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<DIV><FONT face=Arial size=2>When I put the asterisk server inside NAT, I
do not get any SIP-retransmission problem, SIP-trunk seems to work on all
calls.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Why asterisk is occasionally unable to get SIP
packets out to SIP gateway when connected to PUBLIC IP? </FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Did some body had this problem before?</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<BLOCKQUOTE
style="PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV
style="BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: black"><B>From:</B>
<A title=asterisk@prizmcom.com href="mailto:asterisk@prizmcom.com">OMS</A>
</DIV>
<DIV style="FONT: 10pt arial"><B>To:</B> <A
title=Asterisk-Users@lists.digium.com
href="mailto:Asterisk-Users@lists.digium.com">Asterisk-Users@lists.digium.com</A>
</DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Tuesday, August 09, 2005 4:45
PM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> [Asterisk-Users] SIP-Trunk
problem, Please help!!!</DIV>
<DIV><BR></DIV>
<DIV><FONT face=Arial size=2>
<DIV><FONT face=Arial size=2>Hi,</FONT></DIV>
<DIV><FONT face=Arial size=2>We are using VOIP-SIP gateway to route
outbound PSTN calls.</FONT></DIV>
<DIV><FONT face=Arial size=2>Recently, I am getting == No one is
available to answer at this time </FONT></DIV>
<DIV><FONT face=Arial size=2>message, after making 5 SIP attempts
(Retransmitting #5 (no NAT):), </FONT></DIV>
<DIV><FONT face=Arial size=2>and the calls are going out through alternate
Zap-trunk.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>I do not see any hit (sip-debug traffic) on the
voip-gateway for the failed calls.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Strange thing is that this is happening randomly,
half the call I make are able to get through the SIP-Trunk.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT><FONT face=Arial
size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>I will really appreciate any input/suggession on
this.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Obaid.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Here are my conf files, followed by SIP debug
output on asterisk.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>trunk 4= SIP trunk</FONT></DIV>
<DIV>24.XX.XXX.101 ---> Asterisk server on Public IP</DIV>
<DIV>209.XXX.XXX.113 ---> SIP gatway</DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial
size=2>---------------iax_additional.conf--------------</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial
size=2>[20]<BR>username=20<BR>type=friend<BR>secret=XXX<BR>record_out=On-Demand<BR>record_in=On-Demand<BR>qualify=no<BR>notransfer=yes<BR><A
href="mailto:mailbox=20@default">mailbox=20@default</A><BR>host=dynamic<BR>context=from-internal<BR>callerid="512538XXXX"
