NEC Univerge SV8100 - Asterisk 1.8 Sip trunk

system · May 8, 2012, 1:54pm

Hello everyone !

I have a real BIG problem interconnecting this Phillips-NEC to Asterisk via SIP Trunk.
I have created the trunk with the follow settings:

host=192.168.1.11
type=peer
disallow=all
allow=alaw

Incoming settings left blank.

Asterisk box is 192.168.1.16 - Asterisk 1.8.7

Now a can call NEC extensions from Asterisk extensions, everything is ok. The problem appears when I dial an Asterisk ext. from NEC. For example dial from 104 (NEC) to 3801 (Aastra phone - Asterisk).
Ex. 1

3801 is ringing
answer but no voice

Ex. 2

3801 is ringing
hung up 104 but 3801 still ringing

I don’t know where the problem is but it drives me crazy, i must get the solution done (my boss is pressing me about that) and the NEC technical support here in Romania told me that Asterisk is not on their supported equipments list (after he tried to help me out).
Both are configured with g711a - alaw
In the log file at one moment I saw (Ignoring this invite request - I don’t now why…)
Below is the log file (sorry about size) with a call from 104 (nec) to 3801 (elastix) - the nec is calling with CID 100 (don’t know why)

INVITE sip:3801@192.168.1.16 SIP/2.0
From: “100”;tag=338C324631353641000B6B8E
To:
Contact:
Content-Type: application/sdp
Allow: INVITE,ACK,BYE,CANCEL,PRACK,UPDATE
Supported: 100rel,timer
Expires: 180
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Max-Forwards: 70
User-Agent: NEC-i SV8100-GE 06.01
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD
Content-Length: 220

v=0
o=- 0 0 IN IP4 192.168.1.11
s=T059
c=IN IP4 192.168.1.20
t=0 0
m=audio 10026 RTP/AVP 8 2 18 9
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=ptime:30
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/8000
a=ptime:30

[May 8 13:17:27] VERBOSE[10572] chan_sip.c: — (14 headers 12 lines) —
[May 8 13:17:27] VERBOSE[10572] chan_sip.c: Sending to 192.168.1.11:5060 (no NAT)
[May 8 13:17:27] VERBOSE[10572] chan_sip.c: Using INVITE request as basis request - 0201C1A90C81400000000010@192.168.1.11
[May 8 13:17:27] VERBOSE[10572] chan_sip.c: Found peer ‘nec-silf’ for ‘100’ from 192.168.1.11:5060
[May 8 13:17:27] VERBOSE[10572] netsock2.c: == Using SIP RTP TOS bits 184
[May 8 13:17:27] VERBOSE[10572] netsock2.c: == Using SIP RTP CoS mark 5
[May 8 13:17:27] VERBOSE[10572] chan_sip.c: Found RTP audio format 8
[May 8 13:17:27] VERBOSE[10572] chan_sip.c: Found RTP audio format 2
[May 8 13:17:27] VERBOSE[10572] chan_sip.c: Found RTP audio format 18
[May 8 13:17:27] VERBOSE[10572] chan_sip.c: Found RTP audio format 9
[May 8 13:17:27] VERBOSE[10572] chan_sip.c: Found audio description format PCMA for ID 8
[May 8 13:17:27] VERBOSE[10572] chan_sip.c: Found audio description format G726-32 for ID 2
[May 8 13:17:27] VERBOSE[10572] chan_sip.c: Found audio description format G729 for ID 18
[May 8 13:17:27] VERBOSE[10572] chan_sip.c: Found audio description format G722 for ID 9
[May 8 13:17:27] VERBOSE[10572] chan_sip.c: Capabilities: us - 0x8 (alaw), peer - audio=0x1908 (alaw|g726|g729|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
[May 8 13:17:27] VERBOSE[10572] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
[May 8 13:17:27] VERBOSE[10572] chan_sip.c: Peer audio RTP is at port 192.168.1.20:10026
[May 8 13:17:27] VERBOSE[10572] chan_sip.c: Looking for 3801 in from-trunk-sip-nec-silf (domain 192.168.1.16)
[May 8 13:17:27] VERBOSE[10572] chan_sip.c: list_route: hop:
[May 8 13:17:27] VERBOSE[10572] chan_sip.c:

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: “100”;tag=338C324631353641000B6B8E
To:
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Length: 0

