Flowroute, Asterisk and Zoiper Communicator

Asterisk Asterisk General
1

Flowroute provides my external connection for Asterisk. The flowroute folks say it is successfully connected. And I have the Zoiper Communicator softphone sitting on my desktop successfully registered with the Asterisk box. The Asterisk box is still a boat anchor since I still have not been able to connect two “telephones” together which I understand is the primary Asterisk purpose. Correct?

So, what have I done wrong? I used the Elastix distro. (Quite frankly I am disappointed in Elastix since it does not report anything useful in this problem resolution task.) What other Asterisk configuration parameters do I need to make this connection happen? For Zoiper and Flowroute, here are my two set of Asterisk parameters:

Zoiper Communicator softphone

placed in sip_general_custom.conf

[sip.zoiper2]
username=sip.zoiper2
secret=my_pass234
context=zoiper_tests
type=friend
host=dynamic
callerid=8765431

Placed in voicemail.conf

501 => 4242,Example Mailbox,myemail@server.com
502 => 7474,Example Mailbox 2,myemail2@server.com

Placed in extensions_custom.conf

[zoiper_tests]

exten => 1001,1,Dial(SIP/sip.Zoiper2,20)
exten => 1001,2,VoiceMail(501@default)
exten => 1001,n,HangUp()

exten => 111001,1,Answer()
exten => 111001,2,VoiceMailMain(501@default)
exten => 111001,n,HangUp()

Flowroute connection

Added the following to the sip_general_custom.conf
allow=ulaw
allow=g729

register => userNumber:Password@sip.flowroute.com

Add the following to sip_custom_post.conf
[flowroute] ;keep this lowercase, do not change format
type=friend
secret=Password
username=UserNumber
host=sip.flowroute.com
dtmfmode=rfc2833
context=inbound ;change to ‘ext-did’ or ‘from-trunk’ for asterisk@home
canreinvite=no
allow=ulaw
allow=g729
insecure=port,invite
fromdomain=sip.flowroute.com

Added the following to extensions_custom.conf

[outgoing]
exten => _1NXXXXXXXXX,1,Dial(SIP/${EXTEN}@flowroute)
;Send NANPA (USA) as 11 digit
exten => _011.,1,Dial(SIP/${EXTEN:3}@flowroute)
;dialing format - SIP/{countrycode}{number}@flowroute

Thanks in advance for all your help.

2

Asterisk connects two telephones for the vast majority of users, suggesting it cannot won’t win you friends.

You need to start by providing verbose level 3 or more output from the CLI. If that is not obvious, we would want a SIP protocol trace.

Do you have enough G.729 licences?

type=friend and insecure=,port are generally bad security, but won’t stop things working.

You might get more responses on the Asterisk Support forum, as this is as support question.

On re-reading, I see a …custom file, so you actually want the support forum for the Asterisk GUI that you are using.

3

Thanks David,

I guess my attempt at humor directed at Asterisk is a hot-button in the Asterisk community? I see great benefits in using Asterisk. It just seems to be extremely easy to provide configuration parameters and very difficult to provide them correctly.

I will post to the support forum since I make the configuration changes directly on the Linux box.