I googled and search on many VoIP site about that, but find anything yet.
As I state in my subject, here is a more complete description of the problem: I’m trying to transfer a call that I initiate using the transfer button on my phone, not with the Asterisk transfer function because Asterisk is not on the media path. Then I got those line in my verbose console :
== Starting SIP/2018-ed79 at international,2015,0 failed so falling back to exten ‘s’
== Starting SIP/2018-ed79 at international,s,0 still failed so falling back to context 'default’
Jul 14 12:03:04 WARNING: pbx.c:1937 ast_pbx_run: Channel ‘SIP/2018-ed79’ sent into invalid extension ‘s’ in context ‘default’, but no invalid handler
== Spawn extension (macro-sipext, s, 1) exited non-zero on ‘SIP/2010-250a’ in macro ‘sipext’
== Spawn extension (international, 2018, 1) exited non-zero on ‘SIP/2010-250a’
Now when I do that with a received call, everything works fine, I’m able to transfer to another sip/zap/iax.
I tried that with different phones, many model of Grandstream and Zultys they all do the same, so I suppose the problem is not comming from the phone but from Asterisk but probably more exactly from my dialplan.
The way I understand the log from Asterisk, my transfer try to find it’s way thru the default context, after in the Macro used to place my call then in the context of my sip account.
But I can’t figure out what should I do, how I should program my dialplan to make that work. Anybody ever experience this problem ?
If anyone have a clue or a place where I should look, it will be really appreciate.