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 [h@macro-dial-one:1] Macro("SIP/OBITRUNK1-00000013", "hangupcall,") in new stack
pbx.c: -- Executing [s@macro-hangupcall:1] GotoIf("SIP/OBITRUNK1-00000013", "1?theend") in new stack
pbx.c: -- Goto (macro-hangupcall,s,3)
pbx.c: -- Executing [s@macro-hangupcall:3] ExecIf("SIP/OBITRUNK1-00000013", "0?Set(CDR(recordingfile)=)") in new stack
pbx.c: -- Executing [s@macro-hangupcall: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 [s@macro-setmusic:1] Set("SIP/OBITRUNK1-00000018", "CHANNEL(musicclass)=none") in new stack
pbx.c: -- Executing [h@macro-dialout-trunk:1] Macro("SIP/6-00000017", "hangupcall,") in new stack
pbx.c: -- Executing [s@macro-hangupcall:1] GotoIf("SIP/6-00000017", "1?theend") in new stack
pbx.c: -- Goto (macro-hangupcall,s,3)
pbx.c: -- Executing [s@macro-hangupcall:3] ExecIf("SIP/6-00000017", "0?Set(CDR(recordingfile)=)") in new stack
pbx.c: -- Executing [s@macro-hangupcall: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