VoIP Mailing List Archives

[pankajanand18 at gmail...]

hi folks,   I m not able to make SIP calls using external profile. 


 i have added the following lines to the $installdir/conf/dialplan/public.xml 


<extension name="echo">
      <condition field="destination_number" expression="^9996$">
        <action application="answer"/>
        <action application="echo"/>
      </condition>
    </extension>



<extension name="public_extensions">
      <condition field="destination_number" expression="^(10[01][0-9])$">
        <action application="bridge" data="sofia/external/$1@$${domain}"/>
      </condition>
    </extension>


I m able to connect using 1000 and 1001 from public Internet.  I am able to make an echo call. 


when i type : 


$: sofia status  profile external reg


It shows the list of the connected clients and their information.


but when I m trying to make a call from one user to other user, it generates the following error 




2009-09-18 15:41:45.675054 [NOTICE] switch_channel.c:602 New Channel sofia/external/1001@192.168.1.50 (1001@192.168.1.50) [fcb6c23e-bdcd-41dd-b73e-df07b71252be]
2009-09-18 15:41:45.677063 [INFO] mod_dialplan_xml.c:315 Processing 1000->1000 in context public
2009-09-18 15:41:45.679071 [NOTICE] switch_channel.c:602 New Channel sofia/external/1000@192.168.1.50 (1000@192.168.1.50) [1a537865-be53-42ce-b8f5-cc183f4f1306]
2009-09-18 15:41:45.688161 [ERR] sofia_reg.c:1568 No Matching gateway found
2009-09-18 15:41:45.688161 [NOTICE] sofia_reg.c:1588 Hangup sofia/external/1000@192.168.1.50 (1000@192.168.1.50) [CS_CONSUME_MEDIA] [MANDATORY_IE_MISSING]
2009-09-18 15:41:45.688161 [INFO] mod_dptools.c:2093 Originate Failed.  Cause: MANDATORY_IE_MISSING
2009-09-18 15:41:45.689090 [NOTICE] mod_dptools.c:2125 Hangup sofia/external/1001@192.168.1.50 (1001@192.168.1.50) [CS_EXECUTE] [MANDATORY_IE_MISSING]
2009-09-18 15:41:45.690064 [NOTICE] switch_core_session.c:1086 Session 1 (sofia/external/1001@192.168.1.50 (1001@192.168.1.50)) Ended
2009-09-18 15:41:45.690064 [NOTICE] switch_core_session.c:1088 Close Channel sofia/external/1001@192.168.1.50 (1001@192.168.1.50) [CS_DESTROY]
2009-09-18 15:41:45.692078 [NOTICE] switch_core_session.c:1086 Session 2 (sofia/external/1000@192.168.1.50 (1000@192.168.1.50)) Ended
2009-09-18 15:41:45.692078 [NOTICE] switch_core_session.c:1088 Close Channel sofia/external/1000@192.168.1.50 (1000@192.168.1.50) [CS_DESTROY]




with regards
Pankaj anand

[brian at freeswitch.org]

The far end is sending you a challenge and we can't answer it because
we haven't been told what gateway to use.

/b

On Sep 18, 2009, at 5:13 AM, pankaj anand wrote:

[aep.lists at it46.se]

Have you tried with
<action application="bridge"
data="user/${dialed_extension}@${domain_name}"/>

instead?

/aep
--
Stopping junk mailers is good for the environment

[brian at freeswitch.org]

OK pay attention this time.

See this line:

2009-09-18 15:41:45.688161 [ERR] sofia_reg.c:1568 No Matching gateway
found

You sent a call out the profile the far side sent you a challenge
since you're not calling via a gateway we can't answer the challenge
because we do not know HOW.

What is the far end you're calling?

/b


On Sep 18, 2009, at 9:11 AM, pankaj anand wrote:

[pankajanand18 at gmail...]

I m using default configuration of freeswitch.. I m not using any gateway for authentication.
in the $INSTALLDIR/conf/sip_profiles/external/ directory,  there exist only one file which example.xml , this files contains 


