I have the following: Call -> PSTN -> * -> GrandStream 101 1.0.3.81 The GS displays "ohn ro n2600" when the call is past to the GS. If I pass the call to a XTEN client, Caller ID shows up. Any ideas ??
Are those caller ID numeric or have some alpha characters? GS LCD can display only some of those characters. --- John Brown <jmbrown@chagresventures.com> wrote:> I have the following: > > Call -> PSTN -> * -> GrandStream 101 1.0.3.81 > > The GS displays "ohn ro n2600" when the call > is past to the GS. > > If I pass the call to a XTEN client, Caller ID > shows up. > > > Any ideas ?? > > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users==== William Zhang
numeric
${CALLERIDNUM}
On Sat, Aug 23, 2003 at 09:38:22PM -0700, William Zhang
wrote:> Are those caller ID numeric or have some alpha characters? GS LCD can
> display only some of those characters.
>
> --- John Brown <jmbrown@chagresventures.com> wrote:
> > I have the following:
> >
> > Call -> PSTN -> * -> GrandStream 101 1.0.3.81
> >
> > The GS displays "ohn ro n2600" when the call
> > is past to the GS.
> >
> > If I pass the call to a XTEN client, Caller ID
> > shows up.
> >
> >
> > Any ideas ??
> >
> >
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> ====>
> William Zhang
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
On Saturday 23 August 2003 11:09 pm, John Brown wrote:> I have the following: > > Call -> PSTN -> * -> GrandStream 101 1.0.3.81 > > The GS displays "ohn ro n2600" when the call > is past to the GS. > > If I pass the call to a XTEN client, Caller ID > shows up. > > > Any ideas ??Mine GS (callerid) works fine without any special configuration: extensions.conf ; Extension 202 - Grandstream exten => 202,1,Playback,transfer|skip ; "Please hold while..." exten => 202,2,Dial,sip/202|20|t ; Ring, 20 secs max exten => 202,3,Voicemail,u202 ; Send to voicemail... exten => 202,5,Goto,s|6 ; Start over exten => 202,103,Voicemail,b202 ; (2 + 101) "I'm on the phone" exten => 202,104,Goto,5 ; Go to voicemail, etc. sip.conf [general] port = 5060 ; Port to bind to bindaddr = 10.125.65.10 ; Address to bind to context = default ; Default for incoming calls disallow=all allow=ulaw allow=alaw [202] type=friend ;;context=local context=routing host=dynamic insecure=yes defaultip=10.125.65.8 callerid=Steve <202> mailbox=202 dtmfmode=inband canreinvite=no> _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users-- Steve __________________________________________________ You actually need to constantly be alert and willing to handle things, or life will find a way to get you good!