Dear SIr,
please find attached my sip.conf file
Regards
On Wed, Jan 28, 2009 at 5:37 PM, Danny Nicholas <[email protected]> wrote:
> Show us your sip.conf
>
>
> ------------------------------
>
> *From:* [email protected] [mailto:
> [email protected]] *On Behalf Of *michel freiha
> *Sent:* Wednesday, January 28, 2009 9:30 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] FAX
>
>
>
> Hi all,
>
> When trying to send a FAX I got the following error:
>
> Executing [003228949...@micho:1] Dial("SIP/028949469-08466918", "SIP/
> [email protected]|60") in new stack
> [Jan 28 17:19:20] WARNING[10040]: chan_sip.c:3039 sip_call: No audio format
> found to offer. Cancelling call to 003228949469
> -- Couldn't call [email protected]
>
> Where I should define the codec other than the extension in order to
> succeed the call?
>
> Regards
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
{\rtf1\ansi\ansicpg1252\deff0\deflang1033{\fonttbl{\f0\fswiss\fcharset0 Arial;}}
{\*\generator Msftedit 5.41.15.1515;}\viewkind4\uc1\pard\f0\fs20 ;\par
; SIP Configuration example for Asterisk\par
;\par
; Syntax for specifying a SIP device in extensions.conf is\par
; SIP/devicename where devicename is defined in a section below.\par
;\par
; You may also use\par
; SIP/usern...@domain to call any SIP user on the Internet\par
; (Don't forget to enable DNS SRV records if you want to use this)\par
;\par
; If you define a SIP proxy as a peer below, you may call\par
; SIP/proxyhostname/user or SIP/u...@proxyhostname\par
; where the proxyhostname is defined in a section below\par
;\par
; Useful CLI commands to check peers/users:\par
; sip show peers Show all SIP peers (including friends)\par
; sip show users Show all SIP users (including friends)\par
; sip show registry Show status of hosts we register with\par
;\par
; sip debug Show all SIP messages\par
;\par
; module reload chan_sip.so Reload configuration file\par
; Active SIP peers will not be reconfigured\par
;\par
\par
[general]\par
context=default ; Default context for incoming calls\par
;allowguest=no ; Allow or reject guest calls (default is
yes)\par
allowoverlap=no ; Disable overlap dialing support. (Default is
yes)\par
;allowtransfer=no ; Disable all transfers (unless enabled in
peers or users)\par
; Default is enabled\par
;realm=ergatel.net ; Realm for digest authentication\par
; defaults to "asterisk". If you set a system
name in\par
; asterisk.conf, it defaults to that system
name\par
; Realms MUST be globally unique according to
RFC 3261\par
; Set this to your host name or domain name\par
bindport=5060 ; UDP Port to bind to (SIP standard port is
5060)\par
; bindport is the local UDP port that Asterisk
will listen on\par
bindaddr=IP_ADDRESS ; IP address to bind to (0.0.0.0 binds to
all)\par
srvlookup=yes ; Enable DNS SRV lookups on outbound calls\par
; Note: Asterisk only uses the first host\par
; in SRV records\par
; Disabling DNS SRV lookups disables the\par
; ability to place SIP calls based on domain\par
; names to some other SIP users on the
Internet\par
;pedantic=yes ; Enable checking of tags in headers,\par
; international character conversions in
URIs\par
; and multiline formatted headers for strict\par
; SIP compatibility (defaults to "no")\par
\par
; See doc/ip-tos.txt for a description of these parameters.\par
;tos_sip=cs3 ; Sets TOS for SIP packets.\par
;tos_audio=ef ; Sets TOS for RTP audio packets.\par
;tos_video=af41 ; Sets TOS for RTP video packets.\par
\par
;maxexpiry=3600 ; Maximum allowed time of incoming
registrations\par
; and subscriptions (seconds)\par
;minexpiry=60 ; Minimum length of registrations/subscriptions
(default 60)\par
;defaultexpiry=120 ; Default length of incoming/outgoing
registration\par
;t1min=100 ; Minimum roundtrip time for messages to
monitored hosts\par
; Defaults to 100 ms\par
;notifymimetype=text/plain ; Allow overriding of mime type in MWI
NOTIFY\par
;checkmwi=10 ; Default time between mailbox checks for
peers\par
;buggymwi=no ; Cisco SIP firmware doesn't support the MWI
RFC\par
; fully. Enable this option to not get error
messages\par
; when sending MWI to phones with this bug.\par
;vmexten=voicemail ; dialplan extension to reach mailbox sets
