[asterisk-users] Working Sipura 3000 or Linksys 3102 configuration?

[Vincent Delporte]

Hello

	I'm having a problem with the Linksys 3102: With incoming PSTN calls, I 
can hear the caller through the X-Ten softphone, but he can't hear me. The 
problem is worse with Sjphone and the GrandStream 100 hardphone, as I get 
no sound in either direction.

FWIW...

- the SIP client, the PBX and the Linksys are all connected to a switch, 
with no firewall anywhere

- the only way I can get the Linksys to notify the PBX of an incoming PSTN 
call is using the following settings:

* PSTN Line > PSTN-To-VoIP Gateway Setup > PSTN Ring Thru Line 1 = yes
* User 1 > Call Forward Settings > Cfwd All Dest = fxo (where "fxo" is the 
account also used in PSTN Line > Subscriber Information to register with 
the PBX)

Dial plans in either "Line 1" or "PSTN Line" don't make it.

Could someone upload his configuration of the Linksys (File > Save as file) 
so I can compare with what I have?

Since both ends use G711u as their default codec and there's no firewall 
between them, I suspect I'm totally wrong when it comes to configuring the 
Linksys as a simple SIP gateway (no use for the FXS port at this point). 
Possibly some routing issue.

Thank you.


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