Can't make outbound calls but inbound works

I have my system all set up and everything was working fine. I am now having trouble making outbound calls. Although I can make inbound calls. I have been working with my VOIP trunk provide on this but we haven’t been able to figure it out. Can anyone please point me in the right direction with this log?

[2013-09-10 19:17:48] VERBOSE[3336][C-00000000] pbx.c: -- Executing [[email protected]:9] GotoIf("SIP/300-00000000", "0?skipoutcid") in new stack [2013-09-10 19:17:48] VERBOSE[3336][C-00000000] pbx.c: -- Executing [[email protected]:10] Set("SIP/300-00000000", "DIAL_TRUNK_OPTIONS=Tt") in new stack [2013-09-10 19:17:48] VERBOSE[3336][C-00000000] pbx.c: -- Executing [[email protected]:11] Macro("SIP/300-00000000", "outbound-callerid,3") in new stack [2013-09-10 19:17:48] VERBOSE[3336][C-00000000] pbx.c: -- Executing [[email protected]:1] ExecIf("SIP/300-00000000", "0?Set(CALLERPRES()=)") in new stack [2013-09-10 19:17:48] VERBOSE[3336][C-00000000] pbx.c: -- Executing [[email protected]:2] ExecIf("SIP/300-00000000", "0?Set(REALCALLERIDNUM=300)") in new stack [2013-09-10 19:17:48] VERBOSE[3336][C-00000000] pbx.c: -- Executing [[email protected]:3] GotoIf("SIP/300-00000000", "1?normcid") in new stack [2013-09-10 19:17:48] VERBOSE[3336][C-00000000] pbx.c: -- Goto (macro-outbound-callerid,s,6) [2013-09-10 19:17:48] VERBOSE[3336][C-00000000] pbx.c: -- Executing [[email protected]:6] Set("SIP/300-00000000", "USEROUTCID=xxxxxxxxxx") in new stack [2013-09-10 19:17:48] VERBOSE[3336][C-00000000] pbx.c: -- Executing [[email protected]:7] Set("SIP/300-00000000", "EMERGENCYCID=") in new stack [2013-09-10 19:17:48] VERBOSE[3336][C-00000000] pbx.c: -- Executing [[email protected]:8] Set("SIP/300-00000000", "TRUNKOUTCID=xxxxxxxxxx") in new stack [2013-09-10 19:17:48] VERBOSE[3336][C-00000000] pbx.c: -- Executing [[email protected]:9] GotoIf("SIP/300-00000000", "1?trunkcid") in new stack [2013-09-10 19:17:48] VERBOSE[3336][C-00000000] pbx.c: -- Goto (macro-outbound-callerid,s,14) [2013-09-10 19:17:48] VERBOSE[3336][C-00000000] pbx.c: -- Executing [[email protected]:14] ExecIf("SIP/300-00000000", "1?Set(CALLERID(all)=xxxxxxxxxx)") in new stack [2013-09-10 19:17:48] VERBOSE[3336][C-00000000] pbx.c: -- Executing [[email protected]:15] ExecIf("SIP/300-00000000", "1?Set(CALLERID(all)=xxxxxxxxxx)") in new stack [2013-09-10 19:17:48] VERBOSE[3337][C-00000000] app_mixmonitor.c: == Begin MixMonitor Recording SIP/300-00000000 [2013-09-10 19:17:48] VERBOSE[3336][C-00000000] pbx.c: -- Executing [[email protected]:16] ExecIf("SIP/300-00000000", "1?Set(CALLERID(all)=xxxxxxxxxx)") in new stack [2013-09-10 19:17:48] VERBOSE[3336][C-00000000] pbx.c: -- Executing [[email protected]:17] ExecIf("SIP/300-00000000", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack [2013-09-10 19:17:48] VERBOSE[3336][C-00000000] pbx.c: -- Executing [[email protected]:18] Set("SIP/300-00000000", "CDR(outbound_cnum)=xxxxxxxxxx") in new stack [2013-09-10 19:17:48] VERBOSE[3336][C-00000000] pbx.c: -- Executing [[email protected]:19] Set("SIP/300-00000000", "CDR(outbound_cnam)=") in new stack [2013-09-10 19:17:48] VERBOSE[3336][C-00000000] pbx.c: -- Executing [[email protected]:12] GosubIf("SIP/300-00000000", "0?sub-flp-3,s,1()") in new stack [2013-09-10 19:17:48] VERBOSE[3336][C-00000000] pbx.c: -- Executing [[email protected]:13] Set("SIP/300-00000000", "OUTNUM=1xxxxxxxxxx") in new stack [2013-09-10 19:17:48] VERBOSE[3336][C-00000000] pbx.c: -- Executing [[email protected]:14] Set("SIP/300-00000000", "custom=SIP/BPW-Incoming9009") in new stack [2013-09-10 19:17:48] VERBOSE[3336][C-00000000] pbx.c: -- Executing [[email protected]:15] ExecIf("SIP/300-00000000", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)Tt)") in new stack [2013-09-10 19:17:48] VERBOSE[3336][C-00000000] pbx.c: -- Executing [[email protected]:16] ExecIf("SIP/300-00000000", "0?Set(DIAL_TRUNK_OPTIONS=TtM(confirm))") in new stack [2013-09-10 19:17:48] VERBOSE[3336][C-00000000] pbx.c: -- Executing [[email protected]:17] Macro("SIP/300-00000000", "dialout-trunk-predial-hook,") in new stack [2013-09-10 19:17:48] VERBOSE[3336][C-00000000] pbx.c: -- Executing [[email protected]:1] MacroExit("SIP/300-00000000", "") in new stack [2013-09-10 19:17:48] VERBOSE[3336][C-00000000] pbx.c: -- Executing [[email protected]:18] GotoIf("SIP/300-00000000", "0?bypass,1") in new stack [2013-09-10 19:17:48] VERBOSE[3336][C-00000000] pbx.c: -- Executing [[email protected]:19] ExecIf("SIP/300-00000000", "1?Set(CONNECTEDLINE(num,i)=1xxxxxxxxxx)") in new stack [2013-09-10 19:17:48] VERBOSE[3336][C-00000000] pbx.c: -- Executing [[email protected]:20] ExecIf("SIP/300-00000000", "1?Set(CONNECTEDLINE(name,i)=CID:xxxxxxxxxx)") in new stack [2013-09-10 19:17:48] VERBOSE[3336][C-00000000] pbx.c: -- Executing [[email protected]:21] GotoIf("SIP/300-00000000", "0?customtrunk") in new stack [2013-09-10 19:17:48] VERBOSE[3336][C-00000000] pbx.c: -- Executing [[email protected]:22] Dial("SIP/300-00000000", "SIP/BPW-Incoming9009/1xxxxxxxxxx,300,Tt") in new stack [2013-09-10 19:17:48] VERBOSE[3336][C-00000000] netsock2.c: == Using SIP RTP TOS bits 184 [2013-09-10 19:17:48] VERBOSE[3336][C-00000000] netsock2.c: == Using SIP RTP CoS mark 5 [2013-09-10 19:17:48] VERBOSE[3336][C-00000000] app_dial.c: -- Called SIP/BPW-Incoming9009/1xxxxxxxxxx [2013-09-10 19:17:54] WARNING[1977] chan_sip.c: Retransmission timeout reached on transmission [email protected]:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6399ms with no response [2013-09-10 19:17:54] WARNING[1977] chan_sip.c: Hanging up call [email protected]:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [2013-09-10 19:17:54] VERBOSE[3336][C-00000000] app_dial.c: == Everyone is busy/congested at this time (1:0/0/1) [2013-09-10 19:17:54] VERBOSE[3336][C-00000000] pbx.c: -- Executing [[email protected]:23] NoOp("SIP/300-00000000", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 18") in new stack [2013-09-10 19:17:54] VERBOSE[3336][C-00000000] pbx.c: -- Executing [[email protected]:24] GotoIf("SIP/300-00000000", "0?continue,1:s-CHANUNAVAIL,1") in new stack [2013-09-10 19:17:54] VERBOSE[3336][C-00000000] pbx.c: -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1) [2013-09-10 19:17:54] VERBOSE[3336][C-00000000] pbx.c: -- Executing [[email protected]:1] Set("SIP/300-00000000", "RC=18") in new stack [2013-09-10 19:17:54] VERBOSE[3336][C-00000000] pbx.c: -- Executing [[email protected]:2] Goto("SIP/300-00000000", "18,1") in new stack [2013-09-10 19:17:54] VERBOSE[3336][C-00000000] pbx.c: -- Goto (macro-dialout-trunk,18,1) [2013-09-10 19:17:54] VERBOSE[3336][C-00000000] pbx.c: -- Executing [[email protected]:1] Goto("SIP/300-00000000", "s-NOANSWER,1") in new stack [2013-09-10 19:17:54] VERBOSE[3336][C-00000000] pbx.c: -- Goto (macro-dialout-trunk,s-NOANSWER,1) [2013-09-10 19:17:54] VERBOSE[3336][C-00000000] pbx.c: -- Executing [[email protected]:1] NoOp("SIP/300-00000000", "Dial failed due to trunk reporting NOANSWER - giving up") in new stack [2013-09-10 19:17:54] VERBOSE[3336][C-00000000] pbx.c: -- Executing [[email protected]:2] Progress("SIP/300-00000000", "") in new stack [2013-09-10 19:17:54] VERBOSE[3336][C-00000000] pbx.c: -- Executing [[email protected]:3] Playback("SIP/300-00000000", "number-not-answering,noanswer") in new stack [2013-09-10 19:17:54] VERBOSE[3336][C-00000000] file.c: -- Playing 'number-not-answering.gsm' (language 'en') [2013-09-10 19:17:55] VERBOSE[3336][C-00000000] app_macro.c: == Spawn extension (macro-dialout-trunk, s-NOANSWER, 3) exited non-zero on 'SIP/300-00000000' in macro 'dialout-trunk' [2013-09-10 19:17:55] VERBOSE[3336][C-00000000] pbx.c: == Spawn extension (from-internal, xxxxxxxxxx, 6) exited non-zero on 'SIP/300-00000000' [2013-09-10 19:17:55] VERBOSE[3336][C-00000000] pbx.c: -- Executing [[email protected]:1] Hangup("SIP/300-00000000", "") in new stack [2013-09-10 19:17:55] VERBOSE[3336][C-00000000] pbx.c: == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/300-00000000' [2013-09-10 19:17:55] VERBOSE[3337][C-00000000] app_mixmonitor.c: == MixMonitor close filestream (mixed) [2013-09-10 19:17:55] VERBOSE[3337][C-00000000] app_mixmonitor.c: == End MixMonitor Recording SIP/300-00000000

