Hi all,
I am very new to all of this so forgive me if some of my questions don’t make sense 
I experience 1-way audio when I place calls. I can hear them but they can not hear me.
If I bypass asterisk and call the gsm gateway directly, there is absolutely no problem with the audio.
My system:
Ubuntu 12.04 LTS
Asterisk 11.16
Portech mv-378 gsm gateway
Use case:
I want to call in my asterisk server and have asterisk direct the call to the appropriate port depending on the number dialed. Instead of carrying 4 phones, I can leave the other sims at the office and take advantage of in-network dialing using Asterisk routes & trunks for the logic and the gsm gateway to make the calls using a sim from the appropriate network. I initiate the calls from a sip client using an extension that I created in the asterisk server and that registers fine.
Here is my configuration:
The asterisk server and the portech gateway are connected via a switch directly into the isp cable connection, not a router and each has a public IP so there should be no firewall and no NAT.
I created an extension (11116666) that registers into my asterisk from the cellular network.
Here are some files that might be relevant. Please let me know anything else I could show to help troubleshoot.
I have a feeling the problem might have to do with my trunk settings, but I do not know how to set them up properly.
For each trunk, I have the outgoing settings where the port number changes.
type=peer
port=5064
host=my.host.ip.address[b]Do I need to have anything in the trunk Incoming Settings section? (User context, user details) as well as in Registration? Right now all those fields are blank except for the auto-generated User Context…
Does I need to set up Realm in the gsm gateway?[/b]
Please let me know if there is any additional info I could provide to help… I am at a loss!
# cat /etc/asterisk/sip.conf
;--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. All modifications to ;
; this file must be done via the web gui. There are alternative files to make ;
; custom modifications, details at: http://freepbx.org/configuration_files ;
;--------------------------------------------------------------------------------;
[general]
; These files will all be included in the [general] context
;
#include sip_general_additional.conf
;sip_general_custom.conf is the proper file location for placing any sip general
;options that you might need set. For example: enable and force the sip jitterbuffer.
;If these settings are desired they should be set the sip_general_custom.conf file.
;
; jbenable=yes
; jbforce=yes
;
;It is also the proper place to add the lines needed for sip nat'ing when going
;through a firewall. For nat'ing you'd need to add the following lines:
; nat=yes , externip= , localhost= , and optionally fromdomain= .
;
#include sip_general_custom.conf
;sip_nat.conf is here for legacy support reasons and for those that upgrade
;from previous versions. If you have this file with lines in it please make
;sure they are not duplicated in sip_general_custom.conf, if so remove them
;from sip_nat.conf as sip_general_custom.conf will have precedence.
#include sip_nat.conf
;sip_registrations_custom.conf is for any customizations you might need to do to
;the automatically generated registrations that FreePBX makes.
;
#include sip_registrations_custom.conf
#include sip_registrations.conf
; These files should all be expected to come after the [general] context
;
#include sip_custom.conf
#include sip_additional.conf
;sip_custom_post.conf If you have extra parameters that are needed for a
;extension to work to for example, those go here. So you have extension
;1000 defined in your system you start by creating a line [1000](+) in this
;file. Then on the next line add the extra parameter that is needed.
;When the sip.conf is loaded it will append your additions to the end of
;that extension.
;
#include sip_custom_post.conf# cat /etc/asterisk/sip_general_additional.conf
;--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. All modifications to ;
; this file must be done via the web gui. There are alternative files to make ;
; custom modifications, details at: http://freepbx.org/configuration_files ;
;--------------------------------------------------------------------------------;
;
vmexten=*97
context=from-sip-external
callerid=Unknown
notifyringing=yes
notifyhold=yes
tos_sip=cs3
tos_audio=ef
tos_video=af41
alwaysauthreject=yes
useragent=FPBX-2.11.0(11.16.0)
disallow=all
allow=alaw
allow=ulaw
allow=gsm
callevents=no
jbenable=no
minexpiry=60
maxcallbitrate=384
maxexpiry=3600
notifyhold=yes
notifyringing=yes
registerattempts=0
registertimeout=20
rtpholdtimeout=300
rtpkeepalive=0
rtptimeout=30
srvlookup=no
allowguest=yes
checkmwi=10
defaultexpiry=120
videosupport=no
canreinvite=no
g726nonstandard=no
nat=no
externip=my.public.ip (removed for privacy)
localnet=my.public.network (removed for privacy)