[asterisk-bugs] [Asterisk 0012569]: [branch] Receiving Text from res_jabber

Tue Feb 17 14:37:52 CST 2009

A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=12569 
====================================================================== 
Reported By:                eech55
Assigned To:                phsultan
====================================================================== 
Project:                    Asterisk
Issue ID:                   12569
Category:                   Resources/res_jabber
Reproducibility:            always
Severity:                   feature
Priority:                   normal
Status:                     ready for testing
Asterisk Version:           SVN 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 115340 
Request Review:              
====================================================================== 
Date Submitted:             2008-05-02 00:38 CDT
Last Modified:              2009-02-17 14:37 CST
====================================================================== 
Summary:                    [branch] Receiving Text from res_jabber
Description: 
Hello,

So far, chan_gtalk sends text to GTalk clients only. There is no way to
hear input from them which limit us a lot.

It will be awesome if we are able to receive text from GTalk users, to
provide extra services such as:

1) Voice menus. The user can select menus by sending corresponding keys
2) Or, "text" menus by which the chan_gtalk replies by text messages
instead of voice

I am glad that phsultan thought about it, however very disappointed that
it was canceled.
http://bugs.digium.com/bug_view_advanced_page.php?bug_id=8659

======================================================================
Relationships       ID      Summary
----------------------------------------------------------------------
related to          0008659 [patch] Add a jabber text receiver appl...
====================================================================== 

---------------------------------------------------------------------- 
 (0100267) jtodd (administrator) - 2009-02-17 14:37
 http://bugs.digium.com/view.php?id=12569#c100267 
---------------------------------------------------------------------- 
OK, that worked fine for short messages.  Excellent!  Now, on to the next
bug.  :-)

If I send a big block of text at the system, it chokes and shows nothing. 
I cut and pasted a big section of Jabberwocky into my chat client and sent
it to the waiting Asterisk system.  It didn't show anything:

*CLI> 
*CLI> [Feb 17 20:34:55] DEBUG[5622]: chan_sip.c:4573 do_setnat: Setting
NAT on RTP to Off
[Feb 17 20:34:55] DEBUG[5622]: chan_sip.c:6739 sip_alloc: Allocating new
SIP dialog for 00059bf1-a0de004c-13adddad-0b756fc6 at 10.10.3.7 - INVITE (With
RTP)
[Feb 17 20:34:55] DEBUG[5622]: chan_sip.c:4573 do_setnat: Setting NAT on
RTP to Off
[Feb 17 20:34:55] DEBUG[5622]: chan_sip.c:3745 __sip_ack: Stopping
retransmission on '00059bf1-a0de004c-13adddad-0b756fc6 at 10.10.3.7' of
Response 101: Match Found
[Feb 17 20:34:55] DEBUG[5622]: chan_sip.c:4573 do_setnat: Setting NAT on
RTP to Off
[Feb 17 20:34:55] DEBUG[5622]: chan_sip.c:7807 process_sdp: Got
unsupported a:fmtp:18 annexb=no in SDP offer
[Feb 17 20:34:55] DEBUG[5622]: chan_sip.c:7807 process_sdp: Got
unsupported a:fmtp:101 0-15 in SDP offer
[Feb 17 20:34:55] DEBUG[5622]: chan_sip.c:18924 handle_request_invite:
Checking SIP call limits for device 2203
[Feb 17 20:34:55] DEBUG[5622]: pbx.c:3635 pbx_extension_helper: Launching
'JabberSend'
    -- Executing [995 at to-binfone:1] JabberSend("SIP/2203-7e79d8c0",
"jtphone,jtodd at jabber.loligo.com,"Rah rah sis boom bah!"") in new stack

JABBER: jtphone OUTGOING: <message type='chat'
to='jtodd at jabber.loligo.com'
from='jtphone at jabber.loligo.com/asterisk'><body>&quot;Rah rah sis boom
bah!&quot;</body></message>
[Feb 17 20:34:55] DEBUG[5622]: devicestate.c:652 handle_devstate_change:
Processing device state change for 'SIP/2203'
[Feb 17 20:34:55] DEBUG[5622]: devicestate.c:602 process_collection:
Adding per-server state of 'Not in use' for 'SIP/2203'
[Feb 17 20:34:55] DEBUG[5622]: devicestate.c:608 process_collection:
Aggregate devstate result is 1
[Feb 17 20:34:55] DEBUG[5622]: devicestate.c:624 process_collection:
Aggregate state for device 'SIP/2203' has not changed from 'Not in use'

JABBER: jtphone INCOMING: <message type="chat" id="purple7ace65a"
to="jtphone at jabber.loligo.com/asterisk"
from="jtodd at jabber.loligo.com/Adium"><x
xmlns="jabber:x:event"><composing/></x><body>`Twas brillig, and the slithy
toves
Â Â Did gyre and gimble in the wabe:
All mimsy were the borogoves,
Â Â And the mome raths outgrabe.

