At the beginning, I could make calls intermittently back and forth but now for the last couple of months, it just doesn’t work anymore. We can dial from both ends and make the phone ring on the other end, but no audio is coming through. On a few occasions, I have received audio on the remote ext. without audio on the local side.
I suspect it is the Cable company blocking ports, because at the remote location, the cable company also provides VoIP telephone service, but I have attached a log and some of my setting, you find out if any one of you can see something obvious that I am doing wrong.
CentOS 5.2
Asterisk 1.4 (from source)
FreePBX 2.5
This is my setup:
Asterisk PBX behind router with Tomato firmware with ports 5060 UDP and 10000-20000 UDP forwarded to Asterisk. IP 2xx.45.198.xx0
Remote extension Cisco 7912g SIP phone behind Ambit U10C022 (VoIP adapter, cable modem and router) with ports 5060 UDP forwarded to 5060 UDP on the phone and 10000-20000 UDP forwarded to 16384 UDP on the phone. IP 3xx.170.67.xx9
My local extension is 200 (ATA) and my remote CISCO 7912g is extension 400. The local IP of remote ext. is 192.168.0.4.
I have tried to register the phone pointing it to mydyndns.com (my domain) and also to the actual external IP of Asterisk, with the same results.
My extension has:
nat=yes
qualify=yes
host=dynamic ;have also tried with remote static IP
disallow=all
allow=ulaw
My sip_nat.conf has:
nat=yes
externhost=mydyndns.com
externrefresh=30
localnet=192.168.1.0/255.255.255.0
My rtp.conf has:
rtpstart=10000
rtpend=20000
The settings in the phone are:
UID=400
PWD=mysecret
Proxy=2xx.45.198.xx0
AltProxyTimeOut=0
UseLoginID=0
LoginID=0
SIPRegInterval=60
MaxRedirect=5
SIPRegOn=1
NATIP=3xx.170.67.xx9
SIPPort=5060
MediaPort=16384
OutBoundProxy=0
MsgRetryLimits=0x00000000
NatServer=0
NatTimer=0x00000000
DialPlan=911|1>#t8.r9t2-|0>#t811.rat4-|^1t4>#.-
IPDialPlan=1
The following is a tcpdump on Asterisk, first while making a call from ext. 200 to remote ext. 400 and then from 400 to 200.
The actual log shows RTP packets going out (from local) when making the call from 200 to 400 and also when receiving the call, packets are going out from Asterisk’s side but no RTP packets are coming back in either case.
[code]SIP/2.0 200 OK
Via: SIP/2.0/UDP 2xx.45.198.xx0:5060;branch=z9hG4bK6da4b122;rport
From: “John Doe” sip:200@2xx.45.198.xx0;tag=as473f3fd1
To: sip:400@3xx.170.67.xx9:5060;user=phone;transport=udp;tag=2784700350
Call-ID: 21ee464e40297e59208e576b2ba32a2b@2xx.45.198.xx0
CSeq: 102 INVITE
Contact: sip:400@3xx.170.67.xx9:5060;user=phone;transport=udp
Server: Cisco-CP7912/8.0.1-060412A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE
Supported: replaces
Content-Length: 203
Content-Type: application/sdp
v=0
o=400 35745 35745 IN IP4 192.168.0.4
s=Cisco 7912 SIP Call
c=IN IP4 3xx.170.67.xx9
t=0 0
m=audio 16384 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
ACK sip:400@3xx.170.67.xx9:5060;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP 2xx.45.198.xx0:5060;branch=z9hG4bK147fe730;rport
From: “John Doe” sip:200@2xx.45.198.xx0;tag=as473f3fd1
To: sip:400@3xx.170.67.xx9:5060;user=phone;transport=udp;tag=2784700350
Contact: sip:200@2xx.45.198.xx0
Call-ID: 21ee464e40297e59208e576b2ba32a2b@2xx.45.198.xx0
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
REGISTER sip:2xx.45.198.xx0 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK567d8b8e9299827
