Server Fault Stack Exchange
Server Fault Stack Exchange

Questions tagged [asterisk]

Asterisk is software that enables a server to act as an IP PBX system, VoIP gateway, conference server, and more.

see

http://www.asterisk.org/

and

http://en.wikipedia.org/wiki/Asterisk_(PBX)

593 questions
0
votes
2 answers

DAHDI Installation on Slicehost Woes

I'm trying to get DAHDI installed for Asterisk on my Slicehost slice... I start off sudo apt-get install dahdi-dkms dahdi-linux Which partially fails with this in the install log: Setting up dahdi-dkms (1:2.2.1+dfsg-1ubuntu2) ... Loading new…
neezer
  • 810
  • 3
  • 12
  • 29
0
votes
2 answers

NGN detection with Asterisk PBX

Is it possible to detect if a NGN (Non Geographic Number, e.g. 0844, 0845, 0870 etc) number has been used to connect to an Asterisk PBX? Thanks in advance. Kyle
Kyle Hudson
  • 202
  • 2
  • 11
0
votes
0 answers

Asterisk server shows peer UNREACHABLE for Kamailio sipper server. How do I diagnose the issue?

Presently I have 4 sippers, 2 are working using OpenSer and 2 are not working using Kamailio. I have inherited these servers. There is an Asterisk server. When I run sip show peer SIP_SER against one of the servers that is not functioning for…
Jay Ryan
  • 1
0
votes
0 answers

Can't configure Inbound/Outbound on Issabel

I have PRI line connected to E1 card. I installed Issabel with Asterisk11 on it. Whenever I try to call the number given to us by the PRI provider, I find the number busy. I tried to configure inbound route to match all DID and CID and route the…
idk
  • 1
0
votes
0 answers

Postfix using asterisk user to send emails. Mapping did not work, are there any other options?

I am trying to set up a PBX server to send voicemails to emails, which is being triggered properly, however, the from email is not the correct one I want to use. Instead of using support@mydomain.com, it's using asterisk@sip.domain.co.uk So far I…
Ceyntas
  • 1
0
votes
0 answers

Setup Kamailio to accept only register request with FQDN and block IP request

I have set up Kamailio and am quite new to the process, I'm able to register with FQDN and IP directly to Kamailio but my main setup is Kamailio as edge proxy to asterisk to allow register only with FQDN and not IP, route { if…
Mohammad Sarshar
  • 1
0
votes
0 answers

CDR not working with asterisk & odbc

I had ODBC working with asterisk to write to my CDR then recently needed to check my CDR to realise nothing had been written for about 6 months. I had a look at my config, nothing seamed to have changed. but then tried to re-install everything…
Ben
  • 23
  • 4
0
votes
2 answers

Asterisk app_swift Cepstral audio problems

I have installed Asterisk onto OS X (http://www.mezzo.net/asterisk/index.html) with app_swift included as well as a fully licensed Cepstral voice and am getting very choppy audio when using the Swift command. My swift.conf is below and seems to be…
John Duff
  • 101
  • 2
0
votes
0 answers

the freepbx queue.log union in asterisk one directory?

how to merge free pbx queue.log into one directory of asterisk ? I tried to combine them but in myphpadmin queue_name comes queuenumber and I don't want that.
nika gubelidze
  • 1
0
votes
0 answers

Fault tolerant setup of Asterisk and FreePBX using a dedicated line

I am attempting to configure a fault tolerant setup of Asterisk with FreePBX. Going over the docs, I see I need to configure Pacemaker. Not a problem in itself as I've used PCS before. I have 2 servers running RedHat 8 and have Asterisk + FreePBX…
Hussain Akbar
  • 141
  • 1
  • 4
0
votes
0 answers

How can Asterisk voicemail handle call holding?

Given an Asterisk PBX system version 20 or above. Given that a user calls another user, the call is missed and is sent to voicemail. While the caller is directed to voicemail, the caller receives another call and answers it, putting the first call…
Charles Langlois
  • 111
  • 5
0
votes
0 answers

Atxfer does not transfer

I have two agent A and B. Agent A has a call online and wants to transfer the call to agent B. I already have axtfer => *2 configured in the features.conf file and I have alltransfer set to YES on my platform with Magnusbilling, but I still can't…
drop sip
  • 1
0
votes
1 answer

Issabel PBX with call-center addon crashes

I am using Issabel PBX with Asterisk 11.25.3 and Call center module (free edition). Every day once or more the system crashes. We mostly use outgoing calls. There are always 2 list. One with around 300 numbers and one with 5k numbers. The dialer.log…
DzoniPfc
  • 1
  • 1
0
votes
0 answers

Linksys SPA-504G with NAT: RTP stream problems

I've got an asterisk server out in the cloud, and a Linksys SPA-504G behind a nat in my home. My router port forwards TCP 5060-5063, and also UDP 10000-20000 (for RTP) to the sip phone. My phone is set for NAT Mapping Enable = yes and Nat Keep Alive…
Ed Greenberg
  • 306
  • 2
  • 6
0
votes
1 answer

Authentication error in trunking between two asterisk

i'm creating a connection between two asterisk servers. I want only make calls from right side server to left side server, but i receive an authentication error. I can´t find the problem. The Error SIP CONF EXTENSIONS CONF
Tomás Figueiredo
  • 3
  • 1
1 2 3
…
39 40