I am running asterisk 1.0.1 with OH323 compiled in.
We have a 323 trunk to CallManager with a mgcp controlled pri router.
When using sip phones (directly registered with asterisk) to call out the 323 trunnk to PSTN, calls timeout after 3 rings. If I answer b4 3 rings - no problem, otherwise I get "no one is available to answer at this time" on the consoel and it redirects to an extension in extensions.conf under a different context.
Any ideas on where I should be looking:
Thanks,
Greg Oliver
configs follow:
sip.conf----
sip*CLI> sip*CLI> sip*CLI> exit Executing last minute cleanups [EMAIL PROTECTED] asterisk]# cat sip.conf ; ; SIP Configuration for Asterisk ; ; Syntax for specifying a SIP device in extensions.conf is ; SIP/devicename where devicename is defined in a section below. ; ; You may also use ; SIP/[EMAIL PROTECTED] to call any SIP user on the Internet ; (Don't forget to enable DNS SRV records if you want to use this) ; ; If you define a SIP proxy as a peer below, you may call ; SIP/proxyhostname/user or SIP/[EMAIL PROTECTED] ; where the proxyhostname is defined in a section below ; ; Useful CLI commands to check peers/users: ; sip show peers Show all SIP peers (including friends) ; sip show users Show all SIP users (including friends) ; sip show registry Show status of hosts we register with ; ; sip debug Show all SIP messages ;
[general] context=default ; Default context for incoming calls ;realm=mydomain.tld ; Realm for digest authentication ; defaults to "asterisk" ; Realms MUST be globally unique according to RFC 3261 ; Set this to your host name or domain name port=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) srvlookup=no ; Enable DNS SRV lookups on outbound calls ; Note: Asterisk only uses the first host ; in SRV records ; Disabling DNS SRV lookups disables the ; ability to place SIP calls based on domain ; names to some other SIP users on the Internet
;pedantic=yes ; Enable slow, pedantic checking for Pingtel ; and multiline formatted headers for strict ; SIP compatibility ;tos=184 ; Set IP QoS to either a keyword or numeric val ;tos=reliability ; lowdelay,throughput,reliability,mincost,none ;maxexpirey=3600 ; Max length of incoming registration we allow ;defaultexpirey=120 ; Default length of incoming/outoing registration notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY ;videosupport=yes ; Turn on support for SIP video
disallow=all ; First disallow all codecs allow=ulaw ; Allow codecs in order of preference ;allow=ilbc ; Note: codec order is respected only in [general] ;musicclass=default ; Sets the default music on hold class for all SIP calls ; This may also be set for individual users/peers ;language=en ; Default language setting for all users/peers ; This may also be set for individual users/peers ;relaxdtmf=yes ; Relax dtmf handling ;rtptimeout=60 ; Terminate call if 60 seconds of no RTP activity ; when we're not on hold ;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP activity ; when we're on hold (must be > rtptimeout)
; Asterisk can register as a SIP user agent to a SIP proxy (provider) ; Format for the register statement is: ; register => user[:secret[:[EMAIL PROTECTED]:port][/extension] ; ; If no extension is given, the 's' extension is used. The extension ; needs to be defined in extensions.conf to be able to accept calls ; from this SIP proxy (provider) ; ; host is either a host name defined in DNS or the name of a ; section defined below. ; ; Examples: ; ;register => 1234:[EMAIL PROTECTED] ; ; This will pass incoming calls to the 's' extension ; ; ;register => 2345:[EMAIL PROTECTED]/1234 ; ; Register 2345 at sip provider 'sip_proxy'. Calls from this provider connect to local ; extension 1234 in extensions.conf default context, unless you define ; unless you configure a [sip_proxy] section below, and configure a context. ; Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com] ; Tip 2: Use separate type=peer and type=user sections for SIP providers ; (instead of type=friend) if you have calls in both directions
;externip = 200.201.202.203 ; Address that we're going to put in outbound SIP messages ; if we're behind a NAT
