Got a strange problem. Trying to call from a sip client to a normal phone or exetension. This results always in a one way audio connenction.
I use the odbc database, and can't really find the problem. Can anybody help me in the right direction. There seems to be no errors at all.
[general]
context=public
allowguest=no
allowoverlap=no
udpbindaddr=0.0.0.0:15060
tcpenable=no
tcpbindaddr=0.0.0.0
transport=udp
srvlookup=yes
language=ja
externaddr=x.x.x.x
localnet=x.x.x.x/255.255.240.0
nat=force_rport,comedia
rtcachefriends=yes
disallow=all
allow=ulaw
allow=alaw
allow=gsm
/var/log/asterisk/messages
[Apr 12 10:44:36] VERBOSE[23055][C-00000001] netsock2.c: Using SIP RTP CoS mark 5
[Apr 12 10:44:36] VERBOSE[25771][C-00000001] pbx.c: Executing [52431824@context_tok:1] NoOp("SIP/inbound_1_1-00000003", "inbound") in new stack
[Apr 12 10:44:36] VERBOSE[25771][C-00000001] pbx.c: Executing [52431824@context_tok:2] Dial("SIP/inbound_1_1-00000003", "SIP/1_1_1_1/1_1_1_1&SIP/1_1_1_2/1_1_1_2&SIP/1_1_1_3/1_1_1_3&SIP/1_1_1_4/1_1_1_4") in new stack
[Apr 12 10:44:36] VERBOSE[25771][C-00000001] netsock2.c: Using SIP RTP CoS mark 5
[Apr 12 10:44:36] WARNING[25771][C-00000001] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[Apr 12 10:44:36] VERBOSE[25771][C-00000001] netsock2.c: Using SIP RTP CoS mark 5
[Apr 12 10:44:36] WARNING[25771][C-00000001] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[Apr 12 10:44:36] VERBOSE[25771][C-00000001] app_dial.c: Called SIP/1_1_1_1/1_1_1_1
[Apr 12 10:44:36] VERBOSE[25771][C-00000001] app_dial.c: Called SIP/1_1_1_3/1_1_1_3
[Apr 12 10:44:36] VERBOSE[25771][C-00000001] app_dial.c: SIP/1_1_1_1-00000004 is ringing
[Apr 12 10:44:36] VERBOSE[25771][C-00000001] app_dial.c: SIP/1_1_1_3-00000005 is ringing
[Apr 12 10:44:44] VERBOSE[25771][C-00000001] app_dial.c: SIP/1_1_1_3-00000005 answered SIP/inbound_1_1-00000003
[Apr 12 10:44:44] VERBOSE[25846][C-00000001] bridge_channel.c: Channel SIP/1_1_1_3-00000005 joined 'simple_bridge' basic-bridge <16f760ce-43f9-4f36-8aa3-865c4f2e8151>
[Apr 12 10:44:44] VERBOSE[25771][C-00000001] bridge_channel.c: Channel SIP/inbound_1_1-00000003 joined 'simple_bridge' basic-bridge <16f760ce-43f9-4f36-8aa3-865c4f2e8151>
[Apr 12 10:44:52] VERBOSE[25846][C-00000001] bridge_channel.c: Channel SIP/1_1_1_3-00000005 left 'simple_bridge' basic-bridge <16f760ce-43f9-4f36-8aa3-865c4f2e8151>
[Apr 12 10:44:52] VERBOSE[25771][C-00000001] bridge_channel.c: Channel SIP/inbound_1_1-00000003 left 'simple_bridge' basic-bridge <16f760ce-43f9-4f36-8aa3-865c4f2e8151>
[Apr 12 10:44:52] VERBOSE[25771][C-00000001] pbx.c: Spawn extension (context_tok, 52431824, 2) exited non-zero on 'SIP/inbound_1_1-00000003'
Have tried several things, and searched on the net, coudn't find the correct solution.
To help people in the future with the same problem.
In the odbc database we had the standard data on null.
Setting qualify to yes for the users in the database fixed the problem.