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Netural Tandem
by David_Hiers＠adp.com Oct. 19, 2011

Oct. 19, 2011

Hi, Is there anyone from NT on the list? I'm working an issue between ATT and XO, could really use a backchannel contact for a sanity check. Thanks, 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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Toll free SIP termination
by fasterfourier＠gmail.com Oct. 19, 2011

Oct. 19, 2011

I'm looking for a SIP carrier that will provide reliable termination of toll free traffic and pass ANI info correctly (this part is important). Also I want it to be free :) Several companies claim to provide this, who's actually using what and are you happy with it? Low volume--currently about 3500 minutes per month. Thanks!

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Looking for a Sr Network Security Engineer
by jnesheim＠cytek.biz Oct. 14, 2011

Oct. 14, 2011

I'm looking for a Senior Network Security Engineer to join our team at TelePacific and I'm hoping you guys may be able to help with this difficult to fill role. We're looking for someone who is focused on carrier infrastructure and datacenter security more than enterprise IT security so a candidate would need to be well versed in security of core network routing protocols, carrier ethernet, MPLS/VPLS, SIP, and other building blocks of modern networks. If you or someone you know may be interested in joining our team please send me a note at jnesheim at telepacific.com or apply at http://telepacific.submit4jobs.com/index.cfm?fuseaction=85326.viewjobdetail… We also have several other Engineering positions open as well listed here: http://telepacific.submit4jobs.com/index.cfm?fuseaction=85326.viewjobs&CID=… Thanks! -- Jason Nesheim

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AT&T Right of Ways
by carlos＠race.com Oct. 6, 2011

Oct. 6, 2011

Anyone familiar with doing all the paper work and design work for using the right of ways in the AT&T states? We have the row in our ICA amendments and looking for the first time to possibly use them. Carlos Alcantar Race Communications / Race Team Member 101 Haskins Way, So. San Francisco, CA. 94080 Phone: +1 415 376 3314 Fax: +1 650 246 8901 / carlos at race.com / www.race.com<http://www.race.com/>

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Audio latency in DC area, Part 2
by jschnelz＠enterprisepcs.com Oct. 6, 2011

Oct. 6, 2011

Hey David have you tested the following scenarios: 1) Out of area Cell phone roams into the affected market(s) 2) Assigning a Wash DC DID to a phone in your PDX market and testing with a PDX cell phone. In the first case the out of area phone will use a TLDN that is typically a non-cellular DID (it is typically part of the wireline network for the wireless carrier). The second scenario should remove the local market handoff between the cell carriers and level3. You might also try a Wash DC assigned cell phone used in PDX to the Wash DC DID. We had a similar problem with Nextel phones to Level3 DID's which turned out to be a SIP trunk issue between Sprint and Level3 (it was on Sprint's side). We had to get Level3 to open a ticket with Sprint for testing. Since this is all carriers, it points to a Level3 issue (if they are L3 DID's on your VOIP phones) Jon Schnelz -----Original Message----- From: voiceops-bounces at voiceops.org [mailto:voiceops-bounces at voiceops.org] On Behalf Of voiceops-request at voiceops.org Sent: Thursday, October 06, 2011 6:31 AM To: voiceops at voiceops.org Subject: VoiceOps Digest, Vol 28, Issue 4 Send VoiceOps mailing list submissions to voiceops at voiceops.org To subscribe or unsubscribe via the World Wide Web, visit https://puck.nether.net/mailman/listinfo/voiceops or, via email, send a message with subject or body 'help' to voiceops-request at voiceops.org You can reach the person managing the list at voiceops-owner at voiceops.org When replying, please edit your Subject line so it is more specific than "Re: Contents of VoiceOps digest..." Today's Topics: 1. AT&T Right of Ways (Carlos Alcantar) 2. Re: AT&T Right of Ways (Paul Timmins) 3. Re: AT&T Right of Ways (Carlos Alcantar) 4. Audio latency in DC area, Part 2 (David Hiers) ---------------------------------------------------------------------- Message: 1 Date: Wed, 5 Oct 2011 20:17:11 +0000 From: Carlos Alcantar <carlos at race.com> To: VoiceOps <voiceops at voiceops.org> Subject: [VoiceOps] AT&T Right of Ways Message-ID: <CAB20AED.474CC%carlos at race.com> Content-Type: text/plain; charset="us-ascii" Anyone familiar with doing all the paper work and design work for using the right of ways in the AT&T states? We have the row in our ICA amendments and looking for the first time to possibly use them. Carlos Alcantar Race Communications / Race Team Member 101 Haskins Way, So. San Francisco, CA. 94080 Phone: +1 415 376 3314 Fax: +1 650 246 8901 / carlos at race.com / www.race.com<http://www.race.com/>

