search for: noservice

Hi! I have an T1 line for using with IVR AGI. I receive the calls in my T1 but my dialplan has an error but my extensions doesnt have the error that show me asterisk. I dont know from where asterisk take extension 8 and how is playing ss-noservice because in my dialplan is not exist. Any help or any cluees? Verbosity was 5 and is now 7 -- Starting simple switch on 'Zap/1-1' == Unknown extension '8' in context 'from-ptsn' requested -- <Zap/1-1> Playing 'ss-noservice' (language 'en')...

Hello all, I've been playing around with early audio, and I'm able to get some things working We have PSTN calls coming in to asterisk in SIP from a Cisco AS5300. If I do the following: Exten => i,1,Playback(ss-noservice,noanswer) Exten => i,2,Congestion(15) Exten => i,3,Hangup() The PSTN caller does not get an answered call (doesn't get billed) but hears the ss-noservice message. But the early media fails when I try to do the following: Exten => 100,1,Dial(SIP/100,15) Exten => 100,2,Playback(sta...

..."Zap/31-1", "") in new stack [Feb 11 17:45:31] VERBOSE[5764] logger.c: -- Executing [s at from-pstn:3] Wait("Zap/31-1", "2") in new stack [Feb 11 17:45:33] VERBOSE[5764] logger.c: -- Executing [s at from-pstn:4] Playback("Zap/31-1", "ss-noservice") in new stack [Feb 11 17:45:33] VERBOSE[5764] logger.c: -- <Zap/31-1> Playing 'ss-noservice' (language 'en') [Feb 11 17:45:38] VERBOSE[5764] logger.c: -- Executing [s at from-pstn:5] SayAlpha("Zap/31-1", "91905") in new stack [Feb 11 17:45:38]...

...et=testing123 context=home nat=no ; Extensions.conf [general] static=yes writeprotect=no [globals] MAINPHONE=SIP/101 FWDUSERID1=533990 FWD1USERNAME=Norman Zhang FWDPREFIX=* HOMENUMBER=XXXXXXXXXX ; Macros [macro-fastbusy] exten => s,1,Answer exten => s,2,Wait,1 exten => s,3,Playback(ss-noservice) exten => s,4,Wait(30) exten => s,5,Hangup [macro-dialoutsip] exten => s,1,SetCallerID(${FWDUSERID1}) exten => s,2,SetCIDName(${FWD1USERNAME}) exten => s,3,Dial(SIP/${FWDPREFIX}${ARG1}@fwd,70) exten => s,4,Macro(fastbusy) exten => s,5,Hangup exten => s,104,Macro(fastbusy) e...

...=> 1625,n,Goto(digits/1) exten => i,1,NoOp(CallerID is ${CALLERID}) exten => i,n,NoOp(DID is ${DNID}) And the console stuff for a 06xx DID: -- Starting simple switch on 'Zap/1-1' == Unknown extension '0' in context 'incoming' requested -- Playing 'ss-noservice' (language 'en') -- Hungup 'Zap/1-1' For completeness' sake, here's the stuff from /var/log/asterisk/full; note that my 16xx extensions get all four DTMF digits before continuing: Jan 7 22:45:30 VERBOSE[5819] logger.c: -- Starting simple switch on 'Zap/1-1...

...fo #/etc/asterisk/extensions.conf [macro-dialGSM] exten => s,1,Dial(SIP/${ARG1}) exten => s,2,Goto(s-${DIALSTATUS},1) exten => s-CANCEL,1,Hangup exten => s-NOANSWER,1,Hangup exten => s-BUSY,1,Busy(30) exten => s-CONGESTION,1,Congestion(30) exten => s-CHANUNAVAIL,1,playback(ss-noservice) exten => s-CANCEL,1,Hangup [sip-external] exten => 2012,1,Macro(dialSIP,IMSI262428511722625) exten => 2013,1,Macro(dialSIP,IMSI262422146099205)

