August 2009 - VoiceOps - lists.voiceops.org

network jitter tools
by carlos＠race.com Sept. 1, 2009

Sept. 1, 2009

Hey guys just putting a list together of tools to diagnose network jitter what are you guys using out there? -carlos

12 11

network jitter tools
by carlos＠race.com Sept. 1, 2009

Sept. 1, 2009

I?ve heard this is a great product but holds a nice price tag to it as well. -carlos From: voiceops-bounces at voiceops.org [mailto:voiceops-bounces at voiceops.org] On Behalf Of Jason L. Nesheim Sent: Saturday, August 29, 2009 9:31 AM To: Nathan Stratton Cc: voiceops at voiceops.org Subject: Re: [VoiceOps] network jitter tools You could also take a look at the Hammer XMS. http://www.empirix.com/products/hammer_xms_service_assurance.asp -- Jason Nesheim ----- Original Message ----- From: "Nathan Stratton" <nathan at robotics.net> To: "anorexicpoodle" <anorexicpoodle at gmail.com> Cc: voiceops at voiceops.org Sent: Wednesday, August 26, 2009 7:46:59 AM Subject: Re: [VoiceOps] network jitter tools On Tue, 25 Aug 2009, anorexicpoodle wrote: > Acme radius CDR's contain RTP statistics per-call for large scale > statistics collection. Not sure if/what other SBC's make this data > available. Covergence now acme also does. -Nathan _______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops

5 5

[Fwd: Voicemail message from: "UNAVAILABLE" <sip:3073330027@xxx.xxx.xxx.xxx; isup-oli=0; user=phone>]
by abalashov＠evaristesys.com Sept. 1, 2009

Sept. 1, 2009

Anybody know who this clown is? I've been getting a lot of these. -------- Original Message -------- Subject: Voicemail message from: "UNAVAILABLE" <sip:3073330027 at xxx.xxx.xxx.xxx;isup-oli=0;user=phone> From: voicemail at sip.evaristesys.com To: abalashov at evaristesys.com "UNAVAILABLE" <sip:3073330027 at xxx.xxx.xxx.xxx;isup-oli=0;user=phone> left a voice message of 91.80 seconds duration for you. It is attached. -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775

2 1

N11 Services and Regulations for VoIP.
by drooster＠verizon.net Aug. 31, 2009

Aug. 31, 2009

An HTML attachment was scrubbed... URL: <https://puck.nether.net/pipermail/voiceops/attachments/20090828/aee11f22/at…>

2 1

T38 over Sip
by juna＠is.co.za Aug. 31, 2009

Aug. 31, 2009

Howdy Ok so im trying to t38 over sip working with no luck My test lab is as follows: Pangea box connecting to a AS5400 via PRI, which then traverses an Acme Session director, broadsoft and then exits via PGW(SS7) The sniffer capture says ?failure to train? Any ideas? -- junaid Please note: This email and its content are subject to the disclaimer as displayed at the following link http://www.is.co.za/legal/E-mail+Confidentiality+Notice+and+Disclaimer.htm. Should you not have Web access, send a mail to disclaimers at is.co.za and a copy will be emailed to you.

3 10

T38 over Sip
by KevorkB＠francoudi.com Aug. 30, 2009

Aug. 30, 2009

Disable ECM on Cisco gear (AS5400 and the exit gateway of PGW). You can disable ECM on the fax machine also. -----Original Message----- From: voiceops-bounces at voiceops.org [mailto:voiceops-bounces at voiceops.org] On Behalf Of Junaid Hattia Sent: Friday, August 28, 2009 3:03 PM To: Alex Balashov Cc: voiceops at voiceops.org Subject: Re: [VoiceOps] T38 over Sip Attached is a call flow.... The basic prolem is that a fax call cannot be completed... On 2009/08/28 1:29 PM, "Alex Balashov" <abalashov at evaristesys.com> wrote: > There are many things that could be wrong there. Problem description > is not sufficient. > > Junaid Hattia wrote: > >> Howdy >> >> Ok so im trying to t38 over sip working with no luck >> >> My test lab is as follows: >> >> Pangea box connecting to a AS5400 via PRI, which then traverses an >> Acme Session director, broadsoft and then exits via PGW(SS7) >> >> The sniffer capture says ?failure to train? >> >> >> Any ideas? >> >> -- >> junaid >> >> >> >> >> >> Please note: This email and its content are subject to the disclaimer >> as displayed at the following link >> http://www.is.co.za/legal/E-mail+Confidentiality+Notice+and+Disclaimer.htm. >> Should you not have Web access, send an email to >> mailto:disclaimers at is.co.za and a copy will be sent to you. >> >> >> --------------------------------------------------------------------- >> --- >> >> _______________________________________________ >> VoiceOps mailing list >> VoiceOps at voiceops.org >> https://puck.nether.net/mailman/listinfo/voiceops > Please note: This email and its content are subject to the disclaimer as displayed at the following link http://www.is.co.za/legal/E-mail+Confidentiality+Notice+and+Disclaimer.htm. Should you not have Web access, send a mail to disclaimers at is.co.za and a copy will be emailed to you.

