[asterisk-users] asterisk and sipura 3102 (pstn to sip/sip to pstn calls)

Jose Flores Galicia flojose at gmail.com
Wed May 21 11:02:36 CDT 2008


I was seeing your print screen images, and the observation is.

You are not doing any sip registration on the server since your Register
option in the Tab PSTN Line is set to NO.
you should change it to yes. (or add in the sip.conf the host=SPA_ip instead
of dynamic).

regards.

-- 
Jose Flores Galicia
<<FloJoSe at gmail.com>>
BriefCode && Code Based Training
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