search for: email_secur

Displaying 20 results from an estimated 20 matches for "email_secur".

2006 Dec 12
5
Input on Dundi
...;s instock http://www.bochterservices.com/?t=TFdid For Information on PBX Systems for SOHO http://www.bochterservices.com/?j=PBX&t=email BUY Coins, Silver and Gold http://www.bochterservices.com/?j=gold&t=email For new and used security items http://www.bochterservices.com/?j=store&t=email_security
2006 Nov 13
1
SIP Ports (1000 to 2000 works)
...hterservices.com (Cellular) 1-712-432-5401 (Voip PBX) Free World DialUp: 780-217 EXT: 250 WebSite: http://www.freeworlddialup.com/ BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=gold&t=email For new and used security items http://www.bochterservices.com/?j=store&t=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=plating&t=email
2006 Dec 04
5
any possibility of Vonage Integration
Hello, Is there any possibility of integrating plans of vonage with asterisk. Regards Vijay Gandhi
2006 Nov 18
2
AdvancedVoIP Billing ?
Hi after 2 mounth of search, i don't have see a billing solution for my small business.. i see only AdvancedVoIPBilling but i don't know if he can work's with Asterisk. I am search a billing software for the invoice of my custumers, no Calling Card. but i don't see a small and simple product for this. thanks bye
2006 Nov 19
2
Question on CDR Database
Hi I have a small question on CDR Database: It's used by billing software no ? he have only one structure of data or they have multi structure with more information logged ? sample: cdr simple and cdr_extended thanks bye
2006 Nov 22
1
Welcome to Join Asterisk MSN Groups!
:), welcome to join MSN groups: Asterisk-Users@hotmail.com, Asterisk-Dev@hotmail.com, and Asterisk-Biz@hotmail.com! Add to your msn friend, and "/help" for help! Have a good time here ! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061122/87772451/attachment.htm
2006 Dec 03
1
G729 Passthru?
I have a SIP carrier which accepts only G729 connections from my Asterisk server. If all the server does is Dial() (out) two legs of a call which are natively bridged, with no processing the media (and no DTMF detection, etc), do I need to install a G729 codec of my own? All the media from each leg connected to the other is already encoded into G729 by the SIP carrier from which it's coming
2006 Dec 08
3
Vonage SIP access via asterisk?
Does anyone have a working connection to Vonage via asterisk? (SIP, not ATA) I just signed up to test their service and they sent me a Number, Proxy, port and password. Every reference I have tried leaves me with a 404 error coming from Vonage. If you have a working setup, please post some config references. ? Thank You, Steven BerkHolz Soon to be known as HIROTEC AMERICA
2006 Oct 27
2
DTMF detection problem in PABX trunk
Hi for all, i 've installed asterisk with isdn trunk with alcatel pabx. When alcatel users dial for external numbers, a channel on asterisk is allocated for dial. When we call to an number that is an IVR the digits isn't recognized by IVR. In sip.conf i putted dtmfmode as rfc... and info, inband is only for 64k codecs, and still don't work. How can i resolve this issue ?? Thanks.
2006 Nov 16
2
POS Terminals
Hello List - I've got a question regarding POS terminal transactions (credit card machines, ATM, etc...). Currently we setup customers in the following manner: Customer Location --> Data T1 --> DataCenter -> PSTN Termination We are currently using Mediatrix gear for fax transmissions from the customer location, but they don't seem to handle POS modem sales very well. Does
2006 Nov 13
2
STUN with one public and one private IP?
I'm finishing up deploying an Asterisk (Trixbox) box at work. Wow, I thought Asterisk was cool by itself, but Trixbox has made just about everything turnkey. Great stuff! So... we're using Grandstream GXP-2000 handsets to connect to the Trixbox, which sits on our DMZ with a single public IP. I need the phones to work from random places behind NAT, as well as in the office. I'm using
2006 Nov 26
3
Looking for toll-free US did
I am looking for a toll-free US 1800 DID which can be setup quickly . I have seen nufone there quality is very good but they charge for 30 seconds minimum ( others do 6/6 i guess ) . east coast gateway server preffered . . Plz lemme know if you have some suggestions i want it to be setup very quickly :) . Thx . -------------- next part -------------- An HTML attachment was scrubbed... URL:
2006 Dec 06
3
Asterisk freezes when DNS not working: a BUG??
Hi, I'm using Asterisk 1.2.9.1. I have big problem with SIP VoIP providers registrations: Asterisk freezes when it cannot (re-)register with VoIP provider (registration timeout). The problem is related to DNS names resolution: if DNS server is very slow to respond Asterisk stops every activity (no zap or restart commands on CLI). The bad news is VoIP providers usually do not give their IP
2006 Oct 24
10
Meetme... No channel type registered for 'zap'
When I call meetme: exten => 1000,1,Answer exten => 1000,n,Meetme(|||d) Asterisk is complaing with: -- Executing Answer("IAX2/xxx.yyy.142.204:4569-2", "") in new stack -- Executing MeetMe("IAX2/xxx.yyy.142.204:4569-2", "|||d") in new stack -- Playing 'conf-getconfno' (language 'en') Warning, flexible rate not
2006 Oct 22
3
G.729 operating on outgoing only
Greetings list, I have an older Dell Poweredge server running Asterisk 1.2.13. I have installed 5 licenses for G.729 from Digium. I have 5 SIP trunks through a US provider. When my system makes outgoing calls, they go out as G.729. However, when an incoming call comes in, my server does not indicate to the provider's server that G.729 is an option, so the remote server sends the call
2006 Dec 05
0
[Fwd: RE: any possibility of Vonage Integration]
...ME ONLY * * * >>>http://www.bochterservices.com/?t=TF(NM)did >>> >>>BUY Coins, Silver and Gold >>>http://www.bochterservices.com/?j=gold&t=email >>> >>>For new and used security items >>>http://www.bochterservices.com/?j=store&t=email_security >>> >>> >>> >>> Paul wrote: >>> >>>>Brad Templeton wrote: >>>> >>>> >>>> >>>>>On Tue, Dec 05, 2006 at 11:36:12AM -0500, Al Bochter wrote: >>>>> >>>>> >...
2006 Oct 26
6
SIP v IAX2
...-1254 EXT: 250 (Voip PBX) Free World DialUp: 780217 EXT: 250 (Voip) Cellular: 712-432-5401 http://www.BochterServices.com/?t=Email BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=gold&t=email For new and used security items http://www.bochterservices.com/?j=store&t=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=plating&t=email
2006 Oct 18
4
Findme problem
Greetings all, I've been working on having Asterisk put a call through to two different numbers, and give the call to the first one that acknowledges by pressing the 1 key. I found an example on the wiki, but I can't get it working. When I answer the call I hear the message telling me to press 1 to connect, and as soon as the message is done, the call is connected. In other words, it
2006 Oct 23
4
Where to best start looking for voicemail/moh sound quality problem?
I'm running Asterisk 1.2.13 on a Solaris 10 X86 box behind an IPCop firewall on a 5Mbps down/512 up cable connection. I'm having sound quality problems when users call in for voicemail and with music on hold. The sound is choppy and muffled while souding pretty good for calls inside the network. I'd appreciate some pointers as to where to start looking to improve things. I've
2006 Nov 01
6
Java Web Phone
Hello list partners you know about a softphone made in java attachable in a web page? GNU! Thaks in advance!______________________________ Visita http://www.tutopia.com y comienza a navegar m?s r?pido en Internet. Tutopia es Internet para todos. -------------- next part -------------- An HTML attachment was scrubbed... URL: