[not solved] need a pro for hangup detection failure

Hello everyone,
I’m using asterisk 1.6.0.9 on ubuntu 8.04 with asterisk-gui 2.0 and devices connected to an ATA (Grandstream Handytone 502).

When I am placing a call to/from a device connected to the FXS port of the HT502, the calls are not detected to be hanged up when I hangup from a softphone or even from another HT502’s FXS port.

I really need help with this… Thank you very much for any help you could give me.

FYI: my location is indications.conf is set to [be]
and here are my ata’s call progress tones fields:

up

Please turn on SIP history, then post verbose CLI output for a problem call.

If the CLI output indicates that Asterisk still thinks the call is up, please provide the sip history for the incoming leg. If it indicates that the call has ended, please provide the sip history for the outgoing leg.

If you cannot get sip history, because the SIP channel no longer exists, it is very likely that the call has cleared, however, if you still think the problem is with Asterisk, please turn use sip set debug on and supply the SIP traces for incoming and outgoing legs.

Hello thanks for your tips…
Here’s an error message that appears when I tried a call:

I had history on but I don’t know how to check history as I don’t know the call ID…

Thank you very much for the time you’re spending to help me…

You can’t have a hangup problem, as the outgoing call failed because sip.conf was incorrect, not loaded, or the dialplan referenced an unknown SIP channel.

To have a hangup problem, you need to have a call to hang up.

You are going to have to provide VERBOSE console output, sip.conf and extensions.conf/.ael, for anyone to really debug this.

Hello,
here’s my sip.conf:

[quote];!
;! Automatically generated configuration file
;! Filename: sip.conf (/etc/asterisk/sip.conf)
;! Generator: Manager
;! Creation Date: Thu May 28 14:28:28 2009
;!
;
; SIP Configuration example for Asterisk
;
; SIP dial strings
;-----------------------------------------------------------
; In the dialplan (extensions.conf) you can use several
; syntaxes for dialing SIP devices.
; SIP/devicename
; SIP/username@domain (SIP uri)
; SIP/username[:password[:md5secret[:authname[:transport]]]]@host[:port]
; SIP/devicename/extension
;
;
; Devicename
; devicename is defined as a peer in a section below.
;
; username@domain
; Call any SIP user on the Internet
; (Don’t forget to enable DNS SRV records if you want to use this)
;
; devicename/extension
; 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
; This syntax also works with ATA’s with FXO ports
;
; SIP/username[:password[:md5secret[]]]@host[:port]
; This form allows you to specify password or md5secret and authname
; without altering any authentication data in config.
; Examples:
;
; SIP/*98@mysipproxy
; SIP/sales:topsecret::account02@domain.com:5062
; SIP/12345678::bc53f0ba8ceb1ded2b70e05c3f91de4f:myname@192.168.0.1
;
; All of these dial strings specify the SIP request URI.
; In addition, you can specify a specific To: header by adding an
; exclamation mark after the dial string, like
;
; SIP/sales@mysipproxy!sales@edvina.net
;
; CLI Commands
; -------------------------------------------------------------
; 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 set debug on Show all SIP messages
;
; module reload chan_sip.so Reload configuration file
; Active SIP peers will not be reconfigured
;
;------- Naming devices ------------------------------------------------------
;
; When naming devices, make sure you understand how Asterisk matches calls
; that come in.
; 1. Asterisk checks the SIP From: address username and matches against
; names of devices with type=user
; The name is the text between square brackets [name]
; 2. Asterisk checks the IP address (and port number) that the INVITE
; was sent from and matches against any devices with type=peer
;
; Don’t mix extensions with the names of the devices. Devices need a unique
; name. The device name is not used as phone numbers. Phone numbers are
; anything you declare as an extension in the dialplan (extensions.conf).
;
; Note: The parameter “username” is not the username and in most cases is
; not needed at all. Check below. In later releases, it’s renamed
; to “defaultuser” which is a better name, since it is used in
; combination with the “defaultip” setting.
;-----------------------------------------------------------------------------

; ** Deprecated configuration options **
; The “call-limit” configuation option is deprecated. It still works in
; this version of Asterisk, but will disappear in the next version.
; You are encouraged to use the dialplan groupcount functionality
; to enforce call limits instead of using this channel-specific method.
;
; You can still set limits per device in sip.conf or in a database by using
; “setvar” to set variables that can be used in the dialplan for various limits.

