search for: sveasoft

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 Manag...

...t need integrated dsl or cable modem... I've been googling and cruising the list and can find bits and pieces (some using vpn, many using qos), but not a whole lot of anything on folks that are using voip over qos over vpn So far i've come up with this list: Linksys WRT54G(S) running Sveasoft Linksys WRV54G (stock) Draytek 2900 m0n0wall on a Soekris 4801 I do want a small fanless box b/c this will be the primary access router for the remote user (including when they are just at home surfing). I like the soekris because i can have a 'work' interface and a 'home' inte...

I have a SIP trunk to Broadvoice. My Asterisk box (1.4.13) is on public addresses behind a firewall. Originally it was behind a Linksys WRT54G running sveasoft. Sveasoft really can't NOT do NAT even when you turn it off. My Asterisk box is defined as the DMZ box to Sveasoft and it seemed like it was leaving all packets alone. Now I switch to a Centos-based firewall configured with Shorewall (which only compiles Netfilter commands then steps out...

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 stab...

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

...t need integrated dsl or cable modem... I've been googling and cruising the list and can find bits and pieces (some using vpn, many using qos), but not a whole lot of anything on folks that are using voip over qos over vpn So far i've come up with this list: Linksys WRT54G(S) running Sveasoft Linksys WRV54G (stock) Draytek 2900 m0n0wall on a Soekris 4801 I do want a small fanless box b/c this will be the primary access router for the remote user (including when they are just at home surfing). I like the soekris because i can have a 'work' interface and a 'home' inte...

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 pu...

Hi, we're testing Asterisk 1 RC 2 behind ordinary router and NAT. Since we're sharing network with web server it seems like voip packets are not coming through fast enough (Digium demo dies after few seconds...). It's the same if I make direct calls (passing Asterisk) so we conclude it's network problem - it also work normally outside our router... I wonder what solutions can we

I have multiple locations running * where all the phone are on their own lan and all the data is on a separate lan. The problem is they are sharing the same dsl connection. The locations are IAX2 trunked together, but it only takes one data down/up load to just kill the voice. What I am looking for is a small switch with QoS that I can stick in ahead of the dsl modem. Plug in one connection from

Running * with default config files except for sip.conf. Any call made is dropped 5 seconds after connection, with the following messages: Mar 17 16:37:41 WARNING[1009461760]: chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call 6C94C1B1-77C4-11D8-91FB- 000A95DA04DA@192.168.1.152 for seqno 48221 (Response) == Spawn extension (default, s, 5) exited non-zero on 'SIP/2000-6bd7' Mar

Setup: Office A: router: Linksys WRT54GS running SVEASOFT Alchemy-pre7a v3.37.6.8sv Asterisk: v.1.2.4 static IP Office B: router: Linksys WRT54GL running Linksys firmware v4.30.2 Asterisk: v.1.2.7.1 dynamic IP (using dyndns name) Office A is set up with refresh dns and cron job for iax2 reload every 5 minutes. It rarely looses connection to Office B....

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

...me, how do you connect to it? If I have to go digital, which of the TE series do I need? Again, I'm looking for echo cancellation and 1 T1 line. The TE110P offers a single span T1, but no echo cancellation? 2) Any recommendations on a good QOS router? I'm using a Lynksys router running Sveasoft currently for my 15 user TDM system, but I think I'll need something more industrial strength for the new installation. Any recommendations? Thanks, Hugh

Hello all, this is not really an * question but it is somehow related, i am trying to develop a working proposal for cheap and quick telephony services using Voip running over *. By running a wireless network (over 802.11 a/b/g devices), i plan to be able to reach customers directly with eithe table top or handheld 802.11 sip enabled phones. But the disadvantage is that how do i power each radio

...AX (very happy with them, very unhappy with Gafachi). I cannot find any information from LiveVoip that indicates whether they accept G729. Is it likely or is it just dependant upon the provider? My interest is to improve voice quality over DSL and/or Cable Modem connections. I have QoS working (Sveasoft), and it has improved the situation, but the words are still bracketed with distortion. I am hoping the smaller Codec will mitigate this to an acceptable level. George Burt

...ning template 'default' May 18 16:29:49 VERBOSE[3170] logger.c: -- Loaded provisioning template 'default' I'm an admitted newbie, but this seems (to me) that this may be networking/Linux issue and not Asterisk??? SETUP: OFFICE A: -Asterisk 1.2.5 -Linksys WRT54GS router w/ SVEASOFT Alchemy-pre7a v3.37.6.8sv -TW Roadrunner cablemodem (business class) - * box sits behind router with port 4569 forwarded to * box. As noted, I moved * box to DMZ at one time, no help. Internal static IP address of 192.168.1.24 iax.conf: [general] bindport = 4569 ; Port to bind to (IAX is 4569) b...

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 ------------------------------------------------------------

Hi, I succesfully install spandsp chan_misdn and digium card. the rxfax works fine and I get the fax result by email. I would like to do the same using a Patton gw + zaptel but I can't receive fax anymore, the call comes in from ISDN in the Patton gw, patton sends it to asterisk, asterisk run a macro to make a tif file using rxfax, the tif file is correctly created but with a 0 size the call

Does anyone know if sip phones from any of the major IP phone vendors support 802.1x authentication? Any feedback would be greatly appreciated. Thanks in advance. ====================== Jeronimo Romero EUS Networks Email: jromero at euscorp.com Cell: 917-332-7238 Office: 212-624-5943 Web: www.euscorp.com ====================== -------------- next part -------------- An HTML

I want to run dhcp and shorewall on the same computer.It is my gateway and that computer doing NAT for my network.How can I set up shorewall to let only users that get theire static ip address via dhcp, not to let users that had static address.