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Porting Cooling Off Period
by voiceops＠ics-il.net Sept. 22, 2020

Sept. 22, 2020

Is it normal for carriers to have a cooling off period where they don't accept further port requests for an account until x days after the last porting activity completes? ----- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com Midwest Internet Exchange http://www.midwest-ix.com

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Got my first inbound call with a STIR/SHAKEN checkmark
by compuwizz＠gmail.com Sept. 10, 2020

Sept. 10, 2020

I got my first validated call today from a Sprint subscriber to a T-Mobile subscriber. Presents with a tiny check mark in the Missed Calls tab on my iPhone. Unfortunately the originating party misdialed my number. Tested later on T-Mobile to T-Mobile and got a checkmark in the missed calls list but not on the screen during the call notification. Maybe due to the number being in the Contacts list already. ~Jared

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More Question about routing
by David.Zilk＠cdk.com Sept. 9, 2020

Sept. 9, 2020

Having come into telephony from the data networking end, rather than from a PSTN telephony background, where would one find a good basic to detailed tutorial of how routing of VoIP calls into the PSTN and vice versa works? I have so far only been involved with routing from our SIP platform to and from wholesale service providers, and have not yet had to manage our own numbers and peering. With STIR/SHAKEN we may need to get into that, and I need to get up to speed. Thanks, David From: VoiceOps [mailto:voiceops-bounces at voiceops.org] On Behalf Of Richard Jobson Sent: Tuesday, September 8, 2020 12:44 PM To: Glen Gerhard <glen at cognexus.net>; voiceops at voiceops.org Subject: Re: [VoiceOps] Question about SS7 routing Hi Ross So in your original email you were wondering about the role of MTP, the layer 3 protocol. This keeps tabs on the point codes. If you are troubleshooting down to the SS7 messages (MSU?s), SLTM?s & SLTA?s tell you what point codes those links are reaching. The Global Title Translation uses the SCCP/SS7 protocol when connecting to IXC. Local Number Portability uses AIN/TCAP to dip the database to determine the LRN. But many SS7 operations just troubleshoot this by looking at the ISUP protocol where the original calling Party number (CGN) appears in the Generic Address Part GAP and the LRN in the called party number (CDN) field. Cheers Richard From: VoiceOps <voiceops-bounces at voiceops.org<mailto:voiceops-bounces at voiceops.org>> on behalf of Glen Gerhard <glen at cognexus.net<mailto:glen at cognexus.net>> Date: Tuesday, September 8, 2020 at 11:47 AM To: <voiceops at voiceops.org<mailto:voiceops at voiceops.org>> Subject: Re: [VoiceOps] Question about SS7 routing Hi Ross, Unless you have an SS7 trunk to an ILEC you don't need to worry much about the Point Code. For SIP traffic you just dip the call and route on the LRN. The Point Code itself is a special format that is assigned to you when you set up your SS7 capable switch. Unless you have one of them you never need to worry about it. ======== ANSI Point Codes ANSI point codes are made up of three groups of digits called the network indicator (NI), network cluster (NC), and network cluster member (NCM). The values for ANSI point codes depends on the value of the pctype parameter of the chg-sid command, either ansi or other. If the pctype parameter is set to ansi, the ANSI rules for the ANSI point code are used to define the point code. The range of values for an ANSI point code with the pctype=ansi parameter are: NI ? 001-255 NC ? 001-255 (if ni = 001-005) or 000-255, * (if ni = 006-255) NCM ? 000-255, * The pctype=other parameter specifies that the ANSI point codes do not meet ANSI standards. The range of values for ANSI point codes with the pctype=other parameter are: NI ? 000-255 NC ? 000-255, * NCM ? 