January 2013 - VoiceOps - lists.voiceops.org

Conversent Communications rejects port out request for incorrect account number
by orenyny＠gmail.com Jan. 16, 2013

Jan. 16, 2013

Hi, Does anybody have a clue why all of a sudden carriers started rejecting port out requests for an incorrect account number? Specifically done by Conversent Communications, but I think that L3 and Global Crossing also started rejecting requests with this reason. Until now this type of rejection was received for cell phone numbers only. Any idea how to resolve it? Thanks, Oren

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SIPNOC 2013 starts April 22
by lindsey＠e-c-group.com Jan. 15, 2013

Jan. 15, 2013

The SIPForum conference, SIPNOC, starts earlier this year -- April 22. You guys who have opinions and painful experiences -- come and give your talk! SIPNOC is the top engineering meeting for this industry. It's not about sales pitches or vendor demos; it's about ideas and practices from the people making VoIP Networks operate. >>> mark at ecg.co +12293160013 http://ecg.co/lindsey Begin forwarded message: From: "Marc Robins" <marc.robins at sipforum.org> Subject: [Sipnocus2012-attendees] The SIP Forum Opens Call for Presentations for SIPNOC 2013 Date: January 15, 2013 15:12:32 United States Time (New York) To: <sipnocus2012-attendees at sipforum.org> Reply-To: marc.robins at sipforum.org Proposals are now being accepted for workshops, panels, speaker presentations and ?Birds of a Feather? sessions addressing the deployment of SIP in service provider environments; Early Bird registration special has been extended to February 1, 2013 The SIP Forum has announced that it has opened the official Call for Presentations for the third annual SIP Network Operators Conference (SIPNOC 2013), being held at the Hyatt Dulles Hotel in Herndon, VA, April 22 -25, 2013. SIPNOC 2013 has been expanded to a three-day educational conference, focusing on the challenges and opportunities related to the deployment of SIP-based carrier services. Organizers invite companies, executives and industry insiders to propose presentations highlighting the use of session initiation protocol (SIP) in service provider environments. Operators are encouraged to present SIP-related success stories, deployment issues and concerns in their networks, while equipment vendors and other industry stakeholders are invited to work with operators to present real-world deployment experiences. SIPNOC 2013 attendees will include telecommunications providers, major backbone operators, interconnect and wholesale solution providers, ISPs, cable operators, wireless network operators as well as large enterprises deploying major SIP initiatives. Early Bird Registration for attending SIPNOC 2013 is now open, and participants registering before February 1 save $145. Click here to register or visit http://www.regonline.com/sipnoc2013. SIPNOC 2013 Call for Presentations proposals can be submitted by visiting the SIP Forum Call for Presentations site. Possible topics can include SIP peering, SIP trunking, Emergency Services, Congestion Control, Scaling and Capacity Issues, SIP-based applications, Routing, Security, SIP-Network Operations Center Best Practices, IPv6 deployment challenges, User-agent Configuration, Standardization Issues and Progress, HD voice deployments, Video Interoperability, WebRTC and SIP, FoIP/T.38 Deployment, Testing or other issues facing SIP network operations. SIPNOC 2013 offers several different types of speaking opportunities including: ? General Session Talks: A General Session presentation should be on a topic of interest to the general SIPNOC audience, and may be up to 30 minutes long (including time for Q&A). ? General Session Panels: Panels are sessions with a moderator and a team of panelists. The panel moderator should submit an abstract on the panel topic, a list of panelists, and how the panel will be organized. Panel selection is based on the importance, originality, focus and timeliness of the topic, expertise of proposed panelists, as well as the potential for informative and controversial discussion. ? Special Workshops: The SIPNOC 2013 program has been expanded to include special workshops that run for 2-3 hours, providing time to focus in-depth on a variety of issues important to the SIPNOC community. Topics can include a review of SIP RFCs and standards development, the regulatory environment, etc. ? Research Topics: Researchers are invited to present short summaries of their work for operator feedback. Topics may include call routing, network performance, statistical measurement and analysis and protocol development and implementation. Studies presented may be works in progress. Researchers from academia, government, and industry are encouraged to present. ? BOFs: BOFs (Birds of a Feather sessions) are informal sessions on topics which are of interest to a portion of the SIPNOC community. BOFs may be held in break?out areas or in an unscheduled room. Requests for scheduled BOFs can be made at any time, including on site at the show. Acceptance notifications for speaking at SIPNOC 2013 will be sent by the Program Committee in early March. Conference registration fees will be waived for any confirmed speakers. The following is a complete list of key dates for the SIPNOC 2013 speaking program: SIPNOC 2013 Call for Presentation Key Dates Presentation Abstracts January 25, 2013 Draft Presentations Due February 15, 2013 Acceptance Decision and Notifications March 1, 2013 Draft Program Published March 11, 2013 Final Presentations Due March 25, 2013 Final Agenda Published Conference Begins April 1, 2013 April 22, 2013 The SIPNOC 2013 event website is located at www.sipnoc.org. SIPNOC 2013 Call for Presentations General Info To view the official call for presentations, which includes instructions on submitting material and specific SIPNOC policies, please visit www.sipnoc.org. To submit, visithttps://www.easychair.org/conferences/?conf=sipnoc2013. SIPNOC Early Bird Registration Info: Deadline February 1, 2013 Save $145 if you register for SIPNOC 2013 by February 1. A special early-bird registration rate of $850 ? $145 off the regular rate of $995 ? is now in effect! Register at http://www.regonline.com/sipnoc2013. SIPNOC 2013 General Show Info For more information about SIPNOC 2013, please visit www.sipnoc.org or send an email tosipnocinfo at sipforum.org. SIPNOC 2013 Sponsorship Info For information about corporate sponsorship opportunities at SIPNOC 2013, please contact Marc Robins, SIP Forum President and Managing Director, at 203-829-6307 or marc.robins ?at?sipforum.org. ************************* Marc Robins President and Managing Director SIP Forum http://www.sipforum.org Mobile: 203-829-6307 SkypeMe! marcrobins *************************

