search for: trunk1

..., in sequence. I need to detect a busy SIP trunk(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 =...

On 15-04-09 12:06 PM, Chad Wallace wrote: >> but don't know where to put those lines. I have BABY defined as >> >channel variable: >> > >> >BABY = SIP/babytel_out >> > >> >but that seems circular, somehow. > You put them in the context for your clients... From what you show > below, I'd say they go in the "local_200"

Hello, For lab testing, I'm trying to build two differents PJSIP trunks between two Asterisk 13.8.0enabled boxes. I thought I could set up both trunks like this: Box A/port 5060 <------ Trunk1 -----> Box B/port 5060 Box A/port 5062 <------ Trunk2 -----> Box B/port 5062 and declare trunks like this: [foobar1] type=endpoint transport=simpletrans context=from-customer aors=foobar1 allow=!all,alaw [foobar1] type=identify endpoint=foobar1 match=100.66.1.104:5060 [foobar1] type=a...

...but that be pretty easy just to add some more promt 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]...

...st October when I was really new to Asterisk and just accepted the behaviour until now when I'm wondering if it can be refined - I've tried and failed and read all that I could. My dialplan is for the outgoing SIP call is: exten => _00.,1,AbsoluteTimeout(3600) exten => _00.,2,Dial(${TRUNK1}/${EXTEN:2},45,r) exten => _00.,3,Answer exten => _00.,4,Hangup exten => _00.,103,Dial(${TRUNK2}/${EXTEN:2},45,r) exten => _00.,104,Answer exten => _00.,105,Hangup (if call can go through on TRUNK1 send it out, if TRUNK1 is out of capacity and therefore busy then try trunk 2 before...

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

...ppreciate that too. :) What I'm trying to do is set up some 'special' extensions in my internal context to change variables, or change something else in the session before dialling. To be clearer, here's an example. Say I've got this rather simple dial plan: [globals] TRUNK1=Zap/1 TRUNK2=Zap/2 TRUNK=${TRUNK1} [internal] _NXXXXXX,1,Dial(${TRUNK}/${EXTEN}) I'd like to add an extension which I can dial before placing the actual call to change which trunk I'm using, like so: *55,1,SetVar(TRUNK=${TRUNK2}) The problem is that once that's done, asterisk sto...

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 guys. Have 2 external numbers that required registration on provider server, trunk1: 73432260005 at 80.75.132.66 trunk2: 73432260050 at 80.75.132.66 Thing is I can?t figure out how to route them to different IVRs by default Asterisk can?t match endpoint Request from '<sip:+ 73432260005 at 80.75.132.66>' failed for '80.75.132.66:5060' (callid: 50e913276578...

...t on the overall system availability. A lot of questions howevere arise, like: what if one SIP user got REGISTERed at Server 1, and the other on Server 3, so how can they call one another ? Also, outbound registrations can be done from one instance at a time, say it's done from Server1 for Trunk1, so how can users, that got authenticated at Server2, call thru that registration (Trunk1) ? Also, Kamailio itself has to be protected from failing, and probably even from overload... Would be great to read something in-depth about that Thanks!! Kirill Marchuk

...roblem with ASTCC, but may be a problem the way I have set up ASTCC (and possibly the way others have 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...

...plan for outbound calling looks like below. Which I know they are getting to the Congestion part (which explains the busy) but what I can't seem to figure out is the cause for why they are getting sent there. exten => s,1,SetCallerID(${ARG1}) exten => s,2,Wait(2) exten => s,3,Dial(${TRUNK1}/${ARG2}) exten => s,4,Congestion(10) exten => s,104,Congestion(10) The log for a call looked like this May 26 12:21:08 VERBOSE[6997] logger.c: -- Channel 0/4, span 1 got hangup request May 26 12:21:08 VERBOSE[16613] logger.c: -- Zap/4-1 is circuit-busy May 26 12:21:08...

...is as follows. [46.19.209.1] host = 46.19.209.1 type = friend insecure = invite context = from-didww canreinvite=no The other is as follows: [62.180.237.73] host = 62.180.237.73 type = friend insecure = invite context = from-btnet2 canreinvite = no The problem is, i get all calls coming from trunk1(didww) without a problem but, when i receive a call from trunk2(btnet) it tries to authenticate the sip call and denies it. It works only if i allow guest calls. What can be the reason for that? Thank you.

...e same result) How t works in 1.4.20.1 is as follows: We trigger call via Originate action as follows: action:.Originate.. actionid:.1306903_89#AJ_ORIGINATE_25 timeout:.40000 exten:.s async:.true callerid:."".<61211111111> context:.campaignType_5 priority:.1 channel:.SIP/trunk1/61212142321 And the campaignType_5 context looks similar to: [campaignType_5_] exten = s,1,Answer() exten = s,n,Set(timestarted=${STRFTIME(${EPOCH},,%Y%m%d%H%M%S)}) exten = s,n,Set(CALLSTATUS=0) exten = s,n,Background(lyrics-louie-louie) exten = s,n,WaitExten(5) exten = s,n,Set(timefinished=${STR...

...ms. I have a Digium 4 port card with POTS lines plugged into all four ports. How do I play the congestion tone the the caller when they try and dial out but all the lines are in use? should something like this work? [dial-trunklocal] ; Local calls ignorepat => 9 exten => _9NXXXXXX,1,Dial(${TRUNK1}/${EXTEN:1}) exten => _9NXXXXXX,2,Congestion exten => _9NXXXXXX,3,Playtones(congestion) exten => _9NXXXXXX,102,Busy exten => _9NXXXXXX,103,Playtones(busy)     __________________________________ Do you Yahoo!? New and Improved Yahoo! Mail - 100MB free storage! http://promotions....

Hi, I want to do so that extension 501 will always use trunk1 for outbound calls and 502 will use trunk2 for outboud calls. How do I do this. Right now all extensions use the same trunk for outbound calls. -- Zeeshan A Zakaria -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/att...

Hi, In my dial plan I've configured two trunks to make outbound calls (trunk1 and trunk2) to same service provider but I want when any of my exten starts with _2. should goto trunk2 and there should be some kind of disturbance (like some noise or some background noise) when my calls goes to trunk2 to make the call quality bad. Mainly I want to achieve bad call quality on tr...

...DR fields in the 2 versions) . Previously, I would place a call via an AMI Originate action similar to: action:.Originate.. actionid:.1306903_89#AJ_ORIGINATE_25 timeout:.40000 exten:.s async:.true callerid:."".<61211111111> context:.campaignType_5 priority:.1 channel:.SIP/trunk1/61212142321 And the campaignType_5 context would handle the call processing, capturing a failed dial attempt in the failed exten : [campaignType_5_] exten = s,1,Answer() ... exten = h,1,Set(timefinished=${STRFTIME(${EPOCH},,%Y%m%d%H%M%S)}) exten = h,2,GotoIf($["${TIMETOPRESS}foo" = &...

...able IPs, and for this purpose, let'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, What does this message imply? [Apr 29 14:46:30] NOTICE[32175] channel.c: Dropping incompatible voice frame on IAX2/trunk1-9085 of format alaw since our native format has changed to 0x4 (ulaw) If voice frames have been dropped then I suppose that the call quality may be affected? Vieri