search for: reghours

...in, the system seems to detect OK but does ot manage to make the fax to pdf to email leap. Here is what I saw in the CLI when I tested. Any help would be appreciated. Thanks! Wiley -- Starting simple switch on 'Zap/2-1' -- Executing GotoIf("Zap/2-1", "1?from-pstn-reghours|s|1:") in new stack -- Goto (from-pstn-reghours,s,1) -- Executing GotoIf("Zap/2-1", "0?from-pstn-reghours-nofax|s|1:2") in new stack -- Goto (from-pstn-reghours,s,2) -- Executing Answer("Zap/2-1", "") in new stack -- Executing Wait(&q...

...noying to say the least. Here is some log output from the server: Aug 20 11:16:46 VERBOSE[1617]: -- Starting simple switch on 'Zap/4-1' Aug 20 11:16:47 DEBUG[1617]: Expression is '1' Aug 20 11:16:47 VERBOSE[1617]: -- Executing GotoIf("Zap/4-1", "1?from-pstn-reghours|s|1:") in new stack Aug 20 11:16:47 VERBOSE[1617]: -- Goto (from-pstn-reghours,s,1) Aug 20 11:16:47 DEBUG[1617]: Expression is '0' Aug 20 11:16:47 VERBOSE[1617]: -- Executing GotoIf("Zap/4-1", "0?from-pstn-reghours-nofax|s|1:2") in new stack Aug 20 11:16:47...

...s-BUSY, 2) exited non-zero on 'Zap/1-1' in macro 'exten-vm' == Spawn extension (ext-local, 220, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' -- Starting simple switch on 'Zap/1-1' -- Executing GotoIf("Zap/1-1", "1?from-pstn-reghours|s|1:") in new stack -- Goto (from-pstn-reghours,s,1) -- Executing GotoIf("Zap/1-1", "0?from-pstn-reghours-nofax|s|1:2") in new stack -- Goto (from-pstn-reghours,s,2) -- Executing Answer("Zap/1-1", "") in new stack -- Executing Wait(&q...

I am experimenting with an asterisk setup in my office. The last bit I have to test is working with analog lines. I have TE411p digium card, with an ISDN line plugged into the first, a channel bank plugged into the second port, and the last two ports empty. I have the following setup in my zaptel.conf: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 span=2,0,0,d4,ami fxsks=25 And in zapata.conf, I

When I make a call out on the Faktortel number I am then able to transfer to call to my asterisk meetme room of 801 by hitting 'transfer' then '801' then 'send' on my grandstream phone. This connects my faktortel trunk (and who ever is on the other end) to my conference room I can then make another call using my local pstn service and set up a 3 way (or whatever number

...# the calling party gets no ring indication, just silence until either I answer the phone, or the call bounces over to voicemail. below is the console output when a call is recieved. what am i missing here? thanks Bernie -- Executing GotoIf("SIP/2017540019-fe29", "1?from-pstn-reghours|s|1:") in new stack -- Goto (from-pstn-reghours,s,1) -- Executing GotoIf("SIP/2017540019-fe29", "0?from-pstn-reghours-nofax|s|1:2") in new stack -- Goto (from-pstn-reghours,s,2) -- Executing Answer("SIP/2017540019-fe29", "") in new stac...

...Linux/utils/compress/!INDEX.short.html) ; loligo sounds (http://www.loligo.com/asterisk/sounds/) ; mpg123 (http://voip-info.org/wiki-Asterisk+config+musiconhold.conf) ; include part generated from AMP #include extensions_additional.conf [from-pstn] exten => s,1,GotoIf($[${IN_OVERRIDE} = forcereghours]?from-pstn-reghours,s,1:) exten => s,2,GotoIf($[${IN_OVERRIDE} = forceafthours]?from-pstn-afthours,s,1:) exten => s,3,GotoIfTime(${REGTIME}|${REGDAYS}|*|*?from-pstn-reghours,s,1:) exten => s,4,Goto(from-pstn-afthours,s,1) [from-pstn-reghours] exten => s,1,GotoIf($[${FAX_RX} = disabled]...

