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6.2

Cisco and Sip

Asked by arahming in Voice Over IP, Asterisk Open Source Telephony, Network Operations

Wow I guess cisco and SIP are a sore subject here no one is answering anything I post. Anyways I have a setup like this

comcast --- asterisk box ----- LAN  ---- MY PC ---- Cisco 7971G-GE --- Qwest

I have it setup in DHCP that everything goes out through quest except my VOIP which is controlled trough the asterisk box using NAT.

my problem is either I have bad configs or I the stupid phone which no one really seems to like

MY SIP image is SIP 70.8-3-1S

here is my log I'M going to sleep been at this for days

To: <sip:2000@192.168.0.100>;tag=as1dd72ec1
Call-ID: 00152bd2-02640002-0fcf0e0e-f4b9bbbd@192.168.0.12
CSeq: 101 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7eef41bd"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '00152bd2-02640002-0fcf0e0e-f4b9bbbd@192.168.0.12' in 32000 ms (Method: REGISTER)
elastix*CLI>
<--- SIP read from 192.168.0.12:49175 --->
REGISTER sip:192.168.0.100 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.12:5060;branch=z9hG4bKb89e2e2b
From: <sip:2000@192.168.0.100>;tag=00152bd202640002cee21c6c-cafccb93
To: <sip:2000@192.168.0.100>
Call-ID: 00152bd2-02640002-0fcf0e0e-f4b9bbbd@192.168.0.12
Max-Forwards: 70
Date: Fri, 04 May 2007 19:55:02 GMT
CSeq: 102 REGISTER
User-Agent: Cisco-CP7971G-GE/8.3.0
Contact: <sip:7b452e87-4496-4762-e11f-b26751a1884b@192.168.0.12:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-00152bd20264>";+u.sip!model.ccm.cisco.com="119"
Authorization: Digest username="2000",realm="asterisk",uri="sip:192.168.0.100",response="6c4093de66e1e49a0a25524307367e27",nonce="7eef41bd",algorithm=MD5
Supported: (null),X-cisco-xsi-6.0.1
Content-Length: 0
Expires: 3600


<------------->
--- (14 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.0.12 : 5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.0.12:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.12:5060;branch=z9hG4bKb89e2e2b;received=192.168.0.12
From: <sip:2000@192.168.0.100>;tag=00152bd202640002cee21c6c-cafccb93
To: <sip:2000@192.168.0.100>
Call-ID: 00152bd2-02640002-0fcf0e0e-f4b9bbbd@192.168.0.12
CSeq: 102 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<------------>
elastix*CLI>
<--- Transmitting (no NAT) to 192.168.0.12:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.12:5060;branch=z9hG4bKb89e2e2b;received=192.168.0.12
From: <sip:2000@192.168.0.100>;tag=00152bd202640002cee21c6c-cafccb93
To: <sip:2000@192.168.0.100>;tag=as1dd72ec1
Call-ID: 00152bd2-02640002-0fcf0e0e-f4b9bbbd@192.168.0.12
CSeq: 102 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 3600
Contact: <sip:7b452e87-4496-4762-e11f-b26751a1884b@192.168.0.12:5060;transport=udp>;expires=3600
Date: Tue, 02 Jun 2009 02:46:59 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '00152bd2-02640002-0fcf0e0e-f4b9bbbd@192.168.0.12' in 32000 ms (Method: REGISTER)
Scheduling destruction of SIP dialog '29717d5f084401263e0f09365f135f3d@192.168.0.100' in 32000 ms (Method: NOTIFY)
Reliably Transmitting (no NAT) to 192.168.0.12:5060:
NOTIFY sip:7b452e87-4496-4762-e11f-b26751a1884b@192.168.0.12:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.100:5060;branch=z9hG4bK144471ca;rport
From: "Unknown" <sip:Unknown@192.168.0.100>;tag=as12d953b9
To: <sip:7b452e87-4496-4762-e11f-b26751a1884b@192.168.0.12:5060;transport=udp>
Contact: <sip:Unknown@192.168.0.100>
Call-ID: 29717d5f084401263e0f09365f135f3d@192.168.0.100
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 88

Messages-Waiting: no
Message-Account: sip:*97@192.168.0.100
Voice-Message: 0/0 (0/0)

