[SME]

I've been struggling with getting * working with my spa3102 and now I'm just spinning my wheels, so to speak. I've tried different examples from dozens of websites and I even tried the Voxilla SPA3000 Configuration Wizard for Asterisk but still no real results and now I'm not sure what's needed in my configs and what's not needed nor what's wrong.

I managed to get the fxs port going (I think) but the FXO (I think) still wont register and nothing seems to work, except that I get dialtone from my analog phone plugged into the ata. Dialing anything from the ata just gets me a pause, then a busy signal. Calling in via PSTN or sip (using FWD) rings the fwd client on another box but not the * server nor the ata.

Here are my .conf files...
{sip.conf}
[general]
disallow=all
allow=ulaw
allow=alaw
allow=gsm
context=home
maxexpirey=180
defaultexpirey=160
tos=reliability
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
externip = 72.25.124.194
localnet = 10.4.0.0/255.255.0.0
register => myfwd#:myfwdpass@fwd.pulver.com/3000
[fwd-outgoing]
type=friend
allow=ulaw
secret=shhh
username=myfwd#
host=fwd.pulver.com
insecure=very ; needed if we want to allow incoming FWD calls to bypass authentication
[phone01] ; softphone on 10.1.2.3
type=friend
host=dynamic
defaultip=10.1.2.3
username=anyname
secret=shhhh
dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
mailbox=1000 ; Mailbox for message waiting indicator
context=home
callerid="Me" <2124>

[3000]
type=friend
host=dynamic
context=home
secret=shhhhh
mailbox=3000
dtmfmode=rfc2833
disallow=all
allow=ulaw
[4000]
; If you're using Asterisk, this goes into the Incoming settings
; For your Trunk
type=friend
host=dynamic
; If using Asterisk@home, change the below line to context=from-internal
context=home
secret=shhhhhh
dtmfmode=rfc2833
disallow=all
allow=ulaw
insecure=very
[pstn-spa3k]
; If you're using Asterisk, this section goes into the Outgoing Settings
; for your trunk.
type=peer
auth=md5
host=10.1.100.102
port=5061 ; I cant find 5061 listed anywhere but that's what the wizard offered
secret=shhhhhhhhh
username=asterisk
fromuser=asterisk
dtmfmode=rfc2833
; If using Asterisk@home, change the below line to context=from-internal
context=home
insecure=very

*** NOTE: config files continue in next post due to character limits. ***

I'm just completely lost now. I know it shouldn't be THIS hard to configure a spa3102 with * but now I'm completely lost. I think the fxs port is [3000] in sip.conf and I think the fxo port is [pstn-spa3k] but I have no idea what [4000] is nor do I understand dialplans, voicemail or the various extensions. Will somebody, please, translate/help?

Merry Christmas...
SME

[SME]

{voicemail.conf}
[general]
; Default formats for writing Voicemail
;format=g723sf|wav49|wav
format=wav49|gsm|wav
;
; Who the e-mail notification should appear to come from
serveremail=asterisk
;serveremail=asterisk@linux-support.net
; Should the email contain the voicemail as an attachment
attach=yes
; Maximum number of messages per folder. If not specified, a default value
; (100) is used. Maximum value for this option is 9999.
;maxmsg=100
; Maximum length of a voicemail message in seconds
;maxmessage=180
; Minimum length of a voicemail message in seconds for the message to be kept
; The default is no minimum.
;minmessage=3
; Maximum length of greetings in seconds
;maxgreet=60
; How many milliseconds to skip forward/back when rew/ff in message playback
skipms=3000
; How many seconds of silence before we end the recording
maxsilence=10
; Silence threshold (what we consider silence: the lower, the more sensitive)
silencethreshold=128
; Max number of failed login attempts
maxlogins=3
[default]
3000 => 3000,MYCID,me@mine.com,tz=pacific
FWDUSERID=myfwd#
FWDUSERNAME=First Last.
PHONE1=anyextension
PHONE1VM=Voicemail-of-that-extension
FWDEXTEN=1000
;[phone01]
;exten => _X.,1,Answer
;exten => _X.,2,Wait(2)
;exten => _X.,3,Playback(tt-monkeys)
;exten => _X.,4,Hangup

