Hi, I am just about to move out from my parents home and think about how I will phone from now on. In Germany there is a provider (QSC) who offers DSL (1024 down/256 up) with fastpath without volume or time limits. Does anybody know a comercial (or even semi-professional) provider who lets me dial out through H323 (or another protocol) and also offers an number where I can be called from normal phones? If you dont know one in Germany are there others in other coutries who earn money with this service? Might be a way for a poor student to save some bugs... Bye, MartinD: -- +++ GMX - Mail, Messaging & more http://www.gmx.net +++ Bitte l?cheln! Fotogalerie online mit GMX ohne eigene Homepage!
Hi Martin, There is a provider in the US -> www.AddaLine.com, who just launched a SIP service with some great rates for North America [Int'l coming soon]. With your SIP account you also get an actual phone number with several area codes to choose from. I have been using their service for months and I am extremely happy with the service. -Erik -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Martin Dommermuth Sent: Sunday, June 08, 2003 9:21 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] VoIP Provider Hi, I am just about to move out from my parents home and think about how I will phone from now on. In Germany there is a provider (QSC) who offers DSL (1024 down/256 up) with fastpath without volume or time limits. Does anybody know a comercial (or even semi-professional) provider who lets me dial out through H323 (or another protocol) and also offers an number where I can be called from normal phones? If you dont know one in Germany are there others in other coutries who earn money with this service? Might be a way for a poor student to save some bugs... Bye, MartinD: -- +++ GMX - Mail, Messaging & more http://www.gmx.net +++ Bitte l?cheln! Fotogalerie online mit GMX ohne eigene Homepage! _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users
Hi, * Erik Lagerway wrote/schrieb:> > There is a provider in the US -> www.AddaLine.com, who just launched a > SIP> service with some great rates for North America > > I have been using their service for months and I am extremely happy withthe> service.looks like Germany is again laggin behind all others in the communication field. Or I asked at the wrong place. There might not be to many people from Germany in this list. Anyway, thanks for the answer. CU MartinD: -- +++ GMX - Mail, Messaging & more http://www.gmx.net +++ Bitte l?cheln! Fotogalerie online mit GMX ohne eigene Homepage!
Iconnecthere seems to have better rates... -----Original Message----- From: Martin Dommermuth <mailmartin.neu@gmx.de> Date: Thu, 12 Jun 2003 19:48:43 +0200 (MEST) To: asterisk-users@lists.digium.com <asterisk-users@lists.digium.com> Subject: Re: [Asterisk-Users] VoIP Provider Hi, * Erik Lagerway wrote/schrieb:> > There is a provider in the US -> www.AddaLine.com, who just launched a > SIP> service with some great rates for North America > > I have been using their service for months and I am extremely happy withthe> service.looks like Germany is again laggin behind all others in the communication field. Or I asked at the wrong place. There might not be to many people from Germany in this list. Anyway, thanks for the answer. CU MartinD: -- +++ GMX - Mail, Messaging & more http://www.gmx.net +++ Bitte l?cheln! Fotogalerie online mit GMX ohne eigene Homepage! _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users
Hi Everyone, I know this may be off subject but I am not sure who to ask. I am currently looking for voip termination that is closest to replicating U.S. pots service. I run I.V.R. systems and I want to point Sipura 2100's to a voip terminator and have the DTMF tones properly detected. All that I need is outbound service and the problem I run into now is that when the called party presses a key on the phone it does not play it back properly to my system. I have tried to dial through voxee and plain voip and they both have the same problem. Im not sure if this is an asterisk issue or what. When I dial through packet 8, aptella or vonage everything works fine. I think my problems are because I am going through their asterisk servers. If anyone can help I would appreciate it, there is a potential for me using thousands of minutes per day if I could only find compatible service. I use the generic term U.S. Pots service because my dialers work perfectly on normal analog phone lines. I've been looking for service for 2 months and I haven't had any luck. P.S. I do not need any special services, just proper DTMF tone handling. Mark Adams Infinity Marketing 1-800-430-1478 Main 530-579-8856 Fax <http://www.vistaprint.com/vp/gateway.aspx?S=5176697856> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060128/706c76c4/attachment.htm
