SIP Trunk

tsku

Joined
Oct 1, 2010
Messages
62
Likes
0
Points
0
#1
Hi all,

I'm trying to configure Elastix 2.0 to have a SIP trunk for call termination to VoIP Clearing House but no luck to make it work. When try to pass call to clearing house, it is always show circuit busy. However, IAX trunk is working fine for me.

I had used SJPHONE to try the SIP account and it work fine for me, mean the login information and remote server are working fine.

"sip show peers" show as below, looks like my Elastix is not able to connect to remote server. Any idea to make it work? Thanks in advance.


Name/username Host Dyn Nat ACL Port Status
2231120/2231120 202.211.228.41 5060 Unmonitored
 

Lee Sharp

Joined
Sep 28, 2010
Messages
332
Likes
0
Points
0
#2
We can just guess at stuff...

Or you can type "asterisk -r" into a shell and give us the output when you fail to make a call.
 

tsku

Joined
Oct 1, 2010
Messages
62
Likes
0
Points
0
#3
Hi Lee,

verbosity is at least 3
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [990162166917@from-internal:1] Macro("SIP/1231805-0000028a", "user-callerid,SKIPTTL,") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/1231805-0000028a", "AMPUSER=1231805") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/1231805-0000028a", "0?report") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/1231805-0000028a", "1?Set(REALCALLERIDNUM=1231805)") in new stack
-- Executing [s@macro-user-callerid:4] Set("SIP/1231805-0000028a", "AMPUSER=1231805") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/1231805-0000028a", "AMPUSERCIDNAME=508") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/1231805-0000028a", "0?report") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/1231805-0000028a", "AMPUSERCID=1231805") in new stack
-- Executing [s@macro-user-callerid:8] Set("SIP/1231805-0000028a", "CALLERID(all)="508" <1231805>") in new stack
-- Executing [s@macro-user-callerid:9] ExecIf("SIP/1231805-0000028a", "0?Set(CHANNEL(language)=)") in new stack
-- Executing [s@macro-user-callerid:10] GotoIf("SIP/1231805-0000028a", "1?continue") in new stack
-- Goto (macro-user-callerid,s,19)
-- Executing [s@macro-user-callerid:19] NoOp("SIP/1231805-0000028a", "Using CallerID "508" <1231805>") in new stack
-- Executing [990162166917@from-internal:2] Set("SIP/1231805-0000028a", "_NODEST=") in new stack
-- Executing [990162166917@from-internal:3] Macro("SIP/1231805-0000028a", "record-enable,1231805,OUT,") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/1231805-0000028a", "1?check") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] ExecIf("SIP/1231805-0000028a", "0?MacroExit()") in new stack
-- Executing [s@macro-record-enable:5] GotoIf("SIP/1231805-0000028a", "0?Group:OUT") in new stack
-- Goto (macro-record-enable,s,15)
-- Executing [s@macro-record-enable:15] GotoIf("SIP/1231805-0000028a", "0?IN") in new stack
-- Executing [s@macro-record-enable:16] ExecIf("SIP/1231805-0000028a", "1?MacroExit()") in new stack
-- Executing [990162166917@from-internal:4] Macro("SIP/1231805-0000028a", "dialout-trunk,4,0162166917,,") in new stack
-- Executing [s@macro-dialout-trunk:1] Set("SIP/1231805-0000028a", "DIAL_TRUNK=4") in new stack
-- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/1231805-0000028a", "0?sub-pincheck,s,1") in new stack
-- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/1231805-0000028a", "0?disabletrunk,1") in new stack
-- Executing [s@macro-dialout-trunk:4] Set("SIP/1231805-0000028a", "DIAL_NUMBER=0162166917") in new stack
-- Executing [s@macro-dialout-trunk:5] Set("SIP/1231805-0000028a", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing [s@macro-dialout-trunk:6] Set("SIP/1231805-0000028a", "OUTBOUND_GROUP=OUT_4") in new stack
-- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/1231805-0000028a", "1?nomax") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/1231805-0000028a", "0?skipoutcid") in new stack
-- Executing [s@macro-dialout-trunk:10] Set("SIP/1231805-0000028a", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing [s@macro-dialout-trunk:11] Macro("SIP/1231805-0000028a", "outbound-callerid,4") in new stack
-- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/1231805-0000028a", "0?Set(CALLERPRES()=)") in new stack
-- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/1231805-0000028a", "0?Set(REALCALLERIDNUM=1231805)") in new stack
-- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/1231805-0000028a", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,6)
