SPA3000 - تاریخچهٔ ویرایش‌ها

Behnaz در ‏۳ آوریل ۲۰۱۳، ساعت ۰۸:۳۱

2013-04-03T08:31:02Z

Behnaz در ‏۳ آوریل ۲۰۱۳، ساعت ۰۸:۲۳

2013-04-03T08:23:20Z

Behnaz در ‏۱۶ اکتبر ۲۰۱۲، ساعت ۱۱:۳۲

2012-10-16T11:32:49Z

Behnaz: صفحه‌ای جدید حاوی '
SPA3000 First login as admin and click "Advanced Settings" '''1- Sip settings:''' Just make sure the '''SIP Port'''...' ایجاد کرد

2012-10-16T11:22:00Z

صفحه‌ای جدید حاوی '<div dir=ltr lang=en> SPA3000 First login as admin and click "Advanced Settings" '''1- Sip settings:''' Just make sure the '''SIP Port'''...' ایجاد کرد

صفحهٔ جدید

<div dir=ltr lang=en>
SPA3000

First login as admin and click "Advanced Settings"

'''1- Sip settings:'''

Just make sure the '''SIP Port''' is set to port '''5061'''

'''2- Proxy and Registration:'''

'''Proxy''': The numeric IP address of your Asterisk box if both the Sipura and the Asterisk box are on the same local network, or the address of your Asterisk server if it is elsewhere on the Internet.
'''Make Call Without Reg: Yes'''
'''Ans Call Without Reg: Yes'''

'''Register Expires''': 300 (if Asterisk box is on same local network and you used host=dynamic in the PCC Trunk settings), or see discussion note below

'''Notes on Proxy and Registration section''': Some people have reported that they had to set Make Call Without Reg and Ans Call Without Reg to Yes before things would work - it apparently doesn't hurt anything to change those two settings, and it may save you some grief. Also, if you used host=dynamic in the FreePBX Trunk settings (as discussed above), then you will probably want to make the Register Expires: setting something fairly low, especially if the SPA-3000/3102 is on the same local network as the Asterisk box - for example, Register Expires: 300 would make the unit re-register at five minute intervals, while a setting of 900 would probably be a good choice if the device and the Asterisk Server are not on the same local network. The reason for the shortened timeout is that when you use host=dynamic in the PCC Trunk settings, if registration is lost for any reason (such as a server reboot) then the SPA-3000/3102 will be inaccessible until the next time it re-registers. This has led some people to conclude that host=dynamic doesn't work, when in fact it does but is just waiting for the adapter to re-register.

'''3- Subscriber Information:'''

'''Display name''': Put something here that will identify this line - this is only displayed on your phones if you get a call with no Caller ID information (or you don't subscribe to Caller ID). Keep it at 15 characters or shorter. You could use something like LOCAL PSTN CALL.
User ID: 1-pstn ; very important - this must exactly match the PCC Trunk name and username in the trunk configuration[[BR]] Password: XXXXXX (same as you used in PCC Trunk settings).

'''4- Audio Configuration:'''
'''
DTMF Process INFO: Yes
DTMF Process AVT: Yes
DTMF Tx Method: Auto'''

'''Note''': The DTMF Tx Method is the one you especially need to check if your IVR is not receiving DTMF from your callers reliably. Also, in this section you may want to check to make sure that the Preferred Codec is set to the default G711u (assuming you placed "allow=ulaw" in the PCC Trunk configuration as shown above).

'''5- Dial Plans:'''

Under Dial Plans it's important not to change the default (xx.) on any except Dial Plan 2. I put it very simple to go to my inbound so PCC takes care of my calls:

(S0<:1234567890>)

Replace 1234567890 with the telephone number of the PSTN line coming into the device. Note that this must exactly match the DID number in your PCC Inbound Route setting for this device. If the number here and in the Inbound Route don't match exactly, you won't receive incoming calls.

'''6- VoIP-To-PSTN Gateway Setup:'''

This is another important settings section.
'''
VoIP-To-PSTN Gateway Enable: yes
VoIP Caller Auth Method: None
VoIP PIN Max Retry: 3
One Stage Dialing: Yes ; very important
Line 1 VoIP Caller DP: none
VoIP Caller Default DP: none
Line 1 Fallback DP: none
'''

'''7- PSTN-To-VoIP Gateway Setup:'''

PSTN-To-VoIP Gateway Enable: Yes
PSTN Caller Auth Method: none
PSTN Ring Thru Line 1: no
PSTN Pin Max Retry: 3
PSTN CID for VoIP CID: Yes if you subscribe to CallerID service on your PSTN line, otherwise No
PSTN CID Number Prefix: (Leave Blank)
PSTN Caller Default DP: 2 ; important - here is where it sends the calls to.
Off Hook While Calling VoIP: No
Line 1 Signal Hook Flash To PSTN: Disabled
PSTN CID Name Prefix: (Leave Blank

Leave everything else in this section blank. We are almost finished now.

'''8- FXO Timer Values (sec):'''

Just change 2 items here.
'''''
Voip Answer Delay''''': ''0 (The original recommendation was 1, but this can cause a spurious half ring on outgoing calls, before actual ringing from the called line commences, so 0 is now the recommended value'').
PSTN Answer Delay: If you do not subscribe to CallerID service on your PSTN line, this can be set to 0. Most users will want to set it to at least 3 so that the incoming CallerID data is captured. In rare situations you may need a slightly longer delay (5 should be more than enough).

''''''You can find more information here''': http://www.freepbx.org/support/documentation/howtos/howto-linksys-spa-3102-sipura-spa-3000-freepbx'''

</div>

@@ سطر ۱۱: / سطر ۱۱: @@
 '''2- Proxy and Registration:'''
-'''Proxy''': The numeric IP address of your Asterisk box if both the Sipura and the Asterisk box are on the same local network, or the address of your Asterisk server if it is elsewhere on the Internet.
+ '''Proxy''': The numeric IP address of your Asterisk box if both the Sipura and the Asterisk box are on the same local network, or the
-'''Make Call Without Reg: Yes'''
+ your Asterisk server if it is elsewhere on the Internet address of .
-'''Ans Call Without Reg: Yes'''
+ '''Make Call Without Reg''': Yes
 '''Notes on Proxy and Registration section''': Some people have reported that they had to set Make Call Without Reg and Ans Call Without Reg to Yes before things would work - it apparently doesn't hurt anything to change those two settings, and it may save you some grief. Also, if you used host=dynamic in the FreePBX Trunk settings (as discussed above), then you will probably want to make the Register Expires: setting something fairly low, especially if the SPA-3000/3102 is on the same local network as the Asterisk box - for example, Register Expires: 300 would make the unit re-register at five minute intervals, while a setting of 900 would probably be a good choice if the device and the Asterisk Server are not on the same local network. The reason for the shortened timeout is that when you use host=dynamic in the PCC Trunk settings, if registration is lost for any reason (such as a server reboot) then the SPA-3000/3102 will be inaccessible until the next time it re-registers. This has led some people to conclude that host=dynamic doesn't work, when in fact it does but is just waiting for the adapter to re-register.

