7 Nov
2013
7 Nov
'13
12:46 a.m.
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(a)s21sec.com>
Project Manager ACSS
Grupo S21sec Gestión, S.A.
Telefono 628275870
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