可能我少用, 但暫時問題不大
lamsoft-pbx*CLI> sip show registry
Host dnsmgr Username Refresh State Reg.Time
sipgate.com:5060 N 20xxxxxxx 285 Registered Mon, 07 Nov 2011 20:42:59
vpntw.XXXXXX.XXX:5060 N xxxx 285 Registered Mon, 07 Nov 2011 20:42:58
sip.goober.com:5060 N xxxxxxxx 285 Registered Mon, 07 Nov 2011 20:42:59
202.0.179.3:5060 N 8523501xxxx 285 Registered Mon, 07 Nov 2011 20:42:59
sip.pennytel.com:5060 N 888710xxxx 285 Registered Mon, 07 Nov 2011 20:42:59
sip.pennytel.com:5060 N 888919xxxx 285 Registered Mon, 07 Nov 2011 20:42:59
6 SIP registrations.
lamsoft-pbx*CLI> sip set debug ip 202.0.179.3
SIP Debugging Enabled for IP: 202.0.179.3
<--- SIP read from UDP:202.0.179.3:5060 --->
hello
<------------->
<--- SIP read from UDP:202.0.179.3:5060 --->
INVITE sip:8523501XXXX@MY_VOIP_SERVER:5060;user=phone SIP/2.0
From: <sip:6XXXXXX@202.0.179.3;user=phone>;tag=f019ff19
To: <sip:8523501XXXX@MY_VOIP_SERVER;user=phone>
CSeq: 1 INVITE
Call-ID: 001c8aecb8f9a834f43c8cf74dfd9531@sx3000
Via: SIP/2.0/UDP 202.0.179.3:5060;branch=z9hG4bKbff83176c
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REGISTER,PRACK,INFO,UPDATE,SUBSCRIBE,NOTIFY,MESSAGE,REFER
Max-Forwards: 70
Supported: 100rel
Contact: <sip:6XXXXXX@202.0.179.3:5060;user=phone>
Content-Length: 294
Content-Type: application/sdp
v=0
o=HuaweiSoftX3000 2256269 2256269 IN IP4 10.0.1.36
s=Sip Call
c=IN IP4 202.0.179.3
t=0 0
m=audio 19240 RTP/AVP 8 0 18 4 97
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=fmtp:18 annexb=yes
<------------->
--- (12 headers 13 lines) ---
Sending to 202.0.179.3:5060 (no NAT)
Using INVITE request as basis request - 001c8aecb8f9a834f43c8cf74dfd9531@sx3000
Found peer 'COMNET_PSTN' for '6XXXXXX' from 202.0.179.3:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 97
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found audio description format G723 for ID 4
Found audio description format telephone-event for ID 97
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 202.0.179.3:19240
Looking for 8523501XXXX in DID_HKBN_PSTN (domain MY_VOIP_SERVER:5060)
list_route: hop: <sip:6XXXXXX@202.0.179.3:5060;user=phone>
<--- Transmitting (no NAT) to 202.0.179.3:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 202.0.179.3:5060;branch=z9hG4bKbff83176c;received=202.0.179.3
From: <sip:6XXXXXX@202.0.179.3;user=phone>;tag=f019ff19
To: <sip:8523501XXXX@MY_VOIP_SERVER;user=phone>
Call-ID: 001c8aecb8f9a834f43c8cf74dfd9531@sx3000
CSeq: 1 INVITE
Server: Linksys/SPA3102-5.1.10(GW)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:8523501XXXX@MY_VOIP_SERVER:5060>
Content-Length: 0
<------------>
-- Executing [8523501XXXX@DID_HKBN_PSTN:1] NoOp("SIP/COMNET_PSTN-00000017", "Calling from "" <6XXXXXX> 6XXXXXX ") in new stack
-- Executing [8523501XXXX@DID_HKBN_PSTN:2] Goto("SIP/COMNET_PSTN-00000017", "DID_Default_Incoming,s,1") in new stack
-- Goto (DID_Default_Incoming,s,1)
-- Executing [s@DID_Default_Incoming:1] NoOp("SIP/COMNET_PSTN-00000017", "Greeting - Caller ID: 6XXXXXX") in new stack
-- Executing [s@DID_Default_Incoming:2] Set("SIP/COMNET_PSTN-00000017", "CALLERID(name)=6XXXXXX") in new stack
-- Executing [s@DID_Default_Incoming:3] Set("SIP/COMNET_PSTN-00000017", "VOLUME(TX)=3") in new stack
-- Executing [s@DID_Default_Incoming:4] Set("SIP/COMNET_PSTN-00000017", "VOLUME(RX)=3") in new stack
-- Executing [s@DID_Default_Incoming:5] Set("SIP/COMNET_PSTN-00000017", "TIMEOUT(response)=10") in new stack
-- Response timeout set to 10.000
-- Executing [s@DID_Default_Incoming:6] Set("SIP/COMNET_PSTN-00000017", "CHANNEL(musicclass)=default") in new stack
-- Executing [s@DID_Default_Incoming:7] Ringing("SIP/COMNET_PSTN-00000017", "") in new stack
<--- Transmitting (no NAT) to 202.0.179.3:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 202.0.179.3:5060;branch=z9hG4bKbff83176c;received=202.0.179.3
From: <sip:6XXXXXX@202.0.179.3;user=phone>;tag=f019ff19
To: <sip:8523501XXXX@MY_VOIP_SERVER;user=phone>;tag=as4b3b89ee
Call-ID: 001c8aecb8f9a834f43c8cf74dfd9531@sx3000
CSeq: 1 INVITE
Server: Linksys/SPA3102-5.1.10(GW)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:8523501XXXX@MY_VOIP_SERVER:5060>
Content-Length: 0
<------------>
-- Executing [s@DID_Default_Incoming:8] Gosub("SIP/COMNET_PSTN-00000017", "PSTN_CheckCallBackNumber,s,1") in new stack |
發表於 2011-11-6 14:17
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