Re: [asterisk-users] R2 error Seize Timeout

[Gerardo Barajas]

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 
wrote:

> Hi Carlos
>
> On Wed, Mar 9, 2022 at 10:30 AM Carlos Chavez  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  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-004c] 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-004e] 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_

Re: [asterisk-users] R2 error Seize Timeout

[Duncan Turnbull]

Hi Carlos

On Wed, Mar 9, 2022 at 10:30 AM Carlos Chavez  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  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-004c] 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-004e] 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
> >

Re: [asterisk-users] R2 error Seize Timeout

[Carlos Chavez]

    The provider is the timing source.  Both wanpipe1.conf and 
system.conf have the timing sources set to the remote side:


TE_CLOCK     = NORMAL


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.


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  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-004c] 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-004e] 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  

Re: [asterisk-users] R2 error Seize Timeout

[Duncan Turnbull]

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  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-004c] 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-004e] 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
>> 

Re: [asterisk-users] R2 error Seize Timeout

[Steinwendtner]

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-004c] 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-004e] 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_o

[asterisk-users] R2 error Seize Timeout

[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-004c] 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-004e] 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

--
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

aster

Re: [asterisk-users] R2

[David fire]

thanks moises
and Digium's folks put it asap please not until 1.6.3
thanks

2009/1/15 Moises Silva 

> That's Digium's folks decision. It was said they wanted it for 1.6.3,
> but, that's not for sure, as I said, they will decide.
>
> On Thu, Jan 15, 2009 at 11:54 AM, David fire  wrote:
> > thanks for the answer.
> > any idea in wich version it will be merged?
> > thanks
> >
> > 2009/1/15 Moises Silva 
> >>
> >> Is in the process of being merged.
> >>
> >> http://bugs.digium.com/view.php?id=12509
> >> http://reviewboard.digium.com/r/40/
> >> http://www.libopenr2.org/
> >>
> >> Moisés Silva
> >>
> >> On Thu, Jan 15, 2009 at 9:44 AM, David fire  wrote:
> >> > hi i am reading about new codecs and new stuff to be added to
> asterisk.
> >> > (and
> >> > i say thanks to all the guys who are working to add all  the new
> >> > features).
> >> >
> >> > will be R2 added to the main core of asterisk like ISDN?
> >> > Thanks
> >> > David
> >> >
> >> > --
> >> > (\__/)
> >> > (='.'=)This is Bunny. Copy and paste bunny into your
> >> > (")_(")signature to help him gain world domination.
> >> >
> >> >
> >> > ___
> >> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >> >
> >> > asterisk-users mailing list
> >> > To UNSUBSCRIBE or update options visit:
> >> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >> >
> >>
> >>
> >>
> >> --
> >> "I do not agree with what you have to say, but I'll defend to the
> >> death your right to say it." Voltaire
> >>
> >> ___
> >> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >>
> >> asterisk-users mailing list
> >> To UNSUBSCRIBE or update options visit:
> >>   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
> > --
> > (\__/)
> > (='.'=)This is Bunny. Copy and paste bunny into your
> > (")_(")signature to help him gain world domination.
> >
> >
> > ___
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
>
> --
> "I do not agree with what you have to say, but I'll defend to the
> death your right to say it." Voltaire
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
(\__/)
(='.'=)This is Bunny. Copy and paste bunny into your
(")_(")signature to help him gain world domination.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] R2

[Moises Silva]

That's Digium's folks decision. It was said they wanted it for 1.6.3,
but, that's not for sure, as I said, they will decide.

On Thu, Jan 15, 2009 at 11:54 AM, David fire  wrote:
> thanks for the answer.
> any idea in wich version it will be merged?
> thanks
>
> 2009/1/15 Moises Silva 
>>
>> Is in the process of being merged.
>>
>> http://bugs.digium.com/view.php?id=12509
>> http://reviewboard.digium.com/r/40/
>> http://www.libopenr2.org/
>>
>> Moisés Silva
>>
>> On Thu, Jan 15, 2009 at 9:44 AM, David fire  wrote:
>> > hi i am reading about new codecs and new stuff to be added to asterisk.
>> > (and
>> > i say thanks to all the guys who are working to add all  the new
>> > features).
>> >
>> > will be R2 added to the main core of asterisk like ISDN?
>> > Thanks
>> > David
>> >
>> > --
>> > (\__/)
>> > (='.'=)This is Bunny. Copy and paste bunny into your
>> > (")_(")signature to help him gain world domination.
>> >
>> >
>> > ___
>> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> >
>> > asterisk-users mailing list
>> > To UNSUBSCRIBE or update options visit:
>> >   http://lists.digium.com/mailman/listinfo/asterisk-users
>> >
>>
>>
>>
>> --
>> "I do not agree with what you have to say, but I'll defend to the
>> death your right to say it." Voltaire
>>
>> ___
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> --
> (\__/)
> (='.'=)This is Bunny. Copy and paste bunny into your
> (")_(")signature to help him gain world domination.
>
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
"I do not agree with what you have to say, but I'll defend to the
death your right to say it." Voltaire

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] R2

[David fire]

thanks for the answer.
any idea in wich version it will be merged?
thanks

2009/1/15 Moises Silva 

> Is in the process of being merged.
>
> http://bugs.digium.com/view.php?id=12509
> http://reviewboard.digium.com/r/40/
> http://www.libopenr2.org/
>
> Moisés Silva
>
> On Thu, Jan 15, 2009 at 9:44 AM, David fire  wrote:
> > hi i am reading about new codecs and new stuff to be added to asterisk.
> (and
> > i say thanks to all the guys who are working to add all  the new
> features).
> >
> > will be R2 added to the main core of asterisk like ISDN?
> > Thanks
> > David
> >
> > --
> > (\__/)
> > (='.'=)This is Bunny. Copy and paste bunny into your
> > (")_(")signature to help him gain world domination.
> >
> >
> > ___
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
>
> --
> "I do not agree with what you have to say, but I'll defend to the
> death your right to say it." Voltaire
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
(\__/)
(='.'=)This is Bunny. Copy and paste bunny into your
(")_(")signature to help him gain world domination.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] R2

[Moises Silva]

Is in the process of being merged.

http://bugs.digium.com/view.php?id=12509
http://reviewboard.digium.com/r/40/
http://www.libopenr2.org/

Moisés Silva

On Thu, Jan 15, 2009 at 9:44 AM, David fire  wrote:
> hi i am reading about new codecs and new stuff to be added to asterisk. (and
> i say thanks to all the guys who are working to add all  the new features).
>
> will be R2 added to the main core of asterisk like ISDN?
> Thanks
> David
>
> --
> (\__/)
> (='.'=)This is Bunny. Copy and paste bunny into your
> (")_(")signature to help him gain world domination.
>
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
"I do not agree with what you have to say, but I'll defend to the
death your right to say it." Voltaire

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] R2

[David fire]

hi i am reading about new codecs and new stuff to be added to asterisk. (and
i say thanks to all the guys who are working to add all  the new features).

will be R2 added to the main core of asterisk like ISDN?
Thanks
David

-- 
(\__/)
(='.'=)This is Bunny. Copy and paste bunny into your
(")_(")signature to help him gain world domination.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] R2 with Alestra in Mexico...

[Moises Silva]

Two weeks from now I will release a chan_unicall driver that allows to
change the calling party category from the dial plan and configuration
file. Keep posted at http://www.moythreads.com/astunicall/

Regards,

Moisés Silva

On Feb 6, 2008 5:07 PM, Jorge Cisneros <[EMAIL PROTECTED]> wrote:
> Yes, i have the same problem with att a few months ago, the problem is the
> acount (abonado) code, att need 2 and the code of unicall send 1, maybe the
> problem is the same for you, please post the debug unicall code.
>
>In this code, you can see the dial number, but if you see, the last digit
> is 1
>
>
>
>
>
> On Feb 6, 2008 12:21 PM, Moises Silva <[EMAIL PROTECTED]> wrote:
>
> > This is great news :)
> >
> >
> >
> >
> > On Feb 6, 2008 10:56 AM, Carlos Chavez <[EMAIL PROTECTED]> wrote:
> > >
> > > On Wed, 2008-02-06 at 08:17 -0600, Moises Silva wrote:
> > > > Carlos, I have some spare time today in case you want me to check it.
> > > >
> > > > Is this your first time with Alestra?
> > > >
> > > Thank you for the offer.
> > >
> > > Yes this is the first time I use Alestra for R2.  I have another
> > > customer that uses them but with PRI and I do have some problems dialing
> > > certain numbers on that link.
> > >
> > > It turns out that there was a problem with their equipment but
> it took
> > > them almost 24 hours for them to admit it.  It is now working properly.
> > > Calls now go in and out and for now I do not see any other problems.
> > >
> > > My list of tested providers for R2 in Mexico is now: Axtel,
> Alestra,
> > > Maxcom and Telmex.
> > >
> > > --
> > > Telecomunicaciones Abiertas de México S.A. de C.V.
> > > Carlos Chávez Prats
> > > Director de Tecnología
> > > +52-55-91169161 ext 2001
> > >
> >
> >
> >
> > > ___
> > > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> > >
> > > asterisk-users mailing list
> > > To UNSUBSCRIBE or update options visit:
> > >http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> >
> >
> >
> > --
> > "I do not agree with what you have to say, but I'll defend to the
> > death your right to say it." Voltaire
> >
> > ___
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
> > 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 --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
"I do not agree with what you have to say, but I'll defend to the
death your right to say it." Voltaire

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] R2 with Alestra in Mexico...

[Sanjoy Rath]

When I dial one extension to the other, I get the call go into a HOLD music 
instead of rining the other extention. Both extensions are SIP Softphone.
 
Following is the Asterisks CLI commandline log
 
-- Executing [EMAIL PROTECTED]:1] Park("SIP/500-08276430", "") in new stack 
   -- Started music on hold, class 'default', on SIP/500-08276430  == Parked 
SIP/500-08276430 on [EMAIL PROTECTED] Will timeout back to extension 
[from-internal] s, 1 in 45 seconds-- Added extension '701' priority 1 to 
parkedcalls  == Spawn extension (from-internal, s, 1) exited KEEPALIVE on 
'SIP/500-08276430'
Any thoughts how to get the call to successfully dial and get both the 
extensions to talk?
 
Thanks,
SR.
_

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] R2 with Alestra in Mexico...

[Jorge Cisneros]

Yes, i have the same problem with att a few months ago, the problem is the
acount (abonado) code, att need 2 and the code of unicall send 1, maybe the
problem is the same for you, please post the debug unicall code.

  In this code, you can see the dial number, but if you see, the last digit
is 1





On Feb 6, 2008 12:21 PM, Moises Silva <[EMAIL PROTECTED]> wrote:

> This is great news :)
>
> On Feb 6, 2008 10:56 AM, Carlos Chavez <[EMAIL PROTECTED]> wrote:
> >
> > On Wed, 2008-02-06 at 08:17 -0600, Moises Silva wrote:
> > > Carlos, I have some spare time today in case you want me to check it.
> > >
> > > Is this your first time with Alestra?
> > >
> > Thank you for the offer.
> >
> > Yes this is the first time I use Alestra for R2.  I have another
> > customer that uses them but with PRI and I do have some problems dialing
> > certain numbers on that link.
> >
> > It turns out that there was a problem with their equipment but
> it took
> > them almost 24 hours for them to admit it.  It is now working properly.
> > Calls now go in and out and for now I do not see any other problems.
> >
> > My list of tested providers for R2 in Mexico is now: Axtel,
> Alestra,
> > Maxcom and Telmex.
> >
> > --
> > Telecomunicaciones Abiertas de México S.A. de C.V.
> > Carlos Chávez Prats
> > Director de Tecnología
> > +52-55-91169161 ext 2001
> >
> > ___
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
>
> --
> "I do not agree with what you have to say, but I'll defend to the
> death your right to say it." Voltaire
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> 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 --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] R2 with Alestra in Mexico...

