similar to: Redirect with ExtraChannel on Bridged call give AMI event with second channel name AsyncGoto/...<ZOMBIE>

Hi All, I download the app_asyncgoto.c, compile the app_asyncgoto.so. Then according to this page http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO ; when I dial ,there have this warning: -- Executing AsyncGoto("SIP/111-086497c8", "SIP/113-08674628|dynamic-nway|111|1") in new stack Feb 2 16:53:10 DEBUG[4218]: app_asyncgoto.c:95 asyncgoto_exec: Attempting

Hi, I am trying to achieve 3-way conferencing taking hint from wiki link http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO Here is the scenario: 1. user "ua1" calls user "ca1" 2. "ua1" then presses the feature code "*0" to redirect "ca1" to conference room 300 3. "ua1" then dials the user "33" 4. user

Hi, I compiled and am running the latest CVS but strange things are now happening.. it looks like asterisk is randomly declaring my calls to be fax calls, complaining and then sending the calls into a black hole... if I hangup the calls below (soft hangup) asterisk locks up and I have to kill the process. NOTICE[21526]: File chan_zap.c, Line 3520 (zt_read): Fax detected, but no fax

Hi, I setup an asterisk system (version 1.8.0-rc5). While using a SIP only environment I discovered a problem using blind transfer. The phones are SNOM or Aastra and are using the SIP REFER Method. The following is working: User A calls user B, B accepts the call, user A than transfers to user C The following is NOT working: User A calls user B, B accepts the call, user B than transfers to user

Hi, in following of a recent discussion I got to work on MGCP with the Cisco ATA186 again, and got it to work very nicely. However, there is a little thing with transfers I would like to get comments on: Call comes in from PSTN and goes to an ATA186 (MGCP) Call is answered and then, using flash, transferred to another extension If the extension is available, there can be an announcement and

Hello list, i am using: asterisk CVS-10/13/03-11:54:33 chan_capi-0.3.0 ATA-186 V2.16.1.ms over MGCP Situation: ISDN calls ATA ISDN speaks with ATA ATA-Phone presses Flash and speaks to another one (SIP/snom200) ATA-Phone hangs up ISDN talks to SIP/snom200 snom200 hangs up The incoming extension of ATA keeps busy for a time (20 sec?), even its not off-hook anymore! Any ideas? -- Swapping

In my application, I am using AMI to run an Originate command between a channel and a dialplan application (NOT a context). In my case, the application I want to invoke is FastAGI. The Originate AMI command works correctly, but Asterisk generates a very short (0-1s) duration for the CDR that results from this call, regardless of the time spent running the FastAGI application. I want the CDR

hi, i am getting these errors while installing asterisk. i reconfigured kernel and i have all the modules installed. kernel-source readline readline-devel openssl openssl-devel this is the error: (at the last part of the installation) gcc -g -o asterisk -Wl,-E io.o sched.o logger.o frame.o loader.o config.o channel.o translate.o file.o say.o pbx.o cli.o md5.o term.o ulaw.o alaw.o callerid.o

Dear R users, I created a matrix that tells me the first day of use of a category by id. #Calculate time difference test$tdiff<-as.numeric(difftime(as.Date("2002-09-01"), test$ftime, units = "days")) # obtain the index date per person and dcategory index.date.test<-tapply(test$tdiff, list(test$id, test$rcat), max) Nonetheless, at the moment I think will be

I have a data frame with a couple million lines and want to retrieve the largest and second largest values in each row, along with the label of the column these values are in. For example row 1 strongest=-11072 secondstrongest=-11707 strongestantenna=value120 secondstrongantenna=value60 Below is the code I am using and a truncated data.frame. Retrieving the largest value was easy, but I have

The Asterisk.org development team has released version 1.6.0-beta4. Here are some highlights from the changes, with the associated issue numbers from bugs.digium.com if an issue was associated with the change. This release contains the following improvements: - 12020, a CLI formatting improvement - 11964, added the ability to get the original called number on SS7 calls - 11873, Added core API

The Asterisk.org development team has released version 1.6.0-beta4. Here are some highlights from the changes, with the associated issue numbers from bugs.digium.com if an issue was associated with the change. This release contains the following improvements: - 12020, a CLI formatting improvement - 11964, added the ability to get the original called number on SS7 calls - 11873, Added core API

We have a client that has sporadic problems with the *8 pickup facility. The server they are using is 1.8.5 and they are using Snom phones. Every now and then when they try to do a pickup from another phone they get a forbidden message on the phone and I can see the following in the logs. [Aug 8 11:51:53] ERROR[19314] astobj2.c: user_data is NULL [Aug 8 11:51:53] ERROR[19314] astobj2.c:

Hi all, After writing data in MMF(Memory Map File) using SEXP structure, can i reference in R? If input data is larger than 2GB, Can i reference MMF Data in R? my work environment : R version : 2.11.1 OS : WinXP Pro sp3 Thanks and best regards. Park, Young-Ju from Korea. ---------[ ???????? ???????? ???????? ]---------- ???????? : R-help Digest, Vol 89,

Hi! I'm building an application which allows to dial via the Asterisk Manager Interface using the originate command. There should be an optional conferencing feature. The manager commands are basically: --------------------------------- action: login username: sdjklgdsjg secret: xxx events: on action: originate callerid: 3847438609 priority: 1 exten: 4068439865 async: 1 context: out

Hi all, I am trying to setup a three party conference using the Asterisk Manager API. I am using the Redirect action over an established two party call. The procedure I am using is to try to redirect the two existing channels to a third party. I would expect this to connect both channels to the third party. However, one of the two parties gets disconnected. Is this the expected behavior? Is there

I've been scratching my head trying to think of a way to do this, but without success yet. I'm using the Manager API. If I have two channels linked to each other (i.e. direct connection), or even if they are independent channels, I can transfer them both to the same extension by using Action: Redirect and using Channel: for one and ExtraChannel: for the other. This is most useful for

Doe anybody know of a way to redirect both channels from a bridge to different dial plan extensions from the using the AMI. Currently, as soon as I redirect one of the channels the other appears to be dropped and gets reorder tone (congestion, fast busy). I guess what I really need is a way to redirect one of the channels and hold on to the other. Thanks, Neil Cherry

Hello! I am having problems figuring out how to do something, and any help would be much appreciated. I would like to use the manager API to take an existing call on a specific SIP extension, dial and conference in a third party. From what I can tell, the way to do this would be to take the two original parties on the call and stick them in a meetme room using Redirect with ExtraChannel,

Hi all, I'm wanting to take an existing call, and transfer both sides of it into a meetme room (yes I know the phones have a conference ability built-in but humor me). What seems to happen is I can transfer one half of it fine, but as soon as I do that the other half hangs up. Do I have to park it briefly? If so, what does the call ID become once it's parked, so that I can