Please try this and let me know.

extensions.conf: (make sure you dial call with 5 digits since you make 5X.)

[gsmsubscriber]
exten=>_XXXXX,1,Answer()
exten=>_XXXXX,2,Dial(SIP/GSM/${EXTEN})
exten=>_XXXXX,n,Hangup


sip-custom-contexts.conf : change port to 5069

[GSM]
type=friend
host=127.0.0.1
dtmfmode=rfc2833
canreinvite=no
disallow=all
allow=gsm
context=gsmsubscriber
port=5069


osmo-sip-connector.cfg :

 app
 mncc
  socket-path  /tmp/bsc_mncc
 sip
  local 127.0.0.1 5069
  remote 127.0.0.1 5060

This should work actually and you can read here https://osmocom.org/projects/cellular-infrastructure/wiki/OpenBSC_with_Asterisk

ps: sometime you need to adjust port with sip-connector.

Let me know! Thanks.


On Wed, Jan 1, 2020 at 5:34 AM Garrett Allen <garrett.allen1990@gmail.com> wrote:
Yes all works fine when not launching the setup with asterisk the MGW can route calls just fine. Below are the configs for sip-custom-contexts.conf and extensions.conf

[GSM]
type=friend
host=127.0.0.1
dtmfmode=rfc2833
canreinvite=no
disallow=all
allow=gsm
context=gsmsubscriber
port=5062

extensions.conf is quite large so i will omit all that was there by default and included is what i have added directly to the end of the file

[gsmsubscriber]
exten=>_xxxxx,1,Dial(SIP/GSM/${EXTEN})
exten=>_XXXXX,n,Playback(vm-nobodyavail)
exten=>_xxxxx,n,HangUp

sip.conf is again default with just this library include at the end of the file

#include sip-custom-contexts.conf

Thanks for the assistance 





nat=yes

On Tue, 31 Dec 2019 at 16:16, Sandi Suhendro <djks74@gmail.com> wrote:
Can you describe more your configuration and setup with asterisk? your sip connector settings, etc.. ?
Sip.conf, sip-custom-contexts, extensions.conf  ?

You said it it works when using osmo-mgw for voice?

On Tue, Dec 31, 2019 at 9:24 PM Garrett Allen <garrett.allen1990@gmail.com> wrote:
Unfortunately it made no difference adding nat=yes to the asterisk config.

On Tue, 31 Dec 2019, 11:13 Sandi Suhendro, <djks74@gmail.com> wrote:
Have you try adding :

nat=yes

???

:)

regards,
Sandi / DUO

On Tue, Dec 31, 2019, 09:20 Garrett Allen <garrett.allen1990@gmail.com> wrote:
Hi All

I've used osmo-bsc in non standalone mode ie seperate components and all works fine voice and sms work as should. However when introducing. sip connector with asterisk the handsets will not make voice calls they just constantly respond network busy. I'm using osmo bts-trx with osmo-trx-lms on raspbian 10 below is Asterisk config. Any help would be appreciated.

Gar

[GSM]
type=friend
host=127.0.0.1
dtmfmode=rfc2833
canreinvite=no
disallow=all
allow=gsm
context=gsmsubscriber
port=5062



--
Best Regards,
Sandi



--
Best Regards,
Sandi