similar to: Dialing out to 2 clients simultaneously

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 --

I'm running CVS HEAD. The last time I updated was January 7th, at which time everything was fine. Having updated again today, January 17th, I'm having problems with IAX2. I use the "bindaddr" directive for both SIP and IAX2, and while SIP respects it, IAX2 doesn't. It listens on every interface, and uses every one of them for outgoing source addresses. This breaks IAX2

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

Hello all! Has anyone expirienced the following:? With an IAXclient softphone (like diax/iaxcomm/etc) Dialing to the PSTN (zap) or a SIP device has no problems .. but when I make calls between 2 softphones I have weird problems.... in about 4 out of 10 IAX-2-IAX softphone calls I get a big delay .. in the beginning of the call it's all okay... (delay < 0.5 sec) but the longer the call

After updating to the latest CVS Feb 17 15:20:03 WARNING[15317]: config.c:819 read_config_maps: The 'sipfriends' table is obsolete, update your config to use sipusers and sippeers, though they can point to the same table. == Binding sipusers to mysql/asterisk/sip == Binding sippeers to mysql/asterisk/sip Feb 17 15:20:03 WARNING[15317]: config.c:823 read_config_maps: The

Hello all! Is there any possibility to determine the IP address of a caller in my dialplan? I would like to have a predefined channel variable like ${CALLER_IP} but it seems it doesn't exist (http://www.voip-info.org/wiki-Asterisk+Variables) .. is this list complete? Are there any other possibility to store the SIP/IAX caller's IP address on every call? Thanks Niels

Hello maybe someone can help me? I did the CVS checkout and then compiled asterisk Then I tried to compile the addons and got the following (don't understand what's wrong at all and can't find anything about this error on google/wiki) app_addon_sql_mysql.c:164:64: macro "AST_LIST_REMOVE" requires 4 arguments, but only 3 given app_addon_sql_mysql.c: In function

Perhaps stupid question but, is Asterisk a statefull or stateless proxy? Ignace

I was looking at the wiki on 'Asterisk as a voice/fax switch' And was wondering if the extension 'fax' is global to extensions.conf Or just to the context it is in? The reason I ask, is that my PRI might have 5 channels that will be scrictly Fax, and to be functional, I need multiple 'fax' extensions in my various Contexts. Hope that makes sense, Paul Seniuk

Hi, The "all" zone you can use in /etc/shorewall/policy isn''t valid in /etc/shorewall/rules, is this correct? I was entering a rule to (for example) block all TCP port 12345 traffic from all sources to all destinations, and logically thinking I began typing this line. REJECT all all tcp 12345 But it didn''t work :-) If I have to enter the zone names, I would

Hello, I'm looking for a similar feature... Dial a number via ZAP/g1 after the line gets answered wait 10 seconds send DTMF Regards, Marc -- Network Manager Marc Storck LuxAdmin.Org mstorck@luxadmin.org Internet Service Provider http://www.luxadmin.org 15, route d'Esch Phone: +352 2727 3030 L-4544 Belvaux Fax: +352

Hi .. When I use the GoTo statement in realtime to goto a priority only ... E.g. Goto(3) then there's no problem But, If I try to jump to another context ... E.g. Goto(othercontext,${EXTEN},3) then it doesn't work If I process the same statement in extensions.conf things go well Are there things broken regarding GoTo in combination with Realtime Extensions ?