asteriskstuff@ziplip.com
2004-Jul-18 05:13 UTC
[Asterisk-Users] Cisco 7960 SIP V6 and IBM A30P Fedora Asterisk
Hi All
Total noob on the list so all help appreciated....
I've successfully installed Asterisk on an IBM A30P Thinkpad using fedora
Core 2 (I'm looking at having a mobile PBX for conferences and shows).
I've plugged in two Cisco 7960 phones....
The phones register with the Asterisk correctly and I can run the demo's and
even the AIX demo through to digium works correctly.......
but I cannot get the phones to dial each other :(
Initially I was getting a "extension not found in local" message (when
dialling from console...from phone just engaged (busy) tone.
when I add extension XXXX from console I now get a "not found 404"
message....I see that there was an earlier thread on the list that discussed
removing the proxy forwarding from the phone settings and I've tried that
from SIPDefault.cnf but it doesn't fix the problem.....
I've obviously missed something but am too inexperienced to spot it.
P
my files are as follows:-
--------------------------------
sipxxxxxx.cnf
# Lounge Phone Settings
# Line 1 Settings
line1_name: "11" ; Line 1 Extension\User ID
line1_displayname: "Lounge1" ; Line 1 Display Name
line1_authname: "lounge11" ; Line 1 Registration Authentication
line1_password: "lounge" ; Line 1 Registration Password
-------------------------
sipdefault.cnf
# Image Version
image_version: P0S3-06-3-00
# Proxy Server
proxy1_address: ipaddress of A30P ; Can be dotted IP or FQDN
proxy1_port:
5060
# Proxy Registration (0-disable (default), 1-enable)
proxy_register: 0
# Phone Registration Expiration [1-3932100 sec] (Default - 3600)
timer_register_expires: 3600
# Codec for media stream (g711ulaw (default), g711alaw, g729a)
preferred_codec: g711ulaw
# 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
# 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: "137.222.10.60" ; SNTP Server IP Address
sntp_mode: anycast ; unicast, multicast, anycast, or directedbroadcast (default)
time_zone: GMT ; Time Zone Phone is in
dst_offset: 1 ; Offset from Phone's time when BST is in effect
dst_start_month: April ; Month in which BST starts
dst_start_day: "21" ; Day of month in which BST starts
dst_start_day_of_week: Sun ; Day of week in which BST starts
dst_start_week_of_month: 1 ; Week of month in which BST starts
dst_start_time: 02 ; Time of day in which BST starts
dst_stop_month: Oct ; Month in which BST stops
dst_stop_day: "20" ; Day of month in which BST stops
dst_stop_day_of_week: Sunday ; Day of week in which BST stops
dst_stop_week_of_month: 8 ; Week of month in which BST stops 8=last week of
month
dst_stop_time: 2 ; Time of day in which BST stops
dst_auto_adjust: 1 ; Enable(1-Default)/Disable(0) BST automatic adjustment
time_format_24hr: 1 ; Enable(1 - 24Hr Default)/Disable(0 - 12Hr)
dnd_control: 0 ; Default 0 (0=off, 1=on, 2=off no user cntrl, 3=on no user
control)
callerid_blocking: 0 ; Default 0 (Disable sending all calls as anonymous)
anonymous_call_block: 0 ; Default 0 (Disable blocking of anonymous calls)
dtmf_avt_payload: 101 ; Default 101
# Sync value of the phone used for remote reset
sync: 1 ; Default 1
proxy_backup: "" ; Dotted IP of Backup Proxy
proxy_backup_port: 5060 ; Backup Proxy port (default is 5060)
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
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: "" ; restricted to dotted IP or DNS A record only
outbound_proxy_port: 5060 ; default is 5060
# Allow for the bridge on a 3way call to join remaining parties upon hangup
cnf_join_enable : 1 ; 0-Disabled, 1-Enabled (default)
# 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
# XML URLs
;services_url: "http://your.site/services.xml" ; URL for external
Phone Services
services_url: "http://193.113.58.136/bt/" ;bt services
directory_url: "http://your.site/directory.xml" ; URL for external
Directory location
logo_url: "http://your.site/logo.bmp" ; URL for branding logo to be
used on phone display
# HTTP Proxy Support
http_proxy_addr: "http://ipaddress of A30P/SipPhoneProxy/" ; 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: 1 ; 0-Disabled (default), 1-Enabled
# 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
-----------------------------------------------------
sip.conf
;
; SIP Configuration for Asterisk
;
; Syntax for specifying a SIP device in extensions.conf is
; SIP/devicename where devicename is defined in a section below.
;
; You may also use
; SIP/username@domain to call any SIP user on the Internet
; (Don't forget to enable DNS SRV records if you want to use this)
;
; If you define a SIP proxy as a peer below, you may call
; SIP/proxyhostname/user or SIP/user@proxyhostname
; where the proxyhostname is defined in a section below
;
; Useful CLI commands to check peers/users:
; sip show peers Show all SIP peers (including friends)
; sip show users Show all SIP users (including friends)
; sip show registry Show status of hosts we register with
;
; sip debug Show all SIP messages
;
[general]
context=default ; Default context for incoming calls
;recordhistory=yes ; Record SIP history by default (see sip history / sip no
history)
;realm=mydomain.tld ; Realm for digest authentication
; defaults to "asterisk"
; Realms MUST be globally unique according to RFC 3261
; Set this to your host name or domain name
port=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; Note: Asterisk only uses the first host
; in SRV records
; Disabling DNS SRV lookups disables the
; ability to place SIP calls based on domain
; names to some other SIP users on the Internet
;pedantic=yes ; Enable slow, pedantic checking for Pingtel
; and multiline formatted headers for strict
; SIP compatibility
;tos=184 ; Set IP QoS to either a keyword or numeric val
;tos=lowdelay ; lowdelay,throughput,reliability,mincost,none
;maxexpirey=3600 ; Max length of incoming registration we allow
;defaultexpirey=120 ; Default length of incoming/outoing registration
;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY
;videosupport=yes ; Turn on support for SIP video
;disallow=all ; First disallow all codecs
;allow=ulaw ; Allow codecs in order of preference
;allow=ilbc ; Note: codec order is respected only in [general]
;musicclass=default ; Sets the default music on hold class for all SIP calls
; This may also be set for individual users/peers
;language=en ; Default language setting for all users/peers
; This may also be set for individual users/peers
;relaxdtmf=yes ; Relax dtmf handling
;rtptimeout=60 ; Terminate call if 60 seconds of no RTP activity
; when we're not on hold
;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP activity
; when we're on hold (must be > rtptimeout)
;trustrpid = no ; If Remote-Party-ID should be trusted
;progressinband=no ; If we should generate in-band ringing always
;useragent=Asterisk PBX ; Allows you to change the user agent string
;nat=no ; NAT settings
; yes = Always ignore info and assume NAT
; no = Use NAT mode only according to RFC3581
; never = Never attempt NAT mode or RFC3581
support
;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address
; Asterisk can register as a SIP user agent to a SIP proxy (provider)
; Format for the register statement is:
; register => user[:secret[:authuser]]@host[:port][/extension]
;
; If no extension is given, the 's' extension is used. The extension
; needs to be defined in extensions.conf to be able to accept calls
; from this SIP proxy (provider)
;
; host is either a host name defined in DNS or the name of a
; section defined below.
;
; Examples:
;
;register => 1234:password@mysipprovider.com
;
; This will pass incoming calls to the 's' extension
;
;
;register => 2345:password@sip_proxy/1234
;
; Register 2345 at sip provider 'sip_proxy'. Calls from this
provider connect to local
; extension 1234 in extensions.conf default context, unless you define
; unless you configure a [sip_proxy] section below, and configure a context.
; Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com]
; Tip 2: Use separate type=peer and type=user sections for SIP providers
; (instead of type=friend) if you have calls in both
directions
;externip = 200.201.202.203 ; Address that we're going to put in outbound
SIP messages
; if we're behind a NAT
; The externip and localnet is used
; when registering and communicating with other proxies
; that we're registered with
; You may add multiple local networks. A reasonable set of defaults
; are:
;localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks
;localnet=10.0.0.0/255.0.0.0 ; Also RFC1918
;localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation
;localnet=169.254.0.0/255.255.0.0 ;Zero conf local network
;-----------------------------------------------------------------------------------
; Users and peers have different settings available. Friends have all settings,
; since a friend is both a peer and a user
;
; User config options: Peer configuration:
; -------------------- -------------------
; context context
; permit permit
; deny deny
; auth auth
; secret secret
; md5secret md5secret
; dtmfmode dtmfmode
; canreinvite canreinvite
; nat nat
; callgroup callgroup
; pickupgroup pickupgroup
; language language
; allow allow
; disallow disallow
; insecure insecure
; trustrpid trustrpid
; progressinband progressinband
; promiscredir promiscredir
; callerid
; accountcode
; amaflags
; incominglimit
; outgoinglimit
; restrictcid
; mailbox
; username
; template
; fromdomain
; fromuser
; host
; mask
; port
; qualify
; defaultip
; rtptimeout
; rtpholdtimeout
;[sip_proxy]
; For incoming calls only. Example: FWD (Free World Dialup)
;type=user
;context=from-fwd
;[sip_proxy-out]
;type=peer ; we only want to call out, not be called
;secret=guessit
;username=yourusername
;fromuser=yourusername ; Many SIP providers require this!
;host=box.provider.com
;[grandstream1]
;type=friend ; either "friend" (peer+user),
"peer" or "user"
;context=from-sip
;username=grandstream1 ; usually matches the [section] title
;fromuser=grandstream1 ; overrides the callerid, e.g. required by FWD
;callerid=John Doe <1234>
;host=192.168.0.23 ; we have a static but private IP address
;nat=no ; there is not NAT between phone and Asterisk
;canreinvite=yes ; allow RTP voice traffic to bypass Asterisk
;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone
;outgoinglimit=1 ; disable callwaiting signal (2nd call to phone)
;incominglimit=1 ; permit only 1 outgoing call at a time
;mailbox=1234@default ; mailbox 1234 in voicemail context "default"
;disallow=all ; need to disallow=all before we can use
allow;allow=ulaw ; Note: In user sections the order of codecs
; listed with allow= does NOT matter!
;allow=alaw
;allow=g723.1 ; Asterisk only supports g723.1 pass-thru!
;allow=g729 ; Pass-thru only unless g729 license obtained
[phone1]
type=friend
username=phone1
secret=lounge
qualify=100 ; Qualify peer is no more than 200ms away
host=10.131.111.41
defaultip=10.131.111.41 ; This device registers with us
mailbox=1000 ; mailbox for message waiting indicator
context=sip
callerid="Lounge1" <1>
[phone2]
type=friend
username=phone2
secret=kitchen
qualify=100
host=10.131.111.42
defaultip=10.131.111.42
mailbox=2000
context=sip
callerid="Kitchen1" <2>
----------------------------------------
extensions.conf
[default]
;
; By default we include the demo. In a production system, you
; probably don't want to have the demo there.
;
include => demo
;
[sip]
exten => 5511,1,Dial(SIP/phone1,15,t)
exten => 5521,1,Dial(SIP/phone2,15,t)
exten => 1000,1,Dial(SIP/phone1,15,t)
Wiley E. Siler
2004-Jul-18 10:51 UTC
[Asterisk-Users] Cisco 7960 SIP V6 and IBM A30P Fedora Asterisk
I just started out too and I can tell you it is easier to start from
scratch with a good wiki then alter the demo files. Here is a wiki you
can build a good working system with...
http://www.wlug.org.nz/AsteriskSampleSetup
For your ciscos search http://asterisk.xvoip.com/index.php
Wiley
-----Original Message-----
From: asteriskstuff@ziplip.com [mailto:asteriskstuff@ziplip.com]
Sent: Sunday, July 18, 2004 5:13 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Cisco 7960 SIP V6 and IBM A30P Fedora Asterisk
Hi All
Total noob on the list so all help appreciated....
I've successfully installed Asterisk on an IBM A30P Thinkpad using
fedora Core 2 (I'm looking at having a mobile PBX for conferences and
shows).
I've plugged in two Cisco 7960 phones....
The phones register with the Asterisk correctly and I can run the demo's
and even the AIX demo through to digium works correctly.......
but I cannot get the phones to dial each other :(
Initially I was getting a "extension not found in local" message (when
dialling from console...from phone just engaged (busy) tone.
when I add extension XXXX from console I now get a "not found 404"
message....I see that there was an earlier thread on the list that
discussed removing the proxy forwarding from the phone settings and I've
tried that from SIPDefault.cnf but it doesn't fix the problem.....
I've obviously missed something but am too inexperienced to spot it.
P
my files are as follows:-
--------------------------------
sipxxxxxx.cnf
# Lounge Phone Settings
# Line 1 Settings
line1_name: "11" ; Line 1 Extension\User ID
line1_displayname: "Lounge1" ; Line 1 Display Name
line1_authname: "lounge11" ; Line 1 Registration Authentication
line1_password: "lounge" ; Line 1 Registration Password
-------------------------
sipdefault.cnf
# Image Version
image_version: P0S3-06-3-00
# Proxy Server
proxy1_address: ipaddress of A30P ; Can be dotted IP or FQDN
proxy1_port:
5060
# Proxy Registration (0-disable (default), 1-enable)
proxy_register: 0
# Phone Registration Expiration [1-3932100 sec] (Default - 3600)
timer_register_expires: 3600
# Codec for media stream (g711ulaw (default), g711alaw, g729a)
preferred_codec: g711ulaw
# 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
# 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: "137.222.10.60" ; SNTP Server IP Address
sntp_mode: anycast ; unicast, multicast, anycast, or directedbroadcast
(default)
time_zone: GMT ; Time Zone Phone is in
dst_offset: 1 ; Offset from Phone's time when BST is in effect
dst_start_month: April ; Month in which BST starts
dst_start_day: "21" ; Day of month in which BST starts
dst_start_day_of_week: Sun ; Day of week in which BST starts
dst_start_week_of_month: 1 ; Week of month in which BST starts
dst_start_time: 02 ; Time of day in which BST starts
dst_stop_month: Oct ; Month in which BST stops
dst_stop_day: "20" ; Day of month in which BST stops
dst_stop_day_of_week: Sunday ; Day of week in which BST stops
dst_stop_week_of_month: 8 ; Week of month in which BST stops 8=last week
of month
dst_stop_time: 2 ; Time of day in which BST stops
dst_auto_adjust: 1 ; Enable(1-Default)/Disable(0) BST automatic
adjustment
time_format_24hr: 1 ; Enable(1 - 24Hr Default)/Disable(0 - 12Hr)
dnd_control: 0 ; Default 0 (0=off, 1=on, 2=off no user cntrl, 3=on no
user control)
callerid_blocking: 0 ; Default 0 (Disable sending all calls as
anonymous)
anonymous_call_block: 0 ; Default 0 (Disable blocking of anonymous
calls)
dtmf_avt_payload: 101 ; Default 101
# Sync value of the phone used for remote reset
sync: 1 ; Default 1
proxy_backup: "" ; Dotted IP of Backup Proxy
proxy_backup_port: 5060 ; Backup Proxy port (default is 5060)
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
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: "" ; restricted to dotted IP or DNS A record only
outbound_proxy_port: 5060 ; default is 5060
# Allow for the bridge on a 3way call to join remaining parties upon
hangup
cnf_join_enable : 1 ; 0-Disabled, 1-Enabled (default)
# 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
# XML URLs
;services_url: "http://your.site/services.xml" ; URL for external
Phone
Services
services_url: "http://193.113.58.136/bt/" ;bt services
directory_url: "http://your.site/directory.xml" ; URL for external
Directory location
logo_url: "http://your.site/logo.bmp" ; URL for branding logo to be
used
on phone display
# HTTP Proxy Support
http_proxy_addr: "http://ipaddress of A30P/SipPhoneProxy/" ; 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: 1 ; 0-Disabled (default), 1-Enabled
# 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
-----------------------------------------------------
sip.conf
;
; SIP Configuration for Asterisk
;
; Syntax for specifying a SIP device in extensions.conf is ;
SIP/devicename where devicename is defined in a section below.