I had installed Fail2Ban (http://www.voip-info.org/wiki/view/Fail2Ban+(with+iptables)+And+Asterisk) earlier today. We tried adding the VOIP provider IPs to the ignore list and even completely erasing fail2ban and it still didn’t solve the problem. Is there anyway that could have caused my issue?

Thanks,
Brad

Looks like RTP is not allowed through your firewall. How is the system connected to the Internet?

[2013-09-10 19:17:54] WARNING[1977] chan_sip.c: Retransmission timeout reached on transmission
[email protected]:5060 for seqno 102 (Critical Request) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6399ms with no response

The Asterisk is run on a Linux system with CentOS. It’s run through a switch and then a router to the web. My ISP is Century Link. I have forwarded port 5060 and 10000-20000 among a few others. I actually have another Asterisk system currently running on the same LAN and connected to the internet the same way (we are building the new one to replace that one). The other system is working just fine. When I forwarded the ports for the new system I mimicked what the old system has forwarded.

Are the ports forwarded to the new system or the old system?

The ports are forwarded to both systems with separate entries.

Ex.
Port 5060 forwarded to 192.168.1.111
Port 5060 forwarded to 192.168.1.115

How are you forwarding the same port to two different systems?

Is it not possible to forward 5060 to two seperate internal IPs?

I’m no IT expert, I’m a small business owner learning as I go. Maybe this is my issue? I have actually forward several ports to both phone servers.

I have never seen a router that would allow you to forward the same port to two different IP addresses. What brand and model of router are you using?

Try forwardin the ports to the new system and see if it works.

I imagine this is my problem. I’ll try changing the port forwarding tomorrow and report back.

Thanks for the help!

I tried completely removing the old VOIP system and I removed all ports I had forwarded to it. It still didn’t work. I keep seeing this in the logfile when I call, if it helps

[2013-09-11 10:39:16] WARNING[14675][C-0000000d] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) [2013-09-11 10:39:16] VERBOSE[14675][C-0000000d] app_dial.c: == Everyone is busy/congested at this time (1:0/0/1)

And this comes in every 20 seconds when the system is idle.

[2013-09-11 10:39:20] NOTICE[2014] chan_sip.c: -- Registration for '[email protected]' timed out, trying again (Attempt #5) [2013-09-11 10:39:40] NOTICE[2014] chan_sip.c: -- Registration for '[email protected]' timed out, trying again (Attempt #6) [2013-09-11 10:40:00] NOTICE[2014] chan_sip.c: -- Registration for '[email protected]' timed out, trying again (Attempt #7) [2013-09-11 10:40:20] NOTICE[2014] chan_sip.c: -- Registration for '[email protected]' timed out, trying again (Attempt #8) [2013-09-11 10:40:40] NOTICE[2014] chan_sip.c: -- Registration for '[email protected]' timed out, trying again (Attempt #9) [2013-09-11 10:41:00] NOTICE[2014] chan_sip.c: -- Registration for '[email protected]' timed out, trying again (Attempt #10) [2013-09-11 10:41:20] NOTICE[2014] chan_sip.c: -- Registration for '[email protected]' timed out, trying again (Attempt #11) [2013-09-11 10:41:40] NOTICE[2014] chan_sip.c: -- Registration for '[email protected]' timed out, trying again (Attempt #12) [2013-09-11 10:42:00] NOTICE[2014] chan_sip.c: -- Registration for '[email protected]' timed out, trying again (Attempt #13) [2013-09-11 10:42:20] NOTICE[2014] chan_sip.c: -- Registration for '[email protected]' timed out, trying again (Attempt #14) [2013-09-11 10:42:40] NOTICE[2014] chan_sip.c: -- Registration for '[email protected]' timed out, trying again (Attempt #15)

One more thing.

When I re prioritize my outbound routes and make my google voice number primary it calls out like normal.

Could this be an issue on my VOIP provider’s end? Or maybe the way we have the trunks setup?

Quick update. I finally got it to work. looks like an amateur mistake.

I hadn’t added any dial patterns that would use the route in Connectivity>Outbound routes. Once I added those, it worked. I just assumed the default was “all”. I was wrong. Thanks for the help!