similar to: Passing DIALSTATUS between servers

I feel I'm doing something obviously wrong here and will kick myself when I see the answer!!! The scenario: SIP phone -> Asterisk1 -> IAX -> Asterisk2 -> IAX -> Asterisk3 -> PSTN So I place a call from the SIP phone. A1 picks it up and forwards it to A2 which forwards it to A3. A3 sends the call to the PSTN. I control A1 and A2, but not A3. When a call fails (for

I'm running Asterisk CVS-v1-0-12/20/04. I'm using PHP with Manager API Here is the code: #################################################################### # Make call #################################################################### $socket = fsockopen($ask_db,"5038", $errno, $errstr, $timeout); if (!$socket) { echo "$errstr ($errno)<br /\n"; } else {

Hello all, I am working on a creating some intelligent failover dial-plan logic and I'm running into something that I'd like some feedback on. Basically, it appears that if you place a call to an IAX2 peer that refuses the connection, or is unavailable, a NOANSWER dialstatus is returned. Example: -- Executing Macro("IAX2/cubix-19",

Hi. I'm dialling two numbers - one that's unobtainable, one that's busy. ${DIALSTATUS} is coming back ANSWER each time right before the channels hang up. Am using the following dialplan macro to dial out. [macro-advdial] exten => s,1,Dial(${ARG1},20,g) ; Ring the interface, 20 seconds maximum exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status

Hi All! Let me explain the problem. When using the Originate? command from the manager api, the dialstatus variable returns results? for whichever phone picks up first, and in this case it is the IAX/2? connection. It doesn't matter if Zap/G2/XXXXXXX is set as the channel,? or an extension either. What I am ultimately trying to do is get the? dialstatus of the Zap/X/XXXXXXX channel, i.e.,

Hello, I've got some SIP clients, and an IAX2 long distance provider. Ideally, when a the dialed number is busy I will hear a busy signal. Instead, I get Congestion even though * knows it's busy. Is this a bug or am I missing something? The dial plan, in basically this Dial(IAX2/user@provider/19995551234,,) Goto(failedcall-${DIALSTATUS}) failedcall-CONGESTION plays congestion

I'm trying to write a dialplan that will allow me to "stress" test it. I want to be able to dial an extension, or pretend that the extension is busy or out of order (so that I can see what to do) given the dialplan snippet: [outbound] exten => _X.,1,NoOp(${TEST}) exten => _X.,n,Dial(SIP/${EXTEN}) exten => Busy,1,Busy(2) exten => Busy,n,Hangup() exten =>

What am I doing wrong? Goif dialstatus: busy CONGESTION not working. exten => _7NXXXXXX,1,Dial(SIP/7780${EXTEN:1}@pstn-5665,60,tr) exten => _7NXXXXXX,n,GotoIf($[$["${DIALSTATUS}" = "BUSY"] | $["${DIALSTATUS}" = "CONGESTION"]]?line2) exten => _7NXXXXXX,n(line2),Dial(SIP/9780${EXTEN:1}@pstn-1270,60,tr) exten => _7NXXXXXX,n,Hangup() When I try to

Just throwing this out here, hopefully someone can tell me why. *CLI> show version Asterisk CVS-HEAD-11/17/04-10:16:38 built by root@wanderer on a i686 running Linux Zap/g1 is pri_cpe to Bell Canada 5551234 is a normal POTS line I have busied out (handset offhook) exten => 1234,1,Dial(Zap/g1/5551234,,g) exten => 1234,n,NoOp(HANGUPCAUSE is ${HANGUPCAUSE} and DIALSTATUS is

I got two phones on an ATA-186 (601, 602) and two phones on the TDM22B (603, 604). I have two lines on the TDM22B. I cannot figure out some of the problems: 1. 601 dials via ZAP/3-1 to local phone number at PSTN: ringing pickup on PSTN (empty) still ringing in the phone set 601 2. call from PSTN back: 601 picks up ... everything works !!! No caller id shows up 3. For testing I have only one

