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
--
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.
10 Nov
10 Nov
7:59 p.m.
Leonardo,
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(a)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(a)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
10:20 p.m.
Thanks Alexander, I reinstalled all from scratch and now it's working with chan_lcr
and sip connection between LCR and Asterisk. The error was that the opencore-amr
libraries were installed before LCR make.
Leonardo.
El 10/11/2013, a las 19:59, Alexander Chemeris escribió:
Leonardo,
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(a)s21sec.com> wrote:
--
--
Leonardo Nve <lnve(a)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.
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
--
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
3841
days inactive
3845
days old
2 comments
2 participants
participants (2)
-
Alexander Chemeris -
Leonardo Nve