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localcallingguide.com down?
by ivan.kovacevic＠startelecom.ca April 5, 2017

April 5, 2017

I get ?Error loading character set utf8: (1226) User 'lcgread' has exceeded the 'max_questions' resource (current value: 50000)?. Anyone else? (and apologies if this should go to the Outages list, in view of the recent appropriateness discussion there, I felt it?s more relevant to the voice list) Best Regards, Ivan Kovacevic Vice President, Client Services Star Telecom | www.startelecom.ca <http://t.sidekickopen61.com/e1t/c/5/f18dQhb0S7lC8dDMPbW2n0x6l2B9nMJW7t5XYg1…> | SIP Based Services for Contact Centers | LinkedIn <http://t.sidekickopen61.com/e1t/c/5/f18dQhb0S7lC8dDMPbW2n0x6l2B9nMJW7t5XYg1…>

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Avaya IP Office SIP DTMF
by nick＠flhsi.com April 5, 2017

April 5, 2017

Greetings all, Does anyone have any pointers on delivering DTMF reliably to an Avaya IP Office system? We've got a reseller that installs these, And it seems to be a crapshoot regarding if DTMF works well on them. Specifically inbound DTMF (Dialing into the IP Office system, And dialing an extension via an auto attendant..etc.). Most of the time, The system will produce a "That option is invalid" as it's misheard the DTMF. Out of about 30 systems currently serviced, I'd say about 18 of them work perfectly with RFC2833 based DTMF. 6-7 of them work perfectly fine after switching to INBAND DTMF. And a handful (The remaining 6ish) of the remaining don't work well with either. Of the above 30 systems, They range in IP Office version from 7.x to the latest (9.x I think). And there seems to be no correlation with software version and if it functions correctly, Or not. When a DTMF change is made, It's always reflected in the IP office SIP trunk configuration, After we found some wording in the IP Office documentation basically stating it follows whatever the other system presents via DTMF. IP connectivity doesn't seem to be the problem here. As all of the non-functional devices have perfect connectivity, And DTMF can be seen in a packet capture at the port the device plugs into. We're providing SIP to all of these systems with an Asterisk box running 12.8.2. The same box is currently serving about 700 customers. User equipment ranging from just about every known IP phone under the sun, The same with ATA's. Many Digium G-series T1 gateways, Cisco call managers... None of which ever have a problem. Hoping someone has some "magic" they can loan me, Or some relevant settings to tweak (On either system). Nick Olsen Sr. Network Engineer Florida High Speed Internet (321) 205-1100 x106

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Exclusive: Quitting the fast life of open-source SIP
by abalashov＠evaristesys.com April 3, 2017

April 3, 2017

(Filed by UC Celeb Talk - 1 Apr 2017.) BERLIN (Tegel), GERMANY--In the now-infamous breakdown episode on the VIP tarmac at Berlin's Tegel Airport, Alex Balashov shocked the IP telecoms celebrity world last month. Shielding his eyes from the punishingly bright daylight of an overcast day in northern Germany, Balashov climbed out of his Gulfstream G650 jet, clutching a half-finished bottle of Woodford Reserve, and announced his sudden retirement from the "fast life" of open-source SIP consulting to a waiting gathering of tech business reporters. "I'm not with this 'live fast, die young' VoIP s*** anymore," he reportedly shouted, before collapsing and vomiting profusely on the red carpet. UC Celeb Talk's own Benjamin Brant sat down with Balashov in an exclusive interview to learn more about what happened, even as the telecoms industry continues to digest the shock meltdown. -- BRANT: Everyone's dying to know, what was going through your head at that moment? BALASHOV: Funny how you mention dying. I want to live to see 35, man. I can't handle the SIP biz anymore. BRANT: What do you mean? You seemed to have all the makings of an open-source SIP superstar. BALASHOV: I can't keep up with the drinks anymore, the three day-long parties, the paparazzi, always watchin' my back at ClueCon... BRANT: When'd you realise it might be time to call it? BALASHOV: I'll tell you when. It was Kamailio World '16. Man, those vodka + Club Mate drinks they pour down at Fokus are something else. They'll really get you thinking about things. BRANT: So you had a cocktail and thought better of your entire life as a SIP celebrity? The fans, the adulation, the glory? BALASHOV: When you wake up at 3 PM at the Park Inn on Alexanderplatz with a $31,000 room service bill on your AMEX, a goat wandering around, and reporters in your face, you tell me with a straight face you're not gonna take stock of your life and make some changes. BRANT: That really happened? BALASHOV: That's what I'm trying to tell you! That's Kamailio World, dude. That's the SIP software engineering culture. You wanna ITSP and WebRTC? You're gonna have to bro down! BRANT: Surely not everywhere. What about the IETF? BALASHOV: Those guys drink more than ANYONE. Have you seen the SIP Outbound RFC? BRANT: That's the work-product of an alcohol-fuelled creative orgy? BALASHOV: It's the work product of something-something. I don't know if I'm allowed to identify the uh, substance in question. BRANT: How are you going to live without the private jet? If I had a G6, I wouldn't just turn over the keys. BALASHOV: Yeah, I mean, open-source VoIP consulting has its upsides. It's a fine way to make a living. But it's just not worth it. Besides, hourly operating costs are through the roof. BRANT: But you were an icon, and quite a ladies' man too! How do you walk away from that? BALASHOV: Look man, of course women are gonna mob you when you're a SIP consulting star. That's just part of the open-source tech life in general. But it's fake, man. They don't love you. They don't want to know who you are inside. They just want you 'cause you know all about the SIP transaction state machine, loose vs. strict routing, cross-compiling Kamailio. They just got a crush on you 'cause you know presence and parallel forking race conditions. I want something real. BRANT: But that's part of the social bargain of being an open-source SIP celebrity. Didn't you once boast of having uh, lady friends, "in different geographic and non-geographic operator codes"? BALASHOV: Yeah, and that's cool, but I want to come home to a woman who loves me for me, not Mr. 3... ... 02, Moved Temporarily, but not happy necessarily. Feel me? BRANT: What happened at ClueCon? BALASHOV: You talking about the bullets Acme Packet ventilated me with? BRANT: Five of them, as I heard. BALASHOV: Well I'm still standin'. Took those shots and smiled. BRANT: I know you guys have a kind of omert? about this stuff, but I've got to ask: why? BALASHOV: I ain't worried, kids growing up in today's world need to learn the truth. They act like I was dealing transcoding licences on their block. O.G. 729. BRANT: But that wasn't what gave you a wake-up call that the SIP game is sharper than a razor blade? BALASHOV: Hell no, it wasn't their block. That fake-ass east coast channel partner crew can cash me outside howboudah? BRANT: How exclusive is Kamailio World? BALASHOV: It's one of the most exclusive clubs I've ever hit up. Lines wrapping around blocks. And the bouncers. Damn. BRANT: The bouncers? BALASHOV: Those dudes are harsh. BRANT: On a scale of 1 to Berghain, how bad is it? BALASHOV: All I know is, one time some squirrely guy in the audience asked a moderately new-blood question like, "If a Record-Route header has been added by an intermediate proxy, is it proper of a SIP User Agent to populate the Request URI of subsequent in-dialog requests with the Record-Route URI, or should it be the remote Contact?" Half the room got up like it was ready to curb-stomp that little face. Guy didn't know what he got himself into--a room full of elites. BRANT: So what happened? BALASHOV: Someone was like, "Bounce this fool! He don't know how to act." So these three muscular Kaiser Wilhelm-looking dudes--moustaches and all--in sunglasses roll up and drag this guy through the Atrium, out the doors and dump him straight out onto Anna-Louisa-Karsch-Stra?e. BRANT: Damn. Ejected out of the saloon. BALASHOV: Yo, that dude was dismissed from the premises. BRANT: So what do you see in your future post-SIP life? BALASHOV: I'm thinking about starting a bakery. A friend threw it out there as a way to kind of atone for all this and get straight with Jesus. Gonna bake really big cookies. You know, huge ones, like small pizzas? But recognise--they're actually cookies. BRANT: Got any advice for those who bright young upstarts who haven't turned over a new leaf and thrown in the towel yet? BALASHOV: Come on down to Kamailio World #5, May 8-10 2017 in Berlin. You'll see your life in a new way. Registration and more information online at https://www.kamailioworld.com/k05/

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Looking for some help
by brianjenkins＠corp.sonic.net March 30, 2017

