Can´t make calls from a softphone to other (desperated)

Hi all,

I have spent 3 days trying things and reading lot of stuff trying to get my asterisk configuration work.

My configuration is a really easy one. I´m an asterisk newbie and followed a tutorial to create 2 sip users and 2 extensions in extensions.conf. I´m using X-Lite 3.0 as the softphones.

Both phones register ok into asterisk, but when I try to call each other I get a call failure. If anyone could have a look at the sip debug I’ve attached I would be really greateful. Log is at the bottom of the post.

Byte the way:
Asterisk IP:
Machine running phone1:
Machine running phone2:

P.D. If anyone needs any other information let me know.
P.D. Post edited with conf’s files. Hope this helps. I only added users to the end of sip.conf and 2 extensions to the end of extensions.conf

Thanks in advance,


; 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		Reload configuration file
;				Active SIP peers will not be reconfigured

context=default			; Default context for incoming calls
;allowguest=no			; Allow or reject guest calls (default is yes)
allowoverlap=no			; Disable overlap dialing support. (Default is yes)
;allowtransfer=no		; Disable all transfers (unless enabled in peers or users)
				; Default is enabled
;realm=mydomain.tld		; Realm for digest authentication
				; defaults to "asterisk". If you set a system name in
				; asterisk.conf, it defaults to that system name
				; 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)
				; bindport is the local UDP port that Asterisk will listen on
bindaddr=		; IP address to bind to ( 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
;pedantic=yes			; Enable checking of tags in headers, 
				; international character conversions in URIs
				; and multiline formatted headers for strict
				; SIP compatibility (defaults to "no")

; See doc/README.tos for a description of these parameters.
;tos_sip=cs3                    ; Sets TOS for SIP packets.
;tos_audio=ef                   ; Sets TOS for RTP audio packets.
;tos_video=af41                 ; Sets TOS for RTP video packets.

;maxexpiry=3600			; Maximum allowed time of incoming registrations
				; and subscriptions (seconds)
;minexpiry=60			; Minimum length of registrations/subscriptions (default 60)
;defaultexpiry=120		; Default length of incoming/outgoing registration
;t1min=100			; Minimum roundtrip time for messages to monitored hosts
				; Defaults to 100 ms
;notifymimetype=text/plain	; Allow overriding of mime type in MWI NOTIFY
;checkmwi=10			; Default time between mailbox checks for peers
;buggymwi=no			; Cisco SIP firmware doesn't support the MWI RFC
				; fully. Enable this option to not get error messages
				; when sending MWI to phones with this bug.
;vmexten=voicemail		; dialplan extension to reach mailbox sets the 
				; Message-Account in the MWI notify message 
				; defaults to "asterisk"
;disallow=all			; First disallow all codecs
allow=ulaw			; Allow codecs in order of preference
;allow=ilbc			; see doc/rtp-packetization for framing options
; This option specifies a preference for which music on hold class this channel
; should listen to when put on hold if the music class has not been set on the
; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer
; channel putting this one on hold did not suggest a music class.
; This option may be specified globally, or on a per-user or per-peer basis.
; This option specifies which music on hold class to suggest to the peer channel
; when this channel places the peer on hold. It may be specified globally or on
; a per-user or per-peer basis.
;language=en			; Default language setting for all users/peers
				; This may also be set for individual users/peers
;relaxdtmf=yes			; Relax dtmf handling
;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
				; Valid values: yes, no, never Default: never
;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 Asterisk 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.
;videosupport=yes		; Turn on support for SIP video. You need to turn this on
				; in the this section to get any video support at all.
				; You can turn it off on a per peer basis if the general
				; video support is enabled, but you can't enable it for
				; one peer only without enabling in the general section.
;maxcallbitrate=384		; Maximum bitrate for video calls (default 384 kb/s)
				; Videosupport and maxcallbitrate is settable
				; for peers and users as well
;callevents=no			; generate manager events when sip ua 
				; performs events (e.g. hold)
;alwaysauthreject = yes		; When an incoming INVITE or REGISTER is to be rejected,
 		    		; for any reason, always reject with '401 Unauthorized'
 				; instead of letting the requester know whether there was
 				; a matching user or peer for their request

;g726nonstandard = yes		; If the peer negotiates G726-32 audio, use AAL2 packing
				; order instead of RFC3551 packing order (this is required
				; for Sipura and Grandstream ATAs, among others). This is
				; contrary to the RFC3551 specification, the peer _should_
				; be negotiating AAL2-G726-32 instead :-(

