similar to: SIP port assignment for user agents registering to Asterisk.

Asterisk version 1.2.14 I use snom190 and xliteV3 as sip phones. asterisk send the rtp stream never to the xlite softphone. Any hits for me? *CLI> rtp debug RTP Debugging Enabled -- Executing Dial("SIP/xlite-007918f0", "SIP/snom") in new stack -- Called snom -- SIP/snom-00797110 is ringing -- SIP/snom-00797110 is ringing -- SIP/snom-00797110 answered

Hello, I am new in Asterisk Community, i am working on Asterisk 1.6, i need to know how can i monitor the extension status? when i wrote sip show peers on asterisk Extension Domain port Status 111/111 (Unspecified) D 0 Unmonitored 1300/1300 192.168.50.111 D 5060 Unmonitored 222/222

I have an odd problem. I have just installed asterisk on an ubuntu box, and migrated the previous configuration of asterisk (on another ubuntu box) to this new server (scp -pr xxx at oldserver:/etc/asterisk/* /etc/asterisk/) Asterisk worked fine on the old server, but on this server my SIP trunk peer does not login after initial server startup. "sip show peers" shows my phones

Hello, I am having problems getting one xlite clients to communicate through asterisk. I am getting an error message: chan_sip.c:15593 handle_request_register: Registration from '"chinmay chakraborty"<sip:1234 at 10.44.32.193 <sip%3A1234 at 10.44.32.193>>' failed for '10.44.32.193' - No matching peer found sip show peers Name/username Host

Has anyone done any integration work with Covad's hosted solution ? I am considering Covad's hosted solution and want to be able to use Asterisk to develop some other apps. Anyone else tried this ? how did Covad react. I know they use MGCP. Another thing, the Cisco reseller rep tells me if I have a bunch of 7960's setup for MGCP (for use with Covad) I will need to get these

There is an option that you can add to your dhcp server option 150 IIRC. -----Original Message----- From: "Rich Adamson" <radamson@routers.com> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: 2/27/06 4:37 PM > Has anyone done any integration work with Covad's hosted solution ? I > am considering

Hello Everyone - Well, I think I'm getting closer with the asterisk connection. This is my setup and I keep getting this error below in ,my /var/log/asterisk/messages file. I have opened 5060 port on the firewall box. I would this is Warning which I can ignore! But I see the connetcion coming but NO DIAL TONE on mt ata186 box sitting in my 192.168.200.x site! I'm using ATA186(cisco

If you haven't seen it yet, go here with a Flash enabled browser: http://www.theringingmovie.com Chris Coulthurst chris@shuksan.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050721/8ba2fb05/attachment.htm

Hi, there. I have asterisknow beta 5 with the following data: Ip 192.168.0.60 mask 255.255.255.0 gw 192.168.0.1 the router (a linksys) has port forwarded the port udp 5060 and from 16384 to 16482 udp-tcp from the internet to the asterisk machine. the only protocol allowed is g729. Which work fine for the ip phones I already have setup in the LAN. My problem is trying to register to a voip

I recently upgraded a Cisco 7960 to the SIP firmware, it worked fine without a "call-manager". I just put the SIP firmware and associated config files in the TFTP directory of my asterisk server so that the phone could pull the firmware off of my asterisk server via TFTP. It took me about 5 minutes maximum to get the phone working with SIP through asterisk. NO Call-manager was used in

I have asterisk up and running now, and installed XLITE on 2 PC's. Both machines (mistakenly) registered as the same user / extension. Strangely, asterisks allows this and both phones can make calls! But, only the first one to register can receive calls at the extensions. 1. Is this normal behavior? (Why allow 2 phones on same extension) 2. Why is asterisk not showing the second phone

Hello everyone. I've seen postings for connecting asterisk to vonage but I'm still having trouble achieving that. I have a vonage softphone and I'm trying to register to vonage using asterisk. I have not had any luck. I am behind a firewall. I've successfully gotten xlite to connect and work from the same network. When I change the port setting in [general] to 5061, I am able to

I have two sites running Asterisk PBX. Normally the inbound calls go through a 3rd (colocated) server and are routed via IAX to the site (the site registers with the main server) I created a macro that tries to ring one location and then another. Each site explicitly Answer() the call even though it will only ring all the sip phones at the relevant location. When fall back is in effect it goes to

I am trying to set up Hunt Groups and I am having some issues. Here is what I am trying to do. All my users actually register with OpenSIPS. Asterisk is using Realtime and I have set up a MySQL View Table so that Asterisk see's all the SIP users info that OpenSIPS has. This is what I have configured queues.conf ---------------------------------- [irock.com] strategy=leastrecent

hallo all, i would like to know, as i would suspect, nativ bridiging should work also, if only one iax party is behind an nat router and the other has a public ip. when i connect to iax clients, which have both pubic ip's nativ bridging is working. if one of the clients is behind an nat, the iax2 channels always get routed through the asterisk server (latest stable version from cvs) ?? i

Does anyone know if support for the Intel 875P/ICH5 motherboard chipset has yet made it into the stable branch? (Is release 4.9 likely to have it?) I am mainly interested in the IDE and "native" serial ATA devices. There is also a new Intel ethernet controller chip, 82547EI, that is designed to interface directly with the 875P chip. The currently supported chip list only goes up to

Hi all, My scenario is such that I have three users connected to a conference. CLI> meetme list 1234 User #: 01 9176502096 <no name> Channel: Zap/23-1 (unmonitored)00:00:32 User #: 02 john john Channel: SIP/john-b7800468 (unmonitored) 00:00:28 User #: 03 6463875998 <no name> Channel: Zap/22-1 (unmonitored)00:00:19 3 users in that

I'm running into a strange issue with Asterisk + Iaxmodem + hylafax on the same machine. After rebooting the iaxmodems don't register to asterisk. Stoping and starting the relevant services gets it working, but what is the point of using init scripts if it does not work right? I already tried to adjust the init scripts in /etc/rc3.d so I have: S50asterisk s90iaxmodem S95hylafax So it

OK. Now I show my ignorance. What's the difference between a T-1 and a channelised T-1? I see that Covad's voip service (formerly GoBeam) requires a channelised T-1. Then I read recently on the list that many T-1s being installed are actually HDSL....which would be not a T-1 at all...right? Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist