov 30 17:02:03 TSU-R1 pppd[7294]: Plugin zaptel.so loaded. Nov 30 17:02:03 TSU-R1 pppd[7294]: Zaptel Plugin Initialized Nov 30 17:02:03 TSU-R1 pppd[7294]: no device specified and stdin is not a tty In the source code (asterisk) app_zapras.c before running pppd demon a call to dup2 is done to have the zaptel channel descriptor in the STDIN descriptor The patched pppd daemon in /usr/sbin/pppd, I recompiled zaptel and asterisk and all it seem OK. Can someone to help me ? Achille ___ --Bandwidth 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] Live call monitoring
I've noticed that some products, like Fonality's HUD, allow live
monitoring of a VoIP call (not just Zap Barge). The Asterisk {client |
manager} command set only seems to allow recording to a file without the
use of a meetme room. Does anyone have a good solution for this?
What I'd like to implement, ideally, is that once an incoming call is
transferred to a particular operator, the system also calls a manager
who can monitor silently.
Any help is much appreciated!
Yaakov
--
Yaakov Menken
Capalon Communications, Inc.
Ask us about Voice over IP for Business!
http://www.capalon.com
888-CAPALON (227-2566)
410-358-9800 x120
410-510-1053 fax
443-413-1042 cell
[EMAIL PROTECTED]
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Re: [asterisk-users] 200+ analog phones connected to FXS modules
On Thursday 30 November 2006 06:13, Zoa wrote: You could go for 2 quad pri cards + channel banks or for TDMoE or usb channel banks. The last option would be the cheaper and more scalable one imho The scale here is already bordering on unrealistic. I wouldn't expect them to want to make this too much bigger. Scaling aside, 200+ ulaw channels (13Mb/sec not including overhead) over USB? You're joking, right? I know that USB 2.0 is rated at 480Mbps but has this actually been tested and verified? I would imagine USB 2.0 chipset variants and inconsistencies are even worse than that of PCI. And individual RJ11 jacks? I'd strongly suggest the dual TE407P with channel banks. D50 connection, hardware echo cancel/dtmf/etc. Offload all transcoding to another box or set of boxes. You can even use cheapass Carrier Access Access Bank I or IIs off of ebay (sub-$100 range), as this is FXS and you don't need CPD. www.spidermux.org Interesting product, I didn't know about this one until just now. I've heard that TDMoE is more trouble than it's worth, though, and may eventually be phased out of Asterisk. Can anyone from Digium give some more information or suggestions? -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Trouble with regexten
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Joakimsen
Sent: Thursday, November 30, 2006 11:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trouble with regexten
When using autoreg, is there any way to extract the userid
somehow? IE:
SIP.com
regcontext=registrations
[123]
regexten=2125551212
extensions.conf
[phones]
include = registrations
exten = _212NXX,2,Dial(SIP/${VARIABLE}))
exten = _212NXX,3,VoiceMail(u${EXTEN})
Honestly I dont see the point of autoreg unless this can be
done...
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Re: [asterisk-users] Digium through Octasic
On Thursday 23 November 2006 11:44, Heidi Mendoza wrote: We're looking at using 4 or 8 port T1 cards with echo cancellation and are evaluating brands to go with. We know that Sangoma has excellent solutions especially when it comes to echo. But we still have to hear about actual performance of a Digium card using the same Octasic DSP echo canceller. Excellent performance. I had an A104d which was giving some very odd audio artifacting, Sangoma replaced the card but did not test the original to ensure that the card was indeed defective or that the problem was solved with the replacement. I haven't put the replacement in service yet, as I had a TE407P on order and it arrived first. :-) After dealing with the crap that the TE406P was, the TE407P is *heaven*. Highly recommended. -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
[asterisk-users] Asterisk 1.4 : App_Swift (Cepstral) Howto
Hi All, Recent discussions on app_cepstral on the list have led me to believe there's some issues with Asterisk 1.4 I set about creating a small howto for people to get cepstral, with app_swift working. Check it out: http://www.voipphreak.ca/archives/354-Asterisk-1.4-Gentoo-CepstralSwift-Howto-using-App_Swift.html Thanks, Diwelf ___ --Bandwidth 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] Trouble with regexten
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Joakimsen
Sent: Thursday, November 30, 2006 11:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trouble with regexten
When using autoreg, is there any way to extract the userid
somehow? IE:
SIP.com
regcontext=registrations
[123]
regexten=2125551212
extensions.conf
[phones]
include = registrations
exten = _212NXX,2,Dial(SIP/${VARIABLE}))
exten = _212NXX,3,VoiceMail(u${EXTEN})
Honestly I dont see the point of autoreg unless this can be
done...
The answer is no, but I'm not sure what you're expecting. This is no
different than if you weren't using regexten. You would still need a
way to map the DID to the proper device.