[general]
context = default

allowoverlap = no
bindport = 5060
bindaddr = 0.0.0.0

; You can specify port here too, like 123.123.123.123:5080

;
; Note that the TCP and TLS support for chan_sip is currently considered
; experimental. Since it is new, all of the related configuration options are
; subject to change in any release. If they are changed, the changes will
; be reflected in this sample configuration file, as well as in the UPGRADE.txt file.
;
tcpenable = no ; Enable server for incoming TCP connections (default is no)
tcpbindaddr = 0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
srvlookup = yes

allowexternaldomains = no
allowguest = yes
allowsubscribe = yes
allowtransfer = yes
alwaysauthreject = no
autodomain = no
callevents = no
canreinvite = no
checkmwi = 10
compactheaders = no
defaultexpiry = 120
domain = 192.168.1.1
dtmfmode = info
dumphistory = no
externrefresh = 10
fromdomain =
g726nonstandard = no
jbenable = no
jbforce = no
jbimpl =
jblog = no
jbmaxsize =
jbresyncthreshold =
language =
maxcallbitrate = 384
maxexpiry = 3600
minexpiry = 60
mohinterpret = default
mohsuggest =
mwi_from =
nat =
notifyringing = yes
pedantic = no
progressinband = never
promiscredir = no
realm = asterisk
recordhistory = no
registerattempts = 0
registertimeout = 20
relaxdtmf = yes
rtpholdtimeout = 120
rtptimeout = 30
sendrpid = no
sipdebug = no
subscribecontext =
t1min = 100
t38pt_udptl = no
trustrpid = no
useragent = Asterisk PBX
usereqphone = no
videosupport = no

disallow = all
allow = ulaw,alaw,gsm,ulaw,alaw,gsm,ulaw,alaw,gsm,ulaw,alaw,gsm

; 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 checking of tags in headers,
; international character conversions in URIs
; and multiline formatted headers for strict
; SIP compatibility (defaults to “no”)

; See qos.tex or Quality of Service section of asterisk.pdf 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.
tos_text=af41 ; Sets TOS for RTP text packets.

;cos_sip=3 ; Sets 802.1p priority for SIP packets.
;cos_audio=5 ; Sets 802.1p priority for RTP audio packets.
;cos_video=4 ; Sets 802.1p priority for RTP video packets.
;cos_text=3 ; Sets 802.1p priority for RTP text 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
;qualifyfreq=60 ; Qualification: How often to check for the
; host to be up in seconds
; Set to low value if you use low timeout for
; NAT of UDP sessions
;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
;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.
;
;mohinterpret=default
;
; 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.
;
;mohsuggest=default
;
;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
; The default user agent string also contains the Asterisk
; version. If you don’t want to expose this, change the
; useragent string.
;sdpsession=Asterisk PBX ; Allows you to change the SDP session name string, (s=)
; Like the useragent parameter, the default user agent string
; also contains the Asterisk version.
;sdpowner=root ; Allows you to change the username field in the SDP owner string, (o=)
; This field MUST NOT contain spaces
;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 (application/dtmf-relay)
; shortinfo : SIP INFO messages (application/dtmf)
; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
; auto : Use rfc2833 if offered, inband otherwise
;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 an identical response
; equivalent to valid username and invalid password/hash
; instead of letting the requester know whether there was
; a matching user or peer for their request. This reduces
; the ability of an attacker to scan for valid SIP usernames.

;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 :frowning:
;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the devices
;outboundproxy=proxy.provider.domain:8080 ; send outbound signaling to this proxy, not directly to the devices
;outboundproxy=proxy.provider.domain,force ; Send ALL outbound signalling to proxy, ignoring route: headers
;outboundproxy=tls://proxy.provider.domain ; same as ‘=proxy.provider.domain’ except we try to connect with tls
; ; (could also be tcp,udp) - defining transports on the proxy line only
; ; applies for the global proxy, otherwise use the transport= option
;matchexterniplocally = yes ; Only substitute the externip or externhost setting if it matches
; your localnet setting. Unless you have some sort of strange network
; setup you will not need to enable this.

;dynamic_exclude_static = yes ; Disallow all dynamic hosts from registering
; as any IP address used for staticly defined
; hosts. This helps avoid the configuration
; error of allowing your users to register at
; the same address as a SIP provider.

;contactdeny=0.0.0.0/0.0.0.0 ; Use contactpermit and contactdeny to
;contactpermit=172.16.0.0/255.255.0.0 ; restrict at what IPs your users may
; register their phones.

