[asterisk-users] T38 FAX over a Broadsoft
I have a problem with sending a T38 FAX over a Broadsoft server. I am sending to a PSTN FAX so the Broadsoft server is terminating SIP point and it should send me REINVITE for T38 but it does not. It is just accepting the FAX transmission over G711. My question is there some specific advertisement(media attribute or media description) which I should add in my initial INVITE request to the Broadsoft server to make it understand that I support T38? I saw also something called Broadsoft FAX Messaging but did not find any description. -- Best Regards eng. Anatoli Marinov ___ -- 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
Re: [asterisk-users] ringback when the channel is answered
The problem was because my res_indications.so not been loaded. I added it in my modules.conf and now everithing works fine. Thanks a lot 2008/9/5 eng. Anatoli Marinov [EMAIL PROTECTED]: I do not know but I could not set it up. :) bad luck maybe. 2008/9/4 Steve Totaro [EMAIL PROTECTED]: Why is it an option if it should never be used?. Thanks, Steve Totaro On Thu, Sep 4, 2008 at 1:56 PM, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: This has nothing to do with the progressinband setting and you should never use the r option. eng. Anatoli Marinov wrote: Is there any special option which I should enable to activate these tones? My progressinband is yes and I cal Dial app with r option it it right? 2008/9/4 Eric ManxPower Wieling [EMAIL PROTECTED]: It will do so by default if you have a valid /etc/asterisk/indications.conf (only used for inband tones like after an Answer()) eng. Anatoli Marinov wrote: Hi guys, I am trying to configure an asterisk server for our office. Asterisk 1.4.17 SIP only The problem appears when the call comes from external point to our internal network. So when the server receives the call the channel is answered and the remote user hears prompt which invite him to enter internal private number. After that the server starts to wait the extension. After timeout the server executes Dial application and sends invite to sip client from our internal network. The problem is in this point. I want to play ringback tone to remote user when he waits internal user to pick up his phone but I could not instruct Asterisk to generate fake ringback in rtp stream . -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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 -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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 ___ -- 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 -- Best Regards eng. Anatoli Marinov -- Best Regards eng. Anatoli Marinov ___ -- 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] ringback when the channel is answered
Hi guys, I am trying to configure an asterisk server for our office. Asterisk 1.4.17 SIP only The problem appears when the call comes from external point to our internal network. So when the server receives the call the channel is answered and the remote user hears prompt which invite him to enter internal private number. After that the server starts to wait the extension. After timeout the server executes Dial application and sends invite to sip client from our internal network. The problem is in this point. I want to play ringback tone to remote user when he waits internal user to pick up his phone but I could not instruct Asterisk to generate fake ringback in rtp stream . Is there a solution for this? Thanks in advance. -- Best Regards eng. Anatoli Marinov ___ -- 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
Re: [asterisk-users] ringback when the channel is answered
Is there any special option which I should enable to activate these tones? My progressinband is yes and I cal Dial app with r option it it right? 2008/9/4 Eric ManxPower Wieling [EMAIL PROTECTED]: It will do so by default if you have a valid /etc/asterisk/indications.conf (only used for inband tones like after an Answer()) eng. Anatoli Marinov wrote: Hi guys, I am trying to configure an asterisk server for our office. Asterisk 1.4.17 SIP only The problem appears when the call comes from external point to our internal network. So when the server receives the call the channel is answered and the remote user hears prompt which invite him to enter internal private number. After that the server starts to wait the extension. After timeout the server executes Dial application and sends invite to sip client from our internal network. The problem is in this point. I want to play ringback tone to remote user when he waits internal user to pick up his phone but I could not instruct Asterisk to generate fake ringback in rtp stream . -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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 -- Best Regards eng. Anatoli Marinov ___ -- 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
Re: [asterisk-users] ASTERISK supported Video phone