[Moises Silva]

This is great news :)

On Feb 6, 2008 10:56 AM, Carlos Chavez <[EMAIL PROTECTED]> wrote:
>
> On Wed, 2008-02-06 at 08:17 -0600, Moises Silva wrote:
> > Carlos, I have some spare time today in case you want me to check it.
> >
> > Is this your first time with Alestra?
> >
> Thank you for the offer.
>
> Yes this is the first time I use Alestra for R2.  I have another
> customer that uses them but with PRI and I do have some problems dialing
> certain numbers on that link.
>
> It turns out that there was a problem with their equipment but it took
> them almost 24 hours for them to admit it.  It is now working properly.
> Calls now go in and out and for now I do not see any other problems.
>
> My list of tested providers for R2 in Mexico is now: Axtel, Alestra,
> Maxcom and Telmex.
>
> --
> Telecomunicaciones Abiertas de México S.A. de C.V.
> Carlos Chávez Prats
> Director de Tecnología
> +52-55-91169161 ext 2001
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
"I do not agree with what you have to say, but I'll defend to the
death your right to say it." Voltaire

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] R2 with Alestra in Mexico...

[Carlos Chavez]

On Wed, 2008-02-06 at 08:17 -0600, Moises Silva wrote:
> Carlos, I have some spare time today in case you want me to check it.
> 
> Is this your first time with Alestra?
> 
Thank you for the offer.

Yes this is the first time I use Alestra for R2.  I have another
customer that uses them but with PRI and I do have some problems dialing
certain numbers on that link.

It turns out that there was a problem with their equipment but it took
them almost 24 hours for them to admit it.  It is now working properly.
Calls now go in and out and for now I do not see any other problems.

My list of tested providers for R2 in Mexico is now: Axtel, Alestra,
Maxcom and Telmex.

-- 
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001


signature.asc
Description: This is a digitally signed message part
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] R2 with Alestra in Mexico...

[Moises Silva]

Carlos, I have some spare time today in case you want me to check it.

Is this your first time with Alestra?

On Feb 5, 2008 6:50 PM, Carlos Chavez <[EMAIL PROTECTED]> wrote:
> I am trying to set up Astunicall 1.4.16 with a link from Alestra in
> Mexico City.  I have done everything I usually do for other links in
> Mexico but this one simply will not send or receive calls.  I just get
> Protocol error.
>
> Anyone has any experience with R2 and Alestra?
>
> --
> Telecomunicaciones Abiertas de México S.A. de C.V.
> Carlos Chávez Prats
> Director de Tecnología
> +52-55-91169161 ext 2001
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
"I do not agree with what you have to say, but I'll defend to the
death your right to say it." Voltaire

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] R2 with Alestra in Mexico...

[Luis Antonio Prata Barbosa]

Please,

Give us more information about error.

Are you using astunicall ?


2008/2/5, Carlos Chavez <[EMAIL PROTECTED]>:
>
>I am trying to set up Astunicall 1.4.16 with a link from Alestra in
> Mexico City.  I have done everything I usually do for other links in
> Mexico but this one simply will not send or receive calls.  I just get
> Protocol error.
>
>Anyone has any experience with R2 and Alestra?
>
> --
> Telecomunicaciones Abiertas de México S.A. de C.V.
> Carlos Chávez Prats
> Director de Tecnología
> +52-55-91169161 ext 2001
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> 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 --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] R2 with Alestra in Mexico...

[Carlos Chavez]

I am trying to set up Astunicall 1.4.16 with a link from Alestra in
Mexico City.  I have done everything I usually do for other links in
Mexico but this one simply will not send or receive calls.  I just get
Protocol error.

Anyone has any experience with R2 and Alestra? 

-- 
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001


signature.asc
Description: This is a digitally signed message part
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] R2-Unicall Asterisk as CPE and as CO

[Moises Silva]

> Wouldn't it be better if that could be done in unicall.conf? As with the
> other options like protocolvariant and protocolend ??
Yeah, I can add that too :) ... however being able to get/set the
Calling Party Category is needed from the dial plan as well. So I will
allow both things.

> Anyway.. thanks for doing that update ;) I would be glad to know when it
> is available.
Check the astunicall blog in 2 weeks. You can also subscribe to the
blog and probably you will be notified of updates.

> Well.. in the first place.. that pbx is not mine, I didnt configured it
> and I cant even touch it, Im just putting asterisk in between right now.
Well, then try to figure out how it is configured :) ... in anycase,
disabling callerid usually works, then you can start incrementing the
number of callerid digits until fails.

> Im gonna try that.. Thanks for the help!
Good luck!

Moisés Silva

-- 
"I do not agree with what you have to say, but I'll defend to the
death your right to say it." Voltaire

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] R2-Unicall Asterisk as CPE and as CO

[Victor Toofic]

El Sat, Jan 19 de 2008 a las 23:35 -0600, Moises Silva comentaba:
> First, let me say I am confused about this:
> 
> > I've changed the line (chan_unicall.c):
> >
> > uc_callparm_calling_party_category(callparms,
> > UC_CALLER_CATEGORY_NATIONAL_SUBSCRIBER_CALL);
> >
> > to
> >
> > uc_callparm_calling_party_category(callparms,
> > UC_CALLER_CATEGORY_NATIONAL_PRIORITY_SUBSCRIBER_CALL);
> >
> > because without this I cant receive calls from the telco. With or without 
> > this I
> > can't place calls to the pbx.
> 
> I am quite sure you have made a mistake in this statement, why? simply
> because this code is executed when YOU START the call to the far end
> (whatever it is, Telmex or the other PBX), so it makes no sense to say
> that w/o that change you can't receive calls, no sense at all. I am
> sure you messed up somewhere else in the configuration files just like
> possibly you are doing right now for the PBX.

Hmmm.. I was pretty sure that if I dont change that line I cant receive
calls from telmex.. but let try it againg and I will tell you what it was.

> In anycase, I am about
> to make a new release of chan_unicall Asterisk driver that will
> include a way to modify the calling party category from the dialplan
> extensions.conf

Wouldn't it be better if that could be done in unicall.conf? As with the
other options like protocolvariant and protocolend ??

Anyway.. thanks for doing that update ;) I would be glad to know when it
is available.

> Now, regarding your problem when receiving calls from the pbx, I think
> you have configured the PBX to not send ANI digits, and you configured
> chan_unicall to expect ANI digits, hence the timeout. Try configuring
> Asterisk with 0 callerid for the PBX side, or configure the other PBX
> to send the proper number of ANI digits.

Well.. in the first place.. that pbx is not mine, I didnt configured it
and I cant even touch it, Im just putting asterisk in between right now.

Im gonna try that.. Thanks for the help!

--
Regards..
Victor Toofic

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] R2-Unicall Asterisk as CPE and as CO

[Moises Silva]

Hello Victor.

First, let me say I am confused about this:

> I've changed the line (chan_unicall.c):
>
> uc_callparm_calling_party_category(callparms,
> UC_CALLER_CATEGORY_NATIONAL_SUBSCRIBER_CALL);
>
> to
>
> uc_callparm_calling_party_category(callparms,
> UC_CALLER_CATEGORY_NATIONAL_PRIORITY_SUBSCRIBER_CALL);
>
> because without this I cant receive calls from the telco. With or without 
> this I
> can't place calls to the pbx.

I am quite sure you have made a mistake in this statement, why? simply
because this code is executed when YOU START the call to the far end
(whatever it is, Telmex or the other PBX), so it makes no sense to say
that w/o that change you can't receive calls, no sense at all. I am
sure you messed up somewhere else in the configuration files just like
possibly you are doing right now for the PBX. In anycase, I am about
to make a new release of chan_unicall Asterisk driver that will
include a way to modify the calling party category from the dialplan
extensions.conf

Now, regarding your problem when receiving calls from the pbx, I think
you have configured the PBX to not send ANI digits, and you configured
chan_unicall to expect ANI digits, hence the timeout. Try configuring
Asterisk with 0 callerid for the PBX side, or configure the other PBX
to send the proper number of ANI digits.

Regards,

Moises Silva

On Jan 18, 2008 9:41 AM, Victor Toofic <[EMAIL PROTECTED]> wrote:
> Hi!
>
> Im having some troubles trying to configure * as a bridge between a telco
> and a pbx with R2, the scenario is this:
>
> E1/R2-E1/R2
> |   Telco  |-|   *   |-|   PBX|
> | (Telmex) | - |  |
>    
>
> I can receive calls from the telco and can place calls to the pbx, I also
> can place calls to the telco.. but I can't receive any calls from the pbx.
> When receive a call from the pbx I get this:
>
> cause 32771 - T3 timed out
>
> If I connect the pbx directly to the telco there is no problem, the calls
> are stablished correctly.
>
> Im using the package at:
>
> http://www.moythreads.com/astunicall/downloads/
> 
> http://www.moythreads.com/astunicall/files/astunicall-1.2.25-0.1.tar.gz
>
> that contains:
>
> asterisk-1.2.25
> spandsp-0.0.4
> unicall-0.0.5pre1
> libmfcr2-0.0.3
> libsupertone-0.0.2
> libunicall-0.0.3
> zaptel-1.2.22
>
> My zaptel.conf is this:
>
> loadzone=mx
> defaultzone=mx
> span=1,1,0,cas,hdb3
> span=2,1,0,cas,hdb3
> span=3,0,0,cas,hdb3
> span=4,0,0,cas,hdb3
> cas=1-15:1101
> cas=17-31:1101
> cas=32-46:1101
> cas=48-62:1101
> cas=63-77:1101
> cas=79-93:1101
> cas=94-103:1101
> cas=110-124:1101
>
> and unicall.conf is this:
>
> [channels]
> usecallerid=no
> hidecallerid=no
> callwaitingcallerid=no
> threewaycalling=no
> transfer=no
> cancallforward=no
> callreturn=no
> echocancel=yes
> echocancelwhenbridged=yes
> rxgain=0.0
> txgain=0.0
> immediate=yes
> loglevel=255
> protocolclass=mfcr2
>
> protocolvariant=mx,10,4,16
>
> group=1
> protocolend=cpe
> context=incoming1
> channel => 1-15
> channel => 17-31
>
> group=2
> protocolend=cpe
> context=incoming2
> channel => 32-46
> channel => 48-62
>
> protocolvariant=mx,10,8
>
> group=3
> immediate=yes
> usecallerid=yes
> protocolend=co
> context=incoming3
> channel => 63-77
> channel => 79-93
>
> group=4
> protocolend=co
> context=incoming4
> channel => 94-103
> channel => 110-124
>
> The port #1 of a TE405P card is connected to the telco and the port #3 is
> connected to the pbx.
>
> I've changed the line (chan_unicall.c):
>
> uc_callparm_calling_party_category(callparms,
> UC_CALLER_CATEGORY_NATIONAL_SUBSCRIBER_CALL);
>
> to
>
> uc_callparm_calling_party_category(callparms,
> UC_CALLER_CATEGORY_NATIONAL_PRIORITY_SUBSCRIBER_CALL);
>
> because without this I cant receive calls from the telco. With or without 
> this I
> can't place calls to the pbx.
>
> When I receive a call from the telco I place it directly to the pbx.. and
> that works ok:
>
> Jan 16 12:27:01 DEBUG[4136] chan_unicall.c: MFC/R2 UniCall/2  <- 0001
> [1/IDLE/Idle  /Idle ]
> Jan 16 12:27:01 DEBUG[4136] chan_unicall.c: MFC/R2 UniCall/2 Detected
> Jan 16 12:27:01 DEBUG[4136] chan_unicall.c: MFC/R2 UniCall/2 Creating a
> new call with CRN 32770
> Jan 16 12:27:01 DEBUG[4136] chan_u

[asterisk-users] R2-Unicall Asterisk as CPE and as CO

[Victor Toofic]

Hi!

