[asterisk-dev] Dialplan AGI with SIP channels blocking if AGI already running - parallellism not possible?

Johan Sandgren jsa at svep.se
Wed Apr 3 09:58:31 CDT 2013 

Thanks for noticing Michael L. Young.

Retested it with the right address also 2nd SIP MESSAGE.
Still same result, I attach the new logs which show this.
I see it stops after "Looking for 1000 in sipmessage".
Nothing more happens. There's the stopping point. 
Should it work?

==========================================================
2nd Message log again, still same result though - no dialplan is executed.
==========================================================

[Apr  3 16:47:36] DEBUG[18406]: chan_sip.c:8585 parse_request:  Header  0 [ 37]: MESSAGE sip:1000 at a.b.c.d SIP/2.0
[Apr  3 16:47:36] DEBUG[18406]: chan_sip.c:8585 parse_request:  Header  1 [ 89]: Via: SIP/2.0/UDP e.f.g.h:5060;rport;branch=z9hG4bKPjad3799494115439d90d7376909a2854f
[Apr  3 16:47:36] DEBUG[18406]: chan_sip.c:8585 parse_request:  Header  2 [ 16]: Max-Forwards: 70
[Apr  3 16:47:36] DEBUG[18406]: chan_sip.c:8585 parse_request:  Header  3 [ 65]: From: <sip:ip1 at a.b.c.d>;tag=0d92d98a99314139b93b02603b20b8d4
[Apr  3 16:47:36] DEBUG[18406]: chan_sip.c:8585 parse_request:  Header  4 [ 27]: To: <sip:1000 at a.b.c.d>
[Apr  3 16:47:36] DEBUG[18406]: chan_sip.c:8585 parse_request:  Header  5 [ 41]: Call-ID: 2a805f5cbf674ef898e06e41dae4c74a
[Apr  3 16:47:36] DEBUG[18406]: chan_sip.c:8585 parse_request:  Header  6 [ 19]: CSeq: 16537 MESSAGE
[Apr  3 16:47:36] DEBUG[18406]: chan_sip.c:8585 parse_request:  Header  7 [ 29]: Accept: application/scaip+xml
[Apr  3 16:47:36] DEBUG[18406]: chan_sip.c:8585 parse_request:  Header  8 [ 39]: Contact: <sip:ip1 at e.f.g.h:5060;ob>
[Apr  3 16:47:36] DEBUG[18406]: chan_sip.c:8585 parse_request:  Header  9 [ 49]: User-Agent: PJSUA v2.0.1 win32-6.1/i386/msvc-15.0
[Apr  3 16:47:36] DEBUG[18406]: chan_sip.c:8585 parse_request:  Header 10 [156]: Authorization: Digest username="ip1", realm="stt", nonce="0023deb9", uri="sip:1000 at a.b.c.d", response="264566c1aaf02ce65fb518dbeb6e4566", algorithm=MD5
[Apr  3 16:47:36] DEBUG[18406]: chan_sip.c:8585 parse_request:  Header 11 [ 35]: Content-Type: application/scaip+xml
[Apr  3 16:47:36] DEBUG[18406]: chan_sip.c:8585 parse_request:  Header 12 [ 18]: Content-Length: 82
[Apr  3 16:47:36] DEBUG[18406]: chan_sip.c:8585 parse_request:  Header 13 [  0]:
[Apr  3 16:47:36] DEBUG[18406]: chan_sip.c:8622 parse_request:    Body  0 [ 82]: abcdabcdabcdabcdabcdabcdabcdabcdabcdabcdabcdabcdabcdabcdabcdabcdabcdabcdabcdabcdabcdab
--- (13 headers 1 lines) ---
[Apr  3 16:47:36] DEBUG[18406]: chan_sip.c:8147 find_call: = Looking for  Call ID: 2a805f5cbf674ef898e06e41dae4c74a (Checking From) --From tag 0d92d98a99314139b93b02603b20b8d4 --To-tag
[Apr  3 16:47:36] DEBUG[18406]: netsock2.c:134 ast_sockaddr_split_hostport: Splitting 'a.b.c.d' into...
[Apr  3 16:47:36] DEBUG[18406]: netsock2.c:188 ast_sockaddr_split_hostport: ...host 'a.b.c.d' and port ''.
[Apr  3 16:47:36] DEBUG[18406]: netsock2.c:134 ast_sockaddr_split_hostport: Splitting 'a.b.c.d' into...
[Apr  3 16:47:36] DEBUG[18406]: netsock2.c:188 ast_sockaddr_split_hostport: ...host 'a.b.c.d' and port ''.
[Apr  3 16:47:36] DEBUG[18406]: chan_sip.c:25408 handle_incoming: **** Received MESSAGE (11) - Command in SIP MESSAGE
Receiving message!
[Apr  3 16:47:36] DEBUG[18406]: chan_sip.c:16306 receive_message: SIP Text message content-type received: text/plain'
[Apr  3 16:47:36] DEBUG[18406]: chan_sip.c:16316 receive_message: SIP Text message accepted by content-type. 
[Apr  3 16:47:36] DEBUG[18406]: chan_sip.c:16348 receive_message: SIP MESSAGE debug 2
Found peer 'ip1' for 'ip1' from e.f.g.h:5060
Looking for 1000 in sipmessage (domain a.b.c.d)