...e caller is busy it falls through and gets a Congestion... What's the proper syntax for this, reorder tone when there is a reorder and busy when there is a busy... SBC is a T1/PRI. [macro-sbc-outdial] exten => s,1,Dial(${ARG1}/${ARG2}) exten => s,2,Congestion exten => s,102,Playback(noservice) exten => s,103,Congestion

...-- Executing Macro("SIP/101-e528", "fastbusy") in new stack -- Executing Answer("SIP/101-e528", "") in new stack -- Executing Wait("SIP/101-e528", "1") in new stack -- Executing Playback("SIP/101-e528", "ss-noservice") in new stack -- Playing 'ss-noservice' (language 'en') -- Executing Wait("SIP/101-e528", "30") in new stack

...DID or CID Match") in new stack -- Executing [s at from-pstn:2] Answer("DAHDI/2-1", "") in new stack -- Executing [s at from-pstn:3] Wait("DAHDI/2-1", "2") in new stack -- Executing [s at from-pstn:4] Playback("DAHDI/2-1", "ss-noservice") in new stack -- <DAHDI/2-1> Playing 'ss-noservice.ulaw' (language 'en') Is my assumption correct? How can I solve that? TIA, -vcf

...,2,Dial(${OUT}/${EXTEN:1},,) exten => _${DIAL_OUT}011.,3,Congestion exten => _${DIAL_OUT}011.,103,Macro(outisbusy) [outbound-emerg] exten => 911,1,SetGroup(${CALLERIDNUM}) exten => 911,2,Dial(${OUT}/${EXTEN},,${DIAL_OPTIONS}) exten => 911,3,Congestion exten => 911,103,Playback(ss-noservice) exten => 911,104,Congestion exten => _${DIAL_OUT}911,1,SetGroup(${CALLERIDNUM}) exten => _${DIAL_OUT}911,2,Dial(${OUT}/${EXTEN:1},,${DIAL_OPTIONS}) exten => _${DIAL_OUT}911,3,Congestion exten => _${DIAL_OUT}911,103,Playback(ss-noservice) exten => _${DIAL_OUT}911,104,Congestion [...

...Asterisk, but I am not sure. Here's what the console log shows. -- Starting simple switch on 'Zap/1-1' File chan_zap.c, Line 3772 (ss_thread): getdtmf on channel 1: Operation now in progress == Unknown extension 's' in context 'default' requested -- Playing 'ss-noservice' -- Hungup 'Zap/1-1' Incidentally, inbound calls on the PRI are immediately disconnected when inbound caller-id info is present. The E&M trunk group is an attempt to workaround this problem. ================= zapata.conf [channels] group => 1 context => default switchtype...

...st call SetCIDNum on an INBOUND call to get the callerid functions working? Here is what I see in the log when a call comes in: -- Starting simple switch on 'Zap/24-1' == Unknown extension '*9161111111*9162222222' in context 'default' requested -- Playing 'ss-noservice' (language 'en') -- Hungup 'Zap/24-1' 9161111111 is the calling number (Caller ID) 9162222222 is the called number What would be the best way to convert this so I can use just ${EXTEN}? I can get it to work if I do something like: exten => _*XXXXXXXXXX*9162222222,1...

...stack -- Executing [s at from-trunk:3] Wait("SIP/358-106-000000c0", "2") in new stack > 0x2add5020a390 -- Probation passed - setting RTP source address to 217.xxx.xx.xxx:207xx -- Executing [s at from-trunk:4] Playback("SIP/358-106-000000c0", "ss-noservice") in new stack -- <SIP/358-106-000000c0> Playing 'ss-noservice.gsm' (language 'en') -- Executing [s at from-trunk:5] SayAlpha("SIP/358-106-000000c0", "") in new stack -- Executing [s at from-trunk:6] Hangup("SIP/358-106-000000c0",...