; The externip and localnet is used ; when registering and communicating with other proxies ; that we're registered with ; You may add multiple local networks. A reasonable set of defaults ; are: ;localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks ;localnet=10.0.0.0/255.0.0.0 ; Also RFC1918 ;localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation ;localnet=169.254.0.0/255.255.0.0 ;Zero conf local network localnet=206.123.138.0/255.255.255.0
;----------------------------------------------------------------------------------- ; Users and peers have different settings available. Friends have all settings, ; since a friend is both a peer and a user ; ; User config options: Peer configuration: ; -------------------- ------------------- ; context context ; permit permit ; deny deny ; auth auth ; secret secret ; md5secret md5secret ; dtmfmode dtmfmode ; canreinvite canreinvite ; nat nat ; callgroup callgroup ; pickupgroup pickupgroup ; language language ; allow allow ; disallow disallow ; insecure insecure ; callerid ; accountcode ; amaflags ; incominglimit ; outgoinglimit ; restrictcid ; mailbox ; username ; template ; fromdomain ; fromuser ; host ; mask ; port ; qualify ; defaultip ; rtptimeout ; rtpholdtimeout
;[sip_proxy] ; For incoming calls only. Example: FWD (Free World Dialup) ;type=user ;context=from-fwd
;[sip_proxy-out] ;type=peer ; we only want to call out, not be called ;secret=guessit ;username=yourusername ;fromuser=yourusername ; Many SIP providers require this! ;host=box.provider.com
;[grandstream1] ;type=friend ; either "friend" (peer+user), "peer" or "user" ;context=from-sip ;username=grandstream1 ; usually matches the [section] title ;fromuser=grandstream1 ; overrides the callerid, e.g. required by FWD ;callerid=John Doe <1234> ;host=192.168.0.23 ; we have a static but private IP address ;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 ;outgoinglimit=1 ; disable callwaiting signal (2nd call to phone) ;incominglimit=1 ; permit only 1 outgoing call at a time ;[EMAIL PROTECTED] ; 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
;[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 ;username=xlite1 ;callerid="Jane Smith" <5678> ;host=dynamic ;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
;[snom] ;type=friend ; Friends place calls and receive calls ;context=from-sip ; Context for incoming calls from this user ;secret=blah ;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=1234,2345 ; Mailboxes for message waiting indicator ;restrictcid=yes ; To have the callerid restriced -> sent as ANI ;disallow=all ;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw! ;[EMAIL PROTECTED],2345 ; Mailbox(-es) for message waiting indicator
;[pingtel] ;type=friend ;username=pingtel ;secret=blah ;host=dynamic ;insecure=yes ; To match a peer based by IP address only and not peer ;insecure=very ; To allow registered hosts to call without re-authenticating ;qualify=1000 ; Consider it down if it's 1 second to reply ; Helps with NAT session ; qualify=yes uses default value ;callgroup=1,3-4 ; We are in caller groups 1,3,4 ;pickupgroup=1,3-5 ; We can do call pick-p for call group 1,3,4,5 ;defaultip=192.168.0.60 ; IP address to use if peer has not registred
;[cisco1] ;type=friend ;username=cisco1 ;secret=blah ;qualify=200 ; Qualify peer is no more than 200ms away ;nat=yes ; This phone may be natted ; Send SIP and RTP to 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
;[cisco2] ;type=friend ;username=cisco2 ;fromuser=markster ; Specify user to put in "from" instead of callerid ;fromdomain=yourdomain.com ; Specify domain to put in "from" instead of callerid ; fromuser and fromdomain are used when Asterisk ; places calls to this account. It is not used for ; calls from this account. ;secret=blah ;host=dynamic ;defaultip=192.168.0.4 ;amaflags=default ; Choices are default, omit, billing, documentation ;accountcode=markster ; Users may be associated with an accountcode to ease billing
[75000] type=friend secret= auth=md5 nat=yes host=dynamic reinvite=no canreinvite=no qualify=1000 dtmfmode=rfc2833 callerid="Greg R <75000>" disallow=all allow=gsm allow=ulaw context=default mailbox=5000
