Sure, but the point is if he takes the same credentials from the ATA and
puts them into an endpoint in pjsip.conf in Asterisk then it will almost
definitely work.
On 9/23/2020 4:06 PM, Daniel White wrote:
Yes and no.
A trunk is assumed to go to a PBX versus an endpoint so the provider
may enable, disable, or change some features.
The assumption on a trunk is that it will use more minutes than an ATA.
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Daniel White
Co-Founder
phone: +1 (702) 470-2770
direct:+1 (702) 470-2766
Adam Moffett <mailto:dmmoff...@gmail.com>
September 23, 2020 at 13:40
Yeah but there's no functional difference. The SIP registration from
an ATA is just a SIP "trunk" that can only accept one or two calls.
There should be no reason you can't register that SIP user from Asterisk.
On 9/23/2020 2:52 PM, ch...@wbmfg.com wrote:
ch...@wbmfg.com <mailto:ch...@wbmfg.com>
September 23, 2020 at 12:52
The provider charges more for a “trunk”.
And Lewis, I already have this service set up through them for
another system
We have some unique LNP porting issues and really want the voip
provider to have a direct connection to us.
*From:* Brian Webster
*Sent:* Wednesday, September 23, 2020 12:35 PM
*To:* 'AnimalFarm Microwave Users Group'
*Subject:* Re: [AFMUG] OT sip question
It’s been a while since I have done an asterisk config but you still
configure a SIP account much the same as you would for the ATA and it
becomes a trunk as I recall and then you do whatever you need to with
it as a single line trunk. Way back when I was learning it and
playing I had multiple providers for single lines and even an IAX
line too. Then I could use them as part of the dial plan and other
features as to where it would ring and dial out.
Thank you,
Brian Webster
www.wirelessmapping.com
*From:*AF [mailto:af-boun...@af.afmug.com] *On Behalf Of *ch...@wbmfg.com
*Sent:* Wednesday, September 23, 2020 2:01 PM
*To:* af@af.afmug.com
*Subject:* [AFMUG] OT sip quiestion
If I get a voip service from some random provider and I have a sip
device on my end, I am golden. Say an ATA connected to a pots phone.
Now, if I want that number to appear inside an asterisk PBX but I
don’t want to buy a trunk, is there an easy way to do that?
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Brian Webster <mailto:i...@wirelessmapping.com>
September 23, 2020 at 12:35
It’s been a while since I have done an asterisk config but you still
configure a SIP account much the same as you would for the ATA and it
becomes a trunk as I recall and then you do whatever you need to with
it as a single line trunk. Way back when I was learning it and
playing I had multiple providers for single lines and even an IAX
line too. Then I could use them as part of the dial plan and other
features as to where it would ring and dial out.
Thank you,
Brian Webster
www.wirelessmapping.com
*From:*AF [mailto:af-boun...@af.afmug.com] *On Behalf Of *ch...@wbmfg.com
*Sent:* Wednesday, September 23, 2020 2:01 PM
*To:* af@af.afmug.com
*Subject:* [AFMUG] OT sip quiestion
If I get a voip service from some random provider and I have a sip
device on my end, I am golden. Say an ATA connected to a pots phone.
Now, if I want that number to appear inside an asterisk PBX but I
don’t want to buy a trunk, is there an easy way to do that?
--
AF mailing list
AF@af.afmug.com
http://af.afmug.com/mailman/listinfo/af_af.afmug.com
