similar to: Asterisk on Rackspace, My SIP phone behind NAT

Hi all, I'm looking for a solution to be able to back up to a Rackspace Cloud Files account from a CentOS server. I have set up Duplicity but have found out that the back-ups are in GPG-encrypted volumes. There are also GUI clients for Windows and Mac but they're not what I need. I'm looking for something like a command-line tool that will let me push directories on the server (via

Hello. We are looking for a new provider to host our Rails site -- we have outgrown our current provider (for the past 4 years) since our business is beginning to take off in a big way this year. Which provider (e.g. Rackspace, Engine Yard, Heroku, Serverbeach, etc.) would you recommend to run a highly scalable site? FYI, we currently have over 150,000 users that will likely grow to a million

Based on net-next. This patch exports port->port_no port->port_id in the end of IFA_INFO_DATA. Cc: Herbert Xu <herbert at gondor.apana.org.au> Cc: Stephen Hemminger <shemminger at vyatta.com> Cc: "David S. Miller" <davem at davemloft.net> Cc: Thomas Graf <tgraf at suug.ch> Cc: Jesper Dangaard Brouer <brouer at redhat.com> Signed-off-by: Cong Wang

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! I'm trying to connect to ekiga.net through a client connected to my Asterisk server. For it I am being based on this [1] document. Next I put the configurations that I am using. /etc/asterisk/sip.conf: ; Outgoing to ekiga.net [ekiga] type=friend username=MyUser secret=MyPass host=ekiga.net canreinvite=no qualify=300 nat = yes stunaddr =

Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb: > The phone you gave your wife is really old. Are you sure it supports SIP > OPTIONS? Can you make a call in or out to it? If you can, it is more > likely that it just doesn't support that and you can't use a qualify > statement. No, I'm not sure. And no, I can't make any call, right now... At least,

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! I'm trying to test with a friend who has an Asterisk in his office with the Asterisk which I have in my house. Then I have an extension that he is trying to register remotely. Trying with the Twinkle client, I see that it is registered: - --------------------------------------------------------------------------- 400/400

Darryl Moore <darryl at moores.ca> schrieb: > Ahh. Seen that before! That suggests to me that you don't have your > sip.conf records setup right. > > What's your sip.conf look like? Well, here what I wrote in my sip.conf: register => 00493511111111:MYSECRET at pbxluca/00493511111111 register => 00493512222222:MYSECRET at pbxfax/00493512222222 register =>

Hi SamyGo. Thank you for the replay. So, let me explain it better: I knew that I could use something like " same = n,Dial(PJSIP/6001&PJSIP/6002) ". While every extension (called phones) rings and before anyone answers, SIP 183 messages will be sent to Asterisk from callees. If a called phone answer, the others will be hanged up. It is ok for me. I want to connect the caller just

I've noticed for a while that # brctl showstp output is showing 0 for port_no and port_id It seems that somewhere in 2.6 sysfs land the following items got printed in hexadecimal, and brctl code was parsing for decimal only doug:/sys/class/net/eth0/brport# cat port_id 0x8001 doug:/sys/class/net/eth0/brport# cat port_no 0x1 The following patch to bridge-utils (git and 1.2 release) lets

Darryl Moore <darryl at moores.ca> schrieb: > I'd start by turning on sip debugging in asterisk > >sip set debug ip [your_phone_ip] Really destroying SIP dialog '490d1996593c8e11217828b71aae5c4d at 172.16.34.133' Method: OPTIONS Reliably Transmitting (no NAT) to 192.168.200.11:5060: OPTIONS sip:00493512222222 at 192.168.200.11:5060 SIP/2.0 Via: SIP/2.0/UDP

The Gnu Telephony site: http://wiki.gnutelephony.org Has a Centos repo: http://dist.gnutelephony.org/RPMS/ But I caught some text stating that this is for Centos 4.2. Is it really? Is there a difference; i.e. would it be safe to install these on Centos 4.4? Really I am after Twinkle, and it seems there is a lot you need to actually get Twinkle installed...

Hello List, I'm having problems with registrating my Asterisk-server to the SIP-provider. Yesterday all worked fine, this evening I cannot call out. What can be wrong ? This is my registration in sip.conf : register => 092779077:XXXX at 85.119.188.3 This the output of SIP show peers : asterisk*CLI> sip show peers Name/username Host Dyn Nat ACL Port Status

Hi, I have a problem in renaming R object and saving them within a loop. For ex: for (i in 1:length(all_files)) { uncov_GR <- "variable created in loop" filename <- paste0(sample_name[[i]],"_uncov", ".Rdata")) save(uncov_GR,file=filename) } Within the above short code (out of a long program), I want to

Hi Sammy. After answering your last message (please, see my last message), I was thinking about conferences and my main objective. Conferences will not work well for my case, because I it will allows more than one called party answering the call. But, after one answers the call, I need cancel the others ringing callees. In this case, maybe the best thing to do is to let the called party sends

Dear All, Please help me in this regards, Howto Backup Domain Controller (BDC) or Secondary domain controller for the Primary Domain Controller (PDC) in Centos 5.6 Openldap+samba 3.3 Please give the step by step. Regards kamal

any pointers / suggestions re cheapest Rails hosting where they give you full access to Apache (to load modules etc)??? Can be a shared platform, however not sure if there is a shared platform type hosting service where they do give you such access?

I have ekiga registering to a voip provider (skypho) and receiving external call through the stun server. I want to redirect inconditionally all these calls to my asterisk server, but I can't understand how and what should I configure in asterisk in order to accept the redirected call. In asterisk console I can't see nothing when ekiga passes the call. If I turn asterisk's sip

-- Executing [0473775006 at intern:1] NoOp("SIP/twinkle-088e6ea8", "conversation to GSM") in new stack -- Executing [0473775006 at intern:2] Dial("SIP/twinkle-088e6ea8", "SIP/3starsnet/0473775006") in new stack -- Called 3starsnet/0473775006 -- Got SIP response 482 "Loop Detected" back from 85.119.188.3 -- Now forwarding

Hey all, please let's me know how to "Disable Samba Roaming profile" In OpenLDAP+SAMBA SERVER Regards Kamal

With the new fb_sig_request_method provided by Facebook, I''ve patched shanev''s pseudo-resource routes to generate restful routes, minus some exceptions. The announcement: http://www.facebook.com/developers/message.php#msg_126 Patch is submitted here: http://rubyforge.org/tracker/index.php?func=detail&aid=16105&group_id=4187&atid=16132 Blog entry about it: