[Asterisk-Users] Help sought: Cant get Cisco 7960 to register on Asterisk

Sun Feb 6 00:08:10 MST 2005

Take out the "" from the SIP<macADDRESS>.cnf

# Line 1 Registration Authentication
line1_authname: cisco7960
# Line 1 Registration Password
line1_password: grendel

Also for SIPDefault.cnf

# Proxy Server
proxy1_address: 192.168.144.1 ; Can be dotted IP or FQDN
Reset the phone,

"Michael J. Tubby B.Sc." <mike.tubby at thorcom.co.uk> wrote:
Gents,

Following from a previous posting I've switched my 7060G at home
from SCCP/Skinny to SIP but am stuck as I can't get it to register with
Asterisk.

My set up:

- the 7960 has been upgraded to SIP 7.3 (image P003-07-3-00)
which loaded okay after a bit of fiddling around

- phone is on an internal network 192.168.144.0/24

- phone gets its IP address + settings via DHCP ... it happens to
get ip address 192.168.144.182

- Asterisk is version 1.0.5 running on a RH Fedora-Core-3 box
(gate.tubby.org) which has a leg on the inside (private) LAN and
an internet presence

- I have other SIP phones that work just fine including two GrandStream
BT102s and a Cisco 7905 (SIP)

- my extensions.conf for the Cisco 7960 looks like this:

;
; Cisco 7960 in Study
;
exten => 2001,1,Dial(SIP/cisco7960,20,rt)
exten => 2001,2,Voicemail(u2001)
exten => 2001,3,HangUp

- my sip.conf for the Cisco 7960 looks like this:

;;;
;;; My Cisco 7960 in SIP mode (2001) in Study
;;;
[cisco7960]
type=friend
context=default
username=cisco7960
secret=grendel
callerid=Mike Tubby <8202001>
host=dynamic
nat=no
canreinvite=no
incominglimit=1
mailbox=2001 at default
disallow=all
allow=alaw
allow=ulaw
allow=g729

- my SIPDefault.cnf loaded by the phone looks like this:

# SIP Default Generic Configuration File

# Image Version
image_version: P0S3-07-3-00

# Proxy Server
proxy1_address: "192.168.144.1" ; Can be dotted IP or FQDN
proxy2_address: "" ; Can be dotted IP or FQDN
proxy3_address: "" ; Can be dotted IP or FQDN
proxy4_address: "" ; Can be dotted IP or FQDN
proxy5_address: "" ; Can be dotted IP or FQDN
proxy6_address: "" ; Can be dotted IP or FQDN

# Proxy Server Port (default - 5060)
proxy1_port: 5060
proxy2_port: 5060
proxy3_port: 5060
proxy4_port: 5060
proxy5_port: 5060
proxy6_port: 5060

# Proxy Registration (0-disable (default), 1-enable)
proxy_register: 1

# Phone Registration Expiration [1-3932100 sec] (Default - 3600)
timer_register_expires: 3600

# Codec for media stream (g711ulaw (default), g711alaw, g729a)
preferred_codec: g711alaw

# TOS bits in media stream [0-5] (Default - 5)
tos_media: 5

# Inband DTMF Settings (0-disable, 1-enable (default))
dtmf_inband: 1

# Out of band DTMF Settings (none-disable, avt-avt enable (default),
avt_always - always avt )
dtmf_outofband: avt

# DTMF dB Level Settings (1-6dB down, 2-3db down, 3-nominal (default),
4-3db up, 5-6dB up)
dtmf_db_level: 3

# SIP Timers
timer_t1: 500 ; Default 500 msec
timer_t2: 4000 ; Default 4 sec
sip_retx: 10 ; Default 10
sip_invite_retx: 6 ; Default 6
timer_invite_expires: 180 ; Default 180 sec

####### New Parameters added in Release 2.0 #######

# Dialplan template (.xml format file relative to the TFTP root
directory)
dial_template: dialplan

# TFTP Phone Specific Configuration File Directory
tftp_cfg_dir: "" ; Example: ./sip_phone/

- my SIP.cnf for the phone looks like this:

#
# SIP Configuration for Mike's 7960
#

# Line 1 appearance
line1_name: 2001
# Line 1 Registration Authentication
line1_authname: "cisco7960"
# Line 1 Registration Password
line1_password: "grendel"

# Line 2 appearance
#line2_name: football
# Line 2 Registration Authentication
#line2_authname: "UNPROVISIONED"
# Line 2 Registration Password
#line2_password: "UNPROVISIONED"

####### New Parameters added in Release 2.0 #######

# All user_parameters have been removed

# Phone Label (Text desired to be displayed in upper right corner)
phone_label: "Asterisk PBX " ; Has no effect on SIP messaging

