Cannot make outbound calls from public IP, but can from private IP

General Help
1

Hello All!

I am pretty new to the whole PBX Asterisk thing and my apologies, if my questions seems dumb.

My problem is this. After installing the newest version with CentOS 6.5 from Downloads and configure my SIP and extensions, I can make outbound calls. But when I put the machine with Public IP, I cannot make any outbound calls, except to other extensions.

After digging the logs for hours, the only thing I am constantly seeing are those errors.

[2014-04-25 13:22:46] VERBOSE[14969][C-00000001] netsock2.c: == Using SIP RTP TOS bits 184
[2014-04-25 13:22:46] VERBOSE[14969][C-00000001] netsock2.c: == Using SIP RTP CoS mark 5
[2014-04-25 13:22:46] VERBOSE[14969][C-00000001] app_dial.c: – Called SIP/ITDxxxx/0876xxxxxx
[2014-04-25 13:22:46] NOTICE[14766][C-00000001] chan_sip.c: Failed to authenticate on INVITE to ‘sip:[email protected];tag=as6daa5a86’
[2014-04-25 13:22:46] VERBOSE[14969][C-00000001] app_dial.c: – SIP/ITD8378-00000003 is circuit-busy
[2014-04-25 13:22:46] VERBOSE[14969][C-00000001] app_dial.c: == Everyone is busy/congested at this time (1:0/1/0)
[2014-04-25 13:22:46] VERBOSE[14969][C-00000001] pbx.c: – Executing [[email protected]:23] NoOp(“SIP/9112-00000002”, “Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 21”) in new stack
[2014-04-25 13:22:46] VERBOSE[14969][C-00000001] pbx.c: – Executing [[email protected]:24] GotoIf(“SIP/9112-00000002”, “0?continue,1:s-CONGESTION,1”) in new stack
[2014-04-25 13:22:46] VERBOSE[14969][C-00000001] pbx.c: – Goto (macro-dialout-trunk,s-CONGESTION,1)
[2014-04-25 13:22:46] VERBOSE[14969][C-00000001] pbx.c: – Executing [[email protected]:1] Set(“SIP/9112-00000002”, “RC=21”) in new stack
[2014-04-25 13:22:46] VERBOSE[14969][C-00000001] pbx.c: – Executing [[email protected]:2] Goto(“SIP/9112-00000002”, “21,1”) in new stack
[2014-04-25 13:22:46] VERBOSE[14969][C-00000001] pbx.c: – Goto (macro-dialout-trunk,21,1)
[2014-04-25 13:22:46] VERBOSE[14969][C-00000001] pbx.c: – Executing [[email protected]:1] Goto(“SIP/9112-00000002”, “continue,1”) in new stack
[2014-04-25 13:22:46] VERBOSE[14969][C-00000001] pbx.c: – Goto (macro-dialout-trunk,continue,1)
[2014-04-25 13:22:46] VERBOSE[14969][C-00000001] pbx.c: – Executing [[email protected]:1] NoOp(“SIP/9112-00000002”, “TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 21 - failing through to other trunks”) in new stack
[2014-04-25 13:22:46] VERBOSE[14969][C-00000001] pbx.c: – Executing [[email protected]:2] Set(“SIP/9112-00000002”, “CALLERID(number)=9112”) in new stack
[2014-04-25 13:22:46] VERBOSE[14969][C-00000001] pbx.c: – Executing [[email protected]:7] Macro(“SIP/9112-00000002”, “outisbusy,”) in new stack
[2014-04-25 13:22:46] VERBOSE[14969][C-00000001] pbx.c: – Executing [[email protected]:1] Progress(“SIP/9112-00000002”, “”) in new stack
[2014-04-25 13:22:46] VERBOSE[14969][C-00000001] pbx.c: – Executing [[email protected]:2] GotoIf(“SIP/9112-00000002”, “0?emergency,1”) in new stack
[2014-04-25 13:22:46] VERBOSE[14969][C-00000001] pbx.c: – Executing [[email protected]:3] GotoIf(“SIP/9112-00000002”, “0?intracompany,1”) in new stack
[2014-04-25 13:22:46] VERBOSE[14969][C-00000001] pbx.c: – Executing [[email protected]:4] Playback(“SIP/9112-00000002”, “all-circuits-busy-now&pls-try-call-later, noanswer”) in new stack
[2014-04-25 13:22:46] VERBOSE[14969][C-00000001] file.c: – <SIP/9112-00000002> Playing ‘all-circuits-busy-now.gsm’ (language ‘en’)
[2014-04-25 13:22:47] VERBOSE[14969][C-00000001] file.c: – <SIP/9112-00000002> Playing ‘pls-try-call-later.gsm’ (language ‘en’)

