Re: [OpenSIPS-Users] How much of the normal PBX Functions can be implemented using OpenSIPS, Mediaproxy and maybe SEMS
Hi Mike, It is possible to implement all PBX functions using OpenSIPS. It is not easy, it depends on your phones and gateways. They have to support several RFCs such as RFC3515, RFC3891, RFC3892. Check RFC5359 ( http://tools.ietf.org/html/rfc5359) for more details about call flows. I have successfully implemented most of them including Call Pickup, Call Forward, Attended and Unattended Transfers. Asterisk is an excellent gateway when you want to implement all these features. A good phone is also important. Don't try this with X-Lite. Best Regards, Flavio E. Goncalves CEO - V.Office Fone: +554830258590/+554884085000 OpenSIPS Bootcamp (New Jersey, NY Nov. 15-19) 2010/10/24 Mike O'Connor m...@oeg.com.au Hi Guys I've been using OpenSIPS now for about 9 month (after upgrading from OpenSER 1.2 used that for about 2 years) for my core SIP routing and billing. I'm now getting questions from customers about Virtual PBX functionality and I would like the opinion of the group about how well this could be done using OpenSIPS, Mediaproxy and maybe SEMS. My current core system has voicemail, call forwarding and T38 fax using sip forwards to asterisk, but as normal with Asterisk I do get occasional calls issues, mostly related to codec negotiation. I want to be able to have all the normal PBX functions like Auto attendant, Call forwarding on busy or absence, Call Park, Call pickup, Call transfer, Call waiting, Conference Call, Custom Greeting, Voice Mall, Public Addressing, DND, Direct Inward Dial, Busy Lamp. ETC So your comments requested. Thanks Mike ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] How much of the normal PBX Functions can be implemented using OpenSIPS, Mediaproxy and maybe SEMS
As a FreeSWITCH user for about 1yr for conferencing systems, I can assure it works very well as a virtualized PBX on KVM, Xen or OpenVZ virtualization platforms. Fernando Gregianin Testa Voice Technology Ltda ddr +55 11 21752166 cel +55 11 88225531 On 24-10-2010 16:25, Jeff Pyle wrote: Mike, We've been asking much the same questions. We have decided to take a serious look at Freeswitch for the Asterisk-style functions, while leaving the core routing functions to Opensips. - Jeff On Oct 24, 2010, at 1:16 AM, Mark Sayer wrote: Those virtual PBX functions, like your present voicemail, cannot be provided by OpenSIPS. They are Asterisk-style functions. Mark On Sun, Oct 24, 2010 at 2:04 PM, Mike O'Connor m...@oeg.com.au mailto:m...@oeg.com.au wrote: Hi Guys I've been using OpenSIPS now for about 9 month (after upgrading from OpenSER 1.2 used that for about 2 years) for my core SIP routing and billing. I'm now getting questions from customers about Virtual PBX functionality and I would like the opinion of the group about how well this could be done using OpenSIPS, Mediaproxy and maybe SEMS. My current core system has voicemail, call forwarding and T38 fax using sip forwards to asterisk, but as normal with Asterisk I do get occasional calls issues, mostly related to codec negotiation. I want to be able to have all the normal PBX functions like Auto attendant, Call forwarding on busy or absence, Call Park, Call pickup, Call transfer, Call waiting, Conference Call, Custom Greeting, Voice Mall, Public Addressing, DND, Direct Inward Dial, Busy Lamp. ETC So your comments requested. Thanks Mike ___ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] How much of the normal PBX Functions can be implemented using OpenSIPS, Mediaproxy and maybe SEMS
So are you able to integrate FreeSwitch with OpenSIPS like Asterisk is integrated(Usernames and Passwords link up)? On Mon, Oct 25, 2010 at 8:44 AM, Fernando Gregianin Testa te...@voicetechnology.com.br wrote: As a FreeSWITCH user for about 1yr for conferencing systems, I can assure it works very well as a virtualized PBX on KVM, Xen or OpenVZ virtualization platforms. Fernando Gregianin Testa Voice Technology Ltda ddr +55 11 21752166 cel +55 11 88225531 On 24-10-2010 16:25, Jeff Pyle wrote: Mike, We've been asking much the same questions. We have decided to take a serious look at Freeswitch for the Asterisk-style functions, while leaving the core routing functions to Opensips. - Jeff On Oct 24, 2010, at 1:16 AM, Mark Sayer wrote: Those virtual PBX functions, like your present voicemail, cannot be provided by OpenSIPS. They are Asterisk-style functions. Mark On Sun, Oct 24, 2010 at 2:04 PM, Mike O'Connor m...@oeg.com.au mailto:m...