; 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 and have a "regexten=" configuration item.  
; Multiple contexts may be specified by separating them with '&'. The 
; actual extension is the 'regexten' parameter of the registering peer or its
; name if 'regexten' is not provided.  If more than one context is provided,
; the context must be specified within regexten by appending the desired
; context after '@'.  More than one regexten may be supplied if they are 
; separated by '&'.  Patterns may be used in regexten.
;--------------------------- RTP timers ----------------------------------------------------
; These timers are currently used for both audio and video streams. The RTP timeouts
; are only applied to the audio channel.
; The settings are settable in the global section as well as per device
;rtptimeout=60			; Terminate call if 60 seconds of no RTP or RTCP activity
				; on the audio channel
				; when we're not on hold. This is to be able to hangup
				; a call in the case of a phone disappearing from the net,
				; like a powerloss or grandma tripping over a cable.
;rtpholdtimeout=300		; Terminate call if 300 seconds of no RTP or RTCP activity
				; on the audio channel
				; when we're on hold (must be > rtptimeout)
;rtpkeepalive=<secs>		; Send keepalives in the RTP stream to keep NAT open
				; (default is off - zero)
;--------------------------- SIP DEBUGGING ---------------------------------------------------
;sipdebug = yes			; Turn on SIP debugging by default, from
				; the moment the channel loads this configuration
;recordhistory=yes		; Record SIP history by default 
				; (see sip history / sip no history)
;dumphistory=yes		; Dump SIP history at end of SIP dialogue
				; SIP history is output to the DEBUG logging channel

;--------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS) ----------------------------
; You can subscribe to the status of extensions with a "hint" priority
; (See extensions.conf.sample for examples)
; chan_sip support two major formats for notifications: dialog-info and SIMPLE 
; You will get more detailed reports (busy etc) if you have a call limit set
; for a device. When the call limit is filled, we will indicate busy. Note that
; you need at least 2 in order to be able to do attended transfers.
; For queues, you will need this level of detail in status reporting, regardless
; if you use SIP subscriptions. Queues and manager use the same internal interface
; for reading status information.
; Note: Subscriptions does not work if you have a realtime dialplan and use the
; realtime switch.
;allowsubscribe=no		; Disable support for subscriptions. (Default is yes)
;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 (default: no)
;notifyhold = yes		; Notify subscriptions on HOLD state (default: no)
				; Turning on notifyringing and notifyhold will add a lot
				; more database transactions if you are using realtime.
;limitonpeers = yes		; Apply call limits on peers only. This will improve 
				; status notification when you are using type=friend
				; Inbound calls, that really apply to the user part
				; of a friend will now be added to and compared with
				; the peer limit instead of applying two call limits,
				; one for the peer and one for the user.

;----------------------------------------- T.38 FAX PASSTHROUGH SUPPORT -----------------------
; This setting is available in the [general] section as well as in device configurations.
; Setting this to yes, enables T.38 fax (UDPTL) passthrough on SIP to SIP calls, provided
; both parties have T38 support enabled in their Asterisk configuration 
; This has to be enabled in the general section for all devices to work. You can then
; disable it on a per device basis. 
; T.38 faxing only works in SIP to SIP calls, with no local or agent channel being used.
; t38pt_udptl = yes            ; Default false
;----------------------------------------- OUTBOUND SIP REGISTRATIONS  ------------------------
; Asterisk can register as a SIP user agent to a SIP proxy (provider)
; Format for the register statement is:
;       register => user[:secret[]]@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 =>	
;     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 []
;    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

;----------------------------------------- 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 =	; 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
;	; 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=; All RFC 1918 addresses are local networks
;localnet=	; Also RFC1918
;localnet=		; Another RFC1918 with CIDR notation
;localnet= ;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 (;rport)
                                ; never = Never attempt NAT mode or RFC3581 support
				; route = Assume NAT, don't send rport 
				; (work around more UNIDEN bugs)

;----------------------------------- MEDIA HANDLING --------------------------------
; By default, Asterisk tries to re-invite the audio to an optimal path. If there's
; no reason for Asterisk to stay in the media path, the media will be redirected.
; This does not really work with in the case where Asterisk is outside and have
; clients on the inside of a NAT. In that case, you want to set canreinvite=nonat
;canreinvite=yes		; 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).
				; The default setting is YES. If you have all clients
				; behind a NAT, or for some other reason wants Asterisk to
				; stay in the audio path, you may want to turn this off.

				; In Asterisk 1.4 this setting also affect direct RTP
				; at call setup (a new feature in 1.4 - setting up the
				; call directly between the endpoints instead of sending
				; a re-INVITE).