If someone needs more specific logs, I can provide them.
Thank you in advance!

Regards

2

Can you post your Asterisk SIP Settings? Specifically the bit at the top about IP addresses, etc?

3

Hello,

If you are talking about the NAT Settings section in the Sip Settings, I’ve tried with static IP and manually setting my public IP and private network and with and without NAT. I’ve tried selecting only the public IP button, but no luck.

What do you want me to look for?

Thanks in advance!

4

So how did you get the error message above. What where the Asterisk SIP settings at that point. It’s difficult to diagnose a moving target.

5

Hello Lee,

I meant that I’ve tried with every setting mentioned in my previous post, because I really don’t know what is going on.

For example, when I put the machine in the private network, I can make outgoing calls, but when I configure it with the public IP, I cannot make any calls.

If you need specific info, I can provide it.

Regards

6

If you can provide your Asterisk SIP settings it would be useful.

7

There you go. Those are the settings.

Global Settings:

UDP Bindaddress: 0.0.0.0:5060
TCP SIP Bindaddress: Disabled
TLS SIP Bindaddress: Disabled
Videosupport: No
Textsupport: No
Ignore SDP sess. ver.: No
AutoCreate Peer: Off
Match Auth Username: No
Allow unknown access: Yes
Allow subscriptions: Yes
Allow overlap dialing: Yes
Allow promisc. redir: No
Enable call counters: No
SIP domain support: No
Realm. auth: No
Our auth realm asterisk
Use domains as realms: No
Call to non-local dom.: Yes
URI user is phone no: No
Always auth rejects: Yes
Direct RTP setup: No
User Agent: FPBX-2.11.0(11.7.0)
SDP Session Name: Asterisk PBX 11.7.0
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Trust RPID: No
Send RPID: No
Legacy userfield parse: No
Send Diversion: Yes
Caller ID: Unknown
From: Domain: sip.itdnet.net
Record SIP history: Off
Call Events: Off
Auth. Failure Events: Off
T.38 support: No
T.38 EC mode: Unknown
T.38 MaxDtgrm: -1
SIP realtime: Disabled
Qualify Freq : 60000 ms
Q.850 Reason header: No
Store SIP_CAUSE: No

Network QoS Settings:

IP ToS SIP: CS3
IP ToS RTP audio: EF
IP ToS RTP video: AF41
IP ToS RTP text: CS0
802.1p CoS SIP: 4
802.1p CoS RTP audio: 5
802.1p CoS RTP video: 6
802.1p CoS RTP text: 5
Jitterbuffer enabled: No

Network Settings:

SIP address remapping: Enabled using externaddr
Externhost:
Externaddr: 94.xx.xx.156:0
Externrefresh: 10
Localnet: 10.10.0.0/255.255.0.0

Global Signalling Settings:

