Hi zusammen,
ich habe heute dlver-optasterisk300rc7fuerfli4l30xi586.rar auf meinem 3.0.1 fli4l installiert. Der Router war am Anfang auf 75 MHz heruntergetaktet, nun aber auf 120 MHz beschleunigt. Leider will es immer noch nicht mit dem telefonieren über SIP klappen.
Folgende Einstellungen kann ich liefern:
Die sip.conf sieht wie folgt aus:
Wenn ich die Console per asterisk -dddvvvr starte und dann eine Verbindung über SIP herstellen möchte, kommen ohne Unterbrechnung Meldungen wie folgt:
Ich höre beim wählen einen blechernen Ton, allerdings werde weder ich gehört, noch höre ich mein Gegenüber.
Die FW habe ich wie folgt eingestellt:
Wäre schön, wenn sich jemand meinem Problem annehmen könnte.
mfg
Carsten
ich habe heute dlver-optasterisk300rc7fuerfli4l30xi586.rar auf meinem 3.0.1 fli4l installiert. Der Router war am Anfang auf 75 MHz heruntergetaktet, nun aber auf 120 MHz beschleunigt. Leider will es immer noch nicht mit dem telefonieren über SIP klappen.
Folgende Einstellungen kann ich liefern:
Code:
fli4l*CLI> show translation
Translation times between formats (in milliseconds)
Source Format (Rows) Destination Format(Columns)
g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc
g723 - - - - - - - - - - -
gsm - - 90 90 263 98 89 411 - 2640 1409
ulaw - 463 - 1 175 10 1 323 - 2552 1321
alaw - 463 1 - 175 10 1 323 - 2552 1321
g726 - 591 130 130 - 138 129 451 - 2680 1449
adpcm - 468 7 7 180 - 6 328 - 2557 1326
slin - 462 1 1 174 9 - 322 - 2551 1320
lpc10 - 669 208 208 381 216 207 - - 2758 1527
g729 - - - - - - - - - - -
speex - 804 343 343 516 351 342 664 - - 1662
ilbc - 797 336 336 509 344 335 657 - 2886 -Die sip.conf sieht wie folgt aus:
Code:
;
; SIP Configuration example 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@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@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
;
; reload chan_sip.so Reload configuration file
; Active SIP peers will not be reconfigured
;
[general]
context=default ; Default context for incoming calls
;allowguest=no ; Allow or reject guest calls (default is yes, this can also be set to 'osp'
; if asterisk was compiled with OSP support.
;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
bindport=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
;domain=mydomain.tld ; Set default domain for this host
; If configured, Asterisk will only allow
; INVITE and REFER to non-local domains
; Use "sip show domains" to list local domains
;domain=mydomain.tld,mydomain-incoming
; Add domain and configure incoming context
; for external calls to this domain
;domain=1.2.3.4 ; Add IP address as local domain
; You can have several "domain" settings
;allowexternalinvites=no ; Disable INVITE and REFER to non-local domains
; Default is yes
;autodomain=yes ; Turn this on to have Asterisk add local host
; name and local IP to domain list.
;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
;maxexpiry=3600 ; Max length of incoming registration we allow
;defaultexpiry=120 ; Default length of incoming/outoing registration
;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
;checkmwi=10 ; Default time between mailbox checks for peers
;vmexten=voicemail ; dialplan extension to reach mailbox sets the
; Message-Account in the MWI notify message
; defaults to "asterisk"
;videosupport=yes ; Turn on support for SIP video
;recordhistory=yes ; Record SIP history by default
; (see sip history / sip no history)
disallow=all ; First disallow all codecs
allow=alaw ; Allow codecs in order of preference
allow=ulaw
;allow=ilbc ;
;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
;sendrpid = yes ; If Remote-Party-ID should be sent
;progressinband=never ; If we should generate in-band ringing always
; use 'never' to never use in-band signalling, even in cases
; where some buggy devices might not render it
;useragent=Asterisk PBX ; Allows you to change the user agent string
;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.
;usereqphone = no ; If yes, ";user=phone" is added to uri that contains
; a valid phone number
;dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
; Other options:
; info : SIP INFO messages
; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
; auto : Use rfc2833 if offered, inband otherwise
;compactheaders = yes ; send compact sip headers.
