similar to: point to point T hookup?

Hi I need a small PBX for use on the move. This means that outbound calls will need to be made over the cell phone network. Assuming a small hardware PBX with a spare mini-PCI slot or a USB slot then what hardware options do I have to get an outbound cellular channel? Options need to be rock solid, so no bluetooth to a cell phone kind of solutions need apply. Can any of the 3G usb

On 25.05.2015 21:29, Ilia Mirkin wrote: > Commit 8acaf862dfe switched things over to use TEXCOORD instead of > GENERIC, but did not update the nv30 swtnl draw paths. This teaches the > draw logic about TEXCOORD. > > Among other things, this fixes a crash in demos/arbocclude when using > swtnl. Curiously enough, the point-sprite piglit works without this. > > Signed-off-by:

On 26.05.2015 02:49, Ilia Mirkin wrote: > On Mon, May 25, 2015 at 8:37 PM, Tobias Klausmann > <tobias.johannes.klausmann at mni.thm.de> wrote: >> >> On 25.05.2015 21:29, Ilia Mirkin wrote: >>> Commit 8acaf862dfe switched things over to use TEXCOORD instead of >>> GENERIC, but did not update the nv30 swtnl draw paths. This teaches the >>> draw

I have2 Trixbox Servers. Each has an IAX trunks to the other. One works the other fails: -- Executing [s at macro-dialout-trunk:19] Dial("SIP/526-09eec7c8", "IAX2/InterOffice/210,300,tr") in new stack -- Called InterOffice/210 -- Hungup 'IAX2/InterOffice-7578' == Everyone is busy/congested at this time (1:0/0/1) The only difference I am aware of is

Marcus Meissner wrote: > > In article <ax%R6.33895$hV3.53576@newsfeeds.bigpond.com>, > David Rowe <voicet@bigpond.com.au> wrote: > >1) Why does winemaker (after ./configure, make etc) produce a library and > >sym link as the output, e.g. why not an executable. > > This is the usual invokation method of WINE binaries. Do not worry. This is an excellent

Hi I was doing some bandwidth testing, and my incomming usage is 36% more than my outgoing bandwidth. The setup is IAX2 trunking using GSM codec. Is there any obvious reason I am overlooking to figure out why there is such a big difference between the two.? I am using CVS-head September 3rd, maybe there is a version skew? Any suggestions will be appreciated. Thanks Clive

Not sure if this can be done or not, but I can't seem to find it anywhere on the Wiki. When dialing interoffice with Asterisk 1.4.2, I would like to have the caller id of the person I am dialing displayed and not the number I just dialed. Is this possible? So, if extension 4023 is John Doe, and I dial 4023, my display should read John Doe and not 4023. I am using a Polycom 501 by the way in

Hi All, I''ve got an interoffice IPSEC VPN in place that I''m trying to give priority to terminal service (tcp 3389) traffic. I''ve created rules at each end, but have hit a bit of a dillemma. As the data is encrypted I must also give highest priority to protocol 50 otherwise the priority is lost as the packet gets encrypted. When I do this however, I can''t

The benefits of * are obvious so that part of the marketing an * solution is easy. Anybody care to share ideas on how to target companies who would benefit most from */Digium? It seems to me that it would be an easy sell to small/medium companies who need advanced features such as ip trunking, IVR, Conference bridging, etc., etc. I would like to find a way to identify multi-location companies

After about one weeks time I've gone from no VoIP to a completely configured system for two of our offices to be able to page/communicate interoffice as well as handle existing PSTN communications (okay, waiting onf hardware for the PSTN side and I've likely jinxed myself now). I was sweating getting the two boxes talking to each other and I knocked that out in no time without even

I'm rolling out an installation with snom 360s in the near future. Simple SOHO configuration, 3 FXOs hanging off a TDM400B, 4 snom 360s, a snom 200, some variant of IAX softphone, and an IAXy or Sipura 2002. I have the 360's set up to subscribe and notify for the line use lights, which works like a charm for interoffice calling (between the 360's, anyway. The IAXy, 200 and,

I am thinking about a change to our company's phone "layout" and would like to get comments from people who have done something similar. Currently, we have 3 locations - each with their own Asterisk PBX. The corporate office has a PRI. Each remote location has a SIP provider for 5 channels of SIP going to their own PBX. Interoffice calls use the PSTN. Most inbound calls come to

Dear All, We have a customer who is opening a new office in Dubai and we know that VoIP is blocked over there. Has anyone a solution to getting VoIP back out (we want interoffice calls back to the UK)? We we're thinking of IAX trunking, but not sure if that is blocked or just SIP etc. A VPN works, but is not great. We have seen: http://www.speed-voip.com/voiceguard.html At the moment it

[crossposted to isp-clec and asterisk-users] As part of several larger projects, the question of rate importation from termination carriers has come up. If a firm has four different LD partners, as an example, then it is obvious that the firm needs to determine at the origination of a call where that call would be best sent on a cost basis. I'll ignore things like quality of service and

Hello All, I've been investigating and playing with asterisk to see how it would work out as a small-medium business pbx to handle mostly interoffice/branch communication and a possibly communication out to pstn in later stages of implementation. (All communication would be VoIP internally with possibly 4 pstn lines at the HQ with either a TDM04B or spa3000s and 1-2 pstn lines across 7-8

I am using eyebeam and I am happy with it. However, it is boring just to talk to my son in the other room. Whenever I try to convince somebody to buy eyebeam, they are scared of the price. Is there a free video soft phone available, that will work with eyebeam / asterisk? bye Ronald Wiplinger

In a recent interview someone from Digum indicated that the G722 wideband codec was being worked into Asterisk. This will make Asterisk compatible with Polycom's new HDVoice products like the IP650 phone. This is very interesting, potentially exciting, but it brings up certain questions. Who will benefit as long as calls must typically pass into existing PSTN infrstructure, and so be

David, I guess I'm a little confused here. Are you asking me to provide a "pencil copy" of an LOA for your review? I don't understand why you need an LOA from us. We need an LOA from you to order circuits that will be billed to us that will be attached to your CFA. It was also my understanding that you had an LOA ready to be given to us, which had already been reviewed by

I've looked in the obvious places but haven't found a definitive answer to the following: can an IAX extension (an Iaxy phone, for instance) do call pickup via *8? Adolfo

I'm not sure where to check next, so I'm reaching out to those that know this stuff better than I. I've got Asterisk up and running, but I've still got an occasional audio issue. Once in a while (maybe 1 out of every 20-30 calls), the audio becomes heavily distorted, but only on the local side. The party on the other end says the audio is fine. We can hear them, although