"Beware the Jabberwock, my son!
Â Â The jaws that bite, the claws that catch!
Beware the Jubjub bird, and shun
Â Â The frumious Bandersnatch!"
He took his vorpal sword in hand:
Â Â Long time the manxome foe he sought --
So rested he by the Tumtum tree,
Â Â And stood awhile in thought.
And, as in uffish thought he stood,
Â Â The Jabberwock, with eyes of flame,
Came whiffling through the tulgey wood,
Â Â And burbled as it came!
One, two! One, two! And through and through
Â Â The vorpal blade went snicker-snack!
He left it dead, and with its head
Â Â He went galumphing back.
"And, has thou slain the Jabberwock?
Â Â Come to my arms, my beamish boy!
O frabjous day! Callooh! Callay!'
Â Â He chortled in his joy.

`Twas brillig, and the slithy toves
Â Â Did gyre and gimble in the wabe;
All mimsy were the borogoves,
Â Â And the mome raths outgrabe.

</body><html xmlns="http://jabber.org/protocol/xhtml-im"><body
xmlns="http://www.w3.org/1999/xhtml">`Twas brillig, and the slithy
toves<br/>Â Â Did gyre and gimble in the wabe:<br/>All mimsy were the
borogoves,<br/>Â Â And the mome raths
outgrabe.<br/><br/><br/><br/>"Beware the Jabberwock, my
son!<br/>Â Â The jaws that bite, the claws that catch!<br/>Beware the
Jubjub bird, and shun<br/>Â Â The frumious Bandersnatch!"<br/>He took
his vorpal sword in hand:<br/>Â Â Long time the manxome foe he sought
--<br/>So rested he by the Tumtum tree,<br/>Â Â And stood awhile in
thought.<br/>And, as in uffish thought he stood,<br/>Â Â The
Jabberwock, with eyes of flame,<br/>Came whiffling through the tulgey
wood,<br/>Â Â And burbled as it came!<br/>One, two! One, two! And
through and through<br/>Â Â The vorpal blade went snicker-snack!<br/>He
left it dead, and with its head<br/>Â Â He went galumphing
back.<br/>"And, has thou slain the Jabberwock?<br/>Â Â Come to my arms,
my beamish boy!<br/>O frabjous day! Callooh! Callay!'<br/>Â Â He
chortled in his joy.<br/><br/><br/>`Twas brillig, and the slithy
toves<br/>Â Â Did gyre and gimble in the wabe;<br/>All mimsy were the
borogoves,<br/>Â Â And the mome raths
outgrabe.<br/><br/></body></html></message>
[Feb 17 20:34:57] WARNING[5622]: res_jabber.c:1086 aji_recv: XML parsing
failed

*CLI> 
*CLI> [Feb 17 20:35:05] NOTICE[5622]: res_jabber.c:780
acf_jabberreceive_read: Timed out : no message received from
jtodd at jabber.loligo.com
[Feb 17 20:35:05] DEBUG[5622]: pbx.c:3635 pbx_extension_helper: Launching
'Set'
    -- Executing [995 at to-binfone:2] Set("SIP/2203-7e79d8c0", "OPTION=") in
new stack
[Feb 17 20:35:05] DEBUG[5622]: pbx.c:3635 pbx_extension_helper: Launching
'NoOp'
    -- Executing [995 at to-binfone:3] NoOp("SIP/2203-7e79d8c0", ""Wowzers -
we got "") in new stack
[Feb 17 20:35:05] DEBUG[5622]: pbx.c:3635 pbx_extension_helper: Launching
'Hangup'
    -- Executing [995 at to-binfone:4] Hangup("SIP/2203-7e79d8c0", "") in new
stack
[Feb 17 20:35:05] DEBUG[5622]: pbx.c:4228 __ast_pbx_run: Spawn extension
(to-binfone,995,4) exited non-zero on 'SIP/2203-7e79d8c0'
  == Spawn extension (to-binfone, 995, 4) exited non-zero on
'SIP/2203-7e79d8c0'
[Feb 17 20:35:05] DEBUG[5622]: channel.c:1567 ast_softhangup_nolock:
Soft-Hanging up channel 'SIP/2203-7e79d8c0'
[Feb 17 20:35:05] DEBUG[5622]: channel.c:1662 ast_hangup: Hanging up
channel 'SIP/2203-7e79d8c0'
[Feb 17 20:35:05] DEBUG[5622]: chan_sip.c:5571 sip_hangup: Hangup call
SIP/2203-7e79d8c0, SIP callid
00059bf1-a0de004c-13adddad-0b756fc6 at 10.10.3.7
[Feb 17 20:35:05] DEBUG[5622]: chan_sip.c:5511 hangup_cause2sip: AST
hangup cause 16 (no match found in SIP)
[Feb 17 20:35:05] DEBUG[5622]: devicestate.c:652 handle_devstate_change:
Processing device state change for 'SIP/2203'
[Feb 17 20:35:05] DEBUG[5622]: devicestate.c:602 process_collection:
Adding per-server state of 'Not in use' for 'SIP/2203'
[Feb 17 20:35:05] DEBUG[5622]: devicestate.c:608 process_collection:
Aggregate devstate result is 1
[Feb 17 20:35:05] DEBUG[5622]: devicestate.c:624 process_collection:
Aggregate state for device 'SIP/2203' has not changed from 'Not in use'
[Feb 17 20:35:05] DEBUG[5622]: chan_sip.c:3745 __sip_ack: Stopping
retransmission on '00059bf1-a0de004c-13adddad-0b756fc6 at 10.10.3.7' of
Response 102: Match Found

*CLI> 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-02-17 14:37 jtodd          Note Added: 0100267                          
======================================================================