[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [3801@from-trunk-sip-nec-silf:1] Set(“SIP/nec-silf-00000004”, “GROUP()=OUT_4”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [3801@from-trunk-sip-nec-silf:2] Goto(“SIP/nec-silf-00000004”, “from-trunk,3801,1”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Goto (from-trunk,3801,1)
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [3801@from-trunk:1] Macro(“SIP/nec-silf-00000004”, “exten-vm,novm,3801”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [s@macro-exten-vm:1] Macro(“SIP/nec-silf-00000004”, “user-callerid,”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [s@macro-user-callerid:1] Set(“SIP/nec-silf-00000004”, “AMPUSER=100”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [s@macro-user-callerid:2] GotoIf(“SIP/nec-silf-00000004”, “0?report”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [s@macro-user-callerid:3] ExecIf(“SIP/nec-silf-00000004”, “1?Set(REALCALLERIDNUM=100)”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [s@macro-user-callerid:4] Set(“SIP/nec-silf-00000004”, “AMPUSER=”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [s@macro-user-callerid:5] Set(“SIP/nec-silf-00000004”, “AMPUSERCIDNAME=”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [s@macro-user-callerid:6] GotoIf(“SIP/nec-silf-00000004”, “1?report”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Goto (macro-user-callerid,s,10)
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [s@macro-user-callerid:10] GotoIf(“SIP/nec-silf-00000004”, “0?continue”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [s@macro-user-callerid:11] Set(“SIP/nec-silf-00000004”, “__TTL=64”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [s@macro-user-callerid:12] GotoIf(“SIP/nec-silf-00000004”, “1?continue”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Goto (macro-user-callerid,s,19)
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [s@macro-user-callerid:19] Set(“SIP/nec-silf-00000004”, “CALLERID(number)=100”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [s@macro-user-callerid:20] Set(“SIP/nec-silf-00000004”, “CALLERID(name)=100”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [s@macro-user-callerid:21] NoOp(“SIP/nec-silf-00000004”, “Using CallerID “100” “) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [s@macro-exten-vm:2] Set(“SIP/nec-silf-00000004”, “RingGroupMethod=none”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [s@macro-exten-vm:3] Set(“SIP/nec-silf-00000004”, “VMBOX=novm”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [s@macro-exten-vm:4] Set(“SIP/nec-silf-00000004”, “__EXTTOCALL=3801”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [s@macro-exten-vm:5] Set(“SIP/nec-silf-00000004”, “CFUEXT=”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [s@macro-exten-vm:6] Set(“SIP/nec-silf-00000004”, “CFBEXT=”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [s@macro-exten-vm:7] Set(“SIP/nec-silf-00000004”, “RT=”””) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [s@macro-exten-vm:8] Macro(“SIP/nec-silf-00000004”, “record-enable,3801,IN”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [s@macro-record-enable:1] GotoIf(“SIP/nec-silf-00000004”, “1?check”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Goto (macro-record-enable,s,4)
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [s@macro-record-enable:4] ExecIf(“SIP/nec-silf-00000004”, “0?MacroExit()”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [s@macro-record-enable:5] GotoIf(“SIP/nec-silf-00000004”, “0?Group:OUT”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Goto (macro-record-enable,s,15)
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [s@macro-record-enable:15] GotoIf(“SIP/nec-silf-00000004”, “1?IN”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Goto (macro-record-enable,s,20)
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [s@macro-record-enable:20] ExecIf(“SIP/nec-silf-00000004”, “1?MacroExit()”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [s@macro-exten-vm:9] Macro(“SIP/nec-silf-00000004”, “dial-one,”",tr,3801") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [s@macro-dial-one:1] Set(“SIP/nec-silf-00000004”, “DEXTEN=3801”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [s@macro-dial-one:2] Set(“SIP/nec-silf-00000004”, “DIALSTATUS_CW=”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [s@macro-dial-one:3] GosubIf(“SIP/nec-silf-00000004”, “0?screen,1”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [s@macro-dial-one:4] GosubIf(“SIP/nec-silf-00000004”, “0?cf,1”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [s@macro-dial-one:5] GotoIf(“SIP/nec-silf-00000004”, “1?skip1”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Goto (macro-dial-one,s,
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [s@macro-dial-one:8] GotoIf(“SIP/nec-silf-00000004”, “0?nodial”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [s@macro-dial-one:9] GotoIf(“SIP/nec-silf-00000004”, “0?continue”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [s@macro-dial-one:10] Set(“SIP/nec-silf-00000004”, “EXTHASCW=”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [s@macro-dial-one:11] GotoIf(“SIP/nec-silf-00000004”, “1?next1:cwinusebusy”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Goto (macro-dial-one,s,12)
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [s@macro-dial-one:12] GotoIf(“SIP/nec-silf-00000004”, “0?docfu:skip3”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Goto (macro-dial-one,s,16)
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [s@macro-dial-one:16] GotoIf(“SIP/nec-silf-00000004”, “1?next2:continue”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Goto (macro-dial-one,s,17)
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [s@macro-dial-one:17] GotoIf(“SIP/nec-silf-00000004”, “1?continue”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Goto (macro-dial-one,s,25)
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [s@macro-dial-one:25] GotoIf(“SIP/nec-silf-00000004”, “0?nodial”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [s@macro-dial-one:26] GosubIf(“SIP/nec-silf-00000004”, “1?dstring,1:dlocal,1”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [dstring@macro-dial-one:1] Set(“SIP/nec-silf-00000004”, “DSTRING=”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [dstring@macro-dial-one:2] Set(“SIP/nec-silf-00000004”, “DEVICES=3801”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [dstring@macro-dial-one:3] ExecIf(“SIP/nec-silf-00000004”, “0?Return()”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [dstring@macro-dial-one:4] ExecIf(“SIP/nec-silf-00000004”, “0?Set(DEVICES=801)”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [dstring@macro-dial-one:5] Set(“SIP/nec-silf-00000004”, “LOOPCNT=1”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [dstring@macro-dial-one:6] Set(“SIP/nec-silf-00000004”, “ITER=1”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [dstring@macro-dial-one:7] Set(“SIP/nec-silf-00000004”, “THISDIAL=SIP/3801”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [dstring@macro-dial-one:8] GosubIf(“SIP/nec-silf-00000004”, “1?zap2dahdi,1”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [zap2dahdi@macro-dial-one:1] ExecIf(“SIP/nec-silf-00000004”, “0?Return()”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [zap2dahdi@macro-dial-one:2] Set(“SIP/nec-silf-00000004”, “NEWDIAL=”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [zap2dahdi@macro-dial-one:3] Set(“SIP/nec-silf-00000004”, “LOOPCNT2=1”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [zap2dahdi@macro-dial-one:4] Set(“SIP/nec-silf-00000004”, “ITER2=1”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [zap2dahdi@macro-dial-one:5] Set(“SIP/nec-silf-00000004”, “THISPART2=SIP/3801”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [zap2dahdi@macro-dial-one:6] ExecIf(“SIP/nec-silf-00000004”, “0?Set(THISPART2=DAHDI/3801)”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [zap2dahdi@macro-dial-one:7] Set(“SIP/nec-silf-00000004”, “NEWDIAL=SIP/3801&”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [zap2dahdi@macro-dial-one:8] Set(“SIP/nec-silf-00000004”, “ITER2=2”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [zap2dahdi@macro-dial-one:9] GotoIf(“SIP/nec-silf-00000004”, “0?begin2”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [zap2dahdi@macro-dial-one:10] Set(“SIP/nec-silf-00000004”, “THISDIAL=SIP/3801”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [zap2dahdi@macro-dial-one:11] Return(“SIP/nec-silf-00000004”, “”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [dstring@macro-dial-one:9] Set(“SIP/nec-silf-00000004”, “DSTRING=SIP/3801&”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [dstring@macro-dial-one:10] Set(“SIP/nec-silf-00000004”, “ITER=2”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [dstring@macro-dial-one:11] GotoIf(“SIP/nec-silf-00000004”, “0?begin”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [dstring@macro-dial-one:12] Set(“SIP/nec-silf-00000004”, “DSTRING=SIP/3801”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [dstring@macro-dial-one:13] Return(“SIP/nec-silf-00000004”, “”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [s@macro-dial-one:27] GotoIf(“SIP/nec-silf-00000004”, “0?nodial”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [s@macro-dial-one:28] GotoIf(“SIP/nec-silf-00000004”, “1?skiptrace”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Goto (macro-dial-one,s,30)
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [s@macro-dial-one:30] Set(“SIP/nec-silf-00000004”, “D_OPTIONS=tr”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [s@macro-dial-one:31] ExecIf(“SIP/nec-silf-00000004”, “0?SIPAddHeader(Alert-Info: )”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [s@macro-dial-one:32] ExecIf(“SIP/nec-silf-00000004”, “0?SIPAddHeader()”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [s@macro-dial-one:33] ExecIf(“SIP/nec-silf-00000004”, “0?Set(CHANNEL(musicclass)=)”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [s@macro-dial-one:34] GosubIf(“SIP/nec-silf-00000004”, “0?qwait,1”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [s@macro-dial-one:35] Set(“SIP/nec-silf-00000004”, “__CWIGNORE=”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [s@macro-dial-one:36] Set(“SIP/nec-silf-00000004”, “__KEEPCID=TRUE”) in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: – Executing [s@macro-dial-one:37] Dial(“SIP/nec-silf-00000004”, “SIP/3801,”",tr") in new stack
[May 8 13:17:27] VERBOSE[10685] netsock2.c: == Using SIP RTP TOS bits 184
[May 8 13:17:27] VERBOSE[10685] netsock2.c: == Using SIP RTP CoS mark 5
[May 8 13:17:27] VERBOSE[10685] app_dial.c: – Called SIP/3801
[May 8 13:17:27] VERBOSE[10685] chan_sip.c:

SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: “100”;tag=338C324631353641000B6B8E
To: ;tag=as33edd479
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Length: 0

[May 8 13:17:28] VERBOSE[10685] app_dial.c: – SIP/3801-00000005 is ringing
[May 8 13:17:28] VERBOSE[10685] chan_sip.c:

SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: “100”;tag=338C324631353641000B6B8E
To: ;tag=as33edd479
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Length: 0

[May 8 13:17:28] VERBOSE[10685] app_dial.c: – SIP/3801-00000005 is ringing
[May 8 13:17:28] VERBOSE[10572] chan_sip.c:

INVITE sip:3801@192.168.1.16 SIP/2.0
From: “100”;tag=338C324631353641000B6B8E
To:
Contact:
Content-Type: application/sdp
Allow: INVITE,ACK,BYE,CANCEL,PRACK,UPDATE
Supported: 100rel,timer
Expires: 180
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Max-Forwards: 70
User-Agent: NEC-i SV8100-GE 06.01
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD
Content-Length: 220

v=0
o=- 0 0 IN IP4 192.168.1.11
s=T059
c=IN IP4 192.168.1.20
t=0 0
m=audio 10026 RTP/AVP 8 2 18 9
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=ptime:30
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/8000
a=ptime:30

[May 8 13:17:28] VERBOSE[10572] chan_sip.c: — (14 headers 12 lines) —
[May 8 13:17:28] VERBOSE[10572] chan_sip.c: Ignoring this INVITE request
[May 8 13:17:28] VERBOSE[10572] chan_sip.c:

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: “100”;tag=338C324631353641000B6B8E
To:
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Length: 0

[May 8 13:17:29] VERBOSE[10572] chan_sip.c:

INVITE sip:3801@192.168.1.16 SIP/2.0
From: “100”;tag=338C324631353641000B6B8E
To:
Contact:
Content-Type: application/sdp
Allow: INVITE,ACK,BYE,CANCEL,PRACK,UPDATE
Supported: 100rel,timer
Expires: 180
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Max-Forwards: 70
User-Agent: NEC-i SV8100-GE 06.01
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD
Content-Length: 220

v=0
o=- 0 0 IN IP4 192.168.1.11
s=T059
c=IN IP4 192.168.1.20
t=0 0
m=audio 10026 RTP/AVP 8 2 18 9
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=ptime:30
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/8000
a=ptime:30

[May 8 13:17:29] VERBOSE[10572] chan_sip.c: — (14 headers 12 lines) —
[May 8 13:17:29] VERBOSE[10572] chan_sip.c: Ignoring this INVITE request
[May 8 13:17:29] VERBOSE[10572] chan_sip.c:

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: “100”;tag=338C324631353641000B6B8E
To:
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Length: 0

[May 8 13:17:30] VERBOSE[10685] app_dial.c: – SIP/3801-00000005 answered SIP/nec-silf-00000004
[May 8 13:17:30] VERBOSE[10685] chan_sip.c: Audio is at 5060
[May 8 13:17:30] VERBOSE[10685] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[May 8 13:17:30] VERBOSE[10685] chan_sip.c:

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: “100”;tag=338C324631353641000B6B8E
To: ;tag=as33edd479
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Type: application/sdp
Content-Length: 180

v=0
o=root 1201763838 1201763838 IN IP4 192.168.1.16
s=Asterisk PBX 1.8.7.0
c=IN IP4 192.168.1.16
t=0 0
m=audio 11408 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

[May 8 13:17:31] VERBOSE[10572] chan_sip.c: Retransmitting #1 (no NAT) to 192.168.1.11:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: “100”;tag=338C324631353641000B6B8E
To: ;tag=as33edd479
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Type: application/sdp
Content-Length: 180

v=0
o=root 1201763838 1201763838 IN IP4 192.168.1.16
s=Asterisk PBX 1.8.7.0
c=IN IP4 192.168.1.16
t=0 0
m=audio 11408 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

