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    <link>http://gmane.org</link>
  </image>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/31966">
    <title>Re: IMAP_STORAGE issue with app_voicemail</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/31966</link>
    <description>
It's not that it doesn't like it, it just doesn't look for it.  I'll add a log
message.

</description>
    <dc:creator>Sean Bright</dc:creator>
    <dc:date>2008-10-07T14:54:46</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/31965">
    <title>Re: IMAP_STORAGE issue with app_voicemail</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/31965</link>
    <description>


I was able to figure out my problem. I had to install a dovecot server  
purely for testing purposes and then the error changed slightly to the  
point where when I searched it I found the issue because someone else  
had this same problem.

Basically in the voicemail config "imapsecret=password" is incorrect  
it must be "imappassword=password". Very frustrating. If asterisk  
doesn't like imapsecret it should complain about it in the console/ 
logs rather than just failing with some cryptic message.


Should I file a bug report about this?

Brendan Martens


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</description>
    <dc:creator>Brendan Martens</dc:creator>
    <dc:date>2008-10-07T14:26:03</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/31964">
    <title>Re: Proposed change to branch revision blockingpolicy</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/31964</link>
    <description>
I think we need to tow that line a little better than we have been (I might be
one of the more guilty parties in that regard).  For example, this morning I
merged a fix into 1.6.0 (r147051) that, in my mind at least, is borderline.  The
only time it would be an issue would be if someone was using a custom channel
driver that started with "DAH" and was using operator mode.  It's highly
unlikely, but possible, and wouldn't happen in 1.4 or below.  So is that a
regression?  It's a toss up.


Typically, if it's something that requires less work on my part, than I am all
for it :)  Sounds good.

Thanks,
</description>
    <dc:creator>Sean Bright</dc:creator>
    <dc:date>2008-10-07T13:29:09</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/31963">
    <title>Re: ast_pthread_create</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/31963</link>
    <description>_______________________________________________
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    <dc:creator>Steve Murphy</dc:creator>
    <dc:date>2008-10-07T13:13:11</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/31962">
    <title>Re: Proposed change to branch revision blockingpolicy</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/31962</link>
    <description>
+1

</description>
    <dc:creator>Michiel van Baak</dc:creator>
    <dc:date>2008-10-07T13:04:29</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/31961">
    <title>Proposed change to branch revision blocking policy</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/31961</link>
    <description>As it stands right now, when someone commits a change to Asterisk trunk,
it get either merged or blocked in the release branches (1.6.0 and
1.6.1). However, this is resulting in a large number of what appear to
be pointless commits to the 1.6.0 branch, which then have to be pruned
out of the ChangeLog when we make the next 1.6.0.y regression fix release.

Once a 1.6.x branch has reached the release candidate phase (not when
the branch is made), that branch is no longer a candidate for any
changes *except* fixes for regressions in that branch. As such, nobody
is ever going to run 'svnmerge avail' on that branch to see what is
available to be merged, because every fix that would possibly be merged
into the branch during that process is supposed to be already there!

Ideally, the same situation would be true during the time that the 1.6.x
branch exists but has not yet been released, as every bugfix commit to
trunk should be merged to that branch immediately by the developer who
commits it to trunk, but since that doesn't always happen we still have
to run an 'svnmerge avail' before making release candidates to ensure
nothing got missed.

So, I'm proposing that once a 1.6.x branch has reached the release
candidate phase, we stop using svnmerge to block revisions there, as it
results in useless 'noise' commits. When there are multiple 1.6.x
branches that have had releases, this problem would only multiply.

Comments?

</description>
    <dc:creator>Kevin P. Fleming</dc:creator>
    <dc:date>2008-10-07T12:26:37</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/31960">
    <title>Re: [asterisk-commits] jpeeler: trunk r146875 -/trunk/main/features.c</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/31960</link>
    <description>
This change should not be necessary (and should have no effect at all).
When a structure initializer is provided, any fields not included in the
initializer are set to zero, by definition in the C standards. Since
NULL is zero on all modern platforms, these fields already were NULL
before this change.