...#39;' to 's' on channel 0/1, span 1 Mar 26 17:27:37 DEBUG[3570] chan_zap.c: Enabled echo cancellation on channel 1 Mar 26 17:27:37 DEBUG[7284] pbx.c: Expression result is '1' Mar 26 17:27:37 VERBOSE[7284] logger.c: -- Executing GotoIf("Zap/1-1", "1?from-pstn-reghours|s|1:") in new stack Mar 26 17:27:37 VERBOSE[7284] logger.c: -- Goto (from-pstn-reghours,s,1) Mar 26 17:27:37 DEBUG[7284] pbx.c: Expression result is '0' Mar 26 17:27:37 VERBOSE[7284] logger.c: -- Executing GotoIf("Zap/1-1", "0?from-pstn-reghours-nofax|s|1:2&quot...

Hi, maybe a dumb question, but it seems that some calls are directed to our central dial in number despite the extensions the callers say they dialled. E.g. they dial 1234-567, asterisk recognizes 12345, it says this is an unknown extension, where it is right, and redirects the call to the central dial in extension 1234-0. This only seems to happen when the numbers are dialled manually. When

...xecuting Cut("Zap/1-1", "intype=intype|-|1") in new stack Sep 28 13:31:59 DEBUG[14709]: Expression is '0' Sep 28 13:31:59 VERBOSE[14709]: -- Executing GotoIf("Zap/1-1", "0?7:9") in new stack Sep 28 13:31:59 VERBOSE[14709]: -- Goto (from-pstn-reghours,s,9) Sep 28 13:31:59 DEBUG[14709]: Expression is '0' Sep 28 13:31:59 VERBOSE[14709]: -- Executing GotoIf("Zap/1-1", "0?10:12") in new stack Sep 28 13:31:59 VERBOSE[14709]: -- Goto (from-pstn-reghours,s,12) Sep 28 13:31:59 DEBUG[14709]: Expression is '1'...

...thing else is working great. Installed the spandsp patches and software... using the default AMP extensions.conf, I start sending a fax, I hear it pick up and transfer to voicemail after 20s. Fax is set for system... Here is the detail from the extensions.conf [global] FAX_RX = system [from-pstn-reghours] exten => s,1,GotoIf($[${FAX_RX} = disabled]?from-pstn-reghours-nofax,s,1:2) ; if fax detection is disabled, then jump to from-pstn-nofax - else continue exten => s,2,Answer exten => s,3,Wait(1) exten => s,4,SetVar(intype=${INCOMING}) exten => s,5,Cut(intype=intype,-,1) exten =&g...

...el 1/1, span 1 chan_zap.c: Enabled echo cancellation on channel 1 logger.c: -- Executing Goto("Zap/1-1", "s|1") in new stack logger.c: -- Goto (from-pstn,s,1) pbx.c: Expression result is '1' logger.c: -- Executing GotoIf("Zap/1-1", "1?from-pstn-reghours|s|1:") in new stack logger.c: -- Goto (from-pstn-reghours,s,1) pbx.c: Expression result is '1' logger.c: -- Executing GotoIf("Zap/1-1", "1?from-pstn-reghours-nofax|s|1:2") in new stack logger.c: -- Goto (from-pstn-reghours-nofax,s,1) logger.c: -- Exe...

