Hi Tim,
We have renamed / removed our SIP.conf file as per the instuctions on the WIKI.
(http://www.voip-info.org/wiki-Asterisk+RealTime+Static)
I have however, attached a copy of our previous .conf file for your perusal.
I have also included a dump of the sip_buddies table so you can see the entry we have made.
This is driving me nuts, so anything that you can offer would be very much appreciated.
Cheers,
Callum
Tim Connolly wrote:
Let's see your sip.conf and a sip show users.
-----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Callum McGillivray Sent: Wednesday, May 04, 2005 11:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Problem with realtime SIP
Hi Guys,
We have just set up Asterisk (CVS Head) for a realtime enviorment using MySQL & Asterisk Addons.
I have populated the "sip_buddies" table with the same information that is came from our sip.conf, however registration seems to fail for the softphone we have set up.
Does anyone have any idea as to what I should be looking for here? I'm not getting any error messages in debug, and just this line from the command line;
May 5 14:30:18 NOTICE[5063]: chan_sip.c:9020 handle_request_register: Registration from 'Callum McGillivray<sip:[EMAIL PROTECTED]>' failed for '192.168.1.90'
Can someone tell me what I might be missing ? Or can someone give me a dump of their "sip_buddies" table so I can try and see what I might be doing wrong ?
Thanks,
Callum
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sip_buddies.csv
Description: MS-Excel spreadsheet; ; 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] ;register=1001:12345:[EMAIL PROTECTED]/1204 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 (defaults to "no") ;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 MWI NOTIFY ;videosupport=yes ; Turn on support for SIP video ;disallow=all ; First disallow all codecs allow=ulaw ; Allow codecs in order of preference allow=gsm ;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 ; route = Assume NAT, don't send rport (work around more UNIDEN bugs) ;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address ; ; Note that promiscredir when redirects are made to the ; ; local system will cause loops since SIP is incapable ; ; of performing a "hairpin" call. ; ; If regcontext is specified, Asterisk will dynamically ; create and destroy a NoOp priority 1 extension for a given ; peer who registers or unregisters with us. The actual extension ; is the 'regexten' parameter of the registering peer or its ; name if 'regexten' is not provided. More than one regexten may be supplied ; if they are separated by '&'. Patterns may be used in regexten. ; ;regcontext=iaxregistrations ; ; 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 ; 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 ; restrictcid ; mailbox ; username ; template ; fromdomain ; regexten ; 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 ; Authentication user for outbound proxies ;fromuser=yourusername ; Many SIP providers require this! ;host=box.provider.com ;[grandstream1] ;type=friend ; either "friend" (peer+user), "peer" or "user" ;context=from-sip ;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 ;incominglimit=1 ; permit only 1 outgoing call at a time ; from the phone to asterisk ;[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 ;[xlite1] ;Turn off silence suppression in X-Lite ("Transmit Silence"=YES)! ;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed ;type=friend ;regexten=1234 ; When they register, create extension 1234 ;username=xlite1 ;callerid="Jane Smith" <5678> ;host=dynamic ;nat=yes ; X-Lite is behind a NAT router ;canreinvite=no ; Typically set to NO if behind NAT ;disallow=all ;allow=gsm ; GSM consumes far less bandwidth than ulaw ;allow=ulaw ;allow=alaw ;[snom] ;type=friend ; Friends place calls and receive calls ;context=from-sip ; Context for incoming calls from this user ;secret=blah ;language=de ; Use German prompts for this user ;host=dynamic ; This peer register with us ;dtmfmode=inband ; Choices are inband, rfc2833, or info ;defaultip=192.168.0.59 ; IP used until peer