in your wanpipe.conf file change TE_SIG_MODE = CCS to
TE_SIG_MODE = CAS Saludos/Regards - Gerardo Barajas ClearlyIP www.clearlyip.com On Tue, Mar 8, 2022 at 3:43 PM Duncan Turnbull <[email protected]> wrote: > Hi Carlos > > On Wed, Mar 9, 2022 at 10:30 AM Carlos Chavez <[email protected]> wrote: > >> The provider is the timing source. Both wanpipe1.conf and >> system.conf have the timing sources set to the remote side: >> >> TE_CLOCK = NORMAL >> >> Makes sense, I couldn't recall the options but this looks right > >> >> span=1,1,0,CAS,HDB3 >> >> I still have a feeling that the problem is on the providers side as >> during testing we never saw the issue. >> >> I have modified wanpipe1.conf to be CAS but the strange thing is >> that the freepbx gui does show CAS there but sets CCS on the >> configuration file. Now I have to wait and see if the problem persists. >> > > Technically CCS is usually for ISDN and wasn't always on timeslot 16, but > if it was working then perhaps it was good luck. How freepbx sets it is > another question though > > I am not sure what would go wrong on a provider side as they usually > standardise their systems. That said its always possible. > > Your error is a timeout in response to a line seize so either your > provider isn't seeing the signal, they aren't replying for some reason or > you aren't getting it back. That could fit with changes to the signalling > channel. Ideally if you can look at the signalling you can see whats > happening. I can't recall if asterisk will let you do that. CAS signalling > is very simple in that its just reflecting what used to be a physical > change for the line controls. Can you ask your providers to see what they > see or reset the trunk when the issue comes up to see if it matters > > Good luck > > >> On 08/03/22 11:54, Duncan Turnbull wrote: >> > It’s been a r we hike since we used these cards. This example may help >> > >> > >> https://wiki.freepbx.org/plugins/servlet/mobile?contentId=73007457#content/view/73007457 >> > >> > My thinking is it sounds like a timing error. Make sure your provider >> > is the timing source. Once it loses time you will get dropped calls >> > until it resyncs >> > >> > Good luck >> > >> > >> > >> >> On 9/03/2022, at 4:25 AM, Steinwendtner <[email protected]> wrote: >> >> >> >> Hello, >> >> >> >> I must admit that I have never set up an asterisk system with R2 >> >> signalling. But from the config files >> >> >> >> point of view, you stated TE_SIG_MODE in wanpipe1.conf as ccs which >> >> should be cas, right ? >> >> >> >> If this does not help, you need to connect an external E1 Monitor. >> >> >> >> Regards, >> >> >> >> Hans >> >> >> >> Am 08.03.22 um 06:41 schrieb Carlos Chavez: >> >>> Last month we switched a Panasonic pbx with a Freepbx 16 >> >>> appliance. We use a single E1 in MFC/R2 (Mexico) with Telmex as a >> >>> provider. This was connected for a couple of days for testing with no >> >>> problems before the client moved offices to a new location. In the >> new >> >>> location we are now having a problem every few days where we get the >> >>> following error: >> >>> >> >>> [2022-03-07 07:30:11] ERROR[3469][C-0000004c] chan_dahdi.c: Chan 10 - >> >>> Protocol error. Reason = Seize Timeout, R2 State = Seize Transmitted, >> MF >> >>> state = MF Engine Off, MF Group = Forward MF init, CAS = 0x08 >> >>> [2022-03-07 07:30:44] ERROR[29573] chan_dahdi.c: Chan 10 - Protocol >> >>> error. Reason = Seize Timeout, R2 State = Clear Forward Transmitted, >> MF >> >>> state = MF Engine Off, MF Group = Forward MF init, CAS = 0x08 >> >>> [2022-03-07 07:32:15] ERROR[3704][C-0000004e] chan_dahdi.c: Chan 10 - >> >>> Protocol error. Reason = Seize Timeout, R2 State = Seize Transmitted, >> MF >> >>> state = MF Engine Off, MF Group = Forward MF init, CAS = 0x08 >> >>> [2022-03-07 07:32:52] ERROR[29573] chan_dahdi.c: Chan 10 - Protocol >> >>> error. Reason = Seize Timeout, R2 State = Clear Forward Transmitted, >> MF >> >>> state = MF Engine Off, MF Group = Forward MF init, CAS = 0x08 >> >>> >> >>> When we see that error the E1 will no longer send or receive >> >>> calls. Our solution has been to stop and restart Asterisk and >> >>> Wanconfig/Dahdi to restore service. Since restarting solves it I am >> >>> wondering if the problem is on my side and not on