[asterisk-dev] [Code Review]: Truncate host:port when checking SIP domains and make the URI parsing API more explicit about returning host:port and not a domain

[ulogic]

> On Nov. 8, 2011, 7:56 p.m., Simon Perreault wrote:
> >

You might be interested in this patch I submitted almost two months ago but still has not been committed.  I have some Aastra phones that stopped working when transitioning from 1.6 to 1.8 because they send the port number in the To: header in the REGISTER packet.

https://issues.asterisk.org/jira/browse/ASTERISK-18572


- ulogic


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On Nov. 9, 2011, 12:04 a.m., Terry Wilson wrote:
> 
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> https://reviewboard.asterisk.org/r/1574/
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> 
> (Updated Nov. 9, 2011, 12:04 a.m.)
> 
> 
> Review request for Asterisk Developers and wdoekes.
> 
> 
> Summary
> -------
> 
> In reviewing https://reviewboard.asterisk.org/r/1533/ I noticed that going from 1.4 to 1.8, setting domain=x.x.x.x stopped matching when receiving a request to x.x.x.x. The reason is that the parse_uri function returns host:port in what was labeled the "domain" variable in 1.8 when in 1.4 the parsing was done in each individual function and the port truncated manually. For example, peer->fromdomain could be set to a host:port, even though it is specifically split into peer->fromdomain and peer->fromport other places in the code.
> 
> We can't change the parsing function because sometimes we actually use the host:port form of the result (as in when passing it to ast_sockaddr_resolve_first), so we need to do the truncation manually when appropriate. This patch does two things: 1) it renames the "domain" variables in the API functions to "hostport" to more accurately reflect what we are really returning and 2) implements a truncate_hostport function and uses it where appropriate.
> 
> 
> Diffs
> -----
> 
>   /branches/1.8/channels/chan_sip.c 343673 
>   /branches/1.8/channels/sip/include/reqresp_parser.h 343673 
>   /branches/1.8/channels/sip/include/sip.h 343673 
>   /branches/1.8/channels/sip/reqresp_parser.c 343673 
> 
> Diff: https://reviewboard.asterisk.org/r/1574/diff
> 
> 
> Testing
> -------
> 
> Ran the SIP unit tests and SIP-related testsuite tests and saw no difference in behavior. Set domain=127.0.0.1 and verified register requests started working.
> 
> 
> Thanks,
> 
> Terry
> 
>

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