Re: [asterisk-users] Sip and the media path
Hi David, Direct media should work either way. if your phones are behind NAT you will also require the NAT option enabled in asterisk, How ever the tricky part in all this is that you wont be able to acurately keep track of calls on these phones. If or any unforeseen reason the phone goes offline or you dont recieve BYE signal, asterisk wont be able to know that the call has ended. So if call information is critical for you then byepassmedia is not recomended for you. Regards, Qasim On Thu, Apr 25, 2013 at 8:48 PM, David Wessell wrote: > Kevin, > > Thanks for the info. Clarification. The asterisk server is NOT on the > same LAN as the phones. The asterisk server is in a datacenter only > accessible via WAN. > > However, all of the phones are in side of the same LAN. Will directmedia > still function that way? > > Thanks > David > > From: Kevin Larsen > Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion < > asterisk-users@lists.digium.com> > Date: Thursday, April 25, 2013 9:16 AM > > To: Asterisk Users Mailing List - Non-Commercial Discussion < > asterisk-users@lists.digium.com> > Subject: Re: [asterisk-users] Sip and the media path > > You will want to look at the directmedia option. You will want all the > phones on the same lan as the Asterisk server to be directmedia=yes and the > ones on the wan to be directmedia=no. Then, internal calls will send the > media between themselves without involving Asterisk, but ones outside on > the wan will be forced to talk directly to the Asterisk server for > everything. You might also want to look at the nonat option of directmedia. > > Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208 > > > > From:David Wessell > To:Asterisk Users Mailing List - Non-Commercial Discussion < > asterisk-users@lists.digium.com>, > Date:04/25/2013 07:33 AM > Subject:[asterisk-users] Sip and the media path > Sent by:asterisk-users-boun...@lists.digium.com > -- > > > > We're running asterisk 1.8 in the DC on a public IP address. > > Connecting to it are about 200 phones behind a LAN in a remote location. > > Is there a way to reliably keep asterisk out of the media stream on > internal calls inside that LAN? All phones are Polycom Soundpoint phones. > > Asterisk would say in the media stream for any calls that traverse from > LAN to WAN. However it would step out for LAN to LAN calls. > > Thanks > David > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > 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 -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sip and the media path
David, you obviously have to test for your situation, but the short answer is that it should. The connection will start with running through Asterisk, but very quickly the phones will see that they can talk directly and take the Asterisk server out of the media path. There are a couple of gotchas that can happen based on your dial options, so check out this page: http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite canreinvite was renamed to directmedia in Asterisk 1.6.2, but the page is still pretty good with regards to the options that are available. Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208 From: David Wessell To: Asterisk Users Mailing List - Non-Commercial Discussion , Date: 04/25/2013 10:49 AM Subject:Re: [asterisk-users] Sip and the media path Sent by:asterisk-users-boun...@lists.digium.com Kevin, Thanks for the info. Clarification. The asterisk server is NOT on the same LAN as the phones. The asterisk server is in a datacenter only accessible via WAN. However, all of the phones are in side of the same LAN. Will directmedia still function that way? Thanks David From: Kevin Larsen Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion < asterisk-users@lists.digium.com> Date: Thursday, April 25, 2013 9:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion < asterisk-users@lists.digium.com> Subject: Re: [asterisk-users] Sip and the media path You will want to look at the directmedia option. You will want all the phones on the same lan as the Asterisk server to be directmedia=yes and the ones on the wan to be directmedia=no. Then, internal calls will send the media between themselves without involving Asterisk, but ones outside on the wan will be forced to talk directly to the Asterisk server for everything. You might also want to look at the nonat option of directmedia. Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208 From:David Wessell To:Asterisk Users Mailing List - Non-Commercial Discussion < asterisk-users@lists.digium.com>, Date:04/25/2013 07:33 AM Subject: [asterisk-users] Sip and the media path Sent by:asterisk-users-boun...@lists.digium.com We're running asterisk 1.8 in the DC on a public IP address. Connecting to it are about 200 phones behind a LAN in a remote location. Is there a way to reliably keep asterisk out of the media stream on internal calls inside that LAN? All phones are Polycom Soundpoint phones. Asterisk would say in the media stream for any calls that traverse from LAN to WAN. However it would step out for LAN to LAN calls. Thanks David -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sip and the media path
Kevin, Thanks for the info. Clarification. The asterisk server is NOT on the same LAN as the phones. The asterisk server is in a datacenter only accessible via WAN. However, all of the phones are in side of the same LAN. Will directmedia still function that way? Thanks David From: Kevin Larsen mailto:kevin.lar...@pioneerballoon.com>> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com>> Date: Thursday, April 25, 2013 9:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com>> Subject: Re: [asterisk-users] Sip and the media path You will want to look at the directmedia option. You will want all the phones on the same lan as the Asterisk server to be directmedia=yes and the ones on the wan to be directmedia=no. Then, internal calls will send the media between themselves without involving Asterisk, but ones outside on the wan will be forced to talk directly to the Asterisk server for everything. You might also want to look at the nonat option of directmedia. Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208 From:David Wessell mailto:da...@ringfree.biz>> To:Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com>>, Date:04/25/2013 07:33 AM Subject: [asterisk-users] Sip and the media path Sent by: asterisk-users-boun...@lists.digium.com<mailto:asterisk-users-boun...@lists.digium.com> We're running asterisk 1.8 in the DC on a public IP address. Connecting to it are about 200 phones behind a LAN in a remote location. Is there a way to reliably keep asterisk out of the media stream on internal calls inside that LAN? All phones are Polycom Soundpoint phones. Asterisk would say in the media stream for any calls that traverse from LAN to WAN. However it would step out for LAN to LAN calls. Thanks David -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com<http://www.api-digital.com/> -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sip and the media path
You will want to look at the directmedia option. You will want all the phones on the same lan as the Asterisk server to be directmedia=yes and the ones on the wan to be directmedia=no. Then, internal calls will send the media between themselves without involving Asterisk, but ones outside on the wan will be forced to talk directly to the Asterisk server for everything. You might also want to look at the nonat option of directmedia. Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208 From: David Wessell To: Asterisk Users Mailing List - Non-Commercial Discussion , Date: 04/25/2013 07:33 AM Subject:[asterisk-users] Sip and the media path Sent by:asterisk-users-boun...@lists.digium.com We're running asterisk 1.8 in the DC on a public IP address. Connecting to it are about 200 phones behind a LAN in a remote location. Is there a way to reliably keep asterisk out of the media stream on internal calls inside that LAN? All phones are Polycom Soundpoint phones. Asterisk would say in the media stream for any calls that traverse from LAN to WAN. However it would step out for LAN to LAN calls. Thanks David -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sip and the media path
We're running asterisk 1.8 in the DC on a public IP address. Connecting to it are about 200 phones behind a LAN in a remote location. Is there a way to reliably keep asterisk out of the media stream on internal calls inside that LAN? All phones are Polycom Soundpoint phones. Asterisk would say in the media stream for any calls that traverse from LAN to WAN. However it would step out for LAN to LAN calls. Thanks David -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
