December 2011 - VoiceOps - lists.voiceops.org

New Verizon Gateways?
by jared＠compuwizz.net Dec. 14, 2011

Dec. 14, 2011

Did VZB add new gateways lately? I've located what I think are all the new IPs, but if someone from Verizon can contact me with the details of the added IPs, I'd be grateful. My account manager is pretty useless. Regards, Jared Geiger

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T.38 termination provider?
by kris＠kriskinc.com Dec. 13, 2011

Dec. 13, 2011

Recent versions of Asterisk and Freeswitch are also capable of T.38 "gatewaying" (T.30 g711/T.30 interop). However, my guess is that the OP was interested in a pure T.38 path for its perceived reliability. Getting back on topic, you aren't likely to have good fax results from anything but a Tier 1 provider like Level(3). There are just too many LCR shenanigans and who knows what else with anything less. With that being said we've been doing T.38 on Level(3) Vector (Sonus) for over a year. I regularly run loopback tests with multiple geographic and provider diverse PRIs sending/receiving faxes to/from LI and ELS dids on every Level(3) SBC. Sending T.38 faxes to Freeswitch with T.38 regularly results in a %97 success rate. Upon complaining to Level(3) their first suggestion was to switch to ulaw. Keep in mind this is over the public internet. I was skeptical to say the least. Re-running the same tests regularly results in a %98 success rate. We're able to dynamically switch between T.38 and ulaw on a call by call basis. Repeated controlled tests have held with ulaw consistently being about %1 more reliable. My only guess is that T.38, while being more reliable in theory, suffers from implementation bugs like anything else. On a solid connection ulaw is simpler and thus more reliable. On a shaky connection between known endpoints T.38 emerges victorious due to features such as packet redundancy, etc. That's why we still use T.38 down to our ATA devices with great success. Our customer's DSL connections probably couldn't get past tone detection/modem handshake with ulaw. Yes, I really thumbed out this entire message from my BlackBerry near midnight. I've lost way too much sleep from faxing over IP as it is! -- Kristian Kielhofner Sent from a mobile device. ------------------------------ *From*: voiceops-bounces at voiceops.org <voiceops-bounces at voiceops.org> *To*: Nathan Anderson <nathana at fsr.com> *Cc*: voiceops at voiceops.org <voiceops at voiceops.org> *Sent*: Mon Dec 12 21:59:10 2011 *Subject*: Re: [VoiceOps] T.38 termination provider? Slightly off topic but from spec sheet the new transcoding NUIs in the low-end ACME SBCs (3820/4500?) in theory can handle T.38 to T.30(ulaw/alaw) interop, so you might be able to 'wrangle' it yourself. -- Peter Childs - Voice Engineer Internode/Agile Ph: 08 8228 2380 Mb: 0406 388 405 Fax: 08 8235 6801 On 13/12/2011, at 12:26 PM, Nathan Anderson wrote: Hello, New to the list. I work for a regional broadband ISP who is looking to also become an ITSP. We are looking for a good SIP termination provider with reasonable rates to all of North America that also happens to guarantee T.38 gateway support for 100% of calls. Currently, we are using FlowRoute. We are happy with the service, but the odds of us hitting a gateway (through one of their upstreams) that will accept our T.38 re-INVITE are about 50/50...not very good. We tried Gafachi, but that was not a good experience. Their rate table is all kinds of screwed-up. The only other one I've been able to find in my searches is Alcazar Networks. Anybody have any first-hand experience with them who can vouch for them as well as their T.38 support for the wholesale termination product? Other recommendations are also welcome, as well. Although we may someday go down the road of loading an LCR table into our SBC, at this point we are looking for a single solid termination provider that will handle 100% of our outbound call volume (a mix of voice and fax); we can fall-back to FlowRoute as a backup solution if the primary provider goes down. Thanks a million, -- Nathan Anderson First Step Internet, LLC nathana at fsr.com _______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops

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Do Not Call list
by MStappenbeck＠allworx.com Dec. 9, 2011

Dec. 9, 2011

Does anyone know of a SIP product that will block outbound calls to the PSTN from a PBX, based on a database match? It would need to be loaded with the national registry, as well as allow the end user to load numbers themselves. This could reside on a local server, or out in the cloud. Not quite a predictive dialer, since the end user will dial the number from the handset themselves, and receive an appropriate recording notifying them if the number dialed matched a "blocked" number. Thanks for any help -Mark S

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New verizon peers
by David_Hiers＠adp.com Dec. 5, 2011

Dec. 5, 2011

Everyone ready for the three new Verizon peers coming online tomorrow? David Hiers CCIE (R/S, V), CISSP ADP Dealer Services 2525 SW 1st Ave. Suite 300W Portland, OR 97201 o: 503-205-4467 f: 503-402-3277 ###Please note my email address is changing: ###from David_Hiers at adp.com ### to David.Hiers 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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FW: [EMS] EMS Update
by ivank＠rogers.com Dec. 4, 2011

