<DIV>Take out the "" from the SIP<macADDRESS>.cnf</DIV>
<P># Line 1 Registration Authentication<BR>line1_authname: cisco7960<BR># Line 1 Registration Password<BR>line1_password: grendel</P>
<P>Also for SIPDefault.cnf</P>
<P># Proxy Server<BR>proxy1_address: 192.168.144.1 ; Can be dotted IP or FQDN</P>
<DIV>Reset the phone,<BR><BR><B><I>"Michael J. Tubby B.Sc." <mike.tubby@thorcom.co.uk></I></B> wrote:</DIV>
<BLOCKQUOTE class=replbq style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #1010ff 2px solid">Gents,<BR><BR>Following from a previous posting I've switched my 7060G at home<BR>from SCCP/Skinny to SIP but am stuck as I can't get it to register with<BR>Asterisk.<BR><BR><BR>My set up:<BR><BR>- the 7960 has been upgraded to SIP 7.3 (image P003-07-3-00)<BR>which loaded okay after a bit of fiddling around<BR><BR>- phone is on an internal network 192.168.144.0/24<BR><BR>- phone gets its IP address + settings via DHCP ... it happens to<BR>get ip address 192.168.144.182<BR><BR>- Asterisk is version 1.0.5 running on a RH Fedora-Core-3 box<BR>(gate.tubby.org) which has a leg on the inside (private) LAN and<BR>an internet presence<BR><BR>- I have other SIP phones that work just fine including two GrandStream<BR>BT102s and a Cisco 7905 (SIP)<BR><BR>- my extensions.conf for the Cisco 7960 looks like this:<BR><BR>;<BR>; Cisco 7960 in Study<BR>;<BR>exten =>
2001,1,Dial(SIP/cisco7960,20,rt)<BR>exten => 2001,2,Voicemail(u2001)<BR>exten => 2001,3,HangUp<BR><BR>- my sip.conf for the Cisco 7960 looks like this:<BR><BR>;;;<BR>;;; My Cisco 7960 in SIP mode (2001) in Study<BR>;;;<BR>[cisco7960]<BR>type=friend<BR>context=default<BR>username=cisco7960<BR>secret=grendel<BR>callerid=Mike Tubby <8202001><BR>host=dynamic<BR>nat=no<BR>canreinvite=no<BR>incominglimit=1<BR>mailbox=2001@default<BR>disallow=all<BR>allow=alaw<BR>allow=ulaw<BR>allow=g729<BR><BR>- my SIPDefault.cnf loaded by the phone looks like this:<BR><BR># SIP Default Generic Configuration File<BR><BR># Image Version<BR>image_version: P0S3-07-3-00<BR><BR># Proxy Server<BR>proxy1_address: "192.168.144.1" ; Can be dotted IP or FQDN<BR>proxy2_address: "" ; Can be dotted IP or FQDN<BR>proxy3_address: "" ; Can be dotted IP or FQDN<BR>proxy4_address: "" ; Can be dotted IP or FQDN<BR>proxy5_address: "" ; Can be dotted IP or FQDN<BR>proxy6_address: "" ; Can be dotted IP or
FQDN<BR><BR># Proxy Server Port (default - 5060)<BR>proxy1_port: 5060<BR>proxy2_port: 5060<BR>proxy3_port: 5060<BR>proxy4_port: 5060<BR>proxy5_port: 5060<BR>proxy6_port: 5060<BR><BR># Proxy Registration (0-disable (default), 1-enable)<BR>proxy_register: 1<BR><BR># Phone Registration Expiration [1-3932100 sec] (Default - 3600)<BR>timer_register_expires: 3600<BR><BR># Codec for media stream (g711ulaw (default), g711alaw, g729a)<BR>preferred_codec: g711alaw<BR><BR># TOS bits in media stream [0-5] (Default - 5)<BR>tos_media: 5<BR><BR># Inband DTMF Settings (0-disable, 1-enable (default))<BR>dtmf_inband: 1<BR><BR># Out of band DTMF Settings (none-disable, avt-avt enable (default),<BR>avt_always - always avt )<BR>dtmf_outofband: avt<BR><BR># DTMF dB Level Settings (1-6dB down, 2-3db down, 3-nominal (default),<BR>4-3db up, 5-6dB up)<BR>dtmf_db_level: 3<BR><BR># SIP Timers<BR>timer_t1: 500 ; Default 500 msec<BR>timer_t2: 4000 ; Default 4 sec<BR>sip_retx: 10 ; Default 10<BR>sip_invite_retx:
