Re: [asterisk-users] Calling rules
Correcting the line to: exten = _0.,1,Macro(trunkdial-failover-0.3,${trunk_1}/${,${EXTEN:1})},,trunk_1,) problem persists... any other suggestions? Best regards, What does your trunkdial-failover-0.3 look like? Here goes... [macro-trunkdial-failover-0.3] exten = s,1,GotoIf($[${LEN(${FMCIDNUM})} 6]?1-fmsetcid,1) exten = s,2,GotoIf($[${LEN(${GLOBAL_OUTBOUNDCIDNAME})} 1]?1-setgbobname,1) exten = s,3,Set(CALLERID(num)=${IF($[${LEN(${CID_${CALLERID(num)}})} 2]?${CID_${CALLERID(num)}}:)}) exten = s,n,GotoIf($[${LEN(${CALLERID(num)})} 6]?1-dial,1) exten = s,n,Set(CALLERID(all)=${IF($[${LEN(${CID_${ARG3}})} 6]?${CID_${ARG3}}:${GLOBAL_OUTBOUNDCID})}) exten = s,n,Goto(1-dial,1) exten = 1-setgbobname,1,Set(CALLERID(name)=${GLOBAL_OUTBOUNDCIDNAME}) exten = 1-setgbobname,n,Goto(s,3) exten = 1-fmsetcid,1,Set(CALLERID(num)=${FMCIDNUM}) exten = 1-fmsetcid,n,Set(CALLERID(name)=${FMCIDNAME}) exten = 1-fmsetcid,n,Goto(1-dial,1) exten = 1-dial,1,Dial(${ARG1}) exten = 1-dial,n,Gotoif(${LEN(${ARG2})} 0 ?1-${DIALSTATUS},1:1-out,1) exten = 1-CHANUNAVAIL,1,Dial(${ARG2}) exten = 1-CHANUNAVAIL,n,Hangup() exten = 1-CONGESTION,1,Dial(${ARG2}) exten = 1-CONGESTION,n,Hangup() exten = 1-out,1,Hangup() -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calling rules
On Jan 19, 2011, at 5:06 AM, Vitor Carlos Flausino wrote: In other words, which of the following is your situation: 1.) User dials 0X, asterisk sends 0X to the telco. 2.) User dials 0X, asterisk parses 0, strips it, and sends X to the telco. That might narrow it down. Option 2. 0 is to get an external line and XXX is passed to telco. -vcf It seems to me that you are passing the 0 to the telco when the user dials all digits at once. When they dial the 0 first, the call gets sent to one extension (probably extension 0 or _0) and just connects them to the outside line, sending nothing to the telco. When they dial 0X, asterisk matches another extension (probably _0. or another that begins with _0), one that connects them to the outside line and sends everything out to the telco, including the 0. Just a guess, but it sounds right to me. If so, you need to modify the dial command to strip the 0 before sending it. Tom -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Calling rules
Hello. I don't know if this is a problem, but I was expecting a different behavior. Users, have to dial 0 to get an external line, and afterwords the number they want to dial (exe 12345). The thing is: 1-If user dial 012345 there is an error and the call isn't made and the error is handle_request_invite: Call from 'XXX' to extension '012345' rejected because extension not found in context 'DLPN_DialPlanX'. 2-If user dials 0 waits for the signal, and then dials 12345 then it works fine. Should the result be the same? Shouldn't asterisk automatically dial 0, wait and then dial the external number? Best regards, Vitor Flausino -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calling rules
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vitor Carlos Flausino Sent: Tuesday, January 18, 2011 12:10 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Calling rules Hello. I don't know if this is a problem, but I was expecting a different behavior. Users, have to dial 0 to get an external line, and afterwords the number they want to dial (exe 12345). The thing is: 1-If user dial 012345 there is an error and the call isn't made and the error is handle_request_invite: Call from 'XXX' to extension '012345' rejected because extension not found in context 'DLPN_DialPlanX'. 2-If user dials 0 waits for the signal, and then dials 12345 then it works fine. Should the result be the same? Shouldn't asterisk automatically dial 0, wait and then dial the external number? Best regards, Vitor Flausino My best guess is that it is a dialplan inconsistency. The standard for outside line dialing is something like this: - exten = 0.,1,Dial(DAHDI/1,${EXTEN:1}) Where the dialplan chomps the first digit off of the dialed string. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calling rules
My dial plan was generated by asterisk GUI, and the line is: exten = _0,1,Macro(trunkdial-failover-0.3,${trunk_1}/${,${EXTEN:1})},,trunk_1,) where trunk_1 is DAHDI/1 Notice the difference between your 0. and my _0. Is mine correct? Best regards, -vcf - Original Message - From: Danny Nicholas da...@debsinc.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, January 18, 2011 7:21:15 PM Subject: Re: [asterisk-users] Calling rules -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vitor Carlos Flausino Sent: Tuesday, January 18, 2011 12:10 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Calling rules Hello. I don't know if this is a problem, but I was expecting a different behavior. Users, have to dial 0 to get an external line, and afterwords the number they want to dial (exe 12345). The thing is: 1-If user dial 012345 there is an error and the call isn't made and the error is handle_request_invite: Call from 'XXX' to extension '012345' rejected because extension not found in context 'DLPN_DialPlanX'. 2-If user dials 0 waits for the signal, and then dials 12345 then it works fine. Should the result be the same? Shouldn't asterisk automatically dial 0, wait and then dial the external number? Best regards, Vitor Flausino My best guess is that it is a dialplan inconsistency. The standard for outside line dialing is something like this: - exten = 0.,1,Dial(DAHDI/1,${EXTEN:1}) Where the dialplan chomps the first digit off of the dialed string. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calling rules
Un-top-posting and discarding cruft... On Tue, 18 Jan 2011, Vitor Carlos Flausino wrote: Users, have to dial 0 to get an external line, and afterwords the number they want to dial (exe 12345). The thing is: 1-If user dial 012345 there is an error and the call isn't made and the error is handle_request_invite: Call from 'XXX' to extension '012345' rejected because extension not found in context 'DLPN_DialPlanX'. 2-If user dials 0 waits for the signal, and then dials 12345 then it works fine. Should the result be the same? Shouldn't asterisk automatically dial 0, wait and then dial the external number? From: Danny Nicholas da...@debsinc.com My best guess is that it is a dialplan inconsistency. The standard for outside line dialing is something like this: - exten = 0.,1,Dial(DAHDI/1,${EXTEN:1}) Where the dialplan chomps the first digit off of the dialed string. On Tue, 18 Jan 2011, Vitor Carlos Flausino wrote: exten = _0,1,Macro(trunkdial-failover-0.3,${trunk_1}/${,${EXTEN:1})},,trunk_1,) Notice the difference between your 0. and my _0. Is mine correct? Both are 'wrong.' I'm guessing Danny just typed that in off the top of his head -- he forgot the leading underscore in the pattern. Please read up on pattern matching. In particular, what '_' and '.' mean. http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns http://www.voip-info.org/wiki/view/Asterisk+Extension+Matching Should get you started. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calling rules
- Original Message - From: Steve Edwards asterisk@sedwards.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, January 18, 2011 8:06:47 PM Subject: Re: [asterisk-users] Calling rules Un-top-posting and discarding cruft... On Tue, 18 Jan 2011, Vitor Carlos Flausino wrote: Users, have to dial 0 to get an external line, and afterwords the number they want to dial (exe 12345). The thing is: 1-If user dial 012345 there is an error and the call isn't made and the error is handle_request_invite: Call from 'XXX' to extension '012345' rejected because extension not found in context 'DLPN_DialPlanX'. 2-If user dials 0 waits for the signal, and then dials 12345 then it works fine. Should the result be the same? Shouldn't asterisk automatically dial 0, wait and then dial the external number? From: Danny Nicholas da...@debsinc.com My best guess is that it is a dialplan inconsistency. The standard for outside line dialing is something like this: - exten = 0.,1,Dial(DAHDI/1,${EXTEN:1}) Where the dialplan chomps the first digit off of the dialed string. On Tue, 18 Jan 2011, Vitor Carlos Flausino wrote: exten = _0,1,Macro(trunkdial-failover-0.3,${trunk_1}/${,${EXTEN:1})},,trunk_1,) Notice the difference between your 0. and my _0. Is mine correct? Both are 'wrong.' I'm guessing Danny just typed that in off the top of his head -- he forgot the leading underscore in the pattern. Please read up on pattern matching. In particular, what '_' and '.' mean. http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns http://www.voip-info.org/wiki/view/Asterisk+Extension+Matching Should get you started. Correcting the line to: exten = _0.,1,Macro(trunkdial-failover-0.3,${trunk_1}/${,${EXTEN:1})},,trunk_1,) problem persists... any other suggestions? Best regards, -vcf -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calling rules
On Tue, 18 Jan 2011, Vitor Carlos Flausino wrote: 1-If user dial 012345 there is an error and the call isn't made and the error is handle_request_invite: Call from 'XXX' to extension '012345' rejected because extension not found in context 'DLPN_DialPlanX'. 2-If user dials 0 waits for the signal, and then dials 12345 then it works fine. On Tue, 18 Jan 2011, Vitor Carlos Flausino wrote: Correcting the line to: exten = _0.,1,Macro(trunkdial-failover-0.3,${trunk_1}/${,${EXTEN:1})},,trunk_1,) problem persists... How about some console output for a 'good' call and a 'failed' call. Also, a 'show dialplan|dialplan show' for the executed context may yield some clues. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calling rules
