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@192.168.17.13 To: sip:Scott@192.168.17.13 Call-ID: 00115c40-7fa30002-23abef99-5070b845@192.168.17.114 CSeq: 101 REGISTER User-Agent: CSCO/7 Contact: <sip:Scott@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@192.168.17.13 To: sip:Scott@192.168.17.13;tag=as00424045 Call-ID: 00115c40-7fa30002-23abef99-5070b845@192.168.17.114 CSeq: 101 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:Scott@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@192.168.17.13 To: sip:Scott@192.168.17.13;tag=as00424045 Call-ID: 00115c40-7fa30002-23abef99-5070b845@192.168.17.114 CSeq: 101 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:Scott@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@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@192.168.17.13 To: sip:Scott@192.168.17.13 Call-ID: 00115c40-7fa30002-23abef99-5070b845@192.168.17.114 CSeq: 102 REGISTER User-Agent: CSCO/7 Contact: <sip:Scott@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@192.168.17.13 To: sip:Scott@192.168.17.13;tag=as00424045 Call-ID: 00115c40-7fa30002-23abef99-5070b845@192.168.17.114 CSeq: 102 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:Scott@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@192.168.17.13 To: sip:Scott@192.168.17.13;tag=as00424045 Call-ID: 00115c40-7fa30002-23abef99-5070b845@192.168.17.114 CSeq: 102 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 3600 Contact: <sip:Scott@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@192.168.17.114' in 15000 ms 11 headers, 2 lines Reliably Transmitting: NOTIFY sip:Scott@192.168.17.114:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.17.13:5060;branch=z9hG4bK29419b41 From: "asterisk" <sip:asterisk@192.168.17.13>;tag=as42c5efcf To: <sip:Scott@192.168.17.114:5060> Contact: <sip:asterisk@192.168.17.13> Call-ID: 01ff150c37f3e7f946ecc99741e76d52@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@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@192.168.17.13>;tag=as42c5efcf To: <sip:Scott@192.168.17.114:5060> Call-ID: 01ff150c37f3e7f946ecc99741e76d52@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@192.168.17.13' Destroying call '00115c40-7fa30002-23abef99-5070b845@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 ===========================================================================Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK ============================================================================
> 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"Go into the phone and program Line 2 Settings directly, without using the SIP<MAC>.cnf file. If that works, then your .cnf file is wrong. -- Nabeel Jafferali tel: 416.491.9136 (toronto) 646.225.7426 (new york) fwd: 46990 email/msn : nabeel<at>jafferali.net
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@192.168.17.13 > To: sip:Scott@192.168.17.13 > Call-ID: 00115c40-7fa30002-23abef99-5070b845@192.168.17.114 > CSeq: 101 REGISTER > User-Agent: CSCO/7 > Contact: <sip:Scott@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@192.168.17.13 > To: sip:Scott@192.168.17.13;tag=as00424045 > Call-ID: 00115c40-7fa30002-23abef99-5070b845@192.168.17.114 > CSeq: 101 REGISTER > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: <sip:Scott@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@192.168.17.13 > To: sip:Scott@192.168.17.13;tag=as00424045 > Call-ID: 00115c40-7fa30002-23abef99-5070b845@192.168.17.114 > CSeq: 101 REGISTER > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: <sip:Scott@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@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@192.168.17.13 > To: sip:Scott@192.168.17.13 > Call-ID: 00115c40-7fa30002-23abef99-5070b845@192.168.17.114 > CSeq: 102 REGISTER > User-Agent: CSCO/7 > Contact: <sip:Scott@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@192.168.17.13 > To: sip:Scott@192.168.17.13;tag=as00424045 > Call-ID: 00115c40-7fa30002-23abef99-5070b845@192.168.17.114 > CSeq: 102 REGISTER > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: <sip:Scott@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@192.168.17.13 > To: sip:Scott@192.168.17.13;tag=as00424045 > Call-ID: 00115c40-7fa30002-23abef99-5070b845@192.168.17.114 > CSeq: 102 REGISTER > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Expires: 3600 > Contact: <sip:Scott@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@192.168.17.114' in 15000 ms > 11 headers, 2 lines > Reliably Transmitting: > NOTIFY sip:Scott@192.168.17.114:5060 SIP/2.0 > Via: SIP/2.0/UDP 192.168.17.13:5060;branch=z9hG4bK29419b41 > From: "asterisk" <sip:asterisk@192.168.17.13>;tag=as42c5efcf > To: <sip:Scott@192.168.17.114:5060> > Contact: <sip:asterisk@192.168.17.13> > Call-ID: 01ff150c37f3e7f946ecc99741e76d52@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@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@192.168.17.13>;tag=as42c5efcf > To: <sip:Scott@192.168.17.114:5060> > Call-ID: 01ff150c37f3e7f946ecc99741e76d52@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@192.168.17.13' > Destroying call '00115c40-7fa30002-23abef99-5070b845@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. >