;
; You may also use
; SIP/username@domain to call any SIP user on the Internet ; (Don't
forget to enable DNS SRV records if you want to use this) ; ; If you
define a SIP proxy as a peer below, you may call ;
SIP/proxyhostname/user or SIP/user@proxyhostname ; where the
proxyhostname is defined in a section below ; ; Useful CLI commands to
check peers/users:
; sip show peers Show all SIP peers (including friends)
; sip show users Show all SIP users (including friends)
; sip show registry Show status of hosts we register with
;
; sip debug Show all SIP messages
;
[general]
context=default ; Default context for incoming calls
;recordhistory=yes ; Record SIP history by default (see sip
history / sip no history)
;realm=mydomain.tld ; Realm for digest authentication
; defaults to "asterisk"
; Realms MUST be globally unique
according to RFC 3261
; Set this to your host name or domain
name
port=5060 ; UDP Port to bind to (SIP standard port
is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds
to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound
calls
; Note: Asterisk only uses the first
host
; in SRV records
; Disabling DNS SRV lookups disables the
; ability to place SIP calls based on
domain
; names to some other SIP users on the
Internet
;pedantic=yes ; Enable slow, pedantic checking for
Pingtel
; and multiline formatted headers for
strict
; SIP compatibility
;tos=184 ; Set IP QoS to either a keyword or
numeric val
;tos=lowdelay ;
lowdelay,throughput,reliability,mincost,none
;maxexpirey=3600 ; Max length of incoming registration we
allow
;defaultexpirey=120 ; Default length of incoming/outoing
registration
;notifymimetype=text/plain ; Allow overriding of mime type in
NOTIFY
;videosupport=yes ; Turn on support for SIP video
;disallow=all ; First disallow all codecs
;allow=ulaw ; Allow codecs in order of preference
;allow=ilbc ; Note: codec order is respected only in
[general]
;musicclass=default ; Sets the default music on hold class
for all SIP calls
; This may also be set for individual
users/peers
;language=en ; Default language setting for all
users/peers
; This may also be set for individual
users/peers
;relaxdtmf=yes ; Relax dtmf handling
;rtptimeout=60 ; Terminate call if 60 seconds of no RTP
activity
; when we're not on hold
;rtpholdtimeout=300 ; Terminate call if 300 seconds of no
RTP activity
; when we're on hold (must be >
rtptimeout)
;trustrpid = no ; If Remote-Party-ID should be trusted
;progressinband=no ; If we should generate in-band ringing
always
;useragent=Asterisk PBX ; Allows you to change the user agent
string
;nat=no ; NAT settings
; yes = Always ignore info and assume
NAT
; no = Use NAT mode only according to
RFC3581
; never = Never attempt NAT mode or
RFC3581 support
;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP
address
; Asterisk can register as a SIP user agent to a SIP proxy (provider) ;
Format for the register statement is:
; register => user[:secret[:authuser]]@host[:port][/extension]
;
; If no extension is given, the 's' extension is used. The extension ;
needs to be defined in extensions.conf to be able to accept calls ; from
this SIP proxy (provider) ; ; host is either a host name defined in DNS
or the name of a ; section defined below.
;
; Examples:
;
;register => 1234:password@mysipprovider.com
;
; This will pass incoming calls to the 's' extension
;
;
;register => 2345:password@sip_proxy/1234 ;
; Register 2345 at sip provider 'sip_proxy'. Calls from this
provider connect to local
; extension 1234 in extensions.conf default context, unless you
define
; unless you configure a [sip_proxy] section below, and configure a
context.
; Tip 1: Avoid assigning hostname to a sip.conf section like
[provider.com]
; Tip 2: Use separate type=peer and type=user sections for SIP
providers
; (instead of type=friend) if you have calls in
both directions
;externip = 200.201.202.203 ; Address that we're going to put in
outbound SIP messages
; if we're behind a NAT
; The externip and localnet is used
; when registering and communicating
with other proxies
; that we're registered with
; You may add multiple local networks.
A reasonable set of defaults
; are:
;localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local
networks
;localnet=10.0.0.0/255.0.0.0 ; Also RFC1918
;localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation
;localnet=169.254.0.0/255.255.0.0 ;Zero conf local network
;-----------------------------------------------------------------------
------------
; Users and peers have different settings available. Friends have all
settings,
; since a friend is both a peer and a user
;
; User config options: Peer configuration:
; -------------------- -------------------
; context context
; permit permit
; deny deny
; auth auth
; secret secret
; md5secret md5secret
; dtmfmode dtmfmode
; canreinvite canreinvite
; nat nat
; callgroup callgroup
; pickupgroup pickupgroup
; language language
; allow allow
; disallow disallow
; insecure insecure
; trustrpid trustrpid
; progressinband progressinband
; promiscredir promiscredir
; callerid
; accountcode
; amaflags
; incominglimit
; outgoinglimit
; restrictcid
; mailbox
; username
; template
; fromdomain
; fromuser
; host
; mask
; port
; qualify
; defaultip
; rtptimeout
; rtpholdtimeout
;[sip_proxy]
; For incoming calls only. Example: FWD (Free World Dialup)
;type=user
;context=from-fwd
;[sip_proxy-out]
;type=peer ; we only want to call out, not be called
;secret=guessit
;username=yourusername
;fromuser=yourusername ; Many SIP providers require this!
;host=box.provider.com
;[grandstream1]
;type=friend ; either "friend" (peer+user),
"peer" or
"user"
;context=from-sip
;username=grandstream1 ; usually matches the [section] title
;fromuser=grandstream1 ; overrides the callerid, e.g. required
by FWD
;callerid=John Doe <1234>
;host=192.168.0.23 ; we have a static but private IP address
;nat=no ; there is not NAT between phone and
Asterisk
;canreinvite=yes ; allow RTP voice traffic to bypass
Asterisk
;dtmfmode=info ; either RFC2833 or INFO for the
BudgeTone
;outgoinglimit=1 ; disable callwaiting signal (2nd call to
phone)
;incominglimit=1 ; permit only 1 outgoing call at a time
;mailbox=1234@default ; mailbox 1234 in voicemail context "default"
;disallow=all ; need to disallow=all before we can use
allow;allow=ulaw ; Note: In user sections the order of
codecs
; listed with allow= does NOT matter!
;allow=alaw
;allow=g723.1 ; Asterisk only supports g723.1
pass-thru!
;allow=g729 ; Pass-thru only unless g729 license
obtained
[phone1]
type=friend
username=phone1
secret=lounge
qualify=100 ; Qualify peer is no more than 200ms
away
host=10.131.111.41
defaultip=10.131.111.41 ; This device registers with us
mailbox=1000 ; mailbox for message waiting indicator
context=sip
callerid="Lounge1" <1>
[phone2]
type=friend
username=phone2
secret=kitchen
qualify=100
host=10.131.111.42
defaultip=10.131.111.42
mailbox=2000
context=sip
callerid="Kitchen1" <2>
----------------------------------------
extensions.conf
[default]
;
; By default we include the demo. In a production system, you
; probably don't want to have the demo there.
;
include => demo
;
[sip]
exten => 5511,1,Dial(SIP/phone1,15,t)
exten => 5521,1,Dial(SIP/phone2,15,t)
exten => 1000,1,Dial(SIP/phone1,15,t)
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Sean Cheesman
2004-Jul-18 11:07 UTC
[Asterisk-Users] Cisco 7960 SIP V6 and IBM A30P Fedora Asterisk
It doesn't look like you have a context set for phone1. Try putting
context=sip in the phone1 section like you have in phone2. That'll put
both in the same context of your extensions.conf file and should allow
interaction between the two.
-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of
asteriskstuff@ziplip.com
Sent: Sunday, July 18, 2004 7:13 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Cisco 7960 SIP V6 and IBM A30P Fedora Asterisk
Hi All
Total noob on the list so all help appreciated....
I've successfully installed Asterisk on an IBM A30P Thinkpad using
fedora Core 2 (I'm looking at having a mobile PBX for conferences and
shows).
I've plugged in two Cisco 7960 phones....
The phones register with the Asterisk correctly and I can run the demo's
and even the AIX demo through to digium works correctly.......
but I cannot get the phones to dial each other :(
Initially I was getting a "extension not found in local" message (when
dialling from console...from phone just engaged (busy) tone.
when I add extension XXXX from console I now get a "not found 404"
message....I see that there was an earlier thread on the list that
discussed removing the proxy forwarding from the phone settings and I've
tried that from SIPDefault.cnf but it doesn't fix the problem.....
