similar to: PHP/AGI/SetVar Issue

I am running 1.2.34 (also tried on 1.2.32) and whenever I launch an AGI, I get the "avoided deadlock" message below. *CLI> == Spawn extension (CONTEXT3, 6080, 8) exited non-zero on 'SIP/3211-1-081c40a8' -- Executing NoOp("SIP/3211-1-081c40a8", "") in new stack -- Executing AGI("SIP/3211-1-081c40a8", "diallocal.agi") in new

Hi all, is there an easy way to get a 'copy' of a type living in another context into the local context? Background: when calling a function residing in a different module (context2) from a module (context1), we first need to introduce a function declaration of the function with empty body. However, in order to do so, we need the function type. pFuncInContext2->getType gives us the

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 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 =>

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

Would someone be able to give an example of a working use of setvar from chan_dahdi.conf? I am trying to create a custom variable like I use in sip.conf but I have been completely unsuccessful getting any variable set using setvar to appear for a DAHDI channel. I am running 1.8.11-cert8 and am using the newer format (but I have tried using the older [channels] format). Here is an example:

Does anyone know of a way to specify what extension is dialed when 0 is pressed in the voicemail system. I have a situation where there is more than one secretary and they want the 0 to redirect to the appropriate secretary for the two groups of people. So an example would be: 555-1234 -> voicemail -> Secretary 1 555-1235 -> voicemail -> Secretary 2 Any help would be greatly

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

Hi, I had the following odd behaviour in Asterisk 1.2 - We are migrating to 1.6, and I will re-test ASAP, though it is quite hard to replicate, but I am curious to know whether it is a known IAX issue in 1.2. We had 2 users in iax.conf: [user1] username=user1 secret=secret1 context=context1 host=iax.hostname.com [user2] username=user2 secret= context=context2 host=dynamic deny=0.0.0.0/0.0.0.0

Hello users, Recently i have installed the free version of FaxForAsterisk and trying to work with it by sending a fax on T38. My version information is as follows i)Asterisk 1.6.0.20 ii)res_fax-1.6.0.14_1.1.6-x86_32 iii)res_fax_digium-1.6.0.14_1.1.6-i686_32 sip.conf [general] t38pt_udptl=yes extensions.conf [default] exten => _XXXXXXXXXX,1,NoOp(Fax Incoming Call) exten =>

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

Hey I need another pair of eyes on this! I would like to add phones numbers to the blacklist from any handset so I did this: exten => _*66XXXXXXXXXX,1,StripMSD,3 exten => _XXXXXXXXXX,2,DBPut,blacklist/BYEXTENSION/1 exten => _XXXXXXXXXX,3,Hangup However what I get in the database is: /blacklist/BYEXTENSION : 1 And BYEXTENSION is not replaced with the actual number

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

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

For years we've had the following simple context for outgoing calls: [outtrunk] ; match any NANP, and strip leading 1 off exten => _1XXXXXXXXXX,1,StripMSD,1 ; dial outbound on trunk group 1 exten => _XXXXXXXXXX,2,Dial,Zap/g1/${EXTEN} But when I upgraded on Friday to the latest CVSHEAD, this no longer works. If I send 13115552368 to this context, I get a message like pbx.c: Channel

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

Have, i want to create a sip extension to a context in my dialplan. how i can do that?