similar to: Redirecting a channel to Meetme fails with Hangup.

I understand how to control the first local channel, but an having trouble getting the second local channel to enter stasis. I setup have the following extensions.conf to handle 1000 (basically had it setup so if first stasis not there try second, but believe second channel never processes the dial plan so even if second line was hello-world2 it would not matter. [mycontext] exten =>

Hi I am experiencing Asterisk Crash. Log got stopped when asterisk crashed. Please help me to identify the reason and fix this issue. Asterisk: 1.8.20 I am using AMI and fastAGI to control the call. Some part of dial plan is also defined in extensions.conf I am experiencing this crash when app_meetme conference functionality is used with more than 3 parties. I faced this issue with

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 Team, Thanks for your great job an Asterisk new features developments. I installed asterisk-12 Beta and found some changes as well which i notice to put in-front of your knowledge, don't know that bug of new modification into objects or old version (asterisk-11) mistake corrected that time, anyway *Asterisk-12:* Array ( [Event] => ConfbridgeMute [Privilege] => call,all [Conference]

Hi, Thanks for your info, What is the impact of the following line in dialplan, Dial(SIP/19201/19202,300) On Thu, Sep 3, 2015 at 7:20 PM, Vinicius Fontes <vinicius at aittelecom.com.br> wrote: > You might want to use the Originate() application instead. Check its usage > by issuing the command 'core show application originate' on Asterisk CLI. > > 2015-09-03

Hello, I am using Asterisk 11.0.1 and do not notice any changes regarding the Transfer on newer Asterisk 11.x versions. I am using AMI and controlling a channel via AsyncAGI. I send a Transfer cmd (such as the following) Action: AGI ActionID: C8 Channel: SIP/1004-00000002 CommandID: C8 Command: EXEC Transfer SIP/1003 Destination phone starts ringing. If it answers the

Is it possible to simply send a message to appear as an AMI message/event, from the dialplan? For example exten =>123,1,ami(myEvent, param1, param2) and in the AMI a message appears like: Event: myEvent Privilege: call,all Channel: PJSIP/misspiggy-00000001 Uniqueid: 1368479157.3 ChannelState: 3 ChannelStateDesc: Up CallerIDNum: 657-5309 CallerIDName: Miss Piggy

Hello Group, I have a requirement to dialout some external number, once the call is answered the same has to be forwarded to an Internal Queue. Please help me. I have tried calling with two SIP end point forwarding , even that is not working, My dial plan line is , Dial(SIP/19201/19202,300) -- *Best regards,* *Ruban.S* -------------- next part -------------- An HTML attachment

We are communicating with Asterisk via AMI. Running Asterisk version 13.18.5 on an Ubuntu box. If you look at the event response, the Result field is filled with random characters. I'm not sure what to do because that is a completely random result. It makes no sense. We send the following command to asterisk via AMI Action: AGI ActionID: C44415 Channel: SIP/192.168.40.105-00001338

Please help find the cause of strange behavior res_pjsip. Making outgoint call to other sip server (CommuniGatePro), my asterisk suddenly sends BYE after picking up! Partial log of an outgoing call with full debug is attached and on web: http://pastebin.com/tLNCpx4d No diagnostic messages why asterisk suddenly decided to hangup i don't found :( There are suggestions or strong belief

Hi List, I am trying to setup a Asterisk setup in AWS instance Centos6.5 . I have installed Asterisk 13.4 with srtp,pjproject. I have configured two numbers for webRTC clients, when i try to call from a client (sipml5) to another client (sipml5) it throws the following error: "chan_sip.c:5851 dialog_initialize_dtls_srtp: Attempted to set an invalid DTLS-SRTP configuration on RTP

Hello List, Good day, We have faced an issue in Asterisk, the issue is we are not able to make manager connections and we could see two pid's for the service asterisk status command. The sequence of operations in our setup goes here, 1) Linux machine is rebooting , we have issued a service asterisk restart command. 2) Tried to connect from an application to asterisk.

Hi List, The below error caused the Asterisk to crash, if anyone have idea on this please reply,(Asterisk version :1.8.9) [Sep 2 15:59:53] WARNING[24418] channel.c: Codec mismatch on channel SIP/18202-0002d011 setting write format to ilbc from ulaw native formats 0x4 (ulaw) [Sep 2 15:59:53] WARNING[24418] channel.c: Unable to find a codec translation path from 0x4 (ulaw) to

-----BEGIN PGP SIGNED MESSAGE----- ============================================================================= FreeBSD-SA-02:20 Security Advisory FreeBSD, Inc. Topic: syncache/syncookies denial of service Category: core Module: net Announced: 2002-04-16

Dear all, I am relatively new to R and have had some difficulty in understanding an error i get when running a code to solve a system of non-linear equations, with four equations and two variables. This is my code: ALPHA <- c(-0.0985168033402, 0.1, 0.15, 0.2, 0.25, 0.3, 0.35, 0.4) BETA <- c(-0.0985168033402, 0.1, 0.15, 0.2, 0.25, 0.3, 0.35, 0.4) GAMMA <- c(0.3940672148378, 0.1, 0.15,

Is there a form of with (or an equivalent function) that can be applied to objects? I'd like to be able to write something like (drawing from bioconductor + trellis as an example) xyplot(maA ~ maM | maPrintTip, object = swirl[,1]) which would be interpreted as xyplot(swirl[,1]@maA ~ swirl[,1]@maM | swirl[,1]@maPrintTip) (or even better, match functions not slots eg. as

Santi, In the second line of your function you have the following: f <- numeric(length(x)) This sets the length of this numeric vector (i.e., "f") to the length of the vector "x". Later, inside the function you assign to values to 4 elements of the vector "f". This assumes that "f" is at least 4 element in length. However, you define

Hi I've 'inherited' an existing Asterisk with a number of users, and some pstn connections through its Zaptel card. I've recently set up another Asterisk which has no direct pstn access and I've connected the 2 systems with IAX. The original system has an extension number range 1xxx and the new asterisk has the number range 3xxx. I can call both ways, and when required can

I tried to use calcMin with a function that uses a number of ... arguments (all args from resid on) besides the vector of parameters being fit. Same idea as optim, nlm, nlminb for which this form of ... syntax works. But with calcMin I get an error regarding unused arguments. No partial matches to previous arguments that I can see. Anybody know the reason or fix for this?

Hi , I have a problem with a X100, i do a external call to the asterisk server . The dialplan its simple answer and hangup.. when it's done , the telephone which i did the call , is in line but asterisk server is finish. I'll apreciate all your suggestion. Greetings, txus. The asterisk output: -- Executing Hangup("Zap/1-1", "") in new stack == Spawn