Hello, Steve and thank you for replying.
Yes, I know the DISA is going to the correct context because when a user direct-dials into the box (not coming in via FWD) everything works fine -- the tones are understood and acted upon. This only happens when calls are originating from a FWD connection. I'm pretty stumped on this one.
Steve Maroney wrote:
What about your dialplan ? Make sure the DISA app is going to correct context that contains the extensions that you want to dial.
Thank you, Steve Maroney
On Sun, 15 May 2005, Jeffrey Starin wrote:
Tnanks for the reply.
Well, I am only using SIP, I do not have any digium cards or IAX protocols. The Asterisk box only understand the initial dtmf tones if FWD is set to inband, anything else the caller cannot even transfer to the extension that provides DISA. Here is some other information that may help a guru troubleshoot the problem:
using Asterisk CVS-HEAD-5/15/05 on a Centos Operating system kernel 2.4.21-27.0.4.EL
Here are the pertinent entries from the SIP: [general] bindaddr=0.0.0.0 port=5060 externip=x.x.x.x localnet=192.168.1.0/255.255.255.0 disallow=all allow=ulaw canreinvite=no defaultexpirey=160 maxexpiry=180 context=nothing tos=reliability register=bla.bla.bla
[fwd] type=friend host=fwd.pulver.com username=blablabla secret=secretblablabla fromdomain=fwd.pulver.com context=from-fwd insecure=very dtmfmode=inband ;the documenation on wiki says use RFC2833, but in either case when I toggle this value same results...DISA does not accept tones canreinvite=no disallow=all allow=ulaw
Also, another pretty obvious question: I understand that for examle in the [fwd] entries above those are ths settings that take effect in any communication woth FWD. However, I'm confused as to the purpose of the "general" settings -- to what or which connection do they apply? Since the context suggested for the general settings is something like "nothing" to avoid unwanted sip calls, I'm confused as to the purposse of those entries. Can someone shed some light on that for me?
Thanks,
J
Vaniah Voip wrote:
Wilson Pickett wrote:
Any other suggestions?
Have you tried limiting it to ulaw for a test? FWD does ulaw ONLY anyway. _______________________________________________
A good question would be, are you using FWD with SIP or IAX? Anyway I am using one FWD IAX to access an IVR that allow callers to choose which extension they want and that works fine. I am also using another FWD with SIP that allows the caller to access MeetMe and the meetme requires a PIN and that works fine also. I did have a problem, when DtmfMode=Inband, so I just deleted this and it all works fine. What is the default? I have no idea, but it works great without it.
ulaw only applies to IAX channels, not SIP.
Regards Garry Taylor
------------------------------------------------------------------------
_______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
