Hello, Does anyone know a way with chan_sip to tell Asterisk to use a specific IP address for its end of the communication for a specific device? Something like:
[device] type = friend host = 11.22.11.22 ouraddress = 33.44.33.44 This is for use on a server with multiple IP addresses. There is the "extenip" setting, but it's really designed for NAT, and can only appear in the [general] section. Any suggestions would be greatly appreciated. On Sat, 24 Oct 2020 at 09:43, David Cunningham <dcunning...@voisonics.com> wrote: > OK, thank you George. > > > On Sat, 24 Oct 2020 at 03:16, George Joseph <gjos...@digium.com> wrote: > >> >> >> On Thu, Oct 22, 2020 at 4:13 PM David Cunningham < >> dcunning...@voisonics.com> wrote: >> >>> Hi George, >>> >>> Thank you for the response. I'm a little unclear on what you mean by a >>> transport. We're using chan_sip, not pjsip. >>> >>> Do you mean a device in sip.conf, using bindaddr to set the address to >>> bind for that device? We've only used bindaddr in the [general] section >>> before, but if it will work in a device that could be the answer. >>> >> >> Sorry. I just assume chan_pjsip these days. Not sure how you'd do it >> for chan_sip. >> >> >> >>> >>> >>> On Fri, 23 Oct 2020 at 00:13, George Joseph <gjos...@digium.com> wrote: >>> >>>> >>>> >>>> On Wed, Oct 21, 2020 at 9:16 PM David Cunningham < >>>> dcunning...@voisonics.com> wrote: >>>> >>>>> 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? >>>>> >>>> >>>> If termination.com is only on 1.1.1.1 and pstn.com is only on 2.2.2.2, >>>> create 2 transports, one specifically bound to 1.1.1.1, transport-1.1.1.1 >>>> for instance, and another to 2.2.2.2: transport-2.2.2.2. The names >>>> aren't important as long as you can tell the difference. Then explicitly >>>> configure endpoint termination.com's "transport" parameter to >>>> "transport-1.1.1.1" and pstn.com's "transport" parameter to >>>> "transport-2.2.2.2". In your dialplan, you can see which endpoint the >>>> call came in on, and route it out the same endpoint. >>>> >>>> If both providers are available from both interfaces, you can create 2 >>>> endpoint for each provider: termination.com-1.1.1.1, pstn.com-1.1.1.1, >>>> termination.com-2.2.2.2 and pstn.com-2.2.2.2; Then configure each with the >>>> same transports as above. >>>> >>>> >>>> >>>> >>>> >>>>> >>>>> 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 >>>> >>>> >>>> >>>> -- >>>> George Joseph >>>> Asterisk Software Developer >>>> direct/fax +1 256 428 6012 >>>> Check us out at www.sangoma.com and www.asterisk.org >>>> [image: image.png] >>>> -- >>>> _____________________________________________________________________ >>>> -- 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 >>> >>> >>> >>> -- >>> 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 >> >> >> >> -- >> George Joseph >> Asterisk Software Developer >> direct/fax +1 256 428 6012 >> Check us out at www.sangoma.com and www.asterisk.org >> [image: image.png] >> -- >> _____________________________________________________________________ >> -- 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 > > > > -- > David Cunningham, Voisonics Limited > http://voisonics.com/ > USA: +1 213 221 1092 > New Zealand: +64 (0)28 2558 3782 > -- 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