(NOT)Why does Asterisk answer (connect) all inbound calls?

Asterisk Asterisk Support
1

—EDIT : This question is pointless since the problem actually never happened… Please see my last post and accept my apologies ----

Hi,

I just connected my AsteriskNow box to an inbound DID. Everything works well, except that I noticed that the inbound calls were immediately connected, before the destination ext actually answer the call. The drawback to this is that the caller gets charged even if the callee doesn’t answer…

Is it a normal behaviour ? i’m used to ISDN networks which issue a RING information (Q.923 protocol) before the actual CONNECT.

Is there a way to keep this to happen ?

Thanks

2

It is not normal behaviour for Asterisk. It may be normal behavour for AsteriskNow, but this is not the correct forum for AsteriskNow.

3

Thanks…

Would you have a look to this, even if it is an asterisknow install ?

[ext-did-0002]
include => ext-did-0002-custom
exten => fax,1,Goto(ext-fax,in_fax,1)
exten => 028809326,1,Set(__FROM_DID=${EXTEN})
exten => 028809326,n,ExecIf($[ “${CALLERID(name)}” = “” ] ,Set,CALLERID(name)=${CALLERID(num)})
exten => 028809326,n,Ringing()
exten => 028809326,n,Set(_CALLINGPRES_SV=${CALLINGPRES${CALLINGPRES}})
exten => 028809326,n,SetCallerPres(allowed_not_screened)
exten => 028809326,n,Goto(from-did-direct,200,1)

[from-trunk-sip-3STARS]
include => from-trunk-sip-3STARS-custom
exten => _.,1,Set(GROUP()=OUT_3)
exten => _.,n,Goto(from-trunk,${EXTEN},1)

Thanks

Pagaille

4

Ringing() is redundant. If it is actually needed, it will probably answer the call. The call should already be in a ringing state.

The dialplan that this leads to could also answer the call.

You definitely need to provide verbose console output and you may need to provide dahdi debug level output.

5

David,

1000 times sorry. Due to a misinterpretation of my GSM’s indications, I believed that the call was connected. In fact, it was definitvely not, as you can see in this SIP trace I learned to read in the meanwhile :

[code]<------------->

dialparties.agi: Starting New Dialparties.agi
== Parsing ‘/etc/asterisk/manager.conf’: Found
== Parsing ‘/etc/asterisk/manager_additional.conf’: Found
== Parsing ‘/etc/asterisk/manager_custom.conf’: Found
== Manager ‘admin’ logged on from 127.0.0.1
dialparties.agi: Caller ID name is ‘0495267897’ number is '0495267897’
dialparties.agi: Methodology of ring is ‘none’
– dialparties.agi: Added extension 200 to extension map
– dialparties.agi: Extension 200 cf is disabled
– dialparties.agi: Extension 200 do not disturb is disabled
– dialparties.agi: dbset CALLTRACE/200 to 0495267897
– dialparties.agi: Filtered ARG3: 200
== Manager ‘admin’ logged off from 127.0.0.1
– AGI Script dialparties.agi completed, returning 0
– Executing [s@macro-dial:7] Dial(“SIP/3STARS-096615f0”, “SIP/200|30|Ttr”) in new stack
Audio is at myOwnIPadress port 10018
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to myOwnIPadress:9999:
INVITE sip:200@myOwnIPadress:9999 SIP/2.0
Via: SIP/2.0/UDP myOwnIPadress:5060;branch=z9hG4bK34dfa9db;rport
From: “0495267897” sip:0495267897@myOwnIPadress;tag=as4f8c94cd
To: sip:200@myOwnIPadress:9999
Contact: sip:0495267897@myOwnIPadress
Call-ID: 396c9c7762dfd21c7e87b9252bcdc1b2@myOwnIPadress
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 10 Jul 2009 22:18:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 242

v=0
o=root 2716 2716 IN IP4 myOwnIPadress
s=session
c=IN IP4 myOwnIPadress
t=0 0
m=audio 10018 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

-- Called 200

roberto*CLI>
<— Transmitting (NAT) to myProviderIP:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP myProviderIP;branch=z9hG4bK378e.c8940a5.0;received=myProviderIP
Via: SIP/2.0/UDP myProviderIP1:5060;received=myProviderIP1;branch=z9hG4bK65023175;rport=5060
Record-Route: sip:myProviderIP;lr=on;ftag=as6baf07f4
From: “0495267897” sip:0495267897@myProviderIP;tag=as6baf07f4
To: sip:028809326@myProviderIP;tag=as5b7c081d
Call-ID: 2951844526849c910e1d8bff5fb52b17@myProviderIP
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: sip:028809326@myOwnIPadress
Content-Length: 0

<------------>
roberto*CLI>
<— SIP read from myOwnIPadress:9999 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP myOwnIPadress:5060;rport;branch=z9hG4bK34dfa9db
From: “0495267897” sip:0495267897@myOwnIPadress;tag=as4f8c94cd
To: sip:200@myOwnIPadress:9999
Call-ID: 396c9c7762dfd21c7e87b9252bcdc1b2@myOwnIPadress
CSeq: 102 INVITE
Content-Length: 0
Content-Type: application/sdp

<------------->
— (8 headers 0 lines) —
roberto*CLI>
<— SIP read from myOwnIPadress:9999 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP myOwnIPadress:5060;rport;branch=z9hG4bK34dfa9db
From: “0495267897” sip:0495267897@myOwnIPadress;tag=as4f8c94cd
To: sip:200@myOwnIPadress:9999;tag=BBOX6726_v0.89.35-1243312935-ef26bb95-1575738001
Contact: “200” sip:200@myOwnIPadress:9999
Call-ID: 396c9c7762dfd21c7e87b9252bcdc1b2@myOwnIPadress
CSeq: 102 INVITE
Content-Length: 0

<------------->
— (8 headers 0 lines) —
– SIP/200-09651e70 is ringing
roberto*CLI>
<— SIP read from myProviderIP:5060 —>
CANCEL sip:028809326@myOwnIPadress SIP/2.0
Via: SIP/2.0/UDP myProviderIP;branch=z9hG4bK378e.c8940a5.0
From: “0495267897” sip:0495267897@myProviderIP;tag=as6baf07f4
Call-ID: 2951844526849c910e1d8bff5fb52b17@myProviderIP
To: sip:028809326@myProviderIP
CSeq: 102 CANCEL
Max-Forwards: 70
User-Agent: Enswitch SIP proxy
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Sending to myProviderIP : 5060 (NAT)

<— Reliably Transmitting (NAT) to myProviderIP:5060 —>
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP myProviderIP;branch=z9hG4bK378e.c8940a5.0;received=myProviderIP
Via: SIP/2.0/UDP myProviderIP1:5060;received=myProviderIP1;branch=z9hG4bK65023175;rport=5060
From: “0495267897” sip:0495267897@myProviderIP;tag=as6baf07f4
To: sip:028809326@myProviderIP;tag=as5b7c081d
Call-ID: 2951844526849c910e1d8bff5fb52b17@myProviderIP
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0[/code]

As you can see, there is no OK coming from my extension, nor any ACK coming from the provider.
Instead you’ll find the CANCEL sip:028809326@myOwnIPadress SIP/2.0 which shows that I hung up before the call was actually answered.

Sorry again to have make you lose your time and others…

Pagaille (which more or less means “mess” in french :wink:

6

Right, it was an option provided to some IVR-related scenario. I tried it “just to see” if it would help.

I removed this option in the meanwhile.