[VoiceOps] Metaswitch Loopback

[Greg Stone]

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?

[cid:7ddb059d-b6f0-4242-9b24-1c2ccb583118]
Greg Stone
Senior Voice Network Engineer
Race Communications
E : greg.stone at race.com
P : 415-376-3306
Web : Visit Race.com<https://nam04.safelinks.protection.outlook.com/?url=https%3A%2F%2Fwww.race.com%2F&data=04%7C01%7Callracestaff%40race.com%7C0110f233f4ee42d03e8508d9489bf465%7Cc4f7941052bf4601be443afdc7670fe9%7C1%7C0%7C637620656923156401%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwMDAiLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C1000&sdata=YlAIwyP0%2B3z6wavTyMEzOtIg1GK%2BZwwjUzjntvDK9gI%3D&reserved=0>

________________________________
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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