search for: kuj

Has anyone got this to work? Under Idle Display Animation, the administrators guide says "For example, a company logo could be displayed".. In the ipmid.cfg file, I enabled 'ind.idleDisplay.enabled' (ie changed it to 1), and under the IP 500 section, I added an entry for the bitmap that I want to display: bitmap.IP_500.66.name ="arf" but from there I'm not sure

I am using chan_sccp on bristuffed asterisk (0.2RC3 on asterisk 1.0.3). Things seem fine but I am seeing some weird stuff. I have a Kirk IP600 connecting to * with 2 handsets. The weird thing is that for incoming calls the handset that is put second as my dialstring, never rings. This is my dial string: exten => 666666666,4,Dial(${PHONE1}&${DECT1}&${DECT2}),25,tm) Where DECT1

Somewhat off topic. We upgraded a Polycom phone from SIP v1.6.3 to v1.6.6 The phone will no longer send SIP subscription messages for buddies to Asterisk. I have broken the directory file down to make it as simple as possible. Here is what it contains. <?xml version="1.0" encoding="UTF-8" standalone="yes"?> <!-- $Revision: 1.2 $ $Date: 2004/12/21 18:28:05

...l as "LAGRQ" packets. (It doesn't matter though what asterisk version is running on A, 1.2.x or 1.4.x) Does anyone have an RSA authenticated setup working with a 1.4.x based asterisk server receiving calls just fine? Am I missing something here before I try the bug route? Thanks! -- kuj

...--- Fix Queue Log Reporting Every Call COMPLETECALLER With "h" Extension Present (Closes issue ASTERISK-20743. Reported by call) * --- Retain XMPP filters across reconnections so external modules continue to function as expected. (Closes issue ASTERISK-20916. Reported by kuj) * --- Ensure that a declined media stream is terminated with a '\r\n' (Closes issue ASTERISK-20908. Reported by Dennis DeDonatis) * --- Fix pjproject compilation in certain circumstances (Closes issue ASTERISK-20681. Reported by Dinesh Ramjuttun) For a full list of changes in this r...

...--- Fix Queue Log Reporting Every Call COMPLETECALLER With "h" Extension Present (Closes issue ASTERISK-20743. Reported by call) * --- Retain XMPP filters across reconnections so external modules continue to function as expected. (Closes issue ASTERISK-20916. Reported by kuj) * --- Ensure that a declined media stream is terminated with a '\r\n' (Closes issue ASTERISK-20908. Reported by Dennis DeDonatis) * --- Fix pjproject compilation in certain circumstances (Closes issue ASTERISK-20681. Reported by Dinesh Ramjuttun) For a full list of changes in this r...

Hello, I have a mixture of Polycom SP IP 500 and 300 phones. I have been reading through the administration manual to try and solve this problem, but I do not seem to be able to find the answers to my question. I figured I would ask here and see if anyone has some suggestions. The problem is kind of annoying. After dialing 4 digits, the phone seems to pause and miss the 5th digit, often

Outoing calls I make using Motif Google Voice Calls continue ringing even after the other end picks up. I have to restart Asterisk to resolve the issue. I don't see any errors. It's not recognizing that the other party picked up the phone and restarting Asterisk fixes it only for a day. -- Co-op Vacation Rentals www.coopvr.com 15218 Summit Ave Suite #300-354 Fontana, CA 92336

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Hello list. I'm trying to rebuild the 2.6.9.67.0.20.EL kernel, but it fails even without modifications. How did I try it? Created a (non-root) build environment (not a mock ) Installed the kernel.scr.rpm and did a rpmbuild -ba --target=`uname -m` kernel-2.6.spec 2> prep-err.log | tee prep-out.log The build failed at the end: Processing files: kernel-xenU-devel-2.6.9-67.0.20.EL Checking

Quick question for the experts: I'm seeing stuck SIP channel(s) scheduled for destruction stay open. This appears to happen after (apparently successfully) registering with a SIP peer. Any ideas where to start digging into this? I'm running today's CVS, however the problem existed before and does persist. Thanks, Kai-Uwe Example: (registration with iconnecthere.com) ast1000*CLI>