-- Executing [s@macro-outbound-callerid:6] Set("SIP/1231805-0000028a", "USEROUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:7] Set("SIP/1231805-0000028a", "EMERGENCYCID=") in new stack
-- Executing [s@macro-outbound-callerid:8] Set("SIP/1231805-0000028a", "TRUNKOUTCID=oce-sip-outgoing-redtone") in new stack
-- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/1231805-0000028a", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,12)
-- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/1231805-0000028a", "1?Set(CALLERID(all)=oce-sip-outgoing-redtone)") in new stack
-- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/1231805-0000028a", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/1231805-0000028a", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/1231805-0000028a", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
-- Executing [s@macro-dialout-trunk:12] ExecIf("SIP/1231805-0000028a", "0?AGI(fixlocalprefix)") in new stack
-- Executing [s@macro-dialout-trunk:13] Set("SIP/1231805-0000028a", "OUTNUM=0162166917") in new stack
-- Executing [s@macro-dialout-trunk:14] Set("SIP/1231805-0000028a", "custom=SIP/2231120") in new stack
-- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/1231805-0000028a", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^))") in new stack
-- Executing [s@macro-dialout-trunk:16] Macro("SIP/1231805-0000028a", "dialout-trunk-predial-hook,") in new stack
-- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/1231805-0000028a", "") in new stack
-- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/1231805-0000028a", "0?bypass,1") in new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/1231805-0000028a", "0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:19] Dial("SIP/1231805-0000028a", "SIP/2231120/0162166917,300,") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called 2231120/0162166917
-- SIP/2231120-0000028b is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [s@macro-dialout-trunk:20] NoOp("SIP/1231805-0000028a", "Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 21") in new stack
-- Executing [s@macro-dialout-trunk:21] Goto("SIP/1231805-0000028a", "s-CONGESTION,1") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,1)
-- Executing [s-CONGESTION@macro-dialout-trunk:1] Set("SIP/1231805-0000028a", "RC=21") in new stack
-- Executing [s-CONGESTION@macro-dialout-trunk:2] Goto("SIP/1231805-0000028a", "21,1") in new stack
-- Goto (macro-dialout-trunk,21,1)
-- Executing [21@macro-dialout-trunk:1] Goto("SIP/1231805-0000028a", "continue,1") in new stack
-- Goto (macro-dialout-trunk,continue,1)
-- Executing [continue@macro-dialout-trunk:1] GotoIf("SIP/1231805-0000028a", "1?noreport") in new stack
-- Goto (macro-dialout-trunk,continue,3)
-- Executing [continue@macro-dialout-trunk:3] NoOp("SIP/1231805-0000028a", "TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 21 - failing through to other trunks") in new stack
-- Executing [continue@macro-dialout-trunk:4] Set("SIP/1231805-0000028a", "CALLERID(number)=1231805") in new stack
-- Executing [990162166917@from-internal:5] Macro("SIP/1231805-0000028a", "outisbusy,") in new stack
-- Executing [s@macro-outisbusy:1] Progress("SIP/1231805-0000028a", "") in new stack
-- Executing [s@macro-outisbusy:2] GotoIf("SIP/1231805-0000028a", "0?emergency,1") in new stack
-- Executing [s@macro-outisbusy:3] GotoIf("SIP/1231805-0000028a", "0?intracompany,1") in new stack
-- Executing [s@macro-outisbusy:4] Playback("SIP/1231805-0000028a", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
-- <SIP/1231805-0000028a> Playing 'all-circuits-busy-now.gsm' (language 'en')
== Spawn extension (macro-outisbusy, s, 4) exited non-zero on 'SIP/1231805-0000028a' in macro 'outisbusy'
== Spawn extension (from-internal, 990162166917, 5) exited non-zero on 'SIP/1231805-0000028a'
-- Executing [h@from-internal:1] Macro("SIP/1231805-0000028a", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/1231805-0000028a", "1?noautomon") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] NoOp("SIP/1231805-0000028a", "TOUCH_MONITOR_OUTPUT=") in new stack
-- Executing [s@macro-hangupcall:4] GotoIf("SIP/1231805-0000028a", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] GotoIf("SIP/1231805-0000028a", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,10)
-- Executing [s@macro-hangupcall:10] GotoIf("SIP/1231805-0000028a", "1?theend") in new stack
-- Goto (macro-hangupcall,s,12)
-- Executing [s@macro-hangupcall:12] Hangup("SIP/1231805-0000028a", "") in new stack
== Spawn extension (macro-hangupcall, s, 12) exited non-zero on 'SIP/1231805-0000028a' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/1231805-0000028a'
 