;
; 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.
;
;regcontext=sipregistrations
;regextenonqualify=yes ; Default “no”
; If you have qualify on and the peer becomes unreachable
; this setting will enforce inactivation of the regexten
; extension for the peer
;
;--------------------------- SIP timers ----------------------------------------------------
; These timers are used primarily in INVITE transactions.
; The default for Timer T1 is 500 ms or the measured run-trip time between
; Asterisk and the device if you have qualify=yes for the device.
;
;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
; Defaults to 100 ms
;timert1=500 ; Default T1 timer
; Defaults to 500 ms or the measured round-trip
; time to a peer (qualify=yes).
;timerb=32000 ; Call setup timer. If a provisional response is not received
; in this amount of time, the call will autocongest
; Defaults to 64*timert1

;--------------------------- 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= ; Send keepalives in the RTP stream to keep NAT open
; (default is off - zero)

;--------------------------- SIP Session-Timers (RFC 4028)------------------------------------
; SIP Session-Timers provide an end-to-end keep-alive mechanism for active SIP sessions.
; This mechanism can detect and reclaim SIP channels that do not terminate through normal
; signaling procedures. Session-Timers can be configured globally or at a user/peer level.
; The operation of Session-Timers is driven by the following configuration parameters:
;
; * session-timers - Session-Timers feature operates in the following three modes:
; originate : Request and run session-timers always
; accept : Run session-timers only when requested by other UA
; refuse : Do not run session timers in any case
; The default mode of operation is ‘accept’.
; * session-expires - Maximum session refresh interval in seconds. Defaults to 1800 secs.
; * session-minse - Minimum session refresh interval in seconds. Defualts to 90 secs.
; * session-refresher - The session refresher (uac|uas). Defaults to ‘uas’.
;
;session-timers=originate
;session-expires=600
;session-minse=90
;session-refresher=uas
;

;--------------------------- 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 counter enabled
; for a device.
;
; If you set the busylevel, we will indicate busy when we have a number of calls that
; matches the busylevel treshold.
;
; 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 ; Control whether subscriptions already INUSE get sent
; RINGING when another call is sent (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.
;callcounter = yes ; Enable call counters on devices. This can be set per
; device too.
;counteronpeer = yes ; Apply call counting 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 counter instead of applying two call counters,
; one for the peer and one for the user.
; “sip show inuse” will only show active calls on
; the peer side of a “type=friend” object if this
; setting is turned on.

;----------------------------------------- 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 => [transport://]user[:secret[]]@host[:port][/extension]
;
;
;
; domain is either
; - domain in DNS
; - host name in DNS
; - the name of a peer defined below or in realtime
; The domain is where you register your username, so your SIP uri you are registering to
; is username@domain
;
; 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).
;
; A similar effect can be achieved by adding a “callbackextension” option in a peer section.
; this is equivalent to having the following line in the general section:
;
; register => username:secret@host/callbackextension
;
; and more readable because you don’t have to write the parameters in two places
; (note that the “port” is ignored - this is a bug that should be fixed).
;
; Examples:
;
;register => 1234:password@mysipprovider.com
;
; 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

