[asterisk-users] Cancel a ringing SIP call when the other party disconnect
Hi, Here is my scenario. I have a SIP call between two SIP endpoints. A calls B. During the ringing, B disconnects (network cable is unplugged). But A continue ringing forever (until the dial timeout) even if asterisk detects that B is disconnected with the qualify. Is there any setup or asterisk configuration I need to enable to have A close its call ? Note: when A is already talking with B, the call is hanged up on rtp timeout. But not during the Ringing phase. Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how determine mandatory modules to slimming asterisk
Are you using asterisk from source code ? You can run make menuselect after the ./confugure and select the modules you want. The interface will also help you with dependencies between modules. So, if you select chan_sip, it will select everything needed by it. Le 2013-11-11 02:10, s m a écrit : hello guys i want to slimming my asterisk by loading only mandatory modules. in order to do that, i edit my modules.conf file and set autoload=no and load just mandatory modules. my problem is, how should i determine which modules are necessary to asterisk works correctly? i have sip, h323 and dahdi connection on my asterisk. is there any documentation about mandatory modules for asterisk? or anybody has such a list? any comments or hints are appreciated SAM -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk SIP server on windows
Hi all, I need to build an application that will be an SIP server program that will run on Linux and Windows. The sip server need only some features such as be able to : - Register sip endpoints - Answer a call and play a local file - Make a dial from one channel to another. I know asterisk can be stripped to exactly fit my needs. I would like to know if there is a way to build it on windows after it has been stripped. Or do I have other alternatives out there ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Unmute all users in Meetme conference as admin
Hi, I setup an MeetMe conference. So, the admin user calls and enter the conference in talk/listen mode. (Options : dAaxs) Then other users call the same conference and enters in muted mode (options: dlmx) How can the admin user decide, when he is ready to let everybody speaks ? I didn't find such option in the admin menu. Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unmute all users in Meetme conference as admin
I think I found it reading the code. It is *83 to unmute everybody. Thanks From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ruddy Gbaguidi Sent: 2013-12-04 04:07 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Unmute all users in Meetme conference as admin Hi, I setup an MeetMe conference. So, the admin user calls and enter the conference in talk/listen mode. (Options : dAaxs) Then other users call the same conference and enters in muted mode (options: dlmx) How can the admin user decide, when he is ready to let everybody speaks ? I didn't find such option in the admin menu. Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk SIP server on windows
This is about an call center application we are building and that need an embedded PBX. We would then like to have that platform run on Windows and Linux. Are there ways to easy ship linux application embedded in virtual machine so they can run on windows ? Le 2013-12-04 08:02, Dan Journo a écrit : FROM: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] ON BEHALF OF Ruddy Gbaguidi SENT: 04 December 2013 09:08 TO: 'Asterisk Users Mailing List - Non-Commercial Discussion' SUBJECT: [asterisk-users] Asterisk SIP server on windows Hi all, I need to build an application that will be an SIP server program that will run on Linux and Windows. The sip server need only some features such as be able to : - Register sip endpoints - Answer a call and play a local file - Make a dial from one channel to another. I know asterisk can be stripped to exactly fit my needs. I would like to know if there is a way to build it on windows after it has been stripped. Or do I have other alternatives out there ? Servers that can run Asterisk are so cheap nowadays, unless you are talking about huge volumes of traffic. I'd recommend getting a server and putting on Centos which is tried and tested. You'll waste less time that way and avoid any unforeseen problems. Or look for a cloud server to do the job for you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on Windows
I never tought this is become a Linux vs Windows fight. We have been using asterisk on linux from a long time now and happy with it. But some of our customers who has windows in their environment want to use our call center software we developed on top of asterisk. So, the question was : Did anybody ever tried to isolate the asterisk SIP server/module and make it run under Windows ? Since, asterisk 12 is using pjsip (which is cross platform already), I tought it may be possible and wanted advices. I would love that every single customer switch to Linux and Ubuntu tomorrow morning but at the moment, that's not the case. Thanks. Le 2013-12-04 11:31, Patrick Lists a écrit : Probably feeding the trolls but here it goes. On 12/04/2013 04:19 PM, CDR wrote: Digium is 100% lost in the map. If they would come up with a Paid version of Asterisk, one that would use the .NET framework in Windows, something simple to install, they could go public on the product. IIRC Microsoft no longer invests in the .Net framework which makes it a bad idea for a product that would live for up to 10 years. Do you really want to bet your business/company that .Net will be there in 5 to 10 years? Linux has a very steep learning curve. A Windows application that would do exactly the same would be a home run. I find Linux easier than Windows. Installing a package on Linux or Windows is not the issue. How is a simple 'yum install asterisk' any more difficult than double clicking on it in Windows? It's what you do afterwards with the OS and package. Asterisk has a much steeper learning curve than either. It's easy to mess up the config and suffer the consequences if the box is Internet facing. Also, Windows has a terrible reputation when it comes to security. Why would anyone want to use Windows for an Internet facing service? There's a reason that Google, Facebook, Twitter and pretty much the rest of the world are powered by Linux and it's not only because it's cheaper. Just because you find Windows easier does not make it a good idea. Note: I am a Linux expert user, but it took me years to