Lee Sharp

Joined
Sep 28, 2010
Messages
332
Likes
0
Points
0
#4
I see the errors, but do not know why... Dicko? You got a clue?
 

tsku

Joined
Oct 1, 2010
Messages
62
Likes
0
Points
0
#5
Hi Lee,

I saw some threads said Elastix has some issue with SIP trunk but cant find solution on it. Is this true? Anyway, I will try to use asterisk to connect today to confirm whether problem happen to Elastix only.
 

dicko

Joined
Oct 24, 2008
Messages
4,099
Likes
0
Points
0
#6
I would start by adding

qualify=yes

to his trunk called 2231120
 

danardf

Joined
Dec 3, 2007
Messages
8,069
Likes
10
Points
88
#7
Hi all.

That could be a nat issue too. But like Dicko said, first, put qualify=yes.
 

tsku

Joined
Oct 1, 2010
Messages
62
Likes
0
Points
0
#8
Hi,

Thanks for your advice. I tried with qualify=yes in my sip trunk and still cannot work.

Without qualify=yes, "sip show peers" show sip trunk status unmonitored
with qualify=yes, "sip show peers" show sip trunk status unreachable

Below are the log for outgoing call.



-- Executing [990162166917@from-internal:1] Macro("SIP/1231205-0000048c", "user-callerid,SKIPTTL,") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/1231205-0000048c", "AMPUSER=1231205") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/1231205-0000048c", "0?report") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/1231205-0000048c", "1?Set(REALCALLERIDNUM=1231205)") in new stack
-- Executing [s@macro-user-callerid:4] Set("SIP/1231205-0000048c", "AMPUSER=1231205") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/1231205-0000048c", "AMPUSERCIDNAME=CK") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/1231205-0000048c", "0?report") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/1231205-0000048c", "AMPUSERCID=1231205") in new stack
-- Executing [s@macro-user-callerid:8] Set("SIP/1231205-0000048c", "CALLERID(all)="CK" <1231205>") in new stack
-- Executing [s@macro-user-callerid:9] ExecIf("SIP/1231205-0000048c", "0?Set(CHANNEL(language)=)") in new stack
-- Executing [s@macro-user-callerid:10] GotoIf("SIP/1231205-0000048c", "1?continue") in new stack
-- Goto (macro-user-callerid,s,19)
-- Executing [s@macro-user-callerid:19] NoOp("SIP/1231205-0000048c", "Using CallerID "CK" <1231205>") in new stack
-- Executing [990162166917@from-internal:2] Set("SIP/1231205-0000048c", "_NODEST=") in new stack
-- Executing [990162166917@from-internal:3] Macro("SIP/1231205-0000048c", "record-enable,1231205,OUT,") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/1231205-0000048c", "1?check") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] ExecIf("SIP/1231205-0000048c", "0?MacroExit()") in new stack
-- Executing [s@macro-record-enable:5] GotoIf("SIP/1231205-0000048c", "0?Group:OUT") in new stack
-- Goto (macro-record-enable,s,15)
-- Executing [s@macro-record-enable:15] GotoIf("SIP/1231205-0000048c", "0?IN") in new stack
-- Executing [s@macro-record-enable:16] ExecIf("SIP/1231205-0000048c", "1?MacroExit()") in new stack
-- Executing [990162166917@from-internal:4] Macro("SIP/1231205-0000048c", "dialout-trunk,4,0162166917,,") in new stack
-- Executing [s@macro-dialout-trunk:1] Set("SIP/1231205-0000048c", "DIAL_TRUNK=4") in new stack
-- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/1231205-0000048c", "0?sub-pincheck,s,1") in new stack
-- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/1231205-0000048c", "0?disabletrunk,1") in new stack
-- Executing [s@macro-dialout-trunk:4] Set("SIP/1231205-0000048c", "DIAL_NUMBER=0162166917") in new stack
-- Executing [s@macro-dialout-trunk:5] Set("SIP/1231205-0000048c", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing [s@macro-dialout-trunk:6] Set("SIP/1231205-0000048c", "OUTBOUND_GROUP=OUT_4") in new stack
-- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/1231205-0000048c", "1?nomax") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/1231205-0000048c", "0?skipoutcid") in new stack
