SPA3000/3102 integration with asterisk

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2^#

發表於 2011-6-7 10:08 | 只看該作者

我都有如ckleea兄的問題。同求！
不過我是用Elastix 2.0.3版

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3^#

發表於 2011-6-7 16:20 | 只看該作者

HTTP Digest Settings in Asterisk and SPA3102

Asterisk Side

sip.conf
[spa3khttpd]
context = default
type = peer
auth = httpd_name:httpd_secret@SPA3102_IP_Add_or_DDNS:5061 ; md5 authentication with SPA3102 http digest
nat = yes
dtmfmode = rfc2833
host = 192.168.1.123 ; use DDNS or true IP when outside LAN
defaultuser = httpd_name
fromuser = httpd_name
fromdomain = 192.168.1.123 ; use DDNS or true IP when outside LAN
secret = httpd_secret
insecure = port,invite
qualify = yes
canreinvite = no
port = 5061
disallow = all
allow = ulaw,alaw,gsm,g729

extensions.conf
[viaSPA3khttpd]
exten => _135.,1,Dial(SIP/${EXTEN:3}@spa3khttpd,60,tr) ; using remote SPA3102 PSTN GW in HTTP Digest
exten => _135.,n,Playback(the-party-you-are-calling&is-curntly-unavail)
exten => _135.,n,Congestion

SPA3102 Side

SPA3102 PSTN Settings

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4^#

發表於 2011-6-7 16:57 | 只看該作者

本帖最後由 bubblestar 於 2011-6-7 17:29 編輯

Two Stage Dialing via remote SPA3102 from Asterisk

Asterisk Side

[globals]
SPA3KPSTN_GW = 192.168.11.123:5061 ; for SPA3102 PSTN GW. Use DDNS or true IP when outside LAN

extensions.conf
[viaSPA3KPSTN_GW]
exten => s,1,Dial(SIP/${EXTEN}@${SPA3KPSTN_GW}) ; using remote SPA3102 PSTN GW

SPA3102 Side

Create an extension, say 6008 to be registered in SPA3102 PSTN Line

Action: Dial extension 6008# and you will hear the 2nd stage dial tone from SPA3102, then dial the called party via PSTN as you wish. Thats it.

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5^#

發表於 2011-6-7 17:07 | 只看該作者

PSTN dials in directly into asterisk (extensions, AA, or IVR)

It is a very simple and 2 steps configuration in SPA3102 PSTN Line

PSTN Caller Default DP:8 (which corresponding to the extension 6001 in Dial Plan 8 above)

PSTN_dial_in_2.png (5.84 KB)

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6^#

發表於 2011-6-7 17:37 | 只看該作者

回復 3# bubblestar

I use similar setting but not using md5. Do you encrypt the whole into md5?

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7^#

發表於 2011-6-7 18:01 | 只看該作者

回復 5# bubblestar

Can you make both phone and asterisk extension ring at the same time? I understand it can ring after some adjustable delays

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8^#

發表於 2011-6-7 18:03 | 只看該作者

Any one has tried to pass fax in through SPA3000/3102 into asterisk?

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9^#

發表於 2011-6-7 19:40 | 只看該作者

回復 6# ckleea

The md5 method was set up long time ago. I just remember that when I did not use md5, the http digest was not function in my case. Hence, I resolve this by adding md5 as well.

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10^#

發表於 2011-6-7 19:44 | 只看該作者

回復 7# ckleea

As far as I know, you can adjust the VoIP Answer Delay to: 5 or longer to make the phone rings when you dial. After 5 seconds, you are given a 2nd dial tone for further dialing.

In my case, I just make the VoIP Answer Delay to Zero(0) without ringing.

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11^#

發表於 2011-6-7 19:50 | 只看該作者

Just test to confirm that http digest one stage dialing and 2nd stage gateway dialing are mutually exclusive. You can only use either one method instead of both.