similar to: simple dialplan question

Hi. I am trying to pass a variable from one Asterisk PBX to another. I'm using DUNDi with IAX2. Is there a way to do it? I tried the following but it fails. On peer1: [dundi-outgoing] switch => DUNDI/priv exten => s,1,Set(CDR(userfield)=test) exten => s,2,Set(DUNDIVAR=${ARG1}#TEST) exten => s,3,NoOp(Passing ${DUNDIVAR} to DUNDi peer.) exten => s,4,Goto(${DUNDIVAR},1) On

First off, here is what I want to do: SIP Clients -> SER -> Asterisk -> VoIP provider Where SER will handle communications between SIP clients (since I would prefer that my SIP clients not use all of my bandwidth) Asterisk will handle calls to a VoIP provider I have read that people have similar setups working, but I have not seen any documentation of these setups. So far, SIP Clients

Hi, How can I match an extension "ending with 3" (just an example but applicable to any other digit, including * or #)? exten => _ZX.3,n,... exten => _ZX.#,n,... (the above does not work) Can regular expressions be used in the standard dialplan (end with: "$")? Thanks, Vieri

Hi, I'm trying to set a rule to dial out through multiple Zap groups so that, say, g0 is the cheaper POTS lines group and must be used first. However, if g0 is busy or disconnected then try dialing out g1. My g0 group is made up of 4 analog lines connected to a 4-FXO card. I disconnected the RJ-11 wires from the FXO card to simulate a line disconnection. So theoretically all calls should

Sorry for top-posting but my webmail forces me to. I added -W to the APPEND line as suggested but I'm still getting the same result: Booting... Altiris, inc. X86PC PreBoot, PXE-2.x Enhanced Build ID=402 PXEPreZero: Invalid PXE Server list format. and the client PC freezes right there. Here's the full content of my dhcp.conf: max-lease-time 86400; ddns-update-style interim;

Hi, I would like to know if one can set the outgoing caller ID within Asterisk when calls are going out through: 1) an analog POTS line (I suppose not) 2) a telco BRI line (I don't think so) 3) a telco PRI line (maybe) 4) a voip provider (surely) Thanks, Vieri ____________________________________________________________________________________ Moody friends. Drama queens. Your

Hi, I am diverting an incoming call to a mobile phone and a landline using the following:- exten => 0203000000,3,Dial(SIP/442080000000 at sipprovider&SIP/44700000000 at sipprovider,120,r) For billing purposes, i need to be able to work out whether the diverted call was answered by the mobile or whether it was answered by the landline. The CDR only shows the full Dial() information, and

Hi, Suppose you have 2 identical Asterisk servers and 1 alias IP address that you assign to either one, according to system failures, etc. Also suppose that all SIP clients register requests go to the alias IP address. Imagine server1 fails and server2 gets the alias IP address. Correct me if I'm wrong but I would have to wait at least 60 seconds before most SIP clients re-register to

I am trying to find out the best way to replace one of our hardware PBXs. It currently has 200+ analog phones connected to it. The idea is to take advantage of the already installed phone cables (big building) so I'm trying to avoid the use of ethernet adapters (if possible). However, I'm realizing that it's an expensive setup and will definitely require two or more cooperating

I realize this may be slightly off-topic but I'm wondering if someone here can lend me a hand. One of my GXW4008 has gone "unconfigurable" via standard HTTP (refuses connection) and I can't use the built-in IVR because I had previously disabled the "keypad update" feature. So I'm stuck with just telnet, the reset button and RS-232. Telnet commands are very limited

Hi, I'm experiencing "frequent" kernel panics on a system with Asterisk 1.4.26.3. There is no core dump, "just" a kernel panic. This is the only data I could copy from the screen: EIP: 0060: [<f8e248b4>] Tainted: P VLI EFLAGS: 00210297 (2.6.23-gentoo-r8 #1) eax: 00000130 ebx: 00000000 ecx: 00220028 edx: 00000978 esi: 346e5802 edi: 00000000 ebp: c3b45500 esp:

Hi, Before I report a bug on http://bugs.digium.com, I would like to know if someone is seeing the same error message. Personally I am not using wctdm24xxp but other modules such as wcte12xp and wctdm. The latter modules load fine and are compiled with pci_register_driver as expected. The only module that seems to require the deprecated function pci_module_init is wctdm24xxp. Is this normal?

Hi, I have a Linux server at ip address 10.215.144.7 running DHCP, TFTP and syslinux. DHCP config contains the following: next-server 10.215.144.7; filename "/pxe/syslinux/pxelinux.0"; and the 'default' pxelinux.cfg contains: LABEL altiris ??? MENU LABEL ^7. Altiris ??? COM32 pxechn.c32 ??? APPEND 10.215.144.60::/BStrap/x86pc/BStrap.0 When a PXE client boots in my network

I setup two asterisk servers with identical settings (same extensions, same queues, etc). Each one is connected to the same amount of incoming/outgoing links (1 PRI, 4 BRI, 1 IAX friend, etc, on each box). Most extensions are sip and they register via DNS SRV and other methods so that the two servers are load balanced. Incoming PSTN calls (BRI) reach 50% each server so that's load balanced

Hi, I have been trying to configure my Asterisk to use a Sip provider for out and incoming calls. I only have one user and password for connect to my sip provider. My sip.conf is: [general] ;disallow=gsm ;allow=ulaw port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls callerid=No

Hi, Is Asterisk "fully QSIG-compliant"? I currently have an Alcatel 4400 connected to Asterisk 1.2 and 1.4. Zaptel versions are 1.2.26 and 1.4.11. I am using switchtype=euroisdn and all works fine. However, it seems that Alcatel's latest firmware has dropped support for euroisdn which is really despicable. So now I need to see if I can migrate to QSIG which is supported by

Hi, I'm trying to integrate Skype and Asterisk but I'm only interested in these 2 things: 1) allow any Asterisk SIP extension to call any Skype "user". I do not need to call landlines via Skype. 2) allow Internet Skype "users" to call my Asterisk PBX Skype "user" and route the call to a specific Asterisk SIP extension. At first, I thought it would be

Hi, I'm experiencing a few (but meaningful) cases of audio distortion (or bad quality). I can't say yet how often this happens. Please listen to the following sound file: http://213.96.91.201/temp/distorted_audio_1.wav This was recorded by Asterisk while the local SIP caller was dialing out a SIP trunk (so the problem is on my side, definitely, and it doesn't seem to be related to

Hi, I'm trying to find a way to implement the following: I have 1 media source (IceS or MPD) and 1 Icecast stream (say, LAN radio). Once in a while I'd like this stream to be interrupted by short announcements (PA system). Input for these announcements can be from another source (IceS, MPD, Asterisk call). Anyway, to make things simple: I'd have dir1 with ogg music files for

Hi, I have an audio quality problem regarding IAX2. I have 2 Asterisk servers interconnected via 2 LAN trunks at 1Gbps (no nat, no firewall). One trunk is SIP and the other IAX2. Normally, I use IAX2 but have noticed easily reproducible audio quality problems (voice in/out is OK but there's a "third" noise overlapping with a "scratchy sound" as if it were some kind of