similar to: A working SIP Phone for Centos44?

I try to setup networking with a fresh CENTOS4.4 installation. Upto now I have 20 installations, most of them FC3/FC4 and RHEL3. I noticed that the default gateway is not setup properly when using Centos. [root at raaf ~]# route Kernel IP routing table Destination Gateway Genmask Flags Metric Ref Use Iface 192.168.101.0 * 255.255.255.0 U 0 0

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...

I want to set sflphone as extension on asterisk. I have a sip account/DID with vitelity.net. Not sure what to put in the wizard: alias ??? hostname ??? is this the asterisk server hostname, or the hostname where my sflphone is sitting on the lan (it's a home network) username ??? is this the assigned extension number? password ??? is this the assigned extension number password? Any

I thought this was easy, but it turns out to be not so obvious. My problem: I have a database with hour registration. I need to print out about hundreds of pages and have them signed by employees. Of course I want to do this automatically. Up to now I have a collection of html pages (got them using wget). So the final thing I want is to print them all out in one go. How to do that? I search

hi all, can anybody help me to give Opensource SIP client information which can be modified as per our requirment regards Dhaval -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090518/802cc3ac/attachment.htm

This is a A8N32-SLI Deluxe motherboard with a AMD Athlon(tm) 64 X2 Dual Core 4400+ Processor. It freezes and cannot be accessed from the network. Keyboard/mouse/display, everything just stuck. It had an uptime of 17 days after I re-connected all cables and boards after a similar crash. My feeling is, this must be a hardware related problem. I inspected the logs under /var/log, but nothing.

This one puzzles me a lot: [thba at vink layout]$ nc script/xw_functions.ample NEdit: getcwd() fails: No such file or directory NEdit: getcwd() fails: No such file or directory [thba at vink layout]$ ll script/xw_functions.ample -rw-rw-r-- 1 thba thba 16829 Oct 25 16:59 script/xw_functions.ample [thba at vink layout]$ pwd /home/thba/workarea/colibri/design/ana/layout [thba at vink layout]$

-- 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

Hello, I have an error while try to install twinkle: # yum install twinkle [...] Resolving Dependencies --> Running transaction check ---> Package twinkle.i386 0:1.2-1.el5.rf set to be updated --> Processing Dependency: libresolv.so.2(GLIBC_PRIVATE) for package: twinkle --> Finished Dependency Resolution Error: Missing Dependency: libresolv.so.2(GLIBC_PRIVATE) is needed by package

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,

hi wich soft phone do you recomend but i need this feature it must ask for user name and password when it start. i know xline and zoipper but they dont have that i can acomplish this whit twinkle but i need it for Windows :-( any ideas? thanks -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination. -------------- next

Hi, I've successfully setup Asterisk on my local PC and can make call using Twinkle to the server. But, I cannot call to my Asterisk server at Rackspace. I have been trying several things to figure it out, no luck. My PC is behind NAT, so I've set that up in sip.conf (nat=yes). I can ping my Rackspace server so it seems to be Public-static IP. Anyway, I tried with setting externip,

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 =>

-----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

Hey, all. I'm looking -- if possible -- for a decent, multi-platform soft-phone. Specifically, Linux and Windows; that way, I'll go through the same issues my end users do. I've noticed a couple (e.g., minisip, which seems abandoned, and sip-communicator, which, honestly, is probably a great IM client, but has a confusing interface for actual phone calls). So I'm wondering if

I tried to install justvoip several times and I cannot install it. Can somebody tell me how to install it on ubuntu? Meybe next version of WINE will support it?

Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb: > What kind of phone are we talking about, both yours that works and your > wife's that does not? Right! > Can you ping the unreachable phone and does it respond to a ping? I can ping both phones from the VM > Many phones will have a network test function built in to them to help you > determine if the phone

-----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 =

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

Dear all, sorry for the Off-Topic but I have an Astreisk 14 voip server connected to Twinkle and X-Lite clients. I have to use the GSM codec for all of my clients, and it was set up in the sip.conf specifically in "allow=gsm" line. Twinkle has GSM codec built in, but when I open X-Lite audio codecs settings I can't see the GSM codec, being that the official web site and the PDF