Re: [asterisk-users] Cisco vs Asterisk
Hi, I don't use asterisk since 1.2.x version and never deployed an big project with Asterisk, so I don't know if currently Asterisk can replace to Cisco Unity as Voice Mail, but Cisco Unity is not only for voice mail the main objective is to be part of all Unified Communications infrastructure. Then Integration with Active Directory / Exchange (Lotus Notes) and other features is only possible with Cisco Unity. Maybe I'm wrong and Asterisk can do it.. so I would like to read about that... Rgds. On 7/22/08, voip crazy [EMAIL PROTECTED] wrote: Hello all, A client of us, is thinking to migrate their actual PBX to a Cisco CallManager. We want to sell him an asterisk box to complement the Cisco PBX. I think to use asterisk as a Voicemail server (Replazing the Cisco Unity) Has asterisk all the functionalities to replace a CIsco Unity server? Which functionalities Cisco Unity has than asterisk could cover? How could asterisk complement the Cisco Call Manager funcionalities? Thanks. VoipCrazy. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Omar E.P.T - Certified Networking Professionals make better Connections! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Addpac 2620 don't relay DTMF to PSTN
Hi Guys: I'm using Asterisk with Addpac 2620 as gateway, internally I'm using Grandstream BT200, unfortunately when I called to external phones (PSTN), and I have to choose some extensions, the Phone don't dial the extensions, I believe that DTMF relay in ADDPAC is not working well. I'm using RFC 2833 and ALaw for SIP Channel (Between ASterisk and ADDPAC). Someone have any experiencie with Addpac as gateway, or some workaround about this issue. Thanks! -- Omar E.P.T - Certified Networking Professionals make better Connections! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CALL TRANSFER
Hi Guys, I'm implementing my Asterisk step by step, so far the communications between softphones, hardphones with Gateways, voice mail, are working fine. Rightnow I would like to enable Call tranfer (like Traditional PBX) in Blind Transfer and AttendXFER, I'm reading features.conf in accordance to voip-info.org but the transfer doesn't work! Please if you can provide me some examples will be very appreciate. Rgds. -- Omar E.P.T - Certified Networking Professionals make better Connections! http://omarept.blogspot.com/ Usysnet Corp Open Source Solutions www.usysnet.com.pe ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CALL TRANSFER
Thanks!!! I forget Tt option! (too basis!!) On 12/1/06, Damon Estep [EMAIL PROTECTED] wrote: Your dial string must have either the t or T option set. -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *omar parihuana *Sent:* Friday, December 01, 2006 9:10 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] CALL TRANSFER Hi Guys, I'm implementing my Asterisk step by step, so far the communications between softphones, hardphones with Gateways, voice mail, are working fine. Rightnow I would like to enable Call tranfer (like Traditional PBX) in Blind Transfer and AttendXFER, I'm reading features.conf in accordance to voip-info.org but the transfer doesn't work! Please if you can provide me some examples will be very appreciate. Rgds. -- Omar E.P.T - Certified Networking Professionals make better Connections! http://omarept.blogspot.com/ Usysnet Corp Open Source Solutions www.usysnet.com.pe ___ --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 -- Omar E.P.T - Certified Networking Professionals make better Connections! http://omarept.blogspot.com/ Usysnet Corp Open Source Solutions www.usysnet.com.pe ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Call Statistics
Hi Moises, Coul you give more details about how to use Cacti for CDR analysis, there is some special pluggin, additional conf? Your help will be appreciated. Rgds. On 10/31/06, Moises Silva [EMAIL PROTECTED] wrote: of course you can always use http://cacti.net/download_cacti.php On 10/31/06, Andrew Joakimsen [EMAIL PROTECTED] wrote: Check out voip-info.org, there are quite a few GUIS some even generate nice graphs! On 10/31/06, omar parihuana [EMAIL PROTECTED] wrote: Hi Folks, I would like to recover all information about the calls, incoming calls, call time, call history, etc in a Web Format, are there some open source aplication for Asterisk that be easier for use. Pls anything suggestion will be very appreciate. Thanks Rgds. -- Omar E.P.T - Certified Networking Professionals make better Connections! http://omarept.blogspot.com/ Usysnet Corp Open Source Solutions www.usysnet.com.pe ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Omar E.P.T - Certified Networking Professionals make better Connections! http://omarept.blogspot.com/ Usysnet Corp Open Source Solutions www.usysnet.com.pe ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Call Statistics