;----------------------------------------- NAT SUPPORT ------------------------
;
; WARNING: SIP operation behind a NAT is tricky and you really need
; to read and understand well the following section.
;
; When Asterisk is behind a NAT device, the “local” address (and port) that
; a socket is bound to has different values when seen from the inside or
; from the outside of the NATted network. Unfortunately this address must
; be communicated to the outside (e.g. in SIP and SDP messages), and in
; order to determine the correct value Asterisk needs to know:
;
; + whether it is talking to someone “inside” or “outside” of the NATted network.
; This is configured by assigning the “localnet” parameter with a list
; of network addresses that are considered “inside” of the NATted network.
; IF LOCALNET IS NOT SET, THE EXTERNAL ADDRESS WILL NOT BE SET CORRECTLY.
; Multiple entries are allowed, e.g. a reasonable set is the following:
;
; localnet=192.168.0.0/255.255.0.0 ; RFC 1918 addresses
; 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 “externally visible” address and port number to be used when talking
; to a host outside the NAT. This information is derived by one of the
; following (mutually exclusive) config file parameters:
;
; a. “externip = hostname[:port]” specifies a static address[:port] to
; be used in SIP and SDP messages.
; The hostname is looked up only once, when [re]loading sip.conf .
; If a port number is not present, use the “bindport” value (which is
; not guaranteed to work correctly, because a NAT box might remap the
; port number as well as the address).
; This approach can be useful if you have a NAT device where you can
; configure the mapping statically. Examples:
;
; externip = 12.34.56.78 ; use this address.
; externip = 12.34.56.78:9900 ; use this address and port.
; externip = mynat.my.org:12600 ; Public address of my nat box.
;
; b. “externhost = hostname[:port]” is similar to “externip” except
; that the hostname is looked up every “externrefresh” seconds
; (default 10s). This can be useful when your NAT device lets you choose
; the port mapping, but the IP address is dynamic.
; Beware, you might suffer from service disruption when the name server
; resolution fails. Examples:
;
; externhost=foo.dyndns.net ; refreshed periodically
; externrefresh=180 ; change the refresh interval
;
; c. “stunaddr = stun.server[:port]” queries the STUN server specified
; as an argument to obtain the external address/port.
; Queries are also sent periodically every “externrefresh” seconds
; (as a side effect, sending the query also acts as a keepalive for
; the state entry on the nat box):
;
; stunaddr = foo.stun.com:3478
; externrefresh = 15
;
; Note that at the moment all these mechanism work only for the SIP socket.
; The IP address discovered with externip/externhost/STUN is reused for
; media sessions as well, but the port numbers are not remapped so you
; may still experience problems.
;
; NOTE 1: in some cases, NAT boxes will use different port numbers in
; the internal<->external mapping. In these cases, the “externip” and
; “externhost” might not help you configure addresses properly, and you
; really need to use STUN.
;
; NOTE 2: when using “externip” or “externhost”, the address part is
; also used as the external address for media sessions.
; If you use “stunaddr”, STUN queries will be sent to the same server
; also from media sockets, and this should permit a correct mapping of
; the port numbers as well.
;
; In addition to the above, Asterisk has an additional “nat” parameter to
; address NAT-related issues in incoming SIP or media sessions.
; In particular, depending on the 'nat= ’ settings described below, Asterisk
; may override the address/port information specified in the SIP/SDP messages,
; and use the information (sender address) supplied by the network stack instead.
; However, this is only useful if the external traffic can reach us.
; The following settings are allowed (both globally and in individual sections):
;
; nat = no ; default. Use NAT mode only according to RFC3581 (;rport)
; nat = yes ; Always ignore info and assume NAT
; nat = never ; Never attempt NAT mode or RFC3581 support
; nat = route ; 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.

; 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. This will also fail if canreinvite is enabled when
; the device is actually behind NAT.

;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’. Note: realtime peers will
; probably not function across reloads in the way that you expect, if
; you turn this option off.
;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule
; as if it had just registered? (yes|no|)
; 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.
; REGISTER to non-local domains will be automatically denied if a domain
; list is configured.
;
; Domains can be specified using:
; domain=[,]
; 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

;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
;allowexternaldomains=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”.
;-----------------------------------------------------------------------------------

[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 = :@
; auth = #@
; Example:
;auth=mark:topsecret@digium.com
;
; 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
; transport transport
; 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 (deprecated)
; callcounter callcounter
; allowoverlap allowoverlap
; allowsubscribe allowsubscribe
; allowtransfer allowtransfer
; subscribecontext subscribecontext
; videosupport videosupport
; maxcallbitrate maxcallbitrate
; rfc2833compensate mailbox
; session-timers busylevel
; session-expires
; session-minse template
; session-refresher fromdomain
; t38pt_usertpsource regexten
; fromuser
; host
; port
; qualify
; defaultip
; defaultuser
; rtptimeout
; rtpholdtimeout
; sendrpid
; outboundproxy
; rfc2833compensate
; callbackextension
; registertrying
; session-timers
; session-expires
; session-minse
; session-refresher
; timert1
; timerb
; qualifyfreq
; t38pt_usertpsource
; contactpermit ; Limit what a host may register as (a neat trick
; contactdeny ; is to register at the same IP as a SIP provider,
; ; then call oneself, and get redirected to that
; ; same location).

