[Asterisk-Users] SIP-Trunk problem, Please help!!!
OMS
asterisk at prizmcom.com
Tue Aug 9 14:45:04 MST 2005
Hi,
We are using VOIP-SIP gateway to route outbound PSTN calls.
Recently, I am getting == No one is available to answer at this time
message, after making 5 SIP attempts (Retransmitting #5 (no NAT):),
and the calls are going out through alternate Zap-trunk.
I do not see any hit (sip-debug traffic) on the voip-gateway for the failed calls.
Strange thing is that this is happening randomly, half the call I make are able to get through the SIP-Trunk.
I will really appreciate any input/suggession on this.
Obaid.
Here are my conf files, followed by SIP debug output on asterisk.
trunk 4= SIP trunk
24.XX.XXX.101 ---> Asterisk server on Public IP
209.XXX.XXX.113 ---> SIP gatway
---------------iax_additional.conf--------------
[20]
username=20
type=friend
secret=XXX
record_out=On-Demand
record_in=On-Demand
qualify=no
notransfer=yes
mailbox=20 at default
host=dynamic
context=from-internal
callerid="512538XXXX" <20>
-------------------Sip_additional.conf---------------
[23]
username=23
type=friend
secret=XXX
record_out=On-Demand
record_in=On-Demand
qualify=no
port=5060
nat=never
mailbox=23 at default
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid="SIP Lite" <23>
[sip-out]
type=peer
host=209.XXX.XXX.113
-----------------Extensions_additional--------------------------
[outrt-001-sip-out]
include => outrt-001-Prizm-custom
exten => _011.,1,Macro(dialout-trunk,4,${EXTEN},)
exten => _011.,2,Macro(dialout-trunk,1,${EXTEN},)
exten => _011.,3,Macro(outisbusy) ; No available circuits
exten => _1NXXNXXXXXX,1,Macro(dialout-trunk,4,${EXTEN},)
exten => _1NXXNXXXXXX,2,Macro(dialout-trunk,1,${EXTEN},)
exten => _1NXXNXXXXXX,3,Macro(outisbusy) ; No available circuits
exten => _NXXXXXX,1,Macro(dialout-trunk,4,${EXTEN},)
exten => _NXXXXXX,2,Macro(dialout-trunk,1,${EXTEN},)
exten => _NXXXXXX,3,Macro(outisbusy) ; No available circuits
[outrt-002-Local]
include => outrt-002-Local-custom
exten => _9.,1,Macro(dialout-trunk,1,${EXTEN:1},)
exten => _9.,2,Macro(dialout-trunk,2,${EXTEN:1},)
exten => _9.,3,Macro(dialout-trunk,3,${EXTEN:1},)
exten => _9.,4,Macro(outisbusy) ; No available circuits
-----------------------------Sip Debug----------------------------
-- Executing GotoIf("IAX2/20 at 20/4", "1?5:8") in new stack
-- Goto (macro-record-enable,s,5)
-- Executing DBget("IAX2/20 at 20/4", "RecEnable=RECORD-OUT/20") in new stack
-- DBget: varname=RecEnable, family=RECORD-OUT, key=20
-- DBget: Value not found in database.
-- Executing SetVar("IAX2/20 at 20/4", "CALLFILENAME=OUT20-20050809-163643-1123619803.36") in
new stack
-- Executing Goto("IAX2/20 at 20/4", "s|14") in new stack
-- Goto (macro-record-enable,s,14)
-- Executing GotoIf("IAX2/20 at 20/4", "0?15:99") in new stack
-- Goto (macro-record-enable,s,99)
-- Executing NoOp("IAX2/20 at 20/4", "NO RECORDING NEEDED") in new stack
-- Executing GotoIf("IAX2/20 at 20/4", "0?7") in new stack
-- Executing SetCallerID("IAX2/20 at 20/4", "512538XXX") in new stack
-- Executing Goto("IAX2/20 at 20/4", "9") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing SetGroup("IAX2/20 at 20/4", "OUT_4") in new stack
-- Executing CheckGroup("IAX2/20 at 20/4", "5") in new stack
-- Executing SetVar("IAX2/20 at 20/4", "DIAL_NUMBER=484XXX2") in new stack
-- Executing SetVar("IAX2/20 at 20/4", "DIAL_TRUNK=4") in new stack
-- Executing AGI("IAX2/20 at 20/4", "fixlocalprefix") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
fixlocalprefix: Added prefix. New number: 1512484XXX2
-- AGI Script fixlocalprefix completed, returning 0
-- Executing SetVar("IAX2/20 at 20/4", "OUTNUM=1512484XXX2") in new stack
