All pastes #66244 Raw Edit

Someone

public text v1 · immutable
#66244 ·published 2006-06-16 17:29 UTC
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Below is the output of CLI with sip debug on while I call one extension to another and do NOT answer and wait for the NOANSWER voice mail to startup.  The last 20 lines are probably the most significant. The others are there incase they are useful for your analysis.
Rich
----------------------
bx*CLI>
pbx*CLI>
pbx*CLI>
pbx*CLI>
pbx*CLI>
pbx*CLI>
pbx*CLI>
pbx*CLI>
pbx*CLI>
pbx*CLI>
pbx*CLI> sip debug
SIP Debugging enabled
12 headers, 0 lines
Reliably Transmitting (no NAT) to 192.168.101.151:5061:
OPTIONS sip:192.168.101.151:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.101.1:5060;branch=z9hG4bK07aa426b;rport
From: "asterisk" <sip:asterisk@192.168.101.1>;tag=as0147bf2d
To: <sip:192.168.101.151:5061>
Contact: <sip:asterisk@192.168.101.1>
Call-ID: 6aa0277b3fa713391ba8f7b01f64f874@192.168.101.1
CSeq: 102 OPTIONS
User-Agent: figero 2006
Max-Forwards: 70
Date: Fri, 16 Jun 2006 17:23:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
pbx*CLI>
<-- SIP read from 192.168.101.151:5061:
SIP/2.0 200 OK
To: <sip:192.168.101.151:5061>;tag=9413b93b2fe8d347i1
From: "asterisk" <sip:asterisk@192.168.101.1>;tag=as0147bf2d
Call-ID: 6aa0277b3fa713391ba8f7b01f64f874@192.168.101.1
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.101.1:5060;branch=z9hG4bK07aa426b
Server: Sipura/SPA3000-2.0.13(GWg)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura


--- (10 headers 0 lines)---
Destroying call '6aa0277b3fa713391ba8f7b01f64f874@192.168.101.1'
Jun 16 17:23:49 NOTICE[1468]: chan_sip.c:5237 sip_reregister:    -- Re-registration for  3145630068@sunrocket.com
REGISTER 12 headers, 0 lines
Reliably Transmitting (no NAT) to 67.133.234.125:5060:
REGISTER sip:sunrocket.com SIP/2.0
Via: SIP/2.0/UDP 68.38.105.1:5060;branch=z9hG4bK114f5304;rport
From: <sip:3145630068@sunrocket.com>;tag=as110fdf4d
To: <sip:3145630068@sunrocket.com>
Call-ID: 2ed766e550b99fec451eb7d26ace0af1@192.168.101.1
CSeq: 110 REGISTER
User-Agent: figero 2006
Max-Forwards: 70
xpires: 120
Contact: <sip:3145630068@68.38.105.1>
Event: registration
Content-Length: 0


---
pbx*CLI>
<-- SIP read from 67.133.234.125:5060:
SIP/2.0 200 OK
v:SIP/2.0/UDP 68.38.105.1:5060;branch=z9hG4bK114f5304;rport
f:<sip:3145630068@sunrocket.com>;tag=as110fdf4d
t:<sip:3145630068@sunrocket.com>
i:2ed766e550b99fec451eb7d26ace0af1@192.168.101.1
CSeq:110 REGISTER
User-Agent: figero 2006
m:<sip:3145630068@68.38.105.1>;expires=90
Event: registration
l:0


--- (10 headers 0 lines)---
Scheduling destruction of call '2ed766e550b99fec451eb7d26ace0af1@192.168.101.1' in 32000 ms
Jun 16 17:23:49 NOTICE[1468]: chan_sip.c:9661 handle_response_register: Outbound Registration: Expiry for sunrocket.com is 90 sec (Scheduling reregistration in 75 s)
Jun 16 17:23:49 NOTICE[1468]: chan_sip.c:5237 sip_reregister:    -- Re-registration for  2679661066@sunrocket.com
REGISTER 12 headers, 0 lines
Reliably Transmitting (no NAT) to 67.133.234.125:5060:
REGISTER sip:sunrocket.com SIP/2.0
Via: SIP/2.0/UDP 68.38.105.1:5060;branch=z9hG4bK59721511;rport
From: <sip:2679661066@sunrocket.com>;tag=as52781b46
To: <sip:2679661066@sunrocket.com>
Call-ID: 1eef83ea76f0938d5131eadf22d6db7d@192.168.101.1
CSeq: 110 REGISTER
User-Agent: figero 2006
Max-Forwards: 70
xpires: 120
Contact: <sip:2679661066@68.38.105.1>
Event: registration
Content-Length: 0


