On Thu, Nov 27, 2008 at 8:01 PM, Daniel-Constantin Mierla
<miconda@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/>
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Daniel-Constantin Mierla
http://www.asipto.com