I've obviously missed something but am too inexperienced to spot it. P
my files are as follows:-
--------------------------------
sipxxxxxx.cnf
# Lounge Phone Settings
# Line 1 Settings
line1_name: "11" ; Line 1 Extension\User ID
line1_displayname: "Lounge1" ; Line 1 Display Name
line1_authname: "lounge11" ; Line 1 Registration Authentication
line1_password: "lounge" ; Line 1 Registration Password
-------------------------
sipdefault.cnf
# Image Version
image_version: P0S3-06-3-00
# Proxy Server
proxy1_address: ipaddress of A30P ; Can be dotted IP or FQDN
proxy1_port:
5060
# Proxy Registration (0-disable (default), 1-enable)
proxy_register: 0
# Phone Registration Expiration [1-3932100 sec] (Default - 3600)
timer_register_expires: 3600
# Codec for media stream (g711ulaw (default), g711alaw, g729a)
preferred_codec: g711ulaw
# 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
# 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: "137.222.10.60" ; SNTP Server IP Address
sntp_mode: anycast ; unicast, multicast, anycast, or directedbroadcast
(default)
time_zone: GMT ; Time Zone Phone is in
dst_offset: 1 ; Offset from Phone's time when BST is in effect
dst_start_month: April ; Month in which BST starts
dst_start_day: "21" ; Day of month in which BST starts
dst_start_day_of_week: Sun ; Day of week in which BST starts
dst_start_week_of_month: 1 ; Week of month in which BST starts
dst_start_time: 02 ; Time of day in which BST starts
dst_stop_month: Oct ; Month in which BST stops
dst_stop_day: "20" ; Day of month in which BST stops
dst_stop_day_of_week: Sunday ; Day of week in which BST stops
dst_stop_week_of_month: 8 ; Week of month in which BST stops 8=last week
of month
dst_stop_time: 2 ; Time of day in which BST stops
dst_auto_adjust: 1 ; Enable(1-Default)/Disable(0) BST automatic
adjustment
time_format_24hr: 1 ; Enable(1 - 24Hr Default)/Disable(0 - 12Hr)
dnd_control: 0 ; Default 0 (0=off, 1=on, 2=off no user cntrl, 3=on no
user control)
callerid_blocking: 0 ; Default 0 (Disable sending all calls as
anonymous)
anonymous_call_block: 0 ; Default 0 (Disable blocking of anonymous
calls)
dtmf_avt_payload: 101 ; Default 101
# Sync value of the phone used for remote reset
sync: 1 ; Default 1
proxy_backup: "" ; Dotted IP of Backup Proxy
proxy_backup_port: 5060 ; Backup Proxy port (default is 5060)
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
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: "" ; restricted to dotted IP or DNS A record only
outbound_proxy_port: 5060 ; default is 5060
# Allow for the bridge on a 3way call to join remaining parties upon
hangup
cnf_join_enable : 1 ; 0-Disabled, 1-Enabled (default)
# 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
# XML URLs
;services_url: "http://your.site/services.xml" ; URL for external
Phone
Services
services_url: "http://193.113.58.136/bt/" ;bt services
directory_url: "http://your.site/directory.xml" ; URL for external
Directory location
logo_url: "http://your.site/logo.bmp" ; URL for branding logo to be
used
on phone display
# HTTP Proxy Support
http_proxy_addr: "http://ipaddress of A30P/SipPhoneProxy/" ; 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: 1 ; 0-Disabled (default), 1-Enabled
# 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
-----------------------------------------------------
sip.conf
;
; SIP Configuration for Asterisk
;
; Syntax for specifying a SIP device in extensions.conf is
; SIP/devicename where devicename is defined in a section below. ; ; You
may also use
; SIP/username@domain to call any SIP user on the Internet
; (Don't forget to enable DNS SRV records if you want to use this) ;
; If you define a SIP proxy as a peer below, you may call
; SIP/proxyhostname/user or SIP/user@proxyhostname
; where the proxyhostname is defined in a section below
;
; Useful CLI commands to check peers/users:
; sip show peers Show all SIP peers (including friends)
; sip show users Show all SIP users (including friends)
; sip show registry Show status of hosts we register with
;
; sip debug Show all SIP messages
;
[general]
context=default ; Default context for incoming calls
;recordhistory=yes ; Record SIP history by default (see sip
history / sip no history)
;realm=mydomain.tld ; Realm for digest authentication
; defaults to "asterisk"
; Realms MUST be globally unique
according to RFC 3261
; Set this to your host name or domain
name
port=5060 ; UDP Port to bind to (SIP standard port
is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds
to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound
calls
; Note: Asterisk only uses the first
host
; in SRV records
; Disabling DNS SRV lookups disables the
; ability to place SIP calls based on
domain
; names to some other SIP users on the
Internet
;pedantic=yes ; Enable slow, pedantic checking for
Pingtel
; and multiline formatted headers for
strict
; SIP compatibility
;tos=184 ; Set IP QoS to either a keyword or
numeric val
;tos=lowdelay ;
lowdelay,throughput,reliability,mincost,none
;maxexpirey=3600 ; Max length of incoming registration we
allow
;defaultexpirey=120 ; Default length of incoming/outoing
registration
;notifymimetype=text/plain ; Allow overriding of mime type in
NOTIFY
;videosupport=yes ; Turn on support for SIP video
;disallow=all ; First disallow all codecs
;allow=ulaw ; Allow codecs in order of preference
;allow=ilbc ; Note: codec order is respected only in
[general]
;musicclass=default ; Sets the default music on hold class
for all SIP calls
; This may also be set for individual
users/peers
;language=en ; Default language setting for all
users/peers
; This may also be set for individual
users/peers
;relaxdtmf=yes ; Relax dtmf handling
;rtptimeout=60 ; Terminate call if 60 seconds of no RTP
activity
; when we're not on hold
;rtpholdtimeout=300 ; Terminate call if 300 seconds of no
RTP activity
; when we're on hold (must be >
rtptimeout)
;trustrpid = no ; If Remote-Party-ID should be trusted
;progressinband=no ; If we should generate in-band ringing
always
;useragent=Asterisk PBX ; Allows you to change the user agent
string
;nat=no ; NAT settings
; yes = Always ignore info and assume
NAT
; no = Use NAT mode only according to
RFC3581
; never = Never attempt NAT mode or
RFC3581 support
;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP
address
; Asterisk can register as a SIP user agent to a SIP proxy (provider) ;
Format for the register statement is:
; register => user[:secret[:authuser]]@host[:port][/extension]
;
; If no extension is given, the 's' extension is used. The extension ;
needs to be defined in extensions.conf to be able to accept calls ; from
this SIP proxy (provider) ; ; host is either a host name defined in DNS
or the name of a
; section defined below.
;
; Examples:
;
;register => 1234:password@mysipprovider.com
;
; This will pass incoming calls to the 's' extension
;
;
;register => 2345:password@sip_proxy/1234
;
; Register 2345 at sip provider 'sip_proxy'. Calls from this
provider connect to local
; extension 1234 in extensions.conf default context, unless you
define
; unless you configure a [sip_proxy] section below, and configure a
context.
; Tip 1: Avoid assigning hostname to a sip.conf section like
[provider.com]
; Tip 2: Use separate type=peer and type=user sections for SIP
providers
; (instead of type=friend) if you have calls in
both directions
;externip = 200.201.202.203 ; Address that we're going to put in
outbound SIP messages
; if we're behind a NAT
; The externip and localnet is used
; when registering and communicating
with other proxies
; that we're registered with
; You may add multiple local networks.
A reasonable set of defaults
; are:
;localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local
networks
;localnet=10.0.0.0/255.0.0.0 ; Also RFC1918
;localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation
;localnet=169.254.0.0/255.255.0.0 ;Zero conf local network
;-----------------------------------------------------------------------
------------
; Users and peers have different settings available. Friends have all
settings, ; since a friend is both a peer and a user ;
; User config options: Peer configuration:
; -------------------- -------------------
; context context
; permit permit
; deny deny
; auth auth
; secret secret
; md5secret md5secret
; dtmfmode dtmfmode
; canreinvite canreinvite
; nat nat
; callgroup callgroup
; pickupgroup pickupgroup
; language language
; allow allow
; disallow disallow
; insecure insecure
; trustrpid trustrpid
; progressinband progressinband
; promiscredir promiscredir
; callerid
; accountcode
; amaflags
; incominglimit
; outgoinglimit
; restrictcid
; mailbox
; username
; template
; fromdomain
; fromuser
; host
; mask
; port
; qualify
; defaultip
; rtptimeout
; rtpholdtimeout
;[sip_proxy]
; For incoming calls only. Example: FWD (Free World Dialup) ;type=user
;context=from-fwd
;[sip_proxy-out]
;type=peer ; we only want to call out, not be called
;secret=guessit
;username=yourusername
;fromuser=yourusername ; Many SIP providers require this!