# Line 1 Display Name (Display name to use for SIP messaging)
line1_displayname: "Mike Tubby <8202001>"

# Line 2 Display Name (Display name to use for SIP messaging)
#line2_displayname: ""

####### New Parameters added in Release 3.0 ######

# Phone Prompt (The prompt that will be displayed on console and telnet)
phone_prompt: "Cisco 7960" ; Limited to 15 characters (Default -
SIP Phone)

# Phone Password (Password to be used for console or telnet login)
phone_password: "cisco" ; Limited to 31 characters (Default - cisco)

# User classifcation used when Registering [ none(default), phone, ip ]
user_info: none

Symptoms:

- I can place outgoing calls from the 7960 to other local, and wide area,
phones etc. and this works fine

- If I call the phone's extension number I go direct to voice mail
(unavailable)
and the phone does not ring

- the 7960 displays the following against the line #1 button:
2001 (the extension number)
picture of a telephone with small 'x' to bottom right (I presume
means "not registered")

- the 7960 appears to be unable to register with Asterisk

- if I telnet to the 7960 and force a registration, with SIP debugging
enabled,
Asterisk reports the following:

Sip read:
REGISTER sip:192.168.144.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.144.182:5060;branch=z9hG4bK5001cf83
From: sip:2001 at 192.168.144.1
To: sip:2001 at 192.168.144.1
Call-ID: 000a8a2c-8f460003-46a4961a-3ff5c1be at 192.168.144.182
Date: Sat, 05 Feb 2005 21:11:30 GMT
CSeq: 126 REGISTER
User-Agent: CSCO/7
Contact: 
Content-Length: 0
Expires: 3600

11 headers, 0 lines
Using latest request as basis request
Sending to 192.168.144.182 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.144.182:5060;branch=z9hG4bK5001cf83
From: sip:2001 at 192.168.144.1
To: sip:2001 at 192.168.144.1;tag=as0e11d39c
Call-ID: 000a8a2c-8f460003-46a4961a-3ff5c1be at 192.168.144.182
CSeq: 126 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: 
Content-Length: 0

to 192.168.144.182:5060
Feb 5 21:11:29 NOTICE[11276]: chan_sip.c:7654 handle_request:
Registration from 'sip:2001 at 192.168.144.1' failed for '192.168.144.182'
Scheduling destruction of call
'000a8a2c-8f460003-46a4961a-3ff5c1be at 192.168.144.182' in 15000 ms
gate*CLI>

- the same thing logged by Ethereal shows this:

12874.344331 192.168.144.182 -> 192.168.144.1 SIP Request: REGISTER
sip:192.168.144.1

0000 00 0c 6e 77 8c 19 00 0a 8a 2c 8f 46 08 00 45 60 ..nw.....,.F..E`
0010 01 91 24 5e 00 00 40 11 b2 95 c0 a8 90 b6 c0 a8 ..$^.. at .........
0020 90 01 c3 bb 13 c4 01 7d 00 00 52 45 47 49 53 54 .......}..REGIST
0030 45 52 20 73 69 70 3a 31 39 32 2e 31 36 38 2e 31 ER sip:192.168.1
0040 34 34 2e 31 20 53 49 50 2f 32 2e 30 0d 0a 56 69 44.1 SIP/2.0..Vi
0050 61 3a 20 53 49 50 2f 32 2e 30 2f 55 44 50 20 31 a: SIP/2.0/UDP 1
0060 39 32 2e 31 36 38 2e 31 34 34 2e 31 38 32 3a 35 92.168.144.182:5
0070 30 36 30 3b 62 72 61 6e 63 68 3d 7a 39 68 47 34 060;branch=z9hG4
0080 62 4b 30 32 31 30 39 35 62 36 0d 0a 46 72 6f 6d bK021095b6..From
0090 3a 20 73 69 70 3a 32 30 30 31 40 31 39 32 2e 31 : sip:2001 at 192.1
00a0 36 38 2e 31 34 34 2e 31 0d 0a 54 6f 3a 20 73 69 68.144.1..To: si
00b0 70 3a 32 30 30 31 40 31 39 32 2e 31 36 38 2e 31 p:2001 at 192.168.1
00c0 34 34 2e 31 0d 0a 43 61 6c 6c 2d 49 44 3a 20 30 44.1..Call-ID: 0
00d0 30 30 61 38 61 32 63 2d 38 66 34 36 30 30 30 33 00a8a2c-8f460003
00e0 2d 34 36 61 34 39 36 31 61 2d 33 66 66 35 63 31 -46a4961a-3ff5c1
00f0 62 65 40 31 39 32 2e 31 36 38 2e 31 34 34 2e 31 be at 192.168.144.1
0100 38 32 0d 0a 44 61 74 65 3a 20 53 61 74 2c 20 30 82..Date: Sat, 0
0110 35 20 46 65 62 20 32 30 30 35 20 32 31 3a 31 33 5 Feb 2005 21:13
0120 3a 33 30 20 47 4d 54 0d 0a 43 53 65 71 3a 20 31 :30 GMT..CSeq: 1
0130 32 38 20 52 45 47 49 53 54 45 52 0d 0a 55 73 65 28 REGISTER..Use
0140 72 2d 41 67 65 6e 74 3a 20 43 53 43 4f 2f 37 0d r-Agent: CSCO/7.
0150 0a 43 6f 6e 74 61 63 74 3a 20 3c 73 69 70 3a 32 .Contact: 0160 30 30 31 40 31 39 32 2e 31 36 38 2e 31 34 34 2e 001 at 192.168.144.
0170 31 38 32 3a 35 30 36 30 3e 0d 0a 43 6f 6e 74 65 182:5060>..Conte
0180 6e 74 2d 4c 65 6e 67 74 68 3a 20 30 0d 0a 45 78 nt-Length: 0..Ex
0190 70 69 72 65 73 3a 20 33 36 30 30 0d 0a 0d 0a pires: 3600....