-- Executing Cut("IAX2/20 at 20/4", "custom=OUT_4|:|1") in new stack
-- Executing GotoIf("IAX2/20 at 20/4", "0?19") in new stack
-- Executing Dial("IAX2/20 at 20/4", "SIP/sip-out/1512484XXX2") in new stack
We're at 24.XX.XXX.101 port 15202
Answering/Requesting with root capability 0x4 (ulaw)
Answering with preferred capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
12 headers, 11 lines
Reliably Transmitting:
INVITE sip:1512484XXX2 at 209.XXX.XXX.113 SIP/2.0
Via: SIP/2.0/UDP 24.XX.XXX.101:5060;branch=z9hG4bK6adcaf4f
From: "512538XXX" <sip:512538XXXX at 24.XX.XXX.101>;tag=as5329d8fe
To: <sip:1512484XXX2 at 209.XXX.XXX.113>
Contact: <sip:512538XXX at 24.XX.XXX.101>
Call-ID: 03896a42587e3f973b42daad031ea6a3 at 24.XX.XXX.101
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Tue, 09 Aug 2005 20:36:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 242
v=0
o=root 2251 2251 IN IP4 24.XX.XXX.101
s=session
c=IN IP4 24.XX.XXX.101
t=0 0
m=audio 15202 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
(no NAT) to 209.XXX.XXX.113:5060
-- Called sip-out/1512484XXX2
Retransmitting #1 (no NAT):
INVITE sip:1512484XXX2 at 209.XXX.XXX.113 SIP/2.0
Via: SIP/2.0/UDP 24.XX.XXX.101:5060;branch=z9hG4bK6adcaf4f
From: "512538XXX" <sip:512538XXX at 24.XX.XXX.101>;tag=as5329d8fe
To: <sip:1512484XXX2 at 209.XXX.XXX.113>
Contact: <sip:512538XXX at 24.XX.XXX.101>
Call-ID: 03896a42587e3f973b42daad031ea6a3 at 24.XX.XXX.101
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Tue, 09 Aug 2005 20:36:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 242
v=0
o=root 2251 2251 IN IP4 24.XX.XXX.101
s=session
c=IN IP4 24.XX.XXX.101
t=0 0
m=audio 15202 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
to 209.XXX.XXX.113:5060
Retransmitting #2 (no NAT):
INVITE sip:1512484XXX2 at 209.XXX.XXX.113 SIP/2.0
Via: SIP/2.0/UDP 24.XX.XXX.101:5060;branch=z9hG4bK6adcaf4f
From: "512538XXX" <sip:512538XXX at 24.XX.XXX.101>;tag=as5329d8fe
To: <sip:1512484XXX2 at 209.XXX.XXX.113>
Contact: <sip:512538XXX at 24.XX.XXX.101>
Call-ID: 03896a42587e3f973b42daad031ea6a3 at 24.XX.XXX.101
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Tue, 09 Aug 2005 20:36:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 242
v=0
o=root 2251 2251 IN IP4 24.XX.XXX.101
s=session
c=IN IP4 24.XX.XXX.101
t=0 0
m=audio 15202 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
to 209.XXX.XXX.113:5060
Retransmitting #3 (no NAT):
INVITE sip:1512484XXX2 at 209.XXX.XXX.113 SIP/2.0
Via: SIP/2.0/UDP 24.XX.XXX.101:5060;branch=z9hG4bK6adcaf4f
From: "512538XXX" <sip:512538XXX at 24.XX.XXX.101>;tag=as5329d8fe
To: <sip:1512484XXX2 at 209.XXX.XXX.113>
Contact: <sip:512538XXX at 24.XX.XXX.101>
Call-ID: 03896a42587e3f973b42daad031ea6a3 at 24.XX.XXX.101
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Tue, 09 Aug 2005 20:36:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 242
v=0
o=root 2251 2251 IN IP4 24.XX.XXX.101
s=session
c=IN IP4 24.XX.XXX.101
t=0 0
m=audio 15202 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
to 209.XXX.XXX.113:5060
Retransmitting #4 (no NAT):
INVITE sip:1512484XXX2 at 209.XXX.XXX.113 SIP/2.0
Via: SIP/2.0/UDP 24.XX.XXX.101:5060;branch=z9hG4bK6adcaf4f
From: "512538XXX" <sip:512538XXX at 24.XX.XXX.101>;tag=as5329d8fe
To: <sip:1512484XXX2 at 209.XXX.XXX.113>
Contact: <sip:512538XXX at 24.XX.XXX.101>
Call-ID: 03896a42587e3f973b42daad031ea6a3 at 24.XX.XXX.101
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Tue, 09 Aug 2005 20:36:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 242
v=0
o=root 2251 2251 IN IP4 24.XX.XXX.101
s=session
c=IN IP4 24.XX.XXX.101
t=0 0