get here. And still, moving from regular RHEL 6.0 to Fedora 20 (RHEL 7) is a pain in the neck. There is probably a saying about people calling themselves experts and then complain about a move from EL6 to F20 which is puzzling by itself. The .NET framework and Windows server 2012 are miles away in terms of friendliness and on equal footing on performance. I have yet to see a large Telco or ITSP deploy their services on Windows. A while back I have seen some attempts. It was hilarious to hear that the servers had to be restarted every few hours. Performance totally sucked, components would crash and the solution was, even by telco standards, ridiculously expensive. So no, they are not on equal footing when it comes to performance (and other aspects). I don´t mean another slow cygwin port, I man a native Asterisk for windows. In fact, I would invest on the project if somebody wants to do it. If you really want to use Windows then have a look at FreeSWITCH as it's available on Windows too. Then there is also Lync and 3CX. Good luck keeping your Windows boxes from getting hacked with all the financial and other damage it would cause. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to tie orders taken to specific CDR records
You can have your dialplan log and write a data in a specific table as soon as you get the call. Then send that call ID to the agent web interface. And when the agent complete the order, you just update the table with needed information. Ruddy Gbaguidi Micnes - Professional Services 3767 Thimens, Suite 202 Montreal (Quebec), Canada, H4R 1W4 C: +1.514.814.0690 plugwo...@micnes.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mitch Claborn Sent: 2012-10-25 12:19 To: asterisk-users@lists.digium.com Subject: [asterisk-users] How to tie orders taken to specific CDR records Our phone operators work off of an Asterisk queue. They take calls from customers and take orders with our back end systems. What I need to be able to do is tie the orders taken to the specific CDR record that reflects the call from which the order originated. The typical/sample CDR table doesn't have a primary key. I can add an auto-generated PK, but the CDR is not written until the call ends, when the orders have already been placed. (Even if the CDR was written earlier, could I retrieve the generated PK from it in the dialplan somehow?) Is there some combination of fields in the CDR that might uniquely identify a specific call? Open to any and all ideas. -- Mitch -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 passing back and forth variables
I cannot find it
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: 2012-05-21 10:25
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] IAX2 passing back and forth variables
There was a nice thread on this back in April.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ruddy Gbaguidi
Sent: Monday, May 21, 2012 9:23 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] IAX2 passing back and forth variables
No one have an idea ?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ruddy Gbaguidi
Sent: 2012-05-19 15:27
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] IAX2 passing back and forth variables
Sorry, the dialplan is really on server B
exten = s,n,Set(IAXVAR(TESTVAR2)=efgh)
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Noah
Engelberth
Sent: 2012-05-19 14:45
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] IAX2 passing back and forth variables
Uhm, if the dialplan is exactly as you pasted, you're not setting TESTVAR2
to anything. You would need some sort of Set(IAXVAR(TESTVAR2)=.)
Noah
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ruddy Gbaguidi
Sent: Saturday, May 19, 2012 2:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] IAX2 passing back and forth variables
Hi all,
I have two asterisk servers A and B.
And I would like from A, dial to B passing some IAX variables.
Then B handles the calls, setup some other variables that become available
to A which can continue.
So far, I have used IAXVAR function.
It works when sending call from A to B
But variables setup on B are not available on A.
Any idea how I can do it ?
Here are my dialplans.
+++
SERVER A
+++
[contextA]
exten = s,1,Set(IAXVAR(TESTVAR1)=abcd)
exten = s,n,Dial(IAX2/serverb/s,30,g)
exten = s,n,Noop( The out variable is : ${IAXVAR(TESTVAR2)} ) ;
Does not work
+++
SERVER B
+++
[contextB]
exten = s,1,Noop( ${IAXVAR(TESTVAR1)} ) - Does work
exten = s,n,Set(IAXVAR(TESTVAR2))
exten = s,n,Hangup
--
_
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Re: [asterisk-users] IAX2 passing back and forth variables
No one have an idea ?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ruddy Gbaguidi
Sent: 2012-05-19 15:27
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] IAX2 passing back and forth variables
Sorry, the dialplan is really on server B
exten = s,n,Set(IAXVAR(TESTVAR2)=efgh)
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Noah
Engelberth
Sent: 2012-05-19 14:45
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] IAX2 passing back and forth variables
Uhm, if the dialplan is exactly as you pasted, you're not setting TESTVAR2
to anything. You would need some sort of Set(IAXVAR(TESTVAR2)=.)
Noah
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ruddy Gbaguidi
Sent: Saturday, May 19, 2012 2:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] IAX2 passing back and forth variables
Hi all,
I have two asterisk servers A and B.
And I would like from A, dial to B passing some IAX variables.
Then B handles the calls, setup some other variables that become available
to A which can continue.
So far, I have used IAXVAR function.
It works when sending call from A to B
But variables setup on B are not available on A.
Any idea how I can do it ?
Here are my dialplans.
+++
SERVER A
+++
[contextA]
exten = s,1,Set(IAXVAR(TESTVAR1)=abcd)
exten = s,n,Dial(IAX2/serverb/s,30,g)
exten = s,n,Noop( The out variable is : ${IAXVAR(TESTVAR2)} ) ;
Does not work
+++
SERVER B
+++
[contextB]
exten = s,1,Noop( ${IAXVAR(TESTVAR1)} ) - Does work
exten = s,n,Set(IAXVAR(TESTVAR2))
exten = s,n,Hangup
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX2 passing back and forth variables
Hi all,
I have two asterisk servers A and B.
And I would like from A, dial to B passing some IAX variables.
Then B handles the calls, setup some other variables that become available
to A which can continue.
So far, I have used IAXVAR function.
It works when sending call from A to B
But variables setup on B are not available on A.
Any idea how I can do it ?
Here are my dialplans.