Regards,
- Brad
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Re: [asterisk-users] Loosing IAX connection between offices
DM wrote: snip Office A is set up with refresh dns and cron job for iax2 reload every 5 minutes. It rarely looses connection to Office B. Surprisingly, Office B is the one loosing connection with Office A. I'm surprised because Office A is the one with the static IP address. When I do a IAX2 Show Peers, the connection will show as UNKNOWN or UNAVAILABLE. After loosing connection, the only way I can get it to reestablish is to reboot the * box. IAX2 reload doesn't solve it. I haven't been able to establish if it loosing the connection at a specific duration. Though, it seems to be random. snip Any reason you don't have Office B register with Office A and thereby tell Office A what IP office B has? It would remove the need to refresh DNS and reload IAX2 all the time. I use this method to connect my home system (dynamic) to my work system (static) and it works well. -Dave ___ --Bandwidth 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] Live call monitoring
What I'd like to implement, ideally, is that once an incoming call is transferred to a particular operator, the system also calls a manager who can monitor silently. I think you are looking for this : http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanSpy ___ --Bandwidth 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] 200+ analog phones connected to FXS modules
Interesting product, I didn't know about this one until just now. I've heard that TDMoE is more trouble than it's worth, though, and may eventually be phased out of Asterisk. Can anyone from Digium give some more information or suggestions? -A. I'm not from digium but am the proud owner of a preproduction sample of the spidermux, i also took it to Astricon Dallas. (they are already being produced but are not being sold yet). The TDMoE implementation in asterisk works, but is not used by a lot of people or hardware yet, so it needs some work (Especially to make it work with recent kernels). I know the spidermux people already have a bunch of patches ready to be released to fix the issues that exist now. I've never heard something about tdmoe being phased out of asterisk. Zoa. ___ --Bandwidth 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: AGI PHP Issues (AGI script runs but phone hangs up too quickly)
Tony Mountifield wrote: In article [EMAIL PROTECTED], Chris Blunt [EMAIL PROTECTED] wrote: Sorry to re-post this but I'm sure it's something simple that someone has found before. To summarise: Dial plan answers the phone AGI script executes AGI debug in console show phonetics ABC - However no audio at all on the phone and this step is less than 1 second. Dial plan Busy Phone hangs up. Perhaps Asterisk is playing the phonetics in the background (like Background instead of like Playback)? Try putting a Wait(5) after the AGI (or even a sleep(5) within it), to see. General question for anyone that knows: Is there a way in the Dialplan or or in AGI to wait for Background messages to finish playing? Perhaps doing a Playback of a file containing a very short length of silence? Cheers Tony And also, you need to run Answer before Playback. This can be done in the AGI script, or dialplan before execution of AGI. Ove ___ --Bandwidth 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 1.4 : App_Swift (Cepstral) Howto
Great link. After I all you said I get this error loading the module in asterisk via load app_swift The 'load' command is deprecated and will be removed in a future release. Please use 'module load' instead. [Nov 30 13:54:08] WARNING[7825]: loader.c:362 load_dynamic_module: Error loading module 'app_swift': libswift.so.4: cannot open shared object file: No such file or directory [Nov 30 13:54:08] WARNING[7825]: loader.c:607 load_resource: Module 'app_swift' could not be loaded. Any ideas? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Gibson Sent: Thursday, November 30, 2006 1:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk 1.4 : App_Swift (Cepstral) Howto Hi All, Recent discussions on app_cepstral on the list have led me to believe there's some issues with Asterisk 1.4 I set about creating a small howto for people to get cepstral, with app_swift working. Check it out: http://www.voipphreak.ca/archives/354-Asterisk-1.4-Gentoo-CepstralSwift- Howto-using-App_Swift.html Thanks, Diwelf ___ --Bandwidth 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] T1's in St. Lucia
Forum wrote: Does anyone on this list know of a reputable T1/PRI provider in St. Lucia? If so, what monthly costs am I looking at? I do know that Cable and Wireless are the biggest Telco. I think you will find they are the only telco and the cost will be enormous. -- Chris Mason (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth 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 1.4 : App_Swift (Cepstral) Howto
Great link. After I all you said I get this error loading the module in asterisk via load app_swift The 'load' command is deprecated and will be removed in a future release. Please use 'module load' instead. [Nov 30 13:54:08] WARNING[7825]: loader.c:362 load_dynamic_module: Error loading module 'app_swift': libswift.so.4: cannot open shared object file: No such file or directory [Nov 30 13:54:08] WARNING[7825]: loader.c:607 load_resource: Module 'app_swift' could not be loaded. Any ideas? The README file reminds you to do this: Install one of the Cepstral Voices. Use the standard install directory /opt. On Linux don't forget to insert /opt/swift/lib into your /etc/ld.so.conf file and run ldconfig. Make sure you've got /opt/swift/lib in your ld.so.conf file! -MC ___ --Bandwidth 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] Loosing IAX connection between offices
if its a version 5 or higher, that wont be an option, but if its not, give openwrt or ddwrt a try. DM wrote: I assume you are referring to the 54GL in Office B? What about replacing the firmware with SVEASOFT or DDWRT? Would this fix it? On 11/30/06, jason [EMAIL PROTECTED] wrote: The linksys firmware on the WRT54G's on hardware versions 5 and above are notorious for layer 2 problems. Can you swap out that router? DM wrote: Setup: Office A: router: Linksys WRT54GS running SVEASOFT Alchemy-pre7a v3.37.6.8sv Asterisk: v.1.2.4 static IP Office B: router: Linksys WRT54GL running Linksys firmware v4.30.2 Asterisk: v.1.2.7.1 dynamic IP (using dyndns name) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jason The place where you made your stand never mattered, only that you were there... and still on your feet ___ --Bandwidth 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 1.4 : App_Swift (Cepstral) Howto