Dec. 4, 2011

Not sure if anyone here used them. luckily we decided not to. call it gut instinct. From: EMS Financial Services [mailto:admin at emsfinancialservices.com] Sent: December-03-11 6:17 PM Subject: [EMS] EMS Update <http://www.emsfinancialservices.com/> Dear EMS Customers and Friends, You are undoubtedly aware of the difficult situation EMS finds itself in. However you may or may not be aware that a forensic examination of EMS accounts has revealed evidence that the Original Founder, Principal and Managing Member of EMS, Mr. James Nicholson, has misappropriated and mismanaged substantial amounts of EMS funds, having sole and complete access and control of all financial banking throughout the period. Upon the company becoming aware of these matters that were created by Mr. Nicholson, he resigned from his position on September 9, 2011. The remaining EMS employees proceeded with full corporate due diligence, while commencing negotiations with various interested parties to secure the liquidity in order to honor all deposits. We have used all efforts and resources to work with any and all potential investment groups around the world. We are saddened to advise that we have no further resources or opportunities available to restore or correct the actions and mismanagement that have occurred, and are forced to cease operations. The team has been and is in communication with the legal and governmental authorities in order to further pursue all matters related, and would expect them to continue their investigation into the matter. We profusely apologize for this situation, and we will continue to the best of our abilities to keep you informed of any further developments. Kindest regards, The EMS Staff Best regards, <http://www.emsfinancialservices.com/> Ems Financial Services Please note: This e-mail message was sent from a notification-only address that cannot accept incoming e-mail. Please do not reply to this message.

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Problems with Broadworks, call center, hoteling, and a Polycom Soundpoint IP 501
by nelsonh＠socket.net Dec. 1, 2011

Dec. 1, 2011

Hello, I have a ticket in with our software vendor on this, but thought I'd post to see if anyone else had encountered this: We have a call center with Polycom SPIP 501 phones that we are trying to repurpose into a Broadworks call center. We are running Broadworks r17sp4. The agents who answer calls float from desk to desk between and during shifts, and the supervisors want to keep statistics by agent, so we are setting up with call center users as hoteling guests and phones as hoteling hosts. I've run into four problems compared to a similar deployment with Polycom 331s, 550s, and 650s: * The phone does not seem to sync its ACD status from the Broadworks server. When the phone boots, it assumes it is logged out of the queue. If the call center agent hasn't hoteled into the phone yet, the phone will update its status when the user hits the login/logout and available/unavailable button, even though the user the phone is registered as isn't part of the call center itself and can't log in or out. * Once logged in, the phone shows its status as 'Available', even though Broadworks shows the status as 'Sign-in'. If the user hits the available/unavailable button, the phone shows its status as unavailable, but Broadworks shows its status as available. If you press it again, the phone goes to available and Broadworks goes to unavailable, and back and forth. Changing the state in Broadworks directly seems to have no effect on the state the phone displays, and changing the state on the phone sets the state in Broadworks opposite to what the phone shows, even if I try to trick it into synching up by setting the state in Broadworks to the same state the phone thinks it's in. * This is the most confounding: There's a red lamp at the top of the phone that is used for a few things, including the ring indicator, the voicemail indicator, and an error indicator. If a call center agent hotels into the phone as a guest while logged into the call center and takes a call center call, the light starts flashing when the incoming call comes in, but doesn't stop flashing when the agent answers the call, hangs up the call, takes other calls, or places calls. The only way to get the indicator to go off is to disassociate the agent with the phone (or reboot the phone). Then, the light stops blinking. This only happens with this specific combination; if I make the phone user itself a call center agent and disassociate the hoteling guest, the light only blinks until the call is answered. If the phone user itself takes a call without a hoteling guest, it only blinks until the call is answered. If the call center agent is associated and takes a direct call by extension, the light only blinks until the call is answered. I also couldn't duplicate this on a Polycom 550, even using the same Polycom SIP software version (3.1.2 rev B) (the 550 does have separate line indicators from the voicemail/error lamp, so that may be why it behaves different). It's only call center calls to the hoteling guest on a Polycom 501. * The display on the phone doesn't display call statistics when a queue call is ringing in like it does with the newer phones. This may just be a matter of the 501 not supporting the interface that Broadworks uses to display them, or the SIP application version not supporting it. I have not tested this with a 550 using the older SIP application version yet. I have some Broadworks application server logs of the stuck ring indicator, so if anyone wants to see them, let me know and I'll clean them up and send them. If anyone has any other suggestions, I would welcome them. Thanks, Nelson Hicks DNS and Web Administrator SOCKET (573) 817-0000 dns at socket.net [1] 1-800-SOCKET-3 www.socket.net [2] Links: ------ [1] mailto:dns at socket.net [2] http://www.socket.net/

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