[asterisk-users] PortSip and Astericks new install
Charlie Grosvenor
Charlie at cgrosvenor.co.uk
Sat Nov 18 03:29:25 MST 2006
Thanks for you reply, it does not output anything on the console when i
make the call. However i have turned on SIP Debug and get the following:
<-- SIP read from 192.168.2.3:8099:
REGISTER sip:192.168.1.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.3:8099;rport;branch=z9hG4bK32333
From: <sip:4289 at 192.168.1.1:5060>;tag=9948
To: <sip:4289 at 192.168.1.1:5060>
Call-ID: 4619 at 192.168.2.3
CSeq: 1 REGISTER
Contact: <sip:4289 at 192.168.2.3:8099>
Max-Forwards: 70
User-Agent: PortSIP softphone 2.0
Expires: 150
Content-Length: 0
--- (11 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.2.3 : 8099 (NAT)
Transmitting (NAT) to 192.168.2.3:8099:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.2.3:8099;branch=z9hG4bK32333;received=192.168.2.3;rport=8099
From: <sip:4289 at 192.168.1.1:5060>;tag=9948
To: <sip:4289 at 192.168.1.1:5060>
Call-ID: 4619 at 192.168.2.3
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:4289 at 192.168.2.1>
Content-Length: 0
---
Transmitting (NAT) to 192.168.2.3:8099:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
192.168.2.3:8099;branch=z9hG4bK32333;received=192.168.2.3;rport=8099
From: <sip:4289 at 192.168.1.1:5060>;tag=9948
To: <sip:4289 at 192.168.1.1:5060>;tag=as3d203e7d
Call-ID: 4619 at 192.168.2.3
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="5d41ffa1"
Content-Length: 0
---
Scheduling destruction of call '4619 at 192.168.2.3' in 15000 ms
[Kserver1*CLI>
<-- SIP read from 192.168.2.3:8099:
REGISTER sip:192.168.1.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.3:8099;rport;branch=z9hG4bK41
From: <sip:4289 at 192.168.1.1:5060>;tag=9948
To: <sip:4289 at 192.168.1.1:5060>
Call-ID: 4619 at 192.168.2.3
CSeq: 2 REGISTER
Contact: <sip:4289 at 192.168.2.3:8099>
Authorization: Digest username="4289", realm="asterisk",
nonce="5d41ffa1", uri="sip:192.168.1.1:5060",
response="c3f43fd747f5ef51168bbfa2401b680b", algorithm=MD5
Max-Forwards: 70
User-Agent: PortSIP softphone 2.0
[Kserver1*CLI>
Expires: 150
Content-Length: 0
--- (12 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.2.3 : 8099 (NAT)
Transmitting (NAT) to 192.168.2.3:8099:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.2.3:8099;branch=z9hG4bK41;received=192.168.2.3;rport=8099
From: <sip:4289 at 192.168.1.1:5060>;tag=9948
To: <sip:4289 at 192.168.1.1:5060>
Call-ID: 4619 at 192.168.2.3
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:4289 at 192.168.2.1>
Content-Length: 0
---
[Kserver1*CLI>
12 headers, 0 lines
Reliably Transmitting (NAT) to 192.168.2.3:8099:
OPTIONS sip:4289 at 192.168.2.3:8099 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.1:5060;branch=z9hG4bK60c2e2eb;rport
From: "asterisk" <sip:asterisk at 192.168.2.1>;tag=as6aa7255c
To: <sip:4289 at 192.168.2.3:8099>
Contact: <sip:asterisk at 192.168.2.1>
Call-ID: 7aa3a37c7432e2db6163eb1c7c2910fe at 192.168.2.1
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 17 Nov 2006 20:47:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
Transmitting (NAT) to 192.168.2.3:8099:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.2.3:8099;branch=z9hG4bK41;received=192.168.2.3;rport=8099
From: <sip:4289 at 192.168.1.1:5060>;tag=9948
To: <sip:4289 at 192.168.1.1:5060>;tag=as3d203e7d
Call-ID: 4619 at 192.168.2.3
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Expires: 150
Contact: <sip:4289 at 192.168.2.3:8099>;expires=150