Fixed my problem! Note to self... READ EVERYTHING in the instructions! Again thanks for the information! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric M. Sent: Thursday, November 30, 2006 1:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Asterisk 1.4 : App_Swift (Cepstral) Howto Great link. After I all you said I get this error loading the module in asterisk via load app_swift The 'load' command is deprecated and will be removed in a future release. Please use 'module load' instead. [Nov 30 13:54:08] WARNING[7825]: loader.c:362 load_dynamic_module: Error loading module 'app_swift': libswift.so.4: cannot open shared object file: No such file or directory [Nov 30 13:54:08] WARNING[7825]: loader.c:607 load_resource: Module 'app_swift' could not be loaded. Any ideas? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Gibson Sent: Thursday, November 30, 2006 1:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk 1.4 : App_Swift (Cepstral) Howto Hi All, Recent discussions on app_cepstral on the list have led me to believe there's some issues with Asterisk 1.4 I set about creating a small howto for people to get cepstral, with app_swift working. Check it out: http://www.voipphreak.ca/archives/354-Asterisk-1.4-Gentoo-CepstralSwift- Howto-using-App_Swift.html Thanks, Diwelf ___ --Bandwidth 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] zaptel compilation problems with linux 2.6.19
I'm having problems installing ztdummy on my CentOS-2.6.16-rc6/Asterisk-1.2.11 datacenter server (no ZAP HW, SIP only to PSTN). I unpacked the kernel sources and headers in a directory, made (but not re/installed) the kernel, unpacked the zaptel-1.2.11 tarball, then went thru the make sequence. It seemed to proceed OK (without errors, just some warnings), but didn't seem to result in a loadable ztdummy kernel module. Complete (failed) install session transcript is attached to this message; details appended: - # cd path-to-zaptel-1.2.11-source # export KSRC=path-to-kernel-source-root-dir # make clean # make config [... series of shell script conditionals apparently executed OK ...] # make linux26 [... series of CC/LD reports, some warnings, no errors ...] # make install [... series of INSTALL messages, same warnings from (make linux26), no errors ...] # modprobe ztdummy FATAL: Module ztdummy not found. FATAL: Error running install command for ztdummy # modprobe zaptel FATAL: Module zaptel not found. - (make linux26) generated some warnings about various usb_*_dev symbols undefined in [xpp,wcusb]/*.ko, but no actual errors. (make install) repeated those warnings. (modprobe ztdummy) finished with - Building /etc/modprobe.d/zaptel... *** *** WARNING: *** If you had custom settings in /etc/modprobe.d/zaptel, *** they have been moved to /etc/modprobe.d/zaptel.bak. [...] - but seemed to complete without errors. (make install) included a line - INSTALL zaptel-1.2.11-source-root-dir/ztdummy.ko - Complete (failed) install session transcript is attached. On Thu, 2006-11-30 at 12:00 -0700, [EMAIL PROTECTED] wrote: Date: Thu, 30 Nov 2006 19:19:14 +0200 From: Roman Yeryomin [EMAIL PROTECTED] Subject: [asterisk-users] zaptel compilation problems with linux 2.6.19 To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Hello! I have problems compiling zaptel (tried 1.2.11, 1.2.10 and 1.4.0-beta2 -- all give the same error) with 2.6.19 kernel CC [M] /home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/card_fxo.o In file included from /home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/xpd.h:26, from /home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/card_fxo.c:28: /home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/xdefs.h:93: error: conflicting types for 'bool' include/linux/types.h:36: error: previous declaration of 'bool' was here In file included from /home/roman/install/asterisk/zaptel-1.4.0-beta2/zconfig.h:9, from /home/roman/install/asterisk/zaptel-1.4.0-beta2/zaptel.h:34, from /home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/xproto.h:29, from /home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/xpd.h:27, from /home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/card_fxo.c:28: include/linux/config.h:10:3: warning: no newline at end of file make[3]: *** [/home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/card_fxo.o] Error 1 make[2]: *** [/home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp] Error 2 make[1]: *** [_module_/home/roman/install/asterisk/zaptel-1.4.0-beta2] Error 2 make[1]: Leaving directory `/home/roman/install/kernel/linux-2.6.19' make: *** [linux26] Error 2 seems that commenting out typedef int bool; in xpp/xdefs.h on line 93 works that out, but don't know if it's completely right thing to do Roman -- (C) Matthew Rubenstein # make clean rm -f torisatool makefw tor2fw.h radfw.h rm -f ztcfg torisatool makefw ztmonitor ztspeed zttool zttest fxotune rm -f *.o ztcfg tzdriver sethdlc sethdlc-new rm -f zonedata.lo tonezone.lo libtonezone.so *.lo make -C kernel-source-root-dir SUBDIRS=zaptel-1.2.11-source-root-dir clean make[1]: Entering directory `kernel-source-root-dir' CLEAN zaptel-1.2.11-source-root-dir/wct4xxp CLEAN zaptel-1.2.11-source-root-dir/.tmp_versions make[1]: Leaving directory `kernel-source-root-dir' rm -rf .tmp_versions rm -f gendigits tones.h rm -f libtonezone* rm -f tor2ee rm -f fxotune rm -f core rm -f ztcfg-shared fxstest rm -rf misdn* rm -rf mISDNuser* # make config if [ -d /etc/rc.d/init.d ]; then \ install -D -m 755 zaptel.init /etc/rc.d/init.d/zaptel; \ //sbin/chkconfig --add zaptel; \ elif [ -d /etc/init.d ]; then \ install -D -m 755 zaptel.init /etc/init.d/zaptel; \ //sbin/chkconfig --add zaptel; \ fi if [ -d /etc/default ] [ ! -f /etc/default/zaptel ]; then \ install -D -m 644 zaptel.sysconfig /etc/default/zaptel; \ fi if [ -d /etc/sysconfig ] [ ! -f
Re: [asterisk-users] zaptel compilation problems with linux 2.6.19