Im having some troubles trying to configure * as a bridge between a telco
and a pbx with R2, the scenario is this:

E1/R2-E1/R2
|   Telco  |-|   *   |-|   PBX|
| (Telmex) | - |  |
   

I can receive calls from the telco and can place calls to the pbx, I also
can place calls to the telco.. but I can't receive any calls from the pbx.
When receive a call from the pbx I get this:

cause 32771 - T3 timed out

If I connect the pbx directly to the telco there is no problem, the calls
are stablished correctly.

Im using the package at:

http://www.moythreads.com/astunicall/downloads/
http://www.moythreads.com/astunicall/files/astunicall-1.2.25-0.1.tar.gz

that contains:

asterisk-1.2.25
spandsp-0.0.4
unicall-0.0.5pre1
libmfcr2-0.0.3
libsupertone-0.0.2
libunicall-0.0.3
zaptel-1.2.22

My zaptel.conf is this:

loadzone=mx
defaultzone=mx
span=1,1,0,cas,hdb3
span=2,1,0,cas,hdb3
span=3,0,0,cas,hdb3
span=4,0,0,cas,hdb3
cas=1-15:1101
cas=17-31:1101
cas=32-46:1101
cas=48-62:1101
cas=63-77:1101
cas=79-93:1101
cas=94-103:1101
cas=110-124:1101

and unicall.conf is this:

[channels]
usecallerid=no
hidecallerid=no
callwaitingcallerid=no
threewaycalling=no
transfer=no
cancallforward=no
callreturn=no
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
immediate=yes
loglevel=255
protocolclass=mfcr2

protocolvariant=mx,10,4,16

group=1
protocolend=cpe
context=incoming1
channel => 1-15
channel => 17-31

group=2
protocolend=cpe
context=incoming2
channel => 32-46
channel => 48-62

protocolvariant=mx,10,8

group=3
immediate=yes
usecallerid=yes
protocolend=co
context=incoming3
channel => 63-77
channel => 79-93

group=4
protocolend=co
context=incoming4
channel => 94-103
channel => 110-124

The port #1 of a TE405P card is connected to the telco and the port #3 is
connected to the pbx.

I've changed the line (chan_unicall.c):

uc_callparm_calling_party_category(callparms,
UC_CALLER_CATEGORY_NATIONAL_SUBSCRIBER_CALL);

to

uc_callparm_calling_party_category(callparms,
UC_CALLER_CATEGORY_NATIONAL_PRIORITY_SUBSCRIBER_CALL);

because without this I cant receive calls from the telco. With or without this I
can't place calls to the pbx.

When I receive a call from the telco I place it directly to the pbx.. and
that works ok:

Jan 16 12:27:01 DEBUG[4136] chan_unicall.c: MFC/R2 UniCall/2  <- 0001
[1/IDLE/Idle  /Idle ]
Jan 16 12:27:01 DEBUG[4136] chan_unicall.c: MFC/R2 UniCall/2 Detected
Jan 16 12:27:01 DEBUG[4136] chan_unicall.c: MFC/R2 UniCall/2 Creating a
new call with CRN 32770
Jan 16 12:27:01 DEBUG[4136] chan_unicall.c: MFC/R2 UniCall/2 1101  ->
[2/DETECTED/Seize ack /Seize ack]
Jan 16 12:27:01 NOTICE[4136] chan_unicall.c: Unicall/2 event Detected
Jan 16 12:27:01 DEBUG[4136] chan_unicall.c: MFC/R2 UniCall/2  <- 4 on
[2/DETECTED/Seize ack /Seize ack]
Jan 16 12:27:01 DEBUG[4136] chan_unicall.c: MFC/R2 UniCall/2 6 on  ->
[2/DETECTED/Group C   /Category req ]
Jan 16 12:27:01 DEBUG[4136] chan_unicall.c: MFC/R2 UniCall/2  <- 4 off
[2/DETECTED/Group C   /Category req ]
Jan 16 12:27:01 DEBUG[4136] chan_unicall.c: MFC/R2 UniCall/2 6 off ->
[2/DETECTED/Group C   /Category req ]
Jan 16 12:27:01 DEBUG[4136] chan_unicall.c: MFC/R2 UniCall/2  <- 2 on
[2/DETECTED/Group C   /Category req ]
Jan 16 12:27:01 DEBUG[4136] chan_unicall.c: MFC/R2 UniCall/2 1 on  ->
[2/DETECTED/Group C   /ANI request  ]
Jan 16 12:27:01 DEBUG[4136] chan_unicall.c: MFC/R2 UniCall/2  <- 2 off
[2/DETECTED/Group C   /ANI request  ]
Jan 16 12:27:01 DEBUG[4136] chan_unicall.c: MFC/R2 UniCall/2 1 off ->
[2/DETECTED/Group C   /ANI request  ]
Jan 16 12:27:01 DEBUG[4136] chan_unicall.c: MFC/R2 UniCall/2  <- F on
[2/DETECTED/Group C   /ANI request  ]
Jan 16 12:27:01 DEBUG[4136] chan_unicall.c: MFC/R2 UniCall/2 5 on  ->
[2/DETECTED/Group A   /DNIS request ]
Jan 16 12:27:01 DEBUG[4136] chan_unicall.c: MFC/R2 UniCall/2  <- F off
[2/DETECTED/Group A   /DNIS request ]
Jan 16 12:27:01 DEBUG[4136] chan_unicall.c: MFC/R2 UniCall/2 5 off ->
[2/DETECTED/Group A   /DNIS request ]
Jan 16 12:27:01 DEBUG[4136] chan_unicall.c: MFC/R2 UniCall/2  <- 6 on
[2/DETECTED/Group A   /DNIS requ

Re: [asterisk-users] r2 multiframe error - solved

[Steve Underwood]

Jakub Syrek wrote:
> All errors was genereted by physical link.
> Protocolvariant cz,10,6 its ok for me in Poland
> Thanks for help
>
> Regards
> Akron
>   
Thanks. I will make a note of that in the code.

Steve


___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] r2 multiframe error - solved

[Moises Silva]

Good news.

On Nov 20, 2007 7:51 AM, Jakub Syrek <[EMAIL PROTECTED]> wrote:
> All errors was genereted by physical link.
> Protocolvariant cz,10,6 its ok for me in Poland
> Thanks for help
>
> Regards
> Akron
>
> ___
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
"Within C++, there is a much smaller and cleaner language struggling
to get out."

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] r2 multiframe error - solved

[Jakub Syrek]

All errors was genereted by physical link.
Protocolvariant cz,10,6 its ok for me in Poland
Thanks for help

Regards
Akron

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] r2 multiframe error - continue

[Jakub Syrek]

Ok here is some more info.
Currently Elastix 0.9.0 installed and nothing more changed because i dont 
want to create confusion
I'm in Poland, my teleco is Telekomunikacja Polska (TP) and they are using 
Siemens EWSD on my link.
Cas, hdb3, crc4 mfcr2 are in use on link.
My card is from  http://www.phoniceq.com/   TE210P
In my location I allso have Schmid Watson 5 so link from tele looks like 
this:
Siemens EWSD  ---> Schmid Watson 5 ---> TE210P

On their side they are getting a DISA (distance service alarm?) error.

in zttool i get
   TxA 111 111 â  â 
â
   â âTxB 000 000 â  â 
â
   â âTxC 000 000 â  â 
â
   â âTxD 111 111 â 
â
   â â
   â âRxA 000 000 â Loop ââ 
â
   â âRxB 000 000 â 
â
   â âRxC 000 000 â 
â
   â âRxD 000 000

Can any one help me and tell step by step what can i do?
I'm avaible on MSN arkonek at windowslive.com

Arkon

- Original Message - 
From: "Steve Underwood" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Saturday, November 17, 2007 4:52 AM
Subject: Re: [asterisk-users] r2 multiframe error - continue


> Hi Jakub,
>
> Most countries which used to be part of the iron curtain block, back in
> the good old days, use the same protocol. Try the Czech variant. It will
> probably be OK for you. If it works, please report that, and Poland can
> be added to the list of variants.
>
> Steve
>
>
> Jakub Syrek wrote:
>> Im from Poland and there is no pl option, what should i chose?
>> Arkon
>>
>> - Original Message - 
>> From: "Moises Silva" <[EMAIL PROTECTED]>
>> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>> 
>> Sent: Friday, November 16, 2007 7:01 PM
>> Subject: Re: [asterisk-users] r2 multiframe error - continue
>>
>>
>>
>>> So, that means it is succeeded for mx protocolvariant. Now, just
>>> change the protocolvariant 'mx' to whatever fits your country, change
>>> only the country, please leave ANI,DNIS,OPTIONS . If it fails, I think
>>> a bug exists in your particular protocolvariant.
>>>
>>> Let me know the results.
>>>
>>> On Nov 16, 2007 11:11 AM, Jakub Syrek <[EMAIL PROTECTED]> wrote:
>>>
>>>> I was testing my system in local loop for protocolvariant mx,3,3(e1 
>>>> cross
>>>> cable between two spans).
>>>> Here are results:
>>>>
>>>> testcall
>>>> Loading protocol mfcr2
>>>> Thread for channel 0
>>>> MFC/R2 Chan  41: Call control(9)
>>>> MFC/R2 Chan  41: Unblock
>>>> MFC/R2 Chan  41: 1001  ->  [1/BLOCKED /Idle 
>>>>  /Idle ]
>>>> MFC/R2 Chan  41: far_unblocking_expired
>>>> MFC/R2 Chan  41: local_unblocking_expired
>>>> Chan  41: -- Far end unblocked! :-)
>>>> Chan  41: -- Far end unblocked! :-)
>>>> Chan  41: -- Local end unblocked! :-)
>>>> Chan  41: -- Local end unblocked! :-)
>>>> Chan  41: Initiating call
>>>> MFC/R2 Chan  41: Call control(1)
>>>> MFC/R2 Chan  41: Make call
>>>> MFC/R2 Chan  41: Creating a new call with CRN 32769
>>>> MFC/R2 Chan  41: 0001  ->  [1/DIALING /Seize 
>>>> /Idle ]
>>>> Chan  41: -- Dialing on channel 0
>>>> Chan  41: -- Dialing on channel 0
>>>> MFC/R2 Chan  41:  <- 1101  [1/DIALING /Seize 
>>>> /Idle ]
>>>> MFC/R2 Chan  41: 1 on  ->  [2/DIALING /Group I 
>>>> NIS ]
>>>> MFC/R2 Chan  41:  <- 6 on  [2/DIALING /Group I 
>>>> NIS ]
>>>> MFC/R2 Chan  41: 1 off ->  [2/DIALING /Group I 
>>>> NIS ]
>>>> MFC/R2 Chan  41:  <- 6 off [2/DIALING /Group I 
>>>> NIS ]
>>>> MFC/R2 Chan  41: Calling party category 0x0
>>>> MFC/R2 Chan  41: 1 on  ->  [2/DIALING /Group III 
>>>> /Category ]
>>>> MFC/R2 Chan  41:  <- 5 on  [2/DIALING /Group III 
>>>> /Category ]
>>>> MFC/R2 Chan  41: 1 off ->  [2/DIALING /Group III 
>>>> /Category ]
>>>> MFC/R2 Chan  41:  <- 5 off [2/DIALING /Group

Re: [asterisk-users] r2 multiframe error - continue

[Steve Underwood]

Hi Jakub,

Most countries which used to be part of the iron curtain block, back in 
the good old days, use the same protocol. Try the Czech variant. It will 
probably be OK for you. If it works, please report that, and Poland can 
be added to the list of variants.

