search for: pri_causes

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

> On Sun, 27 Mar 2005, Nenad Radosavljevic wrote: >> Only way I have managed to get Zap channel to reject a call on TE110P >> without answering it, is to dial number that is not handled in dialplan >> (I >> have a ISDN PRI with 100 number DID service, and about 30 of them are >> handled by dialplan). So far I didn't manage to reject call that are >>

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?

Hi all, I have to asterisk-1.0.9BRIstuffed-0.2.0n boxes each equipped with a TE410P. Box A is connected with pri1 to the PSTN. Box B is connected with pri1 (cpe) to 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

Hi, I have asterisk connected to PSTN via H.323 gateway via chan_oh323. Incoming calls to SIP extensions work, but SIP message "486 busy here" from a busy extension isn't correctly forwarded to H.323. As a result, a caller from the H.323 side calling a busy SIP extension gets some rings and then an irritating timeout with H.323 message 'no user responding' instead of

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

Hi all, I just installed chan_h323 to interface to a H.323/ISDN gateway. It works really well after two days learning and testing except one thing somebody of you may have an answer to: If I call in from PSTN to a busy asterisk SIP extension I can see a SIP status 486 BUSY, but don't get it passed to the H.323/ISDN side. Asterisk jumps correctly to EXTEN+101 in the dialplan. I tried

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/a...

Hi! For demonstration purposes I try to bridge an incoming video call from a 3G mobile handset to another 3G mobile handset using asterisk as "switch". On the incoming call leg I see all expected bearer capabilities (Digital, 64k Transparent, G.7xx 384k video) but on the outgoing call leg the bearer capability G.7xx 384k video get lost and therefore the call is dropped from the mobile

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

Hi All, the * is working rigth for incoming calls and internal calls, but when trying to call out we got hanged up. The hangup reason is AST_CAUSE_INVALID_IE_CONTENTS I've been searching in the mailing list archive as I thing that some thing similar happens to someone else but did not find. We are runnig asterisk 1.2.4 extensions.conf [default] exten =>

Hi all, it took a while, but on sourceforge.net I added the new release 0.6 of chan_capi-cm driver. Note: dial string and capi.conf has changed. The main changes are: - added 'relaxdtmf'. - more BSD compatibility - correct PROGRESS handling - added verbose text for capi info/reason error messages. - use correct facility-selector for echo-cancel - added application capicommand() for CAPI

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