On Thu, Nov 30, 2006 at 07:19:14PM +0200, Roman Yeryomin wrote: Hello! I have problems compiling zaptel (tried 1.2.11, 1.2.10 and 1.4.0-beta2 -- all give the same error) with 2.6.19 kernel CC [M] /home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/card_fxo.o In file included from /home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/xpd.h:26, from /home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/card_fxo.c:28: /home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/xdefs.h:93: error: conflicting types for 'bool' include/linux/types.h:36: error: previous declaration of 'bool' was here In file included from /home/roman/install/asterisk/zaptel-1.4.0-beta2/zconfig.h:9, from /home/roman/install/asterisk/zaptel-1.4.0-beta2/zaptel.h:34, from /home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/xproto.h:29, from /home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/xpd.h:27, from /home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/card_fxo.c:28: include/linux/config.h:10:3: warning: no newline at end of file make[3]: *** [/home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/card_fxo.o] Error 1 make[2]: *** [/home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp] Error 2 make[1]: *** [_module_/home/roman/install/asterisk/zaptel-1.4.0-beta2] Error 2 make[1]: Leaving directory `/home/roman/install/kernel/linux-2.6.19' make: *** [linux26] Error 2 seems that commenting out typedef int bool; in xpp/xdefs.h on line 93 works that out, but don't know if it's completely right thing to do Simply replacing that int with a _Bool will give several incompatible pointer type warnings. The following is from our internal working copy, with pathes removed for clarity: CC [M] xpp/card_fxo.o xpp/card_fxo.c: In function `__check_report_battery': xpp/card_fxo.c:38: warning: return from incompatible pointer type CC [M] xpp/card_fxs.o xpp/card_fxs.c: In function `__check_poll_digital_inputs': xpp/card_fxs.c:37: warning: return from incompatible pointer type CC [M] xpp/xbus-core.o CC [M] xpp/xpp_zap.o xpp/xpp_zap.c: In function `__check_zap_autoreg': xpp/xpp_zap.c:67: warning: return from incompatible pointer type xpp/xpp_zap.c: In function `__check_prefmaster': xpp/xpp_zap.c:68: warning: return from incompatible pointer type xpp/xpp_zap.c: In function `__check_xpp_ec': xpp/xpp_zap.c:70: warning: return from incompatible pointer type xpp/xpp_zap.c: In function `xpd_read_proc': xpp/xpp_zap.c:437: warning: unused variable `chans' xpp/xpp_zap.c: In function `proc_sync_write': xpp/xpp_zap.c:748: warning: int format, bool arg (arg 5) xpp/xpp_zap.c: In function `proc_xpd_ztregister_write': xpp/xpp_zap.c:816: warning: int format, bool arg (arg 3) Most of them seem to be related to the procfs interface. If you don't need xpp for yourself and can leave with those warnings, go ahead. I'll try to resolve them. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to park calls on a specific extension
On Nov 29, 2006, at 11:40 PM, Lacy Moore - Aspendora wrote: [snip] I went from a Lucent Merlin Legend system to Asterisk. For me, it's a tradeoff for features. To my users, it was a step backward. I also upgraded an office from a Partner system to Asterisk. To the users, it is a huge step backward. They have yet to figure out how to transfer a call. On their old system, they put the call on hold and pressed the line button at another phone. Today, they hold the phone against their leg so the caller doesn't here them yell for the person to come to the phone, and then the person who the call is for comes to that phone and answers the call. It will remain that way for them, because learning how to do it the right way takes more work than the person coming to the phone, or so they say. Sounds like you have stumbled upon one of the truisms of replacing office phone systems: people hate it when you take away their key system lines. So you're right, it would be a good idea for Asterisk to implement that functionality, and they are working on it, IIRC. It also sounds like you need to talk to your customers more before you roll out a system, so they know ahead of time what the interface will be and what changes they should expect. If you let them know ahead of time, and get management on board, you should be OK. I won't be doing another Asterisk install for a while. Customer #2 has made sure of that by telling everyone how their new phone system sucks. Until I can find a suitable solution, I am dead in the water. And yes, I am trying to learn C so that I can write it myself, or modify something else to make it work. Given the flexibility inherent in Asterisk, you really shouldn't have to code your own. It's a great skill, but not necessary. But seriously, the attitude of either write it yourself or deal with it won't cut it for business users. If Asterisk is only for geeks, then fine, it will work perfectly. Well, not to be rude, but if you plan to sell, install, and maintain Asterisk systems, you shouldn't be just a business user, you should be at least a little bit geeky. I would suggest that Asterisk works excellently for business users, but it requires a person who is a bit of a geek to set it up properly for those business users so they don't notice how geeky it really is. If all phones behaved the same, it would help. Cisco, using SIP, has no park button. Cisco, using chan_sccp, has a great parking concept. Polycom has a park button that doesn't appear to work with Asterisk. We use Cisco (SIP) and Polycom. Aastra and SNOM seem to have an easier parking interface. The chan_sccp implementation not only reads back the parking spot, but also displays it on the screen. Why don't you take the specific phone interface out of it? Most of your (and your users') gripes seem to be things that could be resolved with a little research, planning, and a better grasp of Asterisk configuration. for example: In your example above where they can't figure out how to transfer, why don't you edit features.conf and define the transfer key as # or something. Then, when they have a call for Bill across they way, they can do this: 1.) Answer call, determine call is for Bill. 2.) Press #. Asterisk reads back Transfer. 3.) Dial parking extension number (700, for example) 4.) System reads back parking space number (703, for example) 5.) Call or shout to Bill You have a call on 703 This is not really much harder or more complicated than what they are used to with their old key system: 1.) Same as above 2.) Press Hold Button 3.) Look at phone to see which line # 4.) Call or shout to Bill You have a call on Line X This approach also cuts out the press More button, press Transfer Button issue you mention below. Getting users to make that change shouldn't really be that difficult, especially if you let the customer know what to expect from the beginning. Focus on management and stress the advantages they receive as a result of Asterisk being a full-fledged PBX, not a key system. Then explain that minor changes in the user interface are the small price they must pay for those advantages. What I have tried to do is the following scenario. Assign two line keys as Park 720 and Park 721, and using third party patches, been able to monitor those lines (which are actually parking spots) using hints. Also, using third party patches, I can transfer to those lines (transfer directly to a parking spot), but again, that is a several step process (it requires a blind transfer which take pressing transfer, then blind on the Polycom, this method, due to no BLF does not work on the Ciscos) that just won't happen in small businesses. It just takes too many button presses. Plus, as I mentioned, this is third party patches that aren't in the Asterisk main branch, and makes upgrades near
RE: [asterisk-users] Voicemail callback bug?
