VoIP Mailing List Archives
Mailing list archives for the VoIP community |
|
| View previous topic :: View next topic |
| Author |
Message |
nitesh.bansal at gmail...
Guest
|
 Posted: Fri Apr 22, 2016 11:05 am Post subject: [asterisk-users] Dial command for SIP driver with To-header |
 |
|
Hello,
I'm using the following Dial command syntax:
Dial(SIP/peer/exten!sip:xyz@xyz.com ([email]sip%3Axyz@xyz.com[/email])), the SIP URI
after the '!' mark should be set as To-URI in outgoing INVITE
from Asterisk.
It works, but problem is that To-URI formatting is a bit messed up,
It looks as follows:
sip:sip:xyz@xyz.com ([email]sip%3Asip%3Axyz@xyz.com[/email]), it seems that Asterisk added an extra 'sip:' in the
To-header and it breaks.
I'm using Asterisk 13.
I'm wondering if this behaviour is intended or a potential bug?
Thanks,
Nitesh |
|
| Back to top |
|
 |
mjordan at digium.com
Guest
|
| Posted: Tue Apr 26, 2016 7:47 am Post subject: [asterisk-users] Dial command for SIP driver with To-header |
|
|
On Fri, Apr 22, 2016 at 11:04 AM, Nitesh Bansal <nitesh.bansal@gmail.com (nitesh.bansal@gmail.com)> wrote:
| Quote: | Hello,
I'm using the following Dial command syntax:
Dial(SIP/peer/exten!sip:xyz@xyz.com ([email]sip%3Axyz@xyz.com[/email])), the SIP URI
after the '!' mark should be set as To-URI in outgoing INVITE
from Asterisk.
It works, but problem is that To-URI formatting is a bit messed up,
It looks as follows:
sip:sip:xyz@xyz.com ([email]sip%3Asip%3Axyz@xyz.com[/email]), it seems that Asterisk added an extra 'sip:' in the
To-header and it breaks.
I'm using Asterisk 13.
I'm wondering if this behaviour is intended or a potential bug?
|
I would think that it isn't a bug. If you look at the documentation of that dial string option for the chan_sip channel driver in sip.conf.sample, you can see that the URI scheme is left off:
54 ; All of these dial strings specify the SIP request URI.
55 ; In addition, you can specify a specific To: header by adding an
56 ; exclamation mark after the dial string, like
57 ;
58 ; SIP/sales@mysipproxy!sales@edvina.net (sales@edvina.net)
While it might be nice if it didn't always use a scheme of 'sip', that'd probably be categorized as an improvement to this option.
--
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org |
|
| Back to top |
|
|
nitesh.bansal at gmail...
Guest
|
| Posted: Wed Apr 27, 2016 6:34 am Post subject: [asterisk-users] Dial command for SIP driver with To-header |
|
|
Thanks Matt, I adjusted my code to trim the URI scheme.
Regards,
Nitesh
On Tue, Apr 26, 2016 at 2:45 PM, Matthew Jordan <mjordan@digium.com (mjordan@digium.com)> wrote:
| Quote: |
On Fri, Apr 22, 2016 at 11:04 AM, Nitesh Bansal <nitesh.bansal@gmail.com (nitesh.bansal@gmail.com)> wrote:
| Quote: | Hello,
I'm using the following Dial command syntax:
Dial(SIP/peer/exten!sip:xyz@xyz.com ([email]sip%3Axyz@xyz.com[/email])), the SIP URI
after the '!' mark should be set as To-URI in outgoing INVITE
from Asterisk.
It works, but problem is that To-URI formatting is a bit messed up,
It looks as follows:
sip:sip:xyz@xyz.com ([email]sip%3Asip%3Axyz@xyz.com[/email]), it seems that Asterisk added an extra 'sip:' in the
To-header and it breaks.
I'm using Asterisk 13.
I'm wondering if this behaviour is intended or a potential bug?
|
I would think that it isn't a bug. If you look at the documentation of that dial string option for the chan_sip channel driver in sip.conf.sample, you can see that the URI scheme is left off:
54 ; All of these dial strings specify the SIP request URI.
55 ; In addition, you can specify a specific To: header by adding an
56 ; exclamation mark after the dial string, like
57 ;
58 ; SIP/sales@mysipproxy!sales@edvina.net (sales@edvina.net)
While it might be nice if it didn't always use a scheme of 'sip', that'd probably be categorized as an improvement to this option.
--
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://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 |
|
|
|
|
|
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
