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...sk /usr/sbin/asterisk.orig /usr/lib/asterisk /usr/include/asterisk /usr/include/asterisk.h /usr/share/man/man8/asterisk.8 Suggesting that it can see it Does anyone have any advice on what the best practice on this would be? and allowing the user asterisk to connect to the console? Thanks Robert McNaught -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071119/9884fe98/attachment.htm

...I have always tried to persuade customers to go with 2 ethernet drops per workstation to avoid having to use the phone as a switch. I apologize for this question not being directly related to asterisk, but since Polycom phones are used a lot with asterisk, it seems a good place to post ;-) Robert McNaught -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071018/8dddcc4e/attachment.htm

...stclean' and then a fresh configure :-) doing a make menuselect gives this for Resource Modules > res_odbc ODBC Resource Depends on: unixodbc(E), ltdl(E) - Cannot get rid of the XXX? In the yum repository, there is not a unixodbc. Anyone know the secret to the dependencies? Robert McNaught -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071109/f501d1a9/attachment.htm

...s anyone have any thoughts on this BTW, the /dev/zap/ctl file exists and zap is starting with errors or alarms. No Zaptel interface found. [Oct 15 10:31:21] WARNING[7036]: chan_zap.c:10026 zap_show_status: Unable to open /dev/zap/ctl: No such file or directory localhost*CLI> Cheers Robert McNaught -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071015/a416d530/attachment.htm

...ve any advice in how to implement PSTN failover should an internet connection for IAX trunking go down? to route outbound to analog lines Can this be written into the dialplan using a GotoIf statement by testing the whether the internet connection is up, or from a IAX/SIP response? Thanks Robert McNaught -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071015/9914efe8/attachment.htm

...uggest any kind of performance issue with scaling? I am assuming not as all this indicates is that DB queries are pooled from the ODBC connection on the Asterisk Server side rather than the SQL Server? Has anyone done this before in a large implementation? Any advice appreciated. Cheers Robert McNaught -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071112/a2297fa6/attachment.htm

...ith cdr_odbc and dialplan routing based on func_odbc. I have both res_odbc.conf and cdr_odbc.conf pointing to the same DSN in odbc.ini I am starting to think that this limitation in having a single connection would stop this being possible in asterisk - does anyone know otherwise? Thanks Robert McNaught -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071124/8ad59369/attachment.htm

...;,from-disa,"XXXXXXX" > <614xxxxxxxx>) > exten => valid_login,n(end),Hangup > > HTH! > Jason > > -----Original Message----- > From: asterisk-users-bounces at lists.digium.com > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Robert > McNaught > Sent: Monday, April 28, 2008 6:31 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] func_odbc creating records or best practice > > Hi, > > I am trying to write a custom application which will integrate with an > existing MSSQL cr...

Hey Oggers, I have a RedHat 7.1 machine at home, and a 6.2 system at work. After seeing Monty's announcement of the GCC optimize bug, I compiled my ogg-tools package at home with RedHat's 2.96 gcc (the errata version, not the original buggy version). I have no problems playing encoded ogg files on my home machine--they sound great. At work, I have the binary RPMs from vorbis.com,

So I'd like to try out rc3 (been a happy user of rc2 since it came out). I download the RPMS and try to do an upgrade, only to find that libao and vorbis-tools depends on 'libcurl' and 'libasound' which are not part of Red Hat 7.1 (looks like curl is in 7.2, but God knows where to get libasound). In addition, the RPMs want libcrypto.so.0, while 7.1 and 7.2 have .1. So

Hi, Does anyone know anything about the following? In a hosted environment where several area DIDs are provisioned on a single server, how do most carriers establish the origination DID, number. Asterisk allows us to modify the CallerID, name, number and DNID channel variables before dialling out via SIP. Most carriers will allow us to spoof a callerID when placing a call, and pass it forward.

I lurk and comment a little on here and have been playing with * for a short while. I am interested in hearing about the pros and cons for using a database backend to Asterisk. My current setup is simple, out of the box with config files in /etc/asterisk and logs etc going into /var. I notice a great many of the contributors here seem to use a db backend (is this also called Real Time

Hi, Does anyone use the nagios plugin for check_sip against asterisk? Does anyone have a working example of the command definition and service definition in the nagios config files? TIA Robert

Details below. Please contact me for more info. Thanks --Duncan Job Details: Release Automation Engineer CyberSource provides businesses with services and software designed to automate the processing of web, phone and other customer-not-present credit card and check transactions. Our complete suite of hosted transaction services and software products includes real time fraud screening,