[asterisk-users] CDR Posting Delay
We have a situation where it's sometimes taking Asterisk 17-19 minutes to post CDR's, both over the AMI, and over the MySQL socket. It seems however that they are logged locally to /var/log/asterisk/cdr-csv/Master.csv right after the call is terminated. Anyone got any idea what's causing this? It's a problem for us because we (badly IMHO) are using CDR's to maintain call state (if a user is in a call for example). Doug. ___ -- 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] Purchasing Digium IVR Prompts.
Just went to order some IVR prompts from the digium web site From the digium web site: We have created an easy and cost effective way to have customized recordings done quickly and with no hassle. I thought this was rather amusing, as: 1. If you want multiple prompts recorded, you need to submit a new order for each, which means that even prompts of a couple of words are still charged at $12. That is NOT cost effective. You could record all your prompts as a single order, but then you'd need to split up the prompts yourself with audio software. That is NOT hassle free. 2. Since prompts are recorded seperately, each shows up in the shopping cart as a separate item. There is no way to see what the requested prompt is! We're going to have a lot of these (remember, each prompt is different), and keeping track of them NOT hassle free. 3. From the web site Also, you have the ability to upload your own intonation file to ensure a personalized and professional recording every time. what the heck is an intonation file? Is it a text file? Is it an audio recording? What format? The web site doesn't seem to say. Lack of documentation on the web site is NOT hassle free. 4. Of course, when I called customer service, they had no clue. NOT hassle free. Doug. ___ -- 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] New Bridge Command/Event in 1.6?
Thanks Olle. How do I use it? What's the parameters??? Doug. - Original Message From: Johansson Olle E [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, July 20, 2008 1:36:24 AM Subject: Re: [asterisk-users] New Bridge Command/Event in 1.6? 20 jul 2008 kl. 02.55 skrev Douglas Garstang: I just downloaded Asterisk 1.6 beta 9 because I had read that there was a new bridge command. After looking through the doc/* documentation, I see no mention of a bridge application or AMI command. Does it exist? I am trying to take a bridged call, and redirect each to another destination, which I can do with the redirect() AMI command. After doing some dial plan processing, I would like to bridge them back together. How can I do this? The redirect command takes a channel and an extension as an argument, not another channel. Read the CHANGES file: * Added a Bridge action which allows you to bridge any two channels that are currently active on the system. The developer forgot to add documentation to doc/manager_1_1.txt. Adding doc would be helpful. /O ___ -- 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
[asterisk-users] New Bridge App/AMI Command in Asterisk 1.6?
I just downloaded Asterisk 1.6 beta 9 because I had read that there was a new bridge command. After looking through the doc/* documentation, I see no mention of a bridge application or AMI command. Does it exist? I am trying to take a bridged call, and redirect each to another destination, which I can do with the redirect() AMI command. After doing some dial plan processing, I would like to bridge them back together. How can I do this? The redirect command takes a channel and an extension as an argument, not another channel. Doug. ___ -- 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] New Bridge Command/Event in 1.6?
I just downloaded Asterisk 1.6 beta 9 because I had read that there was a new bridge command. After looking through the doc/* documentation, I see no mention of a bridge application or AMI command. Does it exist? I am trying to take a bridged call, and redirect each to another destination, which I can do with the redirect() AMI command. After doing some dial plan processing, I would like to bridge them back together. How can I do this? The redirect command takes a channel and an extension as an argument, not another channel. Doug. ___ -- 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] Tracking Call Time While in Dial()
The person I am working is building a calling card. They want to allow the user to recharge their account when their time runs out (without hanging up the current call). I got no idea how to implement that. In addition, they don't want to charge the user for the time they spend recharging their account. So, they need to track multiple timers for the call. Doug. - Original Message From: Tzafrir Cohen [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, July 12, 2008 1:46:13 AM Subject: Re: [asterisk-users] Tracking Call Time While in Dial() On Fri, Jul 11, 2008 at 10:52:53AM -0700, Douglas Garstang wrote: Wanting to provide a real time call timer on a web page. Can't you get information about other channels through the manager interface without this special AGI? Maybe you just need to somehow mark those channels as interesting before the Dial, or write out start time to a variable before the Dial starts. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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
[asterisk-users] Bridging two Redirected Channels?
All, I was able to use the Redirect AMI command to take two bridged channels and send them elsewhere in the dial plan. Great. Now... how can I bridge them back together again? Looks like Asterisk 1.6 might have a bridge command. What about Asterisk 1.4? Doug. ___ -- 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] Asterisk as an IVR solution
Well I can tell you that it makes a difficult programming environment, just a little more difficult. It means I can't implement a menu as a single reusable piece of code inside a macro. - Original Message From: Tilghman Lesher [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, July 10, 2008 6:07:36 PM Subject: Re: [asterisk-users] Asterisk as an IVR solution On Thursday 10 July 2008 19:13:50 Douglas Garstang wrote: It's a known problem. If you call Background() in a macro, then Asterisk will look for the extensions to jump to in the CALLING Macro/context and NOT the Macro that the Background() app was called in. I wouldn't call it a known problem. It works precisely as it was designed to work. It may not work the way that you want it to, but it works like a Macro: an independent set of instructions, with substitution, that acts as if it were invoked inline with the calling location. That is why Background will match in the context of the calling location: it acts like it never left that original context (and, in a way, it really didn't). Subroutines are a different beast, and they are available with the Gosub/ Return set of routines in app_stack.so. -- Tilghman ___ -- 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
Re: [asterisk-users] Tracking Call Time While in Dial()
Thanks, but that won't do what I need. By calling an AGI before the call starts and after the call ends, all I am doing is accounting the start and the end of the call, not actively monitoring the duration of the call as it occurs. - Original Message From: Cosmin Prund [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, July 11, 2008 3:57:23 AM Subject: Re: [asterisk-users] Tracking Call Time While in Dial() Call an AGI right before the start of the Dial command to record the start time and ether use an manager application (makes use of manager API) or call an DeadAGI once the call has ended (from the h extension). This requires a bit of programming - but then again some programming is required anyway to display the actual talk time somewhere. It might also be that I'm an programmer and I attempt to solve all problems writing programs, so maybe someone else has a better idea! -- Cosmin Prund De la:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] În numele Douglas Garstang Trimis: Thursday, July 10, 2008 7:49 PM Către: asterisk-users@lists.digium.com Subiect: [asterisk-users] Tracking Call Time While in Dial() So, I've been asked if this is possible. Someone wants to actively monitor the duration of a call, while the call is still in progress. Obviously, in Asterisk, once the Dial() application starts, you lose dial plan control until after the call has ended, successful or otherwise. Anyone know if that kind of thing is possible? Doug. ___ -- 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] Asterisk as an IVR solution
Yes, and by doing that your compounding the fact that all your variables are global. - Original Message From: randulo [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, July 11, 2008 12:14:28 AM Subject: Re: [asterisk-users] Asterisk as an IVR solution On Fri, Jul 11, 2008 at 8:28 AM, Douglas Garstang [EMAIL PROTECTED] wrote: Well I can tell you that it makes a difficult programming environment, just a little more difficult. It means I can't implement a menu as a single reusable piece of code inside a macro. I do the IVR stuff in a context and jump to it as needed. The context is reusable from anywhere. ___ -- 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
Re: [asterisk-users] Asterisk as an IVR solution
Well, a macro is the closest thing the dial plan has to a subroutine, and without that, we might as well be programming in Assembler (no subroutines, local variables, lots of goto's... sound familiar?). Doug. - Original Message From: Tilghman Lesher [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, July 11, 2008 7:20:40 AM Subject: Re: [asterisk-users] Asterisk as an IVR solution On Friday 11 July 2008 01:28:34 Douglas Garstang wrote: Well I can tell you that it makes a difficult programming environment, just a little more difficult. It means I can't implement a menu as a single reusable piece of code inside a macro. That's the point. A Macro is NOT a subroutine. It's like saying that you can't effectively hammer a nail with a screwdriver, and therefore you think the screwdriver has a known problem. There's nothing wrong with the screwdriver; it simply is the wrong tool for the job. -- Tilghman ___ -- 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
Re: [asterisk-users] Asterisk as an IVR solution
Ugh. Yes, the variables are local to the current channel. However, they are global to the entire dial plan within the current channel. I have stepped on myself many times because I've had a loop counter called $i for example, jumped somewhere else within that loop, reused the same variable name, $i, and screwed up my logic. Surely you where aware that's the type of thing I was talking about. I'd be surprised if you didn't. Doug. - Original Message From: Tilghman Lesher [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, July 11, 2008 7:36:54 AM Subject: Re: [asterisk-users] Asterisk as an IVR solution On Friday 11 July 2008 09:22:25 Douglas Garstang wrote: Yes, and by doing that your compounding the fact that all your variables are global. No, his variables are local to the channel he's using. Global variables are a completely different beast. -- Tilghman ___ -- 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
Re: [asterisk-users] Tracking Call Time While in Dial()
I want to track call duration while the call is in progress. - Original Message From: Tilghman Lesher [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, July 11, 2008 7:39:40 AM Subject: Re: [asterisk-users] Tracking Call Time While in Dial() On Friday 11 July 2008 09:21:56 Douglas Garstang wrote: Thanks, but that won't do what I need. By calling an AGI before the call starts and after the call ends, all I am doing is accounting the start and the end of the call, not actively monitoring the duration of the call as it occurs. It is unclear from your description what you want to do. Could you be more explicit? -- Tilghman ___ -- 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
Re: [asterisk-users] Asterisk as an IVR solution
A subroutine with arguments? - Original Message From: Tilghman Lesher [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, July 11, 2008 8:58:46 AM Subject: Re: [asterisk-users] Asterisk as an IVR solution On Friday 11 July 2008 09:40:55 Douglas Garstang wrote: Well, a macro is the closest thing the dial plan has to a subroutine, and without that, we might as well be programming in Assembler (no subroutines, local variables, lots of goto's... sound familiar?). I've mentioned Gosub at least twice before in this thread, which implements a subroutine. -- Tilghman ___ -- 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
Re: [asterisk-users] Asterisk as an IVR solution
Fine, I'll call it ${LoopVariable} then... how's that going to fix the problem?
- Original Message
From: Steve Edwards [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, July 11, 2008 8:43:47 AM
Subject: Re: [asterisk-users] Asterisk as an IVR solution
On Fri, 11 Jul 2008, Douglas Garstang wrote:
Ugh. Yes, the variables are local to the current channel. However, they
are global to the entire dial plan within the current channel. I have
stepped on myself many times because I've had a loop counter called $i
for example, jumped somewhere else within that loop, reused the same
variable name, $i, and screwed up my logic.
Ugh indeed. While I sympathize with your local/global name space issues,
you lose credibility with your false economy.
IMNSHO, anybody who uses a single [common] letter for a variable deserves
a bump in the temperature when they reach their final resting place :)
For example, out of the 157 applications on one of my Asterisk servers, 76
contain the letter l.
(absolutetimeout, adsiprog, agentcallbacklogin, agentlogin,
agentmonitoroutgoing, agi, alarmreceiver, appendcdruserfield,
authenticate, changemonitor, chanisavail, congestion, datetime, deadagi,
dial, dictate, digittimeout, directory, disa, dundilookup, eagi, endwhile,
execif, execiftime, externalivr, festival, getcpeid, gosubif, gotoif,
gotoiftime, hasnewvoicemail, hasvoicemail, iax2provision, ices, importvar,
lookupblacklist, lookupcidname, macroexit, macroif, mailboxexists,
meetmeadmin, milliwatt, mixmonitor, monitor, pickup, privacymanager,
readfile, realtime, realtimeupdate, responsetimeout, retrydial, ringing,
saydigits, sayphonetic, sayunixtime, sendimage, setcallerid,
setcdruserfield, setcidname, setcidnum, setrdnis, settransfercapabilit,
sipaddheader, sipdtmfmode, sipgetheader, stopmonitor, testclient,
txtcidname, vmauthenticate, voicemail, voicemailmain, wait, waitexten,
waitforring, waitforsilence, while)
Surely you can come up with a name slightly more descriptive -- maybe
idx?
Take pity on the next sod that has to plod through your dialplan. The
millisecond you spend typing a more meaningful name will be returned to
you (or your employer) a millionfold.
- Original Message
From: Tilghman Lesher [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, July 11, 2008 7:36:54 AM
Subject: Re: [asterisk-users] Asterisk as an IVR solution
On Friday 11 July 2008 09:22:25 Douglas Garstang wrote:
Yes, and by doing that your compounding the fact that all your variables
are global.
No, his variables are local to the channel he's using. Global variables are
a completely different beast.
--
Tilghman
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Thanks in advance,
Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] Tracking Call Time While in Dial()
Wanting to provide a real time call timer on a web page. - Original Message From: Daniel Hazelbaker [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, July 11, 2008 10:17:01 AM Subject: Re: [asterisk-users] Tracking Call Time While in Dial() On Jul 11, 2008, at 10:08 AM, Douglas Garstang wrote: I want to track call duration while the call is in progress. To accomplish what? Are you wanting to beep the channel every 10 seconds? Are you wanting to play a you have 60 seconds left message when they approach some quota? Are you wanting to limit the call to 5 minutes and 23 seconds? Daniel ___ -- 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] Recharge Dial Limit....?
Here's an interesting challange. I need to implement a calling card application, where I call the Dial() command and pass it (L)imit information. Nothing difficult about that. Except it is a requirement that rather than ending the call when the limit is reached, the user gets the option to recharge their account. Now, since the dial() command will just end the call when the limit has been reached, how could I possibly do this? The only way I can think of is to have another system send Asterisk a SIP reinvite before the call ends, and direct the media somewhere else so that we can drop into a new IVR and let them top off their account. A reinvite would have to go to the remote party too, so that they could listen to music on hold while the caller was topping off their account. It just occurred to me that this may not work. The (L)imit information passed to the Dial application has not changed. The Dial() application would still end the call. Ideas? Doug. ___ -- 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] Recharge Dial Limit....?
Thanks, but how does that extend the core functionality of Dial()? If Dial() can't do it, how does a wrapper do it? - Original Message From: Steve Totaro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, July 11, 2008 4:29:50 PM Subject: Re: [asterisk-users] Recharge Dial Limit? On Fri, Jul 11, 2008 at 7:12 PM, Douglas Garstang [EMAIL PROTECTED] wrote: Here's an interesting challange. I need to implement a calling card application, where I call the Dial() command and pass it (L)imit information. Nothing difficult about that. Except it is a requirement that rather than ending the call when the limit is reached, the user gets the option to recharge their account. Now, since the dial() command will just end the call when the limit has been reached, how could I possibly do this? The only way I can think of is to have another system send Asterisk a SIP reinvite before the call ends, and direct the media somewhere else so that we can drop into a new IVR and let them top off their account. A reinvite would have to go to the remote party too, so that they could listen to music on hold while the caller was topping off their account. It just occurred to me that this may not work. The (L)imit information passed to the Dial application has not changed. The Dial() application would still end the call. Ideas? Doug. Use an AGI, dissect ASTCC or ASTPP AGIs, all the goodies you want are in there. Thanks, Steve T ___ -- 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
[asterisk-users] Tracking Call Time While in Dial()
So, I've been asked if this is possible. Someone wants to actively monitor the duration of a call, while the call is still in progress. Obviously, in Asterisk, once the Dial() application starts, you lose dial plan control until after the call has ended, successful or otherwise. Anyone know if that kind of thing is possible? Doug. ___ -- 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] Asterisk as an IVR solution
Admittedly I have not used the ExternalIVR app. Is it any good? I'm not sure I agree that Asterisk is GOOD for building IVR's. Sure, it can do it, but boy it is UGLY. There's also the fact that you can't call Backgound() in a macro, which forces you to use Read() which won't accept a timeout of 1s. There's no DTMF background detection while playing SayDigits so you have to roll your own by calling an external AGI and concatenating sound files. Yuck. By the time you code in logic for handling timeouts and incorrect responses to menu's with all the gotos and what-not, it turns into a god aweful mess. Sure, you can do it. Doug. - Original Message From: Steve Totaro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, July 10, 2008 10:37:55 AM Subject: Re: [asterisk-users] Asterisk as an IVR solution On Thu, Jul 10, 2008 at 1:25 PM, Mark Carpenter [EMAIL PROTECTED] wrote: Hi. We are building an application that will provide users with the ability to call in and report an absence. The caller will have to validate themselves and the call tree will be dynamic, based on data in a MySQL database. We will have many customers, each calling a separate phone number, each having a different call tree. New customers will be added regularly and we do not want a solution that requires extensive programming each time (the call trees are different in subtle ways from each other). Is Asterisk a great solution for this? If not do you know what would? If so, we need someone to help us set it up, can you suggest someone? Thanks in advance. Best. Mark Asterisk certainly is a great solution for this. If you find you need or want extra flexibility, the external IVR app. http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ExternalIVR Thanks, Steve Totaro ___ -- 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] Asterisk as an IVR solution
Don't know about Asterisk 1.4, but in Asterisk 1.2 it expects the input in seconds. If you try and use 0, it seems to drop back to a default of 5s. - Original Message From: MFH [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, July 10, 2008 12:37:31 PM Subject: Re: [asterisk-users] Asterisk as an IVR solution From what I can tell Read allows for a floating point input which uses ast_waitfordigit that accepts milliseconds as input. Douglas Garstang wrote: Admittedly I have not used the ExternalIVR app. Is it any good? I'm not sure I agree that Asterisk is GOOD for building IVR's. Sure, it can do it, but boy it is UGLY. There's also the fact that you can't call Backgound() in a macro, which forces you to use Read() which won't accept a timeout of 1s. There's no DTMF background detection while playing SayDigits so you have to roll your own by calling an external AGI and concatenating sound files. Yuck. By the time you code in logic for handling timeouts and incorrect responses to menu's with all the gotos and what-not, it turns into a god aweful mess. Sure, you can do it. Doug. - Original Message From: Steve Totaro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, July 10, 2008 10:37:55 AM Subject: Re: [asterisk-users] Asterisk as an IVR solution On Thu, Jul 10, 2008 at 1:25 PM, Mark Carpenter [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi. We are building an application that will provide users with the ability to call in and report an absence. The caller will have to validate themselves and the call tree will be dynamic, based on data in a MySQL database. We will have many customers, each calling a separate phone number, each having a different call tree. New customers will be added regularly and we do not want a solution that requires extensive programming each time (the call trees are different in subtle ways from each other). Is Asterisk a great solution for this? If not do you know what would? If so, we need someone to help us set it up, can you suggest someone? Thanks in advance. Best. Mark Asterisk certainly is a great solution for this. If you find you need or want extra flexibility, the external IVR app. http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ExternalIVR Thanks, Steve Totaro ___ -- 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 ___ -- 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] Asterisk as an IVR solution
It's a known problem. If you call Background() in a macro, then Asterisk will look for the extensions to jump to in the CALLING Macro/context and NOT the Macro that the Background() app was called in. Eg: [macro-DoSomething] exten = s,1,Background(Prompt) exten = 1,1,NoOP() [context1] exten = s,1,Macro(DoSomething) If you press 1, Asterisk will look for an extension '1' in the context 'context1', NOT the 'DoSomething' macro. Doug. - Original Message From: Al Baker [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, July 10, 2008 4:50:19 PM Subject: Re: [asterisk-users] Asterisk as an IVR solution Why can't you call Background() from a MACRO ? Isn't is just an Application like any other ? Curious minds want to know ! Quote There's also the fact that you can't call Backgound() in a macro, Douglas Garstang wrote: Don't know about Asterisk 1.4, but in Asterisk 1.2 it expects the input in seconds. If you try and use 0, it seems to drop back to a default of 5s. - Original Message From: MFH [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, July 10, 2008 12:37:31 PM Subject: Re: [asterisk-users] Asterisk as an IVR solution From what I can tell Read allows for a floating point input which uses ast_waitfordigit that accepts milliseconds as input. Douglas Garstang wrote: Admittedly I have not used the ExternalIVR app. Is it any good? I'm not sure I agree that Asterisk is GOOD for building IVR's. Sure, it can do it, but boy it is UGLY. There's also the fact that you can't call Backgound() in a macro, which forces you to use Read() which won't accept a timeout of 1s. There's no DTMF background detection while playing SayDigits so you have to roll your own by calling an external AGI and concatenating sound files. Yuck. By the time you code in logic for handling timeouts and incorrect responses to menu's with all the gotos and what-not, it turns into a god aweful mess. Sure, you can do it. Doug. - Original Message From: Steve Totaro [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com Sent: Thursday, July 10, 2008 10:37:55 AM Subject: Re: [asterisk-users] Asterisk as an IVR solution On Thu, Jul 10, 2008 at 1:25 PM, Mark Carpenter [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi. We are building an application that will provide users with the ability to call in and report an absence. The caller will have to validate themselves and the call tree will be dynamic, based on data in a MySQL database. We will have many customers, each calling a separate phone number, each having a different call tree. New customers will be added regularly and we do not want a solution that requires extensive programming each time (the call trees are different in subtle ways from each other). Is Asterisk a great solution for this? If not do you know what would? If so, we need someone to help us set it up, can you suggest someone? Thanks in advance. Best. Mark Asterisk certainly is a great solution for this. If you find you need or want extra flexibility, the external IVR app. http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ExternalIVR Thanks, Steve Totaro ___ -- 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 ___ -- 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
[asterisk-users] Return VXML vars to Dial Plan
I'm using i6net's vxml browser in Asterisk. I'm trying to work out how I can return the inputs from a menu or form back into the Asterisk dial plan. Is there a variable? The exit tag apparently can be used to return a value (still trying to work out how to do that), but what about multiple values, such as with a form? If you don't return variables back into the dial plan, how do you execute Asterisk applications, such as the dial() command once your inside VXML? It would seem that executing the VXML() app, removes any chance of doing anything else with Asterisk. Isn't that kind of a MAJOR problem? Am I missing something? Doug. ___ -- 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] Building an IVR
So, I need to build a complicated IVR with Asterisk, with a lot of back end hooks. The dial plan itself has a lot of limitations, not the least of which is that the dial plan is ugly, hard to maintain, and full of gotchas like all variables being global etc etc. I've been involved with Asterisk for a couple of years now and this is a problem I have yet to see a good solution for. 1. I looked at VXML but it has too many integration problems. 2. AGI has overhead. 3. Fast AGI has single point of failure problems (we're using Asterisk 1.2 which bombs out the call when an AGI request fails), and has too many moving parts for what should be something fairly simple. 4. I'm aware of res_perl, but am not a fan of the maintainability of perl. 5. I looked for a valid link to res_python, but couldn't find anything. 6. Adhearsion? Looked at it a few months ago but couldn't work it out. There was too much 'voodoo' going on. 7. I'm not a C programmer, so writing a custom module, is both overkill and not feasible. Do I have any other options? Doug. ___ -- 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] Return Vars to Dial Plan from VXML()
I'm using i6net's vxml browser in Asterisk. I'm trying to work out how I can return the inputs from a menu or form back into the Asterisk dial plan. Is there a variable? It's not documented if it is. The exit tag apparently can be used to return a value (still trying to work out how to do that), but what about multiple values, such as with a form? Doug. ___ -- 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] Asterisk VXML... Help.
So, I'm trying to get the Asterisk vxml (from i6net) working. Having no luck with it. My dial plan has: exten = _X.,1,Answer() exten = _X.,n,Wait(1) exten = _X.,n,Vxml(file:///tmp/menu.vxml) The /tmp/menu.vxml file has: ?xml version=1.0? vxml version=1.0 form blockaudio src=tt-monkeys.gsm//block blockHello world!/block /form /vxml The tmp directory also has the tt-monkeys.gsm file: [EMAIL PROTECTED] tmp]# ls -l tt-monkeys.gsm -rw-r--r-- 1 root root 26697 Jul 3 20:57 tt-monkeys.gsm The openvxi daemon is running: [EMAIL PROTECTED] tmp]# ps -ef | grep openvxi root 2076 1 0 18:33 ?00:00:00 /bin/sh /usr/sbin/safe_openvxi root 2114 2076 0 18:33 ?00:00:00 openvxi -channels 100 -config /etc/openvxi/client.cfg root 2606 2409 0 21:00 pts/200:00:00 grep openvxi [EMAIL PROTECTED] tmp]# The /etc/asterisk/vxml.conf file contains: ; VoiceXML Configuration ; [general] wav_codec=gsm videosilence= audiosilence= [license] max=1 video=no key= And, finally here's my console output: -- Executing Vxml(SIP/xxx.201.84.142-b7600c30, file:///tmp/menu.vxml) in new stack VoiceBrowser interface file:///tmp/menu.vxml Initialiting == VXML_URL=(null) == VXML_ID=(null) == VXML_PARAM=(null) == url=file:///tmp/menu.vxml == session=1 == id=0 == param=0 == Opening (url=file:///tmp/menu.vxml, id=(null), param=(null)) == (dnid=1yyy3160157) == (name=1xxx8635808) == (num=1xxx8635808) == remote=1xxx8635808 == local=1yyy3160157 -- open|session=1|module=2|url=file:///tmp/menu.vxml|remote=1xxx8635808|local=1yyy3160157 -- open|session=1|result=ok Waiting -- close|session=1 Exiting == VXML_RESULT= I hear NOTHING. Asterisk drops though to the next command in the dial plan. Shouldn't I hear the tt-monkeys.gsm sound file being played? I tried to keep this as simple as I could. I thought it was interesting too that when I tried this with a web server instead of a local file, if the URL was wrong, the VXML() app still said it connected and got the data ok. ___ -- 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] Asterisk VXML... Help.
Not for file:// access, No... - Original Message From: Alexander Lopez [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, July 3, 2008 2:21:42 PM Subject: Re: [asterisk-users] Asterisk VXML... Help. Does vxml let you use absolute paths? Wouldn’t it have the equivalent of a DocRoot??? Alex From:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: Thursday, July 03, 2008 5:03 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk VXML... Help. So, I'm trying to get the Asterisk vxml (from i6net) working. Having no luck with it. My dial plan has: exten = _X.,1,Answer() exten = _X.,n,Wait(1) exten = _X.,n,Vxml(file:///tmp/menu.vxml) The /tmp/menu.vxml file has: ?xml version=1.0? vxml version=1.0 form blockaudio src=tt-monkeys.gsm//block blockHello world!/block /form /vxml The tmp directory also has the tt-monkeys.gsm file: [EMAIL PROTECTED] tmp]# ls -l tt-monkeys.gsm -rw-r--r-- 1 root root 26697 Jul 3 20:57 tt-monkeys.gsm The openvxi daemon is running: [EMAIL PROTECTED] tmp]# ps -ef | grep openvxi root 2076 1 0 18:33 ?00:00:00 /bin/sh /usr/sbin/safe_openvxi root 2114 2076 0 18:33 ?00:00:00 openvxi -channels 100 -config /etc/openvxi/client.cfg root 2606 2409 0 21:00 pts/200:00:00 grep openvxi [EMAIL PROTECTED] tmp]# The /etc/asterisk/vxml.conf file contains: ; VoiceXML Configuration ; [general] wav_codec=gsm videosilence= audiosilence= [license] max=1 video=no key= And, finally here's my console output: -- Executing Vxml(SIP/xxx.201.84.142-b7600c30, file:///tmp/menu.vxml) in new stack VoiceBrowser interface file:///tmp/menu.vxml Initialiting == VXML_URL=(null) == VXML_ID=(null) == VXML_PARAM=(null) == url=file:///tmp/menu.vxml == session=1 == id=0 == param=0 == Opening (url=file:///tmp/menu.vxml, id=(null), param=(null)) == (dnid=1yyy3160157) == (name=1xxx8635808) == (num=1xxx8635808) == remote=1xxx8635808 == local=1yyy3160157 -- open|session=1|module=2|url=file:///tmp/menu.vxml|remote=1xxx8635808|local=1yyy3160157 -- open|session=1|result=ok Waiting -- close|session=1 Exiting == VXML_RESULT= I hear NOTHING. Asterisk drops though to the next command in the dial plan. Shouldn't I hear the tt-monkeys.gsm sound file being played? I tried to keep this as simple as I could. I thought it was interesting too that when I tried this with a web server instead of a local file, if the URL was wrong, the VXML() app still said it connected and got the data ok. ___ -- 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] Set Language not working!
Argh! I have this...
[ct_start2]
exten = _X.,1,Set(LANGUAGE()=mig33/en/allison-tts)
exten = _X.,n,NoOp(${LANGUAGE()})
exten = _X.,n,Answer()
exten = _X.,n,Wait(1)
exten =
_X.,n,Playback(/var/lib/asterisk/sounds/mig33/en/allison-tts/please-enter-your-pin)
exten = _X.,n,Playback(please-enter-your-pin)
The first playback works, and the second does not. I get:
-- Executing Playback(SIP/xxx.201.84.147-09f11b58,
please-enter-your-pin) in new stack
Jun 27 18:28:27 WARNING[31382]: file.c:512 ast_openstream_full: File
please-enter-your-pin does not exist in any format
Jun 27 18:28:27 WARNING[31382]: file.c:824 ast_streamfile: Unable to open
please-enter-your-pin (format ulaw): No such file or directory
Jun 27 18:28:27 WARNING[31382]: app_playback.c:133 playback_exec:
ast_streamfile failed on SIP/xxx.201.84.147-09f11b58 for please-enter-your-pin
The File exists...
[EMAIL PROTECTED] asterisk]# ls -l /var/lib/asterisk/sounds/mig33/en/allison-tts
total 16
-rw-r--r-- 1 asterisk asterisk 13608 Jun 27 18:00 please-enter-your-pin.ulaw
What is wrong here? The call to set the language should cause Asterisk to look
for sound files in /var/lib/asterisk/sounds/mig33/en/allison-tts, and the file
IS there and IS readable because the first call with the explicit path works.
Asterisk 1.2.
Doug.
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[asterisk-users] Asterisk 1.2 app_vxml
I just downloaded the app_vxml for Asterisk 1.2 from i6net. Couldn't get it to work. We're using Asterisk 1.2 still, and it looks like the app_vxml binary was linked against libstdc_++-5.x (we have libstdc++-6.x). I grabbed the 1.4 version of the module hoping in vain that would work, but it fails with invalid symbols, which isn't surprising. Any ideas on how I can get this to work? Be nice if i6net provided source! Doug. ___ -- 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] Cepstral ... Swift... weird result
Asterisk 1.2, and Cepstral 5, Allison voice. I execute: swift Please enter your pin. -o please-enter-your-pin.ulaw -p audio/channels=1,audio/encoding=ulaw,audio/sampling-rate=8000 then copy it up to /var/lib/asterisk/sounds, and Play() the file. The sound file seems corrupted. All I hear is 'please' or 'please' followed by the rest of the sentence said so fast I almost can't hear it. I've tried other various of the -p option to swift, same results. Also tried generating a wav file and converting to ulaw with sox, same result. I did this once before and it worked. What am I doing wrong? Doug. ___ -- 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] Building a Complex IVR
I don't think anyone did, and I was hoping someone would. :) - Original Message From: Steve Murphy [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, June 24, 2008 3:57:48 PM Subject: Re: [asterisk-users] Building a Complex IVR On Mon, 2008-06-23 at 09:54 -0700, Douglas Garstang wrote: I'm about to build a complex IVR with Asterisk. Having done it a few times with the dial plan, I know it's going to be pretty ugly. What are my other options? I guess I could do it in AGI/FastAGI. What about VxML (about which I know almost nothing...)? Using Asterisk 1.2 Thanks, Doug. Sorry, I tried to peak thru all the stuff in this thread, but I may have missed it; has anyone suggested the externalIVR app? If not, it might be worth consideration...? murf -- Steve Murphy Software Developer Digium ___ -- 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] Building a Complex IVR
I'm about to build a complex IVR with Asterisk. Having done it a few times with the dial plan, I know it's going to be pretty ugly. What are my other options? I guess I could do it in AGI/FastAGI. What about VxML (about which I know almost nothing...)? Using Asterisk 1.2 Thanks, Doug. ___ -- 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] Building a Complex IVR
Right, except now I have to go write a multi-threaded, redundant FastAGI server in python (euww, hate java). That replaces the effort of doing it in the dial-plan with the effort required for a more complex application + the effort required to make it redundant. Asterisk 1.2 also does not recover from a failure to connect to a FastAGI server. When it fails to connect, the current call just bombs out. No recovery possible. Doug. - Original Message From: Steve Totaro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, June 23, 2008 10:02:37 AM Subject: Re: [asterisk-users] Building a Complex IVR On Mon, Jun 23, 2008 at 12:54 PM, Douglas Garstang [EMAIL PROTECTED] wrote: I'm about to build a complex IVR with Asterisk. Having done it a few times with the dial plan, I know it's going to be pretty ugly. What are my other options? I guess I could do it in AGI/FastAGI. What about VxML (about which I know almost nothing...)? Using Asterisk 1.2 Thanks, Doug. FastAGI is a good bet. You can patch it to jump N+101 so you can have failover in case the box hosting the AGI is unreachable, it will jump, instead of the default of just failing and halting. It also offloads the processing to a different box. Thanks, Steve Totaro ___ -- 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
Re: [asterisk-users] Building a Complex IVR
Oh, if only we where installing Asterisk from source. - Original Message From: Steve Totaro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, June 23, 2008 10:52:22 AM Subject: Re: [asterisk-users] Building a Complex IVR Asterisk 1.2 also does not recover from a failure to connect to a FastAGI server. When it fails to connect, the current call just bombs out. No recovery possible. Wrong. If you re-read my initial post, there is a patch for this. http://bugs.digium.com/view.php?id=4029 If you are going to complain about recommended solutions, then why ask in the first place? Just use FreePBX and copy over the pertinent parts of your conf files... Thanks, Steve T On Mon, Jun 23, 2008 at 1:31 PM, Douglas Garstang [EMAIL PROTECTED] wrote: Right, except now I have to go write a multi-threaded, redundant FastAGI server in python (euww, hate java). That replaces the effort of doing it in the dial-plan with the effort required for a more complex application + the effort required to make it redundant. Asterisk 1.2 also does not recover from a failure to connect to a FastAGI server. When it fails to connect, the current call just bombs out. No recovery possible. Doug. - Original Message From: Steve Totaro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, June 23, 2008 10:02:37 AM Subject: Re: [asterisk-users] Building a Complex IVR On Mon, Jun 23, 2008 at 12:54 PM, Douglas Garstang [EMAIL PROTECTED] wrote: I'm about to build a complex IVR with Asterisk. Having done it a few times with the dial plan, I know it's going to be pretty ugly. What are my other options? I guess I could do it in AGI/FastAGI. What about VxML (about which I know almost nothing...)? Using Asterisk 1.2 Thanks, Doug. FastAGI is a good bet. You can patch it to jump N+101 so you can have failover in case the box hosting the AGI is unreachable, it will jump, instead of the default of just failing and halting. It also offloads the processing to a different box. Thanks, Steve Totaro ___ -- 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 ___ -- 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
Re: [asterisk-users] Building a Complex IVR
I would build it this way: 1) Design the dialplan logically so it is understandable and maintainable. 2) Code up the AGIs in whatever language you are comfortable. I would use C, but that's what I'm most comfortable with. 3) Confirm everything works like you think it should. 4) Measure to identify where the real bottlenecks are. 5) Attack the top 1 or 2 bottlenecks. The solution may be: a) Recode an AGI in C. b) Re-implement an AGI as fastagi() on the same server. c) Re-implement an AGI as fastagi() on another server. 6) Go to step 3 Can FastAGI call FastAGI? The application needs to contact another FastAGI server written in Java to lookup various billing information. ___ -- 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] SayNumber while reading DTMF?
I'm using the SayNumber() app to read out a users balance for an IVR. Is there a way I can do that while waiting for DTMF input? Obviously, read() and Background() don't correctly say a number in number format. Thanks, Doug. ___ -- 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] SayNumber while reading DTMF?
