[OpenSIPS-Users] FW: Audio and Video not working for different otuside netrwork

Rajesh Babu rajesh.babu at goodcoresoft.com
Fri Sep 27 07:32:29 CEST 2013


Hi Mike,

 

  I followed your Link and cleard all the issues that where shown on the
config file,

 

Now in /var/log/messages 

 

I am getting 

 

Sep 27 21:30:52 centos64 /usr/local/server/pbx/sbin/opensips[11066]:
INFO:core:evi_publish_event: Registered event <E_ACC_EVENT(3)>

Sep 27 21:30:52 centos64 /usr/local/server/pbx/sbin/opensips[11066]:
INFO:core:evi_publish_event: Registered event <E_ACC_CDR(4)>

Sep 27 21:30:52 centos64 /usr/local/server/pbx/sbin/opensips[11066]:
INFO:core:evi_publish_event: Registered event <E_ACC_MISSED_EVENT(5)>

Sep 27 21:30:52 centos64 /usr/local/server/pbx/sbin/opensips[11066]:
ERROR:core:pv_parse_spec: pvar "" not found

Sep 27 21:30:52 centos64 /usr/local/server/pbx/sbin/opensips[11066]:
ERROR:core:pv_parse_spec: wrong char [$/36] in [$ avp(received_nh)] at [0
(0)]

Sep 27 21:30:52 centos64 /usr/local/server/pbx/sbin/opensips[11066]:
ERROR:nathelper:mod_init: malformed or non AVP $ avp(received_nh) AVP
definition

Sep 27 21:30:52 centos64 /usr/local/server/pbx/sbin/opensips[11066]:
ERROR:core:init_mod: failed to initialize module nathelper

Sep 27 21:30:52 centos64 /usr/local/server/pbx/sbin/opensips[11066]:
ERROR:core:main: error while initializing modules

Sep 27 21:30:52 centos64 /usr/local/server/pbx/sbin/opensips[11066]:
INFO:core:cleanup: cleanup

Sep 27 21:30:52 centos64 opensips: INFO:core:daemonize: pre-daemon process
exiting with -1

 

And the service not getting started

 

From: users-bounces at lists.opensips.org
[mailto:users-bounces at lists.opensips.org] On Behalf Of Mike Tesliuk
Sent: Thursday, 26 September, 2013 10:25 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] FW: Audio and Video not working for different
otuside netrwork

 

When you use the residential script almost all configuration come alredy
working for this

i have a tutorial (in portuguese ( i think that i should translate to
english :)    )) , where you can see a routing script working with nat

http://opensips.com.br/wiki/index.php?title=Opensips_1.9



You can take a look at modules documentation too

http://www.opensips.org/html/docs/modules/1.9.x/nathelper.html
http://www.opensips.org/html/docs/modules/1.9.x/rtpproxy.html

There is on this maillist too a lot of discussions about this, below you can
see one case

http://opensips.org/pipermail/users/2011-January/016130.html

If you get some information from an old version of opensips probably will be
necessary to take a look on the module documentation to check about little
diferences , but i think that this is the start point :)

and if you is new to opensips i recommend to you the book about opensips (
http://www.packtpub.com/building-telephony-systems-with-opensips-1-6/book )



 

 

2013/9/26 Rajesh Babu <rajesh.babu at goodcoresoft.com>

Hi Mike,

 

  Thanks for the response, I am totally new to this world, can you please
help me by directing to on how to configure links. It will be great. 

Thanks in advance

Regards

Rajesh

 

From: users-bounces at lists.opensips.org
[mailto:users-bounces at lists.opensips.org] On Behalf Of Mike Tesliuk
Sent: Thursday, 26 September, 2013 12:25 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] FW: Audio and Video not working for different
otuside netrwork

 

you should configure the nathelper and rtpproxy, this should help in you
issue.

 

2013/9/26 Rajesh Babu <rajesh.babu at goodcoresoft.com>

Hi,

 

   I am new to the OpenSIP world. I have installed a OpenSIP on my network.
If i make a Call inside the network between two users i don't have any
issue, where as from outside the network, even though i can see the user
registered in my server i am not able to call registered user (I see the
user in my UL show listing). The call is established but i am not able to
talk (Mean the audio and video are not getting transffered).

 

Where as messages are going fine without any issue. I guess it is because
message transmit over XMPP where calls on SIP right.

 

 

I am really struck and i don't know how to proceed, please help me out

 

 

 

Thanks

Rajesh


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