Re: [Asterisk-Users] SIP -- PSTN gateways
On Apr 7, 2004, at 10:53 PM, Tom wrote: 1. Do any of the standard companies (Packet8, Broadvox, Vonage, etc.) allow you to use your own SIP device (phone or something like *) instead of the interface hardware they usually provide? Many of the providers use SIP, so even if they don't explicitly tell you the SIP settings to us, it is possible to hookup something else. I've heard that Packet8 gets annoyed when people use self-supplied devices, but it would be possible to hack the code to emulate the device identifier. I've seen an Vonage example posted. I've had great luck with VoicePulse's Connect service. They just opened up a bunch of new rate centers, including Los Angeles/Orange County. $7.99/mo per phone number, 2.9 cents a minute for long distance outbound, no minimums or contract. 3. Finally, do any of the providers deliver more than one call via SIP? In otherwords, if I'm already on a call and another comes in will they attempt to deliver it? Good question. I wonder how many of the providers even have the ability to restrict you to one call! I hope the unlimited providers do, because otherwise people using Asterisk as a SIP UA are going to have a field day. VoicePulse Connect does. I had four people in a MeetMe conference last night, all dialing in via a VPConnect-supplied number. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP -- PSTN gateways
Title: SIP -- PSTN gateways So what are people using these days for SIP or IAX to PSTN gateways. 1. Do any of the standard companies (Packet8, Broadvox, Vonage, etc.) allow you to use your own SIP device (phone or something like *) instead of the interface hardware they usually provide? 2. What about latency and reliability? 3. Finally, do any of the providers deliver more than one call via SIP? In otherwords, if I'm already on a call and another comes in will they attempt to deliver it? Thanks -brian
Re: [Asterisk-Users] SIP -- PSTN gateways
On Wed, 7 Apr 2004, Brian Cuthie wrote: So what are people using these days for SIP or IAX to PSTN gateways. I'm setting up my own gateways. I'm getting a Cisco 2621XM with a HDV module. I had high hopes for the Ovislink gateways, but they discard proxied SIP requests, for some unknown reason. 1. Do any of the standard companies (Packet8, Broadvox, Vonage, etc.) allow you to use your own SIP device (phone or something like *) instead of the interface hardware they usually provide? Many of the providers use SIP, so even if they don't explicitly tell you the SIP settings to us, it is possible to hookup something else. I've heard that Packet8 gets annoyed when people use self-supplied devices, but it would be possible to hack the code to emulate the device identifier. I've seen an Vonage example posted. 2. What about latency and reliability? Well, latency is going to be factor of your network and your provider more than anything. If they are bad, it is going to bad. As far as reliablitity goes, perhaps someone else knows more about that. 3. Finally, do any of the providers deliver more than one call via SIP? In otherwords, if I'm already on a call and another comes in will they attempt to deliver it? Good question. I wonder how many of the providers even have the ability to restrict you to one call! I hope the unlimited providers do, because otherwise people using Asterisk as a SIP UA are going to have a field day. Ultimately, you'll need a business SIP connection. There are some providers out there that can do this for LD. Thanks -brian Tom ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