[May 8 13:17:31] VERBOSE[10572] chan_sip.c:

INVITE sip:3801@192.168.1.16 SIP/2.0
From: “100”;tag=338C324631353641000B6B8E
To:
Contact:
Content-Type: application/sdp
Allow: INVITE,ACK,BYE,CANCEL,PRACK,UPDATE
Supported: 100rel,timer
Expires: 180
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Max-Forwards: 70
User-Agent: NEC-i SV8100-GE 06.01
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD
Content-Length: 220

v=0
o=- 0 0 IN IP4 192.168.1.11
s=T059
c=IN IP4 192.168.1.20
t=0 0
m=audio 10026 RTP/AVP 8 2 18 9
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=ptime:30
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/8000
a=ptime:30

[May 8 13:17:31] VERBOSE[10572] chan_sip.c: — (14 headers 12 lines) —
[May 8 13:17:31] VERBOSE[10572] chan_sip.c: Ignoring this INVITE request
[May 8 13:17:31] VERBOSE[10572] chan_sip.c:

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: “100”;tag=338C324631353641000B6B8E
To:
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Length: 0

[May 8 13:17:31] VERBOSE[10572] chan_sip.c: Audio is at 5060
[May 8 13:17:31] VERBOSE[10572] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[May 8 13:17:31] VERBOSE[10572] chan_sip.c:

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: “100”;tag=338C324631353641000B6B8E
To: ;tag=as33edd479
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Type: application/sdp
Content-Length: 180

v=0
o=root 1201763838 1201763839 IN IP4 192.168.1.16
s=Asterisk PBX 1.8.7.0
c=IN IP4 192.168.1.16
t=0 0
m=audio 11408 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

[May 8 13:17:32] VERBOSE[10572] chan_sip.c: Retransmitting #2 (no NAT) to 192.168.1.11:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: “100”;tag=338C324631353641000B6B8E
To: ;tag=as33edd479
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Type: application/sdp
Content-Length: 180

v=0
o=root 1201763838 1201763838 IN IP4 192.168.1.16
s=Asterisk PBX 1.8.7.0
c=IN IP4 192.168.1.16
t=0 0
m=audio 11408 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

[May 8 13:17:34] VERBOSE[10572] chan_sip.c: Retransmitting #3 (no NAT) to 192.168.1.11:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: “100”;tag=338C324631353641000B6B8E
To: ;tag=as33edd479
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Type: application/sdp
Content-Length: 180

v=0
o=root 1201763838 1201763838 IN IP4 192.168.1.16
s=Asterisk PBX 1.8.7.0
c=IN IP4 192.168.1.16
t=0 0
m=audio 11408 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

[May 8 13:17:34] VERBOSE[10685] pbx.c: – Executing [h@macro-dial-one:1] Macro(“SIP/nec-silf-00000004”, “hangupcall,”) in new stack
[May 8 13:17:34] VERBOSE[10685] pbx.c: – Executing [s@macro-hangupcall:1] GotoIf(“SIP/nec-silf-00000004”, “1?endmixmoncheck”) in new stack
[May 8 13:17:34] VERBOSE[10685] pbx.c: – Goto (macro-hangupcall,s,9)
[May 8 13:17:34] VERBOSE[10685] pbx.c: – Executing [s@macro-hangupcall:9] NoOp(“SIP/nec-silf-00000004”, “End of MIXMON check”) in new stack
[May 8 13:17:34] VERBOSE[10685] pbx.c: – Executing [s@macro-hangupcall:10] GotoIf(“SIP/nec-silf-00000004”, “1?nomeetmemon”) in new stack
[May 8 13:17:34] VERBOSE[10685] pbx.c: – Goto (macro-hangupcall,s,15)
[May 8 13:17:34] VERBOSE[10685] pbx.c: – Executing [s@macro-hangupcall:15] NoOp(“SIP/nec-silf-00000004”, “MEETME_RECORDINGFILE=”) in new stack
[May 8 13:17:34] VERBOSE[10685] pbx.c: – Executing [s@macro-hangupcall:16] GotoIf(“SIP/nec-silf-00000004”, “1?noautomon”) in new stack
[May 8 13:17:34] VERBOSE[10685] pbx.c: – Goto (macro-hangupcall,s,1
[May 8 13:17:34] VERBOSE[10685] pbx.c: – Executing [s@macro-hangupcall:18] NoOp(“SIP/nec-silf-00000004”, “TOUCH_MONITOR_OUTPUT=”) in new stack
[May 8 13:17:34] VERBOSE[10685] pbx.c: – Executing [s@macro-hangupcall:19] GotoIf(“SIP/nec-silf-00000004”, “1?noautomon2”) in new stack
[May 8 13:17:34] VERBOSE[10685] pbx.c: – Goto (macro-hangupcall,s,25)
[May 8 13:17:34] VERBOSE[10685] pbx.c: – Executing [s@macro-hangupcall:25] NoOp(“SIP/nec-silf-00000004”, “MONITOR_FILENAME=”) in new stack
[May 8 13:17:34] VERBOSE[10685] pbx.c: – Executing [s@macro-hangupcall:26] GotoIf(“SIP/nec-silf-00000004”, “1?skiprg”) in new stack
[May 8 13:17:34] VERBOSE[10685] pbx.c: – Goto (macro-hangupcall,s,29)
[May 8 13:17:34] VERBOSE[10685] pbx.c: – Executing [s@macro-hangupcall:29] GotoIf(“SIP/nec-silf-00000004”, “1?skipblkvm”) in new stack
[May 8 13:17:34] VERBOSE[10685] pbx.c: – Goto (macro-hangupcall,s,32)
[May 8 13:17:34] VERBOSE[10685] pbx.c: – Executing [s@macro-hangupcall:32] GotoIf(“SIP/nec-silf-00000004”, “1?theend”) in new stack
[May 8 13:17:34] VERBOSE[10685] pbx.c: – Goto (macro-hangupcall,s,34)
[May 8 13:17:34] VERBOSE[10685] pbx.c: – Executing [s@macro-hangupcall:34] Hangup(“SIP/nec-silf-00000004”, “”) in new stack
[May 8 13:17:34] VERBOSE[10685] app_macro.c: == Spawn extension (macro-hangupcall, s, 34) exited non-zero on ‘SIP/nec-silf-00000004’ in macro ‘hangupcall’
[May 8 13:17:34] VERBOSE[10685] features.c: == Spawn extension (macro-dial-one, h, 1) exited non-zero on ‘SIP/nec-silf-00000004’
[May 8 13:17:34] VERBOSE[10685] app_macro.c: == Spawn extension (macro-dial-one, s, 37) exited non-zero on ‘SIP/nec-silf-00000004’ in macro ‘dial-one’
[May 8 13:17:34] VERBOSE[10685] app_macro.c: == Spawn extension (macro-exten-vm, s, 9) exited non-zero on ‘SIP/nec-silf-00000004’ in macro ‘exten-vm’
[May 8 13:17:34] VERBOSE[10685] pbx.c: == Spawn extension (from-trunk, 3801, 1) exited non-zero on ‘SIP/nec-silf-00000004’
[May 8 13:17:34] VERBOSE[10685] chan_sip.c: Scheduling destruction of SIP dialog ‘0201C1A90C81400000000010@192.168.1.11’ in 32000 ms (Method: INVITE)
[May 8 13:17:35] VERBOSE[10572] chan_sip.c:

INVITE sip:3801@192.168.1.16 SIP/2.0
From: “100”;tag=338C324631353641000B6B8E
To:
Contact:
Content-Type: application/sdp
Allow: INVITE,ACK,BYE,CANCEL,PRACK,UPDATE
Supported: 100rel,timer
Expires: 180
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Max-Forwards: 70
User-Agent: NEC-i SV8100-GE 06.01
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD
Content-Length: 220

v=0
o=- 0 0 IN IP4 192.168.1.11
s=T059
c=IN IP4 192.168.1.20
t=0 0
m=audio 10026 RTP/AVP 8 2 18 9
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=ptime:30
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/8000
a=ptime:30

[May 8 13:17:35] VERBOSE[10572] chan_sip.c: — (14 headers 12 lines) —
[May 8 13:17:35] VERBOSE[10572] chan_sip.c: Ignoring this INVITE request
[May 8 13:17:35] NOTICE[10572] chan_sip.c: Unable to create/find SIP channel for this INVITE
[May 8 13:17:35] VERBOSE[10572] chan_sip.c:

SIP/2.0 503 Unavailable
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: “100”;tag=338C324631353641000B6B8E
To: ;tag=as33edd479
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

