asterisk users - Apr 2006 - outbound calls to sip urls

Hi,
 I wish to use the manager API to make an outbound call to a sip
url,subsequently play a prompt and hangup.Any hints on how to acheive
this/feasability will be much appreciated.
Regards,
Ajit

On Mon, Apr 24, 2006 at 04:46:29PM +0530, Ajit spake
thusly:
> Hi,
>  I wish to use the manager API to make an outbound call to a sip
> url,subsequently play a prompt and hangup.Any hints on how to acheive
> this/feasability will be much appreciated.
I'm no expert, but it looks simple enough to me - just use the originate
action to call with something like this:

Action: Originate
channel: SIP/bla@bla.com
context: testcontext
extension: extensiontosendtheprompt
priority: 1

So that extension will just send the prompt and then hang up.

-- 
Jon-o Addleman - http://redowl.dyndns.org

Hi,
 I am trying to use the manager API to originate a SIP call from one
asterisk extension to another. eg extension 600@myasteriskserver calls
extension 800@myasteriskserver.However this fails with a 482 "Loop
detected".

I beleive this behaviour is incorrect according to rfc3261 :
 The UAS processes the first such request received and responds with a 482
(Loop
      Detected) to the rest of them.

In this case its a sent request colliding with the first recieved request.
Can people on the dev list comment on if this is a bug and where in the code
it may be fixed.

Note that i do not need SIP proxy functionality or NAT  from asterix.
 Any ideas on workarounds?? I think if i can have two SIP stacks on
different ports calling each other this behaviour can be avoided.Is this
possible and if so how can i configure this.

Regards,
Ajit

P.S: Here is the SIP debug (ip addresses modified)
LI> sip debug
SIP Debugging enabled
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'manager' logged on from xxx.yyy.xxx.yyy
We're at myasteriskserver port 10878
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 10 lines
Reliably Transmitting (no NAT) to myasteriskserver:5060:
INVITE sip:800@myasteriskserver SIP/2.0
Via: SIP/2.0/UDP myasteriskserver:5060;branch=z9hG4bK7aa5b7b4;rport
From: "asterisk" <sip:asterisk@myasteriskserver>;tag=as7c2a32b7
To: <sip:800@myasteriskserver>
Contact: <sip:asterisk@myasteriskserver>
Call-ID: 454c631c1c01dff40cc65d314584a1f1@myasteriskserver
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 28 Apr 2006 13:14:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 212

v=0
o=root 5351 5351 IN IP4 myasteriskserver
s=session
c=IN IP4 myasteriskserver
t=0 0
m=audio 10878 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---

<-- SIP read from myasteriskserver:5060:
INVITE sip:800@myasteriskserver SIP/2.0
Via: SIP/2.0/UDP myasteriskserver:5060;branch=z9hG4bK7aa5b7b4;rport
From: "asterisk" <sip:asterisk@myasteriskserver>;tag=as7c2a32b7
To: <sip:800@myasteriskserver>
Contact: <sip:asterisk@myasteriskserver>
Call-ID: 454c631c1c01dff40cc65d314584a1f1@myasteriskserver
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 28 Apr 2006 13:14:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 212

v=0
o=root 5351 5351 IN IP4 myasteriskserver
s=session
c=IN IP4 myasteriskserver
t=0 0
m=audio 10878 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

--- (13 headers 10 lines)---
Transmitting (no NAT) to myasteriskserver:5060:
SIP/2.0 482 Loop Detected
Via: SIP/2.0/UDP
myasteriskserver:5060;branch=z9hG4bK7aa5b7b4;rport;received=myasteriskserver
From: "asterisk" <sip:asterisk@myasteriskserver>;tag=as7c2a32b7
To: <sip:800@myasteriskserver>;tag=as7c2a32b7
Call-ID: 454c631c1c01dff40cc65d314584a1f1@myasteriskserver
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:asterisk@myasteriskserver>
Content-Length: 0


---

<-- SIP read from myasteriskserver:5060:
SIP/2.0 482 Loop Detected
Via: SIP/2.0/UDP
myasteriskserver:5060;branch=z9hG4bK7aa5b7b4;rport;received=myasteriskserver
From: "asterisk" <sip:asterisk@myasteriskserver>;tag=as7c2a32b7
To: <sip:800@myasteriskserver>;tag=as7c2a32b7
Call-ID: 454c631c1c01dff40cc65d314584a1f1@myasteriskserver
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:asterisk@myasteriskserver>
Content-Length: 0


--- (10 headers 0 lines)---
    -- Got SIP response 482 "Loop Detected" back from myasteriskserver
Transmitting (no NAT) to myasteriskserver:5060:
ACK sip:800@myasteriskserver SIP/2.0
Via: SIP/2.0/UDP myasteriskserver:5060;branch=z9hG4bK7aa5b7b4;rport
From: "asterisk" <sip:asterisk@myasteriskserver>;tag=as7c2a32b7
To: <sip:800@myasteriskserver>;tag=as7c2a32b7
Contact: <sip:asterisk@myasteriskserver>
Call-ID: 454c631c1c01dff40cc65d314584a1f1@myasteriskserver
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---

<-- SIP read from myasteriskserver:5060:
ACK sip:800@myasteriskserver SIP/2.0
Via: SIP/2.0/UDP myasteriskserver:5060;branch=z9hG4bK7aa5b7b4;rport
From: "asterisk" <sip:asterisk@myasteriskserver>;tag=as7c2a32b7
To: <sip:800@myasteriskserver>;tag=as7c2a32b7
Contact: <sip:asterisk@myasteriskserver>
Call-ID: 454c631c1c01dff40cc65d314584a1f1@myasteriskserver
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


--- (10 headers 0 lines)---
Destroying call '454c631c1c01dff40cc65d314584a1f1@myasteriskserver'
  == Manager 'manager' logged off from xxx.yyy.xxx.yyy