Re: [asterisk-users] Movistar sip Mexico

introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Kristian Kielhofner

[asterisk-users] Somewhat-OT: Stupid NAT tricks to learn from Apple?

violations ;) More here: http://blog.krisk.org/2013/09/apples-new-facetime-sip-perspective.html As SDP bodies swell more and more can we hope to build significant support for multiplexing and deflate compression in the SIP-focused open source ecosystem? -- Kristian Kielhofner

Re: [asterisk-users] Somewhat OT: Specific SIP packets can cause ethernet controller reset

Update with a response to the statement from Intel: http://blog.krisk.org/2013/02/packets-of-death-update.html On Wed, Feb 6, 2013 at 11:08 AM, Kristian Kielhofner k...@kriskinc.com wrote: While not strictly Asterisk related this issue could certainly affect some of you: http

[asterisk-users] Somewhat OT: Specific SIP packets can cause ethernet controller reset

While not strictly Asterisk related this issue could certainly affect some of you: http://blog.krisk.org/2013/02/packets-of-death.html -- Kristian Kielhofner -- _ -- Bandwidth and Colocation Provided by http://www.api

[asterisk-users] Wireshark AMI Dissector

Hello everyone, Does anyone know of a Wireshark AMI (Asterisk Manager Interface) dissector? Decode as telnet and display filter telnet.data kind of work but TCP reassembly can't happen without a better understanding of the protocol... Thanks! -- Kristian Kielhofner

Re: [asterisk-users] directrtp with SIP + H.323

(directrtp/reinvite).  Is this possible with these to protocols? Thanks Yate claims it can do this: http://yate.null.ro/pmwiki/index.php?n=Main.H323ToSIPSignallingProxy -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com

Re: [asterisk-users] large scale paging

) ;). -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-users] Use of 603 Declined

(configuration option for domains, etc). How much work is this? -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com -- _ -- Bandwidth

Re: [asterisk-users] Use of 603 Declined

there are certainly other cases where it wouldn't fit quite as well (I haven't even looked at those involving REFER) but it looks perfect to me. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com

Re: [asterisk-users] Use of 603 Declined

certainly doesn't do that. In this scenario any forking proxy faced with a 603 coming from Asterisk has to break RFC compliance just to successfully complete the request on another host. Nasty. Are we back to the next-most-generic SIP error, 503 (as originally suggested by Alex)? -- Kristian

[asterisk-users] Use of 603 Declined

any better alternative responses I'm just bothered by the global nature of 6xx failures in the first place. Any thoughts? -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com

Re: [asterisk-users] Use of 603 Declined

can make that change if I like but... A quick poll: Who thinks Asterisk should severely limit the cases it sends 6xx responses? -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com

Re: [asterisk-users] Use of 603 Declined

certifiably unreachable except, perhaps, an invalid IP address, unresolvable host, or something of that sort? Agreed. Even then, on an incoming request, how would it know? -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http

Re: [asterisk-users] Sipgate DTMF not detected by Asterisk 1.6 [dtmfmode=rfc2833]

:101 telephone-event/8000 That's all you need to know. They are configured for RFC2833 and they're sending RFC2833. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com

Re: [asterisk-users] Sipgate DTMF not detected by Asterisk 1.6 [dtmfmode=rfc2833]

system and your carrier I can debug this further. I appreciate this is a 'how long is a piece of string question Kristian, but is there likely to be a way I can fix this? You can try the rfc2833compensate option... Other than that I can't know until I see a packet capture. -- Kristian

Re: [asterisk-users] Sipgate DTMF not detected by Asterisk 1.6 [dtmfmode=rfc2833]

to reverse engineer, not that I have the time for that. I do however, specialize in debugging DTMF. I always make time for interesting cases. I also own a voice service provider so it's unlikely I'm interested in your sipgate credentials :). Good luck. -- Kristian Kielhofner http

Re: [asterisk-users] Sipgate DTMF not detected

RFC2833. Turn on SIP debugging and look in the INVITE from the provider for telephone-event. If you see it, they're configured to use RFC2833. If they are, you need to do a packet capture or other RTP debug to see the out of band RFC2833 events. -- Kristian Kielhofner http://www.astlinux.org

