[Asterisk 10.5] Cannot call out

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.

hi keitex,

I think you have not set port correctly, as it shows first mapped to “host sbc5.a.synergy.voip.co.uk, port 6060” it is trying to reach to 6060 port, but at the end it shows “Retransmission timeout reached on transmission 5017343974158ea2421bb1e67d4958b1@80.x.x.x:5060”

Regards,

Sohaib Khan