similar to: CHANNEL(aor) CHANNEL(contact) return nothing

Hello I have 2 strange errors when using the Background()-application and DTMF-input that is received. First of all, my first 2 lines are not being executed. The first line being executed is the Set() application, thus line 3. Secondly, the received digits (911) is not the same as the EXTEN (which is set to 91). exten => ivr,n,Set(TIMEOUT(digit)=2) exten =>

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Le 20/05/2015 00:50, Joshua Colp a ?crit : > It looks like this is an incoming leg, in which case that information > isn't available. There is no association of an AOR and Contact on > incoming legs (it MAY be possible to deduce but it certainly wouldn't > work in all cases). Since you specify one explicitly on outgoing, that

On Thu, Sep 12, 2019 at 04:19:31PM +0200, Miklos Szeredi wrote: > From: Stefan Hajnoczi <stefanha at redhat.com> > > Michael, > > Here's a v6 of the virtiofs code (fuse.git#virtiofs-v6). I think we've > addressed all your comments. > > Would you mind giving it another look, and if you're satisfied acking this > patch? > > Thanks, > Miklos

On Tue, Sep 03, 2019 at 01:42:02PM +0200, Miklos Szeredi wrote: > From: Stefan Hajnoczi <stefanha at redhat.com> > > Add a basic file system module for virtio-fs. This does not yet contain > shared data support between host and guest or metadata coherency speedups. > However it is already significantly faster than virtio-9p. > > Design Overview > ===============

On Tue, Sep 03, 2019 at 01:42:02PM +0200, Miklos Szeredi wrote: > From: Stefan Hajnoczi <stefanha at redhat.com> > > Add a basic file system module for virtio-fs. This does not yet contain > shared data support between host and guest or metadata coherency speedups. > However it is already significantly faster than virtio-9p. > > Design Overview > ===============

hi all! i have a dsl connection with 1280 kbps for downstream and 256 kbps for upstream, and i want to manage the bandwidth to give high priority to voip traffic and low priority to p2p traffic. i found the script jdg-qos. i readed on this forum (i have a dsl-g604t router with MCMCC firmware) http://www.dslreports.com/forum/remark,16250220 that the two parameters of the jdg-qos script (DWIFLIMIT

Hello I've tried a bunch of things, but still get errors/warnings when using the IF() function: ============== TEST #1 exten => h,n,Set(WAV_FILE=${IF($[ ${STAT(e,/tmp/${CALLTIME}.wav)} ]?${CALLTIME}.wav)}) [Nov 26 21:52:34] WARNING[5074]: func_logic.c:107 acf_if: Syntax IF(<expr>?[<true>][:<false>]) ============== TEST #2 exten =>

Asterisk has this error on compile: flex ast_expr2.fl "ast_expr2.fl", line 50: unrecognized %option: reentrant "ast_expr2.fl", line 51: unrecognized %option: bison-bridge "ast_expr2.fl", line 52: unrecognized %option: bison-locations make: *** [ast_expr2f.c] Error 1 2.6.12-1.1447_FC4smp #1 SMP bison (GNU Bison) 2.0 Written by Robert Corbett and Richard Stallman.

Hi Guys, Using the 2.6.test kernel? Alex and I have done some more tests and this is the patch incase ur using jdg-qos script http://www.digriz.org.uk/jdg-qos-script/index.html#qos-2.6 Thanks K _________________________________________________________________ Winterize your home with tips from MSN House & Home. http://special.msn.com/home/warmhome.armx

Hi. After successfully running ./configure I run make. When running make I get the following error.. [CC] ast_expr2f.c -> ast_expr2f.o [CC] ast_expr2.c -> ast_expr2.o [CC] strcompat.c -> strcompat.o [LD] aelparse.o aelbison.o pbx_ael.o ael_main.o ast_expr2f.o ast_expr2.o strcompat.o -> aelparse aelparse.o(.text+0x3029): In function `ael_yylex':

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

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

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. > >

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...

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

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

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.

Hello. Asterisk 13.2, PJSIP. Problem: I do not get any AMI events when changing the status of the contact. When using chan_sip I got "peerstatus" event. When using res_pjsip and devices (endpoint configuration) I got "peerstatus" event. When using res_pjsip.so and OUTBOUND REGISTRATION WITH OUTBOUND AUTHENTICATION i got "registry" event. When using

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

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