Hi there,
I have just installed Asterisk 10.5 on CentOS 6.2 x64 with Asterisk GUI. I thought I have got everything configured but I am having problems with calling in and calling out. First I would like to sort the calling out problem as it seems more complicating and I cannot figure it out.
Asterisk is installed on VM Machine behind an ASA firewall. I have ports 5060, 6060, 5061, 53, 10000 and 20000 open. Its also NAT’ed from 80.xx.xx.xx to 192.168.10.10
This is what I get when I try to place a call:
== Using SIP RTP CoS mark 5
-- Executing [901525XXXXXX@DLPN_avecsys:1] Macro("SIP/6000-00000009", "trunkdial-failover-0.3,SIP/voip-uk/01525XXXXXX,,voip-uk,") in new stack
-- Executing [s@macro-trunkdial-failover-0.3:1] GotoIf("SIP/6000-00000009", "0?1-fmsetcid,1") in new stack
-- Executing [s@macro-trunkdial-failover-0.3:2] GotoIf("SIP/6000-00000009", "0?1-setgbobname,1") in new stack
-- Executing [s@macro-trunkdial-failover-0.3:3] Set("SIP/6000-00000009", "CALLERID(num)=") in new stack
-- Executing [s@macro-trunkdial-failover-0.3:4] Set("SIP/6000-00000009", "CALLERID(all)=") in new stack
-- Executing [s@macro-trunkdial-failover-0.3:5] GotoIf("SIP/6000-00000009", "0?1-dial,1") in new stack
-- Executing [s@macro-trunkdial-failover-0.3:6] Set("SIP/6000-00000009", "CALLERID(all)=") in new stack
-- Executing [s@macro-trunkdial-failover-0.3:7] Set("SIP/6000-00000009", "CALLERID(all)=") in new stack
-- Executing [s@macro-trunkdial-failover-0.3:8] Goto("SIP/6000-00000009", "1-dial,1") in new stack
-- Goto (macro-trunkdial-failover-0.3,1-dial,1)
-- Executing [1-dial@macro-trunkdial-failover-0.3:1] Dial("SIP/6000-00000009", "SIP/voip-uk/01525XXXXXX") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/voip-uk/01525XXXXXX
> ast_get_srv: SRV lookup for '_sip._udp.proxy.voip.co.uk' mapped to host sbc5.a.synergy.voip.co.uk, port 6060
> ast_get_srv: SRV lookup for '_sip._udp.proxy.voip.co.uk' mapped to host sbc5.b.synergy.voip.co.uk, port 6060
== Spawn extension (macro-trunkdial-failover-0.3, 1-dial, 1) exited non-zero on 'SIP/6000-00000009' in macro 'trunkdial-failover-0.3'
== Spawn extension (DLPN_avecsys, 901525XXXXXX, 1) exited non-zero on 'SIP/6000-00000009'
[Jun 12 14:18:47] WARNING[2325]: chan_sip.c:3663 retrans_pkt: Retransmission timeout reached on transmission 5017343974158ea2421bb1e67d4958b1@80.x.x.x:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 31999ms with no response
Any ideas? I have got NAT settings configured in the GUI side too.