---
pbx*CLI>
<-- SIP read from 67.133.234.125:5060:
SIP/2.0 200 OK
v:SIP/2.0/UDP 68.38.105.1:5060;branch=z9hG4bK59721511;rport
f:<sip:2679661066@sunrocket.com>;tag=as52781b46
t:<sip:2679661066@sunrocket.com>
i:1eef83ea76f0938d5131eadf22d6db7d@192.168.101.1
CSeq:110 REGISTER
User-Agent: figero 2006
m:<sip:2679661066@68.38.105.1>;expires=90
Event: registration
l:0


--- (10 headers 0 lines)---
Scheduling destruction of call '1eef83ea76f0938d5131eadf22d6db7d@192.168.101.1' in 32000 ms
Jun 16 17:23:49 NOTICE[1468]: chan_sip.c:9661 handle_response_register: Outbound Registration: Expiry for sunrocket.com is 90 sec (Scheduling reregistration in 75 s)
Jun 16 17:23:49 NOTICE[1468]: chan_sip.c:5237 sip_reregister:    -- Re-registration for  08122966@sip.stanaphone.com
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 204.147.183.18:5060:
REGISTER sip:sip.stanaphone.com SIP/2.0
Via: SIP/2.0/UDP 68.38.105.1:5060;branch=z9hG4bK73c68cb0;rport
From: <sip:08122966@sip.stanaphone.com>;tag=as62b04a51
To: <sip:08122966@sip.stanaphone.com>
Call-ID: 28b9be0b53b29f866dd04fc34d3a85a4@192.168.101.1
CSeq: 109 REGISTER
User-Agent: figero 2006
Max-Forwards: 70
Authorization: Digest username="08122966", realm="sip.stanaphone.com", algorithm=MD5, uri="sip:sip.stanaphone.com", nonce="4492e7f9cf435d9929fb83d3d36a54520c5b635f", response="b847801cedbda809df8d3f80b456b515", opaque=""
Expires: 120
Contact: <sip:3475584326@68.38.105.1>
Event: registration
Content-Length: 0


---
pbx*CLI>
<-- SIP read from 204.147.183.18:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 68.38.105.1:5060;branch=z9hG4bK73c68cb0;rport=5060
From: <sip:08122966@sip.stanaphone.com>;tag=as62b04a51
To: <sip:08122966@sip.stanaphone.com>;tag=4e487ad2f2fae333606267674b3a79b1.c48e
Call-ID: 28b9be0b53b29f866dd04fc34d3a85a4@192.168.101.1
CSeq: 109 REGISTER
PortaBilling: available-funds:0.00 currency:USD
Contact: <sip:3475584326@68.38.105.1>;expires=115
Server: Sip EXpress router (0.9.5-pre1 (i386/linux))
Content-Length: 0
Warning: 392 204.147.183.18:5060 "Noisy feedback tells:  pid=21601 req_src_ip=68.38.105.1 req_src_port=5060 in_uri=sip:sip.stanaphone.com out_uri=sip:sip.stanaphone.com via_cnt==1"


