[Asterisk-Users] DTMF detection on SIP provider ?

2003-03-09 Thread Mikael Andersson
Hi.. I just wondering why DTMF are not recognized by aterisk on incoming calls from my SIP provider ... ANy suggesteions ?` /Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] DTMF detection on SIP provider ?

2003-03-09 Thread Mikael Andersson
At 14:39 2003-03-09 -0600, Mark Spencer wrote: try the new "dtmfmode" parameters on the user or peer. Note they are not currently valid in the "[general]" section. you can set dtmfmode=inband or dtmfmode=rfc2833 Mark On Sun, 9 Mar 2003, Mikael Andersson wrote: Exactly wher

Re: [Asterisk-Users] DTMF detection on SIP provider ?

2003-03-09 Thread Mikael Andersson
At 00:50 2003-03-10 +0100, Andre Bierwirth wrote: Look into sip.conf.sample [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls ;tos=lowdelay ;tos=184 ;maxexpirey=3600

Re: [Asterisk-Users] DTMF detection on SIP provider ?

2003-03-10 Thread Mikael Andersson
At 00:50 2003-03-10 +0100, Andre Bierwirth wrote: ;[snomsip] ;type=friend ;secret=blah ;host=dynamic ;dtmfmode=inband < Hm.. I think I need more help. My ATAs are working fine, asterisk detects all DTMF input. but on incomming call from my SIP provider, it doesnt seem to work. A

[Asterisk-Users] MSN / ATA

2003-03-10 Thread Mikael Andersson
Hi.. with the new CVS I dont get my ATA to work properly .. they setup the call, but no audio is transmitted. Suggestions ? And the MSN Messenger cant login .? //Regards Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com

[Asterisk-Users] another question

2003-03-10 Thread Mikael Andersson
Sorry for all the newbie questions.. but can somebody explain this ? NOTICE[14351]: File rtp.c, Line 173 (process_rfc3389): RFC3389 support incomplete. Turn off on client if possible /Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://li

[Asterisk-Users] using a Cisco 2920 with E1 as trunk ?

2003-03-10 Thread Mikael Andersson
Hi, anyone know if it is possible to use a Cisco 2920 as my gateway to PSTN with * ? /Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Grandstream v1.0.4.68 firmware

2004-05-18 Thread Mikael Andersson
Thomas Gallaway -- <> wrote on den 17 maj 2004 16:57: > Hmmm well I need to kinda figure out how to get the custom ringtones > to ring on the phone... :-) > ___ Asterisk-Users or how to change them /M _

RE: [Asterisk-Users] Using Cisco AS5350 as pstn GW .. one-way audio problem

2004-07-09 Thread Mikael Andersson
Glen Hinkle wrote: > I assume the pstn is your * system. > Can you get audio both ways if you send the traffic back to *? > > pstn -> as5350 -> pstn ? > > -g > > > Iuse the as5350 for termination at my telco, so it's physicly located there. When I call pstn -> as5350 -> (sip) asterisk, I can

RE: [Asterisk-Users] callerid

2004-12-30 Thread Mikael Andersson
Damon Estep wrote: > Use a separate context for the outbound calls for that customer, > check the caller ID in the dialplan before completing an outbound > call using a PATTERN MATCH, and IF the pattern does not match the > pattern of the customers numbers GOTO a step that sets the caller ID > to t