similar to: Using Asterisk/callerid with "pay as you go"

I am curious how others handle "call out" VOIP and callerid. I have found numerous providers that are cheap and seem to have good voice quality but offer no provisions for callerid. I find it almost useless to use call out when the receiving party gets some bogus callerid number that has no relation to my call. I understand the big thing is spoofing callerid but are there any companies

Asterisk Users, I am thinking about selecting CALLWITHUS as my sip provider. Has anybody ever used them? How was the call quality? DTMF Tones issues? Thanks in advance. -John _________________________________________________________________ Gear up for Halo? 3 with free downloads and an exclusive offer. http://gethalo3gear.com?ocid=SeptemberWLHalo3_MSNHMTxt_1

Being rather new to Asterisk I was wondering what the current status of AEL2 is? I see reference to it back in January but that was many versions ago. Is it in the current code? Doug **************************** * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307 * * * * doug@crompton.com * * http://www.crompton.com * ****************************

Does it matter if you use upper or lowercase rules - I.E. - "x" vs. "X" or mix them? Not that I would do that as a rule but sometimes you make mistakes! Doug **************************** * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307 * * * * doug@crompton.com * * http://www.crompton.com * ****************************

Ok asked this earlier with no response so I will phrase it a different way. I am sure someone had to deal with this and there is a "best way." I want to let Asterisk make the decision on best path based on local exchange - xxx-yyy - where xxx is one of my local area codes and xxx is exchange designator. The problem is that the list is rather large. Maybe 50-100. The idea is that I can

Ok trying this again... is there anyone using the SPA-3000 with * ???? I am not sure if this is a specific problem to it or not. This is something I really need to fix!!! When dialing out using * interfaced to an SPA-3000, fxs,fxo, I cannot access (reliably) DTMF menus at the called party, after call completion. Dialing DTMF is fine. I checked by calling myself. Listening to either end on a

I am getting the following error at the end of 'make install' 1.2.9 I have not tried to find it but I suspect there is just a misplaced punctuation. It runs fine. Doug + program documentation now or later run: + + + + make progdocs + + + + **Note** This requires that

Recently started using * and really am having fun. One problem I encountered... I am using an SPA-3000 3.1.10d When I have transfer enabled - 'T' in the dial string I cannot reliably send DTMF keys to a bank, voicemail, or other service requiring tones. If I disable (remove transfer option) from the dial string all is fine. I would like to be able to use features but the ability to have

Not being very happy with festival I would like ro get a better TTS engine. I looked at the listings at: http://www.voip-info.org/tiki-index.php?page=Asterisk+sound+files+international but I would like to get user input on suggested packages for Linux. Best performance vs. cost ???? Doug **************************** * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307 * *

Hi, While at LCA this year I was talking to James Turnbull, and I think he said it was possible to a) pull files off a puppet client to the puppet master b) push those files out to other puppet clients I''ve been reading up on FileBucket and I don''t understand how I can achieve that, or if I''m on the right track. Whats at

This might be a crazy idea, but it just popped into my head, and I wanted to know if it''s possible. Perhaps not possible right now, but possible in a theoretical sense. Is it possible that puppet could be modified to be used to manage switches that have a command line based interface? When I manage our Allied Telesis switches (which have a CLI similar to cisco IOS) I wonder if I could

Hi, I was trying to register a VoIP trunk in Asterisk , where its keep on sending Register message to the server, where I am not getting any response from server. But whereas if i register in Xlite softphone the account is getting registered. I suspect it could be network related issue, but since in softphone it is getting registered from the same network. Any ideas to isolate things would be

I am trying to use a simple tool in the Dial plan so that if the first number does not connect the logic will go to the second and/or third. Basically, I want the call to ring and connect to the first number Then, if it is not answered I want another number to try to get connected Then, if second number does not answer I want the third to be tried i only list the scenario for the first two

Hi, I've tried to get my sarge desktop booting under the xen hypervisor, and have been having some problems. I've asked about it on the xen-users mailing lists, the #xen irc channel with oftc, and on the sage-au mailing lists, but not had any answers at all. I can boot into the system, but the boot process seems to freeze during the /etc/rc2.d/S* scripts. It used to freeze while

Hello all, I have seen some chatter again about DTMF. I see most of the talk about DTMF around not being able to get an external IVR to recognize digits, not a big issue for me at this time but sill interesting. My issue though, is with talk off on a zap channel. It seems to be getting worse or maybe my patience is thinning. All my calls go out and come in pstn through an FXO as I do not

I have just started working with Asterisk and have run into a road block concerning IAX and an inbound DID from callwithus.com. I am getting nowhere and I don't really know how to isolate the problem. The asterisk version is 1.2.7 on ubuntu, sits behind a firewall with iptables. I can connect and make a call to other internal extensions using zoiper and iax. When I try and use the number,

Hi Everyone, I'm planning my first asterisk box, and I'd like to know what SIP providers everyone likes. Voipjet? Gizmo? Somebody else? Thanks, Alex

Hi, I''m trying to develop a manifest to setup a new puppet master. To solve the SSL certificates I''ve created a root CA outside of puppet, and have generated an intermediate CA for the new puppet master to use. I''ve also configured my puppetmaster daemon to use it''s own ssl directory. So the new puppetmaster is at the same time a client of the old puppet

Hi, I'm running the Debian samba 3.0.10-1 package. I have 'log level = 0'. However my logcheck is showing a lot of noise from samba. Are the following messages normal, or do we have a misconfiguration? (We don't seem to have any problems accessing the shared samba shares): Mar 31 12:13:31 sd01 nmbd[3150]: [2005/03/31 12:13:31, 0]

Is what I am trying to do in this context possible. That is changing the incoming CALLERIDNUM. In this case if the incoming CALLERIDNUM is not preceeded by a "1" I want to add a "1". Often calls come in without the preceeding "1" and this plays havoc with my redial if the 3 digit area code matches a local 3 digit extension. All my outside calls are 10 digits or 1+10