[Asterisk-Users] Cisco 7960G w/ SIP not working properly
Scott A. Henderson
scott at finite-tech.com
Mon Oct 4 22:42:21 MST 2004
I have Asterisk version 1.0-RC1 running on Debian Woody.
I have 1 analog phone working, 2 inbound lines working, X-Lite is working.
The problem that I am having is with Cisco 7960 with SIP version 7.2
software. I can make outbound calls and they work fine, I even get a
notice that I have voice mail on the phone and it seems to register
properly but I can seem to dial to the phone.
Configuration information is:
======================================================================
argon*CLI> sip show peers
Name/username Host Dyn Nat ACL Mask Port
Status
marc/marc (Unspecified) D 255.255.255.255 0
Unmonitored
jarad/jarad (Unspecified) D 255.255.255.255 0
Unmonitored
johnsip/johnsip (Unspecified) D 255.255.255.255 0
Unmonitored
kevinsip/kevins (Unspecified) D 255.255.255.255 0
Unmonitored
scott/scott 192.168.17.114 D 255.255.255.255 5060
Unmonitored <------- This is the phone in question
========================================================================
sip.conf
[scott]
type=friend
host=dynamic
username=scott
secret=scott
context=default
mailbox=6101
callerid=Scott Henderson
==========================================================================
extensions.conf
; Scott Henderson
exten => 6101,1,Dial(SIP/scott/s,20,Ttr)
exten => 6101,2,Dial(Zap/R1/3372860)
exten => 6101,3,Voicemail(u6101)
===========================================================================
sip debug info from the CLI when I dial from one asterisk extension to
the 7960 SIP phone in question
argon*CLI>
Destroying call '06d39dda6546c5a0148ec12a59af7d0a at 127.0.0.1'
We're at 192.168.17.13 port 19478
Answering/Requesting with root capability 4
Answering with capability 0x2(GSM)
Answering with capability 0x8(ALAW)
Answering with non-codec capability 0x1(G723)
12 headers, 12 lines
Reliably Transmitting:
INVITE sip:s at 192.168.17.114 SIP/2.0
Via: SIP/2.0/UDP 192.168.17.13:5060;branch=z9hG4bK4d228686
From: "asterisk" <sip:asterisk at 192.168.17.13>;tag=as73fe6d09
To: <sip:s at 192.168.17.114>
Contact: <sip:asterisk at 192.168.17.13>
Call-ID: 74ee54ba09a3c0d31bc3d6853229181f at 192.168.17.13
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Tue, 05 Oct 2004 05:37:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 263
v=0
o=root 2816 2816 IN IP4 192.168.17.13
s=session
c=IN IP4 192.168.17.13
t=0 0
m=audio 19478 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
(no NAT) to 192.168.17.114:5060
argon*CLI>
Sip read:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.17.13:5060;branch=z9hG4bK4d228686
From: "asterisk" <sip:asterisk at 192.168.17.13>;tag=as73fe6d09
To: <sip:s at 192.168.17.114>;tag=00115c407fa3000948b81d9b-144916e7
Call-ID: 74ee54ba09a3c0d31bc3d6853229181f at 192.168.17.13
Date: Tue, 05 Oct 2004 05:37:05 GMT
CSeq: 102 INVITE
Server: CSCO/7
Contact: <sip:scott at 192.168.17.114:5060>
Content-Length: 0
10 headers, 0 lines
Transmitting:
ACK sip:scott at 192.168.17.114:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.17.13:5060;branch=z9hG4bK4d228686
From: "asterisk" <sip:asterisk at 192.168.17.13>;tag=as73fe6d09
To: <sip:s at 192.168.17.114>;tag=00115c407fa3000948b81d9b-144916e7
Contact: <sip:asterisk at 192.168.17.13>
Call-ID: 74ee54ba09a3c0d31bc3d6853229181f at 192.168.17.13
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0
(no NAT) to 192.168.17.114:5060
Destroying call '74ee54ba09a3c0d31bc3d6853229181f at 192.168.17.13'
argon*CLI>
===================================================================
SIPDefault.cnf
# SIP Default Generic Configuration File
# Image Version
image_version: P0S3-07-2-00 ;
# Proxy Server
proxy1_address: "192.168.17.13" ; Can be dotted IP or FQDN
proxy2_address: "192.168.17.13" ; Can be dotted IP or FQDN
proxy3_address: "192.168.17.13" ; Can be dotted IP or FQDN
proxy4_address: "192.168.17.13" ; Can be dotted IP or FQDN
proxy5_address: "192.168.17.13" ; Can be dotted IP or FQDN
proxy6_address: "192.168.17.13" ; 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: none
# 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/
# Time Server (There are multiple values and configurations refer to
Admin Guide for Specifics)
sntp_server: "192.168.17.11" ; SNTP Server IP Address
sntp_mode: directedbroadcast ; unicast, multicast, anycast, or
directedbroadcast (default)
time_zone: YST ; Time Zone Phone is in
dst_offset: 1 ; Offset from Phone's time when DST is in effect
dst_start_month: April ; Month in which DST starts
dst_start_day: "" ; Day of month in which DST starts
dst_start_day_of_week: Sun ; Day of week in which DST starts
dst_start_week_of_month: 1 ; Week of month in which DST starts
dst_start_time: 02 ; Time of day in which DST starts
dst_stop_month: Oct ; Month in which DST stops
dst_stop_day: "" ; Day of month in which DST stops
