similar to: Executing a dynamic sequence of applications

Hello, I'd like to use the AMI interface to originate a call to a context in a dialplan, and handoff the dial control to the context. Whenever I execute the below action, the recipient does ring, but when I answer it dials the recipient again. I believe this is because once answered the system is going to execute the Context/Exten/Prio in the Originate action? Action: Originate Channel:

Hi all, I need some advice - I have been working on originating multiple calls using AMI and then joining them. What I want to do is: - dial call 1 (where the caller is in a "channel" format, like SIp/1234 or Local/1234 at ext) and "park" it somehow - dial call 2 (where again the caller is in channel format) and join it to the previous call. As a requirement, I cannot use the

I am using AMI to Originate a call. I have been able to get the caller id number to be passed through. However, I can't get the name to be passed through. A person I'm working with has a Freeswitch that is able to pass the caller id name and number through for their call. Comparing the Asterisk SIP messages to the Freeswitch SIP messages, I have narrowed the problem down to a single

Hello, Could someone explain to me what this means? asterisk[30269]: segfault at 0000000000000008 rip 00002aaac8b388f2 rsp 0000000040a75910 error 4 Also, would this segfault crash the whole Asterisk process or will Asterisk continue to run? Is it possible this would affect/disconnect "SOME" DAHDI channels, but not all? At this point, upgrading is not an option, even though I agree we

It means, AMI application is no more running or crashed or lost network connection with asterisk server. In such cases call is neither answered nor disconnected by Asterisk. I want to detect such state and jump to next dial plan to answer or reject the calls Regards Amit Patkar On September 20, 2016 8:07:23 PM GMT+05:30, Matthew Jordan <mjordan at digium.com> wrote: >On Sat, Sep 17,

Hi all, I have an external application commanding asterisk by AMI and AsyncAGI. I also have a dialplan like this: ; AsyncAGI extensions exten => _8.,1,Noop(entering in AGI loop at 8 ${EXTEN}); exten => _8.,n,AGI(agi:async); exten => _8.,n,Hangup(); ; Meetme extensions exten => _1.,1,Noop(Conference ${EXTEN} ${CONTEXT}); exten =>

Hello, I'm noticing strange behavior in one of our Asterisk nodes where the ACK is always sent to the proxy, but RR is not enabled for calls. The proxy drops the ACK. I'm using the AMI interface to originate a call: Action: login Username: myusername Secret: mypassword Events: on Action: Originate Channel: SIP/<SOMENUMBER>@proxy1 CallerID: <SOMENUMBER> Application: Playback

My application fires several calls thru AMI ORIGINATE command. For example if I have 3 operators I do 3 ORIGINATEs. My trouble is when one operator quit for some reason, I should kill the corresponding ORIGINATE. Of course, I could let the call ring and hangup after the customer pick-up. But this is not the case, I do have to kill the corresponding ORIGINATE. I could execute a soft hangup,

Hi all, I'm planning to develop a custom autodialer application which will be dealing with its own model for agents and queues, therefore it won't use neither asterisk agents nor asterisk queues, nor asterisk cdr. The application will supply the whole reporting and agent managing features by itself. The application will command asterisk through an AMI telnet connection using only the

Hi Is there any way to detect inactivity on channel when AsyncAGI is used? I want to detect whether application handling calls using AMI & AGI has stopped responding. Alternatively, how can dialplan check if there is any AMI user connected and decide dial plan execution? Thanks & Regards, Amit Patkar -------------- next part -------------- An HTML attachment was scrubbed... URL:

HI Matt, I am trying to replicate this particular problem. We are seeing more frequently where the Event: AsyncAGIExec is never being sent. The two scenarios I have seen in tests yesterday and today... We sendl an AMI action. For example, play a short file or hangup. AMI Events will indicate it did the work, but we never receive the Event: AsyncAGIExec with a result at all. Asterisk debug

Hi list, I'm trying to create an asterisk 1.8 rpm with SRTP. I found mention of a libsrtp rpm, <http://qutecom.ipex.cz/RPMS/srtp-1.4.4-1.i386.rpm > in these instructions, <http://www.voip-info.org/wiki/view/Asterisk+SRTP> but it is unreachable (by me, anyway). The libSRTP source is here, <http://srtp.sourceforge.net/download.html>. Has this already been packaged for

We have a customer who wants us to record anywhere from 2-4 participants on a call in stereo (as opposed to mono) quality audio. Some background.. We are using asterisk 16.6.1 We are also currently using AMI/AsyncAGI and ConfBridge to bring the parties together. I believe recording in the various file formats (based on extension), it's always recording in mono quality. My one thought is to

Hi, Is there a way to execute next priority in the dialplan if you have called agi:async? I want to play warning message if adhearsion is down. Currently I wasn't able to make it work. The dialplan execution ends after the first priority. [incomming] exten => _X.,1,AGI(agi:async) exten => _X.,2,Answer exten => _X.,3,Playback(some-message) exten => _X.,4,Hangup Regards, Pavel

Hi I am trying to deploy freeswitch with Digium TE121 card for my office setup, but it is continuously showing Signaling is up and channels are down except D channel. Our Architecture is like We have freeswitch installed with libpri1.4 and Dahdi. I am from India and here we are having E1 trunk. Dahdi Configuration is cat system.conf # Autogenerated by /usr/sbin/dahdi_genconf on Wed Aug 7

what is the difference between FreeSwitch and Asterisk , whitch one is more scalable and reliable? _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Please help me understand the following applications and what are its advantages if we compare between each of them. Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. Regards, Kaushal -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120103/ffad2be6/attachment.htm>

Hi, Maybe I missed it while checking it, but which spandsp version is recommended to play with Asterisk 10 and T.38/T.30 gatewaying ? I can see both spandsp-0.0.6pre17.tgz and spandsp-0.0.6pre18.tgz here (http://www.soft-switch.org/downloads/spandsp/) but I couldn't find a changelog documenting differences between them. So I prefer to double check ask for recommendations. Regards

Hello, We have a Kamailio SIP Proxy in front of our Asterisk cluster for incoming calls from our carrier. The sip.conf looks like this: [kamailio1] type=friend host=10.0.0.1 context=incoming disallow=all allow=alaw All calls hit the incoming extension. In the extensions.conf we have multiple extensions configured, but now I have to add one which uses the special h extension to perform a CURL

Hello, I'm trying to execute a stored procedure on a MSSQL Server from the dial plan, but it's not working. I'm getting the following error: Unable to execute query.... Asterisk has been compiled with UnixODBC, and I've done the necessary configurations in func_odbc, res_odbc and odbc.ini. Has anyone done this before with success? -------------- next part -------------- An HTML