6 6

AT&T/Verizon Wholesale Local Phone Service
by gkuri＠ieee.org Aug. 29, 2009

Aug. 29, 2009

Anyone have recommendations of companies that offer wholesale local phone service (traditional POTS) through AT&T and Verizon for resell? We're looking to setup specific customers with an AT&T or Verizon traditional POTS line for specific applications. I suppose we could go to AT&T or Verizon directly, but I'd really rather go through a company that already has the agreements with AT&T and Verizon in place. We want to be able to put in orders with said company for the POTS lines in the respective Verizon and AT&T territories and have the bill come to us directly (ie wholesale solution). Whether or not it matters, these traditional POTS lines do not need long distance, only a phone number with unlimited local phone service and 8xx dialing, that's it. Thanks, Gabe

2 1

FAX over VOIP networks
by kenny.sallee＠gmail.com Aug. 29, 2009

Aug. 29, 2009

So the last thread on T38 made me want to start a thread on FAX over IP via an ATA or something like the Adtran TA900 series. >From our experience, to be most successful w/ FAX over IP of any sort we've found that if you: 1.) Set baud to lowest setting fax can handle 2.) Lower print quality settings if possible 3.) Make sure to force G711 4.) Disable ECM and VAD on fax/modem tone detection And then cross your fingers fax's are successful and consistent. Anyone else have other experiences or suggestions? Kenny

7 7

Carrier choices
by sil＠infiltrated.net Aug. 28, 2009

Aug. 28, 2009

Alrighty all, I'm wondering whose been a successful, competent, reliable provider in your opinion. I don't want to ramble on about any specific company, but I have a gripe with one of my carriers. Currently I use Level3, Global and Verizon. Anyone have other reliable providers who don't make full-blown equipment swapouts without change management (reversion) plans during business hours ;) Difficult to find an all-inclusive list of carriers. I know about XO and Qwest, but anyone else using a reliable peer I'm overlooking. Off-list replies welcomed however, I wouldn't want to hear full-blown gripes. I've had enough from clients this week due to one of my carriers and their desire to yank/change equipment without 1) notifying us of a change(potential outage) not reverting/resolving fast enough. So imagine your entire VoIP ITSP infrastructure down the first time for 8+ hours, only for the same carrier to come back and do the exact same thing within 10 days for 9+ hours more. Yes sir/maam, I've had headaches this week. -- =+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ J. Oquendo SGFA, SGFE, C|EH, CNDA, CHFI, OSCP "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 227C 5D35 7DCB 0893 95AA 4771 1DCE 1FD1 5CCD 6B5E http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x5CCD6B5E

5 7

SIP-I
by russell＠rustlesolutions.ca Aug. 28, 2009

Aug. 28, 2009

Hello, I'm working on a project which requires me to deliver SIP-I signaling to my carrier when originating outbound calls. Right now I have a OpenSER Proxy which is setup as a registrar/location server, and forks calls off-net to my carrier. Initially they asked me to provide them with a JIP (*Jurisdiction* Information Parameter) which I applied to the From header. Now they are asking that I send them all of the ISUP signaling encoded as a MIME attachment in the SIP dialogs. Being that I haven't worked with SIP-I before, I'm not exactly sure where to start, I was looking at the Cisco PGW 2200 which is capable of doing the SIP<->ISUP transaction mapping, do I have any other alternatives? Thanks

3 3