similar to: WSS Socket Configuration

Hi, I have a working WebRTC/SipJS+Asterisk(13.0.1) setup using ws sockets. Now I wanted to switch to wss to have encryption, but cannot find the required configuration parameters. Does Asterisk support wss sockets? How can I configure it? Thanks, Alexej -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I can't mount shares of some servers since I use kernel 2.6.x. As it works with smbclient and also with smbmount on kernel 2.4.x, I assume the problem lies within smbfs. Kernel version: 2.6.8.1 Samba version: 3.0.4 Dist: Debian unstable I get errors when I try to mount a share from OS/2 4.0: 1) smbmount everything's fine 2) cd into the mounted dir smbfs output: ?

Hola! Dovecot should serve virtual mail users. So I've set user_global_uid and user_global_gid in dovecot_ldap.conf to vmail/vmail. Also I've commented auth the user_attrs field. Still Dovecot tries to switch to the uid that is defined in the LDAP entry. It took me some time to figure out, that the only way to prevent this is to set user_attrs = foo=uid,bar=gid or something like this,

Hi, I'm trying to connect to the asterisk pbx via wss, from sipml5.org demo page (http://sipml5.org/call.htm). I used the guide from https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial , to setup the tls. I could make a secure sip call ( SRTP) using the PhonerLite sip client. ( This confirms my sip - tls settings and tls certficates. ( I'd added the tls client certficate

Hi, I've tried unsuccessfully to get printing working under wine. I use wine version 20040309 on Debian unstable. I use cups for printing and it works without problems. According to the wine doc, I did not add any entries to the wine config file or to the registry. My printcap file looks like this: kyocera|Kyocera FS-1500:rm=sbs93:rp=kyocera: phaser|phaser:rm=sbs93:rp=phaser: When I

Hello! I have problems using Samba with Kernel 2.6. With smbmount: mounting works but changing to the mounted directory or doing an 'ls' there leads to a timeout. Syslog says: $ cd /mnt/serv/c Jul 27 10:37:19 sbs93 kernel: smb_setup_bcc: Packet too large 4257 > 4096 Jul 27 10:37:49 sbs93 kernel: smb_add_request: request [ea195e80, mid=0] timed out! $ ls Jul 27 10:38:36 sbs93

Hello, I can't mount shares of some servers since I use kernel 2.6.x. As it works with smbclient and also with smbmount on kernel 2.4.x, I assume the problem lies within smbfs. Kernel version: 2.6.8.1 Samba version: 3.0.4 Dist: Debian unstable I get errors when I try to mount a share from OS/2 4.0: 1) smbmount everything's fine 2) cd into the mounted dir smbfs output: ? smb_setup_bcc:

Hi, I use chrome and sipml5 to connect to asterisk webrtc interface using TLS. The wss connection seems ok and the SIP REGISTER sent from chrome to asterisk and the SIP response received. In the response, I get a "failed: A server must not mask any frames that it sends to the client" error msg and chrome terminates the ws connection. I've checked the asterisk debug logs, and the

Hi, Have anyone tried using SIPML5 to connect to Asterisk over wss? I'm having the error as shown below Connecting to 'wss://54.xxx.xxx.xxx:8080/ws <wss://54.254.228.251:8080/ws>' SIPml-api.js?svn=224:1 ==stack event = starting SIPml-api.js?svn=224:1 __tsip_transport_ws_onerror SIPml-api.js?svn=224:1 __tsip_transport_ws_onclose SIPml-api.js?svn=224:1 ==stack event =

I'm learning about WebRTC clients, and am wondering why Asterisk treats them differently from any other SIP client. The media (RTP) should be no different, so the only difference should be on the signaling side. I noticed that the Asterisk wiki mentions the need for res_pjsip_transport_websocket, so does that mean Asterisk requires the signaling to occur over a websocket? If I used

Hello, Can anyone recommend a particular online WebRTC phone for testing with Asterisk? We tried: - JsSIP, but even with the "enable video" checkbox disabled it sends video options in the INVITE SDP and Asterisk rejects it with "Rejecting secure video stream without encryption details". - sipML5, but it won't register, perhaps something to do with not using the Asterisk

Hello, On a newly re-installed Asterisk 16.7.0 on Debian Buster, I can't find a way to enable HTTPS. Asterisk is running as asterisk:asterisk: asterisk 11097 0.3 6.7 741352 67984 ? Ssl 17:53 0:06 /usr/sbin/asterisk -g -f -p -U asterisk # cat /etc/asterisk/http.conf [general] servername=Asterisk enabled=yes bindaddr=0.0.0.0 bindport=8088 tlsenable=yes tlsbindaddr=0.0.0.0:8089

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.

In the asterisk wiki instructions for Configuring Asterisk for WebRTC clients... https://wiki.asterisk.org/wiki/display/AST/Configuring+Asterisk+for+WebRTC+Clients "To communicate with websocket clients, Asterisk uses its built-in HTTP daemon. Configure /etc/asterisk/http.conf as follows: [general] enabled=yes bindaddr=0.0.0.0 bindport=8088 tlsenable=yes tlsbindaddr=0.0.0.0:8089

Hello, Le lun. 6 janv. 2020 à 19:01, Olivier <oza.4h07 at gmail.com> a écrit : > May I add I could successfully (if pjsip show transports has any meaning) > add a PJSIP TLS-transport with: > > [transport-tls] > type=transport > protocol=tls > bind=0.0.0.0:5061 > cert_file=/etc/asterisk/keys/asterisk.crt > priv_key_file=/etc/asterisk/keys/asterisk.key >

Hello, After countless hours on, this I found the root cause of HTTPS settings on Debian Buster. All this came from ast_tls_cert script using 1024 bits-long keys where Debian's defaut was to require at least 2048-long keys ! Simply passing -b 2048 to ast_tls_cert solved it. 1. May I suggest mentioning explicitly this possibility in wiki page [1] ? 2. What would you say of adding an extra

Hello, I'm trying to get calls working between websocket clients and sip clients. For clients I have sip.js based clients on chrome, Zoipers and a Grandstream phone. Challenge here is I'd like to have Kamailio and rtpengine to handle the bridging between different rtp profiles but Asterisk changes them in the sdp bodies along the way. I'm using Asterisk 11.11.0. Is there a way to

I need to get hooked up with this class, I could have students doing projects for homework :) Interested in RTCP? j On 6/26/23 7:45 PM, TTT wrote: > > I’m in training, so I have to demonstrate something SIP related.  I > figure it would be cool to hack a call, hanging it up while in > progress from outside Asterisk.  Doing so will demonstrate > use/knowledge of ARI, AMI, SIP,

hi, i'm fighting with webrtc for 14 days reporting my experience to minimize number of crazy asterisk users i have working webrtc with simpl5 + asterisk 13 + pjproject 2.4.5 + chan_pjsip + secure websockets + secure audio + audio in both ways problems first, i needed run chan_sip for old hard phones and wss with chan_pjsip only for webrtc. this is possible with patch from

Hello, I was testing with sdp and something came up worth asking: While calling from a webrtc client to another (chrome, sip.js) Asterisk receives the following sdp and rejects it with 488 Not Acceptable. Why does this happen, what's wrong with the sdp? The second sdp body below is accepted instead. Both have rtp profile RTP/SAVPF, difference is that the second one was produced by rtpengine,