search for: outisbusy

...TION HANGUPCAUSE: 34 - failing through to other trunks") in new stack -- Executing [continue at macro-dialout-trunk:2] Set("SIP/306-000000b8", "CALLERID(number)=306") in new stack -- Executing [0149XXXXXX at from-internal:7] Macro("SIP/306-000000b8", "outisbusy,") in new stack -- Executing [s at macro-outisbusy:1] Progress("SIP/306-000000b8", "") in new stack -- Executing [s at macro-outisbusy:2] GotoIf("SIP/306-000000b8", "0?emergency,1") in new stack -- Executing [s at macro-outisbusy:3] GotoIf(&q...

...iling through to > other trunks") in new stack > -- Executing [continue at macro-dialout-trunk:2] Set("SIP/306-000000b8", > "CALLERID(number)=306") in new stack > -- Executing [0149XXXXXX at from-internal:7] Macro("SIP/306-000000b8", > "outisbusy,") in new stack > -- Executing [s at macro-outisbusy:1] Progress("SIP/306-000000b8", "") in > new stack > -- Executing [s at macro-outisbusy:2] GotoIf("SIP/306-000000b8", > "0?emergency,1") in new stack > -- Executing [s at mac...

...by SetGroup. Copy contents to GRP exten => 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-fa...

What is involved in provisioning Asterisk to use a multiline analog service from our local telco? I will only have one twisted pair entering in on a OpenVox card but am not sure how Asterisk interprets and deals with two incoming calls and/or two outgoing calls? Thanks! jlc

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

...er trunks") in new stack > > -- Executing [continue at macro-dialout-trunk:2] > Set("SIP/306-000000b8", > > "CALLERID(number)=306") in new stack > > -- Executing [0149XXXXXX at from-internal:7] Macro("SIP/306-000000b8", > > "outisbusy,") in new stack > > -- Executing [s at macro-outisbusy:1] Progress("SIP/306-000000b8", "") > in > > new stack > > -- Executing [s at macro-outisbusy:2] GotoIf("SIP/306-000000b8", > > "0?emergency,1") in new stack > &...

...TION HANGUPCAUSE: 34 - failing through to other trunks") in new stack -- Executing [continue at macro-dialout-trunk:2] Set("SIP/101-00000103", "CALLERID(number)=101") in new stack -- Executing [0149xxxxxx at from-internal:6] Macro("SIP/101-00000103", "outisbusy,") in new stack -- Executing [s at macro-outisbusy:1] Progress("SIP/101-00000103", "") in new stack -- Executing [s at macro-outisbusy:2] GotoIf("SIP/101-00000103", "0?emergency,1") in new stack -- Executing [s at macro-outisbusy:3] GotoIf(&q...

...o on 'SIP/30-8d25' -- Executing SetGroup("SIP/30-5dde", "30") 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"...

...xten => s,5,Goto(6) exten => s,6,GotoIf($[foo${OUTCID_${ARG1}} = foo]?6) exten => s,7,SetCallerID(${OUTCID_${ARG1}}) exten => s,8,SetVar(length=${LEN(${DIAL_OUT_${ARG1}})}) exten => s,9,Dial(${OUT_${ARG1}}/w${ARG2:${length}}) exten => s,10,Congestion exten => s,108,Macro(outisbusy) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050308/26df53c3/attachment.htm

...ANUNAVAIL,3) -- Executing [s-CHANUNAVAIL at macro-dialout-trunk:3] NoOp("SIP/2022-083c53f0", "TRUNK Dial failed due to CHANUNAVAIL - failing through to other trunks") in new stack -- Executing [90012127773456 at from-internal:5] Macro("SIP/2022-083c53f0", "outisbusy|") in new stack -- Executing [s at macro-outisbusy:1] Playback("SIP/2022-083c53f0", "all-circuits-busy-now|noanswer") in new stack -- <SIP/2022-083c53f0> Playing 'all-circuits-busy-now' (language 'en') -- Executing [s at macro-outisbusy:2]...

Hi All, I am trying to connect my asterisk box with freepbx to PSTN. I have purchased x100p FXO card and installed in my asterisk server. My freepbx detected the x100p FXO card and i can see the card specific details in command line. I have configured the following things. 1. OUTBOUND caller id and Dialing rules in Freepbx. 2. INBOUND route When i call to the PSTN number before

...TION HANGUPCAUSE: 19 - failing through to other trunks") in new stack -- Executing [continue at macro-dialout-trunk:2] Set("SIP/300-00000192", "CALLERID(number)=300") in new stack -- Executing [0176XXXXXX at from-internal:7] Macro("SIP/300-00000192", "outisbusy,") in new stack -- Executing [s at macro-outisbusy:1] Progress("SIP/300-00000192", "") in new stack -- Executing [s at macro-outisbusy:2] GotoIf("SIP/300-00000192", "0?emergency,1") in new stack -- Executing [s at macro-outisbusy:3] GotoIf(&q...