6 ; Default 6<BR>timer_invite_expires: 180 ; Default 180 sec<BR><BR>####### New Parameters added in Release 2.0 #######<BR><BR># Dialplan template (.xml format file relative to the TFTP root<BR>directory)<BR>dial_template: dialplan<BR><BR># TFTP Phone Specific Configuration File Directory<BR>tftp_cfg_dir: "" ; Example: ./sip_phone/<BR><BR><BR>- my SIP<MACADDRESS>.cnf for the phone looks like this:<BR><BR>#<BR># SIP Configuration for Mike's 7960<BR>#<BR><BR># Line 1 appearance<BR>line1_name: 2001<BR># Line 1 Registration Authentication<BR>line1_authname: "cisco7960"<BR># Line 1 Registration Password<BR>line1_password: "grendel"<BR><BR># Line 2 appearance<BR>#line2_name: football<BR># Line 2 Registration Authentication<BR>#line2_authname: "UNPROVISIONED"<BR># Line 2 Registration Password<BR>#line2_password: "UNPROVISIONED"<BR><BR><BR>####### New Parameters added in Release 2.0 #######<BR><BR># All user_parameters have been removed<BR><BR># Phone Label (Text desired to be displayed in
upper right corner)<BR>phone_label: "Asterisk PBX " ; Has no effect on SIP messaging<BR><BR># Line 1 Display Name (Display name to use for SIP messaging)<BR>line1_displayname: "Mike Tubby <8202001>"<BR><BR># Line 2 Display Name (Display name to use for SIP messaging)<BR>#line2_displayname: ""<BR><BR><BR>####### New Parameters added in Release 3.0 ######<BR><BR># Phone Prompt (The prompt that will be displayed on console and telnet)<BR>phone_prompt: "Cisco 7960" ; Limited to 15 characters (Default -<BR>SIP Phone)<BR><BR># Phone Password (Password to be used for console or telnet login)<BR>phone_password: "cisco" ; Limited to 31 characters (Default - cisco)<BR><BR># User classifcation used when Registering [ none(default), phone, ip ]<BR>user_info: none<BR><BR><BR><BR><BR>Symptoms:<BR><BR>- I can place outgoing calls from the 7960 to other local, and wide area,<BR>phones etc. and this works fine<BR><BR>- If I call the phone's extension number I go direct to voice
mail<BR>(unavailable)<BR>and the phone does not ring<BR><BR>- the 7960 displays the following against the line #1 button:<BR>2001 (the extension number)<BR>picture of a telephone with small 'x' to bottom right (I presume<BR>means "not registered")<BR><BR>- the 7960 appears to be unable to register with Asterisk<BR><BR>- if I telnet to the 7960 and force a registration, with SIP debugging<BR>enabled,<BR>Asterisk reports the following:<BR><BR>Sip read:<BR>REGISTER sip:192.168.144.1 SIP/2.0<BR>Via: SIP/2.0/UDP 192.168.144.182:5060;branch=z9hG4bK5001cf83<BR>From: sip:2001@192.168.144.1<BR>To: sip:2001@192.168.144.1<BR>Call-ID: 000a8a2c-8f460003-46a4961a-3ff5c1be@192.168.144.182<BR>Date: Sat, 05 Feb 2005 21:11:30 GMT<BR>CSeq: 126 REGISTER<BR>User-Agent: CSCO/7<BR>Contact: <sip:2001@192.168.144.182:5060><BR>Content-Length: 0<BR>Expires: 3600<BR><BR>11 headers, 0 lines<BR>Using latest request as basis request<BR>Sending to 192.168.144.182 : 5060 (non-NAT)<BR>Transmitting (no
NAT):<BR>SIP/2.0 403 Forbidden<BR>Via: SIP/2.0/UDP 192.168.144.182:5060;branch=z9hG4bK5001cf83<BR>From: sip:2001@192.168.144.1<BR>To: sip:2001@192.168.144.1;tag=as0e11d39c<BR>Call-ID: 000a8a2c-8f460003-46a4961a-3ff5c1be@192.168.144.182<BR>CSeq: 126 REGISTER<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER<BR>Contact: <sip:2001@192.168.144.1><BR>Content-Length: 0<BR><BR>to 192.168.144.182:5060<BR>Feb 5 21:11:29 NOTICE[11276]: chan_sip.c:7654 handle_request:<BR>Registration from 'sip:2001@192.168.144.1' failed for '192.168.144.182'<BR>Scheduling destruction of call<BR>'000a8a2c-8f460003-46a4961a-3ff5c1be@192.168.144.182' in 15000 ms<BR>gate*CLI><BR><BR>- the same thing logged by Ethereal shows this:<BR><BR>12874.344331 192.168.144.182 -> 192.168.144.1 SIP Request: REGISTER<BR>sip:192.168.144.1<BR><BR>0000 00 0c 6e 77 8c 19 00 0a 8a 2c 8f 46 08 00 45 60 ..nw.....,.F..E`<BR>0010 01 91 24 5e 00 00 40 11 b2 95 c0 a8 90 b6 c0 a8 ..$^..@.........<BR>0020
90 01 c3 bb 13 c4 01 7d 00 00 52 45 47 49 53 54 .......}..REGIST<BR>0030 45 52 20 73 69 70 3a 31 39 32 2e 31 36 38 2e 31 ER sip:192.168.1<BR>0040 34 34 2e 31 20 53 49 50 2f 32 2e 30 0d 0a 56 69 44.1 SIP/2.0..Vi<BR>0050 61 3a 20 53 49 50 2f 32 2e 30 2f 55 44 50 20 31 a: SIP/2.0/UDP 1<BR>0060 39 32 2e 31 36 38 2e 31 34 34 2e 31 38 32 3a 35 92.168.144.182:5<BR>0070 30 36 30 3b 62 72 61 6e 63 68 3d 7a 39 68 47 34 060;branch=z9hG4<BR>0080 62 4b 30 32 31 30 39 35 62 36 0d 0a 46 72 6f 6d bK021095b6..From<BR>0090 3a 20 73 69 70 3a 32 30 30 31 40 31 39 32 2e 31 : sip:2001@192.1<BR>00a0 36 38 2e 31 34 34 2e 31 0d 0a 54 6f 3a 20 73 69 68.144.1..To: si<BR>00b0 70 3a 32 30 30 31 40 31 39 32 2e 31 36 38 2e 31 p:2001@192.168.1<BR>00c0 34 34 2e 31 0d 0a 43 61 6c 6c 2d 49 44 3a 20 30 44.1..Call-ID: 0<BR>00d0 30 30 61 38 61 32 63 2d 38 66 34 36 30 30 30 33 00a8a2c-8f460003<BR>00e0 2d 34 36 61 34 39 36 31 61 2d 33 66 66 35 63 31 -46a4961a-3ff5c1<BR>00f0 62 65 40 31 39 32 2e 31 36 38 2e 31 34 34 2e 31
be@192.168.144.1<BR>0100 38 32 0d 0a 44 61 74 65 3a 20 53 61 74 2c 20 30 82..Date: Sat, 0<BR>0110 35 20 46 65 62 20 32 30 30 35 20 32 31 3a 31 33 5 Feb 2005 21:13<BR>0120 3a 33 30 20 47 4d 54 0d 0a 43 53 65 71 3a 20 31 :30 GMT..CSeq: 1<BR>0130 32 38 20 52 45 47 49 53 54 45 52 0d 0a 55 73 65 28 REGISTER..Use<BR>0140 72 2d 41 67 65 6e 74 3a 20 43 53 43 4f 2f 37 0d r-Agent: CSCO/7.<BR>0150 0a 43 6f 6e 74 61 63 74 3a 20 3c 73 69 70 3a 32 .Contact: <sip:2<br>0160 30 30 31 40 31 39 32 2e 31 36 38 2e 31 34 34 2e 001@192.168.144.<BR>0170 31 38 32 3a 35 30 36 30 3e 0d 0a 43 6f 6e 74 65 182:5060>..Conte<BR>0180 6e 74 2d 4c 65 6e 67 74 68 3a 20 30 0d 0a 45 78 nt-Length: 0..Ex<BR>0190 70 69 72 65 73 3a 20 33 36 30 30 0d 0a 0d 0a pires: 3600....<BR><BR>12874.345839 192.168.144.1 -> 192.168.144.182 SIP Status: 403 Forbidden<BR>(1 bindings)<BR><BR>0000 00 0a 8a 2c 8f 46 00 0c 6e 77 8c 19 08 00 45 00 ...,.F..nw....E.<BR>0010 01 8f 00 8a 40 00 40 11 96 cb c0 a8 90 01 c0 a8