@oeg.com.au wrote: Hi Guys I've been using OpenSIPS now for about 9 month (after upgrading from OpenSER 1.2 used that for about 2 years) for my core SIP routing and billing. I'm now getting questions from customers about Virtual PBX functionality and I would like the opinion of the group about how well this could be done using OpenSIPS, Mediaproxy and maybe SEMS. My current core system has voicemail, call forwarding and T38 fax using sip forwards to asterisk, but as normal with Asterisk I do get occasional calls issues, mostly related to codec negotiation. I want to be able to have all the normal PBX functions like Auto attendant, Call forwarding on busy or absence, Call Park, Call pickup, Call transfer, Call waiting, Conference Call, Custom Greeting, Voice Mall, Public Addressing, DND, Direct Inward Dial, Busy Lamp. ETC So your comments requested. Thanks Mike ___ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- -- *--*--*--*--*--* Duane *--*--*--*--*--* -- ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] How much of the normal PBX Functions can be implemented using OpenSIPS, Mediaproxy and maybe SEMS
Mike, We've been asking much the same questions. We have decided to take a serious look at Freeswitch for the Asterisk-style functions, while leaving the core routing functions to Opensips. - Jeff On Oct 24, 2010, at 1:16 AM, Mark Sayer wrote: Those virtual PBX functions, like your present voicemail, cannot be provided by OpenSIPS. They are Asterisk-style functions. Mark On Sun, Oct 24, 2010 at 2:04 PM, Mike O'Connor m...@oeg.com.aumailto:m...@oeg.com.au wrote: Hi Guys I've been using OpenSIPS now for about 9 month (after upgrading from OpenSER 1.2 used that for about 2 years) for my core SIP routing and billing. I'm now getting questions from customers about Virtual PBX functionality and I would like the opinion of the group about how well this could be done using OpenSIPS, Mediaproxy and maybe SEMS. My current core system has voicemail, call forwarding and T38 fax using sip forwards to asterisk, but as normal with Asterisk I do get occasional calls issues, mostly related to codec negotiation. I want to be able to have all the normal PBX functions like Auto attendant, Call forwarding on busy or absence, Call Park, Call pickup, Call transfer, Call waiting, Conference Call, Custom Greeting, Voice Mall, Public Addressing, DND, Direct Inward Dial, Busy Lamp. ETC So your comments requested. Thanks Mike ___ Users mailing list Users@lists.opensips.orgmailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.orgmailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] How much of the normal PBX Functions can be implemented using OpenSIPS, Mediaproxy and maybe SEMS
Hi Guys I've been using OpenSIPS now for about 9 month (after upgrading from OpenSER 1.2 used that for about 2 years) for my core SIP routing and billing. I'm now getting questions from customers about Virtual PBX functionality and I would like the opinion of the group about how well this could be done using OpenSIPS, Mediaproxy and maybe SEMS. My current core system has voicemail, call forwarding and T38 fax using sip forwards to asterisk, but as normal with Asterisk I do get occasional calls issues, mostly related to codec negotiation. I want to be able to have all the normal PBX functions like Auto attendant, Call forwarding on busy or absence, Call Park, Call pickup, Call transfer, Call waiting, Conference Call, Custom Greeting, Voice Mall, Public Addressing, DND, Direct Inward Dial, Busy Lamp. ETC So your comments requested. Thanks Mike ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] How much of the normal PBX Functions can be implemented using OpenSIPS, Mediaproxy and maybe SEMS
Those virtual PBX functions, like your present voicemail, cannot be provided by OpenSIPS. They are Asterisk-style functions. Mark On Sun, Oct 24, 2010 at 2:04 PM, Mike O'Connor m...@oeg.com.au wrote: Hi Guys I've been using OpenSIPS now for about 9 month (after upgrading from OpenSER 1.2 used that for about 2 years) for my core SIP routing and billing. I'm now getting questions from customers about Virtual PBX functionality and I would like the opinion of the group about how well this could be done using OpenSIPS, Mediaproxy and maybe SEMS. My current core system has voicemail, call forwarding and T38 fax using sip forwards to asterisk, but as normal with Asterisk I do get occasional calls issues, mostly related to codec negotiation. I want to be able to have all the normal PBX functions like Auto attendant, Call forwarding on busy or absence, Call Park, Call pickup, Call transfer, Call waiting, Conference Call, Custom Greeting, Voice Mall, Public Addressing, DND, Direct Inward Dial, Busy Lamp. ETC So your comments requested. Thanks Mike ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
