similar to: VON Developers Conference

Hi all, Anyone got a link or the file to Mr. Guy's notes from his speech at Astricon.. I looked for them or relevant info he showed ad FWD's webpage but have not been able to find them. He had a working FWD registered thru Asterisk behing a NAT and a couple of samples of their new scripts, talking weblog (from Mr. Pulver) and other cool new features. If anyone got these please send em

Hi Folks, Just wondering if anyone can give me some pointers, I'm configuring Asterisk to talk to FWD's new IAX service. The asterisk server is behind an iptables NAT Firewall, with port 5036 forwarded: $IPTABLES -t nat -A PREROUTING -p udp -d $EXTERNAL_IP --dport 5036 -j DNAT --to-destination 172.16.20.200:5036 I can make outgoing calls just fine, but when I receive an inbound call

Is this replacing Astricon this year? If so it looks like a pretty poor showing in comparison to Astricon Dallas last year. Cheers, Dean ________________________________ From: Carl Ford [mailto:carlf at vonmag.com] Sent: Wednesday, 18 July 2007 9:09 AM To: Dean Collins Subject: Announcing Digium/Asterisk World's Conference Program

Anyone able to get me a comp/highly discounted ticket to this? $150 just to visit the exhibition halls sounds crazy? Dean > -----Original Message----- > From: Jeff Pulver [mailto:jeff@pulver.com] > Sent: Tuesday, 23 August 2005 11:47 AM > To: mailinglist1 > Subject: Register Today for Fall 2005 VON: "The Destination for IP > Communications" > > Hi There, >

Hi All, My question is directed at those with kernel compiling experience. I have a driver source xxxxx.c file and a Makefile with it. I want to recompile the kernel to include the driver and rebuild the initrd.img. Where do I copy the file in the kernel sources? I currently have it in a sub folder of the drivers directory in the kernel source folder. There is one more question if I may, it does

I'd like to float the idea of a VON panel discussion for large-scale open-source deployments, specifically using Asterisk as an application service and as a gateway service. In order to do that, I'd need to get a list of panelists together who might be interested in speaking. If you manage an installed base of >1000 users, and you're using Asterisk, I'd be interested in

How does one do translation for calls that come in from other pbx's where the incoming caller ID is an internal extension number on their pbx? Eg. when I get a call from Free-World-Dial the CID shows up as "429102" which is essentially their internal extension number sans any routing prefix. To dial the number back I need to dial the extension with FWD's routing prefix

Hi I am a newbie to puppet. I have tried an initial config serving a simple file & dir from the puppetmaster, but the client will not mount. What obvious mistake am I making, I wonder? puppetmaster is a DNS alias for osiris3. Config files: [root@osiris3 puppet]# cat fileserver.conf [config] path /data/puppet/manifests/files allow *.pinnacle.co.uk [root@osiris3 puppet]# cat puppet.conf

My guess Samba 3.6 (smb v1) Windows 8.1 after update, disabled smb v1 probely. ( wannacry crypto leak) Dont know for sure but that was ms its advice also. I think its time to update your samba. Greetz, Louis > -----Oorspronkelijk bericht----- > Van: samba [mailto:samba-bounces at lists.samba.org] Namens > albert samba via samba > Verzonden: woensdag 24 mei 2017 20:13 >

In case you haven't seen it, the program for this year's LLVM Dev Meeting has been posted: http://llvm.org/devmtg/2011-11/ Note there may be some minor changes to the program before the meeting. We are looking for volunteers to serve as session time keepers. There are 18 sessions, so if there are 8 volunteers, each will only need to cover 2 sessions. Please contact me

Having just returned from four days at the VON show in Santa Clara, I thought I would submit a "highlights" message. I hope others who attended the show will take the opportunity to add, as there was far more to see than I can cover on my own. [VoIP IS BIG] First, I have to say that VoIP is BIG. It is the buzz technology of the day. The show was packed, and everybody there was there

Hello, My Samba has been working well until I recently run Windows update for DC. I got the error message below on my client (Windows 8.1 Enterprise Evaluation OS) when accessing the Samba shares. " *\\pinnacle** is not accessible. You might not have permission to use this network resource. Contact the administrator of this server to find out if you have access permissions.* " I can

Registration for the 2011 LLVM Developer Meeting is now open: qcsurvey.qualcomm.com/survey.asp?pub=y&survey_id=7125 The attendance limit has been increased to 225 this year, but with the growing interest in LLVM, registration will likely still fill up quickly; we encourage you to register early. The 2011 meeting will be held on Friday, November 18, 2011 at San Jose Marriott 301 South

Do you want the best performance from your Lustre implementation? Lustre Administration & Support training will be offered June 5-7 in Boulder, Colorado. Attendance is limited! Click here to register Today! http://www.acteva.com/booking.cfm?bevaid=132736 Cluster File Systems offers Lustre? Training programs that provide the all the knowledge required to start building high performance

I have recently started experimenting with Asterisk. I am running the system the other side of the a NAT router and trying to connect to FWD. I have opened UDP ports and have configured sip.conf to handle NAT. The problem: I can call from the FWD phone and the extension on Asterisk rings and there is two way sound so no problem. Now if in the extension.conf file I have, exten =>

Same thing. It will make sense to try Register => <FWDnum>@fwd.pulver.com:<FWDpass>@fwdnat.pulver.com:5082 but in that case Asterisk sends REGISTER sip:fwdnat.pulver.com SIP/2.0 which is not right. It should be sip:fwd.pulver.com but sent thru fwdnat.pulver.com:5082 BR Borut -----Original Message----- Subject: Re: [Asterisk-Users] Using Asterisk with FWD through NAT From:

I have a small HUGE problem with *. I have installed * but I have 2 problems. 1 - Making call to FWD. 2 - Receiving call from FWD More info of the problem at the end. Here is the sip.conf file. ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = sip ;default Default for incoming calls register =>

Extreme N00b, I am getting the error message "a target does not exist" when running the make install inside the zap directory, probably pretty common, possibly a package I didn't install, just need some insight on it. The same occurs with the libpri and asterisk. -----Original Message----- From: asterisk-users-bounces@lists.digium.com

Hi, My configuration: Sipura 2000 Debian/Sarge Asterisk 1.0.1 built by msp@toshiba on a i686 running Linux I am calling 612@fwd.pulver.com which is Daytimephoneline of pulver.com and for the first second the connection seems to be ok and I hear: <thu .. rsda ..> and nothing more, which suppose to mean <Thursday, ...>. The echo line from 613@fwd.pulver.com does work the same. I never

Using DIAX softphone which seems to be working OK can get to VM/echotest etc in the demo context Am trying to setup FWD but get the following problems Can hear it ringing when dialing FWD no 612 for time. Connects but no sound from remote end. Does anyone have any suggestions. Softphone on 192.168.0.2 asterisk on 192.168.0.3 Netgear RP114 doing NAT to the internet port 5060 being forwarded to