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

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129 views

Can a “4 line” Grandstream GXP 2000 handle 8 lines?

We have a whole bunch of GXP 2000 phones here but now we're growing too big for just 4 incoming lines. We use Trixbox (asterisk) for the phone system with 4 analog lines coming into a Dell server. ...
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0answers
497 views

Lync 2010, Kamailio, & Trixbox 2.6.23 (Asterisk 1.4)

I'm having an issue trying to connect Lync 2010 phone calls with our trixbox PBX. I've gotten to the point where Kamailio seems to be functioning properly and acting as a bridge between TCP traffic ...
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2answers
75 views

Asterisk server in Debian linux

I m tried to download sip.conf files in asterisk server with ssh it replies sip.conf is not a regular file,directory or symlink?
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1answer
366 views

asterisk 1.8 poor playback quality

i've just installed an asterisk 1.8 IVR using playback() for playing sln files. i've we have some high quality wav files, that i've converted with sox to sln 16KHz files. (Why aren't higher sampling ...
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1answer
720 views

Asterisk / Elastix Address Book and Call Recording

I am trying to build a small asterisk based PBX using Elastix. It'll be having 4 FXO (2 nos.no ISDN, normal analog POTS, 2 nos. GSM connections using a GSM Terminal) and 4 FXS (2 IP Phones and 2 ...
3
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1answer
879 views

Make call with alternate provider if NOANSWER

I have two voip providers, one free an the other paid. The free provider only allows local calls to certain area codes, so I need to fall back to the the paid provider if a call fails. At the moment, ...
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4answers
12k views

How to stop registration attempts on Asterisk

The main question: My Asterisk logs are littered with messages like these: [2012-05-29 15:53:49] NOTICE[5578] chan_sip.c: Registration from '<sip:912@xx.xx.xx.xx>' failed for '37.75.210.177' - ...
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1answer
7k views

Asterisk SIP/2.0 401 Unauthorized

I'm running into a funny little issue with Asterisk 10.3, but it seems to be applicable to 10.4 as well. The server running Asterisk was relocated from a VPS to dedicated hardware, and now only 1 of ...
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3answers
973 views

Convert a WAV (Asterisk recordings) to MP3

I have a large collection of WAVs. They are: RIFF, little-endian WAVE audio GSM 6.10 Mono 8000Hz effective bitrate of 13.6 I need to convert these to MP3. Current I am forced to send the file ...
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1answer
67 views

Can I include the same file multiple times in Asterisk?

I'm new to Asterisk, and I, would like to include a file that stores all of the common settings for my SIP phones to avoid duplication. Can I use #include more than once in sip.conf to accomplish ...
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3answers
689 views

Is It a Bad Idea to Install the OS GUI on a Server That Will Host Asterisk?

We have a server onto which we plan to install CentOS 6.2. The server will host Asterisk for our office, but is not a webserver. If we install the minimal installation for CentOS - that is, no GUI ...
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2answers
3k views

Asterisk: Authenticated SIP trunk

Let's say I have an Asterisk system with a bunch of connections: there are phones (who register itself with *) and providers (who wish to establish SIP trunks to put a lot of calls over, with ...
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1answer
852 views

Permission Denied trying to execute a .conf file.

I'm using a virtual machine in Windows 7 and I have installed Linux on it and Asterisk software. I want to configure the sip file to create sip user but when I type ./sip.conf in Linux it replies ...
2
votes
3answers
496 views

How to add more incoming/outgoing lines capability to Asterisk?

We have 4 people call us from around the country at once and it causes our phones to be busy. We want to be able to have at least 8 people call us at once without anyone getting the busy signal. ...
2
votes
1answer
144 views

Transfer to Voicemail only when call not answered, not on rejecting a call (Asterisk)

I am working with Asterisk now and observed call transfers to voicemail even though i have rejected it. Is any way to transfer call to voicemail only when on not answered or not available ?
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1answer
266 views

Get Asterisk/Freepbx to ignore digits that were unnecessarily dialed (area code for local calls)

This is the second part of this question, worded better. Remove local area code on outbound calls, and remove "dial 1" for long distance in Trixbox/Asterisk I have removed the NEED to ...
3
votes
2answers
644 views

Server Freeze Up Under Load

I'm having a problem with a debian server that I thought was due to bad RAM, but is persisting. It's a Dell Poweredge 6800 with two dual-core 3.6GHZ Xeon processors and 5GB of DDR2 ECC 333. I've ...
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1answer
161 views

How to add a user who is out of state as if they were just another extension AT our office? [closed]

We want to add a person who is in another state to our phone system as though they were simply in another cubicle here at the office. So, they can dial "302" and get the office assistant's extension ...
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1answer
780 views

How to play and record wav in Asterisk

Please, how could I do these things on asterisk: 1) Externally (with some API) make a call from asterisk and play wav and hangup the call 2) How should I configure the extension/route/whatever on the ...
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1answer
65 views

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, ...
4
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3answers
7k views

How to configure Asterisk to send audio before call is established

Is it possible to configure Asterisk so that it sends RTP packets with audio from the receiver before the remote party actually picks up? This seems to be required for a VoIP compliance test my ...
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1answer
421 views

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 ...
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1answer
483 views

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 ...
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1answer
581 views

Extension not found error on calling an extension

I have setup a SIP server and connected two polycom soundstation duo devices to it. They both register line with the SIP server fine but when I try to place a call from one device to the other, ...
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1answer
474 views

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 ...
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1answer
528 views

How do I get the caller name in Asterisk using Python?

I have an Asterisk (Elastix) installation. I have a MySQL database with about 10,000 customer names and numbers. I would like my call agent to recognize the name associated with the number. I can have ...
2
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1answer
1k views

How to choose an open source, Asterisk friendly firewall?

I'm in pain. We are moving to a SIP based VOIP system and for whatever reason, we could not get our hosted Asterisk solution to work with our Sonicwall. Our VOIP provider gave up and is recommending ...
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1answer
684 views

IM and File Transfers Integrated With VoIP

I have an Asterisk box serving an office of people. I'd like to provide instant messaging and file sharing capabilities alongside the voice and video capabilities provided by Asterisk (ala Windows ...
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2answers
692 views

How do I detect call forwarding in Asterisk?

Basically, I want to do the same thing that Google Voice does. I forward my calls to a DID number that rings my Asterisk server via IAX2, which, if it detects the call has been forwarded, sends it to ...
2
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1answer
6k views

Skype Connect as SIP/Trunk for Asterisk

First off: I'm not sure if this should be on superuser or here. I have recently built a few Asterisk boxes with OpenVOX FXO/FXS ports little or no trouble. My current project is building an ...
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2answers
209 views

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 ...
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2answers
2k views

Asterisk 15-minute drops calls

There seems to be a hard limit on incoming calls set a certain way- a telephone engineer told me about this, but I don't know the specifics. When on a call, the moment the call hits 15 minutes it ...
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3answers
1k views

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?
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2answers
256 views

How do you set up speed dial on Polycom IP6000 phones?

Basically I would like speed dial on this so I can press 1 and it will dial 101 AND I can press 2 and it will dial 555-555-5555. Is there a way to do this?
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2answers
1k views

Remove local area code on outbound calls, and remove “dial 1” for long distance in Trixbox/Asterisk

We have Trixbox CE 2.8 and Asterisk 1.6. I want to remove the 933 area code so you can just dial 555-5555 instead of 933-555-5555 for local calls. I also want to remove the +1 for long distance so ...
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2answers
897 views

FreePBX How to play an announcement for misdialled calls

Our Asterisk-based PBX (FreePBX "Tango" distro) does not require any prefix for outgoing calls. The system is able to recognise external numbers and route them appropriately. Some of our users who ...
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1answer
91 views

Is a PBX responsible for routing calls, or is that handled by the provider?

I am setting up a PBX that external users will connect to and I was wondering how important PBX placement was. Are calls routed through the PBX itself, or do they go directly through the provider ...
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1answer
219 views

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 ...
3
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1answer
718 views

Automated reports with FreePBX 2.8.1.4

FreePBX has a 'Reports' tab that allows you to create a few reports and some of them you can export as a .csv or .pdf, but it doesn't really have any finer features than that. I would ultimately like ...
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1answer
1k views

GUI for Asterisk 1.8.8 on Debian 6 [closed]

Can someone point me towards a tutorial for installing either FreePBX, or a Webmin GUI for Asterisk 1.8.8 on Debian 6? Asterisk Thirdlane Multi Tenant PBX Manager and Asterisk Thirdlane PBX Manager ...
0
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1answer
660 views

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 ...
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4answers
5k views

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 ...
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0answers
331 views

Asterisk 1.8, Dedicated SDSL but SIP Provider Inbound Channel heavily delayed?

We have a dedicated Asterisk 1.8 server running as a front end SIP server for authenticated and internal users only, using SDSL 2MB/2MB, in the UK. The provider of our internet also is the provider of ...
0
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2answers
4k views

Asterisk block incoming call with certain Caller IDs

So I know this is a fairly simple question, but I can't seem to find a good answer to it. I have a DID which is getting hit fairly hard (or at least for me, about 1-3 calls per day) with telemarketers ...
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1answer
380 views

Asterisk IVR to Extension slow

I have a IVR set to pickup our PSTN lines and tell the user to enter the extension OR to press 1 for a representative. If the person enters an extension it transfers within 1 second , but if they ...
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2answers
2k views

Can you please explain the difference between PBX and soft switch in less technical terms?

Asterisk is an open source PBX and FreeSWITCH is an open source soft switch. Source: http://www.freeswitch.org/node/117 Can you please explain the difference between PBX and soft switch in less ...
4
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1answer
2k views

Asterisk isn't playing recording attachted to IVR

I've setup an IVR to pickup up incoming calls to allow them to enter an extension or press one to be dropped into the ring group for our office. The only issue is Asterisk isn't playing the ...
5
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1answer
895 views

Elastix(Asterisk) Slow to repsond on incoming PSTN line

I setup a PBX with asterisk and Elastix as a web GUI. I have all incoming calls set to ring to a ring group of our office personnel in a ringall format. The only issue is the time for Asterisk to ...
0
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1answer
512 views

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 ...
0
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1answer
4k views

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 ...