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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, 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, 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

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 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, 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'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 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 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, 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

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

Is anyone aware of a GSM femtocell that will trunk back to a VoIP softswitch such as Asterisk? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100802/4f58fea7/attachment.htm

Someone has hacked into our system and is making calls overseas. How can I: 1. Find out the where the calls are originating from? 2. Block all calls that are not authorized? Our system is in the USA. Only calls from inside our LAN are allowed. Thank you, Gary Kuznitz -------------- next part -------------- An HTML attachment was scrubbed... URL:

-----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, Has anyone seen INVAL messages on an IAX link before? I'm occasionally getting them from my Gateway provider, and I need to narrow down the potential cause. Symptoms are: Incoming calls fail, I see NEW, AUTHREQ then INVAL messages between the two A*k boxes... then for no reason at all it'll start working ok again.. My Asterisik: 1.2.10, Gateway A*k : 1.2.0

Hi, How do I display/log the results of variables from extenstions.conf? I've several macros, where I'd like to use ${CONTEXT} to help in GotoIf's, but I'm not convinced what value CONTEXT is being set to. I tried using NoOp(${CONTEXT}) and then set debug on for messages, but all I see is: Dec 7 16:09:25 VERBOSE[815] logger.c: -- Executing

All, (Happy new year!) How big can an IAX channel grow to in size? (Realistically) Eg, if I have a 2Mb pipe between two A*k servers, can IAX grow to use the whole 2Mb with no issues, or do I need to create separate IAX channels (and if so, how do you do that in the config). Cheers, Adrian

Hi All, I'm trying to understand how SIP re-invites work, if both parties are NAT'd and firewalled. I can't see how either party can initiate the conversation, and the far-end firewall would surely see any incoming packets as unsolicited, even with a STUN server giving the public IPs.. We recently moved networks, and my boss is telling me this used to work on the old network for