similar to: PJSIP AoR vs Endpoint

Le sam. 18 juil. 2020 à 08:02, Andrew Yager <andrew at rwts.com.au> a écrit : > Hi, > > I realise this is an old question, but I’m struggling to get my head > around it. > > The ERD suggests that endpoints can link to multiple AoRs > > In what situation would you actually use this? Given that mapping of > inbound calls is primary done to the endpoint, > In my

I just realized that you are asking about dynamic AORs, not static Contacts in an AOR. That may be the difference. I have never actually tried giving a dynamic AOR a different name. And you wouldn't want more than one dynamic AOR, you'd just use an AOR that allowed more than 1 contact. On Wed, Apr 1, 2015 at 2:59 PM Trey Hilyard <kctrey at gmail.com> wrote: > I don't know

I am running asterisk 13.1.0 In pjsip.conf, the endpoint section has an aors and an auth field. I can name the auth field anything I want. The key is to set the auth=field accordingly. However, when I try this with the aors field, it never works. It seems I have to name the aors=field to match the name of the endpoint section. Is this correct? Would there ever be a need for multiple aors to

Hello, I am slightly confused by the difference between chan_sip and pjsip. Especially the new (to me) objects aor and contact. I am having trouble mapping them to the typical SIP configuration settings on a phone. Suppose I have a phone with two line buttons, for two extension numbers. Now, I think that means two 'endpoints' in pjsip right? But what exactly is the difference between

Thanks for responding, On Sun, Jan 4, 2015 at 5:45 PM, George Joseph <george.joseph at fairview5.com> wrote: > On Sun, Jan 4, 2015 at 3:29 PM, Antonio G?mez Soto < > antonio.gomez.soto at gmail.com> wrote: > >> Hello, >> >> I am slightly confused by the difference between chan_sip and pjsip. >> Especially the new (to me) objects aor and contact.

Dear List It's probably been more than a year now I switched from chan_sip to pjsip. pjsip works much cleaner than chan_sip. But! I have come across a Problem I was not able to solve with Asterisk Dialplan Logic. With pjsip an endpoint can have multiple AOR, so you need to expand them with ${PJSIP_DIAL_CONTACTS()} to be able to Dial() all of them simultaneously. But there are also

Hi, I'm not sure if I just missed something somewhere along the way, but I noticed while I was going through the CVS logs that there is an option in the wcfxs module to set an "opermode" - which apparently might help with echo issues around the globe (like the ones I'm seeing - some times). So I'd love to use it and give it a go. Apparently, there are some options that

As a follow up to my previous post, I have now identified what is causing the bug with the grandstream phones. When the line faxdetection=incoming is in the zapata.conf file, the grandstream phones will not ring, nor connect a call to the zaptel interface. Can anyone else confirm this bug? I'm going to play with the different options (incoming/outgoing/both) to see if it makes a

Joshua, On Sun, Jan 4, 2015 at 6:39 PM, Joshua Colp <jcolp at digium.com> wrote: [..snip..] > Also I notice, an AOR does seem do be directly correlated with an auth >> record, so why are >> they separate in the configuration, why not unify the aor and the auth >> objects? >> > > They aren't at all. Auth = Authentication. Used to authenticate

Hello everyone, I have started testing the PJSIP stack. I saw that it is possible to setup statically multiple AOR contacts, setup qualify_timeout and attach it to an endpoint, and then dial using this endpoint. When I setup the configuration I used the cli in order to see the status of the contacts, and it worked fine - whenever a contact is unreachable, the status is updated to Unavailable.

Hi, Is there a way to drop the ANSI colour strings from log files? In particular, I've got JSON logging throwing logs over to ES, but they have the ANSI colour escape sequences. Ideally I don't want to lose coloured logs from the console though, and I can't "see" a way to do this. Ast 16 at the moment… Andrew -------------- next part -------------- An HTML attachment was

Hi, Last weekend I was planning to buy a physical PBX system, but instead I have been blown away by the fact that VoIP really works, that Asterisk is so easy to set up and use... and free! We're in Australia, so as I understand it, we aren't allowed to use the Zaptel cards. We need to set up our system to route incoming 56K data calls to the PPP daemon on our Linux box or to our

On Sun, Jan 4, 2015 at 8:48 PM, Joshua Colp <jcolp at digium.com> wrote: > Antonio G?mez Soto wrote: > > <snip> > > >> I did not mean they are the same, I meant that there seems to be a >> one-to-one relationship. >> >> So I am wondering, since the auth does seem useless without an aor, but >> an aor >> can exist without an auth, why

I have the following configuration... [aor3] type = aor max_contacts = 1 remove_existing = yes [auth3] type = auth username = 1004 password = SuperSecretProbation [1004] type = endpoint context = IS transport = transport1 auth = auth3 aors = aor3 accountcode = 3 dtmf_mode = rfc4733 device_state_busy_at = 2 force_rport = no moh_passthrough = yes disallow = all allow = ulaw acl = acl1 When a

I don't know why you have issues using different names. I have multiple AORs assigned to a single endpoint and it works fine. I have to admit that my AORs do contain the endpoint name, though. For example, for endpoint "myswitch" I have two AORs, "myswitch_1" and "myswitch_2", and I assign them to the endpoint with aors=myswitch_1,myswitch_2. When you say that

Dan Cropp wrote: > I am running asterisk 13.1.0 > > In pjsip.conf, the endpoint section has an aors and an auth field. > > I can name the auth field anything I want. The key is to set the > auth=field accordingly. > > However, when I try this with the aors field, it never works. It seems I > have to name the aors=field to match the name of the endpoint section. > >

Antonio G?mez Soto wrote: > > So basically, the 'contact' in the AOR is just an ip address (or > 'dynamic', in which case it accepts > incoming registrations). A contact is a SIP term, it's a way of getting to something. (IP address+port) > So what happens if one endpoint has multiple AOR's which are registered > from different ip addresses. > And

Hi, I'm confused as anything by this bug. I'm hoping that it is just something screwy in my config. I have 1 Cisco 7960 and several Grandstream BT101 & 102's, and a Digium TDM31B. I'm running the latest CVS (CVS-HEAD-05/27/04-17:22:40) of both Asterisk and the Zaptel driver on Fedora Cora 1. When I make an outgoing call on the Cisco phone, everything works fine. I'm

On Sun, Jan 4, 2015 at 3:29 PM, Antonio G?mez Soto < antonio.gomez.soto at gmail.com> wrote: > Hello, > > I am slightly confused by the difference between chan_sip and pjsip. > Especially the new (to me) objects aor and contact. > > I am having trouble mapping them to the typical SIP configuration settings > on a phone. > There's some info on the wiki here...

Hi, I know we've talked about this phone to death. I have pretty good voice quality, with and without WEP enabled, using the G729a codec and DLink & Netgear access points. I am facing one obstacle that is driving me insane. Does anyone have the call hold feature working? If you do... how did you make it work? The instructions say to press the left button to place the call on hold,