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

[muhammad akl]

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

OpenSER : 192.168.200.10

11803 : 192.168.200.222

11801 : 192.168.200.224


The full trace :

http://muhammad.akl.googlepages.com/debug.txt

Regards
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