vicidial.org

GoAutoDial CE 2.0 | Vicidial VERSION: 2.2.1-237 BUILD: 100510-2015 | Asterisk 1.4.27.1-vici | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation

I've got everything working except SIP registration.

From the asterisk CLI whenever I attempt to make a call:

Refreshing DNS lookups.
-- Executing [1863XXXXXXX@default:1] Dial("SIP/9000-00000004", "SIP/1863XXXXXXX@XXXXXXXX") in new stack
-- Called 1863XXXXXXX@XXXXXXXX
-- Got SIP response 604 "Does not exist anywhere" back from XX.XX.XX.XXX
== Everyone is busy/congested at this time (1:0/0/1)
== Auto fallthrough, channel 'SIP/9000-00000004' status is 'CHANUNAVAIL'
-- Executing [h@default:1] DeadAGI("SIP/9000-00000004", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----1-----CHANUNAVAIL----------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0


After speaking to my provider about this here is what I've received from the provider and would appreciate any help anyone can give:

This is what they see on their end when I try to register:

I still see your contact as an "s". Don't remember what setting in
Asterisk fixes this.

URI: sip:s@96.254.XXX.XXX Expiration: Fri Oct 15 16:31:45 EDT
2010

My registration string:

register=>xxxx:xxxx@mars.pbx-change.com

Account entry:

[pbxchange]
type=friend
secret=xxxxxxxxxx
username=xxxxxxxxxx
host=mars.pbx-change.com
dtmfmode=rfc2833
context=inbound
canreinvite=no
allow=ulaw
allow=g729
insecure=port,invite
fromdomain=mars.pbx-change.com
fromuser=xxxxxxxxxx

Dial plan:

exten => _1NXXXXXXXXX,1,Dial(SIP/${EXTEN}@pbxchange)
exten => _011.,1,Dial(SIP/${EXTEN:3}@pbxchange)

Server IP is set correctly as well. There is no NAT, server has dedicated IP on net.

Thanks again in advance.

Edit: Forgot to add that this server actually gets moved around a lot hence the need for using registration.

Are you actually well registred?

sip show registry
sip show users
sip show peers

Is your inbound already working?

Have you tried to do:

exten => _1NXXXXXXXXX,1,Dial(SIP/pbxchange/${EXTEN}) instead?

Have your tried to make separete account for type user and peer instead of using friend?

I hope someone else can REALLY help you, cheers!

Yep, sip show xxx shows good to go.

Tried your dialplan still the same error.

Have your tried to make separate account for type user and peer instead of using friend?

Yeah just tried peer user and friend, no luck same error Thanks for the suggestions though.

Edit:

Inbound is working:

[Nov 5 14:04:51] NOTICE[4336]: chan_sip.c:15147 handle_request_invite: Call from '727xxxxxxx' to extension '727xxxxxxx' rejected because extension not found.

But I have nothing setup for it so that's expected.

Output of:

sip show registry

[pbxchange]
type=friend
secret=xxxxxxxxxx
username=xxxxxxxxxx
host=mars.pbx-change.com
dtmfmode=rfc2833
context=inbound
canreinvite=no
allow=ulaw
allow=g729
insecure=port,invite
fromdomain=mars.pbx-change.com
fromuser=xxxxxxxxxx

Do you have an inbound context somewhere in your dialplan?

I've got everything working except SIP registration.

please post output from asterisk cli when you enter:

Code: Select all

sip show registry

exten => _1NXXXXXXXXX,1,Dial(SIP/${EXTEN}@pbxchange)
exten => _011.,1,Dial(SIP/${EXTEN:3}@pbxchange)

Neither of these is a valid vicidial dialplan entry.

Try this ...

Globals String:

Code: Select all

DIAL9TRUNK = SIP/pbxchange

Dialplan Entry:

Code: Select all

exten =>_91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten =>_91NXXNXXXXXX,2,Dial(${DIAL9TRUNK}/${EXTEN:1},,tTor)
exten =>_91NXXNXXXXXX,3,Hangup
exten =>_9011XXXXXXXX.,1,AGI(agi://127.0.0.1:4577/call_log)
exten =>_9011XXXXXXXX.,2,Dial(${DIAL9TRUNK}/${EXTEN:4},,tTor)
exten =>_9011XXXXXXXX.,3,Hangup

*The above assumes the NXXNXXXXXX US dialplan is appropriate for your area (10 digits, neither 1st nor 4th can be 0 or 1). Change that portion of the dialplan to fit your locale.
*note that "EXTEN:4" will strip of 9011 and send the rest of the number to your provider. This may or may not fit their expected dial string for international numbers. If you post their expectations, we can adjust it to fit.

Campaign Dial Prefix:

Code: Select all

9

Sip Show Registry:

Host Username Refresh State Reg.Time
mars.pbx-change.com:5060 xxxxxxxxxx 104 Registered Mon, 08 Nov 2010 11:12:27


I implemented the dial plan you suggested and I am still getting the Got SIP response 604 "Does not exist anywhere" back message in the CLI when trying to make an outbound call with a softPhone.

Also, I said when I register but I actually meant when I attempt to make an outbound call. Sorry for the confusion.

Can you show us again from your CLI the Dial process with the dialplan entry that williamconley suggested you?

We have to see what are you actually sending to your carrier.

PlexXoniC wrote:I am still getting the Got SIP response 604 "Does not exist anywhere" back message in the CLI

you should post that line of output and the line above it (or two) which presumably cause it.

Here you go and thanks again.

-- Executing [1863676xxxx@default:1] Dial("SIP/9000-0000000e", "SIP/pbxchange/1863676xxxx") in new stack
-- Called pbxchange/1863676xxxx
-- Got SIP response 604 "Does not exist anywhere" back from 63.135.224.130
== Everyone is busy/congested at this time (1:0/0/1)
== Auto fallthrough, channel 'SIP/9000-0000000e' status is 'CHANUNAVAIL'
-- Executing [h@default:1] DeadAGI("SIP/9000-0000000e", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----1-----CHANUNAVAIL----------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0

This seem to me like what you are sending does not match with any entry on your carrier´s dialplan, thats why they are sending the call to the s extension. You should ask them if they need a special prefix.

Currently you are sending 1863 676 xxxx wich is correct, but maybe your VoIP carrier doesnt need the 1, for example one of my providers set a special prefix for each customer for security reasons if a send the wrong prefix they always response: ¨ Everyone is busy/congested at this time¨.

You'll have to try SIP debug or more likely your provider. That is the response they are sending to you. Could be they don't like your caller ID or any other technicality regarding your call. But the call IS going through to your carrier.