[Freeswitch-users] Proto specific hangup cause issue

regs at kinetix.gr regs at kinetix.gr
Tue Dec 9 08:38:21 PST 2008


That approach introduces a third party application to
the setup (in order to capture and parse tha SIP messages)
that adds a lot in terms of complexity and reliability ( and cpu
usage). Also it could become a nightmare when you use a
mix of protocols (iax, sip, h323) and technologies (openzap etc).

In the case of a live debugging session, capturing is the most useful tool
but if you want to troubleshoot based on historical data (CDRs) then you
need some detailing. In addition you don't have to fill your databases
with all the fields that FS gives you in an XML cdr. You could
only pick those which are of interest in a particular application.



Shelby Ramsey wrote:
> Hello,
>
> This is just my 2 cents ... but my experience has been that trying to 
> catch all of the various variables (i.e. from XML_CDR) or otherwise 
> can be a little trying (a row in your CDR database could be over 100 
> fields long!).  
>
> The best option here is to catch the UUID's for the 2 call legs, 
> capture all SIP messaging, parse and dump the messaging, and then 
> correlate the calls from the CDR from there.  
>
> Much easier than trying to do it from FS ... and most folks want to 
> see SIP captures anyway (very broad set of tools to debug).  
>
> Measuring things like ASR, PDD, etc in my opinion is much easier from 
> the raw messaging than trying to do something with FS CDR records.
>
>
>
> On Tue, Dec 9, 2008 at 9:19 AM, Apostolos Pantsiopoulos 
> <regs at kinetix.gr <mailto:regs at kinetix.gr>> wrote:
>
>
>         We are currently in the migration process from our
>     current system to a FS based setup. We are in the process of
>     adapting our billing and routing to FS. All the  CDRs (and variables)
>     related issues that we have been discussing on this mailing list
>     come from the need to extract the same level of information from FS as
>     we do with our current closed source proprietary system. So, we
>     chose FS because of the versatility it provides in every aspect (event
>     handling, config implementation etc.) and we strongly believe that all
>     these additions/fixes would be beneficial to many potential FS users.
>
>         We are at your disposal for more details in case you need
>     more information about what exactly we are trying to do. Basically,
>     our approach is from the VoIP carrier's point of view rather than the
>     PBX user's/implementor's. So, the details that we asked to be
>     introduced
>     to FS come from real life issues that we have faced during the
>     last few years
>     with various platforms and troubleshooting experiences with other
>     VoIP carriers.
>
>
>
>
>     Michael Collins wrote:
>>     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
>>     <regs at kinetix.gr> <mailto:regs at kinetix.gr> wrote:
>>       
>>>     "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 <regs at kinetix.gr> <mailto:regs at kinetix.gr>
>>>     wrote:
>>>         
>>>>     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 <regs at kinetix.gr> <mailto:regs at kinetix.gr>
>>>>     wrote:
>>>>           
>>>>>     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 <regs at kinetix.gr> <mailto:regs at kinetix.gr>
>>>>>     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 <regs at kinetix.gr> <mailto:regs at kinetix.gr>
>>>>>     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: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>>>>     -------------------------------------------
>>>>>
>>>>>
>>>>>     _______________________________________________
>>>>>     Freeswitch-users mailing list
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>>>>>
>>>>>
>>>>>     _______________________________________________
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>>>>>
>>>>>     --
>>>>>     -------------------------------------------
>>>>>     Apostolos Pantsiopoulos
>>>>>     Kinetix Tele.com R & D
>>>>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>>>>     -------------------------------------------
>>>>>
>>>>>     _______________________________________________
>>>>>     Freeswitch-users mailing list
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>>>>>     ________________________________
>>>>>     _______________________________________________
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>>>>>
>>>>>     --
>>>>>     -------------------------------------------
>>>>>     Apostolos Pantsiopoulos
>>>>>     Kinetix Tele.com R & D
>>>>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>>>>     -------------------------------------------
>>>>>
>>>>>     _______________________________________________
>>>>>     Freeswitch-users mailing list
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>>>>>
>>>>>             
>>>>     --
>>>>     Anthony Minessale II
>>>>
>>>>     FreeSWITCH http://www.freeswitch.org/
>>>>     ClueCon http://www.cluecon.com/
>>>>
>>>>     AIM: anthm
>>>>     MSN:anthony_minessale at hotmail.com <mailto:MSN:anthony_minessale at hotmail.com>
>>>>     GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com <mailto:GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com>
>>>>     IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>>>>
>>>>     FreeSWITCH Developer Conference
>>>>     sip:888 at conference.freeswitch.org
>>>>     iax:guest at conference.freeswitch.org/888 <mailto:iax:guest at conference.freeswitch.org/888>
>>>>     googletalk:conf+888 at conference.freeswitch.org <mailto:googletalk:conf+888 at conference.freeswitch.org>
>>>>     pstn:213-799-1400
>>>>
>>>>     ________________________________
>>>>     _______________________________________________
>>>>     Freeswitch-users mailing list
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>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>>>>
>>>>
>>>>     --
>>>>     -------------------------------------------
>>>>     Apostolos Pantsiopoulos
>>>>     Kinetix Tele.com R & D
>>>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>>>     -------------------------------------------
>>>>
>>>>     _______________________________________________
>>>>     Freeswitch-users mailing list
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>>>>     http://www.freeswitch.org
>>>>
>>>>           
>>>     --
>>>     Anthony Minessale II
>>>
>>>     FreeSWITCH http://www.freeswitch.org/
>>>     ClueCon http://www.cluecon.com/
>>>
>>>     AIM: anthm
>>>     MSN:anthony_minessale at hotmail.com <mailto:MSN:anthony_minessale at hotmail.com>
>>>     GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com <mailto:GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com>
>>>     IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>>>
>>>     FreeSWITCH Developer Conference
>>>     sip:888 at conference.freeswitch.org
>>>     iax:guest at conference.freeswitch.org/888 <mailto:iax:guest at conference.freeswitch.org/888>
>>>     googletalk:conf+888 at conference.freeswitch.org <mailto:googletalk:conf+888 at conference.freeswitch.org>
>>>     pstn:213-799-1400
>>>
>>>     ________________________________
>>>     _______________________________________________
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>>>
>>>     --
>>>     -------------------------------------------
>>>     Apostolos Pantsiopoulos
>>>     Kinetix Tele.com R & D
>>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>>     -------------------------------------------
>>>
>>>     _______________________________________________
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>>>
>>>         
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>>       
>
>
>     -- 
>     -------------------------------------------
>     Apostolos Pantsiopoulos
>     Kinetix Tele.com R & D
>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>     ------------------------------------------- 
>
>
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