Thanks Wayne. P
> -----Original Message----- > From: Wayne [mailto:[EMAIL PROTECTED] > Sent: Monday, July 19, 2004, 3:48 PM > To: [EMAIL PROTECTED] > 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! > > > > > > [EMAIL PROTECTED] 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:[EMAIL PROTECTED] > >>Sent: Sunday, July 18, 2004, 11:37 AM > >>To: [EMAIL PROTECTED] > >>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: [EMAIL PROTECTED] > >>[mailto:[EMAIL PROTECTED] On Behalf Of > >>[EMAIL PROTECTED] > >>Sent: Sunday, July 18, 2004 7:13 AM > >>To: [EMAIL PROTECTED] > >>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/[EMAIL PROTECTED] 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/[EMAIL PROTECTED] > >>; 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[:[EMAIL PROTECTED]: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:[EMAIL PROTECTED] > >>; > >>; This will pass incoming calls to the 's' extension > >>; > >>; > >>;register => 2345:[EMAIL PROTECTED]/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 > >>;[EMAIL PROTECTED] ; 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 > >>[EMAIL PROTECTED] > >>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 > >>[EMAIL PROTECTED] > >>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 > >[EMAIL PROTECTED] > >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 > [EMAIL PROTECTED] > 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 [EMAIL PROTECTED] 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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