ckleea

回復 3# bubblestar


Try the siptosis, it worths the money spent

http://www.mhspot.com/sts/siptosis_skype_trunk_howto.html

In an ATOM PC, you can easily get 4 channels working

ckleea

回復 5# bubblestar


    Just US$23 you can get the program and trunk builder. Mine is very old. Just US$9.99 for both

ckleea

回復 7# Qnewbie

I am not sure what you mean. For my setup, the quality is just the same as usual VOIP or PSTN line.
If you look at this for the changes, it just worths the money spent

http://www.mhspot.com/stsforum/viewtopic.php?id=316

Mine is still 2009 Oct one

ckleea

剛用google voice互打，網絡起碼經兩次太平洋，同固網基本上一樣。

ckleea

回復 10# Qnewbie


    I believe so. Just like conference in asterisk

Skype 1 -> skype 2 connected by siptosis to asterisk -> DTMF into conference room
PSTN user -> Asterisk box -> conference room
Skype 3 -> skyper 4 connected by siptosis to asterisk -> conference room
VOIP user  -> Asterisk box -> conference room

I have not tested this before. But as far as I know it should work

ckleea

回復 16# bubblestar


No more incoming call from gizmo5 when using asterisk 1.8. Outgoing works till now.

ckleea

Yes, I still use Gizmo5 to call but I do not register their server all the time. Only when needed.

ckleea

回復 20# bubblestar


建議如下，
register string at sip.conf to remark out not to use，即加";"
the remaining gizmo5 dial out plan reserved at sip.conf so that when needed, can dial out

ckleea

回復 23# Qnewbie
It is just put in the usual

register => 1747XXXXXXXX,password@proxy01.sipphone.net:5060; comment this line out as we do not use incoming call function in asterisk 1.8

[gizmo5]
.
.
.
.

into the sip.conf

& [CallingRule_gizmo5]
exten=> 01!,1 DIAL(SIP/{EXTEN,1}@gizmo5)


So we do not use incoming call function but only outgoing call when dial with 01xxxxxxxxx