;directrtpsetup=yes		; Enable the new experimental direct RTP setup. This sets up
				; the call directly with media peer-2-peer without re-invites.
				; Will not work for video and cases where the callee sends 
				; RTP payloads and fmtp headers in the 200 OK that does not match the
				; callers INVITE.

;canreinvite=nonat		; An additional option is to allow media path redirection
				; (reinvite) but only when the peer where the media is being
				; sent is known to not be behind a NAT (as the RTP core can
				; determine it based on the apparent IP address the media
				; arrives from).

;canreinvite=update		; Yet a third option... use UPDATE for media path redirection,
				; instead of INVITE. This can be combined with 'nonat', as
				; 'canreinvite=update,nonat'. It implies 'yes'.

;----------------------------------------- REALTIME SUPPORT ------------------------
; For additional information on ARA, the Asterisk Realtime Architecture,
; please read realtime.txt and extconfig.txt in the /doc directory of the
; source code.
;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)

;rtsavesysname=yes		; Save systemname in realtime database at registration
				; Default= 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

;----------------------------------------- SIP DOMAIN SUPPORT ------------------------
; 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
; 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

				; Add domain and configure incoming context
				; for external calls to this domain
;domain=			; 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.

; 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. 

;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
; jbenable = yes              ; Enables the use of a jitterbuffer on the receiving side of a
                              ; SIP channel. Defaults to "no". An enabled jitterbuffer will
                              ; be used only if the sending side can create and the receiving
                              ; side can not accept jitter. The SIP channel can accept jitter,
                              ; thus a jitterbuffer on the receive SIP side will be used only
                              ; if it is forced and enabled.

; jbforce = no                ; Forces the use of a jitterbuffer on the receive side of a SIP
                              ; channel. Defaults to "no".

; jbmaxsize = 200             ; Max length of the jitterbuffer in milliseconds.

; jbresyncthreshold = 1000    ; Jump in the frame timestamps over which the jitterbuffer is
                              ; resynchronized. Useful to improve the quality of the voice, with
                              ; big jumps in/broken timestamps, usually sent from exotic devices
                              ; and programs. Defaults to 1000.

; jbimpl = fixed              ; Jitterbuffer implementation, used on the receiving side of a SIP
                              ; channel. Two implementations are currently available - "fixed"
                              ; (with size always equals to jbmaxsize) and "adaptive" (with
                              ; variable size, actually the new jb of IAX2). Defaults to fixed.

; jblog = no                  ; Enables jitterbuffer frame logging. Defaults to "no".

; 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:
; 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
; callingpres		      callingpres
; 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
; allowoverlap		      allowoverlap
; allowsubscribe	      allowsubscribe
; allowtransfer	      	      allowtransfer
; subscribecontext	      subscribecontext
; videosupport		      videosupport
; maxcallbitrate	      maxcallbitrate
; rfc2833compensate           mailbox
;                             username
;                             template
;                             fromdomain
;                             regexten
;                             fromuser
;                             host
;                             port
;                             qualify
;                             defaultip
;                             rtptimeout
;                             rtpholdtimeout
;                             sendrpid
;                             outboundproxy
;                             rfc2833compensate

; 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          			; we only want to call out, not be called
;username=yourusername			; Authentication user for outbound proxies
;fromuser=yourusername			; Many SIP providers require this!
;usereqphone=yes			; This provider requires ";user=phone" on URI
;call-limit=5				; permit only 5 simultaneous outgoing calls to this peer
;outboundproxy=proxy.provider.domain	; send outbound signaling to this proxy, not directly to the peer
					; Call-limits will not be enforced on real-time peers,
					; since they are not stored in-memory
;port=80				; The port number we want to connect to on the remote side
					; Also used as "defaultport" in combination with "defaultip" settings

; Definitions of locally connected SIP devices
; 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 device names, we recommend using only a-z, numerics (0-9) and underscore
; For local phones, type=friend works most of the time
; If you have one-way audio, you probably 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

;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
				; on incoming calls to Asterisk
;host=		; 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
				; This will affect your subscriptions as well.
				; There is no combined call counter for a "friend"
				; so there's currently no way in sip.conf to limit
				; to one inbound or outbound call per phone. Use
				; the group counters in the dial plan for that.
;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=g723.1			; Asterisk only supports g723.1 pass-thru!
;allow=g729			; Pass-thru only unless g729 license obtained
;callingpres=allowed_passed_screen	; Set caller ID presentation
				; See README.callingpres for more information

; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
;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
;allow=gsm			; GSM consumes far less bandwidth than ulaw
;mailbox=1234@default,1233@default	; Subscribe to status of multiple mailboxes

;type=friend			; Friends place calls and receive calls
;context=from-sip		; Context for incoming calls from this user
;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=		; IP used until peer registers
;mailbox=1234@context,2345      ; Mailbox(-es) for message waiting indicator
;subscribemwi=yes		; Only send notifications if this phone 
				; subscribes for mailbox notification
;vmexten=voicemail		; dialplan extension to reach mailbox 
				; sets the Message-Account in the MWI notify message
				; defaults to global vmexten which defaults to "asterisk"
;allow=ulaw			; dtmfmode=inband only works with ulaw or alaw!

;type=friend			; Friends place calls and receive calls
;context=from-sip		; Context for incoming calls from this user
;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
;allow=ulaw                     ; dtmfmode=inband only works with ulaw or alaw!
;progressinband=no		; Polycom phones don't work properly with "never"

;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
; Call group and Pickup group should be in the range from 0 to 63
;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=		; IP address to use if peer has not registered
;deny=		; ACL: Control access to this account based on IP address

;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=		; 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 device

;rfc2833compensate=yes		; Compensate for pre-1.4 DTMF transmission from another Asterisk machine.
				; You must have this turned on or DTMF reception will work improperly.




; extensions.conf - the Asterisk dial plan
; Static extension configuration file, used by
; the pbx_config module. This is where you configure all your 
; inbound and outbound calls in Asterisk. 
; This configuration file is reloaded 
; - With the "extensions reload" command in the CLI
; - With the "reload" command (that reloads everything) in the CLI

; The "General" category is for certain variables.  
; If static is set to no, or omitted, then the pbx_config will rewrite
; this file when extensions are modified.  Remember that all comments
; made in the file will be lost when that happens. 
; XXX Not yet implemented XXX
; if static=yes and writeprotect=no, you can save dialplan by
; CLI command 'save dialplan' too
; If autofallthrough is set, then if an extension runs out of
; things to do, it will terminate the call with BUSY, CONGESTION
; or HANGUP depending on Asterisk's best guess. This is the default.
; If autofallthrough is not set, then if an extension runs out of 
; things to do, Asterisk will wait for a new extension to be dialed 
; (this is the original behavior of Asterisk 1.0 and earlier).
; If clearglobalvars is set, global variables will be cleared 
; and reparsed on an extensions reload, or Asterisk reload.
; If clearglobalvars is not set, then global variables will persist
; through reloads, and even if deleted from the extensions.conf or
; one of its included files, will remain set to the previous value.
; If priorityjumping is set to 'yes', then applications that support
; 'jumping' to a different priority based on the result of their operations
; will do so (this is backwards compatible behavior with pre-1.2 releases
; of Asterisk). Individual applications can also be requested to do this
; by passing a 'j' option in their arguments.
; User context is where entries from users.conf are registered.  The
; default value is 'default'
; You can include other config files, use the #include command
; (without the ';'). Note that this is different from the "include" command
; that includes contexts within other contexts. The #include command works
; in all asterisk configuration files.
;#include "filename.conf"

; The "Globals" category contains global variables that can be referenced
; in the dialplan with the GLOBAL dialplan function:
; ${${GLOBAL(VARIABLE)}} or ${text${GLOBAL(VARIABLE)}} or any hybrid
; Unix/Linux environmental variables can be reached with the ENV dialplan
; function: ${ENV(VARIABLE)}
CONSOLE=Console/dsp				; Console interface for demo
IAXINFO=guest					; IAXtel username/password
TRUNK=Zap/g2					; Trunk interface
; Note the 'g2' in the TRUNK variable above. It specifies which group (defined
; in zapata.conf) to dial, i.e. group 2, and how to choose a channel to use in
; the specified group. The four possible options are:
; g: select the lowest-numbered non-busy Zap channel
;    (aka. ascending sequential hunt group).
; G: select the highest-numbered non-busy Zap channel
;    (aka. descending sequential hunt group).
; r: use a round-robin search, starting at the next highest channel than last
;    time (aka. ascending rotary hunt group).
; R: use a round-robin search, starting at the next lowest channel than last
;    time (aka. descending rotary hunt group).
TRUNKMSD=1					; MSD digits to strip (usually 1 or 0)

