similar to: forward call intended for another domain

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, 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

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 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

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, 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

Getting our own VOIP conference server going will be the 2nd part of the ProgrammingParty until it is accomplished. - The first part will be getting Ekiga ver 3 working on KUbuntu 8.04, & whatever other OSs people have. Come help out on the BTIP conference server, if you like. :) ===== Ekiga(Gnome meeting), Asterisk, Xen, Virtualbox, Debian 15 Years, Free and Open Future, Amarok, ZFS,

Voice over Internet Protocol (VOIP) using Asterisk, Sameer Verma & work on the Programming Party to help get our own Asterisk VOIP conference server working. :) == Join with the friendly productive Global FreeSW HW & Culture community, in the TWICE monthly, Voice over internet Global Conference: BerkeleyTIP-Global: GNU(Linux), BSD, & All Free SW, HW, & Culture TIP = Talks,

Hi, I am setting up an asterisk server for testing purposes and cannot get voicemail to work at all. My host OS is Linux From Scratch 6.3 and the asterisk software versions I built are zaptel-1.4.5.1 and asterisk-1.4.12. I am using the Ekiga softphone on my Ubuntu desktop machine. My asterisk server and client phone are on different computers but are on the same LAN, i.e. no NAT. I have an

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 list, I'm trying to set up a video call from my Ekiga client to a Grandstream GXV3140 IP-phone. The call succeeds but there is no video. I have in sip.conf : videosupport=yes disallow=all allow=alaw allow=g726 allow=g729 allow=gsm allow=h261 allow=h263 allow=h263p allow=h264 The Grandstream peer has codecs (sip.conf) : gsm;alaw;g729;h261;h263;h263p;h264 The Ekiga peer has codecs

I'm trying to setup a demo where my Asterisk box with a TDM01B (FXO) answers an incoming call and forwards that call to a SIP softphone (X-lite.) Seems all is built/installed okay: # ztcfg -vv Zaptel Configuration ====================== Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. I'm pretty new at this and the extensions.conf file is eating my

On Sun, Sep 20, 2020 at 03:13:28PM -0400, Randall S. Becker wrote: > On September 20, 2020 2:02 AM, Damien Miller wrote: > > OpenSSH 8.4p1 is almost ready for release, so we would appreciate testing > > on as many platforms and systems as possible. This is a bugfix release. > > I will be testing this shortly on HPE NonStop platforms. > > Side question: We now have

(Im' answering cc the list, so the knowledge keeps there, and maybe some more qualified answers become). Am Mittwoch, den 15.10.2008, 18:00 -0700 schrieb Francisco del rosario: > Hey Rodolfo... Need some help from you ... > I need to know what hardware do I need to make SIP calls if I set-up > asterisk > So the situation is that I have a PC and configure the software of my PC to

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 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! 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

hi All, its probably very simple but i can't find the way to it. i have some b410p cards and use them to connect to ISDN2, this works OK for calling but i need to have 1 line to be send to the fax machine. the fax machine is a modem connected on another machine with hylafax. as far as i can figure out i need to set 1 of the slots, the one leading to the fax, in the b410p in NT mode by

Dear R users, ?rgamma gives me : rgamma(n, shape, rate = 1, scale = 1/rate) rate: an alternative way to specify the scale. The Gamma distribution with parameters ‘shape’ = a and ‘scale’ = s has density f(x)= 1/(s^a Gamma(a)) x^(a-1) e^-(x/s) Should I understand that scale=1/rate ? Is it written somewhere ? Then

-----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]: