Do I have to have mISDN for LCR even im not going to use any isdn interface ? Im trying to connect the NITB to LCR and LCR to asterisk all on ip.
Thanks for any help.
Rgds Nik
Hi Nik,
On Fri, Mar 02, 2012 at 05:08:45PM +0000, Nik Pakar wrote:
Do I have to have mISDN for LCR even im not going to use any isdn interface ? Im trying to connect the NITB to LCR and LCR to asterisk all on ip.
see http://www.isdn4linux.de/pipermail/isdn4linux/2012-March/005688.html
Kind regards, -Alexander Huemer
I think im little bit lost here now the way it should work. As per this link it only works with isdn interface not sip.
All im trying to do is,
[MS]---[nanoBTS]----[OpenBSC NITB]---(ip)---[LCR]---(ip)---[Asterisk]----(sip)--->
Is this what its been made for or are there any E1/ISDN interfaces in the middle.
On Fri, Mar 2, 2012 at 8:29 PM, Alexander Huemer alexander.huemer@xx.vuwrote:
Hi Nik,
On Fri, Mar 02, 2012 at 05:08:45PM +0000, Nik Pakar wrote:
Do I have to have mISDN for LCR even im not going to use any isdn
interface
? Im trying to connect the NITB to LCR and LCR to asterisk all on ip.
see http://www.isdn4linux.de/pipermail/isdn4linux/2012-March/005688.html
Kind regards, -Alexander Huemer
Hi Carecel,
I also still strugling with no luck. I have a very good understanding of the openbsc and it works well as the NITB.
But LCR and asterisk integration is no where closer.
Im using debian 6.0.4 Asterisk 1.8.9.3 OpenBSC, Osmocore, Osmo-abis : latest git clone LCR : latest git clone mISDN : downloaded from the site
First thing didnt aligned with documentation is the LCR patch doesnt seems apply. I assume the developers has made the patch into it, so its not applying. But when i inspect the source of gsm_bs.cpp, doesnt seems there either.
Passing on - once LCR compiled with asterisk and gsm_bs, it doesnt even create a gsm.conf
So almost no go from that point.
Anyway, now im waiting for some feedback about the actual topology of it since its unclear whether we still need E1/ISDN in the middle to integrate this. If that is the case i wouldnt want to try that. Been all IP solution, there is no point having TDM in the middle at all.
Do you have any idea about that ?
Nik.
On Sat, Mar 3, 2012 at 12:48 PM, Nik Pakar nikpakar@gmail.com wrote:
I think im little bit lost here now the way it should work. As per this link it only works with isdn interface not sip.
All im trying to do is,
[MS]---[nanoBTS]----[OpenBSC NITB]---(ip)---[LCR]---(ip)---[Asterisk]----(sip)--->
Is this what its been made for or are there any E1/ISDN interfaces in the middle.
On Fri, Mar 2, 2012 at 8:29 PM, Alexander Huemer alexander.huemer@xx.vuwrote:
Hi Nik,
On Fri, Mar 02, 2012 at 05:08:45PM +0000, Nik Pakar wrote:
Do I have to have mISDN for LCR even im not going to use any isdn
interface
? Im trying to connect the NITB to LCR and LCR to asterisk all on ip.
see http://www.isdn4linux.de/pipermail/isdn4linux/2012-March/005688.html
Kind regards, -Alexander Huemer
Another attempt with all sources downloaded from this link has failed when building LCR :(
http://www.linux-call-router.de/download/lcr-1.7/
---
g++ -g -O2 -o gentones gentones.o -lpthread -lncurses -lm g++ -DHAVE_CONFIG_H -I. -DWITH_GSM_BS -I./openbsc/include -I./libosmocore/include -I./openbsc -Wall -DCONFIG_DATA=""/usr/local/lcr"" -DSHARE_DATA=""/usr/local/lcr"" -DLOG_DIR=""/usr/local/lcr"" -DEXTENSION_DATA=""/usr/local/lcr/extensions"" -g -O2 -MT genwave.o -MD -MP -MF .deps/genwave.Tpo -c -o genwave.o genwave.c mv -f .deps/genwave.Tpo .deps/genwave.Po g++ -g -O2 -o genwave genwave.o -lpthread -lncurses -lm gcc -DWITH_GSM_BS -I./openbsc/include -I./libosmocore/include -I./openbsc -Wall -DCONFIG_DATA=""/usr/local/lcr"" -DSHARE_DATA=""/usr/local/lcr"" -DLOG_DIR=""/usr/local/lcr"" -DEXTENSION_DATA=""/usr/local/lcr/extensions"" -D_GNU_SOURCE -fPIC -c chan_lcr.c -o chan_lcr.po chan_lcr.c: In function ‘send_setup_to_lcr’: chan_lcr.c:644: error: ‘struct ast_channel’ has no member named ‘cid’ chan_lcr.c:655: error: ‘struct ast_channel’ has no member named ‘cid’ chan_lcr.c: In function ‘lcr_in_setup’: chan_lcr.c:858: warning: passing argument 9 of ‘__ast_channel_alloc’ makes integer from pointer without a cast /usr/include/asterisk/channel.h:1112: note: expected ‘int’ but argument is of type ‘char *’ chan_lcr.c:883: error: ‘struct ast_channel’ has no member named ‘cid’ chan_lcr.c:885: error: ‘struct ast_channel’ has no member named ‘cid’ chan_lcr.c:887: error: ‘struct ast_channel’ has no member named ‘cid’ chan_lcr.c:890: error: ‘struct ast_channel’ has no member named ‘cid’ chan_lcr.c:893: error: ‘struct ast_channel’ has no member named ‘cid’ chan_lcr.c:896: error: ‘struct ast_channel’ has no member named ‘cid’ chan_lcr.c:900: error: ‘struct ast_channel’ has no member named ‘cid’ chan_lcr.c:903: error: ‘struct ast_channel’ has no member named ‘cid’ chan_lcr.c:906: error: ‘struct ast_channel’ has no member named ‘cid’ chan_lcr.c:909: error: ‘struct ast_channel’ has no member named ‘cid’ chan_lcr.c: In function ‘handle_queue’: chan_lcr.c:1707: error: incompatible types when assigning to type ‘union ast_frame_subclass’ from type ‘char’ chan_lcr.c: In function ‘lcr_request’: chan_lcr.c:1820: warning: passing argument 9 of ‘__ast_channel_alloc’ makes integer from pointer without a cast /usr/include/asterisk/channel.h:1112: note: expected ‘int’ but argument is of type ‘char *’ chan_lcr.c: In function ‘lcr_call’: chan_lcr.c:1927: error: ‘struct ast_channel’ has no member named ‘cid’ chan_lcr.c:1927: error: ‘struct ast_channel’ has no member named ‘cid’ chan_lcr.c:1928: error: ‘struct ast_channel’ has no member named ‘cid’ chan_lcr.c:1931: error: ‘struct ast_channel’ has no member named ‘cid’ chan_lcr.c:1931: error: ‘struct ast_channel’ has no member named ‘cid’ chan_lcr.c:1932: error: ‘struct ast_channel’ has no member named ‘cid’ chan_lcr.c:1934: error: ‘struct ast_channel’ has no member named ‘cid’ chan_lcr.c:1934: error: ‘struct ast_channel’ has no member named ‘cid’ chan_lcr.c:1935: error: ‘struct ast_channel’ has no member named ‘cid’ chan_lcr.c: In function ‘lcr_write’: chan_lcr.c:2164: error: wrong type argument to unary exclamation mark chan_lcr.c:2166: error: invalid operands to binary & (have ‘union ast_frame_subclass’ and ‘format_t’) chan_lcr.c: In function ‘lcr_read’: chan_lcr.c:2229: error: incompatible types when assigning to type ‘union ast_frame_subclass’ from type ‘format_t’ chan_lcr.c: In function ‘lcr_indicate’: chan_lcr.c:2274: warning: assignment from incompatible pointer type chan_lcr.c:2289: warning: assignment from incompatible pointer type chan_lcr.c:2316: warning: assignment from incompatible pointer type chan_lcr.c:2381: error: dereferencing pointer to incomplete type chan_lcr.c:2382: error: dereferencing pointer to incomplete type chan_lcr.c: At top level: chan_lcr.c:2602: warning: initialization from incompatible pointer type chan_lcr.c: In function ‘load_module’: chan_lcr.c:2818: warning: passing argument 2 of ‘ast_register_application2’ from incompatible pointer type /usr/include/asterisk/module.h:458: note: expected ‘int (*)(struct ast_channel *, const char *)’ but argument is of type ‘int (*)(struct ast_channel *, void *)’ make[1]: *** [chan_lcr.po] Error 1 make[1]: Leaving directory `/usr/local/src/lcr' make: *** [all] Error 2
On Sat, Mar 3, 2012 at 7:25 PM, Nik Pakar nikpakar@gmail.com wrote:
Hi Carecel,
I also still strugling with no luck. I have a very good understanding of the openbsc and it works well as the NITB.