;sipdebug = yes ; Turn on SIP debugging by default, from
; the moment the channel loads this configuration
;subscribecontext = default ; Set a specific context for SUBSCRIBE requests
; Useful to limit subscriptions to local extensions
; Settable per peer/user also
;notifyringing = yes ; Notify subscriptions on RINGING state
;
; 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=sipregistrations
;
; 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:[email protected]
;
; This will pass incoming calls to the 's' extension
;
;
;register => 2345:password@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 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
;registertimeout=20 ; retry registration calls every 20 seconds (default)
;registerattempts=10 ; Number of registration attempts before we give up
; 0 = continue forever, hammering the other server until it
; accepts the registration
; Default is 0 tries, continue forever
;callevents=no ; generate manager events when sip ua performs events (e.g. hold)
;----------------------------------------- NAT SUPPORT ------------------------
; The externip, externhost and localnet settings are used if you use Asterisk
; behind a NAT device to communicate with services on the outside.
;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
;externhost=foo.dyndns.net ; Alternatively you can specify an
; external host, and Asterisk will
; perform DNS queries periodically. Not
; recommended for production
; environments! Use externip instead
;externrefresh=10 ; How often to refresh externhost if
; used
; 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
; The nat= setting is used when Asterisk is on a public IP, communicating with
; devices hidden behind a NAT device (broadband router). If you have one-way
; audio problems, you usually have problems with your NAT configuration or your
; firewall's support of SIP+RTP ports. You configure Asterisk choice of RTP
; ports for incoming audio in rtp.conf
;
;nat=no ; Global NAT settings (Affects all peers and users)
; 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)
;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
; just like friends added from the config file only on a
; as-needed basis? (yes|no)
;rtupdate=yes ; Send registry updates to database using realtime? (yes|no)
; If set to yes, when a SIP UA registers successfully, the ip address,
; the origination port, the registration period, and the username of
; the UA will be set to database via realtime. If not present, defaults to 'yes'.
;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule
; as if it had just registered? (yes|no|<seconds>)
; If set to yes, when the registration expires, the friend will vanish from
; the configuration until requested again. If set to an integer,
; friends expire within this number of seconds instead of the
; registration interval.
;ignoreregexpire=yes ; Enabling this setting has two functions:
;
; For non-realtime peers, when their registration expires, the information
; will _not_ be removed from memory or the Asterisk database; if you attempt
; to place a call to the peer, the existing information will be used in spite
; of it having expired
;
; For realtime peers, when the peer is retrieved from realtime storage,
; the registration information will be used regardless of whether
; it has expired or not; if it expires while the realtime peer is still in
; memory (due to caching or other reasons), the information will not be
; removed from realtime storage
; Incoming INVITE and REFER messages can be matched against a list of 'allowed'
; domains, each of which can direct the call to a specific context if desired.
; By default, all domains are accepted and sent to the default context or the
; context associated with the user/peer placing the call.
; Domains can be specified using:
; domain=<domain>[,<context>]
; Examples:
; domain=myasterisk.dom
; domain=customer.com,customer-context
;
; In addition, all the 'default' domains associated with a server should be
; added if incoming request filtering is desired.
; autodomain=yes
;
; To disallow requests for domains not serviced by this server:
; allowexternaldomains=no
; fromdomain=mydomain.tld ; When making outbound SIP INVITEs to
; non-peers, use your primary domain "identity"
; for From: headers instead of just your IP
; address. This is to be polite and
; it may be a mandatory requirement for some
; destinations which do not have a prior
; account relationship with your server.
[authentication]
; Global credentials for outbound calls, i.e. when a proxy challenges your
; Asterisk server for authentication. These credentials override
; any credentials in peer/register definition if realm is matched.
;
; This way, Asterisk can authenticate for outbound calls to other
; realms. We match realm on the proxy challenge and pick an set of
; credentials from this list
; Syntax:
; auth = <user>:<secret>@<realm>
; auth = <user>#<md5secret>@<realm>
; Example:
;auth=mark:[email protected]
;
; You may also add auth= statements to [peer] definitions
; Peer auth= override all other authentication settings if we match on realm
;------------------------------------------------------------------------------
; 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
; useclientcode useclientcode
; accountcode accountcode
; setvar setvar
; callerid callerid
; amaflags amaflags
; call-limit call-limit
; restrictcid restrictcid
; subscribecontext subscribecontext
; mailbox
; username
; template
; fromdomain
; regexten
; fromuser
; host
; port
; qualify
; defaultip
; rtptimeout
; rtpholdtimeout
; sendrpid
;[sip_proxy]
; For incoming calls only. Example: FWD (Free World Dialup)
; We match on IP address of the proxy for incoming calls
; since we can not match on username (caller id)
;type=peer
;context=from-fwd
;host=fwd.pulver.com
;[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!
