<html><head><style type="text/css"><!-- DIV {margin:0px;} --></style></head><body><div style="font-family:times new roman,new york,times,serif;font-size:12pt">Not so fast.<br><br>I have tried it with and without pin options. It is the same result.<br><br>It takes anywhere from 10 seconds to 50 seconds after a successful connection for the hangup to occur.<br><br>Perhaps I am making a mistake in my configuration:<br><br>meetme.conf:<br>conf => 1234,1234<br><br>extensions.conf:<br>exten => 600,1,MeetMe(1234,s1)<br><br>Thank you for your help.<br><br>Regards,<br>Peter<br><br><div style="font-family: times new roman,new york,times,serif; font-size: 12pt;"><div style="font-family: arial,helvetica,sans-serif; font-size: 13px;">From: Tilghman Lesher <<a ymailto="mailto:tlesher@digium.com" href="mailto:tlesher@digium.com">tlesher@digium.com</a>><br>Subject: Re: [asterisk-dev] How to debug DAHDI pseudo timer problem..<br>To: Asterisk Developers
Mailing List <<a ymailto="mailto:asterisk-dev@lists.digium.com" href="mailto:asterisk-dev@lists.digium.com">asterisk-dev@lists.digium.com</a>><br>> The problem I am running into is that after a connection is established to<br>> the conference room, I get disconnected in a minute or two. Here is the<br>> output information while running asterisk with a lots of -v option.<br>><br>> == Using SIP RTP CoS mark 5<br>> -- Executing [600@FromCiscoPhone:1] MeetMe("SIP/101-b7b03298",<br>> "1234,s1") in new stack == Parsing '/etc/asterisk/meetme.conf': == Found<br>> -- Created MeetMe conference 1023 for conference '1234'<br>> -- <SIP/101-b7b03298> Playing 'conf-getpin.ulaw' (language 'en')<br>> -- Hungup 'DAHDI/pseudo-1700836616'<br>> == Spawn extension (FromCiscoPhone, 600, 1) exited non-zero on<br>> 'SIP/101-b7b03298'<br><br>Looks fairly
clear-cut to me. Conference 1234 has a pin associated with it;<br>it prompted for the pin, you entered nothing, and it exited for lack of a pin.<br>This has nothing to do with DAHDI and everything to do with a failure to<br>follow instructions.<br><br>-- <br>Tilghman Lesher<br>Digium, Inc. | Senior Software Developer<br>twitter: Corydon76 | IRC: Corydon76-dig (Freenode)<br><span>Check us out at: <a target="_blank" href="http://www.digium.com">www.digium.com</a> & <a target="_blank" href="http://www.asterisk.org">www.asterisk.org</a></span><br><br><br><br>------------------------------<br><br>Message: 4<br>Date: Tue, 24 Nov 2009 08:49:26 +0100<br>From: "Olle E. Johansson" <<a ymailto="mailto:oej@edvina.net" href="mailto:oej@edvina.net">oej@edvina.net</a>><br>Subject: Re: [asterisk-dev] CDR variables<br>To: Asterisk Developers Mailing List <<a ymailto="mailto:asterisk-dev@lists.digium.com"
href="mailto:asterisk-dev@lists.digium.com">asterisk-dev@lists.digium.com</a>><br>Message-ID: <<a ymailto="mailto:2890F02D-7D7E-4DBB-8B4B-6978361800FC@edvina.net" href="mailto:2890F02D-7D7E-4DBB-8B4B-6978361800FC@edvina.net">2890F02D-7D7E-4DBB-8B4B-6978361800FC@edvina.net</a>><br>Content-Type: text/plain; charset=us-ascii<br><br><br>23 nov 2009 kl. 21.05 skrev Brent Thomson:<br><br>> Jamuel P. Starkey wrote:<br>>> Tilghman Lesher wrote:<br>>>> Given that nothing currently has a dash, and given that the dash character is<br>>>> not generally a good character in a database column name, I'd prefer if we<br>>>> eschew the use of any such separator in new CDR column names. Like<br>>>> "astrtpqosaudio".<br>>>> <br>>>> <br>>> <br>>> +1 agreed.<br>> <br>> While we're at it, can we exorcise 'call-limit' in the SIP realtime <br>> peers table as well?<br><br>That's a
good one. It's already replaced in most cases with callcounter=yes,<br>but we can surely implement an alias to "call-limit" for those that still wants an<br>actual limit, not just blinking lamps.<br><br>Thanks.<br><br>/O<br><br><br>------------------------------<br><br>Message: 5<br>Date: Tue, 24 Nov 2009 09:00:19 +0100<br>From: "Olle E. Johansson" <<a ymailto="mailto:oej@edvina.net" href="mailto:oej@edvina.net">oej@edvina.net</a>><br>Subject: Re: [asterisk-dev] CDR variables<br>To: Asterisk Developers Mailing List <<a ymailto="mailto:asterisk-dev@lists.digium.com" href="mailto:asterisk-dev@lists.digium.com">asterisk-dev@lists.digium.com</a>><br>Message-ID: <<a ymailto="mailto:6A973E4A-DAED-4B97-9731-B00457B50645@edvina.net" href="mailto:6A973E4A-DAED-4B97-9731-B00457B50645@edvina.net">6A973E4A-DAED-4B97-9731-B00457B50645@edvina.net</a>><br>Content-Type: text/plain; charset=us-ascii<br><br><br>23 nov 2009 kl. 22.18 skrev Tilghman
Lesher:<br><br>> On Monday 23 November 2009 02:05:19 pm Brent Thomson wrote:<br>>> Jamuel P. Starkey wrote:<br>>>> Tilghman Lesher wrote:<br>>>>> Given that nothing currently has a dash, and given that the dash<br>>>>> character is not generally a good character in a database column name,<br>>>>> I'd prefer if we eschew the use of any such separator in new CDR column<br>>>>> names. Like "astrtpqosaudio".<br>>>> <br>>>> +1 agreed.<br>>> <br>>> While we're at it, can we exorcise 'call-limit' in the SIP realtime<br>>> peers table as well?<br>> <br>> That field is already deprecated in favor of "callcounter={yes|no}". This<br>> change is in 1.6.0 and later. I expect "call-limit" will be removed in the<br>> 1.10 branch.<br>No, call-limit should still be around for those that need a call limit. The issue<br>was that most people
just wanted to enable the call counter to get blinking lamps<br>and did not want any limit at all. Callcounter=yes/no is just a simplification<br>-enabling the counter without enforcing a limit.<br><br>As far as I know, the current policy is not to remove anything if we don't have to, right?<br><br>/O<br><br><br>------------------------------<br><br>Message: 6<br>Date: Tue, 24 Nov 2009 10:09:52 -0000<br>From: "Nick Lewis" <<a ymailto="mailto:Nick.Lewis@atltelecom.com" href="mailto:Nick.Lewis@atltelecom.com">Nick.Lewis@atltelecom.com</a>><br>Subject: Re: [asterisk-dev] we all have our buttons - for asterisk-dev<br> it is political correctness - OT<br>To: <<a ymailto="mailto:vhatz@kinetix.gr" href="mailto:vhatz@kinetix.gr">vhatz@kinetix.gr</a>>, "Asterisk Developers Mailing List"<br> <<a ymailto="mailto:asterisk-dev@lists.digium.com"
href="mailto:asterisk-dev@lists.digium.com">asterisk-dev@lists.digium.com</a>><br>Message-ID: <<a ymailto="mailto:3F97ACBE506F6849A6AEDA62A123928F385AD9@email.atl.local" href="mailto:3F97ACBE506F6849A6AEDA62A123928F385AD9@email.atl.local">3F97ACBE506F6849A6AEDA62A123928F385AD9@email.atl.local</a>><br>Content-Type: text/plain; charset="us-ascii"<br><br>Vlasis Hatzistavrou (KTI) wrote:<br>>We are all sensitive about a few things, we all have our <br>>buttons as we say.<br>+1 for the underdog even though I think he is wrong<br><br>_____________________________________________________________________<br>This message has been checked for all known viruses by Star Internet delivered through the MessageLabs Virus Control Centre.<br>_____________________________________________________________________<br>Disclaimer of Liability<br>ATL Telecom Ltd shall not be held liable for any improper or incorrect use of the information
described and/or contained herein and assumes no responsibility for anyones use of the information. In no event shall ATL Telecom Ltd be liable for any direct, indirect, incidental, special, exemplary, or consequential damages (including, but not limited to, procurement or substitute goods or services; loss of use, data, or profits; or business interruption) however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence or otherwise) arising in any way out of the use of this system, even if advised of the possibility of such damage.<br><br>Registered Office: ATL Telecom Ltd, Fountain Lane, St. Mellons Cardiff, CF3 0FB<br>Registered in Wales Number 4335781<br><br>All goods and services supplied by ATL Telecom Ltd are supplied subject to ATL Telecom Ltd standard terms and conditions, available upon request.<br><br><br><br>------------------------------<br><br>Message:
