similar to: Log interpretation

Hi All, Twice now in the past few weeks I've walked into the office to find that our 1.2.24 Asterisk process is sat at 100%, and that hundreds of thousands of log files in /var/log/asterisk exist, all at 312 bytes, containing: Aug 29 23:22:17 VERBOSE[24303] logger.c: Asterisk Event Logger restarted Aug 29 23:22:17 VERBOSE[24303] logger.c: Asterisk Queue Logger restarted Aug 29 23:22:17

Hi, I'm experimenting with the Meetme feature of Asterisk 1.2, exten => 2095,1,MeetMe(|Ds) This almost gives me what I want, where each employee can create their own on-the-fly conferences with a personal Conference Number and PIN. However, as the PIN is actually set by the first callee, then its subject to problems (first callee might enter the wrong PIN, and then no-one else can

Hi, I'm having complaints from some users about calls into dynamic meetme sessions failing. I'm guessing that they are dialling the wrong DTMF keys, OR that DTMF is hearing the digits entered wrong (or not hearing some). I've put debug => debug into logging.conf, and searched through the file, but I'm not sure how to debug. EG, Jun 1 14:32:33 DEBUG[14820] pbx.c: Function

Anyone have any experience with a Japanese local VoIP termination supplier? I've emailed a few companies looking to setup some PSTN to SIP and SIP to PSTN termination, but no luck so far. Thanks, Adrian -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100505/5068aaab/attachment.htm

Hi All, Anyone know if theres a way to share a Cisco 7940 between hot-desk users? My phones get their setup via SIP .cnf files, that load at boot via tftp, so I'm assuming the configs a failry static. However if I want a phone to be hot-desked, I could have different users sitting there. Is there any concept of "logging on" in these environments? Cheers, Adrian

Hi, How can I have A*k convert a call from +441793xxxxxx to Dial 00441793xxxxxx instead? With the "_+." Below I can "catch" the call, but EXTEN doesn't get set as expected.. and then I need to figure out how to pass the call onto the outgoing-pstn context. Not sure if a Goto would work here... [outgoing-pstn-international] exten => _+.,1,Set(EXTEN=00${EXTEN:+1}) exten

Can anyone tell me if it is possible to encode music in an "interlaced" format.. perhaps I should explain what I mean by this. Say I'm getting 1k/sec off some guy's modem on opennap. It would be sweet if I could listen to the song as it is downloading but obviously the quality is gunna suck. So suppose the song was encoded in this interlaced fashion, I would be able to

Hi All, I've a single 1.2.17 Asterisk system. Gradwell here in the UK is used for PSTN calls via IAX2. Our 'net link is a dedicated 2Mb fibre connection (of which we have ever used 50% max bandwidth). We've no E1/T1 links, everything is IP based. My boss complains that many of the calls he holds with others has a bad quality. He also says that its not just him. My iax.conf file

Hi All, Is there a way to have A*k record calls on-the-fly, at the users request? i.e. a possible scenario: Party A calls Party B During the call, Party A wants to start recording the call, so presses "*", A*k announces "recording.." and starting MixMonitor to a file. Once the call is finished, then A*k emails a copy of the .wav file over... I know that meetme can record

Hi, Can anyone tell me why the below dialplan doesn't filter off dialed numbers for 01793520158, and jump to "local",priority1 If I change it to : exten => 01793520158,1,Goto(local,${EXTEN:-3},1) .... then it works fine (but that's too specific)... exten => _017935201[56][0-9],1,Goto(local,${EXTEN:-3},1) exten =>

Hi, I'm trying to rsync up to some centos repositories, but I only want to pull down the i386 and i386_64 folders with their RPMs, I've tried various combinations and include and exclude, and I'm sure that the below should work, but it doesn't... SOURCE=rsync://mirror.stanford.edu/mirrors/centos rsync -avrt $SOURCE --include=i386/ --include=*/ --exclude=* /var/www/html/centos/

Hi All, I've been reading up on 1.4 snmp integration. When I try and compile asterisk with a -with-netsnmp option it complains about net-snmp installation being broken. However, the net-snmp-devel rpm is installed, and snmpd on the machine runs fine. Anyone have a guide for the pre-requisites needed ? Cheers, Adrian -------------- next part -------------- An HTML attachment

Hi All, I've two asterisk servers on the same LAN, both 1.4, and I keep getting "Got SIP response 489 "Bad event" back from 192.168.3.10" No idea whats causing it. The only references I can find mentions NATing issues, but these are on the same LAN so NAT shouldn't be an issue. 3.10 does authenticate into the server logging the error. The error appears in the log

Hi All, I'm working from home today (DSL -> Internet -> 2MB leased line -> A*K server behind NAT), and trying to pickup voicemail using Zoiper.. I can access the VM system, I hear all the prompts, and I can even hear part of the message playback. But then I get silence on the call (call stays up), and I get: Parsing

Hi All, I'd like to know if anyone has figured out a way to be able to have users logon/logoff manually from Cisco 79xx phones (with SIP firmware loaded)? Scenario is, user walks into office, sits at a random desk, and logs onto the phone. The system would need to "log them off" of the last hardphone they were on, and then configure the new phone for their extension. We're

Hi, I'm trying to add mysql CDR onto a vanilla Asterisk 1.2 install. The add-ons pack has been installed for a while, so now I'm trying to add the Mysql config. I've created a mysql database, added the grants for a user acces, and can run a mysql -u asteriskcdruser -p and can connect to the database. I've been using this as a guide:

Hi All, Just added a Siemens DECT SIP/PSTN S450 phone to login to my A*k server, and I see "Got SIP response 405 "Method Not Allowed" back from 192.168.3.64" but the phone seems to work ok. Any ideas where it falls over in the SIP protocol? I've included this in the debug below. ubiphone*CLI> <-- SIP read from 192.168.3.64:5060: --- (0 headers 0 lines) Nat

-----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of asterisk-users-request at lists.digium.com Sent: Wednesday, August 22, 2007 10:51 PM To: asterisk-users at lists.digium.com Subject: asterisk-users Digest, Vol 37, Issue 88 Send asterisk-users mailing list submissions to asterisk-users at lists.digium.com

All, I've a question on A*k hardware. I'm running 1.2.18 on a Dell DC051 (Intel(R) Celeron(R) CPU 2.80GHz) with 512mb RAM. I'm supporting 60 users (Cisco 7940s each + Xlite PCs). Call loads are low, max of about 10 simultaneous SIP/IAX calls. CPU for A*k rarely goes above 2% as I can tell. Its IP only, no E1/T1 cards present. However, I get complaints of bad voice quality,

Lookin for anyone who has experienced an issue similar to this. It's quite baffling as I'm unable to locate much help when it comes to debugging such an audio oddity. I'm currently running Asterisk 1.2.18 with a T1/E1 PRI. To cause the audio bleed (is audio bleed actually what I should even call this?): I create a meetme conference and have a few people call in to it. Once