Steve


Jakub Syrek wrote:
> Im from Poland and there is no pl option, what should i chose?
> Arkon
>
> - Original Message - 
> From: "Moises Silva" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
> 
> Sent: Friday, November 16, 2007 7:01 PM
> Subject: Re: [asterisk-users] r2 multiframe error - continue
>
>
>   
>> So, that means it is succeeded for mx protocolvariant. Now, just
>> change the protocolvariant 'mx' to whatever fits your country, change
>> only the country, please leave ANI,DNIS,OPTIONS . If it fails, I think
>> a bug exists in your particular protocolvariant.
>>
>> Let me know the results.
>>
>> On Nov 16, 2007 11:11 AM, Jakub Syrek <[EMAIL PROTECTED]> wrote:
>> 
>>> I was testing my system in local loop for protocolvariant mx,3,3(e1 cross
>>> cable between two spans).
>>> Here are results:
>>>
>>> testcall
>>> Loading protocol mfcr2
>>> Thread for channel 0
>>> MFC/R2 Chan  41: Call control(9)
>>> MFC/R2 Chan  41: Unblock
>>> MFC/R2 Chan  41: 1001  ->  [1/BLOCKED /Idle  /Idle ]
>>> MFC/R2 Chan  41: far_unblocking_expired
>>> MFC/R2 Chan  41: local_unblocking_expired
>>> Chan  41: -- Far end unblocked! :-)
>>> Chan  41: -- Far end unblocked! :-)
>>> Chan  41: -- Local end unblocked! :-)
>>> Chan  41: -- Local end unblocked! :-)
>>> Chan  41: Initiating call
>>> MFC/R2 Chan  41: Call control(1)
>>> MFC/R2 Chan  41: Make call
>>> MFC/R2 Chan  41: Creating a new call with CRN 32769
>>> MFC/R2 Chan  41: 0001  ->  [1/DIALING /Seize /Idle ]
>>> Chan  41: -- Dialing on channel 0
>>> Chan  41: -- Dialing on channel 0
>>> MFC/R2 Chan  41:  <- 1101  [1/DIALING /Seize /Idle ]
>>> MFC/R2 Chan  41: 1 on  ->  [2/DIALING /Group I   /DNIS ]
>>> MFC/R2 Chan  41:  <- 6 on  [2/DIALING /Group I   /DNIS ]
>>> MFC/R2 Chan  41: 1 off ->  [2/DIALING /Group I   /DNIS ]
>>> MFC/R2 Chan  41:  <- 6 off [2/DIALING /Group I   /DNIS ]
>>> MFC/R2 Chan  41: Calling party category 0x0
>>> MFC/R2 Chan  41: 1 on  ->  [2/DIALING /Group III /Category ]
>>> MFC/R2 Chan  41:  <- 5 on  [2/DIALING /Group III /Category ]
>>> MFC/R2 Chan  41: 1 off ->  [2/DIALING /Group III /Category ]
>>> MFC/R2 Chan  41:  <- 5 off [2/DIALING /Group III /Category ]
>>> MFC/R2 Chan  41: 2 on  ->  [2/DIALING /Group III /DNIS ]
>>> MFC/R2 Chan  41:  <- 1 on  [2/DIALING /Group I   /DNIS ]
>>> MFC/R2 Chan  41: 2 off ->  [2/DIALING /Group I   /DNIS ]
>>> MFC/R2 Chan  41:  <- 1 off [2/DIALING /Group I   /DNIS ]
>>> MFC/R2 Chan  41: 3 on  ->  [2/DIALING /Group I   /DNIS ]
>>> MFC/R2 Chan  41:  <- 1 on  [2/DIALING /Group I   /DNIS ]
>>> MFC/R2 Chan  41: 3 off ->  [2/DIALING /Group I   /DNIS ]
>>> MFC/R2 Chan  41:  <- 1 off [2/DIALING /Group I   /DNIS ]
>>> Main thread
>>> MFC/R2 Chan  41:  <- 3 on  [2/DIALING /Group I   /Silent   ]
>>> MFC/R2 Chan  41:  <- 3 off [2/DIALING /Group I   /Silent   ]
>>> MFC/R2 Chan  41: 1 on  ->  [2/PROCEED /Group II  /Category ]
>>> Chan  41: -- Proceeding on channel 0
>>> MFC/R2 Chan  41:  <- 1 on  [2/PROCEED /Group II  /Category ]
>>> MFC/R2 Chan  41: 1 off ->  [2/PROCEED /Group II  /Category ]
>>> MFC/R2 Chan  41:  <- 1 off [2/PROCEED /Group II  /Category ]
>>> Chan  41: -- Alerting on channel 0
>>> Chan  41: -- Alerting on channel 0
>>> MFC/R2 Chan  41:  <- 0101  [1/ALERTING/Await answer  /Category ]
>>> Chan  41: -- Connected on channel 0
>>> Chan  41: -- Connected on channel 0
>>> Chan  41: -- '*0001*343*123*#'
>>> Main thread
>>> Main thread
>>> Main thread
>>> MFC/R2 Chan  41:  <- 1101  [1/CONNECTD/Answered  /Category ]
>>> MFC/R2 Chan  41: Far end disconnected(cause=Norma

Re: [asterisk-users] r2 multiframe error - continue

[Moises Silva]

Let's start with something basic, try connecting in loop and using
protocolvariant=mx,0,4,7 then call yourself. That MUST work. Otherwise
you have messed up installing the incorrect libraries, I have seen too
many people complaining about the libraries not working and they just
forgot to install proper spandsp version or something like that. Other
common error is duplicating libraries installed in /usr/local with the
ones in /usr/lib

Regards

On Nov 16, 2007 6:31 AM, Jakub Syrek <[EMAIL PROTECTED]> wrote:
> Im using libs from astunicall-1.4.9-0.1.tar.gz at
> http://www.moythreads.com/astunicall/downloads/  (i have reinstalled
> asterisk, and libs from this package once again)
> No one can call me and i cant call out. Man from teleco still have
> teletransmision error..
> No after starting asterisk im getting in full log something like this:
>
> [Nov 16 13:22:09] NOTICE[3787] chan_unicall.c: Unicall/17 event Detected
> [Nov 16 13:22:09] WARNING[3787] chan_unicall.c: MFC/R2 UniCall/17  <- C
> on  [2/DETECTED/Seize ack /Seize ack]
> [Nov 16 13:22:09] WARNING[3787] chan_unicall.c: MFC/R2 UniCall/17 R2 prot.
> err. [2/DETECTED/Seize ack /Seize ack] cause 32772 - Unexpected MF6
> signal
> [Nov 16 13:22:09] WARNING[3787] chan_unicall.c: MFC/R2 UniCall/17 1001  ->
> [1/IDLE/Idle  /Idle ]
> [Nov 16 13:22:09] NOTICE[3787] chan_unicall.c: Unicall/17 event Protocol
> failure
> [Nov 16 13:22:09] ERROR[3787] chan_unicall.c: Unicall/17 protocol error.
> Cause 32772
> [Nov 16 13:22:09] WARNING[3787] chan_unicall.c: MFC/R2 UniCall/17 Channel
> echo cancel
> [Nov 16 13:22:09] DEBUG[3787] chan_unicall.c: disabled echo cancellation on
> channel 17
>
>
> What can i do?:)
>
> Regards
> Arkon
>
>
> ___
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
"Within C++, there is a much smaller and cleaner language struggling
to get out."

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] r2 multiframe error - continue

[Jakub Syrek]

> Wll, I think you should have started all this thread by mentioning
> that. May be libmfcr2 do not support R2 variant in poland.
Weeell ;) My mistake..

> For you, the quick solution might be just ask your E1 in ISDN-PRI.
>
> If you really want to stay with R2 or you have no choice, we can
> arrange a meeting to start figuring out how to support poland R2
> variant or a work-around for it. However I will not have any time
> before this wednesday.
I dont want to stay with R2 but my teleco force this signalling. I will ask 
them once again.

I will also install new elastix and try to change protocolvariant and 
...change protocolvariant .. and change protocolvariant ;]

Thanks for help once again

> Have a great weekend.
You too
>
> - Moy
>
Arkon 


___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] r2 multiframe error - continue

[Moises Silva]

Wll, I think you should have started all this thread by mentioning
that. May be libmfcr2 do not support R2 variant in poland.

For you, the quick solution might be just ask your E1 in ISDN-PRI.

If you really want to stay with R2 or you have no choice, we can
arrange a meeting to start figuring out how to support poland R2
variant or a work-around for it. However I will not have any time
before this wednesday.

Have a great weekend.

- Moy

On Nov 16, 2007 1:07 PM, Jakub Syrek <[EMAIL PROTECTED]> wrote:
> Im from Poland and there is no pl option, what should i chose?
> Arkon
>
> - Original Message -
> From: "Moises Silva" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> 
> Sent: Friday, November 16, 2007 7:01 PM
> Subject: Re: [asterisk-users] r2 multiframe error - continue
>
>
> > So, that means it is succeeded for mx protocolvariant. Now, just
> > change the protocolvariant 'mx' to whatever fits your country, change
> > only the country, please leave ANI,DNIS,OPTIONS . If it fails, I think
> > a bug exists in your particular protocolvariant.
> >
> > Let me know the results.
> >
> > On Nov 16, 2007 11:11 AM, Jakub Syrek <[EMAIL PROTECTED]> wrote:
> >> I was testing my system in local loop for protocolvariant mx,3,3(e1 cross
> >> cable between two spans).
> >> Here are results:
> >>
> >> testcall
> >> Loading protocol mfcr2
> >> Thread for channel 0
> >> MFC/R2 Chan  41: Call control(9)
> >> MFC/R2 Chan  41: Unblock
> >> MFC/R2 Chan  41: 1001  ->  [1/BLOCKED /Idle  /Idle ]
> >> MFC/R2 Chan  41: far_unblocking_expired
> >> MFC/R2 Chan  41: local_unblocking_expired
> >> Chan  41: -- Far end unblocked! :-)
> >> Chan  41: -- Far end unblocked! :-)
> >> Chan  41: -- Local end unblocked! :-)
> >> Chan  41: -- Local end unblocked! :-)
> >> Chan  41: Initiating call
> >> MFC/R2 Chan  41: Call control(1)
> >> MFC/R2 Chan  41: Make call
> >> MFC/R2 Chan  41: Creating a new call with CRN 32769
> >> MFC/R2 Chan  41: 0001  ->  [1/DIALING /Seize /Idle ]
> >> Chan  41: -- Dialing on channel 0
> >> Chan  41: -- Dialing on channel 0
> >> MFC/R2 Chan  41:  <- 1101  [1/DIALING /Seize /Idle ]
> >> MFC/R2 Chan  41: 1 on  ->  [2/DIALING /Group I   /DNIS ]
> >> MFC/R2 Chan  41:  <- 6 on  [2/DIALING /Group I   /DNIS ]
> >> MFC/R2 Chan  41: 1 off ->  [2/DIALING /Group I   /DNIS ]
> >> MFC/R2 Chan  41:  <- 6 off [2/DIALING /Group I   /DNIS ]
> >> MFC/R2 Chan  41: Calling party category 0x0
> >> MFC/R2 Chan  41: 1 on  ->  [2/DIALING /Group III /Category ]
> >> MFC/R2 Chan  41:  <- 5 on  [2/DIALING /Group III /Category ]
> >> MFC/R2 Chan  41: 1 off ->  [2/DIALING /Group III /Category ]
> >> MFC/R2 Chan  41:  <- 5 off [2/DIALING /Group III /Category ]
> >> MFC/R2 Chan  41: 2 on  ->  [2/DIALING /Group III /DNIS ]
> >> MFC/R2 Chan  41:  <- 1 on  [2/DIALING /Group I   /DNIS ]
> >> MFC/R2 Chan  41: 2 off ->  [2/DIALING /Group I   /DNIS ]
> >> MFC/R2 Chan  41:  <- 1 off [2/DIALING /Group I   /DNIS ]
> >> MFC/R2 Chan  41: 3 on  ->  [2/DIALING /Group I   /DNIS ]
> >> MFC/R2 Chan  41:  <- 1 on  [2/DIALING /Group I   /DNIS ]
> >> MFC/R2 Chan  41: 3 off ->  [2/DIALING /Group I   /DNIS ]
> >> MFC/R2 Chan  41:  <- 1 off [2/DIALING /Group I   /DNIS ]
> >> Main thread
> >> MFC/R2 Chan  41:  <- 3 on  [2/DIALING /Group I   /Silent   ]
> >> MFC/R2 Chan  41:  <- 3 off [2/DIALING /Group I   /Silent   ]
> >> MFC/R2 Chan  41: 1 on  ->  [2/PROCEED /Group II  /Category ]
> >> Chan  41: -- Proceeding on channel 0
> >> MFC/R2 Chan  41:  <- 1 on  [2/PROCEED /Group II  /Category ]
> >> MFC/R2 Chan  41: 1 off ->  [2/PROCEED /Group II  /Category ]
> >> MFC/R2 Chan  41:  <- 1 off [2/PROCEED /Group II  /Category ]
> >> Chan  41: -- Alerting on channel 0
> >> Chan  41: -- Alerting on channel 0
> >> MFC/R2 Chan  41:  <- 0101  [1/ALERTING/Await answer  /Category ]
> >> Chan  41: -- Connected on channel 0
> >> Chan  41: -- Connected on channel 0
> >> Chan  41

