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
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
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
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
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/${EXTEN}) exten=>_XXXXX,n,Playback(vm-nobodyavail) exten=>_xxxxx,n,HangUp
sip.conf is again default with just this library include at the end of the file
#include sip-custom-contexts.conf
Thanks for the assistance
nat=yes
On Tue, 31 Dec 2019 at 16:16, Sandi Suhendro djks74@gmail.com wrote:
Can you describe more your configuration and setup with asterisk? your sip connector settings, etc.. ? Sip.conf, sip-custom-contexts, extensions.conf ?
You said it it works when using osmo-mgw for voice?
On Tue, Dec 31, 2019 at 9:24 PM Garrett Allen garrett.allen1990@gmail.com wrote:
Unfortunately it made no difference adding nat=yes to the asterisk config.
On Tue, 31 Dec 2019, 11:13 Sandi Suhendro, djks74@gmail.com wrote:
Have you try adding :
nat=yes
???
:)
regards, Sandi / DUO
On Tue, Dec 31, 2019, 09:20 Garrett Allen garrett.allen1990@gmail.com wrote:
Hi All
I've used osmo-bsc in non standalone mode ie seperate components and all works fine voice and sms work as should. However when introducing. sip connector with asterisk the handsets will not make voice calls they just constantly respond network busy. I'm using osmo bts-trx with osmo-trx-lms on raspbian 10 below is Asterisk config. Any help would be appreciated.
Gar
[GSM] type=friend host=127.0.0.1 dtmfmode=rfc2833 canreinvite=no disallow=all allow=gsm context=gsmsubscriber port=5062
-- Best Regards, Sandi
Please try this and let me know.
extensions.conf: (make sure you dial call with 5 digits since you make 5X.)
[gsmsubscriber] exten=>_XXXXX,1,Answer() exten=>_XXXXX,2,Dial(SIP/GSM/${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_Aster...
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
On 31/12/2019 23:34, Garrett Allen 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
Garrett, what would probably most help people to help you would be a pcap of traffic on the interface between your asterisk and osmo components. probably the loopback if you are running everything on the same box.
Maybe just bring up the system, start a capture on all interfaces, make the call attempt and stop capture. should be enough. if it's more than a few kB, you can be nice to the mailing list (and the c02 levels!) by posting the pcap someplace and sending a link, rather than sending an attachment to everyone.
best,
k/
Thank you both for your help, I managed to find my mistake while looking through the sip-connector config. All works great now :) have a happy new year :)
Garrett
On Wed, 1 Jan 2020, 15:13 Keith, keith@rhizomatica.org wrote:
On 31/12/2019 23:34, Garrett Allen 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
Garrett, what would probably most help people to help you would be a pcap of traffic on the interface between your asterisk and osmo components. probably the loopback if you are running everything on the same box.
Maybe just bring up the system, start a capture on all interfaces, make the call attempt and stop capture. should be enough. if it's more than a few kB, you can be nice to the mailing list (and the c02 levels!) by posting the pcap someplace and sending a link, rather than sending an attachment to everyone.
best,
k/
-
Garrett Allen -
Keith -
Sandi Suhendro