7<br>Date: Tue, 24 Nov 2009 11:47:32 +0100<br>From: Kai Hoerner <<a ymailto="mailto:kai@ciphron.de" href="mailto:kai@ciphron.de">kai@ciphron.de</a>><br>Subject: Re: [asterisk-dev] CDR variables<br>To: Asterisk Developers Mailing List <<a ymailto="mailto:asterisk-dev@lists.digium.com" href="mailto:asterisk-dev@lists.digium.com">asterisk-dev@lists.digium.com</a>><br>Message-ID: <<a ymailto="mailto:4B0BB9C4.6090602@ciphron.de" href="mailto:4B0BB9C4.6090602@ciphron.de">4B0BB9C4.6090602@ciphron.de</a>><br>Content-Type: text/plain; charset=ISO-8859-1; format=flowed<br><br>Olle E. Johansson wrote:<br>>> While we're at it, can we exorcise 'call-limit' in the SIP realtime <br>>> peers table as well?<br>>> <br>> That's a good one. It's already replaced in most cases with callcounter=yes,<br>> but we can surely implement an alias to "call-limit" for those that still wants an<br>> actual limit, not just
blinking lamps.<br>> <br>+1<br><br><br><br>------------------------------<br><br>Message: 8<br>Date: Tue, 24 Nov 2009 17:41:27 +0530<br>From: Chandrakant Solanki <<a ymailto="mailto:solanki.chandrakant@gmail.com" href="mailto:solanki.chandrakant@gmail.com">solanki.chandrakant@gmail.com</a>><br>Subject: [asterisk-dev] Asterist 1.6.0.13 - MeetMe - can_write ..??<br>To: Asterisk Developers Mailing List <<a ymailto="mailto:asterisk-dev@lists.digium.com" href="mailto:asterisk-dev@lists.digium.com">asterisk-dev@lists.digium.com</a>><br>Message-ID:<br> <<a ymailto="mailto:3bf515fb0911240411k8b31173kc1c04349ad78103b@mail.gmail.com" href="mailto:3bf515fb0911240411k8b31173kc1c04349ad78103b@mail.gmail.com">3bf515fb0911240411k8b31173kc1c04349ad78103b@mail.gmail.com</a>><br>Content-Type: text/plain; charset="iso-8859-1"<br><br>Hi<br><br>what is the purpose of can_write function 'n CONFFLAG_NO_AUDIO_UNTIL_UP
in<br>app_meetme.c file...<br><br>I have setup 2 asterisk machine., on which user1 connect as admin and<br>another as normal.. with 'o' flag into meetme options..<br><br>when 2nd user will connect, it connect with IAX2 channel to user1... but i<br>found audio from admin to normal user i.e. user1 to user2 but vice versa it<br>is not possible...<br><br><br>-- <br>Regards,<br><br>Chandrakant Solanki<br>-------------- next part --------------<br>An HTML attachment was scrubbed...<br>URL: <a href="http://lists.digium.com/pipermail/asterisk-dev/attachments/20091124/5c399f82/attachment-0001.htm" target="_blank">http://lists.digium.com/pipermail/asterisk-dev/attachments/20091124/5c399f82/attachment-0001.htm</a> <br><br>------------------------------<br><br>Message: 9<br>Date: Tue, 24 Nov 2009 07:48:28 -0500<br>From: "Andrew Kohlsmith (Mailing List Account)" <<a ymailto="mailto:aklists@mixdown.ca"
href="mailto:aklists@mixdown.ca">aklists@mixdown.ca</a>><br>Subject: Re: [asterisk-dev] Insulting source code comments in<br> main/channel.c<br>To: <a ymailto="mailto:asterisk-dev@lists.digium.com" href="mailto:asterisk-dev@lists.digium.com">asterisk-dev@lists.digium.com</a><br>Message-ID: <<a ymailto="mailto:200911240748.29113.aklists@mixdown.ca" href="mailto:200911240748.29113.aklists@mixdown.ca">200911240748.29113.aklists@mixdown.ca</a>><br>Content-Type: Text/Plain; charset="iso-8859-1"<br><br>On November 23, 2009 07:37:55 pm Vlasis Hatzistavrou (KTI) wrote:<br>> Was this comment really written by Mark Spencer? Hm... Interesting.<br>> That could explain why very few people commented on the actual comment<br>> itself but instead, many others bashed on those offended by it...<br><br>You really are something else.<br><br>The person who was originally offended has every right to be offended, as <br>we've all
