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
>
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>
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