Inbound SIP Trunk - Doesn't call Extensions

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robkerry
robkerry's picture
Inbound SIP Trunk - Doesn't call Extensions

Hi,

Any help people can offer would be much appreciated.

I have a Trixbox and FreePBX set-up in our office with 9 phones and a 10 external phone numbers (the 10th being a catch-all group for the main number). The phones are successfully logged into the system and can make outgoing calls, however incoming calls are not getting routed properly. We're using a single SIP trunk for the inbound numbers.

Calling the numbers does show up on the "Panel" for the Incoming SIP Trunk although the extension does not ring or show as busy and the caller hears "The number you have dialled is not in service".

Our Extension numbers are 1-9.

I think I may have just missed something in the configuration, could you confirm what settings I need to use in order for this to work?

Many Thanks,

Rob

danswartz
danswartz's picture
it's impossible to say

without knowing what you're trying to do. e.g. which numbers are supposed to get routed to which extensions, or... what? and without seeing your config, even less likely...

rosic
rosic's picture
I have this identical issue.

I have this identical issue. I have only one incoming DID and it is supposed to ring all extensions. I'm using
freedigits for the DID. Per the instructions on their website I have set up a SIP and IAX2 trunk. When the number
it shows as active in the system status, but the caller gets "this number is not in service."
When I look at the trunks, I see a "warning: this trunk is not used in any routes!"
When I look at inboudn routes, I don't see a place to assign a trunk.
Any help anybody can give me with this would be greatly appreciated.
Regards,
Roland

rosic
rosic's picture
Thought I would just update

Thought I would just update this. I was playing with it and changed the settings in PEER Details on my inbound trunks
what was recommended in another thread. Set up my inbound route with the DID telling it to go to extension 220.
Now I get: "the person at extension (did number) is busy. then it beeps for a voice mail." I'm almost 100% certain this
is coming from freedigits.com and not my server. If I look at the call log, it doesn't show this call as incoming.
I think the PEER Details I'm using are bogus.
They are:
allow=all
context=from-trunk
host=freedigits.net
secret=xxx
type=friend
username=5156786975

rubgade
rubgade's picture
SAme issue here

I have Grand central Incoming and it rings the trunk and after about 10 secs drops off. I have configured a DID and Incoming route. Is there a way to configure the Trunk with a Incoming route. What is the Context=from-trunk ?

I did not see any config file for - from-trunk

Here is my configration :

Extension Options :

Direct DID = MY_DID_Number

and Trunk for Incomming settings.

allow=ulaw&ilbc&gsm
canreinvite=no
context=from-trunk
disallow=all
dtmfmode=rfc2833
host=proxy01.sipphone.com
insecure=very
secret=MYSECRET
type=user
username=MUSIPNUMBER

I see that the incoming on the TRunk but it does not get routed to any extension and gets disconnected

rosic
rosic's picture
I finally got mine

I finally got mine working.

My peer details look like:
type=friend
secret=xxx
qualify=yes
nat=yes
insecure=very
host=freedigits.net
fromuser=NxxNxxxxxx
fromdomain=proxy.freedigits.net
dtmfmode=rfc2833
defaultip=proxy.freedigits.net
context=from-pstn
canreinvite=no

and my user details look like:
username=NxxNxxxxxx
type=user
secret=xxxx
nat=yes
insecure=very
host=freedigits.net
context=from-pstn

user-context is NxxNxxxxxxx

and resgistration string:
NxxNxxxxx:secret@freedigits.net:5060

However, the only way I could get this configuration to work is to change my incoming call route to DID any / CID any and have it ring a ring group.

I don't know why you would have your contexts set to from-trunk instead of from-pstn.

netcitizen
netcitizen's picture
thanks

thanks for the post. I spent the last 2 days trying to get my trixbox to accept calls from a DID. Ur config worked perfectly

carlos0330
carlos0330's picture
Can't route calls to desire a2billing accountcode

Hi!

I have a problem routing a call from freedigits to my accountcode 10000000. I sitll don't know what I am missing here. I have a trixbox platform, can anybody help out here? Thank you!