-- Executing [s@macro-dialout-trunk:10] Set("SIP/1231205-0000048c", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing [s@macro-dialout-trunk:11] Macro("SIP/1231205-0000048c", "outbound-callerid,4") in new stack
-- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/1231205-0000048c", "0?Set(CALLERPRES()=)") in new stack
-- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/1231205-0000048c", "0?Set(REALCALLERIDNUM=1231205)") in new stack
-- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/1231205-0000048c", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,6)
-- Executing [s@macro-outbound-callerid:6] Set("SIP/1231205-0000048c", "USEROUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:7] Set("SIP/1231205-0000048c", "EMERGENCYCID=") in new stack
-- Executing [s@macro-outbound-callerid:8] Set("SIP/1231205-0000048c", "TRUNKOUTCID=oce-sip-outgoing-redtone") in new stack
-- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/1231205-0000048c", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,12)
-- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/1231205-0000048c", "1?Set(CALLERID(all)=oce-sip-outgoing-redtone)") in new stack
-- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/1231205-0000048c", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/1231205-0000048c", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/1231205-0000048c", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
-- Executing [s@macro-dialout-trunk:12] ExecIf("SIP/1231205-0000048c", "0?AGI(fixlocalprefix)") in new stack
-- Executing [s@macro-dialout-trunk:13] Set("SIP/1231205-0000048c", "OUTNUM=0162166917") in new stack
-- Executing [s@macro-dialout-trunk:14] Set("SIP/1231205-0000048c", "custom=SIP/2231120") in new stack
-- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/1231205-0000048c", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^))") in new stack
-- Executing [s@macro-dialout-trunk:16] Macro("SIP/1231205-0000048c", "dialout-trunk-predial-hook,") in new stack
-- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/1231205-0000048c", "") in new stack
-- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/1231205-0000048c", "0?bypass,1") in new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/1231205-0000048c", "0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:19] Dial("SIP/1231205-0000048c", "SIP/2231120/0162166917,300,") 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/1231205-0000048c", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 20") in new stack
-- Executing [s@macro-dialout-trunk:21] Goto("SIP/1231205-0000048c", "s-CHANUNAVAIL,1") in new stack
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set("SIP/1231205-0000048c", "RC=20") in new stack
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto("SIP/1231205-0000048c", "20,1") in new stack
-- Goto (macro-dialout-trunk,20,1)
-- Executing [20@macro-dialout-trunk:1] Goto("SIP/1231205-0000048c", "continue,1") in new stack
-- Goto (macro-dialout-trunk,continue,1)
-- Executing [continue@macro-dialout-trunk:1] GotoIf("SIP/1231205-0000048c", "1?noreport") in new stack
-- Goto (macro-dialout-trunk,continue,3)
-- Executing [continue@macro-dialout-trunk:3] NoOp("SIP/1231205-0000048c", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 20 - failing through to other trunks") in new stack
-- Executing [continue@macro-dialout-trunk:4] Set("SIP/1231205-0000048c", "CALLERID(number)=1231205") in new stack
-- Executing [990162166917@from-internal:5] Macro("SIP/1231205-0000048c", "outisbusy,") in new stack
-- Executing [s@macro-outisbusy:1] Progress("SIP/1231205-0000048c", "") in new stack
-- Executing [s@macro-outisbusy:2] GotoIf("SIP/1231205-0000048c", "0?emergency,1") in new stack
-- Executing [s@macro-outisbusy:3] GotoIf("SIP/1231205-0000048c", "0?intracompany,1") in new stack
-- Executing [s@macro-outisbusy:4] Playback("SIP/1231205-0000048c", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
-- <SIP/1231205-0000048c> Playing 'all-circuits-busy-now.gsm' (language 'en')
-- <SIP/1231205-0000048c> Playing 'pls-try-call-later.gsm' (language 'en')
 

