similar to: SIP Warnning Message

Hello guys, anybody here that can help me checking out this xml file, cause I am traying to configure some cisco 7911G phones to asterisk and I can't get it done.... thanks!!!! a paste of the file is here: http://pastebin.ca/1239083 -- http://celord.blogspot.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi list! I?m connecting a Brasilian voip (- gvt.com.br -) provider through my asterisk box and using the register command from sip.conf. What I can?t understand is why my unit sends a new registration message every minute! And every time my asterisk box sends a registration, it gots a sucessful response, and shows de message: "Oct 14 16:48:22 NOTICE[4090]: chan_sip.c:8742

Hi I hope someone can answer something I'm sure is quite basic. I am following the instructions at https://www.centos.org/docs/5/html/Deployment_Guide-en-US/ch-vpn.html On setting up a VPN The part I am having trouble with is when it show the /etc/racoon/racoon.conf file. But it doesn't say whay you have to do with this file. When I bring up my connection ifup bicester I get RTNETLINK

Hi there, I was wondering whay people were using to monitor rsync backups on centos? I have been looking around sourceforge for various programs (IE Backup Monitor), but am hesitant to trust any of them without hearing about some user experiences. We currently just use rsync for backups (like a slightly more "Vanilla" setup), but just want to be sure everything is going as it should

Dear all, I am fitting a nonlinear mixed-effects model from a balanced panel of data using nlme. I would like to know whay would be the best options for formally testing for autocorrelation. Is it possible to carry out a Durbin-Watson test on a nlme object? As far as I've seen, I think the durbin.watson function from the car package just works on lm objects. Thank you very much, Antonio

hello sir im working on text classification using java and r programming i start with exporting a document term matrix (tdm) from my java programme using corpus and now i try to apply knn algorithem using the matrix and r but i cant do that any help sir here my data here my script https://mega.nz/#!Q6J2ibAA!4PadiOKbP7rLodyiRrVsdKl-D2ZP7LYm0gaz94uBmF8 itry to post put icant whay!!

Hi all, After * starts the command "queue show" would not show any of the realtime queues, but just the ones that are in the queues.conf file. In this state de AMI would not send any "QueueMemberStatus" for that queues until a call is received by that realtime queue. Anyone knows any whay to load this information in *'s memory without the need of the queue receiving a

OK - here's whay I need to make it work for my Linuxconf compatible authentication system. First - a question: Will this work? I can find docs on "virtualfile. auth virtualfile { mechanisms = plain digest-md5 userdb = passwd-file /etc/vmail/passwd.%d passdb = passwd-file /etc/vmail/shadow.%d user = root } Then - since the dmain name selects the passwd file, if there were a

hi ppl, i get this error in my asterisk CLI: Aug 31 02:45:31 WARNING[26091]: chan_sip.c:9835 handle_response_register: Got 200 OK on REGISTER that isn't a register Aug 31 10:22:24 WARNING[26091]: chan_sip.c:9835 handle_response_register: Got 200 OK on REGISTER that isn't a register I guess it is related to my problem i have with one of my voip providers: i'm using asterisk 1.2

I second Eero's comment, use a new IPSec daemon. Openswan was forked and became Libreswan. Paul, now a RH employee, was a main developer for the Openswan project before he and others created the Libreswan fork. https://libreswan.org/ EL6 has Openswan EL7 has Libreswan Racoon isn't all that fun to work with. If you have the option, ditch it and EL5 and move to a newer platform

Centos 5 is still soon end of life. Using it as ipsec gateway is .. Eero 21.3.2016 7.25 ip. "Mike - st257" <silvertip257 at gmail.com> kirjoitti: > On Mon, Mar 21, 2016 at 1:17 PM, Mike - st257 <silvertip257 at gmail.com> > wrote: > > > I second Eero's comment, use a new IPSec daemon. > > > > Openswan was forked and became Libreswan. Paul, now

Hello everybody, Just was wondering if somebody can help for G711 fax passthrough w/ asterisk. The issue I have is regarding codec upspeed when the call is already connected using G729 for example. The setup is fax---ATA---asterisk---Cisco---fax When codec upspeed should happen, ATA or Cisco will send a G711 reINVITE causing the codec to be switched over, but asterisk does NOT

Err. Sounds like security nightmare. 21.3.2016 7.47 ip. "Glenn Pierce" <glennpierce at gmail.com> kirjoitti: > Will ask my boss :) We are hosted on memset so not so easy to update > > Thanks > > On 21 March 2016 at 17:36, Eero Volotinen <eero.volotinen at iki.fi> wrote: > > Centos 5 is still soon end of life. Using it as ipsec gateway is .. > >

Hi Guys, My name is Marcias, and I am setting up for the first time an Asterisk PBX, I am learning as I go along. I have been able to download and install Asterisk, Libpri and I have been able to get Asterisk up and running. I have several questions: 1 .I can call the Asterisk server from my Xten phone and it picks up. I have 3 computers (one of them being the asterisk server) I can seem to call

Hello, For the moment I have the following settings in my sip.conf. I want to optimize them to archive the following things: - for the moment all my users will re-register too often. I want that only lagged users to re-register quickly. - check from time to time all users but no too often to see if is logged and can be called. Overall i want only lagged users to reregister and users with good

Martin, If I understand whay you are suggesting, there is a problem in your design: It will only work if you use Hide NAT. The problem is that the ip_src == IP0 rule is wrong: The ip_src is not changed by the router and it is not equal to the IP of any of the machine interfaces. Can you think of a solution that will work in the following reasonabl scenario: Lets say I have two T1 internet

Hello, Please forgive me if I'm repeating this post. I have searched and looked for similar problem with a solution but have not see a similar one. My outgoing SIP and other channels work fine but the incoming/inbound SIP call goes straight to Broadvoice voicemail. I see that Broadvoice is registered when I look at the SIP registry. I have turned on SIP Debug and it is below. Anyone know

I have done additional testing and I haven't been able to figure out why it's failing. Since my original testing we now set the realm on the authentication section to match what we receive from NEC. It's of the format abc at xyz.com I have verified the md5_cred several times and it matches the user:realm:password formula 3016:insiph at something0a646666.com:3016 where username is

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi! So, before a start to code a patch for this, i thught that it whud be a good ide to ask you guys if this feture exists (and if not, is there any reson whay). I was looking in my sshd logs the other day, and found that a user had failed to do a port forwarding, but i can not find any evidence of sucessful port forwarding... This consernes me, i

Hi Shimizu --- Shimizu Miki <> schrieb: > Hello, > > Doesn''t the vircual nic work at all? After the hint of Timo (http://lists.xensource.com/archives/html/xen-users/2006-05/msg00091.html) I got the virtual xen network card show up in yast. However, when I configured it, yast came up with an error message that''no driver module for the device found''