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
1 answer

How to install app_echo.so in FreePBX?

What is the proper way of installing app_echo module when FreePBX is installed ? I cannot find it in Module admin and manual editing of configuration files when freepbx presented is not recommended, is it ? Thanks.
John
  • 1,493
  • 3
  • 26
  • 46
0
votes
1 answer

Asterisk Dial Plan - redial with 44 or 0 (UK numbers)

Have an issue with the redial numbers. Outgoing line is configured to the digit 0, so on the dial plan it is set to "0|." (without the quotes). Now all missed calls from the UK, starts with 44 on the IP Phone. If i hit redial, it refuses the…
Cold T
  • 2,401
  • 2
  • 17
  • 29
0
votes
1 answer

asterisk far end nat traversal not working

We use a obfuscation software to encrypt/mangle both SIP/RTP which sits before asterisk. What happens is sometimes we don't get any voice. after some "rtp set debug" we found out that when received ip of the rtp stream is router's public ip,…
Aftnix
  • 233
  • 2
  • 8
0
votes
1 answer

Voip, connect to asterisk servers over WAN with A580 base station

I have LAN infrastructure with asterisk and SIP phones(Gigaset A580) in it, I have done port forwarding on my public static ip address to my asterisk server in my LAN network, I also have trunk which I have rented so that phones from public PBX can…
Adin M
  • 1
0
votes
2 answers

Asterisk using 100% of the cpu and freezing my debian host

I have a Debian host $ uname -a Linux voip 2.6.25-2-amd64 #1 SMP Mon Jul 14 11:05:23 UTC 2008 x86_64 GNU/Linux Old asterisk asterisk 1:1.4.21.2~dfsg-3 It has been working for a long-long time $uptime 13:50:37 up 1047 days, 21:02, …
Korjavin Ivan
  • 2,250
  • 2
  • 26
  • 41
0
votes
3 answers

Does the Cisco SPA range of phones work with CallManager/CME?

I have about 15 phones and am looking to upgrade from asterisk to an all Cisco solution. Do we need to replace the phones as well?
chrism2671
  • 2,579
  • 9
  • 34
  • 45
0
votes
1 answer

Automatically log into SIP phone when user logs into Windows

We run an asterisk-based PBX, with FreePBX front end, configured in 'device and user mode' to support hot desking users. That is, a user arrives and sits at any desk, keys *11 into the phone (Cisco 7900 series) and logs in with their extension…
Tim Long
  • 1,738
  • 1
  • 21
  • 41
0
votes
1 answer

Sendmail does not allow some Asterisk voicemail notifications to be sent

I have a large asterisk/freepbx installation where some users are unable to leave a voicemail, and have the voicemail sent via email to the callee. In maillog, here's what I see: maillog:Jan 18 15:43:33 telcosrv01 sendmail[21439]: q0ILhXBg021439:…
tsz
  • 229
  • 1
  • 2
  • 9
0
votes
4 answers

Asterisk + SIP 404 not found

I want to make a small Asterisk server in my house. I installed asterisk on my Ubuntu and I use 2 computers, in order to connect to one another. when I connect I can see in Wireshark that registrar ok. here is the output of sip show peers…
Uriel Frankel
  • 103
  • 1
  • 1
  • 4
0
votes
1 answer

Are there parallels to Asterisk AMI and Asterisk AGI in FreeSWITCH?

Asterisk has Asterisk Manager Interface (AMI) and Asterisk Gateway Interface (AGI), using which one can trigger PHP scripts at certain events from Asterisk. Using the same PHP scripts can also instruct Asterisk what to do next to a call/conference.…
jeff musk
  • 247
  • 2
  • 11
0
votes
1 answer

Asterisk configuration for voicemail emails through GMail SMTP server not working

I am trying to configure Asterisk (running on AstLinux) to send emails when it receives voicemails through GMail's SMTP server, but it is not working. I do not receive any emails, and I get the following error in the Asterisk system log: Dec 14…
nickb
  • 153
  • 1
  • 9
0
votes
1 answer

how UAC should handle SIP 183 Session Progress

I have the following scenario: 2 UAC are trying to talk, via a remote SIP server (openSER/Kamailio 3.1.3) = client infrastructure. The UAC software was developed over a local test infrastructure using Asterisk, where it was possible to establish a…
hovanessyan
  • 125
  • 6
0
votes
1 answer

Sip configuration for Elastix, for Terrasip (Provider) with Multiple DIDs

I have just purchased several DIDs with Terrasip I have configured my Elastix (freepbx) SIP trunk according to their suggestion: Please follow this template configuration. (valid for outbound/inbound traffic) Outgoing trunk name…
abutbul
  • 71
  • 6
0
votes
1 answer

AsteriskNow Migration / Shared Extension Space

I am testing the possibility of migrating from an old Avaya phone system to AsteriskNow. The migration would cover several hundred phones--but spread out over several years. (Management wants to move buildings to the new phone system one by one as…
Aaron C. de Bruyn
  • 588
  • 10
  • 30
0
votes
1 answer

Does ENUM make a measurable difference in percentage of calls routed via IP vs PSTN?

I'm considering configuring this, but don't want to waste time if there isn't a significant cost saving to be made. Any advice would also be much appreciated. Thanks, Chris
chrism2671
  • 2,579
  • 9
  • 34
  • 45
1 2 3
…
39 40