;host=box.provider.com
;[grandstream1]
;type=friend ; either "friend" (peer+user),
"peer" or
"user"
;context=from-sip
;username=grandstream1 ; usually matches the [section] title
;fromuser=grandstream1 ; overrides the callerid, e.g. required
by FWD
;callerid=John Doe <1234>
;host=192.168.0.23 ; we have a static but private IP address
;nat=no ; there is not NAT between phone and
Asterisk
;canreinvite=yes ; allow RTP voice traffic to bypass
Asterisk
;dtmfmode=info ; either RFC2833 or INFO for the
BudgeTone
;outgoinglimit=1 ; disable callwaiting signal (2nd call to
phone)
;incominglimit=1 ; permit only 1 outgoing call at a time
;mailbox=1234@default ; mailbox 1234 in voicemail context "default"
;disallow=all ; need to disallow=all before we can use
allow;allow=ulaw ; Note: In user sections the order of
codecs
; listed with allow= does NOT matter!
;allow=alaw
;allow=g723.1 ; Asterisk only supports g723.1
pass-thru!
;allow=g729 ; Pass-thru only unless g729 license
obtained
[phone1]
type=friend
username=phone1
secret=lounge
qualify=100 ; Qualify peer is no more than 200ms
away
host=10.131.111.41
defaultip=10.131.111.41 ; This device registers with us
mailbox=1000 ; mailbox for message waiting indicator context=sip
callerid="Lounge1" <1>
[phone2]
type=friend
username=phone2
secret=kitchen
qualify=100
host=10.131.111.42
defaultip=10.131.111.42
mailbox=2000
context=sip
callerid="Kitchen1" <2>
----------------------------------------
extensions.conf
[default]
;
; By default we include the demo. In a production system, you
; probably don't want to have the demo there.
;
include => demo
;
[sip]
exten => 5511,1,Dial(SIP/phone1,15,t)
exten => 5521,1,Dial(SIP/phone2,15,t)
exten => 1000,1,Dial(SIP/phone1,15,t)
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asteriskstuff@ziplip.com
2004-Jul-18 12:46 UTC
[Asterisk-Users] Cisco 7960 SIP V6 and IBM A30P Fedora Asterisk
Hi Sean Both phones are set for context=sip in the sip.conf file. As I say the phones will both call out OK (I can dial the 500 test number and successfully connect to the remote PBX through my firewall). It's just that when I'm trying to call from phone to phone I'm getting the 404 not found error in the asteris verbose dialog. If anyone has a documented example of their 7960 config sipdefault.cnf and sipxxxxxipadd.cnf files together with their sip.conf and extensions.conf files I could have to test directly on my system I'd be appreciative to test them on my system. While the WiKi's are very useful as example files it would be great (and I may do it myself!!) if there was an up to date example file with all the options for each filed and a verbose description for the rational behind it (although I recognise that this is an 'in development' product and therefore the docs have to be done at the end!!). Part of the problem is there are so many dependencies that can affect the system including how the dhpcd server serves IP address's and associated files (for example the files have to be structured in a particular order on the tftpd server for the cisco's to pick them up correctly). Given this level of dependency I'm not sure where the break could be. The one thing I have noticed from the show sip peers field is that it's showing the phones as having a netmask of 255.255.255.255 although they're actually configyred for 255.255.255.0. P> -----Original Message----- > From: Sean Cheesman [mailto:scheesman@caeveo.com] > Sent: Sunday, July 18, 2004, 11:37 AM > To: asterisk-users@lists.digium.com > Subject: RE: [Asterisk-Users] Cisco 7960 SIP V6 and IBM A30P Fedora Asterisk > > It doesn't look like you have a context set for phone1. Try putting > context=sip in the phone1 section like you have in phone2. That'll put > both in the same context of your extensions.conf file and should allow > interaction between the two. > > -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of > asteriskstuff@ziplip.com > Sent: Sunday, July 18, 2004 7:13 AM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Cisco 7960 SIP V6 and IBM A30P Fedora Asterisk > > > Hi All > > Total noob on the list so all help appreciated.... > > I've successfully installed Asterisk on an IBM A30P Thinkpad using > fedora Core 2 (I'm looking at having a mobile PBX for conferences and > shows). > > I've plugged in two Cisco 7960 phones.... > > The phones register with the Asterisk correctly and I can run the demo's > and even the AIX demo through to digium works correctly....... > > but I cannot get the phones to dial each other :( > > Initially I was getting a "extension not found in local" message (when > dialling from console...from phone just engaged (busy) tone. > > when I add extension XXXX from console I now get a "not found 404" > message....I see that there was an earlier thread on the list that > discussed removing the proxy forwarding from the phone settings and I've > tried that from SIPDefault.cnf but it doesn't fix the problem..... > > I've obviously missed something but am too inexperienced to spot it. P > > my files are as follows:- > > -------------------------------- > > sipxxxxxx.cnf > > > # Lounge Phone Settings > > # Line 1 Settings > line1_name: "11" ; Line 1 Extension\User ID > line1_displayname: "Lounge1" ; Line 1 Display Name > line1_authname: "lounge11" ; Line 1 Registration Authentication > line1_password: "lounge" ; Line 1 Registration Password > > ------------------------- > > sipdefault.cnf > > # Image Version > > image_version: P0S3-06-3-00 > > # Proxy Server > > proxy1_address: ipaddress of A30P ; Can be dotted IP or FQDN > > proxy1_port: > 5060 > # Proxy Registration (0-disable (default), 1-enable) > > proxy_register: 0 > > # Phone Registration Expiration [1-3932100 sec] (Default - 3600) > > timer_register_expires: 3600 > > # Codec for media stream (g711ulaw (default), g711alaw, g729a) > > preferred_codec: g711ulaw > > # 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 > > # 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: "137.222.10.60" ; SNTP Server IP Address > > sntp_mode: anycast ; unicast, multicast, anycast, or directedbroadcast > (default) > > time_zone: GMT ; Time Zone Phone is in > > dst_offset: 1 ; Offset from Phone's time when BST is in effect > > dst_start_month: April ; Month in which BST starts > > dst_start_day: "21" ; Day of month in which BST starts > > dst_start_day_of_week: Sun ; Day of week in which BST starts > > dst_start_week_of_month: 1 ; Week of month in which BST starts > > dst_start_time: 02 ; Time of day in which BST starts > > dst_stop_month: Oct ; Month in which BST stops > > dst_stop_day: "20" ; Day of month in which BST stops > > dst_stop_day_of_week: Sunday ; Day of week in which BST stops > > dst_stop_week_of_month: 8 ; Week of month in which BST stops 8=last week > of month > > dst_stop_time: 2 ; Time of day in which BST stops > > dst_auto_adjust: 1 ; Enable(1-Default)/Disable(0) BST automatic > adjustment > > time_format_24hr: 1 ; Enable(1 - 24Hr Default)/Disable(0 - 12Hr) > > dnd_control: 0 ; Default 0 (0=off, 1=on, 2=off no user cntrl, 3=on no > user control) > > callerid_blocking: 0 ; Default 0 (Disable sending all calls as > anonymous) > > anonymous_call_block: 0 ; Default 0 (Disable blocking of anonymous > calls) > > dtmf_avt_payload: 101 ; Default 101 > > # Sync value of the phone used for remote reset > > sync: 1 ; Default 1 > > proxy_backup: "" ; Dotted IP of Backup Proxy > > proxy_backup_port: 5060 ; Backup Proxy port (default is 5060) > > 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 > > 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: "" ; restricted to dotted IP or DNS A record only > > outbound_proxy_port: 5060 ; default is 5060 > > # Allow for the bridge on a 3way call to join remaining parties upon > hangup > > cnf_join_enable : 1 ; 0-Disabled, 1-Enabled (default) > > # 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 > > # XML URLs > > ;services_url: "http://your.site/services.xml" ; URL for external Phone > Services > > services_url: "http://193.113.58.136/bt/" ;bt services > > directory_url: "http://your.site/directory.xml" ; URL for external > Directory location > > logo_url: "http://your.site/logo.bmp" ; URL for branding logo to be used > on phone display > > # HTTP Proxy Support > > http_proxy_addr: "http://ipaddress of A30P/SipPhoneProxy/" ; 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: 1 ; 0-Disabled (default), 1-Enabled > > # 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 > > ----------------------------------------------------- > sip.conf > > ; > ; SIP