similar to: SIP -> NAT -> *

Hi folks. Please don't flame me but I've been googling around for days, read a tremendous amount, tried everything, and still no go. This is most definitely a typical newbie question. - I sure hope there's somebody(s) out there who'll humble themselves to help me out. I've set up an 'out of the box' basic Asterisk server running on Slackware Linux. - It basically

Hello Everyone, I have one machine (asterisk server) that is DMZ behind my nat firewall on my client end (at home) i have a linksys wrt54g with 16384->32766 forwarded to my cisco 7960 (which works fine) and 16000 -> 16383 forwarded to my sipura 2100 (which is set to these ports) For some reason, the sipura only works some of the time. Both devices are set to register, is there anything

Hi, I'm trying to configure Sipura 2000 (behind NAT) which connects to Asterisk (public IP, no NAT) and having interesting results. When Sipura is behind Linux/NAT firewall it works great and no special NAT settings on Sipura are necessary. The issue I'm having is when Sipura is behind Linksys broadband NAT router. Sipura gets registered with Asterisk just fine, but I can't hear

Hi all, We have put together a firmware for a range of inexpensive routers. It has been configured to provide optimum VoIP performance. We have internally tested it for number of months and it looks very good. You should be able to run it easily with 20+ phones on local network ( we still did not hit the upper limit ) assuming that you have bandwidth. Your VoIP will get prioritized over other

Should a Linksys Sipura 2102 be configured with nat=yes even if it is on the local network? I have been having some troubles with a Linksys Sipura 2100 series, which suffers from NO AUDIO after a few calls.. Because it is on the same subnet as Asterisk it is configured with nat=no. When you think of it because the Sipura 2100 is a broadband router, the voice part may be considered as being behind

Original Post ---------------- I have an Asterisk related problem with mutualphone. I can connect to any number with any softphone that I am using (iaxcomm, SJPhone, and a few others). Also I have a Grandstream BT 101. But I cannot call (via Asterisk) to mutualphone destinations. Other destinations go fine. A working phone call (e.g. from iaxcomm) gives the following on the console: --

Hi, We're installing an * box next week (pbxtra from fonality) and I'm trying to come up with a solution for remote users that want a phone in their home. I need VPN and QoS capability, wireless support would be a nice to have. Ethernet handoff is fine, i don't need integrated dsl or cable modem... I've been googling and cruising the list and can find bits and pieces

Hello, I've been fighting one-way-audio issues with asterisk and SIP extensions for some time..., and I want to share with you my findings ;) My setup: * 1 ADSL router (Zyxel) * 1 Asterisk box with private IP, and interesting ports forwarded to it. * Several extensions, some local some remote The problem: * External extensions behind double nat don't get audio when they

I have a problem that is almost certainly nat-related, but I can't figure out what's happening. Since moving the Sipura behind a NAT server (Linksys), I am no longer able to call between the two lines on the same Sipura. When I dial one extension from the other, it rings, but immediately after I pick up the ringing phone, the call is uncerimoniously dumped. I can tell the call

HI guys i need help i install centos 4.1 and apt-get the rp-pppoe. so i setup the rp-pppoe using adsl-setup and input all the data given by the provider. When i connect to my dsl using adsl-start it keeps on terminating where can i find the logs for rp-pppoe and anyone had this kind experience with bellsouth adsl? anyway when i connect the modem to my linksys wrt54g router it worked fine. --

If this has been covered before - I appologize. We use some Sipura SPA-2000's with the g711 codec and all seems fine (except for the occasional failure to register errors in my asterisk logs - but I will save that for another post). g711 call quality is on par with our Cisco 7960's. However, when using the g729 codec, the call quality on the Sipura device goes downhill on the PSTN side

Hi there everyone! I use asterisk as a home pbx. My internet connection is a DSL one, and I have a Linksys WRT54G that nat things for me in a 192.168.X.X style network. I've installed asterisk on my mac, and tried several examples I've found on the net (voip-info, gizmo, etc.) about how to create a Gizmo and a FWD trunk. However, all my attempts failed. The FWD thing kinda

Hi, On my linksys/sipura phones/ATA, there is a setting called "NAT Mapping Enable" and another called "NAT Keep Alive Enable" These settings must be on in my setup so that my phones/ATA remain connected to my * server. My setup is: Home LAN - Pfsense (NAT, Dynamic Public IP)- Internet - PFsense (1-to-1 NAT, Static public IP) - Asterisk server. I was wondering:

Bob, Have you looked at any of the products by Zyxel? With QOS, VPN & wireless support they have: For ADSL: Prestige 652HW Firewall/Router: Zywall 10W & 30W I'll be honest, I havn't used any of these yet. We were looking for similar products to suuport our VOIP installs. We just ordered some demo units from Zyxel, we shoul have them later this week. I'll let you know

i have been trying to get a newly liberated (from vonage) cisco ata-186 (sip ios v3.1) working properly with asterisk. my client is behind a linksys wrt-54g, which up to this point hasn't proven to be a problem (i have several sipura spa-2000's and polycom phones working just fine behind them). (i'm running cvs-head from yesterday). after looking at the various suggestions,

Hello All, I would be grateful for some help with this issue. I have gotten Asterisk to work with Broadvoice on outgoing calls only. When I try to call my Broadvoice number from another location I get a caller is busy message. My Asterisk box is behind a Linksys WRT54G. Could this be the problem? On the Linksys, I have port 5060 forwarded to the asterisk box 192.168.1.2. I have tried not

I'm looking for a recent asterisk package for the Linksys WRT54G. Has anyone know of a 1.2.X build for this box? Thanks, Frank

Here's the problem my sipura 2000 is setup on Nat Network in my office and my Asterisk Server is setup also on Nat Network at home the sipura can register and get calls but no audio comes in and out of the sipura and when i dial local extensions on the sipura i get this error message. any suggestions on what i can try as work around. *CLI> NOTICE[1158921008]: File chan_sip.c, Line 5394

Ok, I've done some more research and I don't think I want an FXO box... What I'd like to do is use BroadVoice (with their BYOD plan) and then run Asterisk on my WRT54G router. I'd also like to use my regular home phones without having to use a special "SIP" phone... (eg. I like my Vtech normal cordless phones) What do I need to buy to get this working? It sounds like I

I'm using Asterisk (1.2.14, RedHat 9) but I've been having trouble with SIP re-invites. I have a DiD from an ITSP and when someone calls in, Asterisk plays a menu recording and transfers the call to the external line the caller selects. Since both sides of the call are external, I want to use re-invite to avoid the rtp packets from going through my server after the call is bridged. I