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 <[EMAIL PROTECTED] > 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 >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Users mailing list >> [email protected] >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> >> > > -- > Daniel-Constantin Mierla > http://www.asipto.com > >
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