similar to: question about asterisk setup...

I've beaten myself bloody dealing with this one... No luck so far. In summary, incoming calls from Gizmo establish, but neither get nor send sound. Outbound calls to Gizmo work fine (well a bit choppy but work) My thought is that the SIP connection is being made fine, but the RTP is getting stopped / blocked / misdone somewhere. Here is the thing: Asterisk 2.5 on Linux (No hardware

Hi Jorge, > I'm trying to use the PPL (Parma Polyhedra Library, > bugseng.com/products/ppl) in one of my LLVM passes getting the error: > >> Error opening >> '/home/jorge/SvnReps/Systems/llvm/build/Debug+Asserts/lib/MyAnalysis.so': >> /home/jorge/SvnReps/Systems/llvm/build/Debug+Asserts/lib/MyAnalysis.so: >> undefined symbol:

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

On Tuesday, November 15, 2011 at 10:11:25 (+0100), Duncan Sands wrote: >> Hi Jorge, >> >> > I'm trying to use the PPL (Parma Polyhedra Library, >> > bugseng.com/products/ppl) in one of my LLVM passes getting the error: >> > >> >> Error opening >> >> '/home/jorge/SvnReps/Systems/llvm/build/Debug+Asserts/lib/MyAnalysis.so':

Hi all, I'm trying to use the PPL (Parma Polyhedra Library, bugseng.com/products/ppl) in one of my LLVM passes getting the error: > Error opening > '/home/jorge/SvnReps/Systems/llvm/build/Debug+Asserts/lib/MyAnalysis.so': > /home/jorge/SvnReps/Systems/llvm/build/Debug+Asserts/lib/MyAnalysis.so: > undefined symbol: _ZNK23Parma_Polyhedra_Library13PIP_Tree_Node2OKEv >

Hi, I am on amd64 Linux and not really too happy with twinkle, linphone and ekiga. Unfortunately, X-Lite and Zoiper, even though they provide Linux versions (w00t!) have only x86 versions for download. Do you guys know of amd64 versions of those, or can you recommend other softphones that will run on amd64, or which come with source code? Thanks, -- martin | http://madduck.net/ |

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, 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

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

I am presently configuring a test Asterisk 11 server based on CentOS-6 and I need to employ a softphone for testing. The base repo has ekiga. The EPEL repo has twinkle. I lack the knowledge of whether other packages exist or might be better suited. Which of the two do you recommend? Or, alternatively, what other package might be a better choice? -- *** E-Mail is NOT a SECURE

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

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

Abhishek, In reverse order 3/ yes it is freeware, though some of the termination boards are available for sale from www.digium.com 2/ yes you can interface to Cisco handsets running SIP. 1/ Does it have a gui interface - the short answer is no. The longer answer is depending on what you mean, if you mean programming - then no though a number of people have developed sql interfaces. If you

-----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 =

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

Hi, I'm searching for GPLed softphones. I found WengoPhone but actually not available for Asterisk PBX, only for Wengo network. I found Kiax but only for IAX protocol. Did you know a good GPLed softphones which works on Windows ? Thanks Greg -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi there everyone! I use asterisk as a home pbx. My internet connection is a DSL one, and I have a Linksys WRT54G that nat things for me in a 192.168.X.X style network. I've installed asterisk on my mac, and tried several examples I've found on the net (voip-info, gizmo, etc.) about how to create a Gizmo and a FWD trunk. However, all my attempts failed. The FWD thing kinda

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'm getting a compile error when trying to compile the flex output of ppl.l in tools/wpp. This is under Debian (Woody) with latest updates. Can anybody help me out? Enclosed is the output of the build and folling that, the installed packages. I have also tried the latest version (20030408) with identical results. TIA Luke ================================================================