I can't make outgoing external calls

alfgarcas

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#1
Hi,

I have just registered my SIP trunk.
If I type "sip show registry" or "sip show peers", it's all well registered.
I can receive external calls, and can make and receive internal calls.
I followed the manual that was given by my provider, but I don't understand why can not make external outgoing calls.

That's the log:

== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [617848484@from-internal:1] Macro("SIP/100-0000000a", "user-callerid,SKIPTTL,") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/100-0000000a", "AMPUSER=100") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/100-0000000a", "0?report") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/100-0000000a", "1?Set(REALCALLERIDNUM=100)") in new stack
-- Executing [s@macro-user-callerid:4] Set("SIP/100-0000000a", "AMPUSER=100") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/100-0000000a", "AMPUSERCIDNAME=100") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/100-0000000a", "0?report") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/100-0000000a", "AMPUSERCID=100") in new stack
-- Executing [s@macro-user-callerid:8] Set("SIP/100-0000000a", "CALLERID(all)="100" <100>") in new stack
-- Executing [s@macro-user-callerid:9] ExecIf("SIP/100-0000000a", "1?Set(CHANNEL(language)=es)") in new stack
-- Executing [s@macro-user-callerid:10] GotoIf("SIP/100-0000000a", "1?continue") in new stack
-- Goto (macro-user-callerid,s,19)
-- Executing [s@macro-user-callerid:19] NoOp("SIP/100-0000000a", "Using CallerID "100" <100>") in new stack
-- Executing [617848484@from-internal:2] Set("SIP/100-0000000a", "_NODEST=") in new stack
-- Executing [617848484@from-internal:3] Macro("SIP/100-0000000a", "record-enable,100,OUT,") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/100-0000000a", "1?check") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] ExecIf("SIP/100-0000000a", "0?MacroExit()") in new stack
-- Executing [s@macro-record-enable:5] GotoIf("SIP/100-0000000a", "0?Group:OUT") in new stack
-- Goto (macro-record-enable,s,15)
-- Executing [s@macro-record-enable:15] GotoIf("SIP/100-0000000a", "0?IN") in new stack
-- Executing [s@macro-record-enable:16] ExecIf("SIP/100-0000000a", "1?MacroExit()") in new stack
-- Executing [617848484@from-internal:4] Macro("SIP/100-0000000a", "dialout-trunk,2,617848484,,") in new stack
-- Executing [s@macro-dialout-trunk:1] Set("SIP/100-0000000a", "DIAL_TRUNK=2") in new stack
-- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/100-0000000a", "0?sub-pincheck,s,1") in new stack
-- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/100-0000000a", "0?disabletrunk,1") in new stack
-- Executing [s@macro-dialout-trunk:4] Set("SIP/100-0000000a", "DIAL_NUMBER=617848484") in new stack
-- Executing [s@macro-dialout-trunk:5] Set("SIP/100-0000000a", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing [s@macro-dialout-trunk:6] Set("SIP/100-0000000a", "OUTBOUND_GROUP=OUT_2") in new stack
-- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/100-0000000a", "1?nomax") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/100-0000000a", "0?skipoutcid") in new stack
-- Executing [s@macro-dialout-trunk:10] Set("SIP/100-0000000a", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing [s@macro-dialout-trunk:11] Macro("SIP/100-0000000a", "outbound-callerid,2") in new stack
-- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/100-0000000a", "0?Set(CALLERPRES()=)") in new stack
-- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/100-0000000a", "0?Set(REALCALLERIDNUM=100)") in new stack
-- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/100-0000000a", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,6)
-- Executing [s@macro-outbound-callerid:6] Set("SIP/100-0000000a", "USEROUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:7] Set("SIP/100-0000000a", "EMERGENCYCID=") in new stack
-- Executing [s@macro-outbound-callerid:8] Set("SIP/100-0000000a", "TRUNKOUTCID=515647533") in new stack
-- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/100-0000000a", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,12)
-- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/100-0000000a", "1?Set(CALLERID(all)=515XXXXXX)") in new stack
-- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/100-0000000a", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/100-0000000a", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/100-0000000a", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
-- Executing [s@macro-dialout-trunk:12] ExecIf("SIP/100-0000000a", "1?AGI(fixlocalprefix)") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
> fixlocalprefix: Using pattern 9XXXXXXXX
> fixlocalprefix: Using pattern 6XXXXXXXX
== fixlocalprefix: Dialpattern 6XXXXXXXX matched. 617848484 -> 617848484
-- <SIP/100-0000000a>AGI Script fixlocalprefix completed, returning 0
-- Executing [s@macro-dialout-trunk:13] Set("SIP/100-0000000a", "OUTNUM=617848484") in new stack
-- Executing [s@macro-dialout-trunk:14] Set("SIP/100-0000000a", "custom=SIP/voztele") in new stack
-- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/100-0000000a", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^))") in new stack
-- Executing [s@macro-dialout-trunk:16] Macro("SIP/100-0000000a", "dialout-trunk-predial-hook,") in new stack
-- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/100-0000000a", "") in new stack
-- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/100-0000000a", "0?bypass,1") in new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/100-0000000a", "0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:19] Dial("SIP/100-0000000a", "SIP/voztele/617848484,300,") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [s@macro-dialout-trunk:20] NoOp("SIP/100-0000000a", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 20") in new stack
-- Executing [s@macro-dialout-trunk:21] Goto("SIP/100-0000000a", "s-CHANUNAVAIL,1") in new stack
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set("SIP/100-0000000a", "RC=20") in new stack
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto("SIP/100-0000000a", "20,1") in new stack
-- Goto (macro-dialout-trunk,20,1)
-- Executing [20@macro-dialout-trunk:1] Goto("SIP/100-0000000a", "continue,1") in new stack
-- Goto (macro-dialout-trunk,continue,1)
-- Executing [continue@macro-dialout-trunk:1] GotoIf("SIP/100-0000000a", "1?noreport") in new stack
-- Goto (macro-dialout-trunk,continue,3)
-- Executing [continue@macro-dialout-trunk:3] NoOp("SIP/100-0000000a", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 20 - failing through to other trunks") in new stack
-- Executing [continue@macro-dialout-trunk:4] Set("SIP/100-0000000a", "CALLERID(number)=100") in new stack
-- Executing [617848484@from-internal:5] Macro("SIP/100-0000000a", "outisbusy,") in new stack
-- Executing [s@macro-outisbusy:1] Progress("SIP/100-0000000a", "") in new stack
-- Executing [s@macro-outisbusy:2] GotoIf("SIP/100-0000000a", "0?emergency,1") in new stack
-- Executing [s@macro-outisbusy:3] GotoIf("SIP/100-0000000a", "0?intracompany,1") in new stack
-- Executing [s@macro-outisbusy:4] Playback("SIP/100-0000000a", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
-- <SIP/100-0000000a> Playing 'all-circuits-busy-now.gsm' (language 'es')
== Spawn extension (macro-outisbusy, s, 4) exited non-zero on 'SIP/100-0000000a' in macro 'outisbusy'
== Spawn extension (from-internal, 617848484, 5) exited non-zero on 'SIP/100-0000000a'
-- Executing [h@from-internal:1] Macro("SIP/100-0000000a", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/100-0000000a", "1?noautomon") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] NoOp("SIP/100-0000000a", "TOUCH_MONITOR_OUTPUT=") in new stack
-- Executing [s@macro-hangupcall:4] GotoIf("SIP/100-0000000a", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] GotoIf("SIP/100-0000000a", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,10)
-- Executing [s@macro-hangupcall:10] GotoIf("SIP/100-0000000a", "1?theend") in new stack
-- Goto (macro-hangupcall,s,12)
-- Executing [s@macro-hangupcall:12] Hangup("SIP/100-0000000a", "") in new stack
== Spawn extension (macro-hangupcall, s, 12) exited non-zero on 'SIP/100-0000000a' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/100-0000000a'


Any ideas??

