Thanks Matt, I adjusted my code to trim the URI scheme. Regards, Nitesh
On Tue, Apr 26, 2016 at 2:45 PM, Matthew Jordan <[email protected]> wrote: > > On Fri, Apr 22, 2016 at 11:04 AM, Nitesh Bansal <[email protected]> > wrote: > >> Hello, >> >> I'm using the following Dial command syntax: >> Dial*(SIP/peer/exten!sip:[email protected] <sip%[email protected]>*), 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:[email protected] <sip%3asip%[email protected]>*, 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@[email protected] > > 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 >
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