Please help: one-way on receive and doesn't dial out

Hi,

I am trying to make SIP softphone call to PTSN through FreePBX/Asterisk 11 with an Obihai 110’s LINE port set as its FXO.

I followed the instructions from the links below and carefully read the threads; after closely and repeatedly examined each setting on both the device and the pbx.

I am stuck.

http://wiki.freepbx.org/pages/viewpage.action?pageId=4161594
http://www.freepbx.org/support/documentation/howtos/howto-use-an-obi-110-device-to-provide-to-allow-freepbx-to-make-calls-o
http://www.obitalk.com/forum/index.php?topic=1157.msg7261#msg7261
http://tech.iprock.com/?p=3208
http://www.obitalk.com/forum/index.php?topic=57.0

here is the wiring

                lan           --------------------------           |             |           | pstn --- obi 110    asterisk    softphone
(not connected to the internet, no router, no gv, no obitalk, just simple local lan to pstn)

here are the two scenarios(along with obi call history and asterisk logs)

#1 incoming call from the landline rings the softphone,
picks up the caller id. however, when i answer the softphone,
the other party hears me, but i do not hear them

Terminal ID	LINE1	            SP2
Peer Name	<their name>	
Peer Number	1yyyyyyyyyy	    xxxxxxxxxx
Direction          Inbound	            Outbound
21:02:30	        Ringing	
21:02:35		                            Call Connected
21:02:51	        End Call	

netsock2.c:   == Using SIP RTP TOS bits 184
netsock2.c:   == Using SIP RTP CoS mark 5
app_dial.c:     -- Called SIP/6
app_dial.c:     -- SIP/6-00000014 is ringing
app_dial.c:     -- SIP/6-00000014 answered SIP/OBITRUNK1-00000013
pbx.c:     -- Executing [[email protected]:1] Macro("SIP/OBITRUNK1-00000013", "hangupcall,") in new stack
pbx.c:     -- Executing [[email protected]:1] GotoIf("SIP/OBITRUNK1-00000013", "1?theend") in new stack
pbx.c:     -- Goto (macro-hangupcall,s,3)
pbx.c:     -- Executing [[email protected]:3] ExecIf("SIP/OBITRUNK1-00000013", "0?Set(CDR(recordingfile)=)") in new stack
pbx.c:     -- Executing [[email protected]:4] Hangup("SIP/OBITRUNK1-00000013", "") in new stack
app_macro.c:   == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/OBITRUNK1-00000013' in macro 'hangupcall'
pbx.c:   == Spawn extension (macro-dial-one, h, 1) exited non-zero on 'SIP/OBITRUNK1-00000013'
app_macro.c:   == Spawn extension (macro-dial-one, s, 42) exited non-zero on 'SIP/OBITRUNK1-00000013' in macro 'dial-one'
app_macro.c:   == Spawn extension (macro-exten-vm, s, 14) exited non-zero on 'SIP/OBITRUNK1-00000013' in macro 'exten-vm'
pbx.c:   == Spawn extension (from-did-direct, 6, 2) exited non-zero on 'SIP/OBITRUNK1-00000013'

#2 when i dial 81-1-yyy-yyy-yyyy on the softphone, i hear the
softphone rings once and then goes silent.  the call terminates by
itself after 30 seconds.  the other party says their phone didn’t ring.

Terminal ID	SP2	                LINE1
Peer Name		
Peer Number	xxxxxxxxxx	1yyyyyyyyyy
Direction	        Inbound	        Outbound
21:20:01	        Ringing	
21:20:06		                        Call Connected
21:20:22     	End Call

netsock2.c:   == Using SIP RTP TOS bits 184
netsock2.c:   == Using SIP RTP CoS mark 5
app_dial.c:     -- Called SIP/OBITRUNK1/811yyyyyyyyyy
app_dial.c:     -- SIP/OBITRUNK1-00000018 is ringing
app_dial.c:     -- SIP/OBITRUNK1-00000018 answered SIP/6-00000017
pbx.c:     -- Executing [[email protected]:1] Set("SIP/OBITRUNK1-00000018", "CHANNEL(musicclass)=none") in new stack
pbx.c:     -- Executing [[email protected]:1] Macro("SIP/6-00000017", "hangupcall,") in new stack
pbx.c:     -- Executing [[email protected]:1] GotoIf("SIP/6-00000017", "1?theend") in new stack
pbx.c:     -- Goto (macro-hangupcall,s,3)
pbx.c:     -- Executing [[email protected]:3] ExecIf("SIP/6-00000017", "0?Set(CDR(recordingfile)=)") in new stack
pbx.c:     -- Executing [[email protected]:4] Hangup("SIP/6-00000017", "") in new stack
app_macro.c:   == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/6-00000017' in macro 'hangupcall'
pbx.c:   == Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/6-00000017'
app_macro.c:   == Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/6-00000017' in macro 'dialout-trunk'
pbx.c:   == Spawn extension (from-internal, 811yyyyyyyyyy, 5) exited non-zero on 'SIP/6-00000017'

Does this sound familiar? What I should try next?

Jake

32bit freepbx distro 4.211.64-1375214241 with asterisk 11, vanilla setup
obihai 110, hardware 2.8, software 1.3.0 (Build: 2774), full factory reset

Could you enable SIP debug and run another test call?

ahhh! I see it:

SIP is advertising the RTP peer port on a different network interface! Now at least I hear the other side ringing. Let me rewire and debug a bit more.

Derek, thank you for the tip. Will report back soon.

Everything is working! Thank you Derek.

Also changing the DialDelay in the LINE Port on the Obi110 helped to resolve the 3 seconds of MWI stutter tone.