asterisk users - Oct 2007 - Grandstream GXV-3000

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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Hi.

Check the codec allowing, disallow=all and allow=ulaw etc.


At 02:25 p.m. 25/10/2007, 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)
>
>__________________________________________________
>Do You Yahoo!?
>Tired of spam?  Yahoo! Mail has the best spam protection around
>http://mail.yahoo.com
>
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>To UNSUBSCRIBE or update options visit:
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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)
>
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam protection around 
> http://mail.yahoo.com 
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>