On Wednesday 17 Feb 2016, imperium broadcast wrote:
> I kinda have it working with chan_sip.
>
> Dial(SIP/+${EXTEN}\;phone-context=+44@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@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@10.10.10.10:5060;user=phone
> Or
> INVITE sip: 118099@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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