[May 8 13:17:35] VERBOSE[10572] chan_sip.c: Scheduling destruction of SIP dialog ‘0201C1A90C81400000000010@192.168.1.11’ in 32000 ms (Method: INVITE)
[May 8 13:17:38] VERBOSE[10572] chan_sip.c: Retransmitting #4 (no NAT) to 192.168.1.11:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: “100”;tag=338C324631353641000B6B8E
To: ;tag=as33edd479
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Type: application/sdp
Content-Length: 180

v=0
o=root 1201763838 1201763838 IN IP4 192.168.1.16
s=Asterisk PBX 1.8.7.0
c=IN IP4 192.168.1.16
t=0 0
m=audio 11408 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

[May 8 13:17:42] VERBOSE[10572] chan_sip.c: Retransmitting #5 (no NAT) to 192.168.1.11:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: “100”;tag=338C324631353641000B6B8E
To: ;tag=as33edd479
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Type: application/sdp
Content-Length: 180

v=0
o=root 1201763838 1201763838 IN IP4 192.168.1.16
s=Asterisk PBX 1.8.7.0
c=IN IP4 192.168.1.16
t=0 0
m=audio 11408 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

[May 8 13:17:43] VERBOSE[10572] chan_sip.c:

INVITE sip:3801@192.168.1.16 SIP/2.0
From: “100”;tag=338C324631353641000B6B8E
To:
Contact:
Content-Type: application/sdp
Allow: INVITE,ACK,BYE,CANCEL,PRACK,UPDATE
Supported: 100rel,timer
Expires: 180
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Max-Forwards: 70
User-Agent: NEC-i SV8100-GE 06.01
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD
Content-Length: 220

v=0
o=- 0 0 IN IP4 192.168.1.11
s=T059
c=IN IP4 192.168.1.20
t=0 0
m=audio 10026 RTP/AVP 8 2 18 9
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=ptime:30
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/8000
a=ptime:30

[May 8 13:17:43] VERBOSE[10572] chan_sip.c: — (14 headers 12 lines) —
[May 8 13:17:43] VERBOSE[10572] chan_sip.c: Ignoring this INVITE request
[May 8 13:17:46] VERBOSE[10572] chan_sip.c: Retransmitting #6 (no NAT) to 192.168.1.11:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: “100”;tag=338C324631353641000B6B8E
To: ;tag=as33edd479
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Type: application/sdp
Content-Length: 180

v=0
o=root 1201763838 1201763838 IN IP4 192.168.1.16
s=Asterisk PBX 1.8.7.0
c=IN IP4 192.168.1.16
t=0 0
m=audio 11408 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

[May 8 13:17:50] VERBOSE[10572] chan_sip.c: Retransmitting #7 (no NAT) to 192.168.1.11:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: “100”;tag=338C324631353641000B6B8E
To: ;tag=as33edd479
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Type: application/sdp
Content-Length: 180

v=0
o=root 1201763838 1201763838 IN IP4 192.168.1.16
s=Asterisk PBX 1.8.7.0
c=IN IP4 192.168.1.16
t=0 0
m=audio 11408 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

[May 8 13:17:54] VERBOSE[10572] chan_sip.c: Retransmitting #8 (no NAT) to 192.168.1.11:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: “100”;tag=338C324631353641000B6B8E
To: ;tag=as33edd479
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Type: application/sdp
Content-Length: 180

v=0
o=root 1201763838 1201763838 IN IP4 192.168.1.16
s=Asterisk PBX 1.8.7.0
c=IN IP4 192.168.1.16
t=0 0
m=audio 11408 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

[May 8 13:17:58] VERBOSE[10572] chan_sip.c: Retransmitting #9 (no NAT) to 192.168.1.11:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: “100”;tag=338C324631353641000B6B8E
To: ;tag=as33edd479
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Type: application/sdp
Content-Length: 180

v=0
o=root 1201763838 1201763838 IN IP4 192.168.1.16
s=Asterisk PBX 1.8.7.0
c=IN IP4 192.168.1.16
t=0 0
m=audio 11408 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

[May 8 13:17:59] VERBOSE[10572] chan_sip.c:

INVITE sip:3801@192.168.1.16 SIP/2.0
From: “100”;tag=338C324631353641000B6B8E
To:
Contact:
Content-Type: application/sdp
Allow: INVITE,ACK,BYE,CANCEL,PRACK,UPDATE
Supported: 100rel,timer
Expires: 180
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Max-Forwards: 70
User-Agent: NEC-i SV8100-GE 06.01
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD
Content-Length: 220

v=0
o=- 0 0 IN IP4 192.168.1.11
s=T059
c=IN IP4 192.168.1.20
t=0 0
m=audio 10026 RTP/AVP 8 2 18 9
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=ptime:30
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/8000
a=ptime:30

[May 8 13:17:59] VERBOSE[10572] chan_sip.c: — (14 headers 12 lines) —
[May 8 13:17:59] VERBOSE[10572] chan_sip.c: Ignoring this INVITE request

[May 8 13:18:02] VERBOSE[10572] chan_sip.c: Retransmitting #10 (no NAT) to 192.168.1.11:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: “100”;tag=338C324631353641000B6B8E
To: ;tag=as33edd479
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Type: application/sdp
Content-Length: 180

v=0
o=root 1201763838 1201763838 IN IP4 192.168.1.16
s=Asterisk PBX 1.8.7.0
c=IN IP4 192.168.1.16
t=0 0
m=audio 11408 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

[May 8 13:18:02] WARNING[10572] chan_sip.c: Retransmission timeout reached on transmission 0201C1A90C81400000000010@192.168.1.11 for seqno 1 (Critical Response) – See wiki.asterisk.org/wiki/display/ … nsmissions
Packet timed out after 32000ms with no response

Please help me out, guys.
Thank you.

david55 · May 8, 2012, 2:42pm

There is no ACK on what I assume to be a re-invite from the NEC.

There also seem to be protocol violations: To, Contact, etc., containing empty values. If you have removed these for security reasons, there is no point in continuing the analysis, as it is important to be able to match addresses, and tags, even if the addresses are not the real ones, when trying to debug a SIP dialogue.

system · May 8, 2012, 2:57pm

Ok, thank you very much. So you think that NEC is incorrect configured ? What values are missing ?
The ip adresses are real. NEC is passing the voice through 192.168.1.20.

Here is ngrep command on the interface with nec:

From asterisk to nec is generating 2-way traffic:

U 192.168.1.16:11456 -> 192.168.1.20:10028
…://… ."…UT…U…TWU.UTVVT…UVWTU…UTU.UT…UTWWU…U…UUU…UWWT…UUUTVQQQWWWWVVVVUUUTTU.UTUTUUTWTWUTTTWTU…TU…TU…UU…
…U…

U 192.168.1.20:10028 -> 192.168.1.16:11456
…f…U…JT…
…

U 192.168.1.16:11456 -> 192.168.1.20:10028
…;//… ."…UTU…UTU…U…UUTTTTWWWVWUTWWWTUTUUU…WWU…UU…UTT…U…UWVWU…UU.UTTTTWV
VWU…

U 192.168.1.20:10028 -> 192.168.1.16:11456
…g…V@…JT…
…

From nec to asterisk is only one-way traffic:

U 192.168.1.16:15184 -> 192.168.1.20:10020
…qH5…/WUU…UUUTWTUTU…UWU…TTTWT…TWTT…U…UUTTTWTU…UWWUW.U…U…TUQW…U.TT.T…WV.WQ.UTU…UVQT…VW.TV…T…UT.WQ.Q.R.Z.]…5….
.…3.

U 192.168.1.16:15184 -> 192.168.1.20:10020
…q.5…/……"(…(…3,…#’…T…x<;2…>…7…"…>9…l…q…6.HO…h…z…f…S…a…}…m…d6…^…W…1…
.P.`…

U 192.168.1.16:15184 -> 192.168.1.20:10020
…r.5…/zPr…~…Y…`l.f…J…CJ…|.y.p…{G……~…w…7.6…f…RJJ…ca.iC…C…R…f.K…v…j…?..’…d…j’f…d…0.T.
]…f

U 192.168.1.16:15184 -> 192.168.1.20:10020
…s(5…/57…l…2…sn…dz…@…6m…n.1…c…eTX……1.…,……)4……9&…%…1…6…7……%…!..;&.7=…-7)……
.,!.

system · May 8, 2012, 3:15pm

And no, I haven’t removed anything.

david55 · May 8, 2012, 8:16pm

I would assume incorrectly programmed, rather than configured. Although I haven’t double checked the RFC, I’m fairly sure that:

To:

and

Contact:

are invalid.

system · May 9, 2012, 6:01am

Thank you very much. A little bit closer… So I have to tell NEC to program their PBX to fill these fields: To: and Contact: in SDP signaling ?

