similar to: WARNING[4218]: res_features.c:1385 ast_bridge_call: Bridge failed on channels ( when I use asyncgoto)

Originally posted on asterisk-dev with no response for 5 days, so posting it to the wider audience now. Asterisk Release 1.6.1.1 Scenario:- 1. 2 SIP peers (Zoiper softphone, if it matters) registered as 901 and 902 2. Using AMI, 901 is Originated 3. When 901 answers, it is Redirected to an extension "exten => dial,1,Dial(SIP/902)" 4. 902 rings, then answers 5.

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

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

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

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, 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

Hi, i am using the n-way-call dialplan solution found on voip-info. i have added its entry in applicationmap of features.conf file. the problem is......its not working. to activate the n-way call i dial *0 but nothing happens. i have played around with dtmf and codec settings but no success. the extensions and sip configuration is below if you want to have a look. I dont have any clue why its not

is where anyone who knows what is needed to get the pickupexten (*8) running ? gentoo asterisk-stable 1.2.4/zap1.2.4 with bristuff I've activated it in features.conf (default *8) and also tested other extensions res_features.so is loaded show features says: Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind

On my asterisk machines the following features.conf file crashes asterisk (core dump) This happens with 1.2.4, 1.2.10, 1.2.12, with or without bristuff. It's easy to work around, but broken nevertheless. Has anyone else experienced that or is it just me? ;) -------- /etc/asterisk/features.conf ---------------- [applicationmap] # THIS CRASHES asterisk: rateone => #1,caller,Macro,ratecall,1

Hello! I need to use n-way call as it described here: http://habrahabr.ru/sandbox/52259/ It is in russian, but dial plan is quite clear. It works in asterisk 11: -- Blind transferring OOH323/7272-6385 to '0' (context fromtransfer) priority 1 -- Executing [0 at fromtransfer:1] NoOp("OOH323/7272-6385", "") in new stack -- Executing [0 at fromtransfer:1]

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 "user1" calls user "user2" 2. "user1" then presses the feature code "*0" to redirect "user2" to conference room 300 3. "user1" then dials the user "user3" 4.

I'm using asterisk 1.4.5 , on Centos. My kernel version is 2.6.9-55.ELsmp I've configured the nway call. I made entries in extension.conf, feature.conf, as per required. I'm trying to make a 3-way conference with the 1 user myself ( using asterisk), and two others are PSTN line users. I'm making a first call , then putting that person on hold by pressing **( as per feature.conf

Hello, using Asterisk 1.8.12.2 I am having trouble with exiting the conference room by entering a single digit. option X of the Meetme()-application should do this. I have following in extensions.conf : /exten => _1000X,n,Set(MEETME_EXIT_CONTEXT=dynamic-nway-invite)// //exten => _1000X,n,MeetMe(${CONFNO},dMX)// // // //[dynamic-nway-invite]// //exten => 0,1,NoOp(confno =

Hello, I'm trying to implement multi-party calls according to these instructions: http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO They are almost working, except that the Goto at the end of [dynamic-nway-start] doesn't seem to work. When I turn verbosity up a bit, I get something like this in my error log: == Channel 'SIP/SWG-0085a180' jumping out of macro

Hi All, I have a really strange issue occuring where if I run "show dialplan" or "dialplan show" or "dialplan show parkedcalls", then asterisk dumps core. It only appears to happen with contexts that are created within res_features. I am able to display all my other dialplans, but, every time I try to just do a normal "dialplan show" asterisk core dumps

Dear list, I have a dataset as follow and I would like to count the frequencies for the combination of the two variables (f1 and f2) for each id. I know it is should be straight forward, but I just don't know how to do it in R. Here is the SAS code I will use to get the output I want : proc means nway; class id f1 f2; var flag output out=temp; Dataset: id f1 f2 flag 798 1 2

Hi, Can someone set me straight as to how to write formulas in R to indicate: complete independence [A][B][C] joint independence [AB][C] conditional independence [AC][BC] nway interaction [AB][AC][BC] ? For example, if I have 4 factors: hair colour, eye colour, age, sex does > mosaicplot( frequency ~ hair + eye + age + sex) mean that the model fitted is of complete independence of