Configuration for Asterisk > ; > ; Syntax for specifying a SIP device in extensions.conf is > ; SIP/devicename where devicename is defined in a section below. ; ; You > may also use > ; SIP/username@domain to call any SIP user on the Internet > ; (Don't forget to enable DNS SRV records if you want to use this) ; > ; If you define a SIP proxy as a peer below, you may call > ; SIP/proxyhostname/user or SIP/user@proxyhostname > ; where the proxyhostname is defined in a section below > ; > ; Useful CLI commands to check peers/users: > ; sip show peers Show all SIP peers (including friends) > ; sip show users Show all SIP users (including friends) > ; sip show registry Show status of hosts we register with > ; > ; sip debug Show all SIP messages > ; > > [general] > context=default ; Default context for incoming calls > ;recordhistory=yes ; Record SIP history by default (see sip > history / sip no history) > ;realm=mydomain.tld ; Realm for digest authentication > ; defaults to "asterisk" > ; Realms MUST be globally unique > according to RFC 3261 > ; Set this to your host name or domain > name > port=5060 ; UDP Port to bind to (SIP standard port > is 5060) > bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds > to all) > srvlookup=yes ; Enable DNS SRV lookups on outbound > calls > ; Note: Asterisk only uses the first > host > ; in SRV records > ; Disabling DNS SRV lookups disables the > > ; ability to place SIP calls based on > domain > ; names to some other SIP users on the > Internet > > ;pedantic=yes ; Enable slow, pedantic checking for > Pingtel > ; and multiline formatted headers for > strict > ; SIP compatibility > ;tos=184 ; Set IP QoS to either a keyword or > numeric val > ;tos=lowdelay ; > lowdelay,throughput,reliability,mincost,none > ;maxexpirey=3600 ; Max length of incoming registration we > allow > ;defaultexpirey=120 ; Default length of incoming/outoing > registration > ;notifymimetype=text/plain ; Allow overriding of mime type in > NOTIFY > ;videosupport=yes ; Turn on support for SIP video > > ;disallow=all ; First disallow all codecs > ;allow=ulaw ; Allow codecs in order of preference > ;allow=ilbc ; Note: codec order is respected only in > [general] > ;musicclass=default ; Sets the default music on hold class > for all SIP calls > ; This may also be set for individual > users/peers > ;language=en ; Default language setting for all > users/peers > ; This may also be set for individual > users/peers > ;relaxdtmf=yes ; Relax dtmf handling > ;rtptimeout=60 ; Terminate call if 60 seconds of no RTP > activity > ; when we're not on hold > ;rtpholdtimeout=300 ; Terminate call if 300 seconds of no > RTP activity > ; when we're on hold (must be > > rtptimeout) > ;trustrpid = no ; If Remote-Party-ID should be trusted > ;progressinband=no ; If we should generate in-band ringing > always > ;useragent=Asterisk PBX ; Allows you to change the user agent > string > ;nat=no ; NAT settings > ; yes = Always ignore info and assume > NAT > ; no = Use NAT mode only according to > RFC3581 > ; never = Never attempt NAT mode or > RFC3581 support > ;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP > address > ; Asterisk can register as a SIP user agent to a SIP proxy (provider) ; > Format for the register statement is: > ; register => user[:secret[:authuser]]@host[:port][/extension] > ; > ; If no extension is given, the 's' extension is used. The extension ; > needs to be defined in extensions.conf to be able to accept calls ; from > this SIP proxy (provider) ; ; host is either a host name defined in DNS > or the name of a > ; section defined below. > ; > ; Examples: > ; > ;register => 1234:password@mysipprovider.com > ; > ; This will pass incoming calls to the 's' extension > ; > ; > ;register => 2345:password@sip_proxy/1234 > ; > ; Register 2345 at sip provider 'sip_proxy'. Calls from this > provider connect to local > ; extension 1234 in extensions.conf default context, unless you > define > ; unless you configure a [sip_proxy] section below, and configure a > context. > ; Tip 1: Avoid assigning hostname to a sip.conf section like > [provider.com] > ; Tip 2: Use separate type=peer and type=user sections for SIP > providers > ; (instead of type=friend) if you have calls in > both directions > > > ;externip = 200.201.202.203 ; Address that we're going to put in > outbound SIP messages > ; if we're behind a NAT > > ; The externip and localnet is used > ; when registering and communicating > with other proxies > ; that we're registered with > ; You may add multiple local networks. > A reasonable set of defaults > ; are: > ;localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local > networks > ;localnet=10.0.0.0/255.0.0.0 ; Also RFC1918 > ;localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation > ;localnet=169.254.0.0/255.255.0.0 ;Zero conf local network > > ;----------------------------------------------------------------------- > ------------ > ; Users and peers have different settings available. Friends have all > settings, ; since a friend is both a peer and a user ; > ; User config options: Peer configuration: > ; -------------------- ------------------- > ; context context > ; permit permit > ; deny deny > ; auth auth > ; secret secret > ; md5secret md5secret > ; dtmfmode dtmfmode > ; canreinvite canreinvite > ; nat nat > ; callgroup callgroup > ; pickupgroup pickupgroup > ; language language > ; allow allow > ; disallow disallow > ; insecure insecure > ; trustrpid trustrpid > ; progressinband progressinband > ; promiscredir promiscredir > ; callerid > ; accountcode > ; amaflags > ; incominglimit > ; outgoinglimit > ; restrictcid > ; mailbox > ; username > ; template > ; fromdomain > ; fromuser > ; host > ; mask > ; port > ; qualify > ; defaultip > ; rtptimeout > ; rtpholdtimeout > > ;[sip_proxy] > ; For incoming calls only. Example: FWD (Free World Dialup) ;type=user > ;context=from-fwd > > ;[sip_proxy-out] > ;type=peer ; we only want to call out, not be called > ;secret=guessit > ;username=yourusername > ;fromuser=yourusername ; Many SIP providers require this! > ;host=box.provider.com > > ;[grandstream1] > ;type=friend ; either "friend" (peer+user), "peer" or > "user" > ;context=from-sip > ;username=grandstream1 ; usually matches the [section] title > ;fromuser=grandstream1 ; overrides the callerid, e.g. required > by FWD > ;callerid=John Doe <1234> > ;host=192.168.0.23 ; we have a static but private IP address > ;nat=no ; there is not NAT between phone and > Asterisk > ;canreinvite=yes ; allow RTP voice traffic to bypass > Asterisk > ;dtmfmode=info ; either RFC2833 or INFO for the > BudgeTone > ;outgoinglimit=1 ; disable callwaiting signal (2nd call to > phone) > ;incominglimit=1 ; permit only 1 outgoing call at a time > ;mailbox=1234@default ; mailbox 1234 in voicemail context "default" > ;disallow=all ; need to disallow=all before we can use > allow> ;allow=ulaw ; Note: In user sections the order of > codecs > ; listed with allow= does NOT matter! > ;allow=alaw > ;allow=g723.1 ; Asterisk only supports g723.1 > pass-thru! > ;allow=g729 ; Pass-thru only unless g729 license > obtained > > [phone1] > type=friend > username=phone1 > secret=lounge > qualify=100 ; Qualify peer is no more than 200ms > away > host=10.131.111.41 > defaultip=10.131.111.41 ; This device registers with us > mailbox=1000 ; mailbox for message waiting indicator context=sip > callerid="Lounge1" <1> > > [phone2] > type=friend > username=phone2 > secret=kitchen > qualify=100 > host=10.131.111.42 > defaultip=10.131.111.42 > mailbox=2000 > context=sip > callerid="Kitchen1" <2> > > ---------------------------------------- > > extensions.conf > [default] > ; > ; By default we include the demo. In a production system, you > ; probably don't want to have the demo there. > ; > include => demo > ; > [sip] > exten => 5511,1,Dial(SIP/phone1,15,t) > exten => 5521,1,Dial(SIP/phone2,15,t) > exten => 1000,1,Dial(SIP/phone1,15,t) > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
asteriskstuff@ziplip.com
2004-Jul-19 16:21 UTC
[Asterisk-Users] Cisco 7960 SIP V6 and IBM A30P Fedora Asterisk
