[Asterisk-Users] Re: Making a SIP call

[bclark at bwkip.com]

I am still having this problem of only capturing part of the IP address, I
am currently checking into a possible hardware/software issue on the
client side but was wondering if there are any setting I need to set on
the asterisk server to allow an peer to peer call. I have set
dtmfmode=inband.  Is there anything else I need to set?

Brian



> Message: 5
> From: "David J Carter" <david.carter at codepipe.com>
> To: <asterisk-users at lists.digium.com>
> Subject: RE: [Asterisk-Users] Making a SIP call
> Date: Sat, 22 May 2004 00:14:55 +0100
> Reply-To: asterisk-users at lists.digium.com
>
> Check your sip.conf
>
> Make sure the dtmfmode is set the same as the phone.
>
> I had this before.
>
> Usually to dial an IP address you have a keystroke before you enter the
> address.
> I think on a Grandstream phone you press the menu button then the IP
> address.
>
> Dave
>
> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com
> [mailto:asterisk-users-admin at lists.digium.com]On Behalf Of
> bclark at bwkip.com
> Sent: 21 May 2004 21:57
> To: asterisk-users at lists.digium.com
> Subject: [Asterisk-Users] Making a SIP call
>
>
> If someone could point me in the right direction I would much appreciate
> it.  Here is my problem:
>
> My directions for my sip phone says to dial an ip address 12*34*65*78#.
> When I dial that into my phone my asterisk server is only picking up some
> of the numbers in the above example it would pick up 6578.  Then of course
> not find it and ring busy on the phone. The same is true for dialing a
> regular phone number ( it seems to pick up 4 digits or so)
>
> I very new to setting this up so I imagine I need to make a change to a
> config file, but don't know where to start.