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Audio latency in DC area, Part 2
by hiersd＠gmail.com Oct. 6, 2011

Oct. 6, 2011

Hi folks, I want to send out another shout regarding an on-going issue in the DC area. We've got excessive audio delay between our voip phones and cell phones. Packet specs (loss, jitter, etc) are fine, ping times for everything rock. Specifically, these rate centers are involved: DALE CITY HERNDON MANASSAS WSNGTNZN08 WSNGTNZN17 but NOT this nearby one: TRIANGLE Here are some of the voip NPA/NXXs involved: 703647 703880 703508 703647 240283 The issue affects any cell in the area. All our test calls are local; the cell phone is in the same building as the voip phone. The cell phone has and has strong signal. We've measured audio latency of up to 550ms; it is just impossible to hold a conversation. In all cases: 1. VOIP-to-CELL calls are slower than CELL-to-VOIP 2. 3G is slower than GSM 3. PSTN-VOIP calls are good 4. PSTN-CELL calls are good 5 Only VOIP-CELL calls are bad Anyone got any issues in the DC area? Would your users report this kind of problem? The DC area comes in near the bottom of the list on most of the "bad cell phone area" studies, so most cell users there have had tolerance of this sort of thing beat into them over the years. Thanks, David

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Metaswitch forum
by anorexicpoodle＠gmail.com Oct. 3, 2011

Oct. 3, 2011

Who here will be attending the Metaswitch forum in Vegas next week? Anyone interested in gathering over a pint?

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Toll free SIP termination
by gwenzit＠gmail.com Oct. 3, 2011

Oct. 3, 2011

Godaddy Sent from my HTC on the Now Network from Sprint! ----- Reply message ----- From: "Peter Rad." <peter at 4isps.com> Date: Fri, Sep 30, 2011 10:10 am Subject: [VoiceOps] Toll free SIP termination To: "voiceops at voiceops.org" <voiceops at voiceops.org>

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Metaswitch challenging SIP BYE
by mh＠markholloway.com Sept. 30, 2011

Sept. 30, 2011

Avaya Aura platform trunked into Metaswitch. Avaya sends a BYE message towards Metaswitch and it's being challenged by Metaswitch but the Aura is ignoring the challenge. Does anyone know if there is a way to disable this on the Metaswitch platform? Is it something that is enabled by default?

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Weird Adtran TA-900 problem
by matthew＠corp.crocker.com Sept. 28, 2011

Sept. 28, 2011

Hello, I have a weird problem with one TA-900 on my switch. Other TA-900s are working fine. Inbound calls to the TA from Broadworks fail with a SIP timeout. The INVITE is hitting the TA but it isn't responding with a 100 Trying. If I force the TA to re-register the FXS ports inbound calls start working for a short period of time (couple minutes) If a FXS user makes an outbound call it works and inbound calls work for a short period of time. I have packet captures from the adtran (debug ip packet <filter> dump) showing the INVITE coming in from Broadworks (Acme SD actually) When I enable SIP debugging on the TA I don't see anything in the terminal for the INVITE, like the TA isn't treating the traffic as local When phones are working (after outbound call) I see debug traffic like I would expect. This screams of a NAT issue but the TA is directly connected with a T1 and a real IP (no NAT). The registration in the acme shows no NAT and the registration expire timer is 3600 seconds. The Adtran is configured for NAT for its internal eth0/0 port and I have one linksys phone connected internally. The linksys phone appears as behind NAT in the acme with a 30 second registration timer. The linksys phone works fine through the whole process. Anyone see any issues like this before? Firmware Version A4.08.00.E -- Matthew S. Crocker President Crocker Communications, Inc. PO BOX 710 Greenfield, MA 01302-0710 http://www.crocker.com P: 413-746-2760

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