Hello Everyone.
I've got a glitch in my call waiting feature and was hoping someone could suggest a way to fix it or debug it.
The Problem; I have three SIP phones. Two are Grandstream Budgetone 100's and the third is a Cisco ATA 186. The firmware on the Grandstreams is version 1.0.5.11 and the ATA 186 is version 2.16.2
If a call is established between two of the phones and the third phone calls, I do get the call waiting indicator. I can then press my "flash" button to take the call. I can also press "flash" again to return to the original caller.
But if I press "flash" a third time to return to the new caller, audio can be sent but not received. It's almost like one of the RTP channels has been closed.
Could this be an error in my ser.cfg routing? I'm using ser-0.8.99-dev7
Regards, Paul
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On Oct 01, 2004 at 09:40, Java Rockx javarockx@yahoo.com wrote:
Hello Everyone.
I've got a glitch in my call waiting feature and was hoping someone could suggest a way to fix it or debug it.
The Problem; I have three SIP phones. Two are Grandstream Budgetone 100's and the third is a Cisco ATA 186. The firmware on the Grandstreams is version 1.0.5.11 and the ATA 186 is version 2.16.2
If a call is established between two of the phones and the third phone calls, I do get the call waiting indicator. I can then press my "flash" button to take the call. I can also press "flash" again to return to the original caller.
But if I press "flash" a third time to return to the new caller, audio can be sent but not received. It's almost like one of the RTP channels has been closed.
Could this be an error in my ser.cfg routing? I'm using ser-0.8.99-dev7
Are the phones behind NATs? If so make sure you catch the re-INVITEs and force_rtp_proxy for them.
Andrei
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Andrei Pelinescu-Onciul -
Java Rockx