---
elastix*CLI>
<--- SIP read from 192.168.0.12:49183 --->
SIP/2.0 400 Bad Request
Via: SIP/2.0/UDP 192.168.0.100:5060;branch=z9hG4bK144471ca;rport
From: "Unknown" <sip:Unknown@192.168.0.100>;tag=as12d953b9
To: <sip:7b452e87-4496-4762-e11f-b26751a1884b@192.168.0.12:5060;transport=udp>
Call-ID: 29717d5f084401263e0f09365f135f3d@192.168.0.100
Date: Fri, 04 May 2007 19:55:10 GMT
Warning: 399 Bad MWI NOTIFY
CSeq: 102 NOTIFY
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
    -- Got SIP response 400 "Bad Request" back from 192.168.0.12
Really destroying SIP dialog '29717d5f084401263e0f09365f135f3d@192.168.0.100' Method: NOTIFY
Really destroying SIP dialog '00152bd2-02640002-0fcf0e0e-f4b9bbbd@192.168.0.12' Method: REGISTER
Reliably Transmitting (NAT) to 209.249.3.59:5060:
OPTIONS sip:209.249.3.59 SIP/2.0
Via: SIP/2.0/UDP xxx.11.27.1xxx:5060;branch=z9hG4bK39bba087;rport
From: "Unknown" <sip:Unknown@xxx.11.27.1xxx>;tag=as47217107
To: <sip:209.249.3.59>
Contact: <sip:Unknown@xxx.11.27.1xxx>
Call-ID: 6035040551856d385a75ebf365fe216d@xxx.11.27.1xxx
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 02 Jun 2009 02:47:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
elastix*CLI>
<--- SIP read from 209.249.3.59:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxx.11.27.1xxx:5060;branch=z9hG4bK39bba087;rport
To: <sip:209.249.3.59>;tag=3452899551-425375
From: "Unknown" <sip:Unknown@xxx.11.27.1xxx>;tag=as47217107
Call-ID: 6035040551856d385a75ebf365fe216d@xxx.11.27.1xxx
CSeq: 102 OPTIONS
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE
Contact: <sip:209.249.3.59:5060>
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '6035040551856d385a75ebf365fe216d@xxx.11.27.1xxx' Method: OPTIONS
Reliably Transmitting (NAT) to 209.249.3.59:5060:
OPTIONS sip:209.249.3.59 SIP/2.0
Via: SIP/2.0/UDP xxx.11.27.1xxx:5060;branch=z9hG4bK6afdb9c3;rport
From: "Unknown" <sip:Unknown@xxx.11.27.1xxx>;tag=as40079fd2
To: <sip:209.249.3.59>
Contact: <sip:Unknown@xxx.11.27.1xxx>
Call-ID: 4b05409745a3073f5830ea292d11f190@xxx.11.27.1xxx
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 02 Jun 2009 02:48:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
elastix*CLI>
<--- SIP read from 209.249.3.59:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxx.11.27.1xxx:5060;branch=z9hG4bK6afdb9c3;rport
To: <sip:209.249.3.59>;tag=3452899611-484749
From: "Unknown" <sip:Unknown@xxx.11.27.1xxx>;tag=as40079fd2
Call-ID: 4b05409745a3073f5830ea292d11f190@xxx.11.27.1xxx
CSeq: 102 OPTIONS
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE
Contact: <sip:209.249.3.59:5060>
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '4b05409745a3073f5830ea292d11f190@xxx.11.27.1xxx' Method: OPTIONS
Reliably Transmitting (NAT) to 209.249.3.59:5060:
OPTIONS sip:209.249.3.59 SIP/2.0
Via: SIP/2.0/UDP xxx.11.27.1xxx:5060;branch=z9hG4bK4f8bb793;rport
From: "Unknown" <sip:Unknown@xxx.11.27.1xxx>;tag=as2fe9a568
To: <sip:209.249.3.59>
Contact: <sip:Unknown@xxx.11.27.1xxx>
Call-ID: 2ebf3ede51617e4e00dabdc90aa920e2@xxx.11.27.1xxx
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 02 Jun 2009 02:49:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
elastix*CLI>
<--- SIP read from 209.249.3.59:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxx.11.27.1xxx:5060;branch=z9hG4bK4f8bb793;rport
To: <sip:209.249.3.59>;tag=3452899671-547357
From: "Unknown" <sip:Unknown@xxx.11.27.1xxx>;tag=as2fe9a568
Call-ID: 2ebf3ede51617e4e00dabdc90aa920e2@xxx.11.27.1xxx
CSeq: 102 OPTIONS
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE
Contact: <sip:209.249.3.59:5060>
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '2ebf3ede51617e4e00dabdc90aa920e2@xxx.11.27.1xxx' Method: OPTIONS
    -- Remote UNIX connection
    -- Remote UNIX connection disconnected
elastix*CLI>

 
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Solution Provided By: koszegi
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Solution Grade: A
 
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