All pastes #94404 Raw Edit

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public text v1 · immutable
#94404 ·published 2006-07-21 07:30 UTC
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sip debug
SIP Debugging enabled
    -- Starting simple switch on 'Zap/3-1'

<-- SIP read from 10.10.10.105:5060: 
NOTIFY sip:10.10.10.10 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.105:5060;branch=z9hG4bK-ba15c279
From: Sara Hass <sip:11011@10.10.10.10>;tag=2512817a18f3124co0
To: <sip:10.10.10.10>
Call-ID: d741e1ba-38712f0c@10.10.10.105
CSeq: 2319 NOTIFY
Max-Forwards: 70
Event: keep-alive
User-Agent: Sipura/SPA841-3.1.2(d)
Content-Length: 0


--- (10 headers 0 lines)---
Transmitting (no NAT) to 10.10.10.105:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.105:5060;branch=z9hG4bK-ba15c279;received=10.10.10.105
From: Sara Hass <sip:11011@10.10.10.10>;tag=2512817a18f3124co0
To: <sip:10.10.10.10>;tag=as6bf18ef3
Call-ID: d741e1ba-38712f0c@10.10.10.105
CSeq: 2319 NOTIFY
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
Destroying call 'd741e1ba-38712f0c@10.10.10.105'
    -- Executing Macro("Zap/3-1", "stdexten|10|11010")
    -- Executing Dial("Zap/3-1", "SIP/11010|20|trWw") in new stack
    -- SIP Seeding peer from astdb: '11010' at 11010@10.10.10.101:5060 for 3600
We're at 10.10.10.10 port 13354
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 11 lines
Reliably Transmitting (no NAT) to 10.10.10.101:5060:
INVITE sip:11010@10.10.10.101 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.10:5060;branch=z9hG4bK7f34e2f5;rport
From: "asterisk" <sip:asterisk@10.10.10.10>;tag=as185605f3
To: <sip:11010@10.10.10.101>
Contact: <sip:asterisk@10.10.10.10>
Call-ID: 475efa1d3aa20887064887886e6a278b@10.10.10.10
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 21 Jul 2006 07:30:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 16099 16099 IN IP4 10.10.10.10
s=session
c=IN IP4 10.10.10.10
t=0 0
m=audio 13354 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- Called 11010

<-- SIP read from 10.10.10.101:5060: 
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.10.10:5060;branch=z9hG4bK7f34e2f5;rport
From: "asterisk" <sip:asterisk@10.10.10.10>;tag=as185605f3
To: <sip:11010@10.10.10.101>;tag=9AB14489-19E0390E
CSeq: 102 INVITE
Call-ID: 475efa1d3aa20887064887886e6a278b@10.10.10.10
Contact: <sip:11010@10.10.10.101>
User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.6.0036
Content-Length: 0


--- (9 headers 0 lines)---

<-- SIP read from 10.10.10.101:5060: 
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.10.10.10:5060;branch=z9hG4bK7f34e2f5;rport
From: "asterisk" <sip:asterisk@10.10.10.10>;tag=as185605f3
To: <sip:11010@10.10.10.101>;tag=9AB14489-19E0390E
CSeq: 102 INVITE
Call-ID: 475efa1d3aa20887064887886e6a278b@10.10.10.10
Contact: <sip:11010@10.10.10.101>
User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.6.0036
Allow-Events: talk,hold,conference
Content-Length: 0


--- (10 headers 0 lines)---
    -- SIP/11010-007ae680 is ringing
    -- SIP Seeding peer from astdb: '11010' at 11010@10.10.10.101:5060 for 3600
    -- SIP Seeding peer from astdb: '11010' at 11010@10.10.10.101:5060 for 3600

<-- SIP read from 10.10.10.101:5060: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.10:5060;branch=z9hG4bK7f34e2f5;rport
From: "asterisk" <sip:asterisk@10.10.10.10>;tag=as185605f3
To: <sip:11010@10.10.10.101>;tag=9AB14489-19E0390E
CSeq: 102 INVITE
Call-ID: 475efa1d3aa20887064887886e6a278b@10.10.10.10
Contact: <sip:11010@10.10.10.101>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.6.0036
Content-Type: application/sdp
Content-Length: 185

v=0
o=- 978310424 978310424 IN IP4 10.10.10.101
s=Polycom IP Phone
c=IN IP4 10.10.10.101
t=0 0
m=audio 2264 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000

