[OpenSIPS-Users] Calls being disconnected

Thu Jul 5 06:31:16 CEST 2012

Hi,

You don't give the logs and sip traces - its no fun for you or anyone here.
I recommend you to change your FreeSwitch/Asterisk settings to stop
re-invites (canreinvite=no) and in newer version (directmedia=no) - See if
that helps.

I'll again repeat that either you don't engage RTPproxy in the call at all,
or you send the Media via the RTPproxy by setting your A-leg & B-leg
telephony engines. that'll solve your problem

I hope you can handle it from here now.

Regards,
Sammy

On Wed, Jul 4, 2012 at 10:15 PM, Rodrigo Ferreira <
rodrigo.ferreira at vipway.net.br> wrote:

>   I’m trying to debug those calls using wireshark and VoIP calls analisys
> ...
>
> And i guess it might be a network problem, maybe some packets being
> dropped .. Is there any chance of being that?
> Because my Opensips box and my FreePBX (asterisk) they are in the same
> network, just passing through a switch, among with other servers (including
> another opensips who is giving me the same problem)
>
>  *From:* Rodrigo Ferreira <rodrigo.ferreira at vipway.net.br>
> *Sent:* Wednesday, July 04, 2012 9:10 AM
> *To:* OpenSIPS users mailling list <users at lists.opensips.org>
> *Subject:* Re: [OpenSIPS-Users] Calls being disconnected
>
>   That was my bad, its different IPs, but I wrote it wrong.
>
> I will get some new traces today, and I will write the IPs correctly
>
>  *From:* SamyGo <govoiper at gmail.com>
> *Sent:* Wednesday, July 04, 2012 1:54 AM
> *To:* OpenSIPS users mailling list <users at lists.opensips.org>
> *Subject:* Re: [OpenSIPS-Users] Calls being disconnected
>
> I will definitely try to read the SIP traces. I love trying to solve the
> puzzle but the very first line !!
>
> U 201.71.XXX.XXX:5060 -> 201.71.XXX.XXX:5060
>
> Is it the same server originating SIP INVITE sending to the same server !!
> if not please differentiate the two addresses like.
>
> U 201.71.XXX.XXX:5060 -> 201.71.XXX.YYY:5060
>
> Goodness , its FBX,OpenSIPS, gatewaySTS, and even Asterisk 1.8 -- are they
> all using the same IP !!
>
> Caller(FPBX)<--->OpenSIPS+RTPproxy<---->gatewaySTS<---->Asterisk
>    ||<============================RTP===============>||
>
> Please carefully use different IP addresses of all the SIP servers/UACs in
> your particular call scenario. What I'm still suspecting with more
> intensity is that somehow the RTPproxy is called but Asterisk and FPBX
> negotiate media directly in SDPs and bam..RTPproxy gets angry at this
> insulting attitude and disconnects the call !! That's my guess.
>
> OpenSIPS logs along with the new SIP traces will help clarify the real
> cause.
>
> Thanks
> Sammy
>
> On Tue, Jul 3, 2012 at 7:23 PM, Rodrigo Ferreira <
> rodrigo.ferreira at vipway.net.br> wrote:
>
>>   I’m attaching the log that I got from a call that was disconnected
>>
>>  *From:* Rodrigo Ferreira <rodrigo.ferreira at vipway.net.br>
>> *Sent:* Tuesday, July 03, 2012 11:09 AM
>>  *To:* OpenSIPS users mailling list <users at lists.opensips.org>
>> *Subject:* Re: [OpenSIPS-Users] Calls being disconnected
>>
>>    I’m getting the logs, but let me ask you something, force_rport will
>> help me?
>>
>>  *From:* SamyGo <govoiper at gmail.com>
>> *Sent:* Tuesday, July 03, 2012 8:18 AM
>> *To:* OpenSIPS users mailling list <users at lists.opensips.org>
>> *Subject:* Re: [OpenSIPS-Users] Calls being disconnected
>>
>> Specifically talking about one where call gets dropped - what is the
>> network scenario !?  Make one such call and take a sip trace on the