<20></FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial
size=2>-------------------Sip_additional.conf---------------</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial
size=2>[23]<BR>username=23<BR>type=friend<BR>secret=XXX<BR>record_out=On-Demand<BR>record_in=On-Demand<BR>qualify=no<BR>port=5060<BR>nat=never<BR><A
href="mailto:mailbox=23@default">mailbox=23@default</A><BR>host=dynamic<BR>dtmfmode=rfc2833<BR>context=from-internal<BR>canreinvite=no<BR>callerid="SIP
Lite" <23></FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial
size=2>[sip-out]<BR>type=peer<BR>host=209.XXX.XXX.113</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial
size=2>-----------------Extensions_additional--------------------------</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>[outrt-001-sip-out]<BR>include =>
outrt-001-Prizm-custom<BR>exten =>
_011.,1,Macro(dialout-trunk,4,${EXTEN},)<BR>exten =>
_011.,2,Macro(dialout-trunk,1,${EXTEN},)<BR>exten =>
_011.,3,Macro(outisbusy) ; No available circuits<BR>exten =>
_1NXXNXXXXXX,1,Macro(dialout-trunk,4,${EXTEN},)<BR>exten =>
_1NXXNXXXXXX,2,Macro(dialout-trunk,1,${EXTEN},)<BR>exten =>
_1NXXNXXXXXX,3,Macro(outisbusy) ; No available circuits<BR>exten =>
_NXXXXXX,1,Macro(dialout-trunk,4,${EXTEN},)<BR>exten =>
_NXXXXXX,2,Macro(dialout-trunk,1,${EXTEN},)<BR>exten =>
_NXXXXXX,3,Macro(outisbusy) ; No available circuits</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>[outrt-002-Local]<BR>include =>
outrt-002-Local-custom<BR>exten =>
_9.,1,Macro(dialout-trunk,1,${EXTEN:1},)<BR>exten =>
_9.,2,Macro(dialout-trunk,2,${EXTEN:1},)<BR>exten =>
_9.,3,Macro(dialout-trunk,3,${EXTEN:1},)<BR>exten =>
_9.,4,Macro(outisbusy) ; No available circuits</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>-----------------------------Sip
Debug----------------------------</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV><FONT face=Arial size=2>
<DIV><BR> -- Executing GotoIf("<A
href="mailto:IAX2/20@20/4">IAX2/20@20/4</A>", "1?5:8") in new
stack<BR> -- Goto
(macro-record-enable,s,5)<BR> -- Executing DBget("<A
href="mailto:IAX2/20@20/4">IAX2/20@20/4</A>", "RecEnable=RECORD-OUT/20") in
new stack<BR> -- DBget: varname=RecEnable,
family=RECORD-OUT, key=20<BR> -- DBget: Value not found in
database.<BR> -- Executing SetVar("<A
href="mailto:IAX2/20@20/4">IAX2/20@20/4</A>",
"CALLFILENAME=OUT20-20050809-163643-1123619803.36") in </DIV>
<DIV> </DIV>
<DIV>new stack<BR> -- Executing Goto("<A
href="mailto:IAX2/20@20/4">IAX2/20@20/4</A>", "s|14") in new
stack<BR> -- Goto
(macro-record-enable,s,14)<BR> -- Executing GotoIf("<A
href="mailto:IAX2/20@20/4">IAX2/20@20/4</A>", "0?15:99") in new
stack<BR> -- Goto
(macro-record-enable,s,99)<BR> -- Executing NoOp("<A
href="mailto:IAX2/20@20/4">IAX2/20@20/4</A>", "NO RECORDING NEEDED") in new
stack<BR> -- Executing GotoIf("<A
href="mailto:IAX2/20@20/4">IAX2/20@20/4</A>", "0?7") in new
stack<BR> -- Executing SetCallerID("<A
