VoIP Mailing List Archives
Mailing list archives for the VoIP community |
|
| View previous topic :: View next topic |
| Author |
Message |
andrew at convergedgro...
Guest
|
 Posted: Mon Oct 19, 2015 5:45 am Post subject: [asterisk-users] Modify Contact in PJsip |
 |
|
Hi Guys
We are using the wizard to configure our pjsip trunk(see below)
How do we get this setting to work
contact_user=username
We want to change the contact field in the sip invite to display the username of the trunk
[trunk_defaults](!)
type = wizard
transport = transport-udp
endpoint/allow_subscribe = no
endpoint/allow = !all,g729
aor/qualify_frequency = 30
registration/expiration = 1800
contact_pattern=xxx
[xxx](trunk_defaults)
sends_auth = yes
sends_registrations = yes
endpoint/context = extensions
remote_hosts = xxx.xx.xx.xx
accepts_registrations = no
endpoint/send_rpid = yes
endpoint/send_pai = yes
outbound_auth/username = xxx
outbound_auth/password = xxx
contact_pattern=xxx |
|
| Back to top |
|
 |
jcolp at digium.com
Guest
|
| Posted: Mon Oct 19, 2015 6:04 am Post subject: [asterisk-users] Modify Contact in PJsip |
|
|
On 15-10-19 07:41 AM, Andrew Colin wrote:
| Quote: | Hi Guys
We are using the wizard to configure our pjsip trunk(see below)
How do we get this setting to work
contact_user=username
We want to change the contact field in the sip invite to display the
username of the trunk
|
The Contact header can not currently be modified on a per-endpoint basis
and takes its values from the generated From header. On a global scale
it could be controlled using the default_user global option. Otherwise
there's no real way without adding explicit support for it.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
--
_____________________________________________________________________
-- 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 |
|
| Back to top |
|
|
andrew at convergedgro...
Guest
|
| Posted: Mon Oct 19, 2015 6:21 am Post subject: [asterisk-users] Modify Contact in PJsip |
|
|
Hi Joshua
If i put the default_user option per endpoint would it work?
So what exactly does the contact_user option do?
I know that in freeswitch there is the option extension-in-contact.
We basically need to achieve the same functionality
Thanks
-------- Original message --------
From: Joshua Colp <jcolp@digium.com>
Date: 2015/10/19 13:03 (GMT+02:00)
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Modify Contact in PJsip
On 15-10-19 07:41 AM, Andrew Colin wrote:
| Quote: | Hi Guys
We are using the wizard to configure our pjsip trunk(see below)
How do we get this setting to work
contact_user=username
We want to change the contact field in the sip invite to display the
username of the trunk
|
The Contact header can not currently be modified on a per-endpoint basis
and takes its values from the generated From header. On a global scale
it could be controlled using the default_user global option. Otherwise
there's no real way without adding explicit support for it.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
--
_____________________________________________________________________
-- 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 |
|
| Back to top |
|
|
jcolp at digium.com
Guest
|
| Posted: Mon Oct 19, 2015 7:05 am Post subject: [asterisk-users] Modify Contact in PJsip |
|
|
On 15-10-19 08:17 AM, Andrew Colin wrote:
| Quote: | Hi Joshua
If i put the default_user option per endpoint would it work?
|
No, it's a global only option.
| Quote: |
So what exactly does the contact_user option do?
|
It sets the Contact user in an outbound registration so that the URI
dialed by the remote SIP server may contain that user (or may not,
depending on their configuration/deployment).
| Quote: |
I know that in freeswitch there is the option extension-in-contact.
We basically need to achieve the same functionality
|
It would require modifying the code and adding support.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
--
_____________________________________________________________________
-- 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 |
|
| Back to top |
|
|
andrew at convergedgro...
Guest
|
| Posted: Mon Oct 19, 2015 7:17 am Post subject: [asterisk-users] Modify Contact in PJsip |
|
|
Do you know if this can be achieved with the standard sip stack in asterisk?
Kind Regards
Andrew Colin
Converged Telecoms (Pty) Ltd.
Licensed Telecoms Operator : (0258/IECS/JAN/09) (0258/IECNS/JAN/09)
Switchboard: +27 (0)10 591 4600
Email: andrew@convergedgroup.net
Web: http://www.convergedgroup.net
75 Witkoppen Road, Northriding, Johannesburg, 2169
P O Box 7246, Weltevredenpark, 1715
This communication is confidential and intended solely for the addressee(s).
Any unauthorized review, use, disclosure or distribution is prohibited. If
you believe this message has been sent to you in error, please notify the
sender by replying to this transmission and delete the message without
disclosing it. Thank you. E-mail including attachments is susceptible to
data corruption, interception, unauthorized amendment, tampering and
viruses, and we only send and receive emails on the basis that we are not
liable for any such corruption, interception, amendment, tampering or
viruses or any consequences thereof.
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Joshua Colp
Sent: Monday, October 19, 2015 2:05 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Modify Contact in PJsip
On 15-10-19 08:17 AM, Andrew Colin wrote:
| Quote: | Hi Joshua
If i put the default_user option per endpoint would it work?
|
No, it's a global only option.
| Quote: |
So what exactly does the contact_user option do?
|
It sets the Contact user in an outbound registration so that the URI dialed
by the remote SIP server may contain that user (or may not, depending on
their configuration/deployment).
| Quote: |
I know that in freeswitch there is the option extension-in-contact.
We basically need to achieve the same functionality
|
It would require modifying the code and adding support.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
--
_____________________________________________________________________
-- 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 |
|
| Back to top |
|
|
jcolp at digium.com
Guest
|
| Posted: Mon Oct 19, 2015 7:21 am Post subject: [asterisk-users] Modify Contact in PJsip |
|
|
On 15-10-19 09:12 AM, Andrew Colin wrote:
| Quote: | Do you know if this can be achieved with the standard sip stack in asterisk?
|
If you are referring to chan_sip I don't believe so but it is possible
there is some obscure option or method to do it that I am aware of.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
--
_____________________________________________________________________
-- 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 |
|
| Back to top |
|
|
andrew at convergedgro...
Guest
|
| Posted: Mon Oct 19, 2015 7:23 am Post subject: [asterisk-users] Modify Contact in PJsip |
|
|
Ok thanks Joshua
Do you know what this error means when I dial out in pjsip and the call
fails
Unable to create request with auth.No auth credent als for any realms in
challenge
Kind Regards
Andrew Colin
Converged Telecoms (Pty) Ltd.
Licensed Telecoms Operator : (0258/IECS/JAN/09) (0258/IECNS/JAN/09)
Switchboard: +27 (0)10 591 4600
Email: andrew@convergedgroup.net
Web: http://www.convergedgroup.net
75 Witkoppen Road, Northriding, Johannesburg, 2169
P O Box 7246, Weltevredenpark, 1715
This communication is confidential and intended solely for the addressee(s).
Any unauthorized review, use, disclosure or distribution is prohibited. If
you believe this message has been sent to you in error, please notify the
sender by replying to this transmission and delete the message without
disclosing it. Thank you. E-mail including attachments is susceptible to
data corruption, interception, unauthorized amendment, tampering and
viruses, and we only send and receive emails on the basis that we are not
liable for any such corruption, interception, amendment, tampering or
viruses or any consequences thereof.
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Joshua Colp
Sent: Monday, October 19, 2015 2:21 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Modify Contact in PJsip
On 15-10-19 09:12 AM, Andrew Colin wrote:
| Quote: | Do you know if this can be achieved with the standard sip stack in
|
asterisk?
If you are referring to chan_sip I don't believe so but it is possible there
is some obscure option or method to do it that I am aware of.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at:
www.digium.com & www.asterisk.org
--
_____________________________________________________________________
-- 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 |
|
| Back to top |
|
|
|
|
|
You cannot post new topics in this forum
You cannot reply to topics in this forum
You cannot edit your posts in this forum
You cannot delete your posts in this forum
You cannot vote in polls in this forum
|
Powered by phpBB © 2001, 2005 phpBB Group
|
Search