the\par
; Message-Account in the MWI notify message\par
; defaults to "asterisk"\par
disallow=all ; First disallow all codecs\par
allow=ulaw ; Allow codecs in order of preference\par
allow=alaw\par
;allow=g729\par
;allow=g726\par
;allow=ilbc ; see doc/rtp-packetization for framing
options\par
;allow=alaw\par
; This option specifies a preference for which music on hold class this
channel\par
; should listen to when put on hold if the music class has not been set on
the\par
; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the
peer\par
; channel putting this one on hold did not suggest a music class.\par
;\par
; This option may be specified globally, or on a per-user or per-peer basis.\par
;\par
;mohinterpret=default\par
;\par
; This option specifies which music on hold class to suggest to the peer
channel\par
; when this channel places the peer on hold. It may be specified globally or
on\par
; a per-user or per-peer basis.\par
;;\par
;mohsuggest=default\par
;\par
;language=en ; Default language setting for all
users/peers\par
; This may also be set for individual
users/peers\par
;relaxdtmf=yes ; Relax dtmf handling\par
;trustrpid = no ; If Remote-Party-ID should be trusted\par
;sendrpid = yes ; If Remote-Party-ID should be sent\par
;progressinband=never ; If we should generate in-band ringing
always\par
; use 'never' to never use in-band signalling,
even in cases\par
; where some buggy devices might not render
it\par
; Valid values: yes, no, never Default:
never\par
useragent= Ergatel ; Allows you to change the user agent
string\par
;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP
address\par
; Note that promiscredir when redirects are
made to the\par
; local system will cause loops since Asterisk
is incapable\par
; of performing a "hairpin" call.\par
;usereqphone = no ; If yes, ";user=phone" is added to uri that
contains\par
; a valid phone number\par
;dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF.
Default: rfc2833\par
; Other options:\par
; info : SIP INFO messages\par
; inband : Inband audio (requires 64 kbit
codec -alaw, ulaw)\par
; auto : Use rfc2833 if offered, inband
otherwise\par
\par
;compactheaders = yes ; send compact sip headers.\par
;\par
;videosupport=yes ; Turn on support for SIP video. You need to
turn this on\par
; in the this section to get any video support
at all.\par
; You can turn it off on a per peer basis if
the general\par
; video support is enabled, but you can't
enable it for\par
; one peer only without enabling in the
general section.\par
;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384
kb/s)\par
; Videosupport and maxcallbitrate is
settable\par
; for peers and users as well\par
;callevents=no ; generate manager events when sip ua\par
; performs events (e.g. hold)\par
;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be
rejected,\par
; for any reason, always reject with '401
Unauthorized'\par
; instead of letting the requester know
whether there was\par
; a matching user or peer for their request\par
\par
;g726nonstandard = yes ; If the peer negotiates G726-32 audio, use
AAL2 packing\par
; order instead of RFC3551 packing order (this
is required\par
; for Sipura and Grandstream ATAs, among
others). This is ; order instead of RFC3551
packing order (this is required\par
; for Sipura and Grandstream ATAs, among
others). This is\par
; contrary to the RFC3551 specification, the
peer _should_\par
; be negotiating AAL2-G726-32 instead :-(\par
\par
;matchexterniplocally = yes ; Only substitute the externip or externhost
setting if it matches\par
; your localnet setting. Unless you have some
sort of strange network\par
; setup you will not need to enable this.\par
\par
;\par
; If regcontext is specified, Asterisk will dynamically create and destroy a\par
; NoOp priority 1 extension for a given peer who registers or unregisters
with\par
; us and have a "regexten=" configuration item.\par
; Multiple contexts may be specified by separating them with '&'. The\par
; actual extension is the 'regexten' parameter of the registering peer or
its\par
; name if 'regexten' is not provided. If more than one context is provided,\par
