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

asteriskstuff at ziplip.com asteriskstuff at ziplip.com
Mon Jul 19 16:21:44 MST 2004


Thanks Wayne.

P

> -----Original Message-----
> From: Wayne [mailto:Wayne at planetWayne.com]
> Sent: Monday, July 19, 2004, 3:48 PM
> To: asterisk-users at 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 at 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 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)
> >>_______________________________________________
> >>Asterisk-Users mailing list
> >>Asterisk-Users at 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 at 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 at 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
> >  
> >
> _______________________________________________
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