</description>
    <dc:creator>Kevin P. Fleming</dc:creator>
    <dc:date>2008-10-07T12:17:49</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/31959">
    <title>app_vocemail, IMAP, shared voicemail boxes</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/31959</link>
    <description>I have set up a Cyrus server to provide IMAP service to an asterisk 
server. I am using this to provide a shared voicemail facility for a 
number of phones. This is set up with an entry for each phone in the 
voicemail.conf file as follows.

6999 =&gt; 6999,Company shared 
mailbox,company&lt; at &gt;company.com,,|imapuser=companymailbox
6998 =&gt; 6998,Company shared 
mailboz,company&lt; at &gt;company.com,,|imapuser=companymailbox

There is a master user set up for the imap as

authuser = administrator
authpassword = password

This is working and messages can be left for all of the extensions, they 
all appear in the single mailbox and MWI is working. BUT when I try to 
retrieve the messages then I can only retrieve them from the phone that 
the message was left for. I.E. 6999 messages can only be retrieved if I 
dial into 6999 mailbox.

What I would like if to be able to dial in from any of the extensions 
and retrieve all messages in the mailbox.

The problem would seem to stem from the line(s) in app_voicemail.c that 
search for "X-Asterisk-VM-Extension" headers. This uses the mailbox 
number of the dial in user to search the mailbox.

I propose a change that would allow the mailbox number for messages to 
be set when they are being left and when they are being read from a 
configuration option such as 'imapmboxid=6999' to be what this header 
gets set to and gets searched on.

If this looks like the right thing to do then I will prepare a patch and 
submit it to the list. Also, if required I will submit a bug report if 
somebody wants to assign this to me I will then keep up to date any 
experiences reported.

I hope this is the right etiquette for improvements to the asterisk base.

Regards, Howard.


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</description>
    <dc:creator>Howard Wilkinson</dc:creator>
    <dc:date>2008-10-07T11:36:22</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/31958">
    <title>ast_pthread_create</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/31958</link>
    <description>In file util.h is defined :

#define ast_pthread_create(a, b, c, d)                 \
    ast_pthread_create_stack(a, b, c, d,            \
        0, __FILE__, __FUNCTION__, __LINE__, #c)

I want to use it, but I do not what mean :

__FILE__, __FUNCTION__, __LINE__, #c, etc,

Can you help me?

THX

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</description>
    <dc:creator>Gnu Devel</dc:creator>
    <dc:date>2008-10-07T11:20:24</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/31957">
    <title>Re: How to keep the connection withAMIinterfaceactive?</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/31957</link>
    <description>_______________________________________________
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    <dc:creator>caif~</dc:creator>
    <dc:date>2008-10-07T06:16:36</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/31956">
    <title>Re: How to keep the connection withAMIinterfaceactive?</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/31956</link>
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    <dc:creator>caif~</dc:creator>
    <dc:date>2008-10-07T06:06:58</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/31955">
    <title>Re: How to keep the connection with AMIinterfaceactive?</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/31955</link>
    <description>Hi caif,

You may want to look into the "Ping" action in AMI, see
http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Ping

On Tue, Oct 7, 2008 at 09:29, caif~ &lt;caif&lt; at &gt;qq.com&gt; wrote:



</description>
    <dc:creator>Alwin Chan</dc:creator>
    <dc:date>2008-10-07T02:29:08</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/31954">
    <title>Re: How to keep the connection with AMIinterfaceactive?</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/31954</link>
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    <dc:creator>caif~</dc:creator>
    <dc:date>2008-10-07T01:29:15</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/31953">
    <title>IMAP_STORAGE issue with app_voicemail</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/31953</link>
    <description>Hello, I have scoured the internet and asked a few times in IRC but I  
can't figure this out. I'm hoping someone here can shed some light on  
this seemingly odd behavior.



