similar to: Asterisk 1.8.32.3 : no video (h.264)

Hello in sip.conf I have ; videosupport=yes Kind regards. J. On 20-04-17 13:09, Marcelo Terres wrote: > I suppose that you enable the video support on sip.conf, right? > > Regards, > Marcelo H. Terres <mhterres at gmail.com> > IM: mhterres at jabber.mundoopensource.com.br > https://www.mundoopensource.com.br > https://twitter.com/mhterres >

Hello you mean while placing a video call ? What info am I looking for in the debug output ? Kind regards. J. On 21-04-17 12:28, Marcelo Terres wrote: > Did you try to activate DEBUG and set the verbosity to a higher level > (100?) to check what Asterisk tells you about? > > Regards, > Marcelo H. Terres <mhterres at gmail.com> > IM: mhterres at

Am 13.06.2020 um 13:47 schrieb Michael Keuter: Hi > Try "sip show peer <peername>" for a phone. So: mobile phone: bpi*CLI> sip show peer 0049177xxxxxxx * Name : 0049177xxxxxxx Description : Secret : <Set> MD5Secret : <Not set> Remote Secret: <Not set> Context : default Record On feature : automon

Hi. we are experimenting a strange issue in our PBX. By example: if we dial to the 100, the call is answered in 199. We dont have any redirection for that, but the cli show the same issue when request show peers. Aditionally, the user 100 use the ip address 192.168.11.100, and the cli show connected the user from 192.168.11.160 (that ip is assigned to the user 199) PBX*CLI> sip show peers

Am 13.06.2020 09:30, schrieb Luca Bertoncello: Hi again (again) I noticed right now another strange detail... I made a call using my mobile phone (connected to the Asterisk). The quality was top... Maybe is the problem in a codec used from our phones at homes? Could someone suggest me how to check the codec used by my mobile phone and the codec used by the phones at home? Thanks Luca

I wasn't able to get much out of babytel, beyond the fact that I was, apparently, sending options which is why I'm not getting 200 OK. How can I, generally speaking, ping/telnet or otherwise test the connection to get more data? A connection to this peer directly from a softphone, Jitsi, works fine. linux-k7qk*CLI> linux-k7qk*CLI> sip show peer testcarrier * Name :

Hi gurus!!! I have a Freepbx with Asterisk 1.8.25.0 with a sip trunk on the pstn Every minute asterisk sends an OPTION Request, i beleived that it's related to qualify functions. The every minute annoyng answer of the pstn is "403 Forbidden". Some people told that asterisk is not sending the username in the OPTION, required by the pstn. Taking a look of the example of rfc3261.txt

I'm reading the O'Reilly "Asterisk the definitive guide", 4th ed, with a starfish on it. In some ways, astonishing that it's not really that definitive, it's more general -- and it only clocks in at one ream of paper! In any event, I'm having some port problems on my home network: http://security.stackexchange.com/questions/81752/ I need to open ports for

Hi After recently upgrading to 1.8.3 I have noticed that the nat setting for my peer in my sip table is not making it into the realtime cache. For example * Name : 501 Realtime peer: Yes, cached Secret : <Set> MD5Secret : <Not set> Remote Secret: <Not set> Context : pack-local Subscr.Cont. : <Not set> Language : AMA flags :

Hello mailinglist, I want to connect Asterisk with OpenBTS and make a call with a mobile phone. I use: Ubuntu 11.10 + Kernel 3.0.22 GnuRadio 3.3.0 Asterisk 1.8.13 OpenBTS 2.8 Nokia Mobile Phone OpenBTS works and I can send sms from the OpenBTS server to the mobile phone. What I also need is a call between Asterisk and OpenBTS. I have also two soft phones which works with Asterisk. And also

We have a system where calls are coming in from telcos via an opensips server and then being redirected out to a regular sip destination. There is no NAT, DTMF features, call recording, or codec translation being performed so I would expect asterisk to issue a reinvite after the call is answered and switch the audio however it is not happening. Here is the sip peer information for the call

So the call used Alaw as Codec. > Am 13.06.2020 um 17:23 schrieb Luca Bertoncello <lucabert at lucabert.de>: > > Am 13.06.2020 um 13:47 schrieb Michael Keuter: > > Hi > >> Try "sip show peer <peername>" for a phone. > > So: > > mobile phone: > bpi*CLI> sip show peer 0049177xxxxxxx > > > > > * Name :

Turn on sip debugging for this peer and watch for the options sending and response. If you are getting a response to your options asterisk shouldn't be marking the peer as unavailable. is your asterisk behind a firewall? On 20 March 2015 at 13:42, thufir <hawat.thufir at gmail.com> wrote: > I wasn't able to get much out of babytel, beyond the fact that I was, > apparently,

Hello all, I have installed the .deb packages of the Asterisk v1.8.3.3 from the upstream project on my Debian GNU/Linux Squeeze server and bought the Comodo's PossitiveSSL SSL certificate to be used for my SIP/TLS exercise. After setting up everything and trying to fix this problem, I am still getting a 401 Unauthorized SIP message. So as of this writing, I still cannot successfully REGISTER

It looks as if that is more of a question/issue with your router, rather than Asterisk. I have SIP devices working on my LAN, all hardwired, and have no need to open any ports or have the router address SIP in any way My switch is not managed, and the router ports on the LAN side are all unmanaged, just a huge Ethernet "wirenut" You SHOULD be able to communicate between devices on the

I'm trying to configure SIP trunking. Now, I'm referencing "Asterisk the definitive guide", 4th ed. While I don't have the page handy, I was reading the suggestion to try SIP to SIP before proceeding to outside connectivity. I'm aware that SIP trunking is a construct, but am, obviously, learning the system. What I'd like to do is from the CLI "ping"

Asterisk Version: 1.8.1.1 Problem: Multiple Parking Lots Issue: Not redirecting to the right parking lot. Always uses the first parking lot from "parkedcalls show" or "features show" Asterisk Working Version: 1.6.1 Steps Taken: In features.conf added: [parkinglot_test] context => parkedcalls-test parkext => 700 parkpos => 701-710 parkingtime => 120 findslot

I'm having a strange problem recently with a Yealink SIP-T28P phone connected to Asterisk 11.4.0 via openvpn. It was working fine for months, and now when I dial anything from the phone, it shows "Forbidden", and the Asterisk console shows: [May 21 10:47:49] NOTICE[28518][C-00000004]: chan_sip.c:25189 handle_request_invite: Failed to authenticate device "Ext 110" <

I am trying to set up Hunt Groups and I am having some issues. Here is what I am trying to do. All my users actually register with OpenSIPS. Asterisk is using Realtime and I have set up a MySQL View Table so that Asterisk see's all the SIP users info that OpenSIPS has. This is what I have configured queues.conf ---------------------------------- [irock.com] strategy=leastrecent

In the Asterisk log I see: --- [Mar 23 19:25:29] VERBOSE[4067] chan_sip.c: [Mar 23 19:25:29] <--- SIP read from UDP:198.38.7.34:5065 ---> SIP/2.0 200 OK To: <sip:16046289850 at sip.babytel.ca>;tag=sd3D4swKRc From: <sip:16046289850 at sip.babytel.ca>;tag=as07c833c5 Via: SIP/2.0/UDP 96.48.217.39:5060;branch=z9hG4bK13c68eb7;rport Call-ID: