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<br>
We are currently in the migration process from our<br>
current system to a FS based setup. We are in the process of <br>
adapting our billing and routing to FS. All the CDRs (and variables) <br>
related issues that we have been discussing on this mailing list <br>
come from the need to extract the same level of information from FS as<br>
we do with our current closed source proprietary system. So, we<br>
chose FS because of the versatility it provides in every aspect (event<br>
handling, config implementation etc.) and we strongly believe that all<br>
these additions/fixes would be beneficial to many potential FS users.<br>
<br>
We are at your disposal for more details in case you need<br>
more information about what exactly we are trying to do. Basically, <br>
our approach is from the VoIP carrier's point of view rather than the<br>
PBX user's/implementor's. So, the details that we asked to be introduced<br>
to FS come from real life issues that we have faced during the last few
years<br>
with various platforms and troubleshooting experiences with other VoIP
carriers.<br>
<br>
<br>
<br>
Michael Collins wrote:
<blockquote
cite="mid:87f2f3b90812090644i304d72e7u61dc397ab897df50@mail.gmail.com"
type="cite">
<pre wrap="">Thanks for your feedback. It definitely helps to know not only what
you need FS to do but why you need it to do so.
Do you have FS in production right now? Just curious.
Thanks,
MC
On Tue, Dec 9, 2008 at 12:21 AM, Apostolos Pantsiopoulos
<a class="moz-txt-link-rfc2396E" href="mailto:regs@kinetix.gr"><regs@kinetix.gr></a> wrote:
</pre>
<blockquote type="cite">
<pre wrap="">"I already added 2 patches for you right. Just be clear about what you
want."
And I am grateful of that.
"it is protocol neutral, that's why it starts with sip_"
I didn't know that. I thought that the sip_ variables are protocol specific.
So one would expect there to be an iax_hangup_disposition,
woomera_hangup_disposition etc?
"Maybe you should beat around the bush less with your "requirements" for
your application you are expecting me to support for you."
I am just trying to gather statistics for my providers as I would with any
VoIP softswitch. (hangup causes per terminator per destination)
I don't think that this is a specific "application" rather than a general
necessity for VoIP carriers. It is also very useful for troubleshooting
purposes : when I look at my CDRs to find a call that I got a complain for,
I want to be able to tell if it was me or the provider who
hanged up and gave a specific hangup cause, so that I can troubleshoot the
issue better.
"Just be clear about what you want."
I want FS to reach that level of detailing and maturity in all aspects so
that it could be the softswitch of choice by any VoIP entrepreneur
(or hobbyist) and it is my strong belief that this can only be done by the
community giving feedback to the programmers about what
they find useful or not (i.e. experience from real-life situations). The
patches that you made the last few days were not intended for
me exclusively but for anyone that will face the same situations using FS.
If you want the community to stop sending feedback about
features/improvements you may as well close down this mailing list or just
use it as an announcement board.
I wish I was a c programmer and get involved with the project actively. But
I am not. And as far as I can tell most of the registered users
in this list aren't either. So they only way we can help is by testing and
suggesting.
Anthony Minessale wrote:
it is protocol neutral, that's why it starts with sip_
the variable can be any of:
send_bye
recv_bye
send_cancel
send_refuse
using that value you can determine the information you asked. I answered
your specific question which was:
determining "which side hanged up". Maybe you should beat around the bush
less with your "requirements" for your application you are expecting me to
support for you.
I already added 2 patches for you right. Just be clear about what you want.
On Mon, Dec 8, 2008 at 8:13 AM, Apostolos Pantsiopoulos <a class="moz-txt-link-rfc2396E" href="mailto:regs@kinetix.gr"><regs@kinetix.gr></a>
wrote:
</pre>
<blockquote type="cite">
<pre wrap="">Not necessarily. For instance I got a "send_cancel" when the
calling party hanged up before the other party could pick up.
Also, shouldn't something like that be protocol/technology
neutral?
Anthony Minessale wrote:
sip_hangup_disposition will be set to recv_bye on the side that was
hungup.
On Mon, Dec 8, 2008 at 4:11 AM, Apostolos Pantsiopoulos <a class="moz-txt-link-rfc2396E" href="mailto:regs@kinetix.gr"><regs@kinetix.gr></a>
wrote:
</pre>
<blockquote type="cite">
<pre wrap="">Any updates about the "which side hanged up" potential variable?
Michael S Collins wrote:
Makes sense. I will look into this.
-MC
On Dec 5, 2008, at 8:17 AM, Apostolos Pantsiopoulos <a class="moz-txt-link-rfc2396E" href="mailto:regs@kinetix.gr"><regs@kinetix.gr></a>
wrote:
I am sending this second email to request/suggest/enquire about something
relevant :
Wouldn't it be useful to know which end of a specific call leg send the
protocol
specific hangup cause? Otherwise it would be difficult to understand what
really happened.
Michael S Collins wrote:
I will do some research on this and let you know what I find out.
Question: are these internal calls or pstn or ?? Just curious about
your environment.
Thanks,
MC
On Dec 5, 2008, at 4:23 AM, Apostolos Pantsiopoulos <a class="moz-txt-link-rfc2396E" href="mailto:regs@kinetix.gr"><regs@kinetix.gr></a>
wrote:
The proto_specific_hangup_cause is missing on one of the two
call legs. When the caller hangs up it is missing from the a-leg CDR.
When the callee hangs up it is missing from the b-leg CDR. Is this
nornal?
And another question : what piece of info could inform me about who
hanged up?
--
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: <a class="moz-txt-link-abbreviated" href="mailto:regs@kinetix.gr">regs@kinetix.gr</a>
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</pre>
</blockquote>
<pre wrap="">
--
Anthony Minessale II
FreeSWITCH <a class="moz-txt-link-freetext" href="http://www.freeswitch.org/">http://www.freeswitch.org/</a>
ClueCon <a class="moz-txt-link-freetext" href="http://www.cluecon.com/">http://www.cluecon.com/</a>
AIM: anthm
<a class="moz-txt-link-abbreviated" href="mailto:MSN:anthony_minessale@hotmail.com">MSN:anthony_minessale@hotmail.com</a>
<a class="moz-txt-link-abbreviated" href="mailto:GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com">GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com</a>
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
<a class="moz-txt-link-freetext" href="sip:888@conference.freeswitch.org">sip:888@conference.freeswitch.org</a>
<a class="moz-txt-link-abbreviated" href="mailto:iax:guest@conference.freeswitch.org/888">iax:guest@conference.freeswitch.org/888</a>
<a class="moz-txt-link-abbreviated" href="mailto:googletalk:conf+888@conference.freeswitch.org">googletalk:conf+888@conference.freeswitch.org</a>
pstn:213-799-1400
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email: <a class="moz-txt-link-abbreviated" href="mailto:regs@kinetix.gr">regs@kinetix.gr</a>
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</pre>
</blockquote>
<pre wrap="">
--
Anthony Minessale II
FreeSWITCH <a class="moz-txt-link-freetext" href="http://www.freeswitch.org/">http://www.freeswitch.org/</a>
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FreeSWITCH Developer Conference
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</pre>
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<pre wrap=""><!---->
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</blockquote>
<br>
<br>
<pre class="moz-signature" cols="72">--
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Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: <a class="moz-txt-link-abbreviated" href="mailto:regs@kinetix.gr">regs@kinetix.gr</a>
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