- VoiceOps - lists.voiceops.org

T.38 re-invites
by nathana＠fsr.com Jan. 10, 2012

Jan. 10, 2012

This might not be the most appropriate forum for such a question, but given that implementation bugs in either NOC gear or CPE gear can cause headaches for network operators, I offer this here in the hope that perhaps a fellow VoiceOp here may have run into this before and knows the answer... It is my understanding that with SIP, *either* party to a call can send another INVITE (a so-called "re-invite") at any time during an active session; this could be used to change the endpoint of a call or redirect it elsewhere, and is also commonly used to renegotiate the media in use (audio, video, image, and associated codecs) without dropping the call even while the endpoints remain the same. Does this hold true, though, for re-invites whose purpose is to switch from an RTP audio stream to a T.38 UDPTL stream? Can *either* peer send that INVITE? Or is there a rule -- spoken or unspoken -- that states that a T.38 re-invite can only be initiated by the *receiver* of the fax, and not the transmitter? I have some ATAs that appear to send out a T.38 re-invite upon detection of a fax tone regardless of whether the ATA placed the call or answered the call, and if I am understanding what I'm seeing in my packet captures, I think the fact that these ATAs do this even when they are going to play the role of the T.38 transmitter is confusing the heck out of my (Asterisk) SBC and the termination provider. But I can't find any concrete evidence to support the idea that T.38 re-invites should only be initiated by the receiver. -- Nathan Anderson First Step Internet, LLC nathana at fsr.com

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Broadsoft SIP Trunks and ILD Fraud
by zak＠simplesignal.com Jan. 10, 2012

Jan. 10, 2012

Good Morning Voice OPS Is anyone else experiencing anything like this? If so please share what you have done / or will to make it stop We have a series of smaller SIP trunk customers using Broadsoft trunk groups. By design the trunk groups have a concurrent call limitation based off the customer?s order. These smaller SIP trunks groups when compromised are able to run up HUGE fraud bills even tho they only have 5 or 6 SIP trunks. Needing to know if anyone else is seeing this that has Broadsoft and what was done to protect yourselves? Otherwise Happy NYE J Zak Rupas VoIP Engineer *SimpleSignal* 3600 S Yosemite Suite 150 Denver, CO 80237 One Number Rings All My Phones: 303-242-8606 SimpleSignal.com <http://www.simplesignal.com/> | Blog<http://www.simplesignal.com/blog>| Facebook <http://www.facebook.com/SimpleSignal?ref=ts> | Twitter<http://twitter.com/simplesignal>

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Acme Packet/Broadsoft/ENUM
by RDawson＠alliedtelecom.net Jan. 7, 2012

Jan. 7, 2012

Is anyone using ENUM with their Acme/BS deployment that wouldn't mind sharing a snippet of their zone file showing the proper record configuration? Thanks, Rob

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Australian Broadsoft Sales Contact
by nick＠inticon.net.au Jan. 5, 2012

Jan. 5, 2012

While not entirely ontopic Is anyone on this list able to put me in contact with someone who can assist with Broadsoft presales? I've had no luck with the online contact form and was unable to get anyone to call me back after calling through late last year. Cheers Nick Brown.

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please unsubscribe
by andrew.warburton＠aapt.com.au Jan. 5, 2012

Jan. 5, 2012

thanks! *Andrew Warburton* Business Development Manager | Wholesale AAPT Limited, 180 - 188 Burnley St Richmond, VIC 3121 *P +61 3 *8308 3838 | *M **+61 426 779 333* * * This communication, including any attachments, is confidential. If you are not the intended recipient, you should not read it - please contact me immediately, destroy it, and do not copy or use any part of this communication or disclose anything about it.

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Voice Mail retention guidelines
by ujjval＠simplesignal.com Jan. 4, 2012

Jan. 4, 2012

Hi All Do you have any specific guidelines in your SLAs around how long you retain customer voicemails? Is there any legal requirement ITSPs have to fulfill regarding this at federal or state levels? Appreciate your input. Thx, Ujjval

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Global Crossing outage?
by peeip989＠gmail.com Dec. 29, 2011

Dec. 29, 2011

Anyone else experiencing a voip outage with Global Crossing? Any idea how widespread it is?

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Cloudvox and their digits service
by beckman＠angryox.com Dec. 19, 2011

Dec. 19, 2011

Cloudvox used to offer an API for LERG data. http://www.prnewswire.com/news-releases/cloudvox-launches-free-api-bookmark… Unfortunately ifbyphone discontinued Cloudvox which included discontinuing their API service. I've seen the localcallingguide.com API, but I hate XML... Did anyone take over this service? Anyone from ifbyphone here want to put the code out there for someone to take over? Heck, I might be willing to offer it. Beckman --------------------------------------------------------------------------- Peter Beckman Internet Guy beckman at angryox.com http://www.angryox.com/ ---------------------------------------------------------------------------

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T.38 termination provider?
by nathana＠fsr.com Dec. 18, 2011

Dec. 18, 2011

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

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New Verizon Gateways?
by paul＠timmins.net Dec. 18, 2011

Dec. 18, 2011

They gave out finance team 3 days prior notice..... Eric Wieling <EWieling at nyigc.com> wrote: >Yes. VZB has a nasty habit of deploying new gateways but not telling anyone about them. > >-----Original Message----- >From: voiceops-bounces at voiceops.org [mailto:voiceops-bounces at voiceops.org] On Behalf Of Jared Geiger >Sent: Tuesday, December 06, 2011 3:09 PM >To: VoiceOps >Subject: [VoiceOps] New Verizon Gateways? > >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 > > >_______________________________________________ >VoiceOps mailing list >VoiceOps at voiceops.org >https://puck.nether.net/mailman/listinfo/voiceops >

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