Nec guy told me that:

“NEC is not passing the voice because is not receiving OK to SDP sequence”

system · May 9, 2012, 6:18am

It seems like Contact and To filelds are not empty, and asterisk still ignores INVITE:

<------------>
– SIP/3801-00000090 is ringing

<— SIP read from UDP:192.168.1.11:5060 —>
INVITE sip:3801@192.168.1.16 SIP/2.0
From: "100"sip:100@192.168.1.11;tag=2A6132463135364100165109
To: sip:3801@192.168.1.16:5060
Contact: sip:100@192.168.1.11:5060
Content-Type: application/sdp
Allow: INVITE,ACK,BYE,CANCEL,PRACK,UPDATE
Supported: 100rel,timer
Expires: 180
Call-ID: 0201C2C0028140000000002A@192.168.1.11
CSeq: 1 INVITE
Max-Forwards: 70
User-Agent: NEC-i SV8100-GE 06.01
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK00532995F794FB02
Content-Length: 220

v=0
o=- 0 0 IN IP4 192.168.1.11
s=T059
c=IN IP4 192.168.1.20
t=0 0
m=audio 10020 RTP/AVP 8 2 18 9
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=ptime:30
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/8000
a=ptime:30
<------------->
— (14 headers 12 lines) —
Ignoring this INVITE request

david55 · May 9, 2012, 9:10am

Asterisk has sent the OK multiple times to 192.168.1.11:5060, because it didn’t get the ACK back. You need to find out where it is getting lost (which will be outside of Asterisk), or why the NEC is sending that as a contact address, when it isn’t actually at that address.

I am still confused by the missing information in your trace, although maybe that is because it wasn’t properly quoted to the forum.

system · May 9, 2012, 10:08am

Ok. So this is a full log of a call from 165 (nec) to 3801 (Asterisk). No information is extracted or missed. Nec guys solved the problem of CID. Now the call is comming from 165, not 100 as previous examples. But still not working.

<— SIP read from UDP:192.168.1.11:5060 —>
INVITE sip:3801@192.168.1.16 SIP/2.0
From: "165"sip:165@192.168.1.11;tag=A0FA3246313536410018434C
To: sip:3801@192.168.1.16:5060
Contact: sip:100@192.168.1.11:5060
Content-Type: application/sdp
Allow: INVITE,ACK,BYE,CANCEL,PRACK,UPDATE
Supported: 100rel,timer
Expires: 180
Call-ID: 0201C2F1D381400000000034@192.168.1.11
CSeq: 1 INVITE
Max-Forwards: 70
User-Agent: NEC-i SV8100-GE 06.01
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bKD1E31159EC188464
Content-Length: 220

v=0
o=- 0 0 IN IP4 192.168.1.11
s=T059
c=IN IP4 192.168.1.20
t=0 0
m=audio 10022 RTP/AVP 8 2 18 9
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=ptime:30
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/8000
a=ptime:30
<------------->
— (14 headers 12 lines) —
Sending to 192.168.1.11:5060 (no NAT)
Using INVITE request as basis request - 0201C2F1D381400000000034@192.168.1.11
Found peer ‘nec-silf’ for ‘165’ from 192.168.1.11:5060
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
ound RTP audio format 8
Found RTP audio format 2
Found RTP audio format 18
Found RTP audio format 9
Found audio description format PCMA for ID 8
Found audio description format G726-32 for ID 2
Found audio description format G729 for ID 18
Found audio description format G722 for ID 9
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x1908 (alaw|g726|g729|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.1.20:10022
Looking for 3801 in from-trunk-sip-nec-silf (domain 192.168.1.16)
list_route: hop: sip:100@192.168.1.11:5060

<— Transmitting (no NAT) to 192.168.1.11:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bKD1E31159EC188464;received=192.168.1.11
From: "165"sip:165@192.168.1.11;tag=A0FA3246313536410018434C
To: sip:3801@192.168.1.16:5060
Call-ID: 0201C2F1D381400000000034@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:3801@192.168.1.16:5060
Content-Length: 0

