Fw: [Asterisk-Users] Possible Sip logic bug?

Clif Jones ctjones at earthlink.net
Thu Feb 5 06:08:29 MST 2004


Rich,

Try it again after executing: "sip debug" and give us the actual SIP 
messages.  The devil
is usually in the details. 

Rich Adamson wrote:

>Anyone have comments on this? Really could use some suggestions or ideas
>why this is happening.  Thanks.
>Rich
>
>------------------------
>  
>
>>Anyone recognize this as a sip logic bug?
>>
>>Example Case:
>> C7960 -> * -> sip gateway -> pstn
>> (sip gateway config'ed with canreinvite=no, but shouldn't have an
>>  impact on this.)
>>
>>Outgoing call initiated from C7960. Call is completed and conversation
>>is very much normal. All equipment on the same wire; no nat.
>>
>>The C7960 user hangs up the phone. Pkt flows (as observed by sniffer)
>>are:
>>
>>C7960 sends sip BYE packet to *
>>  * returns 200 OK
>>* sends INVITE to sip gateway    <<================ where is BYE?
>>  sip gateway responds with 100 Trying
>>  sip gateway responds with 200 OK
>>  sip gateway responds with 200 OK
>>  sip gateway responds with 200 OK
>>
>>The end result, the sip gateway does not drop the pstn line until the
>>"called" number hangs up.
>>
>>It would appear that asterisk has an issue dropping the call. When the
>>C7960 issues the BYE, I would expect * to send a BYE to the sip g/w.
>>Is this a * logic problem (or my logic problem)?
>>
>>(I'm actually running CVS-12/04/03-14:24:40 and has been very stable
>>in this production environment. Is it time to update this one even
>>though it is 99% sip hardphone based?)
>>
>>Rich
>>
>>
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>
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>
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