[Apr  3 16:47:36] DEBUG[18406]: chan_sip.c:3363 __sip_xmit: Trying to put 'SIP/2.0 202' onto UDP socket destined for 10.135.44.69:5060
Scheduling destruction of SIP dialog '2a805f5cbf674ef898e06e41dae4c74a' in 6400 ms (Method: MESSAGE)
[Apr  3 16:47:42] DEBUG[18406]: chan_sip.c:3903 __sip_autodestruct: Auto destroying SIP dialog '2a805f5cbf674ef898e06e41dae4c74a'
[Apr  3 16:47:42] DEBUG[18406]: chan_sip.c:6058 sip_destroy: Destroying SIP dialog 2a805f5cbf674ef898e06e41dae4c74a
Really destroying SIP dialog '2a805f5cbf674ef898e06e41dae4c74a' Method: MESSAGE


From here, in the normal case, this line would come: "[Apr  3 16:47:07] DEBUG[18395]: pbx.c:4244 pbx_extension_helper: Launching 'Answer'".
But it doesn't.

Any clues why?
Single thread only for all SIP MESSAGES running in the dialplan, need to complete the previous until another MESSAGE can run in the dialplan?

/Johan



-----Ursprungligt meddelande-----
Från: asterisk-dev-bounces at lists.digium.com [mailto:asterisk-dev-bounces at lists.digium.com] För Michael L. Young
Skickat: den 3 april 2013 15:13
Till: Asterisk Developers Mailing List
Ämne: Re: [asterisk-dev] Dialplan AGI with SIP channels blocking if AGI already running - parallellism not possible?

----- Original Message -----
> From: "Johan Sandgren" <jsa at svep.se>
> To: asterisk-dev at lists.digium.com
> Sent: Wednesday, April 3, 2013 8:01:01 AM
> Subject: [asterisk-dev] Dialplan AGI with SIP channels blocking if AGI already running - parallellism not possible?
> 

> ======================================================================
> ========================================================
> Asterisk log 1 - incoming SIP MESSAGE is starting the AGI-java program 
> normally 
> ======================================================================
> ========================================================
> 

>   Header  4 [ 27]: To: <sip:1000 at a.b.c.d>

> Looking for 1000 in sipmessage (domain a.b.c.d)

> ======================================================================
> ========================================================
> Asterisk log 2 - incoming SIP MESSAGE is not starting dialplan 
> ======================================================================
> ========================================================
> 

>   Header  4 [ 22]: To: <sip:a.b.c.d>

> Looking for s in sipmessage (domain a.b.c.d)

I see that in log 1, the incoming message has "To: <sip:1000 at a.b.c.d>", which goes to extension 1000.  In log 2, there is no extension in the To header.  So we fallback to looking for a "s" extension.  Is that extension present in [sipmessage]?  If not, try adding one.

Michael

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