similar to: Nat Traversal help!

Hi there, I have an X-Lite phone on my box and I'm trying to register it with a remote Asterisk box. Both the X-Lite and Asterisk are behind a NAT. I know it's a pain to do because of SIP not working well with NATs, but I know there are ways to do such a thing...moving the Asterisk box outside the NAT is not a possibility at the moment. One thing we tried was setting up a VPN, but

Hello List, Can you please point me to the right resources on making multiple sip phones behind a firewall w/ private address work with asterisk w/c is on a public network. I have seen STUN on the grandstream and Xtunnels on X-lite. What is most deployed by members here with similar setups? Thanks. -- Cheers, Paul P. Pongco

I'm new to Asterisk and have installed it on OS X. I am trying to use an X-Lite softphone to connect through Asterisk to FreeWorld Dialup and am not having any luck. I have looked at several settings pages at voip-info.com but have had no luck. I am on a school network which I assume has some security but the particulars are unknown as they are not very helpful. It is necessary to

Hi there Is it possible to configure ssh so that a session can continue/recover after a change of IP? I have a number of static IP dialup accounts and there is no problem using these because when my dialup connection timesout and later reconnects, it gets the same IP and so any existing ssh sessions are unaffected. However, these accounts with static IP's are metered (i.e. expensive ;-).

Hi- Our PPP RAS server is running Samba 2.2.8a and winbind. We want to authenticate dialup users through the PDC (Win NT 4 SP6). However, in order to do so, the dommainname needs to be prepended to the username, ie, DOMAIN+USERNAME. We don't want to have to teach all our dialup users to add the domainname when they login to RAS, so we are looking for a way to script this. Has anyone tweaked

Hi List, After some research, it seems the only reasonable thing to do in order to get SIP phones behind NAT working reasonably well without fiddling with the DSL router is to have some kind of far end nat traversal mechanism. Is there any way to do this with open source tools? I've seen somewhere that far end nat traversal can be achieved with SER + nathelper does the job... has anybody

Thanks to ww and his patch on bug #104, I have successfully implemented Asterisk behind NAT without using STUN or anything crazy. It's quite straight forward. Until this gets tested enough and put into CVS, you will have to patch your chan_sip.c file to do this. I'm sure within the next few days this will get put merged into CVS if no one finds any problems. I tried this on chan_sip.c

How will this fare? I am planning on putting an asterisk box for my brother in the Philippines but they only have dialup internet. I want them to be able to use a telephone set on a phonejack or linejack card and call me and vice versa via VOIP. My setup in the US is working already with a broadband cable connection. I am thinking that dialup may not work because of the bandwidth required

I'm trying to setup a demo where my Asterisk box with a TDM01B (FXO) answers an incoming call and forwards that call to a SIP softphone (X-lite.) Seems all is built/installed okay: # ztcfg -vv Zaptel Configuration ====================== Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. I'm pretty new at this and the extensions.conf file is eating my

It seems to me that ipsecnat tunnel type is not complete. Latest drafts of ipsec nat-traversal use udp port 4500 for nat-traversal communications. (It''s called port floating). That is needed to get rid of ugly ipsec passthru devices. Now ipsecnat opens port udp/500 from any source port. And I think ipsecnat won''t work at all with gw zone defined? I''m not sure about

Hi, I have a setup that consist of 2 firewalls connected over dialup and PPP. Each side of the ppp are protected by shorewall. One side of the PPP masquerades everything not addressed to the local network to its eth0 (the net). fw1 <---- ppp (dialup) -----> fw0 <----- NET When making an http request to a site on the Internet from the machine not directly connected to the net (fw1), the

Last couple of days some of my hosts were probed for UDP port 500 (IKE daemon, used by IPSec for key exchange) from dialup IPs. Don't remember seeing similar probes before. Some new vaulnerability that script kiddies (and pro crackers) are trying out, or is this some old stuff? I do remember there were some security problems with racoon in the past (that were fixed in current CentOS

Hi, I'm away at a conference in Amsterdam. My home is in Cambridge in the UK. On a whim, I tossed an ATA186 and a phone into my bags before leaving home. I was able to plug my ATA186 into a LAN here at the conference and was connected to my home Asterisk in a few seconds. Total time from unzipping my bag to talking to home no more than 15 seconds. OK, so the kit could be more portable,

Hi, I''m trying to use shorewall for a RAS dialup solution We have networks we need to connect to with the same ranges internally (i.e. 2 separate users with a 192.168.0.0/24 range). We connect to these via a pptp tunnel (or isdn) The problem we have is that we need to access these networks all the time, so allocate them a range from our internal range. This will then be NATed to the

Hello Folks, I noticed that all of the IaxClient based softphones with exception of Firefly only seem to have support for GSM but what about iLBC? The quality is excellent with iLBC even on a dialup connection! Meanwhile while the audio on GSM often sounds scratchy. Is anyone looking to implement iLBC in an IaxClient based softphone soon? Errol Biz4Web Solutions Limited

Hi to all the * people out there, Please kind to me as I am both new to Asterisk and to Linux - But I am learning fast. My config is quite simple, I'm just following examples and the Wiki: I have two PC's running X-Lite phones, these work without problems between each other, and I have a GS BudgeTone-100 registered to Free World Dial UP (working no problem). I have tried to

Hi, after reading about it on the list I decided to set up a Free World Dialup account. For those of you who don't know, that is a sip proxy where you and your friends can singn up free and then you can just connect to it with any sip client and call anybody that is registered for free. Pretty much like iaxtel (I belive that was the name of it) for the iax protocol. It even supports clients

Hi, Currently streaming ogg isn't practical in this situation. That was one of the first things i checked into. WHen i looked i didn't see a streamer that did both ogg and mp3. Thanks. Dave. ----- Original Message ----- From: "Daniel Ballenger" <lpmusix@gmail.com> To: "Dave" <dmehler26@woh.rr.com> Cc: <icecast@xiph.org> Sent: Friday, December

Hi forum, I have been fighting days and days configuring FWD and asterisk with NO success I have the following scenario. My sister in Spain with FWD dialup client My question is if she can dial my FWD dialup number, which is registered in Asterisk and the call being forwarded to ring my IP Phone. Spain LAN FWD

I have Asterisk server providing phone service for my company. The server is behind a PIX-515 FW and is assigned a private address 192.168.11.X/24. With that said what is best to provide remote SIP phones (home offices) securely. If the solution is to put up another Asterisk server with a public IP address I am opposed to that. I am looking for the a secure reliable solution to set up remote SIP