>> opensips server using tcpdump or sipgrep. Share the pcap. As well as the
>> opensips logs please.
>>
>>
>>  On Tue, Jul 3, 2012 at 4:12 PM, Rodrigo Ferreira <
>> rodrigo.ferreira at vipway.net.br> wrote:
>>
>>>   I’ve a lot of different scenarios through my network, I have clients
>>> behind NAT, from other networks, because I work for a telephony company.
>>>
>>>  *From:* SamyGo <govoiper at gmail.com>
>>> *Sent:* Tuesday, July 03, 2012 8:02 AM
>>>  *To:* OpenSIPS users mailling list <users at lists.opensips.org>
>>>  *Subject:* Re: [OpenSIPS-Users] Calls being disconnected
>>>
>>>  RTPproxy - thats what I suspected. What i'm guessing is that RTPproxy
>>> is engaged but somehow the end points receive each other's Direct IPs in
>>> SDP...hmm...tell me if your both end points are on the same LAN and server
>>> is placed somewhere in internet !!
>>>
>>> On Tue, Jul 3, 2012 at 3:58 PM, Rodrigo Ferreira <
>>> rodrigo.ferreira at vipway.net.br> wrote:
>>>
>>>>   I will take a look at my rtp .. I’m using rtpproxy on my box, I will
>>>> try to get some logs today too
>>>>
>>>>  *From:* SamyGo <govoiper at gmail.com>
>>>> *Sent:* Tuesday, July 03, 2012 4:34 AM
>>>> *To:* aamir chougule <aamir_ryu at yahoo.com> ; OpenSIPS users mailling
>>>> list <users at lists.opensips.org>
>>>> *Subject:* Re: [OpenSIPS-Users] Calls being disconnected
>>>>
>>>>  Besides logs and any sip traces as Aamir stated it'd be hard to pin
>>>> point anything. One of the big reason for automatically dropping an
>>>> established call specially after a specific amount of time i.e 30
>>>> seconds..thats because of lack of RTPs flowing through the server and if
>>>> media-relaying tools are used they will send hangup to both ends, even
>>>> though both ends have their RTP flowing *directly*.
>>>>
>>>> Regards,
>>>> Sammy
>>>>
>>>> On Tue, Jul 3, 2012 at 12:21 PM, aamir chougule <aamir_ryu at yahoo.com>wrote:
>>>>
>>>>>  Hi Rodrigo,
>>>>>
>>>>> The only which I know is after the call gets connected i.e. 200OK from
>>>>> the other end there should always be an ACK for that 200OK, if there is no
>>>>> ACK for the 200OK then the call will get disconnected after sometime.
>>>>>
>>>>> Can you please give us the logs for the call by tracing it through
>>>>> ngrep tool because without any sip traces or opensips log we can't tell
>>>>> what is going on within the box.
>>>>>
>>>>> Regards,
>>>>>
>>>>> Aamir Chougule
>>>>> Cell: 09167989111
>>>>>
>>>>>   ------------------------------
>>>>> *From:* Rodrigo Ferreira <rodrigo.ferreira at vipway.net.br>
>>>>> *To:* OpenSIPS users mailling list <users at lists.opensips.org>
>>>>> *Sent:* Monday, 2 July 2012 10:40 PM
>>>>> *Subject:* [OpenSIPS-Users] Calls being disconnected
>>>>>
>>>>>   Hey guys,
>>>>>
>>>>> I’m having problems with calls being “disconnected” and I dont know
>>>>> where I can start to look at, because on my CDR all those calls ends with a
>>>>> 200OK, but that isnt true, because you are talking all the suddenly the
>>>>> call stop.
>>>>>
>>>>> Any ideas where I should start looking at?
>>>>>
>>>>> Engº Rodrigo Ferreira
>>>>> Supervisor de Telefonia
>>>>> VIPWay Telecom
>>>>> Tel.: +55 13 4010-1000
>>>>> Cel.: +55 13 8136-5839
>>>>>
>>>>> _______________________________________________
>>>>> Users mailing list
>>>>> Users at lists.opensips.org
>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
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>>>>>
>>>>>
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