[asterisk-users] [Linksys 3102] Couple of issues

[Vincent Delporte]

Hello

	I'm several days into configuring this thing, and I still have a couple of 
issues. FWIW, I upgraded it to 3.2.10, and will only use this device to 
handle incoming calls from the PSTN, ie. no phone plugged into the Line 
1/FXS port, and no need for outgoing calls through the PSTN Line/FXO port.

Also, following the couple of documents in the PS section never worked: 
Using dial plans in either PSTN Line or PSTN User don't do anything. The 
only way I get the Linksys to notify the PBX of incoming calls is by 
setting "PSTN Line > PSTN-To-VoIP Gateway Setup > PSTN Ring Thru Line 1 = 
yes" and "User 1 > Call Forward Settings > Cfwd All Dest = <extension on 
PBX>".

Hopefully, some people here have used this newer brother of the Sipura 3000 
and know how to solve them:

- when a call comes in through the FXO port, I guess the Linksys first 
notifies the FXS port. Then, thanks to the "Cfwd All Dest" setting, it 
forwards the call to the PBX at the given extension.

Problem is, the notifcation will also make the Linksys go off-hook right 
away while it rings the PBX: If no SIP device actually answers the call, 
the caller will be needlessly charged for the call (and will only hear 
silence while the PBX rings the extension). FWIW, I left "PSTN Line > 
PSTN-To-VoIP Gateway Setup > Off Hook While Calling VoIP = no" as is. Why 
does it do this? Can it be changed?

- when an SIP device does answer the call (using the X-Free softphone, in 
case that matters), I can hear the remote PSTN caller, but he can't hear 
me. The MIC volume is OK. Does it have something to do with RTP and UDP 
ports? How do SIP devices and the Linksys know which ports to use on either 
end?

As a bonus, if someone can confirm the philosophy of this device (I think 
it's really meant for home use, with no PBX : a call comes in from the PSTN 
or VoIP and rings the FXS phone; The FXS phone is used to  make outgoing 
calls through either the PSTN or VoIP line), and tell me what the User 1 
and PSTN Line tabs are really for...

Thank you!

PS: Here are the documents... that didn't work for me:

1. http://mundy.org/blog/index.php?p=65 : changing Dial Plan 1 = (<S0:fxo>) 
doesn't do anything. Calls aren't forwarded. Although the very last comment 
says "This trick is not needed anymore in the lastest firmware release ver. 
3.1.3, have a look at the release notes. just foward to asterisk and the 
sipura wont pickup until asterisk answerd", the Linksys 3102 _does_ pick up 
the call before the PBX actually answers the call.

2. http://voxilla.com/forum-viewtopic-t-1335.html : No need to doctor the 
caller ID number for calls to be forwarded; All it takes is the "PSTN Ring 
Thru Line 1 = yes" and unconditionnal forwarding in "User 1"


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