search for: allison7

...- Goto (macro-dialout-default,s,6) -- Executing Dial("Zap/4-1", "ZAP/g0/0827751492") in new stack == Everyone is busy/congested at this time -- Executing Macro("Zap/4-1", "outisbusy") in new stack -- Executing Playback("Zap/4-1", "allison7/all-circuits-busy-now") in new stack -- Playing 'allison7/all-circuits-busy-now' (language 'en') -- Executing Playback("Zap/4-1", "allison7/pls-try-call-later") in new stack -- Playing 'allison7/pls-try-call-later' (language 'en')...

...quot;) in new stack -- Executing Dial("SIP/30-5dde", "ZAP/g0/19172073420||") in new stack == Everyone is busy/congested at this time -- Executing Macro("SIP/30-5dde", "outisbusy") in new stack -- Executing Playback("SIP/30-5dde", "allison7/all-circuits-busy-now") in ne w stack -- Playing 'allison7/all-circuits-busy-now' (language 'en') -- Executing Playback("SIP/30-5dde", "allison7/pls-try-call-later") in new s tack -- Playing 'allison7/pls-try-call-later' (language 'en...

...w stack -- Goto (macro-dialout-trunk,s-NOANSWER,1) -- Executing NoOp("SIP/231-af2b", "Dial failed due to NOANSWER") in new stack -- Executing Macro("SIP/231-af2b", "outisbusy") in new stack -- Executing Playback("SIP/231-af2b", "allison7/all-circuits-busy-now") in new stack -- Playing 'allison7/all-circuits-busy-now' (language 'en') -- Executing Playback("SIP/231-af2b", "allison7/pls-try-call-later") in new stack -- Playing 'allison7/pls-try-call-later' (language 'en&...

...k -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1) -- Executing NoOp("SIP/200-1661", "Dial failed due to CHANUNAVAIL") in new stack -- Executing Macro("SIP/200-1661", "outisbusy") in new stack -- Executing Playback("SIP/200-1661", "allison7/all-circuits-busy-now") in new stack -- Playing 'allison7/all-circuits-busy-now' (language 'en') -- Executing Playback("SIP/200-1661", "allison7/pls-try-call-later") in new stack -- Playing 'allison7/pls-try-call-later' (language 'en&...

...xten => s,2,SetGroup(${CALLERIDNUM}) exten => s,3,Setvar(FROMCONTEXT=rg-group) exten => s,4,SetCIDName(${PRE}${CALLERIDNAME}) exten => s,5,Macro(dial,${RINGTIMER},${DIAL_OPTIONS},${GRP}) ; ; Outgoing channel(s) are busy ... inform the client ; [macro-outisbusy] exten => s,1,Playback(allison7/all-circuits-busy-now) exten => s,2,Playback(allison7/pls-try-call-later) exten => s,3,Macro(hangupcall) ; What to do on hangup. [macro-hangupcall] exten => s,1,ResetCDR(w) exten => s,2,NoCDR() exten => s,3,Wait(5) exten => s,4,Hangup [macro-faxreceive] exten => s,1,SetVar(FA...

Hi All, I am a new bee to *. I just installed Asterisk@home on FC3. I hv a FXO card. I hv configured two extensions one x-lite and other iaxComm. I configured * using AMP. The following setup works - x-lite (x 200) to iaxComm (x 201) - PSTN to x-lite - PSTN to iaxComm Voice mail, weather etc work fine. When i try to make an external call i am getting message "All routes are busy". In

I just inherited a Cisco 2621 with a VWIC-1MFT-T1 card in it. Can I make this thing into MGCP gateway or even a SIP gateway for asterisk? Seems like it should bee useful for something! I'm perfectly happy to do my homework, but also don't feel thee need to reinvent the wheel! So, links with relevant info would be appreciated. If there is a config for a 2621 being used as a gateway

.../congested at this time Mar 6 21:40:01 DEBUG[21452]: Exiting with DIALSTATUS=CHANUNAVAIL. Mar 6 21:40:01 VERBOSE[21452]: -- Executing Macro("SIP/200-1cf6", "outisbusy") in new stack Mar 6 21:40:01 VERBOSE[21452]: -- Executing Playback("SIP/200-1cf6", "allison7/all-circuits-busy-now") in new stack Mar 6 21:40:01 DEBUG[21452]: Ooh, format changed from unknown to ulaw Mar 6 21:40:01 DEBUG[21452]: Scheduling timer at 160 sample intervals Mar 6 21:40:01 VERBOSE[21452]: -- Playing 'allison7/all-circuits-busy-now' (language 'en') Ma...

...n => s,2,SetGroup(${CALLERIDNUM}) ;exten => s,3,Setvar(FROMCONTEXT=rg-group) ;exten => s,4,SetCIDName(${PRE}${CALLERIDNAME}) ;exten => s,5,Macro(dial,${RINGTIMER},${DIAL_OPTIONS},${GRP}) ; ; Outgoing channel(s) are busy ... inform the client ; [macro-outisbusy] exten => s,1,Playback(allison7/all-circuits-busy-now) exten => s,2,Playback(allison7/pls-try-call-later) exten => s,3,Macro(hangupcall) ; What to do on hangup. [macro-hangupcall] exten => s,1,ResetCDR(w) exten => s,2,NoCDR() exten => s,3,Wait(5) exten => s,4,Hangup [macro-faxreceive] exten => s,1,SetVar(FA...

...;Not Found" back from myfirst.pbx.ip.address -- SIP/myfirstpbx.mydomain.mycountry-d001 is circuit-busy == Everyone is busy/congested at this time -- Executing Macro("SIP/870-7db5", "outisbusy") in new stack -- Executing Playback("SIP/870-7db5", "allison7/all-circuits-busy-now") in new stack -- Playing 'allison7/all-circuits-busy-now' (language 'it') ....... Is there anything to do, i.e. enabling dial from non internal extensions, or could it be a context problem ? any help will be greatly appreciated, Andrea Chi riceves...

...: -- Executing NoOp("SIP/200-361a", "Dial failed due to CHANUNAVAIL") in new stack Sep 16 07:47:46 VERBOSE[2012]: -- Executing Macro("SIP/200-361a", "outisbusy") in new stack Sep 16 07:47:46 VERBOSE[2012]: -- Executing Playback("SIP/200-361a", "allison7/all-circuits-busy-now") in new stack Sep 16 07:47:46 DEBUG[2012]: Ooh, format changed from unknown to ulaw Sep 16 07:47:46 DEBUG[2012]: Scheduling timer at 160 sample intervals Sep 16 07:47:46 VERBOSE[2012]: -- Playing 'allison7/all-circuits-busy-now' (language 'en') Sep 16 07:...

Dear List members- I am trying to configure ASTCC (Asterisk calling card application) but having a hard time to configure it properly. My project deadline is approaching and couldn't figure out how to make ASTCC functional. Here are some details what I have done so far. 1) I have installed ASTCC successfully. 2) I can access astcc-admin.cgi script without any problem. 3) I have created

...9") in new stack == Everyone is busy/congested at this time -- Executing NoOp("SIP/200-3b6b", "dial failed") in new stack -- Executing Macro("SIP/200-3b6b", "outisbusy") in new stack -- Executing Playback("SIP/200-3b6b", "allison7/all-circuits-busy-now") in new stack -- Playing 'allison7/all-circuits-busy-now' (language 'en') == Spawn extension (macro-outisbusy, s, 1) exited non-zero on 'SIP/200-3b6b' in macro 'outisbusy' == Spawn extension (from-internal, 9999, 2) exited non-z...

How many times do you ask for help here before getting a respone? Every single thing I do No matter what I get busy extensions. I am willing to pay someone to help here. Anybody got a clue? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050802/d0d1326c/attachment.htm

...quot;) in new stack -- Local/8521@from-internal-8268,1 answered SIP/XXX.XXX.XXX.XXX-43921110 -- Stopped music on hold on SIP/XXX.XXX.XXX.XXX-43921110 -- Playing 'vm-nobodyavail' (language 'en') -- Executing Playback(" Local/8521@from-internal-8268,2", "allison7/pls-try-call-later") in new stack -- Playing 'allison7/pls-try-call-later' (language 'en') -- Executing Hangup(" Local/8521@from-internal-8268,2", "") in new stack == Spawn extension (macro-exten-vm, novm, 5) exited non-zero on 'Local/8521@from...

Hello all, I have been searching thru all docs that I can find on wiki and such but can not get an answer. I am trying to collect a date from user input in the form of digits dialed from the phone to use in an agi script to do a database look up. I have tried to use "Get Data filename, timeout, maxdigits " in the agi script. In * console I get message saying playing filename but it

The SIP channels have a "call-limit" parameter (which is badly documented and I haven't tested yet) How can I have the same behaviour for IAX channels? I can't see anything related to it. Ah, I'm using Asterisk 1.2.13... maybe there is something in the 1.4 versions... but I can't change to 1.4 right now because of MFC/R2 BarZ

Is it possible using asterisk to allow someone to dial in and remotely change where their call is forwarded to? For example, I'm working from home so I want my calls to go to 555 1234, now I need to go out for a bit so I'd like to phone the office and using DTMF tell the asterisk PBX to now forward my calls to my cell phone 555 3456 Has anyone implimented anything like this? R.

...; 0 during vm message will hangup exten => o,2,goto(from-pstn,s,1) exten => a,1,Goto(app-directory,*411,1) exten => a,2,Hangup exten => novm,1,Macro(dial,120,${DIAL_OPTIONS},${ARG2}) exten => novm,2,Wait(1) exten => novm,3,Playback(vm-nobodyavail) exten => novm,4,Playback(allison7/pls-try-call-later) exten => novm,5,Hangup there's the extension definitions (the same for 201,202,203,204): [20x] username=20x type=friend seret= qualify=200 port=5060 pickupgroup= nat=never mailbox=20x@default host=dynamic dtmfmode=rfc2833 disallow= context=from-internal canreinvite=no c...

All, When calling an invalid number using, I expect to hear: "dooh-deeh-daah We're sorry you have reached a number which has been disconnected ..." And that is indeed what I hear when I dial out from [*] using analog FXO, or VoicePulse or NuPhone. When I dial that same number trough the T1 / PRI interface however, I continually hear ringing, and then the call gets hungup. Any ideas