[asterisk-dev] asterisk-dev Digest, Vol 64, Issue 73
Peter Tap
ptrtap at yahoo.com
Tue Nov 24 13:32:14 CST 2009
Not so fast.
I have tried it with and without pin options. It is the same result.
It takes anywhere from 10 seconds to 50 seconds after a successful connection for the hangup to occur.
Perhaps I am making a mistake in my configuration:
meetme.conf:
conf => 1234,1234
extensions.conf:
exten => 600,1,MeetMe(1234,s1)
Thank you for your help.
Regards,
Peter
From: Tilghman Lesher <tlesher at digium.com>
Subject: Re: [asterisk-dev] How to debug DAHDI pseudo timer problem..
To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
> The problem I am running into is that after a connection is established to
> the conference room, I get disconnected in a minute or two. Here is the
> output information while running asterisk with a lots of -v option.
>
> == Using SIP RTP CoS mark 5
> -- Executing [600 at FromCiscoPhone:1] MeetMe("SIP/101-b7b03298",
> "1234,s1") in new stack == Parsing '/etc/asterisk/meetme.conf': == Found
> -- Created MeetMe conference 1023 for conference '1234'
> -- <SIP/101-b7b03298> Playing 'conf-getpin.ulaw' (language 'en')
> -- Hungup 'DAHDI/pseudo-1700836616'
> == Spawn extension (FromCiscoPhone, 600, 1) exited non-zero on
> 'SIP/101-b7b03298'
Looks fairly clear-cut to me. Conference 1234 has a pin associated with it;
it prompted for the pin, you entered nothing, and it exited for lack of a pin.
This has nothing to do with DAHDI and everything to do with a failure to
follow instructions.
--
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com & www.asterisk.org
------------------------------
Message: 4
Date: Tue, 24 Nov 2009 08:49:26 +0100
From: "Olle E. Johansson" <oej at edvina.net>
Subject: Re: [asterisk-dev] CDR variables
To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
Message-ID: <2890F02D-7D7E-4DBB-8B4B-6978361800FC at edvina.net>
Content-Type: text/plain; charset=us-ascii
23 nov 2009 kl. 21.05 skrev Brent Thomson:
> Jamuel P. Starkey wrote:
>> Tilghman Lesher wrote:
>>> Given that nothing currently has a dash, and given that the dash character is
>>> not generally a good character in a database column name, I'd prefer if we
>>> eschew the use of any such separator in new CDR column names. Like
>>> "astrtpqosaudio".
>>>
>>>
>>
>> +1 agreed.
>
> While we're at it, can we exorcise 'call-limit' in the SIP realtime
> peers table as well?
That's a good one. It's already replaced in most cases with callcounter=yes,
but we can surely implement an alias to "call-limit" for those that still wants an
actual limit, not just blinking lamps.
Thanks.
/O
------------------------------
Message: 5
Date: Tue, 24 Nov 2009 09:00:19 +0100
From: "Olle E. Johansson" <oej at edvina.net>
Subject: Re: [asterisk-dev] CDR variables
To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
Message-ID: <6A973E4A-DAED-4B97-9731-B00457B50645 at edvina.net>
Content-Type: text/plain; charset=us-ascii
23 nov 2009 kl. 22.18 skrev Tilghman Lesher:
> On Monday 23 November 2009 02:05:19 pm Brent Thomson wrote:
>> Jamuel P. Starkey wrote:
>>> Tilghman Lesher wrote:
>>>> Given that nothing currently has a dash, and given that the dash
>>>> character is not generally a good character in a database column name,
>>>> I'd prefer if we eschew the use of any such separator in new CDR column
>>>> names. Like "astrtpqosaudio".
>>>
>>> +1 agreed.
>>
>> While we're at it, can we exorcise 'call-limit' in the SIP realtime
>> peers table as well?
>
> That field is already deprecated in favor of "callcounter={yes|no}". This
> change is in 1.6.0 and later. I expect "call-limit" will be removed in the
> 1.10 branch.
No, call-limit should still be around for those that need a call limit. The issue
was that most people just wanted to enable the call counter to get blinking lamps
and did not want any limit at all. Callcounter=yes/no is just a simplification
-enabling the counter without enforcing a limit.
As far as I know, the current policy is not to remove anything if we don't have to, right?
