thr3ads.net - asterisk users - [Asterisk-Users] SIP-Trunk problem, Please help!!! [Aug 2005]

If this information is useful, please help other people find it:
Share via:

OMS

2005-Aug-09 14:45 UTC

[Asterisk-Users] SIP-Trunk problem, Please help!!!

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@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@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@20/4", "1?5:8") in new
stack
    -- Goto (macro-record-enable,s,5)
    -- Executing DBget("IAX2/20@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@20/4",
"CALLFILENAME=OUT20-20050809-163643-1123619803.36") in

new stack
    -- Executing Goto("IAX2/20@20/4", "s|14") in new stack
    -- Goto (macro-record-enable,s,14)
    -- Executing GotoIf("IAX2/20@20/4", "0?15:99") in new
stack
    -- Goto (macro-record-enable,s,99)
    -- Executing NoOp("IAX2/20@20/4", "NO RECORDING NEEDED")
in new stack
    -- Executing GotoIf("IAX2/20@20/4", "0?7") in new stack
    -- Executing SetCallerID("IAX2/20@20/4", "512538XXX") in
new stack
    -- Executing Goto("IAX2/20@20/4", "9") in new stack
    -- Goto (macro-dialout-trunk,s,9)
    -- Executing SetGroup("IAX2/20@20/4", "OUT_4") in new
stack
    -- Executing CheckGroup("IAX2/20@20/4", "5") in new
stack
    -- Executing SetVar("IAX2/20@20/4",
"DIAL_NUMBER=484XXX2") in new stack
    -- Executing SetVar("IAX2/20@20/4", "DIAL_TRUNK=4") in
new stack
    -- Executing AGI("IAX2/20@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@20/4",
"OUTNUM=1512484XXX2") in new stack
    -- Executing Cut("IAX2/20@20/4", "custom=OUT_4|:|1") in
new stack
    -- Executing GotoIf("IAX2/20@20/4", "0?19") in new stack
    -- Executing Dial("IAX2/20@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@209.XXX.XXX.113 SIP/2.0
Via: SIP/2.0/UDP 24.XX.XXX.101:5060;branch=z9hG4bK6adcaf4f
From: "512538XXX" <sip:512538XXXX@24.XX.XXX.101>;tag=as5329d8fe
To: <sip:1512484XXX2@209.XXX.XXX.113>
Contact: <sip:512538XXX@24.XX.XXX.101>
Call-ID: 03896a42587e3f973b42daad031ea6a3@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@209.XXX.XXX.113 SIP/2.0
Via: SIP/2.0/UDP 24.XX.XXX.101:5060;branch=z9hG4bK6adcaf4f
From: "512538XXX" <sip:512538XXX@24.XX.XXX.101>;tag=as5329d8fe
To: <sip:1512484XXX2@209.XXX.XXX.113>
Contact: <sip:512538XXX@24.XX.XXX.101>
Call-ID: 03896a42587e3f973b42daad031ea6a3@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@209.XXX.XXX.113 SIP/2.0
Via: SIP/2.0/UDP 24.XX.XXX.101:5060;branch=z9hG4bK6adcaf4f
From: "512538XXX" <sip:512538XXX@24.XX.XXX.101>;tag=as5329d8fe
To: <sip:1512484XXX2@209.XXX.XXX.113>
Contact: <sip:512538XXX@24.XX.XXX.101>
Call-ID: 03896a42587e3f973b42daad031ea6a3@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@209.XXX.XXX.113 SIP/2.0
Via: SIP/2.0/UDP 24.XX.XXX.101:5060;branch=z9hG4bK6adcaf4f
From: "512538XXX" <sip:512538XXX@24.XX.XXX.101>;tag=as5329d8fe
To: <sip:1512484XXX2@209.XXX.XXX.113>
Contact: <sip:512538XXX@24.XX.XXX.101>
Call-ID: 03896a42587e3f973b42daad031ea6a3@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@209.XXX.XXX.113 SIP/2.0
Via: SIP/2.0/UDP 24.XX.XXX.101:5060;branch=z9hG4bK6adcaf4f
From: "512538XXX" <sip:512538XXX@24.XX.XXX.101>;tag=as5329d8fe
To: <sip:1512484XXX2@209.XXX.XXX.113>
Contact: <sip:512538XXX@24.XX.XXX.101>
Call-ID: 03896a42587e3f973b42daad031ea6a3@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@209.XXX.XXX.113 SIP/2.0
Via: SIP/2.0/UDP 24.XX.XXX.101:5060;branch=z9hG4bK6adcaf4f
From: "512538XXX" <sip:512538XXX@24.XX.XXX.101>;tag=as5329d8fe
To: <sip:1512484XXX2@209.XXX.XXX.113>
Contact: <sip:512538XXX@24.XX.XXX.101>
Call-ID: 03896a42587e3f973b42daad031ea6a3@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@20/4", "s-NOANSWER|1") in new
stack
    -- Goto (macro-dialout-trunk,s-NOANSWER,1)
    -- Executing NoOp("IAX2/20@20/4", "Dial failed due to
NOANSWER") in new stack
    -- Executing Macro("IAX2/20@20/4",
"dialout-trunk|1|484XXX2|") in new stack
    -- Executing GotoIf("IAX2/20@20/4", "1?3:2)") in new
stack
    -- Goto (macro-dialout-trunk,s,3)
    -- Executing Macro("IAX2/20@20/4",
"record-enable|512538XXX|OUT") in new stack
    -- Executing GotoIf("IAX2/20@20/4", "0 > 0?2:4") in
new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing GotoIf("IAX2/20@20/4", "1?5:8") in new
stack
    -- Goto (macro-record-enable,s,5)
    -- Executing DBget("IAX2/20@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@20/4", 

