similar to: AGI EXEC trouble

Hi, what happens if the dialplan contains something like exten => s,1,AGI(agi://10.0.0.1) exten => s,2,Dial(SIP/phone1|20|tr) etc. - if 10.0.0.1 isn't reachable or doesn't react on the connection? In my test cases, I always got a hangup and no further processing of the dialplan. Any hints? ( the call mustn't go into Nirvana if the AGI server isn't available!) Thanks

Hi, I still have some trouble with the AGI interface: - I can use EXEC now, but it never gives me the error returned by the executed application, if an error occurs - I can use ANSWER, but I have to put something else behind "ANSWER". If I say "ANSWER", I get "510 Invalid or unknown command". If I say "ANSWER ''" or "ANSWER ." or

Hi, I have some trouble with the Monitor() application. I start and stop it via the management interface, giving no special parameters except the channel name. What happens is: - if I specify WAV as the format, the resulting files are exactly 44 bytes big and contain nothing at all - if I specify GSM as the format, the resulting files are of size 0. I did not request mixing of the files or

I can't build it, errors: chan_zap.c:61: #error "You need newer libpri" chan_zap.c: In function `zt_call': chan_zap.c:1806: warning: implicit declaration of function `pri_sr_set_redirecting' chan_zap.c: In function `pri_dchannel': chan_zap.c:7776: structure has no member named `redirectingreason' chan_zap.c:7778: structure has no member named `redirectingreason'

Hi all I've been trying to make * work with IAXtel to no avail, all seems ok in the config but am not getting anywhere This is what I'm getting from console (user/pass/dest # changed for obvious reasons): DEBUG[1133735216]: File chan_sip.c, Line 3841 (check_user): Setting NAT on RTP to 0 DEBUG[1133735216]: File chan_sip.c, Line 4891 (handle_request): Check for res for phone1

Hi everyone, I have a question concerning DNS SRV lookups. The situation is like this: - one central Asterisk server - many domains with SRV records, let's say we have bar.com and doe.com Now the question is: if the SRV lookup is done for foo@bar.com the call is mapped to foo@myasterisk.mydomain.net. Is that correct? If so, I have a problem: if somebody calls foo@bar.com, Asterisk

Hey, whenever I try to load, I get these errors Jan 15 16:37:24 ERROR[7573]: chan_iax2.c:7486 load_module: Unable to bind to 0.0.0.0 port 4569: Address already in use Jan 15 16:37:24 WARNING[7573]: loader.c:345 ast_load_resource: chan_iax2.so: load_module failed, returning -1 == Manager unregistered action IAXpeers == Unregistered channel type 'IAX2' Jan 15 16:37:24 WARNING[7573]:

Hello. I have just arrived to Asterisk. I would like to know if Asterisk can perform some functionalities I am looking for. I want to allow voip over sip to some users. All of them must have their own user name and password to login to Asterisk so only allowed users can login. All calls started by users have to be redirected to one account at our voip provider. I think those functionalities can

Here is an example of a couple of macros that help me where I have a SOHO with a home phone line and a work phone line. If I pick up line 2 my work line I would prefer the call I make to go out my office phone line same with if I pick up line 1 my home phone line I would prefer it go out my home line but want it to roll if needed. So with this little macro it is possible for that to happen.

So I have read and read and read... google is my friend and the wiki is by brother... However, I am still unclear on what the preferred method of using the pound sign is. If the Pound sign is set aside for Transfer.. Then when I make an outbound call to my bank I can not "Enter my PIN followed by Pound" Likewise if I turn off the ability to transfer when initiating a call, my bank pin

Hi all, Is it possible to dial an OUTSIDE SIP address while inside AGI application? For example, within extension context, I could use [from-sip] exten => 7723,1,Dial(SIP/897224@fwd) and this works whereas when I'm inside agi app, $AGI->exec('Dial',"SIP/897224@fwd") and this DOESN'T work. There some errors about invalid argument. If I were to do

I have several DIDs (working well) with LiveVoip and I just signed up for some outbound minutes. Unfortunately they did not send connection instructions. I tried: exten => _1NXXNXXXXXX,2,Dial(IAX2/userid:password@217.160.244.186/${EXTEN}|60|s) but I get the error Feb 19 15:14:09 WARNING[21453]: chan_iax2.c:5546 socket_read: Call rejected by 217.160.244.186: No authority found --

Hello, I'm traying to link 2 * servers using SIP and the following errors was show: "SIP/AsteriskA:AsteriskA@10.0.0.121/100") in new stack Dec 13 22:46:57 WARNING[8767]: chan_sip.c:1398 create_addr: No such host: 10.0.0.121/100 Dec 13 22:46:57 NOTICE[8767]: app_dial.c:759 dial_exec: Unable to create channel of type 'SIP' == Everyone is busy/congested at this time Dec 13

I recently purchased a TDM11B so I could hopefully hook flash the FXO from either the FXS (on the TDM11B) or a SIP device. From the FXS, I've tried hitting # then transferring to an extension that flashes the line then dials the FXS again (3020). This seems to send me to a busy signal and the console tells me no such host of 3020 (the number I'm on). The call on call waiting gets sent

I have 4 Polycom phones with multiple line keys so multiple incoming calls work fine The way I would like the incoming call flow to work is as follows: 1) 2 groups consisting of 2 phones each 2) Incoming call rings the first group, if no answer, the 2nd group is rung 3) However if the first 2 are on a call or busy, it will immediately ring the 2nd group 4) If one

Hmm... Are my questions too simple or too difficult? Anyway, I try it once more... -------------- next part -------------- An embedded message was scrubbed... From: palme at uni-wuppertal.de (Hubert Palme) Subject: R-beta: Printing Date: Tue, 10 Mar 1998 16:58:35 +0100 (MET) Size: 2755 Url: https://stat.ethz.ch/pipermail/r-help/attachments/19980313/12e3bfea/attachment.mht -------------- next

When trying grep(), I get the following: > grep("[a-z]", letters) Error in grep(pattern, x, ignore.case, extended, value) : POSIX regular expressions not available I can't find any pointer in the installation kit. How can I configure R to make it "avaliable"? I run R in an Irix 5.3 system. Thanks in advance! --

I forgot: "make docs" for LaTeX doesn't collect any information about the functions. So in Man.dvi the appendix C is empty. A Bug?? -- ====================================================================== Hubert Palme Bergische Universitaet-Gesamthochschule Wuppertal Computing Center D-42097

Hello, after the line cat("%",as.character(as.name(match.call())), "\n%\n", file=fi, append=T) R complains about Error in as.name(x) : character argument required Is it sufficient to delete as.name()? After doing that, R complains Error: comparison is possible only for vector types Don't know which function call. Can you give me a hint how to debug it? There

I'm trying to fix a problem with the GrandStream Budgetone 102. I've been reading the source code, mailing lists and other resources. Here's the scenario and the approach I have been pursuing. I'm having some problems with the AGI calls and I hope someone can give me some clarification. PSTN <---> T1,PRI * <---> Grandstream BT 102 (12)