RE: [Asterisk-Users] FCC Rules VoIP Must Be Tappable
In trading, we have VOIP sniffers that record ALL calls... Nice systems makes one I believe. -c -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Florin Andrei Sent: Wednesday, August 04, 2004 7:02 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] FCC Rules VoIP Must Be Tappable http://yro.slashdot.org/article.pl?sid=04/08/04/2212251tid=158tid=95tid=1 03 Probably some of you already saw this. Now, beyond discussions regarding the legitimacy of such a ruling (whether they have the legal, moral or whatever right to enforce it), there's the technical aspect. Suppose i provide VoIP services using Asterisk, and i fall under the incidence of the FCC ruling and i have to provide a tap to the guys in the black helicopters. What are the guidelines, what should i do to ensure i won't get spanked because i obstructed the justice or some such. More precisely, what config bits must be put in place to make sure there's always an easy way, with Asterisk, to tap into arbitrary calls? -- Florin Andrei http://florin.myip.org/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FCC Rules VoIP Must Be Tappable
Yep.. using the SPAN port now. I imagine there will be ways to get around this nuisance, by encrypting the traffic, or something.. -c -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Leo Ann Boon Sent: Wednesday, August 04, 2004 9:11 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] FCC Rules VoIP Must Be Tappable AFAIK, these records (Nice, Witness) only work on a LAN (through passive sniffing on a hub or using a span port) or through a conference server. Either way, it'll require the RTP to be proxied through the provider. In reality, the problem might not be as bad, many SIP-based providers are already using ALGs (Application Layer Gateway) to help NAT traversal. These ALGs will proxied RTP if the phone is behind NAT. In theory, it should be possible to record the streams from the ALG. C. Johnson wrote: In trading, we have VOIP sniffers that record ALL calls... Nice systems makes one I believe. -c -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Florin Andrei Sent: Wednesday, August 04, 2004 7:02 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] FCC Rules VoIP Must Be Tappable http://yro.slashdot.org/article.pl?sid=04/08/04/2212251tid=158tid=95; tid=1 03 Probably some of you already saw this. Now, beyond discussions regarding the legitimacy of such a ruling (whether they have the legal, moral or whatever right to enforce it), there's the technical aspect. Suppose i provide VoIP services using Asterisk, and i fall under the incidence of the FCC ruling and i have to provide a tap to the guys in the black helicopters. What are the guidelines, what should i do to ensure i won't get spanked because i obstructed the justice or some such. More precisely, what config bits must be put in place to make sure there's always an easy way, with Asterisk, to tap into arbitrary calls? -- Florin Andrei http://florin.myip.org/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] BT is moving to IP ONLY
Their Syntegra trading turrets already have begun the migration. Now, if I can get my hands on one and getting it to work with *, I'll be set. -cj -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Senad Jordanovic Sent: Thursday, June 10, 2004 12:18 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] BT is moving to IP ONLY Hi, all This is certainly very good news! http://www.neowin.net/comments.php?id=21119category=main ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Network Sniffing Calls for recording
It's in the Bible... http://www.voip-info.org/tiki-index.php Still, I'd be interested to find a solution for recording via sniffing over the net. My solution was to have my asterisk pbx's monitor directory connect to my recording server via a nfs mount, but that's a raggedy ann way of doing it. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of hank smith Sent: Monday, June 07, 2004 6:10 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Network Sniffing Calls for recording how can you record calls with asterisk? I didn't even know this was possible can some one point me to a url for info on this? - Original Message - From: lists [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, June 07, 2004 1:30 PM Subject: [Asterisk-Users] Network Sniffing Calls for recording Ok assuming I don't want to record calls using * but instead want a dedicated server that listens to a mirror port and records calls. Is there a cheap software package out there for doing this for mgcp/sccp? I know if evern cut over to * there is a way but I doubt I will even cut 100% over to * so I was wonder what the list has heard of for call recording via sniffing my gates. I know there are some out there but $100k for 40 users is to high for my blood. Offlist is fine for all flames and answers since this is a bit off topic [EMAIL PROTECTED] OK it's a Monday when it takes 5 tries to get a email to the right list from the right account. Either that or someone switched the coffee pot to decaf again. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIPP Load testing
No, I have not updated since yesterday.. The last * update I did was in March. -cj -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Duane Sent: Monday, May 31, 2004 1:44 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SIPP Load testing C. Johnson wrote: Apparently I'm missing something... Anyone seen this before using SIPP? You updated your asterisk version since yesterday? if so it's the same bug I'm currently trying to work out more details on... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIPP Load testing
