similar to: how to execute something after Dial() ?

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)) <-

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'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

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

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

Hi all, when I do a "grep JERRY *.h" nothing is returned which is what I expect. This is in my source directory... when I do a "grep JERRY *" every file is returned an a line printed even though there is no JERRY on the line. Then if I do a "grep JERRY *.c" just the 4 lines that have JERRY are returned. This is what is prints for "grep JERRY *" src#

Hi Jeremy, I'm afraid this is indeed caused by the talloc tidyup as one can perfectly reproduce it wih the following patch for smbtorture (call bin/smbtorture ncacn_np:lo local.file.file_lines_parse). I guess we need to set up some warning signs around file_lines_parse() expressing that it takes talloced content and that one should not free the separated lines array separately. Cheers,

I've the following setup : sip phone -> ser (auth and routing) -> asterisk with capi isdn when I call a pstn number everything works fine, but I can't hear anything till the called answer. this is the output from a test call : -- Executing Playtones("SIP/2.7.184.61-08152880", "dial") in new stack -- Executing Dial("SIP/2.7.184.61-08152880",

In my application, I am using AMI to run an Originate command between a channel and a dialplan application (NOT a context). In my case, the application I want to invoke is FastAGI. The Originate AMI command works correctly, but Asterisk generates a very short (0-1s) duration for the CDR that results from this call, regardless of the time spent running the FastAGI application. I want the CDR

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

I am seeing some curious behaviour with a 1.2.32 system, which I do not understand and so can't work out how to fix it. I have a PRI routed to context default. Here is the complete default context: [default] exten => _9X.,1,Dial(IAX2/m1peer/${EXTEN:1}) exten => _20XX,1,Dial(IAX2/sipeer/${EXTEN}) exten => _X.,1,Dial(IAX2/m1peer/${EXTEN}) exten =>

hi list, I'm looking for a way to execute commands in my dialplan specifically when a caller has hung up. my curretn dialplan looks like this: exten => s,1,Answer exten => s,n(restart),BackGround(intro) exten => s,n,Read(Enter,4,4) exten => s,n,Voicemail(${Enter},u) exten => s,n,agi(process.php|${Enter}) exten => #,1,Playback(thanks) exten => #,n,Hangup it lets a user

On Wed, Nov 30, 2011 at 10:42 PM, Chris Lattner <clattner at apple.com> wrote: > On Nov 29, 2011, at 12:26 AM, David Blaikie wrote: >> For a bit of an experiment I've been trying to compile LLVM & Clang >> with -Weverything (disabling any errors that seem like more noise/less >> interesting). One warning I've recently hit a few instances of is >>

I'm not sure if this is an Asterisk thing, a handset thing or a telco thing, so please be gentle with me if this is not the right place to ask ..... When placing a call over a SIP channel to a mobile phone, if the phone is engaged, it does not return a BUSY status straightaway. Rather, I get a ringing-out tone for the timeout duration specified in the Dial() statement; *then* I get

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 This is slowly driving me nuts! I have several Cisco 7960s with SIP 8.2/7.5 fw connecting to Asterisk 1.2.5 driving a TE110P on a BT EuroISDN PRI line. On all outgoing calls I get a double ring tone (UK style + US style). I also have a DECT phone on a Sipura SPA-3000 configured with UK tones. This gives me a double ring of UK + UK, so this

Hi List Implementing screening and routing I have stumbled over this issue: [pbx-router] exten => s,1,NoOp(ROUTER FROM: ${CALLERID(Number)} TO: ${DESTINATION}) same => n,Set(SOURCE=${CHANNEL(name)}) same => n,Set(PAI=${PJSIP_HEADER(read,P-Asserted-Identity)}) same => n,Set(FROM=${CALLERID(Number)}) same => n,Set(TO=${DESTINATION}) same

I've just finished an Xcode 3.1 project file for flac and metaflac that builds both tools as 32/64 bit universal binaries. If anyone is interested in either the binaries or the project, I'll be happy to share them. Stephen

On Dec 1, 2011, at 12:08 AM, David Blaikie wrote: > On Wed, Nov 30, 2011 at 10:42 PM, Chris Lattner <clattner at apple.com> wrote: >> On Nov 29, 2011, at 12:26 AM, David Blaikie wrote: >>> For a bit of an experiment I've been trying to compile LLVM & Clang >>> with -Weverything (disabling any errors that seem like more noise/less >>> interesting).

On Mon, 2020-11-09 at 08:55 -0800, Jeremy Allison via samba wrote: > On Mon, Nov 09, 2020 at 04:01:47PM +0100, Benedikt Kale? via samba > wrote: > > Dear List, > > > > I updated to samba 4.12.10-SerNet-Debian-9.buster from 4.12.9 > > and did a > > > > gluster volume set volume performance.write-behind off > > > > Now I get this in

Hi all I am looking at implementing asterisk at a company with two ISDN bricks (60 lines). I know that the VoIP will absorb at least on brick worth of lines but that still leaves me with a need for 30 ISDN lines. As far as I can tell most of the Digicom cards have 4 FXS ports and I've read on this list that at most two could coincide in a box simultaneously without causing an interupt