[carloscampos]

How a can config my Asterisk server and SPa3102 when they are separate by a NAt?
I have my Asterisk server in USA and My spa3201 in México under nat.
local call --> spa3102 --> *server --> VoIP Providers work fine but calls
Asterisk server --> spa3102 -->Localcalls not work.

[hbustillo]

Hello. Did yoiu ever get the sipura configuration to send the right caller id, the one from the outside caller?

I have the same problem, the only caller id I get displayed is the extension name and number configured in the sipura pstn conf.

Your help will be greatly appreciated.

Hector

[IanWorthington]

I've written an article covering configuration of the SPA3000 and trixbox, with particular regard for UK users, at:

Ian Worthington - SPA3000 configuration for trixbox

ian
...

[pcman800]

hello phoneboy;

could you teach me how to configure the VoIP phone and the AXON vitural server. I am new about this. and also about the dial plan. i have no idea about this. I have here a SPA3000, AXON Virtual Server (installed in my pc) and Express talk (installed also).

thanks.



erwin of Philippines (pcman800@yahoo.com)

[pcman800]

hello phoneboy

how to get and editthis things. sip.conf and extensions.conf.?

i see this:

In sip.conf:

[1274]
type=friend
host=dynamic
context=home
secret=1274
mailbox=1274
dtmfmode=rfc2833
disallow=all
allow=ulaw
[802]
; 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=802
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=192.168.1.6
port=5061
secret=123
username=asterisk
fromuser=asterisk
dtmfmode=rfc2833
; If using Asterisk@home, change the below line to context=from-internal
context=home
insecure=very
In voicemail.conf:

[default]
1274 => 1274,Erwin,user.com,user-pager.com,tz=pacific
In extensions.conf (assume calls dialed as 7 digits or 800 calls go out the PSTN line as well as 911 calls):

[home]
exten => 1274,1,Ringing
exten => 1274,2,Dial(SIP/1274,20,T)
exten => 1274,3,Voicemail(u1274)
exten => 1274,4,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

i dont know about this, im in the Philippines

i want to make a call to hongkong or in US. plss help me



erwin of Philippines

[rolando]

Hello. I am Newbie on asterisk.. any idea how to set a SPA3102 on asteriskNOW i got the line 1 talk to x-pro(softphone) but I can not get outbount call go through PSTN LINE.. I also could frwd incoming calls to the autoattendant of asterisk.. but it doesnt release the line after the first call ends.. (the caller hung up) the 2 call goes to VM of Telco..

I ´d appreciate any suggestion

Originally Posted by PhoneBoy

99 must be defined in sip.conf as well.

[TWDM]

Wizard works, added voicepulse.com quickstart files to asterisk. Can get hard line in and out, can get voicepulse in, need help with dial plan (or gateway setting?) for voicepulse out.

Thanks for any help

[MisterQ]

This looks useful, as I have had a devil of a time getting my SPA3K working.


But... I am running PBX in a flash, which of course, runs Freepbx, which keeps all trunk and extension etc details in a DB (as well as the config files) As such it would be better to plug in the values in via the web forms rather than editing sip.conf and extensions.conf.

Has anyone been through that exercise ?? In the interim, I am trying to "reverse engineer" this, based on adding bits via the Various Freepbx options, to see what files they end up placing things in...

Peter