[74678] type=friend username=74678 secret= qualify=200 host=dynamic canreinvite=yes > [EMAIL PROTECTED] asterisk]# cat h323.conf ; The NuFone Network's ; Open H.323 driver configuration ; [general] port = 1720 bindaddr = 0.0.0.0 ;tos=lowdelay ; ; You may specify a global default AMA flag for iaxtel calls. It must be ; one of 'default', 'omit', 'billing', or 'documentation'. These flags ; are used in the generation of call detail records. ; ;amaflags = default ; ; You may specify a default account for Call Detail Records in addition ; to specifying on a per-user basis ; ;accountcode=lss0101 ; ; You can fine tune codecs here using "allow" and "disallow" clauses ; with specific codecs. Use "all" to represent all formats. ; disallow=all ;allow=all ; turns on all installed codecs ;disallow=g723.1 ; Hm... Proprietary, don't use it... allow=gsm ; Always allow GSM, it's cool :) allow=ulaw ; ; User-Input Mode (DTMF) ; ; valid entries are: rfc2833, inband ; default is rfc2833 dtmfmode=rfc2833 ; ; Set the gatekeeper ; DISCOVER - Find the Gk address using multicast ; DISABLE - Disable the use of a GK ; <IP address> or <Host name> - The acutal IP address or hostname of your GK gatekeeper = 192.168.5.20 ; ; ; Tell Asterisk whether or not to accept Gatekeeper ; routed calls or not. Normally this should always ; be set to yes, unless you want to have finer control ; over which users are allowed access to Asterisk. ; Default: YES ; AllowGKRouted = yes ; ; Default context gets used in siutations where you are using ; the GK routed model or no type=user was found. This gives you ; the ability to either play an invalid message or to simply not ; use user authentication at all. ; context=default ; ; H.323 Alias definitions ; ; Type 'h323' will register aliases to the endpoint ; and Gatekeeper, if there is one. ; ; Example: if someone calls [EMAIL PROTECTED] ; Asterisk will send the call to the extension 'time' ; in the context default ; [default] type=h323 context=default
; Keyword's 'prefix' and 'e164' are only make sense when ; used with a gatekeeper. You can specify either a prefix ; or E.164 this endpoint is responsible for terminating. ; ; Example: The H.323 alias 'det-gw' will tell the gatekeeper ; to route any call with the prefix 1248 to this alias. Keyword ; e164 is used when you want to specifiy a full telephone ; number. So a call to the number 18102341212 would be ; routed to the H.323 alias 'time'. ;
; Voice Mail Entry [14000] type=h323 context=default
; Voice Mail on No Answer [14001] type=h323 context=default
; Voice Mail on Busy [14002] type=h323 context=default
[7000] type=h323 context=meetme
[7001] type=h323 context=meetme
[7002] type=h323 context=meetme
[7003] type=h323 context=meetme
[7004] type=h323 context=meetme
[7005] type=h323 context=meetme
[7006] type=h323 context=meetme
[7007] type=h323 context=meetme
[7008] type=h323 context=meetme
[7009] type=h323 context=meetme
[2050] type=h323 context=inbound
[2051] type=h323 context=support
[2052] type=h323 context=conference
[2054] type=h323 context=canada
;[74678] ;type=h323 ;context=default
extensions.conf ----
[EMAIL PROTECTED] asterisk]# cat extensions.conf ; ; Static extension configuration file, used by ; the pbx_config module. This is where you configure all your ; inbound and outbound calls in Asterisk. ;
; ; The "General" category is for certain variables. ; [general] ; ; 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 ; static=yes ; ; if static=yes and writeprotect=no, you can save dialplan by ; CLI command 'save dialplan' too ; writeprotect=no
; 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 ${VARIABLE} or ${ENV(VARIABLE)} for Environmental variable ; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid ; [globals] CONSOLE=Console/dsp ; Console interface for demo ;CONSOLE=Zap/1 ;CONSOLE=Phone/phone0 IAXINFO=guest ; IAXtel username/password ;IAXINFO=myuser:mypass TRUNK=Zap/g2 ; Trunk interface TRUNKMSD=1 ; MSD digits to strip (usually 1 or ;TRUNK=IAX2/user:[EMAIL PROTECTED] DEFTIMEOUT=60
; ; 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) ; ; For example the extension _NXXXXXX would match normal 7 digit dialings, ; while _1NXXNXXXXXX would represent an area code plus phone number ; preceeded by a one. ; ; 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. ; ;[context] ;exten => someexten,priority,application(arg1,arg2,...) ;exten => someexten,priority,application,arg1|arg2... ; ; Timing list for includes is ; ; <time range>|<days of week>|<days of month>|<months> ; ;include => daytime|9:00-17:00|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. ;
