search for: srtpcapable

Displaying 5 results from an estimated 5 matches for "srtpcapable".

2015 Nov 12
3
No sound with internal calls depending on which phones
...3, len 000033) > Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 000033) > == Spawn extension (local, 301, 1) exited non-zero on > 'SIP/hsolutionspf5-00000002' I tried many options to disable SRTP but without success : * canreinvite = no * canreinvite = nonat * srtpcapable=no * encryption=no * directmedia=nonat * ...or noload => res_srtp.so in modules.conf Any help would be GREATLY appreciated ! Denis P. S. We have Asterisk 1.8.4.4 under CentOS release 5.11 (Final) -------------- next part -------------- An HTML attachment was scrubbed... URL: <http:...
2015 Nov 12
3
No sound with internal calls depending on which phones
...nt RTP P2P packet to 192.168.128.231:51350 <http://192.168.128.231:51350> (type 03, len 000033) == Spawn extension (local, 301, 1) exited non-zero on 'SIP/hsolutionspf5-00000002' I tried many options to disable SRTP but without success : * canreinvite = no * canreinvite = nonat * srtpcapable=no * encryption=no * directmedia=nonat * ...or noload => res_srtp.so in modules.conf Any help would be GREATLY appreciated ! Denis P. S. We have Asterisk 1.8.4.4 under CentOS release 5.11 (Final) -- _____________________________________________________________________ -- Bandwidth and Colo...
2013 Mar 31
0
SRTP woes
...oaded: *CLI> module show like res_srtp.so Module Description Use Count res_srtp.so Secure RTP (SRTP) 0 1 modules loaded extensions.conf extract exten => 1002,1,Set(_SIPSRTP=${SIPPEER(1002,srtpcapable)}) exten => 1002,n,Set(CHANNEL(secure_bridge_signaling)=1) exten => 1002,n,Dial(SIP/exten1002,20) exten => 1002,n,Hangup() sip.conf extract: [exten1002] type=friend host=dynamic secret=averygoodone context=users nat=force_rport,comedia encryption=yes transport=tls Thanks, Regards, John...
2014 May 09
1
deactivate SRTP in asterisk 11
Hi all, i try to deactivate SRTP in asterisk 11. In sip.conf: tlsenable=no encryption=no transport=udp srtpcapable=no but when I try to make a call comes following message: [May 9 15:19:03] DEBUG[24745][C-00000086]: sip/sdp_crypto.c:285 sdp_crypto_process: Accepting crypto tag 1 [May 9 15:19:03] DEBUG[24745][C-00000086]: sip/sdp_crypto.c:310 sdp_crypto_offer: Crypto line: a=crypto:1 AES_CM_128_HMAC_SHA1_32...
2014 Aug 12
0
Asterisk 11.11 with TCP/TLS SRTP and Grandstream gxp1450 not working
...ilure WARNING[7421]: tcptls.c:668 handle_tcptls_connection: FILE * open failed! Encryption is configured via ;-------------------------Encryption----- encryption=yes tlsenable=yes tlsbindaddr=:: tlscertfile=/var/lib/asterisk/keys/asterisk.pem tlscafile=/var/lib/asterisk/keys/ca.crt tlscipher=ALL srtpcapable=yes ;tlsclientmethod=tlsv1 tlsdontverifyserver=yes and the phone is sourced by context=default ; Default context for incoming calls allowoverlap=no udpbindaddr=:: tcpenable=yes tcpbindaddr=:: srvlookup=yes and [IPV6](!,my-codecs) dtmfmode=rfc2833 context=sip-out type=friend hos...