Has anyone experienced problems with signalling using LCR/Asterisk?
In my infrastructure everything goes ok calling from MT to external phones using Asterisk, but when the external one (SIP or extension ended with Hangup() ) the call remains activated at the MT (OpenBSC side). I will make a SIP capture tomorrow, and send it.
Here I send RTP & SIP capture, everything is ok there, and a LCR log. The strange thing at the log i saw is a problem setting Ext port in a Release state. I also send interface.conf.
LCR log: 000000 TRACE 12.11.13 01:40:54.228 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 TRACE 12.11.13 01:40:54.359 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 port.cpp/new_state() line 267): PORT(Ext-0-out) new state PORT_STATE_CONNECT --> PORT_STATE_RELEASE 000000 DEBUG (in port.cpp/new_state() line 267): PORT(Ext-0-out) new state PORT_STATE_RELEASE --> PORT_STATE_RELEASE 000000 DEBUG (in remote.cpp/~Premote() line 46): Destroyed Remote process(Ext-0-out). 000000 DEBUG (in port.cpp/~Port() line 215): removing port (2) of type 0x4002, name 'Ext-0-out' interface 'Ext' 000000 DEBUG (in port.cpp/~Port() line 218): Removing us from bridge 1 000000 TRACE 12.11.13 01:40:59.550 EP(2): RELEASE from CH(2) cause value=16 location=1-Local-PBX 000000 TRACE 12.11.13 01:40:59.551 EP(1): TONE to CH(1) directory default name cause_10 000000 TRACE 12.11.13 01:40:59.551 EP(1): DISCONNECT to CH(1) cause value=16 location=1-Local-PBX 000000 DEBUG (in port.cpp/message_epoint() line 617): PORT(GSM-0-in) setting tone 'cause_10' dir '' 000000 DEBUG (in port.cpp/set_tone() line 367): PORT(GSM-0-in) Given Cause 0x10 has no tone, using release tone 000000 TRACE 12.11.13 01:40:59.551 CH(1): MNCC_DISC_REQ LCR<->BSC progress coding=3 location=1 descr=8 cause coding=3 location=1 value=16 000000 DEBUG (in port.cpp/new_state() line 267): PORT(GSM-0-in) new state PORT_STATE_CONNECT --> PORT_STATE_OUT_DISCONNECT 000000 DEBUG (in port.cpp/read_audio() line 497): PORT(GSM-0-in) no tone: /opt/lcr/share/lcr/tones_american/release 000000 DEBUG (in port.cpp/read_audio() line 600): PORT(GSM-0-in) opening tone: /opt/lcr/share/lcr/tones_american/release_loop 000000 DEBUG (in port.cpp/read_audio() line 600): PORT(GSM-0-in) opening tone: /opt/lcr/share/lcr/tones_american/release_loop 000000 DEBUG (in port.cpp/read_audio() line 600): PORT(GSM-0-in) opening tone: /opt/lcr/share/lcr/tones_american/release_loop 000000 DEBUG (in port.cpp/read_audio() line 600): PORT(GSM-0-in) opening tone: /opt/lcr/share/lcr/tones_american/release_loop 000000 DEBUG (in port.cpp/read_audio() line 600): PORT(GSM-0-in) opening tone: /opt/lcr/share/lcr/tones_american/release_loop 000000 DEBUG (in port.cpp/read_audio() line 600): PORT(GSM-0-in) opening tone: /opt/lcr/share/lcr/tones_american/release_loop 000000 DEBUG (in port.cpp/read_audio() line 600): PORT(GSM-0-in) opening tone: /opt/lcr/share/lcr/tones_american/release_loop 000000 DEBUG (in port.cpp/read_audio() line 600): PORT(GSM-0-in) opening tone: /opt/lcr/share/lcr/tones_american/release_loop
Interface.conf # interface.conf ################
[GSM] gsm-bs #hr tones yes earlyb no
# Use chan_lcr (Asterisk PBX interface) as external interface. [Ext] remote asterisk context from-lcr extern earlyb yes tones no
El 11/11/2013, a las 23:58, Leonardo Nve escribió:
Has anyone experienced problems with signalling using LCR/Asterisk?
In my infrastructure everything goes ok calling from MT to external phones using Asterisk, but when the external one (SIP or extension ended with Hangup() ) the call remains activated at the MT (OpenBSC side). I will make a SIP capture tomorrow, and send it.
--
Leonardo Nve lnve@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.
Finally I send the solution myself: At interface.conf GSM section: tones no
<Leo> Thanks Leo! <Leo> No problem Leo! Enjoy!
El 12/11/2013, a las 11:25, Leonardo Nve escribió:
Here I send RTP & SIP capture, everything is ok there, and a LCR log. The strange thing at the log i saw is a problem setting Ext port in a Release state. I also send interface.conf.
