Hello everybody, I have a problem with a FS installed on the cloud (configured following Freeswitch EC2 Wiki). Everything is ok except a one way audio behaviour that i have only if i transfer an outbound call. If I transfer an inbound call I don't have this problem. I'll explain the scenario with a little schema INBOUND CALL -> Extension 100 -> transfer to 101 WORKS. Extension 100 -> OUTBOUND CALL -> transfer to 101 ONE WAY AUDIO. I have one snom 300 (extension 100) and 3 Gigaset A510IP behind a tomato router. Can someone suggest me some way to manage this issue? Thanks in advice crazygabry
use: bin/fs_cli freeswitch@internal> sofia profile internal siptrace on (or whatever is your profile for it) Then, make a call and check for retransmission (ACK normally) or anything wrong. You will have to read some SIP documentation to compare a healthy SIP dialog with your output. Good luck! On Mon, Apr 28, 2014 at 5:36 PM, Mr.Gab <crazygabry@gmail.com> wrote:
Hello everybody,
I have a problem with a FS installed on the cloud (configured following Freeswitch EC2 Wiki). Everything is ok except a one way audio behaviour that i have only if i transfer an outbound call. If I transfer an inbound call I don't have this problem. I'll explain the scenario with a little schema
INBOUND CALL -> Extension 100 -> transfer to 101 WORKS.
Extension 100 -> OUTBOUND CALL -> transfer to 101 ONE WAY AUDIO.
I have one snom 300 (extension 100) and 3 Gigaset A510IP behind a tomato router.
Can someone suggest me some way to manage this issue?
Thanks in advice crazygabry
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-- "Efficiency is doing things right; effectiveness is doing the right things(Peter Drucker) " Miguel Oyarzo DevOps & VoIP Engineer Linux User: # 483188 - counter.li.org http://au.linkedin.com/in/mikeaustralia Melbourne, Australia
use:bin/fs_clifreeswitch <at> internal> sofia profile internal siptrace on (or whatever is your profile for it) Then, make a call and check for retransmission (ACK normally) or anything wrong.
You will have to read some SIP documentation to compare a healthy SIP dialog with your output.
Good luck!
Hi, could it be something wrong with the signaling flow? This is a log console past when i try to transfer the outbound call :) http://pastebin.com/2f39S81c And this is a console past when i transfer succesfully the incoming call http://pastebin.com/v1y2skA7 Thanks in advice
participants (2)
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Miguel Oyarzo
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Mr.Gab