search for: application_dial

Hey guys, I've got a part of my dialplan that dials multiple people: exten => direct,n,Dial(${QUEUEEXTS},${RINGTIME}) Multiple extensions are in the ${QUEUEEXTS} from an external script - e.g. SIP/100&SIP/101&SIP/105 etc This works great, however I want to see if I can find a way to work out (and run an AGI script) when the call is picked up by someone. Thanks all!

I am designing a solution for a hotel booking call center with the following (mandatory) design: After the call from the customer with the booking agent is complete (and the Hotel PBX disconnects from the call), a second PBX takes over to conduct a survey of how the call went. Both PBX's are Asterisk based. So customer phone [C] connects to hotel PBX [H]. Once [H] disconnects, the

i do have asterisk 1.8 (no gui, no distro based) and i would like to enable some features:-call forward (conditional, unconditional,...)-DND-call waiting-attended transfer-follow me all the features i would like to enable/disable them through digit codes such #45# and *45.all these fetures should apply to asterisk only and not use the features from the service provider. i have edited the

Asterisk 11.1.0 I'm trying to use the "b" subroutine of the Dial application so that I can do some stuff with our internal applications that need to have access to the called channel information. I can see that the subroutine is being executed, but the arguments I pass don't see to make it to the subroutine. [callmenow] exten => s,1,NoOp(callmenow: Queue without answer)

Hi there; I didn't see any "G" option in the example above, and the usage for the option parameters is entirely undocumented at https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_Dial The G options are as below G - If the call is answered, transfer the calling party to the specified priority and the called party to the specified priority plus one. context exten priority I think I have something almost there now, with the following: [streamdemo] exten => s,1,Answer exten =&...

hi, I hope someone can help me:-) we’ve got a freepbx server. there are 2 special extensions (2001, 2002). if someone calls this extensions (or a call is forwarded to these extensions) and these extension hangup (not the caller party), then we’d like to put the calls back into a queue (1000) and wouldn’t like to hangup. I read your description about hangup hooks:

Hi, Is it working for anyone? I have tried with trustrpid=yes sendrpid=yes/pai but can not get it working, Asterisk cli shows prevented message like this. Connected line update to SIP/1231-00000200 prevented Regards, Zohair Raza -------------- next part -------------- An HTML attachment was scrubbed... URL:

Thank you - that makes sense. I've seen something about swapping and optimizing channels on the console, but I didn't realise "optimize" meant "not do what you wanted". OK, so here's why I'm dialling anything at all: The first dial is because I MUST limit the incoming call to less than 60 minutes. The second dial, which carries the gH option, is because I

Hi. I'm trying to get Asterisk 13 to authenticate when it sends an INVITE, and for some reason it's simply not doing it. I've even resorted to reading the source code to try and work out what I'm doing wrong... In channels/chan_sip.c I find: * SIP Dial string syntax: * SIP/devicename * or SIP/username at domain (SIP uri) * or

I'd like multiple people to be able to dial in and listen to various live radio streams. I was told that the correct resource-friendly way would be to setup a MoH class, and then select that from the dialplan. This works well, but how do I switch between streams? Someone correct me if I'm wrong, but from previous similar questions a few years ago it seems like once you've entered a