Asterisk: Change number in To header

A long time ago, I wrote about changing the callee ID as seen by the caller using CONNECTEDLINE.
Changing the caller ID as seen by the callee is also pretty obvious using CALLERID.
That leaves two more constellations: changing the caller ID as seen by the caller (which doesn’t make sense because a phone typically doesn’t display its own number on outgoing calls)., and changing the callee ID as seen by the callee, which I’ll talk about here now.

The reason you might want to do this is because you have multiple PSTN phone numbers that ring the same SIP phone. The obvious way to solve this would be to use

exten => _X.,n,SipAddHeader(To: "123456" <sip:123456@server>)
exten => _X.,n,Dial(SIP/${EXTEN})

, but that doesn’t work because SipAddHeader doesn’t overwrite existing headers, it only adds new ones. The Snom forum mentions a hack using the Diversion header, assuming your phone does indeed display that. A much nicer way is the following:

exten => _X.,n,Dial(SIP/${EXTEN}!123456)

. The number after the exclamation mark is simply what Asterisk uses as the local part when it composes the To URI. This features is not well-documented, but from the code I guess it was introduced in Asterisk 1.6. Asterisk 1.8’s (and higher) chan_sip.c gives a short explanation:

 *  SIP Dial string syntax:
 *       SIP/devicename
 *  or   SIP/username@domain (SIP uri)
 * or   SIP/username[:password[:md5secret[:authname[:transport]]]]@host[:port]
 * or   SIP/devicename/extension
 *  or   SIP/devicename/extension/IPorHost
 * or   SIP/username@domain//IPorHost
 * and there is an optional [!dnid] argument you can append to alter the
 *  To: header.

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