Howdy- Trying to run some load tests here using the SIPP tool. Anyone here familiar with creating a custom script to just dial one extension for load testing purposes? I created a uac.xml file, but when I run the load test, I see this when I turn on sip debugging in asterisk: Sip read: SIP/2.0 407 Proxy Authentication Required Apparently I'm missing something... Anyone seen this before using SIPP? Thx, Cedrick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Web service to start a conference and voice mail
is it publicly available? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Serge Mankovski Sent: Tuesday, March 16, 2004 10:45 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Web service to start a conference and voice mail Hi I have written a web service that starts a conference call and then monitors call progress on the manager interface. It works nicely until conference in a voice mail system. It would be better if I could intercept the fact that the answering side is a voice mail and not to conference it in. The same thing might happen with fax or modem, but I have not seen that happening yet. Is there a way to identify when if there is a voice mail machine of the line, fax or a modem? Thank you Serge __ ___ Get business advice and resources to improve your work life, from bCentral. http://special.msn.com/bcentral/loudclear.armx ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk- users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk- users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk PLAR?
Hello- I asked this question a LONG time ago (when I first got started with *), but seem to have lost the answer in between my multiple Windows XP repairs. Has anyone experimented with or achieved PLAR (private line auto ringdown) capability with asterisk? thx, cedrick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cannot find -lXext when building * ?
Howdy.. I am building * on a barebones server, running just the minimum config (no X, etc..) when I build, I get this error, and I'm trying to track it down. Has anyone ran into this before or have a general idea? gcc -shared -Xlinker -x -o pbx_gtkconsole.so pbx_gtkconsole.o `gtk-config --libs gthread` /usr/i386-slackware-linux/bin/ld: cannot find -lXext collect2: ld returned 1 exit status make[1]: *** [pbx_gtkconsole.so] Error 1 Thx, cj ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Has Nufone gone belly-up
I don't think Sathya is ripping Nufone per-se, just trying to figure out what is going on. I'm sure you would be doing the same thing IF you did not get a reply, and did not know where to reach him. Maybe Sathya does NOT know about the chatroom. Lighten up. And I agree, if there is a problem, let it be known to the community. Just don't sweep it under the rug as Roy put it.. -cj -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Roy Sent: Saturday, January 24, 2004 3:23 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Has Nufone gone belly-up I want to hear about problems with VOIP vendors. Sweeping them under the rug isn't going to help. If its a valid problem please post it. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Frankie Gravato Sent: Saturday, January 24, 2004 11:37 AM To: Sathya Subject: Re: [Asterisk-Users] Has Nufone gone belly-up Hello Sathya, Saturday, January 24, 2004, 12:25:43 PM, you wrote: S Folks, S I've ordered a new account from Nufone last month. Transferred money S to Nufone through their paypal account. I had communication with S Nufone sales S up until two weeks back. Since then there were no replies to my emails. S I am afraid with this kind of unresponsiveness how one would run a reliable S service with this company. Have no bad feeling with Jeremy as the S author of S widely used h323 channel, but my concern is about the company NuFone. S Lot of S newcomers when asked for IAX termination/Origination we say NuFone. I just S want to record my experience so far, as it would help anyone wanted S to start S with this company. I can live with the fact that they do not have any S web based interface for customers to do anything with the service as S claimed by S the website. But cannot understand taking two weeks to answer a S freaking email. (Well in the absence of trouble ticketing system or S web based access S to accounts, email is the only way to contact Nufone) S I have services running with Iconnect and Voicepulse etc and I was S just trying to use Nufone being well recommended in this list. S I am not here to tarnish Nufone name but I have no option but to ask S the community since there is no response to my emails or there is no indication S of when my service is available. If they have gone belly-up, well I S can then S concentrate on some other company and consider my money as a cost of S a bad S choice on my part. S If I am a very rare case who just had a bad experience with an S excellent company ( I wish ), Nufone please fix this ASSAP. S Later S Sathya S ___ S Asterisk-Users mailing list S [EMAIL PROTECTED] S http://lists.digium.com/mailman/listinfo/asterisk- users S To UNSUBSCRIBE or update options visit: S http://lists.digium.com/mailman/listinfo/asterisk- users Sathya.. Ripping on Nufone here isn't the greatest thing in the world to do. I get emails from Nufone as soon as i have issue or problem or if i See Jermey on irc he usually answers my questions within couple mins. I've been using nufone since November its been Rocksolid lot better support from them then the all mighty voicepulse which takes sometimes months to get anything out of those fools. Nufone is great dont be trashing them. -- Best regards, Frankie ([EMAIL PROTECTED]) mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk- users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk- users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk- users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk- users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] time to build an open phone?
