[asterisk-users] Hung Lines on GXP2000s?
I'm using Call Files to generate calls for a custom in-house click to dial. Every now and then we get lines or maybe calls on our GXP2000's that we can't hang up. They aren't showing as active channels in Asterisk. They eventually disappear off the line on the phone but I'm not sure what they are. Any ideas? Thanks Gang! - PB ___ -- 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] HUD display?
Did a lot of looking/digging on this and realized I might not have asked clearly. I'm generating the calls using call files, but can't seem to get the info to show in HUD right. Is there something I could do in the call file to get HUD to show the right thing? On Thu, Aug 20, 2009 at 11:02 AM, Danny Nicholas da...@debsinc.com wrote: Users.conf is where you’re going to be able to tweak this. If the line has a Full Name value, that will show up, otherwise the Asterisk code will insert “Unknown”. Since you are Perl’ing, I’d just put the “Unknown” Substitution there; the users.conf change is a YMMV fix. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *J. G. *Sent:* Thursday, August 20, 2009 9:55 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] HUD display? Hey gang, Trying something a little funky with a click to dial app. (Perl AGI) It's working fine but I'd like to change what displays in HUD's screen. Right now it's showing Unknown. I've tried setting various CDR fields but can't seem to change the display. I've tried setting CID and other AGI variables but nothing seems to affect that display. Any ideas? Thanks! PB ___ -- 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 ___ -- 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] HUD display?
Hey gang, Trying something a little funky with a click to dial app. (Perl AGI) It's working fine but I'd like to change what displays in HUD's screen. Right now it's showing Unknown. I've tried setting various CDR fields but can't seem to change the display. I've tried setting CID and other AGI variables but nothing seems to affect that display. Any ideas? Thanks! PB ___ -- 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] ForkCDR and setting the account info?
I've been Googling all morning and searching voip-info.org but not quite finding what I'm looking for. I've read that you can modify the billing/account information on a CDR via AGI but I can't find an example or a how to. I'd like to then assign specific accounts in the CDRs. Possible? Thanks, PB ___ -- 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] ForkCDR and setting the account info?
Gah - I've been trying to find the proper search syntax all day.. I Googled asterisk CDR Function and it's the first thing that comes up... Sometimes I wonder whether or not my brain works right.. Thanks Barry! On Thu, Aug 6, 2009 at 3:03 PM, Barry L. Kline blkl...@attglobal.netwrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 J. G. wrote: I've been Googling all morning and searching voip-info.org http://voip-info.org but not quite finding what I'm looking for. I've read that you can modify the billing/account information on a CDR via AGI but I can't find an example or a how to. I'd like to then assign specific accounts in the CDRs. Possible? Look up the CDR function. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFKeyj/CFu3bIiwtTARAgozAKCPGThtSJzrqs2vlMMvEsfzICDIkgCcD1oE 9oKNSFEDD06hZQ5qa/T9FQI= =YlEU -END PGP SIGNATURE- ___ -- 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 ___ -- 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] Channel Variables in a Call file?
Hey gang, I'm trying to find a) If you can put channel variables into a Call file and b) what the appropriate syntax is. Any ideas? Thanks, PB ___ -- 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] Channel Variables in a Call file?
Yeah, I accidentally pressed Send too early on my question - I knew I was
going to get a google back, was trying to beat the buzzer on a reply!
I am using as stated, but when I attempt to retrieve the channel variable in
an AGI, I get nothing back correctly
In my Call file:
Setvar: LICENSE=test
In my AGI
$AGI-get_full_variable('LICENSE');
(or)
$AGI-get_variable('LICENSE');
AGI Rx GET FULL VARIABLE LICENSE
AGI Tx 200 result=1 (LICENSE)
AGI Rx NOOP -- LICENSE = LICENSE
(instead of 'test')
So, is my Call file syntax incorrect? Is my AGI retrieve incorrect?
Thanks,
PB
On Tue, Jul 21, 2009 at 8:03 AM, Doug Lytle supp...@drdos.info wrote:
J. G. wrote:
Hey gang,
I'm trying to find a) If you can put channel variables into a Call
file and b) what the appropriate syntax is.
