similar to: Sip domains, contexts and CHECKSIPDOMAIN

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>

Hi. I'm trying to set up a bunch of SIP phones to register in various domains on the local subnets connected to an Asterisk appliance, yet be able to support Internet SIP calling. My config looks like this: [general] context=INVALID allowguest=yes autodomain=no ... domain=redfish-solutions.com,redfish-internet ... [internal](!) type=friend qualify=50 nat=yes host=dynamic canreinvite=no

So I saw mention of a way to allow dialing using SIP URI's on Dave McNett's site at http://slacker.com/~nugget/projects/asterisk/page7 Wow, awesome, I can call anywhere now. However, I think there is a more elegant way of figuring out whether or not the local * server should handle a given domain. Specifically, Dave compares a series of domains within extensions.conf to figure out how to

Some of you have unusual SIP configurations, and this SIP perl script may be useful to get remote devices registering with your Asterisk or other SIP server. Most Cisco routers, as an example, are too stupid to REGISTER, so this script would be required to dynamically register them with a remote server. This may not be 100% applicable to Asterisk, since static registrations are possible,

Hi, Asterisk Version : 1.2.15 Card : TDM11B (1 x FXO , 1 x FXS) I have internal dialling working okay SIP->ZAP (analogue phone) and ZAP (analogue phone) -> SIP. The problem comes when I try and make a outbound call. Here is my extensions.conf :- Code: [incoming] exten => s,1,GoToIfTime(17:00-09:00\mon-fri\*\*?outofhours|s,1) exten => s,2,GoToIfTime(*\sat-sun\*\*?outofhours|s,1)

Hopefully I'm not just missing some little detail here. We're trying to set the HANGUPCAUSE on SIP channels to have our softswitch play the proper recording instead of answering the call on Asterisk to play the message. It appears that no matter what the HANGUPCAUSE is set to, Asterisk always just sends "603 Declined". I looked through the source code briefly and it appears

I have several Cisco SIP gateways sending calls to Asterisk. Because the gateways don't have user-agents, they don't authenticate with Asterisk. And because they don't authenticate, they use the default context in the sip.conf file. Is there a way to either: A) identify the inbound gateway with a variable, in channel info, or the manager interface? If there was a ${SIPDOMAIN} for

Asterisk don't running, because show this message WARNING[6949]: chan_sip.c:8865 reload_config: Section 'authentication' lacks type WARNING[6949]: chan_iax2.c:7491 load_module: Unable to open IAX timing interface: No such file or directory WARNING[6949]: chan_skinny.c:2587 reload_config: Unable to get our IP address, Skinny disabled WARNING[6949]: chan_oss.c:239 sound_thread: Read

I have a new Sangoma A102 and I'm trying to get it running in asterisk. A look through the dmesg log shows the card is detected and the various channels created. However, when I start asterisk I get the error below. Any ideas? My zapata.conf is below. Thanks, MD == Registered custom function SIPCHANINFO == Registered custom function CHECKSIPDOMAIN == Manager registered action

Is there a way to pass variables/arguments to the h extension ? for example : [default] exten => _1098933X.,1,NoOp(CARRIER TWT->TIM, EXTEN: ${EXTEN}}, SIPCALLID: ${SIPCALLID}, SIPDOMAIN: ${SIPDOMAIN}) exten => _1098933X.,2,SetVar(_PROVA="bla") [lot of stuff, agi, goto, tricks and magic that happens] exten => _1098933X.,10,Dial(${CHAN_DEST},,L(3600000:3599900)) <-

hi all, how to establish a call between two asterisk servers for the sip users registered for the servers. ----- Original Message ----- From: <asterisk-users-request at lists.digium.com> To: <asterisk-users at lists.digium.com> Sent: Sunday, February 10, 2008 11:30 PM Subject: asterisk-users Digest, Vol 43, Issue 30 > Send asterisk-users mailing list submissions to >

Hi, is there anybody who knows what this warning means?? WARNING[27309]: chan_sip.c:8875 reload_config: Section '10' lacks type TIA Giorgio -- ____________________________________________________________________ GIORGIO INCANTALUPO Tel. +39 02 9350 4780 (104) FG&A Software 20017 Rho - Via Puccini, 8 E-Mail : gincantalupo@fgasoftware.com Internet: http://www.fgasoftware.com

found response from asterisk. =20 =20 On asterisk's log I see messages like: "Looking for conference on conference-context (domain serverIP)" =20 And: "Call from 'conference' to extension 'conference' rejected because extension not found." =20 =20 Does anyone have an ideia of why I'm getting that message? =20 Why does asterisk seem to be using domain

I am using asterisk-oh323-0.7.2-pre and CVS Head of Asterisk. Oh323 Module compiled without errors. But When I try to stary Asterisk with the Oh323.so file in the modules folder, Asterisk is dying with the following error. [chan_oh323.so]Aug 2 14:08:14 NOTICE[18873]: res_musiconhold.c:490 monmp3thread: Request to schedule in the past?!?! => (InAccess Networks OpenH323 Channel Driver) ==

Hi all i have been succesfully using OpenH323 (oh323) for a few months. the versions are: asterisk CVS HEAD 19-07-2005, OpenH323 v1.13.5, PWlib v1.6.6, asterisk-oh323-0.7.2-pre1 I now have moved to Asterisk 1.2.4, so as per the directions i am using: Asterisk 1.2.4 pwlib_Mimas_patch2 openh323_Mimas_rc2 asterisk-oh323-0.7.3 The problem is that when asterisk starts it fails on loading the module

Hi group! Does DTMF inband work with oh323 channel driver ver. 0.6.7? If yes, how to enable it, make it work? I have tried with "inBandDTMF=yes" in general context of oh323.conf, but I get this message when I * is starting. [chan_oh323.so] => (InAccess Networks OpenH323 Channel Driver) == Parsing '/etc/asterisk/rtp.conf': Found == Parsing

Hi, I'd like to call my friends through their SIP URLs. I've found two approaches for doing this in Asterisk: - one is to prepend some numbers at start and catch them - the rest of called string is used for SIP URL - another approach (that I like better) is to use catchall pattern at the end of context _. and then parse string with help of SIPDOMAIN variable. But there is a catch into

Hi all, I am trying to write a program that sends SIP requests to asterisk. My aim is to make asterisk record as many voicemails it can at a time. The design of the program is like this: There are two processes: One main process and a child process (No flames pls. I have very little idea about pthreads and dl modules) The main program asks the user to input the number of test instances. When

When my SIP phones try to register with my asterisk box, this is what I get my log file: Mar 26 14:46:41 NOTICE[3896] chan_sip.c: Registration from '<sip:201@192.168.2.13>' failed for '192.168.3.2' - Not a local SIP domain In sip.conf I have this for my global settings: [general] context=from-sip ; Default context for incoming calls

Hi, I have been experimenting with NAT and Asterisk a bit now. Though I have made progress along the way, I have come across the following problem. I'll really appreciate if anyone can provide me any help or pointers. Thanks! Successful Scenario: ------------------- All sorts of NAT calls are successful with full two-way media when both end points are locally subscribed users. Problem