This is what I have configured;

SIP.CONF

[freedigits]
username=13194413188
type=peer
secret=secret
insecure=very
host=67.55.159.156
fromuser=13194413188
fromdomain=freedigits.net
dtmfmode=info
disallow=all
context=a2billing
canreinvite=no
authuser=13194413188
allow=g729
allow=g723
allow=ulaw
allow=alaw

EXTENSION.CONF

[a2billing]
exten => 13194413188,1,Goto(a2billing-did,${EXTEN},1)
exten => 15672527299,1,Goto(a2billing-did,${EXTEN},1)
exten => _X.,1,Answer
exten => _X.,n,Wait(0.25)
exten => _X.,n,DeadAGI(a2billing.php)
exten => _X.,n,Wait(.25)
exten => _X.,n,Hangup
exten => s,1,Hangup
exten => s,n,Hangup
exten => i,1,Hangup

[a2billing-did]
exten => _X.,1,Noop(a2billing-did ${EXTEN})
exten => _X.,n,DeadAGI(a2billing.php)
exten => _X.,n,Hangup

[a2billing-callback]
exten => _X.,1,Noop(a2billing-callback ${EXTEN})
exten => _X.,n,Answer
exten => _X.,n,DeadAGI(a2billing.php)
exten => _X.,n,Hangup

[a2billing-cid-callback]
exten => _X.,1,Noop(a2billing-cid-callback ${EXTEN})
exten => _X.,n,Ringing
exten => _X.,n,DeadAGI(a2billing.php)
exten => _X.,n,Wait(2)
exten => _X.,n,Hangup

[a2billing-all-callback]
exten => _X.,1,Noop(a2billing-all-callback ${EXTEN})
exten => _X.,n,Ringing
exten => _X.,n,DeadAGI(a2billing.php)
exten => _X.,n,Wait(2)
exten => _X.,n,Hangup

[a2billing-voucher]
exten => _X.,1,Noop(a2billing-voucher ${EXTEN})
exten => _X.,n,DeadAGI(a2billing.php)
exten => _X.,n,Hangup

[callingcard]
exten => _X.,1,Noop(callingcard ${EXTEN})
exten => _X.,n,DeadAGI(a2billing.php)
exten => _X.,n,Hangup

SIPTARWEB_SIPFRIEND.CONF

[10000000]
type=friend
username=10000000
secret=secret
accountcode=10000000
regexten=10000000
callerid=10000000
amaflags=billing
qualify=no
nat=yes
dtmfmode=RFC2833
qualify=yes
canreinvite=no
host=dynamic
context=callingcard
regseconds=0
cancallforward=yes

