similar to: Incoming calls from I-net only for IP-address?

Hi, I've just (earlier today) updated from CVS so that I can apply the dtmf caller id patches. Unfortunately this has had an undesired effect. I have an intertex ix66 which up until the CVS update allowed me to register my * server with the ix66 for my local domain (eg sip.mydomain.com). Now it appears that asterisk gets totally confused and tries to register with itself! Anyone got any

I'm using X-Pro connected to an asterisk server (CVS-HEAD-01/27/05-23:17:07) and after about 15 minutes in a call I get a lot of noise in my end. I don't think the other part of the call hears it. After some 10 seconds or so everything is fine again. In my CLI I get NOTICE[32322]: RTP Transmission error to 85.xxx.xxx.xxx:35162: Operation not permitted. I get it on calls to the PSTN

This week, I've been really busy with the launch of a new Swedish Voip provider, www.bbtele.se, so I haven't been able to follow the Asterisk community and haven't been very responsive either. My apologies if you've tried to contact me and I did not reply quickly or at all. So to cover up (can't report on what is happening :-) I dedicate this issue of Asterisk Sunday News to

Hi, I know this is slightly off topic but I figured the knowlege here is probably the best on the subject.. I want to setup remote offices with 4 to 6 SIP phones (SNOM 200) using ADSL and the internet to connect to the Asterisk box.. These phone will be behind an ADSL router using NAT... I don't want to setup another Asterisk system in each office so IAX is not an option.. I could use

> I have a * box with 2 nics in the following setup: > > Internet > | > 192.168.5.253 (firewall) > | > 192.168.5.xxx network (gw 192.168.5.253) > | > 192.168.5.10 (* nic 1) > 192.168.6.10 (* nic 2) > | > 192.168.6.xxx network > > The netmask for both networks is 255.255.255.0 > > The 192.168.6.xxx networks has a 48 port switch solely for the use

Welcome to a new issue of Asterisk Summer News! The holiday season is coming to an end here in Sweden, people are getting back to work and the kids will start going to school next week. Life is slowly adopting to normal and I have to start dressing more towards a businessman than a beach bum. Guess I have to start going to the gym again as well. Anyway, back to the topic. Asterisk and VoIP.

Hi All, I am attempting to setup Asterisk with sipcall.co.uk. They use Intertex kit to provide the SIP service. Unfortunately Asterisk cannot seem to authenticate against Intertex. Having provided SIP debug info the provider has informed me that Asterisk does not appear to support 'qop', 'nc' and 'cnonce' which are used to stop replay attacks. So, does Asterisk support

Sorry to ask such a basic question: I have a * box with 2 nics in the following setup: Internet | 192.168.5.253 (firewall) | 192.168.5.xxx network (gw 192.168.5.253) | 192.168.5.10 (* nic 1) 192.168.6.10 (* nic 2) | 192.168.6.xxx network The netmask for both networks is 255.255.255.0 The 192.168.6.xxx networks has a 48 port switch solely for the use of cicso 7940 phones, the 192.168.5.xxx is

http://bugzilla.mindrot.org/show_bug.cgi?id=1242 hotz at jpl.nasa.gov changed: What |Removed |Added ---------------------------------------------------------------------------- CC| |hotz at jpl.nasa.gov ------- Comment #2 from hotz at jpl.nasa.gov 2006-11-10 06:20 ------- At our institution machines are SA'd

Hello listers, Does anybody know how to add text to an xyplot without whipping out the existing curve? That's all. For instance, Lets say you generate a graph like this A <- data.frame(x = rnorm(100), y = rnorm(100)) xyplot(y ~ x, data = A) How would you add 'Hello world'? I tried 6.02E23 different partial solutions found on the web and failed. I just need one EXAMPLE that

What I have is: * Asterisk 1.8.10.1~dfsg-1ubuntu1, * SPA112 ATA with analog fax in 1-st FXS port connected, * SIP trunk with provider supporting T.38. My network looks like this: * spa112 (192.168.33.200/24) and Asterisk (192.168.5.253/24) in neighbouring LANs, * Asterisk connects to the provider (80.75.130.136) via router (82.200.7.184). Router has full DNAT to Asterisk server. What happens?

All, Sometimes when I do an rsync, it times out and leaves on orphaned process on the remote machine. I am running rsync over ssh using public key authentication. I synchronize about twenty-five machines, and this problem only appears on one of them. It worked until recently on this machine. I don't know what changed or what is different. I run rsync with a command line like this

I have configured openssh --with-kerberos5=<path> --with-pam --with-tcp-wrapper=<path>, but when I expire my kerberos password, I do not get a challenge to change my password for kerberos... anyone have any thoughts. I tried modifying the Make file to include -DGSSAPI -DHAVE_GSSAPI_GSSAPI_H -DHAVE_GSSAPI_GSSAPI_GENERIC_H because I noticed that the gssapi options for both

The CentOS Project is going to be in the community space at Red Hat Summit this year. Summit's community theme is centered around NASA's JPL poster designs (http://www.jpl.nasa.gov/visions-of-the-future/ ) , and we'd like to make sure that our booth is appropriately dressed, and that our community is appropriately represented. If you enjoy spending hours doing design work so that

Hi fellow traffic-shapers, I am implementing a Diffserv CORE router using Linux kernel 2.4.18. I used this excellent website (http://www.opalsoft.net/qos/DS-38.htm) using AFHTB as a starting basis for my EGRESS diffserv implementation - and all works great so far!!! But my project has an additional requirement. There are services running locally on the router which are sending/receiving IP

Hi all, I'm looking for a reporting solution for Asterisk CDRs. I have a small Asterisk server that will eventually have 4 - 6 trunks. the system is up and the CDRs are being written to a MySQL DB. I tried installing Areski, but had no success .. I assume it's no longer supported... the last update was in March 2005 according to this page..

I'm new to asterisk, but have been at Linux since 1997 .. so kind of a newbie .. I have a good buddy who is supposed to be helping me with this darn install, but you all know how that works. I have a new mid-tower, AMD 64 x2, 4 gigs of memory and spoftware mirror. OpenSuSE 11.2 .. as I said Asterisk 1.4.26-rc6 and Asterisk-GUI 2.0 .. I have an 8 port TDM800P card from Digium. basic install

My third post here. I''m learning Rails. getting Oops! Google Chrome could not connect to localhost:3000 I reinstalled Ruby and Rails. I create the Rails new ???? then run Rails S or Rails server. still cant get the localhost:3000 to start. I''m running Windows 8.1 Ruby version 2.0.0p0 Rails 4.0.0 -- You received this message because you are subscribed to the Google

Friends, SIPit is the main interoperability event for all things SIP. It's organized by the SIP Forum and creates good feedback to the IETF. Asterisk has been participating in SIPit during many years and in many variants - videocaps, Marc Blanchet's IPv6 branch and the standard Digium releases. All these tests has lead to a large amount of improvements for Asterisk and have helped us

Hi all I have 2 snom 100's and an ix66 (sip aware firewall) set up with asterisk. I needed to register a number of lines so what I've done is make asterisk register all the lines i need (attaching them to an extention eg 1000) and then register each phone with asterisk. so for example in sip.conf: register => andy@sip.mydomain1.org/1000 register => andy@sip.mydomain2.org/1000