+++
SERVER A
+++
[contextA]
exten = s,1,Set(IAXVAR(TESTVAR1)=abcd)
exten = s,n,Dial(IAX2/serverb/s,30,g)
exten = s,n,Noop( The out variable is : ${IAXVAR(TESTVAR2)} ) ;
Does not work
+++
SERVER B
+++
[contextB]
exten = s,1,Noop( ${IAXVAR(TESTVAR1)} ) - Does work
exten = s,n,Set(IAXVAR(TESTVAR2))
exten = s,n,Hangup
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 passing back and forth variables
Sorry, the dialplan is really on server B
exten = s,n,Set(IAXVAR(TESTVAR2)=efgh)
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Noah
Engelberth
Sent: 2012-05-19 14:45
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] IAX2 passing back and forth variables
Uhm, if the dialplan is exactly as you pasted, you're not setting TESTVAR2
to anything. You would need some sort of Set(IAXVAR(TESTVAR2)=.)
Noah
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ruddy Gbaguidi
Sent: Saturday, May 19, 2012 2:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] IAX2 passing back and forth variables
Hi all,
I have two asterisk servers A and B.
And I would like from A, dial to B passing some IAX variables.
Then B handles the calls, setup some other variables that become available
to A which can continue.
So far, I have used IAXVAR function.
It works when sending call from A to B
But variables setup on B are not available on A.
Any idea how I can do it ?
Here are my dialplans.
+++
SERVER A
+++
[contextA]
exten = s,1,Set(IAXVAR(TESTVAR1)=abcd)
exten = s,n,Dial(IAX2/serverb/s,30,g)
exten = s,n,Noop( The out variable is : ${IAXVAR(TESTVAR2)} ) ;
Does not work
+++
SERVER B
+++
[contextB]
exten = s,1,Noop( ${IAXVAR(TESTVAR1)} ) - Does work
exten = s,n,Set(IAXVAR(TESTVAR2))
exten = s,n,Hangup
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
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[asterisk-users] Full SIP dial string
Hi All I want to be able to read some sip informations (from a database) like username, password, host and extension number and place a Dial from asterisk. So basicly, I want to dial sip extensions without modifying sip.conf each time. I don't know, in the dialplan, what the dial string should look like. I tried SIP/username:password@host/exten without success Can you help ? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Full SIP dial string
So, I cannot specify username and password ? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of William Stillwell Sent: 2011-06-11 08:03 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Full SIP dial string From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ruddy Gbaguidi Sent: Saturday, June 11, 2011 3:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Full SIP dial string Hi All I want to be able to read some sip informations (from a database) like username, password, host and extension number and place a Dial from asterisk. So basicly, I want to dial sip extensions without modifying sip.conf each time. I don't know, in the dialplan, what the dial string should look like. I tried SIP/username:password@host/exten without success Can you help ? Thanks --- Try SIP/exten@host William Stillwell -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Playback in uplink and recording in downlink
Yes, you can use the Mixmonitor command. But if you want to have only one party on the recording, you should use the Monitor command without the 'm' option. http://www.astblog.com/2011/02/01/asterisk-mixmonitor-cmd/ -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger Sent: 2011-02-01 09:41 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Playback in uplink and recording in downlink On 11-02-01 04:02 AM, Felix Dong wrote: I got a question to asterisk 1.6. Is it possible to playback a Audiofile in uplink and to record the downlink channel in another Audifile at the same time? Yes, look at MixMonitor. *CLI core show application MixMonitor -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip.conf User vs Username
Hi In sip.conf, you generally have something like [name] .. username= secret= What is the difference between the name specified in brackets and the username key ? What the sip client should provide ? What do we use in dialplan when trying to reach this client ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] H323 RTP Transmission error of packet
Nobody on this ? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ruddy Gbaguidi Sent: September-16-09 7:52 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] H323 RTP Transmission error of packet Using H323 to reach another h323 switch, I have no audio and the following error: [Sep 16 15:45:55] DEBUG[10528]: rtp.c:2760 ast_rtp_raw_write: RTP Transmission error of packet 21282 to XXX.XXX.XXX.XXX:6064: Invalid argument [Sep 16 15:45:55] DEBUG[10528]: rtp.c:2760 ast_rtp_raw_write: RTP Transmission error of packet 21283 to XXX.XXX.XXX.XXX:6064: Invalid argument [Sep 16 15:45:55] DEBUG[10528]: rtp.c:2760 ast_rtp_raw_write: RTP Transmission error of packet 21284 to XXX.XXX.XXX.XXX:6064: Invalid argument [Sep 16 15:45:55] DEBUG[10528]: rtp.c:2760 ast_rtp_raw_write: RTP Transmission error of packet 21285 to XXX.XXX.XXX.XXX:6064: Invalid argument [Sep 16 15:45:55] DEBUG[10528]: rtp.c:2760 ast_rtp_raw_write: RTP Transmission error of packet 21286 to XXX.XXX.XXX.XXX:6064: Invalid argument Can you please tell me what I`m missing I`m doing a quick dial like Dial(h323/1514...@xxx.xxx.xxx.xxx) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] H323 RTP Transmission error of packet
Using H323 to reach another h323 switch, I have no audio and the following error: [Sep 16 15:45:55] DEBUG[10528]: rtp.c:2760 ast_rtp_raw_write: RTP Transmission error of packet 21282 to XXX.XXX.XXX.XXX:6064: Invalid argument [Sep 16 15:45:55] DEBUG[10528]: rtp.c:2760 ast_rtp_raw_write: RTP Transmission error of packet 21283 to XXX.XXX.XXX.XXX:6064: Invalid argument [Sep 16 15:45:55] DEBUG[10528]: rtp.c:2760 ast_rtp_raw_write: RTP Transmission error of packet 21284 to XXX.XXX.XXX.XXX:6064: Invalid argument [Sep 16 15:45:55] DEBUG[10528]: rtp.c:2760 ast_rtp_raw_write: RTP Transmission error of packet 21285 to XXX.XXX.XXX.XXX:6064: Invalid argument [Sep 16 15:45:55] DEBUG[10528]: rtp.c:2760 ast_rtp_raw_write: RTP Transmission error of packet 21286 to XXX.XXX.XXX.XXX:6064: Invalid argument Can you please tell me what I`m missing I`m doing a quick dial like Dial(h323/1514...@xxx.xxx.xxx.xxx) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMD Not Working
Maybe the customer hangs up during the AMD analysis or you don't have any audio coming to asterisk through your sip channel. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sam Hawkin Sent: April-23-09 11:00 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] AMD Not Working Hi All, I am trying to use the AMD (Answering Machine Detect). But it is not sending the AMD_Status as either the Human or Machine, it hangs up in middle. can any one suggest us, what might be the problem and possible solution to it. below is the log -- Executing AMD(SIP/sip-ffe0, ) in new stack -- AMD: SIP/sip-ffe0 14082284927 (null) (Fmt: 4) Apr 23 08:00:26 NOTICE[28319]: app_amd.c:134 isAnsweringMachine: AMD using the default parameters. -- AMD: initialSilence [3500] greeting [1500] afterGreetingSilence [300] totalAnalysisTime [5000] minimumWordLength [120] betweenWordsSilence [50] maximumNumberOfWords [5] silenceThreshold [256] -- AMD: HANGUP any help is highly appreciated. Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI PHP script
First run
/var/lib/asterisk/agi-bin/newhire.php
From linux command line to see if you don't have any error and that your AGI
is executable.