....@.@.........<BR>0020 90 b6 13 c4 13 c4 01 7b 07 be 53 49 50 2f 32 2e .......{..SIP/2.<BR>0030 30 20 34 30 33 20 46 6f 72 62 69 64 64 65 6e 0d 0 403 Forbidden.<BR>0040 0a 56 69 61 3a 20 53 49 50 2f 32 2e 30 2f 55 44 .Via: SIP/2.0/UD<BR>0050 50 20 31 39 32 2e 31 36 38 2e 31 34 34 2e 31 38 P 192.168.144.18<BR>0060 32 3a 35 30 36 30 3b 62 72 61 6e 63 68 3d 7a 39 2:5060;branch=z9<BR>0070 68 47 34 62 4b 30 32 31 30 39 35 62 36 0d 0a 46 hG4bK021095b6..F<BR>0080 72 6f 6d 3a 20 73 69 70 3a 32 30 30 31 40 31 39 rom: sip:2001@19<BR>0090 32 2e 31 36 38 2e 31 34 34 2e 31 0d 0a 54 6f 3a 2.168.144.1..To:<BR>00a0 20 73 69 70 3a 32 30 30 31 40 31 39 32 2e 31 36 sip:2001@192.16<BR>00b0 38 2e 31 34 34 2e 31 3b 74 61 67 3d 61 73 31 66 8.144.1;tag=as1f<BR>00c0 32 65 31 39 30 38 0d 0a 43 61 6c 6c 2d 49 44 3a 2e1908..Call-ID:<BR>00d0 20 30 30 30 61 38 61 32 63 2d 38 66 34 36 30 30 000a8a2c-8f4600<BR>00e0 30 33 2d 34 36 61 34 39 36 31 61 2d 33 66 66 35 03-46a4961a-3ff5<BR>00f0 63 31 62 65 40 31 39 32
2e 31 36 38 2e 31 34 34 c1be@192.168.144<BR>0100 2e 31 38 32 0d 0a 43 53 65 71 3a 20 31 32 38 20 .182..CSeq: 128<BR>0110 52 45 47 49 53 54 45 52 0d 0a 55 73 65 72 2d 41 REGISTER..User-A<BR>0120 67 65 6e 74 3a 20 41 73 74 65 72 69 73 6b 20 50 gent: Asterisk P<BR>0130 42 58 0d 0a 41 6c 6c 6f 77 3a 20 49 4e 56 49 54 BX..Allow: INVIT<BR>0140 45 2c 20 41 43 4b 2c 20 43 41 4e 43 45 4c 2c 20 E, ACK, CANCEL,<BR>0150 4f 50 54 49 4f 4e 53 2c 20 42 59 45 2c 20 52 45 OPTIONS, BYE, RE<BR>0160 46 45 52 0d 0a 43 6f 6e 74 61 63 74 3a 20 3c 73 FER..Contact: <S<BR>0170 69 70 3a 32 30 30 31 40 31 39 32 2e 31 36 38 2e ip:2001@192.168.<BR>0180 31 34 34 2e 31 3e 0d 0a 43 6f 6e 74 65 6e 74 2d 144.1>..Content-<BR>0190 4c 65 6e 67 74 68 3a 20 30 0d 0a 0d 0a Length: 0....<BR><BR><BR><BR><BR>So, the problem _appears_ to be that the phone is receiving a "403<BR>Forbidden"<BR>back from Asterisk... but I can't figure out why...<BR><BR>Any SIP gurus care to comment on what I've done wrong or should
change<BR>to fix it???<BR><BR><BR>Regards<BR><BR>Mike<BR><BR>_______________________________________________<BR>Asterisk-Users mailing list<BR>Asterisk-Users@lists.digium.com<BR>http://lists.digium.com/mailman/listinfo/asterisk-users<BR>To UNSUBSCRIBE or update options visit:<BR>http://lists.digium.com/mailman/listinfo/asterisk-users<BR></BLOCKQUOTE></sip:2<br></sip:2001@192.168.144.1></sip:2001@192.168.144.182:5060><BR><BR><p>__________________________________________________<br>Do You Yahoo!?<br>Tired of spam? Yahoo! Mail has the best spam protection around <br>http://mail.yahoo.com