already concluded. The community has also spoken, and decided that <br>it's a ridiculous thing to be offended by. This is also fine.<br><br>Now you, Vlasis, are trying to pick conspiracy theories out of the air. You <br>really are something.<br><br>-A.<br><br><br><br>------------------------------<br><br>Message: 10<br>Date: Tue, 24 Nov 2009 14:41:41 +0100<br>From: Kai Hoerner <<a ymailto="mailto:kai@ciphron.de" href="mailto:kai@ciphron.de">kai@ciphron.de</a>><br>Subject: Re: [asterisk-dev] Insulting source code comments in<br> main/channel.c<br>To: Asterisk Developers Mailing List <<a ymailto="mailto:asterisk-dev@lists.digium.com" href="mailto:asterisk-dev@lists.digium.com">asterisk-dev@lists.digium.com</a>><br>Message-ID: <<a ymailto="mailto:4B0BE295.4010202@ciphron.de" href="mailto:4B0BE295.4010202@ciphron.de">4B0BE295.4010202@ciphron.de</a>><br>Content-Type: text/plain; charset=ISO-8859-1;
format=flowed<br><br>Hi all.<br><br>Jeez.. Can people please stop flaming on the -dev list?<br><br>Andrew:<br>If you would pay more attention to the content of the mails you answer <br>to, you would have noticed that Terry Wilson <<a ymailto="mailto:twilson@digium.com" href="mailto:twilson@digium.com">twilson@digium.com</a>> <br>originally pointed at Mark Spencer, not Vlasis. But this is totally not <br>the point here.<br><br>To the facts:<br><br>If someone really takes offense by this comment, he should move on and <br>do something.<br>Nobody will jump in and "solve" this "problem" for you. You must do it <br>yourself.<br><br>To get the proposed changes into the asterisk source tree, do the following:<br><br>1. stop bothering the mailing list -- we want to discuss technical <br>issues here<br>2. open an issue at <a href="http://bugs.digium.com" target="_blank">http://bugs.digium.com</a><br>3. sign a disclaimer<br>4. provide a description of the
problem<br>5. attach a patch that changes this line to whatever you think is most <br>appropriate.<br><br>Thanks.<br><br><br>Best Regards,<br><br>Kai H?rner<br><br><br><br>Andrew Kohlsmith (Mailing List Account) schrieb:<br>> On November 23, 2009 07:37:55 pm Vlasis Hatzistavrou (KTI) wrote:<br>> <br>>> Was this comment really written by Mark Spencer? Hm... Interesting.<br>>> That could explain why very few people commented on the actual comment<br>>> itself but instead, many others bashed on those offended by it...<br>>> <br>><br>> You really are something else.<br>><br>> The person who was originally offended has every right to be offended, as <br>> we've all already concluded. The community has also spoken, and decided that <br>> it's a ridiculous thing to be offended by. This is also fine.<br>><br>> Now you, Vlasis, are trying to pick conspiracy theories out of the
air. You <br>> really are something.<br>><br>> -A.<br>><br>> _______________________________________________<br>> --Bandwidth and Colocation Provided by <a href="http://www.api-digital.com--" target="_blank">http://www.api-digital.com--</a><br>><br>> asterisk-dev mailing list<br>> To UNSUBSCRIBE or update options visit:<br>> <a href="http://lists.digium.com/mailman/listinfo/asterisk-dev" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-dev</a><br>><br>> <br><br><br><br><br>------------------------------<br><br>_______________________________________________<br>--Bandwidth and Colocation Provided by <a href="http://www.api-digital.com--" target="_blank">http://www.api-digital.com--</a><br><br>asterisk-dev mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-dev"
target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-dev</a><br><br>End of asterisk-dev Digest, Vol 64, Issue 73<br>********************************************<br></div></div>
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