similar to: early-dial SIP 484 "incomplete address", dialplan patterns and international calls

Hi, I am testing some IP phones (eg. GXP2000) and noticed that the "early dial" feature works fine with Asterisk 1.4 but not with 1.2. "early dial" is when digits are sent immediately, one by one, and Asterisk replies with a "484 Address Incomplete" and waits for the next digit until a match is found. This is a very useful feature where no dial patterns have to be

I'm building a dialplan for use with a bunch of GXP2000 desk sets. During testing, we had some user issues surrounding the lack of an on-phone dialplan. Users would hit 9 and sit there waiting for a redial tone, and the GXP would time out, sending just '9' to *, which couldn't do much other than spit back a 404 or play pbx-invalid. I turned on the "early dial" option

Hi, I have a set of Grandstream GXW4008 (units of 8 FXS ATAs) and another set of Linksys SPA8000 (8 FXS ATAs). The GXW4008 has a "neat feature" called "early dial" which allows me to define a "dial pattern" as generic as {*X+,#+,X+} (or something similar; the idea is to "match all digits") and send those digits >>immediately<< as they are

Replying to myself, I've just read in 1.6.1 announcement that a new Incomplete dialplan application is the one that provides what I'm looking for ... 2008/10/3 Olivier <oza-4h07 at myamail.com> > Hi, > > If my memory serves me right, there was thread (in dev mailing list ?) > explaining how we could implement 2 stages dialing with SIP endpoints: > user dials 1234

Hi, If my memory serves me right, there was thread (in dev mailing list ?) explaining how we could implement 2 stages dialing with SIP endpoints: user dials 1234 then asterisk replies 484 Address Incomplete, then user dials 5678 then asterisk begins to treat extension 12345678 as if it had been dialed as a whole. With compliant hardphones, you could get you phone to display a short text invite

I am trying to allow users to place outgoing international calls from a GPX-2000 with "early dial" enabled, connected to Asterisk 1.2.12.1 I have the following extension line: exten => _011.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) When I attempt to place a call to a number in, for instance, Kenya, I dial "011254"...etc. and I get this on the asterisk console: Executing

Hi, We are processing lots of calls and I want to filter these that have incomplete numbers sent with a proper SIP response. These numbers are not in the local dialplan by themselves, so I'm trying to find a way to generate 484 "Address Incomplete" SIP response based on the length of the extension called. Congestion response is too lossy of the original cause and doesn't

Hi, I am searching for the most effective solution for the following scenario: Our users can call into our IVR menu and dial a specific extension and immediately hang up. This event should simply trigger Asterisk to make multiple simultaneous calls through a group of zap channels (5-10 calls). When the called parties answer, Asterisk should simply play a message and hangup. So I was thinking

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;

On Wed, Mar 5, 2014 at 1:55 AM, Vieri <rentorbuy at yahoo.com> wrote: > Sorry for top-posting but my webmail forces me to. Odd. It's been a while since I used Yahoo but I didn't think I had that issue. GMail does default to top-posting but clicking the ellipsis to look at the previous email is enough. > I added -W to the APPEND line as suggested but I'm still getting

I'm new to asterisk and need some help with getting a SIP connection working. I am trying to establish a termination point/DID number in another country. I am currently running Asterisk CVS-HEAD. My foreign provider uses SIP and authenticates via IP address. I am not required to register my SIP connection in order to send or receive calls. Can someone help me with how to understand the

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

I am trying to retrieve the "dialed peer number" but it seems that ${DIALEDPEERNUMBER} is "broken". Also, I know that I could extract the dialed number from the ${CHANNEL} variable but this only works for SIP and maybe IAX (untested). However, it doesn't work for ZAP. All I get when using ZAP is something like "Zap/1-1" (for SIP I would get

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

Sorry for the simple question. I'm trying to match "sipprovider.nocredit" but the following doesn't execute NoOp (it runs "context" but not "context-custom"). What am I doing wrong? [context] include => context-custom exten => _.,1,Set(GROUP()=1) exten => _.,n,Goto(destcontext,${EXTEN},1) [context-custom] exten => sipprovider.nocredit,1,NoOp(No

Hello everybody, i'm from Mexico, at the time i?m working on a production server with asterisk 1.2.25 + spandsp-0.0.4 + libmfcr2-0.0.3+libsupertone-0.0.2+libunicall-0.0.3 and zaptel-1.2.22. I installed this version of astunicall that i downloaded from http://www.moythreads.com/astunicall/ Everything works fine, i'm able to make outgoing calls and recive incoming calls with all ANI 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 just found an old bug report at bugs.digium.com with exactly the same problem. It's really too bad this bug wasn't addressed: http://bugs.digium.com/view.php?id=7315 --- On Fri, 1/9/09, Vieri <rentorbuy at yahoo.com> wrote: > I'm trying to figure out how to reload iax2 (without > breaking existing calls) so it can listen on a new IP > address (like "ip addr

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