"CALLFILENAME=OUT512538XXX-20050809-163649-1123619803.36") in new
stack
    -- Executing Goto("IAX2/20@20/4", "s|14") in new stack
    -- Goto (macro-record-enable,s,14)
    -- Executing GotoIf("IAX2/20@20/4", "0?15:99") in new
stack
    -- Goto (macro-record-enable,s,99)
    -- Executing NoOp("IAX2/20@20/4", "NO RECORDING NEEDED")
in new stack
    -- Executing GotoIf("IAX2/20@20/4", "1?7") in new stack
    -- Goto (macro-dialout-trunk,s,7)
    -- Executing GotoIf("IAX2/20@20/4", "1?9") in new stack
    -- Goto (macro-dialout-trunk,s,9)
    -- Executing SetGroup("IAX2/20@20/4", "OUT_1") in new
stack
    -- Executing CheckGroup("IAX2/20@20/4", "") in new stack
    -- Executing SetVar("IAX2/20@20/4",
"DIAL_NUMBER=484XXX2") in new stack
    -- Executing SetVar("IAX2/20@20/4", "DIAL_TRUNK=1") in
new stack
    -- Executing AGI("IAX2/20@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@20/4", "OUTNUM=484XXX2") in
new stack
    -- Executing Cut("IAX2/20@20/4", "custom=OUT_1|:|1") in
new stack
    -- Executing GotoIf("IAX2/20@20/4", "0?19") in new stack
    -- Executing Dial("IAX2/20@20/4", "ZAP/g0/484XXX2") in
new stack
    -- Called g0/484XXX2
Destroying call '03896a42587e3f973b42daad031ea6a3@24.XX.XXX.101'
    -- Zap/1-1 answered IAX2/20@20/4
    -- Hungup 'Zap/1-1'
  == Spawn extension (macro-dialout-trunk, s, 17) exited non-zero on
'IAX2/20@20/4' in macro

'dialout-trunk'
  == Spawn extension (from-internal, 484XXX2, 2) exited non-zero on
'IAX2/20@20/4'
    -- Executing Macro("IAX2/20@20/4", "hangupcall") in new
stack
    -- Executing ResetCDR("IAX2/20@20/4", "w") in new stack
    -- Executing NoCDR("IAX2/20@20/4", "") in new stack
    -- Executing Wait("IAX2/20@20/4", "5") in new stack
  == Spawn extension (macro-hangupcall, s, 3) exited non-zero on
'IAX2/20@20/4' in macro

'hangupcall'
  == Spawn extension (from-internal, h, 1) exited non-zero on
'IAX2/20@20/4'
    -- Hungup 'IAX2/20@20/4'



-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20050809/228e5537/attachment.htm

Paul Belanger

2005-Aug-09 15:33 UTC

head link

[Asterisk-Users] SIP-Trunk problem, Please help!!!

