search for: did_trunk_1

...[Oct 20 18:49:39] NOTICE[10629] chan_dahdi.c: Got event 17 (Polarity Reversal)... [Oct 20 18:49:42] NOTICE[10629] chan_dahdi.c: Got event 18 (Ring Begin)... [Oct 20 18:49:44] NOTICE[10629] chan_dahdi.c: Got event 2 (Ring/Answered)... [Oct 20 18:49:44] VERBOSE[10629] logger.c: -- Executing [s at DID_trunk_1:1] ExecIf("DAHDI/1-1", "1|SetCallerPres|unavailable") in new stack [Oct 20 18:49:44] VERBOSE[10629] logger.c: -- Executing [s at DID_trunk_1:2] ExecIf("DAHDI/1-1", "1|Set|CALLERID(all)=unknown <0000000>") in new stack The 3 events are always there...

...xxxx at iinetphone.iinet.net.au>>;tag=as0767eb78 -------------------------------------------------------- I can receive incoming call fine, Here is a copy of relevant parts of the configs and other info Trunk Info [trunk_1] disallow = allow = all callerid = 028012xxxx contact = context = DID_trunk_1 dialformat = ${EXTEN:1} fromdomain = iinetphone.iinet.net.au fromuser = 028012xxxx group = hasexten = no hasiax = no hassip = yes host = sip.nsw.iinet.net.au insecure = very port = 5060 provider = registeriax = no registersip = yes secret = xxxxxxxx trunkname = Custom - iinet trunkstyle = customvoi...

...w = all allow = ulaw,gsm subscribecontext = device-hints register => 6087294351:<sip password>@sip.broadvoice.com [trunk_1] type=peer user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=6087294351 secret=<sip password> username=6087294351 insecure=very context=DID_trunk_1 authname=6087294351 dtmfmode=inband dtmf=inband canreinvite=no [guest] type=friend host=dynamic canreinvite=no context=DID_trunk_1 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101103/4e700114/attachm...

...record_vmenu,n,Playback(vm-intro) exten = record_vmenu,n,Record(${var1}) exten = record_vmenu,n,Playback(vm-saved) exten = record_vmenu,n,Playback(vm-goodbye) exten = record_vmenu,n,Hangup exten = play_file,1,Answer exten = play_file,n,Playback(${var1}) exten = play_file,n,Hangup hasbeensetup = Y [DID_trunk_1] include = default [numberplan-custom-1] plancomment = DialPlan1 include = default include = parkedcalls [timebasedrules] *******part of extensions.conf that was added by asterisk-gui (svn)******* *******part of users.conf that was added by asterisk-gui (svn)******* [trunk_1] allow = all contex...

...0:38:07] NOTICE[7151]: chan_zap.c:6469 ss_thread: Got event 2 (Ring/Answered)... [Apr 15 10:38:07] NOTICE[7151]: chan_zap.c:6469 ss_thread: Got event 18 (Ring Begin)... -- Detected ring pattern: 299,290,279 -- Checking 0,0,0 -- Checking 0,0,0 -- Checking 0,0,0 -- Executing [s at DID_trunk_1:1] ExecIf("Zap/1-1", "0|SetCallerPres|unavailable") in new stack -- Executing [s at DID_trunk_1:2] ExecIf("Zap/1-1", "0|Set|CALLERID(all)=unknown <0000000>") in new stack So its not seeing the caller id. What might i have incorrect here? Thanks S...

...e could get the DTMF to pass when we were on the initial server we registered with but when we got pushed to another server the DTMF would fail till I did a sip reload or restarted Astersk. Now we get no DTMF ever. System set up. Asterisk 1.4.22 Asterisk GUI 2.0 users.conf [trunk_1] context = DID_trunk_1 host = galvatron.vtnoc.net username = user name secret = password trunkname = via:talk - galvatron ; GUI metadata hasiax = no registeriax = no hassip = yes registersip = yes trunkstyle = voip hasexten = no fromuser = user name authuser = user name insecure = port,invite dtmf = rfc2833 dtmfmode = r...

...] -- SIP/1000-0895df08 is ringing [Oct 22 23:42:07] -- Agent/10009 is ringing ** PLease see las line with [Oct 22 23:42:05] when the output shows that Called/xxxx and then says DAHDI/4-1 answered. [root at pbx ~]# cat /asterisk/chan_dahdi.conf [trunkgroups] [channels] language=ar context=DID_trunk_1 signalling=fxs_ks callwaiting=yes hidecallerid=no usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=no echocancelwhenbridged=no relaxdtmf=yes usedistinctiveringdetection=yes usecallingpres=yes busydetect=yes callprog...

