similar to: Asterisk Manager API Action Originate

I'm using the Originate Action on the Asterisk Manager to place calls between two extensions in async mode. Is there any way to cancel the Originate Action before I get the OriginateResponse action? I'm unable to perform a Hangup because I can't know the channel name before I get the response... thanks in advance! -- santiago aguiar *netlabs* / Palmar 2548 Montevideo, Uruguay Tel.

I'm no longer on the DEV mailing list, but: # svn checkout http://svn.digium.com/svn/asterisk/branches/1.4 asterisk svn: URL 'http://svn.digium.com/svn/asterisk/branches/1.4' doesn't exist http://svn.digium.com/svn/asterisk/branches/ -- /Nick -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm trying to move my Asterisk deployments under a Virtual IP address and now remember why I dislike this. My primary Asterisk system is now behind a firewall in private address space. My question is what ports are needed to be opened just for the purpose of placing outgoing calls. I would have assumed none, but I can't even get replies on registration from any of my 3 VoIP providers.

hi, all asterisk 1.4.24 , zaptel 1.4.10.1 , E1 Manager API Action : Action: Originate Channel: ZAP/G1/8888888 Callerid: 12345678 Context: callout Exten: s Priority: 1 extensions.conf [callout] exten => s,1,Answer() exten => s,n,Wait(10) exten => s,n,Hangup() when the phone 8888888 pick up , it will come to callout context, after hangup, one cdr generate, but the

An application commanding asterisk with AMI is going to launch lots of concurrent calls in very few seconds using the Originate AMI command but it's also going to need to be able to cancel very quickly any call of them even before each OriginateResponse event comes in. All the calls will be done by the same trunk (a trunking enabled channel). But there's a problem for canceling any call:

I have written an asterisk manager client which creates an outbound call using Asterisk manager API's Originate action. when the call is connected I run 3 applications on it. 1)read a dtmf digit from user 2)A customized application which I have written,(It plays something to user) 3)Hangup If user hangs up while app 2(see above) is executing I get a 'Event Hangup' from asterisk in my

I have 10 SIP trunks that I'd really like to round-robin load balance. Currently I have a macro that switches between available lines, but there really must be a function in Asterisk to do this on its own. So my question is just that, are there any easy ways for Asterisk to either balance between SIP trunks or even just a built in function to find the next available SIP trunk. I think using

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, Has anyone ever seen an Snom320 lose settings? It's been working fine for months and then I got a call this morning saying that it was asking for country, timezone etc. I logged in remotely, and it had lost the server address, username, password, mailbox and ringtone. - -- Kind Regards, Matt Riddell Director

Hello, I am migrating a PABX system based in Asterisk 1.4 to Asterisk 13, with success. I have an application that sends an action Originate to AMI for calling, it's working well, but when i see to Asterisk's CLI, i see 2 calls for just one originate: pftestes40copiabh*CLI> core show channels verbose Channel Context Extension Prio State Application

hello, please any IAX2 ActiveX control that wrap libiax2 or libiaxclient? i want to develope my softphone in delphi thanks __________ Information from ESET NOD32 Antivirus, version of virus signature database 4345 (20090818) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com

Hi All, I am trying to use the AMD (Answering Machine Detect). But it is not sending the AMD_Status as either the Human or Machine, it hangs up in middle. can any one suggest us, what might be the problem and possible solution to it. below is the log -- Executing AMD("SIP/sip-ffe0", "") in new stack -- AMD: SIP/sip-ffe0 14082284927 (null) (Fmt: 4) Apr 23 08:00:26

Hi to all. What can i do if a customer needs to log in the CDR all the dialpan actions related to a call? I mean, not only the lastapp e the lastdata but all the dialpan actions! I know that the actual CDR system store one record for each call (and for billing purposes this can be correct) but in some cases the approach needed is something similar to the queue_log. I know that exists ResetCDR

>Soumya Kat wrote: > What I would like to know is how to get information such as SIP users, > number of SIP connections and traffic associated with those from asterisk > using a C Code. >Russell Bryant > There is actually no good way to do this inside of Asterisk right now. It's > certainly all possible ... it's just software ... but there is no > straightforward

We just completed a new implementation of voicemail for Asterisk. It's much cleaner than Comedian mail and can emulate several voicemail user interfaces, including Audix. It's a great replacement for Audix. All of the sounds/prompts are presently being re-recorded by a professional female voice. If you are interest in the app, let us know at nt_jnewman at yahoo.com. Justin

Hi to all. I'm begin a use the AMI and i have the need to get the uniqueid from the call i have generate using the Action Originate. Anyone can help me? When I generate these commands: action: Originate channel: SIP/101 application: Dial data: SIP/100,120,Ttr The only response I get when the call is answered, is this: Response: Success Message: Originate successfully queued Thanks a

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Has anyone done any integration with this? All I know so far is that it appears to use some non standard form of SIP. Any pointers? - -- Kind Regards, Matt Riddell Director _______________________________________________ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News -

Hi all, I run an Asterisk 1.4.24.1 and face problem with DTMF. When calling from my mobile phone -Nokia E65- in GSM, Asterisk present me a second tone so I can use the GW. For this I use: exten => s,1,NoOp(One of our workers (${CALLERID(number)}) is calling office) ;callerID is the one of the calling mobile phone exten => s,n,Background(silence/1) ; Nokia E65 send digits in

For some reason, I have been unable to find the answer to this online or in books... I want to have a "click-to-connect" feature on my website where the user enters their phone number and then my asterisk server calls their phone and my phone and connects the two calls to each other. All I have are: 1. A Server 2. A DSL connection 3. A Router and DSL Modem 4. A static IP What do i

Hi All, I'm stumped on this and I looking for some clues to fix this. This is a new install of Slackware 12.1 onto an IBM x330 Server. Asterisk 1.4.20.1 plays the wav files and the Cepstral_Allison Swift just fine, but when I play the gsm files the audio quite choppy. And, the files produced from the MixMonitor don't even record any audio other than noise. I have a hard drive from

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all, anyone else seen RX/TXFax crashing Asterisk on latest Asterisk SVN? I've now seen it on two machines I tried to set up - one it seems because the tiff file was malformed, but the other is doing: tiff -> tx fax -> zaptel -> pstn -> ddi -> zaptel -> rx fax -> tiff The above crashes every time. If no one else has