Hi Pavel,
Thanks for the reply,
I tried with Hangup(16) but same result
what I get in console is
-- Executing [01XXX@public:1] Ringing(SIP/10.2.2.75-0004, ) in
new stack
-- Executing [01XXX@public:2] Wait(SIP/10.2.2.75-0004, 1)
in new stack
-- Executing [01XXX@public:3] Set(SIP/10.2.2.75-0004,
vxmlurl=http%3A%2F%2F127.0.0.1:8080%2Ftest%2Findex.jsp?ani=07XX) in
new stack
-- Executing [01XXX@public:4] AGI(SIP/10.2.2.75-0004, agi://
127.0.0.1/url=http%3A%2F%2F127.0.0.1:8080%2Ftest%2Findex.jsp?ani=07XX)
in new stack
-- SIP/10.2.2.75-0004AGI Script agi://
127.0.0.1/url=http%3A%2F%2F127.0.0.1:8080%2Ftest%2Findex.jsp?ani=07XX
completed,
returning 4
[Feb 13 10:25:24] ERROR[10111][C-0004]: 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:07XX@11.2.2.75;user=phone;tag=sbc0403xbxkbx6w-CC-22
To: sip:01XXX@11.2.2.75;user=phone;tag=as02deb6c4
Call-ID: isbcyc60z60kyw17fk6e1y36i7ey7tz33y3i@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:01XXX@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@SoftX3000
From: sip:07XX@11.2.2.75;user=phone;tag=sbc0403xbxkbx6w-CC-22
To: sip:01XXX@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 pat...@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 = 01,1,Ringing()
exten = 01,2,Wait(2)
exten =
01,3,Set(vxmlurl=http%3A%2F%2F127.0.0.1:8080%2Ftest%2Findex.jsp?ani=${CALLERID(num)})
exten = 01,4,Agi(agi://127.0.0.1/url=${vxmlurl})
exten = 01,5,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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