similar to: busy() not setting PRI_CAUSE

I have two servers, each connected to the PTSN via PRI. When I call from site A (951-999-9999) to site B (555-1212) and the phone at site B is on the phone, I hear the normal ring tone for about 20 seconds, then the message "all circuits are busy now. please try your call again latter" followed by the congestion tone. Instead, I want this to busy ring and then hang up without any

Hi, all Is there someway ability to insert custom Header to "SIP 486" message, when HANGUP application is invoked? Our use case is to set that Header, when call-limit is reached, to analyze elsewhere, but we do not want to set some custom causecode in HANGUP application because this can confuse a calling equipment.

Using Dahdi/PRI, I end up with channel names like 'DAHDI/i8/9995551212-4d6B', but when I do a 'core show channels' it cuts off those names to only 'DAHDI/i8/9995551212-'. This is the same for AMI. Is there a way to get the full channel name within AMI? I'm using asterisk 11.7.0 Thanks, -Justin -------------- next part -------------- An HTML attachment was

<div>Is there any chance to create feature request for that useful functionality?</div><div>š</div><div>02.07.2015, 14:03, "Rusty Newton" <rnewton@digium.com>:</div><blockquote type="cite"><div><div><div>On Wed, Jul 1, 2015 at 4:46 AM, <span><<a href="mailto:royj@yandex.ru"

I have an installation that has analog phones connected via T1 channel banks. I'm getting complaints from users that they will enter a partial number (eg 91213), then turn away to get the next few digits, and the system will start dialing before they have a chance to put in the rest of the dialing string. Is there a way to increase this delay? The only use these 4 dialing patterns:

Greeting, I am trying to setup PLAR signalling in asterisk. I have modified the FXSLS TX bits in dahdi-base.c on line 2580, and I can make calls. .sig_type = DAHDI_SIG_FXSLS, .bits[DAHDI_TXSIG_ONHOOK] = DAHDI_BITS_ABCD, /*changed by for PLAR*/ .bits[DAHDI_TXSIG_OFFHOOK] = (0), /*changed by for PLAR*/ .bits[DAHDI_TXSIG_START] = DAHDI_BITS_ABCD, /*changed by for PLAR*/ When I got to change

<div>Thanks for the tip. Our goal is to know that call-limit is triggered. And later analyze this info, maybe do some action.</div><div>Yes, we can parse CDRs or execute AGI script but we do not want inmplement this logic on Asterisk because it can affectš<span>performance.</span></div><div>š</div><div>02.07.2015, 15:31, "jg"

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

Folks! 1) I have modified the original sip.conf and extension.conf file instead of writing mine. This looks like a mistake. 2)I have fired off Asterisk Extensions conf with 2 extensions i.e 2000 and 2001 and made one test call. I forgot to set a time out. The calls between these two extensions were partially successful. After writing my own files, it started working. I went through following

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,

Hi, when I still had ISDN, I was using Hangup(causecode) to send e.g. "Wrong number" to unwelcome callers. Meanwhile, I am only using SIP providers (no PSTN lines any more) and I would like to do similar, i.e. send specific SIP headers. Besides "wrong number", I would especially like to send 302 temp moved with a specified address to deflect certain calls. Is there any way to

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 =>

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

Unsubscribe Elvin G. Nodalo -----Original Message----- From: asterisk-users-request at lists.digium.com Sent: 7/10/2013 1:00 AM To: asterisk-users at lists.digium.com Subject: asterisk-users Digest, Vol 108, Issue 14 Send asterisk-users mailing list submissions to asterisk-users at lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit

Unsubscribe Elvin G. Nodalo -----Original Message----- From: asterisk-users-request at lists.digium.com Sent: 7/10/2013 1:00 AM To: asterisk-users at lists.digium.com Subject: asterisk-users Digest, Vol 108, Issue 14 Send asterisk-users mailing list submissions to asterisk-users at lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit

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

Hi, We have quite a large setup working just fine most of the time. We have 60 outgoing lines on PRI and we never use all of these lines. But sometimes we get the "all circuits busy now" message, seemingly random. Sometimes we get it before the call even goes through to PSTN. Sometimes after 5 or 6 rings etc. It seems that the carrier is signalling something and asterisk always

On Wed, Jul 1, 2015 at 4:46 AM, <royj at yandex.ru> wrote: > Hi, all > > Is there someway ability to insert custom Header to "SIP 486" message, > when HANGUP application is invoked? > > Our use case is to set that Header, when call-limit is reached, to analyze > elsewhere, but we do not want to set some custom causecode in HANGUP > application because this

> Is there any chance to create feature request for that useful functionality? > 02.07.2015, 14:03, "Rusty Newton" <rnewton at digium.com>: >> On Wed, Jul 1, 2015 at 4:46 AM, <royj at yandex.ru <mailto:royj at yandex.ru>> wrote: >> >> Hi, all >> >> Is there someway ability to insert custom Header to "SIP 486"

Hi I've got a script like this exten => s,1,SetVar(CALLFILENAME=/var/www/recordings/${TIMESTAMP:0:8:7}/${UNIQUEID}) exten => s,2,AGI(recordstart.py,${ARG1},${CALLERIDNUM},${CALLFILENAME},Ind) exten => s,3,DIAL(ZAP/g2/${ARG1},70) exten => s,4,AGI(logerror.py,${ARG1},${CALLERIDNUM},${CHANNEL},${DIALSTATUS},${DATETIME}, ${CAUSECODE}) exten => s,5,hangup exten =>