Thanks.
 

jgutierrez

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#2
Yes, I might think, that you are not placing correctly the outbound callerID, or that your provider is expecting another format for outbound calls (for example a prefix).
By the, way is the trunk already registered?

What does your provider tells you about congestion message? What do they see on their end? What recommendations have they told you?
 

alfgarcas

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#3
Hi,

The trunk is registered, as I can receive external calls.
If I type "sip show registry" or "sip show peers", it's all well registered.

The outbound callerID it must be OK if I follow the provider manual.
I tried to change it, and added the international Spain code "34", but it didn´t work.

I will contact the provider to know what they see on their end.
 

alfgarcas

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#4
Ok,

My provider says that the calls don't get out of my LAN.
The don't see any outgoing calls.
 

Bob

Joined
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#5
Alfgarcas,

These are the two important lines in the log

Dial("SIP/100-0000000a", "SIP/voztele/617848484,300,")

"SIP/100-0000000a", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 20") in new stack

The first shows that it is dialing your VSP with the number 617848484. Can you confirm that this is a number that you would normally dial on a standard telephone and get through???

The next line that is important as tell us nothing except it failed.

At this point, we need to look at the SIP messages and responses....it will also give us an idea whether you are reaching their Network (by checking for SIP responses from their system)....and/or why your outbound call is being rejected.

You need to go into the CLI and issue the
sip set debug ip {VSP SERVER IP}
This will allow you to see just the traffic going to the ISP and back....Don't panic if it runs too fast, it is captured in the standard Asterisk logs....

this is your best bet...

Regards

Bob
 

alfgarcas

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#6
Ok, I've got it!

The problem was at the PEER DETAILS, at the configuration of the trunk.
I had a manual that seems wasn't working with Elastix 2.0

I found another post. That's the link:

http://www.elastix.org/en/component/kun ... html#63773

And that is the configuration needed to work properly:

PEER Details:

host=voztele.com
port=5060
username=3400XXXXXX
fromuser=3400XXXXXX
fromusername=3400XXXXXX
fromdomain=voztele.com
secret=
canreinvite=no
dtmfmode=rfc2833
type=peer
defaultexpirey=300
insecure=very
qualify=yes


USERS Details:

type=friend
host=193.22.119.20
username=3400XXXXXX
fromdomain=voztele.com
fromusername=3400XXXXXX


Thanks very much for your help and patient.
 

Bob

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#7
Looks like you found exactly what you needed.

And thanks for closing the thread letting others know your solution :cheer:

Regards

Bob
 

Egor

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#8
Hello all,
I have the same situation with an error: "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 20"
But it appears on trunks between two elastix servers (2.3.0 & 2.2.0) of our company.

I have tried to fix it with instruction bellow but it didn't helps.
Here is an output of of command sip set debug ip {VSP SERVER IP}

Retransmitting #4 (no NAT) to x.x.x.x:5060:
OPTIONS sip:name_of_server SIP/2.0
Via: SIP/2.0/UDP y.y.y.y:5060;branch=z9hG4bK0ce47873
Max-Forwards: 70
From: "Unknown" <sip:Unknown@y.y.y.y>;tag=as3a783c30
To: <sip:name_of_server>
Contact: <sip:Unknown@y.y.y.y:5060>
Call-ID: 1c5141e1779a09ca1e1e0bb73ac41b27@y.y.y.y:5060
CSeq: 102 OPTIONS
User-Agent: pbx-city
Date: Mon, 21 Jan 2013 11:45:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

Thanks in advise
 

jordan.turner1974

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#9
Did you verify with ITSP the details of their settings matches yours? This is the #1 cause of these issues.

Once you verify it, delete it, and recreate it if you have to - to ensure you are indeed using correct spacings, etc - little details will throw it off.
 

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