registers ;username=snom ; Username to use in INVITE until peer registers ;mailbox=1234,2345 ; Mailboxes for message waiting indicator ;restrictcid=yes ; To have the callerid restriced -> sent as ANI ;disallow=all ;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw! ;[EMAIL PROTECTED],2345 ; Mailbox(-es) for message waiting indicator ;[polycom] ;type=friend ; Friends place calls and receive calls ;context=from-sip ; Context for incoming calls from this user ;secret=blahpoly ;host=dynamic ; This peer register with us ;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info ;username=polly ; Username to use in INVITE until peer registers ;disallow=all ;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw! ;progressinband=no ; Polycom phones don't work properly with "never" ;[pingtel] ;type=friend ;username=pingtel ;secret=blah ;host=dynamic ;insecure=yes ; To match a peer based by IP address only and not peer ;insecure=very ; To allow registered hosts to call without re-authenticating ;qualify=1000 ; Consider it down if it's 1 second to reply ; Helps with NAT session ; qualify=yes uses default value ;callgroup=1,3-4 ; We are in caller groups 1,3,4 ;pickupgroup=1,3-5 ; We can do call pick-p for call group 1,3,4,5 ;defaultip=192.168.0.60 ; IP address to use if peer has not registred ;[cisco1] ;type=friend ;username=cisco1 ;secret=blah ;qualify=200 ; Qualify peer is no more than 200ms away ;nat=yes ; This phone may be natted ; Send SIP and RTP to IP address that packet is ; received from instead of trusting SIP headers ;host=dynamic ; This device registers with us ;canreinvite=no ; Asterisk by default tries to redirect the ; RTP media stream (audio) to go directly from ; the caller to the callee. Some devices do not ; support this (especially if one of them is ; behind a NAT). ;defaultip=192.168.0.4 ;[cisco2] ;type=friend ;username=cisco2 ;fromuser=markster ; Specify user to put in "from" instead of callerid ;fromdomain=yourdomain.com ; Specify domain to put in "from" instead of callerid ; fromuser and fromdomain are used when Asterisk ; places calls to this account. It is not used for ; calls from this account. ;secret=blah ;host=dynamic ;defaultip=192.168.0.4 ;amaflags=default ; Choices are default, omit, billing, documentation ;accountcode=markster ; Users may be associated with an accountcode to ease billing [2000] type=friend host=dynamic callerid=Jenn Hales canreinvite=yes dtmfmode=rfc2833 nat=no context=default ;This is a security related setting, that has no security at the moment. disallow=all allow=alaw secret=2000 [2001] type=friend host=dynamic callerid=Callum McGillivray canreinvite=yes dtmfmode=rfc2833 nat=no context=default disallow=all allow=alaw secret=2001 [2002] type=friend host=dynamic callerid=George Pezzutto canreinvite=yes dtmfmode=rfc2833 nat=no context=default disallow=all allow=alaw secret=2002 [2003] type=friend host=dynamic callerid=Hemant Arora canreinvite=yes dtmfmode=rfc2833 nat=no context=default disallow=all allow=alaw secret=2003 [2004] type=friend host=dynamic callerid=George sales canreinvite=yes dtmfmode=rfc2833 nat=no context=default disallow=all allow=alaw secret=2004 [2005] type=friend host=dynamic callerid=Arthur canreinvite=yes dtmfmode=rfc2833 nat=no context=default disallow=all allow=alaw secret=2005 [2006] type=friend host=dynamic callerid=Brigit canreinvite=yes dtmfmode=rfc2833 nat=no context=default disallow=all allow=alaw secret=2006 [2007] type=friend host=dynamic callerid=Tech 1 canreinvite=yes dtmfmode=rfc2833 nat=no context=default disallow=all allow=alaw secret=2007 [2008] type=friend host=dynamic callerid=Jen canreinvite=yes dtmfmode=rfc2833 nat=no context=default disallow=all allow=alaw secret=2008 [2009] type=friend host=dynamic callerid=Tech 2 canreinvite=yes dtmfmode=rfc2833 nat=no context=default disallow=all allow=ulaw secret=2009 [2010] type=friend host=dynamic callerid=Tech 3 canreinvite=no dtmfmode=rfc2833 ;nat=never nat=no context=default disallow=all allow=alaw secret=2010_______________________________________________ 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
