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BUG: 1 incoming call connected to another


mattlandis
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trace.txt** call flow **

 

nexvortex siptrunk -> HuntGroup(extensions a b c) -> snom370

 

 

** How to reproduce issue **

 

extension A B and C are in a HuntgroupA (all snom 370's)

 

call#1 comes through HuntgroupA to extension A

 

extension A is now talking to Call#1

 

call#2 comes within 15seconds of Call#1 (important) is ringing HuntgroupA

 

while ringgroupA is ringing, if extension A presses tranfer button, call#1 and call#2 are connected.

 

 

snom ONE all one versions 3981 and up.

snom 370 - 8.4.32

 

see sip trace attatched.

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** call flow **

 

nexvortex siptrunk -> HuntGroup(extensions a b c) -> snom370

 

 

** How to reproduce issue **

 

extension A B and C are in a HuntgroupA (all snom 370's)

 

call#1 comes through HuntgroupA to extension A

 

extension A is now talking to Call#1

 

call#2 is ringing HuntgroupA

 

while ringgroupA is ringing, if extension A presses tranfer button, call#1 and call#2 are connected.

 

 

snom ONE all one versions 3981 and up.

snom 370 - 8.4.32

 

see sip trace attatched.

 

Good work! we have had customers reporting this off and on for a while now, but I have never been able to recreate the issue. This sounds like it to me. Also seems to have the same issue with a mix of agent groups and hunt groups. The same thing happens if we make call 1 into a agent group..

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I tested this scenario with 2011-4.5.0.1005 Alpha Monocerotids exactly as you mention above using 2 pbx. 1 Pbx has extension 100, 200. PBX 2 has 40,41,42 in the hunt group when the 2nd call comes, I press the transfer button, it does not conjoin the calls instead the phone waits for some input. The phones are using version 8.4.34

 

 

Here is link to the latest snomONE build.

 

http://wiki.snomone.com/index.php?title=Upgrades#snom_ONE.2C_Release_2011-4.3.0.5021.2C_11.2F10.2F2011

 

as usual back up your files.

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  • 3 weeks later...

The ticket has been updated, awhile ago. We have uploaded your domain.tar file on our windows machine and downgraded our phone to 8.4.32 we tested you scenario on 4.2.1.4025 "pbxnsip" and we could not reproduce you scenario. If you can reproduce it, follow the direction on the ticket and update it with the new log file. Thanks

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This can happen, and it's important to ask your SIP provider for the RTP Port ranges they are expecting to be used. Be sure to match these port ranges on you PBX. In the case of Windows PBX's, Microsoft made a few updates in 2011 to battle security breached with DNS, and other services, and the OS will allocate a random range of ports during normal operations. Good reading begins here - http://support.microsoft.com/kb/956188

Based on our analysis, RTP streams were becoming crossed.

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