[asterisk-dev] SIP issue (chan_sip.c:7724 set_address_from_contact) with Nortel CS1K

Christopher Thompson christopher.thompson at zen.co.uk
Sun Apr 1 12:03:26 MST 2007


There was an @ in the section I *BLANKED* originally, the phone-context 
field.

I am not entirely certain any of this will solve the issue anyway. 
Nortel's "phone-context" is basically "for show" in that that most 
enterprise domains etc are not typically accessable from the net anyway. 
Asterisk has already found the media address and port so I am not sure 
what the purpose of any of this is?


Raj Jain wrote:
> <sip:65602093;phone-context=*BLANKED*:5060;maddr=47.160.109.70 
> <http://47.160.109.70/>;transport=udp;user=phone>
>  
> The above URI seems syntactically incorrect to me. This is a tel URI 
> converted to a SIP URI. Agreed that anything before the @ sign 
> (including the semicolon) needs to treated opaquely as userinfo. The 
> host, which is a mandatory part of SIP URI, seems to be missing in the 
> above URI. 
>  
> Someone mentioned that there is an @host in the above URI. I can't see 
> it. What am I missing?
>  
> Raj
>
>  
> On 4/1/07, *Olle E Johansson* <olle at voop.com <mailto:olle at voop.com>> 
> wrote:
>
>
>     1 apr 2007 kl. 13.02 skrev Mikael Magnusson:
>
>     > Olle E Johansson wrote:
>     >> 30 mar 2007 kl. 23.52 skrev Christopher Thompson:
>     >>> Contact: <sip:65602093;phone-context=*BLANKED*:
>     >>> 5060;maddr=47.160.109.70
>     <http://47.160.109.70>;transport=udp;user=phone>
>     >> This is the culprit. There's no @ sign in this contact. nortel has
>     >> a way of implementing their own stuff, not really
>     >> bothering about interoperability.
>     >
>     > Are you sure there isn't a @ in *BLANKED*? That would make it a
>     > valid sip uri. Similar to the following from RFC 4504:
>     >
>     >     sip:5551234;phone-context=+1212 at example.net
>     <http://example.net>;user=phone
>
>     If so, then we're back to us not correctly supporting ;parameters in
>     the username part, which was supposed to be fixed.
>
>     However, I'm sure it is fixed in Asterisk SPE 1.0B.
>
>     /O
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