Hi all,
our asterisk is connected to a sip proxy through a sip trunk. Let's say we
have following dial plan (only an example)
[from_sip_proxy]
exten => 36122512,1,Answer()
exten => 36122512,2,VoiceMailMain()
exten => 3612252,1,Answer()
exten => 3612252,2,MeetMe(313,MI)
exten => 3612252,3,HangUp()
exten => 36122530,1,Answer()
exten => 36122530,2,MusicOnHold()
Overlap from pstn works fine and you can see that asterisk answers with 484
address incomplete as long there is no match.
But if we change our dial plan like the following (we have different
extensions with different length)
[from_sip_proxy]
exten => _36122.,1,Goto(local,${EXTEN:5},1)
[local]
exten => 512,1,Answer()
exten => 512,2,VoiceMailMain()
exten => 52,1,Answer()
exten => 52,2,MeetMe(313,MI)
exten => 52,3,HangUp()
exten => 530,1,Answer()
exten => 530,2,MusicOnHold()
We can notice that incoming calls (e.g for 36122512) are now routed by
asterisk from context [from_sip_proxy] to context [local] and overlap
doesn't work anymore. The answer is 603 Declined.
[CLI]
Sep 4 15:15:21] WARNING[28382]: pbx.c:2450 __ast_pbx_run: Channel
'SIP/192.168.148.186-08c16fe0' sent into invalid extension '5' in context
'local', but no invalid handler
[/CLI]
We think that here the answer for the INVITE 361225 should also be 484
address incomplete and same thing for the next INVITE for 3612251 and finaly
100 Trying for the last INVITE 36122512. Can anyone please confirm this.
thx in advance.
rich
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