search for: trunk3

...I only allow 1 call per trunk) and roll over to a second or third depending on that busy status Here's what I've got for a macro thusfar, but it's not working(fails if the 1st trunk is busy) extensions.conf: [globals] trunk_1 => SIP/trunk1 trunk_2 => SIP/trunk2 trunk_3 => SIP/trunk3 [macro-trunkdial] exten => s,1,Dial(${trunk_1}/${ARG1}) exten => s,2,Hangup() exten => s,102,Dial(${trunk_2}/${ARG1}) exten => s,103,Hangup() exten => s,203,Dial(${trunk_3}/${ARG1}) exten => s,204,Hangup() [from-internal] exten => _NXXNXXXXXX,1,Macro(trunkdial,+1${EXTEN}) ext...

...e set it up as well). The issue is that ASTCC tries to match the pattern *anywhere* in the called number, not necessarily only at the beginning. I have set up ASTCC Routes like this: 1800 Tollfree Trunk1 0 0 100 1416 Canada Trunk2 0 0 400 - other NANPA codes - 1 USA Trunk1 0 0 400 011971 UAE Trunk3 0 0 3000 - other international codes - Now, for "other international codes" I have not included all the countries, just the ones that are important for now. I has expected to add others as they became necessary. However, today somebody called "011966..." which is not one of the...

...9;s say that because of operational restrictions, I cannot just setup all 3 physical interfaces. What would really be nice, is if in the entries for each of the trunks to be able to use the bindaddr setting per trunk. This would allow me to be able to connect trunk1 from eth0, trunk2 from eth0:1, trunk3 from eth0:2. Thanks ~ T

Hi all, I'm trying to resolve a weird issue with SIP routing. I have a number of SIP trunks, from a selection of providers, all of which are registered in sip.conf: [general] context=default allowguest=no allowoverlap=no udpbindaddr=0.0.0.0 tcpenable=yes tcpbindaddr=0.0.0.0 transport=udp bindport=15060 srvlookup=yes allowsubscribe=yes

Hello all, I have no problems setting up trunk groups in general, but is there a way to set up a trunk group for outbound calls that includes channels on multiple servers? I might have missed something somewhere, but I couldn't find any reading about this topic. Thanks! Sean

...nk exten => _1NXXNXXXXXX,2,Macro(nocdial,${TRUNK}/${EXTEN}) exten => _1NXXNXXXXXX,3,ResetCDR ; On Failure, Dispatch Second Trunk exten => _1NXXNXXXXXX,4,Macro(nocdial,${TRUNK2}/${EXTEN}) exten => _1NXXNXXXXXX,5,ResetCDR ; Third time is a charm? exten => _1NXXNXXXXXX,6,Macro(nocdial,${TRUNK3}/${EXTEN}) [macro-nocdial] exten => s,1,Dial(${ARG1},30) exten => s,2,Goto(s-${DIALSTATUS},1) exten => s-NOANSWER,1,Hangup exten => s-BUSY,1,Busy(15) exten => s-BUSY,2,Hangup exten => s-CONGESTION,1,NoOp exten => s-CHANUNAVAIL,1,NoOp exten => s-.,1,Goto(s-NOANSWER,1) --...

...t when determining what channel to call fwd ... This new local channel is realy cool works like a function calls in the extension.conf vs a goto / gotoif whic does not return to the calling context.. ! ROCK on Mark ! [globals] ;outgoing Trunks TRUNK1=Zap/g1 TRUNK2=IAX/myUserName@SomeIAXBox.com TRUNK3=SIP/bah@iconnechre.com ;key associating extension/user to a channel via a DBGet EXT1=1Zap ;see Incoming DBGet Lookup uses this family/key eg. CallFwd/1Zap EXT2=2Zap ; ect ; SPEEDIAL=megacontext ;context for call fwds when using extensions [exts] exten => 11,1,Dial(Zap/1,20,tTrm) ;dial ext on...

...(and they then hear 'busy'). (using Congestion instead of Hangup gives the same behaviour here too but the person calling out still gets the one 'ring' before the busy tone - also removing the ,r makes no difference either) Also, if TRUNK2 were busy would it be possible to go to TRUNK3 by defining context 204 or not ? Hoping someone has the patience to laugh at what I did and suggest how I could fix it if it's fixable ?? maybe ? - please ? Thanks for listening. Have a great rest of weekend. Chris

Hello list, I try to setup an asterisk-server with different SIP-Peers to PSTN. The Peer are working and configured in sip.conf: [peer1] type=peer host=10.10.10.1 [peer2] type=peer host=10.10.10.2 Now dialout is no problem. Extensions.conf says: exten => _0Z.,1,Dial(SIP/49${EXTEN:1}@peer1,30) But how can I setup a failure-route if the SIP-Proxy "peer1" ist not

Is there any way to have users be able to turn on or off call forwarding at the asterisk server, so they can configure their own forwarding number and enable/disable it? Hopefully, with the added benefit that it will remain on between server reloads and restarts? I have written a hack -- a AGI script to do various checking, and if the destination is "ok" set a database variable

Has anyone ever setup Asterisk to pass Feature Group D access while using a CIC code for outbound calls? If so can you please email the configuration you have done? I have tried to get this up and running but with no luck. I have also contacted support and I cant seem to get this going. Thanks in Advance, Jason Miller

I''ve been wrestling with this all night, I''m hoping someone can help. I followed the exact steps in: http://wiki.rubyonrails.org/rails/pages/HowtoUseRailsWithSubversion ..but when I open a new ''Checkout project from SVN'' in RadRails, it opens up the second level dirs as the project dirs (ie. app, log, script, etc), leaving me with a mess of projects. I redid

Hello all, I am looking for a way to restrict users from logging in two separate phones with the same authorization name/password at the same time. Meaning, I only want users to be able to place a call from one phone in one location, but have the ability to move from computer to computer. Has anyone found any sort of solution for this type scenario? Thanks, Bryan Mahin Please visit us @