similar to: problem registering to ekiga.net

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

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

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

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:

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

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

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

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

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

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

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

returning newbie. Trying to register ekiga for the first time to my asterisk server only. [204] user=204 context=internal type=friend secret=xxxxxxx insecure=very canreinvite=no host=dynamic disallow=all allow=ulaw allow=alaw nat=no Can anyone tell me what I am missing? I am not behind NAT or a firewall -------------- next part -------------- An HTML attachment was scrubbed...

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

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

-----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 =

So, I have been testing ekiga 3.0 with Asterisk, and sadly, it don't work. I am told by the ekiga devs in http://bugzilla.gnome.org/show_bug.cgi?id=553595 and http://bugzilla.gnome.org/show_bug.cgi?id=553810 that the problem is that Asterisk does not support SIP forking. The issue is that I have multiple addresses on my workstation: 2: eth0: <BROADCAST,MULTICAST,UP,LOWER_UP> mtu 1500

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

Hello list, I have in sip.conf : /maxexpiry=60 ; Maximum allowed time of incoming registrations ; and subscriptions (seconds) minexpiry=60 ; Minimum length of registrations/subscriptions (default 60) defaultexpiry=120 ; Default length of incoming/outgoing registration ;-----------------------------------------

Hello, I want to optimize my registrations and calls of peers to my asterisk with the following options in sip.conf: ---///--- qualify = yes qualify = 500 qualifyfreq=5 registerattempts = 0 registertimeout = 10 maxexpiry = 60 minexpiry = 20 defaultexpiry = 600 ---///--- Can someone more experienced with these settings to help me to optimize connections from peers with mobile phone that using