similar to: ATA losing registration problems solved by setting tftp

When I try to connect to * using a Cisco ATA 188 configured with a MGPC firmware (v3.1.1), I just keep getting this message every 30 seconds or so : May 10 10:08:21 NOTICE[7913]: chan_mgcp.c:1509 find_subchannel: Gateway '192.168.1.27' (and thus its endpoint '*') does not exist Using tcpdump, I have checked that the ATA188 (with IP 192.168.1.27 and port 2427) sends UDP packets to

Hi, after reading about it on the list I decided to set up a Free World Dialup account. For those of you who don't know, that is a sip proxy where you and your friends can singn up free and then you can just connect to it with any sip client and call anybody that is registered for free. Pretty much like iaxtel (I belive that was the name of it) for the iax protocol. It even supports clients

Hi, I am finally biting the bullet and migrating our NT4 domain to Samba. I am using the following guide: http://vermeulen.ca/linux-windows-nt.html I installed a fresh copy of FC11 and installed samba 3 through yum. Hostname of the linux machine is LEONIDAS. the DOMAIN name is GENOME1. I created an account on the NT4 domain for a backup DC under the server name LEONIDAS. testparm returns that

I have recently started experimenting with Asterisk. I am running the system the other side of the a NAT router and trying to connect to FWD. I have opened UDP ports and have configured sip.conf to handle NAT. The problem: I can call from the FWD phone and the extension on Asterisk rings and there is two way sound so no problem. Now if in the extension.conf file I have, exten =>

Is this replacing Astricon this year? If so it looks like a pretty poor showing in comparison to Astricon Dallas last year. Cheers, Dean ________________________________ From: Carl Ford [mailto:carlf at vonmag.com] Sent: Wednesday, 18 July 2007 9:09 AM To: Dean Collins Subject: Announcing Digium/Asterisk World's Conference Program

Same thing. It will make sense to try Register => <FWDnum>@fwd.pulver.com:<FWDpass>@fwdnat.pulver.com:5082 but in that case Asterisk sends REGISTER sip:fwdnat.pulver.com SIP/2.0 which is not right. It should be sip:fwd.pulver.com but sent thru fwdnat.pulver.com:5082 BR Borut -----Original Message----- Subject: Re: [Asterisk-Users] Using Asterisk with FWD through NAT From:

I have a small HUGE problem with *. I have installed * but I have 2 problems. 1 - Making call to FWD. 2 - Receiving call from FWD More info of the problem at the end. Here is the sip.conf file. ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = sip ;default Default for incoming calls register =>

Hi, My configuration: Sipura 2000 Debian/Sarge Asterisk 1.0.1 built by msp@toshiba on a i686 running Linux I am calling 612@fwd.pulver.com which is Daytimephoneline of pulver.com and for the first second the connection seems to be ok and I hear: <thu .. rsda ..> and nothing more, which suppose to mean <Thursday, ...>. The echo line from 613@fwd.pulver.com does work the same. I never

Using DIAX softphone which seems to be working OK can get to VM/echotest etc in the demo context Am trying to setup FWD but get the following problems Can hear it ringing when dialing FWD no 612 for time. Connects but no sound from remote end. Does anyone have any suggestions. Softphone on 192.168.0.2 asterisk on 192.168.0.3 Netgear RP114 doing NAT to the internet port 5060 being forwarded to

Hi, I've posted a simular message little over a week ago so sorry for reposting. I need to register to freeworld dial up from behind a nat. Using the xten software sip client works fine but with asterisk I don't know how to do it. Last time I posted I got different responses. Some saying I can't register with an outbound proxy from asterisk others said they have done it. If it is

I have problems running poker websites. Ultimate Bet and Absolute Poker will run, but barely, it is like they are "struggling" and they sometimes just get "stuck" and I have to "Force Quit" . And I click on a tab like Cashier or Requests that would take me from the GUI to a website and nothing happens. I can't go from the GUI to another site, but I can go

Can multiple FWD accounts be registered? I have the following output in my sip.conf file: register=74928:xxx@fwd.pulver.com/74928 register=75160:xxx@fwd.pulver.com/75160 register=74573:xxx@fwd.pulver.com/74573 [fwd-74928] type=friend secret=xxx username=74928 host=fwd.pulver.com [fwd-75160] type=friend secret=xxx username=75160 host=fwd.pulver.com [fwd-74573] type=friend secret=xxx

Hey all, I'm trying to get my Asterisk server up and running on fwd.pulver.com just to get the hang of it until I get my FXO card in a couple of days. It seems to connect but that's about it. If I try to dial into it from another fwd # it says user is not online. In sip.conf I have the following added: register => xxxxxx:xxxxxx@fwd.pulver.com/489125 [fwd.pulver.com] type=friend

I'm hoping someone can help me out with this. I am basically just trying to separate my outbound and inbound calls into separate contexts instead of having everything in a single context. Any help would be appreciated. Perhaps I've missed something really obvious.... Here is the network layout: <remote> <--TDM400P--> <nattedbox> <--IAX--> <liveipbox>

Too bad it is August 6th *P.S. Remember, as a member of the Digium Family we have secured a special discount of 50% off of the conference fee for you if you register by July 29, 2007. To take advantage of this limited time offer, please register here <https://secure.pulver.com/cgi-bin/von?mode=gpur&conf=dawfal07&type=g&pricode=billm>!* Digium, Inc wrote: > > If you

I have tried everything and still can't place / receive calls from the fwd network. At one point today I was able to call my test machine on the fwd network, I'd answer the call on the test machine (which stated Call Connected), but then the computer I was calling from, through the Asterisk server would give me a 403 Error. I am using sjphone software. I am able to call various

Hi All, Is there any way to connect (register, initiate and receive calls) with Asterisk to FWD through NAT? Since I own my router port forwarding is not a problem. I tried with Register => <FWDnum>@fwd.pulver.com:<FWDpass>@fwd.pulver.com but since Asterisk still use internal IP in some SIP fields I got "479 We don't accept private IP contacts. Please set your external

Hi all. I have this configuration: Telco <-----(E1)----->TE410P//Dual Xeon Server 2.4Ghz<-----(Ethernet)----->Switch<----->GS//BT The Server is running RedHat Linux 8.0 with kernel 2.4.18-14-smp and we are having the following 2 issues: 1.- When making calls from the GrandStream to the PSTN the audio is choopy, plus theres is a pulsing sound, but when the GS

Hi forum, I have been fighting days and days configuring FWD and asterisk with NO success I have the following scenario. My sister in Spain with FWD dialup client My question is if she can dial my FWD dialup number, which is registered in Asterisk and the call being forwarded to ring my IP Phone. Spain LAN FWD

I've been having issues getting FWD to work. I posted this same Q to the FWD forum (no responses yet), but I was hoping someone here had some insight: http://yabb.pulver.com/cgi-bin/yabb/YaBB.cgi?board=news;action=display;num=1072263468;start=0#0 I just signed up for an FWD account (I know I had one before, but I lost it.. :) I've got it running through Asterisk - all working fine