similar to: VoiceGlo

Hi, I have a deltathree account and I can place calls but I can't receive calls. I use Grandstram sip phones. When I call my deltathree phone # the voicemail is answer :(( I need some help and solutions from the guys who allready are using deltathree. I search on Internet and I try all types of configurations... :( This is my configurations files: - sip.conf - [general] port = 5060

<P>Has anyone managed to get Asterisk to work with Glophone/Voiceglo since this posting.</P> <P><A href="http://lists.digium.com/pipermail/asterisk-users/2004-February/036559.html">http://lists.digium.com/pipermail/asterisk-users/2004-February/036559.html</A></P> <P>I've tried copying the config in this listing with no success. </P>

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

i cannot see caller ID of the call originated from outside zap channel. i hv configured both zapata.conf and extensions.conf. i m right now in india i think asterisk only supports Bellcore enable caller ID. so is it the same bug of BT caller ID problem in UK? or it is the bug of my asterisk configuration? i hv enabled callerID from my TELCO. -------------- next part -------------- An HTML

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

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

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.

No, you don't need a sound card. Do you have ztdummy loaded or zaptel device in your system? Regards, Gus ----- Original Message ----- From: "Chris Hariga" <contact@techselesta.com> To: <asterisk-users@lists.digium.com> Sent: Sunday, October 19, 2003 8:19 PM Subject: [Asterisk-Users] Music on hold... > Hi, > > I need a sound card and mpg123 for music on

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

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

Hi, Can U tell me the Vonage ATA 186 settings? I would like to try to have a web interface on my adapter :-)) Best regards, Chris Hariga

Skipped content of type multipart/alternative-------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 2280 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20031012/70396f74/smime.bin

Skipped content of type multipart/alternative-------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 3179 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20041007/4b01c29c/smime.bin

I am setting up 2 nufone lines. I want to make them both availiable for dial-out. How do you syntax it in extensions.conf so that it figures out which one is avaliable and dials out on it. Also how do you setup the name part of callerid for the outgoing lines?

Hi, si posible on SIP phones to have the dial tone after 9 like on the FXS card? I set ignorepat => 9 on my extensions.conf... Best regards, Chris HARIGA

Hi, I have Fedora Core 2 running with a T1 card. I try to put the log on db but I get the error: Aug 20 15:17:47 ERROR[262160]: cdr_addon_mysql.c:378 my_load_module: Failed to connect to mysql database asteriskcdrdb on localhost. The database exists and I try with "mysqlaccess localhost asteriskcdrdb" and I get: Access-rights for USER 'localhost', from HOST

Skipped content of type multipart/alternative-------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 3179 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20041007/98aec720/smime.bin

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

Hello, I have several * servers behind a SER server (in a local ip range). The SER server is also publicy reachable. On the other site, I have SIP clients that are behind another NAT or in the same NAT range as the * server. Can someone give me some directions/hints etc. on how to make this work. I think I should be using MediaProxy with SER. But do the SIP clients need to register at the SER

Hi all I have been googling and wiki-ing and have found a number of potential solutions to my questions, but I don't want to have to play about for too long and risk messing up my * box now I've just got it working, if one of you kind folk could offer your 2 penneth, (being a Brit I'll have none of this cents business ;] ). I want to send an SMS message whenever I get a voicemail