000-255, * The asterisk (*) point code value indicates a single cluster address for a cluster point code (for example, 20-2-*) or a network routing destination (21-*-*). for more information on cluster point codes, see the Cluster Routing and Management Diversity (CRMD) section. For more information on network routing point codes, see the Network Routing section. A double asterisk (**) and triple asterisk (***) can also be used for the NC and NCM fields of the ANSI point code, but for only the rtrv-dstn, rept-stat-dstn, rtrv-rte, and rept-stat-rte commands. A double asterisk in the NCM field of a point code (for example, 20-2-**) produces a summary report that shows all point code destinations or routes residing in the given cluster (20-2). This does not include the cluster point code, if the cluster point code (for example, 20-2-*) is provisioned. The following examples (rtrv-dstn and rtrv-rte) are reports generated using two asterisks in the NCM field of a point code. ======= ~Glen On 9/3/2020 10:55 AM, Mary Lou Carey wrote: I'll try to make this as short and sweet as possible even though it's pretty complicated. Point Codes are the 10 digit addresses for a particular switch and LRNs are the 10 digit addresses for a particular connection point that switch is associated with. In the PSTN world, all connections are dedicated and implemented by LATA / Tandem area for Local / IntraLATA traffic. When you get your first NPA-NXX for a LATA / tandem area, you enter it in the LERG (National Routing Database) and populate the tandems (Local, IntraLATA and FGD) that you are connecting to. Then you assign a 10 digit phone number from your NXX block to be your LRN. You add that to both the LERG and NPAC (National Porting Database). Once you've published all your switch information in the LERG and NPAC, then you establish your ISUP trunks with each ILEC you're interconnecting with. You can set up additional trunks with other carriers if you want a cheaper option for routing traffic, but the minimum required is the ILEC. Each carrier's switch will have a distinct point code associated with it so you'll order ISUP trunks to each switch (point code route) you need to be connected to. You'll also associate the distinct LRN for that LATA / carrier tandem area with that trunk group. Usually there's multiple trunk groups per LATA / tandem area so you'll program your routing tables with the NPA-NXXs each trunk group serves. That way when a customer originates a call, your switch can do the LNP dip to find the LRN and send it over the route that the NPA-NXX of the LRN is associated with. Routing tables can get complicated depending on how many carriers you're connected to. Companies that operate in more than one ILEC area or LATA usually purchase Least Cost Routing software so they can send their originating traffic out over the cheapest route. IXC traffic is routed a little differently because it is routed by CIC (4 digit code that identifies the IXC) rather than by NPA-NXX. They connect to all the ILEC carriers just like the CLECs do, but they populate their routing information in the SMS database instead of the NPAC database. Once the call is dipped, the traffic is delivered in pretty much the same way.....by dedicated trunk group / tandem area. MARY LOU CAREY BackUP Telecom Consulting Office: 615-791-9969 Cell: 615-796-1111 On 2020-09-02 04:46 PM, Ross Tajvar wrote: Hi all, I'm trying to understand how routing works in SS7-land. I am familiar with portability, and I know (at least in the US) the first step in routing a call is doing an LNP dip to get the LRN. However, it looks like addresses in MTP3 are "point codes" (PCs) which are assigned to switches. Calls are set up with ISDN-UP, which is transported via MTP3. So in order for a call to be set up, the destination switch's PC must be known. How is the destination PC determined from the destination LRN? Thanks, Ross _______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org<mailto:VoiceOps at voiceops.org> https://puck.nether.net/mailman/listinfo/voiceops<https://urldefense.proofpoint.com/v2/url?u=https-3A__puck.nether.net_mailma…> _______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org<mailto:VoiceOps at voiceops.org> https://puck.nether.net/mailman/listinfo/voiceops<https://urldefense.proofpoint.com/v2/url?u=https-3A__puck.nether.net_mailma…> -- Glen Gerhard glen at cognexus.net<mailto:glen at cognexus.net> 858.324.4536 Cognexus, LLC 7891 Avenida Kirjah San Diego, CA 92037 _______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org<mailto:VoiceOps at voiceops.org> https://puck.nether.net/mailman/listinfo/voiceops ---------------------------------------------------------------------- This message and any attachments are intended only for the use of the addressee and may contain information that is privileged and confidential. If the reader of the message is not the intended recipient or an authorized representative of the intended recipient, you are hereby notified that any dissemination of this communication is strictly prohibited. If you have received this communication in error, notify the sender immediately by return email and delete the message and any attachments from your system.

