- VoiceOps - lists.voiceops.org

Automated call completion testing?
by graham.freeman＠cernio.com June 10, 2010

June 10, 2010

Hi, folks, I'm on the hunt for an automated call completion testing solution. Criteria: Must test call completion, not just whether the SIP gateway IP is reachable and has low ping/jitter. Must be able to notify defined points of contact (ideally via email) upon reaching a set failure threshold Must be able to test legacy phone numbers (e.g. +14154622991) Must be able to test via configurable routes Should be open-source Should be compatible with Asterisk Should be something I can integrate with one or more of the following: Zenoss, Nagios, AskoziaPBX, pfSense Should be able to test both SIP URIs (e.g. sip:firstname.lastname at cernio.com) and legacy phone numbers (e.g. +14154622991). Any suggestions? thanks, Graham Freeman Cernio Technology Cooperative www.cernio.com graham.freeman at cernio.com

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DSP issues in 358 with Level 3
by bbridges＠ifbyphone.com June 9, 2010

June 9, 2010

We've seen some strange DSP / jitter issues over the last day or so in LATA 358 (Chicago) with Level 3. Anyone else? -- Brooks R. Bridges Telecommunications Manager Ifbyphone, Inc. Phone: (847) 983-3000 Fax: (847) 676-6553 bbridges at ifbyphone.com http://www.ifbyphone.com

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Number porting service.
by clecny＠gmail.com June 8, 2010

June 8, 2010

Hi, We are a CLEC and are in the process of porting about 5000 number to our network in less then one month. Does anybody know the most cost effective way to do this. Anybody aware of reasonable service providers? Any suggestions...Thanks in advance!

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360 Networks experiences
by jim＠californiatelecom.com June 8, 2010

June 8, 2010

I use them and am very pleased. They helped me through some difficult porting. I use them for my primary trunk service. I highly recommend them. Jim Gurol California Telecom Carlos Alvarez <carlos at televolve.com> wrote: >We're considering buying VoIP services from them, particularly their >"PRI replacement" product. I'd love to hear from anyone who has worked >with them on any voice products. Feel free to reply on-list or >privately as you'd prefer. > >-- >Carlos Alvarez >TelEvolve >602-889-3003 > > >_______________________________________________ >VoiceOps mailing list >VoiceOps at voiceops.org >https://puck.nether.net/mailman/listinfo/voiceops

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The IPv6 Forum Releases the IPv6 SIP Ready Logo Program
by frnkblk＠iname.com June 5, 2010

June 5, 2010

http://www.marketwatch.com/story/the-ipv6-forum-releases-the-ipv6-sip-ready- logo-program-2010-06-03?reflink=MW_news_stmp This press release talks about the IPv6 Forum releasing a test suite for testing SIP using IPv6. It's interesting because I'm not aware of much movement in the existing market space towards using IPv6 for SIP traffic, either on the consumer, business, or inter-carrier side. Frank

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360 Networks experiences
by carlos＠televolve.com June 4, 2010

June 4, 2010

We're considering buying VoIP services from them, particularly their "PRI replacement" product. I'd love to hear from anyone who has worked with them on any voice products. Feel free to reply on-list or privately as you'd prefer. -- Carlos Alvarez TelEvolve 602-889-3003

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My first post to VoiceOps :-)
by d＠d-man.org June 4, 2010

June 4, 2010

Ok, So how do you know when there's an issue? Maybe that's a more relevant question... do you test automatically somehow? - Darren Carlos Alvarez <carlos at televolve.com> wrote: Darren Schreiber wrote: > I am working on some international routing with a client and we are > having a hell of a time finding **reputable** international providers > who aren?t uber-expensive. I am about ready to setup our own routing > system and hand-pick providers. We have very low international volume also, and we are using Level 3 with mixed success. Most of the time it's decent, but we occasionally run into an issue with the smaller countries (today's problem country is Palau). Hopefully others on the list have some ideas for small users like us to find great international termination. Our initial idea is to add a couple more options and then just cycle through them if they fail, though as you state, sometimes the failure isn't apparent. -- Carlos Alvarez TelEvolve 602-889-3003 _______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops

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My first post to VoiceOps :-)
by d＠d-man.org June 3, 2010

June 3, 2010

Hi all, This is my first post to VoiceOps so hopefully my question is appropriate. I am working on some international routing with a client and we are having a hell of a time finding *reputable* international providers who aren't uber-expensive. I am about ready to setup our own routing system and hand-pick providers. We are specifically having problems with calls to France. Sometimes the calls take forever to terminate, and oddly, a couple providers seem to be connecting us with parties the caller swears they didn't call. I chalked these up as misdialed numbers/user error for a while but we just get the complaint too much, I'm starting to wonder. Anyway, any suggestions on how to achieve good quality, international VoIP termination would be helpful (suggested vendors, etc.). Our usage isn't very high but will ramp soon. Thanks much, Darren Schreiber

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Asterisk and Shoretel interop?
by jim＠californiatelecom.com June 3, 2010

June 3, 2010

We normally use Adtran CPE devices <--> asterisk for our customers to do inbound/outbound calling. I am working to configure a customer ShoreTel PBX to use our asterisk for inbound/outbound via SIP trunk so they don't have to use PRI card. Anybody have any sample configs on how this is done? Jim Gurol Engineering California Telecom 213-270-1349

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Anyone using OpenSIPS/OpenSER/Kamailio to interface with L3
by matthew＠corp.crocker.com June 3, 2010

June 3, 2010

OpenSIPS is not an SBC as the members of their mailing list will readily and loudly attest. OpenSIPS is a SIP Proxy. OpenSIPS + rtp-proxy can provide many of the functions of an SBC. Personally I wouldn't allow anything on your private VoIP LAN that I didn't have direct control over. I'm not against using OpenSIPS or Asterisk but I wouldn't let my customers manage it and have direct access to my switch. ----- Original Message ----- > From: "Robert Dawson" <robert.dawson at mindshift.com> > To: VoiceOps at voiceops.org > Sent: Thursday, June 3, 2010 11:39:42 AM > Subject: [VoiceOps] Anyone using OpenSIPS/OpenSER/Kamailio to interface with L3 > > I have a customer that I would like to setup on their own SBC to avoid > adding the overhead/licensing costs to our Acmes. > > Robert Dawson > > > _______________________________________________ > VoiceOps mailing list > VoiceOps at voiceops.org > https://puck.nether.net/mailman/listinfo/voiceops -- 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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