Thanks Wayne. P> -----Original Message----- > From: Wayne [mailto:Wayne@planetWayne.com] > Sent: Monday, July 19, 2004, 3:48 PM > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] Cisco 7960 SIP V6 and IBM A30P Fedora Asterisk > > Hiya! > Looks like you have the same problem as I had... found the answer by > doing a 'debug sip-messages' by telnet'ing into one of my cisco phones... > > The short answer is 'its your "callerid=" line' > you need to remove the quotes around the text part. The cisco's cant > handle it. > eg > where you have for [phone1] in your Sip.conf > callerid="Lounge1" <1> > > what you should have is > callerid=Lounge1 <1> > > etc... > > Threw me for a while but the debug options on the cisco's helped out > there... I think the docs read like you should have the text in quotes - > but as I said - my cisco's didnt like it :) > > anyways - hope this helps :) > Wayne! > > > > > > asteriskstuff@ziplip.com wrote: > > >Hi Sean > > > >Both phones are set for context=sip in the sip.conf file. > > > >As I say the phones will both call out OK (I can dial the 500 test number and > successfully connect to the remote PBX through my firewall). It's just that > when I'm trying to call from phone to phone I'm getting the 404 not found > error in the asteris verbose dialog. > > > >If anyone has a documented example of their 7960 config sipdefault.cnf and > sipxxxxxipadd.cnf files together with their sip.conf and extensions.conf files > I could have to test directly on my system I'd be appreciative to test them on > my system. > > > >While the WiKi's are very useful as example files it would be great (and I > may do it myself!!) if there was an up to date example file with all the > options for each filed and a verbose description for the rational behind it > (although I recognise that this is an 'in development' product and therefore > the docs have to be done at the end!!). > > > >Part of the problem is there are so many dependencies that can affect the > system including how the dhpcd server serves IP address's and associated files > (for example the files have to be structured in a particular order on the > tftpd server for the cisco's to pick them up correctly). Given this level of > dependency I'm not sure where the break could be. > > > >The one thing I have noticed from the show sip peers field is that it's > showing the phones as having a netmask of 255.255.255.255 although they're > actually configyred for 255.255.255.0. > > > >P > > > > > > > > > >>-----Original Message----- > >>From: Sean Cheesman [mailto:scheesman@caeveo.com] > >>Sent: Sunday, July 18, 2004, 11:37 AM > >>To: asterisk-users@lists.digium.com > >>Subject: RE: [Asterisk-Users] Cisco 7960 SIP V6 and IBM A30P Fedora Asterisk > >> > >>It doesn't look like you have a context set for phone1. Try putting > >>context=sip in the phone1 section like you have in phone2. That'll put > >>both in the same context of your extensions.conf file and should allow > >>interaction between the two. > >> > >>-----Original Message----- > >>From: asterisk-users-admin@lists.digium.com > >>[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of > >>asteriskstuff@ziplip.com > >>Sent: Sunday, July 18, 2004 7:13 AM > >>To: asterisk-users@lists.digium.com > >>Subject: [Asterisk-Users] Cisco 7960 SIP V6 and IBM A30P Fedora Asterisk > >> > >> > >>Hi All > >> > >>Total noob on the list so all help appreciated.... > >> > >>I've successfully installed Asterisk on an IBM A30P Thinkpad using > >>fedora Core 2 (I'm looking at having a mobile PBX for conferences and > >>shows). > >> > >>I've plugged in two Cisco 7960 phones.... > >> > >>The phones register with the Asterisk correctly and I can run the demo's > >>and even the AIX demo through to digium works correctly....... > >> > >>but I cannot get the phones to dial each other :( > >> > >>Initially I was getting a "extension not found in local" message (when > >>dialling from console...from phone just engaged (busy) tone. > >> > >>when I add extension XXXX from console I now get a "not found 404" > >>message....I see that there was an earlier thread on the list that > >>discussed removing the proxy forwarding from the phone settings and I've > >>tried that from SIPDefault.cnf but it doesn't fix the problem..... > >> > >>I've obviously missed something but am too inexperienced to spot it. P > >> > >>my files are as follows:- > >> > >>-------------------------------- > >> > >>sipxxxxxx.cnf > >> > >> > >># Lounge Phone Settings > >> > >># Line 1 Settings > >>line1_name: "11" ; Line 1 Extension\User ID > >>line1_displayname: "Lounge1" ; Line 1 Display Name > >>line1_authname: "lounge11" ; Line 1 Registration Authentication > >>line1_password: "lounge" ; Line 1 Registration Password > >> > >>------------------------- > >> > >>sipdefault.cnf > >> > >># Image Version > >> > >>image_version: P0S3-06-3-00 > >> > >># Proxy Server > >> > >>proxy1_address: ipaddress of A30P ; Can be dotted IP or FQDN > >> > >>proxy1_port: > >>5060 > >># Proxy Registration (0-disable (default), 1-enable) > >> > >>proxy_register: 0 > >> > >># Phone Registration Expiration [1-3932100 sec] (Default - 3600) > >> > >>timer_register_expires: 3600 > >> > >># Codec for media stream (g711ulaw (default), g711alaw, g729a) > >> > >>preferred_codec: g711ulaw > >> > >># 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 > >> > >># 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: "137.222.10.60" ; SNTP Server IP Address > >> > >>sntp_mode: anycast ; unicast, multicast, anycast, or directedbroadcast > >>(default) > >> > >>time_zone: GMT ; Time Zone Phone is in > >> > >>dst_offset: 1 ; Offset from Phone's time when BST is in effect > >> > >>dst_start_month: April ; Month in which BST starts > >> > >>dst_start_day: "21" ; Day of month in which BST starts > >> > >>dst_start_day_of_week: Sun ; Day of week in which BST starts > >> > >>dst_start_week_of_month: 1 ; Week of month in which BST starts > >> > >>dst_start_time: 02 ; Time of day in which BST starts > >> > >>dst_stop_month: Oct ; Month in which BST stops > >> > >>dst_stop_day: "20" ; Day of month in which BST stops > >> > >>dst_stop_day_of_week: Sunday ; Day of week in which BST stops > >> > >>dst_stop_week_of_month: 8 ; Week of month in which BST stops 8=last week > >>of month > >> > >>dst_stop_time: 2 ; Time of day in which BST stops > >> > >>dst_auto_adjust: 1 ; Enable(1-Default)/Disable(0) BST automatic > >>adjustment > >> > >>time_format_24hr: 1 ; Enable(1 - 24Hr Default)/Disable(0 - 12Hr) > >> > >>dnd_control: 0 ; Default 0 (0=off, 1=on, 2=off no user cntrl, 3=on no > >>user control) > >> > >>callerid_blocking: 0 ; Default 0 (Disable sending all calls as > >>anonymous) > >> > >>anonymous_call_block: 0 ; Default 0 (Disable blocking of anonymous > >>calls) > >> > >>dtmf_avt_payload: 101 ; Default 101 > >> > >># Sync value of the phone used for remote reset > >> > >>sync: 1 ; Default 1 > >> > >>proxy_backup: "" ; Dotted IP of Backup Proxy > >> > >>proxy_backup_port: 5060 ; Backup Proxy port (default is 5060) > >> > >>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 > >> > >>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: "" ; restricted to dotted IP or DNS A record only > >> > >>outbound_proxy_port: 5060 ; default is 5060 > >> > >># Allow for the bridge on a 3way call to join remaining parties upon > >>hangup > >> > >>cnf_join_enable : 1 ; 0-Disabled, 1-Enabled (default) > >> > >># 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 > >> > >># XML URLs > >> > >>;services_url: "http://your.site/services.xml" ; URL for external Phone > >>Services > >> > >>services_url: "http://193.113.58.136/bt/" ;bt services > >> > >>directory_url: "http://your.site/directory.xml" ; URL for external > >>Directory location > >> > >>logo_url: "http://your.site/logo.bmp" ; URL for branding logo to be used > >>on phone display > >> > >># HTTP Proxy Support > >> > >>http_proxy_addr: "http://ipaddress of A30P/SipPhoneProxy/" ; 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: 1 ; 0-Disabled (default), 1-Enabled > >> > >># 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 > >> > >>----------------------------------------------------- > >>sip.conf > >> > >>; > >>; SIP Configuration for Asterisk > >>; > >>; Syntax for specifying a SIP device in extensions.conf is > >>; SIP/devicename where devicename is defined in a section below. ; ; You > >>may also use > >>; SIP/username@domain to call any SIP user on the Internet > >>; (Don't forget to enable DNS SRV records if you want to use this) ; > >>; If you define a SIP proxy as a peer below, you may call > >>; SIP/proxyhostname/user or SIP/user@proxyhostname > >>; where the proxyhostname is defined in a section below > >>; > >>; Useful CLI commands to check peers/users: > >>; sip show peers Show all SIP peers (including friends) > >>; sip show users Show all SIP users (including friends) > >>; sip show registry Show status of hosts we register with > >>; > >>; sip debug Show all SIP