Phibee Network Operation Center
2009-Nov-21 07:15 UTC
[asterisk-users] Connect two Asterisk Server in IAX ?
Hi My first post get no answer :=<, i post new with new elements. I have two Asterisk server, running on Asterisk 1.6: SRV1 = 192.168.0.5 on Asterisk 1.6.1.4 SRV2 = 192.168.0.20 on Asterisk 1.6.1.8 I want create a link for exchange call. on Srv1: iax.conf: [general] bindport=4569 bindaddr=0.0.0.0 language=fr bandwidth=low jitterbuffer=no forcejitterbuffer=no autokill=yes calltokenoptional=192.168.0.20 [Srv2] type=peer host=192.168.0.20 qualify=yes trunk=no encryption=aes128 disallow=all allow=alaw allow=g729 context=Incoming peercontext=Incoming extension.conf: [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no priorityjumping=no [globals] CONSOLE=Console/dsp ; Console interface for demo [Incoming] exten => _X.,1,Playback(demo-thanks) exten => _X.,2,Hangup [Out] exten => _201X.,1,Dial(IAX2/Srv2/${EXTEN:3},90,r) exten => _201X.,2,Congestion =Srv1*CLI> iax2 show peers Name/Username Host Mask Port Status Srv2 192.168.0.20 (S) 255.255.255.255 4569 (E) OK (39 ms) 1 iax2 peers [1 online, 0 offline, 0 unmonitored] On Srv2 iax.conf [general] bindport=4569 bindaddr=0.0.0.0 language=fr bandwidth=low jitterbuffer=no forcejitterbuffer=no autokill=yes calltokenoptional=192.168.0.5 bandwidth=low [Srv1] type=peer host=192.168.0.5 qualify=yes trunk=no encryption=aes128 disallow=all allow=alaw allow=g729 context=Incoming peercontect=Incoming extensions.conf: [Incoming] exten => _X.,1,Playback(demo-thanks) exten => _X.,2,Hangup [Out] exten => _202X.,1,Dial(IAX2/Srv1/${EXTEN:3},90,r) exten => _202X.,2,Congestion ==trader-voip*CLI> iax2 show peers Name/Username Host Mask Port Status Srv1 192.168.0.5 (S) 255.255.255.255 4569 (E) OK (28 ms) 1 iax2 peers [1 online, 0 offline, 0 unmonitored] == All SIP Poste are connected and have in context in: Out Now, when i call from a post connected on Srv1, i have this error on Srv1: [Nov 21 08:09:44] WARNING[6407]: chan_iax2.c:9018 socket_process: Call rejected by 192.168.0.20: No authority found and on Srv2: [Nov 21 08:09:44] NOTICE[9089]: chan_iax2.c:9785 socket_process: Rejected connect attempt from 192.168.0.5, who was trying to reach '125 at Incoming' 125 are the number called (201125) Dialplan on Srv2 Srv2*CLI> dialplan show Incoming [ Context 'Incoming' created by 'pbx_config' ] '_X.' => 1. Playback(demo-thanks) [pbx_config] 2. Hangup() [pbx_config] -= 1 extension (2 priorities) in 1 context. =- Anyone can help me for know where is my error ? thanks Jerome
Sylvain MEYNELLY (NEWTEK)
2009-Nov-21 15:35 UTC
[asterisk-users] Connect two Asterisk Server in IAX ?
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Hi, maybe this link can be useful: http://www.voip-info.org/wiki/view/IAX+encryption In particular, in your configuration I can't see the authentication method, which must be md5, and a username to authenticate with, in either server. But have a further look at the article, maybe you'll be able to sort out the issue from that :) HTH //Al. -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Phibee Network Operation Center Sent: sabato 21 novembre 2009 8.16 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Connect two Asterisk Server in IAX ? Hi My first post get no answer :=<, i post new with new elements. I have two Asterisk server, running on Asterisk 1.6: SRV1 = 192.168.0.5 on Asterisk 1.6.1.4 SRV2 = 192.168.0.20 on Asterisk 1.6.1.8 I want create a link for exchange call. on Srv1: iax.conf: [general] bindport=4569 bindaddr=0.0.0.0 language=fr bandwidth=low jitterbuffer=no forcejitterbuffer=no autokill=yes calltokenoptional=192.168.0.20 [Srv2] type=peer host=192.168.0.20 qualify=yes trunk=no encryption=aes128 disallow=all allow=alaw allow=g729 context=Incoming peercontext=Incoming extension.conf: [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no priorityjumping=no [globals] CONSOLE=Console/dsp ; Console interface for demo [Incoming] exten => _X.,1,Playback(demo-thanks) exten => _X.,2,Hangup [Out] exten => _201X.,1,Dial(IAX2/Srv2/${EXTEN:3},90,r) exten => _201X.,2,Congestion =Srv1*CLI> iax2 show peers Name/Username Host Mask Port Status Srv2 192.168.0.20 (S) 255.255.255.255 4569 (E) OK (39 ms) 1 iax2 peers [1 online, 0 offline, 0 unmonitored] On Srv2 iax.conf [general] bindport=4569 bindaddr=0.0.0.0 language=fr bandwidth=low jitterbuffer=no forcejitterbuffer=no autokill=yes calltokenoptional=192.168.0.5 bandwidth=low [Srv1] type=peer host=192.168.0.5 qualify=yes trunk=no encryption=aes128 disallow=all allow=alaw allow=g729 context=Incoming peercontect=Incoming extensions.conf: [Incoming] exten => _X.,1,Playback(demo-thanks) exten => _X.,2,Hangup [Out] exten => _202X.,1,Dial(IAX2/Srv1/${EXTEN:3},90,r) exten => _202X.,2,Congestion ==trader-voip*CLI> iax2 show peers Name/Username Host Mask Port Status Srv1 192.168.0.5 (S) 255.255.255.255 4569 (E) OK (28 ms) 1 iax2 peers [1 online, 0 offline, 0 unmonitored] == All SIP Poste are connected and have in context in: Out Now, when i call from a post connected on Srv1, i have this error on Srv1: [Nov 21 08:09:44] WARNING[6407]: chan_iax2.c:9018 socket_process: Call rejected by 192.168.0.20: No authority found and on Srv2: [Nov 21 08:09:44] NOTICE[9089]: chan_iax2.c:9785 socket_process: Rejected connect attempt from 192.168.0.5, who was trying to reach '125 at Incoming' 125 are the number called (201125) Dialplan on Srv2 Srv2*CLI> dialplan show Incoming [ Context 'Incoming' created by 'pbx_config' ] '_X.' => 1. Playback(demo-thanks) [pbx_config] 2. Hangup() [pbx_config] -= 1 extension (2 priorities) in 1 context. =- Anyone can help me for know where is my error ? thanks Jerome _______________________________________________ -- 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
> I have two Asterisk server, running on Asterisk 1.6: > ? ?SRV1 = 192.168.0.5 ? ? on Asterisk 1.6.1.4 > ? ?SRV2 = 192.168.0.20 ? on Asterisk 1.6.1.8 > I want create a link for exchange call.To clarify and expand on Aggio's response. You either need to have a peer and user on both machines, or you can set it up as type=friend, which is the peer and user combined. - Noah