<------------>
– Executing [3801@from-trunk-sip-nec-silf:1] Set(“SIP/nec-silf-000000e0”, “GROUP()=OUT_4”) in new stack
– Executing [3801@from-trunk-sip-nec-silf:2] Goto(“SIP/nec-silf-000000e0”, “from-trunk,3801,1”) in new stack
– Goto (from-trunk,3801,1)
– Executing [3801@from-trunk:1] Macro(“SIP/nec-silf-000000e0”, “exten-vm,novm,3801”) in new stack
– Executing [s@macro-exten-vm:1] Macro(“SIP/nec-silf-000000e0”, “user-callerid,”) in new stack
– Executing [s@macro-user-callerid:1] Set(“SIP/nec-silf-000000e0”, “AMPUSER=165”) in new stack
– Executing [s@macro-user-callerid:2] GotoIf(“SIP/nec-silf-000000e0”, “0?report”) in new stack
– Executing [s@macro-user-callerid:3] ExecIf(“SIP/nec-silf-000000e0”, “1?Set(REALCALLERIDNUM=165)”) in new stack
– Executing [s@macro-user-callerid:4] Set(“SIP/nec-silf-000000e0”, “AMPUSER=”) in new stack
– Executing [s@macro-user-callerid:5] Set(“SIP/nec-silf-000000e0”, “AMPUSERCIDNAME=”) in new stack
– Executing [s@macro-user-callerid:6] GotoIf(“SIP/nec-silf-000000e0”, “1?report”) in new stack
– Goto (macro-user-callerid,s,10)
– Executing [s@macro-user-callerid:10] GotoIf(“SIP/nec-silf-000000e0”, “0?continue”) in new stack
– Executing [s@macro-user-callerid:11] Set(“SIP/nec-silf-000000e0”, “__TTL=64”) in new stack
– Executing [s@macro-user-callerid:12] GotoIf(“SIP/nec-silf-000000e0”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,19)
– Executing [s@macro-user-callerid:19] Set(“SIP/nec-silf-000000e0”, “CALLERID(number)=165”) in new stack
– Executing [s@macro-user-callerid:20] Set(“SIP/nec-silf-000000e0”, “CALLERID(name)=165”) in new stack
– Executing [s@macro-user-callerid:21] NoOp(“SIP/nec-silf-000000e0”, “Using CallerID “165” <165>”) in new stack
– Executing [s@macro-exten-vm:2] Set(“SIP/nec-silf-000000e0”, “RingGroupMethod=none”) in new stack
– Executing [s@macro-exten-vm:3] Set(“SIP/nec-silf-000000e0”, “VMBOX=novm”) in new stack
– Executing [s@macro-exten-vm:4] Set(“SIP/nec-silf-000000e0”, “__EXTTOCALL=3801”) in new stack
– Executing [s@macro-exten-vm:5] Set(“SIP/nec-silf-000000e0”, “CFUEXT=”) in new stack
– Executing [s@macro-exten-vm:6] Set(“SIP/nec-silf-000000e0”, “CFBEXT=”) in new stack
– Executing [s@macro-exten-vm:7] Set(“SIP/nec-silf-000000e0”, “RT=”"") in new stack
– Executing [s@macro-exten-vm:8] Macro(“SIP/nec-silf-000000e0”, “record-enable,3801,IN”) in new stack
– Executing [s@macro-record-enable:1] GotoIf(“SIP/nec-silf-000000e0”, “1?check”) in new stack
– Goto (macro-record-enable,s,4)
– Executing [s@macro-record-enable:4] ExecIf(“SIP/nec-silf-000000e0”, “0?MacroExit()”) in new stack
– Executing [s@macro-record-enable:5] GotoIf(“SIP/nec-silf-000000e0”, “0?Group:OUT”) in new stack
– Goto (macro-record-enable,s,15)
– Executing [s@macro-record-enable:15] GotoIf(“SIP/nec-silf-000000e0”, “1?IN”) in new stack
– Goto (macro-record-enable,s,20)
– Executing [s@macro-record-enable:20] ExecIf(“SIP/nec-silf-000000e0”, “1?MacroExit()”) in new stack
– Executing [s@macro-exten-vm:9] Macro(“SIP/nec-silf-000000e0”, “dial-one,”",tr,3801") in new stack
– Executing [s@macro-dial-one:1] Set(“SIP/nec-silf-000000e0”, “DEXTEN=3801”) in new stack
– Executing [s@macro-dial-one:2] Set(“SIP/nec-silf-000000e0”, “DIALSTATUS_CW=”) in new stack
– Executing [s@macro-dial-one:3] GosubIf(“SIP/nec-silf-000000e0”, “0?screen,1”) in new stack
– Executing [s@macro-dial-one:4] GosubIf(“SIP/nec-silf-000000e0”, “0?cf,1”) in new stack
– Executing [s@macro-dial-one:5] GotoIf(“SIP/nec-silf-000000e0”, “1?skip1”) in new stack
– Goto (macro-dial-one,s,8)
– Executing [s@macro-dial-one:8] GotoIf(“SIP/nec-silf-000000e0”, “0?nodial”) in new stack
– Executing [s@macro-dial-one:9] GotoIf(“SIP/nec-silf-000000e0”, “0?continue”) in new stack
– Executing [s@macro-dial-one:10] Set(“SIP/nec-silf-000000e0”, “EXTHASCW=”) in new stack
– Executing [s@macro-dial-one:11] GotoIf(“SIP/nec-silf-000000e0”, “1?next1:cwinusebusy”) in new stack
– Goto (macro-dial-one,s,12)
– Executing [s@macro-dial-one:12] GotoIf(“SIP/nec-silf-000000e0”, “0?docfu:skip3”) in new stack
– Goto (macro-dial-one,s,16)
– Executing [s@macro-dial-one:16] GotoIf(“SIP/nec-silf-000000e0”, “1?next2:continue”) in new stack
– Goto (macro-dial-one,s,17)
– Executing [s@macro-dial-one:17] GotoIf(“SIP/nec-silf-000000e0”, “1?continue”) in new stack
– Goto (macro-dial-one,s,25)
– Executing [s@macro-dial-one:25] GotoIf(“SIP/nec-silf-000000e0”, “0?nodial”) in new stack
– Executing [s@macro-dial-one:26] GosubIf(“SIP/nec-silf-000000e0”, “1?dstring,1:dlocal,1”) in new stack
– Executing [dstring@macro-dial-one:1] Set(“SIP/nec-silf-000000e0”, “DSTRING=”) in new stack
– Executing [dstring@macro-dial-one:2] Set(“SIP/nec-silf-000000e0”, “DEVICES=3801”) in new stack
– Executing [dstring@macro-dial-one:3] ExecIf(“SIP/nec-silf-000000e0”, “0?Return()”) in new stack
– Executing [dstring@macro-dial-one:4] ExecIf(“SIP/nec-silf-000000e0”, “0?Set(DEVICES=801)”) in new stack
– Executing [dstring@macro-dial-one:5] Set(“SIP/nec-silf-000000e0”, “LOOPCNT=1”) in new stack
– Executing [dstring@macro-dial-one:6] Set(“SIP/nec-silf-000000e0”, “ITER=1”) in new stack
– Executing [dstring@macro-dial-one:7] Set(“SIP/nec-silf-000000e0”, “THISDIAL=SIP/3801”) in new stack
– Executing [dstring@macro-dial-one:8] GosubIf(“SIP/nec-silf-000000e0”, “1?zap2dahdi,1”) in new stack
– Executing [zap2dahdi@macro-dial-one:1] ExecIf(“SIP/nec-silf-000000e0”, “0?Return()”) in new stack
– Executing [zap2dahdi@macro-dial-one:2] Set(“SIP/nec-silf-000000e0”, “NEWDIAL=”) in new stack
– Executing [zap2dahdi@macro-dial-one:3] Set(“SIP/nec-silf-000000e0”, “LOOPCNT2=1”) in new stack
– Executing [zap2dahdi@macro-dial-one:4] Set(“SIP/nec-silf-000000e0”, “ITER2=1”) in new stack
– Executing [zap2dahdi@macro-dial-one:5] Set(“SIP/nec-silf-000000e0”, “THISPART2=SIP/3801”) in new stack
– Executing [zap2dahdi@macro-dial-one:6] ExecIf(“SIP/nec-silf-000000e0”, “0?Set(THISPART2=DAHDI/3801)”) in new stack
– Executing [zap2dahdi@macro-dial-one:7] Set(“SIP/nec-silf-000000e0”, “NEWDIAL=SIP/3801&”) in new stack
– Executing [zap2dahdi@macro-dial-one:8] Set(“SIP/nec-silf-000000e0”, “ITER2=2”) in new stack
– Executing [zap2dahdi@macro-dial-one:9] GotoIf(“SIP/nec-silf-000000e0”, “0?begin2”) in new stack
– Executing [zap2dahdi@macro-dial-one:10] Set(“SIP/nec-silf-000000e0”, “THISDIAL=SIP/3801”) in new stack
– Executing [zap2dahdi@macro-dial-one:11] Return(“SIP/nec-silf-000000e0”, “”) in new stack
– Executing [dstring@macro-dial-one:9] Set(“SIP/nec-silf-000000e0”, “DSTRING=SIP/3801&”) in new stack
– Executing [dstring@macro-dial-one:10] Set(“SIP/nec-silf-000000e0”, “ITER=2”) in new stack
– Executing [dstring@macro-dial-one:11] GotoIf(“SIP/nec-silf-000000e0”, “0?begin”) in new stack
– Executing [dstring@macro-dial-one:12] Set(“SIP/nec-silf-000000e0”, “DSTRING=SIP/3801”) in new stack
– Executing [dstring@macro-dial-one:13] Return(“SIP/nec-silf-000000e0”, “”) in new stack
– Executing [s@macro-dial-one:27] GotoIf(“SIP/nec-silf-000000e0”, “0?nodial”) in new stack
– Executing [s@macro-dial-one:28] GotoIf(“SIP/nec-silf-000000e0”, “1?skiptrace”) in new stack
– Goto (macro-dial-one,s,30)
– Executing [s@macro-dial-one:30] Set(“SIP/nec-silf-000000e0”, “D_OPTIONS=tr”) in new stack
– Executing [s@macro-dial-one:31] ExecIf(“SIP/nec-silf-000000e0”, “0?SIPAddHeader(Alert-Info: )”) in new stack
– Executing [s@macro-dial-one:32] ExecIf(“SIP/nec-silf-000000e0”, “0?SIPAddHeader()”) in new stack
– Executing [s@macro-dial-one:33] ExecIf(“SIP/nec-silf-000000e0”, “0?Set(CHANNEL(musicclass)=)”) in new stack
– Executing [s@macro-dial-one:34] GosubIf(“SIP/nec-silf-000000e0”, “0?qwait,1”) in new stack
– Executing [s@macro-dial-one:35] Set(“SIP/nec-silf-000000e0”, “__CWIGNORE=”) in new stack
– Executing [s@macro-dial-one:36] Set(“SIP/nec-silf-000000e0”, “__KEEPCID=TRUE”) in new stack
– Executing [s@macro-dial-one:37] Dial(“SIP/nec-silf-000000e0”, “SIP/3801,”",tr") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Called SIP/3801

<— Transmitting (no NAT) to 192.168.1.11:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bKD1E31159EC188464;received=192.168.1.11
From: "165"sip:165@192.168.1.11;tag=A0FA3246313536410018434C
To: sip:3801@192.168.1.16:5060;tag=as697fb8c8
Call-ID: 0201C2F1D381400000000034@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:3801@192.168.1.16:5060
Content-Length: 0

<------------>
– SIP/3801-000000e1 is ringing

<— Transmitting (no NAT) to 192.168.1.11:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bKD1E31159EC188464;received=192.168.1.11
From: "165"sip:165@192.168.1.11;tag=A0FA3246313536410018434C
To: sip:3801@192.168.1.16:5060;tag=as697fb8c8
Call-ID: 0201C2F1D381400000000034@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:3801@192.168.1.16:5060
Content-Length: 0

<------------>
– SIP/3801-000000e1 is ringing

<— SIP read from UDP:192.168.1.11:5060 —>
INVITE sip:3801@192.168.1.16 SIP/2.0
From: "165"sip:165@192.168.1.11;tag=A0FA3246313536410018434C
To: sip:3801@192.168.1.16:5060
Contact: sip:100@192.168.1.11:5060
Content-Type: application/sdp
Allow: INVITE,ACK,BYE,CANCEL,PRACK,UPDATE
Supported: 100rel,timer
Expires: 180
Call-ID: 0201C2F1D381400000000034@192.168.1.11
CSeq: 1 INVITE
Max-Forwards: 70
User-Agent: NEC-i SV8100-GE 06.01
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bKD1E31159EC188464
Content-Length: 220

v=0
o=- 0 0 IN IP4 192.168.1.11
s=T059
c=IN IP4 192.168.1.20
t=0 0
m=audio 10022 RTP/AVP 8 2 18 9
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=ptime:30
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/8000
a=ptime:30
<------------->
— (14 headers 12 lines) —
Ignoring this INVITE request

<— Transmitting (no NAT) to 192.168.1.11:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bKD1E31159EC188464;received=192.168.1.11
From: "165"sip:165@192.168.1.11;tag=A0FA3246313536410018434C
To: sip:3801@192.168.1.16:5060
Call-ID: 0201C2F1D381400000000034@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:3801@192.168.1.16:5060
Content-Length: 0

<------------>

<— SIP read from UDP:192.168.1.11:5060 —>
INVITE sip:3801@192.168.1.16 SIP/2.0
From: "165"sip:165@192.168.1.11;tag=A0FA3246313536410018434C
To: sip:3801@192.168.1.16:5060
Contact: sip:100@192.168.1.11:5060
Content-Type: application/sdp
Allow: INVITE,ACK,BYE,CANCEL,PRACK,UPDATE
Supported: 100rel,timer
Expires: 180
Call-ID: 0201C2F1D381400000000034@192.168.1.11
CSeq: 1 INVITE
Max-Forwards: 70
User-Agent: NEC-i SV8100-GE 06.01
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bKD1E31159EC188464
Content-Length: 220

v=0
o=- 0 0 IN IP4 192.168.1.11
s=T059
c=IN IP4 192.168.1.20
t=0 0
m=audio 10022 RTP/AVP 8 2 18 9
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=ptime:30
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/8000
a=ptime:30
<------------->
— (14 headers 12 lines) —
Ignoring this INVITE request

<— Transmitting (no NAT) to 192.168.1.11:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bKD1E31159EC188464;received=192.168.1.11
From: "165"sip:165@192.168.1.11;tag=A0FA3246313536410018434C
To: sip:3801@192.168.1.16:5060
Call-ID: 0201C2F1D381400000000034@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:3801@192.168.1.16:5060
Content-Length: 0

<------------>

<— SIP read from UDP:192.168.1.11:5060 —>
INVITE sip:3801@192.168.1.16 SIP/2.0
From: "165"sip:165@192.168.1.11;tag=A0FA3246313536410018434C
To: sip:3801@192.168.1.16:5060
Contact: sip:100@192.168.1.11:5060
Content-Type: application/sdp
Allow: INVITE,ACK,BYE,CANCEL,PRACK,UPDATE
Supported: 100rel,timer
Expires: 180
Call-ID: 0201C2F1D381400000000034@192.168.1.11
CSeq: 1 INVITE
Max-Forwards: 70
User-Agent: NEC-i SV8100-GE 06.01
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bKD1E31159EC188464
Content-Length: 220

v=0
o=- 0 0 IN IP4 192.168.1.11
s=T059
c=IN IP4 192.168.1.20
t=0 0
m=audio 10022 RTP/AVP 8 2 18 9
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=ptime:30
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/8000
a=ptime:30
<------------->
— (14 headers 12 lines) —
Ignoring this INVITE request

<— Transmitting (no NAT) to 192.168.1.11:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bKD1E31159EC188464;received=192.168.1.11
From: "165"sip:165@192.168.1.11;tag=A0FA3246313536410018434C
To: sip:3801@192.168.1.16:5060
Call-ID: 0201C2F1D381400000000034@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:3801@192.168.1.16:5060
Content-Length: 0

<------------>
– SIP/3801-000000e1 answered SIP/nec-silf-000000e0
Audio is at 5060
Adding codec 0x8 (alaw) to SDP

<— Reliably Transmitting (no NAT) to 192.168.1.11:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bKD1E31159EC188464;received=192.168.1.11
From: "165"sip:165@192.168.1.11;tag=A0FA3246313536410018434C
To: sip:3801@192.168.1.16:5060;tag=as697fb8c8
Call-ID: 0201C2F1D381400000000034@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:3801@192.168.1.16:5060
Content-Type: application/sdp
Content-Length: 180

v=0
o=root 1926494290 1926494290 IN IP4 192.168.1.16
s=Asterisk PBX 1.8.7.0
c=IN IP4 192.168.1.16
t=0 0
m=audio 18844 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

<------------>
Retransmitting #1 (no NAT) to 192.168.1.11:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bKD1E31159EC188464;received=192.168.1.11
From: "165"sip:165@192.168.1.11;tag=A0FA3246313536410018434C
To: sip:3801@192.168.1.16:5060;tag=as697fb8c8
Call-ID: 0201C2F1D381400000000034@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:3801@192.168.1.16:5060
Content-Type: application/sdp
Content-Length: 180

v=0
o=root 1926494290 1926494290 IN IP4 192.168.1.16
s=Asterisk PBX 1.8.7.0
c=IN IP4 192.168.1.16
t=0 0
m=audio 18844 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

Retransmitting #2 (no NAT) to 192.168.1.11:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bKD1E31159EC188464;received=192.168.1.11
From: "165"sip:165@192.168.1.11;tag=A0FA3246313536410018434C
To: sip:3801@192.168.1.16:5060;tag=as697fb8c8
Call-ID: 0201C2F1D381400000000034@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:3801@192.168.1.16:5060
Content-Type: application/sdp
Content-Length: 180

v=0
o=root 1926494290 1926494290 IN IP4 192.168.1.16
s=Asterisk PBX 1.8.7.0
c=IN IP4 192.168.1.16
t=0 0
m=audio 18844 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

Retransmitting #3 (no NAT) to 192.168.1.11:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bKD1E31159EC188464;received=192.168.1.11
From: "165"sip:165@192.168.1.11;tag=A0FA3246313536410018434C
To: sip:3801@192.168.1.16:5060;tag=as697fb8c8
Call-ID: 0201C2F1D381400000000034@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:3801@192.168.1.16:5060
Content-Type: application/sdp
Content-Length: 180

v=0
o=root 1926494290 1926494290 IN IP4 192.168.1.16
s=Asterisk PBX 1.8.7.0
c=IN IP4 192.168.1.16
t=0 0
m=audio 18844 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

Retransmitting #4 (no NAT) to 192.168.1.11:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bKD1E31159EC188464;received=192.168.1.11
From: "165"sip:165@192.168.1.11;tag=A0FA3246313536410018434C
To: sip:3801@192.168.1.16:5060;tag=as697fb8c8
Call-ID: 0201C2F1D381400000000034@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:3801@192.168.1.16:5060
Content-Type: application/sdp
Content-Length: 180

v=0
o=root 1926494290 1926494290 IN IP4 192.168.1.16
s=Asterisk PBX 1.8.7.0
c=IN IP4 192.168.1.16
t=0 0
m=audio 18844 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

Retransmitting #5 (no NAT) to 192.168.1.11:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bKD1E31159EC188464;received=192.168.1.11
From: "165"sip:165@192.168.1.11;tag=A0FA3246313536410018434C
To: sip:3801@192.168.1.16:5060;tag=as697fb8c8
Call-ID: 0201C2F1D381400000000034@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:3801@192.168.1.16:5060
Content-Type: application/sdp
Content-Length: 180

v=0
o=root 1926494290 1926494290 IN IP4 192.168.1.16
s=Asterisk PBX 1.8.7.0
c=IN IP4 192.168.1.16
t=0 0
m=audio 18844 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

<— SIP read from UDP:192.168.1.11:5060 —>
INVITE sip:3801@192.168.1.16 SIP/2.0
From: "165"sip:165@192.168.1.11;tag=A0FA3246313536410018434C
To: sip:3801@192.168.1.16:5060
Contact: sip:100@192.168.1.11:5060
Content-Type: application/sdp
Allow: INVITE,ACK,BYE,CANCEL,PRACK,UPDATE
Supported: 100rel,timer
Expires: 180
Call-ID: 0201C2F1D381400000000034@192.168.1.11
CSeq: 1 INVITE
Max-Forwards: 70
User-Agent: NEC-i SV8100-GE 06.01
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bKD1E31159EC188464
Content-Length: 220

v=0
o=- 0 0 IN IP4 192.168.1.11
s=T059
c=IN IP4 192.168.1.20
t=0 0
m=audio 10022 RTP/AVP 8 2 18 9
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=ptime:30
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/8000
a=ptime:30
<------------->
— (14 headers 12 lines) —
Ignoring this INVITE request

<— Transmitting (no NAT) to 192.168.1.11:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bKD1E31159EC188464;received=192.168.1.11
From: "165"sip:165@192.168.1.11;tag=A0FA3246313536410018434C
To: sip:3801@192.168.1.16:5060
Call-ID: 0201C2F1D381400000000034@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:3801@192.168.1.16:5060
Content-Length: 0

<------------>
Audio is at 5060
Adding codec 0x8 (alaw) to SDP

<— Transmitting (no NAT) to 192.168.1.11:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bKD1E31159EC188464;received=192.168.1.11
From: "165"sip:165@192.168.1.11;tag=A0FA3246313536410018434C
To: sip:3801@192.168.1.16:5060;tag=as697fb8c8
Call-ID: 0201C2F1D381400000000034@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:3801@192.168.1.16:5060
Content-Type: application/sdp
Content-Length: 180

v=0
o=root 1926494290 1926494291 IN IP4 192.168.1.16
s=Asterisk PBX 1.8.7.0
c=IN IP4 192.168.1.16
t=0 0
m=audio 18844 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

