search for: hcb

Hi All, I was wondering, in my effort to show how Asterisk can replace Call Manager, if there is support for call transfers/forwarding from the users Cisco 7940 SIP phone to either another SIP client or through the AS5300 on to the PSTN. I do see some stuff in the docs but seems to be specific to a local PRI board in the PC of which I don't have. Any experiences/comments most appreciated.

Another data point: I updated 2 machines, one a P3-UP machine and the other a P3-SMP machine. Both were fully updated with the CentOS 4.3 errata releases. I did this on each machine: yum clean all yum -y update yum yum clean all yum upgrade No errors or warnings (beyond a few config files installed as .rpmnew) were seen during the upgrade process. I rebooted immediately after the upgrade

...ads_connect: No results returned Joined domain FAK1. It succeeds, the LDAP entry is updated accordingly and nothing is logged Can someone tell me what is wrong here, or where I should have done something differently? (Samba version is 3.0.14a from debian sarge) Best regards -- Helge Bahmann <hcb@chaoticmind.net> /| \__ The past: Smart users in front of dumb terminals /_|____\ _/\ | __) Wer im finally-Block sitzt, sollte nicht \\ \|__/__| mit exceptions werfen. \\/_...

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Thomas, Have spoken to Soren, from my bootlog he believes that the problem is in atapicam causing the system to lock up. He is happy to answer some questions but doesnt have time to delve into atapicam himself. Did you author atapicam, I have seen your name on the sourcecode, can you help me further? Whats next? Thanks Adam. -------------- next part -------------- An embedded message was

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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

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I have a * setup with Grandstream SIP phones dialing out through Nikotel via SIP. When i dial out and the other side picks up, the Grandstream keeps ringing for another seconds and two and the sound coming from the other side is lost. After these two seconds the call is connected find and works flawless. Calling the same path the other way to the Grandstream phone works fine without the cut-off.

Hello, hope somebody can help. I have setup a queue which maps to some Budgetone SIP phones. When a call is answered, the # key to transfer a call does not work. Everything else regarding the queue works fine. Is there a way to activate it? Maybe something like the t option in the Dial application. Thanks in advance, Christian.