danardf

Joined
Dec 3, 2007
Messages
8,069
Likes
10
Points
88
#9
qualify=yes, show only the trunk status.

Try to put every nat parameters. So into your trunk nat=yes, and try to find in this forum, how to set the sip_nat.conf file.

Restart Asterisk and try to make a call.
 

tsku

Joined
Oct 1, 2010
Messages
62
Likes
0
Points
0
#10
Dear danardf,

I tried nat=yes before but it doesn't help. FYI, the server is on Public IP and no NAT involved.

Regards
 

dicko

Joined
Oct 24, 2008
Messages
4,099
Likes
0
Points
0
#11
Indeed Franck it only shows the status,

so rasterisk -x 'sip show peers'

will confirm that he is talking to it it will say ok if it does

then sip debug ip 202.211.228.41

to identify what is not working

reagrds

dicko
 

tsku

Joined
Oct 1, 2010
Messages
62
Likes
0
Points
0
#12
Hi Dicko,

SIP set debug ip 202.211.228.41

Audio is at 202.73.9.180 port 13912
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 202.211.228.41:5060:
INVITE sip:0162166917@sip2.abc.com SIP/2.0
Via: SIP/2.0/UDP 202.73.9.180:5060;branch=z9hG4bK3244bf3a;rport
Max-Forwards: 70
From: "oce-sip-outgoing-abc" <sip:Unknown@202.73.9.180>;tag=as63e10240
To: <sip:0162166917@sip2.abc.com>
Contact: <sip:Unknown@202.73.9.180>
Call-ID: 74bb70c35c8829ae0a30258f1bbae10a@202.73.9.180
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.10
Date: Mon, 25 Oct 2010 06:14:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 2231120 2231120 IN IP4 202.73.9.180
s=Asterisk PBX 1.6.2.10
c=IN IP4 202.73.9.180
t=0 0
m=audio 13912 RTP/AVP 8 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
-- Called 2231120/0162166917

<--- SIP read from UDP:202.211.228.41:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 202.211.228.41:5060;branch=z9hG4bK3244bf3a;rport=5060
From: "oce-sip-outgoing-abc" <sip:Unknown@202.73.9.180>;tag=as63e10240
To: <sip:0162166917@sip2.abc.com>;tag=b27e1a1d33761e85846fc98f5f3a7e58.b7c9
Call-ID: 74bb70c35c8829ae0a30258f1bbae10a@202.73.9.180
CSeq: 102 INVITE
Server: Sip EXpress router (0.9.6 (i386/linux))
Content-Length: 0
Warning: 392 202.211.228.41:5060 "Noisy feedback tells: pid=2447 req_src_ip=202.73.9.180 req_src_port=5060 in_uri=sip:0162166917@sip2.abc.com out_uri=sip:0162166917@sip2.abc.com via_cnt==1"


