[Freeswitch-users] external sip profile

[Chad Vogel]

I was able to confirm that i have one way and two way audio now (I can see it in wireshark), it seems like the echo application doesn't work correctly, it will not send out audio. when i use <action application="delay_echo" data="1000"/> in the dialplan it does seems to send out rtc traffic however the audio is static noise. when i use <action application="echo" />  it doest seem to work. any thoughts on why cant I echo test and why delay_echo sends out static noise. I tried to use the record app it creates a wav file of 44 KB but doesn't grow beyond.


On Sep 12, 2011, at 7:18 PM, Michael Collins wrote:

Can you confirm if you have one-way audio? That is, an echo test won't tell you if you have one-way audio. A simple way to do this is to use the record app. It will play a file (whatever you choose, like "record at the tone..." - there are tons of sound files you can try) and then record the audio from the caller. Another test to do is to get a pcap of that call so you can analyze it in wireshark. If you have RTP going in both directions to/from the FS box then that indicates a NAT issue...

-MC

On Sun, Sep 11, 2011 at 7:00 PM, Chad Vogel <cvogel at lyonl.com<mailto:cvogel at lyonl.com>> wrote:
Level 3 uses port 5070 for their sip server but requires port 5060 to be used on our side; I made changes to the vars.xml config file to support this. I was just able to get the server to answer a call by changing the register value to false and adding an entry in the ACL config for the Level 3 server however now it seems that the audio isn't working correctly. I'm using a simple dial plan to echo the audio back but all I get is dead air.

    <extension name="level3">
      <condition field="destination_number" expression="^13127567000$">
        <action application="answer"/>
        <action application="echo" />
      </condition>
    </extension>



On Sep 11, 2011, at 7:47 PM, Nandy Dagondon wrote:

in addition to peter's advise, take a look at the SIP port 5070. FS is using port 5080 for the external SIP profile. modify the port number at "external.xml" then delete the port numbers in your proxy settings.

On Sun, Sep 11, 2011 at 6:36 PM, Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>> wrote:
You're not giving us much information here. Please post exactly what doesn't work, and also pastebin the actual logs from FreeSWITCH.

/Peter
________________________________________
Från: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] f&#246;r Chad Vogel [cvogel at lyonl.com<mailto:cvogel at lyonl.com>]
Skickat: den 11 september 2011 03:28
Till: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Ämne: [Freeswitch-users] external sip profile

hello,

I'm trying to switch from asterisk to freeswitch; however i'm wondering how can I create a sip profile because the sip profile i created doesn't seem to function with level 3.

here is the sip profile i created that isn't working:

<include>
 <gateway name="Level3">
   <param name="username" value="USERNAME"/>
   <param name="password" value="PASSWORD"/>
   <param name="proxy" value="4.55.35.60:5070<http://4.55.35.60:5070/>"/>
   <param name="expire-seconds" value="3600"/>
   <param name="register-transport" value="udp"/>
   <param name="register" value="true"/>
 </gateway>
</include>

here is my asterisk profile (it works):

[level3_out]
type=peer
nat=no
host=4.55.35.60
username=***Username***
secret=***Password***
dtmfmode=rfc2833
port=5070

[level3_in]
nat=no
insecure=very
dtmfmode=rfc2833
disallow=all
context=from-trunk
canreinvite=no
allow=ulaw&alaw
host=4.55.35.60
type=peer
port=5070

How can I create a sip profile that will function the same in freeswitch?

!DSPAM:4e6c82af32761635315745!


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