Then run 'agi debug' from the asterisk cli, place a call and see what was
send and receive from your agi
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James A.
Shigley
Sent: April-23-09 12:26 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] AGI PHP script
I have the below script that doesn't seem to be working. I don't know if I
have something in the script wrong that I am just missing. Or if I don't
have the php.ini set correctly for emailing
This is the CLI output
-- Executing [4099xxx...@port3_real:1] Goto(DAHDI/50-1, newhire,s,1) in
new stack
-- Goto (newhire,s,1)
-- Executing [...@newhire:1] Ringing(DAHDI/50-1, ) in new stack
-- Executing [...@newhire:2] Answer(DAHDI/50-1, ) in new stack
-- Executing [...@newhire:3] Monitor(DAHDI/50-1,
wav,/var/lib/asterisk/soun
ds/NewHire/Newhire-1240503071.15148-4099819213-s,o) in new stack
-- Executing [...@newhire:4] AGI(DAHDI/50-1, newhire.php) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/newhire.php
-- DAHDI/50-1AGI Script newhire.php completed, returning 0
-- Auto fallthrough, channel 'DAHDI/50-1' status is 'UNKNOWN'
-- Hungup 'DAHDI/50-1'
Here is the script
#!/usr/bin/php5
?php
// Get AGI vars from *
$agivars = array();
while (!feof(STDIN)) {
$agivar = trim(fgets(STDIN));
if ($agivar === '') {
break;
}
$agivar = explode(':', $agivar);
$agivars[$agivar[0]] = trim($agivar[1]);
}
extract($agivars);
// Variable Declarations
$agi_uniqueid;
$agi_callerid;
$agi_calleridname;
$agi_extension;
$agi_uniqueid;
$UNIQUEID = $agi_uniqueid;
$CALLERID = $agi_callerid;
$EXTEN = $agi_extension;
$attachment =
/var/lib/asterisk/sounds/NewHire/Newhire-$UNIQUEID-$CALLERID-$EXTEN.wav;
$from = @xxx.com;
$to =j...@answeringserv.com ;
$subject=New Applicant;
$headers = From: $from;
$message =$UNIQUEID , $CALLERID , $EXTEN , $attachment;
mail($to,$subject,$message,$headers);
?
So is it anything obviously wrong with the script I'm missing?
Besides something not being configured in php.ini correctly any other ideas?
James Shigley
Monroe Telephone Answering Service
409-981-9213
Infinity 5.5,UC 4.02.3803, Blink 3.0.104
Ecreator:2.21, eResponse 1.1.7
Webportal,WebApps,
CONFIDENTIALITY NOTICE: This email, including any attachments, contains
information which may be confidential or privileged. The information is
intended to be for the use of the individual or entity named above. If you
are not the intended recipient, be aware that any disclosure, copying,
distribution or use of the contents of this information is prohibited. If
you have received this email in error, please notify the sender immediately
by reply to sender only message and destroy all electronic and hard copies
of the communication, including attachments.
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Once you can accept the universe as matter expanding into nothing that is
something,wearing stripes with plaid comes easy. -- Albert Einstein
I know a little of everything, but a lot of nothing
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[asterisk-users] Asterisk CDR Error ??
