June 2010 - VoiceOps - lists.voiceops.org

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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Anyone using OpenSIPS/OpenSER/Kamailio to interface with L3
by robert.dawson＠mindshift.com June 3, 2010

June 3, 2010

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

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A challenge with real VoIP attacks
by sjur＠usken.no June 1, 2010

June 1, 2010

hi, For those investigating VoIP attacks and scannings, the Honeynet Project released their VoIP challenge today! It is real attacks and you have to figure out what has happened. It is a good way to learn more about SIP security https://honeynet.org/challenges/2010_4_voip The best answers will win prizes! cheers sjur (and sorry for spamming this mailinglist, but there has been talks about such attacks here... and this can be very helpful!)

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