similar to: IAX Problem

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA256 I have an installation where I'll have a site to site data DS1 for use between two corporate offices. We'll have one asterisk server at each office. I'd like to be able to route calls over the 24 channels on that DS1 between the offices, instead of over the voiceT at each location to maximize savings on interoffice calls. An

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

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

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

I have a list that looks like this ... > have <- list(a=7,b=3,c=1) > have $a [1] 7 $b [1] 3 $c [1] 1 and I want to have a simple way to change it to the following without re-typing the values ... > desire <- list(a=c(7,7),b=c(3,3),c=c(1,1)) > desire $a [1] 7 7 $b [1] 3 3 $c [1] 1 1 In other words, I need to create the list in desire from the list in have. In my

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

Hey all, just compiled xen 3.4.2 and linux 2.6.31.12. I followed the instructions on the Xen wiki to get the install running. my issue here is that in my grub memnu config i set dom0_mem=512M when i login to the dom0 and i do a free -m i get that the system only has 432 under the total section. booting into the stock debian kernel 2.6.26-2-amd64 kernel gives me 8192 free memory for i did

Hi, I have small confusion in my configuration which is I had some DID's like 044578900-04457999. I was configured dial plan below mention. exten => _0XXXXXXXXX,1,NoOp(Int exten:${CALLERID(num)}) exten => _0XXXXXXXXX,2,Set(outgoing_ident=0445789${CALLERID(num):-2}) exten => _0XXXXXXXXX,3,NoOp(Ext ident:${outgoing_ident}) exten =>

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

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

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

Hi, I just installed Samba 2.2.2, using the default installation directories, on a RH 7.2 system. It seems to be up and running fine, but I still have the old 2.2.1a files installed the way RH likes them. I'm not a guru, so should I worry about this? Should I uninstall the 2.2.1a files? Will there be any conflicts? Should I care? Many thanks. Joe -- Joe Follansbee Streaming Media

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