<------------>
– Executing [h@macro-dial-one:1] Macro(“SIP/nec-silf-000000e0”, “hangupcall,”) in new stack
– Executing [s@macro-hangupcall:1] GotoIf(“SIP/nec-silf-000000e0”, “1?endmixmoncheck”) in new stack
– Goto (macro-hangupcall,s,9)
– Executing [s@macro-hangupcall:9] NoOp(“SIP/nec-silf-000000e0”, “End of MIXMON check”) in new stack
– Executing [s@macro-hangupcall:10] GotoIf(“SIP/nec-silf-000000e0”, “1?nomeetmemon”) in new stack
– Goto (macro-hangupcall,s,15)
– Executing [s@macro-hangupcall:15] NoOp(“SIP/nec-silf-000000e0”, “MEETME_RECORDINGFILE=”) in new stack
– Executing [s@macro-hangupcall:16] GotoIf(“SIP/nec-silf-000000e0”, “1?noautomon”) in new stack
– Goto (macro-hangupcall,s,18)
– Executing [s@macro-hangupcall:18] NoOp(“SIP/nec-silf-000000e0”, “TOUCH_MONITOR_OUTPUT=”) in new stack
– Executing [s@macro-hangupcall:19] GotoIf(“SIP/nec-silf-000000e0”, “1?noautomon2”) in new stack
– Goto (macro-hangupcall,s,25)
– Executing [s@macro-hangupcall:25] NoOp(“SIP/nec-silf-000000e0”, “MONITOR_FILENAME=”) in new stack
– Executing [s@macro-hangupcall:26] GotoIf(“SIP/nec-silf-000000e0”, “1?skiprg”) in new stack
– Goto (macro-hangupcall,s,29)
– Executing [s@macro-hangupcall:29] GotoIf(“SIP/nec-silf-000000e0”, “1?skipblkvm”) in new stack
– Goto (macro-hangupcall,s,32)
– Executing [s@macro-hangupcall:32] GotoIf(“SIP/nec-silf-000000e0”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,34)
– Executing [s@macro-hangupcall:34] Hangup(“SIP/nec-silf-000000e0”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 34) exited non-zero on ‘SIP/nec-silf-000000e0’ in macro ‘hangupcall’
== Spawn extension (macro-dial-one, h, 1) exited non-zero on ‘SIP/nec-silf-000000e0’
== Spawn extension (macro-dial-one, s, 37) exited non-zero on ‘SIP/nec-silf-000000e0’ in macro ‘dial-one’
== Spawn extension (macro-exten-vm, s, 9) exited non-zero on ‘SIP/nec-silf-000000e0’ in macro ‘exten-vm’
== Spawn extension (from-trunk, 3801, 1) exited non-zero on 'SIP/nec-silf-000000e0’
Scheduling destruction of SIP dialog ‘0201C2F1D381400000000034@192.168.1.11’ in 6400 ms (Method: INVITE)
Retransmitting #6 (no NAT) to 192.168.1.11:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bKD1E31159EC188464;received=192.168.1.11
From: "165"sip:165@192.168.1.11;tag=A0FA3246313536410018434C
To: sip:3801@192.168.1.16:5060;tag=as697fb8c8
Call-ID: 0201C2F1D381400000000034@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:3801@192.168.1.16:5060
Content-Type: application/sdp
Content-Length: 180

v=0
o=root 1926494290 1926494290 IN IP4 192.168.1.16
s=Asterisk PBX 1.8.7.0
c=IN IP4 192.168.1.16
t=0 0
m=audio 18844 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

Really destroying SIP dialog ‘0201C2F1D381400000000034@192.168.1.11’ Method: INVITE

system · May 9, 2012, 12:16pm

Downgraded to Asterisk 1.4 (trixbox). Now seems to be ok. Thank you very much, although I would like to know what the problem was.
Thank you very much David !

axiallc · October 26, 2012, 7:27pm

skunky,

Can you please provide your installation/configuration steps.
I would like to setup the same.

Thanks in advance.

system · March 1, 2013, 5:59pm

So did anybody got it working with version 1.8?

We are having the same problem with NEC SV8100 version 8.0 and Asterisk version 1.8.

Please let me know if anybody was successful to use the 2 versions together.

david55 · March 1, 2013, 7:35pm

These traces all look as though the NEC is failing to receive or failing to recognize the responses. You need to get traces from the NEC, to see what it thinks it is seeing.

Asterisk is responding 100 Trying to the INVITEs that are being retransmitted, after it has sent a non-100 intermediate response. That is a bug. The RFC 3261 requirement is:

"If a request
retransmission is received while in the “Proceeding” state, the most
recent provisional response that was received from the TU MUST be
passed to the transport layer for retransmission. "

However 1.8.7 is a rather out of date, so one would not be justified in raising a bug report unless you can reproduce it on 1.8.20.1 or later.

In any case, sending 100 inappropriately would not result in the client resending INVITEs, so it is not causing the main problem.

Incidentally, the reference to a incoming entry is a FreePBX concept, not an Asterisk one, and, although this can be treated at an Asterisk level, the very complicated FreePBX dialplan generates a lot of noise. If FreePBX is not prepared to provide a more recent version of Asterisk, that should be taken up with them.

system · March 13, 2013, 8:38am

[quote=“romans4654”]So did anybody got it working with version 1.8?

We are having the same problem with NEC SV8100 version 8.0 and Asterisk version 1.8.

Please let me know if anybody was successful to use the 2 versions together.[/quote]

I would like to add to this issue with a SIP debug:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 111.111.111.111:5060;branch=z9hG4bKEF373A22E8F8D8DB;received=111.111.111.111;rport=5060

From: "19052814429"<sip:19052814429@222.222.222.222>;tag=805E324631353641000A8CD6

To: <sip:4168222773@222.222.222.222:5060>;tag=as4a1106c8

Call-ID: 0203464E478140000000000F@10.0.0.20

CSeq: 2 INVITE

Server: Softswitch

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Session-Expires: 1200;refresher=uas

Contact: <sip:4168222773@222.222.222.222:5060>

Content-Type: application/sdp

Content-Length: 268

 

v=0

o=root 1352808777 1352808778 IN IP4 222.222.222.222

s=Asterisk PBX 1.8.14.0

c=IN IP4 222.222.222.222

t=0 0

m=audio 13538 RTP/AVP 0 110

a=rtpmap:0 PCMU/8000

a=rtpmap:110 telephone-event/8000

a=fmtp:110 0-16

a=silenceSupp:off - - - -

a=ptime:20

a=sendrecv

ACK sip:4168222773@222.222.222.222:5060 SIP/2.0
CSeq: 2 ACK
To: <sip:4168222773@222.222.222.222:5060>;tag=as4a1106c8
From: "19052814429"<sip:19052814429@222.222.222.222>;tag=805E324631353641000A8CD6
Call-ID: 0203464E478140000000000F@10.0.0.20
Max-Forwards: 70
User-Agent: NEC-i SV8100-NA 08.00
Via: SIP/2.0/UDP 111.111.111.111:5060;branch=z9hG4bK06B54C25A399480F
Content-Length: 0
 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 111.111.111.111:5060;branch=z9hG4bKEF373A22E8F8D8DB;received=111.111.111.111;rport=5060
From: "19052814429"<sip:19052814429@222.222.222.222>;tag=805E324631353641000A8CD6
To: <sip:4168222773@222.222.222.222:5060>;tag=as4a1106c8
Call-ID: 0203464E478140000000000F@10.0.0.20
CSeq: 2 INVITE
Server: Softswitch
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1200;refresher=uas
Contact: <sip:4168222773@222.222.222.222:5060>
Content-Type: application/sdp
Content-Length: 268
 
v=0
o=root 1352808777 1352808778 IN IP4 222.222.222.222
s=Asterisk PBX 1.8.14.0
c=IN IP4 222.222.222.222
t=0 0
m=audio 13538 RTP/AVP 0 110
a=rtpmap:0 PCMU/8000
a=rtpmap:110 telephone-event/8000
a=fmtp:110 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
 
ACK sip:4168222773@222.222.222.222:5060 SIP/2.0
CSeq: 2 ACK
To: <sip:4168222773@222.222.222.222:5060>;tag=as4a1106c8
From: "19052814429"<sip:19052814429@222.222.222.222>;tag=805E324631353641000A8CD6
Call-ID: 0203464E478140000000000F@10.0.0.20
Max-Forwards: 70
User-Agent: NEC-i SV8100-NA 08.00
Via: SIP/2.0/UDP 111.111.111.111:5060;branch=z9hG4bK06B54C25A399480F
Content-Length: 0

IPs have been changed, but other from that all the information is the same.

Can anyone give a suggestion why Asterisk cannot match ACK to 200 OK?

dgreengod · June 5, 2013, 11:33pm

Hi.
I am trying to integrate Asterisk (1.8 ) to NEC SV8100 (VER 8 ) by SIP Trunk. Licensing and IPLB are in place, but due the lack of experience I didn’t get that far.
I still can’t establish trunk between 2 systems.

Just wondering if anybody could refer me to appropriate manuals for configuring SV8100 ?
Any information and manuals would be appreciated.

Thank You in advance!

system · November 7, 2013, 11:59am

Hello,
the problem is about the type of “useragent” that Asterisk sends to the NEC pbx.
To solve the problem you has to leave it empty.
into sip_general_custom.conf:
useragent=

Regards,
Gianni

geetha · May 5, 2014, 5:44am

Pls let me know the steps to integrate SC8100 with Asterisk-3.0.0. I am new to this…

Thanks
sagi

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