Which version? Similar issues parsing callback number in 1.2.12 -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Kristian Kielhofner Sent: Thursday, September 28, 2006 10:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Voicemail callback bug? Hello everyone, I'm having a problem with voicemail callback (option 3 after message, option 1 to send a reply). Here is what happens: -- Playing '/var/spool/asterisk/voicemail/default/400/INBOX/msg0001' (language 'en') -- Playing 'vm-prev' (language 'en') -- Playing 'vm-advopts' (language 'en') -- Playing 'vm-toreply' (language 'en') -- Playing 'vm-tohearenv' (language 'en') -- Playing 'vm-starmain' (language 'en') == Parsing '/var/spool/asterisk/voicemail/default/400/INBOX/msg0001.txt': Found -- Leaving voicemail for '3@' in context 'starbox_11' Sep 28 16:14:03 WARNING[1749]: app_voicemail.c:2412 leave_voicemail: No entry in voicemail config file for '3' -- Playing 'vm-prev' (language 'en') -- Playing 'vm-advopts' (language 'en') -- Playing 'vm-toreply' (language 'en') msg0001.txt looks like this: [message] origmailbox=400 context=vm-in macrocontext= exten=vmu priority=106 callerchan=SIP/vm-082b9f78 callerid=Buck Aneer 300 origdate=Thu Sep 28 04:18:36 PM UTC 2006 origtime=1159460316 category= duration=7 No entry in voicemail config file for 3 leads me to think that Asterisk is parsing 300 from the callerid line above as 3, which obviously isn't correct. There is a 300 in the voicemail config file, there just isn't a three. The interesting thing is if you go the other direction (400 leaving vm for 300), Asterisk parses the callerid as 40 instead of 400. Closer, but still no cigar... Any thoughts? Thanks! -- Kristian Kielhofner ___ --Bandwidth 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] zaptel compilation problems with linux 2.6.19
I'm having problems installing ztdummy on my CentOS-2.6.16-rc6/Asterisk-1.2.11 datacenter server (no ZAP HW, SIP only to PSTN). I unpacked the kernel sources and headers in a directory, made (but not re/installed) the kernel, unpacked the zaptel-1.2.11 tarball, then went thru the make sequence. It seemed to proceed OK (without errors, just some warnings), but didn't seem to result in a loadable ztdummy kernel module. Details appended, followed by a complete (failed) install session transcript: - # cd path-to-zaptel-1.2.11-source # export KSRC=path-to-kernel-source-root-dir # make clean # make config [... series of shell script conditionals apparently executed OK ...] # make linux26 [... series of CC/LD reports, some warnings, no errors ...] # make install [... series of INSTALL messages, same warnings from (make linux26), no errors ...] # modprobe ztdummy FATAL: Module ztdummy not found. FATAL: Error running install command for ztdummy # modprobe zaptel FATAL: Module zaptel not found. - (make linux26) generated some warnings about various usb_*_dev symbols undefined in [xpp,wcusb]/*.ko, but no actual errors. (make install) repeated those warnings. (modprobe ztdummy) finished with - Building /etc/modprobe.d/zaptel... *** *** WARNING: *** If you had custom settings in /etc/modprobe.d/zaptel, *** they have been moved to /etc/modprobe.d/zaptel.bak. [...] - but seemed to complete without errors. (make install) included a line - INSTALL zaptel-1.2.11-source-root-dir/ztdummy.ko - but no success. Complete (failed) install session transcript: - # make clean rm -f torisatool makefw tor2fw.h radfw.h rm -f ztcfg torisatool makefw ztmonitor ztspeed zttool zttest fxotune rm -f *.o ztcfg tzdriver sethdlc sethdlc-new rm -f zonedata.lo tonezone.lo libtonezone.so *.lo make -C kernel-source-root-dir SUBDIRS=zaptel-1.2.11-source-root-dir clean make[1]: Entering directory `kernel-source-root-dir' CLEAN zaptel-1.2.11-source-root-dir/wct4xxp CLEAN zaptel-1.2.11-source-root-dir/.tmp_versions make[1]: Leaving directory `kernel-source-root-dir' rm -rf .tmp_versions rm -f gendigits tones.h rm -f libtonezone* rm -f tor2ee rm -f fxotune rm -f core rm -f ztcfg-shared fxstest rm -rf misdn* rm -rf mISDNuser* # make config if [ -d /etc/rc.d/init.d ]; then \ install -D -m 755 zaptel.init /etc/rc.d/init.d/zaptel; \ //sbin/chkconfig --add zaptel; \ elif [ -d /etc/init.d ]; then \ install -D -m 755 zaptel.init /etc/init.d/zaptel; \ //sbin/chkconfig --add zaptel; \ fi if [ -d /etc/default ] [ ! -f /etc/default/zaptel ]; then \ install -D -m 644 zaptel.sysconfig /etc/default/zaptel; \ fi if [ -d /etc/sysconfig ] [ ! -f /etc/sysconfig/zaptel ]; then \ install -D -m 644 zaptel.sysconfig /etc/sysconfig/zaptel; \ fi if [ -d /etc/sysconfig/network-scripts ]; then \ install -D -m 755 ifup-hdlc /etc/sysconfig/network-scripts/ifup-hdlc; \ fi # make linux26 cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o gendigits.o gendigits.c cc -o gendigits gendigits.o -lm ./gendigits tones.h cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\makefw.c -o makefw ./makefw tormenta2.rbt tor2fw tor2fw.h Loaded 69900 bytes from file ./makefw pciradio.rbt radfw radfw.h Loaded 42096 bytes from file ZAPTELVERSION=1.2.11 build_tools/make_version_h version.h.tmp if cmp -s version.h.tmp version.h ; then echo; else \ mv version.h.tmp version.h ; \ fi rm -f version.h.tmp cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o ztcfg.o ztcfg.c cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DBUILDING_TONEZONE -o zonedata.lo zonedata.c cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DBUILDING_TONEZONE -o tonezone.lo tonezone.c ar rcs libtonezone.a zonedata.lo tonezone.lo cc -o ztcfg ztcfg.o libtonezone.a -lm cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o torisatool.o torisatool.c cc -o torisatool torisatool.o cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o ztmonitor.o ztmonitor.c cc -o ztmonitor ztmonitor.o cc -o ztspeed.o -c ztspeed.c cc -o ztspeed ztspeed.o cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o zttool.o zttool.c cc -o zttool
