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...tmp logger -s "Doing asterisk -rx "sip reload"" asterisk -rx "sip reload" fi John Cahill Systems Engineer Services for Asterisk Data Messaging & Communications Ltd Fourth Floor 22 Lever St Manchester M1 1EA Email: john at dmcip.com Telephone: 0800 862 0181 Fax: 0161 850 0126 jabber: gnujunk at jabber.org

One of our clients has a Draytek Vigor 2920- their natted Snom phones behind it are registered to an Asterisk 1.4 server on an external public IP. I've set QOS, bandwidth management and turned off the SIP ALG via telnet but I'm still having some problems with some of the phones losing registration if Asterisk is restarted. I can see the phones sending SIP REGISTER messages, but they

...e can continue to make and receive calls, just not the person with that sip device. So I was just wondering if anyone had had that experience and if it is known to be fixed in later versions. Thanks, Naomi Rosenberg Developer Data Messaging & Communications Ltd t: 0161 850 4005 e: naomi at dmcip.com w: www.servicesforasterisk.co.uk

...attachment was scrubbed... > URL: > <http://lists.digium.com/pipermail/asterisk-users/attachments/20110408/9789f8d3/attachment-0001.htm> > > ------------------------------ > > Message: 2 > Date: Fri, 8 Apr 2011 16:57:27 +0100 (BST) > From: Naomi Rosenberg <naomi at dmcip.com> > Subject: Re: [asterisk-users] Variable inheritance with dialplan > command Originate > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: <28369896.3418.1302278246954.JavaMail.root at pomona> > C...

Hi, Apologies if this is a duplicate - been having mail server issues and I don't think I managed to send it when I tried this morning. It seems there is no .conf syntax highlighting script available for gedit. I'm thinking of putting one together myself, but don't want to reinvent the wheel. So I'm just enquiring if anyone knows of one that already exists that i've missed.

Hi, It seems there is no .conf syntax highlighting script available for gedit. I'm thinking of putting one together myself, but don't want to reinvent the wheel. So I'm just enquiring if anyone knows of one that already exists that i've missed. Thanks Naomi Rosenberg www.servicesforasterisk.co.uk

Hi, I have installed Asterisk 1.8 and am using realtime dialplan contexts from a mysql table. Asterisk keeps segfaulting. When I trace the thread ids associated with the segfaults in the full log, all they have in common is netsock2.c: == Using SIP RTP CoS mark 5 which is probably a red herring since it appears so often in threads that do not segfault. When compiling, I ticked all the

Hi, I would have thought that when spawning a channel using the Originate() dialplan command, variables prefixed with two underscores would be preserved. However this does not work in the following case. Dialplan code: [intern] exten => 200,1,Set(__myvar="foo") exten => 200,n,Originate(Local/123 at test_orig,exten,dummy) [test_orig] exten => 123,1,NoOp(${myvar}) exten =>