;fromdomain=provider.sip.domain
;host=box.provider.com
;usereqphone=yes ; This provider requires ";user=phone" on URI
;call-limit=5 ; permit only 5 simultaneous outgoing calls to this peer
;------------------------------------------------------------------------------
; Definitions of locally connected SIP phones
;
; type = user a device that authenticates to us by "from" field to place calls
; type = peer a device we place calls to or that calls us and we match by host
; type = friend two configurations (peer+user) in one
;
; For local phones, type=friend works most of the time
;
; If you have one-way audio, you propably have NAT problems.
; If Asterisk is on a public IP, and the phone is inside of a NAT device
; you will need to configure nat option for those phones.
; Also, turn on qualify=yes to keep the nat session open
;[grandstream1]
;type=friend
;context=from-sip ; Where to start in the dialplan when this phone calls
;callerid=John Doe <1234> ; Full caller ID, to override the phones config
;host=192.168.0.23 ; we have a static but private IP address
; No registration allowed
;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
;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time
; from the phone to asterisk
; (1 for the explicit peer, 1 for the explicit user,
; remember that a friend equals 1 peer and 1 user in
; memory)
;mailbox=1234@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
;astdb=chan2ext/SIP/grandstream1=1234 ; ensures an astDB entry exists
;[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
;callerid="Jane Smith" <5678>
;host=dynamic ; This device needs to register
;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
;mailbox=1234@default,1233@default ; Subscribe to status of multiple mailboxes
;[snom]
;type=friend ; Friends place calls and receive calls
;context=from-sip ; Context for incoming calls from this user
;secret=blah
;subscribecontext=localextensions ; Only allow SUBSCRIBE for local extensions
;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
;mailbox=1234@context,2345 ; Mailbox(-es) for message waiting indicator
;vmexten=voicemail ; dialplan extension to reach mailbox
; sets the Message-Account in the MWI notify message
; defaults to global vmexten which defaults to "asterisk"
;restrictcid=yes ; To have the callerid restriced -> sent as ANI
;disallow=all
;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
;[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
; Normally you do NOT need to set this parameter
;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
;secret=blah
;host=dynamic
;insecure=port ; Allow matching of peer by IP address without matching port number
;insecure=invite ; Do not require authentication of incoming INVITEs
;insecure=port,invite ; (both)
;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
;secret=blah
;qualify=200 ; Qualify peer is no more than 200ms away
;nat=yes ; This phone may be natted
; Send SIP and RTP to the 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 ; IP address to use until registration
;username=goran ; Username to use when calling this device before registration
; Normally you do NOT need to set this parameter
;setvar=CUSTID=5678 ; Channel variable to be set for all calls from this deviceWenn ich die Console per asterisk -dddvvvr starte und dann eine Verbindung über SIP herstellen möchte, kommen ohne Unterbrechnung Meldungen wie folgt:
Code:
Apr 29 19:49:03 NOTICE[5340]: rtp.c:569 ast_rtp_read: Unknown RTP codec 2 received
Apr 29 19:49:03 NOTICE[5340]: rtp.c:569 ast_rtp_read: Unknown RTP codec 2 received
Apr 29 19:49:03 NOTICE[5340]: rtp.c:569 ast_rtp_read: Unknown RTP codec 2 received
Apr 29 19:49:03 NOTICE[5340]: rtp.c:569 ast_rtp_read: Unknown RTP codec 2 receivedIch höre beim wählen einen blechernen Ton, allerdings werde weder ich gehört, noch höre ich mein Gegenüber.
Die FW habe ich wie folgt eingestellt:
Code:
INPUT_ACCEPT_DEF='yes'
.
.
INPUT_LIST_5='prot:udp 4569 ACCEPT NOLOG'
INPUT_LIST_6='prot:udp 5060 ACCEPT NOLOG'
INPUT_LIST_7='prot:udp 10000-20000 ACCEPT NOLOG'Wäre schön, wenn sich jemand meinem Problem annehmen könnte.
mfg
Carsten