m=audio 15202 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
to 209.XXX.XXX.113:5060
Retransmitting #5 (no NAT):
INVITE sip:1512484XXX2 at 209.XXX.XXX.113 SIP/2.0
Via: SIP/2.0/UDP 24.XX.XXX.101:5060;branch=z9hG4bK6adcaf4f
From: "512538XXX" <sip:512538XXX at 24.XX.XXX.101>;tag=as5329d8fe
To: <sip:1512484XXX2 at 209.XXX.XXX.113>
Contact: <sip:512538XXX at 24.XX.XXX.101>
Call-ID: 03896a42587e3f973b42daad031ea6a3 at 24.XX.XXX.101
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Tue, 09 Aug 2005 20:36:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 242
v=0
o=root 2251 2251 IN IP4 24.XX.XXX.101
s=session
c=IN IP4 24.XX.XXX.101
t=0 0
m=audio 15202 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
to 209.XXX.XXX.113:5060
== No one is available to answer at this time
-- Executing Goto("IAX2/20 at 20/4", "s-NOANSWER|1") in new stack
-- Goto (macro-dialout-trunk,s-NOANSWER,1)
-- Executing NoOp("IAX2/20 at 20/4", "Dial failed due to NOANSWER") in new stack
-- Executing Macro("IAX2/20 at 20/4", "dialout-trunk|1|484XXX2|") in new stack
-- Executing GotoIf("IAX2/20 at 20/4", "1?3:2)") in new stack
-- Goto (macro-dialout-trunk,s,3)
-- Executing Macro("IAX2/20 at 20/4", "record-enable|512538XXX|OUT") in new stack
-- Executing GotoIf("IAX2/20 at 20/4", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing GotoIf("IAX2/20 at 20/4", "1?5:8") in new stack
-- Goto (macro-record-enable,s,5)
-- Executing DBget("IAX2/20 at 20/4", "RecEnable=RECORD-OUT/512538XXX") in new stack
-- DBget: varname=RecEnable, family=RECORD-OUT, key=512538XXX
-- DBget: Value not found in database.
-- Executing SetVar("IAX2/20 at 20/4",
"CALLFILENAME=OUT512538XXX-20050809-163649-1123619803.36") in new stack
-- Executing Goto("IAX2/20 at 20/4", "s|14") in new stack
-- Goto (macro-record-enable,s,14)
-- Executing GotoIf("IAX2/20 at 20/4", "0?15:99") in new stack
-- Goto (macro-record-enable,s,99)
-- Executing NoOp("IAX2/20 at 20/4", "NO RECORDING NEEDED") in new stack
-- Executing GotoIf("IAX2/20 at 20/4", "1?7") in new stack
-- Goto (macro-dialout-trunk,s,7)
-- Executing GotoIf("IAX2/20 at 20/4", "1?9") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing SetGroup("IAX2/20 at 20/4", "OUT_1") in new stack
-- Executing CheckGroup("IAX2/20 at 20/4", "") in new stack
-- Executing SetVar("IAX2/20 at 20/4", "DIAL_NUMBER=484XXX2") in new stack
-- Executing SetVar("IAX2/20 at 20/4", "DIAL_TRUNK=1") in new stack
-- Executing AGI("IAX2/20 at 20/4", "fixlocalprefix") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
-- AGI Script fixlocalprefix completed, returning 0
-- Executing SetVar("IAX2/20 at 20/4", "OUTNUM=484XXX2") in new stack
-- Executing Cut("IAX2/20 at 20/4", "custom=OUT_1|:|1") in new stack
-- Executing GotoIf("IAX2/20 at 20/4", "0?19") in new stack
-- Executing Dial("IAX2/20 at 20/4", "ZAP/g0/484XXX2") in new stack
-- Called g0/484XXX2
Destroying call '03896a42587e3f973b42daad031ea6a3 at 24.XX.XXX.101'
-- Zap/1-1 answered IAX2/20 at 20/4
-- Hungup 'Zap/1-1'
== Spawn extension (macro-dialout-trunk, s, 17) exited non-zero on 'IAX2/20 at 20/4' in macro
'dialout-trunk'
== Spawn extension (from-internal, 484XXX2, 2) exited non-zero on 'IAX2/20 at 20/4'
-- Executing Macro("IAX2/20 at 20/4", "hangupcall") in new stack
-- Executing ResetCDR("IAX2/20 at 20/4", "w") in new stack
-- Executing NoCDR("IAX2/20 at 20/4", "") in new stack
-- Executing Wait("IAX2/20 at 20/4", "5") in new stack
== Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'IAX2/20 at 20/4' in macro
'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'IAX2/20 at 20/4'
-- Hungup 'IAX2/20 at 20/4'
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