; the context must be specified within regexten by appending the desired\par
; context after '@'. More than one regexten may be supplied if they are\par
; separated by '&'. Patterns may be used in regexten.\par
;\par
;regcontext=sipregistrations\par
;\par
;--------------------------- RTP timers
----------------------------------------------------\par
; These timers are currently used for both audio and video streams. The RTP
timeouts\par
; are only applied to the audio channel.\par
; The settings are settable in the global section as well as per device\par
;\par
;rtptimeout=60 ; Terminate call if 60 seconds of no RTP or
RTCP activity\par
; on the audio channel\par
; when we're not on hold. This is to be able to
hangup\par
; a call in the case of a phone disappearing
from the net,\par
; like a powerloss or grandma tripping over a
cable.\par
;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or
RTCP activity\par
; on the audio channel\par
; when we're on hold (must be > rtptimeout)\par
;rtpkeepalive=<secs> ; Send keepalives in the RTP stream to keep NAT
open\par
; (default is off - zero)\par
;--------------------------- SIP DEBUGGING
---------------------------------------------------\par
;sipdebug = yes ; Turn on SIP debugging by default, from\par
; the moment the channel loads this
configuration\par
;recordhistory=yes ; Record SIP history by default\par
; (see sip history / sip no history)\par
;dumphistory=yes ; Dump SIP history at end of SIP dialogue\par
; SIP history is output to the DEBUG logging
channel\par
; SIP history is output to the DEBUG logging
channel\par
\par
\par
;--------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS)
----------------------------\par
; You can subscribe to the status of extensions with a "hint" priority\par
; (See extensions.conf.sample for examples)\par
; chan_sip support two major formats for notifications: dialog-info and
SIMPLE\par
;\par
; You will get more detailed reports (busy etc) if you have a call limit set\par
; for a device. When the call limit is filled, we will indicate busy. Note
that\par
; you need at least 2 in order to be able to do attended transfers.\par
;\par
; For queues, you will need this level of detail in status reporting,
regardless\par
; if you use SIP subscriptions. Queues and manager use the same internal
interface\par
; for reading status information.\par
;\par
; Note: Subscriptions does not work if you have a realtime dialplan and use
the\par
; realtime switch.\par
;\par
;allowsubscribe=no ; Disable support for subscriptions. (Default
is yes)\par
;subscribecontext = default ; Set a specific context for SUBSCRIBE
requests\par
; Useful to limit subscriptions to local
extensions\par
; Settable per peer/user also\par
;notifyringing = yes ; Control whether subscriptions already INUSE
get sent\par
; RINGING when another call is sent (default:
no)\par
;notifyhold = yes ; Notify subscriptions on HOLD state (default:
no)\par
; Turning on notifyringing and notifyhold will
add a lot\par
; more database transactions if you are using
realtime.\par
;limitonpeers = yes ; Apply call limits on peers only. This will
improve\par
; status notification when you are using
type=friend\par
; Inbound calls, that really apply to the user
part\par
; of a friend will now be added to and compared
with\par
; the peer limit instead of applying two call
limits,\par
; one for the peer and one for the user.\par
; "sip show inuse" will only show active calls
on\par
; the peer side of a "type=friend" object if
this\par
; setting is turned on.\par
\par
;----------------------------------------- T.38 FAX PASSTHROUGH SUPPORT
-----------------------\par
;\par
; This setting is available in the [general] section as well as in device
configurations.\par
; Setting this to yes, enables T.38 fax (UDPTL) passthrough on SIP to SIP
calls, provided\par
; both parties have T38 support enabled in their Asterisk configuration\par
; This has to be enabled in the general section for all devices to work. You
can then\par
; disable it on a per device basis.;\par
; T.38 faxing only works in SIP to SIP calls, with no local or agent channel
being used.\par
;\par
t38pt_udptl = yes ; Default false\par
;\par
;----------------------------------------- OUTBOUND SIP REGISTRATIONS
------------------------\par
; Asterisk can register as a SIP user agent to a SIP proxy (provider)\par
; Format for the register statement is:\par
; register => user[:secret[:authuse...@host[:port][/extension]\par
;\par
; If no extension is given, the 's' extension is used. The extension needs
to\par
; be defined in extensions.conf to be able to accept calls from this SIP
proxy\par
; (provider).\par
;\par
; host is either a host name defined in DNS or the name of a section defined\par
; below.\par
;\par
; Examples:\par
;\par
;register => 1234:[email protected]\par
;\par
; This will pass incoming calls to the 's' extension\par
;\par
;\par
;register => 2345:passw...@sip_proxy/1234\par
;\par
; Register 2345 at sip provider 'sip_proxy'. Calls from this provider\par
; connect to local extension 1234 in extensions.conf, default context,\par
; unless you configure a [sip_proxy] section below, and configure a\par
; context.\par
; Tip 1: Avoid assigning hostname to a sip.conf section like
[provider.com]\par
; Tip 2: Use separate type=peer and type=user sections for SIP providers\par
; (instead of type=friend) if you have calls in both directions\par
\par
;registertimeout=20 ; retry registration calls every 20 seconds
(default)\par
;registerattempts=10 ; Number of registration attempts before we
give up\par
; 0 = continue forever, hammering the other
server\par
; until it accepts the registration\par
; Default is 0 tries, continue forever\par
\par
;----------------------------------------- NAT SUPPORT
------------------------\par
; The externip, externhost and localnet settings are used if you use
Asterisk\par
; behind a NAT device to communicate with services on the outside.\par
;externip = 200.201.202.203 ; Address that we're going to put in outbound
SIP\par
; messages if we're behind a NAT\par
\par
; The externip and localnet is used\par
; when registering and communicating with other
proxies\par
; that we're registered with\par
;externhost=foo.dyndns.net ; Alternatively you can specify an\par
; external host, and Asterisk will\par
; perform DNS queries periodically. Not\par
; recommended for production\par
; environments! Use externip instead\par
;externrefresh=10 ; How often to refresh externhost if\par
; used\par
; You may add multiple local networks. A
reasonable\par
; set of defaults are:\par
;localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks\par
;localnet=10.0.0.0/255.0.0.0 ; Also RFC1918\par
;localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation\par
;localnet=169.254.0.0/255.255.0.0 ;Zero conf local network\par
\par
; The nat= setting is used when Asterisk is on a public IP, communicating
with\par
; devices hidden behind a NAT device (broadband router). If you have
one-way\par
; audio problems, you usually have problems with your NAT configuration or
your\par
; firewall's support of SIP+RTP ports. You configure Asterisk choice of RTP\par
; ports for incoming audio in rtp.conf\par
;\par
;nat=no ; Global NAT settings (Affects all peers and
users)\par
; yes = Always ignore info and assume NAT\par
; no = Use NAT mode only according to RFC3581
(;rport)\par
; never = Never attempt NAT mode or RFC3581
support\par
; route = Assume NAT, don't send rport\par
; (work around more UNIDEN bugs)\par
\par
;----------------------------------- MEDIA HANDLING
--------------------------------\par
; By default, Asterisk tries to re-invite the audio to an optimal path. If
there's\par
; no reason for Asterisk to stay in the media path, the media will be
redirected.\par
; This does not really work with in the case where Asterisk is outside and
have\par
; clients on the inside of a NAT. In that case, you want to set
canreinvite=nonat\par
;\par
;canreinvite=yes ; Asterisk by default tries to redirect the\par
; RTP media stream (audio) to go directly
from\par
; the caller to the callee. Some devices do
not\par
; support this (especially if one of them is
behind a NAT).\par
; The default setting is YES. If you have all
clients\par
\par
;directrtpsetup=yes ; Enable the new experimental direct RTP setup.
