--- Tim Connolly <[EMAIL PROTECTED]> wrote: > Is NAT=yes on, are you behind a firewall? Give us > some connectivity details. > Usually when you see maximum retries, its because > you have one-way > communications with the far end for some reason. Are > you setting "externip" > statically?
To answer your questions, yes, I am behind a firewall. The asterisk server is the only device connected to a cheapo Netgear 4-port router/firewall. I'm not setting externip myself, so whatever the default is, it's getting used. I'm also NOT making outgoing calls, and there are no actual SIP devices attached ... I'm just trying to receive incoming calls forwarded from a different provider via SIP. Here is a complete sip.conf file ... do I need to provide anything else? 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 context=unwelcome-calls ; Default context for incoming calls ; After all, we don't want any random ; incoming calls to have access to outbound ; calling ;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=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 ; register for SIP at VoicePulseConnect register => uid:[EMAIL PROTECTED] ;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 ; restrictcid ; mailbox ; username ; template ; fromdomain ; regexten ; fromuser ; host ; mask ; port ; qualify ; defaultip ; rtptimeout ; rtpholdtimeout [888] ; For incoming calls ONLY type=user ; This device takes incoming calls username=uid ; Username on device secret=secret ; Password for device host=srvr.voicepulse.com ; This host doesn't change frequently context=allowed_context ; Inbound calls from this ; host go to the normal ; context ... at the end of this sip.conf are a bunch of commented out details for particular SIP phones. As I have no SIP phones, I've left all of these commented out. Thanks, Maya __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com _______________________________________________ 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