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Summary:ASTERISK-12187: [patch] Page(CHANNEL) doesn't bring 'CHANNEL' into the MeetMe conference.
Reporter:Nick Barnes (bcnit)Labels:
Date Opened:2008-06-13 07:16:47Date Closed:2008-06-13 17:12:30
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Applications/app_page
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) 20080613__bug12850.diff.txt
Description:Using the dialplan code:

exten => 19,1,NoOp(Paging handsets)
exten => 19,n,SIPAddHeader(Call-Info: <sip:81.187.1.1>\;Answer-After=0)
exten => 19,n,Page(${PAGEHS})
exten => 19,n,Hangup()

(where PAGEHS is defined as a global containing one or more SIP resources)

The called handsets answers the call, but no audio is played through the handset. Further investigation (see below) shows that a MeetMe room is created, but that only the caller is a member of that conference and the SIP resources paged are not. When the caller hangs up, the paged resources also hang up.


****** ADDITIONAL INFORMATION ******

The verbose output below shows what happens when ${PAGEHS} is 'SIP/201':

== Using SIP RTP CoS mark 5
-- Executing [19@from-authorised:1] NoOp("SIP/666-095d6e08", "Paging handsets") in new stack
-- Executing [19@from-authorised:2] SIPAddHeader("SIP/666-095d6e08", "Call-Info: <sip:81.187.1.1>\;Answer-After=0") in new stack
-- Executing [19@from-authorised:3] Page("SIP/666-095d6e08", "SIP/201") in new stack
== Using SIP RTP CoS mark 5
-- Called 201
-- <SIP/666-095d6e08> Playing 'beep.gsm' (language 'en')
-- SIP/201-09619ba8 is ringing
-- SIP/201-09619ba8 answered
-- Created MeetMe conference 1023 for conference '216121150d'

pbx09a*CLI> meetme list 216121150d
User #: 01          666 Vitell               Channel: SIP/666-095d6e08     (talking) 00:00:12
1 users in that conference.

pbx09a*CLI> sip show channels
Peer             User/ANR    Call ID          Format           Hold     Last Message
81.187.1.1   201         0fde875c5a7a6ab  0x8 (alaw)       No       Tx: ACK
217.169.1.1  666         3c27ae56e1b8-9j  0x8 (alaw)       No       Rx: ACK
2 active SIP dialogs
Comments:By: Tilghman Lesher (tilghman) 2008-06-13 08:24:34

Please confirm that this patch fixes the problem.
By: Nick Barnes (bcnit) 2008-06-13 08:30:11

Yes, this resolves it. Thank you.
By: Digium Subversion (svnbot) 2008-06-13 09:08:43

Repository: asterisk
Revision: 122557

U   trunk/main/dial.c

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r122557 | tilghman | 2008-06-13 09:08:34 -0500 (Fri, 13 Jun 2008) | 7 lines

Convert one more delimiter to use comma.
(closes issue ASTERISK-12187)
Reported by: bcnit
Patches:
      20080613__bug12850.diff.txt uploaded by Corydon76 (license 14)
Tested by: bcnit

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http://svn.digium.com/view/asterisk?view=rev&revision=122557
By: Digium Subversion (svnbot) 2008-06-13 09:09:19

Repository: asterisk
Revision: 122558

_U  branches/1.6.0/
U   branches/1.6.0/main/dial.c

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r122558 | tilghman | 2008-06-13 09:09:16 -0500 (Fri, 13 Jun 2008) | 15 lines

Merged revisions 122557 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
r122557 | tilghman | 2008-06-13 09:15:07 -0500 (Fri, 13 Jun 2008) | 7 lines

Convert one more delimiter to use comma.
(closes issue ASTERISK-12187)
Reported by: bcnit
Patches:
      20080613__bug12850.diff.txt uploaded by Corydon76 (license 14)
Tested by: bcnit

........