Poo. Thanks Jared. - Original Message From: Jared Smith [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, June 10, 2008 10:24:46 AM Subject: Re: [asterisk-users] SayNumber while reading DTMF? On Tue, 2008-06-10 at 10:03 -0700, Douglas Garstang wrote: I'm using the SayNumber() app to read out a users balance for an IVR. Is there a way I can do that while waiting for DTMF input? Obviously, read() and Background() don't correctly say a number in number format. I don't know of an easy way of doing this, short of writing a routing in an AGI script to read numbers in the proper format. I'd personally love to see SayDigitsBackground, SayNumberBackground, SayAlphaBackground, etc. or even better, the ability to put numbers in the filename parameters to the Read application, such as: Read(some-variable,your-account-balanceis%123%dollars) (Obviously I just chose the percent sign as an arbitrary delimiter there, but you get the idea.) -- Jared Smith Training Manager Digium, Inc. ___ -- 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] Asterisk Wackyness
Here's a weird one. We have a situation where Asterisk seems to be losing it's ODBC database connection during idle periods. A workaround was to have a script connect to AMI and generate a bogus call, which would then generate a CDR and keep the connection alive. We didn't want to be generating actual network traffic for this, so I tried originating a call to [EMAIL PROTECTED] Somehow, magically, Asterisk maps a bogus host name (we have no peer in our config called 'xxx') to a IP of 205.234.182.xxx (not really .xxx...just hiding the IP) and a a host name of unknown.ord.scnet.net. How does that happen? I know it's doing this because I can see the SIP INVITE to out to this address. Seems like other bogus host names also map to the same place. Time to go and grep the source for 'scnet.net' I guess. Doug. ___ -- 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 Database Handling
General Asterisk question. We are sending CDR's to MySQL via odbc. It seems that Asterisk is sometimes dropping CDR's, and they aren't being sent to the database (they ARE in the Master.csv file though). We suspect that when the MySQL socket is idle, it gets disconnected, either by the MySQL server or by our firewall, and when Asterisk goes to send the next CDR over the socket, does not re-open the database, and drops that CDR. Possibly on the next call, it connects ok and sends the next CDR. This was a documented problem with voicemail users in the db in some previous version of Asterisk 1.2, and was patched and fixed in a later version of Asterisk 1.2 We are using Asterisk 1.2.19. So... surely this must be a general problem with ANY Asterisk module that uses the database. Do all modules use the same common database code or do they all use their own? If they all use their own, I guess idle database connection issues may be fixed in some modules and not others. If it's common Asterisk database code, is it all fixed in a newer version? Is it fixed in Asterisk 1.4? Thanks, Doug. ___ -- 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] Asterisk Database Handling
I couldn't find one for cdr_mysql.conf. We're using odbc anyway. MySQL directly might be an option if it works. Don't think we want to modify the server. - Original Message From: Alex Balashov [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, May 21, 2008 3:02:07 PM Subject: Re: [asterisk-users] Asterisk Database Handling Douglas Garstang wrote: We are sending CDR's to MySQL via odbc. It seems that Asterisk is sometimes dropping CDR's, and they aren't being sent to the database (they ARE in the Master.csv file though). We suspect that when the MySQL socket is idle, it gets disconnected, either by the MySQL server or by our firewall, and when Asterisk goes to send the next CDR over the socket, does not re-open the database, and drops that CDR. Possibly on the next call, it connects ok and sends the next CDR. Isn't there a keepalive option somewhere for cdr_mysql.conf, or failing that, a keepalive mechanism that can be enabled for TCP connections on the server side? -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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
Re: [asterisk-users] Asterisk Database Handling
- Original Message From: Alex Balashov [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, May 21, 2008 3:30:06 PM Subject: Re: [asterisk-users] Asterisk Database Handling Douglas Garstang wrote: I couldn't find one for cdr_mysql.conf. We're using odbc anyway. MySQL directly might be an option if it works. Don't think we want to modify the server. Perhaps there's a keepalive option to be set on the UnixODBC DSN, then? I already searched. Could not find one. ___ -- 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] Asterisk Database Handling
So... surely this must be a general problem with ANY Asterisk module that uses the database. Do all modules use the same common database code or do they all use their own? If they all use their own, I guess idle database connection issues may be fixed in some modules and not others. If it's common Asterisk database code, is it all fixed in a newer version? Is it fixed in Asterisk 1.4? Thanks, Doug. I have no answer, just a potential work around if you get no good answers. I guess you could just pipe the master.csv into your database (similar to how queuemetrics does queue_log) Are you saying this is a known problem? ___ -- 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] Asterisk Database Handling
I personally can tell you I've never had a problem with either the PostgreSQL or MySQL cdr apps themselves losing records. However, I can't say personally how well the ODBC method works. I'll just stick to saying that if you're considering using the cdr_mysql addon, I would highly suggest it as I've used it with MUCH success on high load servers. It's interesting you say that Sherwood. Does your MySQL server have some sort of keep alive setting? I suspect this is a general problem that would affect any and all Asterisk database connectivity. Doug. ___ -- 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] Asterisk Database Handling
I personally can tell you I've never had a problem with either the PostgreSQL or MySQL cdr apps themselves losing records. However, I can't say personally how well the ODBC method works. I'll just stick to saying that if you're considering using the cdr_mysql addon, I would highly suggest it as I've used it with MUCH success on high load servers. Oh... Also... does your system have idle periods? You said your servers where under high load. The issue seems to be that Asterisk does not re-establish the database connection after it is disconnected do to being idle. If you don't have idle periods, you'd never see this problem anyway. ___ -- 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] Asterisk Database Handling
Not at all, just offering a workaround. If your master.csv is complete and correct then it makes sense to use that data unless someone can identify your problem and offer a fix. Unfortunately, not really feesible. I didn't design the system but we are using CDR's not only for billing purposes, but also to keep state on users. The state info (in a call, not in a call etc) needs to be updated in as close to real time as possible. Files won't do it. ___ -- 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] Asterisk Database Handling
Looks like an Asterisk 1.4 option? - Original Message From: Tilghman Lesher [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, May 21, 2008 4:39:24 PM Subject: Re: [asterisk-users] Asterisk Database Handling On Wednesday 21 May 2008 17:02:07 Alex Balashov wrote: Douglas Garstang wrote: We are sending CDR's to MySQL via odbc. It seems that Asterisk is sometimes dropping CDR's, and they aren't being sent to the database (they ARE in the Master.csv file though). We suspect that when the MySQL socket is idle, it gets disconnected, either by the MySQL server or by our firewall, and when Asterisk goes to send the next CDR over the socket, does not re-open the database, and drops that CDR. Possibly on the next call, it connects ok and sends the next CDR. Isn't there a keepalive option somewhere for cdr_mysql.conf, or failing that, a keepalive mechanism that can be enabled for TCP connections on the server side? Please check the 'idlecheck' option in res_odbc.conf. -- Tilghman ___ -- 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] Sound Prompt 'per'
Anyone know where I can find an Alison recording of the word 'per'? Seems silly to buy the word 'per' from Digiums web site. And, I'd rather not open up audio editing software and get my 'per' prompt by editing it out of something else. Doug. Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ___ -- 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] Stupid Timeout Question
I haven't done this for a while... yes, that is my excuse. What the heck is wrong with this? [general] autofallthrough=yes exten = s,n(prompt),NoOp() exten = s,n,Background(wish-to-continue) exten = s,n,Background(1-yes-2-no) exten = s,n,WaitExten(5) ; User entered nothing exten = t,1,Playback(yes-dear) exten = t,n,Goto(s,prompt) It never gets to the timeout extension when the user enters nothing. I tried it with autofallthrough set to no as well. No change. Asterisk 1.2. What am I missing? Doug. Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ___ -- 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] REGISTER Outboundproxy
Is it possible to set an outboundproxy for REGISTER from Asterisk? register = xxx:[EMAIL PROTECTED] [foobar] type=peer host=sip99.foobar.com disallow=all allow=g729 canreinvite=no secret=yyy fromuser=xxx port=5099 outboundproxy=xxx.42.149.69 However, SIP REGISTER still goes directly to sip99.foobar.com, not xxx.42.149.69. Thanks, Doug. Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ___ -- 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] REGISTER Outboundproxy
Oops, I got that wrong... should have been register = xxx:[EMAIL PROTECTED] Doug. - Original Message From: Douglas Garstang [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, April 18, 2008 11:56:27 AM Subject: [asterisk-users] REGISTER Outboundproxy Is it possible to set an outboundproxy for REGISTER from Asterisk? register = xxx:[EMAIL PROTECTED] [foobar] type=peer host=sip99.foobar.com disallow=all allow=g729 canreinvite=no secret=yyy fromuser=xxx port=5099 outboundproxy=xxx.42.149.69 However, SIP REGISTER still goes directly to sip99.foobar.com, not xxx.42.149.69. Thanks, Doug. Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ___ -- 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] Error in Callback CDR
Using Asterisk 1.2, still. We are issuing a callback. User rejects the first two calls, but answers the third. For some reason, the Manager Interface outputs a CDR with disposition 'NO ANSWER' for all three attempts, eventhough the 3rd call worked. Is this an asterisk 1.2 bug? Doug. Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping___ -- 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] Post call QoS in Asterisk 1.4
It's time to ask this question again. Maybe I will get a reply one day. :) Asterisk 1.4 has some channel variables that you can inspect after a call is complete that will give you QoS metrics. Stuff like average round trip time, etc. Since there's only one set of variables, and calls will have two channels, which channel is this information for? Is it for one of the channels? Is it an aggregate of both channels? Who added this code and what where they thinking when they wrote it? Thanks, Doug. Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs___ -- 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] LCR in Asterisk
- Original Message From: Jay R. Ashworth [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, February 13, 2008 9:45:34 AM Subject: Re: [asterisk-users] LCR in Asterisk On Wed, Feb 13, 2008 at 11:33:19AM -0600, Tilghman Lesher wrote: On Wednesday 13 February 2008 09:57:59 Alex Balashov wrote: Tilghman Lesher wrote: Uh, why not? You can do LCR quite easily in the dialplan, by using func_odbc for each of the provider lookups, then use SORT() to get the lowest cost. It's quite easy, and you do not need to resort to AGI. You can do almost anything in the dial plan with enough spiritual commitment in about the same way that you can do just about anything you need to do with a bash script, as opposed to Perl, Python, or any toolkits or frameworks. Is that nasty little problem of no local variables in macros fixed yet? That's a pretty big pain in the ass. You have to prefix your variables with the name of the macro it's in to avoid stepping all over yourself. Doug. Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs___ -- 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] Post Call QoS....?
Ok, so I've asked this question before, and didn't get an answer. So here I go again! Asterisk 1.4 has some channel variables that you can inspect after a call is complete that will give you QoS metrics. Stuff like average round trip time, etc. Since there's only one set of variables, and calls will have two channels, which channel is this information for? Is it for one of the channels? Is it an aggregate of both channels? Who added this code and what where they thinking when they wrote it? Thanks, Doug. Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs___ -- 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] Post Call QoS?
Ok, so I've asked this question before, and didn't get an answer. So here I go again! Asterisk 1.4 has some channel variables that you can inspect after a call is complete that will give you QoS metrics. Stuff like average round trip time, etc. Since there's only one set of variables, and calls will have two channels, which channel is this information for? Is it for one of the channels? Is it an aggregate of both channels? Who added this code and what where they thinking when they wrote it? Thanks, Doug. Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs___ -- 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] IAX Registraion Refresh
I have Asterisk 1.4 registering via IAX to another Asterisk machine. How can I change the default registration timeout of 60s? I need my Asterisk box to register every HOUR Anyone? Editting source isn't an option. Doug. Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs___ -- 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] sipsock_read: BAD! BAD! BAD!