I have asterisk 1.6.0 compiled with the IMAP_STORAGE set for  
voicemail, that all seemed to compile up just fine. The error I am  
getting in * cli is this:

[Oct  6 17:28:43] ERROR[20802]: app_voicemail.c:1999 mm_log: IMAP  
Error: Login aborted
[Oct  6 17:28:44] ERROR[20802]: app_voicemail.c:1757 init_mailstream:  
Can't connect to imap server {mail.crosscomm.net:143/imap/notls/user=vmail&lt; at &gt;crosscomm.net 
}INBOX
[Oct  6 17:28:44] ERROR[20802]: app_voicemail.c:1517 messagecount:  
Houston we have a problem - IMAP mailstream is NULL



The output I am seeing on the imap server it's trying to connect to is  
this:

Oct  6 17:29:49 gabriel imapd: LOGIN: DEBUG: ip=[::ffff: 
64.105.202.244], command=LOGOUT
Oct  6 17:29:49 gabriel imapd: LOGOUT, ip=[::ffff:64.105.202.244]



Here is my voicemail.conf, I removed the commented items for  
readability here:

[general]
imapserver=mail.crosscomm.net
imapflags=notls

[imap-voicemail]
brendan =&gt; brendan,System's Mailbox,,,imapuser=vmail&lt; at &gt;crosscomm.net| 
imapsecret=password


I'm quite baffled at this point and have no idea how to troubleshoot  
this. The very odd thing to me is that the imap server seems to be  
receiving a LOGOUT command first?
Thanks for any help you guys may be able to offer figuring this out.

Brendan Martens

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</description>
    <dc:creator>Brendan Martens</dc:creator>
    <dc:date>2008-10-06T21:33:38</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/31952">
    <title>Re: Asterisk on Windows (Subsystem for UNIX)</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/31952</link>
    <description>
That work did not get into 1.4, IIRC. It was in trunk of that time.
Current 1.6.0 should have it.

</description>
    <dc:creator>Tzafrir Cohen</dc:creator>
    <dc:date>2008-10-06T20:59:31</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/31951">
    <title>Re: Asterisk on Windows (Subsystem for UNIX)</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/31951</link>
    <description>
Luigi Rizzo did a bunch of commits a few months back in relation to asterisk
compilation under cygwin, but that isn't being actively maintained and has
probably been broken by subsequent commits.  Probably wouldn't take too much
work to get it up and running.

</description>
    <dc:creator>Sean Bright</dc:creator>
    <dc:date>2008-10-06T20:47:07</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/31950">
    <title>Re: Asterisk on Windows (Subsystem for UNIX)</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/31950</link>
    <description>
You (wrongly) assume nobody here is old enough to remember Xenix with
horror.

There's a reason nobody from MS even mention this sad episode, which
from technical POV makes even WfW look like a technological breakthrough ;-)

But we really digress...

</description>
    <dc:creator>Oron Peled</dc:creator>
    <dc:date>2008-10-06T20:31:11</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/31949">
    <title>Re: Asterisk on Windows (Subsystem for UNIX)</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/31949</link>
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    <dc:creator>Alex Dubinsky</dc:creator>
    <dc:date>2008-10-06T17:23:53</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/31948">
    <title>Re: FastAGI server</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/31948</link>
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    <dc:creator>Eliel Sardañons</dc:creator>
    <dc:date>2008-10-06T13:53:26</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/31947">
    <title>Re: FastAGI server</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/31947</link>
    <description>There's a Java one... Asterisk-java.org. I think there might be a C#
port of it too, if that gets you close to what you need.

Cheers,

Martin Smith, Systems Developer
martins&lt; at &gt;bebr.ufl.edu
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221 

 


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</description>
    <dc:creator>Martin Smith</dc:creator>
    <dc:date>2008-10-06T13:42:17</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/31946">
    <title>Re: How to keep the connection with AMI interface active?</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/31946</link>
    <description>

Yes, and yes.

</description>
    <dc:creator>Kevin P. Fleming</dc:creator>
    <dc:date>2008-10-06T13:12:24</dc:date>
  </item>
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