But LCR and asterisk integration is no where closer.
Im using debian 6.0.4 Asterisk 1.8.9.3 OpenBSC, Osmocore, Osmo-abis : latest git clone LCR : latest git clone mISDN : downloaded from the site
First thing didnt aligned with documentation is the LCR patch doesnt seems apply. I assume the developers has made the patch into it, so its not applying. But when i inspect the source of gsm_bs.cpp, doesnt seems there either.
Passing on - once LCR compiled with asterisk and gsm_bs, it doesnt even create a gsm.conf
So almost no go from that point.
Anyway, now im waiting for some feedback about the actual topology of it since its unclear whether we still need E1/ISDN in the middle to integrate this. If that is the case i wouldnt want to try that. Been all IP solution, there is no point having TDM in the middle at all.
Do you have any idea about that ?
Nik.
On Sat, Mar 3, 2012 at 12:48 PM, Nik Pakar nikpakar@gmail.com wrote:
I think im little bit lost here now the way it should work. As per this link it only works with isdn interface not sip.
All im trying to do is,
[MS]---[nanoBTS]----[OpenBSC NITB]---(ip)---[LCR]---(ip)---[Asterisk]----(sip)--->
Is this what its been made for or are there any E1/ISDN interfaces in the middle.
On Fri, Mar 2, 2012 at 8:29 PM, Alexander Huemer alexander.huemer@xx.vuwrote:
Hi Nik,
On Fri, Mar 02, 2012 at 05:08:45PM +0000, Nik Pakar wrote:
Do I have to have mISDN for LCR even im not going to use any isdn
interface
? Im trying to connect the NITB to LCR and LCR to asterisk all on ip.
see http://www.isdn4linux.de/pipermail/isdn4linux/2012-March/005688.html
Kind regards, -Alexander Huemer
Do I have to have mISDN for LCR even im not going to use any isdn interface ? Im trying to connect the NITB to LCR and LCR to asterisk all on ip.
Thanks for any help.
Rgds Nik
hi nik,
you don't need misdn to use lcr with sip and gsm anymore. also you don't need any patch. the lcr and openbsc compile out of the box supporting each other. the howto is a bit outdated.
try to compile the lcr from the git. don't use chan_lcr, since it still works with isdn only. you need to setup a sip interface in interface.conf:
[sip] sip <local ip> <remote ip>[:<port>] sip 10.0.0.12 10.0.0.34 earlyb no tones no
use asterisk machine for remote ip. if you have asterisk on the same machine, change the sip port of asterisk and use "remoteip:port".
regards,
andreas
Hi Jolly,
On Mon, Mar 5, 2012 at 11:44, jolly andreas@eversberg.eu wrote:
you don't need misdn to use lcr with sip and gsm anymore. also you don't need any patch. the lcr and openbsc compile out of the box supporting each other. the howto is a bit outdated.
try to compile the lcr from the git. don't use chan_lcr, since it still works with isdn only. you need to setup a sip interface in interface.conf:
[sip] sip <local ip> <remote ip>[:<port>] sip 10.0.0.12 10.0.0.34 earlyb no tones no
use asterisk machine for remote ip. if you have asterisk on the same machine, change the sip port of asterisk and use "remoteip:port".
Does it mean that that now we can use LCR with other SIP softswitches/PBX'es, like Freeswitch? I do not follow LCR development closely, but that would be a very interesting development.
Alexander Chemeris wrote:
Does it mean that that now we can use LCR with other SIP softswitches/PBX'es, like Freeswitch? I do not follow LCR development closely, but that would be a very interesting development.
yes, this was my intention. gsm and sip interface of lcr will not rely on misdn anymore. currently i don't support audio transfer via chan_lcr, so chan_lcr will only work with isdn interfaces. the sip interface implementation has not much options, so it can only do sipmple point-to-point sip connections to a gateway or endpoint.
Hi Andreas,
It was apperently compiled on my debian while i had misdn libs isntalled. Now im trying on a fresh debian from the same set of sources which i got it working and without misdn, it fails to compile gsm.
Attached is my config output and compile break.
So should i still install misdn even though its not used.
Rgds Nik
On Mon, Mar 5, 2012 at 10:05 AM, Andreas Eversberg andreas@eversberg.euwrote:
Alexander Chemeris wrote:
Does it mean that that now we can use LCR with other SIP softswitches/PBX'es, like Freeswitch? I do not follow LCR development closely, but that would be a very interesting development.
yes, this was my intention. gsm and sip interface of lcr will not rely on misdn anymore. currently i don't support audio transfer via chan_lcr, so chan_lcr will only work with isdn interfaces. the sip interface implementation has not much options, so it can only do sipmple point-to-point sip connections to a gateway or endpoint.
Hi Andreas, i now have calls coming from external sip->LCR->gsm
But still cant figure out the dial plan to send a call out on sip, gsm->LCR->sip
I tried,
dialing=072333444 : extern interfaces=sip prefix=072333444
But on LCR trace it shows as below. I think im missing some thing small here. Appreciate if you can give a little hint.
Thanks nik
06.03.12 12:40:41.286 EP(1): ACTION (match) action goto line 11 06.03.12 12:40:41.286 EP(1): ACTION goto/menu (change to) ruleset extern dialing 072333444 06.03.12 12:40:41.286 EP(1): ACTION (match) action extern line 28 06.03.12 12:40:41.286 EP(1): ACTION extern (calling) number 072333444 interfaces sip 06.03.12 12:40:41.287 EP(1): SETUP ACKNOWLEDGE to CH(1) 06.03.12 12:40:41.287 EP(2): CHANNEL SELECTION (found given interface) interface sip 06.03.12 12:40:41.287 EP(2): INTERFACE (has no function) interface�@ 06.03.12 12:40:41.287 EP(2): INTERFACE (no free ports found) 06.03.12 12:40:41.287 EP(1): TONE to CH(1) directory default name cause_22 06.03.12 12:40:41.287 EP(1): DISCONNECT to CH(1) cause value=34 location=1-Local-PBX 06.03.12 12:40:41.287 CH(1): MNCC_DISC_REQ LCR<->BSC progress coding=3 location=1 descr=8 cause coding=3 location=1 value=34 06.03.12 12:40:56.246 CH(1): MNCC_REL_IND LCR<->BSC cause coding=3 location=0 value=16 06.03.12 12:40:56.247 EP(1): RELEASE from CH(1) cause value=16 location=0-User 06.03.12 12:40:56.247 EP(1): ACTION hangup
On Tue, Mar 6, 2012 at 7:44 AM, Nik Pakar nikpakar@gmail.com wrote:
Hi Andreas,
It was apperently compiled on my debian while i had misdn libs isntalled. Now im trying on a fresh debian from the same set of sources which i got it working and without misdn, it fails to compile gsm.
Attached is my config output and compile break.
So should i still install misdn even though its not used.
Rgds Nik
On Mon, Mar 5, 2012 at 10:05 AM, Andreas Eversberg andreas@eversberg.euwrote:
Alexander Chemeris wrote:
Does it mean that that now we can use LCR with other SIP softswitches/PBX'es, like Freeswitch? I do not follow LCR development closely, but that would be a very interesting development.
yes, this was my intention. gsm and sip interface of lcr will not rely on misdn anymore. currently i don't support audio transfer via chan_lcr, so chan_lcr will only work with isdn interfaces. the sip interface implementation has not much options, so it can only do sipmple point-to-point sip connections to a gateway or endpoint.
Hi Andreas, got it working both ways now. Many thanks for nice work.
I will now test it further with transcoding.
And start on the documentation.
Rgds Nik
On Tue, Mar 6, 2012 at 12:51 PM, Nik Pakar nikpakar@gmail.com wrote:
Hi Andreas, i now have calls coming from external sip->LCR->gsm
But still cant figure out the dial plan to send a call out on sip, gsm->LCR->sip
I tried,
dialing=072333444 : extern interfaces=sip prefix=072333444
But on LCR trace it shows as below. I think im missing some thing small here. Appreciate if you can give a little hint.
