- OpenBSC - lists.osmocom.org

Re: Meet us at AfricaCom
by Roelf Diedericks 14 Nov '13

14 Nov '13

I met with Alexander today at AfricaCom, and as an interested user of OsmoCOM and the proud owner of a SysmoBTS I can say that the email was certainly of interest to me. It is a very small community, and whilst commercial spamming shouldn't be allowed, I do feel that for the very small special interest group we all participate in, some relaxed attitude may be warranted. The list is currently unobtrusive and if it ever gets as busy as LKML, I'm sure we can do some benevolent dicatatoring. If it were Huawei spamming the list about the conference, then I'd agree, but the Fairwaves guys have generously opened up their hardware designs as well. I don't think it gets more open than that... Regards, Roelf.

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Signalling in OpenBSC - LCR - Asterisk
by Leonardo Nve 12 Nov '13

12 Nov '13

Has anyone experienced problems with signalling using LCR/Asterisk? In my infrastructure everything goes ok calling from MT to external phones using Asterisk, but when the external one (SIP or extension ended with Hangup() ) the call remains activated at the MT (OpenBSC side). I will make a SIP capture tomorrow, and send it. -- -- Leonardo Nve <lnve(a)s21sec.com> Project Manager ACSS Grupo S21sec Gestión, S.A. Telefono 628275870 -- La información contenida en este mail, así como los archivos adjuntos,es CONFIDENCIAL. Grupo S21sec Gestión, S.A. garantiza la adopción de las medidas necesarias para asegurar el tratamiento confidencial de los datos de carácter personal. En el caso de que el destinatario del correo no sea usted, le rogamos envíe una notificación al remitente y lo destruya de forma inmediata. La lectura y/o manipulación de esta información en la situación señalada anteriormente será considerada ilegal, permitiendo a la empresa remitente realizar acciones legales de diferente envergadura.

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[PATCH 1/3] vty: Use new vty_install_default() function
by Jacob Erlbeck 11 Nov '13

11 Nov '13

This patch removes the local 'end' and 'exit' implementations and uses the generic ones provided by libosmocore instead, which are enabled automatically when vty_install_default() is used. --- src/e1_input_vty.c | 2 ++ src/ipa_proxy.c | 26 +------------------------- 2 files changed, 3 insertions(+), 25 deletions(-) diff --git a/src/e1_input_vty.c b/src/e1_input_vty.c index b6e1438..faaebd8 100644 --- a/src/e1_input_vty.c +++ b/src/e1_input_vty.c @@ -276,6 +276,8 @@ int e1inp_vty_init(void) { install_element(CONFIG_NODE, &cfg_e1inp_cmd); install_node(&e1inp_node, e1inp_config_write); + + vty_install_default(L_E1INP_NODE); install_element(L_E1INP_NODE, &cfg_e1_line_driver_cmd); install_element(L_E1INP_NODE, &cfg_e1_line_port_cmd); install_element(L_E1INP_NODE, &cfg_e1_line_name_cmd); diff --git a/src/ipa_proxy.c b/src/ipa_proxy.c index 74cf427..a394b8c 100644 --- a/src/ipa_proxy.c +++ b/src/ipa_proxy.c @@ -644,28 +644,6 @@ static int ipa_cfg_write(struct vty *vty) return CMD_SUCCESS; } -DEFUN(ournode_exit, - ournode_exit_cmd, "exit", "Exit current mode and down to previous mode\n") -{ - switch (vty->node) { - case L_IPA_NODE: - vty->node = CONFIG_NODE; - vty->index = NULL; - break; - } - return CMD_SUCCESS; -} - -DEFUN(ournode_end, - ournode_end_cmd, "end", "End current mode and change to enable mode.\n") -{ - switch (vty->node) { - case L_IPA_NODE: - break; - } - return CMD_SUCCESS; -} - void ipa_proxy_vty_init(void) { tall_ipa_proxy_ctx = @@ -681,9 +659,7 @@ void ipa_proxy_vty_init(void) install_element(CONFIG_NODE, &ipa_cfg_cmd); install_node(&ipa_node, ipa_cfg_write); - install_default(L_IPA_NODE); - install_element(L_IPA_NODE, &ournode_exit_cmd); - install_element(L_IPA_NODE, &ournode_end_cmd); + vty_install_default(L_IPA_NODE); install_element(L_IPA_NODE, &ipa_instance_cfg_add_cmd); install_element(L_IPA_NODE, &ipa_route_cfg_add_cmd); } -- 1.7.9.5