...29 -- Channel 0/23, span 2 got hangup request <----here I get a busy signal -- Hungup 'IAX2/ VoIPServicePrividerOUT-1' [Outbound context] exten => _9011.,1,Macro(dialout-trunk,4,${EXTEN:1},) exten => _9011.,2,Macro(dialout-trunk,2,${EXTEN:1},) exten => _9011.,3,Macro(outisbusy) ; No available circuits exten => _918.,1,Macro(dialout-trunk,2,${EXTEN:1},); 800 numbers to the PRI exten => _918.,2,Macro(outisbusy) ; No available circuits exten => _9Z.,1,Macro(dialout-trunk,4,${EXTEN:1},) exten => _9Z.,2,Macro(dialout-trunk,2,${EXTEN:1},) exten => _9Z.,3,Macro(o...

...- failing through to other trunks") in new stack > -- Executing [continue at macro-dialout-trunk:2] Set("SIP/101-00000103", > "CALLERID(number)=101") in new stack > -- Executing [0149xxxxxx at from-internal:6] Macro("SIP/101-00000103", > "outisbusy,") in new stack > -- Executing [s at macro-outisbusy:1] Progress("SIP/101-00000103", "") > in new stack > -- Executing [s at macro-outisbusy:2] GotoIf("SIP/101-00000103", > "0?emergency,1") in new stack > -- Executing [s at mac...

...1-9_outside include => outrt-002-outgoingFWD [outbound-trunks] include => outbound-trunks-custom exten => _${DIAL_OUT_1}.,1,Macro(dialout,1,${EXTEN}) [outrt-001-9_outside] include => outrt-001-9_outside-custom exten => _9.,1,Macro(dialout-trunk,1,${EXTEN:1}) exten => _9.,2,Macro(outisbusy) ; No available circuits [outrt-002-outgoingFWD] include => outrt-002-outgoingFWD-custom exten => 393,1,Macro(dialout-trunk,2,${EXTEN},) exten => 393,2,Macro(outisbusy) ; No available circuits [internal] exten => 100,1,Dial(SIP/abhijit) exten => abhijit,1,Echo() Can anyone please...

.../0/0) -- Executing Goto("SIP/2211-c1c6", "s-NOANSWER|1") in new stack -- Goto (macro-dialout-trunk,s-NOANSWER,1) -- Executing NoOp("SIP/2211-c1c6", "Dial failed due to NOANSWER") in new stack -- Executing Macro("SIP/2211-c1c6", "outisbusy") in new stack -- Executing Playback("SIP/2211-c1c6", "all-circuits-busy-now") in new stack -- Playing 'all-circuits-busy-now' (language 'en') -- Executing Playback("SIP/2211-c1c6", "pls-try-call-later") in new stack -- Pl...

...- Executing [s-CONGESTION at macro-dialout-trunk:3] > NoOp("SIP/75002-b7705298", "TRUNK Dial failed due to CONGESTION - failing > through to other trunks") in new stack > -- Executing [015921256331 at from-internal:6] > Macro("SIP/75002-b7705298", "outisbusy|") in new stack > -- Executing [s at macro-outisbusy:1] Playback("SIP/75002-b7705298", > "all-circuits-busy-now|noanswer") in new stack > -- <SIP/75002-b7705298> Playing 'all-circuits-busy-now' (language 'en') > Really destroying SI...

You would need to give more information really. Your sip.conf file listing the entries for the phones especially which codecs are permitted. A copy of the 'asterisk -rvvv' console output when you make the call. On 27/03/15 17:05, Salaheddine Elharit wrote: > please no body has som with aastra can help me in this issue > > 2015-03-26 11:02 GMT+00:00 Salaheddine Elharit >

...-dialout-trunk:3] > NoOp("SIP/192.168.0.151-088e7938", "TRUNK Dial failed due to CHANUNAVAIL > (hangupcause: 58) - failing through to other trunks") in new stack > -- Executing [15921256331 at from-internal:6] > Macro("SIP/192.168.0.151-088e7938", "outisbusy|") in new stack > -- Executing [s at macro-outisbusy:1] > Playback("SIP/192.168.0.151-088e7938", "all-circuits-busy-now|noanswer") in > new stack > -- Executing [s at macro-outisbusy:2] > Playback("SIP/192.168.0.151-088e7938", "pls-try-...

Hello everyone! I've had this problem for a while and cant figure it out. When an outside caller calls an extension on my asterisk system, they do not hear any sort of ringing. Inside extensions calling other extensions do hear ringing. We have 3 other asterisk systems that are configured the same way, but do not have this problem. We think it has something to do with asterisk 1.6. The other