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/${EXTEN})
exten=>_XXXXX,n,Hangup
sip-custom-contexts.conf : change port to 5069
[GSM]
type=friend
host=127.0.0.1
dtmfmode=rfc2833
canreinvite=no
disallow=all
allow=gsm
context=gsmsubscriber
port=5069
osmo-sip-connector.cfg :
app
mncc
socket-path /tmp/bsc_mncc
sip
local 127.0.0.1 5069
remote 127.0.0.1 5060
This should work actually and you can read here
https://osmocom.org/projects/cellular-infrastructure/wiki/OpenBSC_with_Aste…
ps: sometime you need to adjust port with sip-connector.
Let me know! Thanks.
On Wed, Jan 1, 2020 at 5:34 AM Garrett Allen <garrett.allen1990(a)gmail.com>
wrote:
Yes all works fine when not launching the setup with
asterisk the MGW can
route calls just fine. Below are the configs for sip-custom-contexts.conf
and extensions.conf
[GSM]
type=friend
host=127.0.0.1
dtmfmode=rfc2833
canreinvite=no
disallow=all
allow=gsm
context=gsmsubscriber
port=5062
extensions.conf is quite large so i will omit all that was there by
default and included is what i have added directly to the end of the file
[gsmsubscriber]
exten=>_xxxxx,1,Dial(SIP/GSM/${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(a)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(a)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(a)gmail.com> wrote:
>>
>>> Have you try adding :
>>>
>>> nat=yes
>>>
>>> ???
>>>
>>> :)
>>>
>>> regards,
>>> Sandi / DUO
>>>
>>> On Tue, Dec 31, 2019, 09:20 Garrett Allen
<garrett.allen1990(a)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