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Twitter Blacklist feed
by sil＠infiltrated.net Jan. 10, 2013

Jan. 10, 2013

So I rebuilt/redesigned/re-deployed a script to add bad hosts to a blacklist. Script is monitoring my SBCs, hosted PBXs, etc., aggregated, sorted, then reported. Tried to remove duplicate addresses. Also, because I deal with forensics and malware, I did a similar script for bad sites that are serving out malware. For VoIP attacks, one can make a script to check for VoIP based attackers and block them on the fly. E.g,: links -dump twitter.com/efensive|awk '/VoIP/' To make say an automated ipfilter rule: links -dump twitter.com/efensive |\ awk '{print "iptables -A INPUT -s "$1" -j DROP"}' |sort -u|\ sh Same goes for any other style rule (ASA, PIX, ScreenOS on the command line) You get the point. Enjoy. (Cross posted to Voice Ops) -- =+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ J. Oquendo SGFA, SGFE, C|EH, CNDA, CHFI, OSCP, CPT, RWSP, GREM "Where ignorance is our master, there is no possibility of real peace" - Dalai Lama 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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by naushit＠yahoo.com Jan. 10, 2013

Jan. 10, 2013

http://whaouh.com/market/inMaking%20money%20from%20home%20has%20never%20bee…!

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Twitter blacklist feed
by joquendo＠e-fensive.net Jan. 9, 2013

Jan. 9, 2013

So I rebuilt/redesigned/re-deployed a script to add bad hosts to a blacklist. Script is monitoring my SBCs, hosted PBXs, etc., aggregated, sorted, then reported. Tried to remove duplicate addresses. Also, because I deal with forensics and malware, I did a similar script for bad sites that are serving out malware. For VoIP attacks, one can make a script to check for VoIP based attackers and block them on the fly. E.g,: links -dump twitter.com/efensive|awk '/VoIP/' To make say an automated ipfilter rule: links -dump twitter.com/efensive |\ awk '{print "iptables -A INPUT -s "$1" -j DROP"}' |sort -u|\ sh You get the point. Enjoy. (Cross posted to Voice Ops) -- =+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ J. Oquendo SGFA, SGFE, C|EH, CNDA, CHFI, OSCP, CPT, RWSP, GREM "Where ignorance is our master, there is no possibility of real peace" - Dalai Lama 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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CTI Group call recording
by David.Hiers＠adp.com Jan. 8, 2013

Jan. 8, 2013

Anyone out there using CTI for call recording? Thanks, David This message and any attachments are intended only for the use of the addressee and may contain information that is privileged and confidential. If the reader of the message is not the intended recipient or an authorized representative of the intended recipient, you are hereby notified that any dissemination of this communication is strictly prohibited. If you have received this communication in error, please notify us immediately by e-mail and delete the message and any attachments from your system.