- Original Message - From: Steve Edwards asterisk@sedwards.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, January 18, 2011 8:54:11 PM Subject: Re: [asterisk-users] Calling rules On Tue, 18 Jan 2011, Vitor Carlos Flausino wrote: 1-If user dial 012345 there is an error and the call isn't made and the error is handle_request_invite: Call from 'XXX' to extension '012345' rejected because extension not found in context 'DLPN_DialPlanX'. 2-If user dials 0 waits for the signal, and then dials 12345 then it works fine. On Tue, 18 Jan 2011, Vitor Carlos Flausino wrote: Correcting the line to: exten = _0.,1,Macro(trunkdial-failover-0.3,${trunk_1}/${,${EXTEN:1})},,trunk_1,) problem persists... How about some console output for a 'good' call and a 'failed' call. Also, a 'show dialplan|dialplan show' for the executed context may yield some clues. -- Here goes... asterisk*CLI dialplan show CallingRule_Outbound_Ch1 [ Context 'CallingRule_Outbound_Ch1' created by 'pbx_config' ] '_0.' = 1. Macro(trunkdial-failover-0.3,${trunk_1}/${,${EXTEN:1})},,trunk_1,) [pbx_config] -= 1 extension (1 priority) in 1 context. =- Log when dialing 0924343424 == Using SIP RTP CoS mark 5 -- Executing [0924343424@DLPN_DialPlan1:1] Macro(SIP/6005-0002, trunkdial-failover-0.3,DAHDI/1/,,trunk_1,) in new stack -- Executing [s@macro-trunkdial-failover-0.3:1] GotoIf(SIP/6005-0002, 0?1-fmsetcid,1) in new stack -- Executing [s@macro-trunkdial-failover-0.3:2] GotoIf(SIP/6005-0002, 1?1-setgbobname,1) in new stack -- Goto (macro-trunkdial-failover-0.3,1-setgbobname,1) -- Executing [1-setgbobname@macro-trunkdial-failover-0.3:1] Set(SIP/6005-0002, CALLERID(name)=Glintt) in new stack -- Executing [1-setgbobname@macro-trunkdial-failover-0.3:2] Goto(SIP/6005-0002, s,3) in new stack -- Goto (macro-trunkdial-failover-0.3,s,3) -- Executing [s@macro-trunkdial-failover-0.3:3] Set(SIP/6005-0002, CALLERID(num)=222355598) in new stack -- Executing [s@macro-trunkdial-failover-0.3:4] GotoIf(SIP/6005-0002, 1?1-dial,1) in new stack -- Goto (macro-trunkdial-failover-0.3,1-dial,1) -- Executing [1-dial@macro-trunkdial-failover-0.3:1] Dial(SIP/6005-0002, DAHDI/1/) in new stack -- Called 1/ -- DAHDI/1-1 answered SIP/6005-0002 -- Hungup 'DAHDI/1-1' == Spawn extension (macro-trunkdial-failover-0.3, 1-dial, 1) exited non-zero on 'SIP/6005-0002' in macro 'trunkdial-failover-0.3' == Spawn extension (DLPN_DialPlan1, 0924343424, 1) exited non-zero on 'SIP/6005-0002' A normal internal call to 2000 is: == Using SIP RTP CoS mark 5 -- Executing [2000@DLPN_DialPlan1:1] Directory(SIP/6005-000a, default,default,f) in new stack == Parsing '/etc/asterisk/voicemail.conf': == Found == Parsing '/etc/asterisk/users.conf': == Found -- SIP/6005-000a Playing 'dir-welcome.ulaw' (language 'en') -- SIP/6005-000a Playing 'dir-pls-enter.ulaw' (language 'en') == Spawn extension (DLPN_DialPlan1, 2000, 1) exited non-zero on 'SIP/6005-000a' Hope helps... Best regards and thanks in advance... -vcf -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calling rules
On 01/18/2011 3:20 PM, Vitor Carlos Flausino wrote: == Spawn extension (DLPN_DialPlan1, 0924343424, 1) exited non-zero on 'SIP/6005-0002' Vitor, Can you please clarify whether the 0 should be received by Asterisk and processed internally, or whether it should be passed to the DAHDI channel by asterisk? In other words, which of the following is your situation: 1.) User dials 0X, asterisk sends 0X to the telco. 2.) User dials 0X, asterisk parses 0, strips it, and sends X to the telco. That might narrow it down. Tom -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calling rules
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vitor Carlos Flausino Sent: Tuesday, January 18, 2011 1:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Calling rules snip Correcting the line to: exten = _0.,1,Macro(trunkdial-failover-0.3,${trunk_1}/${,${EXTEN:1})},,trunk_1,) problem persists... any other suggestions? Best regards, What does your trunkdial-failover-0.3 look like? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calling rules
On Tue, 18 Jan 2011, Vitor Carlos Flausino wrote: Log when dialing 0924343424 [snip] A normal internal call to 2000 is: [snip] These two calls do not demonstrate your issue: 1-If user dial 012345 there is an error and the call isn't made and the error is handle_request_invite: Call from 'XXX' to extension '012345' rejected because extension not found in context 'DLPN_DialPlanX'. 2-If user dials 0 waits for the signal, and then dials 12345 then it works fine. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users