;[fwd-out]
;exten => _7.,1,SetCIDNum(${FWDUSERID})
;exten => _7.,2,SetCIDName(${FWDUSERNAME})
;exten => _7.,3,Dial(SIP/${EXTEN:1}@fwd-outgoing)
;exten => _7.,4,Playback(invalid)
;exten => _7.,5,Hangup
;
;[from-sip]
;exten => ${FWDEXTEN},1,Dial(${PHONE1},30)
;exten => ${FWDEXTEN},2,Voicemail(u${PHONE1VM})
;exten => ${FWDEXTEN},3,Hangup
;exten => ${FWDEXTEN},102,Voicemail(b${PHONE1VM})
;exten => ${FWDEXTEN},103,Hangup
;
;[other]
;pbx => 2468,My Name,me@mine.com

*** NOTE: extensions.conf follows in next post due to character limits ***

I also have some debug output from the CLI:
-- Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from 10.1.2.3

Dialing my cell from the ata give teh following output but nothing happens.

[top] Spawn extension (home, 3000, 3) exited non-zero on 'SIP/3000-086ed000'
-- Executing Dial("SIP/3000-086dd000", "SIP/MYCELL#@pstn-spa3k|60|") in new stack
-- Called MYCELL#@pstn-spa3k

Spawn extension (home, MYCELL#, 1) exited non-zero on 'SIP/3000-086dd000'

BTW, dialing "3000" from the ata will allow me to leave VM.

...continued...

[SME]

