search for: 22092958

Does anyone know how to get any of these VOIP phones to allow me to do menu selections through asterisk, like when accessing voicemail and such. Thanks :P --

...'t. It listens on every interface, and uses every one of them for outgoing source addresses. This breaks IAX2 in my environment. Have I missed a necessary config file change, or is this a bug? -tih -- Tom Ivar Helbekkmo, Senior System Administrator, EUnet Norway Hosting www.eunet.no T +47-22092958 M +47-93013940 F +47-22092901 FWD 484145

Hi When I register a SIP or IAX client to asterisk and I dial to it from another UA then there is no problem at all But, when I register two or more clients to the SAME peer (with the same user/pass) and I call to this peer.. Then only the UA which registered the last will ring.. Others don't ring... What can I do about this?? I would like to register for example 10 UA's to the same

Hello, I'm trying to migrate my system to FreeBSD and the Makefile for asterisk-addons fails in the first make clean: bash-2.05b# make clean "Makefile", line 56: Missing dependency operator "Makefile", line 57: Could not find .depend "Makefile", line 58: Need an operator make: fatal errors encountered -- cannot continue I would like to think there is no

Hi, I'm slowly getting to grips with *. My next quest is to get IAX2/FWD calls working. I've setup a fwd account and added IAX capability to it via the website. I got the email saying it had been done with some welcome text and sample phone numbers to try, such as 10001 for the answer phone. I followed the setup example on the fwd site for configuring * to work with fwd's IAX.

Hi, i just installed latest asterisk on fedora rc2 and on the same machine i installed a sip soft phone called kphone. Kphone complains about /dev/dsp being used and can't place/answer calls (/dev/dsp is obviously used by asterisk) . how can "share" my sound card with these two programs? or can i disable the sound card in asterisk so i can use kphone to place/answer calls? BTW kphone

what is the best way to have 2 asterisk servers communicate with each other?

Hi everybody. I'm trying to find a way to connect two (or more) extensions directly without being kept in the middle during the conversation but it won't happen. The purpose here is to have asterisk running on a low bandwidth (128Kbps) internet connection just as some kind of a proxy between some ip phones with high speed (10Mbps) internet connections. SER is not an option, for now.

Hi, May I ask what ports are necessary for SIP communication through a firewall? I read somewhere that UDP/5060 alone is enough. Some recommends more ports to be opened for RTP. Regards, Norman Zhang

Hi all. Can someone recommend a good UPS for using with an * machine that provides some linux tested software to do managed shutdown in case of power loss? Thanks. Shoval Tomer, IT Manager, SofTov Advanced Systems, Ltd. Office: +972-3-9230686 ext. 179 Fax: +972-3-9216642 Mobile: +972-54-8000200

Got no responses to this, but the list seemed to be down for a while, so here it is again. Sorry for the extra bandwidth! John Hi, I've been messing with getting SIP working for days now, with limited success. I've got Asterisk set up on a remote server with the echo test. Please try it out to verify I've got the server working right: sip:robot at nixon.butchwax.com

Hello all, I am trying to upgrade the firmware on my cisco 7910 without using CCM. I was told that chan skinny is possibly capable of doing that and would like to make sure. I have P00405000600 firmware which I have put in version in skinny.conf. the phone basiclaly stops at verifying load. tcpdump shows nothing happening apart from small amount of traffic to port 2000 (skinny). Does anyone