; ; 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 www.gnophone.com or www.iaxtel.com ; [iaxtel700] exten => _91700XXXXXXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:[EMAIL PROTECTED])
; ; The SWITCH statement permits a server to share the dialplain with ; another server. Use with care: Reciprocal switch statements are not ; allowed (e.g. both A -> B and B -> A), and the switched server needs ; to be on-line or else dialing can be severly delayed. ; [iaxprovider] ;switch => IAX2/user:[EMAIL PROTECTED]/mycontext
[trunkint] ; ; International long distance through trunk ; exten => _9011.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) exten => _9011.,2,Congestion
[trunkld] ; ; Long distance context accessed through trunk ; exten => _91NXXNXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) exten => _91NXXNXXXXXX,2,Congestion
[trunklocal] ; ; Local seven-digit dialing accessed through trunk interface ;
[trunktollfree] ; ; Long distance context accessed through trunk interface ; exten => _91800NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) exten => _91800NXXXXXX,2,Congestion exten => _91888NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) exten => _91888NXXXXXX,2,Congestion exten => _91877NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) exten => _91877NXXXXXX,2,Congestion exten => _91866NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) exten => _91866NXXXXXX,2,Congestion
[international] ; ; Master context for international long distance ; ;ignorepat => 9 include => longdistance include => trunkint
[longdistance] ; ; Master context for long distance ; ;ignorepat => 9 include => local include => trunkld
[local] ; ; Master context for local, toll-free, and iaxtel calls only ; ;ignorepat => 9 ;include => default ;include => corvero ;include => parkedcalls ;include => trunklocal ;include => iaxtel700 ;include => trunktollfree ;include => iaxprovider ; ; 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:[EMAIL PROTECTED]/local
[macro-stdexten]; ; ; 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,Voicemail(u${ARG1}) ; If unavailable, send to voicemail w/ unavail announce exten => s,3,Goto(default,s,1) ; If they press #, return to start exten => s,102,Voicemail(b${ARG1}) ; If busy, send to voicemail w/ busy announce exten => s,103,Goto(default,s,1) ; If they press #, return to start exten => a,1,VoicemailMain(${ARG1}) ; If they press *, send the user into VoicemailMain
[demo] ; ; We start with what to do when a call first comes in. ; ;exten => s,1,Wait,1 ; Wait a second, just for fun ;exten => s,2,Answer ; Answer the line ;exten => s,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds ;exten => s,4,ResponseTimeout,10 ; Set Response Timeout to 10 seconds ;exten => s,5,BackGround(demo-congrats) ; Play a congratulatory message ;exten => s,6,BackGround(demo-instruct) ; Play some instructions
;exten => 2,1,BackGround(demo-moreinfo) ; Give some more information. ;exten => 2,2,Goto(s,6)
;exten => 3,1,SetLanguage(fr) ; Set language to french ;exten => 3,2,Goto(s,5) ; Start with the congratulations
;exten => 1000,1,Goto(default,s,1) ; ; We also create an example user, 1234, who is on the console and has ; voicemail, etc. ; ;exten => 1234,1,Playback(transfer,skip) ; "Please hold while..." ; (but skip if channel is not up) ;exten => 1234,2,Macro(stdexten,1234,${CONSOLE})
;exten => 1235,1,Voicemail(u1234) ; Right to voicemail
;exten => 1236,1,Dial(Console/dsp) ; Ring forever ;exten => 1236,2,Voicemail(u1234) ; Unless busy
; ; # for when they're done with the demo ; ;exten => #,1,Playback(demo-thanks) ; "Thanks for trying the demo" ;exten => #,2,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 => 500,2,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]) ; Call the Asterisk demo ;exten => 500,3,Playback(demo-nogo) ; Couldn't connect to the demo site ;exten => 500,4,Goto(s,6) ; Return to the start over message.
