[Asterisk-Users] Multiple lines on Cisco 7960

[Scott Henderson]

I have been trying to get multiple lines on the 7960 to work for several 
days.  i have read all the posts I can find and have run multiple "sip 
debug" and have gotten no place on this.

Here are the relevant section of the config files:

sip.conf

[scott]
type=friend
host=dynamic
username=scott
secret=scott
context=default
mailbox=6101
callerid=Scott Henderson

[scott1]
type=friend
host=dynamic
username=scott1
secret=scott1
context=default
mailbox=6101
callerid=Scott Henderson 1

macaddress.cnf
# Line 1
line1_name: Scott
line1_authname: "scott"    
line1_password: "scott"

# Line 2
line2_name:  Scott1
line2_authname: "scott1"
line2_password: "scott1"

sip debug output from resetting the phone:
Sip read:
REGISTER sip:192.168.17.13 SIP/2.0
Via: SIP/2.0/UDP 192.168.17.114:5060;branch=z9hG4bK61a1a63a
From: sip:Scott at 192.168.17.13
To: sip:Scott at 192.168.17.13
Call-ID: 00115c40-7fa30002-23abef99-5070b845 at 192.168.17.114
CSeq: 101 REGISTER
User-Agent: CSCO/7
Contact: <sip:Scott at 192.168.17.114:5060>
Content-Length: 0
Expires: 3600


10 headers, 0 lines
Using latest request as basis request
Sending to 192.168.17.114 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.17.114:5060;branch=z9hG4bK61a1a63a
From: sip:Scott at 192.168.17.13
To: sip:Scott at 192.168.17.13;tag=as00424045
Call-ID: 00115c40-7fa30002-23abef99-5070b845 at 192.168.17.114
CSeq: 101 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:Scott at 192.168.17.13>
Content-Length: 0


 to 192.168.17.114:5060
Transmitting (no NAT):
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.17.114:5060;branch=z9hG4bK61a1a63a
From: sip:Scott at 192.168.17.13
To: sip:Scott at 192.168.17.13;tag=as00424045
Call-ID: 00115c40-7fa30002-23abef99-5070b845 at 192.168.17.114
CSeq: 101 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:Scott at 192.168.17.13>
WWW-Authenticate: Digest realm="asterisk", nonce="0045611f"
Content-Length: 0


 to 192.168.17.114:5060
Scheduling destruction of call 
'00115c40-7fa30002-23abef99-5070b845 at 192.168.17.114' in 15000 ms
argon*CLI>

Sip read:
REGISTER sip:192.168.17.13 SIP/2.0
Via: SIP/2.0/UDP 192.168.17.114:5060;branch=z9hG4bK49f2aa87
From: sip:Scott at 192.168.17.13
To: sip:Scott at 192.168.17.13
Call-ID: 00115c40-7fa30002-23abef99-5070b845 at 192.168.17.114
CSeq: 102 REGISTER
User-Agent: CSCO/7
Contact: <sip:Scott at 192.168.17.114:5060>
Authorization: Digest 
username="scott",realm="asterisk",uri="sip:192.168.17.13",response="7b9f392d15161ef76ae35f283e876497",nonce="0045611f",algorithm=md5
Content-Length: 0
Expires: 3600


11 headers, 0 lines
Using latest request as basis request
Sending to 192.168.17.114 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.17.114:5060;branch=z9hG4bK49f2aa87
From: sip:Scott at 192.168.17.13
To: sip:Scott at 192.168.17.13;tag=as00424045
Call-ID: 00115c40-7fa30002-23abef99-5070b845 at 192.168.17.114
CSeq: 102 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:Scott at 192.168.17.13>
Content-Length: 0


 to 192.168.17.114:5060
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.17.114:5060;branch=z9hG4bK49f2aa87
From: sip:Scott at 192.168.17.13
To: sip:Scott at 192.168.17.13;tag=as00424045
Call-ID: 00115c40-7fa30002-23abef99-5070b845 at 192.168.17.114
CSeq: 102 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Expires: 3600
Contact: <sip:Scott at 192.168.17.114:5060>;expires=3600
Date: Fri, 07 Jan 2005 02:56:25 GMT
Content-Length: 0


 to 192.168.17.114:5060
Scheduling destruction of call 
'00115c40-7fa30002-23abef99-5070b845 at 192.168.17.114' in 15000 ms
11 headers, 2 lines
Reliably Transmitting:
NOTIFY sip:Scott at 192.168.17.114:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.17.13:5060;branch=z9hG4bK29419b41
From: "asterisk" <sip:asterisk at 192.168.17.13>;tag=as42c5efcf
To: <sip:Scott at 192.168.17.114:5060>
Contact: <sip:asterisk at 192.168.17.13>
Call-ID: 01ff150c37f3e7f946ecc99741e76d52 at 192.168.17.13
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 36

Messages-Waiting: no
Voicemail: 0/0
 (no NAT) to 192.168.17.114:5060
Scheduling destruction of call 
'01ff150c37f3e7f946ecc99741e76d52 at 192.168.17.13' in 15000 ms
argon*CLI>

Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.17.13:5060;branch=z9hG4bK29419b41
From: "asterisk" <sip:asterisk at 192.168.17.13>;tag=as42c5efcf
To: <sip:Scott at 192.168.17.114:5060>
Call-ID: 01ff150c37f3e7f946ecc99741e76d52 at 192.168.17.13
Date: Fri, 07 Jan 2005 02:56:26 GMT
CSeq: 102 NOTIFY
Content-Length: 0


8 headers, 0 lines
Destroying call '01ff150c37f3e7f946ecc99741e76d52 at 192.168.17.13'
Destroying call '00115c40-7fa30002-23abef99-5070b845 at 192.168.17.114'
argon*CLI>

The result of this configuration is that I always get the first line 
"line_1" but never the second line.  From what I can tell the phone 
never even tries to register the second line.

-- 
Scott Henderson
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