similar to: Strange voicemail things

once u press 9 is there a way to make it so it restores dial tone, like most pbx's do? so dial tone , 9, dialtone, then ur local num -- 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

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:

Can anyone get to www.nufone.net? Is their VoIP down? -Matt

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

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, I had to start using Dovecot on a machine as the new OS does not come with Cyrus IMAP anymore. After multiple problems, I managed to get everything working, including LDAP authentication against the (old) Novell LDAP server. Anyway, the authentication is supposed to be migrated to the new Windows AD. For other tools, I successfully migrated the config to use AD, but somehow Dovecot does not

Hi all, I have been looking at various alternatives for a system that runs several Linux & Windows guests. So far my favorite choice would be OpenSolaris+ZFS+RAIDZ+VirtualBox. Is this combo ready to be a host for Linux & Windows guests? Or is it not 100% stable (yet)? Greetings, Evert This message posted from opensolaris.org

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

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

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

I have a samba PDC with a master openldap server and a samba BDC with a slave openldap server. Replication is done with slurpd with a TLS connection and the slave ldap server has an updateref pointing to the master (I don't use ldaps). On each domain controller my smb.conf contains: passdb backend = ldapsam:ldap://localhost Now I'd like my ldap servers to reject non TLS connections

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

Hey team, I'll be at sea on and off this week, and as such my Internet access will depend on wifi availability while in port. 22a seems stable, but if any critical problems arise, feel free to prepare a release and bug Alfie to upload it. On the brighter side, I expect to be extremely bored while offline so I'll probably get some logcheck work done. <: Cheers, -- [ Todd J.

I am loving Asterisk! I have a Cisco 7960 (Sip) on which I want to try using g729 encoding. I cannot find a setting for this in the phone's interactive screen menu. Do I set it in the sip.conf file? I have also ordered 2 licenses from Digium. My understanding is that because this Cisco phone can handle the encoding, * just passes it thru. Is this correct? Also, I am using LiveVoip for

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

I would like to upgrade my system to a 64 bit machine. I'd like to find a bare bones platform to build on. I'm not looking to spend a lot of money on this as it is a home system. I looked on the CentOS sponsor page but only saw hosting services. I haven't kept up with hardware in years so I'm dumber than dirt on what's out there. I would prefer a desktop so I can stack it.

You need to disable referral following in /etc/ldap/ldap.conf (or whatever applies to your system) Aki On 13.09.2017 14:34, Garry Glendown wrote: > Hi, > > I had to start using Dovecot on a machine as the new OS does not come > with Cyrus IMAP anymore. After multiple problems, I managed to get > everything working, including LDAP authentication against the (old) > Novell LDAP

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.