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. Metaswitch support had me route the call to a SIP loopback trunk, but that also fails because I don't accept transit calls. I saw the warning in the Metaswitch KB about enabling transit calls, so that doesn't excite me. Are there other ways of skinning this cat, aside from rewriting to a non-toll-free number? Why I'm hesitant to do that is the goal here is to dump calls out a TDM provider instead of a SIP provider. These are priority calls and I want a backup to IP-based solutions. We don't have agreements with any TDM termination providers, so that path doesn't seem feasible either. ----- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com Midwest Internet Exchange http://www.midwest-ix.com
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/${EXTEN}@metaswitch01) exten => _X.,n,Hangup - or - Overwrite any called number and send the call to 18007777777 to "metaswitch01": exten => _X.,1,NoOp exten => _X.,n,Dial(SIP/18007777777 at metaswitch01) exten => _X.,n,Hangup (old Asterisk, before pjsip, but not much different) Sample for sip.conf: [metaswitch01] type=peer host=sip.metaswitch.something username=maybe-username-or-leave-empty secret=maybe-password-or-leave-empty disallow=all allow=alaw allow=ulaw canreinvite=no dtmfmode=rfc2833 context=nowhere [my-internal-pbx-or-sbc] type=peer host=10.10.10.10 insecure=port,invite disallow=all allow=alaw allow=ulaw canreinvite=no dtmfmode=rfc2833 context=incoming-calls-from-customers Good luck Markus
I do mean called. It's for 911. If the SIP trunks fail, I'm supposed to route it over TDM to the toll-free number. ----- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com Midwest Internet Exchange http://www.midwest-ix.com ----- Original Message ----- From: "Markus via VoiceOps" <voiceops at voiceops.org> To: voiceops at voiceops.org Sent: Tuesday, November 8, 2022 10:18:29 AM Subject: Re: [VoiceOps] Metaswitch Loopback Am 08.11.2022 um 16:38 schrieb Mike Hammett via VoiceOps:
I'm working a situation where I need to rewrite my called number to a toll-free number. Because the rewriting happens after Metaswitch does the toll-free lookup, the tandem rejects the call as there's no dip.
Did you really mean called number or rather calling number? If you can hook a Asterisk box in between the device where your customers' SIP calls are coming from and Metaswitch you could rewrite either. Overwrite any calls' CLI to calling number 18009999999 and send it out to "metaswitch01" as defined in sip.conf: /etc/asterisk/extensions.conf: [incoming-calls-from-customers] exten => _X.,1,NoOp exten => _X.,n,Set(CALLERID(name)=18009999999) exten => _X.,n,Set(CALLERID(num)=18009999999) exten => _X.,n,Dial(SIP/${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 _______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
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.c...> ________________________________ 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 _______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
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 _______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
participants (3)
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Greg.Stone＠race.com -
universe＠truemetal.org -
voiceops＠ics-il.net