Can you see the INVITE if you put up a trace on your gateway 
(209.XXX.XXX.113)?  Asterisk is not getting anything back that is why it 
retransmits 5 times.

PB

OMS wrote:> INVITE sip:1512484XXX2@209.XXX.XXX.113 SIP/2.0
> Via: SIP/2.0/UDP 24.XX.XXX.101:5060;branch=z9hG4bK6adcaf4f
> From: "512538XXX"
<sip:512538XXXX@24.XX.XXX.101>;tag=as5329d8fe
> To: <sip:1512484XXX2@209.XXX.XXX.113>
> Contact: <sip:512538XXX@24.XX.XXX.101>
> Call-ID: 03896a42587e3f973b42daad031ea6a3@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 - - - -

OMS

2005-Aug-09 15:58 UTC

head link

[Asterisk-Users] SIP-Trunk problem, Please help!!!

I just checked again to make sure. I am not seeing anything at all on
gateway on failed calls.
Again 2 out of 5 test calls were failed to reach gateway.



----- Original Message ----- 
From: "Paul Belanger" <pabelanger@codeslingers.ca>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
Sent: Tuesday, August 09, 2005 5:33 PM
Subject: Re: [Asterisk-Users] SIP-Trunk problem, Please help!!!

> Can you see the INVITE if you put up a trace on your gateway
> (209.XXX.XXX.113)?  Asterisk is not getting anything back that is why it
> retransmits 5 times.
>
> PB
>
> OMS wrote:
> > INVITE sip:1512484XXX2@209.XXX.XXX.113 SIP/2.0
> > Via: SIP/2.0/UDP 24.XX.XXX.101:5060;branch=z9hG4bK6adcaf4f
> > From: "512538XXX"
<sip:512538XXXX@24.XX.XXX.101>;tag=as5329d8fe
> > To: <sip:1512484XXX2@209.XXX.XXX.113>
> > Contact: <sip:512538XXX@24.XX.XXX.101>
> > Call-ID: 03896a42587e3f973b42daad031ea6a3@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 - - - -
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
>

OMS

2005-Aug-10 15:56 UTC

head link

[Asterisk-Users] SIP-Trunk problem, Please help!!!

When I put the asterisk server inside NAT,  I do not get any SIP-retransmission
problem, SIP-trunk seems to work on all calls.

Why asterisk is occasionally unable to get SIP packets out to SIP gateway when
connected to PUBLIC IP?

Did some body had this problem before?

  ----- Original Message ----- 
  From: OMS 
  To: Asterisk-Users@lists.digium.com 
  Sent: Tuesday, August 09, 2005 4:45 PM
  Subject: [Asterisk-Users] SIP-Trunk problem, Please help!!!


  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@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@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@20/4", "1?5:8") in new
stack
      -- Goto (macro-record-enable,s,5)
      -- Executing DBget("IAX2/20@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@20/4",
"CALLFILENAME=OUT20-20050809-163643-1123619803.36") in

  new stack
      -- Executing Goto("IAX2/20@20/4", "s|14") in new stack
      -- Goto (macro-record-enable,s,14)
      -- Executing GotoIf("IAX2/20@20/4", "0?15:99") in new
stack
      -- Goto (macro-record-enable,s,99)
      -- Executing NoOp("IAX2/20@20/4", "NO RECORDING
NEEDED") in new stack
      -- Executing GotoIf("IAX2/20@20/4", "0?7") in new
stack
      -- Executing SetCallerID("IAX2/20@20/4", "512538XXX")
in new stack
      -- Executing Goto("IAX2/20@20/4", "9") in new stack
      -- Goto (macro-dialout-trunk,s,9)
      -- Executing SetGroup("IAX2/20@20/4", "OUT_4") in new
stack
      -- Executing CheckGroup("IAX2/20@20/4", "5") in new
stack
      -- Executing SetVar("IAX2/20@20/4",
"DIAL_NUMBER=484XXX2") in new stack
      -- Executing SetVar("IAX2/20@20/4", "DIAL_TRUNK=4") in
new stack
      -- Executing AGI("IAX2/20@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@20/4",
"OUTNUM=1512484XXX2") in new stack
      -- Executing Cut("IAX2/20@20/4", "custom=OUT_4|:|1")
in new stack
      -- Executing GotoIf("IAX2/20@20/4", "0?19") in new
stack
      -- Executing Dial("IAX2/20@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@209.XXX.XXX.113 SIP/2.0
  Via: SIP/2.0/UDP 24.XX.XXX.101:5060;branch=z9hG4bK6adcaf4f
  From: "512538XXX"
<sip:512538XXXX@24.XX.XXX.101>;tag=as5329d8fe
  To: <sip:1512484XXX2@209.XXX.XXX.113>
  Contact: <sip:512538XXX@24.XX.XXX.101>
  Call-ID: 03896a42587e3f973b42daad031ea6a3@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@209.XXX.XXX.113 SIP/2.0
  Via: SIP/2.0/UDP 24.XX.XXX.101:5060;branch=z9hG4bK6adcaf4f
  From: "512538XXX" <sip:512538XXX@24.XX.XXX.101>;tag=as5329d8fe
  To: <sip:1512484XXX2@209.XXX.XXX.113>
  Contact: <sip:512538XXX@24.XX.XXX.101>
  Call-ID: 03896a42587e3f973b42daad031ea6a3@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@209.XXX.XXX.113 SIP/2.0
  Via: SIP/2.0/UDP 24.XX.XXX.101:5060;branch=z9hG4bK6adcaf4f
  From: "512538XXX" <sip:512538XXX@24.XX.XXX.101>;tag=as5329d8fe
  To: <sip:1512484XXX2@209.XXX.XXX.113>
  Contact: <sip:512538XXX@24.XX.XXX.101>
  Call-ID: 03896a42587e3f973b42daad031ea6a3@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@209.XXX.XXX.113 SIP/2.0
  Via: SIP/2.0/UDP 24.XX.XXX.101:5060;branch=z9hG4bK6adcaf4f
  From: "512538XXX" <sip:512538XXX@24.XX.XXX.101>;tag=as5329d8fe
  To: <sip:1512484XXX2@209.XXX.XXX.113>
  Contact: <sip:512538XXX@24.XX.XXX.101>
  Call-ID: 03896a42587e3f973b42daad031ea6a3@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@209.XXX.XXX.113 SIP/2.0
  Via: SIP/2.0/UDP 24.XX.XXX.101:5060;branch=z9hG4bK6adcaf4f
  From: "512538XXX" <sip:512538XXX@24.XX.XXX.101>;tag=as5329d8fe
  To: <sip:1512484XXX2@209.XXX.XXX.113>
  Contact: <sip:512538XXX@24.XX.XXX.101>
  Call-ID: 03896a42587e3f973b42daad031ea6a3@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@209.XXX.XXX.113 SIP/2.0
  Via: SIP/2.0/UDP 24.XX.XXX.101:5060;branch=z9hG4bK6adcaf4f
  From: "512538XXX" <sip:512538XXX@24.XX.XXX.101>;tag=as5329d8fe
  To: <sip:1512484XXX2@209.XXX.XXX.113>
  Contact: <sip:512538XXX@24.XX.XXX.101>
  Call-ID: 03896a42587e3f973b42daad031ea6a3@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@20/4", "s-NOANSWER|1") in
new stack
      -- Goto (macro-dialout-trunk,s-NOANSWER,1)
      -- Executing NoOp("IAX2/20@20/4", "Dial failed due to
NOANSWER") in new stack
      -- Executing Macro("IAX2/20@20/4",
"dialout-trunk|1|484XXX2|") in new stack
      -- Executing GotoIf("IAX2/20@20/4", "1?3:2)") in new
stack
      -- Goto (macro-dialout-trunk,s,3)
      -- Executing Macro("IAX2/20@20/4",
"record-enable|512538XXX|OUT") in new stack
      -- Executing GotoIf("IAX2/20@20/4", "0 > 0?2:4") in
new stack
      -- Goto (macro-record-enable,s,4)
      -- Executing GotoIf("IAX2/20@20/4", "1?5:8") in new
stack
      -- Goto (macro-record-enable,s,5)
      -- Executing DBget("IAX2/20@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@20/4", 

