similar to: AGI gosub return value

Suppose I have a subroutine (called by Gosub) S that's called from a macro M and there's a goto to an illegal extension in S. That does go to 'i' in S but seems to pop the macro stack so that when there's a later fallthrough in M, the calls hangs up rather than returning to the caller of M. Is this a bug or a feature?

Hi - I have a Macro that contains a GoTo. The documentation indicates: If you GoTo out of the Macro context, the Macro will terminate and control will return at the location refered to by the Goto. I thought I might convert the Macro to a GoSub routine, but the documentation doesn't mention what happens if you GoTo out. It does however mention that the return address gets pushed onto the

How does one go about accessing gosub arguments from Asterisk in extensions.lua? For example, I have the following in extensions.conf: exten => 1000,1,Wait(1) exten => 1000,n,Gosub(functions,mytest,1("123")) exten => 1000,n,Hangup And then the following in extensions.lua: extensions = { functions = { ["mytest"] = function()

Hi, I can't figure out what syntax to use with the Dial() "M" parameter for the AEL parser to interpret properly. Creating an AEL macro named "macro-screen()" partly works as a hack, but it must not turn into a gosub properly, so I get warnings about the "return;". Dial(...,tgM(&screen)) with the ael macro named "screen" does not work

I'm using QSIG between Asterisk and an NEC SV8300. Whenever I make a call from the SV8300, I see: [Oct 4 21:02:49] ERROR[5729]: chan_dahdi.c:12226 dahdi_pri_error: !! < Unknown IE 50 (cs5, len = 3) I see an IE 50 in the Q.932 specification, so I don't understand why this error is occuring.

I'm trying to build it and run into all sorts of problems. First, "make testexpr2" doesn't work at top level, nor in the "main" subdirectory. If I manually try the commands for it in main/Makefile, it doesn't have a "main" and calls "ast_log". If use -DSTANDALONE2 instead, those go away, but then: ast_expr2f.o: In function

Is there an explanation other than the one in the application documentation of exactly what this is for and when you'd want to use it and when you wouldn't? I find the explanation in the documentation a little confusing.

I did a usual "svn update", "./configure" and "make" and got [CC] chan_oss.c -> chan_oss.o gcc: @SDL_INCLUDE@: No such file or directory I don't see any changes to chan_oss recently, so don't understand this. What could be going on?

I'm looking at what would be involved in converting from MeetMe to ConfBridge and there seems to be a lot of missing administrative things, but I hope I'm just missing it. We all know about the missing realtime linkage. That's a major nuisance, but can be worked around. More serious is that the CLI command to display users in a ConfBridge don't show the caller ID information, so

Hello, I've started to play with PJSIP and got stuck at the following problem. I need to retrieve SIP Call-ID associated with PJSIP channel. For inbound channel I can use ${PJSIP_HEADER(read,Call-ID)}, but that doesn't work for outbound channel even in pre-dial or hangup handler. Whatever I do PJSIP_HEADER seem to be unable to read headers for outbound channel. Here's what I do:

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Here's one that may be of interest to any upgraders. If you rely on the behavior of gosub you may want to make note of this change. I have an incoming call context: exten => _XXXX,n,GoSub(incoming,${EXTEN},1(${EXTEN})); that is supposed to gosub into the incoming extension at priority 1. Versions before 1.6.0.6 would drop into the

Hi, Is there some more thorough documentation of this change that has happened in 1.6? The upgrade.txt and changes.txt files mention it, but I have already seen details of this change that do not appear to be documented except in conversations on the mailing list... 1) It appears that it is no longer legal to have: [macro-contaxtA] ...stuff... [contextA] ...stuff... Is this true? Or have I

Hi folks, Is there anywhere I can possibly get a model of the exam itself, maybe possible scenarios for the prac, etc? To people who have done the exam....any helpful hints ? Thanks,

Problem as follows: [default] exten => 777,1,Gosub(sub,1,1) exten => 777,n,Hangup() exten => h,1,NoOp(hung up in 'default' context) [sub] exten => 1,1,NoOp(in sub) exten => 1,n,Playback(tt-monkeys) exten => 1,n,Return() exten => h,1,NoOp(hung up in 'sub' context) This works fine if the caller listens to all the 'tt-monkeys' and let's the system

Good afternoon list. I'm trying to use ${AGISTATUS} after the execution of my script in PHP Agi. But after running the script, it just returns me 0 (true). Thus: -- <SIP/213-00000019>AGI Script check.agi completed, returning 0 I tried putting the lines "return false;" or "return 1;" but did not change anything. Does anyone have a clue? Thanks, Rodrigo Lang.

Does the application PGSQL has been removed from Asterisk? Couldn't find it on Asterisk source and addons. Atenciosamente, Vin?cius Fontes Gerente de Seguran?a da Informa??o Canall Tecnologia em Comunica??es Passo Fundo - RS - Brasil +55 54 2104-7000 Information Security Manager Canall Tecnologia em Comunica??es Passo Fundo - RS - Brazil +55 54 2104-7000

Hi Friends, i have installed and configure asterisk-1.8.0. When i have tried asterisk start get below errors and not able to start asterisk. *FD 32767 exceeds the maximum size of ast_fdset!* Thanks in advance. -- Best Regards, Rajnikant Vanza -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am using asterisk 1.6.1.6 and have been setting up a system to use a Postgresql database as the realtime DB via the ODBC route. I have got extensions and voicemail working but am having trouble with SIP The problem seems to be with using a schema. If I put the table "sip" in the schema "foo" then I add this entry to extconfig.conf sippeers => odbc,psqldb,foo.sip Restart

Hi all, Here is the script i want to make - Caller call to a number to record a message - Asterisk answer and start recording message as following + User press * to start recording + Record is finished if: + User press # + OR message duration reach 60 second + Hangup How do you counting down 60s, and how to detect # (i make a test using Read() but it cant read #) Thanks in advance

Hi guys, Having issue with getting CDR to write to MS-SQL via ODBC. > cdr_odbc: Connected to freetds-connector > cdr_odbc: Error in PREPARE -1 > cdr_odbc: Query FAILED Call not logged! == Spawn extension (cisco, ##########, 2) exited non-zero on 'IAX2/astYYYY-507 Isql test: [xxx at YYYY asterisk]# isql freetds-connector XXXXXXX YYYYYYYYY