search for: ringtimeout

I have a PRI line and I am having problems setting the ringtimeout on the dial application to more than 29. If I set ringtimeout to 29 on the dial application call and I do not answer the ringing phone then I correctly get DIALSTATUS set to NOANSWER. If I set ringtimeout to any value over 29 on the dial application call and I do not answer the ringing phone then...

...h FXO modules on it. Below is the dial plan. When someone can 9123456, CLI will show dialing to 123456 and answered. After answered, the call hangup. I would like to know what will cause the case to happen. Anyone can give me some advice to solve it? exten => _9X.,n,Dial(Zap/g0/${EXTEN:1}|${RINGTIMEOUT}) exten => _9X.,n,Hangup zapata.conf signalling=fxs_ks callerid=asreceived group=0 context=from-pstn ;context=cs channel => 1-8

...yed diferents TDM cards and modules, and my zapata.conf is like, language=en context=from-zaptel switchtype=national usecallerid=yes callerid=asreceived transfer=yes callreturn=yes rxgain=-3.0 txgain=-3.0 immediate=no busydetect=yes busycount=8 answeronpolarityswitch=yes hanguponpolarityswitch=yes ringtimeout=8000 faxdetect=both group=0 signalling=fxs_ks channel => 7 think the problem is not by echo cause I use fxotune, and the problem persist. I made lots of TDM02B instalations and never get this kind of problem. Any clue will be welcomed. Thanks in advance. Regards VoipCrazy -------------- nex...

...1.4.1 built by root @ astpbx on a i686 running Linux on 2007-03-07 11:25:52 UTC extensions.conf ----------------------- [local] ; Incoming calls on DID number, call lands on the home phone exten => 845,1,Answer() exten => 845,n,Dial(Local/2055@local) exten => 2055,1,Dial(SIP/homephone,${RINGTIMEOUT}) exten => 2055,n,VoiceMail(2055@voicemail-context,su) exten => 2055,n,Hangup() exten => 2055,102,VoiceMail(2055@voicemail-context,sb) [level-zero] include => local CLI output with "core set verbose 5" --------------------------------------------------- -- Executing [845@le...

...callwaiting=yes usecallingpres=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=50 immediate=no rxgain=3.0 txgain=4.0 immediate=no busydetect=yes busycount=8 answeronpolarityswitch=yes hanguponpolarityswitch=yes ringtimeout=8000 faxdetect=both signalling=fxs_ks useincomingcalleridonzaptransfer=yes channel => 1-2 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070216/51fe7bbf/attachment.htm

...ecific setting to avoid such incidents ? iam using busycount as 3, signalling=fxs_ks ;toneduration=100 callwaiting=yes threewaycalling=yes callreturn=yes echocancel=128,param1=32,param2=0,param3=14 echocancelwhenbridged=yes echotraining=yes echotraining=800 busycount=3 hanguponpolarityswitch=yes ringtimeout=8000 group=1 context=incoming immediate=yes jitterbuffers=4 jbenable = yes echocancel=yes channel=>1-4 ;overlapdial=yes ;pulsedial=yes dtmfmode=rfc2833 ;relaxdtmf=yes ;rxgain=10.0 ;txgain=8.0 any ideas please! Thanks, Tharanga Abyeseela

...up on 'DAHDI/5-1' -- Hungup 'DAHDI/5-1' *relevant part from chan_dahdi.conf* [outside-line](!) ; A template to hold common options for all phones connected outside signalling = fxs_ks ; tried ls and gs without any success callerid = asreceived ; dialtone_detect=always ringtimeout=8000 faxdetect=incoming context = from-pstn Bart Remmerie -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140526/5e5ee3d6/attachment.html>

...rid=no usecallingpres=yes threewaycalling=yes transfer=yes canpark=yes callreturn=no echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no flash=100 transfertobusy=yes busydetect=yes busycount=4 busypattern=400,400 callprogress=yes progzone=au ringtimeout=6000 faxdetect=incoming musiconhold=default channel=>1 ;X100P channel=>2 ;X100P channel=>3 ;X100P channel=>4 ;X100P & zaptel.conf fxsks=1-4 loadzone = au defaultzone=au & extensions.conf ...relevant part... [pstn-incoming] exten => s,1,Dial(SIP/24|15) exten => s,n,Dial(...

...s that overlap dialing isn't a factor and that asterisk has complete control. As for options in chan_dahdi.conf, I simply can't find any that relate to this problem. I have looked at the page here: http://www.voip-info.org/wiki/view/chan_dahdi.conf and the closest thing I can find is 'ringtimeout' which is obviously not what I want. I would expect to see something like 'dialtimeout' or 'interdigittimeout'. -Justin -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of jg Sent: Monday...

