This is merely a historical archive of years 2008-2021, before the migration to mailman3.
A maintained and still updated list archive can be found at https://lists.osmocom.org/hyperkitty/list/OpenBSC@lists.osmocom.org/.
Sandi Suhendro djks74 at gmail.comPlease 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/${EXTEN})
exten=>_XXXXX,n,Hangup
sip-custom-contexts.conf : change port to 5069
[GSM]
type=friend
host=127.0.0.1
dtmfmode=rfc2833
canreinvite=no
disallow=all
allow=gsm
context=gsmsubscriber
port=5069
osmo-sip-connector.cfg :
app
mncc
socket-path /tmp/bsc_mncc
sip
local 127.0.0.1 5069
remote 127.0.0.1 5060
This should work actually and you can read here
https://osmocom.org/projects/cellular-infrastructure/wiki/OpenBSC_with_Asterisk
ps: sometime you need to adjust port with sip-connector.
Let me know! Thanks.
On Wed, Jan 1, 2020 at 5:34 AM Garrett Allen <garrett.allen1990 at gmail.com>
wrote:
> Yes all works fine when not launching the setup with asterisk the MGW can
> route calls just fine. Below are the configs for sip-custom-contexts.conf
> and extensions.conf
>
> [GSM]
> type=friend
> host=127.0.0.1
> dtmfmode=rfc2833
> canreinvite=no
> disallow=all
> allow=gsm
> context=gsmsubscriber
> port=5062
>
> extensions.conf is quite large so i will omit all that was there by
> default and included is what i have added directly to the end of the file
>
> [gsmsubscriber]
> exten=>_xxxxx,1,Dial(SIP/GSM/${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 at 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 at 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 at gmail.com> wrote:
>>>
>>>> Have you try adding :
>>>>
>>>> nat=yes
>>>>
>>>> ???
>>>>
>>>> :)
>>>>
>>>> regards,
>>>> Sandi / DUO
>>>>
>>>> On Tue, Dec 31, 2019, 09:20 Garrett Allen <garrett.allen1990 at 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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.osmocom.org/pipermail/openbsc/attachments/20200101/fa27076d/attachment.htm>