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I'd like to grab the SIP response code that comes back from an INVITE. The HANGUPCAUSE gives the converted ISDN cause code. Anyone know of a way to get the SIP response code instead? Doug. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com -------------- next part -------------- An HTML

I'm confused about something. It's the way Asterisk handles the A leg (ie the first party dialed) on an originate command via the Manager Interface. Lets say our originate commands looks like this: ACTION: Originate Async: yes Timeout: 60000 Exten: callback Channel: SIP/5551212 at provider Variable: destination=SIP/8675309 at provider Callerid: 5551212 Context: default ActionID: 849120

Is it possible to set an outboundproxy for REGISTER from Asterisk? register => xxx:yyy at sip99.foobar.com [foobar] type=peer host=sip99.foobar.com disallow=all allow=g729 canreinvite=no secret=yyy fromuser=xxx port=5099 outboundproxy=xxx.42.149.69 However, SIP REGISTER still goes directly to sip99.foobar.com, not xxx.42.149.69. Thanks, Doug.

I just downloaded the app_vxml for Asterisk 1.2 from i6net. Couldn't get it to work. We're using Asterisk 1.2 still, and it looks like the app_vxml binary was linked against libstdc_++-5.x (we have libstdc++-6.x). I grabbed the 1.4 version of the module hoping in vain that would work, but it fails with invalid symbols, which isn't surprising. Any ideas on how I can get this to work?

We're doing callback here. Asterisk dials a number, waits for an answer, plays a prompt, dials a second number, and bridges the channels together. Calls are initiated from the AMI. No problems there. Easy stuff. However, I'd like to know if it's possible to have Asterisk dial the same two numbers simultaneously, play the prompt to the first one that answers, dial the second one and

Hope someone can help. I have a situation where asterisk is sending a SIP CANCEL message before the Dial() timeout has hit. It doesn't always do it. Normally, we send an INVITE to the ITSP. They respond with a 100 Trying, then a 180 Ringing, or 183 Session Progress. It seems to be at this point that Asterisk starts the dial timer. Normally, when no more replies have been received by the dial

So... I'm trying to access CDR(duration) and CDR(billsec) inside h... I keep getting 0. Can I access the CDR function inside a hangup extensions? Asterisk 1.4.13 Thanks, Doug. ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.

I'm confused about something.... In Asterisk 1.4 you can collect RTP QoS metrics at the end of a call with: ${CHANNEL(rtpqos,audio,all)} Now, when your using the AMI to do a callout, like this... ACTION: Originate Async: yes Timeout: 60000 Exten: callback Channel: SIP/1000 Variable: callid=849120 Variable: destination=SIP/1001 Variable: timeout=70000 Variable: timeout_warning=60000

Is it well known that setting the ActionID when connecting to AMI has absolutely no effect? Is this fixed in Asterisk 1.4? If you add an ActionID to your Originate command for example, it looks like the only events that come back with an ActionID associated are the initial response, OriginateSuccess and OriginateFailure. That's it. No other events have an ActionID associated. This pretty much

I'm a little confused. I'd like to build an RPM for Asterisk 1.4. Is it better to modify and use the spec file under redhat/asterisk.spec and run a 'make rpm', OR is it better to build a custom spec file from scratch and use 'rpmbuid -ba' <specfile>? How do people normally do it? The problem I see with a custom spec file is that since the source is all contained

I'm a little confused. I'd like to build an RPM for Asterisk 1.4. Is it better to modify and use the spec file under redhat/asterisk.spec and run a 'make rpm', OR is it better to build a custom spec file from scratch and use 'rpmbuid -ba' <specfile>? How do people normally do it? The problem I see with a custom spec file is that since the source is all contained

Does anyone know where I can get an rpm spec file for zaptel 1.4.x? Thanks, Doug. ____________________________________________________________________________________ Be a better sports nut! Let your teams follow you with Yahoo Mobile. Try it now. http://mobile.yahoo.com/sports;_ylt=At9_qDKvtAbMuh1G1SQtBI7ntAcJ -------------- next part -------------- An HTML attachment was

Not strictly an Asterisk question. I've tried to install adhearsion on TWO relatively fresh CentOS 5.x systems, and I get this... [root at localhost rubygems-0.9.5]# gem install adhearsion Bulk updating Gem source index for: http://gems.rubyforge.org ERROR: While executing gem ... (Errno::ENOENT) No such file or directory - /usr/lib/ruby/gems/1.8/gems/adhearsion-0.7.7/bin/ahn The

I have Asterisk 1.4 registering via IAX to another Asterisk machine. How can I change the default registration timeout of 60s? I need my Asterisk box to register every HOUR.... Anyone? Editting source isn't an option. Doug. ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs

Ok, so I've asked this question before, and didn't get an answer. So here I go again! Asterisk 1.4 has some channel variables that you can inspect after a call is complete that will give you QoS metrics. Stuff like average round trip time, etc. Since there's only one set of variables, and calls will have two channels, which channel is this information for? Is it for one of the

Ok, so I've asked this question before, and didn't get an answer. So here I go again! Asterisk 1.4 has some channel variables that you can inspect after a call is complete that will give you QoS metrics. Stuff like average round trip time, etc. Since there's only one set of variables, and calls will have two channels, which channel is this information for? Is it for one of the

It's time to ask this question again. Maybe I will get a reply one day. :) Asterisk 1.4 has some channel variables that you can inspect after a call is complete that will give you QoS metrics. Stuff like average round trip time, etc. Since there's only one set of variables, and calls will have two channels, which channel is this information for? Is it for one of the channels? Is it an

Using Asterisk 1.2, still. We are issuing a callback. User rejects the first two calls, but answers the third. For some reason, the Manager Interface outputs a CDR with disposition 'NO ANSWER' for all three attempts, eventhough the 3rd call worked. Is this an asterisk 1.2 bug? Doug. ____________________________________________________________________________________ Looking

I'm using i6net's vxml browser in Asterisk. I'm trying to work out how I can return the inputs from a menu or form back into the Asterisk dial plan. Is there a variable? It's not documented if it is. The exit tag apparently can be used to return a value (still trying to work out how to do that), but what about multiple values, such as with a form? Doug. --------------

I just downloaded Asterisk 1.6 beta 9 because I had read that there was a new bridge command. After looking through the doc/* documentation, I see no mention of a bridge application or AMI command. Does it exist? I am trying to take a bridged call, and redirect each to another destination, which I can do with the redirect() AMI command. After doing some dial plan processing, I would like to