Re: [asterisk-users] Sipgate DTMF not detected

). The SIP debug, however, will tell you if the remote end is configured to use RFC2833 or not. That's why I was telling you to look for telephone-event in the INVITE from your provider. Keep in mind SIP (most likely) runs over UDP between you and your provider, not TCP. -- Kristian Kielhofner http

[asterisk-users] Semi-OT: Configuring SIP trunks with Cisco UCM 7.0.

to disable this feature (3pcc). Does anyone happen to know how to disable 3pcc on Cisco Unified Communications Manager 7.0? Thanks! -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com

Re: [asterisk-users] delay in processing dtmf

On Mon, Oct 19, 2009 at 2:16 PM, Danny Nicholas da...@debsinc.com wrote: You might want to play with tonedur in dahdi.conf.  This IME effects SIP calls as well. Configuration changes in dahdi.conf do not affect SIP channels. -- Kristian Kielhofner http://www.astlinux.org http

Re: [asterisk-users] Duplicate DTMF

of relaxdtmf in sip.conf in Asterisk 1.4 or later. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com ___ -- Bandwidth and Colocation Provided by http

Re: [asterisk-users] SIP reply CALL-ID from ITSP has internal address in host part

addresses on the outside. Ouch. /O +1 for SIP TLS/SIPS. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com ___ -- Bandwidth and Colocation Provided by http

Re: [asterisk-users] VUC: RE: Friday 11th: Aswath Rao: Trapezoidal VoIP is Evil on VoIP Users Conference at Noon EDT

and disadvantages of the trapezoid although from the title I'm guessing he's going to focus on the disadvantages ;). Then again I could be completely wrong. The SIP trapezoid is real but this speculation is purely my own. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http

Re: [asterisk-users] duration of rfc2833 generated dtmf

that the duration of your DTMF key presses are too short... With that being said AFAIK there is no way to specify a minimum duration for an RFC 2833 DTMF in Asterisk on a bridged channel. -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http

[asterisk-users] SheevaPlug Development Kit

ethernet - USB 2.0 - 5 watt power usage They probably won't be shipping until late March but I thought I'd get my order in early. Of course one of my first tasks will be to get Asterisk running on it... ;) -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http

Re: [asterisk-users] GSM codec is a good choice ???

need meetme or a few other apps that essentially require G.729 transcoding you don't need a license. -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation

Re: [asterisk-users] [asterisk-dev] RFC 2833 DTMF w/ Level 3 Sonus

like an issue on the SONUS side. Anyone else have this issue? Welcome to the club! ;) I'll be blogging about this later today. Look out for that post... -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com

Re: [asterisk-users] E1 conection to a Cisco2600

28 17:32:33] WARNING[6182] chan_dahdi.c: PRI Error on span 0: We think we're the CPE, but they think they're the CPE too. Configure Asterisk for pri_net. -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com

Re: [asterisk-users] RFC -- Improving the quality of the mailinglists

On 1/27/09, Tilghman Lesher tilgh...@mail.jeffandtilghman.com wrote: I would have said Queen's English, but that evokes Freddy Mercury. ...and Freddy Mercury evokes Kevin Fleming. Perfect - we're back on topic! -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http

Re: [asterisk-users] Packet8 hacked

). -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options

Re: [asterisk-users] random Linksys question

On Thu, Jan 22, 2009 at 3:11 PM, Jeff LaCoursiere j...@jeff.net wrote: Can you configure the LAN port on the back of a 2102 to be bridged rather than routed to the WAN port? To my knowledge this is available on all Linksys ATA type devices that offer both ports. -- Kristian Kielhofner http

Re: [asterisk-users] 404 not found from one ip-adress

with the other servers IP/hostname and the ip-only context. -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk

Re: [asterisk-users] [asterisk-biz] Friday VUC 12 Noon ET with Kristian Kielhofner: Identifying Asterisk Quality Issues

the network being tested (again, usually the PSTN). I'm glad we have tools like Asterisk and ecasound that are simple enough for someone like me to slap a shell script on top of to do some pretty cool stuff! -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http

