similar to: Asterisk, SIP and NAT: Help needed!

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,

I am running asterisk 1.0.1 with OH323 compiled in. We have a 323 trunk to CallManager with a mgcp controlled pri router. When using sip phones (directly registered with asterisk) to call out the 323 trunnk to PSTN, calls timeout after 3 rings. If I answer b4 3 rings - no problem, otherwise I get "no one is available to answer at this time" on the consoel and it redirects to an

I am having trouble setting this system up and wonder if some one help me. Does anyone know what is missing if anything to get 2 phones on my asterisk home server to be able to call each other. I have a WRT54GS running OpenWRT/asterisk connected to a PAP2 with 2 extensions 5060/5061, this is on the lan side of my gateway/router WRT54G 192.168.1.1 BusyBox v1.00 (2006.11.07-01:40+0000)

Hi All Total noob on the list so all help appreciated.... I've successfully installed Asterisk on an IBM A30P Thinkpad using fedora Core 2 (I'm looking at having a mobile PBX for conferences and shows). I've plugged in two Cisco 7960 phones.... The phones register with the Asterisk correctly and I can run the demo's and even the AIX demo through to digium works correctly.......

Greetings, I've noticed a problem that might originate from my Asterisk configuration, could use a hand in sorting it out. Problem is a 488 response from Asterisk whenever it gets RTP/SAVPF profile in the SDP. My current setup has Asterisk Kamailio realtime integration, and Kamailio uses dispatcher to route calls for Asterisk to handle. Now I have only one Asterisk, on the same machine as

Sorry if this comes in twice. Wasn't subscribed first time :-( Anyone help me here...... It worked once :-( I have a static IP address which is on my private network.. Phone is 192.192.192.220 and asterisk server is 192.192.192.22 I have the 7690 with a SIP iamge (Whatever latest is ) I have 3 lines setup with Free World Dial up and have the 4th setup to connect to my asterisk server.

Folks, from googling, I see that the dtmfmode parameter is not valid in the [general] context. My problem is that my overseas DID through Libretel seems to want to come into the [general] context! And, having done that, I get my welcome message, but then the DID does not accept the DTMF when I try to dial an extension! It plays the welcome message, waits, and then times out (and hangs up

Anyone help me here? I am a newbie so be gentle ;-)...... It worked once and then I played with the configs. I have a static IP address which is on my private network.. Phone is 192.192.192.220 and asterisk server is 192.192.192.22 I have the 7690 with a SIP iamge (Whatever latest is ) I have 3 lines setup with Free World Dial up and have the 4th setup to connect to my asterisk server. Here

You are reading the instructions for the STABLE 1.0 version of asterisk and are using the CVS version. Goto the wiki and read the instructions for RealTime. -Matthew ----- Original Message ----- From: "Muhammad Rizwan Khan" <rizwan@advcomm.net> To: <Asterisk-Dev@lists.digium.com> Sent: Wednesday, January 05, 2005 12:42 PM Subject: [Asterisk-Dev] Asterisk with MySQL >

Hi everybody Could someone give a tip on how can I configure asterisk to use 2 ATA's 186 to communicate each other using SIP with asterisk. I know this most be a very simple task, however this is the very first aproach I have to asterisk. I set the following config but I don't get dial-tone when I off-hook the phone from any of the two ATAs. Can some one tell what I'm missing in

HELP! Ok, so I have the following SIP.CONF: [general] context=default port=5060 bindaddr=10.1.1.200 externip = XX.XXX.XX.XX localnet=10.0.0.0/255.0.0.0 disallow=all allow=ulaw allow=g729 allow=g726 allow=alaw register => ##########@sip.broadvoice.com:XXXXXXXXX:##########@sip.broadvoice.com/1234 [sip.broadvoice.com] type=peer host=sip.broadvoice.com fromdomain=sip.broadvoice.com

I'm trying to setup a demo where my Asterisk box with a TDM01B (FXO) answers an incoming call and forwards that call to a SIP softphone (X-lite.) Seems all is built/installed okay: # ztcfg -vv Zaptel Configuration ====================== Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. I'm pretty new at this and the extensions.conf file is eating my

Hi, I have a problem with IAX accounts... I set up a few months ago an Asterisk server, with mysql support to load iax accounts. Settings seems fine because apparently the system works as expected. Yesterday I tried to add another iax account in the iax.conf directly. And I have a problem with this new account (named 444). I can authenticate from 444 to the server, and I can receive calls from

I'm having a strange problem recently with a Yealink SIP-T28P phone connected to Asterisk 11.4.0 via openvpn. It was working fine for months, and now when I dial anything from the phone, it shows "Forbidden", and the Asterisk console shows: [May 21 10:47:49] NOTICE[28518][C-00000004]: chan_sip.c:25189 handle_request_invite: Failed to authenticate device "Ext 110" <

hello, I try to set up two lines per ip 300 phone, registration is ok but i get Failure to authenticate 407 for subscribe. Anybody could help me to configure Asterisk in order to set instant message and presence ? I've tried with Ondo sip server it's ok ! Regards D?couvrez le nouveau Yahoo! Mail : 250 Mo d'espace de stockage pour vos mails ! Cr?ez votre Yahoo! Mail

Hello. I have an Asterisk server (ViciDialNow) set up behind NAT. I can manage to make outbound calls, but the communication drops off after 30 seconds or so. I'd really appreciate having some assistance from the mailing list on this issue. So, I'm having an Asterisk server behind a firewall and Zoiper softphones on SIP connecting to Asterisk on the same local area network. The

Hello, I installed 1.2.5 and realtime SIP. The connection to the DB is OK because I can get the values from the CLI. Here are my 3 different cases: 1- If I put an unexisting user, I get 404 and I am not able to dial. 2- If I check "Disable registration" within Firefly it does not register but I am able to dial a destination (...) 3- If I leave registration ON, I get the 404 message

Hello, I'm trying to get calls working between websocket clients and sip clients. For clients I have sip.js based clients on chrome, Zoipers and a Grandstream phone. Challenge here is I'd like to have Kamailio and rtpengine to handle the bridging between different rtp profiles but Asterisk changes them in the sdp bodies along the way. I'm using Asterisk 11.11.0. Is there a way to

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

I'm experimenting with asterisk. This is my environment: - Debian sarge (vanilla kernel 2.4.29) - Asterisk 1.0.7-BRIstuffed-0.2.0-RC8g - Two sip phones (One cisco 7905 and one soft-phones X-Lite) - Digi International Datafire Micro V (Europe) (rev 02) (zaphfc) After two days of work now I can call between the two sip phones and I can call from sip phones to outside numbers via ISDN card.