I had been thinking of doing this, but lack the electronics expertise to do such a thing. I basically need phones that look like trading turrets, so I can sneak them into this one trading firm. Good idea, let's see if there's any traction. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bob Knight Sent: Wednesday, December 24, 2003 1:30 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] time to build an open phone? Open software seems to work. Why don't we try it with hardware. 1. pick an embedded processor. It should have a nice linux gui support (like x jtag debugger). 2. pick a linux based cad system we all have easy access to and place schematics under cvs. 3. pick some type of gpio or serial interface for keyboard/display. 4. pick some basic functionality. 5. code it up. A stripped down *. Let everyone do their own thing with the expensive part. Tooling/packaging. We could let Digium be the distributor, so they are not left out of the loop. A board set would be offered with NO support. If Digium wants no part of it, we just build them on our own for our own use or sell them on ebay. What we would provide is schematics and source code. Everyone can take this to their favorite fab house and crank em out. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk- users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Nortel IP phones?
Howdy- Has anyone on the list implemented Nortel IP telephones (the exact name escapes me at the moment) with *? Doing some work for a small trading firm which needs a phone to have the ability to dial into a improvised hoot-n-holler system (* conference), so it needs speakerphone, and multi-line capability. Also, I'm open to cisco phones, going to look at those now. Thx, -cedrick === Cedrick Johnson www.cedrick.net Market Commentary: http://uranium235.blogspot.com Charts: http://charts.cedrick.net You could starve at a banquet if you were afraid the food was poisioned - Art Cashin, CNBC interview ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] modprobe ztdummy failed
Hello- I've been trying to scour both * and linux user group archives for a solution to this particular problem, but I am just plan stuck. I got the latest zaptel sources from cvs, uncommented ztdummy.o in Makefile, ran make; make install then, did depmod -a. All is well up until now. When i do modprobe ztdummy, i get this error: [EMAIL PROTECTED]:/usr/local/src/zaptel# depmod -a [EMAIL PROTECTED]:/usr/local/src/zaptel# modprobe ztdummy /lib/modules/2.4.20/kernel/drivers/usb/usb-uhci.o.gz: init_module: No such device /lib/modules/2.4.20/kernel/drivers/usb/usb-uhci.o.gz: Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.20/kernel/drivers/usb/usb-uhci.o.gz: insmod /lib/modules/2.4.20/kernel/drivers/usb/usb-uhci.o.gz failed /lib/modules/2.4.20/kernel/drivers/usb/usb-uhci.o.gz: insmod ztdummy failed [EMAIL PROTECTED]:/usr/local/src/zaptel# any ideas? thanks, cj ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Unsubscrip
Go here http://lists.digium.com/mailman/listinfo/asterisk-users -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, October 20, 2003 10:56 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Unsubscrip I would like to remove my mail address from asterisk so pl let me know how to remove from the list. Thanks Venkateswaran ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Meetme question
Ok.. I got * and SIP working internally now .. still wrestling with connecting two remote * pbx's together.. I'll save that for another day though :) I setup Meetme on this new * PBX, but when I try to dial to join the conference, I hear a recording saying the conference is invalid or something to that effect. Here's a copy of my log files: == Parsing '/etc/asterisk/meetme.conf': Found WARNING[24592]: File app_meetme.c, Line 154 (build_conf): Unable to open pseudo channel It then hangs up.. Anyone seen this before?? -cj ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Meetme question