Re: [asterisk-users] zaptel compilation problems with linux 2.6.19
I'm having problems installing ztdummy on my CentOS-2.6.16-rc6/Asterisk-1.2.11 datacenter server (no ZAP HW, SIP only to PSTN). I unpacked the kernel sources and headers in a directory, made (but not re/installed) the kernel, unpacked the zaptel-1.2.11 tarball, then went thru the make sequence. It seemed to proceed OK (without errors, just some warnings), but didn't seem to result in a loadable ztdummy kernel module. Details appended, followed by a complete (failed) install session transcript: - # cd path-to-zaptel-1.2.11-source # export KSRC=path-to-kernel-source-root-dir # make clean # make config [... series of shell script conditionals apparently executed OK ...] # make linux26 [... series of CC/LD reports, some warnings, no errors ...] # make install [... series of INSTALL messages, same warnings from (make linux26), no errors ...] # modprobe ztdummy FATAL: Module ztdummy not found. FATAL: Error running install command for ztdummy # modprobe zaptel FATAL: Module zaptel not found. - (make linux26) generated some warnings about various usb_*_dev symbols undefined in [xpp,wcusb]/*.ko, but no actual errors. (make install) repeated those warnings. (modprobe ztdummy) finished with - Building /etc/modprobe.d/zaptel... *** *** WARNING: *** If you had custom settings in /etc/modprobe.d/zaptel, *** they have been moved to /etc/modprobe.d/zaptel.bak. [...] - but seemed to complete without errors. (make install) included a line - INSTALL zaptel-1.2.11-source-root-dir/ztdummy.ko - but no success. Complete (failed) install session transcript: - # make clean rm -f torisatool makefw tor2fw.h radfw.h rm -f ztcfg torisatool makefw ztmonitor ztspeed zttool zttest fxotune rm -f *.o ztcfg tzdriver sethdlc sethdlc-new rm -f zonedata.lo tonezone.lo libtonezone.so *.lo make -C kernel-source-root-dir SUBDIRS=zaptel-1.2.11-source-root-dir clean make[1]: Entering directory `kernel-source-root-dir' CLEAN zaptel-1.2.11-source-root-dir/wct4xxp CLEAN zaptel-1.2.11-source-root-dir/.tmp_versions make[1]: Leaving directory `kernel-source-root-dir' rm -rf .tmp_versions rm -f gendigits tones.h rm -f libtonezone* rm -f tor2ee rm -f fxotune rm -f core rm -f ztcfg-shared fxstest rm -rf misdn* rm -rf mISDNuser* # make config if [ -d /etc/rc.d/init.d ]; then \ install -D -m 755 zaptel.init /etc/rc.d/init.d/zaptel; \ //sbin/chkconfig --add zaptel; \ elif [ -d /etc/init.d ]; then \ install -D -m 755 zaptel.init /etc/init.d/zaptel; \ //sbin/chkconfig --add zaptel; \ fi if [ -d /etc/default ] [ ! -f /etc/default/zaptel ]; then \ install -D -m 644 zaptel.sysconfig /etc/default/zaptel; \ fi if [ -d /etc/sysconfig ] [ ! -f /etc/sysconfig/zaptel ]; then \ install -D -m 644 zaptel.sysconfig /etc/sysconfig/zaptel; \ fi if [ -d /etc/sysconfig/network-scripts ]; then \ install -D -m 755 ifup-hdlc /etc/sysconfig/network-scripts/ifup-hdlc; \ fi # make linux26 cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o gendigits.o gendigits.c cc -o gendigits gendigits.o -lm ./gendigits tones.h cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\makefw.c -o makefw ./makefw tormenta2.rbt tor2fw tor2fw.h Loaded 69900 bytes from file ./makefw pciradio.rbt radfw radfw.h Loaded 42096 bytes from file ZAPTELVERSION=1.2.11 build_tools/make_version_h version.h.tmp if cmp -s version.h.tmp version.h ; then echo; else \ mv version.h.tmp version.h ; \ fi rm -f version.h.tmp cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o ztcfg.o ztcfg.c cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DBUILDING_TONEZONE -o zonedata.lo zonedata.c cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DBUILDING_TONEZONE -o tonezone.lo tonezone.c ar rcs libtonezone.a zonedata.lo tonezone.lo cc -o ztcfg ztcfg.o libtonezone.a -lm cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o torisatool.o torisatool.c cc -o torisatool torisatool.o cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o ztmonitor.o ztmonitor.c cc -o ztmonitor ztmonitor.o cc -o ztspeed.o -c ztspeed.c cc -o ztspeed ztspeed.o cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o zttool.o zttool.c cc -o zttool
Re: [asterisk-users] Loosing IAX connection between offices
if its a version 5 or higher, that wont be an option, but if its not, give openwrt or ddwrt a try. Actually, this is no longer true (at least for WRT54G), see http://en.wikipedia.org/wiki/DD-WRT for the official list of supported models ___ --Bandwidth 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] Loosing IAX connection between offices
On 11/30/06, Dave Fullerton [EMAIL PROTECTED] wrote: DM wrote: snip Office A is set up with refresh dns and cron job for iax2 reload every 5 minutes. It rarely looses connection to Office B. Surprisingly, Office B is the one loosing connection with Office A. I'm surprised because Office A is the one with the static IP address. When I do a IAX2 Show Peers, the connection will show as UNKNOWN or UNAVAILABLE. After loosing connection, the only way I can get it to reestablish is to reboot the * box. IAX2 reload doesn't solve it. I haven't been able to establish if it loosing the connection at a specific duration. Though, it seems to be random. snip Any reason you don't have Office B register with Office A and thereby tell Office A what IP office B has? It would remove the need to refresh DNS and reload IAX2 all the time. I use this method to connect my home system (dynamic) to my work system (static) and it works well. -Dave I assume you are referring to Method 3 listed here: http://www.voip-info.org/wiki/view/Asterisk+Connect+2+servers I'll have to give it a try. ___ --Bandwidth 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] 200+ analog phones connected to FXS modules