; Any category other than "General" and "Globals" represent 
; extension contexts, which are collections of extensions.  
; Extension names may be numbers, letters, or combinations
; thereof. If an extension name is prefixed by a '_'
; character, it is interpreted as a pattern rather than a
; literal.  In patterns, some characters have special meanings:
;   X - any digit from 0-9
;   Z - any digit from 1-9
;   N - any digit from 2-9
;   [1235-9] - any digit in the brackets (in this example, 1,2,3,5,6,7,8,9)
;   . - wildcard, matches anything remaining (e.g. _9011. matches 
;	anything starting with 9011 excluding 9011 itself)
;   ! - wildcard, causes the matching process to complete as soon as
;       it can unambiguously determine that no other matches are possible
; For example the extension _NXXXXXX would match normal 7 digit dialings, 
; while _1NXXNXXXXXX would represent an area code plus phone number
; preceded by a one.
; Each step of an extension is ordered by priority, which must
; always start with 1 to be considered a valid extension.  The priority
; "next" or "n" means the previous priority plus one, regardless of whether
; the previous priority was associated with the current extension or not.
; The priority "same" or "s" means the same as the previously specified
; priority, again regardless of whether the previous entry was for the
; same extension.  Priorities may be immediately followed by a plus sign
; and another integer to add that amount (most useful with 's' or 'n').  
; Priorities may then also have an alias, or label, in 
; parenthesis after their name which can be used in goto situations
; Contexts contain several lines, one for each step of each
; extension, which can take one of two forms as listed below,
; with the first form being preferred.  One may include another
; context in the current one as well, optionally with a
; date and time.  Included contexts are included in the order
; they are listed.
;exten => someexten,{priority|label{+|-}offset}[(alias)],application(arg1,arg2,...)
;exten => someexten,{priority|label{+|-}offset}[(alias)],application,arg1|arg2...
; Timing list for includes is 
;   <time range>|<days of week>|<days of month>|<months>
; Note that ranges may be specified to wrap around the ends.  Also, minutes are
; fine-grained only down to the closest even minute.
;include => daytime|9:00-17:00|mon-fri|*|*
;include => weekend|*|sat-sun|*|*
;include => weeknights|17:02-8:58|mon-fri|*|*
; ignorepat can be used to instruct drivers to not cancel dialtone upon
; receipt of a particular pattern.  The most commonly used example is
; of course '9' like this:
;ignorepat => 9
; so that dialtone remains even after dialing a 9.

; Sample entries for extensions.conf
; List canonical entries here
;exten => 12564286000,1,Macro(stdexten,6000,IAX2/foo)
;exten => _125642860XX,1,Dial(IAX2/otherbox/${EXTEN:7})

; If you are an ITSP or Reseller, list your customers here.
;exten => _12564286000,1,Dial(SIP/customer1)
;exten => _12564286001,1,Dial(IAX2/customer2)

; If you are freely delivering calls to the PSTN, list them here
;exten => _1256428XXXX,1,Dial(Zap/g2/${EXTEN:7}) ; Expose all of 256-428 
;exten => _1256325XXXX,1,Dial(Zap/g2/${EXTEN:7}) ; Ditto for 256-325

; Context to put your dundi IAX2 or SIP user in for
; full access
include => dundi-e164-canonical
include => dundi-e164-customers
include => dundi-e164-via-pstn

; Just a wrapper for the switch
switch => DUNDi/e164

; Locally to lookup, try looking for a local E.164 solution
; then try DUNDi if we don't have one.
include => dundi-e164-local
include => dundi-e164-switch
; DUNDi can also be implemented as a Macro instead of using 
; the Local channel driver. 
; ARG1 is the extension to Dial
; Extension "s" is not a wildcard extension that matches "anything".
; In macros, it is the start extension. In most other cases, 
; you have to goto "s" to execute that extension.
; For wildcard matches, see above - all pattern matches start with
; an underscore.
exten => s,1,Goto(${ARG1},1)
include => dundi-e164-lookup

; Here are the entries you need to participate in the IAXTEL
; call routing system.  Most IAXTEL numbers begin with 1-700, but
; there are exceptions.  For more information, and to sign
; up, please go to or
exten => _91700XXXXXXX,1,Dial(IAX2/${GLOBAL(IAXINFO)}${EXTEN:1}@iaxtel)

; The SWITCH statement permits a server to share the dialplan with
; another server. Use with care: Reciprocal switch statements are not
; allowed (e.g. both A -> B and B -> A), and the switched server needs
; to be on-line or else dialing can be severly delayed.
;switch => IAX2/user:[key]@myserver/mycontext

; International long distance through trunk
exten => _9011.,1,Macro(dundi-e164,${EXTEN:4})
exten => _9011.,n,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})

