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Hi All, Until now I've only used IAX2 to connect to ITSPs. I've been toying with a SIP connection to Gizmo Project, but not yet successfully. It brings to mind a question. At what point does it make sense to consider a SIP-aware firewall such as those from Ingate? I'd hate to move away from my m0n0wall, which is open source, easy to manage and has served me brilliantly for two years. Thanks, Michael Graves -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist www.pixelpower.com Pixel Power...

...n does this "passthru class" lend bandwidth compared to other classes? And for 3 I have no idea if it''s possible using only HTB? Setting rate to 0 doesn''t work. //Marcus -- ---------------------------------------+-------------------------- Marcus Sundberg <marcus@ingate.com> | Firewalls with SIP & NAT Firewall Developer, Ingate Systems AB | http://www.ingate.com/ _______________________________________________ LARTC mailing list / LARTC@mailman.ds9a.nl http://mailman.ds9a.nl/mailman/listinfo/lartc HOWTO: http://lartc.org/

Is there anything special that anyone here has had to do to get an Aastra phone (on the Internet) to talk to Asterisk behind a PIX firewall? Ports 10000-20000 UDP are open on the PIX and forwarding to the Asterisk server. The Asterisk server's RTP.CONF is set to use 10000-20000. The phone registers, and will place AND receive calls, however, no audio is passed. The phone is an Aastra

Guys. Im writing this because Ive checked the wiki, Xten website and read a lot of docs and still cant figure out a way around the NAT issues. Maybe somebody else can give me some ideas from a fresh perpective. My test setup is this: Asterisk -> 2wire homeportal Firewall -> internet Computer with Xten eyebeam The asterisk box and the computer with xten beam are behind the same

Hello all, I got this problem while trying to shape traffic with iptables MARK and HTB. MARK: targinfosize 8 != 4 --set-mark gives "invalid argument" error message. Kernel version is 2.4.29 (some patches from patch o matic applied) Iptables version 1.3.4 Intel x86 architecture. I saw this problem discussed in a few places, but the discussions didn''t come to a conclusion

I cannot get incoming calls to sip phones behind a PIX to work, outgoing is fine. Asterisk (Public IP) --> Internet --> PIX (NAT) --> Sip Phones I have tried no fixup protocol sip, I have punched a hole in the Pix allowing anything from the Asterisk box into the network, still no incoming. I have done all the Wiki suggests in regarding to NAT. Is their a trick getting the

Hello, I'm searching for a router for our company. Does anybody has a suggestion for a router with a SIP Application Layer Gateway and good working QoS (Upstream AND Downstream). Regards Bastian

Hi, Our guest today is Steven Johnson, President of Ingate Systems. We'll talk with him about the changing role of Session Border Controllers and E-SBC and how this hardware facilitates the use of SIP in "difficult" conditions, about general SIP security considerations, and why you might need such devices on your network. VUC is every Friday...

...out SonicWalls when it comes to breaking SIP traffic. = Most of the "anything but simple PAT" devices I've seen that implement = any SIP specific fixups usually end up breaking something along the = line. Unless the product is from a company where SIP is their core = competency (like Ingate, or /maybe/ Cisco) it's best to stay away and/or = disable the SIP specific fixups wherever possible. I'm looking forward = to the day when SIP-TLS is the norm and these devices have no idea what = kind of traffic is flowing through them!=20 - I sympathize, especially since a client of mine...

...ndors if they attend SIPit and encourage them to participate - If you have collegues that work with SIP development, please forward this mail to them! Short facts: - Date: May 17-21 2010 (very beautiful season in Sweden!) - Location: Kista, Stockholm, Sweden - Host: Edvina and Tandberg - Sponsors: Ingate, Intertex, .se, Telio, Snom We are currently working to set the price and to be able open for registration. SIPit 26 has a Facebook event page at http://www.facebook.com/event.php?eid=340634688354 and a Twitter stream: http://twitter.com/sipit26 where you will get updates and be able to find lin...

I have just finished compiling and installing Asterisk on a test Debian system. All is working well. We are now attempting to get remote offices to test the system I have installed both a SIP and an IAX client at a remote office. Then I connect to our office via Microsoft ISA firewall and the Windows XP VPN client. Neither of the softphones will connect. On the IAX softphone I just get a ringtone

Hi, I'm having an issue where some phones behind a sonicwall are auto-congesting. The status on "sip show peer" shows ping times anywhere from 80ms all the way up to 1100ms. PCs behind the same firewall have a ping time of about 30ms to the PBX itself. Does anyone know if the sonicwall is inserting delay into the SIP signaling path and lagging the OPTIONS messages for qualify?

...ocalnet= networks in sip.conf and those addresses will be excluded from the outbound proxy setting. This patch has been tested with a number of outbound proxies, but may still need some testing. It is really useful if you are using a SIP proxy as a NAT traversal SIP server, like the equipment from Ingate and Intertex. Please test this, and add your comments to the bug tracker! * SIP Outbound Proxy for Asterisk CVS head: http://bugs.digium.com/bug_view_page.php?bug_id=0002859 *** Read the daily news channel ------------------------------- There is a new news source for Asterisk news on the n...

Should the following setup work? SIP UA---NAT---Internet---NAT---SIP UA If both UA's support STUN and report the external IP address in the SIP packet.. I am trying to get away from using canreinvite=no so that traffic can go directly between the UA's and not via the central server but I can't seem to get it to work.. Has anyone set this up and can give me some pointers??