...34 the string gets truncated and the an invalid extension is dialed and the 'not in service' message plays. -- Hungup 'Zap/37-1' -- Starting simple switch on 'Zap/36-1' == Unknown extension '12' in context 'default' requested -- Playing 'ss-noservice' ...depending on how fast I dial the unknown ext will be '1', '12' or '123'. A related thing happens when calling in from fxo - it rings ~2 times before asterisk answers and the call goes into default. Does anyone know how to change this behavior and either make it go...

...ons.conf === [general] static=yes writeprotect=no [globals] SIP_XLITE = SIP/xlite SIP_PHONE = SIP/sipphone H323_SJPHONE = H323/sjphone@192.168.0.1 H323_PHONE = H323/h323phone@192.168.0.101 IAX_FIREFLY = IAX2/firefly ; ; Inbound ; [inbound] exten => s, 1, Answer exten => s, 2, Playback(ss-noservice) exten => s, 3, Hangup ; ; Internal Extensions ; [local] exten => 10,1,Dial(${SIP_XLITE}) exten => 11,1,Dial(${SIP_PHONE}) exten => 20,1,Dial,${H323_SJPHONE} exten => 21,1,Dial,${H323_PHONE} exten => 30,1,Dial(${IAX_FIREFLY}) exten => 0, 1, Answer exten => 0, 2, Playbac...

I need some help understanding SIP dialog. Some actor is trying to access my server, but I can't figure out what he's trying to do ,or how. I'm getting a lot of these warnings. [May 17 10:08:08] WARNING[1532]: chan_sip.c:4068 retrans_pkt: Retransmission timeout reached on transmission _zIr9tDtBxeTVTY5F7z8kD7R.. for seqno 101 With SIP DEBUG I tracked the Call-ID to this INVITE :

...34 the string gets truncated and the an invalid extension is dialed and the 'not in service' message plays. -- Hungup 'Zap/37-1' -- Starting simple switch on 'Zap/36-1' == Unknown extension '12' in context 'default' requested -- Playing 'ss-noservice' ...depending on how fast I dial the unknown ext will be '1', '12' or '123'. A related thing happens when calling in from fxo - it rings ~2 times before asterisk answers and the call goes into default. Does anyone know how to change this behavior and either make it go...

...ately says: Unknown extension 's' in context 'intrunk' requested I checked to make sure immediate is set to no. Full debug output is: -- Starting simple switch on 'Zap/1-1' == Unknown extension 's' in context 'intrunk' requested -- Playing 'ss-noservice' -- Hungup 'Zap/1-1' I don't _think_ I changed anything. zapata.conf has: [channels] context=intrunk signalling=featb rxwink=300 usecallerid=asreceived hidecallerid=no echocancel=yes echocancelwhenbridged=no rxgain=3.0 txgain=0.0 immediate=no amaflags=omit group=1 channe...

...training=400 context=default signalling=em_w musiconhold=default group=1 channel=>1-24 Context that handles incoming from extensions.conf [process-from-trunk] exten=>_2XXX,1,noop exten=>_2XXX,2,Dial(SIP/${EXTEN},20,tr) exten=>_2XXX,3,Voicemail(u${EXTEN}) exten=>_2XXX,4,Playback(ss-noservice) exten=>_2XXX,5,Hangup exten=>_2xxx,103,Voicemail(u${EXTEN}) exten=>_2XXX00,1,StripLSD(2)

...231) where dialSIP is a macro: [macro-dialSIP] exten => s,1,Dial(SIP/${ARG1}) exten => s,2,Goto(s-${DIALSTATUS},1) exten => s-CANCEL,1,Hangup exten => s-NOANSWER,1,Hangup exten => s-BUSY,1,Busy(3000) exten => s-CONGESTION,1,Congestion(3000) exten => s-CHANUNAVAIL,1,playback(ss-noservice) exten => s-CANCEL,2,Hangup