[asterisk-users] Multiple IP addresses and using same IP for outbound calls as inbound
Hello, We have an Asterisk server with two public IP addresses, let's say 1.1.1.1 and 2.2.2.2. Normally calls come in to 1.1.1.1 and are bridged with a call dialled from Asterisk to an external destination. The external destination sees the SIP packet as coming from 1.1.1.1 and the media address in the SDP is 1.1.1.1, which is great. However if we receive a call in to 2.2.2.2 then the call dialled from Asterisk to an external destination still comes from 1.1.1.1, whereas we want it to come from 2.2.2.2. The source of any dialled call (the IP packet and the SDP media address) should be the same as the address the related inbound call was received to. For example: INVITE received to 1.1.1.1:5060 -> Asterisk dials destinat...@termination.com -> INVITE sent from 1.1.1.1:5060 to termination.com INVITE received to 2.2.2.2:5060 -> Asterisk dials destinat...@pstn.com -> INVITE sent from 2.2.2.2:5060 to pstn.com Does anyone know how this can be achieved? Thanks in advance for your help, -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 18.0.0 Now Available
On 21.10.20 at 12:49 Joshua C. Colp wrote: > On Wed, Oct 21, 2020 at 7:46 AM Michael Maier wrote: > >> Hello! >> >> On 20.10.20 at 14:00 Asterisk Development Team wrote: >>> The Asterisk Development Team would like to announce the release of >> Asterisk 18.0.0. >>> This release is available for immediate download at >>> https://downloads.asterisk.org/pub/telephony/asterisk >> >> I just tested the new codec negotiation feature and unfortunately wasn't >> able to get it working as expected. I tried several configurations - but >> none has been working - the result >> has always been the same. >> > > This is expected right now. Foundational aspects were put in, but there is > still work to be done for PJSIP which will land in a future release. Oh - thanks for the information - I missed this :-(. How do I know if this feature is finally enabled? Will it be in asterisk 18 - or will it come in some later major version? > The > complexity of it and the investigation of how things work, interactions, > etc took considerably longer than expected. If there's specific scenarios > that you'd like to ensure are met you can reach out on the asterisk-dev > mailing list and George Joseph will add them to the list if not already > present. Well, I think the scenario I have should be a very easy and basic scenario. I already discussed it in the past. Therefore I think it's not necessary to add it again. Thanks Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 18.0.0 Now Available
Hello! On 20.10.20 at 14:00 Asterisk Development Team wrote: > The Asterisk Development Team would like to announce the release of Asterisk > 18.0.0. > This release is available for immediate download at > https://downloads.asterisk.org/pub/telephony/asterisk I just tested the new codec negotiation feature and unfortunately wasn't able to get it working as expected. I tried several configurations - but none has been working - the result has always been the same. Use case: Alice calls Bob - sends INVITE G722 / alaw / ulaw Configured in Asterisk for this device: G722 / alaw / ulaw / gsm A: codec_prefs_incoming_offer = prefer: configured, operation: intersect, keep: all, transcode: prevent Bob: Configured in Asterisk for this device: alaw / ulaw B: codec_prefs_outgoing_offer = prefer: configured or pending, operation: intersect, keep: first or all, transcode: prevent Asterisk sends INVITE to Bobalaw / ulaw Asterisk receives OK from Bob alaw B: codec_prefs_incoming_answer = prefer: configured or pending, operation: intersect, keep: first or all, transcode: prevent Asterisk sends OK to Alice G722 / alaw / ulaw A: codec_prefs_outgoing_answer = prefer: pending, operation: intersect, keep: first or all, transcode: prevent => I would have expected alaw to be sent to A - but G722 / alaw / ulaw is sent and transcoding is active! What did I do wrong? Could you please add the correct configuration you expect to get the expected result alaw? Thanks Kind regards Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 18.0.0 Now Available
On Wed, Oct 21, 2020 at 8:49 AM Michael Maier wrote: > On 21.10.20 at 12:49 Joshua C. Colp wrote: > > On Wed, Oct 21, 2020 at 7:46 AM Michael Maier > wrote: > > > >> Hello! > >> > >> On 20.10.20 at 14:00 Asterisk Development Team wrote: > >>> The Asterisk Development Team would like to announce the release of > >> Asterisk 18.0.0. > >>> This release is available for immediate download at > >>> https://downloads.asterisk.org/pub/telephony/asterisk > >> > >> I just tested the new codec negotiation feature and unfortunately wasn't > >> able to get it working as expected. I tried several configurations - but > >> none has been working - the result > >> has always been the same. > >> > > > > This is expected right now. Foundational aspects were put in, but there > is > > still work to be done for PJSIP which will land in a future release. > > Oh - thanks for the information - I missed this :-(. How do I know if this > feature is finally enabled? Will it be in asterisk 18 - or will it come in > some later major version? > > It is planned to land in a future Asterisk 18 release, it will be stated in the changelog/release notes and we will likely do a blog post about it as well. -- Joshua C. Colp Asterisk Technical Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Which PCIe cards work with North American BRI & Asterisk?
Which PCIe cards work with North American BRI & Asterisk? Digium & Sangoma don't support it, according to everything I've read (their manuals and their tech support). I think I need National, maybe National 1 or National 2. I already got an NT-1. I'm getting a few ISDN phones to test the circuit. I ordered a Dialogic Diva Diva 4BRI - 8 PCI-E 4 Ports Quad 803-031-02B Gu... but I'm concerned that it does not work out of the box with Asterisk and requires proprietary closed firmware or software. Brad Allen -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 18.0.0 Now Available
On Wed, Oct 21, 2020 at 7:46 AM Michael Maier wrote: > Hello! > > On 20.10.20 at 14:00 Asterisk Development Team wrote: > > The Asterisk Development Team would like to announce the release of > Asterisk 18.0.0. > > This release is available for immediate download at > > https://downloads.asterisk.org/pub/telephony/asterisk > > I just tested the new codec negotiation feature and unfortunately wasn't > able to get it working as expected. I tried several configurations - but > none has been working - the result > has always been the same. > This is expected right now. Foundational aspects were put in, but there is still work to be done for PJSIP which will land in a future release. The complexity of it and the investigation of how things work, interactions, etc took considerably longer than expected. If there's specific scenarios that you'd like to ensure are met you can reach out on the asterisk-dev mailing list and George Joseph will add them to the list if not already present. Even just from AstriDevCon there were some things that individuals brought up. -- Joshua C. Colp Asterisk Technical Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
