similar to: 2 firewalls, different INVITES

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

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

current setup SIP phone 192.168.1.30 --> linksys wrt54g sveasoft -- INTERNET -- (xl0) Firewall (xl2:172.16.0.50)--> (em1:172.16.0.101) Asterisk problem is RTP stream not oging trouhg from * to sip and vice versa. #1 and asterusk is pushing 192.168.1.30 back to linksys with 172 as return address.... or #2 asterisk trying to get back to me as 192.168 on public internet.. got

I just bought two new Linksys WRT54G routers. Sveasoft has loaded Linux on this router and included a bunch of Linux tools, one of which is Bandwidth Management. The QoS aspect of this is supposed to be much more granular than the previous solution (Wonder Shaper). I have not been able to find any suggestions for how to impliment QoS (Bandwidth Management) using the web interface of Alchemy.

Hi all, I've been trying to use BroadVoice as a SIP service provider. They don't officially support * but are helpful when it comes to answering questions for setup parameters. They claim they have no firewalls or access lists that need to be set up so I can get access to their servers. However, something's still not quite right when I use the parameters. It looks like our Asterisk

As seen on my post at: http://www.sveasoft.com/modules/phpBB2/viewtopic.php?p=28112#28112 This works very well... It does NOT work with stable 4.0! sveasoft will be issuing a bug fix for this (4.1) in the near future. Final Rev of working script w/ asterisk support I'm not going to run alchemy on production machines until it is stablish. Remember to set your uplink properly and to set

We are trying to determine how to build out an IVR system we are working on. The system needs to be able to handle probably at most 5-10 concurrent calls at peak times. Other times we are just looking for a reliable service. For incoming calls we've been using BroadVoice VOIP and before that VoicePulse VOIP. VoicePulse's IAX service started dropping DTMF inputs that we were processing

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

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

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

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

Dear All, I am experiencing a problem for which I need some help. I have created a small file containing a simple function File name = smalltest.r Content : smalltest=function(extvar) { itworks=paste('Ca marche',date()) setwd("c:/windows/temp") write.table(itworks,file="resultat.csv", append=FALSE) } This function works when operated from R environment To get

Hello everyone.... Searching the archives and google always comes up with entries regarding the "dyn" dns option in the 7960, but I can't find answers to my specific question.... My 7960 is connected via cable modem and is NAT'ed (everything is working fine). On the 7960 under SIP configuration\NAT Address I have the public IP of my cable connection. Comcast gives me a

- This mail is in HTML. Some elements may be ommited in plain text. - Tel: +46 (0)8 625 46 40 ULTRA-TYST LUFTRENARE FÖR KONTOR OCH HEM Lider du av allergiska besvär på våren? Besväras du av illaluktande ämnen på din arbetsplats eller i hemmet? En luftrenare kan göra underverk på din hälsa! Vi rekommenderar pollenallergiska personer att ställa ett luftfilter vid anslutning till arbetsplatsen och i

- This mail is in HTML. Some elements may be ommited in plain text. - Tel: +46 (0)8 625 46 40 ULTRA-TYST LUFTRENARE FÖR KONTOR OCH HEM Lider du av allergiska besvär på våren? Besväras du av illaluktande ämnen på din arbetsplats eller i hemmet? En luftrenare kan göra underverk på din hälsa! Vi rekommenderar pollenallergiska personer att ställa ett luftfilter vid anslutning till arbetsplatsen och 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

Hello, Here is my setup : Telephone 1 ( GXP 2000 ) Telephone 2 ( SPA942 ) Asterisk 1.4.17 ( same behaviour on Asterisk 1.4.23.1 ) Scenario: I don't like the default asterisk transfer feature, so I am trying to write my own. What I did : 1. Added to dynamic features #3 with AGI pointing to my php script 2. PHP script asks the user to enter his/her extension 3. PHP connects to Asterisk

Besides ranch networks and borderware, what other SIP aware firewalls for the SOHO/medium market exists? -- ------------------------------------------------------------ Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ------------------------------------------------------------

Did anyone see the article? It''s the first time I really noticed that these little Linksys routers are such a fully fledged linux machine with a decent processor and a replacable firmware. I am now itching to get one to replace the multipurpose firewall desktop machine. Has anyone experimented with the current state of the firmware and how advanced you can get with tc rules? For

I'm trying to setup Asterisk on my Linksys WRT54G router but it only has 4MB of storage (and only about 250KB free with OpenWRT linux installed). I'd like to setup voicemail, but I'm afraid I don't have enough storage on my router. Can I somehow store this on my VOIP provider? (eg. BroadVoice) I was also thinking about mounting an NFS partition to store the voicemail, but then