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Mitel 3300 - NAT - Media Issue
by RDawson＠alliedtelecom.net April 4, 2012

April 4, 2012

Working with someone on SIP trunks for a Mitel 3300. The Mitel is sitting behind a NAT on a Cisco 3700 (i.e. no fixup/ALG) and inbound media is failing because the phones private IPs are being targeted directly in the SDP. Even if I force all media back to the NAT address it still fails because there is no translation for the phones. Anyone know if there is a way to force the Mitel to proxy the media or have any insight into getting around this shy of putting a proxy at the customer site(which is what we should probably do anyway)? Thanks, Rob

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Optimum Online's "Fuzzy" VoIP Routing
by joquendo＠e-fensive.net April 2, 2012

April 2, 2012

Anyone else seeing something wrong with this picture? I have a client using Optimum Online as their provider for business VoIP (don't ask), and they're complaining of issues. Aside from them losing registration due to latency (they disconnect frequently), I decided to see if there was anything on my end I may have missed. On this one proxy they're connected to, there are I don't even know how many clients, yet they're the only ones that keep dropping. I know the deductive reasoning here (client's end) just curious how many others have: 1) had issues dealing with Optimum Online 2) seen port based routing on this level out of Optimum [joquendo at fbsd /home/joquendo]# head -n 13 opt-* ==> opt-80 <== tcptraceroute to port 80 6 4.68.63.157 (4.68.63.157) 12.682 ms 9.610 ms 9.953 ms 7 CSC-HOLDING.edge4.NewYork1.Level3.net (4.28.130.62) 14.091 ms 9.132 ms 9.982 ms 8 * * * 9 * * * 10 ool-4353e68e.dyn.optonline.net (67.83.230.142) 6.622 ms 9.577 ms 9.948 ms 11 * * * ==> opt-5060 <== tcptraceroute to port 5060 6 4.68.63.157 (4.68.63.157) 9.960 ms 9.335 ms 9.922 ms 7 CSC-HOLDING.edge4.NewYork1.Level3.net (4.28.130.62) 10.007 ms 9.284 ms 9.940 ms 8 * * * 9 * * * 10 dstswr2-ge3-16.rh.nrwlct.cv.net (67.83.230.134) 6.468 ms 9.886 ms 9.958 ms 11 * * * ==> opt-8080 <== 6 4.68.63.157 (4.68.63.157) 9.342 ms 9.306 ms 10.003 ms 7 CSC-HOLDING.edge4.NewYork1.Level3.net (4.28.130.126) 9.963 ms 9.075 ms 9.987 ms 8 * * * 9 * * * 10 ool-4353e68e.dyn.optonline.net (67.83.230.142) 6.483 ms 9.734 ms 9.998 ms 11 * * * ==> opt-3128 <== 6 4.68.63.157 (4.68.63.157) 4.562 ms 3.935 ms 3.933 ms 7 CSC-HOLDING.edge4.NewYork1.Level3.net (4.28.130.126) 5.955 ms CSC-HOLDING.edge4.NewYork1.Level3.net (4.28.130.10) 11.422 ms 10.366 ms 8 * * * 9 * * * 10 ool-4353e68e.dyn.optonline.net (67.83.230.142) 35.487 ms 28.940 ms 28.441 ms 11 * * * ------------ To make sure it wasn't my machine, jumped on a Linux machine: myvoip-proxy:~# traceroute -p 5060 173.220.xx.xx traceroute to 173.220.xx.xx (173.220.xx.xx), 30 hops max, 40 byte packets 5 64.215.195.230 (64.215.195.230) 3.083 ms 3.330 ms 2.984 ms 6 CSC-HOLDING.edge4.NewYork1.Level3.net (4.28.130.126) 5.004 ms CSC-HOLDING.edge4.NewYork1.Level3.net (4.28.130.62) 4.461 ms CSC-HOLDING.edge4.NewYork1.Level3.net (4.28.130.126) 4.830 ms 7 * * * 8 * * * 9 ool-4353e68e.dyn.optonline.net (67.83.230.142) 6.021 ms 5.850 ms 5.857 ms 10 * * * 11 * * * myvoip-proxy:~# traceroute -p 80 173.220.xx.xx traceroute to 173.220.xx.xx (173.220.xx.xx), 30 hops max, 40 byte packets 5 64.215.195.230 (64.215.195.230) 3.110 ms 3.236 ms 2.986 ms 6 CSC-HOLDING.edge4.NewYork1.Level3.net (4.28.130.126) 4.727 ms CSC-HOLDING.edge4.NewYork1.Level3.net (4.28.130.62) 5.358 ms CSC-HOLDING.edge4.NewYork1.Level3.net (4.28.130.10) 6.252 ms 7 * * * 8 * * * 9 ool-4353e68a.dyn.optonline.net (67.83.230.138) 6.027 ms 5.870 ms 5.858 ms 10 * * * 11 * * * Whenever I send traffic to port 5060 it ALWAYS takes another route. Wondering if Optimum is filtering "reverse" QoS - "no one does VoIP like we do!" -- =+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ J. Oquendo SGFA, SGFE, C|EH, CNDA, CHFI, OSCP, CPT, RWSP, GREM "It takes 20 years to build a reputation and five minutes to ruin it. If you think about that, you'll do things differently." - Warren Buffett 42B0 5A53 6505 6638 44BB 3943 2BF7 D83F 210A 95AF http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x2BF7D83F210A95AF

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Broadworks to Genband C15 Integration
by telephonetoughguy83＠gmail.com March 29, 2012

March 29, 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 23, 2012

March 23, 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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Question about PIC codes
by RDawson＠alliedtelecom.net March 15, 2012

March 15, 2012

We have recently received a few requests to PIC their LD. I honestly don't even think that in our current configuration we are capable of doing PICs - just curious if anyone has any experience/pertinent knowledge related to the topic? Thanks, Rob

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