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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

SPA3102_HTTPD.png

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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

2_stage_dialing.png


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

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

PSTN_dial_in.png



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

PSTN_dial_in_2.png

PSTN_dial_in_2.png (5.84 KB)

PSTN_dial_in_2.png

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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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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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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.

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回復 14# ckleea


   
Oh! really!  Thats good.  Could you share the settings in OBi110?

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