...e could get the DTMF to pass when we were on the initial server we registered with but when we got pushed to another server the DTMF would fail till I did a sip reload or restarted Astersk. Now we get no DTMF ever. System set up. Asterisk 1.4.22 Asterisk GUI 2.0 users.conf [trunk_1] context = DID_trunk_1 host = galvatron.vtnoc.net username = user name secret = password trunkname = via:talk - galvatron ; GUI metadata hasiax = no registeriax = no hassip = yes registersip = yes trunkstyle = voip hasexten = no fromuser = user name authuser = user name insecure = port,invite dtmf = rfc2833 dtmfmode = r...

Hello all. I have a dahdi card with 2 FXO and 1 FXS. I'm able to make calls without any problem. However, when I have an incoming call, I see the following message on the asterisk console: -- Starting simple switch on 'DAHDI/1-1' -- Executing [s at DID_trunk_1:1] ExecIf("DAHDI/1-1", "1?SetCallerPres(unavailable)") in new stack -- Auto fallthrough, channel 'DAHDI/1-1' status is 'UNKNOWN' -- Hungup 'DAHDI/1-1' Notice the "status UNKNOWN". Can someone help me? Best regards, -vcf

...ting re-routed to the boss I call and it goes through. 3. We cannot recieve DTMF from via;talk, have tried auto, rfc2833, and inband without success with any of them, and yes we had via:talk change their end too. Here is the users.conf entry or the connection to via:talk. [trunk_1] context = DID_trunk_1 host = galvatron.vtnoc.net username = <phone number> secret = blablabla trunkname = via:talk ; GUI metadata hasiax = no registeriax = no hassip = yes registersip = yes trunkstyle = voip hasexten = no fromuser = <phone number> authuser = <phone number> insecure = port,invite dtmf =...

...yes busycount = 3 busypattern = 480,620 ringtimeout = 8000 answeronpolarityswitch = no hanguponpolarityswitch = no callprogress = no progzone = in usecallerid = yes cidstart = ring pulsedial = no cidsignalling = v23 flash = 750 rxflash = 1250 mailbox = callerid = asreceived dahdichan = 1 context = DID_trunk_1 group = 1 hasexten = no hasiax = no hassip = no registeriax = no registersip = no trunkstyle = analog disallow = all allow = all gui_volume = 2 ; GUI metadata signalling = fxs_ks gui_fxooffset = 0 ; GUI metadata rxgain = 0 txgain = 0.0 channel = 1 /dahdi/system.conf # Span 1: WCTDM/4 "Wildc...

Hi, Say I wanted to know what context a SIP registration is using to dial out in my dialplan, what would I do? For example, I have phones on a "local-calls-only" context (as defined in sip.conf), others in "unrestricted-calls". In my dialplan, I`d like to act on that knowledge. Mike -------------- next part -------------- An HTML attachment was

...ye) exten = record_vmenu,n,Hangup exten = play_file,1,Answer exten = play_file,n,Playback(${var1}) exten = play_file,n,Hangup hasbeensetup = N [numberplan-custom-1] plancomment = DialPlan1 include = default exten = _00X!,1,Macro(trunkdial,${trunk_4}/${EXTEN:0}) comment = _00X!,1,webcall,standard [DID_trunk_1] include = default [DID_trunk_2] include = default [DID_trunk_3] include = default [DID_trunk_4] include = default [DID_trunk_5] include = default [DID_trunk_6] include = default [DID_trunk_7] include = default [DID_trunk_8] include = default thx for any help jody :) Get news de...

...back(vm-intro) ;exten = record_vmenu,n,Record(${var1}) ;exten = record_vmenu,n,Playback(vm-saved) ;exten = record_vmenu,n,Playback(vm-goodbye) ;exten = record_vmenu,n,Hangup ;exten = play_file,1,Answer ;exten = play_file,n,Playback(${var1}) ;exten = play_file,n,Hangup ;hasbeensetup = Y [DID_trunk_1] include = default exten = s,1,Answer() exten = s,n,NoOp(${CALLERID(num)}) exten = s,n,Directory(default||f) [support] include = default exten = _X.,1,Goto(default|6009|1) exten = s,1,Goto(default|6009|1) [numberplan-custom-1] plancomment = DialPlan1 include = default exten = _...