From: sip:400@2xx.45.198.xx0;user=phone;tag=1530313901
To: sip:400@2xx.45.198.xx0;user=phone
Call-ID: 1213881949@3xx.170.67.xx9
CSeq: 15 REGISTER
Max-Forwards: 70
Contact: sip:400@3xx.170.67.xx9:5060;user=phone;transport=udp;expires=60;+sip.instance="urn:uuid:00000000-0000-0000-0000-0012001441E0"
User-Agent: Cisco-CP7912/8.0.1-060412A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE
Supported: replaces, 100rel
Authorization: Digest username=“400”,realm=“asterisk”,nonce=“4c5325e1”,uri=“sip:2xx.45.198.xx0”,response="0252dfb6d50378a0ae4f2a52802db870"
Content-Length: 0
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK567d8b8e9299827;received=3xx.170.67.xx9
From: sip:400@2xx.45.198.xx0;user=phone;tag=1530313901
To: sip:400@2xx.45.198.xx0;user=phone
Call-ID: 1213881949@3xx.170.67.xx9
CSeq: 15 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK567d8b8e9299827;received=3xx.170.67.xx9
From: sip:400@2xx.45.198.xx0;user=phone;tag=1530313901
To: sip:400@2xx.45.198.xx0;user=phone;tag=as0b1f27b4
Call-ID: 1213881949@3xx.170.67.xx9
CSeq: 15 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="40ff7952"
Content-Length: 0
REGISTER sip:2xx.45.198.xx0 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK6ed2d5fac1b82c57
From: sip:400@2xx.45.198.xx0;user=phone;tag=1530313901
To: sip:400@2xx.45.198.xx0;user=phone
Call-ID: 1213881949@3xx.170.67.xx9
CSeq: 16 REGISTER
Max-Forwards: 70
Contact: sip:400@3xx.170.67.xx9:5060;user=phone;transport=udp;expires=60;+sip.instance="urn:uuid:00000000-0000-0000-0000-0012001441E0"
User-Agent: Cisco-CP7912/8.0.1-060412A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE
Supported: replaces, 100rel
Authorization: Digest username=“400”,realm=“asterisk”,nonce=“40ff7952”,uri=“sip:2xx.45.198.xx0”,response="0d8a01b0c144b40bc7cfe0aa019c4319"
Content-Length: 0
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK6ed2d5fac1b82c57;received=3xx.170.67.xx9
From: sip:400@2xx.45.198.xx0;user=phone;tag=1530313901
To: sip:400@2xx.45.198.xx0;user=phone
Call-ID: 1213881949@3xx.170.67.xx9
CSeq: 16 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0
OPTIONS sip:400@3xx.170.67.xx9:5060;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP 2xx.45.198.xx0:5060;branch=z9hG4bK6fb5e96d;rport
From: “Unknown” sip:Unknown@2xx.45.198.xx0;tag=as037107dc
To: sip:400@3xx.170.67.xx9:5060;user=phone;transport=udp
Contact: sip:Unknown@2xx.45.198.xx0
Call-ID: 706d52c93ce8a22a5f79477f2afa4ce4@2xx.45.198.xx0
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 16 Mar 2011 18:32:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK6ed2d5fac1b82c57;received=3xx.170.67.xx9
From: sip:400@2xx.45.198.xx0;user=phone;tag=1530313901
To: sip:400@2xx.45.198.xx0;user=phone;tag=as0b1f27b4
Call-ID: 1213881949@3xx.170.67.xx9
CSeq: 16 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Expires: 60
Contact: sip:400@3xx.170.67.xx9:5060;user=phone;transport=udp;expires=60
Date: Wed, 16 Mar 2011 18:32:47 GMT
Content-Length: 0
SIP/2.0 200 OK
Via: SIP/2.0/UDP 2xx.45.198.xx0:5060;branch=z9hG4bK6fb5e96d;rport
From: “Unknown” sip:Unknown@2xx.45.198.xx0;tag=as037107dc
To: sip:400@3xx.170.67.xx9:5060;user=phone;transport=udp;tag=2784700350
Call-ID: 706d52c93ce8a22a5f79477f2afa4ce4@2xx.45.198.xx0
CSeq: 102 OPTIONS
Server: Cisco-CP7912/8.0.1-060412A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE
Supported: replaces