; ; Create an extension, 600, for evaulating echo latency. ; ;exten => 600,1,Playback(demo-echotest) ; Let them know what's going on ;exten => 600,2,Echo ; Do the echo test ;exten => 600,3,Playback(demo-echodone) ; Let them know it's over ;exten => 600,4,Goto(s,6) ; Start over
; ; Give voicemail at extension 14000 is the retrieval port ; ;exten => 14000,1,VoicemailMain ;exten => 14000,2,Goto(s,6)
; ; Here's what a phone entry would look like (IXJ for example) ; ;exten => 1265,1,Dial(Phone/phone0,15) ;exten => 1265,2,Goto(s,5)
[mainmenu] ; ; Example "main menu" context with submenu ; ;exten => s,1,Answer ;exten => s,2,Background(thanks) ; "Thanks for calling press 1 for sales, 2 for support, ..." ;exten => 1,1,Goto(submenu,s,1) ;exten => 2,1,Hangup ;include => default ; ;[submenu] ;exten => s,1,Ringing ; Make them comfortable with 2 seconds of ringback ;exten => s,2,Wait,2 ;exten => s,3,Background(submenuopts) ; "Thanks for calling the sales department. Press 1 for steve, 2 for..." ;exten => 1,1,Goto(default,steve,1) ;exten => 2,1,Goto(default,mark,2)
[default]
include => voicemail include => meetme include => sipphones include => inbound include => support include => conference include => canada include => outbound
exten => t,1,Background(pbx-transfer) exten => t,2,Dial(H323/4607,30) ; Send to Line Appearance on Main Reception exten => t,3,Hangup
exten => i,1,Background(pbx-invalid) exten => i,2,Dial(H323/4607,30) ; Send to Line Appearance on Main Reception exten => i,3,Hangup
[voicemail] ; ; this is for the Message Button and for general recall ; exten => 14000,1,NoOp(Message button ${CALLERIDNUM} pressed) exten => 14000,2,VoicemailMain(s${CALLERIDNUM})
; ; this is when the call is redirected to voice mail ; the problem is that we do not know what extension redirected the call. ; exten => 14001,1,NoOp(Voice Mail Capture for ${CALLERIDNUM}) exten => 14001,2,Voicemail(u${CALLERIDNUM}) ; If unavailable, send to voicemail w/ unavail announce
exten => 14002,1,NoOp(Voice Mail Capture for ${CALLERIDNUM}) exten => 14002,2,Voicemail(b${CALLERIDNUM}) ; If unavailable, send to voicemail w/ busy
[meetme]
; ; Or a conference room (you'll need to edit meetme.conf to enable this room) ; exten => 7000,1,Meetme(70000) exten => 7001,1,Meetme(70010) exten => 7002,1,Meetme(70020) exten => 7003,1,Meetme(70030) exten => 7004,1,Meetme(70040) exten => 7005,1,Meetme(70050) exten => 7006,1,Meetme(70060) exten => 7007,1,Meetme(70070) exten => 7008,1,Meetme(70080) exten => 7009,1,Meetme(70090)
[sipphones] exten => _7XXXX,1,NoOp("Call for "${EXTEN}) exten => _7XXXX,2,Dial(SIP/${EXTEN},60,tr) exten => _7XXXX,3,Congestion
;exten => 4XXX,1,NoOp("Call for "${EXTEN}) ;exten => 4XXX,2,Dial(H323/${EXTEN},60,tr) ;exten => 4XXX,3,Congestion