;[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
;defaultuser=yourusername ; Authentication user for outbound proxies
;fromuser=yourusername ; Many SIP providers require this!
;fromdomain=provider.sip.domain
;host=box.provider.com
;transport=udp,tcp ; This sets the transport type to udp for outgoing, and will
; ; accept both tcp and udp. Default is udp. The first transport
; ; listed will always be used for outgoing connections.
;usereqphone=yes ; This provider requires “;user=phone” on URI
;callcounter=yes ; Enable call counter
;busylevel=2 ; Signal busy at 2 or more calls
;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the peer
;port=80 ; The port number we want to connect to on the remote side
; Also used as “defaultport” in combination with “defaultip” settings

;— sample definition for a provider
;[provider1]
;type=peer
;host=sip.provider1.com
;fromuser=4015552299 ; how your provider knows you
;secret=youwillneverguessit
;callbackextension=123 ; Register with this server and require calls coming back to this extension
;transport=udp,tcp ; This sets the transport type to udp for outgoing, and will
; ; accept both tcp and udp. Default is udp. The first transport
; ; listed will always be used for outgoing connections.

;------------------------------------------------------------------------------
; 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
;
; Because you might have a large number of similar sections, it is generally
; convenient to use templates for the common parameters, and add them
; the the various sections. Examples are below, and we can even leave
; the templates uncommented as they will not harm:

basic-options; a template
dtmfmode = rfc2833
context = from-office
type = friend

natted-phone; another template inheriting basic-options
nat = yes
canreinvite = no
host = dynamic

public-phone; another template inheriting basic-options
nat = no
canreinvite = yes

my-codecs; a template for my preferred codecs
disallow = all
allow = ilbc
allow = g729
allow = gsm
allow = g723
allow = ulaw

ulaw-phone; and another one for ulaw-only
disallow = all
allow = ulaw
; and finally instantiate a few phones
;
; 2133
; secret = peekaboo
; 2134
; secret = not_very_secret
; 2136
; secret = not_very_secret_either
; …
;

; Standard configurations not using templates look like this:
;
;[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
; on incoming calls to Asterisk
;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 (deprecated)
; 1 for the explicit peer, 1 for the explicit user,
; remember that a friend equals 1 peer and 1 user in
; memory
; 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=alaw
;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

;[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
;registertrying=yes ; Send a 100 Trying when the device registers.

;[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
;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”
;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
;defaultuser=polly ; Username to use in INVITE until peer registers
;defaultip=192.168.40.123
; 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
;qualifyfreq=60 ; Qualification: How often to check for the
; host to be up in seconds
; Set to low value if you use low timeout for
; NAT of UDP sessions
;
; 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=192.168.0.60 ; IP address to use if peer has not registered
;deny=0.0.0.0/0.0.0.0 ; ACL: Control access to this account based on IP address
;permit=192.168.0.60/255.255.255.0

;[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
;defaultuser=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

;[pre14-asterisk]
;type=friend
;secret=digium
;host=dynamic
;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.
;t38pt_usertpsource=yes ; Use the source IP address of RTP as the destination IP address for UDPTL packets
; if the nat option is enabled. If a single RTP packet is received Asterisk will know the
; external IP address of the remote device. If port forwarding is done at the client side
; then UDPTL will flow to the remote device.[/quote]

Here’s extensions.conf: (My dialpan is the one at the end (DLPN_CH2O] and i also use the lst ring group)

…and my extensions.ael:

What generated these? Where are your SIP lines and trunks defined? What context and extension are you trying to dial when it goes wrong?

I turned on SIP debug and here’s what happens when I place a call dialing 6400 (which is a dialgroup), talk a few seconds then Hang Up. The caller, after the other party hangs up, can hear a continuous tone (like the one I can hear before I place a call). then, after a few second, there is a tone like an alarm: BIP BIP BIP BIP BIP…

Here’s the debug output of the entire call:

[quote]<— SIP read from UDP://192.168.1.3:5060 —>
INVITE sip:6400@192.168.1.1;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK921439626;rport
From: “6000” sip:6000@192.168.1.1;user=phone;tag=1272071100
To: sip:6400@192.168.1.1;user=phone
Call-ID: 141878049-5060-6@192.168.1.3
CSeq: 50 INVITE
Contact: sip:6000@192.168.1.3:5060;user=phone
Max-Forwards: 70
User-Agent: Grandstream HT-502 V1.1B 1.0.1.21
Privacy: none
P-Asserted-Identity: “6000” sip:6000@192.168.1.1;user=phone
Supported: replaces, path, timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 429

v=0
o=6000 8000 8000 IN IP4 192.168.1.3
s=SIP Call
c=IN IP4 192.168.1.3
t=0 0
m=audio 5004 RTP/AVP 0 8 4 18 2 97 103 102 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:103 AAL2-G726-40/8000
a=rtpmap:102 G729E/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32-36,54

