[VoiceOps] Metaswitch Loopback

[Mike Hammett]

That seems to work in testing. 


Call goes out the tandem trunk and hits the remote system with the right CID. 




----- 
Mike Hammett 
Intelligent Computing Solutions 
http://www.ics-il.com 



Midwest Internet Exchange 
http://www.midwest-ix.com 



----- Original Message -----

From: "Greg Stone" <Greg.Stone at race.com> 
To: "Markus" <universe at truemetal.org>, "Mike Hammett" <voiceops at ics-il.net> 
Cc: voiceops at voiceops.org 
Sent: Tuesday, November 8, 2022 10:25:55 AM 
Subject: Re: Metaswitch Loopback 


What if you were to build a subscriber with a call forward unconditional that the number routes to, then you can put the toll free in as the called number in the UCON Forward? 






	
	Greg Stone 
Senior Voice Network Engineer 
Race Communications 
E : greg.stone at race.com 
P : 415-376-3306 
Web : Visit Race.com 



From: VoiceOps <voiceops-bounces at voiceops.org> on behalf of Mike Hammett via VoiceOps <voiceops at voiceops.org> 
Sent: Tuesday, November 8, 2022 8:23 AM 
To: Markus <universe at truemetal.org> 
Cc: voiceops at voiceops.org <voiceops at voiceops.org> 
Subject: Re: [VoiceOps] Metaswitch Loopback 



CAUTION: This email originated from outside of the organization. Do not click links or open attachments unless you recognize the sender and know the content is safe. 

I do mean called. 

It's for 911. If the SIP trunks fail, I'm supposed to route it over TDM to the toll-free number. 




----- 
Mike Hammett 
Intelligent Computing Solutions 
http://www.ics-il.com 



Midwest Internet Exchange 
http://www.midwest-ix.com 





From: "Markus via VoiceOps" <voiceops at voiceops.org> 
To: voiceops at voiceops.org 
Sent: Tuesday, November 8, 2022 10:18:29 AM 
Subject: Re: [VoiceOps] Metaswitch Loopback 

Am 08.11.2022 um 16:38 schrieb Mike Hammett via VoiceOps: 
> I'm working a situation where I need to rewrite my called number to a 
> toll-free number. Because the rewriting happens after Metaswitch does 
> the toll-free lookup, the tandem rejects the call as there's no dip. 

Did you really mean called number or rather calling number? If you can 
hook a Asterisk box in between the device where your customers' SIP 
calls are coming from and Metaswitch you could rewrite either. 

Overwrite any calls' CLI to calling number 18009999999 and send it out 
to "metaswitch01" as defined in sip.conf: 

/etc/asterisk/extensions.conf: 

[incoming-calls-from-customers] 

exten => _X.,1,NoOp 
exten => _X.,n,Set(CALLERID(name)=18009999999) 
exten => _X.,n,Set(CALLERID(num)=18009999999) 
exten => _X.,n,Dial(SIP/${EXTEN}@metaswitch01) 
exten => _X.,n,Hangup 

- or - Overwrite any called number and send the call to 18007777777 to 
"metaswitch01": 

exten => _X.,1,NoOp 
exten => _X.,n,Dial(SIP/18007777777 at metaswitch01) 
exten => _X.,n,Hangup 

(old Asterisk, before pjsip, but not much different) 

Sample for sip.conf: 

[metaswitch01] 
type=peer 
host=sip.metaswitch.something 
username=maybe-username-or-leave-empty 
secret=maybe-password-or-leave-empty 
disallow=all 
allow=alaw 
allow=ulaw 
canreinvite=no 
dtmfmode=rfc2833 
context=nowhere 

[my-internal-pbx-or-sbc] 
type=peer 
host=10.10.10.10 
insecure=port,invite 
disallow=all 
allow=alaw 
allow=ulaw 
canreinvite=no 
dtmfmode=rfc2833 
context=incoming-calls-from-customers 

Good luck 
Markus 
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