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

Ron Wellsted ron at wellsted.org.uk
Wed Aug 23 12:57:48 MST 2006


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incent Delporte wrote:
> 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.

Here are mine (with UK regional settings/A-law).

http://www.wellsted.org.uk/spa3102router.html for the router
configuration as bridge and
http://www.wellsted.org.uk/spa3102voice.html for my voice configuration
with UK regionalisation (A-law, UK tones/cadences).




- --
Ron Wellsted
ron at wellsted.org.uk http://www.wellsted.org.uk
N 52.567623, W 2.137621 Linux Counter No. 202120
FWD:519961
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