<------------->
— (16 headers 19 lines) —
Sending to 192.168.1.3 : 5060 (NAT)
Using INVITE request as basis request - 141878049-5060-6@192.168.1.3
Found user ‘6000’ for '6000’
WaterTowerServer*CLI>
<— Reliably Transmitting (no NAT) to 192.168.1.3:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK921439626;received=192.168.1.3;rport=5060
From: “6000” sip:6000@192.168.1.1;user=phone;tag=1272071100
To: sip:6400@192.168.1.1;user=phone;tag=as526afc98
Call-ID: 141878049-5060-6@192.168.1.3
CSeq: 50 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="2957c7f4"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘141878049-5060-6@192.168.1.3’ in 32000 ms (Method: INVITE)
WaterTowerServer*CLI>
<— SIP read from UDP://192.168.1.3:5060 —>
ACK sip:6400@192.168.1.1;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK921439626;rport
From: “6000” sip:6000@192.168.1.1;user=phone;tag=1272071100
To: sip:6400@192.168.1.1;user=phone;tag=as526afc98
Call-ID: 141878049-5060-6@192.168.1.3
CSeq: 50 ACK
Content-Length: 0

<------------->
— (7 headers 0 lines) —
WaterTowerServer*CLI>
<— SIP read from UDP://192.168.1.3:5060 —>
INVITE sip:6400@192.168.1.1;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK1567133396;rport
From: “6000” sip:6000@192.168.1.1;user=phone;tag=1272071100
To: sip:6400@192.168.1.1;user=phone
Call-ID: 141878049-5060-6@192.168.1.3
CSeq: 51 INVITE
Contact: sip:6000@192.168.1.3:5060;user=phone
Authorization: Digest username=“6000”, realm=“asterisk”, nonce=“2957c7f4”, uri="sip:6400@192.168.1.1;user=phone", response=“5e76bf3b1dfccca0b9bbb1d47bac1dbc”, algorithm=MD5
Max-Forwards: 70
User-Agent: Grandstream HT-502 V1.1B 1.0.1.21
Privacy: none
P-Asserted-Identity: “6000” sip:6000@192.168.1.1;user=phone
Supported: replaces, path, timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 429

v=0
o=6000 8000 8000 IN IP4 192.168.1.3
s=SIP Call
c=IN IP4 192.168.1.3
t=0 0
m=audio 5004 RTP/AVP 0 8 4 18 2 97 103 102 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:103 AAL2-G726-40/8000
a=rtpmap:102 G729E/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32-36,54

<------------->
— (17 headers 19 lines) —
Sending to 192.168.1.3 : 5060 (NAT)
Using INVITE request as basis request - 141878049-5060-6@192.168.1.3
Found user ‘6000’ for '6000’
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 2
Found RTP audio format 97
Found RTP audio format 103
Found RTP audio format 102
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.3:5004
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G723 for ID 4
Found audio description format G729 for ID 18
Found audio description format G726-32 for ID 2
Found audio description format iLBC for ID 97
Found unknown media description format AAL2-G726-40 for ID 103
Found unknown media description format G729E for ID 102
Found audio description format telephone-event for ID 101
Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0xd0d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.1.3:5004
Looking for 6400 in DLPN_CH2O (domain 192.168.1.1)
list_route: hop: sip:6000@192.168.1.3:5060;user=phone

<— Transmitting (no NAT) to 192.168.1.3:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK1567133396;received=192.168.1.3;rport=5060
From: “6000” sip:6000@192.168.1.1;user=phone;tag=1272071100
To: sip:6400@192.168.1.1;user=phone
Call-ID: 141878049-5060-6@192.168.1.3
CSeq: 51 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: sip:6400@192.168.1.1
Content-Length: 0