--- (11 headers 0 lines)---
Scheduling destruction of call '28b9be0b53b29f866dd04fc34d3a85a4@192.168.101.1' in 32000 ms
Jun 16 17:23:49 NOTICE[1468]: chan_sip.c:9661 handle_response_register: Outbound Registration: Expiry for sip.stanaphone.com is 115 sec (Scheduling reregistration in 100 s)
12 headers, 3 lines
Reliably Transmitting (no NAT) to 192.168.101.135:5061:
NOTIFY sip:1025@192.168.101.135:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.101.1:5060;branch=z9hG4bK2a764dc8;rport
From: "asterisk" <sip:asterisk@192.168.101.1>;tag=as21af82df
To: <sip:1025@192.168.101.135:5061>
Contact: <sip:asterisk@192.168.101.1>
Call-ID: 2990a2ac35689a3219a06ccd2cc64825@192.168.101.1
CSeq: 102 NOTIFY
User-Agent: figero 2006
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 80

Messages-Waiting: no
Message-Account: sip:asterisk@
Voice-Message: 0/0 (0/0)

---
Scheduling destruction of call '2990a2ac35689a3219a06ccd2cc64825@192.168.101.1' in 15000 ms
12 headers, 3 lines
Reliably Transmitting (no NAT) to 192.168.101.135:5060:
NOTIFY sip:1024@192.168.101.135:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.101.1:5060;branch=z9hG4bK4817e49f;rport
From: "asterisk" <sip:asterisk@192.168.101.1>;tag=as0b78daca
To: <sip:1024@192.168.101.135:5060>
Contact: <sip:asterisk@192.168.101.1>
Call-ID: 5c1a9e52528c63761d32530c680b7b54@192.168.101.1
CSeq: 102 NOTIFY
User-Agent: figero 2006
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 80

Messages-Waiting: no
Message-Account: sip:asterisk@
Voice-Message: 0/0 (0/0)

---
Scheduling destruction of call '5c1a9e52528c63761d32530c680b7b54@192.168.101.1' in 15000 ms
pbx*CLI>
<-- SIP read from 192.168.101.135:5060:
SIP/2.0 200 OK
Call-ID: 5c1a9e52528c63761d32530c680b7b54@192.168.101.1
From: asterisk<sip:asterisk@192.168.101.1>;tag=as0b78daca
To: <sip:1024@192.168.101.135:5060>
CSeq: 102 NOTIFY
Via: SIP/2.0/UDP 192.168.101.1:5060;branch=z9hG4bK4817e49f
Content-Length: 0


--- (7 headers 0 lines)---
Destroying call '5c1a9e52528c63761d32530c680b7b54@192.168.101.1'
pbx*CLI>
<-- SIP read from 192.168.101.135:5061:
SIP/2.0 200 OK
Call-ID: 2990a2ac35689a3219a06ccd2cc64825@192.168.101.1
From: asterisk<sip:asterisk@192.168.101.1>;tag=as21af82df
To: <sip:1025@192.168.101.135:5061>
CSeq: 102 NOTIFY
Via: SIP/2.0/UDP 192.168.101.1:5060;branch=z9hG4bK2a764dc8
Content-Length: 0


--- (7 headers 0 lines)---
Destroying call '2990a2ac35689a3219a06ccd2cc64825@192.168.101.1'
pbx*CLI>
<-- SIP read from 192.168.101.135:5060:
INVITE sip:1023@asterisk SIP/2.0
Call-ID: abc4dad49d-11902221-62be4-c0a8794b@asterisk
From: 1024<sip:1024@asterisk>;tag=36afc-29eeb4
To: <sip:1023@asterisk>
CSeq: 100 INVITE
Via: SIP/2.0/UDP 192.168.101.135:5060
Contact: 1024<sip:1024@192.168.101.135:5060>
Max-Forwards: 70
User-Agent: Telco Systems AC-211 v4.57.41
Supported: timer,replaces
Allow: REFER, UPDATE, INFO, OPTIONS, ACK, INVITE, BYE, CANCEL, NOTIFY
Expires: 9999
Content-Type: application/sdp
Content-Length: 131

v=0
o=1024 2748083 2748083 IN IP4 192.168.101.135
s=-
c=IN IP4 192.168.101.135
t=0 0
m=audio 16384 RTP/AVP 2 0 8
a=ptime:30

