<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta content="text/html;charset=ISO-8859-1" http-equiv="Content-Type">
</head>
<body bgcolor="#ffffff" text="#000000">
Hi All<br>
<br>
Agter a bit of logging to a syslog server, I found a peculiar entry
today, ironically right after a call failed to transfer. They key
sequence and call path used until it gets transferred is as follows<br>
<ul>
<li>Phone rings on Asterisk</li>
<li>Asterisk transferres to the receptionists phone (GXP 2000)<br>
</li>
<ul>
<li>Receptionist doensnt answer for 15 seconds, and the call gets
routed to the bosses secrataries phone</li>
</ul>
<li>Bosses secretary answers the phone and tries to transfer it to
the boss with the keysequence "flash", extention "315", talks,
"transfer" but the transfer is the one that fails</li>
</ul>
Message in the log is <br>
<blockquote type="cite">Feb 22 09:55:22 10.219.127.102 GS_LOG:
[00:0B:82:13:02:CF][000][FFFD][01010412] Received SIP message: 407<br>
Feb 22 09:55:22 10.219.127.102 GS_LOG:
[00:0B:82:13:02:CF][000][FFFD][01010412] SIP dialog matched to channel 0<br>
Feb 22 09:55:22 10.219.127.102 GS_LOG:
[00:0B:82:13:02:CF][000][FFFD][01010412] Send SIP message: ACK To
10.219.127.7:5060, sip_handle: 0x0046F09C<br>
Feb 22 09:55:22 10.219.127.102 GS_LOG:
[00:0B:82:13:02:CF][000][FFFD][01010412] sip_len: 553, sip_handle:
0x0046F09C, ACK <a class="moz-txt-link-freetext" href="sip:00123463409@10.219.127.7;user=p">sip:00123463409@10.219.127.7;user=p</a><br>
hone SIP/2.0 Via: SIP/2.0/UDP
10.219.127.102:5060;branch=z9hG4bK623e473ec5e8c5e8 From: "Wanda"
<a class="moz-txt-link-rfc2396E" href="sip:312@10.219.127.7;user=phone"><sip:312@10.219.127.7;user=phone></a>;tag=1a7b934ecd3e23f7 To:<br>
<a class="moz-txt-link-rfc2396E" href="sip:00123463409@10.219.127.7;user=phone"><sip:00123463409@10.219.127.7;user=phone></a>;tag=as07aa3c42
Contact: <a class="moz-txt-link-rfc2396E" href="sip:312@10.219.127.102:5060;transport=udp;user=phone"><sip:312@10.219.127.102:5060;transport=udp;user=phone></a>
Supported: path Call-ID: 1138f5f7<br>
<a class="moz-txt-link-abbreviated" href="mailto:40c8d06c@10.219.127.102">40c8d06c@10.219.127.102</a> CSeq: 58150 ACK User-Agent: Grandstream BT200
1.1.4.18 Max-Forwards: 70 Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SU<br>
BSCRIBE,UPDATE,PRACK Content-Length: 0<br>
</blockquote>
<br>
The message that got my worried is the one saying "Recieved SIP message
407", can that be the ghost I am looking for?<br>
<br>
Extentions.conf<br>
[incoming_calls]<br>
exten => s,1,NoOp(${CALLERID(name)} skakel Luzaan)<br>
exten => s,n,dial(SIP/300,15)<br>
exten => s,n,Set(CALLERID(name)=deur)<br>
exten => s,n,Set(CALLERID(num)=deur)<br>
exten => s,n,NoOp(${CALLERID(name)} skakel Wanda)<br>
exten => s,n,dial(SIP/312)<br>
;exten => s,n,dial(SIP/317)<br>
exten => s,n,Hangup()<br>
<br>
[internal]<br>
exten => 900,1,Verbose(1|Echo test application)<br>
exten => 900,n,Echo()<br>
exten => 900,n,Hangup()<br>
;interne oproepe<br>
<br>
exten => _3XX,1,NoOp(${CALLERID} skakel ${EXTEN})<br>
exten => _3XX,n,Dial(SIP/${EXTEN},30)<br>
;exten => _3XX,n,execif(${CALLERID} != _3XX|goto|incoming_calls/s/1)<br>
exten => _3XX,n,goto(incoming_calls,s,1)<br>
exten => _3XX,n,Hangup()<br>
<br>
Sip.conf<br>
[general]<br>
context=default<br>
allowoverlap=no<br>
bindport=5060<br>
bindaddr=0.0.0.0<br>
srvlookup=yes<br>
limitonpeers=yes<br>
allowtransfer=yes<br>
callevents=yes<br>
regcontext=GXP_BLF<br>
<br>
[sets](!)<br>
type=friend<br>
context=internal<br>
host=dynamic<br>
;disallow=all<br>
;allow=speex<br>
secret=test<br>
dtmfmode=info<br>
callgroup=1<br>
pickupgroup=1<br>
call-limit=20<br>
subscribecontext=GXP_BLF<br>
canreinvite=yes<br>
nat=no<br>
<br>
[300](sets) ;Luzaan<br>
regexten=300<br>
<br>
Any help will be apreciated<br>
<br>
Thanks<br>
Ian<br>
<br>
Ian said the following on 22-Feb-08 10:06 AM:
<blockquote cite="mid:47BE8273.3090204@iancoetzee.za.net" type="cite">
<meta content="text/html;charset=ISO-8859-1" http-equiv="Content-Type">
<title></title>
Hi,<br>
<br>
Mojo with Horan & Company, LLC said the following on 20-Feb-08
09:31 PM:
<blockquote cite="mid:47BC8015.6060304@horanappraisals.com"
type="cite">
<pre wrap="">Is it AFTER you have parked a call? Meaning, for example, you transfer
an incoming call to 700. No problem. Later, when it's picked up from
701, can it NOT be transferred again?
Moj
</pre>
</blockquote>
No I don't park the call.<br>
<br>
The call comes in, and gets redirected to our receptionists phone, from
there it gets transferred to another extension (the bosses secratary)
and then gets transferred (to the boss). now the problem, sometimes
that transfer fails, other times the call dont even want to leave the
receptionists phone.<br>
<br>
The big thing about this problem is that it comes and goes, like
yesterday we didn't have a problem, and I did not change a thing.<br>
<br>
Ian<br>
<blockquote cite="mid:47BC8015.6060304@horanappraisals.com"
type="cite">
<pre wrap="">Ian wrote:
</pre>
<blockquote type="cite">
<pre wrap="">Hi All
Sorry to be a bother again but seems like I just cant get away from
the problems.
This time my problem is that *sometimes* a user cant transfer a call
from one extension to another, I have narrowed down the problem to it
only happening to calls from outside the internal system.
The wierd thing about the problem is that it comes and goes one moment
the user can transfer, and the next call he can't.
I am running:
* Asterisk 1.4.17
* Zaptel 1.4.7.1
* Libpri 1.4.3
Using the following phones and firmware
* Grandstream GXP2000 (with ext pad) : 1.1.4.14
* Grandstream BT200 : 1.1.4.18
I have set up the phones to log debug logs to a syslog server, I am
still trying to figure out what exactly the log says.
Is it an * problem, or Grandstream problem
Does anyone know if I am able to see the keysequence the user types
into the phone (just in case it might even be a user made problem), I
have tried scanning though the logs of a failed call, but could not
see any lines that can be a keypress, or maybe I am looking in the
incorrect spot?
Your help will be greatly appreciated.
Let me know if, in any way, I can shed some more light on the subject.
Thanks in advance
Ian
--
<a moz-do-not-send="true" class="moz-txt-link-abbreviated"
href="http://www.vddi.co.za">www.vddi.co.za</a> <a
moz-do-not-send="true" class="moz-txt-link-rfc2396E"
href="http://www.vddi.co.za/"><http://www.vddi.co.za/></a>
I Coetzee
IT Tegnikus
Telefoon         :         012 664 2300
Selfoon         :         079 522 6519
Faks         :         012 644 2902
E-pos         :         <a moz-do-not-send="true" class="moz-txt-link-abbreviated"
href="mailto:ian@vddi.co.za">ian@vddi.co.za</a>
Skype         :         vddb_igcoetzee
------------------------------------------------------------------------
_______________________________________________
-- Bandwidth and Colocation Provided by <a moz-do-not-send="true"
class="moz-txt-link-freetext" href="http://www.api-digital.com">http://www.api-digital.com</a> --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
<a moz-do-not-send="true" class="moz-txt-link-freetext"
href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a>
</pre>
</blockquote>
<pre wrap=""><!---->
_______________________________________________
-- Bandwidth and Colocation Provided by <a moz-do-not-send="true"
class="moz-txt-link-freetext" href="http://www.api-digital.com">http://www.api-digital.com</a> --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
<a moz-do-not-send="true" class="moz-txt-link-freetext"
href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a>
</pre>
</blockquote>
<br>
<pre wrap="">
<hr size="4" width="90%">
_______________________________________________
-- Bandwidth and Colocation Provided by <a class="moz-txt-link-freetext" href="http://www.api-digital.com">http://www.api-digital.com</a> --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
<a class="moz-txt-link-freetext" href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a></pre>
</blockquote>
<br>
</body>
</html>