imperium broadcast
2016-Feb-16 20:03 UTC
[asterisk-users] SIP URI set 'telephone-context='
Thanks for the reply Trey, should of said I'm using chan_sip. Regards Mick On 16 Feb 2016 18:03, "Trey Hilyard" <kctrey at gmail.com> wrote:> Are you using res_pjsip or chan_sip? > > For PJSIP, it's as easy as passing the parameters to the Dial. For example: > Dial(PJSIP/${ARG1}\;phone-context=mydomain.com at pjsippeer,60) > > I am pretty sure it was easy in chan_sip, too. If you are using chan_sip, > I'll try and find an example. > > On Tue, Feb 16, 2016 at 11:03 AM imperium broadcast < > imperium.broadcast at gmail.com> wrote: > >> Hi all, I am currently using asterisk 11, and I am trying to figure out >> how to set the uri parameter telephone-context. >> I need to set it for outbound calls for a specific carrier when making >> emergency calls and don't seem able to find the option to set it. >> >> Regards >> Impy >> aka Mick >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160216/7d7318ae/attachment.html>
imperium broadcast
2016-Feb-17 11:37 UTC
[asterisk-users] SIP URI set 'telephone-context='
I kinda have it working with chan_sip.
Dial(SIP/+${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.
On 16 February 2016 at 20:03, imperium broadcast <
imperium.broadcast at gmail.com> wrote:
> Thanks for the reply Trey, should of said I'm using chan_sip.
>
> Regards
> Mick
> On 16 Feb 2016 18:03, "Trey Hilyard" <kctrey at gmail.com>
wrote:
>
>> Are you using res_pjsip or chan_sip?
>>
>> For PJSIP, it's as easy as passing the parameters to the Dial. For
>> example:
>> Dial(PJSIP/${ARG1}\;phone-context=mydomain.com at pjsippeer,60)
>>
>> I am pretty sure it was easy in chan_sip, too. If you are using
chan_sip,
>> I'll try and find an example.
>>
>> On Tue, Feb 16, 2016 at 11:03 AM imperium broadcast <
>> imperium.broadcast at gmail.com> wrote:
>>
>>> Hi all, I am currently using asterisk 11, and I am trying to figure
out
>>> how to set the uri parameter telephone-context.
>>> I need to set it for outbound calls for a specific carrier when
making
>>> emergency calls and don't seem able to find the option to set
it.
>>>
>>> Regards
>>> Impy
>>> aka Mick
>>> --
>>>
_____________________________________________________________________
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com
--
>>> New to Asterisk? Join us for a live introductory webinar every
Thurs:
>>> http://www.asterisk.org/hello
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>> http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
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On Wednesday 17 Feb 2016, imperium broadcast wrote:> I kinda have it working with chan_sip. > > Dial(SIP/+${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 .