December 2010 - VoiceOps - lists.voiceops.org

Polycom Ghost Ringing
by clec＠cygnustel.com Jan. 3, 2011

Jan. 3, 2011

We are having the strangest issue that I have seen for some time. A customer of ours with Polycom phones (4x ip335, 2x ip550) will occasionally (maybe 1 in 50 calls) hear ringing on the line along with the other party. It has happened on both incoming and outgoing calls across apparently all of the phones. We use ip550 in our office and have never had such a problem (we run the same firmware as well, although their hardware is newer). It can be fixed if the person using the phone puts the other person on hold and then takes them off hold. They have just been doing this by double pressing hold so the other party doesn't even realize it. Call setup on these calls is all normal: invite from pbx to phone 100 trying from phone to pbx 180 ringing from phone to pbx 200 ok from phone to pbx ack from pbx to phone <start rtp> I have listened to the RTP and there is absolutely no ringing! At this point I am entirely out of ideas! Does anyone have anything to suggest or has possibly encountered this before? Thanks for any ideas or help! -Eric

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Is call forwarding in Polycom phones fundamentally broken?
by jdupuy-list＠socket.net Jan. 3, 2011

Jan. 3, 2011

I've encountered this problem in the past, many years ago, when working with Asterisk farms. I'm encountering it again now working with the Broadsoft platform: Polycom Soundpoint IP phones can do call forwarding in two ways: client-side and server-side. In client-side mode, the phone itself does the forwarding. If the phone is unreachable for any reason, the call forwarding never actually happens. That is because the call server doesn't know to forward the call and the phone isn't reachable to do the forwarding. By default, the call usually goes to VM, which is not what the user wanted. So, it works most of the time, but not predictably. So, it is broken by design. (Plus with the half-broken NAT support on the Polycoms, any forwarded calls needlessly burn two audio streams because the call is not likely to transfer off the handset. But, I'll refrain on Polycom NAT commentary for the moment...) So, we go with server-side... In server-side mode, the phone refuses to show that the phone is forwarded anywhere on the screen. It is not uncommon for the user to "wonder why they aren't getting calls" if they forget to cancel the forwarding. So, it is also broken by design. (Albeit, not as badly as client-side, since at least it will always function.) This does not appear to be a bug, since it is well documented by Polycom. The only work-around I've come up with is writing a customized hack that modifies the idle screen, but that is far less than ideal. :) In general, I really like Polycom phones, but may have no choice but to switch to cisco/aastra/etc. But, before I do, has anyone come up with a better work-around? John

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Mexico DID's
by sil＠infiltrated.net Dec. 30, 2010

Dec. 30, 2010

Hey all, while not related to operational issues, still VoIP related! Lest I forget, happy new years to everyone here, so far I seem to enjoy and appreciate being able to learn from the others on the list and share when I can. Anyhow, down to "bidness," can anyone recommend a company providing DID's in Mexico, specifically the central region (Queretaro). I'm aware that GLBX has a presence there, but does anyone know of an alternative/failover? -- =+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ J. Oquendo SGFA, SGFE, C|EH, CNDA, CHFI, OSCP, CPT "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

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Dynamic T1 Faxing over IP
by ciscoplumber＠gmail.com Dec. 28, 2010

Dec. 28, 2010

We are experiencing a fair amount of heart burn attempting to support faxing over our Dynamic T1 product. Some customers it works great and others have frequent problems. We currently support dynamic T1's using both Adtran TA900 series and Cisco IAD2400 services routers. The customers fax machines will connect to their PBX that has a PRI to an IAD/TA. We have the IAD/TA configured for SIP signaling to an Acme SBC selecting T.38 as the preferred codec with fallback to g.711 pass-through. We can force a slower speed, disable error correction, disabled VAD but in the end still have very inconsistent results. From a network perspective the QoS is matching and queuing correctly, the circuits are clean and the IP SLA gives us no reason to believe they are having any network drops or jitter. I was wondering if others who have experience with faxing over IP could share what they have learned works best or anything else that may be helpful. Any help is greatly appreciated. Thanks, Jeff

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AOC for SIP
by Kevork.Belian＠spidernet.com Dec. 28, 2010

Dec. 28, 2010

Hi folks, Does anybody know about AOC of charge implementation on SIP Proxy. What I am looking for is a SIP server that can be configured as Advice of Charge designation point and send AOC messages via SIP INFO. Astersik has this capability with Sirrix PCI cards. But I am looking for something without any card dependency. thanks Kevork Belian PrimeTel Senior Network Engineer The Maritime Centre, 141 Omonoia Avenue, CY-3506, Limassol, Cyprus e-mail: kevork at francoudi.com www.thunderworx.com<outbind://192/www.thunderworx.com> tel.: (+357) 25867136 fax: (+357) 25568131

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VoIPSwitch
by sil＠infiltrated.net Dec. 27, 2010

Dec. 27, 2010

Anyone on the list using VoIPSwitch? If so, I have maybe one or two THOUSAND questions. Alright, well one or two. -- =+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ J. Oquendo SGFA, SGFE, C|EH, CNDA, CHFI, OSCP, CPT "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

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Provisioning in Montana
by garrett＠bridgeband.net Dec. 21, 2010