/O
------------------------------
Message: 6
Date: Tue, 24 Nov 2009 10:09:52 -0000
From: "Nick Lewis" <Nick.Lewis at atltelecom.com>
Subject: Re: [asterisk-dev] we all have our buttons - for asterisk-dev
it is political correctness - OT
To: <vhatz at kinetix.gr>, "Asterisk Developers Mailing List"
<asterisk-dev at lists.digium.com>
Message-ID: <3F97ACBE506F6849A6AEDA62A123928F385AD9 at email.atl.local>
Content-Type: text/plain; charset="us-ascii"
Vlasis Hatzistavrou (KTI) wrote:
>We are all sensitive about a few things, we all have our
>buttons as we say.
+1 for the underdog even though I think he is wrong
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------------------------------
Message: 7
Date: Tue, 24 Nov 2009 11:47:32 +0100
From: Kai Hoerner <kai at ciphron.de>
Subject: Re: [asterisk-dev] CDR variables
To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
Message-ID: <4B0BB9C4.6090602 at ciphron.de>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Olle E. Johansson wrote:
>> While we're at it, can we exorcise 'call-limit' in the SIP realtime
>> peers table as well?
>>
> That's a good one. It's already replaced in most cases with callcounter=yes,
> but we can surely implement an alias to "call-limit" for those that still wants an
> actual limit, not just blinking lamps.
>
+1
------------------------------
Message: 8
Date: Tue, 24 Nov 2009 17:41:27 +0530
From: Chandrakant Solanki <solanki.chandrakant at gmail.com>
Subject: [asterisk-dev] Asterist 1.6.0.13 - MeetMe - can_write ..??
To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
Message-ID:
<3bf515fb0911240411k8b31173kc1c04349ad78103b at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Hi
what is the purpose of can_write function 'n CONFFLAG_NO_AUDIO_UNTIL_UP in
app_meetme.c file...
I have setup 2 asterisk machine., on which user1 connect as admin and
another as normal.. with 'o' flag into meetme options..
when 2nd user will connect, it connect with IAX2 channel to user1... but i
found audio from admin to normal user i.e. user1 to user2 but vice versa it
is not possible...
--
Regards,
Chandrakant Solanki
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Message: 9
Date: Tue, 24 Nov 2009 07:48:28 -0500
From: "Andrew Kohlsmith (Mailing List Account)" <aklists at mixdown.ca>
Subject: Re: [asterisk-dev] Insulting source code comments in
main/channel.c
To: asterisk-dev at lists.digium.com
Message-ID: <200911240748.29113.aklists at mixdown.ca>
Content-Type: Text/Plain; charset="iso-8859-1"
On November 23, 2009 07:37:55 pm Vlasis Hatzistavrou (KTI) wrote:
> Was this comment really written by Mark Spencer? Hm... Interesting.
> That could explain why very few people commented on the actual comment
> itself but instead, many others bashed on those offended by it...
You really are something else.
The person who was originally offended has every right to be offended, as
we've all already concluded. The community has also spoken, and decided that
it's a ridiculous thing to be offended by. This is also fine.
Now you, Vlasis, are trying to pick conspiracy theories out of the air. You
really are something.
-A.
------------------------------
Message: 10
Date: Tue, 24 Nov 2009 14:41:41 +0100
From: Kai Hoerner <kai at ciphron.de>
Subject: Re: [asterisk-dev] Insulting source code comments in
main/channel.c
To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
Message-ID: <4B0BE295.4010202 at ciphron.de>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Hi all.
Jeez.. Can people please stop flaming on the -dev list?
Andrew:
If you would pay more attention to the content of the mails you answer
to, you would have noticed that Terry Wilson <twilson at digium.com>
originally pointed at Mark Spencer, not Vlasis. But this is totally not
the point here.
To the facts:
If someone really takes offense by this comment, he should move on and
do something.
Nobody will jump in and "solve" this "problem" for you. You must do it
yourself.
To get the proposed changes into the asterisk source tree, do the following:
1. stop bothering the mailing list -- we want to discuss technical
issues here
2. open an issue at http://bugs.digium.com
3. sign a disclaimer
4. provide a description of the problem
5. attach a patch that changes this line to whatever you think is most
appropriate.
Thanks.
Best Regards,
Kai H?rner
Andrew Kohlsmith (Mailing List Account) schrieb:
> On November 23, 2009 07:37:55 pm Vlasis Hatzistavrou (KTI) wrote:
>
>> Was this comment really written by Mark Spencer? Hm... Interesting.
>> That could explain why very few people commented on the actual comment
>> itself but instead, many others bashed on those offended by it...
>>
>
> You really are something else.
>
> The person who was originally offended has every right to be offended, as
> we've all already concluded. The community has also spoken, and decided that
> it's a ridiculous thing to be offended by. This is also fine.
>
> Now you, Vlasis, are trying to pick conspiracy theories out of the air. You
> really are something.
>
> -A.
>
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