Content-Length: 278
Content-Type: application/sdp
v=0
o=400 36716 36716 IN IP4 192.168.0.4
s=Cisco 7912 SIP Call
c=IN IP4 3xx.170.67.xx9
t=0 0
m=audio 16384 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
BYE sip:200@2xx.45.198.xx0 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK547cb3462f02355b
From: sip:400@3xx.170.67.xx9;user=phone;transport=udp;tag=2784700350
To: “John Doe” sip:200@2xx.45.198.xx0;tag=as473f3fd1
Call-ID: 21ee464e40297e59208e576b2ba32a2b@2xx.45.198.xx0
CSeq: 1 BYE
Max-Forwards: 70
User-Agent: Cisco-CP7912/8.0.1-060412A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE
Supported: replaces, 100rel
Content-Length: 0
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK547cb3462f02355b;received=3xx.170.67.xx9
From: sip:400@3xx.170.67.xx9;user=phone;transport=udp;tag=2784700350
To: “John Doe” sip:200@2xx.45.198.xx0;tag=as473f3fd1
Call-ID: 21ee464e40297e59208e576b2ba32a2b@2xx.45.198.xx0
CSeq: 1 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0
NOTIFY sip:400@3xx.170.67.xx9:5060;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP 2xx.45.198.xx0:5060;branch=z9hG4bK4536d65f;rport
From: “Unknown” sip:Unknown@2xx.45.198.xx0;tag=as52a7c7cf
To: sip:400@3xx.170.67.xx9:5060;user=phone;transport=udp
Contact: sip:Unknown@2xx.45.198.xx0
Call-ID: 17b617ed4359421953b05b0c712a0434@2xx.45.198.xx0
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 89
Messages-Waiting: no
Message-Account: sip:*97@2xx.45.198.xx0
Voice-Message: 0/0 (0/0)
SIP/2.0 200 OK
Via: SIP/2.0/UDP 2xx.45.198.xx0:5060;branch=z9hG4bK4536d65f;rport
From: “Unknown” sip:Unknown@2xx.45.198.xx0;tag=as52a7c7cf
To: sip:400@3xx.170.67.xx9:5060;user=phone;transport=udp;tag=2784700350
Call-ID: 17b617ed4359421953b05b0c712a0434@2xx.45.198.xx0
CSeq: 102 NOTIFY
Server: Cisco-CP7912/8.0.1-060412A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE
Supported: replaces
Content-Length: 0
INVITE sip:200@2xx.45.198.xx0;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK1000cbe7bbc24227
From: sip:400@2xx.45.198.xx0;user=phone;tag=2784700350
To: sip:200@2xx.45.198.xx0;user=phone
Call-ID: 322380413@3xx.170.67.xx9
CSeq: 1 INVITE
Max-Forwards: 70
Contact: sip:400@3xx.170.67.xx9:5060;user=phone;transport=udp
User-Agent: Cisco-CP7912/8.0.1-060412A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE
Supported: replaces, 100rel
Expires: 300
Content-Length: 278
Content-Type: application/sdp
v=0
o=400 38794 38794 IN IP4 192.168.0.4
s=Cisco 7912 SIP Call
c=IN IP4 3xx.170.67.xx9
t=0 0
m=audio 16384 RTP/AVP 0 18 8 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK1000cbe7bbc24227;received=3xx.170.67.xx9
From: sip:400@2xx.45.198.xx0;user=phone;tag=2784700350
To: sip:200@2xx.45.198.xx0;user=phone;tag=as4117842f
Call-ID: 322380413@3xx.170.67.xx9
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="5d905ae6"
Content-Length: 0
ACK sip:200@2xx.45.198.xx0;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK1000cbe7bbc24227
From: sip:400@2xx.45.198.xx0;user=phone;tag=2784700350
To: sip:200@2xx.45.198.xx0;user=phone;tag=as4117842f
Call-ID: 322380413@3xx.170.67.xx9
CSeq: 1 ACK
Max-Forwards: 70
User-Agent: Cisco-CP7912/8.0.1-060412A
Content-Length: 0
INVITE sip:200@2xx.45.198.xx0;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK3282ce62635f24
From: sip:400@2xx.45.198.xx0;user=phone;tag=2784700350
To: sip:200@2xx.45.198.xx0;user=phone
