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[Freeswitch-users] Two Users Registered ..But calls only going one way


 
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davidwaf at gmail.com
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PostPosted: Thu Feb 11, 2016 4:47 am    Post subject: [Freeswitch-users] Two Users Registered ..But calls only goi Reply with quote

Hi all,I have two users who registered in the same domain: user A and B.
A can call B just fine. When B tries to call A, there is silence (no ringback)..then after sometime the call goes into voice mail. A never receives the call.  Please see the call trace here:




http://pastebin.com/gWrrS4zw


Am not sure what is causing it.


Regards

--
David W
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mike at jerris.com
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PostPosted: Thu Feb 11, 2016 11:22 am    Post subject: [Freeswitch-users] Two Users Registered ..But calls only goi Reply with quote

Sip trace would help... is call forwarding turned on on the phone?
Quote:
On Feb 11, 2016, at 3:46 AM, David Wafula <davidwaf@gmail.com (davidwaf@gmail.com)> wrote:
Hi all,I have two users who registered in the same domain: user A and B.
A can call B just fine. When B tries to call A, there is silence (no ringback)..then after sometime the call goes into voice mail. A never receives the call. Please see the call trace here:


http://pastebin.com/gWrrS4zw

Am not sure what is causing it.


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ahabiba at gmail.com
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PostPosted: Thu Feb 11, 2016 1:07 pm    Post subject: [Freeswitch-users] Two Users Registered ..But calls only goi Reply with quote

Are you using TLS? is GS configures with nat configuration correctly?
Quote:
From: Michael Jerris <mike@jerris.com (mike@jerris.com)>
Subject: Re: [Freeswitch-users] Two Users Registered ..But calls only going one way
Date: February 11, 2016 at 7:21:11 PM GMT+3
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)>
Reply-To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)>
Sip trace would help... is call forwarding turned on on the phone?
Quote:
On Feb 11, 2016, at 3:46 AM, David Wafula <davidwaf@gmail.com (davidwaf@gmail.com)> wrote:
Hi all,I have two users who registered in the same domain: user A and B.
A can call B just fine. When B tries to call A, there is silence (no ringback)..then after sometime the call goes into voice mail. A never receives the call. Please see the call trace here:


http://pastebin.com/gWrrS4zw

Am not sure what is causing it.




From: mohammed shafeeque <shafeeq.v@gmail.com (shafeeq.v@gmail.com)>
Subject: Re: [Freeswitch-users] Oneway audio issues in freeswitch
Date: February 11, 2016 at 7:28:45 PM GMT+3
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)>
Reply-To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)>
Surprised that no one else experienced this problem. Can anyone give any hint. Really Dont want to move back to 1.4.x
On Thu, Feb 11, 2016 at 7:44 AM, Luis Daniel Lucio Quiroz <luis.daniel.lucio@gmail.com (luis.daniel.lucio@gmail.com)> wrote:
Quote:
As a rule of dumb, try turning on rport
Le 10 févr. 2016 8:55 PM, "Ítalo Rossi" <italo@freeswitch.org (italo@freeswitch.org)> a écrit :
Quote:
You need to look at the sip signaling to see what's going on
On Wed, Feb 10, 2016 at 5:59 PM, mohammed shafeeque <shafeeq.v@gmail.com (shafeeq.v@gmail.com)> wrote:
Quote:
Hello AllWe are getting one way audio issues with some softphones and grandstream phones behind nat registerd to our freeswitch server. Here is scenario: Grandstream call any extensions (one way audio) Any extension call Grandstream ( Audio works just fine)We have tried multiple softphones and the result is same. Everything was working fine with 1.4.18 and 1.6.2. We were having DTMF issue with 1.6.2, upgraded to 1.6.5 and now one way audio issue started with an upgrade to freeswitch.
Any help or hint will be much appreciated.
Thank you,

_________________________________________________________________________Professional FreeSWITCH Consulting Services:consulting@freeswitch.org (consulting@freeswitch.org)FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org


-- Ítalo Rossiitalo@freeswitch.org (italo@freeswitch.org)



_________________________________________________________________________Professional FreeSWITCH Consulting Services:consulting@freeswitch.org (consulting@freeswitch.org)FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org



_________________________________________________________________________Professional FreeSWITCH Consulting Services:consulting@freeswitch.org (consulting@freeswitch.org)FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org



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steveayre at gmail.com
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PostPosted: Thu Feb 11, 2016 5:43 pm    Post subject: [Freeswitch-users] Two Users Registered ..But calls only goi Reply with quote

See the contents of the Contact header in A's REGISTER messages. They tell FreeSWITCH where to send the call to.

NAT can confuse matters when picking the correct value though. It needs to be the external IP & port of the NAT router that the internal IP/port of the SIP messages are mapped to. Sometimes the phone will put the internal details instead which aren't routable externally. Sometimes it can detect it correctly (eg via STUN). If it can't some routers will contain a SIP ALG that will rewrite the header for a phone sending the internal ip/port to the external ip/port, but sometimes this can cause more problems than it solves if it doesn't do this correctly and it can't modify the packet if you're using TLS.


On top of that that internal to external port mapping will expire on that NAT router if you don't re-REGISTER frequently enough so that could stop the INVITE getting through even if you're sending to the correct place.


If you're having issues like that getting the SIP packets through then it's likely 


If you can't fix it on the phone/router then you can also look at the NDLB (no device left behind) options. For example there's one that'll use the address the REGISTER is received from instead of the Contact header. This differs from how SIP is supposed to work but works in most cases (usually a phone will ask you to call it directly not via a proxy or route you elsewhere).



On 11 February 2016 at 09:46, David Wafula <davidwaf@gmail.com (davidwaf@gmail.com)> wrote:
Quote:
Hi all,I have two users who registered in the same domain: user A and B.
A can call B just fine. When B tries to call A, there is silence (no ringback)..then after sometime the call goes into voice mail. A never receives the call.  Please see the call trace here:




http://pastebin.com/gWrrS4zw


Am not sure what is causing it.


Regards

--
David W



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http://www.freeswitchsolutions.com

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