[Freeswitch-users] external sip profile

[Michael Collins]

Open the pcap file in wireshark and see if the destination port in the SDP
matches the actual port number to which the incoming RTP packets are being
sent. My guess is the router/NAT device is messing with your RTP traffic.

-MC

On Mon, Sep 12, 2011 at 6:31 PM, Chad Vogel <cvogel at lyonl.com> wrote:

>  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> 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> 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 [
>>> freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Chad Vogel [
>>> cvogel at lyonl.com]
>>> Skickat: den 11 september 2011 03:28
>>> Till: 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"/>
>>>    <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!
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
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>>> http://www.freeswitch.org
>>>
>>
>>
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