[Asterisk-Users] Multiple lines on Cisco 7960

[Nathan Alberti]

Do you have:

# Proxy Server
proxy1_address: "x.x.x.x"
proxy2_address: "x.x.x.x"

Unsure if this is required, does your phone list the correct server ? 
(settings | 4 | 2 | 6)


Nathan.


Scott Henderson wrote:

> 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.
>