On Jan 28, 2006, at 6:50 AM, Mark Adams wrote:> > > Hi Everyone, > > I know this may be off subject but I am not sure who to ask. I am > currently looking for voip termination that is closest to replicating > U.S. pots service. I run I.V.R. systems and I want to point Sipura > 2100?s to a voip terminator and have the DTMF tones properly detected. > All that I need is outbound service and the problem I run into now is > that when the called party presses a key on the phone it does not play > it back properly to my system. I have tried to dial through voxee and > plain voip and they both have the same problem. Im not sure if this is > an asterisk issue or what. When I dial through packet 8, aptella or > vonage everything works fine. I think my problems are because I am > going through their asterisk servers. If anyone can help I would > appreciate it, there is a potential for me using thousands of minutes > per day if I could only find compatible service. > > I use the generic term U.S. Pots service because my dialers work > perfectly on normal analog phone lines. I?ve been looking for service > for 2 months and I haven?t had any luck. > > P.S. I do not need any special services, just proper DTMF tone > handling. >This might be a codec negotiation issue with the termination service. I am using Teliax with my asterisk server to terminate my SIP and IAX calls from several ATAs and softphones. All of that works fine with DTMF. I am using the G729 codec exclusively for my Teliax calls. You also need to be sure that the extensions for each ATA/phone have the DTMF configured righteously. HTH, Marty -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: text/enriched Size: 2150 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060128/33aba271/attachment.bin
Is somebody here using a RoadRunner/Time Warner connection and able to successfully with SIP (or IAX2)? We are experiencing high latency up to the point that the voice conversation is not understandable anymore. This goes for both SIP and IAX2. Is anybody willing to share experiences or give tips? Rene Kluwen Chimit
> Is somebody here using a RoadRunner/Time Warner connection and able to > successfully with SIP (or IAX2)? > > We are experiencing high latency up to the point that the voice conversation > is not understandable anymore. This goes for both SIP and IAX2. > > Is anybody willing to share experiences or give tips?I have an employee using a Cisco 7960 over RoadRunner, 15 hops away, working just fine with g711. Some cable companies are known to use rate-limiting devices to reduce inbound/outbound Internet traffic. You might ask their tech support folks if they are using such as box.
Hi, I am looking for voip providers in bay area, any suggestions? My requirements are: handling around 2000 calls a day (incoming) and around 1000 calls a day outgoing. I have a Asterisk PBX server to take care of routing calls to appropriate deparment. So I am looking mainly for IAX2 or SIP protocol support from VOIP provider. Also a dedicated t1 line in case provider can provide this too. Thanks, Nitin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060510/8bc9420b/attachment.htm
Have you looked at CBeyond? I like their T1 SIPConnect product. _____ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Nitin Gupta Sent: Wednesday, May 10, 2006 7:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] VOIP provider Hi, I am looking for voip providers in bay area, any suggestions? My requirements are: handling around 2000 calls a day (incoming) and around 1000 calls a day outgoing. I have a Asterisk PBX server to take care of routing calls to appropriate deparment. So I am looking mainly for IAX2 or SIP protocol support from VOIP provider. Also a dedicated t1 line in case provider can provide this too. Thanks, Nitin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060510/4109929b/attachment.htm
Thanks for the information, I will surely look into it! Nitin On 5/10/06, Kerry Garrison <support@techdatapros.com> wrote:> > Have you looked at CBeyond? I like their T1 SIPConnect product. > > ------------------------------ > *From:* asterisk-users-bounces@lists.digium.com [mailto: > asterisk-users-bounces@lists.digium.com] *On Behalf Of *Nitin Gupta > *Sent:* Wednesday, May 10, 2006 7:04 PM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* [Asterisk-Users] VOIP provider > > > > Hi, > I am looking for voip providers in bay area, any suggestions? > My requirements are: > handling around 2000 calls a day (incoming) and around 1000 calls a day > outgoing. I have a Asterisk PBX server to take care of routing calls to > appropriate deparment. So I am looking mainly for IAX2 or SIP protocol > support from VOIP provider. Also a dedicated t1 line in case provider can > provide this too. > > Thanks, > Nitin > > > > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com <http://easynews.com/>-- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060511/3b4244bc/attachment.htm