similar to: ISDN Cause codes for unanswered calls

Well, it works for me .. YMMV. Yesterday I had a problem where I had a meridian talking to * via a PRI card, and from * to the pstn via an isdn30 link. The problem was that if the number was bad, or engaged then the meridian line simply dropped, not giving the operator any indication of what occurred. With much help from this list, I managed to construct a dialplan which solved our issues.

Hi, I was trying to use 2 of asterisk servers and interconnected, one of them as a peer to other sever (configured in sip.conf), so all the calls to server 1 will just be passed to server 2 (has PRI Card, TE 412P, only one PRI connected), i was sending calls to server 1 and that would send to server 2 and then dial out using Dahdi, but the problem that i got was the hangup cause codes, i was not

hello, i'm testing sending ISDN cause codes to customer pbx (test scenario for unallocated number) topology: PSTN-E1-AsteriskA-AsteriskB-SOMEPBX INVITE from SOMEPBX to PSTN AsteriskA sends to AsteriskB Status-Line: SIP/2.0 503 Service Unavailable X-Asterisk-HangupCause: Unallocated (unassigned) number X-Asterisk-HangupCauseCode: 1 how can i resend HangupCauseCode from AsteriskB to

Hello All, Just got one general question on ISDN error code 42. As per Cisco docs and Wiki (http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause) it says "Switch equipment Congestion" with explanation "The destination cannot be reached because the network switching equipment is temporarily overloaded"... My question is are the calls treated as

Hi everyone, Currectly I'm having some troubles to get correct status of my calls throug ISDN lines, when outbound calls don't get its destination I always receive NO ANSWER as ${DIALSTATUS} despite the fact I know the target number doesn't exists or is busy at that time. Maybe there is something I must change in my zaptel.conf or zapata.conf, current configs follows: ####

Apologies if this messages appears more than once, somehow sending it via gmane's nntp-server doesn't work as expected. Been trying to send this message for 2 days now, without success so far... --- Hello, I'm exploring the capablities of Asterisk and must say I'm really impressed! However, I don't need most of the options, but can't figure out the things that appear

Keywords T1 Q.931 isdn disconnect cause codes itu standard libpri Dont know if anyone wondered what q.931 cause codes are but i wishwe could get these back into the dial plan as a var Standard Q931 Codes Decimal Value Hexadecimal Value Definition 1 01 Unallocated (unassigned) number. This number is not in the routing table or it has no path across the ISDN cloud (network). 1. Check routing

Hi, I am experimenting with getting hold of the sip cause and sip response from outgoing call. How could i make a userevent printing the sip cause and/or sip response. I have tried using hangupcause, sip_cause and such , but i am not getting any data. I would at least like to use the q.850 reason codes in the dialplan which i now am unable to do. Any help appreciated. [Beskrivning: Fogwise -

I am trying to retrieve the cause code of a outgoing call over a PRI where the number called is out of service. When an out service number is called I get a recording that the number dialed is not a working number. I see cause code 1 in in the CLI as soon as the call is dialed the Telco recording goes on for 30 sec. then hangs up. Any idea on how retrieve info that the called number is is

Hi All Just been looking at stats for one of my sites, and I'm conserned about the number of error cause codes being returned from the telco for example 12000 calls processed 131 are cause code 31* normal. unspecified.* 139 are cause code 28 * invalid number format (address incomplete).* 112 are cause code 1 *Unallocated (unassigned) number. *this adds up to about 3% of calls not

I have a legacy ISDN PBX (Network Alchemy Argent Office) connected to Span 2 of a Digium Wildcard TE205P. Recently Calls from this PBX have been failing with ISDN Cause Code 38 (Network Out of Order!). The problem seems to be getting worse and is now effecting more calls than not (although this could just be because I'm aware of it). Once the ISDN PBX has decided the Network's Out of

Hi, I am using Asterisk: 1.6.1.20 LibPRI: 1.4.11.4 DAHDI: 2.3.0.1 Echo Canceller: MG2 Wanpipe-Driver: 3.5.15 Sangoma-Firmware: 43 (A104d) I handle some calls with my own PHP-AGI-Script. After a dial-command I use "GET FULL VARIABLE ${answeredtime}" or "GET FULL VARIABLE ${dialstatus}" and get valid information. Sometimes "dialstatus" has the value

Hello, On an Asterisk 1.4.33.1 in a simple scenario: [test] exten => _X.,1,Dial(SIP/12345 at peer01,,,) exten => i,1,Hangup(${HANGUPCAUSE}) exten => t,1,Hangup(${HANGUPCAUSE}) exten => h,1,Hangup(${HANGUPCAUSE}) I have noticed that no matter what value we set in the Hangup(<cause code>) commands, if the call is not answered by peer01 for any reason, the actual cause code

Steve Casto escribi?: >/ I am trying to retrieve the cause code of a outgoing call over a PRI />/ where the number called is out of service. When an out service number is />/ called I get a recording that the number dialed is not a working />/ number. I see cause code 1 in in the CLI as soon as the call is dialed />/ the Telco recording goes on for 30 sec. then hangs up. Any

Hi, I've been debuging the call disconnection problem in our architecture: PSTN---E1---OldPBX---E1---Asterisk This is our problem: -SIP user agent "A" calls a pstn phone "B". -"B" hangs up the call. -SIP user agent "A" starts listenning busytones... But the call still on. (and being payed). - Call only ends when it is correctly hanged up in the

Dear friends, ? I am not able to capture the CDR records for unanswered calls. Only one record per call is coming in the CDR table. In the cdr.conf, I have enabled it by setting : unanswered = yes ?

Hi all, first off all - sorry for the cross posting - i did already posted this message to asterisk-dev - after that i realized that it isn't really a -dev related question - more a -users questions. So ignore it on -dev .... we have the following setup PSTN 3 PRI Lines <---> Asterisk (1.4.22) <---> Siemens HiCom <---> Bosch Integral The Asterisk Machine

Hi list, I'm wondering if it's possible to transfer a call that is still ringing??? I Have some Grandstream GXP-2000 and with the TRNF button it's impossible. So, I've configured some keys to transfer the calls like this: [featuremap] blindxfer => #2 ; Blind transfer (default is #) disconnect => *0 ; Disconnect (default is *) ;automon => *1

I am one of the few people left in the US with a ISDN connection to the internet. (Internet cable has been promised "Real Soon Now" for a couple years). Worse, my home network runs on a decrepit Cisco 2500 series router, and is double-natted via a firewall, which is why I currently use iax to call anywhere. I have been making ever-increasing use of my asterisk PBX, and I'm ready

Just throwing this out here, hopefully someone can tell me why. *CLI> show version Asterisk CVS-HEAD-11/17/04-10:16:38 built by root@wanderer on a i686 running Linux Zap/g1 is pri_cpe to Bell Canada 5551234 is a normal POTS line I have busied out (handset offhook) exten => 1234,1,Dial(Zap/g1/5551234,,g) exten => 1234,n,NoOp(HANGUPCAUSE is ${HANGUPCAUSE} and DIALSTATUS is