09-22-2009 04:02 PM - edited 03-21-2019 09:19 AM
Hi All,
I have used Patrick's helpful doument here
https://www.myciscocommunity.com/docs/DOC-7654
To get inbound calls from the PSTN to be routed to my Asterisk PBX.
My question is how do I go the other way and route calls from Asterisk to the SPA8800 and out an FXO port to the PSTN?
Thanks
Roland
Solved! Go to Solution.
09-23-2009 03:41 PM
Hi Roland,
I'm glad that you found the document useful. I appreciate the feedback. Take a look at the Asterisk phone config documents if you want to deploy SPA phones with your Asterisk server. We now have guides on using the web-ui to configure SPA phones with Asterisk and automatic provisioning, aka zero touch configuration of SPA phones with Asterisk. Available from here: https://www.myciscocommunity.com/docs/DOC-10647
Here's how to call from IP phone > Asterisk > SPA8800 FXO > PSTN
I'll refer to https://www.myciscocommunity.com/docs/DOC-7654 in this explanation:
This section defines lines 2 and 3 of the SPA8800. I know that they are lines 2 and 3 by looking at the port number, 5060= line 1, ... 5063= line 4.
Page 7 shows a snippet of /etc/asterisk/sip.conf where the FXO ports (lines) are declared:
;define SPA8800 pstn2 user
[pstn2]
type=friend
host=192.168.2.237 ;IP address of the SPA8800
port=5161 ;5161 is the default SIP port for line 2 on the SPA8800
dtmfmode=rfc2833
context=pstn2
insecure=very
;
;define SPA8800 pstn3 user
[pstn3]
type=friend
host=192.168.2.237 ;IP address of the SPA8800
port=5261 ;5261 is the default SIP port for line 2 on the SPA8800
dtmfmode=rfc2833
context=pstn3
insecure=very
;
This section uses two SPA8800 FXO ports (lines 2 and 3) and trunks them together. This allows the trunk to be shared among many users. The trunk is oversubscribed, so there can never be more than 2 simultaneous calls...
Page 8 shows a snippet of /etc/asterisk/extensions.conf where outbound call routing is defined:
;
; dial 7 to explicitly use FXO3
exten => _7.,1,Dial(SIP/${EXTEN:1}@pstn3,60,r)
;
; dial 8 as a steering digit:
; if FXO2 is not available, FXO3 will be used.
; if FXO3 is not available, the user hears congestion
exten => _8.,1,Dial(SIP/${EXTEN:1}@pstn2,60,r)
exten => _8.,2,Dial(SIP/${EXTEN:1}@pstn3,60,r)
;
Page 15 in the Configuring FXO Line Ports on the SPA8800configures SPA8800 FXO ports 2 and 3 to point to the Asterisk server:
Once you've saved the changes and reloaded the SIP module on the Asterisk server, you should be able to make outbound calls by using either 7 or 8 as a steering digit where 7 will only use line 3 and 8 will first try line 2 and then line 3 if line 2 is not available.
Does this make sense to you?
Regards,
Patrick
-----------
09-23-2009 03:41 PM
Hi Roland,
I'm glad that you found the document useful. I appreciate the feedback. Take a look at the Asterisk phone config documents if you want to deploy SPA phones with your Asterisk server. We now have guides on using the web-ui to configure SPA phones with Asterisk and automatic provisioning, aka zero touch configuration of SPA phones with Asterisk. Available from here: https://www.myciscocommunity.com/docs/DOC-10647
Here's how to call from IP phone > Asterisk > SPA8800 FXO > PSTN
I'll refer to https://www.myciscocommunity.com/docs/DOC-7654 in this explanation:
This section defines lines 2 and 3 of the SPA8800. I know that they are lines 2 and 3 by looking at the port number, 5060= line 1, ... 5063= line 4.
