similar to: PJSIP - sessions-timers support not working on 13.X

Ok , digging more into this i could see that (timers=no) and (timers=forced) not work asterisk not pay attention to this options when is reloaded cli not say anything and when the pjsip show endpoint <endpoint> it show always timers=yes when (timers=no) and (timers=forced) to that endpoint. I wonder to force asterisk to refresh the session in some cases when is needed . pjsip is able to

Hi guys i was trying to get working sessions-timer over PJSIP channel i was trying to see what is supported or not about this features on the new pjsip channel since chan_sip was kind of flexible on this , at the moment since wiki says pjsip support 4 modes of operation (forced, no, required, yes) but if i try to change any of the timers parameters (timers, timers_min_se or timers_sess_expiries)

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:

Using Asterisk 13.3.2 on CentOS7 and pjproject 2.4, I can't make asterisk try to send a register. I have configured my pjsip.conf similar to https://wiki.asterisk.org/wiki/display/AST/res_pjsip+Configuration+Examples#res_pjsipConfigurationExamples-ASIPtrunktoyourserviceprovider,includingoutboundregistration my pjsip.conf: http://pastebin.com/raw.php?i=EA0PEcrb using tcpdump, I never even

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.

On 06/05/2017 at 11:30 AM, Joshua Colp wrote: > On Sun, Jun 4, 2017, at 10:40 AM, Michael Maier wrote: >> On 06/04/2017 at 01:41 PM Telium Technical Support wrote: >>> Just a guess (without knowing about your network), but are the two ends >>> points on public networks and visible to one another? If not the reinvite >>> may be passing an internal (nat'ed)

Microsip (Windows) is free and small. 2.5Mb download, 10Mb RAM usage, does everything I need and configuring TLS is a doddle. http://www.microsip.org/ On 16 February 2017 at 13:04, Max Grobecker <max.grobecker at ml.grobecker.info> wrote: > Hello, > > I'm a big fan of PhonerLite. > It's more poplar in Germany, but also available in English language. > This client

Hi list , I'm doing some tests with asterisk 13.4 and tls, and failed to make it work, all my terminals spa Cisco 5XX look my cli [Jul 8 11:09:16] ERROR[14733]: pjsip:0 <?>: tlsc0x7f539801 TLS connect() error: Connection refused [code=120111] [Jul 8 11:09:16] WARNING[14733]: pjsip:0 <?>: tsx0x7f53a8008 Failed to send Request msg OPTIONS/cseq=48024 (tdta0x7f53c000dcb0)!

If this is a small site, I recommend you download the free version of SecAst (www.telium.ca <http://www.telium.ca> ) and replace fail2ban. SecAst does NOT use the log file, or regexes, to match etc.instead it talks to Asterisk through the AMI to extract security information. Messing with regexes is a losing battle, and the lag in reading logs can allow an attacker 100+ registration

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

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

Greetings ! My goal is to get Twilio trunking working, and with TLS/SRTP. I see this concerning message in my log: [Feb 7 16:50:26] ERROR[20596] res_sorcery_config.c: Could not create an object of type 'endpoint' with id ?twilio' from configuration file ?pjsip.conf? Thus, ?pjsip show endpoints? does not show the endpoint for the Twilio trunk. Hoping for a sanity check of

Hi All, I installed Astreisk 13beta with pjproject 2.3(2.2.1) from source code on CentOS7. -- https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject The installation is OK. But the connected SIP cilents (both Linphone on Windows7) cannot communicate. I hope your comment such as the testing for resolving the problem. My status is the following(1 and 2). Why 'Everyone

> On 27Apr, 2015, at 16:39, Motty Cruz <motty.cruz at gmail.com> wrote: > > forgot to mentioned I am running Asterisk 1.8.22.0 on CentOS. > > Thanks, > > > On 04/27/2015 02:38 PM, Motty Cruz wrote: >> here is what I have: >> exten => _9XXXXXXX,1,Set(l_HomeAreaCode=381) >> >> exten =>

I am following the instructions in https://wiki.asterisk.org/wiki/display/AST/Basic+PBX+Functionality and I am trying to make a call from extension Alice (6001) to extension for Bob (6002). When I make the call, I can hear the ringing on Alice's phone (caller), but Bob's phone (callee) doesn't ring, or show a call coming in from Alice. My setup and environment is as follows: Alice, Bob

>> 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

I'm trying to set up pjsip to work with an obi202 and google voice. But I can't configure the endpoint. pjsip: [obi202-auth](!) type = auth auth_type = userpass password = <mypass> [obi202-aor](!) type = aor max_contacts = 2 ; ===== endpoints ======== [gv-voice](obi202-endpoint) auth = gv-voice aors = gv-voice identify_by=auth_username ;identify_by=username ; I also tried

I have setup my Asterisk 13.1.0 server with PJSIP on AWS/EC2. My basic configuration works, and I am connected to a SIP trunk using SIP.US, and have set up my inbound calling which works correctly (when I call my PBX DID, the call does come into my PBX network). The issue is that I am not able to make outbound calls, because the call fails with the error:

On Sat, Apr 6, 2019, at 10:04 AM, sean darcy wrote: > On 4/5/19 10:36 AM, sean darcy wrote: > > I'm trying to set up pjsip to work with an obi202 and google voice. But > > I can't configure the endpoint. > > > > pjsip: > > > > [obi202-auth](!) > > type = auth > > auth_type = userpass > > password = <mypass> > > >

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