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.
Check the dial-plan of your SIP device. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of bclark@bwkip.com Sent: Friday, May 21, 2004 3:57 PM To: asterisk-users@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. _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of bclark@bwkip.com Sent: 21 May 2004 21:57 To: asterisk-users@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. _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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@codepipe.com> > To: <asterisk-users@lists.digium.com> > Subject: RE: [Asterisk-Users] Making a SIP call > Date: Sat, 22 May 2004 00:14:55 +0100 > Reply-To: asterisk-users@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@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of > bclark@bwkip.com > Sent: 21 May 2004 21:57 > To: asterisk-users@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.
bclark@bwkip.com wrote:> 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?dtmfmode=inband only works with the ulaw and alaw codecs. If you use any other codec you MUST use rfc2833 or info DTMF modes (set on the phone AND on Asterisk)
Well I am getting the phones to ring but have no voice. When someone dials an IP number does this circumvent the * server? I was trying to make a capture of the call with ethereal but saw no traffic at the server for the call. Unfortunatly I have no way to set the dtmfmode on the phone side so I am stuck with inband. Is there something I am missing that is causing the lack of voice on the line. Brian Date: Mon, 24 May 2004 16:20:36 -0500 From: Eric Wieling <eric@fnords.org> To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Re: Making a SIP call Reply-To: asterisk-users@lists.digium.com bclark@bwkip.com wrote:> 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?dtmfmode=inband only works with the ulaw and alaw codecs. If you use any other codec you MUST use rfc2833 or info DTMF modes (set on the phone AND on Asterisk)