search for: trunk2

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=aor contact=sip:100.66.1.104:5060 (In my above example,simple...

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 trunk2 by add...

...t 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 stops looking fo...

Hello, I can not seem to be able to get StripMSD and Prefix to work for me in extensions.conf. This is an example of what I have: exten => _050.,1,StripMSD,1 exten => _50.,Prefix,01051 exten => _001051.,1,Dial(${TRUNK2}/${EXTEN}) exten => _001051.,2,Busy exten => _001051.,102,Busy What I want to achieve is to call 001051501657887 on TRUNK2 when dialing 0501657887. dialing 0501657887: -- Executing StripMSD("SIP/3267915-7e9a", "1") in new stack ...this is the only thing I get on the cons...

...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 => _NXXNXXXXXX,1,Macro(...

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: 50e9132765782741404408k2469rmwp) - No matching e...

...ring 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 giving up) if that is busy (therefore it is likely the number really is busy then grab the caller and han...

...secure = 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.

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/attachments/20061028/0879628c/attachment.htm

...voice frame". Anyone encounter this before? Asterisk is bridging two SIP lines together, so the technology should be the same. Maybe I'll try allowing only ULAW. ************************************** Asterisk Standard debug (level 3) *************************************** -- Called trunk2/12095387895 [Nov 26 13:49:37] WARNING[7744]: channel.c:3021 set_format: Unable to find a codec translation path from unknown to unknown [Nov 26 13:49:37] WARNING[7744]: channel.c:3402 ast_channel_make_compatible: Unable to set read format on channel SIP/trunk2-0990c538 to 524288 -- SIP/trunk1-098dc...

...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

...hem could return a SIP error message (maybe caused by DOWN E1s) Googling I found a few possible solutions: 1. Using DIALSTATUS variable. 2. Dialing in sequence: exten => _X.,1,Dial(SIP/${TRUNK1}/${EXTEN}) exten => _X.,2,Dial(SIP/${TRUNK2}/${EXTEN}) 3. ChanIsAvail Using the first method it's possible to get the CONGESTION and CHANUNAVAIL status which pretty much solves my problem but it takes more than 2 lines of dialplan(I like one liners). The second solution requires less space in the dialplan but it sho...

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

...retty 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] exten => 1...

...e 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 they became necessary. Howe...

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

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

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 @

...ions.conf logic that I am using [e164] ; Dundi exten => _1NXXNXXXXXX,1,Macro(dundi-e164,${EXTEN}) ; Dispatch First Trunk exten => _1NXXNXXXXXX,2,Macro(nocdial,${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-...

Hello! However I'm a newbie in Asterisk/VOIP/CM I would like to make sure that this system design can work: Cisco 2811 Voice Gateway - sip trunk1 - asterisk on linux box - s?p trunk2 - Cisco Call Manager 6.0 There is also a Siemens Hicom old pbx connected with QSIG to the Voice Gateway. I would like to record all calls with mixmon going through Asterisk. Is it possible? Also if there is call forward/find me/call transfer etc. in Cisco CM, how can I find the change in Asteris...