This is what I get in cli

a2billing.php:
a2billing.php: A2Billing AGI internal configuration:
a2billing.php: Array
a2billing.php: (
a2billing.php: [debug] => 2
a2billing.php: [answer_call] =>
a2billing.php: [logger_enable] => 1
a2billing.php: [log_file] => /tmp/a2billing.log
a2billing.php: [setlanguage_deprecate] => 1
a2billing.php: [say_goodbye] =>
a2billing.php: [play_menulanguage] =>
a2billing.php: [force_language] => EN
a2billing.php: [intro_prompt] =>
a2billing.php: [len_cardnumber] => 8
a2billing.php: [len_voucher] => 8
a2billing.php: [min_credit_2call] => 0
a2billing.php: [use_dnid] => 1
a2billing.php: [no_auth_dnid] => Array
a2billing.php: (
a2billing.php: [0] => 2400
a2billing.php: [1] => 2300
a2billing.php: )
a2billing.php:
a2billing.php: [number_try] => 1
a2billing.php: [say_balance_after_auth] =>
a2billing.php: [say_balance_after_call] =>
a2billing.php: [say_timetocall] =>
a2billing.php: [cid_enable] =>
a2billing.php: [cid_askpincode_ifnot_callerid] =>
a2billing.php: [cid_auto_create_card] =>
a2billing.php: [cid_auto_create_card_typepaid] => PREPAY
a2billing.php: [cid_auto_create_card_credit] => 0
a2billing.php: [cid_auto_create_card_credit_limit] => 1000
a2billing.php: [cid_auto_create_card_tariffgroup] => 6
a2billing.php: [sip_iax_friends] =>
a2billing.php: [sip_iax_pstn_direct_call_prefix] => 00
a2billing.php: [sip_iax_pstn_direct_call] =>
a2billing.php: [dialcommand_param] => |60|HL(%timeout%:61000:45000)
a2billing.php: [dialcommand_param_sipiax_friend] => |60|HL(1200000:61000:45000)
a2billing.php: [switchdialcommand] =>
a2billing.php: [record_call] =>
a2billing.php: [monitor_formatfile] => gsm
a2billing.php: [base_currency] => usd
a2billing.php: [agi_force_currency] =>
a2billing.php: [currency_association] => Array
a2billing.php: (
a2billing.php: [0] => usd:prepaid-dollar
a2billing.php: [1] => mxn:pesos
a2billing.php: [2] => eur:euro
a2billing.php: [3] => all:credit
a2billing.php: )
a2billing.php:
a2billing.php: [file_conf_enter_destination] => prepaid-enter-dest
a2billing.php: [file_conf_enter_menulang] => prepaid-menulang2
a2billing.php: [debugshell] => 0
a2billing.php: [currency_association_internal] => Array
a2billing.php: (
a2billing.php: [usd] => prepaid-dollar
a2billing.php: [mxn] => pesos
a2billing.php: [eur] => euro
a2billing.php: [all] => credit
a2billing.php: )
a2billing.php:
a2billing.php: )
a2billing.php:
a2billing.php: AGI Request:
a2billing.php: Array
a2billing.php: (
a2billing.php: [agi_request] => a2billing.php
a2billing.php: [agi_channel] => SIP/13194413188-08329b60
a2billing.php: [agi_language] => en
a2billing.php: [agi_type] => SIP
a2billing.php: [agi_uniqueid] => 1226175496.0
a2billing.php: [agi_callerid] => FreeDigits
a2billing.php: [agi_calleridname] => anonymous
a2billing.php: [agi_callingpres] => 0
a2billing.php: [agi_callingani2] => 0
a2billing.php: [agi_callington] => 0
a2billing.php: [agi_callingtns] => 0
a2billing.php: [agi_dnid] => 13194413188
a2billing.php: [agi_rdnis] => unknown
a2billing.php: [agi_context] => callingcard
a2billing.php: [agi_extension] => 13194413188
a2billing.php: [agi_priority] => 1
a2billing.php: [agi_enhanced] => 0.0
a2billing.php: [agi_accountcode] =>
a2billing.php: )
a2billing.php:
a2billing.php: FreeDigits ; SIP/13194413188-08329b60 ; 1226175496.0 ; ; 13194413188
a2billing.php: FORCE LANGUAGE : EN
a2billing.php: Requesting DTMF ::> Len-8
FOP Server Started
a2billing.php: RES DTMF : -1
a2billing.php: CARDNUMBER ::> -1
a2billing.php: SELECT credit, tariff, activated, inuse, simultaccess, typepaid, creditlimit, language, removeinterprefix, redial, enableexpire, UNIX_TIMESTAMP(expirationdate), expiredays, nbused, UNIX_TIMESTAMP(firstusedate), UNIX_TIMESTAMP(cc_card.creationdate), cc_card.currency FROM cc_card LEFT JOIN cc_tariffgroup ON tariff=cc_tariffgroup.id WHERE username='-1'
a2billing.php: 0
a2billing.php: PREPAID-AUTH-FAIL

fskrotzki
fskrotzki's picture
carlos0330, Please do not

carlos0330, Please do not hijack a thread about one thing with another.

This is clearly a different topic and in respect to the original author who is probably getting copies of updates to his question please start a new forum topic and post your question there. You'll also have a better shoot at getting the answers you are looking for then under the subject of "Inbound SIP Trunk - Doesn't call Extensions".