Does anyone know the cause of these BAD BAD BAD messages? I think I lost all my calls when it happened too. We have nagios running against IAX and nagios reports that IAX is down. It would seem that the entire application locks up when this happens and calls are dropped. Connected to Asterisk 1.2.14 currently running on flexo (pid = 26846) Verbosity is at least 3 flexo*CLI show channels Channel Location State Application(Data) SIP/teleglobe-09f887 (None) Down(None) 1 active channel 12 active calls Jan 30 03:28:13 WARNING[2671]: channel.c:781 channel_find_locked: Avoided deadlock for '0x9f6a6f0', 10 retries! Jan 30 03:28:14 ERROR[26983]: chan_sip.c:11451 sipsock_read: We could NOT get the channel lock for SIP/teleglobe-09f83250 - Call ID [EMAIL PROTECTED] Jan 30 03:28:14 ERROR[26983]: chan_sip.c:11452 sipsock_read: SIP MESSAGE JUST IGNORED: BYE Jan 30 03:28:14 ERROR[26983]: chan_sip.c:11453 sipsock_read: BAD! BAD! BAD! Jan 30 03:28:15 ERROR[26983]: chan_sip.c:11451 sipsock_read: We could NOT get the channel lock for SIP/teleglobe-09f83250 - Call ID [EMAIL PROTECTED] Jan 30 03:28:15 ERROR[26983]: chan_sip.c:11452 sipsock_read: SIP MESSAGE JUST IGNORED: BYE Jan 30 03:28:15 ERROR[26983]: chan_sip.c:11453 sipsock_read: BAD! BAD! BAD! Jan 30 03:28:16 ERROR[26983]: chan_sip.c:11451 sipsock_read: We could NOT get the channel lock for SIP/teleglobe-09f83250 - Call ID [EMAIL PROTECTED] Jan 30 03:28:16 ERROR[26983]: chan_sip.c:11452 sipsock_read: SIP MESSAGE JUST IGNORED: BYE Jan 30 03:28:16 ERROR[26983]: chan_sip.c:11453 sipsock_read: BAD! BAD! BAD! Jan 30 03:28:19 ERROR[26983]: chan_sip.c:11451 sipsock_read: We could NOT get the channel lock for SIP/teleglobe-09f83250 - Call ID [EMAIL PROTECTED] Jan 30 03:28:19 ERROR[26983]: chan_sip.c:11452 sipsock_read: SIP MESSAGE JUST IGNORED: BYE Doug. Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping___ -- 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] Simultaneous Callback?!
We're doing callback here. Asterisk dials a number, waits for an answer, plays a prompt, dials a second number, and bridges the channels together. Calls are initiated from the AMI. No problems there. Easy stuff. However, I'd like to know if it's possible to have Asterisk dial the same two numbers simultaneously, play the prompt to the first one that answers, dial the second one and bridge the two channels together.? I'm not even sure how this would work within the limits of the dial plan. Normally, the dialing of the first leg is implicit (the channel in an AMI originate command), that is, there is no dial plan code for it, although you can specify a Local channel and asterisk will then jump into the dial plan to dial the first number. Once the first one answers, Asterisk jumps to the location specified by the second number (ie [EMAIL PROTECTED]) and calls it, and bridges them together. How would this work with simultaneous numbers? Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ -- 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] Help! channel_find_deadlocked: Avoided initial deadlock for ...
Replying to myself. :) I just noticed the deadlock message still displayed on the console at the end of a normal call, so the the deadlock message is not related to the early CANCEL - Original Message From: Douglas Garstang [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, January 8, 2008 5:31:12 PM Subject: [asterisk-users] Help! channel_find_deadlocked: Avoided initial deadlock for ... Hope someone can help. I have a situation where asterisk is sending a SIP CANCEL message before the Dial() timeout has hit. It doesn't always do it. Normally, we send an INVITE to the ITSP. They respond with a 100 Trying, then a 180 Ringing, or 183 Session Progress. It seems to be at this point that Asterisk starts the dial timer. Normally, when no more replies have been received by the dial timeout, Asterisk sends a CANCEL message. That's all fine, and when this happens, this is what appears on the console: -- Called [EMAIL PROTECTED] -- SIP/teleglobe-09879188 is making progress passing it to SIP/teleglobe-09876568 -- Nobody picked up in 4 ms -- Executing PlayTones(SIP/teleglobe-09876568, congestion) in new stack However, when asterisk sends the CANCEL earlier then this, this is what appears on the console: -- SIP/teleglobe-09879188 is making progress passing it to SIP/teleglobe-09876568 == Spawn extension (default, callback, 7) exited non-zero on 'SIP/teleglobe-09876568' Jan 9 01:16:34 WARNING[5719]: channel.c:781 channel_find_locked: Avoided initial deadlock for '0x97f24d8', 10 retries! Does anyone know what the deadlock message is all about? It is ocurring quite frequently. This is Asterisk 1.2.14. Thanks, Doug Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs___ -- 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] Help! channel_find_deadlocked: Avoided initial deadlock for ...
Hope someone can help. I have a situation where asterisk is sending a SIP CANCEL message before the Dial() timeout has hit. It doesn't always do it. Normally, we send an INVITE to the ITSP. They respond with a 100 Trying, then a 180 Ringing, or 183 Session Progress. It seems to be at this point that Asterisk starts the dial timer. Normally, when no more replies have been received by the dial timeout, Asterisk sends a CANCEL message. That's all fine, and when this happens, this is what appears on the console: -- Called [EMAIL PROTECTED] -- SIP/teleglobe-09879188 is making progress passing it to SIP/teleglobe-09876568 -- Nobody picked up in 4 ms -- Executing PlayTones(SIP/teleglobe-09876568, congestion) in new stack However, when asterisk sends the CANCEL earlier then this, this is what appears on the console: -- SIP/teleglobe-09879188 is making progress passing it to SIP/teleglobe-09876568 == Spawn extension (default, callback, 7) exited non-zero on 'SIP/teleglobe-09876568' Jan 9 01:16:34 WARNING[5719]: channel.c:781 channel_find_locked: Avoided initial deadlock for '0x97f24d8', 10 retries! Does anyone know what the deadlock message is all about? It is ocurring quite frequently. This is Asterisk 1.2.14. Thanks, Doug Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping___ -- 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] is Power fail transfer possible with asterisk?
When I saw the subject I thought the poster was maybe asking if was possible to transfer the live RTP stream from one Asterisk system to another in the event that power was lost - Original Message From: MatsK [EMAIL PROTECTED] To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, January 2, 2008 4:37:56 PM Subject: Re: [asterisk-users] is Power fail transfer possible with asterisk? John covici wrote: OK, to clarify a bit, he wants to fix things so that all we are depending on are the pots lines -- I know if they go out you are gone. So what can we do in that case? There is ATA boxes with a port that you connect to a analog line and when SIP fails to register will it use the analog port. Some ATA boxes routes emergency calls direct to the analog port. I think that Sipura SPA-3000, http://www.sipura.com/products/spa3000.htm will cover your needs. on Wednesday 01/02/2008 Tilghman Lesher([EMAIL PROTECTED]) wrote On Wednesday 02 January 2008 17:10:05 John covici wrote: Hi. I have a client who wants some way that his analog phones can call out even after the power is out and the UPS has died -- some way that a phone can connect directly to an fxo or some such when power is gone. Any hardware around which can do this? I have heard of some ATA's which do this, do any of the channel banks have this capability? 1) If his phones are this critical, he needs a triple redundant generator. 2) Ask him what he would like to do after 36 hours of power outage, when even the telco stops being able to provide battery on their POTS lines. If your provider is out, there's very little you can do. Perhaps a ham radio attached to a car battery? Speed costs; how fast would you like to go? -- Tilghman No need to be rude Mr Tilghman, try to be constructive. ___ --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 Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping___ --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] Setting Multiple Values via func_odbc ...?
Alex, Yes, but the issue isn't MySQL. The issue is func_odbc and passing multiple values to it. Douglas. - Original Message From: Alex Balashov [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, December 6, 2007 10:23:02 AM Subject: Re: [asterisk-users] Setting Multiple Values via func_odbc ...? On Thu, 6 Dec 2007, Douglas Garstang wrote: I need to insert/update multiple MySQL columns in a single row with the func_odbc function at the SAME TIME. If I understand your question correctly, this can be done using a standard SQL UPDATE query. UPDATE tblname SET col1 = val1, col2 = val2, .. WHERE field = criterion; -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --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 Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs___ --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] Setting Multiple Values via func_odbc ...?
I need to insert/update multiple MySQL columns in a single row with the
func_odbc function at the SAME TIME.
Someone showed me how to use ARRAY to retrieve multiple values at the same
time, but I need to SET multiple values.
Can this be done?
If not, I will just stick with MySQL, but that's a pain in the ass because the
asterisk-addons package has no default rpm spec file for building an RPM.
I had something like this in func_odbc.conf:
[VOX_LOG_CALL_LEG]
dsn=MySQL
write=INSERT into CallLog (Source,IDDCode,AreaCode,ProviderId,SIPReply) values
(${VAL1},${VAL2},${VAL3},${VAL4},${VAL5})
but it doesn't like it. In order for this to work, I'd have to have several
LOG_CALL_LEG functions, each taking one parameter, and then requiring several
database updates!
Doug.
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[asterisk-users] CDR Function in Hangup Channel
So... I'm trying to access CDR(duration) and CDR(billsec) inside h... I keep getting 0. Can I access the CDR function inside a hangup extensions? Asterisk 1.4.13 Thanks, Doug. Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ --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] CDR Function in Hangup Channel
Ok, this is a little crazy...
billsec and duration are 0, but disposition is ANSWERED.
Huh?
h = {
NoOp(*** LEG B HANGUP ${CDR(duration)} ${CDR(billsec)}
${CDR(disposition)});
AddCallLeg(${LEGB_SOURCE},${LEGB_DEST},1,2,${HANGUPCAUSE});
};
- Original Message
From: Douglas Garstang [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, December 6, 2007 12:04:29 PM
Subject: CDR Function in Hangup Channel
So... I'm trying to access CDR(duration) and CDR(billsec) inside h...
I keep getting 0. Can I access the CDR function inside a hangup extensions?
Asterisk 1.4.13
Thanks, Doug.
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Re: [asterisk-users] CDR Function in Hangup Channel
Oh Crap. So there's no way to get the duration and billsec from the dial plan then? - Original Message From: Tilghman Lesher [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, December 6, 2007 1:19:59 PM Subject: Re: [asterisk-users] CDR Function in Hangup Channel On Thursday 06 December 2007 14:54:14 Douglas Garstang wrote: Ok, this is a little crazy... billsec and duration are 0, but disposition is ANSWERED. Huh? That's correct. Both of those values depend upon the call be ENDED. If the call is not yet ended, neither of those values has yet been determined. -- Tilghman ___ --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 Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ --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] CDR Function in Hangup Channel
Got it! endbeforehexten=yes Wooo! - Original Message From: Steve Edwards [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, December 6, 2007 2:31:54 PM Subject: Re: [asterisk-users] CDR Function in Hangup Channel On Thu, 6 Dec 2007, Joshua Colp wrote: There is an option which can be enabled in the general section of cdr.conf, endbeforehexten, which will cause the values to be calcuated before entering the h extension. A 1.4-ism? Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --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 Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping___ --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] Adhearsion Install Fails.
Not strictly an Asterisk question. I've tried to install adhearsion on TWO relatively fresh CentOS 5.x systems, and I get this... [EMAIL PROTECTED] rubygems-0.9.5]# gem install adhearsion Bulk updating Gem source index for: http://gems.rubyforge.org ERROR: While executing gem ... (Errno::ENOENT) No such file or directory - /usr/lib/ruby/gems/1.8/gems/adhearsion-0.7.7/bin/ahn The directory /usr/lib/ruby/gems/1.8/gems/adhearsion-0.7.7 exists, but it has no 'bin' directory. The ahn binary is located in /usr/lib/ruby/gems/1.8/gems/adhearsion-0.7.7. A google search apparently tells me that no one else in the known universe has ever had this problem, and I get it on two systems? I must be doing something fundamentally wrong! [EMAIL PROTECTED] adhearsion]# rpm -qa | grep ruby ruby-1.8.5-5.el5_1.1 ruby-libs-1.8.5-5.el5_1.1 ruby-devel-1.8.5-5.el5_1.1 ruby-rdoc-1.8.5-5.el5_1.1 ruby-irb-1.8.5-5.el5_1.1 [EMAIL PROTECTED] adhearsion]# cat /etc/issue CentOS release 5 (Final) Kernel \r on an \m [EMAIL PROTECTED] adhearsion]# uname -a Linux localhost.localdomain 2.6.18-8.1.15.el5 #1 SMP Mon Oct 22 08:32:04 EDT 2007 i686 i686 i386 GNU/Linux Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ --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] Multiple Return Values from func_odbc
Thanks... That was just what I needed.
