similar to: Adding Subscribe Handlers in PJSIP

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

Hello - I am trying to decide if I have stumbled across a bug in PJSIP or I am just missing something. My Asterisk has two interfaces, an "internal" eth0 and an "external" eth1. In pjsip.conf, I define the following transports: [trusted] type=transport protocol=udp bind=10.xx.yy.zz:5060 [untrusted] type=transport protocol=udp bind=12.4.aa.bb:5060 My internal endpoints use

On Thu, Apr 2, 2015 at 10:43 AM, Rusty Newton <rnewton at digium.com> wrote: > On Wed, Apr 1, 2015 at 9:08 AM, Trey Hilyard <kctrey at gmail.com> wrote: > >> Hello - >> >> I am trying to decide if I have stumbled across a bug in PJSIP or I am >> just missing something. My Asterisk has two interfaces, an "internal" eth0 >> and an

I found an issue with how PJSIP handles a typo in the Dial application. If the Channel is mistakenly typed with two slashes (i.e Dial(PJSIP//xxxx...), the Dial applications fails (obviously), but it also kills the server. I put some code in my pbx_config to check for that string and not let the dialplan reload, but it seems like there should be a better way to handle in in the PJSIP stack or Dial

Thanks for the reply Trey, should of said I'm using chan_sip. Regards Mick On 16 Feb 2016 18:03, "Trey Hilyard" <kctrey at gmail.com> wrote: > Are you using res_pjsip or chan_sip? > > For PJSIP, it's as easy as passing the parameters to the Dial. For example: > Dial(PJSIP/${ARG1}\;phone-context=mydomain.com at pjsippeer,60) > > I am pretty sure it was

On Fri, Mar 18, 2016 at 10:49 AM Administrator TOOTAI <admin at tootai.net> wrote: > Le 18/03/2016 16:20, Trey Hilyard a ?crit : > > I am trying to set up my Asterisk server so that it will recognize an > > incoming call to the Asterisk's own Location Routing Number (LRN), > > validating the "rn" in the INVITE and then using the Called Number from >

On Fri, 18 Mar 2016, Trey Hilyard wrote: > I thought this would be as easy as > exten => _XXXXXXXXXX\;rn=+19136630000,1,Goto(from_pstn,${EXTEN:0:10}) Have you tried the '_!.' pattern? -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST

Hi all, I am currently using asterisk 11, and I am trying to figure out how to set the uri parameter telephone-context. I need to set it for outbound calls for a specific carrier when making emergency calls and don't seem able to find the option to set it. Regards Impy aka Mick -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

I am trying to set up my Asterisk server so that it will recognize an incoming call to the Asterisk's own Location Routing Number (LRN), validating the "rn" in the INVITE and then using the Called Number from the INVITE as the extension in the dialplan. The INVITE R-URI looks like: INVITE sip:+19135041291;rn=+19136630000;npdi at 12.4.240.200:5060;user=phone;transport=udp SIP/2.0

On Wed, Apr 1, 2015 at 9:08 AM, Trey Hilyard <kctrey at gmail.com> wrote: > Hello - > > I am trying to decide if I have stumbled across a bug in PJSIP or I am > just missing something. My Asterisk has two interfaces, an "internal" eth0 > and an "external" eth1. In pjsip.conf, I define the following transports: > > [trusted] > type=transport >

Hello, I am currently trying to set up pjsip realtime and would like to have outbound-publish, inbound-publication, and asterisk-publication sorcery object types in ODBC realtime. Is that currently supported? I know that some object types are known working and others are not. I was curious what the status of those objects are. Thanks! Matt Hoskins | NPG Corp | Systems Architect

Hello List I am in the progress of migrating from chan_sip to pjsip. I fear I have missed something on how hints need to be specified for pjsip. For chan_sip I have configured sip.conf subscribecontext = localuser and in the dialplan I set: [localuser] exten => 11,hint,SIP/11 Now if a phone subscribes to '11' this works. Now I try to get the same working for pjsip. I understood

Hi Joshua thank you for the quick reply > Have you checked the Asterisk console when PJSIP is loaded to see if > the endpoint did not load for some reason? Does it show up in "pjsip > show endpoints"? Yes, the endpoint shows up. Endpoint: 11/(scrubbed from mail) Not in use 0 of inf InAuth: 11/11 Aor: 11

Excellent. I was using ast-13.1.0 with no luck. I upgraded to 13.2.0 and have made it further, but am having a little difficulty. The outbound-publish object types seems to be working in realtime now. But the asterisk-publication object is only reading from sorcery.conf. I know you said that it *should* work, with no guarantee, which I'm fine with. I just want to make sure I don't

Hi Joshua > The chan_pjsip module doesn't prevent that. You'd need to provide the > full SUBSCRIBE now that it is actually finding the endpoint and coming > in. Ok, let's see if we can solve the mystery.. pjsip.conf [endpt-home](!) type=endpoint disallow=all allow=g722 allow=alaw allow=gsm ice_support=yes context=from-home allow_subscribe=yes

The call initiated from internal extension. I have made two test call: Successful call from device on res_pjsip via endpoint on chan_sip: http://pastebin.com/LWeDYstj Unsuccessful call from device on res_pjsip via endpoint on res_pjsip: http://pastebin.com/hepVb6Nu And ones again i don't see anything that would make asterisk send BYE. I would be grateful for any ideas. 11.02.2016 1:47,

i noticed that when i active the voicemail in the IP-phone where the number 0033149xxxxxx is configured i can call this number without issue Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called SIP/FD/0033149xxxxxx == Begin MixMonitor Recording SIP/101-0000010d -- SIP/FD-0000010e is making progress passing it to SIP/101-0000010d > 0x2b393cfc2610 -- Probation passed

On 16-08-16 04:38, George Joseph wrote: > > > On Mon, Aug 15, 2016 at 1:24 PM, Jonas Kellens > <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote: > > Hello > > using pjproject 2.5.5 > using asterisk-certified-13.8-cert1 > > > IIRC there were API changes in pjproject 2.5 that aren't accounted for > in

Asterisk Project Security Advisory - AST-2014-005 Product Asterisk Summary Remote Crash in PJSIP Channel Driver's Publish/Subscribe Framework Nature of Advisory Denial of Service Susceptibility