Codecs: (gsm|ulaw|alaw)
Codec Order: ulaw:20,alaw:20,gsm:20
Relax DTMF: No
RFC2833 Compensation: No
Symmetric RTP: Yes
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 30
RTP Hold Timeout: 300
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: No
Pedantic SIP support: Yes
Reg. min duration 60 secs
Reg. max duration: 3600 secs
Reg. default duration: 120 secs
Sub. min duration 60 secs
Sub. max duration: 3600 secs
Outbound reg. timeout: 20 secs
Outbound reg. attempts: 0
Outbound reg. retry 403:0
Notify ringing state: Yes
Include CID: No
Notify hold state: Yes
SIP Transfer mode: open
Max Call Bitrate: 384 kbps
Auto-Framing: No
Outb. proxy:
Session Timers: Accept
Session Refresher: uas
Session Expires: 1800 secs
Session Min-SE: 90 secs
Timer T1: 500
Timer T1 minimum: 100
Timer B: 32000
No premature media: Yes
Max forwards: 70

Default Settings:

Allowed transports: UDP
Outbound transport: UDP
Context: from-sip-external
Record on feature: automon
Record off feature: automon
Force rport: Yes
DTMF: rfc2833
Qualify: 0
Keepalive: 0
Use ClientCode: No
Progress inband: Never
Language:
Tone zone:
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: *97

Hope it helps

Thank you in advance!

8

One more thing. When I try to make an outbound call, those messages appear in the logs.

[2014-04-28 18:49:28] NOTICE[14766][C-00000573]: chan_sip.c:22978 handle_response_invite: Failed to authenticate on INVITE to ‘sip:[email protected];tag=as23769ed3’

9

Here is one more log. Hope it helps.

== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Executing [[email protected]:1] Macro(“SIP/913-00000651”, “user-callerid,LIMIT,EXTERNAL,”) in new stack
– Executing [[email protected]:1] Set(“SIP/913-00000651”, “TOUCH_MONITOR=1398700493.1657”) in new stack
– Executing [[email protected]:2] Set(“SIP/913-00000651”, “AMPUSER=913”) in new stack
– Executing [[email protected]:3] GotoIf(“SIP/913-00000651”, “0?report”) in new stack
– Executing [[email protected]:4] ExecIf(“SIP/913-00000651”, “1?Set(REALCALLERIDNUM=913)”) in new stack
– Executing [[email protected]:5] Set(“SIP/913-00000651”, “AMPUSER=913”) in new stack
– Executing [[email protected]:6] GotoIf(“SIP/913-00000651”, “0?limit”) in new stack
– Executing [[email protected]:7] Set(“SIP/913-00000651”, “AMPUSERCIDNAME=Sergey3”) in new stack
– Executing [[email protected]:8] GotoIf(“SIP/913-00000651”, “0?report”) in new stack
– Executing [[email protected]:9] Set(“SIP/913-00000651”, “AMPUSERCID=913”) in new stack
– Executing [[email protected]:10] Set(“SIP/913-00000651”, “__DIAL_OPTIONS=Ttr”) in new stack
– Executing [[email protected]:11] Set(“SIP/913-00000651”, “CALLERID(all)=“Sergey3” <913>”) in new stack
– Executing [[email protected]:12] GotoIf(“SIP/913-00000651”, “0?limit”) in new stack
– Executing [[email protected]:13] ExecIf(“SIP/913-00000651”, “1?Set(GROUP(concurrency_limit)=913)”) in new stack
– Executing [[email protected]:14] GosubIf(“SIP/913-00000651”, “7?sub-ccss,s,1(from-internal,0876xxx868)”) in new stack
– Executing [[email protected]:1] ExecIf(“SIP/913-00000651”, “0?Return()”) in new stack
– Executing [[email protected]:2] Set(“SIP/913-00000651”, “CCSS_SETUP=TRUE”) in new stack
– Executing [[email protected]:3] GosubIf(“SIP/913-00000651”, “0?monitor_config,1(from-internal,0876xxx868):monitor_default,1(from-internal,0876xxx868)”) in new stack
– Executing [[email protected]:1] GotoIf(“SIP/913-00000651”, “0?is_exten”) in new stack
– Executing [[email protected]:2] StackPop(“SIP/913-00000651”, “”) in new stack
– Executing [[email protected]:3] Return(“SIP/913-00000651”, “FALSE”) in new stack
– Executing [[email protected]:15] GotoIf(“SIP/913-00000651”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,28)
– Executing [[email protected]:28] Set(“SIP/913-00000651”, “CALLERID(number)=913”) in new stack
– Executing [[email protected]:29] Set(“SIP/913-00000651”, “CALLERID(name)=Sergey3”) in new stack
– Executing [[email protected]:30] Set(“SIP/913-00000651”, “CDR(cnum)=913”) in new stack
– Executing [[email protected]:31] Set(“SIP/913-00000651”, “CDR(cnam)=Sergey3”) in new stack
– Executing [[email protected]:32] Set(“SIP/913-00000651”, “CHANNEL(language)=en”) in new stack
– Executing [[email protected]:2] Gosub(“SIP/913-00000651”, “sub-record-cancel,s,1()”) in new stack
– Executing [[email protected]:1] Set(“SIP/913-00000651”, “__REC_POLICY_MODE=”) in new stack
– Executing [[email protected]:2] ExecIf(“SIP/913-00000651”, “1?Return()”) in new stack
– Executing [[email protected]:3] Set(“SIP/913-00000651”, “__REC_POLICY_MODE=never”) in new stack
– Executing [[email protected]:4] Set(“SIP/913-00000651”, “ROUTE_CIDSAVE=“Sergey3” <913>”) in new stack
– Executing [[email protected]:5] Set(“SIP/913-00000651”, “MOHCLASS=default”) in new stack
– Executing [[email protected]:6] Set(“SIP/913-00000651”, “_NODEST=”) in new stack
– Executing [[email protected]:7] Gosub(“SIP/913-00000651”, “sub-record-check,s,1(out,0876xxx868,)”) in new stack
– Executing [[email protected]:1] Set(“SIP/913-00000651”, “REC_POLICY_MODE_SAVE=never”) in new stack
– Executing [[email protected]:2] GotoIf(“SIP/913-00000651”, “1?check”) in new stack
– Goto (sub-record-check,s,7)
– Executing [[email protected]:7] Set(“SIP/913-00000651”, “__MON_FMT=wav”) in new stack
