search for: scenarion

...? Do I have to transfer > some > other ports on Firewall2 because of Grandstream RTP connection ? If > everything works on outgoing calls there must be some little condition > that > gets changed when I try to reach grandstream - what could that be ? > > If anyone has working scenarion, please be so kind to help. > > > Regards, > > Rob. > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update opt...

...& unable to resolve it as per mail reply from one of our colgn niranjan.ashok. Following command is not working on my system ie $ svn co http://svn.easysw.com/public/cups/branches/branch-1.3/. Its showing error as " could not resolve hostname, host not found". Following is the actual scenarion :- I have a very good query related to printer sharing using samba. >> Following >>>> is the scenario... >>>> I had installed HP laserjet 9040Dn printer on redhat Enterprise linux > 5. >>>> Shared it using samba. Used winbind protocol to integrate linux...

...a that is passed > to the socket doesn't include the prompt itself, like '>'. The socket > is a redirekt from the client's > stdout/stderr. > If I use a standard client like OpenSSH the client echoes back the > prompt after a command is executed, > but in my scenarion it's lost and never written to stdout/stderr(my > socket). > Does the SSH client remove the prompt in some way before data is > passed to stdout/stderr ?? > Many thanks in advance ! > /Niclas > --- Jeremy McMillan

...chine does Grandstream connect RTP ? Do I have to transfer some other ports on Firewall2 because of Grandstream RTP connection ? If everything works on outgoing calls there must be some little condition that gets changed when I try to reach grandstream - what could that be ? If anyone has working scenarion, please be so kind to help. Regards, Rob.

Hi List, I am making outgoing call from asterisk to GSM network with the help of VoIP trunk(SIP trunk) then I am not geting any caller ID at destination end. Is this the asterisk issue or VoIP trunk issue? Is this is due to asterisk then how we solve it? I already user Set(CALLERID(num)=XXXXXXXXXXX) in dialplan. ----- Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer

Hello Someone who has working lmhosts please tell me where it is (i.e., the path). I RTFMd man lmhosts and: 1) man hosts tells: SYNOPSIS: /etc/hosts. 2) man lmhosts doesn't tell anything about the path. 3) man lmhosts: "It is very similar to the /etc/hosts file format" - does it mean that the similarity is also in the placement of the file? 4) All config files are in

...Hash: SHA1 All, are there any plans to allow FreeBSD jails to bind to more than one IP address? My scenario (virtual hosting) : 3 front-end hosts with 2 interfaces each, one on the public network, the other on a private subnet. 1 back-end host, providing NFS mounts for the front-ends. This scenarion is not uncommon in ISP environments, usually with a big Netapp Filer in the backend... I'd like to be able to bind each jail to both public and private IPs, so the frontend hosts can mount filesystems off the backend. Cheers, s. - -- It strikes me that cats are like soft-tipped ammo, th...

...not encrypted, so the server restarts unattended in case of a reboot. I have already looked at man pages and a few online tutorials, but frankly they are not clear on what to do to achieve all and _only_ what I wrote above. Most documentation, when not outdated, seems targeted at much more complex scenarions. Is this sequence of actions and commands correct and complete for my case, or not: 1) cd /usr/share/ssl 2) modify openssl.cnf to have your Common Name and other parameters 3) run: ./CA -newca ./CA -newreq-nodes 4) move the private key from the .pem file to a separate file 5) put the...

...some > > other ports on Firewall2 because of Grandstream RTP connection ? If > > everything works on outgoing calls there must be some little condition > > that > > gets changed when I try to reach grandstream - what could that be ? > > > > If anyone has working scenarion, please be so kind to help. > > > > > > Regards, > > > > Rob. > > > > > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinf...

Hi All ! I have a little question about the shell that is run when establishing a connection towards an SSH server. The client(OpenSSH) displays a prompt(as usual) when a command is executed, but my question is, where does the prompt come from. Is it sent by the remote shell or is it handled in the client ?? The reason I ask is that we have developed a product that redirects stdin/stdout/stderr

https://bugs.freedesktop.org/show_bug.cgi?id=57278 Priority: medium Bug ID: 57278 Assignee: nouveau at lists.freedesktop.org Summary: [xf86-video-nouveau] flightgear crash when loading scenary Severity: critical Classification: Unclassified OS: Linux (All) Reporter: king.infet at gmail.com

Hi all, As the resident Linux guru, I've just been tasked with costing a webmail setup for about 600 000 users. They each have 10MiB (small, I know) mailboxes. The current setup has about 40 million web page accesses per month. Has anyone here any experience with this kind of thing? If so, any pointers as to software and hardware used, and any other advice would be appreciated. TIA --

Hi Can someone please help with the following, We are using asterisk@home 1.5 and SIP trunks to communicate to the PSTN network. We are having some problems with the call quality. Although we can hear the other person's voice quite clear when making or receiving a call, we get complaints from the people on the other end saying that our voices sound very unclear, low and that the voice

Hello, we want to setup the following scenario: - each user has a softphone AND a hardphone - the softphone is started with the operating system - the hardphone is connected all the time using SIP - only ONE extension for each user Both phones should ring when the user is called. We've setup an asterisk 1.4.18 and at the moment only the last registered client rings. In Asterisk 1.2 the