Hi On Thu, Nov 30, 2006 at 12:59:13PM -0500, Andrew Kohlsmith wrote: On Thursday 30 November 2006 06:13, Zoa wrote: You could go for 2 quad pri cards + channel banks or for TDMoE or usb channel banks. [ disclaimer: I work for the company that makes the USB channel bank which was mentioned by Zoa ] The last option would be the cheaper and more scalable one imho The scale here is already bordering on unrealistic. I wouldn't expect them to want to make this too much bigger. Scaling aside, 200+ ulaw channels (13Mb/sec not including overhead) over USB? You're joking, right? I know that USB 2.0 is rated at 480Mbps but has this actually been tested and verified? I would imagine USB 2.0 chipset variants and inconsistencies are even worse than that of PCI. USB2 actually seems to be a well-established standard. It is used by many devices. For instance, people with high-capacity USB storage devices would be rather upset to find their chipset acting up. And if one USB2 card is not enough, get a USB2 PCI adapter for 10$ from your local hardware store. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth 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] 2nd attempt - Return code - How to?
Can anyone give me some insight on this? If I am not making myself clear
please let me know.
At voip-info.org they show the following example
exten = s,1,Set(foo=${STAT(s,/var/t3)})
which I guess is suppose to work and make foo = size of t3
I did the following
exten = 542,1,Set(s1=${STAT(e,/var/lib/asterisk/t1)})
which should set s1 = 1 if the file exists and 0 if not.
but I get
Nov 30 00:19:06 ERROR[23493]: pbx.c:1382 ast_func_read: Function STAT not
registered
-- Executing Set(SIP/grandstream406-22e9, s1=0) in new stack
and in general I am confused about return codes. How would you use a
return code from the following
exten = s,1,System(somescript arg1 arg2)
Can someone give me a working example??? I keep getting the above error
Doug
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Re: [asterisk-users] 2nd attempt - Return code - How to?
Nov 30 00:19:06 ERROR[23493]: pbx.c:1382 ast_func_read: Function STAT not registered from http://voip-info.org/wiki/view/Asterisk+functions : Functions in the below list are marked in red if they are only available in version 1.4 and higher. And STAT is marked in red so I guess you're not running 1.4 ___ --Bandwidth 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] Pickup *8 with CallerID
Hi list ! I implemented *8 to pickup any call on my asterisk system. But after the pickup callerid is missing, so there is no way to see from where the call originated. How can this callerid be passed on. Nik ___ --Bandwidth 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] voicemailmain
When I call to VoicemailMain it just sits. ; Retrieve Voice Mail exten = 2500,1,Wait(2) exten = 2500,2,VoicemailMain(s100) exten = 2500,3,Macro(endcall) 1.4.3 latest SVN. voicemail(100) works and the mwi systems works. I am not using ODBC or SQL. Voice mail to email works ok. I just cannot retrieve it by the application. I'm not sure when this quite we get little voice mail traffic. Thanks --john ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Trouble using 2 IAX2 DiDs provided by different ITSPs
Asterisk 1.2.7 Redhat 9 I have DiDs from two different ITSP both set up as IAX2. Each one works when it's the only one in my iax.conf, but when I have them both defined in iax.conf at the same time, only one will work. My iax.conf is provided below. Any ideas how to fix? I'd like to use both DiDs! Thanks, H My iax.conf is below. When I dial the DiD provided by ITSP_B, the other ITSP seems to reject it. For example when I call the ITSP_B DiD, I get the following error message: Nov 29 21:50:17 NOTICE[23106]: chan_iax2.c:7203 socket_read: Host IP failed to authenticate as ITSP_A iax.conf == [general] register = my UserID:my password@ITSP A Server #1 domain register = my UserID:my password@ITSP A Server #2 domain register = my UserID:my password@ITSP B #1 domain notransfer=yes bindport=4569 bindaddr=0.0.0.0 bandwidth=low disallow=all allow=ulaw allow=g729 jitterbuffer=yes forcejitterbuffer=no tos=lowdelay autokill=yes [ITSP_B] context=incoming-iax type=friend qualify=2000 host=ITSP B #1 domain user=my UserID username=my UserID auth=md5 secret=my password disallow=all allow=ulaw ; ; *** ITSP_A Inbound *** [ITSP_A] context=incoming-iax type=user auth=md5 username=my UserID secret=my password disallow=all allow=ulaw ; ; *** ITSP_A Outbound *** [ITSP_A-Out] type=peer host=ITSP A Server #1 domain auth=md5 username=my UserID secret=my password disallow=all qualify=yes allow=ulaw ; [ITSP_A-Out2] type=peer host=ITSP A Server #2 domain auth=md5 username=my UserID secret=my password disallow=all qualify=yes allow=ulaw ___ --Bandwidth 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] 200+ analog phones connected to FXS modules
You can get a basic VOIP phone for the same price that it will cost you for a FXS port. As far as wiring you can go with a bit more expensive phone and get a dual port with POE (if they have an existing computer network). - Original Message - From: Vieri [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, November 30, 2006 11:15 AM Subject: [asterisk-users] 200+ analog phones connected to FXS modules I am trying to find out the best way to replace one of our hardware PBXs. It currently has 200+ analog phones connected to it. The idea is to take advantage of the already installed phone cables (big building) so I'm trying to avoid the use of ethernet adapters (if possible). However, I'm realizing that it's an expensive setup and will definitely require two or more cooperating Asterisk servers (cluster) mainly due to PCI slot availability. I am aware of the TDM2400P card. One could put 6 FXS uqad-modules and would serve 24 analog phones. However, I would need at least 9 of these PCI cards which could be placed in 2 or 3 servers. Is there another way of doing this (hopefully cheaper and more convenient)? Thank you for your suggestions. Vieri Yahoo! Music Unlimited Access over 1 million songs. http://music.yahoo.com/unlimited ___ --Bandwidth 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] incominglimit and outgoinglimit