I have two internal analogue extensions off a TDM400P. If the first is busy, I'd like to ring the second. So: [incoming] exten =>s,1,Answer() exten =>s,n,Dial(${mainline},60) exten =>s,n,ExecIf($["${DIALSTATUS}" = "BUSY"]?Dial(${secondline},30)) But it doesn't work because * first tries Call Waiting on the main line. Here I dial out: -- Starting

I am experiencing a very wired problem. Some of my cdr are lost. I use logging cdr to csv, mysql and odbc. But some of them are lost. They miss in csv mysql and odbc, so i'm pretty sure it is related to asterisk functioning. I am running asterisk 1.0.7; this is simple configuration file: extensions.conf [general] static=yes writeprotect=no [macro-gw-voipjet] exten =>

Hi. I'm dialling two numbers - one that's unobtainable, one that's busy. ${DIALSTATUS} is coming back ANSWER each time right before the channels hang up. Am using the following dialplan macro to dial out. [macro-advdial] exten => s,1,Dial(${ARG1},20,g) ; Ring the interface, 20 seconds maximum exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status

I have recently built my first asterisk system and am very impressed with its capabilities. However, I have run into one problem that hopefully someone can help me with. I am trying to use the DIALSTATUS function to route incoming calls to the appropriate Voice Mail (busy or unavailable) or to an Unavailable Number recording if the number is not assigned. However, I find that DIALSTATUS

On machine A I have something like the following in extensions.conf: [iax-extensions] exten => _9.,1,Dial(IAX2/machineB/${EXTEN:1}@mycontext) exten => _9.,2,NoOp(DIALSTATUS=${DIALSTATUS}) exten => _9.,3,Hangup On machineB I have something like this: [mycontext] exten => 2002,1,Dial(SIP/2002,60) exten => 2002,2,NoOp(DIALSTATUS=${DIALSTATUS}) exten => 2002,3,Hangup If I use a

hi, i have a problem, and i am completely stuck with it, i hope someone can point out where is my config wrong. I have three server, connect together with IAX trunking. The server are at romania (10.0.4.23, V1.4.22), hungary (10.0.1.23, V1.4.20) and serbia (10.0.3.4, V1.4.22). I have a hardphone (6251) connected to the romanian server, i dial a hungarian telephone number, the call goes to the

I have a list of VSPs that I use. Some are not able to terminate to different locations. It appears they are returning this error message: Sep 13 00:01:43 WARNING[22093]: chan_iax2.c:6835 socket_read: Call rejected by x.x.x.x: No such context/extension I would like to find out what the dialstatus is on this so I can try a different VSP that is able to terminate the call. Right now I have this

> On Jan 2, 2024, at 23:17, thelma at sys-concept.com wrote: > > On 1/2/24 15:13, asterisk at phreaknet.org wrote: >>> On 1/2/2024 4:03 PM, thelma at sys-concept.com wrote: >>> I'm using asterisk-16.30.1 >>> >>> When I try to call another asterisk server over IAX I get a busy signal, >>> >>> chan_iax2.c:4739 __auto_congest:

I have the following pair of macros defined to handle outbound calls from *. Rather than specifying full dialstrings in the main body of extensions.conf, outbound dial commands are made using a macro call as follows: Macro (outbound,number_to_dial,callerid_to_present,gateway1,gateway2,gateway3,gate way4) The final gateway defined is nearly always a fallback to PSTN if none of the IAX or SIP

I can dial from 601 to a public number. The public number rings. I pickup and hear nothing, while on 601 it keeps ringing. (BTW, is it right to say "ringing" on the active phone?) The *CLI> doesn't show me anything useful: Executing Macro("SIP/601-8238", "dial-pstn|88097880|") in new stack Executing SetGlobalVar("SIP/601-8238",