Re: [asterisk-users] Friday VUC 12 Noon ET with Kristian Kielhofner: Identifying Asterisk Quality Issues

they be IP, TDM, or any combination. In this Friday's VoIP User's Conference, Kristian Kielhofner will give an introduction to Recqual and provide several examples of its usage in real world contexts. More on Recqual here on Kristian's blog : http://tr.im/recqual2 or on Voip-Info: http

Re: [asterisk-users] why does users.conf generate SIP peer and SIP user?

has been on HBO lately. Am I correct? -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users

Re: [asterisk-users] Using Asterisk to measure call quality: Introducing Recqual

On Tue, Dec 23, 2008 at 6:31 AM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Mon, Dec 22, 2008 at 10:37:01AM -0500, Kristian Kielhofner wrote: Hey everyone, A while back I worked on a project to measure call quality. I've finally gotten around to releasing it and I'm calling

Re: [asterisk-users] Using Asterisk to measure call quality: Introducing Recqual

decisions we had to make periodic calls for at least a day to detect the majority of the bad hosts/trunks/etc. -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth

Re: [asterisk-users] Using Asterisk to measure call quality: Introducing Recqual

is only scratching the surface of what it can do! -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com

[asterisk-users] Using Asterisk to measure call quality: Introducing Recqual

this with a great deal of success but I think there is still a lot to be done. More on my blog here: http://blog.krisk.org Thoughts? -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com

Re: [asterisk-users] Application Layer Gateway for SIP protocol

connection tracking module for iptables/netfilter: http://www.calivia.com/iptables-sip-conntrack-nat -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth

Re: [asterisk-users] Linux Software to monitor quality of bandwidth for carrying voip traffic - suggestions please?

for the source but it's GPL licensed. Let your mind ponder that for a minute... There are some interesting docs, whitepapers, etc on the site (nProbe/PF_RING) if you are interested. -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http

Re: [asterisk-users] Stability unmatched!

/ -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update

Re: [asterisk-users] Stability unmatched!

On Mon, Dec 8, 2008 at 3:37 PM, Jeff LaCoursiere [EMAIL PROTECTED] wrote: On Mon, 8 Dec 2008, Kristian Kielhofner wrote: That much uptime at The Planet in Dallas? I guess you're lucky: http://www.thewhir.com/marketwatch/060208_The_Planet_Explosion_Causes_Outages.cfm http

Re: [asterisk-users] directrtpsetup without reinvite

systems. -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE

Re: [asterisk-users] changing the size of voice packets

On Mon, Nov 10, 2008 at 2:40 PM, Kristian Kielhofner [EMAIL PROTECTED] wrote: Depends? What is the status of maxptime in Asterisk? ... or the remote end, for that matter... -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http

Re: [asterisk-users] changing the size of voice packets

On Mon, Nov 10, 2008 at 2:24 PM, Alex Balashov [EMAIL PROTECTED] wrote: If the packetisation durations are different between endpoints, the SDP offer/answer should fail with a 488 Not Acceptable Here. Right? Depends? What is the status of maxptime in Asterisk? -- Kristian Kielhofner http

Re: [asterisk-users] changing the size of voice packets

. - The Genuine Asterisk Experience (TM) Yep, that's how it's supposed to work. Are you confirming our understanding of the spec or Asterisk's implementation of the spec? -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com

Re: [asterisk-users] changing the size of voice packets

On Mon, Nov 10, 2008 at 4:52 PM, Kevin P. Fleming [EMAIL PROTECTED] wrote: Kristian Kielhofner wrote: Are you confirming our understanding of the spec or Asterisk's implementation of the spec? Well the former, and I hope the latter too, since it should match the former :-) -- Kevin P

Re: [asterisk-users] [Astlinux-users] VoIP Users Conference Call Friday Nov 7 On Wideband Voice Conferencing

most conferences. Come on guys, don't let it happen to the VoIP User's Conference!! ;) -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided

Re: [asterisk-users] VoIP traffic shaping

traffic and gives the option to use HFSC or HTB. Not only do I use it myself for AstLinux and Star2Star, most of the reports I've (we've) had have been favorable. Try it out! -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com

