[asterisk-users] Moving User Agent To Remote Location

[Nick Khamis]

Oh that's so smart!!! So, if I did not misunderstand you, for this one
call, have:
rtpstart=10004
rtpend=1008

Just for testing purposes, and deduce my way from there? Right now I
am trying to call the phone from my softphone. That being said, I
currently I am not able to reach the remote extension from my location
here. I think this is the root of the problem here:

    -- Executing [1003 at context-from-toronto:1]
Dial("SIP/OpenSIPS-00000009", "SIP/1003, 20") in new stack
Really destroying SIP dialog
'06775f8653ff88b47cfa9ec123abdd89 at 127.0.0.1:0' Method: INVITE
[Dec 12 15:35:54] WARNING[1736]: app_dial.c:2198 dial_exec_full:
Unable to create channel of type 'SIP' (cause 20 - Unknown)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [1003 at context-from-toronto:2]
Wait("SIP/OpenSIPS-00000009", "1") in new stack
    -- Executing [1003 at context-from-toronto:3]
Answer("SIP/OpenSIPS-00000009", "") in new stack
Audio is at 5060
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP


It's actually not able to create the SIP channel between the two UA? I
will try taking opensips out of the picture and work outwards...

N.

On 1/3/13, Danny Nicholas <danny at debsinc.com> wrote:
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Nick Khamis
> Sent: Thursday, January 03, 2013 11:47 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Moving User Agent To Remote Location
>
> Hello Everyone,
>
> Before getting into SIP and RTP traces, I wanted to clarify some of the
> sip.conf settings that may to some seem redundant or have a misconception
> with. I do apologize if this has been discussed time and time again as I
> would imagine. If anything, this email would make google search results
> that
> much stronger :).
>
> With the UA local to my network I had tested two way audio, and now with
> the
> phone outside of network, we have no way audio. Before discussing NAT
> (which
> is enabled on the peer), and port forwarding (which is setup on the remote
> location), I would like to make sure I fully understand all the sip.conf
> settings. We are using Asterisk realtime via sip_buddies, and the fields in
> question are:
>
> (Enclosed in brackets are an example value for the setting)
>
> * host (dynamic): No problem here. Just wanted to mention that it's set as
> such....
> * nat (yes): No problem here either....
> * defaultuser (1003 at example.com): Does the "@example.com" have to point to
> the UA (i.e., (1003 at ua-public-ip), or is it just a name type field?
> * fullcontact: What to put here for a UA that is running at a remote
> location with dynamic external IP?
> * ipaddr (ua-public-ip): I did try setting it to the public ip of the UA,
> but is that really practical?
> What if I don't know where the initial registration request is coming from?
> I am guessing "host=dynamic" takes care of that.
> * defaultip??
> * dynamic: Should this be set to yes, or is "host=dynamic" sufficient?
>
> The phone registers fine, and terminates a call through our providers.
> Just no audio both ways, which would suggest something more that an RTP
> issue which should at least have one way outgoing audio.
>
> Things that have been attempted:
> * Port forwarding to the phone
> * Changing defaultuser to 1003 at ua-public-ip. This made our OpenSIPS sip
> proxy through a fit.
>
> Things I will attempt today:
> Calling the UA extension from an extension here SIP trace
>
> Your help is greatly appreciated!!!
>
> Nick.
>
> I'm going to vote for the RTP issue.  If you are establishing a call but
> have no audio, you are getting the 5060 port, but not the 10000-20000 range
> that RTP normally expects. A "better" practice is to allocate 4 ports per
> line you expect to use in rtp.conf (10000-20000 would allow 2500 lines;
> much
> more that most folks need and more "holes" to monitor).
>
>
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