similar to: PJSIP: Failed to create outgoing session to endpoint

On Mon, Apr 25, 2016 at 11:11 AM, Olivier <oza.4h07 at gmail.com> wrote: > > > > 2016-04-25 18:14 GMT+02:00 George Joseph <gjoseph at digium.com>: > >> >> >> On Mon, Apr 25, 2016 at 10:00 AM, George Joseph <gjoseph at digium.com> >> wrote: >> >>> >>> >>> On Mon, Apr 25, 2016 at 9:29 AM, Olivier <oza.4h07

On Mon, Apr 25, 2016 at 10:00 AM, George Joseph <gjoseph at digium.com> wrote: > > > On Mon, Apr 25, 2016 at 9:29 AM, Olivier <oza.4h07 at gmail.com> wrote: > >> Hello, >> >> I've just discovered PJSIP 's support of set_var setting in pjsip.conf. >> Is this setting also supported in pjsip_wizard.conf ? >> On a fresh 13.8.2, it

On 1 July 2016 at 17:41, Joshua Colp <jcolp at digium.com> wrote: > > >> exten => 1234,Set(CALLERID(all)="Jon Doe" <+123456789>) >> same => n,Dial(PJSIP/phone123, 30) >> > > Your exten line has no priority, is that how it is in your dialplan? > Actually no, I stole that line from an earlier email to this list. Mine has a priority.

Hello, I've just discovered PJSIP 's support of set_var setting in pjsip.conf. Is this setting also supported in pjsip_wizard.conf ? On a fresh 13.8.2, it doesn't seem but I may have missed somthing. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello! Oh, wise ones, ponder with me over two of the surprises that populate the universe! I have a phone, that I sometimes cannot reach, connected via pjsip. It can call other extensions just fine, it can call out over a trunk to my cell, all is well, but getting a call? Forget it most of the time. Here is all the config relevant to that phone: [murftest12] type=aor qualify_frequency=1992

Hi, all Can someone tell me where to tell asterisk no to use 127.0.0.1 IP (localhost)? When I am registering with VoIP providers, they get my info as s@127.0.0.1. (This is SIP registration). Also, in SIP logs, when calling I am getting things like this: Executing SetCallerID("SIP/phone2-22c3", ""CID Name" <CIDNUMBER>") > in new stack > -- Executing

Dear All, Just hope your guys out there can help me through..since I've been playing for serval hours....and still not able to resolve it... The workflow as I've created an .call file for Asterisk to call out and it's working fine with outdial, passing variable to asterisk..But the problem is when the calls reached Context and execute AGI script, the script didn't get any

Hello. I've been pulling together a Debian package out of Maxence Guesdon's OCaml bindings for R. Will be available from my website as soon as I get my router to obey me. Here's Maxence's bindings: http://pauillac.inria.fr/~guesdon/ocaml-r.en.html The purpose of this software is to access R from OCaml programs. However, my issue is that after having pulled things to a Debian

Hello. I've been pulling together a Debian package out of Maxence Guesdon's OCaml bindings for R. Will be available from my website as soon as I get my router to obey me. Here's Maxence's bindings: http://pauillac.inria.fr/~guesdon/ocaml-r.en.html The purpose of this software is to access R from OCaml programs. However, my issue is that after having pulled things to a Debian

Hi, Got problems with incoming SIP calls. Scenario: Server1: 3cx or any other server Server2: Asterisk 16.2.1 . PJPROJECT 2.8 Server2 registers on Server1 with SIP ID 1121. Registration is OK. Server2 outgoing calls are OK. INVITE, unauthorized, INVITE with password, OK, RINGING,... Troubles with incoming calls / incoming INVITE's . I can not identify endpoint by IP, I have multiple

I'm doing an upgrade from Asterisk 11 to 13. I'm following the guide at https://wiki.asterisk.org/wiki/display/ast/setting+up+PJSIP+REaltime to setup realtime, as I use realtime on Asterisk 11 too. I'm getting the following error when trying to connect the peer to the server. Help? :) Thanks, Travis [Jun 15 16:20:03] NOTICE[5116] res_odbc.c: res_odbc: Connected to laf [laf] [Jun

I am having some problems with variable scoping. I need to be able to set a variable that is accessable by other methods within the class (or instance) (i.e not global). An example is 2 pages that change a class variable: class AjtestController < ApplicationController def initialize @@variable = "init" end def show @display = @@variable @@variable = "change 1"

Hello, I've started to play with PJSIP and got stuck at the following problem. I need to retrieve SIP Call-ID associated with PJSIP channel. For inbound channel I can use ${PJSIP_HEADER(read,Call-ID)}, but that doesn't work for outbound channel even in pre-dial or hangup handler. Whatever I do PJSIP_HEADER seem to be unable to read headers for outbound channel. Here's what I do:

Hello, I have had numerous issues with PJSIP outbound calling in Asterisk 13.1.0 and SIP.US SIP trunk. My Asterisk server is on EC2 and I have opened up the appropriate ports. The SIP clients can be anywhere on the Internet, including behind NATs. I am able to get to my Asterisk server's internal extensions via the DID (and appropriate dialplans) but I am not able to make outbound calls to

Running those versions of code, my 7960 will not register with Asterisk. The same 7960 is authenticating against another * server on line 2 just fine though - with the same settings in sip.conf. On the failing * server I am just getting 401 unauthorized errors on the console. From the phone's shell I get that t is registering, but not authenticated .1. from show reg. Any ideas would be

Hello, Anyone know how to set the "To:" in an invite for PJSIP to custom settings. I got the "from" to be the way I need it. From: <sip:e04f43a2ed59 at xaccel.net;tag=44l1nRmW2 To: "TEST" <sip:5tf2f2s0rbtdj-20d14fl6n65t0o-0u03 at 34.221.174.202> I have tried a lot of changes to get to this but nothing works. I am getting this From: sip:109643183 at

Asterisk 13.8 Is CALLERID(all) supposed to wok for pjsip? When I do this: exten => 1234,Set(CALLERID(all)="Jon Doe" <+123456789>) same => n,Dial(PJSIP/phone123, 30) I expect the callerid to be as set, but is always seems to be "phone123", the name of the endpoint. Andrew -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I'm trying to port our configuration form sip to pjsip channel and have following issue. Sip.conf has a host parameter that sets the RURI to a given value. This functionality is needed in some of our scenarios where we need to send requests to specific IP address with specific domain in RURI. I did not found an equivalent to the host parameter in pjsip configuration. Did I

-----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 =>

I am trying to convert my 7905G to be SIP based and seem to be running into a few hassles. Below are all the config files and logs from the server. I have tried to follow the pdf's from cisco and some posts from other mailing lists that google turnedup, but it seems that nothing is working. Am I somehow missing a fundamental step in trying to upgrade from Call Manager to SIP? Any help is