[asterisk-dev] [Code Review] SIP: peer matching by callbackextension

David Vossel dvossel at digium.com
Thu Aug 27 17:54:53 CDT 2009


-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/344/
-----------------------------------------------------------

Review request for Asterisk Developers.


Summary
-------

If there are a number of peers with different callbackextension parameters and the same host address.  The first peer found matching the address is used regardless if that peer's callbackextension matches the incoming extension or not.

Now, to better match peers with incoming calls, if an incoming call's address can match multiple peers by address, we check each of those peer's callbackextension against the incoming extension for the best possible match.

It is possible that my implementation may be too expensive and only serve to address a minor edge case in the usage of chan_sip.  I do not fully understand the impact my changes may have upon performance when a large number of peers are present.  This patch assumes the new parse_uri() change has been made.

-------------------------------------------
for example with two peers as follows
[trunk1]
host=sip.myitsp.com
callbackextension=9991
...
[trunk2]
host=sip.myitsp.com
callbackextension=9992
...

incoming calls to 9991 and to 9992 are both matched to the peer trunk1
--------------------------------------------


This addresses bug 14340.
    https://issues.asterisk.org/view.php?id=14340


Diffs
-----

  /trunk/channels/chan_sip.c 214359 

Diff: https://reviewboard.asterisk.org/r/344/diff


Testing
-------

Tested multiple peers with the same address containing different callbackextensions. Verified the correct peers were matched with incoming calls.


Thanks,

David




More information about the asterisk-dev mailing list