  "CALLFILENAME=OUT512538XXX-20050809-163649-1123619803.36") in new
stack
      -- Executing Goto("IAX2/20@20/4", "s|14") in new stack
      -- Goto (macro-record-enable,s,14)
      -- Executing GotoIf("IAX2/20@20/4", "0?15:99") in new
stack
      -- Goto (macro-record-enable,s,99)
      -- Executing NoOp("IAX2/20@20/4", "NO RECORDING
NEEDED") in new stack
      -- Executing GotoIf("IAX2/20@20/4", "1?7") in new
stack
      -- Goto (macro-dialout-trunk,s,7)
      -- Executing GotoIf("IAX2/20@20/4", "1?9") in new
stack
      -- Goto (macro-dialout-trunk,s,9)
      -- Executing SetGroup("IAX2/20@20/4", "OUT_1") in new
stack
      -- Executing CheckGroup("IAX2/20@20/4", "") in new
stack
      -- Executing SetVar("IAX2/20@20/4",
"DIAL_NUMBER=484XXX2") in new stack
      -- Executing SetVar("IAX2/20@20/4", "DIAL_TRUNK=1") in
new stack
      -- Executing AGI("IAX2/20@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@20/4", "OUTNUM=484XXX2")
in new stack
      -- Executing Cut("IAX2/20@20/4", "custom=OUT_1|:|1")
in new stack
      -- Executing GotoIf("IAX2/20@20/4", "0?19") in new
stack
      -- Executing Dial("IAX2/20@20/4", "ZAP/g0/484XXX2") in
new stack
      -- Called g0/484XXX2
  Destroying call '03896a42587e3f973b42daad031ea6a3@24.XX.XXX.101'
      -- Zap/1-1 answered IAX2/20@20/4
      -- Hungup 'Zap/1-1'
    == Spawn extension (macro-dialout-trunk, s, 17) exited non-zero on
'IAX2/20@20/4' in macro

  'dialout-trunk'
    == Spawn extension (from-internal, 484XXX2, 2) exited non-zero on
'IAX2/20@20/4'
      -- Executing Macro("IAX2/20@20/4", "hangupcall") in
new stack
      -- Executing ResetCDR("IAX2/20@20/4", "w") in new
stack
      -- Executing NoCDR("IAX2/20@20/4", "") in new stack
      -- Executing Wait("IAX2/20@20/4", "5") in new stack
    == Spawn extension (macro-hangupcall, s, 3) exited non-zero on
'IAX2/20@20/4' in macro

  'hangupcall'
    == Spawn extension (from-internal, h, 1) exited non-zero on
'IAX2/20@20/4'
      -- Hungup 'IAX2/20@20/4'






------------------------------------------------------------------------------


  _______________________________________________
  Asterisk-Users mailing list
  Asterisk-Users@lists.digium.com
  http://lists.digium.com/mailman/listinfo/asterisk-users
  To UNSUBSCRIBE or update options visit:
     http://lists.digium.com/mailman/listinfo/asterisk-users
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20050810/831a5c20/attachment.htm

Apparently Analagous Threads

Search for more apparently analagous threads

asterisk users - Aug 2005 - SIP-Trunk problem, Please help!!!

[Asterisk-Users] SIP-Trunk problem, Please help!!!

[Asterisk-Users] SIP-Trunk problem, Please help!!!

[Asterisk-Users] SIP-Trunk problem, Please help!!!

[Asterisk-Users] SIP-Trunk problem, Please help!!!

Apparently Analagous Threads