– Executing [[email protected]:8] GotoIf(“SIP/913-00000651”, “1?next”) in new stack
– Goto (sub-record-check,s,11)
– Executing [[email protected]:11] ExecIf(“SIP/913-00000651”, “0?Return()”) in new stack
– Executing [[email protected]:12] ExecIf(“SIP/913-00000651”, “0?Set(__REC_POLICY_MODE=)”) in new stack
– Executing [[email protected]:13] GotoIf(“SIP/913-00000651”, “0?out,1”) in new stack
– Executing [[email protected]:14] Set(“SIP/913-00000651”, “__REC_STATUS=INITIALIZED”) in new stack
– Executing [[email protected]:15] Set(“SIP/913-00000651”, “NOW=1398700493”) in new stack
– Executing [[email protected]:16] Set(“SIP/913-00000651”, “__DAY=28”) in new stack
– Executing [[email protected]:17] Set(“SIP/913-00000651”, “__MONTH=04”) in new stack
– Executing [[email protected]:18] Set(“SIP/913-00000651”, “__YEAR=2014”) in new stack
– Executing [[email protected]:19] Set(“SIP/913-00000651”, “__TIMESTR=20140428-185453”) in new stack
– Executing [[email protected]:20] Set(“SIP/913-00000651”, “__FROMEXTEN=913”) in new stack
– Executing [[email protected]:21] Set(“SIP/913-00000651”, “__CALLFILENAME=out-0876xxx868-913-20140428-185453-1398700493.1657”) in new stack
– Executing [[email protected]:22] Goto(“SIP/913-00000651”, “out,1”) in new stack
– Goto (sub-record-check,out,1)
– Executing [[email protected]:1] ExecIf(“SIP/913-00000651”, “0?Set(__REC_POLICY_MODE=dontcare)”) in new stack
– Executing [[email protected]:2] GosubIf(“SIP/913-00000651”, “0?record,1(exten,0876xxx868,913)”) in new stack
– Executing [[email protected]:3] Return(“SIP/913-00000651”, “”) in new stack
– Executing [[email protected]:8] Macro(“SIP/913-00000651”, “dialout-trunk,2,0876xxx868,off”) in new stack
– Executing [[email protected]:1] Set(“SIP/913-00000651”, “DIAL_TRUNK=2”) in new stack
– Executing [[email protected]:2] GosubIf(“SIP/913-00000651”, “0?sub-pincheck,s,1()”) in new stack
– Executing [[email protected]:3] GotoIf(“SIP/913-00000651”, “0?disabletrunk,1”) in new stack
– Executing [[email protected]:4] Set(“SIP/913-00000651”, “DIAL_NUMBER=0876xxx868”) in new stack
– Executing [[email protected]:5] Set(“SIP/913-00000651”, “DIAL_TRUNK_OPTIONS=Ttr”) in new stack
– Executing [[email protected]:6] Set(“SIP/913-00000651”, “OUTBOUND_GROUP=OUT_2”) in new stack
– Executing [[email protected]:7] GotoIf(“SIP/913-00000651”, “1?nomax”) in new stack
– Goto (macro-dialout-trunk,s,9)
– Executing [[email protected]:9] GotoIf(“SIP/913-00000651”, “0?skipoutcid”) in new stack
– Executing [[email protected]:10] Set(“SIP/913-00000651”, “DIAL_TRUNK_OPTIONS=Tt”) in new stack
– Executing [[email protected]:11] Macro(“SIP/913-00000651”, “outbound-callerid,2”) in new stack
– Executing [[email protected]:1] ExecIf(“SIP/913-00000651”, “0?Set(CALLERPRES()=)”) in new stack
– Executing [[email protected]:2] ExecIf(“SIP/913-00000651”, “0?Set(REALCALLERIDNUM=913)”) in new stack
– Executing [[email protected]:3] GotoIf(“SIP/913-00000651”, “1?normcid”) in new stack
– Goto (macro-outbound-callerid,s,6)
– Executing [[email protected]:6] Set(“SIP/913-00000651”, “USEROUTCID=”) in new stack
– Executing [[email protected]:7] Set(“SIP/913-00000651”, “EMERGENCYCID=”) in new stack
– Executing [[email protected]:8] Set(“SIP/913-00000651”, “TRUNKOUTCID=3592439xxxx”) in new stack
– Executing [[email protected]:9] GotoIf(“SIP/913-00000651”, “1?trunkcid”) in new stack
– Goto (macro-outbound-callerid,s,14)
– Executing [[email protected]:14] ExecIf(“SIP/913-00000651”, “1?Set(CALLERID(all)=3592439xxxx)”) in new stack
– Executing [[email protected]:15] ExecIf(“SIP/913-00000651”, “0?Set(CALLERID(all)=)”) in new stack
– Executing [[email protected]:16] ExecIf(“SIP/913-00000651”, “1?Set(CALLERID(all)=3592439xxxx)”) in new stack
