[asterisk-users] Dial command for SIP driver with To-header config

Nitesh Bansal nitesh.bansal at gmail.com
Fri Apr 22 11:04:31 CDT 2016


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?

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