similar to: asterisk newbie and phones which don't want to comunicate

hi do u have the sip phones extensions in the extension.conf and are they in the right context (sip-incoming)??? are the sip phone registering to asterisk?? try stop asterisk and reconnect as asterisk -vvvvvvvc to check see them registering... ________________________________ From: asterisk-users-bounces@lists.digium.com on behalf of Sistemista WebSolvingJaa Sent: Fri 7/1/2005 6:43 PM

Dear, I have XEN source 3.3/ Centos 5.4 64 bits running DELL power 1800. My big problem is with vritual machine running Windows Server 64 bits for roles Sharepoint 3.0 and SQL Server 2005. The virtual network of virtual machine stop comunicate but others all virtual machine still comunicate. I don´t understand why when network interface stop comunicate, the device manager of network interface

Hi list, i've problems with my * server and the 4 phones which are linked to it. i've 2 grandstream bt100 with the firmware upgraded to 101, a wi-fi phone (i don't know its brand) and another ip phone i don't know its brand. with this sip.conf : [general] port = 5060 bindaddr = 192.168.100.229 context = default ;x changed from default to sip localnet = 192.168.100.0/24

Recently I've noticed two bits of odd behavior with respect to regcontext/regexten in CVS HEAD & 1.2 Beta1, and I was wondering if anyone could shed some light on this. I've set up a regcontext in sip.conf. I've set up two users with regexten entries, one in sip.conf and one in a mysql realtime table. The first bit of oddness is that regexten seems to work somewhat as described

Dear all. I have what appears to be a configuration error but I cannot for the life of me see what it is. (I am a newbie) I have searched the wikki and google etc but still none the wiser. Any help would be very gratefully received. Problem: Unable to make outgoing calls via E1 euroISDN Digium TE110p card, given congestion signal as per config, unable to open zap channel. All incoming calls work

Hey all, Has anyone had any issues with phones having multiple lines that are in different contexts? We've got a couple phones that we're testing intercom functionality for, and I'm noticing that for some strange reason, no matter what line we use, the phones tend to be completely in one context or another, not segregated like I would expect. Our contexts look like this: context

Can anyone help with the use of regexten? (* 1.4.3) I've got Asterisk creating extensions for my SIP phones using regexten but I can't seem to figure out how to make use of them once they're registered. Here's my dialplan for from-sip (the SIP's default context): asterisk*CLI> dialplan show from-sip [ Context 'from-sip' created by 'pbx_config' ]

I've google for hours trying to find a discussion of a similar problem as the one I'm having, so forgive me if this has come up before. If it has, please point me in the right direction! The problem occurs when a caller (A) is transferred by an intermediary party (B) to voicemail (Voicemail or VoicemailMain), either directly or by being taken to voicemail when the callee (C) doesn't

Hello Thanks for the answer... Just realized it myself, as your mail arrived :) Could be a nice feature though. Jon -----Oprindelig meddelelse----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne af Olle E Johansson Sendt: 8. juni 2006 12:09 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [Asterisk-Users] Using

I've been googling and talking with Libretel about my setup and the fact that incoming calls to my asterisk box through the Libretel number reach my box (I hear the greeting being played) but then don't accept DTMF. Here is a rough diagram of my setup: Asterisk | server | NAT <------------ Libretel | router | Note that there are NO SIP

I'm in the process of testing a TLS/SRTP install. My experience is improving with each new challenge, but this one is a great test of my 2 month experience with Asterisk. When I dial 6003 from 6001, it takes 35 seconds until I get the error message that 6003 is circuit-busy. Any help would greatly be appreciated. Below is the error message and the extensions and sip.conf files. *CLI>

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

All, I've set up tinc yesterday (Win32 -> Linux), the connection comes up but I can't reach a host behind the tinc "Server". Maybe I did something completely wrong, so let me first describe my network.... The "Server" has multiple Interfaces, be we just care for two of them here. One NIC "eth2" ist to the LAN (192.168.100.x/24) and the oher one is

I am new to asterisk and need to dig up some info on how to set it all up. It looks a bit daunting especially all the options available in the .conf files. I have 2 SIP phones, GXP2000 and a budgettone 100. phone1 - 192.168.0.160/24 extension 1000 phone2 - 192.168.0.161/24 extension 1001 Server - 192.168.0.57 I get the following all the time, but can make calls between the 2 extensions,

(Im' answering cc the list, so the knowledge keeps there, and maybe some more qualified answers become). Am Mittwoch, den 15.10.2008, 18:00 -0700 schrieb Francisco del rosario: > Hey Rodolfo... Need some help from you ... > I need to know what hardware do I need to make SIP calls if I set-up > asterisk > So the situation is that I have a PC and configure the software of my PC to

Hi all, In sip.conf: [general] regcontext = autoreg [devabc] regexten = 543 creates "exten=> 543,1,Noop(devabc)" in context autoreg when devabc registers. But I can't use "exten=> _5XX,2,Dial(SIP/${EXTEN})" in the dialplan, because there's no device SIP/543. Now I know I can add a line like "exten=> 543,2,Dial(SIP/devabc)" for each and

Hello-- I've just written and submitted a new module for asterisk, to the asterisk bug database. See http://bugs.digium.com/view.php?id=6021 There is a file there you can download, AEL2v0.3.patch.bz2 and I created a wiki page: http://www.voip-info.org/wiki/view/Asterisk+AEL2 Why did I do it? Because I was very impressed with AEL, but the current AEL compiler isn't real good at

Ok, I could have sworn this was fixed in Asterisk 1.2, but it seems in Asterisk 1.4.4, that doing a reload, or even an 'extensions reload' will clear any extensions that have been created by regexten. This is VERY bad! Doug. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Anyone know how to use regcontext et regexten parameter from sip.conf and can give an example ? thx regards Harry -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090807/ef9ba45e/attachment.htm

Hello, I'm having a problem in voicemail check attempts from SIP-based phones. I've searched a ton of docs but don't see anyone else having a similar issue. I have a TDM22B with two non-sip phones connected to it as well as several SIP phones including a GXP-2000 and some X-Lites. Users of the real phones in the same context can pick up and dial *8 to get to VoiceMailMain() just