[asterisk-users] SIP URI set 'telephone-context='

[imperium broadcast]

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 <kctrey at gmail.com> wrote:

> 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-
>> list, change address to asterisk1list at earthshod dot co dot uk .
>>
>> --
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