<------------>
Really destroying SIP dialog ‘0815856e652125427791bec83f7023dd@192.168.1.1’ Method: INVITE
[May 28 15:59:33] WARNING[8207]: app_dial.c:1468 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 20 - Unknown)
Audio is at 192.168.1.1 port 10408
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Reliably Transmitting (no NAT) to 192.168.1.3:5062:
INVITE sip:6001@192.168.1.3:5062;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK6a82b148;rport
Max-Forwards: 70
From: “Fasttel” sip:6000@192.168.1.1;tag=as68f32060
To: sip:6001@192.168.1.3:5062;user=phone
Contact: sip:6000@192.168.1.1
Call-ID: 598f4fe52f46234f5e090f75049315e6@192.168.1.1
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Thu, 28 May 2009 13:59:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 226

v=0
o=root 150744708 150744708 IN IP4 192.168.1.1
s=Asterisk PBX 1.6.0.9
c=IN IP4 192.168.1.1
t=0 0
m=audio 10408 RTP/AVP 0 3
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv


WaterTowerServer*CLI>
<— SIP read from UDP://192.168.1.3:5062 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK6a82b148;rport=5060
From: “Fasttel” sip:6000@192.168.1.1;tag=as68f32060
To: sip:6001@192.168.1.3:5062;user=phone
Call-ID: 598f4fe52f46234f5e090f75049315e6@192.168.1.1
CSeq: 102 INVITE
Supported: replaces, path, timer
User-Agent: Grandstream HT-502 V1.1B 1.0.1.21
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

<------------->
— (10 headers 0 lines) —
WaterTowerServer*CLI>
<— SIP read from UDP://192.168.1.3:5062 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK6a82b148;rport=5060
From: “Fasttel” sip:6000@192.168.1.1;tag=as68f32060
To: sip:6001@192.168.1.3:5062;user=phone;tag=96226728
Call-ID: 598f4fe52f46234f5e090f75049315e6@192.168.1.1
CSeq: 102 INVITE
Contact: sip:6001@192.168.1.3:5062;user=phone
Supported: replaces, path, timer
User-Agent: Grandstream HT-502 V1.1B 1.0.1.21
Allow-Events: talk, hold
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

<------------->
— (12 headers 0 lines) —
WaterTowerServer*CLI>
<— Transmitting (no NAT) to 192.168.1.3:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK1567133396;received=192.168.1.3;rport=5060
From: “6000” sip:6000@192.168.1.1;user=phone;tag=1272071100
To: sip:6400@192.168.1.1;user=phone;tag=as40fbc396
Call-ID: 141878049-5060-6@192.168.1.3
CSeq: 51 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: sip:6400@192.168.1.1
Content-Length: 0

<------------>
WaterTowerServer*CLI>
<— SIP read from UDP://192.168.1.3:5062 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK6a82b148;rport=5060
From: “Fasttel” sip:6000@192.168.1.1;tag=as68f32060
To: sip:6001@192.168.1.3:5062;user=phone;tag=96226728
Call-ID: 598f4fe52f46234f5e090f75049315e6@192.168.1.1
CSeq: 102 INVITE
Contact: sip:6001@192.168.1.3:5062;user=phone
Supported: replaces, path, timer
User-Agent: Grandstream HT-502 V1.1B 1.0.1.21
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Type: application/sdp
Content-Length: 180

v=0
o=6001 8002 8000 IN IP4 192.168.1.3
s=SIP Call
c=IN IP4 192.168.1.3
t=0 0
m=audio 5012 RTP/AVP 0
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=silenceSupp:off - - - -

<------------->
— (12 headers 10 lines) —
Found RTP audio format 0
Peer audio RTP is at port 192.168.1.3:5012
Found audio description format PCMU for ID 0
Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.1.3:5012
list_route: hop: sip:6001@192.168.1.3:5062;user=phone
set_destination: Parsing sip:6001@192.168.1.3:5062;user=phone for address/port to send to
set_destination: set destination to 192.168.1.3, port 5062
Transmitting (no NAT) to 192.168.1.3:5062:
ACK sip:6001@192.168.1.3:5062;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK3e471337;rport
Max-Forwards: 70
From: “Fasttel” sip:6000@192.168.1.1;tag=as68f32060
To: sip:6001@192.168.1.3:5062;user=phone;tag=96226728
Contact: sip:6000@192.168.1.1
Call-ID: 598f4fe52f46234f5e090f75049315e6@192.168.1.1
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0