I use it with n800 device - nokia internet tablet and standard nokia soft phone I have video call. The codec that I use is h263 and it works great. 2008/9/4 Steve Repo [EMAIL PROTECTED]: On Thu, Sep 4, 2008 at 1:48 PM, Gordon Henderson [EMAIL PROTECTED] wrote: On Thu, 4 Sep 2008, Tharanga wrote: Hi folks, Can some one recommend a good video phone for asterisk (SIP.IAX2). I need better quality, duarability. and should support various video codec's .(Codec auto negotiation support id prefferble) I suspect that the choices are so limited right now that good or bad is going to be very subjective. Grandstream GXP3000's appear to work from what I've heard others say, as does X-Lite (eyebeam?) softphone, Ekega (softphone) and ATL video phones... Then theres Polycoms with an extra zero aded to the price... Some people have reported good results with the BT Videophone 1000 units too.. (avalable for £60 a pair, but they need to have the early s/w release on them) I'm just about to order up a paid of Grandstreams for a project... (Hm. Can I trunk video over IAX?) Dlink has launched one in india. http://www.techgadgets.in/misc-gadgets/2008/13/d-link-gvc-3000-ip-videophone-and-glv-540-ip-phones-announced-in-india/ ___ -- 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 -- Best Regards eng. Anatoli Marinov ___ -- 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
Re: [asterisk-users] ringback when the channel is answered
So as I understand the only thing that I can do is to set up indications.conf. Ok I will try it tomorrow and will write again with my results. Thanks a lot. 2008/9/4 Eric ManxPower Wieling [EMAIL PROTECTED]: This has nothing to do with the progressinband setting and you should never use the r option. eng. Anatoli Marinov wrote: Is there any special option which I should enable to activate these tones? My progressinband is yes and I cal Dial app with r option it it right? 2008/9/4 Eric ManxPower Wieling [EMAIL PROTECTED]: It will do so by default if you have a valid /etc/asterisk/indications.conf (only used for inband tones like after an Answer()) eng. Anatoli Marinov wrote: Hi guys, I am trying to configure an asterisk server for our office. Asterisk 1.4.17 SIP only The problem appears when the call comes from external point to our internal network. So when the server receives the call the channel is answered and the remote user hears prompt which invite him to enter internal private number. After that the server starts to wait the extension. After timeout the server executes Dial application and sends invite to sip client from our internal network. The problem is in this point. I want to play ringback tone to remote user when he waits internal user to pick up his phone but I could not instruct Asterisk to generate fake ringback in rtp stream . -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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 -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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 -- Best Regards eng. Anatoli Marinov ___ -- 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
Re: [asterisk-users] ringback when the channel is answered
I do not know but I could not set it up. :) bad luck maybe. 2008/9/4 Steve Totaro [EMAIL PROTECTED]: Why is it an option if it should never be used?. Thanks, Steve Totaro On Thu, Sep 4, 2008 at 1:56 PM, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: This has nothing to do with the progressinband setting and you should never use the r option. eng. Anatoli Marinov wrote: Is there any special option which I should enable to activate these tones? My progressinband is yes and I cal Dial app with r option it it right? 2008/9/4 Eric ManxPower Wieling [EMAIL PROTECTED]: It will do so by default if you have a valid /etc/asterisk/indications.conf (only used for inband tones like after an Answer()) eng. Anatoli Marinov wrote: Hi guys, I am trying to configure an asterisk server for our office. Asterisk 1.4.17 SIP only The problem appears when the call comes from external point to our internal network. So when the server receives the call the channel is answered and the remote user hears prompt which invite him to enter internal private number. After that the server starts to wait the extension. After timeout the server executes Dial application and sends invite to sip client from our internal network. The problem is in this point. I want to play ringback tone to remote user when he waits internal user to pick up his phone but I could not instruct Asterisk to generate fake ringback in rtp stream . -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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 -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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 ___ -- 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 -- Best Regards eng. Anatoli Marinov ___ -- 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
Re: [asterisk-users] ASTERISK supported Video phone
SIP supporst video :) I am sure because I use it. 2008/9/5 bilal ghayyad [EMAIL PROTECTED]: And he can use Vidoe with SIP? As I know that SIP still does not support video. Regards Bilal ___ -- 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 -- Best Regards eng. Anatoli Marinov ___ -- 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