--- (11 headers 8 lines)---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 10.10.10.101:2264
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:11010@10.10.10.101>
set_destination: Parsing <sip:11010@10.10.10.101> for address/port to send to
set_destination: set destination to 10.10.10.101, port 5060
Transmitting (no NAT) to 10.10.10.101:5060:
ACK sip:11010@10.10.10.101 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.10:5060;branch=z9hG4bK1f1edf5c;rport
From: "asterisk" <sip:asterisk@10.10.10.10>;tag=as185605f3
To: <sip:11010@10.10.10.101>;tag=9AB14489-19E0390E
Contact: <sip:asterisk@10.10.10.10>
Call-ID: 475efa1d3aa20887064887886e6a278b@10.10.10.10
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
    -- SIP/11010-007ae680 answered Zap/3-1
    -- SIP Seeding peer from astdb: '11010' at 11010@10.10.10.101:5060 for 3600
    -- SIP Seeding peer from astdb: '11010' at 11010@10.10.10.101:5060 for 3600

<-- SIP read from 10.10.10.101:5060: 
BYE sip:asterisk@10.10.10.10 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.101;branch=z9hG4bK2b2fea1c9DA71C87
From: <sip:11010@10.10.10.101>;tag=9AB14489-19E0390E
To: "asterisk" <sip:asterisk@10.10.10.10>;tag=as185605f3
CSeq: 1 BYE
Call-ID: 475efa1d3aa20887064887886e6a278b@10.10.10.10
Contact: <sip:11010@10.10.10.101>
User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.6.0036
Max-Forwards: 70
Content-Length: 0


--- (10 headers 0 lines)---
Sending to 10.10.10.101 : 5060 (non-NAT)
Transmitting (no NAT) to 10.10.10.101:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.101;branch=z9hG4bK2b2fea1c9DA71C87;received=10.10.10.101
From: <sip:11010@10.10.10.101>;tag=9AB14489-19E0390E
To: "asterisk" <sip:asterisk@10.10.10.10>;tag=as185605f3
Call-ID: 475efa1d3aa20887064887886e6a278b@10.10.10.10
CSeq: 1 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:asterisk@10.10.10.10>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing


---
  == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'Zap/3-1' in macro 'stdexten'
  == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'Zap/3-1'
    -- SIP Seeding peer from astdb: '11010' at 11010@10.10.10.101:5060 for 3600
    -- SIP Seeding peer from astdb: '11010' at 11010@10.10.10.101:5060 for 3600
    -- Hungup 'Zap/3-1'
Jul 21 03:30:46 NOTICE[16111]: chan_sip.c:5336 sip_reregister:    -- Re-registration for  7322761242@sip.sipmedia.com
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 69.1.236.33:5060:
REGISTER sip:sip.sipmedia.com SIP/2.0
Via: SIP/2.0/UDP 192.168.2.125:5060;branch=z9hG4bK462d159f;rport
From: <sip:7322761242@sip.sipmedia.com>;tag=as3062afb7
To: <sip:7322761242@sip.sipmedia.com>
Call-ID: 7e1ee1647f15a5566ec2f4857f56660b@192.168.2.125
CSeq: 106 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="7322761242", realm="sip1.xchangetele.com", algorithm=MD5, uri="sip:sip.sipmedia.com", nonce="44c088066a22ad6ad167418dac1706be8251e333", response="9e7f28f4592e8d667e357448ff34da7d", opaque=""
Expires: 120
Contact: <sip:7322761242@192.168.2.125>
Event: registration
Content-Length: 0


---
Destroying call '475efa1d3aa20887064887886e6a278b@10.10.10.10'