Re: [asterisk-users] fax / t38 gateway

for the HFSC qdisc: http://astlinux.svn.sourceforge.net/viewvc/astlinux/trunk/package/iproute2/astshape -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth

Re: [asterisk-users] network design philosophy and practice

-vlan on Cisco switches. Or... you can install cdp-tools on a Linux box and have it advertise a voice vlan for you! http://gpl.internetconnection.net/ I added the voice vlan support to cdp-tools. ;) -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org

[asterisk-users] CDP (was Re: network design philosophy and practice)

that's practical) to disable it in a config file.* * Classic chicken or the egg... -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided

Re: [asterisk-users] fax / t38 gateway

really help those of us trying to make do with the internet, for example. -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-users] Fring: Open VPN client to be installed on the mobile, which mobile?

-sip.sourceforge.net/ -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE

Re: [asterisk-users] Sonicwall potentially causing long ping times to SIP phones

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! -- Kristian Kielhofner http

Re: [asterisk-users] How Secure Is Asterisk

has experimental support for TLS and I know SDES/SRTP is on the roadmap. -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-users] How Secure Is Asterisk

experimental. -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE

Re: [asterisk-users] Asterisk Problem

On 10/19/08, Ahmed torinto [EMAIL PROTECTED] wrote: After installing a new box and asterisk. i have got these errors [EMAIL PROTECTED] ~]# asterisk Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory mkdir /var/run/asterisk -- Kristian Kielhofner

Re: [asterisk-users] [Asterisk-users] asterisk +heartbeat (Wilton Helm)

need other IPs, add an alias to your bonded interface! -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-users] SER + Asterisk

On 10/17/08, Alex Balashov [EMAIL PROTECTED] wrote: SER is defunct. Kamailio / OpenSIPS (formerly OpenSER) is the thing to do. Slight clarification: Kamailio (formerly OpenSER) / OpenSIPS is the thing to do. -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http

Re: [asterisk-users] Is there a way to test SIP credentials withoutmaking a call?

Chamberlain Most implementations (including Asterisk) don't challenge OPTIONS, at least I don't think they do... -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth

Re: [asterisk-users] Asterisk SIP calls stop working having more than 300 calls (more than 600 channels)

. What distro are you on? Are you using the Asterisk startup scripts? In later versions this is done for you automatically if you are running Asterisk as root. Have a look at this: http://www.voip-info.org/wiki/view/file+descriptors -- Kristian Kielhofner http://blog.krisk.org http

Re: [asterisk-users] Asterisk SIP calls stop working having more than 300 calls (more than 600 channels)

unlimited ulimit -s 244 ulimit -l unlimited Make sure these are in your Asterisk startup scripts before Asterisk starts. -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com

Re: [asterisk-users] Inbound Calls From AS5300 Rejected with 488 Code

, the underlying problem (multipart messages) is the same. Let us know what happens. -- Kristian Kielhofner http://blog.krisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE

[asterisk-users] SIP problems?

and it's free (of course) and you can sign up for an account if you feel like helping me out... :) Thoughts? Tips? -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com

Re: [asterisk-users] Troubleshooting one-way voice... how to peek into SIP RTP?

were able to get the carrier to correct the offending pieces of equipment. I'm looking into a way to do this in real time but for now this collection of scripts works pretty well. It's not ready for release but I could get it to you shortly for some testing. -- Kristian Kielhofner http

Re: [asterisk-users] Transcoding G.729 files

. -- Kristian Kielhofner http://blog.krisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE

Re: [asterisk-users] OT: Cisco 1841 - Can it be made SIP aware?

for this router? Thanks, Bart Bart, IMNSHO, the less SIP aware the better... I have to disable SIP inspection on every IOS/PIX device I come across. Fix the one-way audio problems on your proxy, registrar, etc (in the case, Asterisk). Most SIP ALGs are broken. -- Kristian Kielhofner http

Re: [asterisk-users] OT: Cisco 1841 - Can it be made SIP aware?

or bodies including SDPs) tells me that SIP ALGs are not the best solution in most cases, certainly not long term. -- Kristian Kielhofner http://blog.krisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September

Re: [asterisk-users] SIP to IAX?

