similar to: modifying a dialed exension before dialplan processing

Just a quick listing of tested, and updated, steps from my notes. Enjoy ! http://asterisk-notes.blogspot.com/2007/07/installing-asterisk-from-source-on.html -baji. --

Dear All: I have several boxes that up and running just great, then we changed internet equipment due to a lightning strike, now all my inbound IAX connections (iax2 show peers) have unknown status. If I log into the remote boxes, it says "Request sent." The authentications haven't changed at all, and all the iax.conf settings are correct. It looks like a firewall issue, but

hello, running asterisk 1.4.11 on CentOS 4.5 I am getting no response on function STRPTIME() the system just hangs, STRFTIME() is working fine as seen below. Same thing happens whether I called in from a softphone or via teliax. While executing the following code : ; exten => s,n,Set(v_ts=) exten => s,n,Set(v_ts=${STRFTIME(|America/New_York|%Y-%m-%d)}) exten =>

May be slightly off topic, but I was wondering what everyone thinks of this latest ruling against Vonage? Does anyone really know what Verizon hold patents for, and could those patents possible affect anything in Asterisk? Who knows who Verizon will go after next. Brent -------------- next part -------------- An HTML attachment was scrubbed... URL:

Calling from PSTN let extension 601 ring twice, hang up and starts over again to ring twice, ... If I pickup I do not hear on extension 601, and on the PSTN it is still signaling to ring. Can anybody enlighten me, please? extension.conf [incoming_88097074] exten => s,1,Wait(1) ;wait to get caller ID in. exten => s,2,Dial(SIP/102,20) exten => s,3,Voicemail(u102) exten =>

I'm trying to find the right Digium card for the Dell 2950 Dell 2850 HP DL380 G3 HP DL360 G3 Are these 3.3v or 5.0v machines ? I am out of the office, and need to buy a card today. I am looking at either the TE407 or TE412, and would appreciate any help. :) Julian

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"

Hello All, For some reason my asterisk server is not registering a port number with my VSPs. This is causing problems where people are not able to dial in from any of my SIP or IAX VSPs. I do have one VSP that has hard coded my IP and port so I can get incoming calls but this still leaves a problem with my other VSPs. Hose can I get asterisk to register my IP and port? I have been

Hi, I'm having trouble getting Dial to exit when the caller hangs up in Asterisk 1.4.21.2. I use a POTS line to call into the DiD given to me by VOIP service provider. When the call comes in, I have the VOIP provider send it to another POTS line. All this works fine however when the caller (me) hangs up, the Dial command does not exit. The callee stays connected (and my billing

Hi i have the running i both sides, client and server sides the puppet 0.25.4 Get this error on server side: puppetmasterd[5363]: Report puppet_dashboard failed: wrong Content-Length format And receive this error on my client side: warning: Value of ''preferred_serialization_format'' (pson) is invalid for report, user default (b64_zlib_yaml) I am getting any reports on my

On Thu, Feb 13, 2014 at 5:52 PM, Renato Golin <renato.golin at linaro.org> wrote: > On 13 February 2014 13:47, Evgeniy Stepanov <eugenis at google.com> wrote: >> Hm, I see that -funwind-tables on arm-linux-androideabi target >> replaces this "cantunwind" with a proper unwind table. >> Hence http://llvm-reviews.chandlerc.com/D2762. > > If Android is

I'm sure this might not be the correct place to ask and I have done a Google but I can't seem to find anything that says there is a VSP that will work with * in the Ukraine. I have a friend that lives in Kiev and basically want a phone number there to be able to talk to him and have him call me. If anyone has any information on it and they are willing to share please advise.

Has anyone tried using BroadVoice for VSP? I have Asterisk configured for a home office & I've been trying to decide which VoIP provider to go with for a little while now. I had heard you could get sub $.01 calls but I have not found that to be true yet (not saying it's not possible, I just haven't found it!). Also I'm not sure if BV will support multiple lines. Any

Due to an issue I'm having with 7.x, and trying to track it down, I spent tonight getting my server setup to allow my to break into the debugger when it hangs, and hopefully dump core ... But, although I *think* I've got it all, I'm obviously missing something, as it isn't breaking ... First ... I'm running a proliant server, and when I connect via SSH to ILO on that

Hi, I've Sipura 1001 phone adapter. In the settings it has separate Line 1 and Line 2 tabs, which apparently means it can control two separate phone lines. I've Asterisk@Home server and want to setup two different extensions for two phones, i.e. 201 and 202. After doing all this, I can see in Info tab that both lines are registered but only one phone gets the dials tone. Am I doing

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Hi, I am trying to do my first puppet configuration, already installed the puppetserver and client, in this link show my configuration and my puppet structure: http://paste.pocoo.org/show/212227/ But when i run the client side daemon i get this message: info: /Class[main]/Node[basenode]/Class[inittab]/File[inittab]/source: No specified sources exist err:

Bear with me this is a bit long winded. I am having some issues making automated outbound calls over Broadvoice from my Asterisk 1.4.2 server. For reference, none of the below issues happen when I make the calls to VoIP phones attached to the Asterisk server. What I am trying to do is call, using a .call file, out via the SIP trunk we have setup, and when the party picks up use AMD to

I am trying to build cross compiler for custom processor (say XYZ) but on compilation it is giving error llvm-build: error: invalid native target: XYZ (not in project) I have tried configuring like these 1. ./configure --target=XYZ 2. ./configure --target=XYZ --enable-targets=XYZ 3. ./configure --enable-targets=XYZ But every time it is not recognising the XYZ processor. What could be the

Hi all, I've just been told from an ex workmate that my VSP (who I used to work for) has put an anti flooding limit of 80 SIP messages per IP per minute in place. I run the phone system for a facility that has a lot of extensions, but would rarely have more than 4 or 5 simultaneous external calls. Am I in danger of tripping over this limit? It sounds dangerously low to me.