search for: ondo

HI All, I am new to asterisk. i want to make outbound calls from asterisk. I tried with many times with the given settings but in vain This is my scenario: I have a *user A* who has registered with sip server(ONDO), I made asterisk to register as a sip client with ONDO, I want to make a call to user A from an extension. My configurations sip.config [general] context=default register => raja:adsi6677@*192.xxx.xxx.xxx*/1234 bindport=5060 ; UDP Port to bind to (SIP standard port is...

...ic canreinvite=no context=default allow=ulaw [1001] type=friend nat=yes host=dynamic canreinvite=no context=default allow=ulaw extensions.conf ============ [default] exten => 1000,1,Dial(SIP/1000) exten => 1001,1,Dial(SIP/1001) *Case B:* I have register other phone with ondo sip server running on other PC [192.168.1.12] with number as 6001. Now, Want to make calls between this two [asterisk PBX [1000/1001] on 192.168.1.11 and ondo [6001] on 192.168.1.12]. Please suggest how to configure sip trunk in conf files. Thanks in advance. Regards, Garge. ------------...

I cannot get incoming calls to sip phones behind a PIX to work, outgoing is fine. Asterisk (Public IP) --> Internet --> PIX (NAT) --> Sip Phones I have tried no fixup protocol sip, I have punched a hole in the Pix allowing anything from the Asterisk box into the network, still no incoming. I have done all the Wiki suggests in regarding to NAT. Is their a trick getting the

...ll my PST lines. The main problem that I have is with the outgoing calls from the 1204. I can only make one call and then after the outgoing call ends, I have to reset the 1204 in order to call again. Most of the calls (85%) we make from external lines (PST) work. I am using a mediatrix 1204 and ondo sip server in one of our offices and a mediatrix 2102 in the other office. Have you ever experienced this problem before? I read that some of you installed mediatrix devices with a SIP server and it worked OK. Could you please tell me which SIP server you used, and how did you configure it on th...

...very compatible so I will use a SIP FXO gateway then. Untill I find one, im just trying to get my 2 cisco SIP phones talking to my * server. just as a learning experience for now. heres what I have so far: 2 Cisco 7960's both using DHCP and both registering with my SIP proxy server (Brekeke OnDo on windows XP) they can call eachothers extension and chat no problems. Fedora Core 2, and The latest Asterisk from CVS compiled and running. and one useless modem card :) what im trying to do is get * to register with my proxy server and then be able call * from my SIP phones and my SIP phone...

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 sur http://fr.mail.yahoo.com/ -------------- next part -------------- ; ; Static extension configuration file, used by ; the pbx_config modu...

...AMP and have several 7960's connected to it. The asterisk server has a public ip and all the 7960's are behind nat'd routers. When I try to call from extension to extension I get directed straight to voicemail. I do not have any cards installed and instead direct everything to an Ondo server. I have been told it's not an AMP problem, I'm just totally lost on where to look for a solution. I have noticed when running ztcfg -v I have 0 channels configured, do I require some type of pseudo channels to make this work? If I dial extension 2346 this is what my logfile lo...