Thanks nik
06.03.12 12:40:41.286 EP(1): ACTION (match) action goto line 11 06.03.12 12:40:41.286 EP(1): ACTION goto/menu (change to) ruleset extern dialing 072333444 06.03.12 12:40:41.286 EP(1): ACTION (match) action extern line 28 06.03.12 12:40:41.286 EP(1): ACTION extern (calling) number 072333444 interfaces sip 06.03.12 12:40:41.287 EP(1): SETUP ACKNOWLEDGE to CH(1) 06.03.12 12:40:41.287 EP(2): CHANNEL SELECTION (found given interface) interface sip 06.03.12 12:40:41.287 EP(2): INTERFACE (has no function) interface�@ 06.03.12 12:40:41.287 EP(2): INTERFACE (no free ports found) 06.03.12 12:40:41.287 EP(1): TONE to CH(1) directory default name cause_22 06.03.12 12:40:41.287 EP(1): DISCONNECT to CH(1) cause value=34 location=1-Local-PBX 06.03.12 12:40:41.287 CH(1): MNCC_DISC_REQ LCR<->BSC progress coding=3 location=1 descr=8 cause coding=3 location=1 value=34 06.03.12 12:40:56.246 CH(1): MNCC_REL_IND LCR<->BSC cause coding=3 location=0 value=16 06.03.12 12:40:56.247 EP(1): RELEASE from CH(1) cause value=16 location=0-User 06.03.12 12:40:56.247 EP(1): ACTION hangup
On Tue, Mar 6, 2012 at 7:44 AM, Nik Pakar nikpakar@gmail.com wrote:
Hi Andreas,
It was apperently compiled on my debian while i had misdn libs isntalled. Now im trying on a fresh debian from the same set of sources which i got it working and without misdn, it fails to compile gsm.
Attached is my config output and compile break.
So should i still install misdn even though its not used.
Rgds Nik
On Mon, Mar 5, 2012 at 10:05 AM, Andreas Eversberg andreas@eversberg.euwrote:
Alexander Chemeris wrote:
Does it mean that that now we can use LCR with other SIP softswitches/PBX'es, like Freeswitch? I do not follow LCR development closely, but that would be a very interesting development.
yes, this was my intention. gsm and sip interface of lcr will not rely on misdn anymore. currently i don't support audio transfer via chan_lcr, so chan_lcr will only work with isdn interfaces. the sip interface implementation has not much options, so it can only do sipmple point-to-point sip connections to a gateway or endpoint.
Hi Andreas,
Call signalling all works fine right through out from NITB to LCR and out on SIP. how ever im getting some strange media behaviour.
My setup is,
[MS]---[nano.BTS]---[NITB/LCR]----[SIP Softswitch]
LCR is setup to bridge two interfaces, so what ever comes from gsm goes to sip and what ever comes from sip goes to gsm.
Now a test call from a mobile to mobile, should go all the way to the softswitch and come back.
All works fine in terms of signalling.
But in media, LCR seems sending initial SDP to the softswitch as PCMA:8 not gsm FR.
So softswitch expect the media as PCMA and not transcoding.
Same if the call goes out from softswitch, still no medial as it think incoming media from LCR is on PCMA.
Any idea about this ?
This is the LCR trace - http://pastebin.com/5PNKYc5m This is the sip trace from softswitch - http://pastebin.com/cVtx1mFB
Rgds Nik
On Tue, Mar 6, 2012 at 3:38 PM, Nik Pakar nikpakar@gmail.com wrote:
Hi Andreas, got it working both ways now. Many thanks for nice work.
I will now test it further with transcoding.
And start on the documentation.
Rgds Nik
On Tue, Mar 6, 2012 at 12:51 PM, Nik Pakar nikpakar@gmail.com wrote:
Hi Andreas, i now have calls coming from external sip->LCR->gsm
But still cant figure out the dial plan to send a call out on sip, gsm->LCR->sip
I tried,
dialing=072333444 : extern interfaces=sip prefix=072333444
But on LCR trace it shows as below. I think im missing some thing small here. Appreciate if you can give a little hint.
Thanks nik
06.03.12 12:40:41.286 EP(1): ACTION (match) action goto line 11 06.03.12 12:40:41.286 EP(1): ACTION goto/menu (change to) ruleset extern dialing 072333444 06.03.12 12:40:41.286 EP(1): ACTION (match) action extern line 28 06.03.12 12:40:41.286 EP(1): ACTION extern (calling) number 072333444 interfaces sip 06.03.12 12:40:41.287 EP(1): SETUP ACKNOWLEDGE to CH(1) 06.03.12 12:40:41.287 EP(2): CHANNEL SELECTION (found given interface) interface sip 06.03.12 12:40:41.287 EP(2): INTERFACE (has no function) interface�@ 06.03.12 12:40:41.287 EP(2): INTERFACE (no free ports found) 06.03.12 12:40:41.287 EP(1): TONE to CH(1) directory default name cause_22 06.03.12 12:40:41.287 EP(1): DISCONNECT to CH(1) cause value=34 location=1-Local-PBX 06.03.12 12:40:41.287 CH(1): MNCC_DISC_REQ LCR<->BSC progress coding=3 location=1 descr=8 cause coding=3 location=1 value=34 06.03.12 12:40:56.246 CH(1): MNCC_REL_IND LCR<->BSC cause coding=3 location=0 value=16 06.03.12 12:40:56.247 EP(1): RELEASE from CH(1) cause value=16 location=0-User 06.03.12 12:40:56.247 EP(1): ACTION hangup
On Tue, Mar 6, 2012 at 7:44 AM, Nik Pakar nikpakar@gmail.com wrote:
Hi Andreas,
It was apperently compiled on my debian while i had misdn libs isntalled. Now im trying on a fresh debian from the same set of sources which i got it working and without misdn, it fails to compile gsm.
Attached is my config output and compile break.
So should i still install misdn even though its not used.
Rgds Nik
On Mon, Mar 5, 2012 at 10:05 AM, Andreas Eversberg <andreas@eversberg.eu
wrote:
Alexander Chemeris wrote:
Does it mean that that now we can use LCR with other SIP softswitches/PBX'es, like Freeswitch? I do not follow LCR development closely, but that would be a very interesting development.
yes, this was my intention. gsm and sip interface of lcr will not rely on misdn anymore. currently i don't support audio transfer via chan_lcr, so chan_lcr will only work with isdn interfaces. the sip interface implementation has not much options, so it can only do sipmple point-to-point sip connections to a gateway or endpoint.
Hi Andreas, is the LCR actually transcoding gsm-fr to alaw towards sip side ?
Rgds Nik
On Wed, Mar 7, 2012 at 3:17 PM, Nik Pakar nikpakar@gmail.com wrote:
Hi Andreas,
Call signalling all works fine right through out from NITB to LCR and out on SIP. how ever im getting some strange media behaviour.
My setup is,
[MS]---[nano.BTS]---[NITB/LCR]----[SIP Softswitch]
LCR is setup to bridge two interfaces, so what ever comes from gsm goes to sip and what ever comes from sip goes to gsm.
Now a test call from a mobile to mobile, should go all the way to the softswitch and come back.
All works fine in terms of signalling.
But in media, LCR seems sending initial SDP to the softswitch as PCMA:8 not gsm FR.
So softswitch expect the media as PCMA and not transcoding.
Same if the call goes out from softswitch, still no medial as it think incoming media from LCR is on PCMA.
Any idea about this ?
This is the LCR trace - http://pastebin.com/5PNKYc5m This is the sip trace from softswitch - http://pastebin.com/cVtx1mFB
Rgds Nik
On Tue, Mar 6, 2012 at 3:38 PM, Nik Pakar nikpakar@gmail.com wrote:
Hi Andreas, got it working both ways now. Many thanks for nice work.
I will now test it further with transcoding.
And start on the documentation.
Rgds Nik
On Tue, Mar 6, 2012 at 12:51 PM, Nik Pakar nikpakar@gmail.com wrote:
Hi Andreas, i now have calls coming from external sip->LCR->gsm
But still cant figure out the dial plan to send a call out on sip, gsm->LCR->sip
I tried,
dialing=072333444 : extern interfaces=sip prefix=072333444
But on LCR trace it shows as below. I think im missing some thing small here. Appreciate if you can give a little hint.
