[asterisk-dev] Reject incoming call

[Raj Roy Ghandhi]

Hi Pavel,
Thanks for the reply,
I tried with Hangup(16) but same result

what I get in console is

 -- Executing [01XXXXXXX at public:1] Ringing("SIP/10.2.2.75-00000004", "") in
new stack
    -- Executing [01XXXXXXX at public:2] Wait("SIP/10.2.2.75-00000004", "1")
in new stack
    -- Executing [01XXXXXXX at public:3] Set("SIP/10.2.2.75-00000004",
"vxmlurl=http%3A%2F%2F127.0.0.1:8080%2Ftest%2Findex.jsp?ani=07XXXXXX") in
new stack
    -- Executing [01XXXXXXX at public:4] AGI("SIP/10.2.2.75-00000004", "agi://
127.0.0.1/url=http%3A%2F%2F127.0.0.1:8080%2Ftest%2Findex.jsp?ani=07XXXXXX")
in new stack
    -- <SIP/10.2.2.75-00000004>AGI Script agi://
127.0.0.1/url=http%3A%2F%2F127.0.0.1:8080%2Ftest%2Findex.jsp?ani=07XXXXXX
completed,
returning 4
[Feb 13 10:25:24] ERROR[10111][C-00000004]: utils.c:1321 ast_carefulwrite:
write() returned error: Broken pipe


<--- Reliably Transmitting (no NAT) to 11.200.1.53:9131 --->
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 11.200.1.53:9131
;branch=z9hG4bKkcw3bikyw336f07x6xt6wbbc6;received=10.200.1.53
From: <sip:07XXXXXX at 11.2.2.75;user=phone>;tag=sbc0403xbxkbx6w-CC-22
To: <sip:01XXXXXXX at 11.2.2.75;user=phone>;tag=as02deb6c4
Call-ID: isbcyc60z60kyw17fk6e1y36i7ey7tz33y3i at SoftX3000
CSeq: 1 INVITE
Server: Asterisk PBX 11.8.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:11.200.1.53:9131 --->
ACK sip:01XXXXXXX at 11.2.2.75:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 11.200.1.53:9131
;branch=z9hG4bKkcw3bikyw336f07x6xt6wbbc6;received=11.200.1.53
Call-ID: isbcyc60z60kyw17fk6e1y36i7ey7tz33y3i at SoftX3000
From: <sip:07XXXXXX at 11.2.2.75;user=phone>;tag=sbc0403xbxkbx6w-CC-22
To: <sip:01XXXXXXX at 11.2.2.75;user=phone>;tag=as02deb6c4
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0

I am running Asterisk 11.8.0 with voiceglue 0.14

When I try the .jsp page with wget, it works fine without any error.

Regards,
Roy.


On Thu, Feb 12, 2015 at 4:58 PM, Pavel Troller <patrol at sinus.cz> wrote:

> Hi Roy,
>
> > Hi Friends,
> >
> > I am trying to implement a simple dial plan with asterisk.
> > 1. Ring the inbound call
> > 2. wait for 2 seconds
> > 3. call agi script with cli
> > 4. hangup
> >
> > But when it gets hangup I see the Declined is passed from the asterisk.
> > But my PSTN provider keep waiting in dialing state with no noise until 60
> > seconds.
> >
> > any idea what I have done wrong.
> >
> > here is my dial plan
> >
> > exten => 01XXXX,1,Ringing()
> > exten => 01XXXX,2,Wait(2)
> > exten =>
> >
> 01XXXX,3,Set(vxmlurl=http%3A%2F%2F127.0.0.1:8080%2Ftest%2Findex.jsp?ani=${CALLERID(num)})
> > exten => 01XXXX,4,Agi(agi://127.0.0.1/url=${vxmlurl})
> > exten => 01XXXX,5,Hangup()
>
> Did you try to put a specific Clear Cause code to the Hangup command ?
> For example, Hangup(17) means User Busy, i.e. 486 Busy Here on SIP, or
> Hangup(1) means Unallocated Number or 404 Not Found etc. If you want to
> find more codes, search for ISDN Cause Codes. Your PSTN provider will
> probably react better to other causes than the default one, which is
> probably 16 Normal Call Clearing.
>
> With regards,
>   Pavel
>
> >
> > Please advice,
> >
> > Best Regards,
> > Roy.
>
>
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