[Freeswitch-users] external sip profile

[Michael Collins]

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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>>
>
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