[Asterisk-Users] Cisco 7960 SIP V6 and IBM A30P Fedora Asterisk

asteriskstuff at ziplip.com asteriskstuff at ziplip.com
Sun Jul 18 12:46:00 MST 2004


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 at caeveo.com]
> Sent: Sunday, July 18, 2004, 11:37 AM
> To: asterisk-users at 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 at lists.digium.com
> [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of
> asteriskstuff at ziplip.com
> Sent: Sunday, July 18, 2004 7:13 AM
> To: asterisk-users at 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 at 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 at 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 at mysipprovider.com	
> ;
> ;     This will pass incoming calls to the 's' extension
> ;
> ;
> ;register => 2345:password at 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 at 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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