similar to: IAX2 doesn't respect bindaddr?

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 --

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 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

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, I build dovecot tih --with-rawlog and created a folder dovecot.rawlog inside the root-dir of the mailbox (same level as new, cur and tmp). For testing purposes I set the env MAIL to the maildir and started libexec/imap. This generally works, but I can't find the promised logs of input and output. Where do I have to look for them? Regards Marten

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I know this is a bit off topic but we are using Asterisk :) Since this list is full of tech gurus w/ all different sorts of backgrounds, I thought I would get the best opinions here. We have several different switches and other telecom equipment at our facilities which all have their own proprietary cdr platforms, which are rather limited. The company I work for is looking to develop their

Hello everyone, ff123 compiled Monty's branch of the RC1 encoder, see his post on r3mix.net forum: http://66.96.216.160/cgi-bin/YaBB.pl?board=c&action=display&num=994299736&start=30 Anyway it only supports ~128kbps mode, so I did a quick listening test with some files that bugged vorbis beta4. grace.wav - the right channel is still a bit watery, and I think this can be seen

There was an interruption in mail service and your memo was not sent. Please re-send your message. Thank-you -----Original Message----- From: samba@samba.anu.edu.au <samba@samba.anu.edu.au> To: Multiple recipients of list <samba@samba.anu.edu.au> Date: Sunday, November 01, 1998 1:24 AM Subject: SAMBA digest 1861 > SAMBA Digest 1861 > >For information on unsubscribing

Hi, I'm trying to configure a dual homed asterisk server with iax accepting connection on all address. from iax.conf.example bindaddr = 0.0.0.0 Address to bind to ( all addresses on machine ) but if i register a client using the second ip address i will receive the response from the first ip address and obviously the client discard this. let me explain more: Client : 192.168.0.4

Hey everyone, I'm an * newbie and I'm trying to get things going here, and please forgive me if you've heard this problem before. I have my * server on OS X behind a NAT, and I'm trying to connect to SIP soft phones (X-Lite) on various computers, some behind the same NAT and others with public addresses. I have successfully configured port forwarding and a few minutes ago I was

Hi all, is it possible to bind SIP protokol not to all but to more that one interfaces. I did try use bindaddr, but i don't know right syntax. Could anyone help me. Thanks, Bob.

Hello all, I am running asterisk on VMs with standby heartbeat configuration, Heartbeat assigns a virtual IP 172.20.255.40 on machine afterwards asterisk is started. In the sip.conf, I have explicitly define bindaddr=172.20.255.40 but sometimes I see packets coming from physical IP 172.20.255.41 I have both tcp and udp transport enabled Here is the lsof -ni :5060 output asterisk 2878 asterisk