Thanks nik
06.03.12 12:40:41.286 EP(1): ACTION (match) action goto line 11 06.03.12 12:40:41.286 EP(1): ACTION goto/menu (change to) ruleset extern dialing 072333444 06.03.12 12:40:41.286 EP(1): ACTION (match) action extern line 28 06.03.12 12:40:41.286 EP(1): ACTION extern (calling) number 072333444 interfaces sip 06.03.12 12:40:41.287 EP(1): SETUP ACKNOWLEDGE to CH(1) 06.03.12 12:40:41.287 EP(2): CHANNEL SELECTION (found given interface) interface sip 06.03.12 12:40:41.287 EP(2): INTERFACE (has no function) interface�@ 06.03.12 12:40:41.287 EP(2): INTERFACE (no free ports found) 06.03.12 12:40:41.287 EP(1): TONE to CH(1) directory default name cause_22 06.03.12 12:40:41.287 EP(1): DISCONNECT to CH(1) cause value=34 location=1-Local-PBX 06.03.12 12:40:41.287 CH(1): MNCC_DISC_REQ LCR<->BSC progress coding=3 location=1 descr=8 cause coding=3 location=1 value=34 06.03.12 12:40:56.246 CH(1): MNCC_REL_IND LCR<->BSC cause coding=3 location=0 value=16 06.03.12 12:40:56.247 EP(1): RELEASE from CH(1) cause value=16 location=0-User 06.03.12 12:40:56.247 EP(1): ACTION hangup
On Tue, Mar 6, 2012 at 7:44 AM, Nik Pakar nikpakar@gmail.com wrote:
Hi Andreas,
It was apperently compiled on my debian while i had misdn libs isntalled. Now im trying on a fresh debian from the same set of sources which i got it working and without misdn, it fails to compile gsm.
Attached is my config output and compile break.
So should i still install misdn even though its not used.
Rgds Nik
On Mon, Mar 5, 2012 at 10:05 AM, Andreas Eversberg < andreas@eversberg.eu> wrote:
Alexander Chemeris wrote:
Does it mean that that now we can use LCR with other SIP softswitches/PBX'es, like Freeswitch? I do not follow LCR development closely, but that would be a very interesting development.
yes, this was my intention. gsm and sip interface of lcr will not rely on misdn anymore. currently i don't support audio transfer via chan_lcr, so chan_lcr will only work with isdn interfaces. the sip interface implementation has not much options, so it can only do sipmple point-to-point sip connections to a gateway or endpoint.
It seems BSC is sending payload type GSM to LCR, but LCR send payload type PCMA on the sip channel.
07.03.12 22:45:03.907 CH(69): New call ref LCR<->BSC callref new=0x80000029 07.03.12 22:45:03.907 CH(69): Codec negotiation LCR<->BSC bearer capa='given by MS' speech version='AMR given' ignored='Not suitable for LCR' version='5 given' ignored='Not supported by LCR' version='EFR given' ignored='Not suitable for LCR' version='Full Rate given' version='Half Rate given' ignored='Not suitable for LCR' 07.03.12 22:45:03.908 CH(69): MNCC_SETUP_IND LCR<->BSC calling number=07777201 imsi=413011492012312 dialing number=4290080001 07.03.12 22:45:03.908 CH(69): MNCC_LCHAN_MODIFY LCR<->BSC speech version='Full Rate given' mode 0x01 07.03.12 22:45:03.908 CH(69): MNCC_CALL_PROC_REQ LCR<->BSC progress coding=3 location=1 descr=8 07.03.12 22:45:03.908 CH(69): unknown LCR<->BSC 07.03.12 22:45:03.908 CH(70): NEW handle handle new=0x8d65cc0 07.03.12 22:45:03.908 CH(70): INVITE from uri=sip:07777201@192.168.1.30 to uri= sip:4290080001@192.168.1.25:4757 rtp ip=103.10.172.30 port=30026,30027 payload=PCMA:8 07.03.12 22:45:03.930 CH(70): RESPOND respond value=183 07.03.12 22:45:03.930 CH(70): Payload received rtp payload=PCMA:8 payload=telephone-event:101 07.03.12 22:45:13.117 CH(69): MNCC_DISC_IND LCR<->BSC cause coding=3 location=0 value=16 07.03.12 22:45:13.148 CH(69): MNCC_REL_REQ LCR<->BSC 07.03.12 22:45:13.148 CH(70): CANCEL cause value=16 07.03.12 22:45:13.169 CH(70): RESPOND respond value=487
On Wed, Mar 7, 2012 at 3:43 PM, Nik Pakar nikpakar@gmail.com wrote:
Hi Andreas, is the LCR actually transcoding gsm-fr to alaw towards sip side ?
Rgds Nik
On Wed, Mar 7, 2012 at 3:17 PM, Nik Pakar nikpakar@gmail.com wrote:
Hi Andreas,
Call signalling all works fine right through out from NITB to LCR and out on SIP. how ever im getting some strange media behaviour.
My setup is,
[MS]---[nano.BTS]---[NITB/LCR]----[SIP Softswitch]
LCR is setup to bridge two interfaces, so what ever comes from gsm goes to sip and what ever comes from sip goes to gsm.
Now a test call from a mobile to mobile, should go all the way to the softswitch and come back.
All works fine in terms of signalling.
But in media, LCR seems sending initial SDP to the softswitch as PCMA:8 not gsm FR.
So softswitch expect the media as PCMA and not transcoding.
Same if the call goes out from softswitch, still no medial as it think incoming media from LCR is on PCMA.
Any idea about this ?
This is the LCR trace - http://pastebin.com/5PNKYc5m This is the sip trace from softswitch - http://pastebin.com/cVtx1mFB
Rgds Nik
On Tue, Mar 6, 2012 at 3:38 PM, Nik Pakar nikpakar@gmail.com wrote:
Hi Andreas, got it working both ways now. Many thanks for nice work.
I will now test it further with transcoding.
And start on the documentation.
Rgds Nik
On Tue, Mar 6, 2012 at 12:51 PM, Nik Pakar nikpakar@gmail.com wrote:
Hi Andreas, i now have calls coming from external sip->LCR->gsm
But still cant figure out the dial plan to send a call out on sip, gsm->LCR->sip
I tried,
dialing=072333444 : extern interfaces=sip prefix=072333444
But on LCR trace it shows as below. I think im missing some thing small here. Appreciate if you can give a little hint.
Thanks nik
06.03.12 12:40:41.286 EP(1): ACTION (match) action goto line 11 06.03.12 12:40:41.286 EP(1): ACTION goto/menu (change to) ruleset extern dialing 072333444 06.03.12 12:40:41.286 EP(1): ACTION (match) action extern line 28 06.03.12 12:40:41.286 EP(1): ACTION extern (calling) number 072333444 interfaces sip 06.03.12 12:40:41.287 EP(1): SETUP ACKNOWLEDGE to CH(1) 06.03.12 12:40:41.287 EP(2): CHANNEL SELECTION (found given interface) interface sip 06.03.12 12:40:41.287 EP(2): INTERFACE (has no function) interface�@ 06.03.12 12:40:41.287 EP(2): INTERFACE (no free ports found) 06.03.12 12:40:41.287 EP(1): TONE to CH(1) directory default name cause_22 06.03.12 12:40:41.287 EP(1): DISCONNECT to CH(1) cause value=34 location=1-Local-PBX 06.03.12 12:40:41.287 CH(1): MNCC_DISC_REQ LCR<->BSC progress coding=3 location=1 descr=8 cause coding=3 location=1 value=34 06.03.12 12:40:56.246 CH(1): MNCC_REL_IND LCR<->BSC cause coding=3 location=0 value=16 06.03.12 12:40:56.247 EP(1): RELEASE from CH(1) cause value=16 location=0-User 06.03.12 12:40:56.247 EP(1): ACTION hangup
On Tue, Mar 6, 2012 at 7:44 AM, Nik Pakar nikpakar@gmail.com wrote:
Hi Andreas,
It was apperently compiled on my debian while i had misdn libs isntalled. Now im trying on a fresh debian from the same set of sources which i got it working and without misdn, it fails to compile gsm.
Attached is my config output and compile break.
So should i still install misdn even though its not used.
Rgds Nik
On Mon, Mar 5, 2012 at 10:05 AM, Andreas Eversberg < andreas@eversberg.eu> wrote:
Alexander Chemeris wrote: > Does it mean that that now we can use LCR with other SIP > softswitches/PBX'es, like Freeswitch? I do not follow LCR development > closely, but that would be a very interesting development. > yes, this was my intention. gsm and sip interface of lcr will not rely on misdn anymore. currently i don't support audio transfer via chan_lcr, so chan_lcr will only work with isdn interfaces. the sip interface implementation has not much options, so it can only do sipmple point-to-point sip connections to a gateway or endpoint.
