hi luca,
it seems that there is a change in the API. therefore i prepared a "snapshot" at www.linux-call-router.de to be downloaded. this version of openbsc and lcr is older, but works together. as soon as the current openbsc has been tested more (channel ressource handling has changed for example), i will make it work again.
regards,
andreas
#0 0x0805b009 in gsm0408_rcvmsg (msg=0x864ffb8, link_id=0 '\0') at
openbsc/src/bsc_api.c:111
111 rc = api->compl_l3(lchan->conn, msg,
0);
The problem seems to be the variable api, which is NULL.
(gdb) print api $1 = (struct bsc_api *) 0x0
Am Wed, 7 Jul 2010 15:53:27 +0200 schrieb "Andreas.Eversberg" Andreas.Eversberg@versatel.de:
it seems that there is a change in the API. therefore i prepared a "snapshot" at www.linux-call-router.de to be downloaded. this version of openbsc and lcr is older, but works together. as soon as the current openbsc has been tested more (channel ressource handling has changed for example), i will make it work again.
Hi,
I tried this version unfortunately unsuccessfully...
When I try to call a mobile (from Twinkle or other VoIP-phone) I get this error:
== Using SIP RTP CoS mark 5 -- Executing [12345@btsctrl:1] Dial("SIP/debian503-00000006", "SIP/12345@btsctrl,120") in new stack == Using SIP RTP CoS mark 5 [Jul 7 16:06:24] WARNING[24859]: chan_sip.c:5340 create_addr: No such host: btsctrl [Jul 7 16:06:24] WARNING[24859]: app_dial.c:1747 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Auto fallthrough, channel 'SIP/debian503-00000006' status is 'CHANUNAVAIL' == Using SIP RTP CoS mark 5
In my sip.conf I have this:
[debian503] type=friend username=debian503 md5secret=02e540ea7754341606ad6cf1b3c9e618 host=static context=btsctrl dtmfmode=inband
and in extensions.conf this:
[btsctrl] ;;exten => _X.,1,Set,CALLERID(num)=5552342 exten => _X.,1,dial(SIP/${EXTEN}@btsctrl,120)
Can someone help me?
Thanks a lot!
Dear Luca,
On Mon, Jul 12, 2010 at 04:28:19PM +0200, Luca Bertoncello wrote:
When I try to call a mobile (from Twinkle or other VoIP-phone) I get this error:
== Using SIP RTP CoS mark 5 -- Executing [12345@btsctrl:1] Dial("SIP/debian503-00000006", "SIP/12345@btsctrl,120") in new stack == Using SIP RTP CoS mark 5 [Jul 7 16:06:24] WARNING[24859]: chan_sip.c:5340 create_addr: No such host: btsctrl [Jul 7 16:06:24] WARNING[24859]: app_dial.c:1747 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Auto fallthrough, channel 'SIP/debian503-00000006' status is 'CHANUNAVAIL' == Using SIP RTP CoS mark 5
1) Those error messages do not seem to relate to OpenBSC at all 2) Please look at them carefully Dial("SIP/debian503-00000006", "SIP/12345@btsctrl,120" No such host: btsctrl
Your program seems to want to establish a connection to the host "btsctrl" but that host does not seem to have a DNS or other name resolving entry, it thus fails to obtain an IP address and does not connect to wherever you want to connect to.
Please keep this list on-topic, i.e. OpenBSC related, including OpenBSC + lcr. But people here have little interest in discussing Asterisk configuration. Thanks, Harald
Am Tue, 13 Jul 2010 16:11:01 +0200 schrieb Harald Welte laforge@gnumonks.org:
- Those error messages do not seem to relate to OpenBSC at all
- Please look at them carefully Dial("SIP/debian503-00000006", "SIP/12345@btsctrl,120" No such host: btsctrl
Your program seems to want to establish a connection to the host "btsctrl" but that host does not seem to have a DNS or other name resolving entry, it thus fails to obtain an IP address and does not connect to wherever you want to connect to.
After the HowTo from http://openbsc.osmocom.org/trac/wiki/OpenBSC_LCR it should be the Asterisk-context. But I tried to add btsctrl in my /etc/hosts (with the IP of my PC), too. It makes a loop, and it does not work again.
Please keep this list on-topic, i.e. OpenBSC related, including OpenBSC + lcr. But people here have little interest in discussing Asterisk configuration. Thanks,
I know, and I'm very sorry to annoying you with this question, but my Asterisk works (I can call VoIP-phones). It seems to be a problem of the integration between OpenBSC and LCR, or a problem of LCR.
Has someone a running integration OpenBSC/Asterisk and can send to the configuration?
Thanks
Am Wed, 7 Jul 2010 15:53:27 +0200 schrieb "Andreas.Eversberg" Andreas.Eversberg@versatel.de:
it seems that there is a change in the API. therefore i prepared a "snapshot" at www.linux-call-router.de to be downloaded. this version of openbsc and lcr is older, but works together. as soon as the current openbsc has been tested more (channel ressource handling has changed for example), i will make it work again.
regards,
andreas
#0 0x0805b009 in gsm0408_rcvmsg (msg=0x864ffb8, link_id=0 '\0') at
openbsc/src/bsc_api.c:111
111 rc = api->compl_l3(lchan->conn, msg,
0);
The problem seems to be the variable api, which is NULL.
(gdb) print api $1 = (struct bsc_api *) 0x0
Hi, List and Andreas!
Today I debugged your program and I found the problem. OpenBSC, in his main(), initializes the struct bsc_api, but LCR does NOT make it!
I wrote a patch. Here is it! Now it is possible to compile the last version of OpenBSC and LCR (from git) and to get them together working.
Besides, the HowTo at http://openbsc.osmocom.org/trac/wiki/OpenBSC_LCR is for a OLD version of Asterisk and LCR/OpenBSC. 1) It is not necessary to change Makefile.am and gsm_audio.c 2) The option for configure is no more --with-gsm but --with-gsm-bs 3) In extensions.conf the right syntax is:
[btsctrl] exten => _X.,1,Set,CALLERID(num)=5552342 exten => _X.,n,Dial(LCR/GSM/${EXTEN},120)
I hope, this will help someone other trying to get OpenBSC and Asterisk working together!
Regards
-
Andreas.Eversberg -
Harald Welte -
Holger Hans Peter Freyther -
Luca Bertoncello