that tell them how to use the 0 menu and do
this by hand... but users are lazy and resent documentation.
Thanks!
Jeremy Wadhams
Yahoo Inc
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List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Force a new user to configure Comedian
mail?
Jeremy Wadhams wrote:
I know I can publish docs that tell them how to use the 0 menu and
do this by hand... but users are lazy and resent documentation.
As are Asterisk administrators
Have you tried turning off SIP fixup on the PIX? On any version below 7.x it should beno fixup protocol sip 5060no fixup protocol sip udp 5060I've had all kinds of problems with PIX firewalls peeking inside of packets (more with SCCP than SIP) and deciding not to open a port...--JW- Original
This is the code I wrote to do that job, without getting into the full phpagi:function readManagerResponse($resp, $lookFor) { $value_start = strpos($resp, $lookFor) + strlen($lookFor); $value_stop = strpos($resp, "\r\n", $value_start); if (strpos($resp, $lookFor) === FALSE ){ echo "ERROR:
Have you tried this guy's suggestion? (I have not, yet)http://slacker.com/~nugget/projects/asterisk/page7--JW- Original Message From: Joao Pereira [EMAIL PROTECTED]To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.comSent: Thursday, April 6, 2006
In both SCCP and SIP loads, dialtone comes from the phone itself, so if you're not getting that, it's probably a firmware problem. Does it affect all three sound systems (speaker, headset, handset) or just one of them?--JW- Original Message From: Paul A Brown [EMAIL PROTECTED]To: Asterisk
I had the same problem on a script, I suspect this is the first time you're using the "holdopt" variable? Try setting it to zero before the read.It looks like if holdopt is "NULL" (the user doesn't input anything and you haven't got something in the variable to start with) asterisk interprets that
