Re: [asterisk-users] asterisk gateway
Try the CT-G1000 from cyber-telecom.net it is 39.99 GBP atm. Sam _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Rojas Sent: Wednesday, January 30, 2008 10:35 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] asterisk gateway Hello everybody Anyone, to know a gateway that works with nextel simm cards? I'm looking for them, in internet, but I did'n look. Best regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk gateway
On Thu, Jan 31, 2008 at 05:59:03AM +0800, Sam Tam wrote: Try the CT-G1000 from cyber-telecom.net it is 39.99 GBP atm. And how do they compare to others? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk gateway
On Thu, Jan 31, 2008 at 05:59:03AM +0800, Sam Tam wrote: Try the CT-G1000 from cyber-telecom.net it is 39.99 GBP atm. err biz again ... Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk gateway
Hello everybody Anyone, to know a gateway that works with nextel simm cards? I'm looking for them, in internet, but I did'n look. Best regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk gateway
Any GSM 900/1800 gateway will work with a Nextel (US) SIM card. However I assume you actually want to register on a local iDEN network and not be roaming internationally (Nextel does not have any GSM roaming partners in the US) That is not possible. On Jan 29, 2008 9:34 PM, Carlos Rojas [EMAIL PROTECTED] wrote: Hello everybody Anyone, to know a gateway that works with nextel simm cards? I'm looking for them, in internet, but I did'n look. Best regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk gateway to other gateways
Though I've had implementations of Asterisk, I havent encountered this one yet, so i'd like to seek your advise if this possible. I would want asterisk to be stand between the dialer the destination. The dialer will now dial asterisk access number. Asterisk will acknowledge user by using CallerID and check against its database for authentication and then sends out a DTMF A tone for second to enable the dialer to send the whole overseas digit. Assume the caller is not in database, asterik could give user a busy tone, IVR or just leave it and sends out a DTMF A tone anyway. Once the overseas digit are sent from dialer to asterik, asterik will then decide which telco/carrier/Voip to send the traffic to using LCR. Please note that we need to assign at least 5-10 telco/carrier/Voip access number for backup purposes. Once the least cost destination is selected by asterik, asterik will pick up the PRI line and dial a local access number and waits for a DTMF A tone. Once the A tone is heard from telco/carrier/Voip, it will send the overseas digit which was sent by the dialer earlier on. Also, can asterik sends out a musical tone or IVR while connecting to other telco to advice user that the call is connecting, else it would be dead air from there on. The whole process takes less than 5 seconds while the user stays on the line for this whole thing to happen. Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk gateway to other gateways
I was wondering why bother with this dialer thing at all :) When you implement a new thing, it is a good idea to go back to basics and think what you want to do. You have multiple voip carriers and you have bunch of users. Based on the number dialed you want to pick a carrier and dial out. May be authenticate a certain caller for certain path. Then end of the day look for CDR's. You do not need to set your call setup this fragmented. How about this; (a) User dials a international number (b) * looks at the CallerID and Destination Number (at one database lookup) (c) This caller is not allowed to dial this destination, so play a message for that extent (d) If caller is allowed, then * finds the least cost root and dial through that path and connect the call. * can do all these. So I would first sit and right down what I want my system to do (in my users perspective). Then try to find a way to implement. I wouldn't try to replace a box with another box. my two cents SW Message: 4 From: lito lampitoc [EMAIL PROTECTED] To: [EMAIL PROTECTED] Date: Fri, 02 Jan 2004 01:04:40 +0800 Subject: [Asterisk-Users] asterisk gateway to other gateways Reply-To: [EMAIL PROTECTED] Though I've had implementations of Asterisk, I havent encountered this one yet, so i'd like to seek your advise if this possible. I would want asterisk to be stand between the dialer the destination. The dialer will now dial asterisk access number. Asterisk will acknowledge user by using CallerID and check against its database for authentication and then sends out a DTMF A tone for ½ second to enable the dialer to send the whole overseas digit. Assume the caller is not in database, asterik could give user a busy tone, IVR or just leave it and sends out a DTMF A tone anyway. Once the overseas digit are sent from dialer to asterik, asterik will then decide which telco/carrier/Voip to send the traffic to using LCR. Please note that we need to assign at least 5-10 telco/carrier/Voip access number for backup purposes. Once the least cost destination is selected by asterik, asterik will pick up the PRI line and dial a local access number and waits for a DTMF A tone. Once the A tone is heard from telco/carrier/Voip, it will send the overseas digit which was sent by the dialer earlier on. Also, can asterik sends out a musical tone or IVR while connecting to other telco to advice user that the call is connecting, else it would be dead air from there on. The whole process takes less than 5 seconds while the user stays on the line for this whole thing to happen. Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