href="mailto:IAX2/20@20/4">IAX2/20@20/4</A>", "512538XXX") in new
stack<BR> -- Executing Goto("<A
href="mailto:IAX2/20@20/4">IAX2/20@20/4</A>", "9") in new
stack<BR> -- Goto
(macro-dialout-trunk,s,9)<BR> -- Executing SetGroup("<A
href="mailto:IAX2/20@20/4">IAX2/20@20/4</A>", "OUT_4") in new
stack<BR> -- Executing CheckGroup("<A
href="mailto:IAX2/20@20/4">IAX2/20@20/4</A>", "5") in new
stack<BR> -- Executing SetVar("<A
href="mailto:IAX2/20@20/4">IAX2/20@20/4</A>", "DIAL_NUMBER=484XXX2") in new
stack<BR> -- Executing SetVar("<A
href="mailto:IAX2/20@20/4">IAX2/20@20/4</A>", "DIAL_TRUNK=4") in new
stack<BR> -- Executing AGI("<A
href="mailto:IAX2/20@20/4">IAX2/20@20/4</A>", "fixlocalprefix") in new
stack<BR> -- Launched AGI Script
/var/lib/asterisk/agi-bin/fixlocalprefix<BR> fixlocalprefix: Added
prefix. New number: 1512484XXX2<BR> -- AGI Script
fixlocalprefix completed, returning 0<BR> -- Executing
SetVar("<A href="mailto:IAX2/20@20/4">IAX2/20@20/4</A>", "OUTNUM=1512484XXX2")
in new stack<BR> -- Executing Cut("<A
href="mailto:IAX2/20@20/4">IAX2/20@20/4</A>", "custom=OUT_4|:|1") in new
stack<BR> -- Executing GotoIf("<A
href="mailto:IAX2/20@20/4">IAX2/20@20/4</A>", "0?19") in new
stack<BR> -- Executing Dial("<A
href="mailto:IAX2/20@20/4">IAX2/20@20/4</A>", "SIP/sip-out/1512484XXX2") in
new stack<BR>We're at 24.XX.XXX.101 port 15202<BR>Answering/Requesting with
root capability 0x4 (ulaw)<BR>Answering with preferred capability 0x8
(alaw)<BR>Answering with non-codec capability 0x1 (telephone-event)<BR>12
headers, 11 lines<BR>Reliably Transmitting:<BR>INVITE
sip:1512484XXX2@209.XXX.XXX.113 SIP/2.0<BR>Via: SIP/2.0/UDP
24.XX.XXX.101:5060;branch=z9hG4bK6adcaf4f<BR>From: "512538XXX"
<sip:512538XXXX@24.XX.XXX.101>;tag=as5329d8fe<BR>To:
<sip:1512484XXX2@209.XXX.XXX.113><BR>Contact:
<sip:512538XXX@24.XX.XXX.101><BR>Call-ID: <A
href="mailto:03896a42587e3f973b42daad031ea6a3@24.XX.XXX.101">03896a42587e3f973b42daad031ea6a3@24.XX.XXX.101</A><BR>CSeq:
102 INVITE<BR>User-Agent: Asterisk PBX<BR>Date: Tue, 09 Aug 2005 20:36:43
GMT<BR>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER<BR>Content-Type:
application/sdp<BR>Content-Length: 242</DIV>
<DIV> </DIV>
<DIV>v=0<BR>o=root 2251 2251 IN IP4 24.XX.XXX.101<BR>s=session<BR>c=IN IP4
24.XX.XXX.101<BR>t=0 0<BR>m=audio 15202 RTP/AVP 0 8 101<BR>a=rtpmap:0
PCMU/8000<BR>a=rtpmap:8 PCMA/8000<BR>a=rtpmap:101
telephone-event/8000<BR>a=fmtp:101 0-16<BR>a=silenceSupp:off - - -
-<BR> (no NAT) to 209.XXX.XXX.113:5060<BR> -- Called
sip-out/1512484XXX2<BR>Retransmitting #1 (no NAT):<BR>INVITE
sip:1512484XXX2@209.XXX.XXX.113 SIP/2.0<BR>Via: SIP/2.0/UDP
24.XX.XXX.101:5060;branch=z9hG4bK6adcaf4f<BR>From: "512538XXX"
<sip:512538XXX@24.XX.XXX.101>;tag=as5329d8fe<BR>To:
<sip:1512484XXX2@209.XXX.XXX.113><BR>Contact:
<sip:512538XXX@24.XX.XXX.101><BR>Call-ID: <A
href="mailto:03896a42587e3f973b42daad031ea6a3@24.XX.XXX.101">03896a42587e3f973b42daad031ea6a3@24.XX.XXX.101</A><BR>CSeq:
