rendered paste body*CLI> sip debug ip xxx.yyy.89.135
SIP Debugging Enabled for IP: xxx.yyy.89.135
The 'sip debug' command is deprecated and will be removed in a future release. Please use 'sip set debug' instead.
*CLI>
<--- SIP read from xxx.yyy.89.135:1024 --->
INVITE sip:5192922001@xxx.yyy.89.134 SIP/2.0
Via: SIP/2.0/UDP xxx.yyy.89.135:5060;branch=z9hG4bK632edb90;rport
Max-Forwards: 70
From: "MR ME" <sip:1112223456@xxx.yyy.89.135>;tag=as7e9d839a
To: <sip:5192922001@xxx.yyy.89.134>
Contact: <sip:1112223456@xxx.yyy.89.135>
Call-ID: 50332f3d496972bd1434da9c0d99b7a3@xxx.yyy.89.135
CSeq: 102 INVITE
User-Agent: Asterisk PBX SVN-trunk-r67164
Date: Fri, 22 Jun 2007 00:10:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 251
v=0
o=root 1577940051 1577940051 IN IP4 xxx.yyy.89.135
s=session
c=IN IP4 xxx.yyy.89.135
t=0 0
m=audio 11670 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 12 lines) ---
Sending to xxx.yyy.89.135 : 1024 (NAT)
Using INVITE request as basis request - 50332f3d496972bd1434da9c0d99b7a3@xxx.yyy.89.135
Found no matching peer or user for 'xxx.yyy.89.135:1024'
Found RTP audio format 3
Found RTP audio format 101
Peer audio RTP is at port xxx.yyy.89.135:11670
Found description format GSM for ID 3
Found description format telephone-event for ID 101
Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x2 (gsm)/video=0x0 (nothing), combined - 0x2 (gsm)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port xxx.yyy.89.135:11670
Looking for 5192922001 in default (domain xxx.yyy.89.134)
list_route: hop: <sip:1112223456@xxx.yyy.89.135>
<--- Transmitting (NAT) to xxx.yyy.89.135:1024 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP xxx.yyy.89.135:5060;branch=z9hG4bK632edb90;received=xxx.yyy.89.135;rport=1024
From: "MR ME" <sip:1112223456@xxx.yyy.89.135>;tag=as7e9d839a
To: <sip:5192922001@xxx.yyy.89.134>
Call-ID: 50332f3d496972bd1434da9c0d99b7a3@xxx.yyy.89.135
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:5192922001@xxx.yyy.89.134>
Content-Length: 0
<------------>
-- Executing [5192922001@default:1] NoOp("SIP/xxx.yyy.89.135-082be860", "default context hit "5192922001"| this is wrong!") in new stack
-- Executing [5192922001@default:2] Hangup("SIP/xxx.yyy.89.135-082be860", "") in new stack
== Spawn extension (default, 5192922001, 2) exited non-zero on 'SIP/xxx.yyy.89.135-082be860'
-- Executing [h@default:1] NoOp("SIP/xxx.yyy.89.135-082be860", "default context hit h exten| this is wrong!") in new stack
Scheduling destruction of SIP dialog '50332f3d496972bd1434da9c0d99b7a3@xxx.yyy.89.135' in 32000 ms (Method: INVITE)
<--- Reliably Transmitting (NAT) to xxx.yyy.89.135:1024 --->
SIP/2.0 603 Declined
Via: SIP/2.0/UDP xxx.yyy.89.135:5060;branch=z9hG4bK632edb90;received=xxx.yyy.89.135;rport=1024
From: "MR ME" <sip:1112223456@xxx.yyy.89.135>;tag=as7e9d839a
To: <sip:5192922001@xxx.yyy.89.134>;tag=as04a4fedb
Call-ID: 50332f3d496972bd1434da9c0d99b7a3@xxx.yyy.89.135
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:5192922001@xxx.yyy.89.134>
Content-Length: 0
<------------>
<--- SIP read from xxx.yyy.89.135:1024 --->
ACK sip:5192922001@xxx.yyy.89.134 SIP/2.0