Nik,
You could check whether this is PCMA or GSM-FR with RTP stream bitrate. Capture the stream with Wireshark and look at RTP payload size or use a voice call information dialog (Telephony->VoIP Calls). PCMA bitrate is 64kbit (often 160bytes per 20ms), while bitrate GSM-FR is 13.2kbit (usually 33bytes per 20ms).
On Wed, Mar 7, 2012 at 21:36, Nik Pakar nikpakar@gmail.com wrote:
It seems BSC is sending payload type GSM to LCR, but LCR send payload type PCMA on the sip channel.
07.03.12 22:45:03.907 CH(69): New call ref LCR<->BSC callref new=0x80000029 07.03.12 22:45:03.907 CH(69): Codec negotiation LCR<->BSC bearer capa='given by MS' speech version='AMR given' ignored='Not suitable for LCR' version='5 given' ignored='Not supported by LCR' version='EFR given' ignored='Not suitable for LCR' version='Full Rate given' version='Half Rate given' ignored='Not suitable for LCR' 07.03.12 22:45:03.908 CH(69): MNCC_SETUP_IND LCR<->BSC calling number=07777201 imsi=413011492012312 dialing number=4290080001 07.03.12 22:45:03.908 CH(69): MNCC_LCHAN_MODIFY LCR<->BSC speech version='Full Rate given' mode 0x01 07.03.12 22:45:03.908 CH(69): MNCC_CALL_PROC_REQ LCR<->BSC progress coding=3 location=1 descr=8 07.03.12 22:45:03.908 CH(69): unknown LCR<->BSC 07.03.12 22:45:03.908 CH(70): NEW handle handle new=0x8d65cc0 07.03.12 22:45:03.908 CH(70): INVITE from uri=sip:07777201@192.168.1.30 to uri=sip:4290080001@192.168.1.25:4757 rtp ip=103.10.172.30 port=30026,30027 payload=PCMA:8 07.03.12 22:45:03.930 CH(70): RESPOND respond value=183 07.03.12 22:45:03.930 CH(70): Payload received rtp payload=PCMA:8 payload=telephone-event:101 07.03.12 22:45:13.117 CH(69): MNCC_DISC_IND LCR<->BSC cause coding=3 location=0 value=16 07.03.12 22:45:13.148 CH(69): MNCC_REL_REQ LCR<->BSC 07.03.12 22:45:13.148 CH(70): CANCEL cause value=16 07.03.12 22:45:13.169 CH(70): RESPOND respond value=487
On Wed, Mar 7, 2012 at 3:43 PM, Nik Pakar nikpakar@gmail.com wrote:
Hi Andreas, is the LCR actually transcoding gsm-fr to alaw towards sip side ?
Rgds Nik
On Wed, Mar 7, 2012 at 3:17 PM, Nik Pakar nikpakar@gmail.com wrote:
Hi Andreas,
Call signalling all works fine right through out from NITB to LCR and out on SIP. how ever im getting some strange media behaviour.
My setup is,
[MS]---[nano.BTS]---[NITB/LCR]----[SIP Softswitch]
LCR is setup to bridge two interfaces, so what ever comes from gsm goes to sip and what ever comes from sip goes to gsm.
Now a test call from a mobile to mobile, should go all the way to the softswitch and come back.
All works fine in terms of signalling.
But in media, LCR seems sending initial SDP to the softswitch as PCMA:8 not gsm FR.
So softswitch expect the media as PCMA and not transcoding.
Same if the call goes out from softswitch, still no medial as it think incoming media from LCR is on PCMA.
Any idea about this ?
This is the LCR trace - http://pastebin.com/5PNKYc5m This is the sip trace from softswitch - http://pastebin.com/cVtx1mFB
Rgds Nik
On Tue, Mar 6, 2012 at 3:38 PM, Nik Pakar nikpakar@gmail.com wrote:
Hi Andreas, got it working both ways now. Many thanks for nice work.
I will now test it further with transcoding.
And start on the documentation.
Rgds Nik
On Tue, Mar 6, 2012 at 12:51 PM, Nik Pakar nikpakar@gmail.com wrote:
Hi Andreas, i now have calls coming from external sip->LCR->gsm
But still cant figure out the dial plan to send a call out on sip, gsm->LCR->sip
I tried,
dialing=072333444 : extern interfaces=sip prefix=072333444
But on LCR trace it shows as below. I think im missing some thing small here. Appreciate if you can give a little hint.
Thanks nik
06.03.12 12:40:41.286 EP(1): ACTION (match) action goto line 11 06.03.12 12:40:41.286 EP(1): ACTION goto/menu (change to) ruleset extern dialing 072333444 06.03.12 12:40:41.286 EP(1): ACTION (match) action extern line 28 06.03.12 12:40:41.286 EP(1): ACTION extern (calling) number 072333444 interfaces sip 06.03.12 12:40:41.287 EP(1): SETUP ACKNOWLEDGE to CH(1) 06.03.12 12:40:41.287 EP(2): CHANNEL SELECTION (found given interface) interface sip 06.03.12 12:40:41.287 EP(2): INTERFACE (has no function) interface�@ 06.03.12 12:40:41.287 EP(2): INTERFACE (no free ports found) 06.03.12 12:40:41.287 EP(1): TONE to CH(1) directory default name cause_22 06.03.12 12:40:41.287 EP(1): DISCONNECT to CH(1) cause value=34 location=1-Local-PBX 06.03.12 12:40:41.287 CH(1): MNCC_DISC_REQ LCR<->BSC progress coding=3 location=1 descr=8 cause coding=3 location=1 value=34 06.03.12 12:40:56.246 CH(1): MNCC_REL_IND LCR<->BSC cause coding=3 location=0 value=16 06.03.12 12:40:56.247 EP(1): RELEASE from CH(1) cause value=16 location=0-User 06.03.12 12:40:56.247 EP(1): ACTION hangup
On Tue, Mar 6, 2012 at 7:44 AM, Nik Pakar nikpakar@gmail.com wrote:
Hi Andreas,
It was apperently compiled on my debian while i had misdn libs isntalled. Now im trying on a fresh debian from the same set of sources which i got it working and without misdn, it fails to compile gsm.
Attached is my config output and compile break.
So should i still install misdn even though its not used.
Rgds Nik
On Mon, Mar 5, 2012 at 10:05 AM, Andreas Eversberg andreas@eversberg.eu wrote: > > Alexander Chemeris wrote: > > Does it mean that that now we can use LCR with other SIP > > softswitches/PBX'es, like Freeswitch? I do not follow LCR > > development > > closely, but that would be a very interesting development. > > > yes, this was my intention. gsm and sip interface of lcr will not > rely > on misdn anymore. currently i don't support audio transfer via > chan_lcr, > so chan_lcr will only work with isdn interfaces. the sip interface > implementation has not much options, so it can only do sipmple > point-to-point sip connections to a gateway or endpoint.
Hi Alexander,
I will check that tomorrow and update. But the problem here is LCR get GSM-FR from MNCC and forward SDP as PCMA on SIP. So it will confuse the softswitch as it see the stream as PCMA even it comes on FR. and if its actually PCMA, then LCR must have transcode it.
Either way no audio once call connected.
On openbsc console i see some warning on lchan with no rtp.
Rgds Nik
On Wed, Mar 7, 2012 at 6:32 PM, Alexander Chemeris < alexander.chemeris@gmail.com> wrote:
Nik,
You could check whether this is PCMA or GSM-FR with RTP stream bitrate. Capture the stream with Wireshark and look at RTP payload size or use a voice call information dialog (Telephony->VoIP Calls). PCMA bitrate is 64kbit (often 160bytes per 20ms), while bitrate GSM-FR is 13.2kbit (usually 33bytes per 20ms).
On Wed, Mar 7, 2012 at 21:36, Nik Pakar nikpakar@gmail.com wrote:
It seems BSC is sending payload type GSM to LCR, but LCR send payload
type
PCMA on the sip channel.
