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
