[Asterisk-Users] Multiple lines on Cisco 7960

Fri Jan 7 09:31:50 MST 2005

I did set these to the correct poxy serveras well in the SIPDefault.cnf 
file.

This is very frustrating problem, I have ready dozens of posts that 
refer to how to set this up and I see mto have followed all the suggestions.

I had not looked at the phones settings yet, thanks for the suggestion.  
The setting indicate that there is no configuration on the second line 
it is listed as "UNPROVISIONED"

Scott

Nathan Alberti wrote:

> 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.
>>
>
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-- 
Scott Henderson
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