similar to: Get channel name of originated channel

Hello, Is it possible to use the asterisk manager interface to originate multiple channels? like Action: Originate Channel: SIP/101&SIP/102 So that both extensions 101 and 102 rings simultaneously. I am using asterisk manager interface over http. Thanks

Hello asterisk-users, I setup my asterisk to support several features like automon,blindxfer,atxfer,parkcall etc. by using features.conf and the global variable DYNAMIC_FEATURES=automon#blindxfer#atxfer#parkcall#disconnect in extension.conf. Every Dial() command in my diaplan has the appropriate parameters out of {tTkWwW}. For calls from my SIP phones everything works fine. Pressing #1 will

2013/3/26 Richard Mudgett <rmudgett at digium.com> > > On 03/25/2013 05:17 PM, Olivier wrote: > > > Hello, > > > > > > I'm giving hangup-handlers a try on a new Asterisk 11.2.1 setup. > > > My plan is to use this handler to update my CDRs with values such > > > as > > > Asterish and Tech cause (see function HANGUP_CAUSE). >

Dear All, I am originating the call directly to the SIP Provider using the maganger interface + originate (ASYNC) command. Here is the PHP-AGI Script. $call = $asm->send_request('Originate', array('Channel'=>"SIP/416XXXXXXX at ABC/n", 'Context'=>'ORIG',

Hello, Is it possible to detect a hook flash in asterisk. I want to be able to perform some functions an hook flash. I have the following entry in features.conf which executes a Macro on detecting key press '**'. [applicationmap] test => **,caller,Macro,testflash Is it possible to do this action on hook flash? -------------- next part -------------- An HTML attachment was

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi list, I'm trying to use CEL to display channel information in real time. It works fine for simple calls, blind transfers, or SIP attended transfers (initiated from the SIP phone). My problem is for Asterisk attended transfers (atxfer as configured in features.conf). The scenario is: . phone 107 calls phone 100, . 100 dials the atxfer code,

The Asterisk Development Team has announced the release of Asterisk 13.2.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.2.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

The Asterisk Development Team has announced the release of Asterisk 13.2.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.2.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

Hi All, I am in the process of migrating from 1.4.20 to 1.6.2.x and have stumbled across a number of "differences" between the 2 versions that are forcing me to use local channels in my dialplan (mainly centered around the different behavior of CDR fields in the 2 versions) . Previously, I would place a call via an AMI Originate action similar to: action:.Originate..

Hello, I'm giving hangup-handlers a try on a new Asterisk 11.2.1 setup. My plan is to use this handler to update my CDRs with values such as Asterish and Tech cause (see function HANGUP_CAUSE). I want to have my custom hangup-handler be run only once and when "the second channel" hangs up. At the moment, I'm issuing a couple of "channel originate Local/1 at mycontext1

Hi everyone, I'm trying the new atxfer functionality. All seems to work fine at the beginning, but there is no audio between the party at the end of the transfer. Plus, after that, even normal calls won't work well (they can't hangup!). I'm using the Asterisk CVS from 2005-01-10 with Asterisk-OH323. Here is my dialplan: [default] exten => h,1,NoOp(bye) exten =>

Hi Is there a way to disable blind and attended transfer during a call. I am trying this configuration but unfortunately with no luck: - in features.conf [applicationmap] disabletransfer => 9*9,self,GoSub(disabletransfer,s,1) - in extensions.conf [incoming] exten => 99,1,Set(__DYNAMIC_FEATURES=disabletransfer) exten => 99,n,Dial(Sip/alice,120,tT) exten => 99,n,Hangup()

Hello, I want originate a call to some destination, and when B side answes to play a prompt. Asterisk version is 1.6.0.5. But also I need to insert a SIP header to Invite, that's why I'm using Local Channel. This is my extension.ael: context autodialer-local { _X. => { SipAddHeader(P-Asserted-Identity: <sip:${CALLERID(num)}@xxx.xxx.xxx.xxx;user=phone>);

Hi all, I'm having an odd problem with my polycom 301. I am initiating a call to it with AMI Originate() function: Action: Originate Channel: local/112 at Management Context: to_meetme Exten: s Priority: 1 Variable: dropped_conf=111 The "to_meetme" context is very simple: [to_meetme] exten=>s,1,MeetMe(${dropped_conf},id) If I specify every other device I have to test: *

I am sending this again since I haven't get it back for twelve hours: When I originate call to IAXComm, more or less one of tree calls fails for no aparent reason. Originating calls to SIP clients works as expected. Anybody has similar problems? Is it asterisk or client problem? Asterisk log: Jul 15 00:00:04 DEBUG[1179663]: manager.c:1018 process_message: Manager received command

Good...it almost works fine! I just have 't' in command Dial, but i also have 'T'. Is it a problem ? This is my Dial() exten => 605,1,Dial(${GIORDANO NAT},60,Ttr) I have only a problem: A and B are speaking, B calls C and ask it if wants speak with A, C accept but if B hang up A is waiting and C get busy tone. To make it works B don't have to hangup but habe to press

Hi, is there a way to know if originate call channel ended the call *before* connecting to context/extension/priority? DIALSTATUS is empty, HANGUPCAUSE is always 16, nothing in SIP Headers nor in AST_CONTROL_FRAME_[HANGUP|ANSWER] Asterisk is 1.6.2.16 Thanks for any hint -- Daniel

While on conversation with another party, I dial the atxfer key sequence. Asterisk says "Transfer" then gives you a dial tone, while put the other party on hold music. I dial the transferee number and talk with the transferee, then I hang up and the other party must be connected with the transferee. But this doesn't work the transferee hears a beep. -- Playing 'beep'

I've isolated this problem the furthest that I can, and I'm now convinced this is a bug in asterisk. I have a context in extensions.conf like so: [my_context] exten => _X.,1,AGI(my_agi|${EXTEN}|${CHANNEL}) exten => _X.,2,GOTO(my_other_context|${EXTEN}|1) exten => h,1,DeadAGI(my_agi_cleanup) For the purposes of this scenario, my_agi simply will try to HANGUP the channel to

Hi folks, I repeat "as is" the title of a post someone did a few months ago, since I am facing the same problem and did not see one single answer to his post. Maybe I'll be a little bit more lucky. When I'm trying to issue an Atxfer AMI command, in the asterisk 1.8 branch, what happens is that some DTMF's are sent, like this : [Mar 22 15:46:27] DTMF[5910]: channel.c:3900