similar to: SIP user registration and Asterisk Realtime

Hello We have recently upgraded to Realtime engine (sip buddies and extensions) and now have problems with calling local SIP users. I have rtcachefriends=yes but tried with 'no' and it's even worse. (asterisk 1.8.1.1 + realtime mysql) Here's an example: User 1000 registers successfully and can then be called with Dial(SIP/1000,30) successfully After some time when I try to call

I am running CVS Head i686 running Linux on 2005-06-30 22:55:14. I have SIP Buddies installed using MySQL. If I try to set up a ATA that has 2 two phone lines (resulting in 2 lines on 1 IP address), my second line can never authenticate to dial out. I ran ethereal and found that Asterisk is "looking at the IP the request came from" and then, apparently looking up the IP address in

I've got two Polycom 500's that I'm playing with, and I want view the status of either phone, (busy/on the phone/etc.) from the other. I've got this cute little 'Buddies' button, and I can add contacts to that. But the status doesnt actually update. Do I need to setup realtime for asterisk? Can anyone point me to some documentation or give me some hints? Thanks, Nihal

A follow up (late better than never) Finally had time to sit down and look at this sip.cfg <keys key.scrolling.timeout="1" key.IP_500.31.function.prim="BuddyStatus"/> This turns the Services key which I never use on a 501 into the Buddy Status. It even works while on a call. One touch! Bill ________________________________ From: Bill Gibbs

Hello! I have 2 table: users and buddies User: id, name, ... Buddy: id, user_id, user_buddy_id, ... So if you have 2 users 1,jack and 2,fred and fred is a buddy of jack, there is a Buddy object: id=1, user_id=1, user_buddy_id=2 I can declare only one belongs_to in Buddy and one has_many in User. And there is conflict if I had the second one (the first one is discarded) class User has_many

I know this is not asterisk specific but we all know this group is used for Polycom issues as well... I have hints working ok on Asterisk. However the Polycom phone will only show the buddies key if there is not a call. This defeats the purpose of using the buddies to see if you can transfer a call to another extension (using the Buddy key to see if they are on the phone). Polycom sip

I simply found it at: http://getgnuwin32.sourceforge.net/ Seung ---- Original message ---- >Date: Tue, 12 Feb 2008 21:11:04 -0800 >From: "Ted Neward" <ted at tedneward.com> >Subject: Re: [LLVMdev] Is there someone tried LLVM 2.1 on Visual Studio 2005? >To: "'LLVM Developers Mailing List'" <llvmdev at cs.uiuc.edu> > >I'm sorry,

Hello list, Can I do something like this for BLF functionality : [test-blf] exten => _XX,hint,Macro(GetSIPaccount,${EXTEN}) exten => _XX,hint,SIP/${SIPACCOUNT} GetSIPaccount is a macro that looks at a MySQL-database for the realtime table sip_buddies where the SIPusername is taken from. It works great for internal calls... but how about BLF-functionality ?? Feedback is appreciated !

>From a quick look at the source code, it looks like onComplete is only called when the whole thing is done - in the case of Updater, right after the update, in the case of PeriodicalUpdater, right after it has been stopped. It looks like you need to pass your callback method as an ''insertion'' method in the options. This takes two parameters, the first will probably be the

Hi all, We are looking for information about Dynamic Realtime Asterisk, We have install some Snom phone 360 (SIP) for our customer , but we have a problem when we want to register 2 accounts on the same phone and on the same Asterisk PBX. The problem when we register two phone line in realtime it doesn't work, we can't make a call the registration failed when we place a call. Can

Apparently, the realtime system in asterisk is faulty. Implementing realtime, begins a host of seeding messages along with registration messages visible at the CLI prompt. This is not the case with .conf file configuration Unfortunately, it is not clear where the bug originates but is shows it's head while calling the register_verify function, (which there are 2 one in chan_sip.c and one in

Dear All, I am currently working on asterisk cvs-head version in order to use realtime with mysql, 2 asterisk servers with duplicate mysql databases, one asterisk server is serving the sip phones and the data is logged to the database and replicated to the other asterisk database, when the first server fails though it has the sip phones data in it's database the sip phones need to re-register

Hello there I'd like to use sip users and peers realtime. I think I done all I need to get asterisk works fine in realtime: res_odbc.conf configuration. extconfig.conf sippeers => odbc,asterisk,sipclient sipusers => odbc,asterisk,sipclient sip.conf [general] rtcachefriends=yes The sipclient table as suggest in this article: SIP Realtime, MySQL table structure (

Hi! I use the following configuration to register my asterisk server to my SIP provider: register => 12345:passwd@sip.provider.com/12345 sip.conf: [sipout-test] type=peer username=12345 fromuser=12345 fromdomain=provider.com secret=passwd insecure=very host=sip.provider.com qualify=yes context=test-incoming extensions.conf: exten => 12345,1,Dial(SIP/10) exten =>

I have read and followed as much as I can find but I am missing something. What I want to do is get as much as I can running from mysql and keep the *.conf files for static things. So I have setup a SIP users/peers table in a mysql database and I have populated it with a few peers. I have configured asterisk addons and from the asterisk CLI I am able to search the sip users / peers tables using

I have set up realtime for Asterisk just as the instruction provide. Everything works, except it apearer that SIP devices do not regisert correctly. I can place a call from a SIP device, but not place a call to a SIP device. If a I use sip.conf everything seems to work. I have not posted all the configurations here because I'm just looking for a set of checks to follow. I noted that

If you are using the Express versions of Visual Studio, the Platform SDK(windows.h) is a seperate install that you have to download. Kevin Tew Ted Neward wrote: > Thanks--I was offline when I wrote it, couldn't Google. Found it in about 5 > seconds once I was back online. > > Second question: I'm getting various build errors relating (it seems) to > configuration:

A user wants to time a command/program/application and ensure the full contents of that command's data management has been flushed from RAM/cache and saved to disk. Is there a way to ensure the cache/RAM is flushed to disk, or is the flushing a function of the kernel or the application? This happens to be on a 64-bit Intel system with 64-bit CentOS 5, fully patched. But even if the

Greetings, Recently there has been talk of the presence/buddy feature with asterisk and Polycom phones. I have it setup, and working as expected, however I can only get 7 buddies to appear on the screen at any given time. Has anyone gotten more than 7 buddies to appear? I'm just trying to find out if this is some polycom limitation, bug, or my error. Thanks, Matt -- Matt Gibson VOIP

Thanks--I was offline when I wrote it, couldn't Google. Found it in about 5 seconds once I was back online. Second question: I'm getting various build errors relating (it seems) to configuration: can't find windows.h and so forth. Where (or to whom) is the best place to report these and iterate until we fix them? Ted Neward Java, .NET, XML Services Consulting, Teaching, Speaking,