– Executing [[email protected]:17] ExecIf(“SIP/913-00000651”, “0?Set(CALLERPRES()=prohib_passed_screen)”) in new stack
– Executing [[email protected]:18] Set(“SIP/913-00000651”, “CDR(outbound_cnum)=3592439xxxx”) in new stack
– Executing [[email protected]:19] Set(“SIP/913-00000651”, “CDR(outbound_cnam)=”) in new stack
– Executing [[email protected]:12] GosubIf(“SIP/913-00000651”, “0?sub-flp-2,s,1()”) in new stack
– Executing [[email protected]:13] Set(“SIP/913-00000651”, “OUTNUM=0876xxx868”) in new stack
– Executing [[email protected]:14] Set(“SIP/913-00000651”, “custom=SIP/ITDSIP9”) in new stack
– Executing [[email protected]:15] ExecIf(“SIP/913-00000651”, “0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)Tt)”) in new stack
– Executing [[email protected]:16] ExecIf(“SIP/913-00000651”, “0?Set(DIAL_TRUNK_OPTIONS=TtM(confirm))”) in new stack
– Executing [[email protected]:17] Macro(“SIP/913-00000651”, “dialout-trunk-predial-hook,”) in new stack
– Executing [[email protected]:1] MacroExit(“SIP/913-00000651”, “”) in new stack
– Executing [[email protected]:18] GotoIf(“SIP/913-00000651”, “0?bypass,1”) in new stack
– Executing [[email protected]:19] ExecIf(“SIP/913-00000651”, “1?Set(CONNECTEDLINE(num,i)=0876xxx868)”) in new stack
– Executing [[email protected]:20] ExecIf(“SIP/913-00000651”, “1?Set(CONNECTEDLINE(name,i)=CID:3592439xxxx)”) in new stack
– Executing [[email protected]:21] GotoIf(“SIP/913-00000651”, “0?customtrunk”) in new stack
– Executing [[email protected]:22] Dial(“SIP/913-00000651”, “SIP/ITDSIP9/0876xxx868,300,Tt”) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Called SIP/ITDSIP9/0876xxx868
[2014-04-28 18:54:53] NOTICE[14766][C-00000578]: chan_sip.c:22978 handle_response_invite: Failed to authenticate on INVITE to ‘sip:[email protected];tag=as43ff1bec’
– SIP/ITDSIP9-00000652 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
– Executing [[email protected]:23] NoOp(“SIP/913-00000651”, “Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 21”) in new stack
– Executing [[email protected]:24] GotoIf(“SIP/913-00000651”, “0?continue,1:s-CONGESTION,1”) in new stack
– Goto (macro-dialout-trunk,s-CONGESTION,1)
– Executing [[email protected]:1] Set(“SIP/913-00000651”, “RC=21”) in new stack
– Executing [[email protected]:2] Goto(“SIP/913-00000651”, “21,1”) in new stack
– Goto (macro-dialout-trunk,21,1)
– Executing [[email protected]:1] Goto(“SIP/913-00000651”, “continue,1”) in new stack
– Goto (macro-dialout-trunk,continue,1)
– Executing [[email protected]:1] NoOp(“SIP/913-00000651”, “TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 21 - failing through to other trunks”) in new stack
– Executing [[email protected]:2] Set(“SIP/913-00000651”, “CALLERID(number)=913”) in new stack
– Executing [[email protected]:9] Set(“SIP/913-00000651”, “CALLERID(all)=“Sergey3” <913>”) in new stack
– Executing [[email protected]:10] Set(“SIP/913-00000651”, “_KEEPCID=TRUE”) in new stack
– Executing [[email protected]:11] Goto(“SIP/913-00000651”, “app-announcement-1,s,1”) in new stack
– Goto (app-announcement-1,s,1)
– Executing [[email protected]:1] GotoIf(“SIP/913-00000651”, “0?begin”) in new stack
– Executing [[email protected]:2] Answer(“SIP/913-00000651”, “”) in new stack
> 0x7f6d8406b270 – Probation passed - setting RTP source address to 10.10.11.1:51676
– Executing [[email protected]:3] Wait(“SIP/913-00000651”, “1”) in new stack
– Executing [[email protected]:4] NoOp(“SIP/913-00000651”, “Playing announcement Outbound Error”) in new stack
– Executing [[email protected]:5] Playback(“SIP/913-00000651”, “custom/OutBoundGreshka,noanswer”) in new stack
– <SIP/913-00000651> Playing ‘custom/OutBoundGreshka.slin’ (language ‘en’)
== Spawn extension (app-announcement-1, s, 5) exited non-zero on ‘SIP/913-00000651’

10