Hi ! as the wiki says there is only the possibility to set incominglimit and outgoinglimit to type peer, how can I accomplish this with the type friend? nik ___ --Bandwidth 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] PAP2 and Asterisk
I have a Linksys PAP2 connected to Asterisk. Have one of the FXS ports working fine. I am unable to get the other to work. Does anybody have an example configuration to make both work. Both are registering fine but there's just no dialtone on the non working port. TIA ___ --Bandwidth 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:voicemailmain
I looked at the voicemail.c code and you must have the res.adsi module loaded. I was not loading it. Now it works. Something to remember. Thanks --john ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] VoIP GSM Gateways
We do have @cough VoIP GSM Gateway for sell as well @ cough Try to search on ebay for gsm voip gateway and you will see some in there As far as I am concern it is cheaper than 2n. And if you are looking for multi ports then it will come off as RJ11 ports rather than voip and they are £100 per port with a max of 16 ports in 1 chassis. Sam -Original Message- From: Matteo Brancaleoni [mailto:[EMAIL PROTECTED] Sent: Monday, October 30, 2006 12:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] VoIP GSM Gateways Hi, On Sun, 2006-10-29 at 13:46 +0200, Tzafrir Cohen wrote: Is vISDN (extra kernel modules, extra non-standard Asterisk channel) required? The page on vGSM there suggests it is. no, vgsm uses only a part of visdn (timer system and streamport), so you need only chan_vgsm, visdn_streamport (for audio) and visdn_timer_system for timing. Other visdn things, like chan_visdn, complex visdn pci conf etc etc is not needed. Nothing more. The card is in production since months on various systems and is running very smooth :) 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 ___ --Bandwidth 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] Pickup *8 with CallerID
Where are you looking for the caller id at? On 11/30/06, Nik Engel [EMAIL PROTECTED] wrote: Hi list ! I implemented *8 to pickup any call on my asterisk system. But after the pickup callerid is missing, so there is no way to see from where the call originated. How can this callerid be passed on. Nik ___ --Bandwidth 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] voicemailmain
What do you get in the CLI ? - Original Message - From: John Hill [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, November 30, 2006 11:24 PM Subject: [asterisk-users] voicemailmain When I call to VoicemailMain it just sits. ; Retrieve Voice Mail exten = 2500,1,Wait(2) exten = 2500,2,VoicemailMain(s100) exten = 2500,3,Macro(endcall) 1.4.3 latest SVN. voicemail(100) works and the mwi systems works. I am not using ODBC or SQL. Voice mail to email works ok. I just cannot retrieve it by the application. I'm not sure when this quite we get little voice mail traffic. Thanks --john ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth 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] Cisco 7970 SIP upgrade issues
Hi Thanks for the advice but it really is more fundamental. I have an old (v5) sccp phone. I need to upgrade it to v7 sccpbefore I can load the Sip image. I downloaded the V7 sccp file from the cisco website but it seems to want call manager to load. Does anyone have any experience of upgrading a V5 7970? Please please :-) - Original Message - From: Alfred Nagl [EMAIL PROTECTED] To: Paul [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Sent: Wednesday, November 29, 2006 2:26 PM Subject: Re: [asterisk-users] Cisco 7970 SIP upgrade issues Paul writes: I am having problems putting a SIP image on a 7970. Hi! Two weeks ago I loaded a recent SIP Image, SIP70.8-0-4SR1S, on a 7970, but I started from a relatively new SCCP Image. ( the phone has Boot Load ID 7970_64060118.bin) I did the following: .) configured a tftp server on the phone, to unlock I had to type star star numbersign (**#), and then I could save that configuration .) Got cmterm-7970_7971-sip.8-0-4SR1.zip from cisco website and unzipped it in tftp Directory .) Created file SEPMAC.cnf.xml with the following entry: loadInformationSIP70.8-0-4SR1S/loadInformation Most of the content of my SEPMAC.cnf.xml is from the follwing webpage http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIPview_comment_id=12387 and also from http://www.reub.net/files/cisco-7941/SEP-my-mac.cnf.xml If you are in a hurry, I could try to send you a sanitized / shorted working version of my SEPMAC.cnf.xml. regards, --alfred P.S.: I have tried to find some Documentation about the Meaning of all these XML Tags in the cnf.xml file, but was only partly successfull: http://www.voip-info.org/wiki-Asterisk+Cisco+79XX+XML+Services PP.S: there is a docoment about converting from SCCP to SIP and back (but it does not mention the 7970) http://www.cisco.com/warp/public/788/voip/handset_to_sip.html -- Alfred Nagl ([EMAIL PROTECTED]) Fax +43 (1) 31336-904811 University of Economics, A-1090 Vienna, Austria, EUROPE ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
