8/3/2023 0 Comments Session avaya tls versionsReliably Transmitting (no NAT) to 147.x.x.43:5060: Auto fallthrough, channel 'SIP/p3-0074db00' status is 'CONGESTION' = Everyone is busy/congested at this time (1:0/1/0) SIP/avaya.proxy-00751ec0 is circuit-busy This is cli output = Using SIP RTP CoS mark 5 When calling P1 from P3 I use it just like you have written. It may be not that clear, but I have successfully make the call, but I have no audio and I keep getting ICMP port unreachable for rtp packets… If it can help, I will post sip debug output as well… SIP Options: replaces replace 100rel timer join histinfoĪnd when running tshark I see ICMP packets with ‘port unreachable’ for 147.x.x.57(the media server) This is a result of ‘sip show channel’ for both channels: * SIP Call In my nf I have this line: exten => s,1,Dial(SIP/p3) set_address_from_contact host '147.x.x.50' I have set verbose output with ‘core set verbose 9’ and that’s what I got: = Using SIP RTP CoS mark 5 I would have suspected misconfiguration of avaya server, but because P2 can connect with P1 I guess there must be something wrong with asterisk configuration. P2 can call with P1, but P3 and P4 cannot.Īlso I can call Asterisk from P1 without problems. P2,P3 and P4 can make calls together easily. Rtp transport and the sound exchange is handled by media server. P1 is normal phone, P2-4 are VoIP phones.Īsterisk is listening on two interfaces - one is local and one is opened to the net. I have some problem running asterisk together with avaya sip server.
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