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Alexander Chemeris alexander.chemeris at gmail.comLeonardo, Look into the Wireshark capture of SIP traffic - which codecs are offered and how does the codec negotiation goes. On Thu, Nov 7, 2013 at 3:46 AM, Leonardo Nve <lnve at s21sec.com> wrote: > 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 > -- > > La información contenida en este mail, así como los archivos adjuntos,es > CONFIDENCIAL. Grupo S21sec Gestión, S.A. garantiza la adopción de las > medidas necesarias para asegurar el tratamiento confidencial de los datos > de carácter personal. En el caso de que el destinatario del correo > no sea usted, le rogamos envíe una notificación al remitente y lo destruya > de forma inmediata. La lectura y/o manipulación de esta información en la > situación señalada anteriormente será considerada ilegal, permitiendo a la > empresa remitente realizar acciones legales de diferente envergadura. > > > > > -- Regards, Alexander Chemeris. CEO, Fairwaves LLC / ООО УмРадио http://fairwaves.ru