similar to: adding IAX headers

My application fires several calls thru AMI ORIGINATE command. For example if I have 3 operators I do 3 ORIGINATEs. My trouble is when one operator quit for some reason, I should kill the corresponding ORIGINATE. Of course, I could let the call ring and hangup after the customer pick-up. But this is not the case, I do have to kill the corresponding ORIGINATE. I could execute a soft hangup,

Hi list, is there any existing IAX command to add information to a call like SipAddHeader? Another solution is sending text frame (0x07) frame type, but I don know how do it in a dialplan. Thanks for any hint. -- Daniel

I have an asterisk box using a SIP provider and IAX2 softphones clients. Audio is low and I need to apply some gain on it. How can I configure such gains - in/out on sip and iax2 channels? Thanks, Valter -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101106/85417a17/attachment.htm

I have been running af Asterisk server Version 0.7.2 for a while now But I I wanted to upgrade my version to the new 0.9.0 or the CVS 1.0 Stable. But when I install one of the new asterisk servers I having lots of troubles with the IAX connection between my servers. When I start the 0.7.2 asterisk server it shows me something lige this == Parsing '/etc/asterisk/iax.conf': Found ==

Prezado Fernando, Muito obrigado por sua complementa??o na resposta! Surgiram algumas d?vidas agora: A ?nica forma de retornar os dados num header field, como o Rafael dos Santos Saraiva sugeriu envolve criar outro channel? Ou seja, o que eu preciso ? que a mesma execu??o do dia plan obtenha um valor recebido do Sip Client, execute uma query num banco de dados e em seguida inclua a resposta

I'm wondering if anyone has a solution to this before I begin looking at making some changes to the SIP channel. Basically when calling SIPAddHeader() twice from the Dialplan or an AGI script with the same header name it adds duplicate headers instead of overwriting the existing one. Here's a practical situation where this applies. A call is to be terminated via SIP and we have two

HI ALL: I am newbie to IAX, my iax.conf is as follows: [general] port=5036 ..... but I donot why it doesnot listen on UDP PROT 5035, instead it listens on 4569 Asterisk CLI debug says: [chan_iax2.so] => (Inter Asterisk eXchange (Ver 2)) == Manager registered action IAXpeers == Parsing '/etc/asterisk/iax.conf': Found Nov 30 11:52:12 WARNING[1076220544]:

Hi all ! I don't know what else to do but I absolutely can't get it to work !!! :( I've installed 2.2.8a on my SGI 6.5.17 workstation and all the tests I've done failed. I'm getting quite desperate. :( Ok. Point by point I'll try to explain what I did. - Installed the 2.2.8a successfully and started it successfully. - Logged in the samba (almost) successfully from the

Hi, I am setting up a small call center using *. I have ZAP setup for incoming calls and IAX setup for agents. Agents login using AgentCallbackLogin. When customers call, it's getting picked up and when queue is trying to call back the agents, I am getting error. I am using CVS HEAD, and updated just now. The error is: -- Executing Answer("Zap/1-1", "") in new

Hi All, I am having trouble with receiving calls from FWD via IAX. I know this isn't a FWD support forum, but I suspect the problem is my asterisk setup. The problem is that I can dial out to fwd subscribers, even myself but they can't dial me using my FWD number. I don't know much about IAX, but it would seem to me like a registration problem, but I get no errors or warnings in

Please can someone help me with the following two error messages: Error 1. I have loaded the Zaptel dirvers and everything is ok with ztcfg. I have configured Zapata.conf and everthing looks good but it apears the Zap channels dont load when starting Asterisk. When I make a call to one of the fxs port I get the following error message. -- Executing Dial("SIP/39-b204",

Hi ! I'm sending you all my test smb.conf and the log file generated with a 2.2.5rc4 installation. The trouble is that I cannot create new directories (I'm using a WinXX Italian version). Then under Win98 I get the most strange behaviour. In addition to the above, using win98 I can't copy files from the share to the local disk (error 1026). Please help Thanks Valter Simo Sorce

Hi all, I'm trying to get IAX2 trunking between two * boxes and am having extreme difficulty :) What happens is when the sending * server (the one initiating the call) receives the ACCEPT back from the receiving server it immediately replies with INVAL. I've checked the code and it seems to be not matching the accept packet with the relevant item in the iaxs array due to the following

Hello, Right now there is not a table build script at: http://www.voip-info.org/wiki-Asterisk+RealTime+IAX Therefore I have taken the SIP build script and added a few fields that I use from my iax.conf (could be more out there, please see the complete build script below): `dbsecret` varchar(100) default '', `notransfer` varchar(100) default '', `inkeys` varchar(100)

If a call from one Asterisk box to another comes in, it always goes to the default context because it is treated as IAX guest. I don't understand why ?! Asterisk box 1 iax.conf : [BOX2] type=user auth=rsa inkeys=public_key_box2 context=from-BOX2 [BOX2] type=peer host=dynamic auth=rsa outkey=private_key_box1 Asterisk box 2 iax.conf : register => BOX2:BOX2 at IP_BOX1 [BOX1] type=user

On 7/23/2023 12:32 PM, Dirk-Willem van Gulik wrote: >> On 22 Jul 2023, at 23:40, asterisk at phreaknet.org wrote: >> >> I'm assuming you mean at the device level, and that you want to send >> only the relevant header to each device? >> Use pre-dial handlers; a unique handler runs on each destination >> channel. With PJSIP, you're forced to do this

I have an Ubuntu system and went into Synaptic and checked asterisk for installation. Once installed, I started it with /usr/sbin/asterisk -vvvgc and got the following output with several errors and notices. Do I need to do more or are these ok? I expected to have some conf files in /etc/asterisk but there is nothing there. Thanks! Created by Mark Spencer <markster@digium.com>

We have a couple of parallel ring settings (and this has worked well for eons). Either in the form of same => n,Dial(SIP/1001 & SIP/1002 & SIP/1003 …..) Or via a subroutine (below) that has a bit of extra logic: FOO = 1010 & 1019 & 1017 & 1033 ... same => n,gosub(sub-callout,s,1,(${FOO},”Ringing all class FOO telefons")) Now I have two types of phones

> On 22 Jul 2023, at 23:40, asterisk at phreaknet.org wrote: > > On 7/22/2023 4:51 PM, Dirk-Willem van Gulik wrote: >> We have a couple of parallel ring settings (and this has worked well for eons). >> >> Either in the form of >> >> same => n,Dial(SIP/1001 & SIP/1002 & SIP/1003 …..) >> >> Or via a subroutine (below) that has a bit

Hello list, I'm looking for a SIP/IAX client that would hook its audio on the JACK audio server on the client host. I first tried to use the Asterisk JACK application, installing an Asterisk instance on the client host, but I got mitigated results : the host runs a RT kernel (necessary for my use of JACK) but each time a new call is started, it triggers Xruns. Not good. Also, during the call,