Gregory Wiktor - ADCom Corp.
2005-Apr-28 19:05 UTC
[Asterisk-Users] Help to configure asterisk to dial to an PSTNline
Hello Tomek, I suspect I may have located my problem. The Hisax driver supports NI-1, but I believe my CO is sending signalling in 5ESS, which does not seem to be supported by the hisax driver.... Looks like it's back to the drawing board... Greg -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tomasz Chmielewski Sent: Thursday, April 28, 2005 5:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Help to configure asterisk to dial to an PSTNline Amit Singla wrote:> Hi Everyone, > > I have bought a Digium TDM400P and am using asterisk on RedHat 9.0. I > was able to configure Asterisk and SJPhone, so I have been able to > call from IP to IP and also from IP to a analog phone which is > attached to the digium card. > > My problem now is to dial from an IP phone to an PSTN line or any > telecom line and reverse. I don't know what changes or addition I have> to make in which files (sip.conf,extension.conf,zapata.conf etc). I > have tried to search for it but all in vain.It depends on the hardware you have. This is for an ISDN line. This means that if you dial 01234567 on your sip phone (_0.), Asterisk will dial Modem/group1 (which is configured in modem.conf), and dial your extension (${EXTEN) 01234567, without the first number (:1}) exten => _0.,1,Dial(Modem/g1:${EXTEN:1}) ; can be also Modem/ttyI0; will call through ; first available /dev/ttyI though exten => _0.,2,Congestion Dialing 6712 on your sip phone will call 12. exten => 6712,1,Dial(Modem/ttyI0:12) Try reading Asterisk Handbook Project, it should be explained there for your configuration I think. Tomek _______________________________________________ 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
All, I have the following config problem with dtmfmode I use ANTEK gw which only support dtmfmode=info but it is not supported in Asterisk voicemail. I wonder if it is possilbe to setup config that is runtime determined. I mean say, if I dial to voicemail then the asterisk can choose dtmfmode=inband or rfc2833 while switch to dtmfmode=info when I outdail to pstn. Cheers. Raymond -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050428/221b0b2c/attachment.htm