similar to: Remote Phone and Server Behind NAT

Hi Everyone, Is it possible to have a SIP Phone work remotely if it's behind a Router performing NAT without connecting the Router to a VPN? The Asterisk Box will be in the DMZ. Thanks Dan CYTEXONE ---- Dan Levine dan@cytexone.com CYTEXONE | Your Technology Specialists R 877.CYTEXONE x 810 212.477.0990 x 810 212.208.6889 FAX 502 Laguardia Place, Suite 2B New York, NY 10012

Andy Shipman wrote: > When running spork on a merb application, whenever a spec is run I get > the following error from the Spork server. > > /opt/local/lib/ruby/gems/1.8/gems/merb-core-1.0.11/lib/merb-core/bootloader.rb:1358: > [BUG] rb_gc_mark(): unknown data type 0x3c(0x2203d0) non object > ruby 1.8.7 (2009-06-08 patchlevel 173) [i686-darwin9] > > Which crashes the

On Thu, Aug 7, 2008 at 5:46 PM, David Shipman <dshipman at gmail.com> wrote: > I've just tried building r54494 on 64bit linux and had the same (no > include path) > error. Any idea what's happening here? > No. :-( Could you try Mike's suggestion? (Replicated here) Try adding: #define _GCC_LIMITS_H_ to limitx.h like so: #ifdef _GCC_NEXT_LIMITS_H +#define

Hi, I am new to R plot. I am trying to increase the data point observation when duplicate data points exist x y 1 10 1 10 2 3 4 5 9 8 in the about example 1, 10 would be displayed larger than the other data points. Could someone give me some assistance with this problem 757-864-7114 LARC/J.L.Shipman/jshipman Jeffery.L.Shipman at nasa.gov

Mike's suggestion seems to fix the problem - happy to use it, although my impression is that it shouldn't be necessary? -David Shipman On Fri, Aug 8, 2008 at 12:59 PM, Bill Wendling <isanbard at gmail.com> wrote: > On Thu, Aug 7, 2008 at 5:46 PM, David Shipman <dshipman at gmail.com> wrote: >> I've just tried building r54494 on 64bit linux and had the same (no

Hi all, A few things. Firstly, I've got a working implementation of the X86ThisCall calling convention, but I'm unsure how to go about submitting it. (I'm not really sure how to go about creating patch files etc, but would like to contribute to the project). Also, I'm using MS Visual C++ Express, and there are a few things that stop llvm1.9 (and the current CVS release) from

Something tells me this was covered recently, but i didn't find it again in gmail. The nodes see each other but if i have something misconfigured that is decreasing my speed. All my nodes are behind NATs. i have a remote node that connects to a central node via port forwarding from public port 7777 to hp821 running on port 655. remoteNode ----> public ip 1.2.3.4:7777 ---->

I currently do this for about 30 different cisco 79xx's connecting to some hosted Asterisk servers. Asterisk listens by default for any SIP connection on UDP port 5060. And will use RTP UDP port 10000 to 20000 The phones use UDP Port 5061 for incoming connections (from Asterisks or other SIP Devices) and use for RTP, UDP port 10000 to 20000. Now, if you are going to have the two remote

Greetings: I have a working configuration for SCCP on our LANS which doesn't route RTP correctly to a skinny phone behind NAT registering from a remote public IP. Configuration: asterisk 1.4.35 servicing only skinny phones trunked to asterisk 1.2.40 which services chan_phone FXS, zap FXO and SIP phones; both instances of asterisk are behind NAT and run on the same host (using different base

Hi all, -------- I have a Asterisk at a public Network (official IP address). In the local network I have isntalled a Snom 200 IP phone and in my home network (behind NAT) a Snom 100 device. I can dial the Snom200 device from my home location without any problems but the Snom200 can not dial me. It always gets a "we do not rely". I tried to forward the SIP Port (5060) UDP via UPnP

I have been working with this for a wile and I have been watching the list for about a month on this subject, to no avail. I am wondering if anyone has successfully configured asterisk for clients to connect to it when the clients are behind nat. I mean successfully because I can do everything except for audio, my audio is only one way. I am asking so I can determin if I will be continuing

Lance, Have you configured your sip.conf to use these aprameters under General? ;externip=66.213.227.66 ;localnet=192.168.1.0 ;localmask=255.255.255.0 -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Lance Grover Sent: Thursday, June 02, 2005 9:39 AM To: Asterisk Users Mailing List - Non-Commercial

I'm running samba 2.2.3a-6 under Debian (testing) and I am having a heck of a time getting Windows 2000 to print to my printer. I keep getting that "Access denied, unable to connect" message. At one point I had it so you could print test pages, but as soon as you tried to print from an application, it would break. However, that doesn't even work anymore. I've been reading

Hi, I have the following problem. I have 5 phones behind the same nat (canreinvite=yes). it works fine to receive calls and to make calls. sound quality is good, so everything works fine. The poblem is that the phone behind nat cant call each other. It works if canreinvite=no. But i want to do this. Does anyone have an idea? Regards, cjk.

[outgoing-fax] exten => _0XXXXXXXXX,1,SetVar(NumberCalled=${EXTEN}) exten => _0XXXXXXXXX,2,Wait(10) exten => fax,1,SetCallerid(${FAX_CALLERID}) exten => fax,2,Dial(Zap/g1/${NumberCalled},60) exten => fax,3,Hangup exten => t,1,Busy exten => i,1,Busy -----Oorspronkelijk bericht----- Van: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]

Hi Matthijs, >> I did put in a hack, but it was horrible. It might be a good idea to >> test out Mike's suggestion to see if it's a better way of doing it. > I just tried building llvm-gcc without your hack, and it still works (even > without the fix Mike suggested). > > So, it seems that r54245 can be reverted again? > > I didn't test bootstrap, however,

Hello, I have an Asterisk 15 with PJSIP behind NAT (Amazon EC2). Now I would like to get Early Media Video working between clients in different NATed networks. The 183 signalling goes trough perfectly, but asterisk doesn't forward the Early Media RTP stream from the caller to the recipent. I have the following configuration: [6001] type = endpoint context = internal rewrite_contact = yes

My understanding based on Wireshark analysis is that the signaling works (also the recipent phone is displaying the video frame before accepting the call), also the calling phone send video (i see that also via Wireshark) but the recipent phone doesn't get any video from the Asterisk before the call. 2018-04-09 17:02 GMT+02:00 Joshua Colp <jcolp at digium.com>: > On Mon, Apr 9,

Yes, media is flowing through Asterisk because both client's are behind different NAT's. Do I need to do something special in the Call Flow? Or anything additional to the pjsip.conf? 2018-04-09 16:50 GMT+02:00 Joshua Colp <jcolp at digium.com>: > On Mon, Apr 9, 2018, at 11:42 AM, Benjamin Marty wrote: > > Hello, > > > > I have an Asterisk 15 with PJSIP behind

wohoo, so if I unterstand it correctly with that patch early media video works over the Asterisk server? In other words the Asterisk server get's able to (process/)forward the early media video stream with that patch? 2018-04-09 17:57 GMT+02:00 Joshua Colp <jcolp at digium.com>: > On Mon, Apr 9, 2018, at 12:04 PM, Benjamin Marty wrote: > > My understanding based on Wireshark