Page 7 shows a snippet of /etc/asterisk/sip.conf where the FXO ports (lines) are declared:
;define SPA8800 pstn2 user
[pstn2]
type=friend
host=192.168.2.237 ;IP address of the SPA8800
port=5161 ;5161 is the default SIP port for line 2 on the SPA8800
dtmfmode=rfc2833
context=pstn2
insecure=very
;
;define SPA8800 pstn3 user
[pstn3]
type=friend
host=192.168.2.237 ;IP address of the SPA8800
port=5261 ;5261 is the default SIP port for line 2 on the SPA8800
dtmfmode=rfc2833
context=pstn3
insecure=very
;
This section uses two SPA8800 FXO ports (lines 2 and 3) and trunks them together. This allows the trunk to be shared among many users. The trunk is oversubscribed, so there can never be more than 2 simultaneous calls...
Page 8 shows a snippet of /etc/asterisk/extensions.conf where outbound call routing is defined:
;
; dial 7 to explicitly use FXO3
exten => _7.,1,Dial(SIP/${EXTEN:1}@pstn3,60,r)
;
; dial 8 as a steering digit:
; if FXO2 is not available, FXO3 will be used.
; if FXO3 is not available, the user hears congestion
exten => _8.,1,Dial(SIP/${EXTEN:1}@pstn2,60,r)
exten => _8.,2,Dial(SIP/${EXTEN:1}@pstn3,60,r)
;
Page 15 in the Configuring FXO Line Ports on the SPA8800configures SPA8800 FXO ports 2 and 3 to point to the Asterisk server:
Once you've saved the changes and reloaded the SIP module on the Asterisk server, you should be able to make outbound calls by using either 7 or 8 as a steering digit where 7 will only use line 3 and 8 will first try line 2 and then line 3 if line 2 is not available.
Does this make sense to you?
Regards,
Patrick
-----------
10-01-2009 12:29 PM
Hi Patrick !
We are about to buy an SPA8800 and I'd like to ask you about its Line Ports for outbound calls.
I don't want to have to dial the steering digit (8 in you example) to make an outbound call.
I want SPA8800 to decide for a free FXO Line Port to make it, via round robin or something like that.
So, is it possible to have the following for outbound calls ?
In my sip.conf:
[192.168.2.237]
type=friend
host=192.168.2.237 ;IP address of the SPA8800
port=5060
dtmfmode=rfc2833
context=default
insecure=very
; -------------------------------
And in extensions.conf:
exten => XXXXXXXX,1,Dial(SIP/${EXTEN}@192.168.2.237,60,r)
; -------------------------------
Regards,
André Lemos
10-01-2009 04:57 PM
Hi André
The only reason that steering digit is used is because Asterisk is acting as a PBX and may have many different outbound routes and it wants you to choose.
For example, just by using the SPA8800, outbound routes could be SIP or PSTN-based.
Each SPA8800 port is explicitly registered to the Asterisk server at ports 5060-5063.
I bet that there are Asterisk experts that can offer a suggestion as to how you can make an outbound call without select a route and automatically route the call out of any one of 4 possible routes.
Not much help, I'm afraid.
Regards,
Patrick
-----------
10-01-2009 05:36 PM
Hi André,
You can just drop the steering digit '_8' and leave the '_.', this will match on any number. Then setup 4 outbound routes one after the other with the port definition on each one lining up with the port declared on each of the FXO lines on the SPA8800.
In theory, when the call to the first FXO port fails, it will try the next then the next and so on until you run out of ports to try.
Roland
10-02-2009 07:14 AM
Roland and Patrick,
thank you for the answers.
Another question: and what if I dial directly to the SPA8800 address like above?
exten => XXXXXXXX,1,Dial(SIP/${EXTEN}@192.168.2.237,60,r)
Have you ever tested this ?
Andre
10-02-2009 03:33 PM
Hi Andre,
I've not tested this.
I'll run the test for you next week. What are you hoping will happen?
Regards,
Patrick
-----------
10-05-2009 05:20 AM
Hi!
I hope SPA8800 makes the outgoing call to XXXXXXXX, choosing the most apropriated free port line.
André
10-02-2009 03:31 PM
Hi Roland,
Thanks for contributing to the community. Your expertise is most appreciated.
Regards,
Patrick
-----------
Discover and save your favorite ideas. Come back to expert answers, step-by-step guides, recent topics, and more.
New here? Get started with these tips. How to use Community New member guide