[asterisk-users] R2 error Seize Timeout
Duncan Turnbull
duncan at turnbull.co.nz
Tue Mar 8 15:43:40 CST 2022
Hi Carlos
On Wed, Mar 9, 2022 at 10:30 AM Carlos Chavez <cursor at telecomab.mx> 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 <steinwendtner at gmx.net> 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
> >>
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> >
> --
> 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
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