search for: callingani2

...llerID changed: unknown callerID changed: unknown callerID changed: unknown callerID changed: unknown callerID changed: unknown callerID changed: unknown callerID changed: unknown callerID changed: unknown AGI Environment Dump: -- accountcode = -- callerid = unknown -- calleridname = unknown -- callingani2 = 0 -- callingpres = 0 -- callingtns = 0 -- callington = 0 -- channel = IAX2/u32218094-1 -- context = default -- dnid = unknown -- enhanced = 0.0 -- extension = 8365 -- language = en -- priority = 3 -- rdnis = unknown -- request = agi-VDADtransfer.agi -- type = IAX2 -- uniqueid = 1139...

With the help of one of the providers we terminate on, I've found the source of the problem of getting busy even when the called isn't really busy in the absence of ANI codes in sip headers generated by asterisk. If I put a NoOp(${CALLINGANI2}) in the dialplan before the dial I can see it holds the value '0', but seems that value won't find the way to the sip header. Is this an error for asterisk to not put the code or a misconfiguration of the remote switches to drop calls without it ? (Have I to open a bug or to reques...

...f("Zap/1-1", "0?5:4") in new stack -- Goto (macro-dial,s,4) -- Executing AGI("Zap/1-1", "dialparties.agi") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi -- dialparties.agi: priority = 4 -- dialparties.agi: callingani2 = 0 -- dialparties.agi: accountcode = -- dialparties.agi: channel = Zap/1-1 -- dialparties.agi: callerid = unknown -- dialparties.agi: context = macro-dial -- dialparties.agi: callington = 0 -- dialparties.agi: dnid = unknown -- dialparties.agi: request = dialpart...

...f("Zap/1-1", "1?4:2") in new stack -- Goto (macro-dial,s,4) -- Executing AGI("Zap/1-1", "dialparties.agi") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi -- dialparties.agi: priority = 4 -- dialparties.agi: callingani2 = 0 -- dialparties.agi: accountcode = -- dialparties.agi: channel = Zap/1-1 -- dialparties.agi: callerid = 5122311245 -- dialparties.agi: context = macro-dial -- dialparties.agi: callington = 33 -- dialparties.agi: dnid = 5126873305 -- dialparties.agi: request =...

Hello, i've recently switched over from 1.0.9 to 1.2.3. I've experienced some (to me) weird behaviour. This is the config for an example queue.conf: [654] wrapuptime=30 timeout=20 strategy=ringall retry=5 queue-youarenext=queue-youarenext queue-thereare=queue-thereare queue-thankyou=queue-thankyou queue-callswaiting=queue-callswaiting music=default monitor-join=yes monitor-format=

...sterHouse.agi,"CallerID": -- accountcode = MisterHouse.agi,"CallerID": -- arg_1 = CallerID MisterHouse.agi,"CallerID": -- callerid = 8884732963 MisterHouse.agi,"CallerID": -- calleridname = UNAVAILABLE MisterHouse.agi,"CallerID": -- callingani2 = 0 MisterHouse.agi,"CallerID": -- callingpres = 0 MisterHouse.agi,"CallerID": -- callingtns = 0 MisterHouse.agi,"CallerID": -- callington = 0 MisterHouse.agi,"CallerID": -- channel = Zap/4-1 MisterHouse.agi,"CallerID": -- contex...

Hi, i have a HT 488 and I want using this like an FXO for Asterisk. I have find some configuration in the list archive & google but my HT with these config not work. my sip.conf [HT-488] username=400 type=peer secret=wowowow qualify=yes port=5062 nat=no host=192.168.1.157 fromuser=400 disallow=all context=from-pstn allow=g711u allow=ulaw allow=alaw my sip debug:

Hi to all I've mada up my mind and i tried to change from i4l to chan_capi, following some councelling from the gurus. I compiled it up, and when i try to load it in modules.conf, i get that wonderful message and Asterisk does not start: [chan_capi.so]Feb 17 09:21:40 WARNING[16384]: loader.c:239 ast_load_resource: /usr/lib/asterisk/modules/chan_capi.so: undefined symbol: ast_get_group Feb

Hi, maybe a dumb question, but it seems that some calls are directed to our central dial in number despite the extensions the callers say they dialled. E.g. they dial 1234-567, asterisk recognizes 12345, it says this is an unknown extension, where it is right, and redirects the call to the central dial in extension 1234-0. This only seems to happen when the numbers are dialled manually. When