similar to: PJSIP realtime: lots of problems

Displaying 20 results from an estimated 8000 matches similar to: "PJSIP realtime: lots of problems"

2015 Feb 02
2
Asterisk 13 - realtime + static modes
Hello, In Asterisk 11 it is possible to set extensions on DB table (sipppers) and also in sip.conf. But in Asterisk 13 apparently this is not possible: as I tried to set in ps_endpoints and also in pjsip.conf but only the realtime endpoints are loaded. Is there a way to use realtime + static modes at the same time for the ps_endpoints lookup using PJSIP. Thanks -------------- next part
2016 Jun 13
2
PJSIP does not qualify contacts after starting Asterisk
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
2016 Jan 21
2
Mixing PJSIP realtime and flat files
Hello, Is it possible to mix PJSIP realtime and flat file configuration in pjsip,conf? What we want is to set up endpoints in the ps_endpoints table with some columns set but most being NULL, and then allow end-customers to optionally add configuration by adding a pjsip.conf section. For example, in ps_endpoinds might be an endpoint with id "asterisk-1" with the transport, aors, auth,
2016 Jun 13
2
PJSIP does not qualify contacts after starting Asterisk
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..............................................>
2005 Jul 25
1
Fortran function name not in load table
Using R 2.0.1 on Windows XP, I am getting an error msg: Error in .Fortran("conic", nxy = nxy, npt = npt, CP = cp, EP1 = ep1, EP2 = ep2, : Fortran function name not in load table I am wondering if there is a way to see what function names are in the load table? Maybe the function name has been altered? The first thing I do in my analysis script is to load a DLL, conic.dll,
2015 Feb 18
3
Asterisk 13 - sorcery realtime for pjsip publish objects
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
2015 Feb 02
2
Asterisk 13 - realtime + static modes
On 2 February 2015 at 15:12, Joshua Colp <jcolp at digium.com> wrote: > Sunny wrote: > >> Hello, >> >> >> In Asterisk 11 it is possible to set extensions on DB table (sipppers) >> and also in sip.conf. >> >> But in Asterisk 13 apparently this is not possible: as I tried to set in >> ps_endpoints and also in pjsip.conf but only the
2015 Jan 04
2
Confused by concepts behind pjsip: endpoint, aor, contact
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.
2016 Jul 12
2
Asterisk 13 MWI
I am still a little confused about how to activate MWI with PJSIP on Asterisk 13.9.1. I use realtime for configuration. So far I have tried setting both the mailboxes field on ps_endpoints and the mailboxes field in ps_aors but I cannot get the indicator lamp to blink on any of my phones (Digium, Aastra and Yealink). I have tried just the number of the mailbox and also adding the context.
2015 Jan 04
3
Confused by concepts behind pjsip: endpoint, aor, contact
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
2011 Jul 21
2
Printing the loop number for each iteration
Hi all, I have a lengthy 'for' loop and for each loop I want to track the iteration number that is currently going on. For this, I have tried following: > for (i in 1:10) { + DumDat <- rnorm(1000) + cat("iteration:", i, " \n") + } iteration: 1 iteration: 2 iteration: 3 iteration: 4 iteration: 5 iteration: 6 iteration: 7 iteration: 8
2012 Feb 09
1
Constraint on one of parameters.
Dear all, I have a function to optimize for a set of parameters and want to set a constraint on only one parameter. Here is my function. What I want to do is estimate the parameters of a bivariate normal distribution where the correlation has to be between -1 and 1. Would you please advise how to revise it? ex=function(s,prob,theta1,theta,xa,xb,xc,xd,t,delta) { expo1=
2004 Aug 19
0
SIP reinvite code negotiation
Hi, We're routing SIP calls through Asterisk and we want to be able to reinvite calls without Asterisk performing codec conversion. We've performed the following test: Asterisk has license for G.729 installed sip.conf [general] context=default autocreatepeer=yes disallow=all allow=alaw allow=g729 canreinvite=yes nat=no We have configured two endpoints: EP1, preferred codec order
2015 Feb 18
2
Asterisk 13 - sorcery realtime for pjsip publish objects
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
2023 Jul 05
3
Setting codec on originating (calling) channel with chan_pjsip (SIP_CODEC_INBOUND equivalent)
Hello, Anyone? I have hard time to believe this is not possible with chan_pjsip. Anyway, may I ask how people handle the following scenario which I imagine should be quite common: - I have internal extensions talk to each other using g722. so their codec setting (with chan_sip now) is "allow=g722,ulaw" - I have carriers trunks that handle ulaw only (allow=ulaw) - calls between
2016 Jul 02
3
Registration server with PJSIP
Hello, I am moving from realtime chan_sip to pjsip and one of the problem I am facing is the lack of "sipregs". With chan_sip, when an extension registers, the server where it has registered to is stored in sipregs. Is there something similar in pjsip? How can I find on which server the pjsip extension has registered to? Leandro -------------- next part -------------- An HTML
2002 Oct 30
1
restricting interfaces.
Hello, I've got samba running on a FreeBSD box that has two interfaces, ep0 which is an external interface, and ep1 which is for internal use only. I only want samba to listen on ep1 so if i'm ever portscanned port 137/139 will not show up as open on the external interface. I've added these lines to the global section of my smb.conf file: hosts allow=192.168.0.
2015 Oct 06
2
PJSIP: how to retrieve underlying SIP Call-ID
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:
2023 Jul 06
1
Setting codec on originating (calling) channel with chan_pjsip (SIP_CODEC_INBOUND equivalent)
On Thu, Jul 6, 2023 at 1:43 PM Michael Ulitskiy <mulitskiy at acedsl.com> wrote: > Hello, > > After I have re-read the "PJSIP Advanced Codec negotiation" document, it > occurred to me that the desired behavior should actually happen > automatically, just due to the codec negotiation logic, but it looks like > asterisk doesn't actually follow the described
2015 Feb 02
0
Asterisk 13 - realtime + static modes
Sunny wrote: > Hello, > > > In Asterisk 11 it is possible to set extensions on DB table (sipppers) > and also in sip.conf. > > But in Asterisk 13 apparently this is not possible: as I tried to set in > ps_endpoints and also in pjsip.conf but only the realtime endpoints are > loaded. > > Is there a way to use realtime + static modes at the same time for the >