Re: [asterisk-users] r2 multiframe error - continue

[Jakub Syrek]

Im from Poland and there is no pl option, what should i chose?
Arkon

- Original Message - 
From: "Moises Silva" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Friday, November 16, 2007 7:01 PM
Subject: Re: [asterisk-users] r2 multiframe error - continue


> So, that means it is succeeded for mx protocolvariant. Now, just
> change the protocolvariant 'mx' to whatever fits your country, change
> only the country, please leave ANI,DNIS,OPTIONS . If it fails, I think
> a bug exists in your particular protocolvariant.
>
> Let me know the results.
>
> On Nov 16, 2007 11:11 AM, Jakub Syrek <[EMAIL PROTECTED]> wrote:
>> I was testing my system in local loop for protocolvariant mx,3,3(e1 cross
>> cable between two spans).
>> Here are results:
>>
>> testcall
>> Loading protocol mfcr2
>> Thread for channel 0
>> MFC/R2 Chan  41: Call control(9)
>> MFC/R2 Chan  41: Unblock
>> MFC/R2 Chan  41: 1001  ->  [1/BLOCKED /Idle  /Idle ]
>> MFC/R2 Chan  41: far_unblocking_expired
>> MFC/R2 Chan  41: local_unblocking_expired
>> Chan  41: -- Far end unblocked! :-)
>> Chan  41: -- Far end unblocked! :-)
>> Chan  41: -- Local end unblocked! :-)
>> Chan  41: -- Local end unblocked! :-)
>> Chan  41: Initiating call
>> MFC/R2 Chan  41: Call control(1)
>> MFC/R2 Chan  41: Make call
>> MFC/R2 Chan  41: Creating a new call with CRN 32769
>> MFC/R2 Chan  41: 0001  ->  [1/DIALING /Seize /Idle ]
>> Chan  41: -- Dialing on channel 0
>> Chan  41: -- Dialing on channel 0
>> MFC/R2 Chan  41:  <- 1101  [1/DIALING /Seize /Idle ]
>> MFC/R2 Chan  41: 1 on  ->  [2/DIALING /Group I   /DNIS ]
>> MFC/R2 Chan  41:  <- 6 on  [2/DIALING /Group I   /DNIS ]
>> MFC/R2 Chan  41: 1 off ->  [2/DIALING /Group I   /DNIS ]
>> MFC/R2 Chan  41:  <- 6 off [2/DIALING /Group I   /DNIS ]
>> MFC/R2 Chan  41: Calling party category 0x0
>> MFC/R2 Chan  41: 1 on  ->  [2/DIALING /Group III /Category ]
>> MFC/R2 Chan  41:  <- 5 on  [2/DIALING /Group III /Category ]
>> MFC/R2 Chan  41: 1 off ->  [2/DIALING /Group III /Category ]
>> MFC/R2 Chan  41:  <- 5 off [2/DIALING /Group III /Category ]
>> MFC/R2 Chan  41: 2 on  ->  [2/DIALING /Group III /DNIS ]
>> MFC/R2 Chan  41:  <- 1 on  [2/DIALING /Group I   /DNIS ]
>> MFC/R2 Chan  41: 2 off ->  [2/DIALING /Group I   /DNIS ]
>> MFC/R2 Chan  41:  <- 1 off [2/DIALING /Group I   /DNIS ]
>> MFC/R2 Chan  41: 3 on  ->  [2/DIALING /Group I   /DNIS ]
>> MFC/R2 Chan  41:  <- 1 on  [2/DIALING /Group I   /DNIS ]
>> MFC/R2 Chan  41: 3 off ->  [2/DIALING /Group I   /DNIS ]
>> MFC/R2 Chan  41:  <- 1 off [2/DIALING /Group I   /DNIS ]
>> Main thread
>> MFC/R2 Chan  41:  <- 3 on  [2/DIALING /Group I   /Silent   ]
>> MFC/R2 Chan  41:  <- 3 off [2/DIALING /Group I   /Silent   ]
>> MFC/R2 Chan  41: 1 on  ->  [2/PROCEED /Group II  /Category ]
>> Chan  41: -- Proceeding on channel 0
>> MFC/R2 Chan  41:  <- 1 on  [2/PROCEED /Group II  /Category ]
>> MFC/R2 Chan  41: 1 off ->  [2/PROCEED /Group II  /Category ]
>> MFC/R2 Chan  41:  <- 1 off [2/PROCEED /Group II  /Category ]
>> Chan  41: -- Alerting on channel 0
>> Chan  41: -- Alerting on channel 0
>> MFC/R2 Chan  41:  <- 0101  [1/ALERTING/Await answer  /Category ]
>> Chan  41: -- Connected on channel 0
>> Chan  41: -- Connected on channel 0
>> Chan  41: -- '*0001*343*123*#'
>> Main thread
>> Main thread
>> Main thread
>> MFC/R2 Chan  41:  <- 1101  [1/CONNECTD/Answered  /Category ]
>> MFC/R2 Chan  41: Far end disconnected(cause=Normal Clearing [16]) - state
>> 0x400
>>
>> and asterisk log
>> [Nov 16 18:06:51] NOTICE[28848] chan_unicall.c: Unicall/25 event Local 
>> end
>> unblocked
>> [Nov 16 18:06:51] NOTICE[28848] chan_unicall.c: Unicall/26 event Local 
>> end
>> unblocked
>> [Nov 16 18:06:51] NOTICE[28848] chan_unicall.c: Unicall/27 event Local 
>> end
>> unblocked
>> [Nov 16 18:06:51] NOTICE[28848] chan_unicall.c: Unicall/28 event Local 
>> end
>> unblocked
>> [Nov 16 18:06:51] NOTICE[28848] chan_unicall.c: Unicall/29 event Local 
>> end
>> unblo

Re: [asterisk-users] r2 multiframe error - continue

[Moises Silva]

: 3 index = 0,
> normal = 11, callwait = -1, thirdcall = -1
> [Nov 16 18:07:14] DEBUG[28886] chan_unicall.c: Updated conferencing on 3,
> with 0 conference users
> [Nov 16 18:07:14] NOTICE[28886] chan_unicall.c: Unicall/10 event Answered
> [Nov 16 18:07:14] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel
> echo cancel
> [Nov 16 18:07:14] DEBUG[28886] chan_unicall.c: Enabled echo cancellation on
> channel 10
> [Nov 16 18:07:14] DEBUG[28886] chan_unicall.c: DTMF digit: * on UniCall/10-1
> [Nov 16 18:07:14] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel
> switching
> [Nov 16 18:07:14] DEBUG[28886] chan_unicall.c: DTMF digit: 0 on UniCall/10-1
> [Nov 16 18:07:14] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel
> switching
> [Nov 16 18:07:14] DEBUG[28886] chan_unicall.c: DTMF digit: 0 on UniCall/10-1
> [Nov 16 18:07:14] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel
> switching
> [Nov 16 18:07:15] DEBUG[28886] chan_unicall.c: DTMF digit: 0 on UniCall/10-1
> [Nov 16 18:07:15] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel
> switching
> [Nov 16 18:07:15] DEBUG[28886] chan_unicall.c: DTMF digit: 0 on UniCall/10-1
> [Nov 16 18:07:15] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel
> switching
> [Nov 16 18:07:15] DEBUG[28886] chan_unicall.c: DTMF digit: 0 on UniCall/10-1
> [Nov 16 18:07:15] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel
> switching
> [Nov 16 18:07:15] DEBUG[28886] chan_unicall.c: DTMF digit: 0 on UniCall/10-1
> [Nov 16 18:07:15] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel
> switching
> [Nov 16 18:07:15] DEBUG[28886] chan_unicall.c: DTMF digit: 0 on UniCall/10-1
> [Nov 16 18:07:15] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel
> switching
> [Nov 16 18:07:15] DEBUG[28886] chan_unicall.c: DTMF digit: 1 on UniCall/10-1
> [Nov 16 18:07:15] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel
> switching
> [Nov 16 18:07:15] DEBUG[28886] chan_unicall.c: DTMF digit: * on UniCall/10-1
> [Nov 16 18:07:15] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel
> switching
> [Nov 16 18:07:15] DEBUG[28886] chan_unicall.c: DTMF digit: 3 on UniCall/10-1
> [Nov 16 18:07:15] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel
> switching
> [Nov 16 18:07:15] DEBUG[28886] chan_unicall.c: DTMF digit: 4 on UniCall/10-1
> [Nov 16 18:07:15] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel
> switching
> [Nov 16 18:07:15] DEBUG[28886] chan_unicall.c: DTMF digit: 3 on UniCall/10-1
> [Nov 16 18:07:15] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel
> switching
> [Nov 16 18:07:16] DEBUG[28886] chan_unicall.c: DTMF digit: * on UniCall/10-1
> [Nov 16 18:07:16] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel
> switching
> [Nov 16 18:07:16] DEBUG[28886] chan_unicall.c: DTMF digit: 1 on UniCall/10-1
> [Nov 16 18:07:16] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel
> switching
> [Nov 16 18:07:16] DEBUG[28886] chan_unicall.c: DTMF digit: 2 on UniCall/10-1
> [Nov 16 18:07:16] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel
> switching
> [Nov 16 18:07:16] DEBUG[28886] chan_unicall.c: DTMF digit: 3 on UniCall/10-1
> [Nov 16 18:07:16] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel
> switching
> [Nov 16 18:07:16] DEBUG[28886] chan_unicall.c: DTMF digit: * on UniCall/10-1
> [Nov 16 18:07:16] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel
> switching
> [Nov 16 18:07:16] DEBUG[28886] chan_unicall.c: DTMF digit: # on UniCall/10-1
> [Nov 16 18:07:16] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel
> switching
> [Nov 16 18:07:31] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel
> gains
> [Nov 16 18:07:31] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel
> switching
> [Nov 16 18:07:31] DEBUG[28886] chan_unicall.c: Hangup: channel: 10 index =
> 0, normal = 18, callwait = -1, thirdcall = -1
> [Nov 16 18:07:31] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Call
> control(7)
> [Nov 16 18:07:31] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Drop
> call(cause=Normal Clearing [16])
> [Nov 16 18:07:31] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 1101  ->
> [1/CONNECTD/Clear back/Accepted Paid]
>
>
> Thanks for your help and patience Moy
>
> - Original Message -
> From: "Moises Silva" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> 
> Sent: Friday, November 16, 2007 3:39 PM
> Subject: Re: [asterisk-users] r2 multiframe error - continue
>
>
> > Let's start with something basic, try connecting in loop and using
> > protocolvariant=mx,0,4,7 then call yourself. That MUST work. Otherwise
> > you have messed up in

Re: [asterisk-users] r2 multiframe error - continue

[Luis Antonio Prata Barbosa]

rmal = 11, callwait = -1, thirdcall = -1
> [Nov 16 18:07:14] DEBUG[28886] chan_unicall.c: Updated conferencing on 3,
> with 0 conference users
> [Nov 16 18:07:14] NOTICE[28886] chan_unicall.c: Unicall/10 event Answered
> [Nov 16 18:07:14] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel
> echo cancel
> [Nov 16 18:07:14] DEBUG[28886] chan_unicall.c: Enabled echo cancellation
> on
> channel 10
> [Nov 16 18:07:14] DEBUG[28886] chan_unicall.c: DTMF digit: * on
> UniCall/10-1
> [Nov 16 18:07:14] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel
> switching
> [Nov 16 18:07:14] DEBUG[28886] chan_unicall.c: DTMF digit: 0 on
> UniCall/10-1
> [Nov 16 18:07:14] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel
> switching
> [Nov 16 18:07:14] DEBUG[28886] chan_unicall.c: DTMF digit: 0 on
> UniCall/10-1
> [Nov 16 18:07:14] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel
> switching
> [Nov 16 18:07:15] DEBUG[28886] chan_unicall.c: DTMF digit: 0 on
> UniCall/10-1
> [Nov 16 18:07:15] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel
> switching
> [Nov 16 18:07:15] DEBUG[28886] chan_unicall.c: DTMF digit: 0 on
> UniCall/10-1
> [Nov 16 18:07:15] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel
> switching
> [Nov 16 18:07:15] DEBUG[28886] chan_unicall.c: DTMF digit: 0 on
> UniCall/10-1
> [Nov 16 18:07:15] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel
> switching
> [Nov 16 18:07:15] DEBUG[28886] chan_unicall.c: DTMF digit: 0 on
> UniCall/10-1
> [Nov 16 18:07:15] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel
> switching
> [Nov 16 18:07:15] DEBUG[28886] chan_unicall.c: DTMF digit: 0 on
> UniCall/10-1
> [Nov 16 18:07:15] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel
> switching
> [Nov 16 18:07:15] DEBUG[28886] chan_unicall.c: DTMF digit: 1 on
> UniCall/10-1
> [Nov 16 18:07:15] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel
> switching
> [Nov 16 18:07:15] DEBUG[28886] chan_unicall.c: DTMF digit: * on
> UniCall/10-1
> [Nov 16 18:07:15] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel
> switching
> [Nov 16 18:07:15] DEBUG[28886] chan_unicall.c: DTMF digit: 3 on
> UniCall/10-1
> [Nov 16 18:07:15] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel
> switching
> [Nov 16 18:07:15] DEBUG[28886] chan_unicall.c: DTMF digit: 4 on
> UniCall/10-1
> [Nov 16 18:07:15] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel
> switching
> [Nov 16 18:07:15] DEBUG[28886] chan_unicall.c: DTMF digit: 3 on
> UniCall/10-1
> [Nov 16 18:07:15] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel
> switching
> [Nov 16 18:07:16] DEBUG[28886] chan_unicall.c: DTMF digit: * on
> UniCall/10-1
> [Nov 16 18:07:16] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel
> switching
> [Nov 16 18:07:16] DEBUG[28886] chan_unicall.c: DTMF digit: 1 on
> UniCall/10-1
> [Nov 16 18:07:16] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel
> switching
> [Nov 16 18:07:16] DEBUG[28886] chan_unicall.c: DTMF digit: 2 on
> UniCall/10-1
> [Nov 16 18:07:16] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel
> switching
> [Nov 16 18:07:16] DEBUG[28886] chan_unicall.c: DTMF digit: 3 on
> UniCall/10-1
> [Nov 16 18:07:16] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel
> switching
> [Nov 16 18:07:16] DEBUG[28886] chan_unicall.c: DTMF digit: * on
> UniCall/10-1
> [Nov 16 18:07:16] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel
> switching
> [Nov 16 18:07:16] DEBUG[28886] chan_unicall.c: DTMF digit: # on
> UniCall/10-1
> [Nov 16 18:07:16] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel
> switching
> [Nov 16 18:07:31] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel
> gains
> [Nov 16 18:07:31] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel
> switching
> [Nov 16 18:07:31] DEBUG[28886] chan_unicall.c: Hangup: channel: 10 index =
> 0, normal = 18, callwait = -1, thirdcall = -1
> [Nov 16 18:07:31] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Call
> control(7)
> [Nov 16 18:07:31] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Drop
> call(cause=Normal Clearing [16])
> [Nov 16 18:07:31] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10
> 1101  ->
> [1/CONNECTD/Clear back/Accepted Paid]
>
>
> Thanks for your help and patience Moy
>
> - Original Message -
> From: "Moises Silva" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> 
> Sent: Friday, November 16, 2007 3:39 PM
> Subject: Re: [asterisk-users] r2 multiframe error - continue
>
>
> > Let's start with something basic, try connecting in loop and using
> > protoc

Re: [asterisk-users] r2 multiframe error - continue

[Jakub Syrek]

16 18:07:15] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel 
switching
[Nov 16 18:07:15] DEBUG[28886] chan_unicall.c: DTMF digit: 0 on UniCall/10-1
[Nov 16 18:07:15] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel 
switching
[Nov 16 18:07:15] DEBUG[28886] chan_unicall.c: DTMF digit: 0 on UniCall/10-1
[Nov 16 18:07:15] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel 
switching
[Nov 16 18:07:15] DEBUG[28886] chan_unicall.c: DTMF digit: 0 on UniCall/10-1
[Nov 16 18:07:15] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel 
switching
[Nov 16 18:07:15] DEBUG[28886] chan_unicall.c: DTMF digit: 1 on UniCall/10-1
[Nov 16 18:07:15] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel 
switching
[Nov 16 18:07:15] DEBUG[28886] chan_unicall.c: DTMF digit: * on UniCall/10-1
[Nov 16 18:07:15] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel 
switching
[Nov 16 18:07:15] DEBUG[28886] chan_unicall.c: DTMF digit: 3 on UniCall/10-1
[Nov 16 18:07:15] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel 
switching
[Nov 16 18:07:15] DEBUG[28886] chan_unicall.c: DTMF digit: 4 on UniCall/10-1
[Nov 16 18:07:15] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel 
switching
[Nov 16 18:07:15] DEBUG[28886] chan_unicall.c: DTMF digit: 3 on UniCall/10-1
[Nov 16 18:07:15] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel 
switching
[Nov 16 18:07:16] DEBUG[28886] chan_unicall.c: DTMF digit: * on UniCall/10-1
[Nov 16 18:07:16] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel 
switching
[Nov 16 18:07:16] DEBUG[28886] chan_unicall.c: DTMF digit: 1 on UniCall/10-1
[Nov 16 18:07:16] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel 
switching
[Nov 16 18:07:16] DEBUG[28886] chan_unicall.c: DTMF digit: 2 on UniCall/10-1
[Nov 16 18:07:16] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel 
switching
[Nov 16 18:07:16] DEBUG[28886] chan_unicall.c: DTMF digit: 3 on UniCall/10-1
[Nov 16 18:07:16] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel 
switching
[Nov 16 18:07:16] DEBUG[28886] chan_unicall.c: DTMF digit: * on UniCall/10-1
[Nov 16 18:07:16] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel 
switching
[Nov 16 18:07:16] DEBUG[28886] chan_unicall.c: DTMF digit: # on UniCall/10-1
[Nov 16 18:07:16] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel 
switching
[Nov 16 18:07:31] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel 
gains
[Nov 16 18:07:31] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Channel 
switching
[Nov 16 18:07:31] DEBUG[28886] chan_unicall.c: Hangup: channel: 10 index = 
0, normal = 18, callwait = -1, thirdcall = -1
[Nov 16 18:07:31] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Call 
control(7)
[Nov 16 18:07:31] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 Drop 
call(cause=Normal Clearing [16])
[Nov 16 18:07:31] WARNING[28886] chan_unicall.c: MFC/R2 UniCall/10 1101  -> 
[1/CONNECTD/Clear back/Accepted Paid]


Thanks for your help and patience Moy

- Original Message - 
From: "Moises Silva" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Friday, November 16, 2007 3:39 PM
Subject: Re: [asterisk-users] r2 multiframe error - continue


> Let's start with something basic, try connecting in loop and using
> protocolvariant=mx,0,4,7 then call yourself. That MUST work. Otherwise
> you have messed up installing the incorrect libraries, I have seen too
> many people complaining about the libraries not working and they just
> forgot to install proper spandsp version or something like that. Other
> common error is duplicating libraries installed in /usr/local with the
> ones in /usr/lib
>
> Regards
>
> On Nov 16, 2007 6:31 AM, Jakub Syrek <[EMAIL PROTECTED]> wrote:
>> Im using libs from astunicall-1.4.9-0.1.tar.gz at
>> http://www.moythreads.com/astunicall/downloads/  (i have reinstalled
>> asterisk, and libs from this package once again)
>> No one can call me and i cant call out. Man from teleco still have
>> teletransmision error..
>> No after starting asterisk im getting in full log something like this:
>>
>> [Nov 16 13:22:09] NOTICE[3787] chan_unicall.c: Unicall/17 event Detected
>> [Nov 16 13:22:09] WARNING[3787] chan_unicall.c: MFC/R2 UniCall/17  <- 
>> C
>> on  [2/DETECTED/Seize ack /Seize ack]
>> [Nov 16 13:22:09] WARNING[3787] chan_unicall.c: MFC/R2 UniCall/17 R2 
>> prot.
>> err. [2/DETECTED/Seize ack /Seize ack] cause 32772 - Unexpected 
>> MF6
>> signal
>> [Nov 16 13:22:09] WARNING[3787] chan_unicall.c: MFC/R2 UniCall/17 
>> 001  ->
>> [1/IDLE/Idle  /Idle ]
>> [Nov 16 13:22:09] NOTICE[3787] chan_unicall.c: Unicall/17 event Protocol
>> failure
>> [Nov 16 13:22:09] ERROR[3787] chan_unicall.c: Unicall/17 protocol error.
&

Re: [asterisk-users] r2 multiframe error - continue

[Jakub Syrek]

Im using libs from astunicall-1.4.9-0.1.tar.gz at 
http://www.moythreads.com/astunicall/downloads/  (i have reinstalled 
asterisk, and libs from this package once again)
No one can call me and i cant call out. Man from teleco still have 
teletransmision error..
No after starting asterisk im getting in full log something like this:

