[Kamailio-Users] Attendant call transfer between asterisk gateway and OpenSER problem

[Daniel-Constantin Mierla]

Hello,

openser does just routing job in this case. If the sip requests reach 
the endpoints properly, then the issue is probably in on one of them.

As I can get from your summary, Asterisk does not have active the call 
to be replaced.

Cheers,
Daniel

On 11/27/08 10:03, muhammad akl wrote:
> I have the following scenario :
>
>
> Pstn Number(1234567) <-----------> Asterisk GW <----------------> 
> Openser | <-------------->11803
>                                                                       
>                                            |
>                                                                                                                  
> |
>                                                                                                                  
> | <--------------> 11801
>
> Firstly extension 11803 will call the pstn number and this works fine  
> without no problem , after that 11803 will put 1234567 on hold and 
> will call 11801 , then 11803 will transfer 1234567 to 11801 (<---- the 
> problem now started ), what is happening now is that both 123456 and 
> 11801 will be on hold with 11803 after the transfer is done
>
> I've traced the full dialog between the three extensions and found an 
> interesting part , which a NOTIFY message came from asterisk and 
> contains this sentence : SIP/2.0 481 Call leg/transaction does not exist
>
> The addresses of the devices as follows :
>
> Asterisk Gw : 192.168.200.202 <http://192.168.200.202/>
>
> OpenSER : 192.168.200.10 <http://192.168.200.10/>
>
> 11803 : 192.168.200.222 <http://192.168.200.222/>
>
> 11801 : 192.168.200.224 <http://192.168.200.224/>
>
>
> The full trace :
>
> http://muhammad.akl.googlepages.com/debug.txt
>
> Regards
> ------------------------------------------------------------------------
>
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-- 
Daniel-Constantin Mierla
http://www.asipto.com