Problem with BSC<>LCR(SIP)<>Asterisk

Ulrich Meckel <[email protected]>
Newsgroups gmane.linux.isdn.i4l.user
Organization Netzing Solutions AG
Message-ID <[email protected]> 
Hi List
here my Situation:


MS A : 47809
MS B : 23906

MS A called MS B with number 2 23906
asterisk/extensions.cfg:
exten => _2X.,1,Dial(SIP/${EXTEN}@192.168.8.160)

LCR LOG:

30.03.12 16:26:08.870 CH(1): New call ref LCR<->BSC  callref new=0x8000000b
30.03.12 16:26:08.870 CH(1): Codec negotiation LCR<->BSC  bearer 
capa='given by MS'  speech version='AMR given' version='EFR given' 
version='Full Rate given' version='5 given' ignored='Not supported by 
LCR' version='Half Rate given'
30.03.12 16:26:08.870 CH(1): MNCC_SETUP_IND LCR<->BSC  calling 
number=47809 imsi=262032310454035  dialing number=223906
30.03.12 16:26:08.870 CH(1): MNCC_CALL_PROC_REQ LCR<->BSC  progress 
coding=3 location=1 descr=8
30.03.12 16:26:08.870 CH(1): unknown LCR<->BSC
30.03.12 16:26:08.880 CH(2): NEW handle  handle new=0x96a30c8
30.03.12 16:26:08.881 CH(2): INVITE  from uri=sip:[email protected]  
to uri=sip:[email protected]:5061  rtp ip=192.168.8.160 
port=30036,30037 payload=AMR:96 payload=GSM-EFR:97 payload=GSM:3 
payload=GSM-HR:98
30.03.12 16:26:08.891 CH(3): Payload received  rtp payload=GSM:3 
payload=PCMA:8 payload=PCMU:0 payload=telephone-event:101
30.03.12 16:26:08.892 CH(3): NEW handle  handle new=0x96abde8
30.03.12 16:26:08.892 CH(3): INVITE  rtp port=18178  calling 
present=allowed number=47809  dialing number=23906  bearer capa=speech 
mode=circuit
30.03.12 16:26:08.902 CH(4): Codec negotiation LCR<->BSC  rtp 
payload='GSM:3 supported' payload='PCMA:8 unsupported' payload='PCMU:0 
unsupported'
30.03.12 16:26:08.902 CH(4): New call ref LCR<->BSC  callref new=0x1
30.03.12 16:26:08.903 CH(4): MNCC_SETUP_REQ LCR<->BSC  calling type=0 
plan=1 present=0 screen=3 number=47809  dialing number=23906
30.03.12 16:26:11.878 CH(4): MNCC_CALL_CONF_IND LCR<->BSC
30.03.12 16:26:11.878 CH(4): Codec negotiation LCR<->BSC  bearer 
capa='given by MS'  speech version='AMR given' version='EFR given' 
version='Full Rate given' version='5 given' ignored='Not supported by 
LCR' version='Half Rate given'
30.03.12 16:26:11.878 CH(4): MNCC_LCHAN_MODIFY LCR<->BSC  speech 
version='Full Rate given'  mode 0x01
30.03.12 16:26:12.151 CH(4): MNCC_ALERT_IND LCR<->BSC
30.03.12 16:26:12.152 CH(3): RESPOND  respond value='180 Ringing'
30.03.12 16:26:12.162 CH(2): RESPOND  respond value=180
30.03.12 16:26:12.172 CH(1): MNCC_ALERT_REQ LCR<->BSC  progress coding=3 
location=1 descr=8


At this Point MS B is ringing ( all fine )
Now MS B accept the call:


30.03.12 16:26:35.079 CH(4): MNCC_SETUP_CNF LCR<->BSC  connect 
number=23906 imsi=238016110003824
30.03.12 16:26:35.079 CH(4): MNCC_SETUP_COMPL_REQ LCR<->BSC
30.03.12 16:26:35.079 CH(4): unknown LCR<->BSC
30.03.12 16:26:35.100 CH(3): RESPOND  respond value='200 OK'  reason 
'call connected'  rtp ip=192.168.8.20 port=30040,30041 payload=GSM:3
30.03.12 16:26:35.110 CH(2): RESPOND  respond value=200
30.03.12 16:26:35.111 CH(2): Payload received  rtp payload=GSM:3
30.03.12 16:26:35.111 CH(5): Payload received  rtp payload=GSM:3
30.03.12 16:26:35.111 CH(5): NEW handle  handle new=0x96abde8
30.03.12 16:26:35.112 CH(5): INVITE  rtp port=30036  calling 
present=allowed number=47809  dialing number=23906  bearer capa=speech 
mode=circuit
30.03.12 16:26:35.122 CH(1): MNCC_SETUP_RSP LCR<->BSC  connected type=0 
plan=1 present=0 screen=3 number=
30.03.12 16:26:35.123 CH(1): MNCC_LCHAN_MODIFY LCR<->BSC  speech 
version='Full Rate given'  mode 0x01
30.03.12 16:26:35.123 CH(6): Codec negotiation LCR<->BSC  rtp 
payload='GSM:3 supported'
30.03.12 16:26:35.123 CH(6): New call ref LCR<->BSC  callref new=0x2
30.03.12 16:26:35.123 CH(6): MNCC_SETUP_REQ LCR<->BSC  calling type=0 
plan=1 present=0 screen=3 number=47809  dialing number=23906
30.03.12 16:26:35.123 CH(7): Payload received  rtp payload=GSM:3
30.03.12 16:26:35.124 CH(7): NEW handle  handle new=0x96a30c8
30.03.12 16:26:35.124 CH(7): INVITE  rtp port=30040  calling 
present=allowed number=223906  dialing number=47809  bearer capa=speech 
mode=circuit
30.03.12 16:26:35.134 CH(8): Codec negotiation LCR<->BSC  rtp 
payload='GSM:3 supported'
30.03.12 16:26:35.134 CH(8): New call ref LCR<->BSC  callref new=0x3
30.03.12 16:26:35.135 CH(8): MNCC_SETUP_REQ LCR<->BSC  calling type=0 
plan=1 present=0 screen=3 number=223906  dialing number=47809
30.03.12 16:26:35.236 CH(1): MNCC_SETUP_COMPL_IND LCR<->BSC
30.03.12 16:26:35.357 CH(6): MNCC_CALL_CONF_IND LCR<->BSC  cause 
coding=3 location= value=
30.03.12 16:26:35.357 CH(6): Codec negotiation LCR<->BSC  bearer 
capa='given by MS'  speech version='AMR given' version='EFR given' 
version='Full Rate given' version='5 given' ignored='Not supported by 
LCR' version='Half Rate given'
30.03.12 16:26:35.358 CH(6): MNCC_LCHAN_MODIFY LCR<->BSC  speech 
version='Full Rate given'  mode 0x01
30.03.12 16:26:35.459 CH(8): MNCC_CALL_CONF_IND LCR<->BSC  cause 
coding=3 location= value=
30.03.12 16:26:35.459 CH(8): Codec negotiation LCR<->BSC  bearer 
capa='given by MS'  speech version='AMR given' version='EFR given' 
version='Full Rate given' version='5 given' ignored='Not supported by 
LCR' version='Half Rate given'
30.03.12 16:26:35.459 CH(8): MNCC_LCHAN_MODIFY LCR<->BSC  speech 
version='Full Rate given'  mode 0x01
30.03.12 16:26:35.460 CH(6): MNCC_ALERT_IND LCR<->BSC
30.03.12 16:26:35.460 CH(5): RESPOND  respond value='180 Ringing'
30.03.12 16:26:35.571 CH(8): MNCC_ALERT_IND LCR<->BSC
30.03.12 16:26:35.571 CH(7): RESPOND  respond value='180 Ringing'


Now I think the call is established and some piece of voice will be 
transmitted but also now B try to call A parallel!???
( Kann mich auf english schwer ausdrücken : Man kann nun bei MS B , den 
Anrufer von MS A "hören" aber der Audiokanal ist wie zerhäckselt. Zudem 
versucht auf einmal während des Gesprächs B nun A zurückanzurufen... ?)

I tested also to set up a Call from a Softphone to an MS and this works 
fine.

Did anybody know thats the problem here?

Here my Configurations:

LCR Interface.cfg
[GSM]
gsm-bs
tones yes
earlyb no
rtp-bridge
bridge SIP

[SIP]
sip 192.168.8.160 192.168.8.20:5061
earlyb yes
tones no
rtp-bridge
bridge GSM


Asterisk CLI Output
     -- Executing [223906@default:1] Dial("SIP/192.168.8.160-00000000", 
"SIP/[email protected]") in new stack
   == Using SIP RTP CoS mark 5
     -- Called SIP/[email protected]
     -- SIP/192.168.8.160-00000001 is ringing
     -- SIP/192.168.8.160-00000001 answered SIP/192.168.8.160-00000000
     -- Remotely bridging SIP/192.168.8.160-00000000 and 
SIP/192.168.8.160-00000001
     -- Remotely bridging SIP/192.168.8.160-00000000 and 
SIP/192.168.8.160-00000001
     -- Remotely bridging SIP/192.168.8.160-00000000 and 
SIP/192.168.8.160-00000001



Regards,

Ulrich

-- 
_______________________________________________________________________
Ulrich Meckel
Entwicklung                               Mail:  [email protected]


NETZING Solutions AG                      Tel.:  0351/41381 -  0
Fröbelstr. 57, 01159 Dresden              Fax:   0351/41381 - 12
_______________________________________________________________________

Impressum:
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Sitz der Gesellschaft Amtsgericht Dresden HRB 18926
Vorstand  Dieter Schneider  -  Aufsichtsratsvorsitzender Volker Kanitz
USt.Id DE211326547  Mail:  [email protected]
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