[asterisk-users] Dial command for SIP driver with To-header config
nitesh.bansal at gmail.com
Wed Apr 27 06:34:01 CDT 2016
Thanks Matt, I adjusted my code to trim the URI scheme.
On Tue, Apr 26, 2016 at 2:45 PM, Matthew Jordan <mjordan at digium.com> wrote:
> On Fri, Apr 22, 2016 at 11:04 AM, Nitesh Bansal <nitesh.bansal at gmail.com>
>> I'm using the following Dial command syntax:
>> Dial*(SIP/peer/exten!sip:xyz at xyz.com <sip%3Axyz at xyz.com>*), 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 at xyz.com <sip%3Asip%3Axyz at xyz.com>*, 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 at mysipproxy!sales at 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
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