@@ سطر ۲۵: / سطر ۲۵: @@
 '''Display name''': Put something here that will identify this line - this is only displayed on your phones if you get a call with no Caller ID information (or you don't subscribe to Caller ID). Keep it at 15 characters or shorter. You could use something like LOCAL PSTN CALL.
 User ID: 1-pstn ; very important - this must exactly match the PCC Trunk name and username in the trunk configuration[[BR]] Password: XXXXXX (same as you used in PCC Trunk settings).
 '''4- Audio Configuration:'''
 '''
-DTMF Process INFO: Yes
+ '''DTMF Process INFO''': Yes
-DTMF Process AVT: Yes
+ '''DTMF Process AVT''': Yes
-DTMF Tx Method: Auto'''
+ '''DTMF Tx Method''': Auto
 '''Note''': The DTMF Tx Method is the one you especially need to check if your IVR is not receiving DTMF from your callers reliably. Also, in this section you may want to check to make sure that the Preferred Codec is set to the default G711u (assuming you placed "allow=ulaw" in the PCC Trunk configuration as shown above).
@@ سطر ۴۸: / سطر ۴۹: @@
 This is another important settings section.
 '''
-VoIP-To-PSTN Gateway Enable: yes
+ '''VoIP-To-PSTN Gateway Enable''': yes
-VoIP Caller Auth Method: None
+ '''VoIP Caller Auth Method''': None
-VoIP PIN Max Retry: 3
+ '''VoIP PIN Max Retry''': 3
-One Stage Dialing: Yes ; very important
+ '''One Stage Dialing''': Yes ; very important
-Line 1 VoIP Caller DP: none
+ '''Line 1 VoIP Caller DP''': none
-VoIP Caller Default DP: none
+ '''VoIP Caller Default DP''': none
-Line 1 Fallback DP: none
+ '''Line 1 Fallback DP''': none
 '''
 '''7- PSTN-To-VoIP Gateway Setup:'''
-PSTN-To-VoIP Gateway Enable: Yes
+ '''PSTN-To-VoIP Gateway Enable''': Yes
-PSTN Caller Auth Method: none
+ '''PSTN Caller Auth Method''': none
-PSTN Ring Thru Line 1: no
+ '''PSTN Ring Thru Line 1''': no
-PSTN Pin Max Retry: 3
+ '''PSTN Pin Max Retry''': 3
-PSTN CID for VoIP CID: Yes if you subscribe to CallerID service on your PSTN line, otherwise No
+ '''PSTN CID for VoIP CID''': Yes if you subscribe to CallerID service on your PSTN line, otherwise No
-  PSTN CID Number Prefix: (Leave Blank)
+  '''PSTN CID Number Prefix''': (Leave Blank)
-PSTN Caller Default DP: 2 ; important - here is where it sends the calls to.
+ '''PSTN Caller Default DP: 2''' ; important - here is where it sends the calls to.
-Off Hook While Calling VoIP: No
+ '''Off Hook While Calling VoIP''': No
-Line 1 Signal Hook Flash To PSTN: Disabled
+ '''Line 1 Signal Hook Flash To PSTN''': Disabled
-PSTN CID Name Prefix: (Leave Blank
+ '''PSTN CID Name Prefix''': (Leave Blank
 Leave everything else in this section blank. We are almost finished now.
@@ سطر ۷۸: / سطر ۷۸: @@
 Just change 2 items here.
-'''''
-Voip Answer Delay''''': ''0 (The original recommendation was 1, but this can cause a spurious half ring on outgoing calls, before actual ringing from the called line commences, so 0 is now the recommended value'').
+'''Voip Answer Delay''':0 (The original recommendation was 1, but this can cause a spurious half ring on outgoing calls, before actual ringing from the called line commences, so 0 is now the recommended value'').
 PSTN Answer Delay: If you do not subscribe to CallerID service on your PSTN line, this can be set to 0. Most users will want to set it to at least 3 so that the incoming CallerID data is captured. In rare situations you may need a slightly longer delay (5 should be more than enough).

@@ سطر ۱: / سطر ۱: @@
 <div dir=ltr lang=en>
-SPA3000
+:::::::::::::<big><big><big>'''SPA3000'''</big></big></big>
 '''1- Sip settings:'''

SPA3000 - تاریخچهٔ ویرایش‌ها

Behnaz در ‏۳ آوریل ۲۰۱۳، ساعت ۰۸:۳۱

Behnaz در ‏۳ آوریل ۲۰۱۳، ساعت ۰۸:۲۳

Behnaz در ‏۱۶ اکتبر ۲۰۱۲، ساعت ۱۱:۳۲

Behnaz: صفحه‌ای جدید حاوی ' SPA3000 First login as admin and click "Advanced Settings" '''1- Sip settings:''' Just make sure the '''SIP Port'''...' ایجاد کرد

Behnaz: صفحه‌ای جدید حاوی '
SPA3000 First login as admin and click "Advanced Settings" '''1- Sip settings:''' Just make sure the '''SIP Port'''...' ایجاد کرد