Dec. 21, 2010

I am looking for someone who does provisioning in Montana for Verizon or AT&T. I am pretty sure at this time it is the same person or group until the Altell swap is complete. If anyone can get me some contact info I would be great full. You can contact me directly garrett at bridgeband.net -- Garrett Talbot Montana Opticom, LLC. Bozeman (406)586-1102 Billings (406)256-1773 garrett at bridgeband.net

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Voicemail-to-text API Service
by beckman＠angryox.com Dec. 15, 2010

Dec. 15, 2010

I'm interested in Voice-to-text services that offer an API. I've looked at a few. * Quicktate -- accurate, humans, expensive: $0.015 per word retail! * MyCaption -- I have no real data on them yet * Twilio -- Does not offer the service as an API, just within their own services; could be hacked to do so * VoiceCloud -- Have an API, pricing and quality unknown (haven't returned call/email). * Tropo -- Does not offer the service as an API, just within their own services; could be hacked to do so Is anyone using any of these or other services with great success and at affordable rates? Both Twilio and Tropo offer $0.005/minute to $0.01/minute for transcription, but I don't know about the quality yet. The cost may be enough for me to write an app that my system calls their service and plays the message for transcription. It's a hack, but if the quality is good, may be worth the effort. So what do you know? Beckman --------------------------------------------------------------------------- Peter Beckman Internet Guy beckman at angryox.com http://www.angryox.com/ ---------------------------------------------------------------------------

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[outages] VoIP - complete outage at DASH Carrier Services
by abarratt＠socket.net Dec. 15, 2010

Dec. 15, 2010

We've been looking at the possibility of using DASH for some of our 911 Trunking -- is this a common problem? Has it been resolved yet? Adam E Barratt Sales Engineer SOCKET (573) 817-0000 abarratt at socket.net 1-800-SOCKET-3 www.socket.net ********************************************************************** Privacy Notice: The information in this message is confidential and may be legally privileged. It is intended for use only by the individual or entity to which it is addressed. Access to this message by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, distribution or any action taken or omitted to be taken in reliance on it, is prohibited and may be unlawful. If you have received this e-mail in error please notify the sender immediately and delete it from your system without copying or forwarding. ********************************************************************** On 12/15/2010 11:00 AM, voiceops-request at voiceops.org wrote: > Send VoiceOps mailing list submissions to > voiceops at voiceops.org > > To subscribe or unsubscribe via the World Wide Web, visit > https://puck.nether.net/mailman/listinfo/voiceops > or, via email, send a message with subject or body 'help' to > voiceops-request at voiceops.org > > You can reach the person managing the list at > voiceops-owner at voiceops.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of VoiceOps digest..." > > > Today's Topics: > > 1. FW: [outages] VoIP - complete outage at DASH Carrier Services > (Frank Bulk - iName.com) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Tue, 14 Dec 2010 17:23:41 -0600 > From: "Frank Bulk - iName.com"<frnkblk at iname.com> > To: "VoiceOps"<voiceops at voiceops.org> > Subject: [VoiceOps] FW: [outages] VoIP - complete outage at DASH > Carrier Services > Message-ID: > <!&!AAAAAAAAAAAuAAAAAAAAAKTyXRN5/+lGvU59a+P7CFMBAN6gY+ZG84BMpVQcAbDh1IQAAAATbSgAABAAAAB0bejhCULgT4l2FNg6jueSAQAAAAA=(a)iname.com> > > Content-Type: text/plain; charset="us-ascii" > > FYI > > -----Original Message----- > From: outages-bounces at outages.org [mailto:outages-bounces at outages.org] > Sent: Tuesday, December 14, 2010 4:57 PM > To: outages at outages.org > Subject: [outages] VoIP - complete outage at DASH Carrier Services > > > From their email notification: > > "Please be advised that dash Carrier Services has a complete outage for both > inbound and outbound call routing. dash is working to resolve the issue and > will provide an update on an hourly basis or as soon as the issue is > resolved." > > Been going on for about 45 mins so far..... > > Chris > > _______________________________________________ > Outages mailing list > Outages at outages.org > https://puck.nether.net/mailman/listinfo/outages > > > > ------------------------------ > > _______________________________________________ > VoiceOps mailing list > VoiceOps at voiceops.org > https://puck.nether.net/mailman/listinfo/voiceops > > > End of VoiceOps Digest, Vol 18, Issue 6 > *************************************** >

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The romanian hackers busted
by sjur＠usken.no Dec. 15, 2010

Dec. 15, 2010

There were several mails about INVITES coming from IPs all over, but special from Romania. Action was taken by the National Organisation for preventing organized crime and terrorism which raided 42 places, 32 people busted. approx 11,5 mill euro was the loss of the hacked parties, the bad guys got about 10% of this. more info: http://blog.sipvicious.org/2010/12/11-million-euro-loss-in-voip-fraud-and.h… http://www.usken.no/2010/12/voip-hacking-is-a-big-industry-42-people-appreh… (yes, it is wrong title about Russia, should be Romania ;-) This was also the hackers who got trapped in the Honeypot in the Forensic Challenge earlier this year by the Honeynet Project. https://honeynet.org/challenges/2010_4_voip If any have more news on it or would like more info, contact me or Sandro. cheers sjur

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