07.03.12 22:45:03.907 CH(69): New call ref LCR<->BSC callref
new=0x80000029
07.03.12 22:45:03.907 CH(69): Codec negotiation LCR<->BSC bearer capa='given by MS' speech version='AMR given' ignored='Not suitable for LCR' version='5 given' ignored='Not supported by LCR' version='EFR given' ignored='Not suitable for LCR' version='Full Rate given' version='Half
Rate
given' ignored='Not suitable for LCR' 07.03.12 22:45:03.908 CH(69): MNCC_SETUP_IND LCR<->BSC calling number=07777201 imsi=413011492012312 dialing number=4290080001 07.03.12 22:45:03.908 CH(69): MNCC_LCHAN_MODIFY LCR<->BSC speech version='Full Rate given' mode 0x01 07.03.12 22:45:03.908 CH(69): MNCC_CALL_PROC_REQ LCR<->BSC progress coding=3 location=1 descr=8 07.03.12 22:45:03.908 CH(69): unknown LCR<->BSC 07.03.12 22:45:03.908 CH(70): NEW handle handle new=0x8d65cc0 07.03.12 22:45:03.908 CH(70): INVITE from uri=sip:07777201@192.168.1.30 to uri=sip:4290080001@192.168.1.25:4757 rtp ip=103.10.172.30
port=30026,30027
payload=PCMA:8 07.03.12 22:45:03.930 CH(70): RESPOND respond value=183 07.03.12 22:45:03.930 CH(70): Payload received rtp payload=PCMA:8 payload=telephone-event:101 07.03.12 22:45:13.117 CH(69): MNCC_DISC_IND LCR<->BSC cause coding=3 location=0 value=16 07.03.12 22:45:13.148 CH(69): MNCC_REL_REQ LCR<->BSC 07.03.12 22:45:13.148 CH(70): CANCEL cause value=16 07.03.12 22:45:13.169 CH(70): RESPOND respond value=487
On Wed, Mar 7, 2012 at 3:43 PM, Nik Pakar nikpakar@gmail.com wrote:
Hi Andreas, is the LCR actually transcoding gsm-fr to alaw towards sip side ?
Rgds Nik
On Wed, Mar 7, 2012 at 3:17 PM, Nik Pakar nikpakar@gmail.com wrote:
Hi Andreas,
Call signalling all works fine right through out from NITB to LCR and
out
on SIP. how ever im getting some strange media behaviour.
My setup is,
[MS]---[nano.BTS]---[NITB/LCR]----[SIP Softswitch]
LCR is setup to bridge two interfaces, so what ever comes from gsm goes to sip and what ever comes from sip goes to gsm.
Now a test call from a mobile to mobile, should go all the way to the softswitch and come back.
All works fine in terms of signalling.
But in media, LCR seems sending initial SDP to the softswitch as PCMA:8 not gsm FR.
So softswitch expect the media as PCMA and not transcoding.
Same if the call goes out from softswitch, still no medial as it think incoming media from LCR is on PCMA.
Any idea about this ?
This is the LCR trace - http://pastebin.com/5PNKYc5m This is the sip trace from softswitch - http://pastebin.com/cVtx1mFB
Rgds Nik
On Tue, Mar 6, 2012 at 3:38 PM, Nik Pakar nikpakar@gmail.com wrote:
Hi Andreas, got it working both ways now. Many thanks for nice work.
I will now test it further with transcoding.
And start on the documentation.
Rgds Nik
On Tue, Mar 6, 2012 at 12:51 PM, Nik Pakar nikpakar@gmail.com
wrote:
Hi Andreas, i now have calls coming from external sip->LCR->gsm
But still cant figure out the dial plan to send a call out on sip, gsm->LCR->sip
I tried,
dialing=072333444 : extern interfaces=sip prefix=072333444
But on LCR trace it shows as below. I think im missing some thing
small
here. Appreciate if you can give a little hint.
Thanks nik
06.03.12 12:40:41.286 EP(1): ACTION (match) action goto line 11 06.03.12 12:40:41.286 EP(1): ACTION goto/menu (change to) ruleset extern dialing 072333444 06.03.12 12:40:41.286 EP(1): ACTION (match) action extern line 28 06.03.12 12:40:41.286 EP(1): ACTION extern (calling) number
072333444
interfaces sip 06.03.12 12:40:41.287 EP(1): SETUP ACKNOWLEDGE to CH(1) 06.03.12 12:40:41.287 EP(2): CHANNEL SELECTION (found given
interface)
interface sip 06.03.12 12:40:41.287 EP(2): INTERFACE (has no function) interface�@ 06.03.12 12:40:41.287 EP(2): INTERFACE (no free ports found) 06.03.12 12:40:41.287 EP(1): TONE to CH(1) directory default name cause_22 06.03.12 12:40:41.287 EP(1): DISCONNECT to CH(1) cause value=34 location=1-Local-PBX 06.03.12 12:40:41.287 CH(1): MNCC_DISC_REQ LCR<->BSC progress
coding=3
location=1 descr=8 cause coding=3 location=1 value=34 06.03.12 12:40:56.246 CH(1): MNCC_REL_IND LCR<->BSC cause coding=3 location=0 value=16 06.03.12 12:40:56.247 EP(1): RELEASE from CH(1) cause value=16 location=0-User 06.03.12 12:40:56.247 EP(1): ACTION hangup
On Tue, Mar 6, 2012 at 7:44 AM, Nik Pakar nikpakar@gmail.com
wrote:
> > Hi Andreas, > > It was apperently compiled on my debian while i had misdn libs > isntalled. Now im trying on a fresh debian from the same set of
sources
> which i got it working and without misdn, it fails to compile gsm. > > Attached is my config output and compile break. > > http://pastebin.com/W6UHn4Lc > > So should i still install misdn even though its not used. > > Rgds > Nik > > > On Mon, Mar 5, 2012 at 10:05 AM, Andreas Eversberg > andreas@eversberg.eu wrote: >> >> Alexander Chemeris wrote: >> > Does it mean that that now we can use LCR with other SIP >> > softswitches/PBX'es, like Freeswitch? I do not follow LCR >> > development >> > closely, but that would be a very interesting development. >> > >> yes, this was my intention. gsm and sip interface of lcr will not >> rely >> on misdn anymore. currently i don't support audio transfer via >> chan_lcr, >> so chan_lcr will only work with isdn interfaces. the sip interface >> implementation has not much options, so it can only do sipmple >> point-to-point sip connections to a gateway or endpoint. > >
-- Regards, Alexander Chemeris.
hi nik,
since lcr uses alaw (optionally ulaw) internally, it transcodes the audio between alaw and gsm full rate.
alternatively you can bridge rtp directly between freeswitch and openbsc, if you use the "bridge <interface>" feature at interface.conf. additionally you need to add a line "rtp-bridge" to your interface definition. this rtp-bridge feature only works with the bridge feature, since lcr will not be able to handle audio streams then. since it is still developed, you need to use the "jolly/rtpmux" branch of openbsc. then it will only offer the codec to freeswitch which the phone supports. on the other direction, lcr selects the commonly supported codec. if more codecs are supported, the upper most (prefered) is used.
anyway it should work in both cases. (with and without rtp-bridge feature)
regards,
andreas
Thank you very andreas, for the help and nice integration work.
Im going to try this branch now.
Is it only this patch to be applied to the rtp_proxy.c for the git clone i have from openbsc ?
http://cgit.osmocom.org/cgit/openbsc/commit/?h=jolly/rtpmux
On Thu, Mar 8, 2012 at 8:09 AM, jolly andreas@eversberg.eu wrote:
hi nik,
since lcr uses alaw (optionally ulaw) internally, it transcodes the audio between alaw and gsm full rate.
alternatively you can bridge rtp directly between freeswitch and openbsc, if you use the "bridge <interface>" feature at interface.conf. additionally you need to add a line "rtp-bridge" to your interface definition. this rtp-bridge feature only works with the bridge feature, since lcr will not be able to handle audio streams then. since it is still developed, you need to use the "jolly/rtpmux" branch of openbsc. then it will only offer the codec to freeswitch which the phone supports. on the other direction, lcr selects the commonly supported codec. if more codecs are supported, the upper most (prefered) is used.
anyway it should work in both cases. (with and without rtp-bridge feature)
regards,
andreas
Thanks andreas,
Im bit new on GIT, thats why asked.
I tried git checkout jolly/rtpmux from the openbsc clone directory but i didnt seems synced with the jolly/rtpmux branch.
Would you mind give me a little hint on how to clone the rtpmux branch.
Thanks and sorry for basic question on git.
Rgds Nik
On Thu, Mar 8, 2012 at 2:07 PM, jolly andreas@eversberg.eu wrote:
Is it only this patch to be applied to the rtp_proxy.c for the git clone i have from openbsc ?
no, you need most of this branch, because the patch affects the mncc-interface between openbsc and lcr, too.
Nik Pakar wrote:
Would you mind give me a little hint on how to clone the rtpmux branch.
