search for: caused

Anyone know how to let asterisk deal with the progress codes coming from the carrier? The problem I am having is when a customer calls an invalid number the carrier tells me the call is invalid via a progress code but doesn't route me to a recording (this number is invalid). Instead they hang up on me causing a fast busy or sometimes hold up the call with dead air for 15 to 30 seconds then a

Hi all, I have begun the release of my MFC/R2 protocol software. At http://www.opencall.org/installing-mfcr2.html there are instructions for installing what I have released so far. This is the MFC/R2 protocol software, and a test program. The software to interface Asterisk to the MFC/R2 code will be released shortly. It used to work, but it hasn't been touched for a while, and Asterisk

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, After I''ve saved, I want to display the Description from the Codes table that corresponds with the ID in the Cause table. Rails doesn''t seem to like that, because on my form, I have two sets of data in combo boxes that come from the Codes table. So, in the show template, I can''t do this: Cause of Death: <%= @cause.code.description %> Any ideas? The

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

Hello everyone, (This is my second question posted today on the R list). I am carrying out a competing risks analysis using the cuminc function...this takes the form: cuminc(ftime,fstatus,group) In my study, fstatus has 3 different causes of failure (1,2,3) there are also censored cases (0). "group" has two levels (0 and 1). I therefore have 6 different cumulative incidence curves:

...=AVC msg=audit(1398199187.646:29332): avc: denied { getattr } for pid=23387 comm="smtp" path="/var/spool/postfix/active/546AA6099F" dev=dm-0 ino=395679 scontext=unconfined_u:system_r:postfix_smtp_t:s0 tcontext=unconfined_u:object_r:postfix_spool_maildrop_t:s0 tclass=file Was caused by: Missing type enforcement (TE) allow rule. You can use audit2allow to generate a loadable module to allow this access. type=AVC msg=audit(1398199187.646:29333): avc: denied { read write } for pid=23387 comm="smtp" name="546AA6099F" dev=dm-0 ino=395679 scontext=unconfi...

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

Hello, when picking up an incoming call from one ip phone on another ip phone, the call terminates after about 5 to 10 seconds. When reading out the hangup cause variable in the h-extention of the dialplan, the hangup cause seems to be 111. In the dialplan output, you can see that SIP-peer sipacc3 picks up the incoming channel SipAgenT01-00001454, and the call is answered. After 7 seconds,

[2006/11/29 08:53:39, 0] smbd/server.c:(383) open_sockets_smbd: accept: Software caused connection abort [2006/11/29 10:21:20, 0] lib/util_sock.c:(202) Failed to set socket option SO_KEEPALIVE (Error Invalid argument) [2006/11/29 10:21:20, 0] lib/util_sock.c:(202) Failed to set socket option TCP_NODELAY (Error Invalid argument) [2006/11/29 10:21:20, 0] lib/util_sock.c:(1000) get...

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[Apr 26 10:47:01] NOTICE[32151]: chan_sip.c:16223 sip_poke_noanswer: Peer '3516533812' is now UNREACHABLE! Last qualify: 86 [Apr 26 10:47:11] NOTICE[32151]: chan_sip.c:12723 handle_response_peerpoke: Peer '3516533812' is now Reachable. (98ms / 2000ms) [Apr 26 12:08:49] WARNING[32273]: app_dial.c:1242 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -

Hi all, I tried to connect the bristuffed(0.3.0-PRE-1i) * to an Alcatel PBX via BRI (zaphfc) and Q.SIG. The Alcatel PBX is connected to the outside world and should forward our calls to the telco. This setup works correctly as far as I use euroisdn as the switchtype. The first problem was that it is only possible to run the * side in CPE-mode -- I wanted NET. Anyway, I configured * this way:

Hi Guys, I am calling out 416-999-1111 on Channel 1 of PRI and then calling 416-999-2222 on Channel 2 of PRI. When the two channels are going to be ZAP native bridged, both channels hangup and CLI show PRI cause (16). Asterisk Verbose *(Channel 1 already connected to party)*: -- Requested transfer capability: 0x00 - SPEECH -- Called g0/4169992222 -- Zap/2-1 is proceeding passing it

We have 2 PRI from AT&T And all is well but only few numbers having following issue. We are getting hangup cause 18 do you guys have any idea ? We have just migrate 1.2 to 1.8 and this issue raised [Jun 7 17:57:10] VERBOSE[23717] sig_pri.c: -- Span 2: Channel 0/3 got hangup request, cause 18 [Jun 7 17:57:10] DEBUG[24856] sig_pri.c: Not yet hungup... Calling hangup once with icause,

Hello. Someone knows what cause 127 mean. The phone that i'm calling rings once and than the connection interrupts: P[ 5] --> l3id:10040 P[ 5] --> cause:127 P[ 5] --> out_cause:127 P[ 5] --> state:ALERTING P[ 5] --> Channel: mISDN/5-1 hanguped new state:CLEANING P[ 5] $$$ CLEANUP CALLED pid:3 best regards -- Thomas Stein knowledgeTools? ....damit Sie sehen, was Sie

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Hi, developers, Recently, I read two papers [1], [2] about finding the root causes of compiler bugs. However, I do not find any information in these paper about how compiler developers find the root causes of compiler bugs in practice. So I am curious whether these techniques are useful in practice. For my experience, the outputs of compilers are always used to isolate the causes of compiler

Hello, Following my first mail about this issue [1], I think I know now what the problem is. When I have both lines being used and a third call comes in, the person calling doesn't get a busy tone, he gets something like line unavailable. I've been debugging mISDN and I think the reason is because asterisk is sending the release cause as 0. P[ 3] --> channel:0 mode:TE cause:0