a support contract with Voice System you should use OpenSIPS. Otherwise you are free to chose either Kamailio or OpenSIPS for whatever reasons you like. There is no OpenSER anymore. If you are confused you can always just SER (the original from iptel/FOKUS) ;). -- Kristian Kielhofner http

Re: [asterisk-users] SIP to IAX?

is to use some type of VPN to traverse these firewalls/NATs. IPSEC, OpenVPN, etc. -- Kristian Kielhofner http://blog.krisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona

Re: [asterisk-users] SCCP port numbers used for audio stram?

for the audio stream? 2. Is there a way to force SCCP and the phone to use a different port range for audio? Thanks MD SCCP (like SIP, MGCP, etc) uses RTP for audio transport. You will need to modify rtp.conf to change the port range Asterisk uses. -- Kristian Kielhofner http

Re: [asterisk-users] Does X-Lite 'remember' Congestion state? (halfway OT)

but haven't provided a fix yet... If you restart the phone or wait about 10 minutes (I think) it should be able to make outbound calls again. -- Kristian Kielhofner http://blog.krisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-users] Problematic Trunk SIP: Got SIP response 405 Method not allowed

should point out that REGISTER failures would not necessarily prevent a successful call setup but because it does not support REGISTER you will have to create a trunk (or whatever their terminology is) on the Samsung to point to your Asterisk system's IP address. -- Kristian Kielhofner http

[asterisk-users] Semi-OT: ServerBeach for VoIP

of small packets! :) Other hardware specs don't matter much either. I'd rather have a Pentium 2 running on an awesome network than have an Athlon 5000 with nothing but dirty bandwidth. Any ideas? -- Kristian Kielhofner http://blog.krisk.org

Re: [asterisk-users] multiple asterisk approach

://www.asteriskcookbook.com/wiki/index.php/Main_Page -- Kristian Kielhofner http://blog.krisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users

Re: [asterisk-users] Experience with Vicidial

... ;) Again, thanks and WELL DONE! -- Kristian Kielhofner http://blog.krisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users

Re: [asterisk-users] MagicJack quality

to me why business desk phones are so expensive? I'm not knocking my friends over at Polycom, Snom, or any other manufacturer but in some cases you can buy a cheap but usable laptop for less than you can buy a phone. What gives? -- Kristian Kielhofner http://blog.krisk.org

Re: [asterisk-users] How to monitor Asterisk logs ?

: http://www.syslog.org/wiki/Syslog-ng/Syslog-ngWiki -- Kristian Kielhofner http://blog.krisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http

Re: [asterisk-users] Cisco 7940 not getting PoE from Linksys SLM224P

(it was a refurb unit from CDW) w/out having a PoE device? Thanks- Matt The switch you're using supports IEEE 802.3af standard PoE. The phone (7940) supports pre-standard Cisco in-line power. There are some various hacks for this situation scattered around the internet. -- Kristian

Re: [asterisk-users] The S word: Asterisk security

it could protect from a true DoS, it would offer some protection at the application level and that could make all the difference in some instances... As far as wider Asterisk/security issues I think J. Oquendo would be a great guest (hint, hint). -- Kristian Kielhofner NOT sent from my iPhone

Re: [asterisk-users] The S word: Asterisk security

is the same protections (script) would work for any other SIP application or network device. -- Kristian Kielhofner NOT sent from my iPhone or Blackberry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22

Re: [asterisk-users] sip extension compromised, need help blocking brute force attempts

to jump into userland too much in iptables/netfilter. Does anyone want to write a kernel module? ;) -- Kristian Kielhofner NOT sent from my iPhone or Blackberry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008

Re: [asterisk-users] sip extension compromised, need help blocking brute force attempts

and 407s, for instance) and I don't like using the string match. The all other SIP methods rule is dicey too because of things like OPTIONS, SUBSCRIBE, etc. -- Kristian Kielhofner NOT sent from my iPhone or Blackberry ___ -- Bandwidth and Colocation

Re: [asterisk-users] Asterisk 1.2 app_vxml

version of the module hoping in vain that would work, but it fails with invalid symbols, which isn't surprising. Any ideas on how I can get this to work? Be nice if i6net provided source! Doug. Doug, Use the 1.2 module and install a libstdc++-compat library. -- Kristian Kielhofner NOT sent

Re: [asterisk-users] SIP over TCP

compatibility with NAT devices (it seems some of them don't do UDP well) - Support for TLS - Support for packet fragmentation (to support large/diverse SDPs, headers, etc) I'm sure there are other ones but that's all I can think of this early on a Sunday morning... -- Kristian Kielhofner

Re: [asterisk-users] 911 via MAX TNT ??

the 484 Address Incomplete. I bet your TNT is doing something similar However: _NXXNXX _911 Would work just fine (in Asterisk). You need to figure out how to do something similar on your TNT. -- Kristian Kielhofner NOT sent from my iPhone or Blackberry

Re: [asterisk-users] E911 Recommendations?

know many people who have had good experiences with Dash911, now known as Dash Carrier Services (I believe). Good luck Doug! -- Kristian Kielhofner ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list

Re: [asterisk-users] E911 Recommendations?

On Mon, Apr 14, 2008 at 3:43 PM, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Mon, Apr 14, 2008 at 03:23:45PM -0400, Kristian Kielhofner wrote: Wow, that response was completely unnecessary. I think most people (myself included) know what he meant. Clearly, no, *I* don't. Or I wouldn't

Re: [asterisk-users] E911 Recommendations?

. We didn't like their API or they way they handled calls. I don't remember all of the details but you could certainly check out both and see what you think. -- Kristian Kielhofner ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-users] ZD Net article

On 4/11/08, Dean Collins [EMAIL PROTECTED] wrote: Lol – though nothing is going to top the Forbes article about Mark Spencer this week http://www.forbes.com/forbes/2006/0410/063.html That article is over two years old... -- Kristian Kielhofner

Re: [asterisk-users] ISPBX Announces COGOBLUE Interface and PBX Appliances

! -- Kristian Kielhofner ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] ISPBX Announces COGOBLUE Interface andPBX Appliances

(CogoBlue). Where would you be without the free software projects (Asterisk, Linux, etc) ispbx uses? Where would you be without the Asterisk community (hint - you wouldn't have a market for CogoBlue). I'm usually not one to feed the trolls but this comment is over the top. -- Kristian Kielhofner

Re: [asterisk-users] Microsoft Office Communications Server

Riddell Director Matt, I believe OCS only supports SIP over TCP. You'll either need to use Asterisk 1.6/trunk with SIP TCP or install SER/OpenSER as a UDP-TCP proxy. -- Kristian Kielhofner ___ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-users] Microsoft Office Communications Server

packet to grow larger than the smallest MTU in the path between your two endpoints it doesn't work (no fragmentation support with SIP over UDP). SIP over TCP does not have this problem. Also, you need SIP TCP support for TLS... -- Kristian Kielhofner

[asterisk-users] Maybe OT: SIP - Missing 407 messages

in a couple of places) with a description of the problem. If you think something is relevant, feel free to copy and paste it into your reply for completeness in the archives: ttph://blog.krisk.org/2008/02/missing-sip-traffic.html Thanks! -- Kristian Kielhofner

Re: [asterisk-users] Verficar VoiceMail

2008/1/14 Gilberto Nunes [EMAIL PROTECTED]: A Monday 14 January 2008 16:01:27, Shane D escreveu: Yes, i can! In fact, I really do! :-) Sorry! thanks Sorry this is an English-only list. Have you tried asteriskbrasil.org? -- Kristian Kielhofner

Re: [asterisk-users] OT: Traffic Shaping

around as much as I used to and I STILL haven't been able to really compare HTB and HFSC for VoIP traffic. What have you found? How do you like the other changes to AstShape (iptables CLASSIFY, etc)? -- Kristian Kielhofner ___ -- Bandwidth

Re: [asterisk-users] asterisk 1.4 with around 230 SIP connections

Source Team Lead Digium, Inc. Russell, What are your thoughts on SIP/RTP multicast, if any? It's been discussed before. Seems like a great solution for paging (f the phones support it). Anyone interested in a bounty? -- Kristian Kielhofner

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