Via: SIP/2.0/UDP xxx.yyy.89.135:5060;branch=z9hG4bK632edb90;rport
Max-Forwards: 70
From: "MR ME" <sip:1112223456@xxx.yyy.89.135>;tag=as7e9d839a
To: <sip:5192922001@xxx.yyy.89.134>;tag=as04a4fedb
Contact: <sip:1112223456@xxx.yyy.89.135>
Call-ID: 50332f3d496972bd1434da9c0d99b7a3@xxx.yyy.89.135
CSeq: 102 ACK
User-Agent: Asterisk PBX SVN-trunk-r67164
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
sip no denbugReliably Transmitting (no NAT) to xxx.yyy.89.135:5060:
OPTIONS sip:xxx.yyy.89.135 SIP/2.0
Via: SIP/2.0/UDP xxx.yyy.89.134:5060;branch=z9hG4bK7936ee6e;rport
From: "asterisk" <sip:asterisk@xxx.yyy.89.134>;tag=as166183ee
To: <sip:xxx.yyy.89.135>
Contact: <sip:asterisk@xxx.yyy.89.134>
Call-ID: 5360daef0022147d1a3008e660f78df7@xxx.yyy.89.134
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 22 Jun 2007 00:04:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
---
<--- SIP read from xxx.yyy.89.135:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxx.yyy.89.134:5060;branch=z9hG4bK7936ee6e;rport
From: "asterisk" <sip:asterisk@xxx.yyy.89.134>;tag=as166183ee
To: <sip:xxx.yyy.89.135>;tag=B77C89E0-228A4B1F
CSeq: 102 OPTIONS
Call-ID: 5360daef0022147d1a3008e660f78df7@xxx.yyy.89.134
Contact: <sip:843@192.168.77.146>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.7.0098
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '5360daef0022147d1a3008e660f78df7@xxx.yyy.89.134' Method: OPTIONS Reliably Transmitting (NAT) to xxx.yyy.89.135:5060:
OPTIONS sip:843@192.168.77.146 SIP/2.0
Via: SIP/2.0/UDP xxx.yyy.89.134:5060;branch=z9hG4bK199022ab;rport
From: "asterisk" <sip:asterisk@xxx.yyy.89.134>;tag=as756087c7
To: <sip:843@192.168.77.146>
Contact: <sip:asterisk@xxx.yyy.89.134>
Call-ID: 42a8d8b820b459b153fb4f4e72fcbc08@xxx.yyy.89.134
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 22 Jun 2007 00:04:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
---
<--- SIP read from xxx.yyy.89.135:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxx.yyy.89.134:5060;branch=z9hG4bK199022ab;rport
From: "asterisk" <sip:asterisk@xxx.yyy.89.134>;tag=as756087c7
To: <sip:843@192.168.77.146>;tag=2FA17F4B-A0805606
CSeq: 102 OPTIONS
Call-ID: 42a8d8b820b459b153fb4f4e72fcbc08@xxx.yyy.89.134
Contact: <sip:843@192.168.77.146>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.7.0098
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '42a8d8b820b459b153fb4f4e72fcbc08@xxx.yyy.89.134' Method: OPTIONS sip no debug
SIP Debugging Disabled
The 'sip no debug' command is deprecated and will be removed in a future release. Please use 'sip set debug off' instead.
*CLI>
Machine A (the one placing the call)
[machineB]
type=peer
host=xxx.yyy.89.134
deny=0.0.0.0/0
permit=xxx.yyy.89.134/32
dtmfmode=rfc2833
context=incoming
disallow=all
allow=gsm
nat=no
canreinvite=no
qualify=1000
Machine B (the one receiving and putting the call in [default] instead of where it should be:
[machineA]
type=peer
host=xxx.yyy.89.135
deny=0.0.0.0/0
permit=xxx.yyy.89.135/32
context=from_andrew
qualify=500
qualifysmoothing=yes
disallow=all
allow=gsm
dtmfmode=rfc2833
canreinvite=no
nat=no