But what about going the other way? How can I pass multiple values to a
function in func_odbc?
I can't use ARRAY as it can only be used to set variables, not read form them!
Doug.
- Original Message
From: Tilghman Lesher [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, November 27, 2007 9:08:50 PM
Subject: Re: [asterisk-users] Multiple Return Values from func_odbc
On Tuesday 27 November 2007 20:05:55 Douglas Garstang wrote:
Is there any way to return multiple values from functions defined in
func_odbc.conf? It appears that you can only return one value.
Use the ARRAY function:
func_odbc.conf:
read=SELECT foo,bar FROM tablename WHERE baz='${ARG1}'
extensions.conf:
Set(ARRAY(foo,bar)=${ODBC_WHATEVER(bax)})
--
Tilghman
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Re: [asterisk-users] Multiple Return Values from func_odbc
Thanks... That was just what I needed.
But what about going the other way? How can I pass multiple values to a
function in func_odbc?
I can't use ARRAY as it can only be used to set variables, not read form them!
Doug.
- Original Message
From: Tilghman Lesher [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, November 27, 2007 9:08:50 PM
Subject: Re: [asterisk-users] Multiple Return Values from func_odbc
On Tuesday 27 November 2007 20:05:55 Douglas Garstang wrote:
Is there any way to return multiple values from functions defined in
func_odbc.conf? It appears that you can only return one value.
Use the ARRAY function:
func_odbc.conf:
read=SELECT foo,bar FROM tablename WHERE baz='${ARG1}'
extensions.conf:
Set(ARRAY(foo,bar)=${ODBC_WHATEVER(bax)})
--
Tilghman
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[asterisk-users] Multiple Return Values from func_odbc
Is there any way to return multiple values from functions defined in func_odbc.conf? It appears that you can only return one value. True? Hope not Doug. Be a better pen pal. Text or chat with friends inside Yahoo! Mail. See how. http://overview.mail.yahoo.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] Building an Asterisk 1.4 RPM
I'm a little confused. I'd like to build an RPM for Asterisk 1.4. Is it better to modify and use the spec file under redhat/asterisk.spec and run a 'make rpm', OR is it better to build a custom spec file from scratch and use 'rpmbuid -ba' specfile? How do people normally do it? The problem I see with a custom spec file is that since the source is all contained within a tar.gz file, there's no way to interactively run a 'make menuselect' first and customise or remove what you don't need. For example, if I don't do this, the ogg vorbis module is installed by default, and then when I go to install my rpm, there's complaints all round if the ogg vorbis libs aren't already installed. Doug. Get easy, one-click access to your favorites. Make Yahoo! your homepage. http://www.yahoo.com/r/hs ___ --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] Zaptel 1.4 spec file
Does anyone know where I can get an rpm spec file for zaptel 1.4.x? Thanks, Doug. Be a better sports nut! Let your teams follow you with Yahoo Mobile. Try it now. http://mobile.yahoo.com/sports;_ylt=At9_qDKvtAbMuh1G1SQtBI7ntAcJ___ --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] Building an Asterisk 1.4 RPM.
I'm a little confused. I'd like to build an RPM for Asterisk 1.4. Is it better to modify and use the spec file under redhat/asterisk.spec and run a 'make rpm', OR is it better to build a custom spec file from scratch and use 'rpmbuid -ba' specfile? How do people normally do it? The problem I see with a custom spec file is that since the source is all contained within a tar.gz file, there's no way to interactively run a 'make menuselect' first and customise or remove what you don't need. For example, if I don't do this, the ogg vorbis module is installed by default, and then when I go to install my rpm, there's complaints all round if the ogg vorbis libs aren't already installed. Doug. Get easy, one-click access to your favorites. Make Yahoo! your homepage. http://www.yahoo.com/r/hs ___ --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] AEL2 and Callbacks
- Original Message
From: Richard Lyman [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, November 1, 2007 8:47:28 AM
Subject: Re: [asterisk-users] AEL2 and Callbacks
Douglas Garstang wrote:
I am originating a command via the AMI with this...
Action: Login
Username: xxx
Secret: yyy
ACTION: Originate
Async: yes
Timeout: 6
Exten: callback
Channel: Local/[EMAIL PROTECTED]
Callerid: 849120
Context: default
ActionID: 849120
My LegA context:
---
context LegA {
_X. = {
Dial(SIP/[EMAIL PROTECTED]);
}
}
And my default context:
--
context default {
callback = {
NoCDR();
Wait(1);
Dial(${destination},60,oL(${timeout}:${timeout_warning}:${timeout_warning_repeat}));
}
}
The A leg is established, and once Asterisk goes to dial the B leg...
-- Executing [EMAIL PROTECTED]:1]
Dial(Local/[EMAIL PROTECTED],2, SIP/[EMAIL PROTECTED]) in new stack
-- Called [EMAIL PROTECTED]
-- SIP/Provider-09a8cff8 is making progress passing it to
Local/[EMAIL PROTECTED],2
-- SIP/Provider-09a8cff8 answered Local/[EMAIL PROTECTED],2
== Starting Local/[EMAIL PROTECTED],1 at default,callback,1
failed so falling back to exten 's'
== Starting Local/[EMAIL PROTECTED],1 at default,s,1 still
failed so falling back to context 'default'
[Oct 31 01:57:07] WARNING[29795]: pbx.c:2450 __ast_pbx_run: Channel
'Local/[EMAIL PROTECTED],1' sent into invalid extension 's' in
context 'default', but no invalid handler
Uhm, why? I have a default context with a callback extension. Of
course I have no explicit priority 1 though... this is AEL2
What's it complaining for?
Doug.
originates have always had an issue where it falls back to an 's'
extension. and since you do not have one, nor an 'i' for invalid
extension... it bombs out.
Yes... but I DO have a default context and I DO have a callback extension.
What's it whining about?
Doug.
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[asterisk-users] AEL2 and Callbacks
I am originating a command via the AMI with this...
Action: Login
Username: xxx
Secret: yyy
ACTION: Originate
Async: yes
Timeout: 6
Exten: callback
Channel: Local/[EMAIL PROTECTED]
Callerid: 849120
Context: default
ActionID: 849120
My LegA context:
---
context LegA {
_X. = {
Dial(SIP/[EMAIL PROTECTED]);
}
}
And my default context:
--
context default {
callback = {
NoCDR();
Wait(1);
Dial(${destination},60,oL(${timeout}:${timeout_warning}:${timeout_warning_repeat}));
}
}
The A leg is established, and once Asterisk goes to dial the B leg...
-- Executing [EMAIL PROTECTED]:1] Dial(Local/[EMAIL PROTECTED],2,
SIP/[EMAIL PROTECTED]) in new stack
-- Called [EMAIL PROTECTED]
-- SIP/Provider-09a8cff8 is making progress passing it to Local/[EMAIL
PROTECTED],2
-- SIP/Provider-09a8cff8 answered Local/[EMAIL PROTECTED],2
== Starting Local/[EMAIL PROTECTED],1 at default,callback,1 failed so falling
back to exten 's'
== Starting Local/[EMAIL PROTECTED],1 at default,s,1 still failed so falling
back to context 'default'
[Oct 31 01:57:07] WARNING[29795]: pbx.c:2450 __ast_pbx_run: Channel
'Local/[EMAIL PROTECTED],1' sent into invalid extension 's' in context
'default', but no invalid handler
Uhm, why? I have a default context with a callback extension. Of course I have
no explicit priority 1 though... this is AEL2
What's it complaining for?
Doug.
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[asterisk-users] MySQL() timeout
Anyone know if the MySQL() application has a configurable timeout? If it tries to connect to a bogus IP, it's timeout seems to be a few minutes. I'd like to cut it down to a few seconds. Doug. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] MySQL() timeout
I guess... it shouldn't be too hard to find the time out value in the source and change it - Original Message From: Doug Lytle [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, October 30, 2007 5:23:35 PM Subject: Re: [asterisk-users] MySQL() timeout Douglas Garstang wrote: Anyone know if the MySQL() application has a configurable timeout? If it tries to connect to a bogus IP, it's timeout seems to be a few minutes. I never got a response on that question myself. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] A Leg Control on Asterisk Callback
I'm confused about something. It's the way Asterisk handles the A leg (ie the first party dialed) on an originate command via the Manager Interface. Lets say our originate commands looks like this: ACTION: Originate Async: yes Timeout: 6 Exten: callback Channel: SIP/[EMAIL PROTECTED] Variable: destination=SIP/[EMAIL PROTECTED] Callerid: 5551212 Context: default ActionID: 849120 Priority: 1 Asterisk first goes and dials the Channel parameter, SIP/[EMAIL PROTECTED] This is where it gets confusing. You have no control over what happens here. The actions don't even appear on the Asterisk console debug. It isn't until this party has picked up, and control jumps to the 'callback' extension, that Asterisk shows you what it is doing. So, I went and changed the Channel parmeter to Channel: Local/[EMAIL PROTECTED], and made a LegA context: [LegA] exten = _X.,1,Dial(SIP/[EMAIL PROTECTED]) exten = _X.,n,Playback(tt-monkeys) I wanted to have control over the call both before and after it is placed. I wanted to be able to play a prompt to the caller before the call is placed to the destination number. However, since we've dialled the A party already, we have no control over the dial plan anymore after they have answered, and I can't play prompts. What can I do here? Doug. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] Asterisk 1.4 from RPM
I'm trying to build an Asterisk rpm from the supplied asterisk.spec file. Made numerous changes to get it to work. The architecture of the system I am building on is x86_64. I'd like to build for i686 though. I added a --target i686 to the rpmbuild line in the Makefile, but it looks like it's still requiring 64bit system libraries. When I try to install the rpm on the i686 machine, it complains it doesn't have the 64 bit libraries. How can I build with 32 bit libraries? Doug. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] Asterisk 1.4 from RPM
Since I'm executing a 'make rpm' from within the Asterisk 1.4.13 distribution source, I'd say it's an Asterisk question. - Original Message From: Philip Prindeville [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, October 29, 2007 6:24:06 PM Subject: Re: [asterisk-users] Asterisk 1.4 from RPM That's really a question for [EMAIL PROTECTED] The short and generally not very helpful answer is that there are a lot of poorly packaged software releases out there that don't play well with cross-development environments. -Philip Douglas Garstang wrote: I'm trying to build an Asterisk rpm from the supplied asterisk.spec file. Made numerous changes to get it to work. The architecture of the system I am building on is x86_64.. I'd like to build for i686 though. I added a --target i686 to the rpmbuild line in the Makefile, but it looks like it's still requiring 64bit system libraries. When I try to install the rpm on the i686 machine, it complains it doesn't have the 64 bit libraries. How can I build with 32 bit libraries? Doug. ___ --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 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] Getting SIP Response Code from HANGUPCAUSE
Ah jeez. All I wanted to do was connect to a carrier and then perform fail over logic based on their SIP response. Not supposed to be difficult. This is what Asterisk is supposed to be good at. We have a SIP module, why not have SIP responses available to the module. Now, I have to look at the lossy HANGUPCAUSE variable and make a best guess. Not an ideal situation. We're trying to improve the ASR's we get from providers. They are low, and often they fail calls for no particular reason. They all do it, even the big ones like Verizon. Checking their responses for purpose of trying another carrier on the fly, and reporting is pretty critical. Doug. - Original Message From: Raj Jain [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, October 27, 2007 11:29:21 AM Subject: Re: [asterisk-users] Getting SIP Response Code from HANGUPCAUSE The only place where it is reasonable to customize is in the specification of the channel in the configuration file. That is where you would customize, for example, whether DTMF is inband, SIP INFO, or RFC 2833, as well as what codecs will be negotiated for that particular user/peer. But you already have the SIP_HEADER function, which is quite contradictory to what you say. This allows users who know what they are doing to examine headers directly. We use this a lot. What would be the harm in having a SIP_RESPONSE function or something alike? I'd agree that SIP response code should be accessible from the dial plan. Knowing the exact SIP response code could be critical for making call processing decisions. The conversion of SIP response codes to Q.931 codes (HANGUPCAUSE) is just too lossy. Building a truly protocol agnostic dial plan API is a worthy goal. But, I think it is somewhat of an unsolvable problem. The signaling protocols are very different and for various reasons people have always wanted access to native information elements carried in the protocol. Perhaps, a very simple solution for this problem could be to support a keyword such as TOPLINE in the SIP_HEADER function to fetch the topmost line in a SIP message. This will not only get the caller the response code for SIP response messages, but will also have the nice byproduct of making the Request-URI available if the message in question is a SIP request. - Raj ___ --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 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] Getting SIP Response Code from HANGUPCAUSE
Thanks. I am quite familiar with ngrep. I was asking how I could get the SIP response code from the dial plan. Doug. - Original Message From: Rizwan Hisham [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, October 26, 2007 6:18:50 AM Subject: Re: [asterisk-users] Getting SIP Response Code from HANGUPCAUSE I think you can use the 'ngrep' command to see the sip packets coming in using the sip listening port. I dont know the exact command though, you will have to lookit up urself. you will see the sip packets coming into ur system and in those packets you can see the response code. On 10/25/07, Douglas Garstang [EMAIL PROTECTED] wrote: I'd like to grab the SIP response code that comes back from an INVITE. The HANGUPCAUSE gives the converted ISDN cause code. Anyone know of a way to get the SIP response code instead? Doug. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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 -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] Getting SIP Response Code from HANGUPCAUSE
I'd like to grab the SIP response code that comes back from an INVITE. The HANGUPCAUSE gives the converted ISDN cause code. Anyone know of a way to get the SIP response code instead? Doug. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] AstManProxy Host Prefix?