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Issue with Adtran TA9xx/Broadworks/Session Audit Timers
by RDawson＠alliedtelecom.net Jan. 7, 2013

Jan. 7, 2013

Having an issue where occasionally a TA9xx doesn?t see an ack for a session audit timer invite event and will then reply with a 500 "re-invite inside a re-invite" error on the next invite. I am investigating the possibility of a network side issue now but wondering if anyone else has dealt with this? Thanks, Rob

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VoIP on 10mb Cat 3 switched network
by carlos＠televolve.com Jan. 6, 2013

Jan. 6, 2013

Has anyone had experience successfully running VoIP over 10mb Cat 3 cabling? A new customer has asked, because that's what they have in place. I'm not interested in an unreliable solution just to save a few bucks, but in theory it should work reliably to have a dedicated 10mb to each phone and then gigabit out of the main switches. -- Carlos Alvarez TelEvolve 602-889-3003

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IAD and SIP PBX Hardening Guidelines
by lindsey＠e-c-group.com Jan. 5, 2013

Jan. 5, 2013

Are there any published hardening guidelines or howtos on IADs, SIP PBXs, or and any other SIP devices on the public Internet? If not, we should put some together. For IADs, the list might start with some of the same rules you'd use on any other router. But for IADs or SIP PBXs, it seems like these two are pretty important: Restricting remote login; Restricting SIP. (1) Restrict telnet, ssh, snmp to only Administrator locations, by IP. Many of the recent attacks happening at service providers are IAD or SIP PBX compromises. These devices are often on the public Internet, and they have poor passwords. BadGuy just scans for telnet-accessible devices, logs in, and then sets up the IAD to route calls. Or he may just simply steal the SIP credentials. I can't think of good reasons that these IADs should allow telnet, ssh, or snmp from the public Internet at large. Of course, the administrators of the device (often the voice SP itself) needs to be able to login. As an alternative, you might just ensure that strong passwords are used. Lots of folks are moving to RADIUS authentication for IADs, on the hopes that they'll use better usernames and passwords. (2) Restrict SIP & RTP to only the Service Provider's SBCs, by IP. On some IADs, you can send SIP to the device's SIP stack just by sending it SIP to port UDP/5060. A lot of IADs are CPU-poor anyway, so giving them extra SIP parsing isn't going to help. In fact, just sending in bogus SIP can be enough to DoS the device. But in our world of (a) buffer overflows and code injection and (b) complex parsing of SIP, it seems intrinsically risky to allow BadGuy to send SIP in to a CPE. It seems very likely that a determined attacker could use specially-crafted SIP messages to cause a SIP IAD or phone to do anything they want: -- Make fraudulent calls -- Play advertisements -- Send a copy of the RTP to a mysterious destination on the Internet Some of the same risks are present with RTP. But fortunately, RTP is simple enough that hardening the implementation may be easier. And often RTP is handed directly to a DSP, where (I would guess) the opportunities for code injection are less interesting. But there's a real downside here: What if you need to manage SIP traffic, and allow IADs to register to some other location? If you're doing it RFC3263 style, then you might just be using DNS to route IADs and PBXs to the appropriate SBC. If you need to add additional SBCs, then you have to be sure the IAD will have ACLs that allow traffic from those other SBCs in advance. Finally: What about the poor souls who put SIP phones directly on the Public Internet? Maybe you can change the administrator password, and make it harder to login and extract the SIP authentication credentials. But beyond that, SIP phones don't typically have firewall-type features. I think the better bet here is to put them behind a stateful firewall/NAT device. Any other opinions? >>> mark at ecg.co +12293160013 http://ecg.co/lindsey

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