similar to: Please help parse this GotoIf line

I set up a context to allow me to call in to my * server (via Teliax in this case using IAX2) from my cellphone, and let me do a number of things, including dial other extensions, AND dial outbound again so callers could see my proper work CallerID when I use this service. Is there a way to be able to transfer calls to other extensions of my asterisk server FROM the cell phone/ This isn't a

Hello Instead of using PrivacyManager, I'd rather use my own dialplan to prompt the user for a ten-digit number if they called while blocking CID. This code does prompt the user, but 1) hangs up if the user didn't type the ten digits before the timeout 2) if the user did type the right number of digits, it still hangs up instead of Returning and then jumping forth to the "cid"

I set up a context to allow me to call in to my * server (via Teliax in this case using IAX2) from my cellphone, and let me do a number of things, including dial other extensions, AND dial outbound again so callers could see my proper work CallerID when I use this service. Is there a way to be able to transfer calls to other extensions of my asterisk server FROM the cell phone/ This isn't a

I have installed asterisk @ home 2.6. I am using a Telasip VOIP account. When I make outbound or inbound calls the calls seem to connect and then get hung up. I was wondering if there was something that I am misisng. I have tried several different sip.conf configurations. Here is what they are currently. telasip-gw canreinvite=yes context=telasip-in dtmfmode=rfc2833 fromuser=jrasxxx

Hi, why would Asterisk core dump with the following test dialplan extension ? exten => 8100,1,Answer() exten => 8100,n,Set(CALLERID(all)="") exten => 8100,n,PrivacyManager() exten => 8100,n,GotoIf(${[${PRIVACYMGRSTATUS} = FAILED]}?:nocid) exten => 8100,n,NoOp(Number is ${CALLERID(num)}) exten => 8100,n,Hangup() exten => 8100,n(nocid),Playback(vm-goodbye) exten

Hi all, I've just setup a sip line with Telasip and when they route the calls to my asterisk box, they include an extension along with the context that is defined in sip.conf for that DID. At first, I couldn't figure why they were getting 404 error from my asterisk box, but then figured out that they are sending the call to an extension that matches my number with them, in the

Any recommendations for a VoIP provider in the UK? I have a few guys in a field office in the UK with SIP phones and a VPN tunnel back to a working Asterisk setup in the US. The Asterisk setup has an IAX trunk with TelaSIP/VoipXpress with local DiD's for US offices, so they can call vendors, customers etc in the US at local rates. I'd like to get the same thing for the UK, so that UK

Probably just a simple syntax issue, but does anyone know how to dial a number and the once phone has been answered, play DTMF tones and then disconnect. I am trying to use this for page notification. Ive been trying the following string with out luck: exten => s,2,Dial(SIP/TelaSip-gw4/5198881212|D(12345678) Any help would be greatly appreciated! -------------- next part -------------- An

I have downloaded the source (wine-20011004), compiled and installed with ~/wine-20011004/tools/install.sh and am attempting to start running it. My first try, as suggested in the docs, is /windows/sol.exe from a terminal in KDE and the error message I get is: Gtk-WARNING **: cannot open display: I compiled and installed as root - that may have been a mistake. However, I have a /root/.wine

I am having trouble understanding how to use this. I want to transfer certain incoming calls from an IAX ITSP based on caller ID. From what I can make of the docs, I thought I need to do something like this... exten => _NXXNXXXXXX,n(nocid),transfer(1000) exten => _NXXNXXXXXX,n,noop(boo,${TRANSFERSTATUS}) exten => _NXXNXXXXXX,n,hangup exten =>

I can't seem to get consistant outbound caller ID working correctly. I have set the fromuser and callerid field in my sip.conf for my TelaSIP peer, but half the time it shows up as "No Caller ID" on my cell phone, other times it shows it correctly. Using asterisk CVS. Any ideas? Doug

It _appears_ that the only way to create valid [context] is by a context = line in sip.conf. Is there another way to create a [new_context] in extensions.conf so I can dial from it? Right now most of my extens are in [default] and I'd like to avoid that. Larry -- Larry Alkoff N2LA - Austin TX Using Thunderbird on Slackware Linux

Is it possible for a template to know the extension number in sip.conf? In the example extension below, I'd like to _not_ have to type the lines username=410 and callerid=410 in each extension. Instead, the template should be able to pick up the extension name from the [410] extension specified. In the [grandstream] template example below, I'd like to specify in the template that it

II noticed that Caller ID presentation is not making it to my cell phone through outound Telasip calls and I don't know why. It may very well have been this way for awhile (or always, not sure I called my cell phone during telasip testing). Does Telasip expect a different format than SetCIDNum(NXXNXXXXXX) ? It has always worked for the Teliax lines. BUT--- It doesn't have a problem

I would like to show a remark that would show call progress and appear on the CLI screen. The remark should be in the code of a sip [channel] or extentions [context] If I can't send my own remark, what little used 'show' command could I insert in the code? Can this be done? -- Larry Alkoff N2LA - Austin TX Using Thunderbird on Linux

I'm new to the list and to Wine. I loaded wine wine-0.9.7-i486-S10.2 into Slackware 10.1 in order to run the Forte Agent newsreader. It was setup and running under Win4lin which I have discontinued. What would be the syntax to load Forte Agent and specify the old "start-in" directory used under Win4lin? wine ./agent something_for_start-in_directory ?? Does Wine have the

I am trying to setup a simple Samba server using the instructions in the O'Reilly book "Using Samba" first edition on a Slackware 9.1 system which uses Samba 2.2.8a. The windows machine uses windows 98SE. The instructions under Server Configuration starting page 93 and Disk Share Configuration starting page 96 had me setup a very simple smb.conf which should then appear in Network

Problem: SPA3000 phone does not ring for incoming PSTN call although I can dial out. I set up my Sipura with the Voxilla Wizard which is pretty good but leaves out some important details. The Voxilla Wizard for Supura SPA3000 gave me a setting for PSTN Tab -> Dial Plans -> Dial Plan 8 (<S0:66610>) Should I put extension [66610] in sip.conf with a context in extensions.conf that

I would like to access my shell environment variable MYIP from within sip.conf to put in externip. I've tried some variations of syntax after reading "The Future of Telephony" but it's not working yet. Should it be externip=${ENV{$MYIP}} or some other syntax?? Larry -- Larry Alkoff N2LA - Austin TX Using Thunderbird on Linux

I am trying to page my Grandstream GXP-2000 phones and keep getting the error message: Jan 27 12:55:04 WARNING[30401]: app_page.c:183 page_exec: Incomplete destination '' supplied. How can I fix this error? The two contexts below do either one-way paging or two-way paging to all Grandstream phones in a list. [One_Way_Page_GROUP] ; one to many page exten =>