<html><head></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">Hi, <div><br></div><div>I have installed an infrastructure IPACCESS - OpenBSC - LCR (without misdn) - Asterisk</div><div><br></div><div>LCR si connected via SIP to Asterisk.</div><div><br></div><div><br></div><div>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:</div><div><br></div><div>Configured TCH/F FR for MT and we tried different codecs (alaw,gsm,ulaw, etc etc).</div><div>Also I tried bridging GSM and SIP interfaces on LCR config and putting rtp-bridge.</div><div><br></div><div>On LCR debug I see this error:</div><div><br></div><div><div>000000 DEBUG (in port.cpp/new_state() line 267): PORT(SIP-0-out) new state PORT_STATE_OUT_ALERTING --> PORT_STATE_CONNECT</div><div>nua: nua_application_event: entering</div><div>000000 DEBUG (in sip.cpp/sip_callback() line 1775): Event 7 from stack received (handle=0x94906d8)</div><div>000000 DEBUG (in sip.cpp/sip_callback() line 1825): state change received</div><div>nua: nua_application_event: entering</div><div>000000 DEBUG (in sip.cpp/sip_callback() line 1775): Event 7 from stack received (handle=0x94906d8)</div><div>000000 DEBUG (in sip.cpp/sip_callback() line 1825): state change received</div><div>nua: nua_application_event: entering</div><div>000000 DEBUG (in sip.cpp/sip_callback() line 1775): Event 5 from stack received (handle=0x94906d8)</div><div>000000 DEBUG (in sip.cpp/sip_callback() line 1837): active received</div><div>000000 TRACE 06.11.13 20:10:06.434 EP(2): CONNECT from CH(2) connect id number= present='not available'</div><div>000000 TRACE 06.11.13 20:10:06.435 EP(1): CONNECT to CH(1) connect id number= present='not available'</div><div>000000 TRACE 06.11.13 20:10:06.435 EP(1): TONE to CH(1) off </div><div>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=</div><div>000000 DEBUG (in port.cpp/new_state() line 267): PORT(GSM-0-in) new state PORT_STATE_IN_ALERTING --> PORT_STATE_CONNECT_WAITING</div><div>000000 DEBUG (in port.cpp/message_epoint() line 617): PORT(GSM-0-in) setting tone '' dir ''</div><div>000000 DEBUG (in gsm.cpp/audio_send() line 487): no valid CMR yet.</div><div>000000 DEBUG (in gsm.cpp/audio_send() line 487): no valid CMR yet.</div><div>000000 DEBUG (in gsm.cpp/audio_send() line 487): no valid CMR yet.</div><div>000000 DEBUG (in gsm.cpp/audio_send() line 487): no valid CMR yet.</div><div>000000 DEBUG (in gsm.cpp/audio_send() line 487): no valid CMR yet.</div><div>000000 DEBUG (in gsm.cpp/audio_send() line 487): no valid CMR yet.</div><div>000000 TRACE 06.11.13 20:10:06.563 CH(1): MNCC_SETUP_COMPL_IND LCR<->BSC</div><div>000000 DEBUG (in port.cpp/new_state() line 267): PORT(GSM-0-in) new state PORT_STATE_CONNECT_WAITING --> PORT_STATE_CONNECT</div><div>000000 DEBUG (in gsm.cpp/audio_send() line 487): no valid CMR yet.</div><div>000000 DEBUG (in gsm.cpp/audio_send() line 487): no valid CMR yet.</div><div>000000 DEBUG (in gsm.cpp/audio_send() line 487): no valid CMR yet.</div><div>000000 DEBUG (in gsm.cpp/audio_send() line 487): no valid CMR yet.</div><div>000000 DEBUG (in gsm.cpp/audio_send() line 487): no valid CMR yet.</div><div>000000 DEBUG (in gsm.cpp/audio_send() line 487): no valid CMR yet.</div></div><div><--...--></div><div><-- THIS ERROR REPEATED CONTINUOUSLY DURING THE CALL --></div><div><div><--...--></div></div><div><br></div><div>Configurations:</div><div><br></div><div>OpenBSC trx 0 config:</div><div><br></div><div><div>trx 0</div><div> rf_locked 0</div><div> arfcn 636</div><div> nominal power 23</div><div> max_power_red 0</div><div> rsl e1 tei 0</div><div> timeslot 0</div><div> phys_chan_config CCCH+SDCCH4</div><div> hopping enabled 0</div><div> timeslot 1</div><div> phys_chan_config SDCCH8</div><div> hopping enabled 0</div><div> timeslot 2</div><div> phys_chan_config TCH/F</div><div> hopping enabled 0</div><div> timeslot 3</div><div> phys_chan_config TCH/F</div><div> hopping enabled 0</div><div> timeslot 4</div><div> phys_chan_config TCH/F</div><div> hopping enabled 0</div><div> timeslot 5</div><div> phys_chan_config TCH/F</div><div> hopping enabled 0</div><div> timeslot 6</div><div> phys_chan_config TCH/F</div><div> hopping enabled 0</div><div> timeslot 7</div><div> phys_chan_config TCH/F</div><div> hopping enabled 0</div><div>mncc-int</div><div> default-codec tch-f fr</div><div><br></div><div>LCR config:</div><div><br></div><div><br></div><div>interfaces.conf</div><div><br></div><div><div>[GSM]</div><div>gsm-bs</div><div>tones yes</div><div>earlyb no</div><div><br></div><div>[SIP]</div><div>extern</div><div>sip localhost:5059 localhost:5060</div><div>tones yes</div><div>earlyb yes</div></div><div><br></div><div>Asterisk User conf:</div><div><br></div><div>user.conf (one user)</div><div><br></div><div><div>[6001]</div><div>fullname = SIPPhone2</div><div>registersip = no</div><div>host = dynamic</div><div>callgroup = 1</div><div>mailbox = 6001</div><div>call-limit = 100</div><div>type = peer</div><div>username = 6001</div><div>secret = nomypasshere</div><div>transfer = yes</div><div>nat = yes</div><div>context = openBSC_Integration</div><div>dtmfmode = rfc2833</div><div>cid_number = 6001</div><div><div>disallow = all</div><div>allow = alaw,gsm ; I Tried different codecs</div></div><div><br></div><div>callcounter = no</div><div>hasvoicemail = no</div><div>vmsecret = </div><div>email = </div><div>threewaycalling = no</div><div>hasdirectory = no</div><div>callwaiting = no</div><div>hasmanager = no</div><div>hasagent = no</div><div>hassip = yes</div><div>hasiax = no</div><div>canreinvite = no</div><div>insecure = no</div><div>pickupgroup = 1</div><div>autoprov = yes</div><div>label = 6001</div><div>linenumber = 1</div><div>LINEKEYS = 1</div></div><div><br></div><div><br></div><div>Other configs seem irrelevant...</div><div><br></div><div><br></div><div><br></div><div>
<div style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; "><div><div><div style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; "><div>-- <br>--<br>Leonardo Nve <<a href="mailto:lnve@s21sec.com">lnve@s21sec.com</a>><br>Project Manager ACSS<br>Grupo S21sec Gestión, S.A.<br>Telefono 628275870</div><div>--<br><br>La información contenida en este mail, así como los archivos adjuntos,es<br>CONFIDENCIAL. Grupo S21sec Gestión, S.A. garantiza la adopción de las<br>medidas necesarias para asegurar el tratamiento confidencial de los datos<br>de carácter personal. En el caso de que el destinatario del correo<br>no sea usted, le rogamos envíe una notificación al remitente y lo destruya<br>de forma inmediata. La lectura y/o manipulación de esta información en la<br>situación señalada anteriormente será considerada ilegal, permitiendo a la<br>empresa remitente realizar acciones legales de diferente envergadura.</div><div><br></div></div></div></div><div><br></div></div><br class="Apple-interchange-newline"><br class="Apple-interchange-newline">
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