similar to: Display SIP useragents

Hello, I have one question regarding *. Default configuration for asterisk is to keep configuration(s) in ordinary text based config files. My question is simple: is it possible to keep those config info (at least, to start from - sip.conf, extensions.conf and voicemail.conf) on a database, which asterisk access via ODBC. If it is possible, I'd appreciate if someone points me where I can

Hmmm... Folks, I beg you pardon, if I'm telling something which was said before, but actually I have not found this anywhere, neither on Voip-info.org or in several Asterisk's docs. So, here is the statement: If SIP extensions are in DIFFERENT CONTEXTS, then RTP traffic between them will ALWAYS go via Asterisk. I.e. Asterisk WILL NOT issue Re-INVITE even if: 1. Both UAs have

Hello, I have some reports from users that occasionally DTMF will stop working in voicemail and they will have to exit the system to get it to work again. The useragents are Polycom IP 500's and I am using dtmfmode=rfc2833, with Ulaw codec. This is all on an internal switched 100mb lan. Has anyone else seen anything like this? Thanks, - Brent

Hi Yves.. This may be silly... but what is the useragent of your sip configuration? In the case that useragent has some special characters like "(.", please remove it and tell us if there is any change!!. Regards. rv 2014-11-22 14:50 GMT-03:00 Eric Wieling <EWieling at nyigc.com>: > Try setting directmedia=no in sip.conf. > > -----Original Message----- > From:

Hello, I'm running an Asterisk 1.4.14 on a linux machine. Serving SIP Snom users. I've noticed that each time Asterisk is restarted, for the first 5-10 minutes, the SIP users can dial but cannot be dialed until each phone re-registers itself against the server. So only after the "Saved useragent...for peer 111" line appears on the Asterisk console, then the 111 user can be

Guys. What do I need to configure in order to let my Asterisk receive calls from sip phones, etc not registered with my server on my extension? For example, let people use their asterisks or sip phones to call blah111@server.com?

Hi list ! Is it possible to show the used Useragent of a peer that registered with Asterisk? It's being saved obviously because the console says so when a phone is registering but sip show peers doesn't show it? Is there any other way to view it? Thanks!

What a great way to end the day! This one has me laughing my ass off.... I am hoping you actually meant "guys". You may want to look up the meaning of the word you used... Here are the resources you should look into... Make samples at the command line will create your asterisk sample scripts. This is covered in many of the "Getting Started" types of docs. This is the Wiki

I met a strange SIP problem recently. In an ordinary procedure, when asterisk loads sip module, a series of functions are called sequentially: load_module()->restart_monitor()->ast_pthread_create()->pthread_create()->do_monitor() However in my system, pthread_create() failed to create a child thread to execute do_monitor(), (though pthread_create() returns a successful signal to

I am trying to upload a directory structure with rsync via ssh from one domain to another. I would like the target files (which may or may not already exist on the target machine) to assume the security permissions of the target directory they are placed into, since the target machine lies in a different domain. Currently, when I upload these files they seem to assume some strange random

All, Is there a way I can get a list of all my users' useragents? I basically want to know what firmware each of my phones have without having to list them individually. Thanks David. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070214/fed26ed2/attachment.htm

Hello All! Is there any chance of showing the real useagent in the "listclients.xsl" I like to keep track of what players and version is used. Winamp 2.x or 5.x only shows Winamp not "WinampMPEG/5.0" Now xmms useragents works like it should! Thanks John

Hi all, Right now our queue system is working quite well. One thing that I would like to have the option of doing would be to play a "ring" for the customer when they are connected to an agent. Right now they go from music on hold to an agent with no indication that their call is "going through." Our agents are played around a five second message when they pick up a call,

Greetings list,this question is rather a pain in my side.. i have been trying to figure it out.. it could be simple.i have a customer with a callcenter .. we developed a CRM "Customer Relations Management" with an SIP dialers built in.the question is the following.. is it possible to force the agents (users) to use a certain UserAgent which is the one built-in our system? this way will

I am trying to get remote extensions to work correctly with agents. I have ackcall=yes and have agents logged in to extension 101 using agentcallbacklogin with extension 101 defined as: exten => 101,1,Dial(Zap/3/18165551234,20,tTA(custom/presspoundtoanswer)) This setup works great on local and/or voip channels, but on zap channels, the zap channel answers immediately as soon as it goes off

I have connected asterisk 1.0.7 with Avaya Definity via E1 with a TE100P Digium Card. Inbound calls are working perfectly and I dont have any problem. But when I try to make an outgoing call with my softphone (xlite) I am getting the following messages. Hungup 'Zap/13-1' Executing Dial("SIP/IZ-bc0a", "Zap/g1/3118") in new stack Called g1/3118 Channel 0/1, span 1 got

Dear All, I am having some problems with connecting with a UA. Sometimes there is not sound in the call made, sometimes the caller would near no sound, while the callee can hear the caller. I have attached the rtp debug and sip debug for you comments. Please help me. Thank you all. Asterisk Version is 1.2.1 Asterisk RTP Range is 10000 to 20000 UA Listen RTP Port is 15000 Below is the the

I have a simple 4 person ACD queue using the AgentCallback function. No matter what strategy I use, anytime someone calls into the queue asterisk dials the agents in the order that they are listed in the agents.conf file. This doesn't seem right to me, or am I wrong.

I don't know...now I have a _X. in my CDR. -----Original Message----- From: Matthew Boehm [mailto:mboehm@cytelcom.com] Sent: Saturday, March 12, 2005 8:05 PM To: Asterisk Developers Mailing List <asterisk-dev@lists.digium.com> Subject: Re: [Asterisk-Dev] SetVarCDR You must have some fux0red config 'cause using _X. works fine here. I haven't had an 's' in my CDRs for

We recently deployed 10 SNOMs as part of a PBX hosted solution. We have one phone setup as the receptionist phone, using hints to show busy office lines. This all works as expected. This is a new installation, and people are just starting to setup their phones. For those of you not familiar with SNOM phones, there is a row of keys on the right side of the phone which SNOM calls function keys. In