[asterisk-users] SIP URI set 'telephone-context='
Trey Hilyard
kctrey at gmail.com
Wed Feb 17 07:13:38 CST 2016
Agree. All you have to do is:
Dial(SIP/+${EXTEN}\;phone-context=+44 at 10.10.10.10\;user=phone)
I am actually surprised that the dialplan reload would work without it...
On Wed, Feb 17, 2016 at 5:51 AM A J Stiles <asterisk_list at earthshod.co.uk>
wrote:
> On Wednesday 17 Feb 2016, imperium broadcast wrote:
> > I kinda have it working with chan_sip.
> >
> > Dial(SIP/+${EXTEN}\;phone-context=+44 at 10.10.10.10;user=phone)
> > But it doesn't include the user=phone at the end when dialling out.
> >
> > "To: <sip:+4499999999999;phone-context=+44 at 10.10.10.10>".
> >
> > 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;phone-context=+44 at 10.10.10.10:5060;user=phone
> > Or
> > INVITE sip: 118099 at 10.10.10.10:5060; 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-
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