Hi Folks, I would like to recover all information about the calls, incoming calls, call time, call history, etc in a Web Format, are there some open source aplication for Asterisk that be easier for use. Pls anything suggestion will be very appreciate. Thanks Rgds. -- Omar E.P.T - Certified Networking Professionals make better Connections! http://omarept.blogspot.com/ Usysnet Corp Open Source Solutions www.usysnet.com.pe ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bandwidth requirements
For bandwidth requeriments don't forget Layer 2 overhead. I.e Frame-relay overhead is lower than Ethernet overhead. Rgds. On 10/5/06, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Thu, Oct 05, 2006 at 09:44:56AM +0100, Brian Candler wrote: A 20ms packet duration means that 20ms of audio is stuffed into one IP packet. Since each packet carries 1/50th of a second of audio, that means you're generating 50 packets per second for each channel. With g729 your audio is 8000 bits per second. The overhead on each packet is 20 bytes (IP) + 8 bytes (UDP) + 12 bytes (RTP) = 40 bytes or 320 bits. So your bandwidth requirement per channel is: - 8000 bits per second for payload - 320x50 = 16000 bits per second for overhead making a total of 24000 bits per second. 20 simultaneous calls is therefore 480,000 bits per second. A reminder: much equipment, particularly low end/consumer equipment, chokes *much* faster on high PPS than it does on high BPS. While short packets are good for latency, they do impose stricter engineering evaluation requirements on the other links in your chains. Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Omar E.P.T - Certified Networking Professionals make better Connections! http://omarept.blogspot.com/ Usysnet Corp Open Source Solutions www.usysnet.com.pe ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Bandwidth requirements
Unfortunately cRTP is not supported by Asterisk, and normally it is implemented between routers that support it. On 10/5/06, Dan Austin [EMAIL PROTECTED] wrote: J. Oquendo wrote: Benny Amorsen wrote: rJ == raphael Jacquot [EMAIL PROTECTED] writes: rJ ATM cell tax is actually 10% as there's 5 header bytes for each 53 rJ bytes cell, For VoIP the cell tax is much larger. In the example, each RTP packet contains 20 useful bytes and 40 bytes IP overhead. 60 bytes doesn't fit in one cell, so you end up with 106 bytes at the ATM layer to transport 20 bytes of G.729. The ATM-caused overhead is thus 46 bytes per voice packet, thereby making the needed bandwidth 77% larger. CRTP solves this issue (40byte waste) cRTP is a great idea that is not widely implimented. Not very many endpoint support it, and Asterisk currently does not. In version 1.4, Asterisk will support configurable RTP packetization. So you could use 40ms for your G729 payload and reach a 50/50 split between payload and overhead (before ATM encapsulation for DSL). At 50ms for g729, you get a nice fit into two ATM cells, while a small bump to 60ms will force you into three ATM cells. Of course you need to be mindful of the latency between endpoints when you start increasing the payload, but a healthy network and tuning the payload can greatly reduce bandwidth requirements. AND most endpoints do support configurable RTP packetization. If Asterisk and more endpoints supported cRTP, then combining that feature with larger RTP payloads would result in very low overhead numbers. Dan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Omar E.P.T - Certified Networking Professionals make better Connections! http://omarept.blogspot.com/ Usysnet Corp Open Source Solutions www.usysnet.com.pe ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Newbie question about meetme
Hi Folks, I'm reading about meetme feature, but in accordance to voip-info it say: A zaptel interface must be installed for conferencing to work. Unfortunately I don't have a Zaptel Card, I'm using Asterisk like SIP server then I would like to implement meetme function. What can I do? Is possible use meetme feature without Zaptel card? (ztdummy will be the solution? ) Thanks in advanced.. -- Omar E.P.T - Certified Networking Professionals make better Connections! http://omarept.blogspot.com/ Usysnet Corp Open Source Solutions www.usysnet.com.pe ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
