Hi Carlos: I'm not using SIP endpoints, the problem is with FXS/DAHDI endpoints. Anyway I could see I'm using RFC2833 for SIP extensions. Tks.
Regards, *Ramiro PAZ MASTERLINE LOGISTICS *** On Mon, Oct 24, 2011 at 3:33 PM, Carlos Rojas <[email protected]> wrote: > Hello, > > That sound a tones problem, what do you seting, dtmf in your sip.conf? > > Regards > > > On Mon, Oct 24, 2011 at 2:15 PM, Ramiro Paz < > [email protected]> wrote: > >> Hi everibody: >> >> Sorry, I want to relive this issue. I still have the problem, if somebody >> could help me will be appreciated. Tks. >> >> *Ramiro PAZ >> MASTERLINE LOGISTICS >> * >> ** >> On Wed, Oct 19, 2011 at 3:25 PM, Ramiro Paz < >> [email protected]> wrote: >> >>> Hi Danny, Warren: >>> >>> This is what I found in extensions_additional.conf: >>> >>> [from-internal-additional] >>> include => from-internal-additional-custom >>> include => app-dialvm >>> include => app-vmmain >>> include => app-recordings >>> include => app-callwaiting-cwoff >>> include => app-callwaiting-cwon >>> include => ext-group >>> include => grps >>> include => ext-queues >>> include => app-queue-toggle >>> include => app-calltrace >>> include => app-directory >>> include => app-echo-test >>> include => app-speakextennum >>> include => app-speakingclock >>> include => app-cf-busy-off >>> include => app-cf-busy-off-any >>> include => app-cf-busy-on >>> include => app-cf-off >>> include => app-cf-off-any >>> include => app-cf-on >>> include => app-cf-unavailable-off >>> include => app-cf-unavailable-on >>> include => app-cf-toggle >>> include => app-fmf-toggle >>> include => ext-findmefollow >>> include => fmgrps >>> include => app-userlogonoff >>> include => ext-local-confirm >>> include => findmefollow-ringallv2 >>> include => app-pickup >>> include => app-zapbarge >>> include => app-chanspy >>> include => ext-test >>> include => ext-local >>> include => outbound-allroutes >>> exten => h,1,Hangup >>> >>> ; end of [from-internal-additional] >>> >>> There is nothing for [from-internal-custom]. I mean >>> extensions_custom.conf is empty. >>> >>> Just in case, Warren is right, we're using FXS/DAHDI endpoints. Thanks >>> for your time. >>> * >>> Ramiro PAZ >>> **MASTERLINE LOGISTICS* >>> >>> >>> On Wed, Oct 19, 2011 at 12:59 PM, Warren Selby <[email protected]>wrote: >>> >>>> On Wed, Oct 19, 2011 at 11:28 AM, Danny Nicholas <[email protected]>wrote: >>>> >>>>> Or you could just add these lines to [from-internal-xfer] >>>>> >>>>> Exten => _X,1,Dial(SIP/${EXTEN},30,iKkTtt)**** >>>>> >>>>> Exten => _XX,1,Dial(SIP/${EXTEN},30,iKkTt)**** >>>>> >>>>> ** ** >>>>> >>>>> If you have 3 or 4 digit extensions you would need these lines**** >>>>> >>>>> Exten => _XXX,1,Dial(SIP/${EXTEN},30,iKkTtt)**** >>>>> >>>>> Exten => _XXXX,1,Dial(SIP/${EXTEN},30,iKkTt)**** >>>>> >>>>> >>>>> >>>> Except he's not sending to SIP endpoints, he's sending to FXS / DAHDI >>>> endpoints. So the syntax would be a bit more specific based on which >>>> extension was being dialed and which port it was hooked up to on the card. >>>> >>>> -- >>>> Thanks, >>>> --Warren Selby, dCAP >>>> http://www.SelbyTech.com <http://www.selbytech.com> >>>> >>>> >>>> -- >>>> _____________________________________________________________________ >>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>>> New to Asterisk? Join us for a live introductory webinar every Thurs: >>>> http://www.asterisk.org/hello >>>> >>>> asterisk-users mailing list >>>> To UNSUBSCRIBE or update options visit: >>>> http://lists.digium.com/mailman/listinfo/asterisk-users >>>> >>> >>> >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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