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

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imperium.broadcast at ...
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Posted: Tue Feb 16, 2016 12:03 pm Post subject: [asterisk-users] SIP URI set 'telephone-context='

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

kctrey at gmail.com
Guest

Posted: Tue Feb 16, 2016 1:03 pm Post subject: [asterisk-users] SIP URI set 'telephone-context='

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@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@gmail.com (imperium.broadcast@gmail.com)> wrote:

Quote:

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

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imperium.broadcast at ...
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Posted: Tue Feb 16, 2016 3:04 pm Post subject: [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@gmail.com (kctrey@gmail.com)> wrote:

Quote:

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imperium.broadcast at ...
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Posted: Wed Feb 17, 2016 6:38 am Post subject: [asterisk-users] SIP URI set 'telephone-context='

I kinda have it working with chan_sip.

Dial(SIP/+${EXTEN}\;phone-context=+44@10.10.10.10 (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 (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.

On 16 February 2016 at 20:03, imperium broadcast <imperium.broadcast@gmail.com (imperium.broadcast@gmail.com)> wrote:

Quote:

Thanks for the reply Trey, should of said I'm using chan_sip.
Regards
Mick On 16 Feb 2016 18:03, "Trey Hilyard" <kctrey@gmail.com (kctrey@gmail.com)> wrote:

Quote:

asterisk_list at earth...
Guest

Posted: Wed Feb 17, 2016 6:51 am Post subject: [asterisk-users] SIP URI set 'telephone-context='

On Wednesday 17 Feb 2016, imperium broadcast wrote:

Quote:

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 .

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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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kctrey at gmail.com
Guest

Posted: Wed Feb 17, 2016 8:14 am Post subject: [asterisk-users] SIP URI set 'telephone-context='

Agree. All you have to do is:

Dial(SIP/+${EXTEN}\;phone-context=+44@10.10.10.10 (44@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@earthshod.co.uk (asterisk_list@earthshod.co.uk)> wrote:

Quote:

On Wednesday 17 Feb 2016, imperium broadcast wrote:

Quote:

I kinda have it working with chan_sip.

Dial(SIP/+${EXTEN}\;phone-context=+44@10.10.10.10 (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 (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.

imperium.broadcast at ...
Guest

Posted: Wed Feb 17, 2016 8:50 am Post subject: [asterisk-users] SIP URI set 'telephone-context='

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@gmail.com (kctrey@gmail.com)> wrote:

Quote:

On Wednesday 17 Feb 2016, imperium broadcast wrote:

Quote:

I kinda have it working with chan_sip.

Dial(SIP/+${EXTEN}\;phone-context=+44@10.10.10.10 (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 (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.

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