102 INVITE<BR>User-Agent: Asterisk PBX<BR>Date: Tue, 09 Aug 2005 20:36:43
GMT<BR>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER<BR>Content-Type:
application/sdp<BR>Content-Length: 242</DIV>
<DIV> </DIV>
<DIV>v=0<BR>o=root 2251 2251 IN IP4 24.XX.XXX.101<BR>s=session<BR>c=IN IP4
24.XX.XXX.101<BR>t=0 0<BR>m=audio 15202 RTP/AVP 0 8 101<BR>a=rtpmap:0
PCMU/8000<BR>a=rtpmap:8 PCMA/8000<BR>a=rtpmap:101
telephone-event/8000<BR>a=fmtp:101 0-16<BR>a=silenceSupp:off - - - -</DIV>
<DIV> </DIV>
<DIV> to 209.XXX.XXX.113:5060<BR>Retransmitting #2 (no NAT):<BR>INVITE
sip:1512484XXX2@209.XXX.XXX.113 SIP/2.0<BR>Via: SIP/2.0/UDP
24.XX.XXX.101:5060;branch=z9hG4bK6adcaf4f<BR>From: "512538XXX"
<sip:512538XXX@24.XX.XXX.101>;tag=as5329d8fe<BR>To:
<sip:1512484XXX2@209.XXX.XXX.113><BR>Contact:
<sip:512538XXX@24.XX.XXX.101><BR>Call-ID: <A
href="mailto:03896a42587e3f973b42daad031ea6a3@24.XX.XXX.101">03896a42587e3f973b42daad031ea6a3@24.XX.XXX.101</A><BR>CSeq:
102 INVITE<BR>User-Agent: Asterisk PBX<BR>Date: Tue, 09 Aug 2005 20:36:43
GMT<BR>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER<BR>Content-Type:
application/sdp<BR>Content-Length: 242</DIV>
<DIV> </DIV>
<DIV>v=0<BR>o=root 2251 2251 IN IP4 24.XX.XXX.101<BR>s=session<BR>c=IN IP4
24.XX.XXX.101<BR>t=0 0<BR>m=audio 15202 RTP/AVP 0 8 101<BR>a=rtpmap:0
PCMU/8000<BR>a=rtpmap:8 PCMA/8000<BR>a=rtpmap:101
telephone-event/8000<BR>a=fmtp:101 0-16<BR>a=silenceSupp:off - - - -</DIV>
<DIV> </DIV>
<DIV> to 209.XXX.XXX.113:5060<BR>Retransmitting #3 (no NAT):<BR>INVITE
sip:1512484XXX2@209.XXX.XXX.113 SIP/2.0<BR>Via: SIP/2.0/UDP
24.XX.XXX.101:5060;branch=z9hG4bK6adcaf4f<BR>From: "512538XXX"
<sip:512538XXX@24.XX.XXX.101>;tag=as5329d8fe<BR>To:
<sip:1512484XXX2@209.XXX.XXX.113><BR>Contact:
<sip:512538XXX@24.XX.XXX.101><BR>Call-ID: <A
href="mailto:03896a42587e3f973b42daad031ea6a3@24.XX.XXX.101">03896a42587e3f973b42daad031ea6a3@24.XX.XXX.101</A><BR>CSeq:
102 INVITE<BR>User-Agent: Asterisk PBX<BR>Date: Tue, 09 Aug 2005 20:36:43
GMT<BR>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER<BR>Content-Type:
application/sdp<BR>Content-Length: 242</DIV>
<DIV> </DIV>
<DIV>v=0<BR>o=root 2251 2251 IN IP4 24.XX.XXX.101<BR>s=session<BR>c=IN IP4
24.XX.XXX.101<BR>t=0 0<BR>m=audio 15202 RTP/AVP 0 8 101<BR>a=rtpmap:0
PCMU/8000<BR>a=rtpmap:8 PCMA/8000<BR>a=rtpmap:101
telephone-event/8000<BR>a=fmtp:101 0-16<BR>a=silenceSupp:off - - - -</DIV>
<DIV> </DIV>
<DIV> to 209.XXX.XXX.113:5060<BR>Retransmitting #4 (no NAT):<BR>INVITE
sip:1512484XXX2@209.XXX.XXX.113 SIP/2.0<BR>Via: SIP/2.0/UDP
24.XX.XXX.101:5060;branch=z9hG4bK6adcaf4f<BR>From: "512538XXX"
<sip:512538XXX@24.XX.XXX.101>;tag=as5329d8fe<BR>To:
<sip:1512484XXX2@209.XXX.XXX.113><BR>Contact:
<sip:512538XXX@24.XX.XXX.101><BR>Call-ID: <A
href="mailto:03896a42587e3f973b42daad031ea6a3@24.XX.XXX.101">03896a42587e3f973b42daad031ea6a3@24.XX.XXX.101</A><BR>CSeq:
102 INVITE<BR>User-Agent: Asterisk PBX<BR>Date: Tue, 09 Aug 2005 20:36:43
