similar to: Queue PJSIP, not all contacts rings

Nick Awesome wrote: > Hay guys, have question. > > When I do regular dial I use > $this->AGI->get_fullvariable('${PJSIP_DIAL_CONTACTS('.$callObj.')}',false,true); > > to get all contacts of current endpoint and so I dial to all phones > at once, > > but if I exec QUEUE, I have just one phone rings, seems like it take > first one as Dial app by

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.

>> So which option is preferred? >> >> A) Have a softphone aor/auth_user/password for a particular human, and >> expect them to configure it on multiple devices. Do not worry that 1) >> multiple are registered at once (because that's normal in SIP) and 2) >> asterisk has no idea which is which (because the intent is to place a >> call to

Hi, I'm facing a strange dialplan issue with a PJSIP_DIAL_CONTACTS. When I try to call an offline endpoint with PJSIP_DIAL_CONTACTS, the dial command breaks and the call control go to hangup block instead of next priority. The error in CLI says "*Dial requires an argument (technology/resource)*". This error seems legit as there are no contacts for an offline endpoint. The dialplan

    Is there a solution to dial multiple contacts for a Queue agent?  Since the pandemic started many of our customers have begun to move agents off site.  Since most of them were using softphones we did not have any problems but now we have one case where the agents have a desk phone in their cubicle and are using a softphone from home.  For regular calls there is no problem as

I have an endpoint with multiple phones registered as aor contacts. When I attempt to originate a call it will only ring one of the phones. Is it possible to ring multiple phones as a single endpoint. First phone to answer wins the call and all others terminated? Again, using AMI. Dan -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

Ok I've got multiple phone sets registered with the same extension/secret. However, this causes a strange problem. If I have 3 phone sets registered on extension 123, and I then call extension 123 (from extension 456), only a SINGLE phone set will ring. Is this by design or a bug? Does only the most recently registered phone set ring when I call the extension? Seems odd...is there a way

Hi list(...), I've narrowed down a weird bug. It's like a ghost in the machine, in that functions seem to remember things that they should not be able to. In the example below, the result of the second (and subseqent) calls depend on what was given in the first call. foo <- function(given = NULL) { callObj <- quote(callFunc()) if (!is.null(given)) callObj$given

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.

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

Is there any less cumbersome way of doing conditionalized/branching in extensions.conf other than something like: exten => s,n,GotoIf($["${SIP}" = "PJSIP" ]?pjsip) exten => s,n,Dial(${ARG2},20,TtWw) exten => s,n,Goto(afterdial) exten => s,n(pjsip),Dial(${PJSIP_DIAL_CONTACTS(${STRREPLACE(ARG2,"PJSIP/","")})},20,TtWw) exten =>

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

On Wed, May 27, 2020 at 5:30 PM Dan Cropp <dan at amtelco.com> wrote: > I have an endpoint with multiple phones registered as aor contacts. > > > > When I attempt to originate a call it will only ring one of the phones. > > > > Is it possible to ring multiple phones as a single endpoint. First phone > to answer wins the call and all others terminated? >

On Mon, Jun 15, 2015 at 12:43 AM, Nathan Anderson <nathana at fsr.com> wrote: > What you want is called SIP call forking, and unfortunately, last time I checked (before Asterisk 12 and the advent of PJSIP), Asterisk's SIP channel driver does not support it, and I would be shocked if Asterisk 12+ changes this situation. You can even see that people have written and submitted patches

In other words. I there a way that both phones are ring with only one extension? On 06.02.19 15:05, basti wrote: > both phones are in the same net. > when the soft phone is shut down, on hardware phone only an led is > flashing to show an incoming call but no sound. > > both phones use the same extension. that is the reason why I use pjsip. > > On 06.02.19 14:59, Antony

(I'm new to Asterisk, after having started VOIP with vat on the mbone in the 90s.) I am setting up my first Asterisk system, and trying to read docs/guidance and follow best practices. I have read the 5th Edition of "Asterisk: The Definitive Guide" and like the 3rd Edition on the web it recommends that hardphones and softphones both have a unique name distinct from any concept of

Hello. I am having a lot of problems with SIP through NAT. So, I decided to adopt PJSIP. However, I am not able to make the extensions ring when receiving a call from the queue. I'm using telnet to include the extension and on the asterisk console, it even shows Called PJSIP/6001, but the extension doesn't ring. If I call from extension to extension, it works normally. telenet:

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Thanks for your reply Larry. Le 27/07/2015 01:22, Larry Moore a ?crit : > I think the "488 Not acceptable here" is occurring because the channel > connecting through is not T.38 capable, that will be the IAX channel > from iaxmomdem. This is what T38gateway is supposed to do. And I'm very happy to report that after one more

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..............................................>