This sets up\par
; the call directly with media peer-2-peer
without re-invites.\par
; Will not work for video and cases where the
callee sends\par
; RTP payloads and fmtp headers in the 200 OK
that does not match the\par
; callers INVITE. This will also fail if
canreinvite is enabled when\par
; the device is actually behind NAT.\par
\par
;canreinvite=nonat ; An additional option is to allow media path
redirection\par
; (reinvite) but only when the peer where the
media is being\par
; sent is known to not be behind a NAT (as the
RTP core can\par
; determine it based on the apparent IP address
the media\par
; arrives from).\par
\par
;canreinvite=update ; Yet a third option... use UPDATE for media
path redirection,\par
; instead of INVITE. This can be combined with
'nonat', as\par
; 'canreinvite=update,nonat'. It implies
'yes'.\par
\par
;----------------------------------------- REALTIME SUPPORT
------------------------\par
; For additional information on ARA, the Asterisk Realtime Architecture,\par
; please read realtime.txt and extconfig.txt in the /doc directory of the\par
; source code.\par
;\par
rtcachefriends=yes ; Cache realtime friends by adding them to the
internal list\par
; just like friends added from the config file
only on a\par
; as-needed basis? (yes|no)\par
\par
;rtsavesysname=yes ; Save systemname in realtime database at
registration\par
; Default= no\par
\par
;rtupdate=yes ; Send registry updates to database using
realtime? (yes|no)\par
; If set to yes, when a SIP UA registers
successfully, the ip address,\par
; the origination port, the registration
period, and the username of\par
; the UA will be set to database via
realtime.\par
; If not present, defaults to 'yes'. Note:
realtime peers will\par
; probably not function across reloads in the
way that you expect, if\par
; you turn this option off.\par
;rtautoclear=yes ; Auto-Expire friends created on the fly on the
same schedule\par
; as if it had just registered?
(yes|no|<seconds>)\par
; If set to yes, when the registration expires,
the friend will\par
; vanish from the configuration until requested
again. If set\par
; to an integer, friends expire within this
number of seconds\par
; instead of the registration interval.\par
\par
;ignoreregexpire=yes ; Enabling this setting has two functions:\par
\par
;domain=mydomain.tld,mydomain-incoming\par
; Add domain and configure incoming context\par
; for external calls to this domain\par
;domain=1.2.3.4 ; Add IP address as local domain\par
; You can have several "domain" settings\par
;allowexternaldomains=no ; Disable INVITE and REFER to non-local
domains\par
; Default is yes\par
;autodomain=yes ; Turn this on to have Asterisk add local
host\par
; name and local IP to domain list.\par
\par
; fromdomain=mydomain.tld ; When making outbound SIP INVITEs to\par
; non-peers, use your primary domain
"identity"\par
; for From: headers instead of just your IP\par
; address. This is to be polite and\par
; it may be a mandatory requirement for some\par
; destinations which do not have a prior\par
; account relationship with your server.\par
\par
;------------------------------ JITTER BUFFER CONFIGURATION
--------------------------\par
; jbenable = yes ; Enables the use of a jitterbuffer on the
receiving side of a\par
; SIP channel. Defaults to "no". An enabled
jitterbuffer will\par
; be used only if the sending side can create and
the receiving\par
; side can not accept jitter. The SIP channel can
accept jitter,\par
; thus a jitterbuffer on the receive SIP side
will be used only\par
; if it is forced and enabled.\par
\par
; jbforce = no ; Forces the use of a jitterbuffer on the receive
side of a SIP\par
; channel. Defaults to "no".\par
\par
; jbmaxsize = 200 ; Max length of the jitterbuffer in
milliseconds.\par
\par
; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the
jitterbuffer is\par
; resynchronized. Useful to improve the quality
of the voice, with\par
; big jumps in/broken timestamps, usually sent
from exotic devices\par
; and programs. Defaults to 1000.\par
\par
; jbimpl = fixed ; Jitterbuffer implementation, used on the
receiving side of a SIP\par
; channel. Two implementations are currently
available - "fixed"\par
; (with size always equals to jbmaxsize) and
"adaptive" (with\par
; variable size, actually the new jb of IAX2).
Defaults to fixed.\par
\par
; jblog = no ; Enables jitterbuffer frame logging. Defaults to
"no".\par
;-----------------------------------------------------------------------------------\par
}
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
