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David Brazier

2006-Feb-02 05:27 UTC

[Asterisk-Users] Outbound Call & SIP Results

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<BODY>
<DIV id=idOWAReplyText35301 dir=ltr>
<DIV dir=ltr><FONT face=Arial size=2><SPAN
class=046321023-01022006>We make outbound calls via our Asterisk (*1), then
via SIP to&nbsp;a 3rd party Asterisk (*2), which then routes to PSTN, again
via SIP.&nbsp; If the called number is invalid or out of service, *2
gets&nbsp;a&nbsp;404 Not Found, which seems appropriate.&nbsp;
However, *2 then passes on a 403 Forbidden to *1, which is not really the right
response. &nbsp;*1&nbsp;then returns a 486 Busy Here, which also seems
wrong, as it would generally mean the called number was busy (engaged), I
think.</SPAN></FONT></DIV></DIV>
<DIV>
<DIV><FONT face=Arial size=2><SPAN
class=046321023-01022006></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=Arial size=2><SPAN class=046321023-01022006>Is
it possible to vary this behaviour?&nbsp; The ${DIALSTATUS} variable
doesn't seem to be fine-grained enough to
help.</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN
class=046321023-01022006></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=Arial size=2><SPAN
class=046321023-01022006>Thanks for any
thoughts,</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN
class=046321023-01022006></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=Arial size=2><SPAN
class=046321023-01022006>David</SPAN></FONT></DIV></DIV></BODY></HTML>

Olle E Johansson

2006-Feb-02 05:49 UTC

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[Asterisk-Users] Outbound Call & SIP Results

David Brazier wrote:> We make outbound calls via our Asterisk (*1), then via SIP to a 3rd 
> party Asterisk (*2), which then routes to PSTN, again via SIP.  If the 
> called number is invalid or out of service, *2 gets a 404 Not Found, 
> which seems appropriate.  However, *2 then passes on a 403 Forbidden to 
> *1, which is not really the right response.  *1 then returns a 486 Busy 
> Here, which also seems wrong, as it would generally mean the called 
> number was busy (engaged), I think.
>  
> Is it possible to vary this behaviour?  The ${DIALSTATUS} variable 
> doesn't seem to be fine-grained enough to help.
>  Would it be possible to see som log files?


/O

David Brazier

2006-Feb-02 07:14 UTC

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[Asterisk-Users] RE: Outbound Call & SIP Results

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<P>Here is part of the log from *1, showing a 403 received and 486 passed
on (IP addresses, host names and telephone number
changed):<BR><BR>--- (10 headers 0 lines)---<BR><-- SIP
read from ip.of.asterisk.2:5060:<BR>SIP/2.0 403 Forbidden<BR>Via:
SIP/2.0/UDP ip.of.asterisk.1:5060;branch=z9hG4bK596fad3e<BR>From:
"1000" <sip:12345@ip.of.asterisk.1>;tag=as46f716d4<BR>To:
<sip:01234567890@asterisk.2.com>;tag=as0474d5e5<BR>Call-ID: <A
href="mailto:049978b95e32737b4f61c07856176f87@ip.of.asterisk.1"
target=_blank>049978b95e32737b4f61c07856176f87@ip.of.asterisk.1</A><BR>CSeq:
103 INVITE<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK,
CANCEL, OPTIONS, BYE, REFER<BR>Contact:
<sip:01234567890@ip.of.asterisk.2><BR>Content-Length:
0<BR><BR>--- (10 headers 0 lines)---<BR>Transmitting (no NAT)
to ip.of.asterisk.2:5060:<BR>ACK sip:01234567890@asterisk.2.com
SIP/2.0<BR>Via: SIP/2.0/UDP
ip.of.asterisk.1:5060;branch=z9hG4bK596fad3e;rport<BR>From:
"1000" <sip:12345@ip.of.asterisk.1>;tag=as46f716d4<BR>To:
<sip:01234567890@asterisk.2.com>;tag=as0474d5e5<BR>Contact:
<sip:12345@ip.of.asterisk.1><BR>Call-ID: <A
href="mailto:049978b95e32737b4f61c07856176f87@ip.of.asterisk.1"
target=_blank>049978b95e32737b4f61c07856176f87@ip.of.asterisk.1</A><BR>CSeq:
103 ACK<BR>User-Agent: Asterisk PBX<BR>Max-Forwards:
70<BR>Content-Length: 0<BR><BR>---<BR>Feb&nbsp; 1
14:48:12 WARNING[1436]: chan_sip.c:9532 handle_response_invite: Forbidden -
wrong password on authentication for INVITE to '"1000"
<sip:12345@ip.of.asterisk.1>;tag=as46f716d4'<BR>&nbsp;&nbsp;&nbsp;
-- SIP/sip_channel-a404 is circuit-busy<BR>&nbsp; == Everyone is
busy/congested at this time (1:0/1/0)<BR>&nbsp;&nbsp;&nbsp; --
Executing Busy("SIP/vcs-7b2e", "") in new
stack<BR>Transmitting (no NAT) to ip.of.client:5060:<BR>SIP/2.0 486
Busy Here<BR>Via: SIP/2.0/UDP
ip.of.client:5060;received=ip.of.client<BR>From:
<sip:unknown@0.0.0.0>;tag=d6e314d4-13c4-43e0ca2b-39df1197-4823<BR>To:
<sip:01234567890@asterisk.1.com>;tag=as2d609cdc<BR>Call-ID:
d6e314d4-13c4-43e0ca2b-39df1197-29-1a214c0<BR>CSeq: 1
INVITE<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL,
OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<BR>Contact:
<sip:01234567890@ip.of.asterisk.1><BR>Content-Length:
0<BR>X-Asterisk-HangupCause: Call Rejected</P>
<P>The dial plan is a basic:</P>
<P>exten ==> _0Z.,1,Dial(SIP/sip_channel/${EXTEN},30,j)<BR>exten
==> _0Z.,2,Congestion<BR>exten ==> _0Z.,102,Busy</P>
<P>David</P>
<P>&nbsp;</P>
<P>&nbsp;</P></BODY></HTML>

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