[asterisk-users] Grandstream GXV-3000
Mojo with Horan & Company, LLC
mojo at horanappraisals.com
Thu Oct 25 14:54:28 CDT 2007
Is your snippet from extensions.conf in the [default] context?
Are spaces OK around the '=' in sip.conf? They might be, just an idea.
Now I notice that for friend 112, you say codecs 'allow=all' and for
friend 113 you say 'allow=h263' -- maybe you need to explicitly allow
something like ulaw on friend [113] (thinking maybe they're just not
agreeing on codecs...?) Try adding 'disallow=all' to [113]'s
definition, just before the 'allow=h263' and the 'allow=ulaw' i'm
suggesting. If, theoretically, friend 113 will ONLY use h263, does
X-Lite support this codec?
Have you tried kicking the verbose level at the console up a little bit?
Moj
hin lee wrote:
> I am trying to set up a Grandstream GXV-3000 Video
> phone to Asterisk ver 1.2.21.1. The problem I'm
> having is that it can call other SIP phones, but not
> vice versa. Can someone tell me where is the problem?
> TIA!
>
> Here's part of my configurations:
>
> ----------
> sip.conf
> ----------
> ; 113 is the Grandstream phone
> [113]
> type=friend
> username=113
> secret=secret
> context=default
> dtmfmode = rfc2833
> host = dynamic
> qualify = yes
> allow = h263
> video=yes
> videosupport=yes
>
> ; 112 is the X-Lite phone
> [112]
> type=friend
> host=dynamic
> user=112
> username=112
> secret=secret
> allow=all
> nat=no
>
> -------------
> extensions.conf
> -------------
> exten => 112,1,Dial(SIP/112)
> exten => 112,2,Playback(vm-nobodyavail)
> exten => 112,102,Playback(tt-allbusy)
> exten => 112,103,Voicemail(b112 at default)
>
> exten => 113,1,Dail(SIP/113)
> exten => 113,2,Playback(vm-nobodyavail)
> exten => 113,102,Playback(tt-allbusy)
> exten => 113,103,Voicemail(b113 at default)
>
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