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Outsourcing STIR/SHAKEN Setup
by dovid＠telecurve.com Sept. 9, 2020

Sept. 9, 2020

Hi, Does anyone have a recommendation for a company that get us everything needed for STIR/SHAKEN setup? By setup I mean helping us file to get a cert etc. From the small research I have done there is a lot of fragmented information out there and it would be easier for us to pay someone else to do this then invest our own time to take care of this. TIA. Regards, Dovid

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TNS Outage?
by rkohutek＠gmail.com Sept. 9, 2020

Sept. 9, 2020

We have redundant A-Links to TNS and both are hard down. Ticket is open, but no ETR. Anybody have an idea what's happening?

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Question about SS7 routing
by ross＠tajvar.io Sept. 8, 2020

Sept. 8, 2020

Hi all, I'm trying to understand how routing works in SS7-land. I am familiar with portability, and I know (at least in the US) the first step in routing a call is doing an LNP dip to get the LRN. However, it looks like addresses in MTP3 are "point codes" (PCs) which are assigned to switches. Calls are set up with ISDN-UP, which is transported via MTP3. So in order for a call to be set up, the destination switch's PC must be known. How is the destination PC determined from the destination LRN? Thanks, Ross

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Google Fi
by shripald＠gmail.com Sept. 8, 2020

Sept. 8, 2020

Hi All, Is there anyone from Google Fi on this list? we have a slammed number situation and they only have email support to their porting/orders department. thanks, Shri

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Mobile HD Voice interop status?
by hartnett.tom＠gmail.com Sept. 3, 2020

Sept. 3, 2020

Does anyone have any updates or experiences in bridging AT&T or VZW HD Voice calls over to any non-mobile (VoIP) service and keeping the call wideband? It's discussed from time-to-time (with no real success reported) but it's been a while.

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Outsourcing STIR/SHAKEN Setup
by marylou＠backuptelecom.com Sept. 2, 2020

Sept. 2, 2020

I'm a Carrier Consultant who's been helping CLEC, IXC, Paging, Wireless and VOIP carriers install and maintain their PSTN networks for the the last 20 years. I can help clients get their FCC Certification to become a STIR/SHAKEN carrier as well as Numbering Resources, NPAC / LSR training, etc (if you need those pieces). Once my clients get their certification, I refer them to TransNexus. Jim and his team can help you with the process of turning your STIR/SHAKEN services up. MARY LOU CAREY BackUP Telecom Consulting Office: 615-791-9969 Cell: 615-796-1111 On 2020-08-31 05:37 AM, Dovid Bender wrote: > Hi, > > Does anyone have a recommendation for a company that get us everything > needed for STIR/SHAKEN setup? By setup I mean helping us file to get a > cert etc. From the small research I have done there is a lot of > fragmented information out there and it would be easier for us to pay > someone else to do this then invest our own time to take care of this. > > TIA. > > Regards, > > Dovid > _______________________________________________ > VoiceOps mailing list > VoiceOps at voiceops.org > https://puck.nether.net/mailman/listinfo/voiceops

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Looking for SIP/PRI provider
by ctaloi＠gmail.com Sept. 2, 2020

Sept. 2, 2020

We have a customer who needs a PRI to SIP gateway to handle some legacy applications and a service provider to handle their calls (1T!1, ~25k/month). It is not a market we deal with but I am looking to refer the customer to a provider who does. Any recommendations? I would just direct the customer to the provider, not looking to stay in the middle of this. Thanks, Chris

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