similar to: registering Asterisk on SIP/Nortel MCS server

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

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 Sirs, I did the following command (against ldapsam backend): net groupmap add rid=3002 unixgroup=wheel type=local ntgroup=Marketoids comment=Mm -d 10 I just wanted to add new group. But instead of that I saw many-many-many records: lib/smbldap.c:smbldap_search(963) passdb/pdb_ldap.c:ldapsam_getgroup(2008) they all wanted to find group with gidNumber=4294967295, yes, sure, there's

-----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 =

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

Can You please advice me free softphone which supports SIP registrations ? Cheers, Kate -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070807/539fb7a2/attachment.htm

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

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

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 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

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, 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

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, 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

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

Hi, Is the following an (unknown) bug? Rhythmbox (RB) has an option for using a icon-tray. Restoring/minimizing to the tray in combination with switching viewports (VP) seems to confuse compiz. Steps to reproduce: 1. Start RB on VP1. 2. Minimize to tray. 3. Change to VP2. 4. Click the tray icon to restore. Result: RB is restored to VP2, while compiz changes to VP1. Subsequent clicking on the

Hi friends. I am working in a TELCO, we have a trouble with our very old Alcatel Voicemail system (and now we dont have support and worst this system was forgotten for Alcatel) I've used Asterisk for just small jobs, but I've proposed use it and tomorrow begins with the tests :) ... so: we have a thousands of users, how do I have to configure my server Asterisk to works like

Using two Asterisk connected between they, How do I can check the voicemail in a remote system but working like *97? I mean dont want ask the voicemail box, just the password and go to the voicemail of caller. If I have the same extensions in the two Asterisk it doesn't work. Thanks. -- Linux User Registered #232544 Jabber : pepo at jabberes.org Ekiga : pepo at