March 30, 2017

Hello, We are experiencing an intermittent buzzing sound on only inbound calls. Only the originator hears the buzzing. Calls come into our class 5 switch via ISUP trunks directly connect to an AT&T tandem. Looking at our EMI report all the calls with the buzz get billed to Century Link. We have tested directly at our hand off to AT&T and were unable to hear any buzzing on an active call where the originator was hearing it. I have been trying to work with Century Link but since we are not a direct customer it has been difficult. I was hoping to find a Century Link Voice Engineer out there I could work with. Please contact me directly. Thank You -- Brian Jenkins Network Operation

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Caller ID name discrepancies between carriers
by caalvarez＠gmail.com March 28, 2017

March 28, 2017

On occasion we have a problem where CNAM is shown incorrectly by one or two destination carriers. Most often it has been Cox and T-Mobile, because they seem to cache CNAM "forever." Today it's Verizon wireless. They are showing an ancient name for one client. We changed it at least three years ago. Our carrier for that number is Onvoy. Should we open a ticket with them? I'm guessing I'd have no success calling Verizon. I have been able to call Cox when they are screwed up, but probably only because I have a direct NOC number.

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T.38 version 3 industry support?
by nathana＠fsr.com March 27, 2017

March 27, 2017

Way back when we started looking into FoIP, it was hard to find good ATAs with a quality T.38 implementation (or at least it seemed that way to us). And that's even before touching on all of the interop issues due in part to a poorly-written spec that can be interpreted many different ways. Over the years, things seemed to have mostly settled when it comes to T.38 version "0". ATAs are fairly abundant, and it's not hard to find direct-IP orig and term carriers that can gateway T.38, either. But there are a lot more "Super G3" (V.34) fax terminals out there in the wild these days, and I'm getting tired of us having to explain to end-users that they need to turn that crap off. And my perception (whether accurate or inaccurate) is that most carriers still by-and-large only support version 0 of the protocol, which has no support for V.34 modulation. Is this accurate, or are things gradually changing on this front as well? I haven't even bothered to shop around for and test T.38 version 3 TAs because it's pointless to go to that trouble if nobody can carry that traffic (and we are not really interested in buying a bunch of circuits and gatewaying it ourselves), though even there it sure feels like there are few options. Not even the venerable Adtran line will do anything other than v0 (last I checked). -- Nathan

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Sending CNAM to T-Mobile
by caalvarez＠gmail.com March 24, 2017

March 24, 2017

One of our customers is claiming that when they call their T-Mobile cell phones, the CLID is correct, but they get CNAM such as "Reception1" and "Lab2". Those are the internal names of the phones. This seems impossible. Anyone heard of this being possible? The calls were sent via Onvoy.

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Multicast Paging Device
by robert.j＠bendtel.com March 24, 2017

March 24, 2017

Does anyone have recommendations on a SIP Multicast paging device that would connect to a pre-existing paging amplifier.

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Maximum latency
by caalvarez＠gmail.com March 22, 2017

March 22, 2017

One of our larger customers is about to launch a new call center in Malaysia. The connection there is fast, the trace looks surprisingly clean and short, but latency is consistently at 239-245ms. We've never knowingly had a connection over 130ms. Does anyone have experiences, good or bad, with latency approaching a quarter second? The jitter level seems fine, so I believe they'll just have a delay but decent call quality.

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SecureLogix
by ivan.kovacevic＠startelecom.ca March 17, 2017

March 17, 2017

Anyone using SecureLogix and can comment on their ability to ?Prevent service abuse and malicious or unwanted voice traffic to lower expenses and boost your security and compliance efforts. ? http://www.securelogix.com/Telephony-Denial-Of-Service/index.htm It sounds like it isn?t just an SBC-type prevention of SIPVicious-style attacks, but also TDM blocking based on ? policy fields? call type; absence or blocking of caller ID, call source and destination numbers; call direction; time of call; call length; and certain VoIP call attributes, such as excessive media rate? I understand this can sound appealing, but having had to deal with a few TDOS attacks, I am not sure how well does this would work in practice. Anyone able to comment? Best Regards, Ivan Kovacevic Vice President, Client Services Star Telecom | www.startelecom.ca | SIP Based Services for Contact Centers | LinkedIn <https://www.linkedin.com/company/star-telecom-www-startelecom-ca-?trk=biz-c…>

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