<------------->
--- (9 headers 0 lines) ---
Transmitting (NAT) to 202.211.228.41:5060:
ACK sip:0162166917@sip2.abc.com SIP/2.0
Via: SIP/2.0/UDP 202.73.9.180:5060;branch=z9hG4bK3244bf3a;rport
Max-Forwards: 70
From: "oce-sip-outgoing-abc" <sip:Unknown@202.73.9.180>;tag=as63e10240
To: <sip:0162166917@sip2.abc.com>;tag=b27e1a1d33761e85846fc98f5f3a7e58.b7c9
Contact: <sip:Unknown@202.73.9.180>
Call-ID: 74bb70c35c8829ae0a30258f1bbae10a@202.73.9.180
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.10
Content-Length: 0
 

dicko

Joined
Oct 24, 2008
Messages
4,099
Likes
0
Points
0
#13
I don't think you have enough sip traffic there to be very usuful but it looks like your authorities don't agree.

you might want to try

insecure=very

in the trunk settings
 

tsku

Joined
Oct 1, 2010
Messages
62
Likes
0
Points
0
#14
Hi Dicko,

I'm seeing message below and from is always show unknown, is this the root cause why I can't pass call to clearing house because from is unknown?

Retransmitting #2 (NAT) to 202.211.228.41:5060:
OPTIONS sip:sip2.abc.com SIP/2.0
Via: SIP/2.0/UDP 202.73.9.180:5060;branch=z9hG4bK06bd20f8;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@202.73.9.180>;tag=as6baa4c5f
To: <sip:sip2.abc.com>
Contact: <sip:Unknown@202.73.9.180>
Call-ID: 0730d30319f6a2bf7f7332d10b2daae2@202.73.9.180
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.10
Date: Mon, 25 Oct 2010 08:28:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
 

danardf

Joined
Dec 3, 2007
Messages
8,069
Likes
10
Points
88
#15
Don't forget in your router, to forward every SIP and RTP ports to Elastix server.
5060 UDP and 10000 to 20000 UDP.
 

tsku

Joined
Oct 1, 2010
Messages
62
Likes
0
Points
0
#16
Hi Danardf,

This server is in DMZ with its own Public IP address and no NAT involve, firewall allow all traffic origin from this server and I don't think firewall or router will caused this issue.
 

danardf

Joined
Dec 3, 2007
Messages
8,069
Likes
10
Points
88
#17
And why not? And if it was the case?

Anyway, show me your config trunk. I can check it if you could have any problem.
 

tsku

Joined
Oct 1, 2010
Messages
62
Likes
0
Points
0
#18
[2231120]
disallow=all
username=2231120
type=peer
secret=xjnjnvknliok
host=202.211.228.41
allow=alaw
allow=gsm
context=from-trunk-sip-2231120

I tried to stop the firewall rules but still not working and it proved that firewall is not the root caused for this,
 

danardf

Joined
Dec 3, 2007
Messages
8,069
Likes
10
Points
88
#19
Try to set this config like that:

disallow=all
allow=alaw
allow=gsm
username=2231120
type=peer
secret=xjnjnvknliok
host=202.211.228.41
context=from-trunk-sip-2231120 (I guess that's a custom context, else it's from-trunk)
qualify=yes
insercuce=very


If the context from-trunk-sip-2231120 not exist, you couldn't call, and be called.

If your operator needed a register string information, you must put it.

Code:
account:password@202.211.228.41
To verify your conf, 2 commands:
sip show peers
sip show registry
(in the case where you used the register string.)
 

tsku

Joined
Oct 1, 2010
Messages
62
Likes
0
Points
0
#20
after add in the register string, sip show registry show status 'registered', however outgoing call still show circuit busy, with and without qualify and insecure in place.
 

Members online

No members online now.

Latest posts

Forum statistics

Threads
30,913
Messages
130,917
Members
17,589
Latest member
cristian.saiz
Top