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" <[email protected]> To: "Markus" <[email protected]>, "Mike Hammett" <[email protected]> Cc: [email protected] 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 : [email protected] P : 415-376-3306 Web : Visit Race.com From: VoiceOps <[email protected]> on behalf of Mike Hammett via VoiceOps <[email protected]> Sent: Tuesday, November 8, 2022 8:23 AM To: Markus <[email protected]> Cc: [email protected] <[email protected]> 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" <[email protected]> To: [email protected] 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@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 _______________________________________________ VoiceOps mailing list [email protected] https://puck.nether.net/mailman/listinfo/voiceops
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