the providers. So >> far >> >>> it happens once or twice a week. When we report this to the provider >> >>> they simply state that the problem is on our side (it is their default >> >>> position) unless we can provide evidence to the contrary. Any >> >>> recommendations on how to debug this? >> >>> >> >>> Here is wanpipe1.conf: >> >>> [devices] >> >>> wanpipe1 = WAN_AFT_TE1, Comment >> >>> >> >>> [interfaces] >> >>> w1g1 = wanpipe1, , TDM_VOICE, Comment >> >>> >> >>> [wanpipe1] >> >>> CARD_TYPE = AFT >> >>> S514CPU = A >> >>> CommPort = PRI >> >>> AUTO_PCISLOT = NO >> >>> PCISLOT = 4 >> >>> PCIBUS = 8 >> >>> FE_MEDIA = E1 >> >>> FE_LCODE = HDB3 >> >>> FE_FRAME = NCRC4 >> >>> FE_LINE = 1 >> >>> TE_CLOCK = NORMAL >> >>> TE_REF_CLOCK = 0 >> >>> TE_SIG_MODE = CCS >> >>> TE_HIGHIMPEDANCE = NO >> >>> TE_RX_SLEVEL = 430 >> >>> HW_RJ45_PORT_MAP = DEFAULT >> >>> LBO = 120OH >> >>> FE_TXTRISTATE = NO >> >>> MTU = 1500 >> >>> UDPPORT = 9000 >> >>> TTL = 255 >> >>> IGNORE_FRONT_END = NO >> >>> TDMV_SPAN = 1 >> >>> TDMV_DCHAN = 16 >> >>> TE_AIS_MAINTENANCE = NO #NO: defualt YES: Start port in AIS >> >>> Blue Alarm and keep line down >> >>> #wanpipemon -i w1g1 -c Ttx_ais_off to >> >>> disable AIS maintenance mode >> >>> #wanpipemon -i w1g1 -c Ttx_ais_on to >> >>> enable AIS maintenance mode >> >>> TDMV_HW_DTMF = NO # YES: receive dtmf events from hardware >> >>> TDMV_HW_FAX_DETECT = NO # YES: receive fax 1100hz events >> >>> from hardware >> >>> HWEC_OPERATION_MODE = OCT_NORMAL # OCT_NORMAL: echo cancelation >> >>> enabled with nlp (default) >> >>> # OCT_SPEECH: improves software >> >>> tone detection by disabling NLP (echo possible) >> >>> # OCT_NO_ECHO:disables echo >> >>> cancelation but allows VQE/tone functions. >> >>> HWEC_DTMF_REMOVAL = NO # NO: default YES: remove dtmf out of >> >>> incoming media (must have hwdtmf enabled) >> >>> HWEC_NOISE_REDUCTION = NO # NO: default YES: reduces noise on the >> >>> line - could break fax >> >>> HWEC_ACUSTIC_ECHO = NO # NO: default YES: enables acustic echo >> >>> cancelation >> >>> HWEC_NLP_DISABLE = NO # NO: default YES: guarantees software >> >>> tone detection (possible echo) >> >>> HWEC_TX_AUTO_GAIN = 0 # 0: disable -40-0: default tx audio >> >>> level to be maintained (-20 default) >> >>> HWEC_RX_AUTO_GAIN = 0 # 0: disable -40-0: default tx audio >> >>> level to be maintained (-20 default) >> >>> HWEC_TX_GAIN = 0 # 0: disable -24-24: db values to >> >>> be applied to tx signal >> >>> HWEC_RX_GAIN = 0 # 0: disable -24-24: db values to >> >>> be applied to tx signal >> >>> >> >>> [w1g1] >> >>> ACTIVE_CH = ALL >> >>> TDMV_HWEC = NO >> >>> MTU = 8 >> >>> >> >>> Here is system.conf >> >>> >> >>> span=1,1,0,CAS,HDB3 >> >>> cas=1-10,11-15,17-31:1101 >> >>> echocanceller=oslec,1-10,11-15,17-31 >> >>> loadzone=mx >> >>> defaultzone=mx >> >>> >> >>> Here is chan_dahdi.conf >> >>> >> >>> signalling=mfcr2 >> >>> mfcr2_variant=mx >> >>> mfcr2_get_ani_first=no >> >>> mfcr2_max_ani=10 >> >>> mfcr2_max_dnis=4 >> >>> mfcr2_category=national_priority_subscriber >> >>> mfcr2_call_files=no >> >>> mfcr2_mfback_timeout=-1 >> >>> mfcr2_metering_pulse_timeout=-1 >> >>> mfcr2_allow_collect_calls=yes >> >>> mfcr2_double_answer=no >> >>> mfcr2_immediate_accept=no >> >>> mfcr2_accept_on_offer=yes >> >>> mfcr2_skip_category=no >> >>> mfcr2_forced_release=no >> >>> mfcr2_charge_calls=yes >> >>> group=0 >> >>> context=from-digital >> >>> channel=>1-10 >> >>> >> >> >> >> -- >> >> _____________________________________________________________________ >> >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> >> >> Check out the new Asterisk community forum at: >> >> https://community.asterisk.org/ >> >> >> >> New to Asterisk? Start here: >> >> https://wiki.asterisk.org/wiki/display/AST/Getting+Started >> >> >> >> asterisk-users mailing list >> >> To UNSUBSCRIBE or update options visit: >> >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > >> -- >> Telecomunicaciones Abiertas de México S.A. de C.V. >> Carlos Chávez >> +52 (55)8116-9161 >> >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> Check out the new Asterisk community forum at: >> https://community.asterisk.org/ >> >> New to Asterisk? Start here: >> https://wiki.asterisk.org/wiki/display/AST/Getting+Started >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
