[asterisk-users] Moving User Agent To Remote Location

[Nick Khamis]

To Answer Some of You Questions:

Please not that I replace the true domain wtih "example", and the true
ip for the remote UA with "public-ip". Nothing against no one here,
just don't know who else would read this email in the future!!!

PS: The public IP of the remote UA is correct.

SIP Show Peers:

Name/username                     Host            Dyn Forcerport ACL
Port Status     Realtime
1002/1002 at toronto.example. 192.168.2.13          N    5060
UNKNOWN    Cached RT
1003/1003 at toronto.example. -public-ip-       D     N    5060     OK
(86 ms) Cached RT


Peers look registered correctly. This has now become a sip proxy issue :S.

Thank you so much for your time guys!!!

N.


On 1/3/13, Nick Khamis <symack at gmail.com> wrote:
> Oooops yes of course 10004-10007!! Simple math does not come easy
> anymore... Anyhow, I singled out Opensips and I have two way audio
> form UA(local) -> UA(remote) but not from UA -> Siptrunk. That being
> said maybe a small diagram of the architecture. Please don't laugh!!!
> :) I know having a block of static IPs would make like easier
> however....
>
> UA (Remote) -> Router (Remote) -> Internet -> Router (Local) ->
> OpenSIPS+RTPProxy -> Asterisk
>
> Port forwarding (Remote): 5060, and 10000-50000 to UA
> Port Forwarding (Local): 5060. and 10000-50000 to OpenSIPS)   "No Audio"
> Port Forwarding (Local): 5060. and 10000-50000 directly to Asterisk
> "Two Way Audio"
>
> Cheers Guys!
>
> Nick
>
> 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 Jason
>> Parker
>> Sent: Thursday, January 03, 2013 2:26 PM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [asterisk-users] Moving User Agent To Remote Location
>>
>> On 01/03/2013 02:23 PM, Markus Weiler wrote:
>>> Am 03.01.2013 21:21, schrieb Nick Khamis:
>>>> Oh that's so smart!!! So, if I did not misunderstand you, for this
>>>> one call, have:
>>>> rtpstart=10004
>>>> rtpend=1008
>>
>> The rtpend should be 10008 and rtpstart should be 10005.  A SIP call in
>> Asterisk originates on 5060 (5061 for TLS) and then spawns two RTP
>> channels
>> for audio.  AFAIK the odd channel is send and the even channel is receive
>> (smarter folks than me like Tzafir can give you the specifics; this was
>> covered at least twice in 2012 threads).  If you open 5060 on your
>> NAT/firewall, but open no RTP channels, you will establish a call with no
>> sound.
>>
>>
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>