search for: bochterservices

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 Systems for SOHO http://www.bochterservices.com/?j=PBX&t=email BUY Coins,...

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

...0 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 Services http://www.BochterServices.com/?t=Email Are you outside of the US? Do you need to call US Toll Free Numbers? We can help you save money on calling US toll free numbers. Email for information: usTollFree@bochterservices.com (Cellular) 1-712-432-5401 (Voip PBX) Free World DialUp: 780-217 EXT: 250 WebSite: http://www.freewo...

To everyone on the list I put a site on line the URL is *http://bochterservices.com/phpbb/ *This is for any information on Good or Bad ITSP You can post any problems you had with the provider You can Vote on the provider This is for allowing multiple viewpoints to be heard. If a provider receives a bad review, they are more than welcome to post So long as the exchange is fa...

...------ > An HTML attachment was scrubbed... > URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061217/d90792e3/attachment-0001.htm > > ------------------------------ > > Message: 2 > Date: Sun, 17 Dec 2006 23:58:42 -0500 > From: Al Bochter <Al.Bochter@bochterservices.com> > Subject: Re: [asterisk-users] Good Commercial Grade Service Provider? > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <45862002.7040803@bochterservices.com> > Content-Type: text/plain; charset=&quo...

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

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

Hello, Is there any possibility of integrating plans of vonage with asterisk. Regards Vijay Gandhi

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

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

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

...of working on G729 or GSM codec, because linksys boxes by default use G711, which consumes hell lot of B/w. >>> >>> >>> Regards >>> >>> Vijay Gandhi >>> >>> -----Original Message----- >>> *From:* Al Bochter [mailto:Al.Bochter@bochterservices.com] >>> *Sent:* Tuesday, December 05, 2006 4:06 PM >>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion >>> *Subject:* Re: [asterisk-users] any possibility of Vonage Integration >>> >>>Brad Templeton, >>> >>>Thats a very g...

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

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.

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:

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

...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 for others to use code. With help from the list. -- Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email

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

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

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