similar to: outbound SIP trunk hunting (or any fxo for that matter)

I've got 4 SIP phone lines with a call-limit of 2 for each. I've written a handy macro to allow my users to dial a phone number and the macro will figure out the next available line to use by first checking if the GROUP() is over 2 and then checking to see if ChanIsAvail() as a backup, and if it can't use the line for either reason it goes to the next line. The problem is that there

I'm trying to move my Asterisk deployments under a Virtual IP address and now remember why I dislike this. My primary Asterisk system is now behind a firewall in private address space. My question is what ports are needed to be opened just for the purpose of placing outgoing calls. I would have assumed none, but I can't even get replies on registration from any of my 3 VoIP providers.

Has anyone else had issues with Originate returning the wrong error code? According to the docs, the following errors are supposed to be returned: 0 = no such extension or number 1 = no answer 4 = answered 8 = congested or not available Now in Asterisk 1.4.23 I get some error code 5's but since they're so few I tend not to worry. But what is concerning is the number of Error 0's I

I'm trying to figure out how to listen in to a channel that I specify. I have the impression I've seen this done via Flash web controls, but I'm trying to write something myself and I can't figure out what command would be used. ChanSpy looks great, but I don't see how to specify the channel. I have a channel identifier like "SIP/provider-08748db0" which is what I

Hi, I have been headbanging with asterisk and Jack for a while, decited to ask other linuxists for an advice. The problem is that Jack is compiled from source (0.118) in /usr/local/, but menuselect says "XXX" for it (cannot enable it). I need jack... Otherwise I will inotify Monitor WAVs, what is bad :`( After installing jack from sources: Add system-wide

Hello, I need to capture calee's audio in real-time in order to capture operator messages (I've written sound recognition software that works with Jack: http://github.com/Motiejus/SoundPatty/). Jack does the following: Incoming call audio -> audio in to jack, audio out from jack -> current Asterisk application Outgoing call audio <- current Asterisk application However, I need

Hi, List, I am looking for a cheapest (and therefore most funny) way to attach GSM card to my asterisk home box. Needed features: Calls+SMS in/out one or two SIM cards (ports) Should I try looking for a GSM PCI card that is compatible with linux/asterisk, or GSM USB card, or modern full-blown SIP GSM gateway (with ethernet)? Maybe an ordinary cell phone with USB interface and mangling with

I've set up plenty of Asterisk boxes but never one that had to deal with a proxy server to be able to use a line. Using "X-Lite" I have no issue with settings as follows: Display Name: Any Name User name: 00575000010XXXX Password: 00575000010XXXX Authorization user name: <blank> Domain: directnationalloan.com Checked "Register with domain" and "Send outbound

Greetings: I did some quick searching of my history of this list, and I tried a quick Google search as well, but perhaps someone on the list can quickly answer this question. I have a very nicely working Asterisk system at home with two Digium X100P FXO cards. When my SIP phones want to dial-out I have them setup to grab the first analog card (Zap/1) with the following extensions.conf segment:

Hi All, I want to track a call that is originated using originate AMI command through AstManProxy server. I m using AstManProxy server and I developed an AstManProxy client. By using my AstManClient program I can able to login AstManProxy server. Now I can able to issue/send originate command to generate a call but I m very confuse that I cannot able to track my call. The AMI events were

Hi List, Our company has several small distributed offices we would like to inter-connect with bridged VPN a single subnet (last example in http://www.shorewall.net/OPENVPN.html). We have SIP phones in every office (up to 5) so we can use SIP without any NATing and securely. Max theoretical simultaneous calls possible ~30, but we have ~5-10 @ regular basis. OpenVPN server would be in the same

Hello there, I have successfully installed and configured asterisk for use as an office PBX using SIP trucks and Voip handsets (using g.729 codec) which works great. Now I wish to try and configure asterisk to do a HTTP request and submit callerID to an external website when a call is missed. eg Someone calls PBX and rings extension 100 -> Call is not answered -> HTTP request is initiated

Sorry if this appears twice - I originally sent it nearly 18 hours ago and never saw it .. I have a need to have a unique integer number that can be used by a dynamic meetme room (I am wanting to redirect a call into a meeting room, and need a unique number to make sure I don't put two people together !) I was going to use a global variable ${NEXTMEETME}, and add one every time I

Ok, here's the challenge: I would like to be able to find, match - and then react - upon prompts that are presented by the outbound/remote side of a call. Think mobile phone and "This user is temporarily unavailable". Collecting a limited number of known prompt snippets should not be a problem, but how would you then detect their presence in a longer recording (or live audio

Hi All, I defined 3 trunks for a client: [trunk-100] ... [trunk-101] ... [trunk-102] ... How can i do a trunk hunting ability like this: [dial-out] exten => _1.,1,Dial(SIP/${EXTEN}@trunk-100) if busy try: exten => _1.,1,Dial(SIP/${EXTEN}@trunk-102) if busy try: exten => _1.,1,Dial(SIP/${EXTEN}@trunk-103) How can i make it dynamic,coz different clients will have different number of

I'm running Asterisk without FreePBX or any of the other managers. I'm trying to figure out how to set the maximum number of channels allowed on a single line? I'd just rather not have Asterisk try the line when I know I'll recieve a CONGESTION back from the SIP phone provider (ViaTalk in this case). Is there a configuration option I can't find that sets the maximum number

I've got a very nice PHP AGI script but I want to be able to do some database cleanup when the user hangs up the phone. I wish everyone would hang up when they were suposed to, but some people don't. So what does Asterisk send to an AGI file when the line has been disconnected? If I remember reading somewhere correctly, I don't need to use DeadAGI. Instead I'm able to use

Hello asterisk users! I, like many people, have a cell phone. I also have some SIP phone devices (software and hardware). I'd like to have one number that rings all my phones and routes the call to wherever I pick up. However, my cell phone has its own call forwarding voicemail. I can't just turn that off, because then direct-to-cell calls wouldn't ever get to voicemail - that

I have 10 SIP trunks that I'd really like to round-robin load balance. Currently I have a macro that switches between available lines, but there really must be a function in Asterisk to do this on its own. So my question is just that, are there any easy ways for Asterisk to either balance between SIP trunks or even just a built in function to find the next available SIP trunk. I think using

I've heard about this, but I really can't seem to find anything on it. I've got a strange setup that exists only because of firewall issues, and everything about it seems fine. The setup: SIP clients -> Asterisk (office) -> IAX -> Asterisk (colocation) -> SIP PSTN Termination All the extensions I want to be able to dial are on the colocation box. What I'd really