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...