search for: pri_caus

Okay, I think I need a sanity check here - If I call a person that's on the phone, I should get a busy signal. Now more specifically, a call comes into the pbx via PRI. The destination dialplan runs busy(20). Now, the PRI causecode should get set to 17 (user busy) so that the originating end can play a busy tone, correct? -Justin -------------- next part -------------- An HTML attachment

...ID service, and about 30 of them are >> handled by dialplan). So far I didn't manage to reject call that are >> handled >> in dialplan, except by Congestion command which answers the call first. > > There are two methods: > > * use the Hangup app after setting the PRI_CAUSE variable. This is the > general way of sending a specified disconnect code. See > http://www.voip-info.org/wiki-Asterisk+variable+PRI_CAUSE Tried this with the Stable 1.0.6 connected to Panasonic D500 and it indeed hung up a call from panasonics EXT or CO line but it doesn't g...

Just turned up a new PRI with DID's in the US. I'm receiving 5 digits of the DID numbers as I requested. Assuming I have 100 DID numbers but only define 50 of those in extensions.conf, is there an easy way to send the incoming calls for the 20 undefined numbers to a common resource (ivr, operator, or canned message) without having to define each one?

...o the Box A at pri2 (net). Now I want Box B to dial out to the PSTN tunneled thru Box A and have it get all ISDN indications in case of call failure, eg. unallocated destination number etc. But currently Box B always gets only "normal clearing". Interestingly on Box A the NoOp Output of PRI_CAUSE and CAUSECODE after the dial attempt to an unallocated number is also empty. How to access/query the reason for a failed ISDN dial attempt from the dial plan, I expected at last PRI_CAUSE to be filled. Is it possible at all with asterisk - what modification to my attached configs would you sugge...

...no user responding' instead of 'user busy'. Asterisk knows the user is busy and jumps to the prio+101 extension. The CDR also logs the call as 'busy'. With Zap ISDN channels the following works nicely: exten => 12345,1,Dial(SIP/${EXTEN},45,r) exten => 12345,102,SetVar(PRI_CAUSE=17) exten => 12345,103,Hangup Can we do something similar with chan_oh323? Many thanks, Jan

Hello, i have succesfullu setup asterisk with Sangoma E1 card, evrything works well but i don't know how to pass indications from telco switch to the user - when users call bad number telco switch shuld talk "unallocated number" but its only send PRI_CAUSE 1. How to pass voice indications thru asterisk to clients? My /etc/zaptel.conf: span=1,0,0,CCS,HDB3,CRC4 dchan=16 bchan=1-10 alaw=1-10 loadzone=pl defaultzone=pl My /etc/asterisk/zapata.conf: [channels] language=en context=from-pstn switchtype=euroisdn signalling=pri_cpe pridialplan=local usecal...

...ialplan. I tried different Apps there (Hangup, Busy, Congestion) They deliver different cause codes to the H.323/ISDN side (normal call clearing or call rejected) but none of them returns 'user busy' as expected. In Zaptel with Q.931 PRI (euroisdn) you can do exten => 123,102,SetVar(PRI_CAUSE=17) exten => 123,103,Hangup to explicitely set the RELEASE cause code. Is something similiar also possible with H.323? Thank you and regards, Jan

I'm trying to build a test suite so that I can run "calls" through and verify the call results. I've made a cross over cable and linked my 2 ISDN30 ports together. So now I can dial out on span 1 , and to receive the call on span 2. in the context for span 2, I have the following: <snip> ; #1 "answer" a call and play music 000XXX : ring for a random period,

...0 switch) I would like to find a way of informing Asterisk that the call is progressing or something like that, not ringing until it gets the correct message from SER . I am using Asterisk CVS-04/06/04-10:46:21 on Red Hat 9 and Sip Express Router version 12 on Red Hat 9. I tried to use PRI_causes and "r" extension in extension.conf but still the problem is there. Any idea on how I can solve this problem? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040611/b56e05a6...

