OPENBSC/LCR/Asterisk

historical

Hi, 

I have installed an infrastructure IPACCESS - OpenBSC - LCR (without misdn) - Asterisk

LCR si connected via SIP to Asterisk.

The problem is that i can call MT <-> softphone, soft and MT <-> MT BUT i don't hear anything in any side ( softphone <-> softphone works well). I think is a codec problem:

Configured TCH/F FR for MT and we tried different codecs (alaw,gsm,ulaw, etc etc).
Also I tried bridging GSM and SIP interfaces on LCR config and putting rtp-bridge.

On LCR debug I see this error:

000000 DEBUG (in port.cpp/new_state() line 267): PORT(SIP-0-out) new state PORT_STATE_OUT_ALERTING --> PORT_STATE_CONNECT
nua: nua_application_event: entering
000000 DEBUG (in sip.cpp/sip_callback() line 1775): Event 7 from stack received (handle=0x94906d8)
000000 DEBUG (in sip.cpp/sip_callback() line 1825): state change received
nua: nua_application_event: entering
000000 DEBUG (in sip.cpp/sip_callback() line 1775): Event 7 from stack received (handle=0x94906d8)
000000 DEBUG (in sip.cpp/sip_callback() line 1825): state change received
nua: nua_application_event: entering
000000 DEBUG (in sip.cpp/sip_callback() line 1775): Event 5 from stack received (handle=0x94906d8)
000000 DEBUG (in sip.cpp/sip_callback() line 1837): active received
000000 TRACE 06.11.13 20:10:06.434 EP(2): CONNECT  from CH(2)  connect id number= present='not available'
000000 TRACE 06.11.13 20:10:06.435 EP(1): CONNECT  to CH(1)  connect id number= present='not available'
000000 TRACE 06.11.13 20:10:06.435 EP(1): TONE  to CH(1)  off 
000000 TRACE 06.11.13 20:10:06.436 CH(1): MNCC_SETUP_RSP LCR<->BSC  connected type=0 plan=1 present=0 screen=3 number=
000000 DEBUG (in port.cpp/new_state() line 267): PORT(GSM-0-in) new state PORT_STATE_IN_ALERTING --> PORT_STATE_CONNECT_WAITING
000000 DEBUG (in port.cpp/message_epoint() line 617): PORT(GSM-0-in) setting tone '' dir ''
000000 DEBUG (in gsm.cpp/audio_send() line 487): no valid CMR yet.
000000 DEBUG (in gsm.cpp/audio_send() line 487): no valid CMR yet.
000000 DEBUG (in gsm.cpp/audio_send() line 487): no valid CMR yet.
000000 DEBUG (in gsm.cpp/audio_send() line 487): no valid CMR yet.
000000 DEBUG (in gsm.cpp/audio_send() line 487): no valid CMR yet.
000000 DEBUG (in gsm.cpp/audio_send() line 487): no valid CMR yet.
000000 TRACE 06.11.13 20:10:06.563 CH(1): MNCC_SETUP_COMPL_IND LCR<->BSC
000000 DEBUG (in port.cpp/new_state() line 267): PORT(GSM-0-in) new state PORT_STATE_CONNECT_WAITING --> PORT_STATE_CONNECT
000000 DEBUG (in gsm.cpp/audio_send() line 487): no valid CMR yet.
000000 DEBUG (in gsm.cpp/audio_send() line 487): no valid CMR yet.
000000 DEBUG (in gsm.cpp/audio_send() line 487): no valid CMR yet.
000000 DEBUG (in gsm.cpp/audio_send() line 487): no valid CMR yet.
000000 DEBUG (in gsm.cpp/audio_send() line 487): no valid CMR yet.
000000 DEBUG (in gsm.cpp/audio_send() line 487): no valid CMR yet.
<--...-->
<-- THIS ERROR REPEATED CONTINUOUSLY DURING THE CALL -->
<--...-->

Configurations:

OpenBSC trx 0 config:

trx 0
   rf_locked 0
   arfcn 636
   nominal power 23
   max_power_red 0
   rsl e1 tei 0
    timeslot 0
     phys_chan_config CCCH+SDCCH4
     hopping enabled 0
    timeslot 1
     phys_chan_config SDCCH8
     hopping enabled 0
    timeslot 2
     phys_chan_config TCH/F
     hopping enabled 0
    timeslot 3
     phys_chan_config TCH/F
     hopping enabled 0
    timeslot 4
     phys_chan_config TCH/F
     hopping enabled 0
    timeslot 5
     phys_chan_config TCH/F
     hopping enabled 0
    timeslot 6
     phys_chan_config TCH/F
     hopping enabled 0
    timeslot 7
     phys_chan_config TCH/F
     hopping enabled 0
mncc-int
 default-codec tch-f fr

LCR config:

interfaces.conf

[GSM]
gsm-bs
tones yes
earlyb no

[SIP]
extern
sip localhost:5059 localhost:5060
tones yes
earlyb yes

Asterisk User conf:

user.conf (one user)

[6001]
fullname = SIPPhone2
registersip = no
host = dynamic
callgroup = 1
mailbox = 6001
call-limit = 100
type = peer
username = 6001
secret = nomypasshere
transfer = yes
nat = yes
context = openBSC_Integration
dtmfmode = rfc2833
cid_number = 6001
disallow = all
allow = alaw,gsm ; I Tried different codecs

callcounter = no
hasvoicemail = no
vmsecret = 
email = 
threewaycalling = no
hasdirectory = no
callwaiting = no
hasmanager = no
hasagent = no
hassip = yes
hasiax = no
canreinvite = no
insecure = no
pickupgroup = 1
autoprov = yes
label = 6001
linenumber = 1
LINEKEYS = 1

Other configs seem irrelevant...

-- 
--
Leonardo Nve <lnve at s21sec.com>
Project Manager ACSS
Grupo S21sec Gestión, S.A.
Telefono 628275870
--

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