similar to: 1.2 AGI Deadlock

Is there a limitation to the number of variables you can set from a PHP agi script? I have a simple example and I can't get it to let me set more than 1. I am pretty sure I am just missing something, but I've searched all over an can't find the answer. Here is the extensions.conf part: exten => _XXXXXXXXXX,1,AGI,diallocal.agi exten => _XXXXXXXXXX,n,NoOp(${ISLOCALCONTEXT})

Hi all In my application I''ve some named routes defined this way... map.label_context1 '':context1/label'', :controller => ''mycontroller'' map.label_context2 '':context1/:context2/label'', :controller => ''mycontroller'' map.label_context3 '':context1/:context2/:context3/label'', :controller =>

Hi, I have a little situation with my dialplan, and I am wondering if what I want is even possible. Here it is: I have three contexts, context1 includes contexts2, and context2 includes context3. In other words, in context1 all extensions of context2 and context3 are valid (and actually working, so that's good). I am using those context for the sake of code clarity and reuse, and for

Hi, BRIEF RESUME: Is there any other way to obtain the same result but being easier to configure?? Thanks! EXTENDED RESUME: i've configured a, rather difficult, macro that even for me without being documented is difficult. I ask for the help of the experts to know if the functionality it apports can be achieved better in another way. What i'm trying is to enable call a channel (e.g.

when i wnat to working with realtime and mysql for any context i have to insert (switch => Realtiem/context at extensions) statment into extensions.conf for example if i want to have 10 context, i have to insert these lines into extension.conf : [context1] switch => Realtiem/context1 at extensions [context2] switch => Realtiem/context2 at extensions [context3] switch =>

Hello, Is it possible to use the switch => statement in extensions.conf (http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions) to point to a database and in the database use the include-statement ? In extconfig.conf I would have : extensions => mysql,asterisk,extensions_table In extensions.conf I would then have : [includecontext] switch => Realtime/includecontext at

I am running Asterisk 1.8.10.1. I am trying to set up some routing in my dial plans and having some issues (the issue being that I don't quite understand all of the syntax and patterns that can be used: Examples: DID1 = 6140000000 DID2 = 6140000001 CNAME1 = 6149999999 CNAME2 = 6149999998 CNAME3 = 6149999997 context1 context2 context3 I have looked at several examples (patterns) and I

I have several of incoming numbers on IAX from voiptalk and magrathea but have a problem with IAX.conf. If I follow the example from voiptalk [VoIPTalk Incoming Number] type=friend username=VoIPTalk Incoming Number context=[XXXXXXXX] and make incoming entries in IAX.conf for the numbers like below with a different entry for each number pointing to a different context, incoming numbers always

I am running 1.4.10.1. I have a macro that is called from default for a certain extension (both below). I added NoCDR to s to try and stop extra CDR records, but I am still getting them. Any idea how to stop them? extensions.conf: [macro-STDEXT] exten =s,1,NoCDR() exten =s,2,Dial(${ARG1},30,Tt) exten =s,3,Goto(s-${DIALSTATUS},1) exten =s-NOANSWER,1,Voicemail(${ARG2}|u) exten

What are these? Nov 22 09:35:23 WARNING[7127]: channel.c:787 channel_find_locked: Avoided deadlock for '0xf6c06778', 10 retries! Nov 22 09:35:24 WARNING[7127]: channel.c:787 channel_find_locked: Avoided deadlock for '0xf6c06778', 10 retries! Nov 22 09:35:24 WARNING[7127]: channel.c:787 channel_find_locked: Avoided deadlock for '0xf6c06778', 10 retries! Nov 22 09:35:25

My Asterisk is Asterisk 1.2.30.2 currently running on viciserver the problem is : Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided deadlock for '0x861f6d8', 9 retries! Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided deadlock for '0x85a6420', 9 retries! Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided

Is there any way to see the called number when a call gets redirected to the 'a' extension from voicemail? Say x123 calls x456 and it rolls to voicemail. x123 hits * and gets dumped into the 'a' extension in the original context. I need some logic in 'a' to do a database lookup based on the original called number (x456). Any ideas? When I do a test, it appears

Hello! I clear remarks in Makefile: DEBUG_THREADS = -DDEBUG_THREADS -DDETECT_DEADLOCKS But same things in CLI: Aug 30 18:16:31 WARNING[22292]: channel.c:780 channel_find_locked: Avoided initial deadlock for '0x82f2fe0', 9 retries! -- Zap/32-1 is proceeding passing it to Zap/31-1 -- Zap/32-1 is ringing -- Accepting call from '2177' to '7141278' on channel

Package: logcheck Version: 1.2.32 Severity: minor "Ghandi" should be "Gandhi" in README.how.to.interpret, assuming that you mean the Indian freedom fighter M.K. Gandhi a.k.a. Mahatma Gandhi. -- System Information: Debian Release: 3.1 APT prefers unstable APT policy: (500, 'unstable') Architecture: i386 (i686) Kernel: Linux 2.6.7 Locale: LANG=C, LC_CTYPE=C

I have 20-30 GXP2000's connected to * over a T1 line. Neither end is NAT'd and there is plenty of bandwidth available over the line. The GXP's are 1.1.5.15, which is the latest. I have a problem where the phones keep dropping off of * and I get a "failed to register" message in the log of *. Sometimes they eventually connect and sometimes, I have to reboot them to

In asterisk 1.2 branch SVN 51363 zaptel svn 1980 libpri svn 393 addons svn 332 My equipment is a Via EPIA minit-itx CN10000 1.2ghz, 1gb ram and a tdm400p (4fxo). A call comes from zap, a SIP ulaw receives the call, talks for a while and when SIP users tries to park the call, then dozens of... WARNING[3853]: channel.c:781 channel_find_locked: Avoided initial deadlock for '0x91bb840', 10

Hi i have 20 people calling agents calling when ever they calling i get this below error May 30 00:46:57 WARNING[2840]: channel.c:785 channel_find_locked: Avoided deadlock for '0x8b2f50', 10 retries! and the voice go choppy, and voice breakages iam using Latest SVN, any suggestion to come over this problem ram -------------- next part -------------- An HTML attachment was scrubbed...

We've had a few people complain that the "beep" before leaving a voicemail is not loud enough and too short. Does anybody have a recorded beep that they can share, that is a little louder and a little longer? We've had this box in production for 2+ years, so I hate to mess with the gain on the PRI or anything like that because everything else works fine. I know nothing

Is anybody using 1.2 (or 1.2.1) in a production network using Realtime (voicemail, sip or extensions) with 100+ SIP phones? If so, what are your experiences? We've been running 1.0.3 for about a year and it's been rock-solid. We'd like to upgrade to Realtime and 1.2, but I'm afraid of killing our stability. Obviously, we'd do it in stages (upgrade to 1.2, then realtime

I am using realtime on two boxes, one running 1.4.10.1 and one running 1.4.11. Everything works fine except for when I make a database change, such as a phones password. I change the DB, I prune the peer, I see it is gone and then I see it show up again in "sip show peer xxxx", but everything is not being updated. The phone will not register even though the DB and the phone have