...root @ pbx-test.bb.ic3s.de on a i686 running Linux on 2006-06-16 10:17:00 UTC with A Quad ZAP Pri Card ---- dialplan ;exten => 297,1,SetTransferCapability(VIDEO) exten => 297,1,Noop() exten => 297,2,Dial(${TRUNK}/0175234567) exten => 297,3,Hangup exten => 297,104,SetVar(PRI_CAUSE=17) ; Indicate Busy exten => 297,105,Hangup ---- Log pbx-test*CLI> == Primary D-Channel on span 2 down Oct 31 09:05:03 WARNING[6771]: chan_zap.c:2506 pri_find_dchan: No D-channels available! Using Primary channel 47 as D-channel anyway! 1 < Protocol Discriminator: Q.931 (8) len=45...

hi, well, most of the things work right now due to the help of peter svensson, but after heavy use of our ericsson BP250 today several problems appeared. i split into several mails as they are seperate problems. * i can't signal Busy to the calling party. asterisk receives busy from the ericsson PBX but does not forward this to the external caller. i tried with exten =>

...the switch that the call is going and the phone is ringing while it is not the case. I cant find a way of informing Asterisk that the call is progressing or something like that . I am using Asterisk CVS-03/22/04-15:45:54 on Red Hat 9 and Sip Express Router version 12 on Red Hat 9 I tried to use PRI_causes but still the problem is there. Any idea on how I can solve this problem? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040607/3526b80d/attachment.htm

...but did not find. We are runnig asterisk 1.2.4 extensions.conf [default] exten => _0.,1,Set(CALLERID(Number)=971288612) exten => _0.,2,Set(CALLERID(Name)='Gil Estarellas') exten => _0.,3,Dial(Zap/g1/${EXTEN:1}||rf) exten => _0.,4,NoOp(${HANGUPCAUSE}) exten => _0.,5,NoOp(${PRI_CAUSE}) ______________________________________________ LLama Gratis a cualquier PC del Mundo. Llamadas a fijos y m?viles desde 1 c?ntimo por minuto. http://es.voice.yahoo.com

...|...) and added stationid... - added config option 'immediate' to start pbx if no dnid has been received yet. - support 'type of number' (numbering-plan). - U-Law setting is now done in capi.conf instead of Makefile define. - on hangup, use hangupcause from other channel or from var PRI_CAUSE. - capi.conf structure changes: one own section for each interface, no global 'interfaces' any more. Section name will be interface name. - dial string changed: parameters like 'b' not as part of number any more. - send alert on alerting only (busy() and congestion() work now). -...

I know this topic came up many months back and some discussions were being had on how to do this within the Zaptel drivers. However, I'm looking for even a crude hack that someone has put together to get this done. We have PRI's and LD T1's that are load balanced on two boxes. The hunt order goes from box to box as far as the spans are concerned. There are times that I would like to

Hi, I am trying to send "404 Not found" reply, without any luck with the following: exten => 555,1,Playback(you-dialed-wrong-number,noanswer) exten => 555,n,Playback(check-number-dial-again,noanswer) exten => 555,n,Congestion() However the above results in "500 Service Unavailable" being send out. What would be the correct application/function to generate "404

Hi, I've an PRI coming to my asterisk ,calls are coming fine and my agents are able to answer no prob. but I've an agreement with my telco with some incoming no if the no of calls on these no are more then 3 then send to another no. they use busy signal to divert call on another number so I'm sending the call to Congestion() if no of calls in this group are more then 3. But my

Hi all, I have the following in my extensions.conf: exten => s,4,GotoIf($["${CALLERID(number)}" = "8585979857" | "8585970327"]?15:5) The numbers listed above are known spammer numbers. However, when I call from any other CALLERID, it still directs me to s,15 which is the Hangup() application. Here are logs from the asterisk CLI: -- Executing

Hi all, I have a Wildcard TE110P connected to a E1 line an I want to reserve channels in the following way: channels 1-15 and 17-21 for incoming calls channels 22-28 for outgoing calls channels 29-31 for emergency calls My zaptel.conf looks like this: ; incoming group = 1 signalling=pri_cpe context=from-zaptel channel => 1-15 channel => 17-21 ; outgoing group = 2 signalling=pri_cpe