July 2011 - VoiceOps - lists.voiceops.org

Wholesale termination time-of-day routing
by abalashov＠evaristesys.com Aug. 6, 2011

Aug. 6, 2011

Our principal product offering is a proxy-based "Class 4" trunking / wholesale routing/accounting/billing platform. We are internally debating implementing time-of-day routing, and are curious to know if there is a use case for time-of-day routing on this level. This is a popular feature on on PBXs, for obvious reasons. However, what we want to know is whether it is thought useful in an intra-industrial termination trunking context. It seems to have little utility for those doing domestic LD in NANPA land. However, there are international providers that have peak/off-peak rates in certain jurisdictions. Our experience with it has been mainly in the UK, but perhaps there are others. Opinions appreciated. Thanks a lot! -- Alex -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

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Letting a phone ring forever?
by carlos＠televolve.com Aug. 1, 2011

Aug. 1, 2011

We have a customer who has asked us to let his main line ring "forever" or at least five minutes. As a standard practice we forcefully hang up on an unanswered call at 90 seconds if the customer has turned off voicemail and auto-attendants. It seems wrong to let a line ring forever or even for minutes at a time, and I recall something in the back of my head about a traditional industry-standard limit. Your opinions requested. -- Carlos Alvarez TelEvolve 602-889-3003

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Helluva time with international
by d＠d-man.org July 29, 2011

July 29, 2011

We have a client who does sales in international locations and is constantly calling random countries. They are very sensitive to call quality. Countries called are generally in Europe, specifically lots of calls to France, Germany, Italy. We have been using a carrier whom gets 60%-70% ASR on each route and gave us three routes. The problem is that when calls fail on one route, sometimes there's a long delay before we get a failure code back. If we hangup quickly and move on to the next carrier, often the phone becomes "busy" because the first attempt was actually completing, just slowly. Sometimes we also get improper early media. So waiting for a response is hit/miss on speed, and not waiting for a response is hit/miss on call completion/quality. We're only doing $500-1,000 in international right now per month. I'm looking for a reliable carrier to move this traffic to. I was looking at PacWest for this. We're too low on $s to go direct to Global Crossing. I'm hyper-sensitive to running through extra media servers for no reason ? I want the RTP stream to go direct to the final carrier if possible. I wonder if anyone can make a recommendation on whom I can use? Or how I might get access to iBasis / Global Crossing routes directly in terms of RTP? I don't meet their minimums yet. - Darren

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Letting a phone ring forever?
by carlos＠televolve.com July 26, 2011

July 26, 2011

That's an interesting and useful suggestion. This customer is a local service company so nearly all calls are local, not toll. I've seen no evidence that he's using any sort of ring group with another carrier, as we hold all of his published numbers and nothing odd shows up in the logs. Jasper van Beusekom wrote: > > > Carlos Alvarez wrote, on 7/26/2011 3:32 PM: >> and made us remove the one we had implemented before (answer, tell the >> caller to please hold, then ring all his phones again). > > I suppose you could answer just to play your own ring tone, without the > prompt, but that is misleading to the caller who is starting to pay. > > The only theory I can think of is if he has you as part of a ring group > at some other provider. If you handle the call, that will impact the > other members of the parent ring group. > > Jasper -- Carlos Alvarez TelEvolve 602-889-3003

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AT&T Wholesale TF Origination and Termination contact
by compuwizz＠gmail.com July 23, 2011

July 23, 2011

Does anyone have a sales contact that they can share for AT&T Wholesale SIP? I haven't been able to find a contact on the AT&T website to call. Thanks! Jared Geiger

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Allworx Followme - no audio issue
by ujjval＠simplesignal.com July 22, 2011

July 22, 2011

Hi All: We have a customer with Allworx PBX that is setup for followme on an extension that forwards to a call Phone. When the call gets forwarded, back out the SIP Turnk intous, the allworx has answer confirmation which the followme/cellphone hears fine, when the followme cell phone presses 1 to accept the call, all audio is lost both ways. Call stays up until disconnected. The Allworx sends a REINVITE after the ?1? is pressed?too Anyone seen this behavior with ACME and Allworx and any remedies. Ujjval Karihaloo

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Any Metaswitch / Metasphere EAS users?
by lindsey＠e-c-group.com July 20, 2011

