March 2012 - VoiceOps - lists.voiceops.org

TCPDump on an Adtran TA908E
by zak＠simplesignal.com Nov. 9, 2012

Nov. 9, 2012

Does anyone know the TCP Dump commands for the Adtran TA908e CLI? *Thanks-* Zak Rupas Tier 3 Engineer Support: 303-242-8606 option 1 Like SimpleSignal on *Facebook <http://www.facebook.com/SimpleSignal> !* SimpleSignal Inc. 3600 S. Yosemite Street Suite 150 Denver, CO 80237 [image: Description: Description: cid:image001.png at 01CBDE2D.5E1CC730]

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SIPRoutes
by BrandonB＠netins.com July 6, 2012

July 6, 2012

Does anyone have any experience with SIPRoutes (siproutes.com)? It seems too good to be true, so of course I'm worried. Feel free to contact me offline or on. Thanks, --- Brandon P. Buckner Switching Technician / VoIP Admin Iowa Network Services brandonb at netins.com<mailto:brandonb at netins.com>

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Broadworks Customer Portal
by RDawson＠alliedtelecom.net April 27, 2012

April 27, 2012

Looking to see what solutions are popular, particularly interested if anyone has done any customization of the native portal and if there are any recommendations for developers specializing in that type of work. Also interested to hear experiences with Loki. Thanks, Rob

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Broadworks to Genband C15 Integration
by telephonetoughguy83＠gmail.com March 28, 2012

March 28, 2012

Does anyone have experience integrating a Genband C15 (formerly Nortel CS1500) with Broadworks? There is a partner config guide from Broadworks in regards to a CS2000, but nothing on C15. I'm trying to use a partnering Telco's C15 as a routing NE, but I'm having signalling issues for both incoming and outgoing calls. On incoming calls, the INVITE message I receive on my AS looks standard except for a few entries in the TO header (there is NPDI and CIC information in the TO header that I don't usually see in INVITE messages). My AS responds "604 Not Found Anywhere". On outbound calls, my AS sends and INVITE to the C15. Like any other INVITE sent by the AS, the source port is an ephemeral port, like 35438, and the destination port is 5060. For some reason, the TRYING message sent back by the C15 has the destination port set as 35438, instead of 5060. Any insight you could provide would be appreciated.

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Hiring voip engineer in Philadelphia area
by voiceops-01＠jeremykister.com March 28, 2012

March 28, 2012

If you've ever set up an Acme SBC or are very familiar with how to manage it's capabilities, I have a full time position that I'd like you to apply for. Feel free to email me for details or just send me your resume in pdf/text format. -- Jeremy Kister http://jeremy.kister.net./

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Experience with Innomedia MTA 6328 series
by RDawson＠alliedtelecom.net March 28, 2012

March 28, 2012

Anyone working with the Innomedia MTA 6328s? I purchased two for testing and had some mixed results, one was an out of box failure. The other unit seemed to work well enough. Just wanted to get some opinions from anyone with substantial numbers deployed. [cid:image001.png at 01CC5068.C9623860] Robert Dawson Voice Systems Engineer (o) 202-349-0477 | (f) 202-465-4165 rdawson at alliedtelecom.net<mailto:rdawson at alliedtelecom.net> | www.alliedtelecom.net<http://www.alliedtelecom.net/> follow us: [cid:image002.png at 01CC5068.C9623860] facebook<http://www.facebook.com/alliedtelecom> | [cid:image003.png at 01CC5068.C9623860] LinkedIn<http://www.linkedin.com/company/allied-telecom-group> | [cid:image004.png at 01CC5068.C9623860] twitter<http://twitter.com/#!/allied_telecom> 1120 20th Street, NW, Suite 500-South, Washington, DC 20036

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Broadworks: Preserve sip call-ID
by abalashov＠evaristesys.com March 23, 2012