<include>
  <!--<gateway name="asterlink.com">-->
  <!--/// account username *required* ///-->
  <!--<param name="username" value="cluecon"/>-->
  <!--/// auth realm: *optional* same as gateway name, if blank ///-->
  <!--<param name="realm" value="asterlink.com"/>-->
  <!--/// username to use in from: *optional* same as  username, if blank ///-->
  <!--<param name="from-user" value="cluecon"/>-->
  <!--/// domain to use in from: *optional* same as  realm, if blank ///-->
  <!--<param name="from-domain" value="asterlink.com"/>-->
  <!--/// account password *required* ///-->
  <!--<param name="password" value="2007"/>-->
  <!--/// extension for inbound calls: *optional* same as username, if blank ///-->
  <!--<param name="extension" value="cluecon"/>-->
  <!--/// proxy host: *optional* same as realm, if blank ///-->
  <!--<param name="proxy" value="asterlink.com"/>-->
  <!--/// send register to this proxy: *optional* same as proxy, if blank ///-->
  <!--<param name="register-proxy" value="mysbc.com"/>-->
  <!--/// expire in seconds: *optional* 3600, if blank ///-->
  <!--<param name="expire-seconds" value="60"/>-->
  <!--/// do not register ///-->
  <!--<param name="register" value="false"/>-->
  <!-- which transport to use for register -->
  <!--<param name="register-transport" value="udp"/>-->
  <!--How many seconds before a retry when a failure or timeout occurs -->
  <!--<param name="retry-seconds" value="30"/>-->
  <!--Use the callerid of an inbound call in the from field on outbound calls via this gateway -->
  <!--<param name="caller-id-in-from" value="false"/>-->
  <!--extra sip params to send in the contact-->
  <!--<param name="contact-params" value="tport=tcp"/>-->
  <!--send an options ping every x seconds, failure will unregister and/or mark it down-->
  <!--<param name="ping" value="25"/>-->
  <!--</gateway>-->
</include>




as you can see, all the lines are commented. So i m not using any gateways.






On Fri, Sep 18, 2009 at 3:43 PM, pankaj anand <pankajanand18@gmail.com (pankajanand18@gmail.com)> wrote:

[tculjaga at gmail.com]

in other works,

what are you trying to achieve?
where do you want send calls?
what is 192.168.1.50?
where/how are you originating calls from?

basically can you please tell us what is your call flow scenario otherwise we can't help?

T.


On Fri, Sep 18, 2009 at 4:15 PM, Brian West <brian@freeswitch.org (brian@freeswitch.org)> wrote:

[pankajanand18 at gmail...]

@Tihomir Culjaga

HI folks,
 thanx for such a quick reply.
 
Q. what I want to achieve with FreeSwitch ?
A: I want to enable the outside users ( from internet) to have video chat on peer2peer using freeSwitch for signaling. External Profile is being used to for this. External profile is using 5080 port. That port is forwarded on the NAT server. Users are able to connect using 5080 port. They get  registered with no issues. 
 
Q. where do you want to send calls ?
A. I want to send call from one extension to another extension ( both extension exist on the are on public internet). Right now i m trying with 1000 and 1001 user available in the default directory.


1. What is 192.168.1.50 ?
   Ans: well , this is my domain name which is by default the local-ip address of the machine. My current setup is like this:
   FreeSwitch ( 192.168.1.50) ---->NAT(122.162.153.224)-->Internet<----(122.80.0.180)NAT<--(192.168.1.15)1001(user)
    
2. 
   Where/how are you originating calls from ?
    
   
   1. I am using X-lite, Phoner , LinPhone to make calls. All these  phones have stun server enabled .
   
    
   For the public dial plan I have added these lines in the file public.xml which is used by the external profile 
    
    <extension name="public_extensions">
         <condition field="destination_number" expression="^(10[01][0-9])$">
           <action application="bridge" data="sofia/external/$1@$${domain}"/>
           <action application="echo"/>
         </condition>
       </extension>
    
   <extension name="echo">
         <condition field="destination_number" expression="^9996$">
           <action application="answer"/>
           <action application="echo"/>
         </condition>
       </extension>
    
   Now the echo calls works through the external profile. But when a call is being made to some other user, for example if user 1000 makes a call to the 1001 it reaches to the "public_extensions "  but it generates the error which I have already mentioned. For the gateway thing , not gateway is being used.
    
    

On Fri, Sep 18, 2009 at 7:41 PM, pankaj anand <pankajanand18@gmail.com (pankajanand18@gmail.com)> wrote:

[tculjaga at gmail.com]

check this: http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Dialing_A_Registered_User

dial registered user:         <action application="bridge" data="sofia/external/$1%$${domain}"/>
dial external endpoint:     <action application="bridge" data="sofia/external/$1@$${domain}"/>


another issue you might have with RTP so check the wiki for NAT config as well.

T.

On Sat, Sep 19, 2009 at 7:50 AM, pankaj anand <pankajanand18@gmail.com (pankajanand18@gmail.com)> wrote:

[brian at freeswitch.org]

If you refer to the latest internal.xml in the default config for sip
profiles you'll see an example of how to use a single profile for
phones inside and outside of NAT. So you no longer have to have two
profiles thus cutting the confusion level to almost zero when you
setup FreeSWITCH to talk inside and outside of nat.

Key elements are local-network-acl, ext-sip-ip and ext-rtp-ip and
you're all set.

/b

On Sep 19, 2009, at 2:04 AM, Tihomir Culjaga wrote: