<div>Hey,</div><div><br></div>So far the Dialplan execution is ok, despite the conflicts and some other mistakes like repeating priorities in it but they're not involved in this call.<div><br></div><div>You'r A-leg is H323 endpoint and Destination is on SIP. I'm now thinking about codec mismatch on first try Tell me this happens every time? Like first call fails for sure and second call goes through?</div>
<div><br></div><div>if so please post the tcpdump/wireshark traces for a failed call as well as successful call separately. Also NAting could be the reason like Rube suspects - but the call should be failing everytime. Anyways post the complete h323 as well as SIP traces combined for each failed & successful call. </div>
<div><br></div><div>Regards.</div><div>-Sammy<br><br><div class="gmail_quote">On Wed, Sep 28, 2011 at 10:59 AM, Malvin Rito <span dir="ltr"><<a href="mailto:mrito@mail.altcladding.com.ph">mrito@mail.altcladding.com.ph</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div bgcolor="#FFFFFF" text="#000000">
Thanks Sam. Please see below CLI log:<br>
<br>
<i>[root@localhost ~]# asterisk -rvvvv<br>
Asterisk 1.6.2.7, Copyright (C) 1999 - 2010 Digium, Inc. and
others.<br>
Created by Mark Spencer <a href="mailto:markster@digium.com" target="_blank"><markster@digium.com></a><br>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show
warranty' for detail <br>
<br>
s.<br>
This is free software, with components licensed under the GNU
General Public<br>
License version 2 and other licenses; you are welcome to
redistribute it under<br>
certain conditions. Type 'core show license' for details.<br>
=========================================================================<br>
== Parsing '/etc/asterisk/asterisk.conf': == Found<br>
Connected to Asterisk 1.6.2.7 currently running on localhost (pid
= 11533)<br>
Verbosity is at least 4<br>
== Spawn extension (avaya-internal, <a href="tel:15707088788" value="+15707088788" target="_blank">15707088788</a>, 2) exited
non-zero on 'OOH323 <br>
<br>
/(null)-b7798910'<br>
-- Executing [s@avaya-internal:1]
Answer("OOH323/(null)-b7798910", "") in
ne <br>
<br>
w stack<br>
-- Executing [s@avaya-internal:2]
BackGround("OOH323/(null)-b7798910",
"pls- <br>
<br>
entr-num-uwish2-call") in new stack<br>
-- <OOH323/(null)-b7798910> Playing
'pls-entr-num-uwish2-call.gsm'
(language <br>
<br>
'en')<br>
== CDR updated on OOH323/(null)-b7798910<br>
-- Executing <a href="tel:%5B15707088788" value="+15707088788" target="_blank">[15707088788</a>@avaya-internal:1]
Authenticate("OOH323/(null)-b779
<br>
<br>
8910", "/etc/asterisk/passcode.txt,a") in new stack<br>
-- <OOH323/(null)-b7798910> Playing 'agent-pass.ulaw'
(language 'en')<br>
-- <OOH323/(null)-b7798910> Playing 'auth-thankyou.ulaw'
(language 'en')<br>
-- Executing <a href="tel:%5B15707088788" value="+15707088788" target="_blank">[15707088788</a>@avaya-internal:2]
Dial("OOH323/(null)-b7798910",
" <br>
<br>
SIP/<a href="tel:15707088788" value="+15707088788" target="_blank">15707088788</a>@cordia") in new stack<br>
== Using SIP RTP TOS bits 184<br>
== Using SIP RTP CoS mark 5<br>
-- Called <a href="tel:15707088788" value="+15707088788" target="_blank">15707088788</a>@cordia<br>
-- SIP/cordia-00000017 answered OOH323/(null)-b7798910<br>
== Spawn extension (avaya-internal, <a href="tel:15707088788" value="+15707088788" target="_blank">15707088788</a>, 2) exited
non-zero on 'OOH323 <br>
<br>
/(null)-b7798910'<br>
-- Executing [s@avaya-internal:1]
Answer("OOH323/(null)-0a389388", "") in
ne <br>
<br>
w stack<br>
-- Executing [s@avaya-internal:2]
BackGround("OOH323/(null)-0a389388",
"pls- <br>
<br>
entr-num-uwish2-call") in new stack<br>
-- <OOH323/(null)-0a389388> Playing
'pls-entr-num-uwish2-call.gsm'
(language <br>
<br>
'en')<br>
-- Executing [s@avaya-internal:3]
WaitExten("OOH323/(null)-0a389388", "")
in <br>
<br>
new stack<br>
== CDR updated on OOH323/(null)-0a389388<br>
-- Executing <a href="tel:%5B18772281023" value="+18772281023" target="_blank">[18772281023</a>@avaya-internal:1]
Authenticate("OOH323/(null)-0a38
<br>
<br>
9388", "/etc/asterisk/passcode.txt,a") in new stack<br>
-- <OOH323/(null)-0a389388> Playing 'agent-pass.ulaw'
(language 'en')<br>
-- <OOH323/(null)-0a389388> Playing 'auth-thankyou.ulaw'
(language 'en')<br>
-- Executing <a href="tel:%5B18772281023" value="+18772281023" target="_blank">[18772281023</a>@avaya-internal:2]
Dial("OOH323/(null)-0a389388",
" <br>
<br>
SIP/<a href="tel:18772281023" value="+18772281023" target="_blank">18772281023</a>@cordia") in new stack<br>
== Using SIP RTP TOS bits 184<br>
== Using SIP RTP CoS mark 5<br>
-- Called <a href="tel:18772281023" value="+18772281023" target="_blank">18772281023</a>@cordia<br>
-- SIP/cordia-00000018 is making progress passing it to
OOH323/(null)-0a3893 <br>
<br>
88<br>
-- SIP/cordia-00000018 is ringing<br>
-- SIP/cordia-00000018 is making progress passing it to
OOH323/(null)-0a3893 <br>
<br>
88<br>
-- SIP/cordia-00000018 answered OOH323/(null)-0a389388<br>
== Spawn extension (avaya-internal, <a href="tel:18772281023" value="+18772281023" target="_blank">18772281023</a>, 2) exited
non-zero on 'OOH323 <br>
<br>
/(null)-0a389388'<br>
== Manager 'admin' logged on from 127.0.0.1<br>
== Manager 'admin' logged off from 127.0.0.1<br>
== Manager 'admin' logged on from 127.0.0.1<br>
== Manager 'admin' logged off from 127.0.0.1<br>
localhost*CLI><br>
Disconnected from Asterisk server<br>
Executing last minute cleanups<br>
[root@localhost ~]# nano /etc/asterisk/extensions_custom.conf<br>
[root@localhost ~]# asterisk -rvvvv<br>
Asterisk 1.6.2.7, Copyright (C) 1999 - 2010 Digium, Inc. and
others.<br>
Created by Mark Spencer <a href="mailto:markster@digium.com" target="_blank"><markster@digium.com></a><br>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show
warranty' for details.<br>
This is free software, with components licensed under the GNU
General Public<br>
License version 2 and other licenses; you are welcome to
redistribute it under<br>
certain conditions. Type 'core show license' for details.<br>
=========================================================================<br>
== Parsing '/etc/asterisk/asterisk.conf': == Found<br>
Connected to Asterisk 1.6.2.7 currently running on localhost (pid
= 11533)<br>
Verbosity is at least 4<br>
-- Remote UNIX connection<br>
localhost*CLI> sip set debug peer cordia<br>
SIP Debugging Enabled for IP: <a href="http://66.148.120.167:5060" target="_blank">66.148.120.167:5060</a><br>
localhost*CLI> core set verbose 0<br>
Verbosity is now OFF<br>
Audio is at 192.168.254.15 port 19144<br>
Adding codec 0x4 (ulaw) to SDP<br>
Adding codec 0x8 (alaw) to SDP<br>
Reliably Transmitting (no NAT) to <a href="http://66.148.120.167:5060" target="_blank">66.148.120.167:5060</a>:<br>
INVITE <a>sip:15707088788@66.148.120.167</a> SIP/2.0<br>
Via: SIP/2.0/UDP 192.168.254.15:5060;branch=z9hG4bK1e0698f8;rport<br>
Max-Forwards: 70<br>
From: "10.1.129.247"
<a><sip:1105@192.168.254.15></a>;tag=as4f38e456<br>
To: <a><sip:15707088788@66.148.120.167></a><br>
Contact: <a><sip:1105@192.168.254.15></a><br>
Call-ID: <a href="mailto:1a2d18961fc1c50a50ecad427e9f350c@192.168.254.15" target="_blank">1a2d18961fc1c50a50ecad427e9f350c@192.168.254.15</a><br>
CSeq: 102 INVITE<br>
User-Agent: Asterisk PBX 1.6.2.7<br>
Date: Wed, 28 Sep 2011 02:43:24 GMT<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY, INFO<br>
Supported: replaces, timer<br>
Content-Type: application/sdp<br>
Content-Length: 235<br>
<br>
v=0<br>
o=root <a href="tel:2082067001" value="+12082067001" target="_blank">2082067001</a> <a href="tel:2082067001" value="+12082067001" target="_blank">2082067001</a> IN IP4 192.168.254.15<br>
s=Asterisk PBX 1.6.2.7<br>
c=IN IP4 192.168.254.15<br>
t=0 0<br>
m=audio 19144 RTP/AVP 0 8<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:8 PCMA/8000<br>
a=silenceSupp:off - - - -<br>
a=ptime:20<br>
a=sendrecv<br>
<br>
---<br>
localhost*CLI><br>
<--- SIP read from UDP:<a href="http://66.148.120.167:5060" target="_blank">66.148.120.167:5060</a> ---><br>
SIP/2.0 100 Trying<br>
CSeq: 102 INVITE<br>
Via: SIP/2.0/UDP 222.127.178.113:5060;branch=z9hG4bK1e0698f8;rport<br>
From: "10.1.129.247"
<a><sip:1105@192.168.254.15></a>;tag=as4f38e456<br>
Call-ID: <a href="mailto:1a2d18961fc1c50a50ecad427e9f350c@192.168.254.15" target="_blank">1a2d18961fc1c50a50ecad427e9f350c@192.168.254.15</a><br>
To:
<a><sip:15707088788@66.148.120.167></a>;tag=28091911111014129494936357<br>
Contact: <a><sip:66.148.120.167:5060;transport=udp></a><br>
Content-Length: 0<br>
<br>
<br>
<-------------><br>
--- (8 headers 0 lines) ---<br>
localhost*CLI><br>
<--- SIP read from UDP:<a href="http://66.148.120.167:5060" target="_blank">66.148.120.167:5060</a> ---><br>
SIP/2.0 200 OK<br>
CSeq: 102 INVITE<br>
Via: SIP/2.0/UDP 222.127.178.113:5060;branch=z9hG4bK1e0698f8;rport<br>
From: "10.1.129.247"
<a><sip:1105@192.168.254.15></a>;tag=as4f38e456<br>
Call-ID: <a href="mailto:1a2d18961fc1c50a50ecad427e9f350c@192.168.254.15" target="_blank">1a2d18961fc1c50a50ecad427e9f350c@192.168.254.15</a><br>
To:
<a><sip:15707088788@66.148.120.167></a>;tag=28091911111014129494936357<br>
Contact: <a><sip:66.148.120.167:5060;transport=udp></a><br>
Content-Type: application/sdp<br>
Content-Length: 225<br>
<br>
v=0<br>
o=VoipSwitch 6356 7356 IN IP4 66.148.120.167<br>
s=VoipSIP<br>
i=Audio Session<br>
c=IN IP4 66.148.120.167<br>
t=0 0<br>
m=audio 6356 RTP/AVP 0 101<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-15<br>
a=sendrecv<br>
<br>
<-------------><br>
--- (9 headers 11 lines) ---<br>
Found RTP audio format 0<br>
Found RTP audio format 101<br>
Found audio description format PCMU for ID 0<br>
Found audio description format telephone-event for ID 101<br>
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4
(ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4
(ulaw)<br>
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1
(telephone-event), combined - 0x0 (nothing)<br>
Peer audio RTP is at port <a href="http://66.148.120.167:6356" target="_blank">66.148.120.167:6356</a><br>
list_route: hop: <a><sip:66.148.120.167:5060;transport=udp></a><br>
set_destination: Parsing
<a><sip:66.148.120.167:5060;transport=udp></a> for address/port to
send to<br>
set_destination: set destination to 66.148.120.167, port 5060<br>
Transmitting (no NAT) to <a href="http://66.148.120.167:5060" target="_blank">66.148.120.167:5060</a>:<br>
ACK <a>sip:66.148.120.167:5060;transport=udp</a> SIP/2.0<br>
Via: SIP/2.0/UDP 192.168.254.15:5060;branch=z9hG4bK28aa746f;rport<br>
Max-Forwards: 70<br>
From: "10.1.129.247"
<a><sip:1105@192.168.254.15></a>;tag=as4f38e456<br>
To:
<a><sip:15707088788@66.148.120.167></a>;tag=28091911111014129494936357<br>
Contact: <a><sip:1105@192.168.254.15></a><br>
Call-ID: <a href="mailto:1a2d18961fc1c50a50ecad427e9f350c@192.168.254.15" target="_blank">1a2d18961fc1c50a50ecad427e9f350c@192.168.254.15</a><br>
CSeq: 102 ACK<br>
User-Agent: Asterisk PBX 1.6.2.7<br>
Content-Length: 0<br>
<br>
<br>
---<br>
localhost*CLI></i><br>
<br>
Regards,<br><font color="#888888">
Malvin</font><div><div></div><div class="h5"><br>
<br>
On 9/28/2011 1:46 PM, Sam Govind wrote:
<blockquote type="cite">I see a couple of conflicting extensions as well as
something I assume copy-paste malfunction. Please paste the CLI
logs of the call. <br>
<br>
<div class="gmail_quote">On Wed, Sep 28, 2011 at 8:26 AM, Malvin
Rito <span dir="ltr"><<a href="mailto:mrito@mail.altcladding.com.ph" target="_blank">mrito@mail.altcladding.com.ph</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div bgcolor="#FFFFFF" text="#000000"> Thanks All. Here is my
config:<br>
<br>
<b>On my Firewall NAT:</b><br>
<br>
<i>I allowed the following ports: 4569,5004-5082,
10000-20000</i><br>
<b><br>
On Asterisk Box:</b><br>
<br>
Here is the extensions.conf:<br>
<i>[general]<br>
static=yes<br>
autofallthrough=yes<br>
<br>
[avaya-internal]<br>
exten => s,1,Answer()<br>
exten => s,2,background(pls-entr-num-uwish2-call)<br>
exten => s,3,WaitExten()<br>
exten => s,4,Dial(SIP/${EXTEN})<br>
exten => s,5,Dial(H323/${EXTEN})<br>
exten => s,6,PlayBack(vm-nobodyavail)<br>
exten => s,7,HangUp()<br>
<br>
exten => 1000,1,Dial(SIP/1000)<br>
exten => 1000,1,Answer()<br>
<br>
exten => 1000,2,PlayBack(vm-goodbye)<br>
exten => 1000,3,HangUp()<br>
<br>
#Extension for recording<br>
exten => 9000,1,Answer()<br>
exten => 9000,2,Background(pm-to-record-phrase)<br>
exten => 9000,3,Hangup()<br>
#exten => 9000,3,Wait(2)<br>
exten => 9000,4,Record(alt_recording%d:ulaw)<br>
exten => 9000,5,Wait(2)<br>
exten => 9000,6,Playback(${RECORDED_FILE})<br>
exten => 9000,7,Wait(2)<br>
exten => 9000,8,Hangup<br>
<br>
exten => _XXXX,1,Dial(SIP/${EXTEN}@Avaya)<br>
exten => _11XX,1,Dial(H323/${EXTEN}@Avaya)<br>
<br>
exten =>
_XXXXXXXXXXXXX,1,Authenticate(/etc/asterisk/passcode.txt,a)<br>
exten => _XXXXXXXXXXXXX,2,Dial(SIP/${EXTEN}@cordia)<br>
<br>
exten =>
_XXXXXXXXXXXX,1,Authenticate(/etc/asterisk/passcode.txt,a)<br>
exten => _XXXXXXXXXXXX,2,Dial(SIP/${EXTEN}@cordia)</i><br>
<br>
<br>
<br>
Regards,<br>
<font color="#888888"> Malvin</font>
<div>
<div><br>
<br>
On 9/26/2011 9:56 PM, Ruben Rögels wrote:
<blockquote type="cite">
<pre>Am <a href="tel:26.09.2011%2013" value="+12609201113" target="_blank">26.09.2011 13</a>:18, schrieb Malvin Rito:
</pre>
<blockquote type="cite">
<pre>Hi list,
My call does not pass through on the first dial, I have to redial again
the number for the call to pass through. I'm not sure if the problem is
on my voip proovider or my asterisk server setup. Any thoughts pls?
Regards,
Malvin
</pre>
</blockquote>
<pre>Hi,
could be a NAT related issue.
Please be more specific about your setup.
best regards,
Ruben
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