search for: aor

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 ne...

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...

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...

...ge.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. >> >> I am having trouble mapping them to the typical SIP configuration >> settings on a phone. >> > > There's some info on the wiki here... > > https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip > > >> &g...

Hi all, (sending this again from the correct address) I'm running Asterisk 13.8.0 (I need to check if that happens with 13.9.1 too when I have the time to build it) with PJSIP realtime config. I've defined several aors in the table ps_aors, like this (real url replaced by myurl): *CLI> pjsip show aor pbx-node-1 Aor: <Aor..............................................> <MaxContact> Contact: <Aor/ContactUri............................> <Hash....> <Status> <RTT(ms)....

Hi, Yes, we're implementing the dialplan in realtime too. Here the contents of sorcery.conf: [res_pjsip] endpoint=realtime,ps_endpoints aor=realtime,ps_aors contact=realtime,ps_contacts [res_pjsip_endpoint_identifier_ip] identify=realtime,ps_endpoint_id_ips Cheers, Francisco. From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Annus Fictus Sent: 13 June 2016 14:11 To: As...

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 aor and contact?...

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....

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 incoming > calls/registrations/oth...

...allow_reload=yes [twilio](!) type=endpoint transport=transport-tls context=from-twilio disallow=all allow=ulaw dtmf_mode=inband media_encryption=sdes rtp_symmetric=yes rewrite_contact=yes force_rport=yes canreinvite=no tlsdontverifyserver=yes [auth-out](!) type=auth auth_type=userpass [twilio] aors=twilio-aors [twilio-aors] type=aor contact=sips:trunkname.pstn.twilio.com:5061 ;tried with sip: also [twilio] type=identify endpoint=twilio match=54.172.60.0 match=54.172.60.1 match=54.172.60.2 match=54.172.60.3 [endpoint-basic](!) type=endpoint transport=transport-tls context=from-phones disal...

...interesting part of it. We took our configs and database > tables from our working 13.12.2 deployments and tried to use them with > our > new 13.17.1 deployments and we are having issues where the tables are not > working. On the new server asterisk keeps saying it can't find the AORS > entries when we purge the sorcery memory cache it starts finding the aors > but then it says it cant find the auths. > > The wired thing is when it says it can't find the aors and auths entries > it does not show it is looking for the values in the aors and auth fields...

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 addr...

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. > > Is thi...

On Fri, Feb 14, 2020 at 3:04 PM Dan Cropp <dan at amtelco.com> wrote: > 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 = t...

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi list, I'm trying to use CHANNEL(aor) and CHANNEL(contact) on PJSIP channel, on asterisk-13.3.2, but they don't return anything. Is this a bug, or did I miss something? Here is my test dialplan: exten => *98,1,Answer same => n,NoOp(Channel=<${CHANNEL(name)}>,type= <${CHANNEL(channeltype)}>) same => n,NoOp(A...

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, it looks to me like most of the scenarios where this might be beneficial are actually not possible? One example I had envisaged was being able to have multiple distinct auth entrie...

...hua 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 was the auth object created in the first >> place, >> instead of extending the aor object with username/password/etc fields? >> > > Auth is useless on its own but is used by many things - in fact it's not >...

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... https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip > > Suppose I have a phone with two line buttons...

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 was the auth object created in the first > place, > instead of extending the aor object with username/password/etc fields? Auth is useless on its own but is used by many things - in fact it's not even used directly by an AOR. It's...

I just finished installing Asterisk 13 on our test server and I can now use PJSIP to register phones and make and receive calls. The only problem I am having is that when I register multiple phones to a single account only one of them rings. The AOR for the account has maxcontacts at 3. If I do a pjsip show endpoints I can see two "Contact" entries which I take to mean that both phones have registered: Endpoint: 101 Not in use 0 of inf InAuth: 101/101 Aor:...