similar to: SPA-3102 Registration Failed .. need advise

Dear The Expert, I am very new with this, I don't understand why my Windows 2000 Prof PC failed when trying to register as member of domain "LINUX".. but I am able to login by using Windows 9x client , below is my /etc/samba/smb.conf I need advise.. thanks a lot in advance [global] workgroup = LINUX server string = Samba Server printcap name = /etc/printcap load printers = yes

Hi all, I am very new with samba, I am trying to configure my samba as PDC, my samba is running on FC2, I have configured my /etc/samba/smb.conf as below: now,I cannot login to my samba .. I am very sure that I already missed something.. please advise what's am I missing? many thanks in advance Regards Winanjaya [global] workgroup = edp netbios name = portal server

Hello, i have a SPA3102 and asterisk v. 1.2.18. I also hev a mysql database with cdr. Well all I want is to receive incoming calls from pstn on specified sip account (suppose 8000), and to initiate outgoing calls from all my asterisk sip accounts through SPA3102 device. Someone can explain me what may i set on SPA and asterisk to do this thing. Thank you for your support. -------------- next part

Dear The Expert, I am very new with this, I have installed AsteriskNow, X-Lite as my SoftPhone, I am using SPA-3102. I have 3 extensions, me at 250, 998 is my Linksys SPA-3102 and 999 for PSTN Line (see below) My problem is, I am unable to call 998, I thought this is registration problem, (because the Linksys screen info said Registration Failed) Could any body please help? Many thanks in

I've got PBXNSIP running on a windows box and it is trying to register with my Asterisk box. I can set up one trunk and it works fine, however if I try to setup a second trunk from the same box, there is some sort of authentication issue where Asterisk appears to be confusing which trunk is which. Here is the chunk from my sip.conf: [TEST1] context=STUFF-LD type=friend

Asterisk Users, I recently ran into some problems with the quality of service with Teliax. This occurred on August 1, 2007 with a dropped outbound call, audio quality isse on the callee side- not hearing me well on callee side, and sending DTMF tones (configured for RFC2833). Am I the only Teliax customer having this problem? It seems like when I am ready to go live with my Asterisk

Hello, I am wondering how to send a call to a specific IP address that is different than the host of the URI. For example, an invite to the URI is " john at phone.com" needs to be sent to the IP address 123.456.789.255, not to the IP address of phone.com. How is this done? Thanks, Elliot -------------- next part -------------- An HTML attachment was scrubbed... URL:

Good Day list I am looking to buy a few Wooksung Video phones to try with my asterisk box.... http://www.wooksung.com/eng/html/pro/pro_001.html has anyone had any experience using these with asterisk? Thanks Ron

Can anyone please direct me to UK based suppliers of equipment. Website URL's would be appreciated. TIA -- ***** Not everyone is touched by an Angel.... .... Those that are, never forget the experience *****

The versions of avr-gcc in linux repositories mostly do not work. I've been trying to build avr-gcc from sources, using scripts provided by Bingo600. So far the effort has not succeeeded. See here http://www.avrfreaks.net/index.php?name=PNphpBB2&file=viewtopic&t=42631 and here http://www.avrfreaks.net/index.php?name=PNphpBB2&file=viewtopic&t=124056 for some of what has gone

Hello: I am testing Asterisk with oh323. My question is: can Asterisk route some calls thru a second h323 gateway (a h323 <-> PSTN gw)? - Asterisk ip: 192.168.1.10 - h323<->PSTN gw: 192.168.1.20 I've tried: exten => _9XXXXXXXX,1,Dial(OH323/192.1.1.20) or exten => _9XXXXXXXX,1,Dial(OH323/BYEXTENSION@192.1.1.20) but it does not work at all. If my h323 client

I am hoping some of you can help me troubleshoot this problem I am having with my home asterisk machine. I have incoming POTS service using a SPA-3000 (extension 119). Calls on that line go to an attendant recording that offers a menu choice: press 1 for Nancy, press 2 for the rest of us. In reality, pressing anything other than 1 sends the call to the rest of us by dialing both extensions 101

Hi, Is FAXing with Asterisk a practical option ? Or is it better just to use a plain fax connected to an FXS and just switch with Asterisk. I specifically wanted to know if there was any experience using just the fax scanner to send faxes and receive them via asterisk and the to e-mail. My idea was to take my old fax connect it to an FXS port and send faxes with the fax machine (using the fax

if anyone ever serches here for using responsa in linux then you should know that: it works! for more info look here: http://www.whatsup.org.il/index.php?name=PNphpBB2&file=viewtopic&t=31072&start=30&postdays=0&postorder=asc&highlight=

Hey Guys! We had migrate asterisk 1.2 to 1.8 now big issue is queue system. Before we had 3 queues and we were using AgentCallbackLogin but now its quite difficult to use AddQueueMember. Is there any easy way to logged into multiple queue using AddQueueMember ? and restrict agent for specific queue ? -S -------------- next part -------------- An HTML attachment was scrubbed...

Hi All, What is the going rate for the Snom 100 in the UK? I've found a couple of suppliers with prices around the ?170 (exc vat) mark. Regards, Nathan.

Hi All, Can anyone recommend a supplier and/or a particular a model of SIP voip phones in the UK? I don't seem to be having much luck locating any. Thanks, Nathan.

Hello I just read this article about a kid in England who built a box with a 3G SIM card: www.dailymail.co.uk/sciencetech/article-1394448/Doorbell-tricks-burglars-thinking-youre-home-invented-schoolboy-Laurence-Rook-13.html When someone rings your intercom, the box will call your cellphone so you can answer just like you were home. I don't know anything about electronics and would like to

Hi, I am running asterisk 1.8.5.0 and have compiled in the srtp module All but Snom phones are working. I have set the srtp tag on the snoms to 80 and RTP/SAVP to mandatory and they worked for a few hours. This morning all snoms are reporting this when trying to make a call (this is snom calling snom). ---------snip------------------ == Using SIP RTP CoS mark 5 -- Executing [10000 at

My wife and daughter, and to lesser extent myself and my daughters boyfriend would like a communications system which allowed us to talk to each other, both on a one on one basis, but also occassionally in conference. My wife and I live in a house with an internal LAN with each of us with a desktop machine (hers in Windows XP, mine runs Linux) and a Linux server acting as firewall and NAT