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[PATCH 1/2] gprs: Fix VTY NSVC initialisation bug by changing gprs_nsvc_create()
by Jacob Erlbeck 11 Nov '13

11 Nov '13

Currently the field nsvci_is_valid is set to 0 in the NSVC object returned by gprs_nsvc_create(). This was a semantic change probably introduced by commit 5e6d679d. As a result, NSVC created via the VTY have this flag set to 0 causing RESET_ACK messages to be rejected. This patch changes the default behaviour of gprs_nsvc_create() to always set this flag. So it must be set to 0 explicitely if needed which is more intuitive and thus less error prone. It fixes breaking connections from the Gbproxy to the SGSN. Ticket: OW#874 Sponsored-by: On-Waves ehf --- src/gb/gprs_ns.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/src/gb/gprs_ns.c b/src/gb/gprs_ns.c index 55535ad..b500d9a 100644 --- a/src/gb/gprs_ns.c +++ b/src/gb/gprs_ns.c @@ -200,6 +200,7 @@ struct gprs_nsvc *gprs_nsvc_create(struct gprs_ns_inst *nsi, uint16_t nsvci) nsvc = talloc_zero(nsi, struct gprs_nsvc); nsvc->nsvci = nsvci; + nsvc->nsvci_is_valid = 1; /* before RESET procedure: BLOCKED and DEAD */ nsvc->state = NSE_S_BLOCKED; nsvc->nsi = nsi; @@ -794,7 +795,6 @@ static int gprs_ns_rx_reset(struct gprs_nsvc **nsvc, struct msgb *msg) gprs_ns_ll_str(*nsvc)); orig_nsvc = *nsvc; *nsvc = gprs_nsvc_create((*nsvc)->nsi, nsvci); - (*nsvc)->nsvci_is_valid = 1; (*nsvc)->nsei = nsei; } } @@ -1193,6 +1193,7 @@ int gprs_ns_vc_create(struct gprs_ns_inst *nsi, struct msgb *msg, existing_nsvc = gprs_nsvc_by_nsvci(nsi, nsvci); if (!existing_nsvc) { *new_nsvc = gprs_nsvc_create(nsi, 0xffff); + (*new_nsvc)->nsvci_is_valid = 0; log_set_context(GPRS_CTX_NSVC, *new_nsvc); gprs_ns_ll_copy(*new_nsvc, fallback_nsvc); LOGP(DNS, LOGL_INFO, "Creating NS-VC for BSS at %s\n", @@ -1327,6 +1328,7 @@ struct gprs_ns_inst *gprs_ns_instantiate(gprs_ns_cb_t *cb, void *ctx) /* Create the dummy NSVC that we use for sending * messages to non-existant/unknown NS-VC's */ nsi->unknown_nsvc = gprs_nsvc_create(nsi, 0xfffe); + nsi->unknown_nsvc->nsvci_is_valid = 0; llist_del(&nsi->unknown_nsvc->list); return nsi; @@ -1527,8 +1529,6 @@ struct gprs_nsvc *gprs_ns_nsip_connect(struct gprs_ns_inst *nsi, nsvc = gprs_nsvc_create(nsi, nsvci); nsvc->ip.bts_addr = *dest; nsvc->nsei = nsei; - nsvc->nsvci = nsvci; - nsvc->nsvci_is_valid = 1; nsvc->remote_end_is_sgsn = 1; gprs_nsvc_reset(nsvc, NS_CAUSE_OM_INTERVENTION); -- 1.7.9.5

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OPENBSC/LCR/Asterisk
by Leonardo Nve 10 Nov '13

10 Nov '13

Hi, I have installed an infrastructure IPACCESS - OpenBSC - LCR (without misdn) - Asterisk LCR si connected via SIP to Asterisk. The problem is that i can call MT <-> softphone, soft and MT <-> MT BUT i don't hear anything in any side ( softphone <-> softphone works well). I think is a codec problem: Configured TCH/F FR for MT and we tried different codecs (alaw,gsm,ulaw, etc etc). Also I tried bridging GSM and SIP interfaces on LCR config and putting rtp-bridge. On LCR debug I see this error: 000000 DEBUG (in port.cpp/new_state() line 267): PORT(SIP-0-out) new state PORT_STATE_OUT_ALERTING --> PORT_STATE_CONNECT nua: nua_application_event: entering 000000 DEBUG (in sip.cpp/sip_callback() line 1775): Event 7 from stack received (handle=0x94906d8) 000000 DEBUG (in sip.cpp/sip_callback() line 1825): state change received nua: nua_application_event: entering 000000 DEBUG (in sip.cpp/sip_callback() line 1775): Event 7 from stack received (handle=0x94906d8) 000000 DEBUG (in sip.cpp/sip_callback() line 1825): state change received nua: nua_application_event: entering 000000 DEBUG (in sip.cpp/sip_callback() line 1775): Event 5 from stack received (handle=0x94906d8) 000000 DEBUG (in sip.cpp/sip_callback() line 1837): active received 000000 TRACE 06.11.13 20:10:06.434 EP(2): CONNECT from CH(2) connect id number= present='not available' 000000 TRACE 06.11.13 20:10:06.435 EP(1): CONNECT to CH(1) connect id number= present='not available' 000000 TRACE 06.11.13 20:10:06.435 EP(1): TONE to CH(1) off 000000 TRACE 06.11.13 20:10:06.436 CH(1): MNCC_SETUP_RSP LCR<->BSC connected type=0 plan=1 present=0 screen=3 number= 000000 DEBUG (in port.cpp/new_state() line 267): PORT(GSM-0-in) new state PORT_STATE_IN_ALERTING --> PORT_STATE_CONNECT_WAITING 000000 DEBUG (in port.cpp/message_epoint() line 617): PORT(GSM-0-in) setting tone '' dir '' 000000 DEBUG (in gsm.cpp/audio_send() line 487): no valid CMR yet. 000000 DEBUG (in gsm.cpp/audio_send() line 487): no valid CMR yet. 000000 DEBUG (in gsm.cpp/audio_send() line 487): no valid CMR yet. 000000 DEBUG (in gsm.cpp/audio_send() line 487): no valid CMR yet. 000000 DEBUG (in gsm.cpp/audio_send() line 487): no valid CMR yet. 000000 DEBUG (in gsm.cpp/audio_send() line 487): no valid CMR yet. 000000 TRACE 06.11.13 20:10:06.563 CH(1): MNCC_SETUP_COMPL_IND LCR<->BSC 000000 DEBUG (in port.cpp/new_state() line 267): PORT(GSM-0-in) new state PORT_STATE_CONNECT_WAITING --> PORT_STATE_CONNECT 000000 DEBUG (in gsm.cpp/audio_send() line 487): no valid CMR yet. 000000 DEBUG (in gsm.cpp/audio_send() line 487): no valid CMR yet. 000000 DEBUG (in gsm.cpp/audio_send() line 487): no valid CMR yet. 000000 DEBUG (in gsm.cpp/audio_send() line 487): no valid CMR yet. 000000 DEBUG (in gsm.cpp/audio_send() line 487): no valid CMR yet. 000000 DEBUG (in gsm.cpp/audio_send() line 487): no valid CMR yet. <--...--> <-- THIS ERROR REPEATED CONTINUOUSLY DURING THE CALL --> <--...--> Configurations: OpenBSC trx 0 config: trx 0 rf_locked 0 arfcn 636 nominal power 23 max_power_red 0 rsl e1 tei 0 timeslot 0 phys_chan_config CCCH+SDCCH4 hopping enabled 0 timeslot 1 phys_chan_config SDCCH8 hopping enabled 0 timeslot 2 phys_chan_config TCH/F hopping enabled 0 timeslot 3 phys_chan_config TCH/F hopping enabled 0 timeslot 4 phys_chan_config TCH/F hopping enabled 0 timeslot 5 phys_chan_config TCH/F hopping enabled 0 timeslot 6 phys_chan_config TCH/F hopping enabled 0 timeslot 7 phys_chan_config TCH/F hopping enabled 0 mncc-int default-codec tch-f fr LCR config: interfaces.conf [GSM] gsm-bs tones yes earlyb no [SIP] extern sip localhost:5059 localhost:5060 tones yes earlyb yes Asterisk User conf: user.conf (one user) [6001] fullname = SIPPhone2 registersip = no host = dynamic callgroup = 1 mailbox = 6001 call-limit = 100 type = peer username = 6001 secret = nomypasshere transfer = yes nat = yes context = openBSC_Integration dtmfmode = rfc2833 cid_number = 6001 disallow = all allow = alaw,gsm ; I Tried different codecs callcounter = no hasvoicemail = no vmsecret = email = threewaycalling = no hasdirectory = no callwaiting = no hasmanager = no hasagent = no hassip = yes hasiax = no canreinvite = no insecure = no pickupgroup = 1 autoprov = yes label = 6001 linenumber = 1 LINEKEYS = 1 Other configs seem irrelevant... -- -- Leonardo Nve <lnve(a)s21sec.com> Project Manager ACSS Grupo S21sec Gestión, S.A. Telefono 628275870 -- La información contenida en este mail, así como los archivos adjuntos,es CONFIDENCIAL. Grupo S21sec Gestión, S.A. garantiza la adopción de las medidas necesarias para asegurar el tratamiento confidencial de los datos de carácter personal. En el caso de que el destinatario del correo no sea usted, le rogamos envíe una notificación al remitente y lo destruya de forma inmediata. La lectura y/o manipulación de esta información en la situación señalada anteriormente será considerada ilegal, permitiendo a la empresa remitente realizar acciones legales de diferente envergadura.

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Openbsc support for new BTS
by Jason DSouza 07 Nov '13

07 Nov '13

Hello Harald and everyone out there! I have been going through your Osmocom projects especially Openbsc (osmo-nitb) and Osmobts projects and I must say it's a real great work! I'm a lead engineer at Octasic and after looking at your work on OpenBSC and OsmoBTS, we decided to use your OpenBSC stack for internal testing of our Octasic PHY. I know currently openbsc supports few BTS models like ip.access, sysmocom etc. I have gone through the wiki and some of the openbsc archives. I could find an archive regarding Octasic SDR, but I guess it has not gone any further. http://lists.osmocom.org/pipermail/openbsc/2010-November/002196.html I will be interested in osmo-nitb which I can use for network part of GSM and use osmo-bts for BTS L2/L3 stuff, which already has abis-over-ip interface to openBSC(osmo-nitb). I have Octasic SDR hardware that has the layer1 (DSP image), so all I need to implement is the interfaces between Octasic Layer1 and osmobts L2/L3 to communicate between them, which I'm working on. I have already installed openbsc (osmo-nitb) and also osmobts on my Debian 7.1.0. I'm currently in a process of writing interfaces between Octasic L1 and osmobts. I'm using the default config file openbsc.cfg for osmo-nitb & using ip.access config file for osmobts (with changes in the OML ip & band GSM900) At present I'm trying to get TRX config request from osmo-nitb. However though I'm receiving few OML messages like NM_MT_SET_BTS_ATTR (pcap attached), I guess they are specific to ip.access BTS, I'm not able to see any TRX config request coming from osmo-nitb (Or by any chance is that what I'm receiving is the TRX config one?) I'm simply sending ack for those OML messages and eventually expecting a TRX config request from osmo-nitb. I'm not sure if osmo-nitb expects something to se sent from BTS side, looks like it is simply waiting for something(see below). I have tried to run osmo-nitb program and it look like: ./osmo-nitb -c openbsc.cfg <0019> input/ipaccess.c:934 enabling ipaccess BSC mode DB: Database initialized. DB: Database prepared. <001d> sms_queue.c:220 Attempting to send 20 SMS <0019> input/ipa.c:323 accept()ed new link from 127.0.0.1 to port 3002 (I do not see RSL link becoming up here) I tried turn on debug: ./osmo-nitb -c openbsc.cfg -C --debug=DRLL:DCC:DMM:DRR:DRSL:DNM DB: Database initialized. DB: Database prepared. <0005> abis_nm.c:315 OC=SITE-MANAGER(00) INST=(ff,ff,ff) STATE CHG: OP_STATE=NULL AVAIL=Power off(02) <0005> abis_nm.c:315 OC=BTS(01) INST=(00,ff,ff) STATE CHG: OP_STATE=NULL AVAIL=Dependency(05) <0005> abis_nm.c:1442 Set BTS Attr (bts=0) <0005> abis_nm.c:1763 OC=BTS(01) INST=(00,ff,ff) Sending OPSTART <0005> abis_nm.c:315 OC=GPRS-NSE(f0) INST=(00,ff,ff) STATE CHG: OP_STATE=NULL AVAIL=Dependency(05) <0005> abis_nm.c:315 OC=GPRS-CELL(f1) INST=(00,ff,ff) STATE CHG: OP_STATE=NULL AVAIL=Dependency(05) <0005> abis_nm.c:315 OC=GPRS-NSVC(f2) INST=(00,00,ff) STATE CHG: OP_STATE=NULL AVAIL=Dependency(05) <0005> abis_nm.c:315 OC=GPRS-NSVC(f2) INST=(00,01,ff) STATE CHG: OP_STATE=NULL AVAIL=Dependency(05) <0005> abis_nm.c:315 OC=RADIO-CARRIER(02) INST=(00,00,ff) STATE CHG: OP_STATE=NULL AVAIL=Power off(02) <0005> abis_nm.c:315 OC=BASEBAND-TRANSCEIVER(04) INST=(00,00,ff) STATE CHG: OP_STATE=NULL AVAIL=Power off(02) <0005> abis_nm.c:315 OC=CHANNEL(03) INST=(00,00,00) STATE CHG: OP_STATE=NULL AVAIL=Power off(02) <0005> abis_nm.c:315 OC=CHANNEL(03) INST=(00,00,01) STATE CHG: OP_STATE=NULL AVAIL=Power off(02) <0005> abis_nm.c:315 OC=CHANNEL(03) INST=(00,00,02) STATE CHG: OP_STATE=NULL AVAIL=Power off(02) <0005> abis_nm.c:315 OC=CHANNEL(03) INST=(00,00,03) STATE CHG: OP_STATE=NULL AVAIL=Power off(02) <0005> abis_nm.c:315 OC=CHANNEL(03) INST=(00,00,04) STATE CHG: OP_STATE=NULL AVAIL=Power off(02) <0005> abis_nm.c:315 OC=CHANNEL(03) INST=(00,00,05) STATE CHG: OP_STATE=NULL AVAIL=Power off(02) <0005> abis_nm.c:315 OC=CHANNEL(03) INST=(00,00,06) STATE CHG: OP_STATE=NULL AVAIL=Power off(02) <0005> abis_nm.c:315 OC=CHANNEL(03) INST=(00,00,07) STATE CHG: OP_STATE=NULL AVAIL=Power off(02) <0005> abis_nm.c:315 OC=BTS(01) INST=(00,ff,ff) STATE CHG: OP_STATE=Enabled AVAIL=Dependency(05) <0005> abis_nm.c:1442 Set BTS Attr (bts=0) <0005> abis_nm.c:1763 OC=BTS(01) INST=(00,ff,ff) Sending OPSTART And then BSC keeps waiting here though at osmobts - osmo-nitb interface IPACCESS ping-pong goes on for keep-alive of the abis OML link and after some time, Osmobts reports link down(see attached osmo-bts debug). At present, I have just created socket interface from osmo-bts to octasic L1 and I'm in the process of writing interfaces for the messages. I'm expecting 1st message, TRX config request from osmo-nitb and do not see it. I was thinking, once the abis interface is up (OML link), osmo-nitb will trigger TRX configuration and then based on the response from BTS, other configurations. Does osmo-nitb require some kind of handshake from the BTS side to send the TRX configuration? I shall be really grateful to hear advices from the experts out there. I can give you more information if required. Thanks, Jason D'Souza

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New handover algorithm with Nokia BTS
by Sipos Csaba 06 Nov '13

06 Nov '13

I just wanted to ask before I spend a lot of time with it. Does the new HO algorithm also works with non-IP based BTSes like Nokia metro or insite? Or at least in theory it should work? Because the handover is not working with the current algo for Nokia, I want to give the new one a try. Csaba

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[PATCH 1/2] sysmobts: Only set RC state to LOCK if all channels are muted
by Jacob Erlbeck 05 Nov '13

05 Nov '13

Currently only mute_state[0] (refers to ts[0]) is inspected to determine, whether the Radio Carrier's state is set to LOCK. This patch changes this by looking at all channels and using LOCK if (and only if) all channels have been muted successfully. Sponsored-by: On-Waves ehf --- src/osmo-bts-sysmo/oml.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/src/osmo-bts-sysmo/oml.c b/src/osmo-bts-sysmo/oml.c index c87acb6..4c63fd5 100644 --- a/src/osmo-bts-sysmo/oml.c +++ b/src/osmo-bts-sysmo/oml.c @@ -342,9 +342,15 @@ int oml_mo_rf_lock_chg(struct gsm_abis_mo *mo, uint8_t mute_state[8], int success) { if (success) { - /* assume mute_state[i] == mute_state[k] */ + int i; + int is_locked = 1; + + for (i = 0; i < ARRAY_SIZE(mute_state); ++i) + if (!mute_state[i]) + is_locked = 0; + mo->nm_state.administrative = - mute_state[0] ? NM_STATE_LOCKED : NM_STATE_UNLOCKED; + is_locked ? NM_STATE_LOCKED : NM_STATE_UNLOCKED; mo->procedure_pending = 0; return oml_mo_statechg_ack(mo); } else { -- 1.7.9.5

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OpenBSC + LCR + Asterisk = half sided calls in one direction
by Sipos Csaba 05 Nov '13

05 Nov '13

Hi, I am setting up openbsc with LCR and Asterisk to provide external connectivity for the GSM clients, and I encountered and interesting problem. If the GSM phone initiates a voice call towards a SIP phone it works perfectly, the voice goes both ways, the quality is OK, everything is fine. But when the SIP phones initiates a voice call towards the GSM phone, only the SIP phone can hear the voice of the GSM phone, and not the other way around (half sided call). The connection setup works both ways just fine. The GSM and the SIP phone can also call the asterisk test numbers, and even can do echo test just fine. The two GSM phones can call each other too without any problem. Does somebody has any idea what could go wrong? Config: E1 based Nokia BTS with DAHDI card and TCH/F FR codec. The SIP phone uses Alaw, LCR also set to use Alaw. The LCR is bridged to Asterisk, interface.conf: [GSM] gsm-bs tones yes earlyb no bridge ast [ast] remote asterisk context from-lcr earlyb no tones yes bridge GSM routing.conf is completely empty. Astrisk SIP.conf: [general] context=incoming disallow=all allow=alaw allow=ulaw allow=gsm transport=udp udpbindaddr=0.0.0.0 tcpenable=no tcpbindaddr=0.0.0.0 tlsenable=no tlsbindaddr=0.0.0.0 [5010] type=friend username=5010 secret=123456 dtmfmode=rfc2833 callerid="5010" <5010> host=dynamic canreinvite=no context=myphones Asterisk extensions.conf [general] static=yes writeprotect=no clearglobalvars=no [globals] ; Global variables goes here [incoming] ; Nothing should land here yet, but every context should end in ; a Hangup(), so we do that. exten => s,1,Hangup() [myphones] ; When we dial something from the phones we just added in ; sip.conf, Asterisk will look for a matching extension here, ; in this context. ; SIP phone exten => 5010,1,Dial(SIP/5010) exten => 5010,n,Hangup() ; Another SIp phone ;exten => 1001,1,Dial(SIP/1001) ;exten => 1001,n,Hangup() ; GSM phone 1 exten => 12346,1,Dial(LCR/ast/${EXTEN:0},60) exten => 12346,n,Hangup() ; GSM phone 2 exten => 12345,1,Dial(LCR/ast/${EXTEN:0},60) exten => 12345,n,Hangup() ; Testing extension exten => 201,1,Answer() exten => 201,n,Playback(levan_polka.mp3) exten => 201,n,Hangup() ; Echo-test, it is good to test if we have sound in both directions. ; The call is answered exten => 202,1,Answer() ; Welcome message is played exten => 202,n,Playback(dir-welcome) ; Play information about the echo test exten => 202,n,Playback(demo-echotest) ; Do the echo test, end with the # key exten => 202,n,Echo() ; Plays information that the echo test is done exten => 202,n,Playback(demo-echodone) ; Goodbye message is played exten => 202,n,Playback(vm-goodbye) ; Hangup() ends the call, hangs up the line exten => 202,n,Hangup() [from-lcr] include => myphones exten => _X.,1,Dial(LCR/ast/${EXTEN:0},60) exten => 5010,1,Dial(SIP/5010) exten => 5010,n,Hangup() BR, Csaba

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fix subscriber not seen as authorized
by Kevin Redon 05 Nov '13

05 Nov '13

here a small patch for a bug which prevent phones to register. more info is available in the patch note: the same mistake may exist at other places. I only looked at how the users are read kevin

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