Bruin08
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Posts posted by Bruin08
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For anyone looking - I reset the unit to the factory defaults and that fixed the problem - although why I should have to do that is beyond me.
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Hi,
i have multiple SIP endpoints configured to register and dial into a conference configured on the cs-425.
This was all working fine until a couple of days ago. Nothig has changed. The units are still trying to register but it looks like there are errors in the log. Any help is appreciated. Here is a partial log for one of the units.
[7] 2008/12/15 10:55:10: SIP Tx udp:192.168.0.100:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.100:5060;rport=5060;branch=z9hG4bK1395208270
From: <sip:401@192.168.0.119>;tag=93843748
To: <sip:700@192.168.0.119>;tag=286c37868a
Call-ID: 1045378098@192.168.0.100
CSeq: 20 INVITE
Contact: <sip:700@192.168.0.119:5060>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: pbxnsip-PBX/3.0.1.3023
Content-Length: 0
[9] 2008/12/15 10:55:10: Resolve 2194: aaaa udp 192.168.0.100 5060
[9] 2008/12/15 10:55:10: Resolve 2194: a udp 192.168.0.100 5060
[9] 2008/12/15 10:55:10: Resolve 2194: udp 192.168.0.100 5060
[7] 2008/12/15 10:55:10: SIP Tx udp:192.168.0.100:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.100:5060;rport=5060;branch=z9hG4bK1395208270
From: <sip:401@192.168.0.119>;tag=93843748
To: <sip:700@192.168.0.119>;tag=286c37868a
Call-ID: 1045378098@192.168.0.100
CSeq: 20 INVITE
Contact: <sip:700@192.168.0.119:5060>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: pbxnsip-PBX/3.0.1.3023
Content-Length: 0
[7] 2008/12/15 10:55:10: SIP Rx udp:192.168.0.100:5060:
ACK sip:700@192.168.0.119:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.100:5060;rport;branch=z9hG4bK1395208270
From: <sip:401@192.168.0.119>;tag=93843748
To: <sip:700@192.168.0.119>;tag=286c37868a
Call-ID: 1045378098@192.168.0.100
CSeq: 20 ACK
Content-Length: 0
[7] 2008/12/15 10:55:10: Last message repeated 2 times
[9] 2008/12/15 10:55:10: Message repetition, packet dropped
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Hi,
We are using the cs-425 in a unique way. We have our own manufactured SIP endpoints that are configured to auto register with the cs-425 and auto call into a preconfigured conference.
There are 11 devices - x401 through x411. All devices are calling int x700 so they can conference with eachother.
So, all devices are listed in the call status (401-411). Everything is good.
The problem I am having is this: if the endpoint loses and regains power, or is rebooted, it calls back into the conference and shows up twice in the call status. This can create a problem as the cs-425 thinks the ports are being used when they are not in reality.
Is there a way for the call to be disconnected by the cs-425 if the device is no longer connected, or 'unregisters ungracefully'?
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If you don't have to, don't upgrade. Release notes for 3.1 are here: http://wiki.pbxnsip.com/index.php/Release_Notes_3.1. Maybe you wait for 3.1.1; then you will probably have no upgrade problem.
Thanks!
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New to this forum and I just purchased a cs-425 for a SIP application. The unit is currently running version 3.0.1.3023. I noticed an uprade is available to 3.1. Are there any enhancements to this upgrade that make it worthwhile at this time?
Lost Configuration and License
in Embedded
Posted
Been having issues with this new cs-425. I reset to factory defaults and it looked like it took care of the problem I was having (endpoints not able to register).
I checked it this morning and my domain is gone completely, along with the accounts. The static IP I gave the unit is still there, but when I go to create a new Domain I receive the message: "There are no more licenses available. The domain has not been created".
Currently running version 3.0.1.3023.