similar to: making * more like a normal pbx

When I call an extension (say my extension 1000)and it goes directly to voicemail the first time, it does exactly what it should do (plays announcement and then records, the second time when i call back (within about a minute), it goes directly to a beep (for recording), no announcement. Another thing, during this time when I call 0 (my voicemail access number) it gives me a fast busy. any

Hi everyone, I've got my * system up and running and I'm really pleased. I've gone with G.711 (alaw) and I've stumbled across a problem; when people place calls internally some people think they have been cut off if the line is quiet for a few seconds. Is there a way of getting comfort noise on the call? I'm using the STABLE release and cisco 7960 phones under FC-1 Cheers

anyone know a company that will terminate did 831/408 area codes in california. FYI i already checked voicepulse, negative. -- Gafachi.com - referal code hunter81 instant iax termination - 2 cents a minute Also they have a great referal program, tell them jacob, hunter81 sent you -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type:

I've done something similar at home, but made my dialplan such that I can dial either 10 or 11 digits locally. I don't use a "throw away" digit at all. Any 7, 10, or 11 digit call will be appropriately mangled and sent out the PSTN / VoIP provider. ________________________________ From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On

Hi, I'm putting together a fax solution for my company that utilizes T.38. I wanted to throw out my plan and get some feedback if anyone is doing something similar or sees a blatant problem with it. We're currently rolling out SPA-942 phones for the standard desk phone with vanilla Asterisk 1.4.15 (just upgraded it today) on the back end. Most calls for satellite offices are handled by

Hello Asterisk-Users, For some reason my CDR records for disposition and billsec are not working correctly. I always receive a 0 for billsec and the disposition is always at "NO ANSWER', even when I grab the calls. I experience this with Asterisk 1.6.0.1 and Asterisk 1.4.22. Here is information on how I do the call: -----------------------------------------------------------------

Hi, folks. I'm having a heck of a time trying to get outgoing T38 faxing (I don't need inbound right now) working with FFA and Gafachi. G711 faxing works (as well as can be expected over the internet), but I want the higher reliability of T38. I'm running Asterisk 10-beta1. When I drop my callfile in to make the call, I get this: -- Attempting call on SIP/18884732963 at

We're having an issue where we call ReceiveFax in a context that includes a hangup extension and half the time dialplan execution doesn't continue after the fax is received successfully. Am I missing something simple here? Below is a sample call where this happened: The last log line for this channel/call is: [Feb 21 09:10:53] VERBOSE[13730] res_fax_digium.c: -- Channel

I saw a couple of posts about this in the archive, but none seemed specifically to address the problem I am having. If I missed something please let me know. Right now I would classify myself as "novice," and there is probably really nothing so trivial that I couldn't possibly have screwed it up. :-) I'm trying to use the AMD command to detect answering machines, and have

I just signed up with livevoip for 800 DID and have very choppy audio. From PSTN to my asterisk is ok but asterisk to PSTN is terrible. I am using IAX and was assigned to server iax01.nyc.*. I do not believe it is a bandwidth problem on my end and I have no problems using iax with gafachi. I opened a ticket with livevoip but no response yet. Would I be better off using sip with them? Is there

Hello ppl, Am looking at some PSTN termination providers in US. If this question has been repeated, please point me to the correct link, as I've tried searching the archives but have been unsuccesful so far. I have come across quite a few companies which provide the same, such as : Iconnecthere <http://www.iconnecthere.com> Vonage <http://www.vonage.com> Teliax

Hi list, we want to connect asterisk to an traditionnal PBX (EADS 6550/Matra). People from telco told that they can't connect two PBX's using E1/T1 or only with QSig signaling. I wanted to use EuroISDN. In this case, it was me told that VN6-VN7 would be used. The PBX has a spare ADQ card installed on which we would connect. Has someone a such working setup? Is it working well? More

Dear All, I have a setup where three errision PBX's (head office & 2 branch offices ) are @ three different location. Head of has a errision BP250 where the branch officers has Errision BP50's. I wana connect all three PBX's through IP link's. My idea is to connect BP250 ( head office ) to asterisk. This has to be done using analog interfacing coz there is no E1 / or T1

Hi, all. I've got an old Telrad PBX with an Emagen(?) voicemail box. The VM box, itself, is beginning to show its age. Big-time. We're thinking it might be time to look for a replacement. I'd love to install Asterisk with an FXO card or something, but I don't think it supports whatever protocol legacy PBX's used to speak to VM systems. If someone can tell me I'm

Hi All, I have been reading through the list quite a bit, and I am going to post this more as a poll than anything else. I am working on setting up a very small business with something that resembles a professional voice system. My idea is to use Asterisk with a SIP provider and SIP clients. I currently have a Vonage account already. So adding the 9.99 a month Soft Phone would be easy.

Hi All, i want to make an extension from pbx1 able to tlak to another extension from pbx2 or use pbx2's trunk to dial outside calls. so i edited in both servers accordinally the iax.conf: register => pbx1:pass at 172.16.200.175 <pbx1%3Apass at 172.16.200.175> [pbx2] type=friend host=dynamic trunk=yes sercret=pass context=[default] ; i used the biggest context to avoid confusion as

Does anyone have experience with melding Asterisk with an older Executone PBX? I have a client whose existing voicemail server(repartee) has become bonkers and we need to stick a VM system in there asap. I thought asterisk would be a good thing to use. Does anyone have experience with the older Executone PBX's and asterisk? Any caveats, any tips, any things I should be aware of? Thanks,

Hi, We are using MixMonitor to record the call. When the call is bridged, the latency is significant. We tried to increase the internet speed and the server RAM and processor speed and still we are having that issue. We use VoiceTrading and Gafachi's Termination minutes to make calls. As we are in US and VoiceTrading in Europe, somebody suggested to move the termination minute provider

Hi All, We recently had an old office building burn down. The office housed maybe 20-30 people. Only about 10 or so of those had their own extensions. We had a standard pbx from an area communications company, and I'm not quite sure about what kind of phone lines were there, I only know that their were actually 3 phone numbers, but everyone could get an outside line if they needed

Dear all, I wanna know what can I do to get the PBX's clock from -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090428/3218b8b0/attachment.htm