July 20, 2011

I'm looking for Metaswitch / Metasphere EAS users who would be willing to share some packet captures of the Shared-Line-Appearance and Busy-Lamp-Field features from their platform. If you can share, please let me know. Background: I'm looking at the effects of fancy signaling features on network capacity. E.g., "Shared Line Appearance" generates NOTIFY and INVITE transactions on every device where the line is shared. That means a lot more signaling. But what does that mean for the network access links and the various QoS policies in use (e.g., LLQ, CBWFQ, Ethernet strict-priority queues)? The conventional wisdom used to be that SIP signaling was so minute compared to RTP that you could ignore it. But is that still true? For those interested -- You can see the results of the Network Design survey, announced on this list in January, at the upcoming ACM IPTComm meeting in Chicago. The network designs of 36 VoIP service providers are considered. mark at ecg.co | +1-229-316-0013 | http://ecg.co/lindsey On Jan 11, 2011, at 12:20 PM, Mark R Lindsey wrote: > It's best for the Science and Art to have a public discussion about VoIP network designs. What was once an incidental installation task is often recognized as a key activity with serious cost implications. But still, many networks are never designed -- they just happen. IP Networks do not -- by default -- work well for VoIP. > > We the industry should document the current state of the art for the design of VoIP Networks. The need is greatest in carriers and large enterprises networks, but the design of networks for VoIP at all scales is interesting. > > Please help me document the standard industry practices in VoIP Network Design, both of new networks and also the expansion and evolution of existing networks. Your responses will be analyzed and compiled into a publication that will be submitted for conference publication. > > I hope to hear from vendors such as Acme Packet, Audiocodes, BroadSoft, Cisco, Digium, Genband, Metaswitch, Sonus, Taqua, and others. But I especially want feedback from individual designers & integrators. > > Everything you provide will be published online and credited. > > Responses received by February 15, 2011 will be used for the paper submission. > > My employer, ECG, Inc. is a for-profit entity, but this project is NOT a commercial enterprise. We wish to foster and sponsor innovation in VoIP networks in an open, free academic dialog. > > Mark R Lindsey, ECG Inc. > mark at ecg.co | +1-229-316-0013 | http://ecg.co/lindsey > > > ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ > > VoIP Network Design Survey. > Please respond to lindsey at e-c-group.com > > > > A. If you would be willing to submit reference network designs for publication, please send them in PDF format. [VNDS1-A] > > > > B. Please provide answers to the following questions -- as much as possible. Even if you only have time to answer one question. [VNDS1-B] > > > > Q-1. What application-layer (VoIP) network devices do you regularly design with? What are special network requirements of each device? [VNDS1-B-1] > > > > Q-2. Where do you use public IP addresses, and where do you use private IP addresses? In each case: why do you use that type of address? [VNDS1-B-2] > > > > Q-3. How do you handle customer-premise access devices behind NAT? [VNDS1-B-3] > > > > Q-4. How do you handle peering ("SIP trunks") between carriers? (e.g., SBC? Separate IP assignment per endpoint? Separate Port assign nment? Authentication?) [VNDS1-B-4] > > > > Q-5. How do you guarantee network capacity for VoIP Media traffic? ...for VoIP signaling traffic? [VNDS1-B-5] > > > > Q-6. Are you designing to support SIP over TCP or SCTP? If so, why? If not, do you address fragmentation of large SIP? [VNDS1-B-6] > > > > Q-7. How do you describe your approach to network security boundaries? How do you enforce these boundaries? (Common approaches: firewalls, SBCs, VRFs/routing domains, server packet filters, private IPs and routing policies, etc.) [VNDS1-B-7] > > > > Q-8. What are the key security or operational threats that you design to protect against? [VNDS1-B-8] > > > > Q-9. What are the key network design principles for VoIP Networks that you follow? [VNDS1-B-9] > > > > Q-10. What are the key mistakes you try to avoid in VoIP Network design? [VNDS1-B-10] > > > > Q-11. What future trends in network design and requirements do you anticipate? [VNDS1-B-11] > > > > Q-12. What are your tightest constraints in your network designs? (e.g., Design time? Testing/Stability/Change Control? Rack space? Network link capacity? Money? CPU Capacity? CAM space?) [VNDS1-B-12] > > > > > >

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Porting Cell Numbers
by marylou＠backuptelecom.com July 15, 2011

