similar to: Question on CDR Database

Displaying 20 results from an estimated 600 matches similar to: "Question on CDR Database"

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 Dec 12
5
Input on Dundi
Ok, I am looking for some input on using dundi. Is anyone using dundi? And how is it working out? -- Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VOIP PBX) 1-866-638-1254 (Voip PBX) Free World DialUp: 780-217 WebSite: http://www.freeworlddialup.com/ We have Toll Free DID's instock http://www.bochterservices.com/?t=TFdid For Information on PBX
2006 Nov 13
1
SIP Ports (1000 to 2000 works)
I was reading the posts and someone said about the default 1000 to 2000 I see in the .conf the default is 10000 to 20000 I found a service that gives inbound DID's in the firewall 5060 and 10000 - 20000 is setup no workie on the DID But when I set 5060 , 10000 - 20000 and (Unblocked) 1000 - 2000 Now the DID works fine. So you me what the standard is -- Best regards, Al Bochter Bochter
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 Dec 04
5
any possibility of Vonage Integration
Hello, Is there any possibility of integrating plans of vonage with asterisk. Regards Vijay Gandhi
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 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 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 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 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:
2006 Dec 20
13
Need quality toll free 800 number over IAX?
Hi List I need a quality US 800 DID over IAX for my Asterisk server, preferably one that doesn't cost the earth. Any suggestions please? Thanks -- Chris Blunt Entropy IT Ltd -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061220/4919f3cb/attachment.htm
2006 Dec 15
2
Bandwidth requirements for 1, 000, 000 minutes a month
This may expose my ignorance, but here goes :) I've been asked to figure out how much bandwidth would be needed to handle 1,000,000 minutes a month. Here's the environment: ) All calls are received via SIP. ) All calls use the ulaw codec. ) Calls average 10 minutes in duration. ) The "busiest" hour will account for 10% of the daily total. This is how I'm figuring
2007 Feb 25
7
Sending Email From the dialplan
I have looked around with no luck. Does anyone know of a way to send an email from the dialplan. The system that I am working on has none thing to do with VoiceMail. This is something like the SMS command but using sending email I am working on a prepaid alarm dispatch program for Asterisk if anyone has any input please let me know. I will be more than happy to write the code as Open Source