--- (14 headers 7 lines)---
Using INVITE request as basis request - abc4dad49d-11902221-62be4-c0a8794b@asterisk
Sending to 192.168.101.135 : 5060 (non-NAT)
Reliably Transmitting (no NAT) to 192.168.101.135:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.101.135:5060;received=192.168.101.135
From: 1024<sip:1024@asterisk>;tag=36afc-29eeb4
To: <sip:1023@asterisk>;tag=as5fa2abd4
Call-ID: abc4dad49d-11902221-62be4-c0a8794b@asterisk
CSeq: 100 INVITE
User-Agent: figero 2006
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:1023@192.168.101.1>
Proxy-Authenticate: Digest realm="asterisk", nonce="75cd5a0e"
Content-Length: 0


---
Scheduling destruction of call 'abc4dad49d-11902221-62be4-c0a8794b@asterisk' in 15000 ms
Found user '1024'
pbx*CLI>
<-- SIP read from 192.168.101.135:5060:
ACK sip:1023@asterisk SIP/2.0
Call-ID: abc4dad49d-11902221-62be4-c0a8794b@asterisk
From: 1024<sip:1024@asterisk>;tag=36afc-29eeb4
To: <sip:1023@asterisk>;tag=as5fa2abd4
CSeq: 100 ACK
Via: SIP/2.0/UDP 192.168.101.135:5060
Max-Forwards: 70
Content-Length: 0


--- (8 headers 0 lines)---
pbx*CLI>
<-- SIP read from 192.168.101.135:5060:
INVITE sip:1023@asterisk SIP/2.0
Call-ID: abc4dad49d-11902221-62be4-c0a8794b@asterisk
From: 1024<sip:1024@asterisk>;tag=36afc-29eeb4
To: <sip:1023@asterisk>
CSeq: 101 INVITE
Via: SIP/2.0/UDP 192.168.101.135:5060
Contact: 1024<sip:1024@192.168.101.135:5060>
Max-Forwards: 70
User-Agent: Telco Systems AC-211 v4.57.41
Supported: timer,replaces
Allow: REFER, UPDATE, INFO, OPTIONS, ACK, INVITE, BYE, CANCEL, NOTIFY
Expires: 9999
Proxy-Authorization: Digest username="1024",realm="asterisk",uri="sip:1023@asterisk",response="e6856c469c373086e52156f4a06ede5b",nonce="75cd5a0e"
Content-Type: application/sdp
Content-Length: 131

v=0
o=1024 2748083 2748083 IN IP4 192.168.101.135
s=-
c=IN IP4 192.168.101.135
t=0 0
m=audio 16384 RTP/AVP 2 0 8
a=ptime:30

--- (15 headers 7 lines)---
Using INVITE request as basis request - abc4dad49d-11902221-62be4-c0a8794b@asterisk
Sending to 192.168.101.135 : 5060 (non-NAT)
Found user '1024'
Found RTP audio format 2
Found RTP audio format 0
Found RTP audio format 8
Peer audio RTP is at port 192.168.101.135:16384
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Looking for 1023 in home (domain asterisk)
list_route: hop: <sip:1024@192.168.101.135:5060>
Transmitting (no NAT) to 192.168.101.135:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.101.135:5060;received=192.168.101.135
From: 1024<sip:1024@asterisk>;tag=36afc-29eeb4
To: <sip:1023@asterisk>
Call-ID: abc4dad49d-11902221-62be4-c0a8794b@asterisk
CSeq: 101 INVITE
User-Agent: figero 2006
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:1023@192.168.101.1>
Content-Length: 0


