rendered paste bodyAllow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.3.0067
Content-Type: application/sdp
Content-Length: 189
v=0
o=- 1172423141 1172423141 IN IP4 192.168.1.109
s=Polycom IP Phone
c=IN IP4 192.168.1.109
t=0 0
m=audio 2268 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 192.168.1.109:2268
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0x6 (gsm|ulaw), 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:102@192.168.1.109>
set_destination: Parsing <sip:102@192.168.1.109> for address/port to send to
set_destination: set destination to 192.168.1.109, port 5060
Transmitting (no NAT) to 192.168.1.109:5060:
ACK sip:102@192.168.1.109 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.120:5060;branch=z9hG4bK1d974b11;rport
From: "Sioux City IA" <sip:7128723003@192.168.1.120>;tag=as164147aa
To: <sip:102@192.168.1.109>;tag=16058633-7BC334EC
Contact: <sip:7128723003@192.168.1.120>
Call-ID: 24651c7b17cd020414caec454ec7c0bc@192.168.1.120
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
-- SIP/102-09f967e0 answered SIP/66.225.202.80-09f8fba8
-- Attempting native bridge of SIP/66.225.202.80-09f8fba8 and SIP/102-09f967e0
localhost*CLI>
<-- SIP read from 66.225.202.76:5060:
--- (0 headers 0 lines) Nat keepalive ---
Retransmitting #3 (no NAT) to 66.225.202.76:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 66.225.202.76;branch=z9hG4bK711c.376de652.0;received=66.225.202.76
Via: SIP/2.0/UDP 66.225.202.80:5060;branch=z9hG4bK51442a7e
Record-Route: <sip:66.225.202.76;ftag=as2ce4b118;lr=on>
From: "Sioux City IA" <sip:7128723003@66.225.202.80>;tag=as2ce4b118
To: <sip:carpenter1@proxy-1.nufone.net>;tag=as4d20d894
Call-ID: 1daa55511cf58d9f7ac84b5f163f164e@66.225.202.80
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:8002081258@192.168.1.120>
Content-Type: application/sdp
Content-Length: 241
v=0
o=root 10823 10824 IN IP4 192.168.1.120
s=session
c=IN IP4 192.168.1.120
t=0 0
m=audio 14878 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
Retransmitting #4 (no NAT) to 66.225.202.76:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 66.225.202.76;branch=z9hG4bK711c.376de652.0;received=66.225.202.76
Via: SIP/2.0/UDP 66.225.202.80:5060;branch=z9hG4bK51442a7e
Record-Route: <sip:66.225.202.76;ftag=as2ce4b118;lr=on>
From: "Sioux City IA" <sip:7128723003@66.225.202.80>;tag=as2ce4b118
To: <sip:carpenter1@proxy-1.nufone.net>;tag=as4d20d894
Call-ID: 1daa55511cf58d9f7ac84b5f163f164e@66.225.202.80
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:8002081258@192.168.1.120>
Content-Type: application/sdp
Content-Length: 241
v=0
o=root 10823 10824 IN IP4 192.168.1.120
s=session
c=IN IP4 192.168.1.120
t=0 0
m=audio 14878 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
Destroying call '402a6dfb11c882c50c7d4a5c3ca324ba@127.0.0.1'
localhost*CLI>
<-- SIP read from 192.168.1.109:5060:
BYE sip:7128723003@192.168.1.120 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.109;branch=z9hG4bK6f972936B8851365
From: <sip:102@192.168.1.109>;tag=16058633-7BC334EC
To: "Sioux City IA" <sip:7128723003@192.168.1.120>;tag=as164147aa
CSeq: 1 BYE
Call-ID: 24651c7b17cd020414caec454ec7c0bc@192.168.1.120
Contact: <sip:102@192.168.1.109>
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.3.0067
Max-Forwards: 70
Content-Length: 0
--- (10 headers 0 lines) ---
Sending to 192.168.1.109 : 5060 (non-NAT)
Transmitting (no NAT) to 192.168.1.109:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.109;branch=z9hG4bK6f972936B8851365;received=192.168.1.109
From: <sip:102@192.168.1.109>;tag=16058633-7BC334EC
To: "Sioux City IA" <sip:7128723003@192.168.1.120>;tag=as164147aa
Call-ID: 24651c7b17cd020414caec454ec7c0bc@192.168.1.120
CSeq: 1 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:7128723003@192.168.1.120>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing
---
== Spawn extension (inbound, 8002081258, 3) exited non-zero on 'SIP/66.225.202.80-09f8fba8'
Scheduling destruction of call '1daa55511cf58d9f7ac84b5f163f164e@66.225.202.80' in 32000 ms
Destroying call '24651c7b17cd020414caec454ec7c0bc@192.168.1.120'
Retransmitting #5 (no NAT) to 66.225.202.76:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 66.225.202.76;branch=z9hG4bK711c.376de652.0;received=66.225.202.76
Via: SIP/2.0/UDP 66.225.202.80:5060;branch=z9hG4bK51442a7e
Record-Route: <sip:66.225.202.76;ftag=as2ce4b118;lr=on>
From: "Sioux City IA" <sip:7128723003@66.225.202.80>;tag=as2ce4b118
To: <sip:carpenter1@proxy-1.nufone.net>;tag=as4d20d894
Call-ID: 1daa55511cf58d9f7ac84b5f163f164e@66.225.202.80
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:8002081258@192.168.1.120>
Content-Type: application/sdp
Content-Length: 241
v=0
o=root 10823 10824 IN IP4 192.168.1.120
s=session
c=IN IP4 192.168.1.120
t=0 0
m=audio 14878 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
localhost*CLI>
<-- SIP read from 192.168.1.106:3040:
--- (0 headers 1 lines) ---
Retransmitting #6 (no NAT) to 66.225.202.76:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 66.225.202.76;branch=z9hG4bK711c.376de652.0;received=66.225.202.76
Via: SIP/2.0/UDP 66.225.202.80:5060;branch=z9hG4bK51442a7e
Record-Route: <sip:66.225.202.76;ftag=as2ce4b118;lr=on>
From: "Sioux City IA" <sip:7128723003@66.225.202.80>;tag=as2ce4b118
To: <sip:carpenter1@proxy-1.nufone.net>;tag=as4d20d894
Call-ID: 1daa55511cf58d9f7ac84b5f163f164e@66.225.202.80
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:8002081258@192.168.1.120>
Content-Type: application/sdp
Content-Length: 241
v=0
o=root 10823 10824 IN IP4 192.168.1.120
s=session
c=IN IP4 192.168.1.120
t=0 0
m=audio 14878 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
Feb 25 15:55:07 WARNING[10835]: chan_sip.c:1228 retrans_pkt: Maximum retries exceeded on transmission 1daa55511cf58d9f7ac84b5f163f164e@66.225.202.80 for seqno 103 (Non-critical Response)
localhost*CLI>
<-- SIP read from 66.225.202.76:5060:
--- (0 headers 0 lines) Nat keepalive ---
Destroying call '1daa55511cf58d9f7ac84b5f163f164e@66.225.202.80'
localhost*CLI>
<-- SIP read from 192.168.1.106:3040:
--- (0 headers 1 lines) ---
localhost*CLI>