...s that overlap dialing isn't a factor and that asterisk has complete control. As for options in chan_dahdi.conf, I simply can't find any that relate to this problem. I have looked at the page here: http://www.voip-info.org/wiki/view/chan_dahdi.conf and the closest thing I can find is 'ringtimeout' which is obviously not what I want. I would expect to see something like 'dialtimeout' or 'interdigittimeout'. -Justin -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of jg Sent: Monday...

...v file which contains my telco's disconnect supervision. I am using , asterisk-1.4.33.1 dahdi-linux-complete-2.3.0.1+ 2.3.0 OS => Debian-lenny 5 users.conf ------------- [trunk_1] trunkname = pstn ; GUI metadata busydetect = 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 hass...

...extensions.conf channel.dadhi.conf [channels] signalling=fxs_ks ;toneduration=100 callwaiting=yes threewaycalling=yes callreturn=yes echocancel=128,param1=32,param2=0,param3=14 echocancelwhenbridged=yes echotraining=yes echotraining=800 busydetect=yes busycount=2 hanguponpolarityswitch=yes ringtimeout=8000 group=1 context=sip immediate=yes jitterbuffers=4 jbenable = yes echocancel=yes channel=>1-4 ;overlapdial=yes ;pulsedial=yes dtmfmode=rfc2833 ;relaxdtmf=yes ;rxgain=10.0 ;txgain=8.0 Many thanks Tharanga

...nclude /etc/asterisk/dahdi-channels.conf signalling=fxs_ks usecallerid=yes callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=no echocancelwhenbridged=no group=1 callgroup=1 pickupgroup=1 immediate=yes ringtimeout=8000 signalling=fxs_ks callerid=asreceived group=0 context=from-pstn channel => 1 [general] static=yes writeprotect=no clearglobalvars=no [globals] CONSOLE=Console/dsp IAXINFO=guest TRUNK=DAHDI/g1 TRUNKMSD=0 [default] exten => 1205,1,Wait(2) exten => 1205,2,Record(/tmp/ast...

...rid=no usecallingpres=yes threewaycalling=yes transfer=yes canpark=yes callreturn=no echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no flash=100 transfertobusy=yes busydetect=yes busycount=4 busypattern=400,400 callprogress=yes progzone=au ringtimeout=6000 faxdetect=incoming musiconhold=default channel=>1 ;X100P channel=>2 ;X100P channel=>3 ;X100P channel=>4 ;X100P & zaptel.conf fxsks=1-4 loadzone = au defaultzone=au & extensions.conf ...relevant part... [pstn-incoming] exten => s,1,Dial(SIP/24|15) exten => s,n,Dial(...

My SIP phones can dial out through Sipura SPA3k to POTS for local and 911 calls _but_ incoming POTS calls are being swallowup somehow. Am I on the right track with the code snippit below? sip.conf: --------- In sip.conf the following code is _supposed_ to ring the SIP phones when a POTS line call comes in through Sipuara to Asterisk. [spa3k-pstn-in] ; Pots-line-in from Sipura ; If

...oneduration=250 usecallerid=yes callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=yes rxgain=10 txgain=10 group=1 callgroup=1 pickupgroup=1 immediate=yes ringtimeout=8000 signalling=fxs_ls callerid=asreceived channel => 1 # cat /etc/asterisk/extensions.conf [general] static=yes writeprotect=no clearglobalvars=no [globals] CONSOLE=Console/dsp TRUNK=DAHDI/g1 TRUNKMSD=0 [trunktollfree] exten => _91800NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${...

...000P?zA\000\000\000\000?@ \017\000\000\000\000`\2074?7\000\000\000??u\000\000\000\000\000P?D\017\000\000\000\000?\234B\000\000\000\000\000?`zA\000\000\000\000??G\000\000\000\000" runningdata = "SIP/${fromsip_sippeers_hostid}${billsip_rt_routeprefix}${tosip}@${outbound_${billsip_rt_gw}}|${RINGTIMEOUT}|L(${availCallLimit}:120000)${DialOpt}\000zA\000\000\000\000\000\200a@\000\000\000\000P?zA\000\000\000\000?FM\000\000\000\000\000P?zA\000\000\000\000\020]zA", '\0' <repeats 12 times>, "?`"... oldargs = {0x0 <repeats 81 times>} argc = 1 x = 1098539312 res = 0...

Folks, I've seen a few reports that people have had problems with hang up detection on UK cable phone lines. I have a TDM400P with two FXO ports, one connected to my BT line and the other connected to my Telewest/Virgin Media cable line. If I ring the BT line and then clear down, Asterisk detects this and acts accordingly. If I ring the Telewest line, the clear down is not detected, hence