[Devel] userinfo part of the SIP

[Bogdan-Andrei Iancu]

Hi Di-Shi,

the uri parser will blindly accept the username part. If you want to 
interpret the username, you have to do it "by hand" from the script 
(maybe based on regexp, etc).

regards,
bogdan

Di-Shi Sun wrote:

>Hi Bogdan,
>
>Thank you very much for your reply. I found I made a mistake, the SIP header
>should be "To: <sip:8001234567; phone-context=+ at 172.17.2.3:5060;
>user=phone>". It did have "sip:".
>
>Does that mean openser support this format?
>
>Thanks,
>
>Di-Shi Sun.
>
>----- Original Message ----- 
>From: "Bogdan-Andrei Iancu" <bogdan at voice-system.ro>
>To: "Di-Shi Sun" <di-shi at transnexus.com>
>Cc: <devel at openser.org>
>Sent: Tuesday, January 24, 2006 3:12 PM
>Subject: Re: [Devel] userinfo part of the SIP
>
>
>  
>
>>Hi,
>>
>>looks that the userpart is an telephone-subscriber as defined in
>>rfc2806, but there is no "sip:" at the beginning of the URI....so , in
>>this format it will not work.
>>
>>regards,
>>bogdan
>>
>>Di-Shi Sun wrote:
>>
>>    
>>
>>>Hi All,
>>>
>>>Our customer's gateway sent SIP INVITE messages with userinfo part.
>>>For example, "To: <8001234567; phone-context=+ at 172.17.2.3:5060
>>><mailto:phone-context=+ at 172.17.2.3:5060>; user=phone>". Would any one
>>>please tell me if openser fully supports this format of SIP INVITE
>>>messages? If it does, how can I get the called/calling number from the
>>>INVITE messages using openser functions?
>>>
>>>Thanks,
>>>
>>>Di-Shi Sun.
>>>
>>>------------------------------------------------------------------------
>>>
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>>>Devel mailing list
>>>Devel at openser.org
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>>>
>>>
>>>      
>>>
>>
>>    
>>
>
>
>  
>