hi all do you guys know why asterisk sometimes, in the cdr put the dst (the extension) number in the src ?? I have 4 digit extensions (DID) and sometimes, the same values if found in the src that usually have the calling user caller id. Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Files
Local channel will help you send your call through the dialplan. You can make all your decision there. If it answers, then the specified application will be execute. Check this example http://www.astblog.com/2008/09/18/use-the-power-of-local-channels/ David Klaverstyn wrote: I have successfully created call files and I can get Asterisk to make calls based on those files. The problem I have is that it seems you need to use a Channel for the first leg of the call file. This means I have to use either a ZAP, SIP or IAX2 channel. What I would prefer to do is send the first leg of the call to a context and extension so I can send the call using DUNDi rather than a predefined channel. Once the call has been established then is should go to context, extension so the second leg of the call can be completed. Is it possible to send the first leg of a call file to DUNDi and if not aviable send over IAX2 or then ZAP? The call files seem to be limited to a channel and not allow the first leg of the call to be decided by the path of a context, extension. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Internal Virus Database is out of date. Checked by AVG. Version: 8.0.100 / Virus Database: 269.23.16/1448 - Release Date: 5/16/2008 7:42 PM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] giving a user asterisk CLI access: how bad could it get
Did you know that any commandyou type in asterisk cli starting with exclamation point (!) is execute in the shell by asterisk ?? Example : running !ls will run 'ls' in your current directory So, be aware because your user can do whatever we want then. Dima wrote: On Sat, Nov 01, 2008 at 12:38:52AM +0100, Dima wrote: Setting the user's shell to /usr/sbin/rasterisk works. On login user gets into asterisk CLI if asterisk is running (user just has to have write permission to /var/lib/asterisk.*). How does that user login? client$ ssh [EMAIL PROTECTED] password: Asterisk SVN-branch-1.4-r137138, Copyright (C) 1999 - 2008 Digium, Inc. and others. ... Verbosity is at least 9 asterisk.machine*CLI CLI has the ability to create extensions, extensions which could execute the System application, pick up his phone, dial the extension, execute the command, and even cover his tracks by putting NoCDR in the extension path and removing the incriminating extension afterwards (again with the CLI). In 1.4, it's even easier: he can originate a call from the command line, perhaps even to a phone of a person he wanted to take the fall for his exploit. The person I'm giving the access to is an admin of that asterisk. It's up to him to do evil stuff with asterisk itself. as long as he can't get a shell and do rm -rf / I'm safe. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Internal Virus Database is out of date. Checked by AVG. Version: 8.0.100 / Virus Database: 269.23.16/1448 - Release Date: 5/16/2008 7:42 PM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk src=dst
Hi all I saw in the CDR stocked in mysql as well as those in the csv file that some time, the src field is the same as the dst field which is the extension. When does it happens. Here, we have 4 dgits extensions and most of the time the dst field is the extension and the src field is the 10 digit customer phone number. Do you know when does this happens ?? Thanks Ruddy Gbaguidi http://www.astblog.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] credit card processing
Hi Guys We have a service that can be use by our customer via a website and also via telephone. On the website, we already accept credit card by sending users to paypal website where we have an account. Now, we want to do the same with an IVR where people can call a number, enter their credit card number and expiration date. But I don't see any service or credit card procession company that offers this. What I want basicly is a service where I can send the credit card number I collected and expiration that and their charge the number and give me a status back. Do you know any company that do this ?? Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] credit card processing
Yes, we can do that. But : 1. we are not too confortabe about keeping users credit card informations in our databases 2. we are now targeting the 50, 60+ people and their are not confortable about a website. So, we want to be able to register people by phone, and they can make payments by phone. We provide long distance service, so the website is only for payments for now. It will be more easier if people can pay by phone as well. Chris Bagnall wrote: Most credit card processing gateways require you to have the user's name and address for AVS verification when you perform customer not present transactions. Easy enough to do over a website, but a bit more tricky on the phone. If these are for repeat orders, how about getting the user to register via the website first, entering a payment card to be used for future orders, then give then a customer number and PIN that can be used by telephone for future top-up orders? Something like that would be fairly easy to query against a database. I'd have thought. Regards, Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Internal Virus Database is out of date. Checked by AVG. Version: 8.0.100 / Virus Database: 269.23.16/1448 - Release Date: 5/16/2008 7:42 PM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Read one or X DTMF
Hi all I'm just having a problem now and I don't have any idea how to do this. It is pretty simple. When a customer calls, to speed up the navigation in the dialplan, I want something like Welcome. Please enter your 10 digit customer number or press * to register So, I want to read up to 10 digits, and if the user press *, I want to go to the next extension. Do you have an idea ?? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Read one or X DTMF
Thanks for the hint. Sorry about that. If I use your soution, I cannot make any difference between a user pressing * and a user that reach the timeout because he didn't enter any digit. In both cases, I will have an empty string Karsten Wemheuer wrote: Hi, Am Freitag, den 12.09.2008, 11:03 -0400 schrieb Ruddy Gbaguidi: Hi all I'm just having a problem now and I don't have any idea how to do this. It is pretty simple. When a customer calls, to speed up the navigation in the dialplan, I want something like Welcome. Please enter your 10 digit customer number or press * to register So, I want to read up to 10 digits, and if the user press *, I want to go to the next extension. Do you have an idea ?? You can use the read application to get some digits. This application returns the number a user entered in a variable. If the user enters '*' the variable is set to an empty string. You can than proceed in Your dialplan. To distinguish the answers, You can use the function len. The read application is able to play a audio file. (see the doc with 'core show application read') One little hint: If You start a new thread, create a new message instead of using an old one. Your question is now part of the thread about application jack and its runtime, what is probably not what You want. Maybe some people ignore Your mail, because they are not interessted in jack... Regards, Karsten ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Internal Virus Database is out of date. Checked by AVG. Version: 8.0.100 / Virus Database: 269.23.16/1448 - Release Date: 5/16/2008 7:42 PM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Read one or X DTMF
