similar to: Glophone/Voiceglo and Asterisk

Hi, VoiceGlo is comercial version of Asterisk? :))) loooooooooollllllllllllllllllll Take a loock on http://www.voiceglo.com/ The softphone is IAX :) Best regards, Chris HARIGA Techselesta Inc. http://www.techselesta.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031201/307c10e9/attachment.htm

The call quality is really pretty good. I think better than Vonage over an FXO bridge. If you are looking for a home provider with direct SIP support and local phone numbers this is a good choice. If anyone has questions or comments about my configuration please pass them along. I have noticed that if you don't put fromuser=phone# then the extension caller id passes through. Also the

I found a message in the archives from Cameron Palmer on 23 Dec regarding his voiceglo SIP configuration. Unfortunately (for me), the archive has his email address removed. So, Cameron -- or anybody else using voiceglo with their * box -- please reply to me so that I can get your email address and ask you a question about your setup. Thanks, Greg

Hello all, I've been trying to get a simple PBX up and running with asterisk. I decided to sign up with Voiceglo so I could have a PSTN gateway. The problem is that I can't seem to get Asterisk to handle dtmf decoding reliably. I tried inband and the rfc decoding. inband tried to work and the rfc mode didn't do anything. By try to work I mean that it rarely properly

>> Somehow sysvolcheck is using a LOWER CASE 'f' in the GUID folder name >> for the default GPO! >> >> Where is this coming from? Of course, in Windows this doesn't matter. >>> But in linux it is a showstopper. > I think it is coming from AD. > You will probably have to rename the GPO in AD, possibly along with the > 'name'

I feel dirty. I have some graphs I'm building to communicate chargeback rates and service usage for our backup system here at the University of Florida. These come down to daily data points on a graph of number-of-bytes transferred and stored. Since we chargeback on the same basis (price per MB this, price per KB that) the same chart with a different scale can be used to communicate bytes

Sprint Nextel is sueing vonage, voiceglo and theglobe.com for infringing on VoIP patents. Sprint Nextel claims to have about 100 patents on VoIP technologies. Does anyone know which ones this article is talking about, and if so does asterisk have any of those features? The reason I am asking is that the article is vague, Vonage uses a fairly standard codec set, I dont know about the others.

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Hi All, I'm at the end of my tether here and would really appreciate some help. I'm trying to implement DTMF based pause/resume of call recording. I'm using Asterisk 1.4.22.1. Here's the scenario: The caller (SIP or ISDN, doesn't matter) dials into the asterisk which executes the following code: exten => _X.,1,Monitor(wav,${CALLDIR}${UNIQUEID},mb)

Is anyone signed up with Vonage and using an asterisks box?? Also what VOIP providers would anyone recommend? -- James Moran Potential Technologies http://www.potentialtech.com

I am new to Asterisk and currently work mainly with Cisco Callmanager. With Callmanager I can setup partitions and call search spaces to determine where a given phone can and can't dial. Does Asterisk offer this type of functionality, and if so how? Blake Parker -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I've been trying to use BroadVoice as a SIP service provider. They don't officially support * but are helpful when it comes to answering questions for setup parameters. They claim they have no firewalls or access lists that need to be set up so I can get access to their servers. However, something's still not quite right when I use the parameters. It looks like our Asterisk

<P>I am running on Core 3 also with a voicepulse account.</P> <P>I found this document quite helpful....www.voip-info.org/tiki-print.php?page=Asterisk+Fedora+Core+3</P> <P>I did deviate in that I ran my make of Asterisk itself as follows</P> <P>cd /usr/src/asterisk</P> <P>make clean <BR>make linux26 <BR>make install <BR>make

I constantly get the following error messages in /var/log/asterisk/messages: Dec 8 10:52:57 WARNING[1009521664]: File chan_iax.c, Line 3324 (iax_ack_registry): Received unsolicited registry ack from '192.168.0.1' Dec 8 10:52:57 WARNING[1009521664]: File chan_iax.c, Line 4181 (socket_read): Registration failure Where 192.168.0.1 is another asterisk server. Below are the local and

A few months ago, the friendly folks from liveVOIP went under. We had some discussion on how to limit our losses, and my recommendation was a chargeback, since "FTTP Services" -- their CC merchant -- wasn't affected by the bankruptcy, as far as we could tell. Today, I received this from my CC company: http://muware.com/asterisk/livevoip.pdf Anyone else got lucky?

Hi Anyone know how to use regcontext et regexten parameter from sip.conf and can give an example ? thx regards Harry -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090807/ef9ba45e/attachment.htm

I set up an asterisk box with a broadvoice sip connection for incoming connections it works great when I use a cell phone, vonage line, calling card to call the asterisk box, but when I try to call it from our verizon land line it is busy and asterisk logs do not show incoming call. Any ideas on what the issue is? Thanks! Randy

So it looks like Livevoip went Bankrupt ------------------------------------------- There is a Federal Court Order in place and has been since Friday early a.m. ALL Suppliers are now under a Court Order that prevents them from terminating any and all services to LiveVoip LLC. If they take such any action they will be in direct violation of a U.S. Federal Court Order. If you have any questions

I just installed tapi and some app called identapop pro. I havent tested incoming calls yet but so far, I cant get calls out using outlooks. I configured TAPI for asterisk inside outlooks and I set TAPI to these configs: TAPI connects using the manager to asterisk without problems. As channels I configure this: User channel: SIP/myphone and the phone actually rings when I tell outlook to dial

Hi all, I have to questions about queues. Member is a phone like SIP/myphone and only one member in the queue. At first, DIALSTATUS doesn't return any status. How to now if a call in queue has been answered or if caller just hangup? Second, how to deal with timeout, I have strange behaviors. If I put timeout=60 in queue.conf and I call the queue passing also 60 as timeout value,