July 15, 2011

So sorry....when you said Verizon I was thinking ILEC not Verizon Wireless. You shouldn't need an agreement with a carrier to port a number from them. As long as the potential customer's TN homes on the same ILEC tandems that you are connected to, you can port their number to your network. You may want to talk to the other carrier about setting up a provisioning interface with them because you're supposed to issue an LSR to them to request the number be ported, but you shouldn't need an agreement to submit an LSR to them. Mary Lou Carey BackUP Telecom Consulting CLEC Consultant OFF: 615-791-9969 -----Original Message----- From: Eric Hiller [mailto:clec at cygnustel.com] Sent: Tuesday, July 12, 2011 11:54 AM To: Mary Lou Carey Subject: RE: [VoiceOps] Porting Cell Numbers Doesn't it just matter what tandem they have the number homed off of? ... and don't the cellular carriers typically home of the ILECs tandems? On Tue, July 12, 2011 8:26 am, Mary Lou Carey wrote: > You may or may not be able to do this, but I think the bigger question is > how would the call get back to your switch if you don't have any > interconnection trunks with them. > > > Mary Lou Carey > BackUP Telecom Consulting > CLEC Consultant > OFF: 615-791-9969 > > > > -----Original Message----- > From: voiceops-bounces at voiceops.org [mailto:voiceops-bounces at voiceops.org] > On Behalf Of Eric Hiller > Sent: Monday, July 11, 2011 12:54 PM > To: VoiceOps at voiceops.org > Subject: [VoiceOps] Porting Cell Numbers > > We are a CLEC and regularly port numbers over from the ILEC (ATT) who we > have an ICA with. Would we be able to port cellular/wireless numbers from > say Verizon or Sprint over to us as a landline CLEC without an ICA? I know > we could do it from the Neustar/NPAC side of things, but would > Verizon/Sprint/etc. release the number? > > Thanks! > -Eric > > > > _______________________________________________ > VoiceOps mailing list > VoiceOps at voiceops.org > https://puck.nether.net/mailman/listinfo/voiceops > > >

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T38 delay management
by fasterfourier＠gmail.com July 13, 2011

July 13, 2011

I'm running into occasional problems with failed fax transmissions due to what appears to be excessive delay between receipt of the DIS from the receiving fax and sending of the DCS by the transmitting machine. Here's the setup: -SPA2102 ATA running 5.2.12 with the following fax-related configuration bits: <FAX_CED_Detect_Enable_1_ ua="na">Yes</FAX_CED_Detect_Enable_1_> <FAX_CNG_Detect_Enable_1_ ua="na">Yes</FAX_CNG_Detect_Enable_1_> <FAX_Passthru_Codec_1_ ua="na">G711u</FAX_Passthru_Codec_1_>  <FAX_Codec_Symmetric_1_ ua="na">Yes</FAX_Codec_Symmetric_1_> <FAX_Passthru_Method_1_ ua="na">ReINVITE</FAX_Passthru_Method_1_>  <FAX_Process_NSE_1_ ua="na">No</FAX_Process_NSE_1_> <FAX_Disable_ECAN_1_ ua="na">Yes</FAX_Disable_ECAN_1_> <FAX_Enable_T38_1_ ua="na">Yes</FAX_Enable_T38_1_> <FAX_T38_Redundancy_1_ ua="na">1</FAX_T38_Redundancy_1_>  <FAX_T38_V29_Only_1_ ua="na">No</FAX_T38_V29_Only_1_> <FAX_Tone_Detect_Mode_1_ ua="na">callee only</FAX_Tone_Detect_Mode_1_>  -ATA registered to [commercial asterisk-based multitenant PBX system] running asterisk 1.6 and kamamilio 1.5 -Calls terminated through Level3, Global Crossing, and T38faxing.com Fax calls are set up properly, ReINVITE is issued, RTP stops and T38 session starts as usual. Once the T38 session starts, it appears that it takes about 2.8 seconds for the ATA / fax combination to return a DCS message once the DIS message hits from the terminating gateway. This, when combined with other delays (a few tens of milliseconds between the ATA and terminating gateway) seems to cause the T1 timer (I think that's the one...) on some receiving faxes to expire. A few more DIS messages are sent and then they give up. What is a normal amount of time for a DCS to be returned by the ATA when hit with a DIS? Is there anything that can be done to prune the delay down? Thanks in advance.

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Voip Origination Partners
by Marty_Sorensen＠adp.com July 13, 2011

July 13, 2011

Need some Voip origination provider recommendations domestically in the US besides Level 3 and Verizon that can support hosted pbx and sip trunking. You can contact me off list if you would like to. Martin Sorensen Director, IPNS 3rd Floor West ADP Dealer Services 2525 SW First Avenue, Portland, OR 97201 Phone: (503) 205-4438, Fax: (503) 402-3277 Email: Marty_Sorensen at adp.com<mailto:Joe_Bell at adp.com> 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, please notify us immediately by e-mail and delete the message and any attachments from your system.

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