{extensions.conf}
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no
[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 0)
;TRUNK=IAX2/userass@provider
[dundi-e164-canonical]
;
; List canonical entries here
;
;exten => 12564286000,1,Macro(std-exten,6000,IAX2/foo)
;exten => _125642860XX,1,Dial(IAX2/otherbox/${EXTEN:7})
[dundi-e164-customers]
;
; If you are an ITSP or Reseller, list your customers here.
;
;exten => _12564286000,1,Dial(SIP/customer1)
;exten => _12564286001,1,Dial(IAX2/customer2)
[dundi-e164-via-pstn]
;
; If you are freely delivering calls to the PSTN, list them here
;
;exten => _1256428XXXX,1,Dial(Zap/g2/${EXTEN:7}) ; Expose all of 256-428
;exten => _1256325XXXX,1,Dial(Zap/g2/${EXTEN:7}) ; Ditto for 256-325
[dundi-e164-local]
;
; Context to put your dundi IAX2 or SIP user in for
; full access
;
include => dundi-e164-canonical
include => dundi-e164-customers
include => dundi-e164-via-pstn
[dundi-e164-switch]
;
; Just a wrapper for the switch
;
switch => DUNDi/e164
[dundi-e164-lookup]
;
; Locally to lookup, try looking for a local E.164 solution
; then try DUNDi if we don't have one.
;
include => dundi-e164-local
include => dundi-e164-switch
;
; DUNDi can also be implemented as a Macro instead of using
; the Local channel driver.
;
[macro-dundi-e164]
;
; ARG1 is the extension to Dial
;
exten => s,1,Goto(${ARG1},1)
include => dundi-e164-lookup
;
; Here are the entries you need to participate in the IAXTEL
; call routing system. Most IAXTEL numbers begin with 1-700, but
; there are exceptions. For more information, and to sign
; up, please go to GnoPhone, by linux support services or IAXTel.com | Connecting Asterisk Users Since abs(-1872)
;
[iaxtel700]
exten => _91700XXXXXXX,1,Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel)
;
; The SWITCH statement permits a server to share the dialplan with
; another server. Use with care: Reciprocal switch statements are not
; allowed (e.g. both A -> B and B -> A), and the switched server needs
; to be on-line or else dialing can be severly delayed.
;
[iaxprovider]
;switch => IAX2/user:[key]@myserver/mycontext
[trunkint]
;
; International long distance through trunk
;
exten => _9011.,1,Macro(dundi-e164,${EXTEN:4})
exten => _9011.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunkld]
;
; Long distance context accessed through trunk
;
exten => _91NXXNXXXXXX,1,Macro(dundi-e164,${EXTEN:1})
exten => _91NXXNXXXXXX,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunklocal]
;
; Local seven-digit dialing accessed through trunk interface
;
exten => _9NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunktollfree]
;
; Long distance context accessed through trunk interface
;
exten => _91800NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91888NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91877NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91866NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[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 => parkedcalls
include => trunklocal
include => iaxtel700
include => trunktollfree
include => iaxprovider
;
[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,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten => s-NOANSWER,1,Voicemail(u${ARG1}) ; If unavailable, send to voicemail w/ unavail announce
exten => s-NOANSWER,2,Goto(default,s,1) ; If they press #, return to start
exten => s-BUSY,1,Voicemail(b${ARG1}) ; If busy, send to voicemail w/ busy announce
exten => s-BUSY,2,Goto(default,s,1) ; If they press #, return to start
exten => _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no answer
exten => a,1,VoicemailMain(${ARG1}) ; If they press *, send the user into VoicemailMain
[macro-stdPrivacyexten];
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
; ${ARG3} - Optional DONTCALL context name to jump to (assumes the s,1 extension-priority)
; ${ARG4} - Optional TORTURE context name to jump to (assumes the s,1 extension-priority)`
;
exten => s,1,Dial(${ARG2},20|p) ; Ring the interface, 20 seconds maximum, call screening option (or use P for databased call screening)
exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten => s-NOANSWER,1,Voicemail(u${ARG1}) ; If unavailable, send to voicemail w/ unavail announce
exten => s-NOANSWER,2,Goto(default,s,1) ; If they press #, return to start
exten => s-BUSY,1,Voicemail(b${ARG1}) ; If busy, send to voicemail w/ busy announce
exten => s-BUSY,2,Goto(default,s,1) ; If they press #, return to start
exten => s-DONTCALL,1,Goto(${ARG3},s,1) ; Callee chose to send this call to a polite "Don't call again" script.
exten => s-TORTURE,1,Goto(${ARG4},s,1) ; Callee chose to send this call to a telemarketer torture script.
exten => _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no answer
exten => a,1,VoicemailMain(${ARG1}) ; If they press *, send the user into VoicemailMain
[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,n,Answer ; Answer the line
exten => s,n,Set(TIMEOUT(digit)=5) ; Set Digit Timeout to 5 seconds
exten => s,n,Set(TIMEOUT(response)=10) ; Set Response Timeout to 10 seconds
exten => s,n(restart),BackGround(demo-congrats) ; Play a congratulatory message
exten => s,n(instruct),BackGround(demo-instruct) ; Play some instructions
exten => s,n,WaitExten ; Wait for an extension to be dialed.
exten => 2,1,BackGround(demo-moreinfo) ; Give some more information.
exten => 2,n,Goto(s,instruct)
exten => 3,1,Set(LANGUAGE()=fr) ; Set language to french
exten => 3,n,Goto(s,restart) ; Start with the congratulations
exten => 1000,1,Goto(default,s,1)
;
; We also create an example user, 1234, who is on the console and has
; voicemail, etc.
;
exten => 1234,1,Playback(transfer,skip) ; "Please hold while..."
; (but skip if channel is not up)
exten => 1234,n,Macro(stdexten,1234,${CONSOLE})
exten => 1235,1,Voicemail(u1234) ; Right to voicemail
exten => 1236,1,Dial(Console/dsp) ; Ring forever
exten => 1236,n,Voicemail(u1234) ; Unless busy
;
; # for when they're done with the demo
;
exten => #,1,Playback(demo-thanks) ; "Thanks for trying the demo"
exten => #,n,Hangup ; Hang them up.
;
; A timeout and "invalid extension rule"
;
exten => t,1,Goto(#,1) ; If they take too long, give up
exten => i,1,Playback(invalid) ; "That's not valid, try again"
;
; Create an extension, 500, for dialing the
; Asterisk demo.
;
exten => 500,1,Playback(demo-abouttotry); Let them know what's going on
exten => 500,n,Dial(IAX2/guest@misery.digium.com/s@default) ; Call the Asterisk demo
exten => 500,n,Playback(demo-nogo) ; Couldn't connect to the demo site
exten => 500,n,Goto(s,6) ; Return to the start over message.
;
; Create an extension, 600, for evaluating echo latency.
;
exten => 600,1,Playback(demo-echotest) ; Let them know what's going on
exten => 600,n,Echo ; Do the echo test
exten => 600,n,Playback(demo-echodone) ; Let them know it's over
exten => 600,n,Goto(s,6) ; Start over
;
; Give voicemail at extension 8500
;
exten => 8500,1,VoicemailMain
exten => 8500,n,Goto(s,6)