Audio is at 192.168.1.1 port 10758
Adding codec 0x4 (ulaw) to SDP

<— Reliably Transmitting (no NAT) to 192.168.1.3:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK1567133396;received=192.168.1.3;rport=5060
From: “6000” sip:6000@192.168.1.1;user=phone;tag=1272071100
To: sip:6400@192.168.1.1;user=phone;tag=as40fbc396
Call-ID: 141878049-5060-6@192.168.1.3
CSeq: 51 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: sip:6400@192.168.1.1
Content-Type: application/sdp
Content-Length: 203

v=0
o=root 535139182 535139182 IN IP4 192.168.1.1
s=Asterisk PBX 1.6.0.9
c=IN IP4 192.168.1.1
t=0 0
m=audio 10758 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
WaterTowerServer*CLI>
<— SIP read from UDP://192.168.1.3:5060 —>
ACK sip:6400@192.168.1.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK1545443768;rport
From: “6000” sip:6000@192.168.1.1;user=phone;tag=1272071100
To: sip:6400@192.168.1.1;user=phone;tag=as40fbc396
Call-ID: 141878049-5060-6@192.168.1.3
CSeq: 51 ACK
Contact: sip:6000@192.168.1.3:5060;user=phone
Max-Forwards: 70
Supported: replaces, path, timer
User-Agent: Grandstream HT-502 V1.1B 1.0.1.21
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

<------------->
— (12 headers 0 lines) —
WaterTowerServer*CLI>
<— SIP read from UDP://192.168.1.3:5062 —>
BYE sip:6000@192.168.1.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5062;branch=z9hG4bK1397535960;rport
From: sip:6001@192.168.1.3:5062;user=phone;tag=96226728
To: “Fasttel” sip:6000@192.168.1.1;tag=as68f32060
Call-ID: 598f4fe52f46234f5e090f75049315e6@192.168.1.1
CSeq: 103 BYE
Contact: sip:6001@192.168.1.3:5062;user=phone
Max-Forwards: 70
Supported: replaces, path, timer
User-Agent: Grandstream HT-502 V1.1B 1.0.1.21
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Sending to 192.168.1.3 : 5062 (NAT)

<— Transmitting (NAT) to 192.168.1.3:5062 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.3:5062;branch=z9hG4bK1397535960;received=192.168.1.3;rport=5062
From: sip:6001@192.168.1.3:5062;user=phone;tag=96226728
To: “Fasttel” sip:6000@192.168.1.1;tag=as68f32060
Call-ID: 598f4fe52f46234f5e090f75049315e6@192.168.1.1
CSeq: 103 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘141878049-5060-6@192.168.1.3’ in 32000 ms (Method: ACK)
set_destination: Parsing sip:6000@192.168.1.3:5060;user=phone for address/port to send to
set_destination: set destination to 192.168.1.3, port 5060
Reliably Transmitting (no NAT) to 192.168.1.3:5060:
BYE sip:6000@192.168.1.3:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK6ee5cf3a;rport
Max-Forwards: 70
From: sip:6400@192.168.1.1;user=phone;tag=as40fbc396
To: “6000” sip:6000@192.168.1.1;user=phone;tag=1272071100
Call-ID: 141878049-5060-6@192.168.1.3
CSeq: 102 BYE
User-Agent: Asterisk PBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


WaterTowerServer*CLI>
<— SIP read from UDP://192.168.1.3:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK6ee5cf3a;rport=5060
From: sip:6400@192.168.1.1;user=phone;tag=as40fbc396
To: “6000” sip:6000@192.168.1.1;user=phone;tag=1272071100
Call-ID: 141878049-5060-6@192.168.1.3
CSeq: 102 BYE
Contact: sip:6000@192.168.1.3:5060;user=phone
Supported: replaces, path, timer
User-Agent: Grandstream HT-502 V1.1B 1.0.1.21
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

<------------->
— (11 headers 0 lines) —
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog ‘598f4fe52f46234f5e090f75049315e6@192.168.1.1’ Method: BYE
Really destroying SIP dialog ‘141878049-5060-6@192.168.1.3’ Method: ACK
[/quote]

[quote]<— SIP read from UDP://192.168.1.3:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK6ee5cf3a;rport=5060
From: sip:6400@192.168.1.1;user=phone;tag=as40fbc396 [/quote]

The called party has acknowledged the hangup from Asterisk. Look elsewhere.

How is hangup signalled over the FXS lines (by default there is no mechanism to do that).

Thank you very much already, I have now the confirmation that my Asterisk config is OK.
It means that the ATA configuration is the source of the problem.
I’m using a Grandstream Handytone 502 (ATA). The device that does not detect the hangup (and has extension 6000) is connected to FXS port 1.

Now I need to find how to update the ATA (latest firmware already) so that it detect the hangup…