Thanks and sorry for basic question on git.
When you know that you need to apologize you obviously know that you are asking a question that you are not supposed to be asking (here) in the first place.
Study the tools! There is amazing material about Git online so you have absolutely no excuse for not trying much harder on your own to accomplish what you are obviously getting paid a lot of money to do.
Did you already look around, or are you asking (i.e. expecting) others to help you without trying to find the answer on your own? That is quite unacceptable and will make people hate you, because as you know everyone is very busy with other things than educating you about git. I think you understand this.
I strongly recommend http://progit.org/book/ which has very good coverage of git, and is easy to read.
You can't expect any success whatsoever in a project where you are new to the development tools, if you aren't willing, eager and able to learn those tools. You will just make people angry in general and angry with you in particular.
You must understand that open source communities are a completely different social construct than what you may be used to from commercial customer/supplier relationships.
Your behavior on this mailing list (and apparently others) is as if the open source community was a commercial supplier to you and you are rapidly making one faux-pas after another, which naturally results in an incredibly bad standing for you within the community.
If you are going to be successful in using an open source component then you must change your mindset and become a part of the community.
The only way to do that, really, is to contribute something.
//Peter
Hi Andreas, thousand times thanks for the correct hint.
Its indeed crystal clear now.
I have tried the git clone from git://git.osmocom.org/openbsc.git but noticed there is still misdn requirement and also no sip support.
Then found from the maillist that i should clone git://git.osmocom.eu/openbsc.git git://git.osmocom.org/openbsc.git instead of .org which shows sip interfaces on the source.
How ever there is no configure file on this source.
Am i still checking out from the wrong git ?
Once again highly appreciate some light on the final bit.
Thanks Nik
On Mon, Mar 5, 2012 at 7:44 AM, jolly andreas@eversberg.eu wrote:
Do I have to have mISDN for LCR even im not going to use any isdn interface ? Im trying to connect the NITB to LCR and LCR to asterisk all on ip.
Thanks for any help.
Rgds Nik
hi nik,
you don't need misdn to use lcr with sip and gsm anymore. also you don't need any patch. the lcr and openbsc compile out of the box supporting each other. the howto is a bit outdated.
try to compile the lcr from the git. don't use chan_lcr, since it still works with isdn only. you need to setup a sip interface in interface.conf:
[sip] sip <local ip> <remote ip>[:<port>] sip 10.0.0.12 10.0.0.34 earlyb no tones no
use asterisk machine for remote ip. if you have asterisk on the same machine, change the sip port of asterisk and use "remoteip:port".
regards,
andreas
Hi Andreas, extreamlly sorry for the blind question.
I didnt do autogen to create the configure.
Extreamly sorry again.
Rgds Nik
On Mon, Mar 5, 2012 at 10:19 AM, Nik Pakar nikpakar@gmail.com wrote:
Hi Andreas, thousand times thanks for the correct hint.
Its indeed crystal clear now.
I have tried the git clone from git://git.osmocom.org/openbsc.git but noticed there is still misdn requirement and also no sip support.
Then found from the maillist that i should clone git://git.osmocom.eu/openbsc.git instead of .org which shows sip interfaces on the source.
How ever there is no configure file on this source.
Am i still checking out from the wrong git ?
Once again highly appreciate some light on the final bit.
Thanks Nik
On Mon, Mar 5, 2012 at 7:44 AM, jolly andreas@eversberg.eu wrote:
Do I have to have mISDN for LCR even im not going to use any isdn interface ? Im trying to connect the NITB to LCR and LCR to asterisk all on ip.
Thanks for any help.
Rgds Nik
hi nik,
you don't need misdn to use lcr with sip and gsm anymore. also you don't need any patch. the lcr and openbsc compile out of the box supporting each other. the howto is a bit outdated.
try to compile the lcr from the git. don't use chan_lcr, since it still works with isdn only. you need to setup a sip interface in interface.conf:
[sip] sip <local ip> <remote ip>[:<port>] sip 10.0.0.12 10.0.0.34 earlyb no tones no
use asterisk machine for remote ip. if you have asterisk on the same machine, change the sip port of asterisk and use "remoteip:port".
regards,
andreas
Hi Andreas,
Every thing works fine.
Just one last question. I cant find any sample dialplan option towards gsm and sip interface. tried quite a few didnt worked.
Appreciate if you can just give the last hint how would the dial plan should looks like towards gsm and sip.
Thanks again. Nik
On Mon, Mar 5, 2012 at 10:23 AM, Nik Pakar nikpakar@gmail.com wrote:
Hi Andreas, extreamlly sorry for the blind question.
I didnt do autogen to create the configure.
Extreamly sorry again.
Rgds Nik
On Mon, Mar 5, 2012 at 10:19 AM, Nik Pakar nikpakar@gmail.com wrote:
Hi Andreas, thousand times thanks for the correct hint.
Its indeed crystal clear now.
I have tried the git clone from git://git.osmocom.org/openbsc.git but noticed there is still misdn requirement and also no sip support.
Then found from the maillist that i should clone git://git.osmocom.eu/openbsc.git instead of .org which shows sip interfaces on the source.
How ever there is no configure file on this source.
Am i still checking out from the wrong git ?
Once again highly appreciate some light on the final bit.
Thanks Nik
On Mon, Mar 5, 2012 at 7:44 AM, jolly andreas@eversberg.eu wrote:
Do I have to have mISDN for LCR even im not going to use any isdn interface ? Im trying to connect the NITB to LCR and LCR to asterisk all on ip.
Thanks for any help.
Rgds Nik
hi nik,
you don't need misdn to use lcr with sip and gsm anymore. also you don't need any patch. the lcr and openbsc compile out of the box supporting each other. the howto is a bit outdated.
try to compile the lcr from the git. don't use chan_lcr, since it still works with isdn only. you need to setup a sip interface in interface.conf:
[sip] sip <local ip> <remote ip>[:<port>] sip 10.0.0.12 10.0.0.34 earlyb no tones no
use asterisk machine for remote ip. if you have asterisk on the same machine, change the sip port of asterisk and use "remoteip:port".
regards,
andreas
Dear Nik,
i am trying to finalize the build of lcr w/ gsm/sip support
*latest lcr version (in /usr/src/lcr) *latest openbsc version (in /usr/src/openbsc/openbsc) *./configure --prefix=/usr/src/lcr/ --with-gsm-bs --with-sip works ok (with libsofia-sip-ua-dev and libncurses5-dev) *make does break gsm_audio.c:20: error: 'gsm' was not expected in this scope
Can you tell what option you gave to configure and how did you organize your dirs ?
My best,
Xavier.
On Mon, Mar 05, 2012 at 11:39:28AM +0000, Nik Pakar wrote:
Hi Andreas,
Every thing works fine.
Just one last question. I cant find any sample dialplan option towards gsm and sip interface. tried quite a few didnt worked.
Appreciate if you can just give the last hint how would the dial plan should looks like towards gsm and sip.
Thanks again. Nik
On Mon, Mar 5, 2012 at 10:23 AM, Nik Pakar nikpakar@gmail.com wrote:
Hi Andreas, extreamlly sorry for the blind question.
I didnt do autogen to create the configure.
Extreamly sorry again.
Rgds Nik
On Mon, Mar 5, 2012 at 10:19 AM, Nik Pakar nikpakar@gmail.com wrote:
Hi Andreas, thousand times thanks for the correct hint.
Its indeed crystal clear now.
I have tried the git clone from git://git.osmocom.org/openbsc.git but noticed there is still misdn requirement and also no sip support.
Then found from the maillist that i should clone git://git.osmocom.eu/openbsc.git instead of .org which shows sip interfaces on the source.
How ever there is no configure file on this source.
Am i still checking out from the wrong git ?
Once again highly appreciate some light on the final bit.
Thanks Nik
On Mon, Mar 5, 2012 at 7:44 AM, jolly andreas@eversberg.eu wrote:
Do I have to have mISDN for LCR even im not going to use any isdn interface ? Im trying to connect the NITB to LCR and LCR to asterisk all on ip.
Thanks for any help.
Rgds Nik
hi nik,
you don't need misdn to use lcr with sip and gsm anymore. also you don't need any patch. the lcr and openbsc compile out of the box supporting each other. the howto is a bit outdated.
try to compile the lcr from the git. don't use chan_lcr, since it still works with isdn only. you need to setup a sip interface in interface.conf:
[sip] sip <local ip> <remote ip>[:<port>] sip 10.0.0.12 10.0.0.34 earlyb no tones no
use asterisk machine for remote ip. if you have asterisk on the same machine, change the sip port of asterisk and use "remoteip:port".
regards,
andreas
Did you install libgsm ?