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http://svn.digium.com/view/asterisk?view=rev&revision=122558
By: Digium Subversion (svnbot) 2008-06-13 17:12:30

Repository: asterisk
Revision: 122724

_U  team/seanbright/resolve-shadow-warnings/
U   team/seanbright/resolve-shadow-warnings/channels/chan_iax2.c
U   team/seanbright/resolve-shadow-warnings/include/asterisk/channel.h
U   team/seanbright/resolve-shadow-warnings/include/asterisk/timing.h
U   team/seanbright/resolve-shadow-warnings/main/asterisk.c
U   team/seanbright/resolve-shadow-warnings/main/autoservice.c
U   team/seanbright/resolve-shadow-warnings/main/channel.c
U   team/seanbright/resolve-shadow-warnings/main/dial.c
U   team/seanbright/resolve-shadow-warnings/main/file.c
U   team/seanbright/resolve-shadow-warnings/main/timing.c
U   team/seanbright/resolve-shadow-warnings/res/res_config_pgsql.c
A   team/seanbright/resolve-shadow-warnings/res/res_timing_dahdi.c

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r122724 | seanbright | 2008-06-13 17:12:27 -0500 (Fri, 13 Jun 2008) | 87 lines

Merged revisions 122523,122526,122557,122616,122664,122714,122716 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
r122523 | russell | 2008-06-13 08:45:50 -0400 (Fri, 13 Jun 2008) | 11 lines

Merge changes from timing branch

- Convert chan_iax2 to use the timing API
- Convert usage of timing in the core to use the timing API instead of
  using DAHDI directly
- Make a change to the timing API to add the set_rate() function
- change the timing core to use a rwlock
- merge a timing implementation, res_timing_dahdi

Basic testing was successful using res_timing_dahdi

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r122526 | russell | 2008-06-13 08:53:08 -0400 (Fri, 13 Jun 2008) | 3 lines

Do not allow res_timing_dahdi to be unloaded.  We can re-enable this once we
add automatic use count handling for timing modules.

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r122557 | tilghman | 2008-06-13 10:15:07 -0400 (Fri, 13 Jun 2008) | 7 lines

Convert one more delimiter to use comma.
(closes issue ASTERISK-12187)
Reported by: bcnit
Patches:
      20080613__bug12850.diff.txt uploaded by Corydon76 (license 14)
Tested by: bcnit

................
r122616 | jpeeler | 2008-06-13 13:38:28 -0400 (Fri, 13 Jun 2008) | 13 lines

Blocked revisions 122613 via svnmerge

........
r122613 | jpeeler | 2008-06-13 12:36:56 -0500 (Fri, 13 Jun 2008) | 6 lines

(closes issue ASTERISK-12184)
Reported by: Netview
Tested by: jpeeler

Use correct location to search for tonezone.

........

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r122664 | jpeeler | 2008-06-13 14:58:29 -0400 (Fri, 13 Jun 2008) | 8 lines

Blocked revisions 122663 via svnmerge

........
r122663 | jpeeler | 2008-06-13 13:57:24 -0500 (Fri, 13 Jun 2008) | 1 line

fixed dahdi compatability header from assuming either dahdi or zaptel is installed (may not have either)
........

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r122714 | mmichelson | 2008-06-13 17:45:21 -0400 (Fri, 13 Jun 2008) | 17 lines

Merged revisions 122713 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r122713 | mmichelson | 2008-06-13 16:44:53 -0500 (Fri, 13 Jun 2008) | 9 lines

Short circuit the loop in autoservice_run if there are no channels to poll.
If we continued, then the result would be calling poll() with a NULL
pollfd array. While this is fine with POSIX's poll(2) system call, those
who use Asterisk's internal poll mechanism (Darwin systems) would have
a failed assertion occur when poll is called.

(related to issue ASTERISK-9989)


........

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r122716 | tilghman | 2008-06-13 17:50:28 -0400 (Fri, 13 Jun 2008) | 2 lines

Properly detect the size of char/varchar fields

................

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http://svn.digium.com/view/asterisk?view=rev&revision=122724