similar to: extensions not being detected consistently

Hello, My client customers complaining that their calls suddenly get hung-up, I am just investigating if the problem from my side, I had a log of a hang-up case, Does it help to know if there is a problem that can be resolved from my side? elastix*CLI> -- Hungup 'IAX2/99999-6813' == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on

Hi List, I have another issue on allowing outgoing calls to PSTN on Asterisk via Avaya Phones, I hope that anyone could help me fix this issue: *When I dial through Avaya phone i just here a "good bye message" reply from asterisk server. And here is the log:* == Starting OOH323/(null)-b7db8aa0 at internal,s,1 failed so falling back to exten 's' == Starting

Hi list I'm kinda new to asterisk and I'm woriking for a company that sells Asterisk solutions and appliances. I installed TrixBox on a litle PC @ home and a x100p card which is recognized as a Zaptel card, I made some in/outbound routes and they seem to work but I have a problem with SIP softphones. I created 2 estensions 1000 and 1001 they're both in different cities, when I 1000

Hello, I have setup the n-way conferencing with Asterisk and it's working when I use with my budgetone 100 phone but it doesn't work for any of the voip software or other ATA that I have. When I turned the debug on, I see that the correct keys (*0) were entered but asterisk doesn't detect the signal to trigger the features event. I have set a test extension to get the input dtmf key

Hey all, I wrote a very basic script to determine if cassandra db is running. I'm setting a variable called 'pid' to the output of a ps | grep like to grab the pid of the cassandra process. #!/bin/bash pid=$(ps -ef | grep cassandra | grep -v grep | grep -i -v -e grep -e screen -e s3fs|awk '{print $2}') if [[ -e $pid ]] then echo "Cassandra is running with pid:

i have posted this question couple of times and never really got any hits i wasn't able to provide any debug info Connected to Asterisk 1.6.0.21 currently running on phoneserver (pid = 3309) Verbosity is at least 4 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 == Using SIP VRTP TOS bits 136 == Using SIP VRTP CoS mark 6 == Extension Changed 117[ext-local] new

I just had a user report that they called out to someone on a cell phone this morning, and after a short conversation, the call was dropped/lost. The person on the cell phone says that this is very rare, and would not suspect the dropped/lost call to be on their side. I have looked at the asterisk/full log as thoroughly as I can, and have pasted the lines which seem relevant to that call below.

> > "-e" means "if file exists". You should use "-n" That did it!! [root at web1:~] #./bin/check-cass.sh Cassandra is running with pid: 26979 This is what the script looks like now: #!/bin/bash pid=$(ps -ef | grep cassandra | grep -v grep | grep -i -v -e grep -e screen -e s3fs|awk '{print $2}') if [[ -n $pid ]] then echo "Cassandra is

Hi, With Asterisk 1.8.3, I can't figure out how to pass an argument to a macro which is used within an originate command. Here is my sample dialplan to illustrate: exten => 123,1,Answer() exten => 123,n,Originate(SIP/20,app,Macro,foo,bar) exten => 123,n,NoOp(This is the NoOp after the originate command) exten => 123,n,Wait(30) exten => 123,n,Hangup() [macro-foo] exten =>

Hi people, I have a problem with some extensions. The calls are ended after 31/35 seconds, also, it depends on the number which I call. This is the log, but I've not been able to find something wrong... Any ideas? [Jun 11 15:50:46] DEBUG[26071] app_macro.c: Executed application: ExecIf [Jun 11 15:50:46] VERBOSE[26071] logger.c: -- Executing [s at macro-dialout-trunk:16]

I've got a piece of CPE equipment that has an FXS port that I have tied to an FXO port on a TDM400 clone card. Normally, if I go off-hook with a standard telephone connected to it, I get a dialtone. If I dial a digit, and send a hookflash, the device will provide a dialtone back for the next available channel on the device. I'm trying to recreate this same behavior with Asterisk,

I am trying to record calls on demand both inbound and outbound calls. I can record outbound calls just fine but not inbound calls or calls from an internally between extensions. I am using the latest asterisk 1.8.x certified version. On an outbound call I see: == Using SIP RTP CoS mark 5 -- Called SIP/ BVTrunk /7190000000 -- SIP/BVTrunk-00000163 is making progress passing it to

Hello, This is my first asterisk installation, and having read up on the documentation, and trying several tutorials i'm unable to get my outbound route working. I'm certain it's an issue with my configuration and my inexperience with asterisk. So i have my POTS phone connected to my digium card, and when i make a call, I receive the "cannot be completed as dialed" message.

[also posted on Trixbox trunk forum] I am also working with Sangoma directly to debug this, but so far no real luck. TrixBox 2.6.1, A102d card with V33 firmware (latest) and WANPIPE 3.2.6 (3.2.7 is out, but nothing has changed that would affect this problem). The system gets about 200 calls inbound on the trunk, which is not very heavily used, and of those calls one or two a day is randomly

I have a .csv table named mailing.csv as below. It consist a receiver, subject and sender. Receiver subject sender 1 Adrian Cole RE: [WHIRR-117] Composable services Tom White 2 Adrian Cole RE: [WHIRR-117] Composable services Tom White 3 Adrian Cole RE: [WHIRR-117] Composable services Adrian Cole 4 Adrian Cole RE: [WHIRR-117]

Hi All, I have searched the various post and not able to find the solution. I am using Asterisk 1.4.21.2 for outgoing calls. Earlier i used ZAP trunk and it works fine. Now i need to move to SIP trunk and configured the same. When i try from softphone i got error as "Call rejected" and in the asterisk i got error as

I am having problems sending and receiving calls over the T1. They never seem to connect - outbound keeps ringing, inbound gets busy. The T1 looks ok - no errors on the line. Any ideas on what is wrong? I have tried a variety of fxsks and fxoks configurations without avail. This is a single asterisk@home system with a single T1 card. Robbed Bit T1 ami, d4. ------------------inbound call

Hello Guys, I'm somewhat of a newbie and am desperately seeking for some help... I've managed to get asterisk up and running on my server, and signed up with a broadvoice account... I'm having no problem dialing and communicating between extensions, but whenever anyone tries to call my broadvoice account, they are greeted by no ring or anything, but rather simply a direct to

Hello List, I have few installations out there based on 1.6.1 or above. I'm trying to play different voice mail messages based on certain criteria's. For example, I want during office hours to play (in short): "we are not available to take your call, please leave a message", during off-hours and weekends I would play: "we are closed, our opening hours xx:xx-yy:yy, please

Please check out this PNG image from attachment or from Google docs: http://docs.google.com/drawings/pub?id=1P3jdSddseG1oSYrtjREWcajizxmxoRIhUHCEw4sDi3k&w=771&h=624So, what I want to do is something like a private cloud storage solution.I belive the http servers and application servers should be set up on virtual VMs, but what about the Cassandra and Hadloop servers, should their be set up