Date: Fri, 17 Nov 2006 20:47:18 GMT
Content-Length: 0
---
Scheduling destruction of call '4619 at 192.168.2.3' in 15000 ms
[Kserver1*CLI>
<-- SIP read from 192.168.2.3:8099:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.1:5060;branch=z9hG4bK60c2e2eb;rport=5060
From: "asterisk" <sip:asterisk at 192.168.2.1>;tag=as6aa7255c
To: <sip:4289 at 192.168.2.3:8099>;tag=18467
Call-ID: 7aa3a37c7432e2db6163eb1c7c2910fe at 192.168.2.1
CSeq: 102 OPTIONS
User-Agent: PortSIP softphone 2.0
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE,
INFO, REFER, UPDATE
Content-Length: 0
--- (9 headers 0 lines) ---
Destroying call '7aa3a37c7432e2db6163eb1c7c2910fe at 192.168.2.1'
[Kserver1*CLI>
<-- SIP read from 192.168.2.3:8099:
INVITE sip:500 at 192.168.1.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.3:8099;rport;branch=z9hG4bK31747
From: <sip:4289 at 192.168.1.1:5060>;tag=30024
To: <sip:500 at 192.168.1.1:5060>
Call-ID: 19841 at 192.168.2.3
CSeq: 20 INVITE
Contact: <sip:4289 at 192.168.2.3:8099>
Max-Forwards: 70
User-Agent: PortSIP softphone 2.0
Subject: call
Expires: 120
Allow: INVITE, ACK, UPDATE, INFO, CANCEL, BYE, OPTIONS, REFER,
SUBSCRIBE, NOTIFY, MESSAGE
Content-Type: application/sdp
Content-Length: 325
v=0
o=- 2177823 2177823 IN IP4 192.168.2.3
s=PortSIP VOIP SDK 2.0
c=IN IP4 192.168.2.3
t=0 0
m=audio 51636 RTP/AVP 0 3 8 97 4 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
--- (14 headers 14 lines) ---
Using INVITE request as basis request - 19841 at 192.168.2.3
Sending to 192.168.2.3 : 8099 (NAT)
Reliably Transmitting (NAT) to 192.168.2.3:8099:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
192.168.2.3:8099;branch=z9hG4bK31747;received=192.168.2.3;rport=8099
From: <sip:4289 at 192.168.1.1:5060>;tag=30024
To: <sip:500 at 192.168.1.1:5060>;tag=as6aa8059e
Call-ID: 19841 at 192.168.2.3
CSeq: 20 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="6301f109"
Content-Length: 0
---
Scheduling destruction of call '19841 at 192.168.2.3' in 15000 ms
Found user '4289'
[Kserver1*CLI>
<-- SIP read from 192.168.2.3:8099:
ACK sip:500 at 192.168.1.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.3:8099;rport;branch=z9hG4bK31747
From: <sip:4289 at 192.168.1.1:5060>;tag=30024
To: <sip:500 at 192.168.1.1:5060>;tag=as6aa8059e
Call-ID: 19841 at 192.168.2.3
CSeq: 20 ACK
Content-Length: 0
--- (7 headers 0 lines) ---
[Kserver1*CLI>
<-- SIP read from 192.168.2.3:8099:
INVITE sip:500 at 192.168.1.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.3:8099;rport;branch=z9hG4bK6334
From: <sip:4289 at 192.168.1.1:5060>;tag=30024
To: <sip:500 at 192.168.1.1:5060>
Call-ID: 19841 at 192.168.2.3
CSeq: 21 INVITE
Contact: <sip:4289 at 192.168.2.3:8099>
Proxy-Authorization: Digest username="4289", realm="asterisk",
nonce="6301f109", uri="sip:500 at 192.168.1.1:5060",
response="50d2e5940d8c3970109f8efeec310f16", algorithm=MD5
Max-Forwards: 70
User-Agent: PortSIP softphone 2.0
Subject: call
Expires: 120
Allow: INVITE, ACK, UPDATE, INFO, CANCEL, BYE, OPTIONS, REFER,
SUBSCRIBE, NOTIFY, MESSAGE
Content-Type: application/sdp
Content-Length: 325
v=0
o=- 2177823 2177823 IN IP4 192.168.2.3
s=PortSIP VOIP SDK 2.0
c=IN IP4 192.168.2.3
t=0 0
m=audio 51636 RTP/AVP 0 3 8 97 4 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
--- (15 headers 14 lines) ---
Using INVITE request as basis request - 19841 at 192.168.2.3
Sending to 192.168.2.3 : 8099 (NAT)
Found user '4289'
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 8
