similar to: Question about how Asterisk works with RTP ports

Hello, I have defined that I want to receive audio (RTP) on port 11500 till 11954 (rtp.conf). The same range I have defined in my firewall. I now see that an IP-address gets blocked by my firewall because there are packets coming onto port 11955. How come the client sends audio on port 11955 when I clearly define in my SDP-body that I want to receive audio on port range 11500 till 11954 ?

Hello thank you for your answer. I don't understand how there are many tutorials and examples on the web where every time the outcome is a working setup. Very strange I feel now after my personal experience with Asterisk 11 and webRTC. You also say Asterisk 13. How about Asterisk 12 then ?? Kind regards. On 10-08-16 21:53, Matt Fredrickson wrote: > I don't see an ice-ufrag or

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

Hello I am trying to set up webRTC video calls from my Chrome webbrowser (Fedora) to my Chrome webbrowser (Windows 10). There is local video input (I can see myself), but never video on the receiving side. This is the case in both directions (so it makes no difference which peer is calling which peer). Both webRTC SIP peers have opus and H264 codec in their peer definition :   Video

Hello list, this is the communication between an Aastra 5000 PBX and Asterisk, where the Aastra makes a call to Asterisk. For some reason, Asterisk responds with 401-Unauthorized and I don't know why. Calls go well with Panasonic PBX, Avaya PBX, Alcatel-Lucent PBX but NOT with this Aastra. A1.A1.A1.A1 = IP-address Asterisk PBX AS.AS.AS.AS = IP-address Aastra PBX Aastra PBX makes a call

Hello list, I have in sip.conf : /maxexpiry=60 ; Maximum allowed time of incoming registrations ; and subscriptions (seconds) minexpiry=60 ; Minimum length of registrations/subscriptions (default 60) defaultexpiry=120 ; Default length of incoming/outgoing registration ;-----------------------------------------

Hello list when trying to set up webRTC communications with sipjs client package (tried 0.7.0, 0.10.0 and 0.19.0), I see in the asterisk debug log-file the following : DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP c=IN IP4 99.88.77.66... OK. DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtcp:9 IN IP4 0.0.0.0... UNSUPPORTED OR FAILED.

Hello, I want to place an Asterisk-server A in front of 2 other Asterisk-servers (B1 & B2). This first Asterisk-server A needs to send incoming calls to one of the 2 available Asterisk-servers (B1 or B2) behind it. So I want the first Asterisk-server A to accept the call, and based upon some checks in the dialplan send the call through to one of the other Asterisk-servers (B1 or B2)

On 11-08-16 18:03, Matt Fredrickson wrote: > On Thu, Aug 11, 2016 at 9:40 AM, Jonas Kellens <jonas.kellens at telenet.be> wrote: >> My main reason not to upgrade to Ast 13 is because I'm afraid of losing >> functionality as there are certain functions deprecated/replaced. This can >> also cause headache :-) >> >> I will do so if there is no other option.

Hello setting "nat=no" or omitting "nat=" in peer definition does not help either. Still no audio. Why do you think this is a NAT issue ? IP and port information in SDP-body is correct. Kind regards. On 12-08-16 09:25, ????? ?????? wrote: > > Try delete nat from 770000wrtc settings ice should do the same > > > On Aug 11, 2016 10:00 PM, "Jonas

Hello list, I notice that the dialplan method SIPAddHeader is not working : in dialplan : /exten => s,n,SIPAddHeader(Privacy: id)/ in SIP invite no trace of this header : /INVITE sip:0473 at sip.domain.be SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:5063;branch=z9hG4bK-5b2b1b97 From: "VC" <sip:voip2 at sip.domain.be>;tag=729476652f511c67o2 To: <sip:0473 at sip.domain.be>

Hello Using Asterisk 12.8.2. I now have the "via ICE" messages in the RTP debug (see below). If you look in the SIP debug (see below), you also now see the "ice-ufrag" and "ice-pwd" in the 200 OK SIP-message from Asterisk to the webRTC client. But still no audio ! None at all ! In both directions. You can see in the SIP debug that the IP-address in de

hi, i have following topology PSTN - Asterisk ---- internet -----  router - jssip client (wss) Asterisk 13.29.1 on public IP, chan_pjsip for wss, chan_sip/udp for SIP connection to PSTN router - public IP/private IP (NAT) jssip client - private IP - sip over websocket to Asterisk PJSIP ~30% of calls has problem with no audio. reason is that Asterisk is sending RTP to private IP of jssip

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 >

Hi: I have a linksys voip gateway connecting to an asterisk server ,when i dial a call from the linksys gateway to asterisk , i see repeated messages of a RTP errors ,and at same time i hear fake ring on the linksys?, This is wht i see on asterisk console?: ? -- Executing [9613070741 at direct:1] Set("SIP/03070741-088bd470", "CALLERID(number)=96170707070") in new stack ??? --

with wireshark i need decrypt traffic every call which is time consuming. get debug from pjnat through asterisk is not possible because of technical reasons or nobody did it? in my case its strange that ice candidates are the same good call v=0 o=- 3669976329745317845 2 IN IP4 127.0.0.1 s=- t=0 0 a=msid-semantic: WMS EoNIdKcMZvWBLULGqGPJTDe12ujjFEemeapo m=audio 52421 RTP/SAVPF 8 0 101 c=IN

Hey i have an interesting topic to discuss here. The main goal here is to be able to make a video call between two WebRTC endpoints registered on asterisk 13 it is a feature that definitely asterisk 13 should support . the problems that i faced with this is the following and i hope i could get an advise here. asterisk 13 vanilla version has some issues marking the video packets this complain

How can I set the prefered codec between 2 calling parties ?? My Grandstream supports G729, alaw and gsm... in this order. The Zoiper softphone has alaw and gsm as codecs... in that order. Although there should be a matching codec found, my Grandstream can not call the Zoiper softphone. CLI shows : [Mar 11 17:47:21] WARNING[22367]: channel.c:3340 ast_channel_make_compatible: No path to

Hello, are there any known reasons why Asterisk would disconnect random calls ? My server uses 1,5 GB out of 8 GB RAM My server uses up to 35% CPU at peak There are about 40 concurrent calls. I have 300 RTP-ports available. I just see the call ending, as if one of the connected parties hung up but that is not the case ! So what could be a bottleneck ? Any known reasons for random hangup ?

Hello I get the following error when using our Let's Encrypt ssl certificate for webRTC calls : [Jun 2 14:29:28] == DTLS ECDH initialized (secp256r1), faster PFS enabled [Jun 2 14:29:28] ERROR[27360][C-00000ae5]: res_rtp_asterisk.c:1441 ast_rtp_dtls_set_configuration: Specified certificate file '/etc/letsencrypt/live/ws.mydomain.tld/privkey.pem' for RTP instance