Im in the last bit of getting it done. Once all done i will redo a documentation.
On Mon, Mar 5, 2012 at 1:10 PM, carcelle carcellelist@free.fr wrote:
Dear Nik,
i am trying to finalize the build of lcr w/ gsm/sip support
*latest lcr version (in /usr/src/lcr) *latest openbsc version (in /usr/src/openbsc/openbsc) *./configure --prefix=/usr/src/lcr/ --with-gsm-bs --with-sip works ok (with libsofia-sip-ua-dev and libncurses5-dev) *make does break gsm_audio.c:20: error: 'gsm' was not expected in this scope
Can you tell what option you gave to configure and how did you organize your dirs ?
My best,
Xavier.
On Mon, Mar 05, 2012 at 11:39:28AM +0000, Nik Pakar wrote:
Hi Andreas,
Every thing works fine.
Just one last question. I cant find any sample dialplan option towards
gsm
and sip interface. tried quite a few didnt worked.
Appreciate if you can just give the last hint how would the dial plan should looks like towards gsm and sip.
Thanks again. Nik
On Mon, Mar 5, 2012 at 10:23 AM, Nik Pakar nikpakar@gmail.com wrote:
Hi Andreas, extreamlly sorry for the blind question.
I didnt do autogen to create the configure.
Extreamly sorry again.
Rgds Nik
On Mon, Mar 5, 2012 at 10:19 AM, Nik Pakar nikpakar@gmail.com wrote:
Hi Andreas, thousand times thanks for the correct hint.
Its indeed crystal clear now.
I have tried the git clone from git://git.osmocom.org/openbsc.git but noticed there is still misdn requirement and also no sip support.
Then found from the maillist that i should clone git://git.osmocom.eu/openbsc.git instead of .org which shows sip interfaces on the source.
How ever there is no configure file on this source.
Am i still checking out from the wrong git ?
Once again highly appreciate some light on the final bit.
Thanks Nik
On Mon, Mar 5, 2012 at 7:44 AM, jolly andreas@eversberg.eu wrote:
Do I have to have mISDN for LCR even im not going to use any isdn interface ? Im trying to connect the NITB to LCR and LCR to
asterisk
all on ip.
Thanks for any help.
Rgds Nik
hi nik,
you don't need misdn to use lcr with sip and gsm anymore. also you
don't
need any patch. the lcr and openbsc compile out of the box supporting each other. the howto is a bit outdated.
try to compile the lcr from the git. don't use chan_lcr, since it
still
works with isdn only. you need to setup a sip interface in interface.conf:
[sip] sip <local ip> <remote ip>[:<port>] sip 10.0.0.12 10.0.0.34 earlyb no tones no
use asterisk machine for remote ip. if you have asterisk on the same machine, change the sip port of asterisk and use "remoteip:port".
regards,
andreas
Hi Andreas, searched out every possible place to look for a sample routing expression for sofia and for gsm interfaces. But couldnt found any.
Would highly appreciate if you can shed little light on that.
Thanks nik
On Mon, Mar 5, 2012 at 11:39 AM, Nik Pakar nikpakar@gmail.com wrote:
Hi Andreas,
Every thing works fine.
Just one last question. I cant find any sample dialplan option towards gsm and sip interface. tried quite a few didnt worked.
Appreciate if you can just give the last hint how would the dial plan should looks like towards gsm and sip.
Thanks again. Nik
On Mon, Mar 5, 2012 at 10:23 AM, Nik Pakar nikpakar@gmail.com wrote:
Hi Andreas, extreamlly sorry for the blind question.
I didnt do autogen to create the configure.
Extreamly sorry again.
Rgds Nik
On Mon, Mar 5, 2012 at 10:19 AM, Nik Pakar nikpakar@gmail.com wrote:
Hi Andreas, thousand times thanks for the correct hint.
Its indeed crystal clear now.
I have tried the git clone from git://git.osmocom.org/openbsc.git but noticed there is still misdn requirement and also no sip support.
Then found from the maillist that i should clone git://git.osmocom.eu/openbsc.git instead of .org which shows sip interfaces on the source.
How ever there is no configure file on this source.
Am i still checking out from the wrong git ?
Once again highly appreciate some light on the final bit.
Thanks Nik
On Mon, Mar 5, 2012 at 7:44 AM, jolly andreas@eversberg.eu wrote:
Do I have to have mISDN for LCR even im not going to use any isdn interface ? Im trying to connect the NITB to LCR and LCR to asterisk all on ip.
Thanks for any help.
Rgds Nik
hi nik,
you don't need misdn to use lcr with sip and gsm anymore. also you don't need any patch. the lcr and openbsc compile out of the box supporting each other. the howto is a bit outdated.
try to compile the lcr from the git. don't use chan_lcr, since it still works with isdn only. you need to setup a sip interface in interface.conf:
[sip] sip <local ip> <remote ip>[:<port>] sip 10.0.0.12 10.0.0.34 earlyb no tones no
use asterisk machine for remote ip. if you have asterisk on the same machine, change the sip port of asterisk and use "remoteip:port".
regards,
andreas
Nik Pakar wrote:
searched out every possible place to look for a sample routing expression for sofia and for gsm interfaces. But couldnt found any.
Wrong approach. You can not depend on examples and help from others to accomplish what you want to do. You must understand that YOU must do the job. YOU must learn how to create the routing that YOU need.
Would highly appreciate if you can shed little light on that.
I recommend that you look at the source code to learn how to control routing. That may not be the kind of documentation you are used to, but you can on the other hand know with certainty that what you learn will be relevant for the running code. After you have gained this understanding, I strongly recommend that you then immediately STOP and create documentation for the benefit of others, so that they do not have to spend as much time as you on learning the neccessary things.
This is a good way to contribute back to a community which has spent many thousand hours on developing software which you can simply install in order to reach your goal, without any cost.
//Peter
Hello Andreas,
What changes need to be done in routing.conf to use pipe via SIP instead of chan_lcr?
Thanks, -Don
On Sun, Mar 4, 2012 at 11:44 PM, jolly andreas@eversberg.eu wrote:
Do I have to have mISDN for LCR even im not going to use any isdn interface ? Im trying to connect the NITB to LCR and LCR to asterisk all on ip.
Thanks for any help.
Rgds Nik
hi nik,
you don't need misdn to use lcr with sip and gsm anymore. also you don't need any patch. the lcr and openbsc compile out of the box supporting each other. the howto is a bit outdated.
try to compile the lcr from the git. don't use chan_lcr, since it still works with isdn only. you need to setup a sip interface in interface.conf:
[sip] sip <local ip> <remote ip>[:<port>] sip 10.0.0.12 10.0.0.34 earlyb no tones no
use asterisk machine for remote ip. if you have asterisk on the same machine, change the sip port of asterisk and use "remoteip:port".
regards,
andreas
Hi Don,
You can use interface bridge option on interface.conf, so no need of any routing. sip and gsm interfaces will be bridged and calls coming on one interface will route to the other.
Just use bridge sip on interace gsm and bridge gsm interface sip.
Rgds Nik
On Wed, Mar 14, 2012 at 4:20 AM, Don Fanning don@00100100.net wrote:
Hello Andreas,
What changes need to be done in routing.conf to use pipe via SIP instead of chan_lcr?
Thanks, -Don
On Sun, Mar 4, 2012 at 11:44 PM, jolly andreas@eversberg.eu wrote:
Do I have to have mISDN for LCR even im not going to use any isdn interface ? Im trying to connect the NITB to LCR and LCR to asterisk all on ip.
Thanks for any help.
Rgds Nik
hi nik,
you don't need misdn to use lcr with sip and gsm anymore. also you don't need any patch. the lcr and openbsc compile out of the box supporting each other. the howto is a bit outdated.
try to compile the lcr from the git. don't use chan_lcr, since it still works with isdn only. you need to setup a sip interface in interface.conf:
[sip] sip <local ip> <remote ip>[:<port>] sip 10.0.0.12 10.0.0.34 earlyb no tones no
use asterisk machine for remote ip. if you have asterisk on the same machine, change the sip port of asterisk and use "remoteip:port".
regards,
andreas
-
Alexander Chemeris -
Alexander Huemer -
Andreas Eversberg -
carcelle -
Don Fanning -
jolly
-
Nik Pakar -
Peter Stuge