Can the Asterisk Manager Proxy, AstManProxy, prefix the host name that output applies to, to the start of each line? If you are proxying multiple systems, how can it uniquely identify the output from each system? Thanks, Doug. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] AstManProxy Host Prefix?
Thanks, just realised that... - Original Message From: Richard Lyman [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, October 24, 2007 10:45:25 AM Subject: Re: [asterisk-users] AstManProxy Host Prefix? Douglas Garstang wrote: Can the Asterisk Manager Proxy, AstManProxy, prefix the host name that output applies to, to the start of each line? If you are proxying multiple systems, how can it uniquely identify the output from each system? Thanks, Doug. each Event block should have a Server: . appended to it. ___ --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 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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 ActionID.... Doesn't work
Is it well known that setting the ActionID when connecting to AMI has absolutely no effect? Is this fixed in Asterisk 1.4? If you add an ActionID to your Originate command for example, it looks like the only events that come back with an ActionID associated are the initial response, OriginateSuccess and OriginateFailure. That's it. No other events have an ActionID associated. This pretty much makes the AMI useless. What about all the other events? Newcallerid, Newstate, Link, Unlink and REALLY importantly the CDR events. Really... someone please tell me it's fixed in 1.4? Thanks, Doug. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] Asterisk Redundancy
- Original Message From: SIP [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Wednesday, September 26, 2007 4:31:08 AM Subject: Re: [asterisk-users] Asterisk Redundancy Per Jessen wrote: Atis Lezdins wrote: This seems nice way of sharing settings, however it wouldn't take over calls in progress. For us, currently the greatest problem is that whenever Asterisk crashes, calls are lost, and that means - lost money. Are there any ideas? Perhaps investigate/diagnose the craches? Software instability is not solved with a high-availability solution. IMHO. /Per Jessen, Zürich No. It's not. But there still exists the possibility even in a relatively stable situation that the software could crash or that hardware could fail. It's best, when planning a highly-available solution, to plan for the unforeseen and not assume you can avoid all mishaps. Let's assume, for the sake of argument, that the software will NEVER fail. Hardware still might, and that would still mean a lost call unless there's a way to switch running calls over to a new server seamlessly. Are there such ways? IP calls are especially troublesome in that regard. Don't set your goals too high. I've worked for a few companies with Asterisk now and just having an architecture that can recover within a few seconds and process new calls almost seamlessly is a workable goal. Having an architecture that can seamlessly fail over and keep calls up is kinda like the whole grail of redundancy with Asterisk. Hint... you might be able to do it with SIP reinvites... Doug. Yahoo! oneSearch: Finally, mobile search that gives answers, not web links. http://mobile.yahoo.com/mobileweb/onesearch?refer=1ONXIC___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Asterisk Redundancy
- Original Message From: Scott Moseman [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, September 26, 2007 6:07:06 AM Subject: Re: [asterisk-users] Asterisk Redundancy On 9/26/07, SIP [EMAIL PROTECTED] wrote: No. It's not. But there still exists the possibility even in a relatively stable situation that the software could crash or that hardware could fail. It's best, when planning a highly-available solution, to plan for the unforeseen and not assume you can avoid all mishaps. Let's assume, for the sake of argument, that the software will NEVER fail. Hardware still might, and that would still mean a lost call unless there's a way to switch running calls over to a new server seamlessly. Also be sure that you have a very redundant network configuration. Too often I see people spend a great deal of time and money to get redundant servers when their switches, firewalls, routers, etc are not even capable of handling a failed network element. You can achieve this at the application level. Moody friends. Drama queens. Your life? Nope! - their life, your story. Play Sims Stories at Yahoo! Games. http://sims.yahoo.com/ ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Asterisk Redundancy
It's nice to see Asterisk redundancy being discussed. A year and half ago, when I posed the question of Asterisk redundancy, I was looked at like I was from outer space. - Original Message From: Jared Smith [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, September 25, 2007 7:27:37 AM Subject: Re: [asterisk-users] Asterisk Redundancy On Tue, 2007-09-25 at 15:59 +0200, Per Jessen wrote: I haven't looked into it in any detail, but how about the standard Linux HA solution with a heartbeat monitor, a shared file-system and IP take-over? It's been my experience that this usually works fairly well for stateless protocols like HTTP, but doesn't do so well on stateful protocols like SIP and IAX, and in general is a much more difficult problem to solve. Most people tend to use some combination of SIP proxies (such as SER and OpenSER), DUNDi, shared storage, redundant databases with replication, T1/E1 failover boxes, and horizontal scaling to make Asterisk more highly-available. Of course, I haven't really gone into much detail here, but hopefully it helps answer your question. (It's also my personal experience that people who know how to build such solutions are making enough money off of selling their solution that they aren't real eager to give away all their secrets.) In reality though, you say the word cluster and it means five different things to five different people. To really be able to answer the original poster's question, we'd really have to know a lot more about his architecture and his potential points of failure. -- Jared Smith Community Relations Manager Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 Check out the hottest 2008 models today at Yahoo! Autos. http://autos.yahoo.com/new_cars.html___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Asterisk Redundancy
Nagios that's not redundancy. - Original Message From: Dave Walker [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, September 25, 2007 9:09:46 AM Subject: Re: [asterisk-users] Asterisk Redundancy On Tue, 2007-09-25 at 18:01 +0200, Philipp Kempgen wrote: Adrian Marsh wrote: so maybe it's a case of looking at Linux-HA. If I remember this correctly a normal ping is all Linux HA can do. It does not check whether Asterisk or other services are alive and respond to queries. Have you looked at: http://www.voip-info.org/wiki-Asterisk+monitoring My personal favourite would be nagios (not that I have used the SIP plugin, but do use nagios for other services) Kind Regards, Dave Walker Tonight's top picks. What will you watch tonight? Preview the hottest shows on Yahoo! TV. http://tv.yahoo.com/ ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Asterisk Redundancy
- Original Message From: Atis Lezdins [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, September 25, 2007 2:11:10 PM Subject: Re: [asterisk-users] Asterisk Redundancy On 9/25/07, Philipp Kempgen [EMAIL PROTECTED] wrote: Adrian Marsh wrote: I'm interested in how people are clustering Asterisk, if that's possible, or how you might be achieving a redundant solution. I've a single Asterisk server driving the company. Its well backed-up, and I've a cloned machine that (in theory) with a DNS change could take over operations. However I'd like to achieve something more automated if possible.. Maybe my post at http://lists.digium.com/pipermail/asterisk-users/2007-August/195339.html could provide you with some answers. Hi, This seems nice way of sharing settings, however it wouldn't take over calls in progress. For us, currently the greatest problem is that whenever Asterisk crashes, calls are lost, and that means - lost money. Are there any ideas? You might want to take Asterisk out of the media path then. If it crashes, calls will stay up, although your CDR's will be screwed. If screwed CDR's still means lost money... your still screwed! Doug. Pinpoint customers who are looking for what you sell. http://searchmarketing.yahoo.com/___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Polycom 501 Phones Rebooting
Wow. Polycom phones are STILL doing that? I haven't been involved with Polycom phones since before January, and it was a problem back then too. Jeez - Original Message From: Gregory Boehnlein [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, September 21, 2007 2:36:02 PM Subject: [asterisk-users] Polycom 501 Phones Rebooting Hello, At one of our locations, we have started to see Polycom 501s (running 1.6.7 firmware) randomly reboot. We have taken packet traces of the phones to determine if there is something odd in the Layer 2 or 3 of the network that might cause it, and have not seen anything strange. There are no errors on the ports. This appears to be affecting POE powered as well as AC powered phones. The Polycom Logs for the phones don't seem to provide any clarity. Where should I troubleshoot this next? ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 Take the Internet to Go: Yahoo!Go puts the Internet in your pocket: mail, news, photos more. http://mobile.yahoo.com/go?refer=1GNXIC___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Confused about Asterisk 1.4 RTPQOS...
I'm confused about something
In Asterisk 1.4 you can collect RTP QoS metrics at the end of a call with:
${CHANNEL(rtpqos,audio,all)}
Now, when your using the AMI to do a callout, like this...
ACTION: Originate
Async: yes
Timeout: 6
Exten: callback
Channel: SIP/1000
Variable: callid=849120
Variable: destination=SIP/1001
Variable: timeout=7
Variable: timeout_warning=6
Variable: timeout_warning_repeat=3
Callerid: 5551212
Context: default
ActionID: 849120
Priority: 1
and you have in your dialplan...
[default]
; Callback
exten = callback,1,Wait(1)
exten = callback,2,Playback(please-wait)
exten =
callback,3,Dial(${destination},40,gjoL(${timeout}:${timeout_warning}:${timeout_warning_repeat}))
exten = callback,4,Noop(*** [${CHANNEL(rtpqos,audio,all)}]);
exten = callback,5,Playtones(congestion)
exten = callback,6,Wait(5)
exten = callback,7,Hangup
exten = callback,104,Playtones(busy)
exten = callback,105,Wait(5)
exten = callback,106,Hangup
Just which leg exactly are the metrics for? Are they for the A-leg or the
B-leg? How can I get the metrics for BOTH legs of the call?...
Thanks,
Douglas.
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[asterisk-users] Dial() Command Parameter L Overflow?
