28 Nov
2008
28 Nov
'08
5:58 p.m.
Thanks miconda for your response , I know OpenSER does his job and the
problem is particularly found in Asterisk . I'll try to search about that .
Regards
On Thu, Nov 27, 2008 at 8:01 PM, Daniel-Constantin Mierla <miconda(a)gmail.com
wrote:
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