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Is there a way to pass variables/arguments to the h extension ? for example : [default] exten => _1098933X.,1,NoOp(CARRIER TWT->TIM, EXTEN: ${EXTEN}}, SIPCALLID: ${SIPCALLID}, SIPDOMAIN: ${SIPDOMAIN}) exten => _1098933X.,2,SetVar(_PROVA="bla") [lot of stuff, agi, goto, tricks and magic that happens] exten => _1098933X.,10,Dial(${CHAN_DEST},,L(3600000:3599900)) <-

let's suppose I have this dialplan : exten => _X.,1,Playtones(ring) exten => _X.,2,Dial(CAPI/contr1/${EXTEN},,g) exten => _X.,3,AGI(update) where "update" updates some db tables we have based on the type of extension Now, from the wiki : If the /g/ option is specified, and the called party hangs up before the calling party, then Dial exits with a return code of 0 to

I've an asterisk 1.2 connecting to a quescom gateway via SIP, the caller (asterisk) can hear the called, but the called hears nothing. Since both machines are on public ip, what other problem can it be ?

I have strange peaks of machine load on my asterisk servers, looking at top the load is very high even if cpu usage is low and no swap memory is used. This happens on all the machines, some of them have asterisk, mysql, agi and digium cards on them, so I thought I was only asking too much, but yesterday I noticed the same behaviour on an asterisk machine with only two digium in it, no other

I've the following installation : |asterisk client| --- > |asterisk server| --- > |other asterisk server| all the connections are made in IAX, the client and first server allows 711 and 729 the other server only allows 729 since it has low bandwidth at disposal all the numbers but a few are routed to a digium card in the first server, the others are routed to the other server, this

Hi, Just downloaded the latest cvs from zaptel on my sparking new Athlon64 Centos4.2 system, but hitting a stumbling block... (sorry for the long post) #make cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -m64 -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -c -o gendigits.o gendigits.c cc -o gendigits gendigits.o -lm ./gendigits cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -m64

we got this installation : WinSip(demo version) -> ser(radius accounting) -> asterisk(from sip to h323 channel) -> gsm gateway(with 32 sims in it) we configured winsip to make 28 calls like from 28 different sip accounts, to 28 different cellular phones numbers after the first ten : -- Executing Dial("SIP/5060-081925b0", "OH323/33xxxxxx@gsm.gateway.ip") in

I've this setup : CiscoAta186 -> asterisk with oh323 chan -> gsmgateway dtmf doesn't work, tryed inband, with g711a and g729 codecs CiscoAta186 -> gsmgateway works, even with g729, so it seems the problem is in * oh323.conf has inBandDTMF=yes, what else may I need to tweak ?

for the particular configuration of software/hardware that connects to my asterisk pstn gateway I need to do something like the following : [...] exten => _X,3,Dial(CAPI/02xxx.b${EXTEN},60,M(senddtmf)) [...] [macro-senddtmf] exten => s,1,SendDTMF(*) but the DTMF must be sended to the caller channel, and not the called : SIP -> * -> ISDN SIP calls some ISDN number, when ISDN picks

We've got this configuration : Cisco as5400 --- asterisk main server ---- asterisk for cells ---- gsm gateway cisco and the gsm gateway are connected to asterisk via sip, the two asterisk servers are connected via iax. On a succesful call the cisco (not always, 60% of the times) will keep sending a ringtone to the connected phone, even if the call is answered, actually if the user behind

While I wait for the call to be answered I hear a "double ringing tone", like : expected tone : tuuu tuuu tuuu tuuu what I hear : tuuu tuuu tuuu tuuu tuuu tuuu tuuu tuuu the second "tuuu" I think is generated somewhere and not "true", since it sounds slightly different and the lambda between the first and the second is always

sorry, just a test, seems I'm no more receiving mails ...

I've an asterisk 1.2 connecting to a quescom gateway via SIP, the caller (asterisk) can hear the called, but the called hears nothing. Since both machines are on public ip, what other problem can it be ? There's one configuration working : lynksys pap -sip-> asterisk server -sip-> quescom this way both sides can hear voice but with : lynksys pap connected to a switch -sip->

I terminate some calls on a h323 device (a quescom gsmgateway) from asterisk 1.2.3 with ooh323, the customer is complayining about choppy sound on most of the calls, the only warning message I can see is : src/chan_h323.c:944 ooh323_indicate: Don't know how to indicate condition -1 on ooh323c_102 (the calls sounds perfectly with iax/zap termination and the quescom seems to work fine with

After reading a description of apparently the same problem by Juan J. Sierralta more detailed than mine "tuuu tuuu instead of tuuu" we've solved the problem changing the call progress tone of sip phones to something not udible.

If I call a cellular phone while it's off, I can't hear the voice saying "called number is unreachable", but only if I'm passing trough a iax channel. SIP client ---> Asterisk ---> SIP gateway, works SIP client ---> Asterisk client ---> Asterisk server ---> SIP gateway, doesn't work (I can't put an explicit Answer in the extension for billing

I have a strange problem when calling some numbers with asterisk, I get an hangup for busy condition even if the phone at the other end isn't busy. I can route the calls via SIP to another carrier and then I have a SIP code 486 or I can terminate them on digium cards (E1) and I have an Hangup code 17 I know for sure that one of the numbers is hosted by a different provider than the one

I know there's a variable for the IP of a SIP channel, but I can't find if such a variable is avaliable for a generic voip cahnnel, or at least h323 channels (ooh323)

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

I've setup this simple configuration to test the new mediaonly iax feature in 1.4 : Input (client) -> Server (routing) -> Termination transfer=no transfer=mediaonly transfer=no all the machines are in the same 192.168.0.x net the routing Server in the middle has iaxusers realtime backend on mysql the call is originated with a sip phone registered on the Input client