Found RTP audio format 97
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 192.168.2.3:51636
Found description format PCMU
Found description format GSM
Found description format PCMA
Found description format iLBC
Found description format G723
Found description format G729
Found description format telephone-event
Capabilities: us - 0x4c (ulaw|alaw|slin), peer - audio=0x50f
(g723|gsm|ulaw|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0xc
(ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Looking for 500 in managers (domain 192.168.1.1)
list_route: hop: <sip:4289 at 192.168.2.3:8099>
Transmitting (NAT) to 192.168.2.3:8099:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.2.3:8099;branch=z9hG4bK6334;received=192.168.2.3;rport=8099
From: <sip:4289 at 192.168.1.1:5060>;tag=30024
To: <sip:500 at 192.168.1.1:5060>
Call-ID: 19841 at 192.168.2.3
CSeq: 21 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:500 at 192.168.2.1>
Content-Length: 0
---
[Kserver1*CLI>
We're at 192.168.2.1 port 11784
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.2.3:8099:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.2.3:8099;branch=z9hG4bK6334;received=192.168.2.3;rport=8099
From: <sip:4289 at 192.168.1.1:5060>;tag=30024
To: <sip:500 at 192.168.1.1:5060>;tag=as7c73e159
Call-ID: 19841 at 192.168.2.3
CSeq: 21 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:500 at 192.168.2.1>
Content-Type: application/sdp
Content-Length: 236
v=0
o=root 4616 4616 IN IP4 192.168.2.1
s=session
c=IN IP4 192.168.2.1
t=0 0
m=audio 11784 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 - - - -
---
[Kserver1*CLI>
<-- SIP read from 192.168.2.3:8099:
ACK sip:500 at 192.168.2.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.3:8099;rport;branch=z9hG4bK26500
From: <sip:4289 at 192.168.1.1:5060>;tag=30024
To: <sip:500 at 192.168.1.1:5060>;tag=as7c73e159
Call-ID: 19841 at 192.168.2.3
CSeq: 21 ACK
Contact: <sip:4289 at 192.168.2.3:8099>
Proxy-Authorization: Digest username="4289", realm="asterisk",
nonce="6301f109", uri="sip:500 at 192.168.1.1:5060",
response="50d2e5940d8c3970109f8efeec310f16", algorithm=MD5
Max-Forwards: 70
User-Agent: PortSIP softphone 2.0
Content-Length: 0
--- (11 headers 0 lines) ---
[Kserver1*CLI> ex
12 headers, 3 lines
Reliably Transmitting (NAT) to 192.168.2.3:8099:
NOTIFY sip:4289 at 192.168.2.3:8099 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.1:5060;branch=z9hG4bK794df374;rport
From: "asterisk" <sip:asterisk at 192.168.2.1>;tag=as44c6a747
To: <sip:4289 at 192.168.2.3:8099>
Contact: <sip:asterisk at 192.168.2.1>
Call-ID: 2add5bd90dc94ddb47e46a133b1dea27 at 192.168.2.1
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 91
Messages-Waiting: no
Message-Account: sip:asterisk at 192.168.2.1
Voice-Message: 0/0 (0/0)
---
Scheduling destruction of call
'2add5bd90dc94ddb47e46a133b1dea27 at 192.168.2.1' in 15000 ms
[Kserver1*CLI> ex
<-- SIP read from 192.168.2.3:8099:
SIP/2.0 481 Subscription Does Not Exist
Via: SIP/2.0/UDP 192.168.2.1:5060;branch=z9hG4bK794df374;rport=5060
From: "asterisk" <sip:asterisk at 192.168.2.1>;tag=as44c6a747
To: <sip:4289 at 192.168.2.3:8099>;tag=41
Call-ID: 2add5bd90dc94ddb47e46a133b1dea27 at 192.168.2.1
CSeq: 102 NOTIFY
User-Agent: PortSIP softphone 2.0
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE,
INFO, REFER, UPDATE
Content-Length: 0
--- (9 headers 0 lines) ---
Destroying call '2add5bd90dc94ddb47e46a133b1dea27 at 192.168.2.1'
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