March 23, 2012

INSIPID? Nice! Made my day, and it's only 03:30. But that's going to be hard to top. -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Atlanta, GA 30030 Tel: +1-678-954-0671 Web: http://www.evaristesys.com/, http://www.alexbalashov.com Hadriel Kaplan <HKaplan at acmepacket.com> wrote: >There's good news on that front - >We've finally got a new IETF Working Group created for just that purpose: the "INtermediary-safe SIP session ID" Working Group, a.k.a. INSIPID. > >It's first meeting is in just a few days: this Monday, March 26th, at the IETF meeting. > >The meeting's in France, but the IETF has free remote participation - you can listen to the audio broadcast and be in a jabber chat room for the working group meeting. And if you want to say something at the microphone, you just type what you want said into the jabber room and someone will say it for you in the physical room. >All of that you can do remotely for free, without having to register, be a member, or sign up for anything. > >The current agenda is here: >http://www.ietf.org/proceedings/83/agenda/agenda-83-insipid.txt >The meeting's only an hour long, at 9:10-10:10am Eastern Daylight Time US. > >To join the audio and jabber session, go to the following page, search for "insipid", and click the little speaker icon for an audio stream, and the jabber lightbulb icon for a jabber URL. >http://tools.ietf.org/agenda/83/ >Note you do need a jabber client to use that, but you can get them for free on all platforms; go to http://www.ietf.org/jabber/ for info on that. > >I urge anyone who cares about the issue to please join the meeting. The more feedback we get, positive or negative, the faster we'll get somewhere. (well, I suppose if you say the whole idea is useless, then we won't get anywhere real fast :) > >Also, there's a new Working Group mailing list you can join, where the real action happens, here: >https://www.ietf.org/mailman/listinfo/insipid >Again, it's open to anyone who wants to join it. > >-hadriel > > >On Mar 17, 2012, at 9:23 PM, Howard Hart wrote: > >They've been trying to push through a Session-id spec in the SIP Dispatch IETF working group for some time. This is a globally unique identifier that is carried across B2BUA's unmolested. Seems to always fade away, but if it were actually ever passed as an RFC, you could go to your vendor and ask them to implement it with some level of confidence they'd actually do it in a way that's compatible with all the other vendors. > >The last attempt was http://tools.ietf.org/html/draft-kaplan-sip-session-id-03, expired March 2011. Some talk lately in the IETF of reviving it or a slight permutation, though nothing specific that I'm aware of. > >Howard Hart > > >On 03/15/2012 01:20 PM, Keith Croxford wrote: >I'm matching by multiple headers (to, from, pai, others) and these are nested within a specific block of time. It's basically a few modifications from the default call merging algorithm. I can't show my exact modifications, but that should lead you in the proper path. > >This works OK other than in one scenario. > >Imagine that 4801001234 at domain1 makes a call. Within this set time window 1234 at domain2 initiates a call as well. Even though I am set to match on the last 6 characters of the caller/callee, Palladion is merging the calls from 1234 at domain2 and 4801001234 at domain1. > >I'm still working with IPTEGO to get a few of these bugs worked out. If I knew how to (((write code in LISP)) it would be much easier accomplish.) > >Keith > > >_______________________________________________ >VoiceOps mailing list >VoiceOps at voiceops.org >https://puck.nether.net/mailman/listinfo/voiceops

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Broadworks: Preserve sip call-ID
by rakeshunni＠gmail.com March 23, 2012

March 23, 2012

Hi guys, We are running r17 at the moment and currently testing a monitoring platform for both signalling and media. They use sip callID to identify call legs to merge and i would like to have all call legs on one call (from the originating access side to where we send the traffic to for PSTN). Since Broadworks act as a B2BUA it rewrites the originating callID to BW-xxxxx. I've made changes on the SBC to preserve the callID but now need Broadworks to preserve it. Is it possible on Broadworks to transparent proxy the callID from the access side to the terminating side ? Thanks Rakesh

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Looking for number for AT&T translations department
by jason＠thebaughers.com March 22, 2012

March 22, 2012

We're an ILEC in West-Central Illinois having trouble originating 800-number calls to CIC 288 (AT&T).Our Central Office has been going around in circles trying to open a trouble with AT&T's translations department. We keep getting the runaround, passed back and forth from one group to another. Does anyone have a good contact number? Thanks

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Latency info using RTCP from SIP end-points
by feby.francis＠crosstel.com March 19, 2012

March 19, 2012

Hello All-, Has anyone played with RTCP parameter on the Linksys phones. I am not able to capture packets with Latency info from them. Thanks, Feby Francis

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