GMT<BR>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER<BR>Content-Type:
application/sdp<BR>Content-Length: 242</DIV>
<DIV> </DIV>
<DIV>v=0<BR>o=root 2251 2251 IN IP4 24.XX.XXX.101<BR>s=session<BR>c=IN IP4
24.XX.XXX.101<BR>t=0 0<BR>m=audio 15202 RTP/AVP 0 8 101<BR>a=rtpmap:0
PCMU/8000<BR>a=rtpmap:8 PCMA/8000<BR>a=rtpmap:101
telephone-event/8000<BR>a=fmtp:101 0-16<BR>a=silenceSupp:off - - - -</DIV>
<DIV> </DIV>
<DIV> to 209.XXX.XXX.113:5060<BR>Retransmitting #5 (no NAT):<BR>INVITE
sip:1512484XXX2@209.XXX.XXX.113 SIP/2.0<BR>Via: SIP/2.0/UDP
24.XX.XXX.101:5060;branch=z9hG4bK6adcaf4f<BR>From: "512538XXX"
<sip:512538XXX@24.XX.XXX.101>;tag=as5329d8fe<BR>To:
<sip:1512484XXX2@209.XXX.XXX.113><BR>Contact:
<sip:512538XXX@24.XX.XXX.101><BR>Call-ID: <A
href="mailto:03896a42587e3f973b42daad031ea6a3@24.XX.XXX.101">03896a42587e3f973b42daad031ea6a3@24.XX.XXX.101</A><BR>CSeq:
102 INVITE<BR>User-Agent: Asterisk PBX<BR>Date: Tue, 09 Aug 2005 20:36:43
GMT<BR>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER<BR>Content-Type:
application/sdp<BR>Content-Length: 242</DIV>
<DIV> </DIV>
<DIV>v=0<BR>o=root 2251 2251 IN IP4 24.XX.XXX.101<BR>s=session<BR>c=IN IP4
24.XX.XXX.101<BR>t=0 0<BR>m=audio 15202 RTP/AVP 0 8 101<BR>a=rtpmap:0
PCMU/8000<BR>a=rtpmap:8 PCMA/8000<BR>a=rtpmap:101
telephone-event/8000<BR>a=fmtp:101 0-16<BR>a=silenceSupp:off - - - -</DIV>
<DIV> </DIV>
<DIV> to 209.XXX.XXX.113:5060<BR> == No one is available to answer
at this time<BR> -- Executing Goto("<A
href="mailto:IAX2/20@20/4">IAX2/20@20/4</A>", "s-NOANSWER|1") in new
stack<BR> -- Goto
(macro-dialout-trunk,s-NOANSWER,1)<BR> -- Executing NoOp("<A
href="mailto:IAX2/20@20/4">IAX2/20@20/4</A>", "Dial failed due to NOANSWER")
in new stack<BR> -- Executing Macro("<A
href="mailto:IAX2/20@20/4">IAX2/20@20/4</A>", "dialout-trunk|1|484XXX2|") in
new stack<BR> -- Executing GotoIf("<A
href="mailto:IAX2/20@20/4">IAX2/20@20/4</A>", "1?3:2)") in new
stack<BR> -- Goto
(macro-dialout-trunk,s,3)<BR> -- Executing Macro("<A
href="mailto:IAX2/20@20/4">IAX2/20@20/4</A>", "record-enable|512538XXX|OUT")
in new stack<BR> -- Executing GotoIf("<A
href="mailto:IAX2/20@20/4">IAX2/20@20/4</A>", "0 > 0?2:4") in new
stack<BR> -- Goto
(macro-record-enable,s,4)<BR> -- Executing GotoIf("<A
href="mailto:IAX2/20@20/4">IAX2/20@20/4</A>", "1?5:8") in new
stack<BR> -- Goto
(macro-record-enable,s,5)<BR> -- Executing DBget("<A
href="mailto:IAX2/20@20/4">IAX2/20@20/4</A>",
"RecEnable=RECORD-OUT/512538XXX") in new stack<BR> -- DBget:
varname=RecEnable, family=RECORD-OUT, key=512538XXX<BR> --
DBget: Value not found in database.<BR> -- Executing
SetVar("<A href="mailto:IAX2/20@20/4">IAX2/20@20/4</A>", </DIV>
<DIV> </DIV>
<DIV>"CALLFILENAME=OUT512538XXX-20050809-163649-1123619803.36") in new
stack<BR> -- Executing Goto("<A
href="mailto:IAX2/20@20/4">IAX2/20@20/4</A>", "s|14") in new
stack<BR> -- Goto
(macro-record-enable,s,14)<BR> -- Executing GotoIf("<A
