similar to: Accessing to ekiga.net through 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

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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! I am trying to redirect to a local extension the incoming calls that I receive to an account which I have in iptel.org, but when receiving I'm obtaining this error: alderamin*CLI> -- Executing [300 at from-internal:1] Dial("SIP/danib-089f8820", "SIP/300|30|tTrm") in new stack [Feb 19 19:22:50] WARNING[19254]:

sip.conf.sample: ;register => 2345:password at sip_proxy/1234 ; ; Register 2345 at sip provider 'sip_proxy'. Calls from this provider ; connect to local extension 1234 in extensions.conf, default context, ; unless you configure a [sip_proxy] section below, and configure a ; context. sip.conf: [general] context=default allowoverlap=no udpbindaddr=0.0.0.0 tcpenable=no

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

Hi all! I have recently purchased an Athlon X2 3800+ and I should like to install FreeBSD or OpenBSD in a HVM domU on Xen 3.0.3-1 dom0 (Debian packages). Is it possible? In such case, can someone recommend me a guide/howto to accomplish it? Thanks in advance. Regards, Daniel -- Daniel Bareiro - System Administrator Fingerprint: BFB3 08D6 B4D1 31B2 72B9 29CE 6696 BF1B 14E6 1D37 Powered by

Hello, I get some problem installing asterisk + ekiga on my debian etch: ii asterisk 1.2.13~dfsg-2 Open Source Private Branch Exchange (PBX) ii ekiga 2.0.3-4 H.323 and SIP compatible VOIP client $: asterisk -U asterisk -vgc give me some WARNING like : ,---- | WARNING[21806]: res_musiconhold.c:852 moh_register: Unable to open | pseudo channel for timing... Sound

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

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 guys, I'm trying to write hangup causes from asterisk into the CDR record. Using version 1.4.24.1 at the moment, but no joy so far. Has anyone implemented this? Neeraj Chand Support Analyst Fiji Islands Australia T: +6793342526 T: +61388924326 M:+6799344012 New Zealand www.ocis.com.au T: +649 980 7022 -----Original Message----- From: asterisk-users-bounces at

Hi, I'm curious if anyone knows of any possibility to use video VOIP client (like Ekiga or Linphone or...) under Linux that could be operated by touchscreen friendly GUI (bigger buttons, large keypad, etc...) ? I like Ekiga, but GUI is small and cannot be operated via touchscreen... But maybe there are some skins for existing clients that are more touchscreen friendly ? Thanks in

I will install Asterisk on my home server, I want to be able to route video calls, but I need the Windows and Linux clients to be interoperable. On Linux, it looks like Ekiga is a good candidate. But how about Windows? Anyone using Kapanga in an Asterisk network that includes Ekiga? Are these two interoperable? I'm not necessarily looking for open source software, free for personal use is

Hi, I have configured a certified asterisk 13 server with chan_mobile and res_pjsip. I have a Cisco 7940 hardphone and I use ekiga as softphone client. Now the problem is, using the hardphone I'm able to call the softphone and hear everything properly. But when I call from the hardphone to some number that has to be dialed via chan_mobile, I'm not able to hear what the other side says (I

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

Hello list, I'm hoping that you could read through this mail and give me some tips on how to improve my setup (functionality, security, really anything). It's my first Asterisk installation and meant for simple home use. I installed Asterisk 11 on an OpenWrt Barrier Breaker router. Currently it's configured for Ekiga so I can test. In a few weeks I'll change to a Telco SIP

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

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

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

I run CentOS at home, not just at work... Anyway, I've got a friend in Chicago who recently mentioned that they'd like to do videocalling. Now, I've heard of skype, but a quick google says there's some problems on Linux. I also see ekiga, and aMSN. Anyone here run such a beast, and have any recommendations or comments? Obviously, must work on CentOS, not Ubuntu, or Fedora, or

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