Call-ID: 322380413@3xx.170.67.xx9
CSeq: 2 INVITE
Max-Forwards: 70
Contact: sip:400@3xx.170.67.xx9:5060;user=phone;transport=udp
User-Agent: Cisco-CP7912/8.0.1-060412A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE
Supported: replaces, 100rel
Proxy-Authorization: Digest username=“400”,realm=“asterisk”,nonce=“5d905ae6”,uri=“sip:200@2xx.45.198.xx0”,response="d0ed51a539f45abd67f2942966180389"
Expires: 300
Content-Length: 278
Content-Type: application/sdp
v=0
o=400 38809 38809 IN IP4 192.168.0.4
s=Cisco 7912 SIP Call
c=IN IP4 3xx.170.67.xx9
t=0 0
m=audio 16384 RTP/AVP 0 18 8 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK3282ce62635f24;received=3xx.170.67.xx9
From: sip:400@2xx.45.198.xx0;user=phone;tag=2784700350
To: sip:200@2xx.45.198.xx0;user=phone
Call-ID: 322380413@3xx.170.67.xx9
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: sip:200@2xx.45.198.xx0
Content-Length: 0
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK3282ce62635f24;received=3xx.170.67.xx9
From: sip:400@2xx.45.198.xx0;user=phone;tag=2784700350
To: sip:200@2xx.45.198.xx0;user=phone;tag=as2f169cf8
Call-ID: 322380413@3xx.170.67.xx9
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: sip:200@2xx.45.198.xx0
Content-Length: 0
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK3282ce62635f24;received=3xx.170.67.xx9
From: sip:400@2xx.45.198.xx0;user=phone;tag=2784700350
To: sip:200@2xx.45.198.xx0;user=phone;tag=as2f169cf8
Call-ID: 322380413@3xx.170.67.xx9
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: sip:200@2xx.45.198.xx0
Content-Type: application/sdp
Content-Length: 242
v=0
o=root 8551 8551 IN IP4 2xx.45.198.xx0
s=session
c=IN IP4 2xx.45.198.xx0
t=0 0
m=audio 10684 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
ACK sip:200@2xx.45.198.xx0 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK611bd62e7cc429ef
From: sip:400@2xx.45.198.xx0;user=phone;tag=2784700350
To: sip:200@2xx.45.198.xx0;user=phone;tag=as2f169cf8
Call-ID: 322380413@3xx.170.67.xx9
CSeq: 2 ACK
Max-Forwards: 70
User-Agent: Cisco-CP7912/8.0.1-060412A
Proxy-Authorization: Digest username=“400”,realm=“asterisk”,nonce=“5d905ae6”,uri=“sip:200@2xx.45.198.xx0”,response="d0ed51a539f45abd67f2942966180389"
Content-Length: 0
BYE sip:200@2xx.45.198.xx0 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK8521a031f7750eee
From: sip:400@2xx.45.198.xx0;user=phone;tag=2784700350
To: sip:200@2xx.45.198.xx0;user=phone;tag=as2f169cf8
Call-ID: 322380413@3xx.170.67.xx9
CSeq: 3 BYE
Max-Forwards: 70
User-Agent: Cisco-CP7912/8.0.1-060412A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE
Supported: replaces, 100rel
Proxy-Authorization: Digest username=“400”,realm=“asterisk”,nonce=“5d905ae6”,uri=“sip:200@2xx.45.198.xx0”,response="6071ac25b458f4a8e532322a237b9312"
Content-Length: 0
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK8521a031f7750eee;received=3xx.170.67.xx9
From: sip:400@2xx.45.198.xx0;user=phone;tag=2784700350
To: sip:200@2xx.45.198.xx0;user=phone;tag=as2f169cf8
Call-ID: 322380413@3xx.170.67.xx9
CSeq: 3 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0
REGISTER sip:2xx.45.198.xx0 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK5d1a60f76c2b6f97
From: sip:400@2xx.45.198.xx0;user=phone;tag=1530313901
To: sip:400@2xx.45.198.xx0;user=phone
Call-ID: 1213881949@3xx.170.67.xx9
CSeq: 17 REGISTER
Max-Forwards: 70