Google is your friend:
http://www.the-asterisk-book.com/unstable/call-file.html
Parameters
These parameters may be used in call files:
|Setvar: var=value|
/|Setvar:|/ lets you set one or more channel variables.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
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Re: [asterisk-users] Channel Variables in a Call file?
ok, did that and added the Concise option to try as well
I see the channel but not the variable in either option
Local/s...@obtest2-2059,1:macro-stdexten:s:8:Up:Dial:Local/s...@obtest
/n|:191922::3:80:Local/s...@obtest-91a2,1
Wonder if my syntax is slightly off on Setvar...
On Tue, Jul 21, 2009 at 9:39 AM, Miguel Molina mmol...@millenium.com.cowrote:
J. G. escribió:
Yeah, I accidentally pressed Send too early on my question - I knew
I was going to get a google back, was trying to beat the buzzer on a
reply!
I am using as stated, but when I attempt to retrieve the channel
variable in an AGI, I get nothing back correctly
In my Call file:
Setvar: LICENSE=test
In my AGI
$AGI-get_full_variable('LICENSE');
(or)
$AGI-get_variable('LICENSE');
AGI Rx GET FULL VARIABLE LICENSE
AGI Tx 200 result=1 (LICENSE)
AGI Rx NOOP -- LICENSE = LICENSE
(instead of 'test')
So, is my Call file syntax incorrect? Is my AGI retrieve incorrect?
Thanks,
PB
To be sure of the values of the variables you set on the call file,
while the call is on you can do a CLI core show channel tech/chan.
That will give you a complete information, including channel variables
and their actual values. If the call happens too fast, add a
Wait(seconds) on your dialplan to keep the channel alive while you
check its info with the command.
Cheers,
--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center
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[asterisk-users] Originate (Executing a System Command)
I know I'm doing something simple and wrong, but I can't quite figure it out: Example (executing system command): Action: Originate Channel: Local/1...@dummy Application: System http://www.voip-info.org/wiki/view/Asterisk+cmd+System Data: /path/to/script I keep getting a Unable to request channel and am not sure what it is looking for in place of Local/1...@dummy. The script is an internal voice delivery to my agents (among other things I'd like to do) Thanks! PB ___ -- 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] AGI Transfer?
I've been trying to get an AGI transfer to work for several weeks now. It isn't error-ing out, but it isn't working either. I can't use dial in this case due to what I'm trying to accomplish. Does an AGI Transfer actually work? -= Info about application 'Transfer' =- [Synopsis] Transfer caller to remote extension [Description] Transfer([Tech/]dest[|options]): Requests the remote caller be transferred to a given destination. If TECH (SIP, IAX2, LOCAL etc) is used, only an incoming call with the same channel technology will be transfered. Note that for SIP, if you transfer before call is setup, a 302 redirect SIP message will be returned to the caller. The result of the application will be reported in the TRANSFERSTATUS channel variable: SUCCESS Transfer succeeded FAILURE Transfer failed UNSUPPORTED Transfer unsupported by channel driver The option string many contain the following character: 'j' -- jump to n+101 priority if the channel transfer attempt fails Thanks!! PB ___ -- 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] Skype for Asterisk. Any return of experience ?
I received a phone call asking for specs, how I'd use, etc, etc and they said they'd be turning my beta account up in another 6 weeks. That was 3 weeks ago. PB On Mon, Jun 29, 2009 at 1:31 AM, randulo spamsucks2...@gmail.com wrote: Though they have written me back twice to say coming soon I am still waiting for the software... So you'd rather have it even when it hasn't been finished? Umm, no, but then when a company says looking for beta testers - please sign up now! and then four months later has nothing to let me beta test, I am a bit put off. The beta was limited. Digium wants to open it but says Skype themselves are delaying the operation. I have compelling reasons to believe this, even though I can't put them out in public. I was surprised too at the apparent slowness, but I think it will happen in good time. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AMI Transfer?
Trying to accomplish something that seems simple enough but I've tried
everything I can think of but I cannot get an AGI Transfer to work.
Seems simple enough
$AGI-exec('transfer','SIP/101');
and here's the resultant Debug:
AGI Rx EXEC transfer SIP/101
-- AGI Script Executing Application: (transfer) Options: (SIP/101)
AGI Tx 200 result=0
But nothing happens.