[default]
;
; By default we include the demo. In a production system, you
; probably don't want to have the demo there.
;
include => demo
[home]
exten => 3000,1,Ringing
exten => 3000,2,Dial(SIP/3000,20,T)
exten => 3000,3,Voicemail(u3000)
exten => 3000,4,Hangup
exten => 911,1,Dial(SIP/911@pstn-spa3k,60,)
exten => 911,2,Congestion
exten => _XXXXXXX,1,Dial(SIP/${EXTEN}@pstn-spa3k,60,)
exten => _XXXXXXX,2,Congestion
exten => _1800XXXXXXX,1,Dial(SIP/${EXTEN}@pstn-spa3k,60,)
exten => _1800XXXXXXX,2,Congestion
exten => _1888XXXXXXX,1,Dial(SIP/${EXTEN}@pstn-spa3k,60,)
exten => _1888XXXXXXX,2,Congestion
exten => _1877XXXXXXX,1,Dial(SIP/${EXTEN}@pstn-spa3k,60,)
exten => _1877XXXXXXX,2,Congestion
exten => _1866XXXXXXX,1,Dial(SIP/${EXTEN}@pstn-spa3k,60,)
exten => _1866XXXXXXX,2,Congestion

Last file...

Any help?

[SME]

OK, OK, let's not have everyone helping at once...

I managed to fix *part* of the problem by changing the extension used in the PSTN dialplan to one that works. Now, when the PSTN line rings, * rings the analog phone and even puts it in VM but dialing out from the ata results in silence and no ring on the other end. So inbound PSTN works but outbound PSTN is still broke. Also, how do I retrieve my VM in *?

Thanks again...

SME

[rvtango]

I have no problem with outbound ... did you follow the guide?
VM retrieval on analog phone is done by punching *98 or *97
Just make sure you have right setting for DTMF on analog phone registration

[SME]

Originally Posted by rvtango

I have no problem with outbound ... did you follow the guide?
VM retrieval on analog phone is done by punching *98 or *97
Just make sure you have right setting for DTMF on analog phone registration

Yes, I followed the instructions on the wizard and the various incomplete tutorials that I've found but I'm still having outbound issues. I fixed the FXO inbound issue by correcting the PSTN dialplan which WAS routed to a non-existant extension so I edited the dialplan (to ext 3000) instead of creating a new extension (2000), which may or may not have been the correct way to fix it but it works better, now, than it did.

*98/97 didn't work, I just get a busy signal after pressing either. My DTMF Tx Method for line1 and PSTN are both set to "auto" and Mode is set to "strict," is that correct?

Also, based on my pasted configs, I think there are too many superfluous entries but I don't know what's NOT needed and what is. Obviously anything commented out is not needed.

[rvtango]

As for incoming: it was clearly stated in a guide that you need to change SPA dialplan to point to existing extension.

VM issues: post the Line1 dialplan in a thread (you can find it on the bottom of the Line1 config page). Most likely you have default DP which won't work with Trixbox)

[SME]

Originally Posted by rvtango

As for incoming: it was clearly stated in a guide that you need to change SPA dialplan to point to existing extension.

VM issues: post the Line1 dialplan in a thread (you can find it on the bottom of the Line1 config page). Most likely you have default DP which won't work with Trixbox)

I must have missed that part.

([2-79]11<:@gw0>|xx.|*xx.|**xx.|<#,:>xx.<:@gw0>|<#,:>*xx< :@gw0>) but it's * not trixbox.

[rvtango]

look at Linksys SPA Dial Plan Guide to begin with ...

[SME]

I started off reading various tutorials but had no luck, then I used the "wizard" to configure my SPA and give me added entries in my configs. I have read your guide, that's how I found out the wizard gave me the wrong ext for the PSTN dialplan. Everything else basically looked the same to me. Have I missed someting else?