[outbound] exten => _4XXX,1,NoOp("Route to CCM1" ${EXTEN} ) exten => _4XXX,2,Dial(H323/${EXTEN}) exten => _4XXX,3,Congestion
exten => _5XXX,1,NoOp("Route to CCM1" ${EXTEN} ) exten => _5XXX,2,Dial(H323/${EXTEN}) exten => _5XXX,3,Congestion
exten => _9NXXXXXXXXX,1,NoOp("Route to CCM1" ${EXTEN}) exten => _9NXXXXXXXXX,2,Dial(H323/${EXTEN})
exten => _91NXXXXXXXXX,1,NoOp("Route to CCM1" ${EXTEN}) exten => _91NXXXXXXXXX,2,Dial(H323/${EXTEN})
[inbound] exten => s,1,Ringing ; Make them comfortable with 2 seconds of ringback exten => s,2,Wait,2 exten => s,3,Answer ; Answer the line exten => s,4,Wait,2 exten => s,5,DigitTimeout,5 ; Set Digit Timeout to 5 seconds exten => s,6,ResponseTimeout,10 ; Set Response Timeout to 10 seconds exten => s,7,BackGround(Rec-Main-1) ; Play intro message exten => s,8,BackGround(Rec-Main-3) ; Play intro message
exten => 0,1,Playback(pbx-transfer) exten => 0,2,Dial(H323/4607,30)
exten => 1,1,Playback(pbx-transfer) exten => 1,2,Goto(support,2051,1)
[support] exten => 2051,1,Ringing ; Make them comfortable with 2 seconds of ringback exten => 2051,2,Wait,2 exten => 2051,3,Answer ; Answer the line exten => 2051,4,Wait,2 exten => 2051,5,DigitTimeout,5 ; Set Digit Timeout to 5 seconds exten => 2051,6,ResponseTimeout,10 ; Set Response Timeout to 10 seconds exten => 2051,7,Playback(Rec_Supp_Ame_1) ; Play intro message exten => 2051,8,BackGround(Rec_Supp_Ame_3) ; Play intro message exten => 2051,9,BackGround(Rec_Supp_Ame_4) ; Play intro message
[conference] exten => 2052,1,Ringing ; Make them comfortable with 2 seconds of ringback exten => 2052,2,Wait,2 exten => 2052,3,Answer ; Answer the line exten => 2052,4,DigitTimeout,5 ; Set Digit Timeout to 5 seconds exten => 2052,5,ResponseTimeout,10 ; Set Response Timeout to 10 seconds exten => 2052,6,BackGround(conf-usermenu) ; Play intro message
[canada]
exten => 0,1,Background(pbx-transfer) exten => 0,2,Dial(H323/4608,30) ; Send to Line Appearance on Main Reception exten => 0,3,Hangup
exten => 1,1,Goto(support,2051,1)
exten => 2054,1,Ringing ; Make them comfortable with 2 seconds of ringback exten => 2054,2,Setvar(op=4608) exten => 2054,3,Wait,2 exten => 2054,4,Answer ; Answer the line exten => 2054,5,Wait,3 exten => 2054,6,Playback(Rec-Can-Main-1) exten => 2054,7,BackGround(Rec-Can-Main-3) exten => 2054,8,Dial(H323/${op},30) ; Send to Line Appearance on Main Reception exten => 2054,9,Hangup
exten => t,1,Background(pbx-transfer) exten => t,2,Dial(H323/${op},30) ; Send to Line Appearance on Main Reception exten => t,3,Hangup
exten => i,1,Background(pbx-invalid) exten => i,2, Goto(canada,2054,5)
callerid=9723814678 nat=yes disallow=all allow=ulaw context=default mailbox=4678 dtmfmode=inband
h323.conf ---
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