I can't use dial (which does work incidentally) because dial creates
channels and messes with the CDR info.
Any ideas?
Thanks!
PB
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Re: [asterisk-users] AMI + AGI for outbound click to dial
Ok - after a lot of playing I'm still a bit stuck. I'd like to accomplish the following - can't get it to work as it should (at least in my head! LOL) I've got an app that initiates an AMI call for Originate. I want to click a number onscreen, send the Originate, then (this is the part I can't figure out) have the box call my extension, adding in the SIPAddHeader info for answer-after:0 (so my phone auto-picks up) as well as spawning a MixMonitor to record the call in a specified format. (I have the AGI working for the mixmonitor) Make sense? maybe? ... chuckle Thanks for the help! PB ___ -- 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] AMI + AGI for outbound click to dial
Ping-ponging the call? That's a good idea.. Now, to try to accomplish that in an AGI script. Thanks Jim! PB On Wed, May 6, 2009 at 11:18 AM, Jimmy Godbout s...@inbox.com wrote: Send your call to a different extension that will set the header before calling your phone. -Original Message- From: pallet...@gmail.com Sent: Wed, 6 May 2009 10:51:30 -0400 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] AMI + AGI for outbound click to dial Ok - after a lot of playing I'm still a bit stuck. I'd like to accomplish the following - can't get it to work as it should (at least in my head! LOL) I've got an app that initiates an AMI call for Originate. I want to click a number onscreen, send the Originate, then (this is the part I can't figure out) have the box call my extension, adding in the SIPAddHeader info for answer-after:0 (so my phone auto-picks up) as well as spawning a MixMonitor to record the call in a specified format. (I have the AGI working for the mixmonitor) Make sense? maybe? ... chuckle Thanks for the help! PB ___ -- 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 -- - Jason Gehman General Manager North Voice Communications www.NorthVC.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
Re: [asterisk-users] AMI + AGI for outbound click to dial
Would this return during the ring or only after the remote party has picked up? On Wed, May 6, 2009 at 12:51 PM, Jimmy Godbout s...@inbox.com wrote: In your AMI portion, you set the outgoing call first, then the extension you want to be reached at: Action: Originate Channel: Zap/g2/8135551212 Context: default Exten: 101 Priority: 1 Timeout: 3 In the dialplan: [default] exten = 101,1,SIPAddHeader(... exten = 101,n,Dial(... exten = 101,n,... J. -Original Message- From: pallet...@gmail.com Sent: Wed, 6 May 2009 11:42:43 -0400 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] AMI + AGI for outbound click to dial Ping-ponging the call? That's a good idea.. Now, to try to accomplish that in an AGI script. Thanks Jim! PB On Wed, May 6, 2009 at 11:18 AM, Jimmy Godbout s...@inbox.com wrote: Send your call to a different extension that will set the header before calling your phone. -Original Message- From: pallet...@gmail.com Sent: Wed, 6 May 2009 10:51:30 -0400 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] AMI + AGI for outbound click to dial Ok - after a lot of playing I'm still a bit stuck. I'd like to accomplish the following - can't get it to work as it should (at least in my head! LOL) I've got an app that initiates an AMI call for Originate. I want to click a number onscreen, send the Originate, then (this is the part I can't figure out) have the box call my extension, adding in the SIPAddHeader info for answer-after:0 (so my phone auto-picks up) as well as spawning a MixMonitor to record the call in a specified format. (I have the AGI working for the mixmonitor) Make sense? maybe? ... chuckle Thanks for the help! PB ___ -- 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 -- - Jason Gehman General Manager North Voice Communications www.NorthVC.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 -- - Jason Gehman General Manager North Voice Communications www.NorthVC.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] AMI + AGI for outbound click to dial
Hey Gang, Trying to figure out how I can do the following (have each part working individually but drawing a blank on combining) 1) click on-screen which sends an AMI originate (works fine) 2) the originated call is to an internal extension that looks up the number to be dialed (works) 3) then via Perl, adding in a SIPAddHeader for answer-after=0.. (works separate from the above) What I can't figure out is how I can click on screen, either send an AMI originate or something, then have the SIP header added in and have it ring back my grandstream to auto-dial back out. Any ideas? ___ -- 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
