similar to: Spawn extension.....exited non-zero

in sip.conf [general] port = 5060 ; The TCP/IP port for SIP communiations bindaddr = 0.0.0.0 ; Address to bind to. 0.0.0.0 all addresses on server. context=other ; Default for incoming calls disallow=all allow=ulaw allow=gsm in extensions.conf [general] static=yes ; These two lines prevent the command-line interface writeprotect=yes ; from overwriting the config file. Leave them here

We upgraded our asterisk server to 1.2.18 last night to pick up the security update. Since then, any calls coming in on IAX2 links get dropped if they try to enter a MeetMe conference room. The log shows this: Apr 26 08:33:16 NOTICE[27362]: chan_iax2.c:3167 iax2_read: I should never be called! Hanging up. I've temporarily worked around it by switching our inbound provider to use SIP

All, After installing asterisk on Linux, I run "asterisk -vvvc". But I got the following warning message: chan_oss.so] => (OSS Console Channel Driver) May 26 00:37:58 WARNING[-1084845952]: chan_oss.c:980 load_module: XXX I don't work right with non-full duplex sound cards XXX == Registered channel type 'Console' (OSS Console Channel Driver) == Parsing

Hi i generate a call from the dialplan in this mode: exten => 1002,1,Answer() exten => 1002,2,Dial(SIP/user at host) the call is generated, but after some seconds it is interrupted, here the asterisk log: *CLI> -- Executing Answer("SIP/host1-0819d0d0", "") in new stack -- Executing Dial("SIP/host1-0819d0d0", "SIP/caller at host") in new

Hi All, I'm having a problem after reinstalling the operating system. Festival works fine for SIP, but when IAX users are calling the same extension they don't hear the festival and I see the next message on console: NOTICE[3996]: chan_iax2.c:2995 iax2_read: I should never be called! I googled and couldn't find a solution, if somebody can help.... neobase*CLI>

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Hi, I'm trying to call a SIP server while providing the SIP server username/password in dial string but it's not working ... Dial(SIP/gs102:test at 192.168.2.81); User on sip server (192.168.2.81): [gs102] disallow=all allow=ulaw allow=alaw type=friend username=gs102 secret=test host=dynamic dtmfmode=inband defaultip=192.168.2.1 qualify=1000 mailbox=102 context=context-gs102

Does anyone know how to get CID working in AEL2 ? In extensions.conf you can do: exten => 111/666,1,PlayBack(demo-congrats) exten => 111/666,2,Hangup() exten => 111,1,PlayBack(demo-moreinfo) exten => 111,2,Hangup() and if callerid 666 dialed 111, they would get demo-congrats, everyone else gets demo-moreinfo. In AEL: 111 => { Playback(demo-moreinfo);

Hi all, Couldn't find a place to search the list archives... I'm having issues in getting any sound using a fresh asterisk install and a SJPhone to connect to it. I went by the instructions pointed at the "10 minute guide", located here: http://www.voip-info.org/tiki-index.php?page=Asterisk+quickstart. I installed it on a Slackware 10.1, by using no more than "make

Hi all, Couldn't find a place to search the list archives... I'm having issues in getting any sound using a fresh asterisk install and a SJPhone to connect to it. I went by the instructions pointed at the "10 minute guide", located here: http://www.voip-info.org/tiki-index.php?page=Asterisk+quickstart. I installed it on a Slackware 10.1, by using no more than "make

Hi, I have cvs updated all my modules (zapata, libpri, zaptel and asterisk). I have also read in the archives & seems that no-one has run into this problem. What I'm trying to do is simple. Just make and outbound call using the /var/spool/asterisk/outgoing directory. I copied /usr/src/asterisk/sample.call and only changed the context & extension. I configured my Zap1 to the same

Hi! I just installed Asterisk 0.4.0 with all the default options, and the configuration samples it has. When I try to dial from an h323 client (gnomemeeting) I get this message on the messages file: Jun 19 11:48:45 WARNING[15375]: File file.c, Line 410 (ast_openstream): File demo-congrats does not exist in any format Jun 19 11:48:45 WARNING[15375]: File file.c, Line 553 (ast_streamfile):

actually, i do have a workaround which bypasses the exec command entirely: system("asterisk -r -x 'add extension s,3,Playback(demo-congrats) into local'"); but it's ugly. seems like it should be possible to do this with exec. .t ---------- Forwarded message ---------- Date: Sun, 23 Nov 2003 21:17:50 -0500 (EST) From: tad <tad@media.mit.edu> To:

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I try to get callers forwarded to by mobile phone when they dial a certain digit. In my extensions.conf I have defined the following: [279xxxx] exten => s,1,SetLanguage(de) exten => s,2,Wait,5 exten => s,3,BackGround(demo-congrats) exten => s,4,Goto(boksa,#,1) exten => 3,1,VoiceMail,u1 exten => 4,1,VoicemailMain exten =>

as i said, right now i'm just getting my feet wet. but, i will be needing to build dialplans on the fly. 'add extension' seems like the right call to make. .t > What is the goal of this? It doesn't make much sense to me. Care to > share some insite into what your goal is? > > bkw > > On Sun, 23 Nov 2003, tad wrote: > > > actually, i do have a

hi folks. (apologies in advance if this is a particularly stupid question) just getting my feet wet with asterisk / agi, and am a little stuck using EXEC. it works fine for applicaitons that take simple arguments, but chokes on applications that require multiple words as arguments. for example, this works fine: EXEC Playback(demo-congrats) but this doesn't: EXEC add extension

Hi list, I have been looking around for a while now, and cant seem to get to the bottom of my problem. My setup is that I have a separate SIP server that servers my SIP subscribers, and I want to use Asterisk purely for voicemail for now. So I set up a common SIP extension at my SIP server, and made Asterisk register it, so that normal users can forward calls to that common extension, and

My first Asterisk install: Debian sarge with the 2.6 kernel, and two X-Lite soft-phones. I followed the online how-to documents and was calling between the two soft-phones and calling the demo system with no problems and had full audio. I then went on to configure the TDM400P's two FXS modules. I got into that a ways and was having some success, but no dial-tone when I was off the

Hi, the first thing I did with Asterisk is listening to `demo-congrats' by Xlite on the same machine. This works perfectly. The config files are those shipped with the package. Now I want to listen to it over ISDN/Capi but I don't succeed. My `capi.conf' is like show in many tutorial on the web. In `extensions.conf' I just added the following lines: [capi-in] exten =>