[asterisk-users] Asterisk and alternate RTP ports

[Todd R .]

Been working with Asterisk for a long time but this is the first time I have 
dealt with this issue.
I am setting up an Asterisk box (FreePBX not my choice) to interface with an 
e911 provider.
They say their switches only listen for RTP on ports 2-21001 which is 
outside the normal range Asterisk listens on 1-2.
I wish I knew more about this topic but since I have never had an issue 
interfacing with providers, ITSP etc., I just haven't had a need to know.
I get audio on some calls and others not so much.
How do I deal with this?
I don't really want to change the RTP ports that Asterisk listens on because 
this is a production system with trunks pointing to several other providers etc.
The 911 provider says I don't need to listen on 2-21001, that's just what 
they listen on.
In fact, they say this exactly You can listen on what you want, as long as the 
RTP port is sent to us in the INVITE SDP info..
Any assistance with solving this issue would be greatly appreciated, I have 
done my digging in Google etc before asking here as always.
Thanks in advance.







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Re: [asterisk-users] Asterisk and alternate RTP ports

[Chad Mothersell]

Hi Todd,
From my experience one-way or no-way audio is typically caused by the Asterisk 
system being behind a NAT or improperly configured firewall (usually NAT). 

The provider you are dealing with is correct when they say it doesn’t matter if 
they listen on different RTP ports. That all gets negotiated at the beginning 
of the call. 

Chad

On Jul 2, 2014, at 1:49 PM, Todd R. tjrl...@live.com wrote:

 Been working with Asterisk for a long time but this is the first time I have 
 dealt with this issue.
 
 I am setting up an Asterisk box (FreePBX not my choice) to interface with an 
 e911 provider.
 
 They say their switches only listen for RTP on ports 2-21001 which is 
 outside the normal range Asterisk listens on 1-2.
 
 I wish I knew more about this topic but since I have never had an issue 
 interfacing with providers, ITSP etc., I just haven't had a need to know.
 
 I get audio on some calls and others not so much.
 
 How do I deal with this?
 
 I don't really want to change the RTP ports that Asterisk listens on because 
 this is a production system with trunks pointing to several other providers 
 etc.
 
 The 911 provider says I don't need to listen on 2-21001, that's just what 
 they listen on.
 
 In fact, they say this exactly You can listen on what you want, as long as 
 the RTP port is sent to us in the INVITE SDP info..
 
 Any assistance with solving this issue would be greatly appreciated, I have 
 done my digging in Google etc before asking here as always.
 
 Thanks in advance.
 -- 
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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   http://www.asterisk.org/hello
 
 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users