; Long distance context accessed through trunk
exten => _91NXXNXXXXXX,1,Macro(dundi-e164,${EXTEN:1})

; Local seven-digit dialing accessed through trunk interface

; Long distance context accessed through trunk interface
exten => _91800NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
exten => _91888NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
exten => _91877NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
exten => _91866NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})

; Master context for international long distance
ignorepat => 9
include => longdistance
include => trunkint

; Master context for long distance
ignorepat => 9
include => local
include => trunkld

; Master context for local, toll-free, and iaxtel calls only
ignorepat => 9
include => default
include => trunklocal
include => iaxtel700
include => trunktollfree
include => iaxprovider

;Include parkedcalls (or the context you define in features conf)
;to enable call parking.
include => parkedcalls
; You can use an alternative switch type as well, to resolve
; extensions that are not known here, for example with remote 
; IAX switching you transparently get access to the remote
; Asterisk PBX
; switch => IAX2/user:password@bigserver/local
; An "lswitch" is like a switch but is literal, in that
; variable substitution is not performed at load time
; but is passed to the switch directly (presumably to
; be substituted in the switch routine itself)
; lswitch => Loopback/12${EXTEN}@othercontext
; An "eswitch" is like a switch but the evaluation of
; variable substitution is performed at runtime before
; being passed to the switch routine.
; eswitch => IAX2/context@${CURSERVER}

; Standard trunk dial macro (hangs up on a dialstatus that should 
; terminate call)
;   ${ARG1} - What to dial
exten => s,1,Dial(${ARG1})
exten => s,n,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,Hangup
exten => s-BUSY,1,Hangup
exten => _s-.,1,NoOp

; Standard extension macro:
;   ${ARG1} - Extension  (we could have used ${MACRO_EXTEN} here as well
;   ${ARG2} - Device(s) to ring
exten => s,1,Dial(${ARG2},20)			; Ring the interface, 20 seconds maximum
exten => s,2,Goto(s-${DIALSTATUS},1)		; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)

exten => s-NOANSWER,1,Voicemail(${ARG1},u)	; If unavailable, send to voicemail w/ unavail announce
exten => s-NOANSWER,2,Goto(default,s,1)		; If they press #, return to start

exten => s-BUSY,1,Voicemail(${ARG1},b)		; If busy, send to voicemail w/ busy announce
exten => s-BUSY,2,Goto(default,s,1)		; If they press #, return to start

exten => _s-.,1,Goto(s-NOANSWER,1)		; Treat anything else as no answer

exten => a,1,VoicemailMain(${ARG1})		; If they press *, send the user into VoicemailMain

; Standard extension macro:
;   ${ARG1} - Extension  (we could have used ${MACRO_EXTEN} here as well
;   ${ARG2} - Device(s) to ring
;   ${ARG3} - Optional DONTCALL context name to jump to (assumes the s,1 extension-priority)
;   ${ARG4} - Optional TORTURE context name to jump to (assumes the s,1 extension-priority)`
exten => s,1,Dial(${ARG2},20|p)			; Ring the interface, 20 seconds maximum, call screening 
						; option (or use P for databased call screening)
exten => s,2,Goto(s-${DIALSTATUS},1)		; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)

exten => s-NOANSWER,1,Voicemail(${ARG1},u)	; If unavailable, send to voicemail w/ unavail announce
exten => s-NOANSWER,2,Goto(default,s,1)		; If they press #, return to start

exten => s-BUSY,1,Voicemail(${ARG1},b)		; If busy, send to voicemail w/ busy announce
exten => s-BUSY,2,Goto(default,s,1)		; If they press #, return to start

exten => s-DONTCALL,1,Goto(${ARG3},s,1)		; Callee chose to send this call to a polite "Don't call again" script.

exten => s-TORTURE,1,Goto(${ARG4},s,1)		; Callee chose to send this call to a telemarketer torture script.

exten => _s-.,1,Goto(s-NOANSWER,1)		; Treat anything else as no answer

exten => a,1,VoicemailMain(${ARG1})		; If they press *, send the user into VoicemailMain

; Paging macro:
;       Check to see if SIP device is in use and DO NOT PAGE if they are
;   ${ARG1} - Device to page

exten => s,1,ChanIsAvail(${ARG1}|js)			; j is for Jump and s is for ANY call
exten => s,n,GoToIf([${AVAILSTATUS} = "1"]?autoanswer:fail)
exten => s,n(autoanswer),Set(_ALERT_INFO="RA")			; This is for the PolyComs
exten => s,n,SIPAddHeader(Call-Info: Answer-After=0)	; This is for the Grandstream, Snoms, and Others
exten => s,n,NoOp()					; Add others here and Post on the Wiki!!!!
exten => s,n,Dial(${ARG1}||)
exten => s,n(fail),Hangup

; We start with what to do when a call first comes in.
exten => s,1,Wait(1)			; Wait a second, just for fun
exten => s,n,Answer			; Answer the line
exten => s,n,Set(TIMEOUT(digit)=5)	; Set Digit Timeout to 5 seconds
exten => s,n,Set(TIMEOUT(response)=10)	; Set Response Timeout to 10 seconds
exten => s,n(restart),BackGround(demo-congrats)	; Play a congratulatory message
exten => s,n(instruct),BackGround(demo-instruct)	; Play some instructions
exten => s,n,WaitExten			; Wait for an extension to be dialed.

exten => 2,1,BackGround(demo-moreinfo)	; Give some more information.
exten => 2,n,Goto(s,instruct)

exten => 3,1,Set(LANGUAGE()=fr)		; Set language to french
exten => 3,n,Goto(s,restart)		; Start with the congratulations

exten => 1000,1,Goto(default,s,1)
; We also create an example user, 1234, who is on the console and has
; voicemail, etc.
exten => 1234,1,Playback(transfer,skip)		; "Please hold while..." 
					; (but skip if channel is not up)
exten => 1234,n,Macro(stdexten,1234,${GLOBAL(CONSOLE)})

exten => 1235,1,Voicemail(1234,u)		; Right to voicemail

exten => 1236,1,Dial(Console/dsp)		; Ring forever
exten => 1236,n,Voicemail(1234,b)		; Unless busy

; # for when they're done with the demo
exten => #,1,Playback(demo-thanks)	; "Thanks for trying the demo"
exten => #,n,Hangup			; Hang them up.

; A timeout and "invalid extension rule"
exten => t,1,Goto(#,1)			; If they take too long, give up
exten => i,1,Playback(invalid)		; "That's not valid, try again"

; Create an extension, 500, for dialing the
; Asterisk demo.
exten => 500,1,Playback(demo-abouttotry); Let them know what's going on
exten => 500,n,Dial(IAX2/	; Call the Asterisk demo
exten => 500,n,Playback(demo-nogo)	; Couldn't connect to the demo site
exten => 500,n,Goto(s,6)		; Return to the start over message.

; Create an extension, 600, for evaluating echo latency.
exten => 600,1,Playback(demo-echotest)	; Let them know what's going on
exten => 600,n,Echo			; Do the echo test
exten => 600,n,Playback(demo-echodone)	; Let them know it's over
exten => 600,n,Goto(s,6)		; Start over

;	You can use the Macro Page to intercom a individual user
exten => 76245,1,Macro(page,SIP/Grandstream1)
; or if your peernames are the same as extensions
exten => _7XXX,1,Macro(page,SIP/${EXTEN})
; System Wide Page at extension 7999
exten => 7999,1,Set(TIMEOUT(absolute)=60)
exten => 7999,2,Page(Local/Grandstream1@page&Local/Xlite1@page&Local/1234@page/n|d)

; Give voicemail at extension 8500
exten => 8500,1,VoicemailMain
exten => 8500,n,Goto(s,6)
; Here's what a phone entry would look like (IXJ for example)
;exten => 1265,1,Dial(Phone/phone0,15)
;exten => 1265,n,Goto(s,5)

;	The page context calls up the page macro that sets variables needed for auto-answer
;	It is in is own context to make calling it from the Page() application as simple as 
;	Local/{peername}@page
exten => _X.,1,Macro(page,SIP/${EXTEN})

; Example "main menu" context with submenu
;exten => s,1,Answer
;exten => s,n,Background(thanks)		; "Thanks for calling press 1 for sales, 2 for support, ..."
;exten => s,n,WaitExten
;exten => 1,1,Goto(submenu,s,1)
;exten => 2,1,Hangup
;include => default
;exten => s,1,Ringing					; Make them comfortable with 2 seconds of ringback
;exten => s,n,Wait,2
;exten => s,n,Background(submenuopts)	; "Thanks for calling the sales department.  Press 1 for steve, 2 for..."
;exten => s,n,WaitExten
;exten => 1,1,Goto(default,steve,1)
;exten => 2,1,Goto(default,mark,2)

; By default we include the demo.  In a production system, you 
; probably don't want to have the demo there.
include => demo