LCR log: 000000 TRACE 12.11.13 01:40:54.228 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 TRACE 12.11.13 01:40:54.359 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 port.cpp/new_state() line 267): PORT(Ext-0-out) new state PORT_STATE_CONNECT --> PORT_STATE_RELEASE 000000 DEBUG (in port.cpp/new_state() line 267): PORT(Ext-0-out) new state PORT_STATE_RELEASE --> PORT_STATE_RELEASE 000000 DEBUG (in remote.cpp/~Premote() line 46): Destroyed Remote process(Ext-0-out). 000000 DEBUG (in port.cpp/~Port() line 215): removing port (2) of type 0x4002, name 'Ext-0-out' interface 'Ext' 000000 DEBUG (in port.cpp/~Port() line 218): Removing us from bridge 1 000000 TRACE 12.11.13 01:40:59.550 EP(2): RELEASE from CH(2) cause value=16 location=1-Local-PBX 000000 TRACE 12.11.13 01:40:59.551 EP(1): TONE to CH(1) directory default name cause_10 000000 TRACE 12.11.13 01:40:59.551 EP(1): DISCONNECT to CH(1) cause value=16 location=1-Local-PBX 000000 DEBUG (in port.cpp/message_epoint() line 617): PORT(GSM-0-in) setting tone 'cause_10' dir '' 000000 DEBUG (in port.cpp/set_tone() line 367): PORT(GSM-0-in) Given Cause 0x10 has no tone, using release tone 000000 TRACE 12.11.13 01:40:59.551 CH(1): MNCC_DISC_REQ LCR<->BSC progress coding=3 location=1 descr=8 cause coding=3 location=1 value=16 000000 DEBUG (in port.cpp/new_state() line 267): PORT(GSM-0-in) new state PORT_STATE_CONNECT --> PORT_STATE_OUT_DISCONNECT 000000 DEBUG (in port.cpp/read_audio() line 497): PORT(GSM-0-in) no tone: /opt/lcr/share/lcr/tones_american/release 000000 DEBUG (in port.cpp/read_audio() line 600): PORT(GSM-0-in) opening tone: /opt/lcr/share/lcr/tones_american/release_loop 000000 DEBUG (in port.cpp/read_audio() line 600): PORT(GSM-0-in) opening tone: /opt/lcr/share/lcr/tones_american/release_loop 000000 DEBUG (in port.cpp/read_audio() line 600): PORT(GSM-0-in) opening tone: /opt/lcr/share/lcr/tones_american/release_loop 000000 DEBUG (in port.cpp/read_audio() line 600): PORT(GSM-0-in) opening tone: /opt/lcr/share/lcr/tones_american/release_loop 000000 DEBUG (in port.cpp/read_audio() line 600): PORT(GSM-0-in) opening tone: /opt/lcr/share/lcr/tones_american/release_loop 000000 DEBUG (in port.cpp/read_audio() line 600): PORT(GSM-0-in) opening tone: /opt/lcr/share/lcr/tones_american/release_loop 000000 DEBUG (in port.cpp/read_audio() line 600): PORT(GSM-0-in) opening tone: /opt/lcr/share/lcr/tones_american/release_loop 000000 DEBUG (in port.cpp/read_audio() line 600): PORT(GSM-0-in) opening tone: /opt/lcr/share/lcr/tones_american/release_loop
Interface.conf # interface.conf ################
[GSM] gsm-bs #hr tones yes earlyb no
# Use chan_lcr (Asterisk PBX interface) as external interface. [Ext] remote asterisk context from-lcr extern earlyb yes tones no
<sipRTP.cap>
El 11/11/2013, a las 23:58, Leonardo Nve escribió:
Has anyone experienced problems with signalling using LCR/Asterisk?
In my infrastructure everything goes ok calling from MT to external phones using Asterisk, but when the external one (SIP or extension ended with Hangup() ) the call remains activated at the MT (OpenBSC side). I will make a SIP capture tomorrow, and send it.
--
Leonardo Nve lnve@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.
--
Leonardo Nve lnve@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.
Because the communication is better with chan_lcr than using SIP, I also see less CPU usage.
Are there any special reason to use SIP? (apart integration with other providers)
-- Leonardo Nve lnve@s21sec.com Project Manager departamento Auditoria 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.
El 12/11/2013, a las 16:56, Peter Stuge peter@stuge.se escribió:
Leonardo Nve wrote:
# Use chan_lcr (Asterisk PBX interface) as external interface. [Ext] remote asterisk
Why not use sip instead of chan_lcr?
//Peter
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Leonardo Nve -
Peter Stuge