I swear I tested it like that and it didn't work. But its working now so thanks guys for your help.
On 17 February 2016 at 13:13, Trey Hilyard <[email protected]> wrote: > Agree. All you have to do is: > > Dial(SIP/+${EXTEN}\;[email protected]\;user=phone) > > I am actually surprised that the dialplan reload would work without it... > > On Wed, Feb 17, 2016 at 5:51 AM A J Stiles <[email protected]> > wrote: > >> On Wednesday 17 Feb 2016, imperium broadcast wrote: >> > I kinda have it working with chan_sip. >> > >> > Dial(SIP/+${EXTEN}\;[email protected];user=phone) >> > But it doesn't include the user=phone at the end when dialling out. >> > >> > "To: <sip:+4499999999999;[email protected]>". >> > >> > even adding >> > usereqphone=yes >> > to the sip.conf doesn't add the user=phone to the end unless I remove >> the >> > the sip uri stuff out of the dial string. >> > >> > Ideally I would like it to look like this >> > INVITE sip:118099;[email protected]:5060;user=phone >> > Or >> > INVITE sip: [email protected]:5060; user=phone; phone-context=+44 >> > >> > It doesn't matter which way I do it I can only include one extra >> parameter >> > and not the two (user=phone;phone-context) as Asterisk ignores the >> second >> > one. >> >> That's because in the Asterisk dialplan, a semicolon is used to denote a >> comment (on account of the comment mark being a valid DTMF digit). So >> you >> will have to insert a backslash before the semicolon before user=phone . >> >> -- >> AJS >> >> Note: Originating address only accepts e-mail from list! If replying >> off- >> list, change address to asterisk1list at earthshod dot co dot uk . >> >> -- >> _____________________________________________________________________ >> -- 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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