messages > >>; > >> > >>[general] > >>context=default ; Default context for incoming calls > >>;recordhistory=yes ; Record SIP history by default (see sip > >>history / sip no history) > >>;realm=mydomain.tld ; Realm for digest authentication > >> ; defaults to "asterisk" > >> ; Realms MUST be globally unique > >>according to RFC 3261 > >> ; Set this to your host name or domain > >>name > >>port=5060 ; UDP Port to bind to (SIP standard port > >>is 5060) > >>bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds > >>to all) > >>srvlookup=yes ; Enable DNS SRV lookups on outbound > >>calls > >> ; Note: Asterisk only uses the first > >>host > >> ; in SRV records > >> ; Disabling DNS SRV lookups disables the > >> > >> ; ability to place SIP calls based on > >>domain > >> ; names to some other SIP users on the > >>Internet > >> > >>;pedantic=yes ; Enable slow, pedantic checking for > >>Pingtel > >> ; and multiline formatted headers for > >>strict > >> ; SIP compatibility > >>;tos=184 ; Set IP QoS to either a keyword or > >>numeric val > >>;tos=lowdelay ; > >>lowdelay,throughput,reliability,mincost,none > >>;maxexpirey=3600 ; Max length of incoming registration we > >>allow > >>;defaultexpirey=120 ; Default length of incoming/outoing > >>registration > >>;notifymimetype=text/plain ; Allow overriding of mime type in > >>NOTIFY > >>;videosupport=yes ; Turn on support for SIP video > >> > >>;disallow=all ; First disallow all codecs > >>;allow=ulaw ; Allow codecs in order of preference > >>;allow=ilbc ; Note: codec order is respected only in > >>[general] > >>;musicclass=default ; Sets the default music on hold class > >>for all SIP calls > >> ; This may also be set for individual > >>users/peers > >>;language=en ; Default language setting for all > >>users/peers > >> ; This may also be set for individual > >>users/peers > >>;relaxdtmf=yes ; Relax dtmf handling > >>;rtptimeout=60 ; Terminate call if 60 seconds of no RTP > >>activity > >> ; when we're not on hold > >>;rtpholdtimeout=300 ; Terminate call if 300 seconds of no > >>RTP activity > >> ; when we're on hold (must be > > >>rtptimeout) > >>;trustrpid = no ; If Remote-Party-ID should be trusted > >>;progressinband=no ; If we should generate in-band ringing > >>always > >>;useragent=Asterisk PBX ; Allows you to change the user agent > >>string > >>;nat=no ; NAT settings > >> ; yes = Always ignore info and assume > >>NAT > >> ; no = Use NAT mode only according to > >>RFC3581 > >> ; never = Never attempt NAT mode or > >>RFC3581 support > >>;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP > >>address > >>; Asterisk can register as a SIP user agent to a SIP proxy (provider) ; > >>Format for the register statement is: > >>; register => user[:secret[:authuser]]@host[:port][/extension] > >>; > >>; If no extension is given, the 's' extension is used. The extension ; > >>needs to be defined in extensions.conf to be able to accept calls ; from > >>this SIP proxy (provider) ; ; host is either a host name defined in DNS > >>or the name of a > >>; section defined below. > >>; > >>; Examples: > >>; > >>;register => 1234:password@mysipprovider.com > >>; > >>; This will pass incoming calls to the 's' extension > >>; > >>; > >>;register => 2345:password@sip_proxy/1234 > >>; > >>; Register 2345 at sip provider 'sip_proxy'. Calls from this > >>provider connect to local > >>; extension 1234 in extensions.conf default context, unless you > >>define > >>; unless you configure a [sip_proxy] section below, and configure a > >>context. > >>; Tip 1: Avoid assigning hostname to a sip.conf section like > >>[provider.com] > >>; Tip 2: Use separate type=peer and type=user sections for SIP > >>providers > >>; (instead of type=friend) if you have calls in > >>both directions > >> > >> > >>;externip = 200.201.202.203 ; Address that we're going to put in > >>outbound SIP messages > >> ; if we're behind a NAT > >> > >> ; The externip and localnet is used > >> ; when registering and communicating > >>with other proxies > >> ; that we're registered with > >> ; You may add multiple local networks. > >>A reasonable set of defaults > >> ; are: > >>;localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local > >>networks > >>;localnet=10.0.0.0/255.0.0.0 ; Also RFC1918 > >>;localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation > >>;localnet=169.254.0.0/255.255.0.0 ;Zero conf local network > >> > >>;----------------------------------------------------------------------- > >>------------ > >>; Users and peers have different settings available. Friends have all > >>settings, ; since a friend is both a peer and a user ; > >>; User config options: Peer configuration: > >>; -------------------- ------------------- > >>; context context > >>; permit permit > >>; deny deny > >>; auth auth > >>; secret secret > >>; md5secret md5secret > >>; dtmfmode dtmfmode > >>; canreinvite canreinvite > >>; nat nat > >>; callgroup callgroup > >>; pickupgroup pickupgroup > >>; language language > >>; allow allow > >>; disallow disallow > >>; insecure insecure > >>; trustrpid trustrpid > >>; progressinband progressinband > >>; promiscredir promiscredir > >>; callerid > >>; accountcode > >>; amaflags > >>; incominglimit > >>; outgoinglimit > >>; restrictcid > >>; mailbox > >>; username > >>; template > >>; fromdomain > >>; fromuser > >>; host > >>; mask > >>; port > >>; qualify > >>; defaultip > >>; rtptimeout > >>; rtpholdtimeout > >> > >>;[sip_proxy] > >>; For incoming calls only. Example: FWD (Free World Dialup) ;type=user > >>;context=from-fwd > >> > >>;[sip_proxy-out] > >>;type=peer ; we only want to call out, not be called > >>;secret=guessit > >>;username=yourusername > >>;fromuser=yourusername ; Many SIP providers require this! > >>;host=box.provider.com > >> > >>;[grandstream1] > >>;type=friend ; either "friend" (peer+user), "peer" or > >>"user" > >>;context=from-sip > >>;username=grandstream1 ; usually matches the [section] title > >>;fromuser=grandstream1 ; overrides the callerid, e.g. required > >>by FWD > >>;callerid=John Doe <1234> > >>;host=192.168.0.23 ; we have a static but private IP address > >>;nat=no ; there is not NAT between phone and > >>Asterisk > >>;canreinvite=yes ; allow RTP voice traffic to bypass > >>Asterisk > >>;dtmfmode=info ; either RFC2833 or INFO for the > >>BudgeTone > >>;outgoinglimit=1 ; disable callwaiting signal (2nd call to > >>phone) > >>;incominglimit=1 ; permit only 1 outgoing call at a time > >>;mailbox=1234@default ; mailbox 1234 in voicemail context "default" > >>;disallow=all ; need to disallow=all before we can use > >>allow> >>;allow=ulaw ; Note: In user sections the order of > >>codecs > >> ; listed with allow= does NOT matter! > >>;allow=alaw > >>;allow=g723.1 ; Asterisk only supports g723.1 > >>pass-thru! > >>;allow=g729 ; Pass-thru only unless g729 license > >>obtained > >> > >>[phone1] > >>type=friend > >>username=phone1 > >>secret=lounge > >>qualify=100 ; Qualify peer is no more than 200ms > >>away > >>host=10.131.111.41 > >>defaultip=10.131.111.41 ; This device registers with us > >>mailbox=1000 ; mailbox for message waiting indicator context=sip > >>callerid="Lounge1" <1> > >> > >>[phone2] > >>type=friend > >>username=phone2 > >>secret=kitchen > >>qualify=100 > >>host=10.131.111.42 > >>defaultip=10.131.111.42 > >>mailbox=2000 > >>context=sip > >>callerid="Kitchen1" <2> > >> > >>---------------------------------------- > >> > >>extensions.conf > >>[default] > >>; > >>; By default we include the demo. In a production system, you > >>; probably don't want to have the demo there. > >>; > >>include => demo > >>; > >>[sip] > >>exten => 5511,1,Dial(SIP/phone1,15,t) > >>exten => 5521,1,Dial(SIP/phone2,15,t) > >>exten => 1000,1,Dial(SIP/phone1,15,t) > >>_______________________________________________ > >>Asterisk-Users mailing list > >>Asterisk-Users@lists.digium.com > >>http://lists.digium.com/mailman/listinfo/asterisk-users > >>To UNSUBSCRIBE or update options visit: > >> http://lists.digium.com/mailman/listinfo/asterisk-users > >>_______________________________________________ > >>Asterisk-Users mailing list > >>Asterisk-Users@lists.digium.com > >>http://lists.digium.com/mailman/listinfo/asterisk-users > >>To UNSUBSCRIBE or update options visit: > >> http://lists.digium.com/mailman/listinfo/asterisk-users > >> > >> > >_______________________________________________ > >Asterisk-Users mailing list > >Asterisk-Users@lists.digium.com > >http://lists.digium.com/mailman/listinfo/asterisk-users > >To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users