Contact: sip:400@3xx.170.67.xx9:5060;user=phone;transport=udp;expires=60;+sip.instance="urn:uuid:00000000-0000-0000-0000-0012001441E0"
User-Agent: Cisco-CP7912/8.0.1-060412A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE
Supported: replaces, 100rel
Authorization: Digest username=“400”,realm=“asterisk”,nonce=“40ff7952”,uri=“sip:2xx.45.198.xx0”,response="0d8a01b0c144b40bc7cfe0aa019c4319"
Content-Length: 0
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK5d1a60f76c2b6f97;received=3xx.170.67.xx9
From: sip:400@2xx.45.198.xx0;user=phone;tag=1530313901
To: sip:400@2xx.45.198.xx0;user=phone
Call-ID: 1213881949@3xx.170.67.xx9
CSeq: 17 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK5d1a60f76c2b6f97;received=3xx.170.67.xx9
From: sip:400@2xx.45.198.xx0;user=phone;tag=1530313901
To: sip:400@2xx.45.198.xx0;user=phone;tag=as1b6e1592
Call-ID: 1213881949@3xx.170.67.xx9
CSeq: 17 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="062b4dc0"
Content-Length: 0
REGISTER sip:2xx.45.198.xx0 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK99ada71dbc721779
From: sip:400@2xx.45.198.xx0;user=phone;tag=1530313901
To: sip:400@2xx.45.198.xx0;user=phone
Call-ID: 1213881949@3xx.170.67.xx9
CSeq: 18 REGISTER
Max-Forwards: 70
Contact: sip:400@3xx.170.67.xx9:5060;user=phone;transport=udp;expires=60;+sip.instance="urn:uuid:00000000-0000-0000-0000-0012001441E0"
User-Agent: Cisco-CP7912/8.0.1-060412A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE
Supported: replaces, 100rel
Authorization: Digest username=“400”,realm=“asterisk”,nonce=“062b4dc0”,uri=“sip:2xx.45.198.xx0”,response="709a7eb6fdff875d34f9b407240f1afd"
Content-Length: 0
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK99ada71dbc721779;received=3xx.170.67.xx9
From: sip:400@2xx.45.198.xx0;user=phone;tag=1530313901
To: sip:400@2xx.45.198.xx0;user=phone
Call-ID: 1213881949@3xx.170.67.xx9
CSeq: 18 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0
OPTIONS sip:400@3xx.170.67.xx9:5060;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP 2xx.45.198.xx0:5060;branch=z9hG4bK10e603e9;rport
From: “Unknown” sip:Unknown@2xx.45.198.xx0;tag=as666bb023
To: sip:400@3xx.170.67.xx9:5060;user=phone;transport=udp
Contact: sip:Unknown@2xx.45.198.xx0
Call-ID: 53579afc01673dc1369ba8fe2a7ed4e0@2xx.45.198.xx0
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 16 Mar 2011 18:33:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK99ada71dbc721779;received=3xx.170.67.xx9
From: sip:400@2xx.45.198.xx0;user=phone;tag=1530313901
To: sip:400@2xx.45.198.xx0;user=phone;tag=as1b6e1592
Call-ID: 1213881949@3xx.170.67.xx9
CSeq: 18 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Expires: 60
Contact: sip:400@3xx.170.67.xx9:5060;user=phone;transport=udp;expires=60
Date: Wed, 16 Mar 2011 18:33:39 GMT
Content-Length: 0
SIP/2.0 200 OK
Via: SIP/2.0/UDP 2xx.45.198.xx0:5060;branch=z9hG4bK10e603e9;rport
From: “Unknown” sip:Unknown@2xx.45.198.xx0;tag=as666bb023
To: sip:400@3xx.170.67.xx9:5060;user=phone;transport=udp;tag=2784700350
Call-ID: 53579afc01673dc1369ba8fe2a7ed4e0@2xx.45.198.xx0
CSeq: 102 OPTIONS
Server: Cisco-CP7912/8.0.1-060412A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE
Supported: replaces
Content-Length: 278
Content-Type: application/sdp
v=0
o=400 41949 41949 IN IP4 192.168.0.4
s=Cisco 7912 SIP Call
c=IN IP4 3xx.170.67.xx9
t=0 0
m=audio 16384 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
[/code]
Anything I am doing wrong?