[Nov 16 13:22:09] NOTICE[3787] chan_unicall.c: Unicall/17 event Detected
[Nov 16 13:22:09] WARNING[3787] chan_unicall.c: MFC/R2 UniCall/17  <- C 
on  [2/DETECTED/Seize ack /Seize ack]
[Nov 16 13:22:09] WARNING[3787] chan_unicall.c: MFC/R2 UniCall/17 R2 prot. 
err. [2/DETECTED/Seize ack /Seize ack] cause 32772 - Unexpected MF6 
signal
[Nov 16 13:22:09] WARNING[3787] chan_unicall.c: MFC/R2 UniCall/17 1001  -> 
[1/IDLE/Idle  /Idle ]
[Nov 16 13:22:09] NOTICE[3787] chan_unicall.c: Unicall/17 event Protocol 
failure
[Nov 16 13:22:09] ERROR[3787] chan_unicall.c: Unicall/17 protocol error. 
Cause 32772
[Nov 16 13:22:09] WARNING[3787] chan_unicall.c: MFC/R2 UniCall/17 Channel 
echo cancel
[Nov 16 13:22:09] DEBUG[3787] chan_unicall.c: disabled echo cancellation on 
channel 17


What can i do?:)

Regards
Arkon


___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] r2 multiframe error

[Jakub Syrek]

>> I downloaded packages for 1.4.
>> Shoudl I now install asterisk, zaptel, libs for unicall as it is typed at
>> http://www.voip-info.org/tiki-index.php?page=Asterisk+MFC+R2 ?
> astunicall package already include zaptel, asterisk and all the Steve
> underwood libraries. However, personally, since Elastix Asterisk and
> Zaptel was working just fine. I just removed libunicall, libmfcr2,
> libsupertone , spandsp and chan_unicall.so packages and installed
> those ones from the astunicall tar leaving Asterisk and Zaptel
> included with Elastix untouched.
>
>> I will get acces to my asterisk machine tomorrow or on wednesday so i 
>> will
>> do it then and let you now.
> Perfect, feedback will be appreciated.
>
> Moy

Hi,
I've downloaded and installed packages as it was told. Everything seems to 
by ok but i still can't dial or recive  calls.  I have connected two ports 
of my card with E1 cross cable and  try to connect from asterisk to 
testcall.

Here are results:
[EMAIL PROTECTED] ~]# cat /var/log/asterisk/full | grep unicall
[Nov 14 22:42:30] DEBUG[4530] chan_unicall.c: unicall_call called - '2/825'
[Nov 14 22:42:30] DEBUG[4530] chan_unicall.c: unicall_call caller id - '200'
[Nov 14 22:42:30] WARNING[4530] chan_unicall.c: MFC/R2 UniCall/2 Call 
control(1)
[Nov 14 22:42:30] WARNING[4530] chan_unicall.c: MFC/R2 UniCall/2 Make call
[Nov 14 22:42:30] WARNING[4530] chan_unicall.c: MFC/R2 UniCall/2 Making a 
new call with CRN 32770
[Nov 14 22:42:30] WARNING[4530] chan_unicall.c: MFC/R2 UniCall/2 0001  -> 
[1/   1/Idle  /Idle ]
[Nov 14 22:42:30] NOTICE[4530] chan_unicall.c: Exception on 10, channel 2
[Nov 14 22:42:30] NOTICE[4530] chan_unicall.c: Unicall/2 event Dialing
[Nov 14 22:42:30] WARNING[4530] chan_unicall.c: MFC/R2 UniCall/2  <- 
1101  [1/  40/Seize /Idle ]
[Nov 14 22:42:30] WARNING[4530] chan_unicall.c: MFC/R2 UniCall/2 8 on  -> 
[2/  40/Group I   /Idle ]
[Nov 14 22:42:35] WARNING[4530] chan_unicall.c: MFC/R2 UniCall/2 R2 prot. 
err. [2/  40/Group I   /DNIS ] cause 32769 - T1 timed out
[Nov 14 22:42:35] WARNING[4530] chan_unicall.c: MFC/R2 UniCall/2 8 off -> 
[1/   1/Idle  /Idle ]
[Nov 14 22:42:35] WARNING[4530] chan_unicall.c: MFC/R2 UniCall/2 1001  -> 
[1/   1/Idle  /Idle ]
[Nov 14 22:42:35] NOTICE[4530] chan_unicall.c: Exception on 10, channel 2
[Nov 14 22:42:35] NOTICE[4530] chan_unicall.c: Unicall/2 event Protocol 
failure
[Nov 14 22:42:35] ERROR[4530] chan_unicall.c: Unicall/2 protocol error. 
Cause 32769
[Nov 14 22:42:35] WARNING[4530] chan_unicall.c: MFC/R2 UniCall/2  <- 
1001  [1/   1/Idle  /Idle ]
[Nov 14 22:42:35] WARNING[4530] chan_unicall.c: MFC/R2 UniCall/2 1001  -> 
[1/   1/Idle  /Idle ]
[Nov 14 22:42:35] WARNING[4530] chan_unicall.c: MFC/R2 UniCall/2 Channel 
gains
[Nov 14 22:42:35] WARNING[4530] chan_unicall.c: MFC/R2 UniCall/2 Channel 
switching
[Nov 14 22:42:35] DEBUG[4530] chan_unicall.c: Hangup: channel: 2 index = 0, 
normal = 10, callwait = -1, thirdcall = -1
[Nov 14 22:42:35] DEBUG[4530] chan_unicall.c: Updated conferencing on 2, 
with 0 conference users

from testcall
MFC/R2 Chan  33:  <- 0001  [1/IDLE/Idle  /Idle ]
MFC/R2 Chan  33: Detected
MFC/R2 Chan  33: Creating a new call with CRN 32769
MFC/R2 Chan  33: 1101  ->  [2/DETECTED/Seize ack /Seize ack]
Chan  33: -- Detected on channel 0, CRN 32769
Chan  33: -- Detected on channel 0, CRN 32769
Main thread
MFC/R2 Chan  33:  <- 1001  [2/DETECTED/Seize ack /Seize ack]
MFC/R2 Chan  33: Far end disconnected(cause=Normal, unspecified cause 
[31]) - st 
ate 0x2
Chan  33: -- Far end disconnected on channel 0
Chan  33: -- Far end disconnected on channel 0
MFC/R2 Chan  33: Call control(7)
MFC/R2 Chan  33: Drop call(cause=Normal Clearing [16])
MFC/R2 Chan  33: Call disconnected(cause=Normal, unspecified cause [31]) - 
state 
0x800
Chan  33: -- Drop call on channel 0
Chan  33: -- Drop call on channel 0
MFC/R2 Chan  33: Call control(8)
MFC/R2 Chan  33: Release call
MFC/R2 Chan  33: 1001  ->  [1/ALL DISC/Release guard /Seize ack]
MFC/R2 Chan  33: Release guard expired
MFC/R2 Chan  33: Destroying call with CRN 32769
Chan  33: -- Release call on channel 0
Chan  33: -- Release call on channel 0

Files:
[EMAIL PROTECTED] ~]# cat /etc/asterisk/unicall.conf
[channels]
loglevel=255
context=default
usecallerid=no
threewaycalling=no
transfer=no
cancallforward=yes
callreturn=yes
echocancel=no
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=yes
protocolclass=mfcr2
protocolvariant=cz,3,3
protocolend=co
supertones=pl
channel => 1-15
channel => 17-31

[EMAIL PROTECTED] ~]# cat /etc/zaptel.conf
# Autogenerated by /usr/sbin/zapconf on Wed Nov  7 18:27:13 2007 -- do not 
hand edit
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#
# Span 1: ZTDUMMY/1 "ZTDUMMY/1 1"

# Glo

Re: [asterisk-users] r2 multiframe error

[Moises Silva]

> I downloaded packages for 1.4.
> Shoudl I now install asterisk, zaptel, libs for unicall as it is typed at
> http://www.voip-info.org/tiki-index.php?page=Asterisk+MFC+R2 ?
astunicall package already include zaptel, asterisk and all the Steve
underwood libraries. However, personally, since Elastix Asterisk and
Zaptel was working just fine. I just removed libunicall, libmfcr2,
libsupertone , spandsp and chan_unicall.so packages and installed
those ones from the astunicall tar leaving Asterisk and Zaptel
included with Elastix untouched.

> I will get acces to my asterisk machine tomorrow or on wednesday so i will
> do it then and let you now.
Perfect, feedback will be appreciated.

Moy

-- 
"Within C++, there is a much smaller and cleaner language struggling
to get out."

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] r2 multiframe error

[Jakub Syrek]

Hi,
Thanks for your fast replay.
I downloaded packages for 1.4.
Shoudl I now install asterisk, zaptel, libs for unicall as it is typed at 
http://www.voip-info.org/tiki-index.php?page=Asterisk+MFC+R2 ?
I will get acces to my asterisk machine tomorrow or on wednesday so i will 
do it then and let you now.
Regard
Arkon


> Hi Arkon,
>
> I run the blog http://www.moythreads.com/astunicall/ where you can
> find packages that are known to work for Unicall/R2.
>
> I also recently ( 2 weeks ago ) joined Elastix development to help
> them to support R2. However, this weekend was the first time I
> downloaded elastix and actually tried it. And I found that R2 did not
> worked for me. I am not sure where they got the R2 libraries from, but
> the fact is that as soon I downloaded the packages I have in the blog
> I mentioned, it worked as I expected. So go ahead and try downloading
> and installing that package and let me know if it works for you.
>
> I will discuss with the other Elastix developers to see where they got
> the R2 versions and if they actually tested R2 signaling. I will try
> to persuade them to include the newest packages in Elastix for this
> release.
>
> Regards,
>
> Moy
>


___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] r2 multiframe error

[Moises Silva]

Hi Arkon,

I run the blog http://www.moythreads.com/astunicall/ where you can
find packages that are known to work for Unicall/R2.

I also recently ( 2 weeks ago ) joined Elastix development to help
them to support R2. However, this weekend was the first time I
downloaded elastix and actually tried it. And I found that R2 did not
worked for me. I am not sure where they got the R2 libraries from, but
the fact is that as soon I downloaded the packages I have in the blog
I mentioned, it worked as I expected. So go ahead and try downloading
and installing that package and let me know if it works for you.

I will discuss with the other Elastix developers to see where they got
the R2 versions and if they actually tested R2 signaling. I will try
to persuade them to include the newest packages in Elastix for this
release.

Regards,

Moy


On Nov 10, 2007 2:17 PM,  <[EMAIL PROTECTED]> wrote:
> Hello
> I have conected line from telecom company (TP - Poland) and im forced to
> use mfc/r2 signaling. Everything seems to be ok (gren light on card, in
> zttool status OK) but i cant recive nor dial calls. Men from telecom
> company told me that they heve multiframe errors from my card all the time.
> I was searching solution few days but i havent met simmilar problem.. Can
> someone help me?
>
> My system is elastix v0.9 - beta2
> Card is from phoniceq end is recognise as TE210p
>
> Conf files:
> zaptel.conf
>
> span=1,1,0,cas,hdb3
> cas=1-15:1101
> cas=17-31:1101
> defaultzone=pl
> loadzone=pl
>
> unicall.conf
> protocolclass=mfcr2
> protocolend=co
> protocolvariant=cz,10,6
> channel=1-15
> channel=17-31
>
> Regards
> Arkon
>
>
> ___
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
"Within C++, there is a much smaller and cleaner language struggling
to get out."

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] r2 multiframe error

[arkon]

Hello
I have conected line from telecom company (TP - Poland) and im forced to 
use mfc/r2 signaling. Everything seems to be ok (gren light on card, in 
zttool status OK) but i cant recive nor dial calls. Men from telecom 
company told me that they heve multiframe errors from my card all the time. 
I was searching solution few days but i havent met simmilar problem.. Can 
someone help me?