---
    -- Executing Macro("SIP/1024-e33e", "voicemail|SIP/1023") in new stack
    -- Executing Dial("SIP/1024-e33e", "SIP/1023|20") in new stack
We're at 192.168.101.1 port 10680
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 12 lines
Reliably Transmitting (no NAT) to 192.168.101.151:5060:
INVITE sip:1023@192.168.101.151:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.101.1:5060;branch=z9hG4bK6a21bdb8;rport
From: "AC211-1" <sip:1024@192.168.101.1>;tag=as324fb104
To: <sip:1023@192.168.101.151:5060>
Contact: <sip:1024@192.168.101.1>
Call-ID: 4c2b857814eef94546818b9c2d4f50c1@192.168.101.1
CSeq: 102 INVITE
User-Agent: figero 2006
Max-Forwards: 70
Date: Fri, 16 Jun 2006 17:23:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 2594 2594 IN IP4 192.168.101.1
s=session
c=IN IP4 192.168.101.1
t=0 0
m=audio 10680 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- Called 1023
pbx*CLI>
<-- SIP read from 192.168.101.151:5060:
SIP/2.0 100 Trying
To: <sip:1023@192.168.101.151:5060>
From: "AC211-1" <sip:1024@192.168.101.1>;tag=as324fb104
Call-ID: 4c2b857814eef94546818b9c2d4f50c1@192.168.101.1
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.101.1:5060;branch=z9hG4bK6a21bdb8
Server: Sipura/SPA3000-2.0.13(GWg)
Content-Length: 0


--- (8 headers 0 lines)---
pbx*CLI>
<-- SIP read from 192.168.101.151:5060:
SIP/2.0 180 Ringing
To: <sip:1023@192.168.101.151:5060>;tag=2c35d737b3d0dfd9i0
From: "AC211-1" <sip:1024@192.168.101.1>;tag=as324fb104
Call-ID: 4c2b857814eef94546818b9c2d4f50c1@192.168.101.1
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.101.1:5060;branch=z9hG4bK6a21bdb8
Server: Sipura/SPA3000-2.0.13(GWg)
Content-Length: 0


--- (8 headers 0 lines)---
    -- SIP/1023-c56f is ringing
Transmitting (no NAT) to 192.168.101.135:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.101.135:5060;received=192.168.101.135
From: 1024<sip:1024@asterisk>;tag=36afc-29eeb4
To: <sip:1023@asterisk>;tag=as11fd6592
Call-ID: abc4dad49d-11902221-62be4-c0a8794b@asterisk
CSeq: 101 INVITE
User-Agent: figero 2006
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:1023@192.168.101.1>
Content-Length: 0


---
Destroying call '518b57c99-2e09bd50-70310-c0a8794b@asterisk'
Destroying call '53258c5ec-86c96e4-8163c-c0a8794c@asterisk'
    -- Nobody picked up in 20000 ms
Reliably Transmitting (no NAT) to 192.168.101.151:5060:
CANCEL sip:1023@192.168.101.151:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.101.1:5060;branch=z9hG4bK6a21bdb8;rport
From: "AC211-1" <sip:1024@192.168.101.1>;tag=as324fb104
To: <sip:1023@192.168.101.151:5060>
Contact: <sip:1024@192.168.101.1>
Call-ID: 4c2b857814eef94546818b9c2d4f50c1@192.168.101.1
CSeq: 102 CANCEL
User-Agent: figero 2006
Max-Forwards: 70
Content-Length: 0


---
Scheduling destruction of call '4c2b857814eef94546818b9c2d4f50c1@192.168.101.1' in 15000 ms
    -- Executing Goto("SIP/1024-e33e", "s-NOANSWER|1") in new stack
    -- Goto (macro-voicemail,s-NOANSWER,1)
    -- Executing VoiceMail("SIP/1024-e33e", "u1023") in new stack
We're at 192.168.101.1 port 12222
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Reliably Transmitting (no NAT) to 192.168.101.135:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.101.135:5060;received=192.168.101.135
From: 1024<sip:1024@asterisk>;tag=36afc-29eeb4
To: <sip:1023@asterisk>;tag=as11fd6592
Call-ID: abc4dad49d-11902221-62be4-c0a8794b@asterisk
CSeq: 101 INVITE
User-Agent: figero 2006
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:1023@192.168.101.1>
Content-Type: application/sdp
Content-Length: 207

v=0
o=root 2594 2594 IN IP4 192.168.101.1
s=session
c=IN IP4 192.168.101.1
t=0 0
m=audio 12222 RTP/AVP 0 8 3
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=silenceSupp:off - - - -

---
pbx*CLI>
Disconnected from Asterisk server
Executing last minute cleanups
pbx root # 
============= machine is dead now =============