search for: setup_srtp

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

2015 Nov 12
3
No sound with internal calls depending on which phones
...erfectly, * internal calls between some phones too, * but internal call between two similar phones don't work !!! (Snom 710) When we have sound, there are no errors in asterisk. When we do not have sound, there is the following error : * [Nov 10 17:51:47] ERROR[21480]: chan_sip.c:28306 setup_srtp: No SRTP module loaded, can't setup SRTP session. This is a working internal call : > == Using SIP RTP CoS mark 5 > -- Executing [301 at local:1] Dial("SIP/dbucher-00000000", > "SIP/phone1") in new stack > == Using SIP RTP CoS mark 5 > -- Cal...
2015 Nov 12
3
No sound with internal calls depending on which phones
...work perfectly, * internal calls between some phones too, * but internal call between two similar phones don't work !!! (Snom 710) When we have sound, there are no errors in asterisk. When we do not have sound, there is the following error : * [Nov 10 17:51:47] ERROR[21480]: chan_sip.c:28306 setup_srtp: No SRTP module loaded, can't setup SRTP session. This is a working internal call : == Using SIP RTP CoS mark 5 -- Executing [301 at local:1] Dial("SIP/dbucher-00000000", "SIP/phone1") in new stack == Using SIP RTP CoS mark 5 -- Called phone1 -- SIP/phone...
2010 Dec 01
0
<solved!> Asteris 1.8 and mISDN - 'mISDN' (cause 66 - Channel not implemented)
...c/init.d/asterisk start", I can't load chan_misdn.so If I run asterisk 1.8 as root via "asterisk -vvvc" I can access my ISDN-card and I be able to dial out to my PSTN provider! ;) Example: *CLI> == Using SIP RTP CoS mark 5 [Dec 1 10:49:47] ERROR[16779]: chan_sip.c:27876 setup_srtp: No SRTP module loaded, can't setup SRTP session. -- Executing [089216750916 at default:1] Dial("SIP/14-00000000", "mISDN/g:Mnet/089216750916") in new stack -- Called g:Mnet/089216750916 -- mISDN/1-u1 is proceeding passing it to SIP/14-00000000 -- mIS...
2010 Nov 30
2
Asteris 1.8 and mISDN - 'mISDN' (cause 66 - Channel not implemented)
...centos5.i386 I've configured ma testhost lieke my production-server (asterisk 1.6), but if I try to dial out to my local PSTN, I receive an error: [root at asterisk-testing ~]# asterisk -r Verbosity is at least 3 == Using SIP RTP CoS mark 5 [Nov 30 10:35:53] ERROR[5043]: chan_sip.c:27876 setup_srtp: No SRTP module loaded, can't setup SRTP session. -- Executing [08936046666 at default:1] Dial("SIP/14-00000003", "mISDN/g:Mnet/08936046666") in new stack [Nov 30 10:35:53] WARNING[7281]: channel.c:5353 ast_request: No channel type registered for 'mISDN' [...
2012 Aug 17
2
How to test Websocket support in SIP in Asterisk trunk?
I see no indication of how to do this in sip.conf, and when I start Asterisk, it doesn't wait on port 80. Greetings, -- Juan Carlos Castro y Castro Instant Solutions - Telefonia Gerando Resultado http://www.instant.com.br Principais capitais: 4063-6100 Demais regi?es: (11)4063-6100