Any idea how to setup a zap_dummy since I do not have a zap device?? Thanks :) -cj -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kim C. Callis Sent: Thursday, September 25, 2003 2:46 AM To: Asterisk User Mailing List Subject: Re: [Asterisk-Users] Meetme question You need to either have a zap channel available or zap_dummy in place to get this working... SIP setup only requires a zap channel for meetme... K. Callis On Thu, 2003-09-25 at 00:43, C. Johnson wrote: Ok.. I got * and SIP working internally now .. still wrestling with connecting two remote * pbx's together.. I'll save that for another day though :) I setup Meetme on this new * PBX, but when I try to dial to join the conference, I hear a recording saying the conference is invalid or something to that effect. Here's a copy of my log files: == Parsing '/etc/asterisk/meetme.conf': Found WARNING[24592]: File app_meetme.c, Line 154 (build_conf): Unable to open pseudo channel It then hangs up.. Anyone seen this before?? -cj ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/aster isk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Meetme question
No problem.. I am editing the makefile for zaptel now, and uncommenting ztdummy depmod: *** Unresolved symbols in /lib/modules/2.4.20/misc/ztdummy.o Hmm... Ok.. I give up.. I've been here for over 18 hours now, I'll try it with a fresh head tomorrow Thanks to all who replied (Kim, Chee, WipeOut, Josh...) -cj -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kim C. Callis Sent: Thursday, September 25, 2003 3:07 AM To: Asterisk User Mailing List Subject: RE: [Asterisk-Users] Meetme question I was thinking faster than I type... That should have been ztdummy On Thu, 2003-09-25 at 01:02, C. Johnson wrote: Any idea how to setup a zap_dummy since I do not have a zap device?? Thanks :) -cj -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kim C. Callis Sent: Thursday, September 25, 2003 2:46 AM To: Asterisk User Mailing List Subject: Re: [Asterisk-Users] Meetme question You need to either have a zap channel available or zap_dummy in place to get this working... SIP setup only requires a zap channel for meetme... K. Callis On Thu, 2003-09-25 at 00:43, C. Johnson wrote: Ok.. I got * and SIP working internally now .. still wrestling with connecting two remote * pbx's together.. I'll save that for another day though :) I setup Meetme on this new * PBX, but when I try to dial to join the conference, I hear a recording saying the conference is invalid or something to that effect. Here's a copy of my log files: == Parsing '/etc/asterisk/meetme.conf': Found WARNING[24592]: File app_meetme.c, Line 154 (build_conf): Unable to open pseudo channel It then hangs up.. Anyone seen this before?? -cj ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/aster isk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/aster isk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP + NAT Howto?
Hello Folks- Pretty new to the list here, got a lot of reading to do.. Does anyone know where I can find a decent HOWTO or set of instructions for running Asterisk and SIP clients thru firewall/NAT systems? I have a Asterisk box sitting behind a linux firewall at a remote location and have the 5060 and etc ports open as well at 16381-16391 UDP open and routed to the Asterisk box as well. I have a bunch of clients at another location which are also sitting behind a Linux ipchains/tables firewall So far, I'm able to get the clients (Xten Lite) to ring each other, but they ring, and one will say it's connected, while the other one just hangs up. -cj ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP + NAT Howto?
Ok so if I understand correctly: For IAX, just open up the IAX ports on the firewall (the exact numbers escape me right at the moment), and let it fly? -cj -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Varga Sent: Friday, September 19, 2003 5:27 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] SIP + NAT Howto? Client Server XTEN -- */Firewall(NAT) ---IAX--- Firewall(NAT)/* If you are going to use IAX, I don't think you have to put * on the firewall boxes, only if you wish to use SIP. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/aster isk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