I just signed up for the BroadVoice service a few hours ago, but for the life of me I can't get any incoming voice. The incoming connection is fine as it rings my extension from outside, but I can't hear anyone talking. Outgoing voice is working fine though. I've been looking through the archives, but I haven't found a solution to the problem yet. I even tried another router

...es in, and if it's on the X100 card it will get the caller id information collected by my cid daemon and return to asterisk in the CALLERID variable. get_cid.pl should be but in asterisk agi-bin directory, and the following modifications should be implemented in extensions.conf: [from-pstn-reghours] exten => s,1,GotoIf($[${FAX_RX} = disabled]?from-pstn-reghours-nofax,s,1:2) exten => s,2,Answer exten => s,3,agi,getcid.pl ;exten => s,3,Wait(1) [from-pstn-afthours] exten => s,1,GotoIf($[${FAX_RX} = disabled]?from-pstn-afthours-nofax,s,1:2) exten => s,2,Answer exten =...

...e some way to debug what file/line is being executed by asterisk? My custom context is this: [from-pstn-nofax] include => from-pstn-custominclude customizations include => ext-did exten => _.,1,Goto(s,1) exten => s,1,SetVar(INCOMING=GRP-1) exten => s,2,GotoIf($[${IN_OVERRIDE} = forcereghours]?from-pstn-reghours-nofax,s,1:2) exten => s,3,GotoIf($[${IN_OVERRIDE} = forceafthours]?from-pstn-afthours-nofax,s,1:3) exten => s,4,GotoIfTime(${REGTIME}|${REGDAYS}|*|*?from-pstn-reghours-nofax,s,1:2) exten => s,5,Goto(from-pstn-afthours-nofax,s,1) ext-did context is this (made by amp):...

...#39; in macro 'hangupcall' == Spawn extension (default, s, 2) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' ; include part generated from AMP #include extensions_additional.conf [from-pstn] include => ext-did exten => s,1,GotoIf($[${IN_OVERRIDE} = forcereghours]?from-pstn-reghours,s,1:) exten => s,2,GotoIf($[${IN_OVERRIDE} = forceafthours]?from-pstn-afthours,s,1:) exten => s,3,GotoIfTime(${REGTIME}|${REGDAYS}|*|*?from-pstn-reghours,s,1:) exten => s,4,Goto(from-pstn-afthours,s,1) [from-pstn-reghours] exten => s,1,GotoIf($[${FAX_RX} = disabled]...

...include => ext-did include => from-pstn-timecheck ; this has to be included otherwise it overrides ext-did [from-pstn-timecheck] exten => .,1,Goto(s,1) ; catch-all matching for calls that have DID info (if a DID route hasn't matched them) exten => s,1,GotoIf($[${IN_OVERRIDE} = forcereghours]?from-pstn-reghours,s,1 exten => s,2,GotoIf($[${IN_OVERRIDE} = forceafthours]?from-pstn-afthours,s,1 exten => s,3,GotoIfTime(${REGTIME}|${REGDAYS}|*|*?from-pstn-reghours,s,1 exten => s,4,Goto(from-pstn-afthours,s,1) [from-pstn-reghours] exten => s,1,GotoIf($[${FAX_RX} = disabled]?from-...

Hi, I'm struggling to get busydetect to work. I'm using asterisk 1.2.1 and a digium TDM04B (4 port FXO) card. I've set busydetect=yes, busycount=6 and busypattern=300,200 in zapata.conf and i've modified zondata.c with a busy setting of 620+480, 300/200 which is the busysignal received from Korea Telecom. Asterisk isn't detecting the busy signal and doesn't hangup.

...t; <1252@postaetrustsi> changed : Name: Robert Rozman| Number: 1252@postaetrustsi") in new stack -- Executing Goto("IAX2/guest@252/2", "from-pstn|s|1") in new stack -- Goto (from-pstn,s,1) -- Executing GotoIf("IAX2/guest@252/2", "1?from-pstn-reghours|s|1:") in new stack So in Variable ${CALLERID} is right (with dots in iax address) but not the case with ${CALLERIDNUM} that has dots stripped off. So now I cannot callback cause proper address is lost. On outgoing part I have: [outbound-iax] exten => _2.,1,NoOp(Outbound IAX call from lo...

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