similar to: ekiga register problems

Hello all, Normally I just use pri's with our asterisk systems, but a request came in to add some normal pots lines to the setup. We have 3 lines, and they run into the fxs ports. They hit the dialplan just fine, and they always dial the "s" extension. However, my question would be... Is there a way to determine what number was dialed and have it forward to a specific phone? With a

Hello. I'm a bit puzzled at the moment. I have a x100P and TDM400 with 4 modules (extensions). Asterisk CVS-02/01/04-06:55:30 Part of my extensions.conf says this: ; Zap Phone #1 ; exten => 204,1,Dial(Zap/2,20) ; Ring for 20 seconds exten => 204,2,Voicemail(u${EXTEN}) exten => 204,3,Hangup ; Unavail voicemail if extension doesn't answer exten

greetings, using: CentOS 6.7 current KDE 4.3.4 ekiga 3.2.6 i am having problems getting ekiga to make any type of connection. i have gone thru documentation and troubleshooting manuals with out finding reason other than; ~]$ ekiga -d 4 2>&1 | grep "PDU is likely too large" ~]$ echo 3600 > /proc/sys/net/ipv4/netfilter/ip_conntrack_udp_timeout \ bash:

I'm having a problem getting a good clear output sidnal from Ekiga to a VOIP conference call using the Ekiga.net free conference call system. I'm told that each time I speak, my voice is clear & intelligible for about .5 - 2 seconds, but then it starts to be garbled, sounding like the sounds R2D2 makes. I've used 2 or three mic/headsets - two plug into my audio I/O sockets on my

Hello, does anyone else have a problem with Asterisk crashing right after a valid password/PIN is entered when trying to access voicemail in the 1.4b3 version? Not sure if this is anything to do with "realtime" per se but I keep getting the asterisk process bail on me as soon as a valid PIN is entered. Anyone? Cheers Ranj

ekiga registration fails. I've set nat = yes ( also blank ) and i've set externip. Anybody have a sip.conf that works? Here's the sip debug: Reliably Transmitting (NAT) to 86.64.162.35:5060: REGISTER sip:ekiga.net SIP/2.0 Via: SIP/2.0/UDP 10.10.11.180:5060;branch=z9hG4bK17818198;rport Max-Forwards: 70 From: <sip:test at ekiga.net>;tag=as64618445 To: <sip:test at

Dear All I have installed Asterisk on my centos 5.0 and I have two other centos 6.0 and centos 5.6 with ekiga sip client. The centos 6.0 can make successful sip calls but centos 5.6 cannot. Among the Asterisk logs, I found that the centos 6.0 has ekiga 3.2.6 but centos 5.6 has ekiga 2.0.2 . How can I install/upgrade my older ekiga? Thank you

I think I'm able to connect with Ekiga, at least it reports "registered". Curiously, when I exit Ekiga and switch to SFLphone, it isn't able to connect with the exact same parameters; it just says "trying" and never resolves. I'm not able to test outside connectivity because of too many hops: thufir at doge:~$ thufir at doge:~$ sudo sipsak -vv -s sip:thufir

Hi all, after having developed a theora video codec plugin for Ekiga/OPAL based on libtheora I would like to ask if someone could please commit the enclosed patch to the theora svn trunk. It - fixes cross-compilation for windows using minGW - allows to disable the build of the examples via configure switch This is required for our cross-compiled win32 build of Ekiga since some dependencies of

Hi! I want my PBX to be reachable at my ekiga.net account. It seems I am registered: vajna2*CLI> sip show registry Host Username Refresh State Reg.Time ekiga.net:5060 magwas 585 Registered Sat, 13 Nov 2010 13:48:22 However when others try to call magwas at ekiga.net, they find me unavailable. My asterisk

I am experiencing a "606 not Acceptable" error trying to set up an Asterisk server as an ekiga.net client. My server is behind a firewall with NAT routing. I have googled this problem and read about Asterisk feeding its local ip address to ekiga.net. That seems to be my problem. I tried putting stunaddr=stun.ekiga.net into the sip.conf file under [ekiga]. I also tried

Hello, Asterisk 1.4.26.2, FreeBSD 8.0-RELEASE We have a very simple setup, using SIP softphones and a simple diaplan as follows in the examples below. When I dial the 700 extension it asks me for the extension and password, and it always says "login incorrect". The mail system send the email ok and Ekiga shows that I have vaoicemail, so the only thing that is failing is the actual

My wife bought an inexpensive store brand web cam, so she can do video conferencing with people using Microsoft NetMeeting. Probably it is not supported in Linux and we will need to buy a supported web cam. I installed the Ekiga package. Is there something easier to configure and use or is Ekiga the best way to go? TIA! -- Lanny --------------------------------------------------------- Over 800

dear All, when first installing CentOs some 6 months ago, I noticed this strange thing called Ekiga. Now, some guys at work use Skype and I heard that's owned by Microsoft. So enter James. I bought a webcam, actualy 2, to test Ekiga. Logitech Webcam c210. I saw this url :http://www.ideasonboard.org/uvc/ on this url :http://wiki.centos.org/AdditionalResources/HardwareList/Webcams and in

As usual, the BerkeleyTIP group programming project is to learn about & work toward implementing our own Asterisk VOIP conference server, during our meeting. 10A-6P Pacific USA (-8H GMT) = 1P-9P Eastern US = 6P-2AM GMT. http://sites.google.com/site/berkeleytip/ Join the VOIP online conference & help out, or chat with your buddies. :)

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi All, Has anyone been successful in making ekiga's chat functionality work with Asterisk. This is a really neat feature and it would be awesome to finally see it working. - -- Alan Hanley FSF Member 4949 "No matter where you go , you're probably lost" -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using

Hello, I have a Sip phone (Siemens C470IP) which works perfectly with different VoIP providers (iptel, betamax, ovh...). It also worked well with my testing server (ubuntu and inside the LAN). But now the problem i have is that the hardphone doesn't connect to my dedicated server (debian lenny / Asterisk 1.6.2.13). The strange thing is that ekiga can connect to the same asterisk server with

Question was off topic for the thread, but I'm feeling helpful today. More of a 1234... make install modprobe usb-uhci modprobe zaptel modprobe ztdummy -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Brad Sumrall Sent: Monday, March 19, 2007 13:17 To: 'Asterisk Users Mailing List - Non-Commercial

I'm trying to get this zap dial to work. I want to send DNIS and ANI to other system (ZAP/g2) at answer, while the caller hears ring (RBT). I hear the RBT, but callerid and exten is not sent to other T1 - The ZAP/g2 T1 is standard D4, SF, E&M Wink Start. - At ZAP/g2 wink, asterisk should send DTMF "*ANI*DNIS*" exten => _XXXX,1,NoOp,${CALLERID} exten =>

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