similar to: usegmtime=yes for cdr_custom

Displaying 20 results from an estimated 9000 matches similar to: "usegmtime=yes for cdr_custom"

2009 Mar 06
5
work around the 64 pickupgroups limit
Hi! What are the typical ways to work around the 64 groups limit? thanks klaus
2009 Mar 11
2
how to configure for incoming message-summary SUBSCRIBE
Hi! AFAIS the incoming SUBSCRIBE is handled in the same context as INVITE - but how should I handle the SUBSCRIBE in the context? thanks klaus SUBSCRIBE sip:u+431234567 at foobar.at:5160 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.82:39982;branch=z9hG4bK-d8754z-3116e1207913aa4e-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:u+431234567 at 11.111.11.11:39982> To:
2014 Feb 06
0
cdr_custom.conf in V12
Hello Asterisk, I just got V12 running and all seems well but just now I looked at my CDR logs and they were messed up so I copied over the sample cdr_custom.conf and uncommented the first master line and the simple line and the logs look like: Simple.csv: "1391652220","","" Master.csv
2006 May 11
3
sangoma A102 installation question
Hi! I've went through the READMEs and could not answer this question: During installation, the Setup program asks: Would you like update/upgrade wanpipe drivers? (y/n) For a pure Asterisk TDM installation - is it required to patch the kernel or is this only when using the sangoma cards as WAN router? regards klaus
2009 Nov 10
2
looking for an Asterisk supervision (status viewer) tool
Hi! I am looking for a tool (application or webinterface) which shows me the current status of an Asterisk server, e.g.: - Status of the SIP peers (registered/offline) - current incoming and outgoing calls - start-time, numbers, some history - history (calls stopped in the last 15 minutes, who hang up?) - should be possible to link those calls to the relevant SIP peers -
2005 Sep 22
0
cdr_custom?
I have a need to use cdr_custom and would like to know if anyone has gotten it to work with a mysql cdr backend, and any examples if possible -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050922/bca01301/attachment.htm
2009 Jun 08
3
T.38 pass-through 488 handling problem
Hi! I have the following problem with Asterisk 1.4.23: ATA w/ T.38 Asterisk ATA w/o T.38 --------INVITE--------> --------INVITE--------> <-------200OK---------- <-------200OK---------- --------ACK-----------> --------ACK-----------> --------INVITE
2009 Jan 14
8
evaluate SIP response codes in dialplan
Hi! Is it somehow possible to evaluate the SIP response code inside the dialplan? I have an Asterisk server which forwards requests to various PSTN gateways with SIP. If the Dial() attempt is not successful I want to differ at least these 3 options: - called destination is busy (486): e.g. activate auto-redial - called destination does not exist, unassigned number (404) - gateway is broken,
2009 Jan 08
3
AEL and };
Hi! All the AEL examples have a semicolon after the closing curly bracket, e.g: context test { 1 => Hangup(); }; but without ; it works fine too, e.g: context test { 1 => Hangup(); } So - what is the reason for the ; after the closing curly bracket? thanks klaus
2010 Sep 22
5
http://www.asterisk.org/downloads naming schema
Hi! Since some time the download of the newest Asterisk does not contains the version number anymore, but is just called "asterisk-1.4-current.tar.gz" This gives me a tarball where I do not know the version without looking into the tarball. Thus, IMO it would be very useful to switch back to old schema war the download contained the version number. Thanks Klaus
2009 Feb 25
3
Asterisk with Internet connectivity
Hi! I have a setup with Asterisk in front of a PBX connected with ISDN to the PSTN and to the PBX. This Asterisk (a old 1.2 instance) is doing ENUM for outgoing calls and allows incoming calls per SIP. Recently the IP connectivity for this location was down the whole telephony was down too - not even incoming calls did work. This is really strange as incoming calls from PSTN are routed
2008 Dec 23
2
why does users.conf generate SIP peer and SIP user?
Hi! I wonder why users.conf generates a SIP user and a SIP peer? Why is it not possible to set type=... in users.conf? (Asterisk 1.4.22) thanks klaus
2010 Feb 08
2
conferencing without DAHDI
Hi! IIRC there was an announcement some time ago that it is possible now to make conferences without the need for DAHDI anymore - but I can not remember the name of this feature anymore, and google didn't solved my problem. Thus, any references to this new system are appreciated. thanks klaus
2009 Mar 04
0
Access sip.conf's mailbox from dialplan ? [SOLVED]
2009/3/4 Klaus Darilion <klaus.mailinglists at pernau.at> > core show function SIPPEER Thanks : that's exactly what I was looking for !! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090304/7d05f172/attachment.htm
2009 Feb 24
7
multiple asterisks in a server
Hi all, Is it possible to install more than 1 asterisk in a single server? If yes, what do I need to set and take care? Rgds, ango
2013 Sep 14
1
Asterisk-1.8.23.1 mysql cdr
Hi list, I am using Asterisk1.6.2 form a long time and upgarding to Asterisk-1.8.23.1. I am using mysql backend for cdr. in asterisk-1.6.2 i have usegmtime=yes and it works as expected insert cdr date in GMT0. now i tested Asterisk-1.8.23.1 and asterisk-11.5 with same results no matter what i configure in cdr_mysql.conf "timezone=UTC usegmtime=yes" cdr always inserted in local time. I
2009 Jan 20
5
the FXS ports of Digium and damaging if connected to Tel Line
Hi All; I am facing a problem that always the users confused and connect the telephone line coming from the telephone service provider to the FXS port and cause it to be damaged, specially if the card was 2 fxs and 2 fxo, so they make mistake and connect the line to fxs while it should be connected to fxo. What is the solution for this disaster? Regards Bilal
2009 Aug 04
4
CDR Problem - No CDRs when call is not bridged
Hi! I just found out that Asterisk (1.4) does not write CDRs if the incoming call was not forwarded but handled internally without answering the call. E.g.: [from_pstn] exten => 997,1,Answer() exten => 997,2,Playback(tt-weasels) exten => 997,3,Hangup() exten => 999,1,Playback(tt-weasels|noanswer) exten => 999,4,Hangup() For incoming calls to 997 a CDR will be written, but not
2009 Feb 26
0
[cdr_odbc] error: Cannot insert the value NULL into column 'calldate'
Hi, I am trying to get * log to mssql server. I have odbc and freetds configured, but my insert query is missing calldate which is a NOT NULL field in database schema. cdr_adaptive_odbc: Insert failed on 'sqlserver:cdr'. CDR failed: INSERT INTO cdr (clid,src,dst,dcontext,channel,lastapp,lastdata,duration,billsec,disposition,amaflags,uniqueid) VALUES
2024 Jan 11
1
support for ALIAS records
While SVCB/HTTPS provides a better solution for the browsing use case, I see other use cases where ALIAS/ANAME would be ideal, notably in apex RRs. So while fostering SVCB/HTTPS deployment is a good thing, I wouldn?t mind name server software implementing ALIAS. Including NSD, but I reckon it?s much more challenging to do due to NSD architecture than it was to implement it in PowerDNS. But if