12874.345839 192.168.144.1 -> 192.168.144.182 SIP Status: 403 Forbidden
(1 bindings)

0000 00 0a 8a 2c 8f 46 00 0c 6e 77 8c 19 08 00 45 00 ...,.F..nw....E.
0010 01 8f 00 8a 40 00 40 11 96 cb c0 a8 90 01 c0 a8 .... at .@.........
0020 90 b6 13 c4 13 c4 01 7b 07 be 53 49 50 2f 32 2e .......{..SIP/2.
0030 30 20 34 30 33 20 46 6f 72 62 69 64 64 65 6e 0d 0 403 Forbidden.
0040 0a 56 69 61 3a 20 53 49 50 2f 32 2e 30 2f 55 44 .Via: SIP/2.0/UD
0050 50 20 31 39 32 2e 31 36 38 2e 31 34 34 2e 31 38 P 192.168.144.18
0060 32 3a 35 30 36 30 3b 62 72 61 6e 63 68 3d 7a 39 2:5060;branch=z9
0070 68 47 34 62 4b 30 32 31 30 39 35 62 36 0d 0a 46 hG4bK021095b6..F
0080 72 6f 6d 3a 20 73 69 70 3a 32 30 30 31 40 31 39 rom: sip:2001 at 19
0090 32 2e 31 36 38 2e 31 34 34 2e 31 0d 0a 54 6f 3a 2.168.144.1..To:
00a0 20 73 69 70 3a 32 30 30 31 40 31 39 32 2e 31 36 sip:2001 at 192.16
00b0 38 2e 31 34 34 2e 31 3b 74 61 67 3d 61 73 31 66 8.144.1;tag=as1f
00c0 32 65 31 39 30 38 0d 0a 43 61 6c 6c 2d 49 44 3a 2e1908..Call-ID:
00d0 20 30 30 30 61 38 61 32 63 2d 38 66 34 36 30 30 000a8a2c-8f4600
00e0 30 33 2d 34 36 61 34 39 36 31 61 2d 33 66 66 35 03-46a4961a-3ff5
00f0 63 31 62 65 40 31 39 32 2e 31 36 38 2e 31 34 34 c1be at 192.168.144
0100 2e 31 38 32 0d 0a 43 53 65 71 3a 20 31 32 38 20 .182..CSeq: 128
0110 52 45 47 49 53 54 45 52 0d 0a 55 73 65 72 2d 41 REGISTER..User-A
0120 67 65 6e 74 3a 20 41 73 74 65 72 69 73 6b 20 50 gent: Asterisk P
0130 42 58 0d 0a 41 6c 6c 6f 77 3a 20 49 4e 56 49 54 BX..Allow: INVIT
0140 45 2c 20 41 43 4b 2c 20 43 41 4e 43 45 4c 2c 20 E, ACK, CANCEL,
0150 4f 50 54 49 4f 4e 53 2c 20 42 59 45 2c 20 52 45 OPTIONS, BYE, RE
0160 46 45 52 0d 0a 43 6f 6e 74 61 63 74 3a 20 3c 73 FER..Contact: 0170 69 70 3a 32 30 30 31 40 31 39 32 2e 31 36 38 2e ip:2001 at 192.168.
0180 31 34 34 2e 31 3e 0d 0a 43 6f 6e 74 65 6e 74 2d 144.1>..Content-
0190 4c 65 6e 67 74 68 3a 20 30 0d 0a 0d 0a Length: 0....

So, the problem _appears_ to be that the phone is receiving a "403
Forbidden"
back from Asterisk... but I can't figure out why...

Any SIP gurus care to comment on what I've done wrong or should change
to fix it???

Regards

Mike

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