; An extension like the one below can be used for FWD, Nikotel, sipgate etc.
; Note that you must have a [sipprovider] section in sip.conf
;exten => _41X.,1,Dial(SIP/${EXTEN:2}@sipprovider,,r)

; Real extensions would go here. Generally you want real extensions to be
; 4 or 5 digits long (although there is no such requirement) and start with a
; single digit that is fairly large (like 6 or 7) so that you have plenty of
; room to overlap extensions and menu options without conflict.  You can alias
; them with names, too, and use global variables

;exten => 6245,hint,SIP/Grandstream1&SIP/Xlite1,Joe Schmoe ; Channel hints for presence
;exten => 6245,1,Dial(SIP/Grandstream1,20,rt)	; permit transfer
;exten => 6245,n(dial),Dial(${HINT},20,rtT)	; Use hint as listed
;exten => 6245,n,Voicemail(6245,u)		; Voicemail (unavailable)
;exten => 6245,s+1,Hangup			; s+1, same as n
;exten => 6245,dial+101,Voicemail(6245,b)	; Voicemail (busy)
;exten => 6361,1,Dial(IAX2/JaneDoe,,rm)		; ring without time limit
;exten => 6389,1,Dial(MGCP/aaln/1@
;exten => 6390,1,Dial(JINGLE/caller/callee) ; Dial via jingle using labels
;exten => 6391,1,Dial(JINGLE/ ;Dial via jingle using asterisk as the transport and calling mogorman.
;exten => 6394,1,Dial(Local/6275/n)		; this will dial ${MARK}

;exten => 6275,1,Macro(stdexten,6275,${MARK})	; assuming ${MARK} is something like Zap/2
;exten => mark,1,Goto(6275|1)			; alias mark to 6275
;exten => 6536,1,Macro(stdexten,6236,${WIL})	; Ditto for wil
;exten => wil,1,Goto(6236|1)

;If you want to subscribe to the status of a parking space, this is
;how you do it. Subscribe to extension 6600 in sip, and you will see
;the status of the first parking lot with this extensions' help
;exten => 6600,hint,park:701@parkedcalls
;exten => 6600,1,noop
; Some other handy things are an extension for checking voicemail via
; voicemailmain
;exten => 8500,1,VoicemailMain
;exten => 8500,n,Hangup
; Or a conference room (you'll need to edit meetme.conf to enable this room)
;exten => 8600,1,Meetme(1234)
; Or playing an announcement to the called party, as soon it answers
;exten = 8700,1,Dial(${MARK},30,A(/path/to/my/announcemsg))
; For more information on applications, just type "show applications" at your
; friendly Asterisk CLI prompt.
; 'show application <command>' will show details of how you
; use that particular application in this file, the dial plan. 
; 'show functions" will list all dialplan functions
; 'show function <COMMAND>' will show you more information about
; one function. Remember that function names are UPPER CASE.


exten => 9250,1,Dial(SIP/9250,20)
exten => 9250,2,Hangup

exten => 9251,1,Dial(SIP/9251,20)
exten => 9251,2,Hangup

can you edit your post, remove the sip debug, and post your sip.conf and extensions.conf instead ?

it’s hard to work through those with all the comments, but from what i can see it looks fine. when you start x-lite, so you see the registration on the Asterisk console ? what does “sip show peers” return ? do you get any activity on the console when you try to place a call to the other user ?

This is what Show peers show:

Name/username              Host            Dyn Nat ACL Port     Status          
9251/9251             D          24430    Unmonitored     
9250/9250             D          21078    Unmonitored     
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]

I see activity on the console when I try to make a call. The activity was the debug information I posted first time, I can repost it if you want.

And well, I see lota of info when the phone is registering. At last, the phone shows taht it has registered :S. The problem seems to be that when I phone the other phone, it is not seen, because I get same error with the other phone not connected to asterisk, the error in X-Lite is: Call failed, request timeout.

Again if you need any other information please let me know, and thanks for your time.


I have been doing some more tests and I have read in the asterisk book I can call back my phone, and I have done this, and I´m having the same result than calling the other soft phone. I have tried this both dial plans with the same result:

exten => 100,1,Dial(SIP/9250)
exten => 611,1,Echo()

I dial 100 and doesn´t seem to get back to the phone, and same with extension number 611 and the echo thing.

Odd, isn´t it?

Well at last it worked… My computer has “something” that is not letting my softphone see the other softphone. Have tried from my laptor (brought it this morning) and now I can call the other one. What a waste of time (3 days), but well I´m happy it worked.

Thanks anyway for all the help,