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

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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, ...
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2answers
4k 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
201 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
356 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
358 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
301 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
295 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 ...
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1answer
687 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
334 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
360 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 ...
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1answer
4k 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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164 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
911 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
774 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
167 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
906 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
420 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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83 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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125 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 ...
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1answer
323 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
809 views

GUI for Asterisk 1.8.8 on Debian 6

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 ...
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460 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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3k 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
290 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 ...
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2k 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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247 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 ...
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1answer
944 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 ...
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1answer
629 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 ...
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1answer
377 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 ...
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2k 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 ...
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335 views

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 ...
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1answer
94 views

Asterisk - which protocol/technology is using? [closed]

Asterisk installed on Linux server by other admin. I am trying to find out which communication protocol/technology is currently using on Asterisk? Eg: isdn, voip, acd, etc. I dont know which one is ...
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1answer
679 views

Using Asterisk and FreePBX how can I map extensions to outbound routes

I have a Trixbox server (Asterisk and FreePBX) that has multiple tenants on it. I need these tenants calls to go out via different outbound routes in order to split the bills at the SIP trunk provider ...
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1answer
820 views

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 ...
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132 views

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 ...
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2answers
315 views

Asterisk and sip station codecs

we have a small instance running on ec2 of voxilla/FreePBX-Asterisk-1.6.0.5-i386.manifest which has been updated as far as i can see We signed up for sipstation (because it removes half the ...
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24 views

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
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1answer
203 views

how is the voip network working when server is outside nat and clients are inside nat?

I want to install an asterisk on a server with a public ip. The sip clients are behind different NATs. How is the signaling and the rtp packets (the voice) working in this case? Will rtp be peer to ...
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2answers
108 views

how to associate an ip with a name?

I have installed an asterisk on my private network(so it s behind a router ). Is there anu way to set the asterisk on a public ip though it's installed in a server with private ip? As Ive read it ...
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2answers
273 views

Setting Polycom device.net options in the config file

I have several hundred Polycom SoundPoint IP phones and I need to make a config change to them. I would like to effect this config change through the XML config files that are loaded when the phone ...
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2answers
2k views

How can I configure my asterisk to work with meetme

I try to use commend MeetMe(...), but i get from the CLI this response "pbx.c:3695 pbx_extension_helper: No application 'MeetMe' for extension (test_main, 600, 1)" I try to use ztdummy but i can not ...
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0answers
212 views

Is it possible to split Asterisk SIP signaling and voice traffic?

How do I configure Asterisk to send SIP signaling to one IP address and the RTSP stream, voice traffic, to another address?
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2answers
3k views

Asterisk - inbound calls from SIP DID trunk “rejected because extension not found”

I have a DID with DIDforSale that's pointed at my Asterisk server. When I call it from my landline phone I get the AT&T disconnected line recording. The Asterisk CLI shows the error message: [Oct ...
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2answers
805 views

Asterisk DID incoming rules =s

So I have a 10 way SIP trunk from a UK provider. Calls in and out work fine, but as I have multiple numbers for different purposes I set up the inbound rules for each number but they never work, only ...
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236 views

Uninstalled asterisk, still gets email about error logs

/etc/cron.daily/logrotate: error: error accessing /var/log/asterisk: No such file or directory error: asterisk:1 glob failed for /var/log/asterisk/*_log error: found error in /var/log/asterisk/debug ...
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3answers
211 views

Share phones but not extensions in Asterisk

I'm setting up a new Asterisk PBX, and I'm on the process of buying IP phones. I plan to add phones as needed, as we can't afford to buy them all at once. However, I do want to create a single ...
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1answer
465 views

match 2 numbers in extensions at asterisks dialplan

is there a simple way to match something like this: exten => _+380XXXXXXXXX,1,NoOp(hint that) exten => _0XXXXXXXXX,1,NoOp(hint that) this two extensions matches same number with or without ...
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1answer
385 views

SIP ( Asterisk ) CoS solutions with no VLAN's?

I have a Asterisk box actually two running pure asterisk. No GUI i.e. FreePBX, TL etc. The asterisk box is connected to a an EX4200 VC Stack, the EX4200's then connect to our headquarters via a Qwest ...
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2answers
170 views

Calling out from Asterisk Server

I have an Asterisk server connected with a Legacy PBX. This setup is connected to another Asterisk Server (In another country) over the Internet with an IAX2 Trunk. Inter Office Extensions are ...