<-- SIP read from 69.1.236.33:5060: 
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.2.125:5060;branch=z9hG4bK462d159f;rport=65415
From: <sip:7322761242@sip.sipmedia.com>;tag=as3062afb7
To: <sip:7322761242@sip.sipmedia.com>;tag=91791a7492f328483c7efb39681bf0a4.241c
Call-ID: 7e1ee1647f15a5566ec2f4857f56660b@192.168.2.125
CSeq: 106 REGISTER
WWW-Authenticate: Digest realm="sip1.xchangetele.com", nonce="44c0894155ef19a0f40546e7be36909fd1ef3df8", stale=true
Server: Platform Experts BBT (Version 2.1a)
Content-Length: 0
Warning: 392 69.1.236.33:5060 "Noisy feedback tells:  pid=7913 req_src_ip=72.68.108.196 req_src_port=65415 in_uri=sip:sip.sipmedia.com out_uri=sip:sip.sipmedia.com via_cnt==1"


--- (10 headers 0 lines)---
Responding to challenge, registration to domain/host name sip.sipmedia.com
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 69.1.236.33:5060:
REGISTER sip:sip.sipmedia.com SIP/2.0
Via: SIP/2.0/UDP 192.168.2.125:5060;branch=z9hG4bK79724411;rport
From: <sip:7322761242@sip.sipmedia.com>;tag=as7b899a4d
To: <sip:7322761242@sip.sipmedia.com>
Call-ID: 7e1ee1647f15a5566ec2f4857f56660b@192.168.2.125
CSeq: 107 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="7322761242", realm="sip1.xchangetele.com", algorithm=MD5, uri="sip:sip.sipmedia.com", nonce="44c0894155ef19a0f40546e7be36909fd1ef3df8", response="025eb7c28a8f59b95b65ae831e840ff9", opaque=""
Expires: 120
Contact: <sip:7322761242@192.168.2.125>
Event: registration
Content-Length: 0


---

<-- SIP read from 69.1.236.33:5060: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.125:5060;branch=z9hG4bK79724411;rport=65415
From: <sip:7322761242@sip.sipmedia.com>;tag=as7b899a4d
To: <sip:7322761242@sip.sipmedia.com>;tag=91791a7492f328483c7efb39681bf0a4.a64d
Call-ID: 7e1ee1647f15a5566ec2f4857f56660b@192.168.2.125
CSeq: 107 REGISTER
Contact: <sip:7322761242@72.68.108.196:65415>;q=0.00;expires=120
Server: Platform Experts BBT (Version 2.1a)
Content-Length: 0
Warning: 392 69.1.236.33:5060 "Noisy feedback tells:  pid=7913 req_src_ip=72.68.108.196 req_src_port=65415 in_uri=sip:sip.sipmedia.com out_uri=sip:sip.sipmedia.com via_cnt==1"


--- (10 headers 0 lines)---
Scheduling destruction of call '7e1ee1647f15a5566ec2f4857f56660b@192.168.2.125' in 32000 ms
Jul 21 03:30:46 NOTICE[16111]: chan_sip.c:9828 handle_response_register: Outbound Registration: Expiry for sip.sipmedia.com is 120 sec (Scheduling reregistration in 105 s)

<-- SIP read from 10.10.10.105:5060: 
NOTIFY sip:10.10.10.10 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.105:5060;branch=z9hG4bK-5deabd7d
From: Sara Hass <sip:11011@10.10.10.10>;tag=2512817a18f3124co0
To: <sip:10.10.10.10>
Call-ID: d741e1ba-38712f0c@10.10.10.105
CSeq: 2320 NOTIFY
Max-Forwards: 70
Event: keep-alive
User-Agent: Sipura/SPA841-3.1.2(d)
Content-Length: 0


--- (10 headers 0 lines)---
Transmitting (no NAT) to 10.10.10.105:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.105:5060;branch=z9hG4bK-5deabd7d;received=10.10.10.105
From: Sara Hass <sip:11011@10.10.10.10>;tag=2512817a18f3124co0
To: <sip:10.10.10.10>;tag=as362dc3f0
Call-ID: d741e1ba-38712f0c@10.10.10.105
CSeq: 2320 NOTIFY
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
Destroying call 'd741e1ba-38712f0c@10.10.10.105'