Hi thanks for the hint. That will works I think. But now, if I'm in an AGI script and I want to stay in there and don't want to jump from an extension to other in the dialplan, how can I do it ?? Tony Mountifield wrote: In article [EMAIL PROTECTED], Ruddy Gbaguidi [EMAIL PROTECTED] wrote: Hi all I'm just having a problem now and I don't have any idea how to do this. It is pretty simple. When a customer calls, to speed up the navigation in the dialplan, I want something like Welcome. Please enter your 10 digit customer number or press * to register So, I want to read up to 10 digits, and if the user press *, I want to go to the next extension. Do you have an idea ?? One possibility: [getnumber] exten = s,1,Background(please-enter-num-or-star) exten = s,n,Waitexten(30) exten = *,1,Goto(register,s,1) exten = _X*,1,Goto(register,s,1) exten = _XX*,1,Goto(register,s,1) exten = _XXX*,1,Goto(register,s,1) exten = _*,1,Goto(register,s,1) exten = _X*,1,Goto(register,s,1) exten = _XX*,1,Goto(register,s,1) exten = _XXX*,1,Goto(register,s,1) exten = _*,1,Goto(register,s,1) exten = _X*,1,Goto(register,s,1) exten = _XX,1,Do whatever exten = _XX,n,You want to do with exten = _XX,n,A 10-digit customer number [register] exten = s,1,Start registration process Hope that helps Cheers Tony Internal Virus Database is out of date. Checked by AVG. Version: 8.0.100 / Virus Database: 269.23.16/1448 - Release Date: 5/16/2008 7:42 PM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Read one or X DTMF
But user just needs to enter * instead of *# We are doing this because 80% of the callers already have an account, so, instead of playing : If you have an account, press 1, if not press 2 we prefer to play Enter you account now or press * if you don't have any Karsten Wemheuer wrote: Hi Ruddy, Am Freitag, den 12.09.2008, 13:22 -0400 schrieb Ruddy Gbaguidi: Thanks for the hint. Sorry about that. If I use your soution, I cannot make any difference between a user pressing * and a user that reach the timeout because he didn't enter any digit. In both cases, I will have an empty string You can use the variable EPOCH to get a timestamp before and after execution of the read application. If the difference of the two values evaluates to the timeout, the user enters nothing. Otherwise the user enters '*#' or directly the #-key without anything more. I don't know how to distinguish this two cases. Regards, Karsten ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Internal Virus Database is out of date. Checked by AVG. Version: 8.0.100 / Virus Database: 269.23.16/1448 - Release Date: 5/16/2008 7:42 PM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Read one or X DTMF
Thanks for your help. This can be add to Read command as feature Tony Mountifield wrote: In article [EMAIL PROTECTED], Ruddy Gbaguidi [EMAIL PROTECTED] wrote: Hi thanks for the hint. That will works I think. But now, if I'm in an AGI script and I want to stay in there and don't want to jump from an extension to other in the dialplan, how can I do it ?? Ah, you didn't say anything about AGI, so I gave you a solution just using the dialplan. If you are writing an AGI program to do this, you might just as well have a loop around a WAIT FOR DIGIT command and check each digit as it comes, collecting numeric digits until you have ten of them, or jumping to the registration section if you get a *. Cheers Tony Tony Mountifield wrote: In article [EMAIL PROTECTED], Ruddy Gbaguidi [EMAIL PROTECTED] wrote: Hi all I'm just having a problem now and I don't have any idea how to do this. It is pretty simple. When a customer calls, to speed up the navigation in the dialplan, I want something like Welcome. Please enter your 10 digit customer number or press * to register So, I want to read up to 10 digits, and if the user press *, I want to go to the next extension. Do you have an idea ?? One possibility: [getnumber] exten = s,1,Background(please-enter-num-or-star) exten = s,n,Waitexten(30) exten = *,1,Goto(register,s,1) exten = _X*,1,Goto(register,s,1) exten = _XX*,1,Goto(register,s,1) exten = _XXX*,1,Goto(register,s,1) exten = _*,1,Goto(register,s,1) exten = _X*,1,Goto(register,s,1) exten = _XX*,1,Goto(register,s,1) exten = _XXX*,1,Goto(register,s,1) exten = _*,1,Goto(register,s,1) exten = _X*,1,Goto(register,s,1) exten = _XX,1,Do whatever exten = _XX,n,You want to do with exten = _XX,n,A 10-digit customer number [register] exten = s,1,Start registration process Hope that helps Cheers Tony Internal Virus Database is out of date. Checked by AVG. Version: 8.0.100 / Virus Database: 269.23.16/1448 - Release Date: 5/16/2008 7:42 PM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Internal Virus Database is out of date. Checked by AVG. Version: 8.0.100 / Virus Database: 269.23.16/1448 - Release Date: 5/16/2008 7:42 PM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] After Dial execution, using DIALEDTIME, ANSWEREDTIME
First, if you want to use that, you may want pass the call tracknum to
the myagi.agi,
so you will know which call the dialedtime and answeredtime belongs to.
But you can use the Dial 'g' option that doesn't hangup up both legs of
the call when the called party hangs up.
selmak se wrote:
Hi,
I noticed that when dial terminates it does not return to the
dialplan, and therefore can not execute any entry after Dial().
Is there any trick to overcome this limitation ?
How I am supposed to handle the returned vales DIALEDTIME,
ANSWEREDTIME if I can not execute anything after Dial()?
I made a workaround with DeadAGI (below) but it is unreliable: if 2
calls end nearly at the same time I do not know to whom belongs the
ANSWEREDTIME value :
exten = h,1,DeadAGI(myagi.agi,0,${DIALEDTIME},${ANSWEREDTIME},00)
Any comments?
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Internal Virus Database is out of date.
Checked by AVG.
Version: 8.0.100 / Virus Database: 269.23.16/1448 - Release Date: 5/16/2008
7:42 PM
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Re: [asterisk-users] Perl AGI defunct process
Worked like a charm
Thanks you all
Darren Sessions wrote:
Ruddy,
I've used deadagi for years with perfect success.