I have two Asterisk Systems. One on of those, when I execute this: Dial(SIP/teleglobe-007931d0, SIP/[EMAIL PROTECTED]|60|oL(400752:6:3)) ... It causes Asterisk to immediately read out the time limit of the call (66,792 minutes), as soon as the other end answers, even though we aren't down to 60s remaining yet. Asterisk then goes into an infinite loop and reads out the time limit over and over again! On ANOTHER system, with the same Dial command, this does not happen. Both versions of Asterisk are the same. Anyone got any idea what might be causing this? Maybe one was compiled in 32 bit mode and an Integer value is overflowing? How do I check this? Douglas. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Building an RPM from Asterisk 1.4
Ok, so I'm no rpm expert, but Asterisk 1.4.11 comes with an asterisk.spec file. Running rpmbuild against it yields errors, the first one being that the 'Copyright' tag is unknown, and that I need a License tag instead. Fixed that, and... Processing files: asterisk-CVS-1 error: File not found: /tmp/asterisk/etc/asterisk error: File not found by glob: /tmp/asterisk/etc/asterisk/*.conf error: File not found by glob: /tmp/asterisk/etc/asterisk/*.adsi error: File not found: /tmp/asterisk/etc/asterisk/extensions.ael error: File not found: /tmp/asterisk/etc/rc.d/init.d/asterisk And so on. What am I missing here? Why doesn't rpm build the /tmp/asterisk directory itself? Why is it looking for files that obviously don't exist? Doug. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Building an RPM from Asterisk 1.4
I'd like to know why the spec file is even included at all then? I think we'd prefer to build our own, rather than trust someone elses build. On 9/19/07 3:22 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Wed, Sep 19, 2007 at 02:54:17PM -0700, Douglas Garstang wrote: Ok, so I'm no rpm expert, but Asterisk 1.4.11 comes with an asterisk.spec file. Running rpmbuild against it yields errors, the first one being that the 'Copyright' tag is unknown, and that I need a License tag instead. Fixed that, and... Processing files: asterisk-CVS-1 error: File not found: /tmp/asterisk/etc/asterisk error: File not found by glob: /tmp/asterisk/etc/asterisk/*.conf error: File not found by glob: /tmp/asterisk/etc/asterisk/*.adsi error: File not found: /tmp/asterisk/etc/asterisk/extensions.ael error: File not found: /tmp/asterisk/etc/rc.d/init.d/asterisk And so on. What am I missing here? Why doesn't rpm build the /tmp/asterisk directory itself? So you'll catch those errors. Why is it looking for files that obviously don't exist? That spec uses quite a few discourged methods for rpm packages. There are a number of well-maintained RPM packages of Asterisk. Use one of them or modify one of them. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Measuring Jitter in Asterisk
Oh jeez. Another GUI... -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of lenz Sent: Thursday, August 09, 2007 6:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Measuring Jitter in Asterisk I have used this freeware tool in the past: http://sineapps.com/sinestatiax.php maybe you can have a look at it as well l. In data Thu, 09 Aug 2007 02:07:49 +0200, John Todd [EMAIL PROTECTED] ha scritto: At 3:33 PM -0700 2007/8/3, Douglas Garstang wrote: At 12:31 PM -0700 2007/8/3, Douglas Garstang wrote: How can I objectively measure jitter in Asterisk on a SIP channel? I don't just want to turn the new 1.4 jitter buffer on. I want to measure jitter. Thanks, Doug. You could look at the txjitter and rxjitter values (and other values) stored in the CHANNEL() function, and those values you're looking for were previously known as RTPAUDIOQOS. Or is this not sufficient? Are txjitter and rxjitter working reliably? These calls are going to be placed from AMI and bridged together. Do you think the variables would be correctly set for each leg of the call? Doug. I think the best way to determine this would be to compare the numbers provided by CHANNEL() versus the numbers provided by something with a little more reliability, such as wireshark, in a controlled set of circumstances. Please post your results here - it would be an interesting test. JT ___ --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 -- Home of QueueMetrics - http://queuemetrics.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 ___ --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] Measuring Jitter in Asterisk
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of John Todd
Sent: Wednesday, August 08, 2007 5:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Measuring Jitter in Asterisk
At 3:33 PM -0700 2007/8/3, Douglas Garstang wrote:
At 12:31 PM -0700 2007/8/3, Douglas Garstang wrote:
How can I objectively measure jitter in Asterisk on a SIP
channel?
I don't just want to turn the new 1.4 jitter buffer on. I want to
measure jitter.
Thanks,
Doug.
You could look at the txjitter and rxjitter values (and other
values)
stored in the CHANNEL() function, and those values you're looking
for
were previously known as RTPAUDIOQOS. Or is this not sufficient?
Are txjitter and rxjitter working reliably? These calls are going to
be
placed from AMI and bridged together. Do you think the variables
would
be correctly set for each leg of the call?
Doug.
I think the best way to determine this would be to compare the
numbers provided by CHANNEL() versus the numbers provided by
something with a little more reliability, such as wireshark, in a
controlled set of circumstances.
Please post your results here - it would be an interesting test.
No comparisons yet, but I may not need to.
I'm not feeling too confident with the figures in Asterisk to begin
with.
I had an Asterisk box, bridging two channels, where the media was going
to two different ITSP's.
Upon hangup of the call, I was printing out the QoS stats available with
the CHANNEL(rtpqos) command. That seems to be what's implemented in
Asterisk 1.4.8.
h = {
Noop(local_ssrc = ${CHANNEL(rtpqos,audio,local_ssrc)});
Noop(local_lostpackets =
${CHANNEL(rtpqos,audio,local_lostpackets)});
Noop(local_jitter =
${CHANNEL(rtpqos,audio,local_jitter)});
Noop(local_count= ${CHANNEL(rtpqos,audio,local_count)});
Noop(remote_ssrc= ${CHANNEL(rtpqos,audio,remote_ssrc)});
Noop(remote_lostpackets =
${CHANNEL(rtpqos,audio,remote_lostpackets)});
Noop(remote_jitter =
${CHANNEL(rtpqos,audio,remote_jitter)});
Noop(remote_count =
${CHANNEL(rtpqos,audio,remote_count)});
Noop(rtt= ${CHANNEL(rtpqos,audio,rtt)});
}
When the call is hung up, I only see the output from this once. I'd
never thought about it before, but when you hang up a call, where two
channels are bridged, the hangup extension only gets called once for the
call, not once for each channel. Correct?
So, my output looked like this...
Connected to Asterisk 1.4.8 currently running on a1 (pid = 30914)
Verbosity is at least 3
a1*CLI show channels
Channel Location State Application(Data)
SIP/edge1-09bad778 (None) Up Bridged
Call(SIP/edge1-09baf35
SIP/edge1-09baf358 [EMAIL PROTECTED] Up
Dial(SIP/edge1/13033372500|60|
2 active channels
1 active call
== Spawn extension (Outbound, 13033372500, 2) exited non-zero on
SIP/edge1-09baf358'
-- Executing [EMAIL PROTECTED]:1] NoOp(SIP/edge1-09baf358, local_ssrc
= 891055531) in new stack
-- Executing [EMAIL PROTECTED]:2] NoOp(SIP/edge1-09baf358,
local_lostpackets = 1215) in new stack
-- Executing [EMAIL PROTECTED]:3] NoOp(SIP/edge1-09baf358, local_jitter
= 3) in new stack
-- Executing [EMAIL PROTECTED]:4] NoOp(SIP/edge1-09baf358, local_count
= 1124) in new stack
-- Executing [EMAIL PROTECTED]:5] NoOp(SIP/edge1-09baf358, remote_ssrc
= 59917798) in new stack
-- Executing [EMAIL PROTECTED]:6] NoOp(SIP/edge1-09baf358,
remote_lostpackets = 1) in new stack
-- Executing [EMAIL PROTECTED]:7] NoOp(SIP/edge1-09baf358,
remote_jitter = 0) in new stack
-- Executing [EMAIL PROTECTED]:8] NoOp(SIP/edge1-09baf358, remote_count
= 1123) in new stack
-- Executing [EMAIL PROTECTED]:9] NoOp(SIP/edge1-09baf358, rtt
= 0) in new stack
So, what do the totals represent? We're getting stats for two channels
added together it seems. Is local_jitter local jitter on both channels?
If so, it's completely useless. We need to be able to see stats for EACH
CHANNEL, otherwise they mean nothing.
Also, rtt is always 0. Man... the internet is fast today. Also,
local_lostpackets looks bogus. It's always some huge number, larger than
local_count.
Don't know if it's relevant, but this Asterisk box sent the call to an
edge router, than would sent the call onto the ITSP, and then drop out
of the RTP path. This Asterisk box was in the media, but the edge router
was not.
Doug.
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Re: [asterisk-users] Measuring Jitter in Asterisk
I also just plugged a NoOp(${CHANNEL}) in the output. It does not matter
WHICH channel hangs up the call. The ${CHANNEL} variable is always set
to the second, outgoing call leg.
What does this mean? Why is that the case?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Thursday, August 09, 2007 12:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Measuring Jitter in Asterisk
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of John Todd
Sent: Wednesday, August 08, 2007 5:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Measuring Jitter in Asterisk
At 3:33 PM -0700 2007/8/3, Douglas Garstang wrote:
At 12:31 PM -0700 2007/8/3, Douglas Garstang wrote:
How can I objectively measure jitter in Asterisk on a SIP
channel?
I don't just want to turn the new 1.4 jitter buffer on. I want to
measure jitter.
Thanks,
Doug.
You could look at the txjitter and rxjitter values (and other
values)
stored in the CHANNEL() function, and those values you're looking
for
were previously known as RTPAUDIOQOS. Or is this not sufficient?
Are txjitter and rxjitter working reliably? These calls are going to
be
placed from AMI and bridged together. Do you think the variables
would
be correctly set for each leg of the call?
Doug.
I think the best way to determine this would be to compare the
numbers provided by CHANNEL() versus the numbers provided by
something with a little more reliability, such as wireshark, in a
controlled set of circumstances.
Please post your results here - it would be an interesting test.
No comparisons yet, but I may not need to.
I'm not feeling too confident with the figures in Asterisk to begin
with.
I had an Asterisk box, bridging two channels, where the media was going
to two different ITSP's.
Upon hangup of the call, I was printing out the QoS stats available with
the CHANNEL(rtpqos) command. That seems to be what's implemented in
Asterisk 1.4.8.
h = {
Noop(local_ssrc = ${CHANNEL(rtpqos,audio,local_ssrc)});
Noop(local_lostpackets =
${CHANNEL(rtpqos,audio,local_lostpackets)});
Noop(local_jitter =
${CHANNEL(rtpqos,audio,local_jitter)});
Noop(local_count= ${CHANNEL(rtpqos,audio,local_count)});
Noop(remote_ssrc= ${CHANNEL(rtpqos,audio,remote_ssrc)});
Noop(remote_lostpackets =
${CHANNEL(rtpqos,audio,remote_lostpackets)});
Noop(remote_jitter =
${CHANNEL(rtpqos,audio,remote_jitter)});
Noop(remote_count =
${CHANNEL(rtpqos,audio,remote_count)});
Noop(rtt= ${CHANNEL(rtpqos,audio,rtt)});
}
When the call is hung up, I only see the output from this once. I'd
never thought about it before, but when you hang up a call, where two
channels are bridged, the hangup extension only gets called once for the
call, not once for each channel. Correct?
So, my output looked like this...
Connected to Asterisk 1.4.8 currently running on a1 (pid = 30914)
Verbosity is at least 3
a1*CLI show channels
Channel Location State Application(Data)
SIP/edge1-09bad778 (None) Up Bridged
Call(SIP/edge1-09baf35
SIP/edge1-09baf358 [EMAIL PROTECTED] Up
Dial(SIP/edge1/13033372500|60|
2 active channels
1 active call
== Spawn extension (Outbound, 13033372500, 2) exited non-zero on
SIP/edge1-09baf358'
-- Executing [EMAIL PROTECTED]:1] NoOp(SIP/edge1-09baf358, local_ssrc
= 891055531) in new stack
-- Executing [EMAIL PROTECTED]:2] NoOp(SIP/edge1-09baf358,
local_lostpackets = 1215) in new stack
-- Executing [EMAIL PROTECTED]:3] NoOp(SIP/edge1-09baf358, local_jitter
= 3) in new stack
-- Executing [EMAIL PROTECTED]:4] NoOp(SIP/edge1-09baf358, local_count
= 1124) in new stack
-- Executing [EMAIL PROTECTED]:5] NoOp(SIP/edge1-09baf358, remote_ssrc
= 59917798) in new stack
-- Executing [EMAIL PROTECTED]:6] NoOp(SIP/edge1-09baf358,
remote_lostpackets = 1) in new stack
-- Executing [EMAIL PROTECTED]:7] NoOp(SIP/edge1-09baf358,
remote_jitter = 0) in new stack
-- Executing [EMAIL PROTECTED]:8] NoOp(SIP/edge1-09baf358, remote_count
= 1123) in new stack
-- Executing [EMAIL PROTECTED]:9] NoOp(SIP/edge1-09baf358, rtt
= 0) in new stack
So, what do the totals represent? We're getting stats for two channels
added together it seems. Is local_jitter local jitter on both channels?
If so, it's completely useless. We need to be able to see stats for EACH
CHANNEL, otherwise they mean nothing.
Also, rtt is always 0. Man... the internet is fast today. Also,
local_lostpackets looks bogus. It's always some huge number, larger than
local_count.
Don't know if it's
