similar to: Asterisk on Proxy

I am trying to make 1 soft SIP UA behind NAT connect to a public hard CISCO UA via a public asterisk server. The CISCO UA can hear the voice from the SIP UA but not vice versa. I do set nat to yes for the soft phone. Any help would be greatly appreciated. Below is my sip.conf [general] port = 8060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all

All, I have my * box NAT'd with all ports forwarded that are SIP related (based on Wiki). I also have nat=yes, externalip=WAN address of firewall, internalip=LAN network of *. I have my Xten soft phone on a PC which is NAT'd behind firewall with ports forwarded. I have also followed instructions on Wiki for Xten. I can authenticate fine, and sip show peers shows my extension is OK,

Wayne, I have been using the rsync snapshot from 1/25 for the last few days and everything seems quite solid so far. I include below a few nit-picks in case you're looking for thing to tidy up. I'm also looking forward to hearing news about the --filter option. I'm thinking it is going to be very useful for backup purposes. What I would want to be able to do is, on a

Hi all, Love tinc by the way. It's a great VPN. I'm having issues with 2 nodes always talking through an intermediate node. My set up is a VPS in a cloud somewhere that's running tinc and 2 other nodes - one a roaming laptop (always NAT'd) and the other a server behind a dynamic IP home broadband connection (Not NAT'd but firewalled). Neither the laptop nor the home

System = CentOS / Apache 2.0 / HTML files location = /var/www/html Nat'd IP - ports 80, 20, 21 already opened & routed to internal ip. I desperately need the quick & dirty solution. I need a setup where one user can ftp from another desktop (windows) to upload new website files. I currently have CuteFTP, but will use anything needed at this point. I need to try to test this today.

Good morning all, I'm setting up Asterisk for the first time with no prior PBX experience. I'm following Andy Powell's 'Getting Started with Asterisk' (http://www.automated.it/guidetoasterisk.htm). This is my second time through that document - as I did something weird the first time and really upset it somehow - and I wanted to ask a few general questions of the list.

I have 20-30 GXP2000's connected to * over a T1 line. Neither end is NAT'd and there is plenty of bandwidth available over the line. The GXP's are 1.1.5.15, which is the latest. I have a problem where the phones keep dropping off of * and I get a "failed to register" message in the log of *. Sometimes they eventually connect and sometimes, I have to reboot them to

Here is my setup: * is on a NAT'd subnet, but also has an externally routable IP address. I have a Snom360 that's external to this and behind NAT. The Snom360 can call other phones in * subnet (by their internal extension numbers) and voice is transmitted fine; however, when I attempt to check voicemail (or any * voice recordings for that matter) I can't hear them. The phone just

Ok, I have a challenge - get two services working over a single port. (!) Sometimes it's a web service, sometimes it's a VNC service. Here's what I had in mind: 1) Server is normally a webserver listening on port 80, and normally has the following rules active: # iptables -A INPUT -m state --state ESTABLISHED,RELATED -j ACCEPT # iptables -t nat -A POSTROUTING -j MASQUERADE 2)

I have found that the NTP server is not contacted when the phone (Budgetone 100) comes back from a power down. I must reboot the phone without powering down to get the phone to contact the NTP server for the time. It doesn't matter whether I reboot from the phone's web site or using the menu reset function: either works. I have only tested this with a private NTP server. This may

I've searched and tried several ideas (na.action. and other things), but I can't see to figure this out. I'm guessing this is so simple I'll feel foolish for asking, but here goes. Thanks, L.A. Dataset$Rcil=with(Dataset, ifelse(Rpr >= .95, Dataset[,"percentchgn"], NA)) Dataset$LLCI<-with(Dataset, ave(Rcil, LEAID, Property, FUN=function(x)max(x))) LEAID

We have a handful of nodes set up. Some are NAT'd but a few have direct access to the Internet. Sample confs: HostA: Name = HostA AddressFamily = any Interface = tap0 Mode = switch Connectto = HostB GraphDumpFile = /tmp/mesh HostB: Name = HostB AddressFamily = any Interface = tap0 Mode = switch Connectto = HostA GraphDumpFile = /tmp/mesh And so on. If I use HostA as the main meta sever.

I'm afraid I'm quite confused by what I've found on the Wiki. I have the following dial plan that works: exten => 2201,1,Dial(sip/2201@gs1.uucp,20,) exten => 2201,2,Voicemail(u2201) exten => 2201,3,Hangup exten => 2201,102,voicemail(b2201) exten => 2201,104,hangup When the phone is in use it goes to voice mail as busy. When not picked up, as

I created my first Xen VM that didn't use a static IP address. It grabs an IP from the 198.168.122.0 subnet (from I don't know where). I would rather want it to use my DHCP server. How can I achieve this? Kai -- Kai Sch?tzl, Berlin, Germany Get your web at Conactive Internet Services: http://www.conactive.com

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

For anyone who is interested, I have a working tftpd (modified wvtftpd) capable of serving configuration, dialplans, and ringtones to Cisco 7960/7940 and ATA-186 devices that are located behind NAT firewalls. As TFTP is not a very firewall/NAT friendly protocol, I had to break some rules to get it to work with these cisco devices. It might cause problems for other TFTP clients, but it works with

Hello all, I am looking for a way to restrict users from logging in two separate phones with the same authorization name/password at the same time. Meaning, I only want users to be able to place a call from one phone in one location, but have the ability to move from computer to computer. Has anyone found any sort of solution for this type scenario? Thanks, Bryan Mahin Please visit us @

Hi, Sorry for my bad english but I?ll try to explain my problem I got an Asterisk running in my house with ADSL... I?m using X100P and TDM400P cards.... My intention is get calls via PSTN to my house and Redirect to my computer in my work using X-Lite by SIP... Here?s the map with Firewalls Call for anyone to my house => PSTN => X100P => EXTENSIONS => SIP/RTP => ISA MICROSOFT

Hi, We got the following error messages when we ran rsync 2.5.2 from Solaris-5.5.1 to Solaris-5.8. vl121909/jagcdms_ch+_diags/sys/sys_l/ch_sys/rtl/systop.v vl121909/jagcdms_ch+_diags/sys/sys_l/ch_sys/rtl/sysup_esat.v vl121909/jagcdms_ch+_diags/sys/sys_l/ch_sys/rtl/upesat_defines.h wrote 63878001383 bytes read 1529124 bytes 1246928.64 bytes/sec total size is 148182624008 speedup is 2.32 rsync

I am experiencing similar issues as noted in this bug id: http://bugzilla.mindrot.org/show_bug.cgi?id=510 I am ssh'ing from a dchp'd address to a nat'd address (tried both hostname & ip). after a successful login, I launch an X app. Shortly thereafter I get: "Disconnecting: Corrupted MAC on input." I was not experiencing this problem w/3.7, but I cannot place full