If it's a perl agi module, you need to make absolutely sure that
you're using 'use strict' and 'use warnings' in the main agi file -as
well- as any includes. You'll need to test your agi while in console
mode, so any of the perl warning messages that get sent to the console
are visible. You'll want to get rid of any errors and warnings.
In addition, I've setup my agi scripts to execute cleanup functions
when they detect any kind of sig message just for good measure.
$SIG{INT} = 'cleanup';
$SIG{TERM} = 'cleanup';
$SIG{QUIT} = 'cleanup';
$SIG{HUP} = IGNORE;
With this approach, as I said before, I've ran perl agi apps in very
high call volumes at various companies for years without any issues.
Hope this helps.
- Darren
_
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
http://www.darrensessions.com
_
On Aug 19, 2008, at 10:20 PM, Igor A. Goncharovsky wrote:
Hi!
Ruddy Gbaguidi wrote:
I'm using DeadAgi and has set AGISIGHUP to no because I don't want my
script to stop if the user hangs up.
But when it reach the end of the script, the child process should die.
And I don't see why I only have this trouble with perl agis.
Can you check if your script realy don't get SIGHUP?
Some time ago I have problem with that setting AGISIGHUP to 'no' have no
effect.
--
Best regards,
Igor A. Goncharovsky
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Internal Virus Database is out of date.
Checked by AVG.
Version: 8.0.100 / Virus Database: 269.23.16/1448 - Release Date: 5/16/2008
7:42 PM
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[asterisk-users] Perl AGI defunct process
Hi all. I'm using asterisk 1.4.21.2 and when I run ps -ef |grep defunct, I can see a lot of my perl agi still pending there. The channel has been cleaned up in asterisk. I don't have this kind of problem with python or php. I'm using ubuntu ... Anyone has an idea ? I've tried export LD_ASSUME_KERNEL=2.4.1 but after that I fail to even start asterisk. Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Perl AGI defunct process
I'm using DeadAgi and has set AGISIGHUP to no because I don't want my script to stop if the user hangs up. But when it reach the end of the script, the child process should die. And I don't see why I only have this trouble with perl agis. Eric ManxPower Wieling wrote: Your script is not catching SIGHUP, which is what Asterisk uses to tell the AGI the channel went away. Ruddy Gbaguidi wrote: Hi all. I'm using asterisk 1.4.21.2 and when I run ps -ef |grep defunct, I can see a lot of my perl agi still pending there. The channel has been cleaned up in asterisk. I don't have this kind of problem with python or php. I'm using ubuntu ... Anyone has an idea ? I've tried export LD_ASSUME_KERNEL=2.4.1 but after that I fail to even start asterisk. Internal Virus Database is out of date. Checked by AVG. Version: 8.0.100 / Virus Database: 269.23.16/1448 - Release Date: 5/16/2008 7:42 PM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX2 variable sharing
Hi all Back in the 1.2 days I think, there were some discussions about how two asterisk servers can share channel variables through an IAX protocol. I don't see anything in 1.4 at least to be able to make it done. Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 variable sharing
It doesn't seems to be working ... What I wanted to do is on the first server, Set a channel variable... then dial the number. When I received the call on the remote server, use that variable ... Is it possible ? Richard Lyman wrote: Ruddy Gbaguidi wrote: Hi all Back in the 1.2 days I think, there were some discussions about how two asterisk servers can share channel variables through an IAX protocol. I don't see anything in 1.4 at least to be able to make it done. Thanks Back in 1.2 you had to use type 'friend' to pass vars, as the 'peer' structure didn't have vars, only 'user' did. It is/was probably the same with 1.4 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Internal Virus Database is out of date. Checked by AVG. Version: 8.0.100 / Virus Database: 269.23.16/1448 - Release Date: 5/16/2008 7:42 PM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] channel variables not kept
You are using AGI or DeadAGI ? Paradise Dove wrote: hi, i'm using asterisk 1.4.21.2, and i use channel variables in my agi scripts. the problem is that some variables (and maybe all, not sure) like ANSWEREDTIME does not kept if the caller hangs up. my agi script continues to run after caller/callee hangup but the variables are not set properly if callers hangs up. is there anything i should to to avoid this or it's a bug. thanks, paradise dove ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Internal Virus Database is out of date. Checked by AVG. Version: 8.0.100 / Virus Database: 269.23.16/1448 - Release Date: 5/16/2008 7:42 PM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] channel variables not kept
Try DeadAGI and it should work.. Paradise Dove wrote: I'm using AGI and set AGISIGHUP=no to make it keep on running on channel hangup On Fri, Aug 8, 2008 at 10:24 PM, Ruddy Gbaguidi [EMAIL PROTECTED] wrote: You are using AGI or DeadAGI ? Paradise Dove wrote: hi, i'm using asterisk 1.4.21.2, and i use channel variables in my agi scripts. the problem is that some variables (and maybe all, not sure) like ANSWEREDTIME does not kept if the caller hangs up. my agi script continues to run after caller/callee hangup but the variables are not set properly if callers hangs up. is there anything i should to to avoid this or it's a bug. thanks, paradise dove ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Internal Virus Database is out of date. Checked by AVG. Version: 8.0.100 / Virus Database: 269.23.16/1448 - Release Date: 5/16/2008 7:42 PM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Internal Virus Database is out of date. Checked by AVG. Version: 8.0.100 / Virus Database: 269.23.16/1448 - Release Date: 5/16/2008 7:42 PM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] channel variables not kept