dst_stop_day_of_week: Sunday ; Day of week in which DST stops
dst_stop_week_of_month: 8 ; Week of month in which DST stops 8=last
week of month
dst_stop_time: 2 ; Time of day in which DST stops
dst_auto_adjust: 1 ; Enable(1-Default)/Disable(0) DST automatic
adjustment
time_format_24hr: 0 ; Enable(1 - 24Hr Default)/Disable(0 - 12Hr)
# Do Not Disturb Control (0-off, 1-on, 2-off with no user control, 3-on
with no user control)
dnd_control: 0 ; Default 0 (Do Not Disturb feature is off)
# Caller ID Blocking (0-disbaled, 1-enabled, 2-disabled no user control,
3-enabled no user control)
callerid_blocking: 0 ; Default 0 (Disable sending all calls as
anonymous)
# Anonymous Call Blocking (0-disabled, 1-enabled, 2-disabled no user
control, 3-enabled no user control)
anonymous_call_block: 0 ; Default 0 (Disable blocking of
anonymous calls)
# DTMF AVT Payload (Dynamic payload range for AVT tones - 96-127)
dtmf_avt_payload: 101 ; Default 101
# Sync value of the phone used for remote reset
sync: 1 ; Default 1
####### New Parameters added in Release 2.1 #######
# Backup Proxy Support
proxy_backup: "" ; Dotted IP of Backup Proxy
proxy_backup_port: 5060 ; Backup Proxy port (default is 5060)
# Emergency Proxy Support
proxy_emergency: "" ; Dotted IP of Emergency Proxy
proxy_emergency_port: 5060 ; Emergency Proxy port (default is 5060)
# Configurable VAD option
enable_vad: 0 ; VAD setting 0-disable (Default), 1-enable
####### New Parameters added in Release 2.2 ######
# NAT/Firewall Traversal
nat_enable: 0 ; 0-Disabled (default), 1-Enabled
nat_address: "" ; WAN IP address of NAT box (dotted IP or
DNS A record only)
voip_control_port: 5060 ; UDP port used for SIP messages
(default - 5060)
start_media_port: 16384 ; Start RTP range for media (default - 16384)
end_media_port: 32766 ; End RTP range for media (default - 32766)
nat_received_processing: 0 ; 0-Disabled (default), 1-Enabled
# Outbound Proxy Support
outbound_proxy: "" ; restricted to dotted IP or DNS A record only
outbound_proxy_port: 5060 ; default is 5060
####### New Parameter added in Release 3.0 #######
# Allow for the bridge on a 3way call to join remaining parties upon hangup
cnf_join_enable : 1 ; 0-Disabled, 1-Enabled (default)
####### New Parameters added in Release 3.1 #######
# Allow Transfer to be completed while target phone is still ringing
semi_attended_transfer: 1 ; 0-Disabled, 1-Enabled (default)
# Telnet Level (enable or disable the ability to telnet into the phone)
telnet_level: 1 ; 0-Disabled (default), 1-Enabled, 2-Privileged
####### New Parameters added in Release 4.0 #######
# XML URLs
services_url: "" ; URL for external Phone Services
directory_url: "" ; URL for external Directory location
logo_url: "" ; URL for branding logo to be used on phone display
# HTTP Proxy Support
http_proxy_addr: "" ; Address of HTTP Proxy server
http_proxy_port: 80 ; Port of HTTP Proxy Server (80-default)
# Dynamic DNS/TFTP Support
dyn_dns_addr_1: "" ; restricted to dotted IP
dyn_dns_addr_2: "" ; restricted to dotted IP
dyn_tftp_addr: "" ; restricted to dotted IP
# Remote Party ID
remote_party_id: 0 ; 0-Disabled (default), 1-Enabled
####### New Parameters added in Release 4.4 #######
# Call Hold Ringback (0-disabled, 1-enabled, 2-disabled no user control,
3-enabled no user control)
call_hold_ringback: 0 ; Default 0 (Disable ringback of held call)
=====================================================================
SIP00115C407FA3.conf
# SIP Configuration Generic File
# Line 1
line1_name: scott
line1_authname: "scott"
line1_password: "scott"
# Line 2
line2_name: Line 2
line2_authname: "UNPROVISIONED"
line2_password: "UNPROVISIONED"
# Line 3
line2_name: "Line 2"
line2_authname: "UNPROVISIONED"
line2_password: "UNPROVISIONED"
# Line 4
line2_name: "Line 4"
line2_authname: "UNPROVISIONED"
line2_password: "UNPROVISIONED"
# Line 5
line2_name: "Line 5"
line2_authname: "UNPROVISIONED"
line2_password: "UNPROVISIONED"
# Line 6
line2_name: "Line 6"
line2_authname: "UNPROVISIONED"
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: "" ; Has no effect on SIP messaging
# Line 1 Display Name (Display name to use for SIP messaging)
line1_displayname: "User ID"
# 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: "SIP Phone" ; 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
========================================================================
The result of this process is that the 6101 fails to dial so the system
then dial 337-2860 and I can complete the call.
Any help would be appreciated on this
--
Scott Henderson
==========================================
Finite Technologies Incorporated
3763 Image Drive, Anchorage, Alaska 99504
Phone: 907.337.2860, Fax: 907.333.4482
http://www.finite-tech.com
http://www.chillywall.com
http://www.virtuale.cc
http://www.mphage.com
==========================================
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