href="mailto:IAX2/20@20/4">IAX2/20@20/4</A>", "0?15:99") in new
stack<BR> -- Goto
(macro-record-enable,s,99)<BR> -- Executing NoOp("<A
href="mailto:IAX2/20@20/4">IAX2/20@20/4</A>", "NO RECORDING NEEDED") in new
stack<BR> -- Executing GotoIf("<A
href="mailto:IAX2/20@20/4">IAX2/20@20/4</A>", "1?7") in new
stack<BR> -- Goto
(macro-dialout-trunk,s,7)<BR> -- Executing GotoIf("<A
href="mailto:IAX2/20@20/4">IAX2/20@20/4</A>", "1?9") in new
stack<BR> -- Goto
(macro-dialout-trunk,s,9)<BR> -- Executing SetGroup("<A
href="mailto:IAX2/20@20/4">IAX2/20@20/4</A>", "OUT_1") in new
stack<BR> -- Executing CheckGroup("<A
href="mailto:IAX2/20@20/4">IAX2/20@20/4</A>", "") in new
stack<BR> -- Executing SetVar("<A
href="mailto:IAX2/20@20/4">IAX2/20@20/4</A>", "DIAL_NUMBER=484XXX2") in new
stack<BR> -- Executing SetVar("<A
href="mailto:IAX2/20@20/4">IAX2/20@20/4</A>", "DIAL_TRUNK=1") in new
stack<BR> -- Executing AGI("<A
href="mailto:IAX2/20@20/4">IAX2/20@20/4</A>", "fixlocalprefix") in new
stack<BR> -- Launched AGI Script
/var/lib/asterisk/agi-bin/fixlocalprefix<BR> -- AGI Script
fixlocalprefix completed, returning 0<BR> -- Executing
SetVar("<A href="mailto:IAX2/20@20/4">IAX2/20@20/4</A>", "OUTNUM=484XXX2") in
new stack<BR> -- Executing Cut("<A
href="mailto:IAX2/20@20/4">IAX2/20@20/4</A>", "custom=OUT_1|:|1") in new
stack<BR> -- Executing GotoIf("<A
href="mailto:IAX2/20@20/4">IAX2/20@20/4</A>", "0?19") in new
stack<BR> -- Executing Dial("<A
href="mailto:IAX2/20@20/4">IAX2/20@20/4</A>", "ZAP/g0/484XXX2") in new
stack<BR> -- Called g0/484XXX2<BR>Destroying call <A
href="mailto:'03896a42587e3f973b42daad031ea6a3@24.XX.XXX.101'">'03896a42587e3f973b42daad031ea6a3@24.XX.XXX.101'</A><BR>
-- Zap/1-1 answered <A
href="mailto:IAX2/20@20/4">IAX2/20@20/4</A><BR> -- Hungup
'Zap/1-1'<BR> == Spawn extension (macro-dialout-trunk, s, 17) exited
non-zero on <A href="mailto:'IAX2/20@20/4'">'IAX2/20@20/4'</A> in macro </DIV>
<DIV> </DIV>
<DIV>'dialout-trunk'<BR> == Spawn extension (from-internal, 484XXX2, 2)
exited non-zero on <A
href="mailto:'IAX2/20@20/4'">'IAX2/20@20/4'</A><BR> --
Executing Macro("<A href="mailto:IAX2/20@20/4">IAX2/20@20/4</A>",
"hangupcall") in new stack<BR> -- Executing ResetCDR("<A
href="mailto:IAX2/20@20/4">IAX2/20@20/4</A>", "w") in new
stack<BR> -- Executing NoCDR("<A
href="mailto:IAX2/20@20/4">IAX2/20@20/4</A>", "") in new
stack<BR> -- Executing Wait("<A
href="mailto:IAX2/20@20/4">IAX2/20@20/4</A>", "5") in new stack<BR> ==
Spawn extension (macro-hangupcall, s, 3) exited non-zero on <A
href="mailto:'IAX2/20@20/4'">'IAX2/20@20/4'</A> in macro </DIV>
<DIV> </DIV>
<DIV>'hangupcall'<BR> == Spawn extension (from-internal, h, 1) exited
non-zero on <A
href="mailto:'IAX2/20@20/4'">'IAX2/20@20/4'</A><BR> --
Hungup <A href="mailto:'IAX2/20@20/4'">'IAX2/20@20/4'</A></DIV>
<DIV> </DIV>
<DIV> </DIV>
<DIV> </DIV>
<DIV></FONT> </DIV></FONT></DIV>
<P>
<HR>
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