I don't really know :) I run DeadAGI with that error for many months now and nothing ever happened Paradise Dove wrote: Thanks, It works now! but i get this warning as well: Running DeadAGI on a live channel will cause problems, please use AGI is it serious? what problems will occur!?? On Fri, Aug 8, 2008 at 11:30 PM, Ruddy Gbaguidi [EMAIL PROTECTED] wrote: Try DeadAGI and it should work.. Paradise Dove wrote: I'm using AGI and set AGISIGHUP=no to make it keep on running on channel hangup On Fri, Aug 8, 2008 at 10:24 PM, Ruddy Gbaguidi [EMAIL PROTECTED] wrote: You are using AGI or DeadAGI ? Paradise Dove wrote: hi, i'm using asterisk 1.4.21.2, and i use channel variables in my agi scripts. the problem is that some variables (and maybe all, not sure) like ANSWEREDTIME does not kept if the caller hangs up. my agi script continues to run after caller/callee hangup but the variables are not set properly if callers hangs up. is there anything i should to to avoid this or it's a bug. thanks, paradise dove ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Internal Virus Database is out of date. Checked by AVG. Version: 8.0.100 / Virus Database: 269.23.16/1448 - Release Date: 5/16/2008 7:42 PM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Internal Virus Database is out of date. Checked by AVG. Version: 8.0.100 / Virus Database: 269.23.16/1448 - Release Date: 5/16/2008 7:42 PM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Internal Virus Database is out of date. Checked by AVG. Version: 8.0.100 / Virus Database: 269.23.16/1448 - Release Date: 5/16/2008 7:42 PM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange beep during calls
maybe you are using the L option in Dial app to limit the conversation time. Check those channel variables (just a wild guess) *LIMIT_PLAYAUDIO_CALLER **LIMIT_PLAYAUDIO_CALLEE **LIMIT_TIMEOUT_FILE **LIMIT_CONNECT_FILE **LIMIT_WARNING_FILE * Felippe Silvestre wrote: Hi all, Our users are complaining about beeps that happen in the middle of some calls. They are similar to the sound heard you are in a call and press any button in your phone. Please find bellow some examples of these beeps(the recordings are in Portuguese, but the beeps are easy to identify): http://www.katizak.locaweb.com.br/asterisk/beep.mp3 http://www.katizak.locaweb.com.br/asterisk/beep2.mp3 http://www.katizak.locaweb.com.br/asterisk/beep3.mp3 http://www.katizak.locaweb.com.br/asterisk/beep4.mp3 We are sure that our users are not pressing any button in the softphones during the conversations. Do you guys are able to identify where these beeps are coming from? Maybe an * functionality that we need to turn off... We are using Asterisk 1.4.21.2. Thanks. Felippe Silvestre ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Internal Virus Database is out of date. Checked by AVG. Version: 8.0.100 / Virus Database: 269.23.16/1448 - Release Date: 5/16/2008 7:42 PM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] vars in Macros called by DIAL with option M
I don't think you can do that because, asterisk, in the caller thread
will only read MACRO_RESULT to know if he has to connect the call or not.
A workaround will be to :
1. before the dial, add a row in a database table and retrieve an id
2. pass the id to test_connect and test_connect will then write his
variable value into the database
3. after the dial,. use the id to retrieve the needed variable.
Hope this will help.
Thomas Winter wrote:
Hi all,
Iam using an DIAL Command wird Macro if callee is answer the call.
exten = 123,n,DIAL(SIP/[EMAIL PROTECTED],180,gM(test_connect))
exten = 123,n,NoOp( ${var_from_macro})
In Macro test_connect Iam generating an new variable var_from_macro and would
like to use this var in the original dialplan.
I tried also __var_from_macro but didnt work. How can I set vars in macros
called by DIAL so that I can use these vars in the Dialplan or in the h
extention.
best regards
Thomas
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Internal Virus Database is out of date.
Checked by AVG.
Version: 8.0.100 / Virus Database: 269.23.16/1448 - Release Date: 5/16/2008
7:42 PM
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Re: [asterisk-users] vars in Macros called by DIAL with option M
And if you use DIALSTATUS and ANSWERTIME to check the last dial status,
you need to take care of the following bug
http://bugs.digium.com/view.php?id=13216
Thomas Winter wrote:
Hi all,
Iam using an DIAL Command wird Macro if callee is answer the call.
exten = 123,n,DIAL(SIP/[EMAIL PROTECTED],180,gM(test_connect))
exten = 123,n,NoOp( ${var_from_macro})
In Macro test_connect Iam generating an new variable var_from_macro and would
like to use this var in the original dialplan.
I tried also __var_from_macro but didnt work. How can I set vars in macros
called by DIAL so that I can use these vars in the Dialplan or in the h
extention.
best regards
Thomas
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Internal Virus Database is out of date.
Checked by AVG.
Version: 8.0.100 / Virus Database: 269.23.16/1448 - Release Date: 5/16/2008
7:42 PM
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Re: [asterisk-users] list of minutes spent on SIP phone calls?! any advice?!
You can check asterisk CDR (call detail records). You should have a csv file in /var/log/asterisk/cdr-csv/Master.csv You can also configure it to write the CDR in a database http://www.voip-info.org/wiki-Asterisk+cdr+mysql Then you can just write a script that will look at your database and send you a report every x day RoLaNd RoLaNd wrote: Hi All, i have asterisk with 9 SIP accounts on it. i was wondering if theres a way to setup asterisk, to send the amount of minutes each SIP account have spent incoming as well as outgoing and if possible the number it called! any advice?! any help would truly be appreciated..! thanks in advance and best regards, Connect to the next generation of MSN Messenger Get it now! http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-ussource=wlmailtagline ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Internal Virus Database is out of date. Checked by AVG. Version: 8.0.100 / Virus Database: 269.23.16/1448 - Release Date: 5/16/2008 7:42 PM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
