[asterisk-users] Chan_h323 isn`t dropping calls comming with wrong codecs
I` using chan_h323 on my asterisk-1.4 to receive incomings calls. I need to set just two codecs to receive this call (g723 and g729), but I`m using disallow=all allow=g729 allow=g723.1 In h323.conf, but when I received a call using codec g711 for example, the call is answered, but doesn`t have audio. I made a test today using just disallow=all In h323.conf, but the call was answered too!! the log of this test: [Feb 21 23:44:56] DEBUG[4264]: chan_h323.c:2112 setup_incoming_call: Setting up incoming call for ip$189.0.24.69:4020/28391 -- Setting up Call -- CLICall token: [ip$189.0.24.69:4020/28391] -- CLICalling party name: [200] -- CLICalling party number: [200] -- CLICalled party name: [30144588] -- CLICalled party number: [30144588] -- CLICalling party IP: [189.0.24.69] [Feb 21 23:44:56] DEBUG[4264]: chan_h323.c:1611 find_user: Could not find user by name 200 or address 189.0.24.69 [Feb 21 23:44:56] DEBUG[4264]: chan_h323.c:2177 setup_incoming_call: Sending [EMAIL PROTECTED] to context [ss7] extension 30144588 [Feb 21 23:44:56] DEBUG[4264]: chan_h323.c:2478 set_local_capabilities: Setting capabilities for connection ip$189.0.24.69:4020/28391 Setting capabilities to 0x0 (nothing) Capabilities in preference order is () DTMF mode is 1 Allowed Codecs for ip$189.0.24.69:4020/28391 (ip$201.7.99.242:1720): Zap/33-1 answered H323/ip$189.0.24.69:4020/28391 [Feb 21 23:44:57] DEBUG[4264]: chan_h323.c:666 oh323_answer: Answering on H323/ip$189.0.24.69:4020/28391 Answering call ip$189.0.24.69:4020/28391 Receiving RFC2833 on payload 101 Peer capability is G.711-uLaw-64k 1 Found peer capability G.711-uLaw-64k 1, Asterisk code is 4, frame size (in ms) is 20 Peer capability is G.711-ALaw-64k 2 Found peer capability G.711-ALaw-64k 2, Asterisk code is 8, frame size (in ms) is 20 Peer capabilities = 0xc (ulaw|alaw), ordered list is (ulaw|alaw) [Feb 21 23:44:57] DEBUG[4264]: chan_h323.c:2448 set_peer_capabilities: Got remote capabilities from connection ip$189.0.24.69:4020/28391 [Feb 21 23:44:57] DEBUG[4264]: chan_h323.c:2462 set_peer_capabilities: prefs[0]=ulaw:20 [Feb 21 23:44:57] DEBUG[4264]: chan_h323.c:2462 set_peer_capabilities: prefs[1]=alaw:20 =-= In OnConnectionEstablished for call 28391 -- Connection Established with 200 [189.0.24.69] [Feb 21 23:44:57] DEBUG[4264]: chan_h323.c:2055 connection_made: Call ip$189.0.24.69:4020/28391 answered Some one knows why isn`t asterisk droping the call? Andre Luiz Martins ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Stress test
Hello peoples, I need to do a test of urgent stress. It know as much as connections simultaneous my equipment is going to do passing codec g729 and g723. Someone knows say me as obtain does him? Andre Luiz Martins mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] problem in ooh323
Hello everbody, I have a problem, I installed ooh323 in mine * but when I try to dial appears like itself be contacting the gatekeeper but nothing happens. follows my ooh323.conf [general] port=1720 bindaddr=0.0.0.0 gateway=no h323id=XXX e164=1234567890 callerid=h323id gatekeeper = ipgatekeeper context=voip-h323 disallow=all ;Note order of disallow/allow is important.allow=g729allow=gsmallow=ulawdtmfmode=rfc2833 in debug mode i have --- ooh323_request - data 55XX format 0x4 (ulaw)--- find_peer+++ find_peer+++ ooh323_request--- ooh323_call- 55XX+++ ooh323_call -- Called 55XX ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ooh323 error
Hello everbody I have a new error in connection with gatekeeper: 19:08:39:219 Enabled RFC2833 DTMF capability for (outgoing, ooh323c_o_13)19:08:39:219 Parsing destination 55218702233619:08:39:219 Destination is parsed as dialed digits 55218702233619:08:39:219 Trying to connect to remote endpoint(:0) to setup H2250 channel (outgoing, ooh323c_o_13) 19:08:39:219 ERROR:Failed to connect to remote destination for transmit H2250 channel(outgoing, ooh323c_o_13)19:08:39:219 ERROR:Failed to create H225 connection to :019:08:39:319 Cleaning Call (outgoing, ooh323c_o_13)- reason:OO_REASON_NOUSER 19:08:39:319 Closing H.245 connection (outgoing, ooh323c_o_13)19:08:39:319 Removed call (outgoing, ooh323c_o_13) from list my ooh323.conf is: [general] port=1720 bindaddr=0.0.0.0 gateway=no h323id=XXX e164=1234567890 callerid=h323id gatekeeper = ipgatekeeper context=voip-h323 disallow=all ;Note order of disallow/allow is important.allow=g729allow=gsmallow=ulawdtmfmode=rfc2833 What the problem?? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] help in create user group
Hello to everybody.I need of urgent help. I need to create a group in that this have access barely the connection between extensions. In hypothesis some those persons can do connection by pstn or voip. Someone has some hint of as I should proceed? Andre Luiz Martins ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