My system is elastix v0.9 - beta2
Card is from phoniceq end is recognise as TE210p

Conf files:
zaptel.conf

span=1,1,0,cas,hdb3
cas=1-15:1101
cas=17-31:1101
defaultzone=pl
loadzone=pl

unicall.conf
protocolclass=mfcr2
protocolend=co
protocolvariant=cz,10,6
channel=1-15
channel=17-31

Regards
Arkon


___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] R2 Argentina

[Oscar Carriles]

Dear Folks,

I have found that Argentine variant ar libmfcr2.0.0.3 is not set
correctly
Regarding ANI restriction signal.

Argentine regulations since 1999 have swaped SIG_12 with SIG_15 in order
To restrict ANI presentation to the user.
I dont know if it has been patched in later releases of mfcr2 lib but
this
Simple patch works for me in mfcr2.c:

/*
 * patch de Oscar Carriles
 *
   mfcr2->group_i_end_of_ANI_restricted = R2_SIGI_12;*/
mfcr2->group_i_end_of_ANI_restricted = R2_SIGI_15;
mfcr2->group_i_end_of_ANI = R2_SIGI_12;

Thanks to Steve UnderWood for his excelent work!




Ing. Oscar Andrés Carriles

Director de Ingeniería

Eolix Technologies S.R.L.

Tel 54 11 50 32 33 52 ext. 2002

www.eolix.com.ar

 

AVISO LEGAL: Esta información es privada y confidencial y está dirigida
únicamente a su destinatario. Si usted no es el destinatario original de
este mensaje y por este medio pudo acceder a dicha información por favor
elimine el mensaje. La distribución o copia de este mensaje está
estrictamente prohibida. Esta comunicación es sólo para propósitos de
información y no debe ser considerada como propuesta, aceptación ni como
una declaración de voluntad oficial de Eolix Technologies. La
transmisión de E-mails no garantiza que el correo electrónico sea seguro
o libre de error. Por consiguiente, no manifestamos que esta información
sea completa o precisa. Toda información está sujeta a alterarse sin
previo aviso


-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Steve
Underwood
Enviado el: Lunes, 24 de Julio de 2006 07:33 p.m.
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [asterisk-users] Clocking Multiple T1 Cards

Andrew Kohlsmith wrote:

>On Monday 24 July 2006 12:11, Shaw Terwilliger wrote:
>  
>
>>Thank you; this is the kind of information I was looking for.  The
wiki
>>and other documents told me exactly what the configuration options
did,
>>but I didn't know what kind of timing configuration was right for
>>multiple cards.
>>
>>
>
>Essentially the timing is ONLY for the hardware on the card.  The
Digium cards 
>use a quad framer chip (maybe a dual for the TE210 but I don't think
so) and 
>it's a hardware limitation of the framer that all spans must share the
same 
>clock source.  Sangoma's cards use individual framers and don't have
this 
>limitation.  (essentially I think it was a cost/space tradeoff.)
>
>Once the data is on the PCI bus, the clock source is irrelevant.
They're all 
>close enough that it doesn't matter anymore.
>
This statement is very very wrong. The timing matters enormously. If the

timing doesn't match, there will be frame slips, and things like modems 
will not work. The snag is, right now neither Asterisk or the cards it 
uses have the ability to lock their clocks together.

>  Those framers want exact 
>lock-step timing though, which is why your clocking settings are so
very 
>important, and why with telephony in general it is crucial to think
about 
>your clocking before throwing hardware at a solution.
>  
>
Steve

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


-- 
No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.1.394 / Virus Database: 268.10.4/396 - Release Date:
24/07/2006
 

No virus found in this outgoing message.
Checked by AVG Free Edition. 
Version: 7.5.472 / Virus Database: 269.8.13/842 - Release Date:
09/06/2007 10:46 a.m.
 

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] R2/MFC Configuration.

[Moises Silva]

Fernando: In the following URL you can find some sample files that I
have created. Actually they are mi configuration files for some test
server I use.

http://phpmexic.u33.0web-hosting.com/wordpress/misc/miscfiles.tar.bz2

It includes:

testcall.c (small change to the original to receive the configuration
file as argument)

testcall.conf (sample configuration for testcall)

unicall.conf (sample for MFCR2 in Mexico)

zaptel.conf (SPAN 1 configured for MFCR2, SPAN2 configured for HDLC networking)

Additionaly, if you speak spanish, or at least you are capable of
basic understanding (Im able to read docs in portuguese, so I guess
you can read spanish) here is a document I wrote for troubleshooting
in MFCR2.

http://phpmexic.u33.0web-hosting.com/wordpress/misc/mfcr2-asterisk-unicall.pdf

2 weeks ago I gave consultory to jefferson networks, via SSH, in
Brazil. We solved the problem (his tormenta card was the problem).

If you are interested please email me off-list to give you my quote.

Best Regards

On 5/19/06, Fernando Lujan <[EMAIL PROTECTED]> wrote:

Moises Silva wrote:
> Fernando: There are few or no people that will give you an Answer with
> that information. The list is usually for people that already have
> tried something and is experimienting some kind of specific problem.
> Your question seems like "ahhh it does not work, help me!". As far as
> I can see you have 2 options. Please search in google for information
> about how to configure, try something and then come back with a more
> meaningfull question, or 2, hire some Asterisk consultant to make it
> work.
>
> Do you know Unicall?
>
> please check http://www.voip-info.org/wiki/index.php?page=Asterisk+MFC+R2

Hi Moises,


I found this url later. I already download and install following the
instructions in the URL above. The problem is that I can use the
testcall program.

I don't know how to set up a configuration file. :( I try to create a
testcall.conf file without success. So I stopped.

Thanks in advance.

Fernando Lujan
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users




--
"Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org";
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] R2/MFC Configuration.

[Fernando Lujan]

Moises Silva wrote:

Fernando: There are few or no people that will give you an Answer with
that information. The list is usually for people that already have
tried something and is experimienting some kind of specific problem.
Your question seems like "ahhh it does not work, help me!". As far as
I can see you have 2 options. Please search in google for information
about how to configure, try something and then come back with a more
meaningfull question, or 2, hire some Asterisk consultant to make it
work.

Do you know Unicall?

please check http://www.voip-info.org/wiki/index.php?page=Asterisk+MFC+R2


Hi Moises,


I found this url later. I already download and install following the 
instructions in the URL above. The problem is that I can use the 
testcall program.


I don't know how to set up a configuration file. :( I try to create a 
testcall.conf file without success. So I stopped.


Thanks in advance.

Fernando Lujan
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] R2/MFC Configuration.

[Moises Silva]

Fernando: There are few or no people that will give you an Answer with
that information. The list is usually for people that already have
tried something and is experimienting some kind of specific problem.
Your question seems like "ahhh it does not work, help me!". As far as
I can see you have 2 options. Please search in google for information
about how to configure, try something and then come back with a more
meaningfull question, or 2, hire some Asterisk consultant to make it
work.

Do you know Unicall?

please check http://www.voip-info.org/wiki/index.php?page=Asterisk+MFC+R2

Regards

On 5/18/06, Fernando Lujan <[EMAIL PROTECTED]> wrote:

I'm trying to put asterisk working with a proprietary pbx system.

I'm doing it using a T1 crossover cable. The pbx system uses the R2/MFC
specification. And the don't inform if it uses cas, ccs, ami or hbd3.

My digium card is flashing a red light.

How can I put this working with the R2/MFC system?

Thanks.

Fernando Lujan
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users




--
"Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org";
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] R2/MFC Configuration.

[Fernando Lujan]

I'm trying to put asterisk working with a proprietary pbx system.

I'm doing it using a T1 crossover cable. The pbx system uses the R2/MFC 
specification. And the don't inform if it uses cas, ccs, ami or hbd3.


My digium card is flashing a red light.

How can I put this working with the R2/MFC system?

Thanks.

Fernando Lujan
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] R2 protocol error

[Dennis Nacino]

Hi,


Thanks a lot, guys! The problem is now fixed by updating the libmfcr2-0.0.3 to 
pre9 and setting
the span timing correctly.


Dennis




__
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] R2 protocol error

[Oscar Carriles]

I have one Sangoma A101 in Argentina running MFCR2 with no synching
problem
What about the timing source?
I believe you have to set

span=1,1,0,cas,hdb3

to be the slave of the source timing clock from your carrier
hope it helps

Ing. Oscar Andrés Carriles


-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Moises
Silva
Enviado el: Martes, 04 de Abril de 2006 08:49 p.m.
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [Asterisk-Users] R2 protocol error

a mirror to soft-switch can be found at:
http://zarzamora.com.mx/mirror/www.soft-switch.org/

regards

On 4/3/06, Steve Underwood <[EMAIL PROTECTED]> wrote:
> Hi Dennis,
>
> Update to libmfcr2-0.0.3 pre9. I made a slip in pre8. Sorry.
>
> Steve
>
>
> Dennis Nacino wrote:
>
> >Hi,
> >
> >I have three R2 installation on different carriers, all shows the
same inconsistency at varying
> >degree. But, on most test calls we made, it reaches T3. The worst
part of these, the carrier
> >claims that it's my R2 box that is not responding in time. Please,
check the attached file and
> >take note of the timestamp, you'll find that in some call, it already
contradict what the carrier
> >claims but they too have logs to counter my claim. So, I hope people,
please give me a good
> >insight and direction to resolve this problem.
> >
> >I have the following for my R2 box:
> >unicall-0.0.3pre8
> > libmfcr2-0.0.3
> > libsupertone-0.0.2
> > libunicall-0.0.3
> >spandsp-0.0.2pre25
> >
> >asterisk-1.2.6
> >zaptel-1.2.5
> >wanpipe-2.3.3-2
> >
> >2.6.11-1.1369_FC4smp
> >sangoma A101
> >
> >in my zaptel.conf I got the following:
> >
> >span=1,0,0,cas,hdb3
> >loadzone = us
> >defaultzone=us
> >cas=1-15:1101
> >cas=17-31:1101
> >
> >in my unicall.conf I got these lines:
> >
> >[channels]
> >context=default
> >usecallerid=yes
> >hidecallerid=no
> >callwaitingcallerid=yes
> >threewaycalling=yes
> >transfer=yes
> >cancallforward=yes
> >callreturn=yes
> >echocancel=yes
> >echocancelwhenbridged=yes
> >rxgain=0.0
> >txgain=0.0
> >group=1
> >callgroup=1
> >pickupgroup=1
> >immediate=no
> >supertones=ph
> >loglevel=255
> >protocolclass=mfcr2
> >protocolvariant=ph,10,3,12
> >protocolend=co
> >group = 1
> >channel => 1-15
> >channel => 17-31
> >
> >
> >Thanks a lot.
> >
> >Dennis
> >
> >
> >
> >
> >
> >
> >__
> >Do You Yahoo!?
> >Tired of spam?  Yahoo! Mail has the best spam protection around
> >http://mail.yahoo.com
> >
>
>---
-
> >
> >Apr  3 11:34:54 WARNING[17334]: chan_unicall.c:627 unicall_report:
MFC/R2 UniCall/1  <- 0001  [1/   1/Idle  /Idle ]
> >Apr  3 11:34:54 WARNING[17334]: chan_unicall.c:627 unicall_report:
MFC/R2 UniCall/1 Detected
> >Apr  3 11:34:54 WARNING[17334]: chan_unicall.c:627 unicall_report:
MFC/R2 UniCall/1 Making a new call with CRN 32769
> >Apr  3 11:34:54 WARNING[17334]: chan_unicall.c:627 unicall_report:
MFC/R2 UniCall/1 1101  ->  [2/   2/Idle  /Idle ]
> >Apr  3 11:34:54 WARNING[17334]: chan_unicall.c:2644 handle_uc_event:
Unicall/1 event Detected
> >Apr  3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report:
MFC/R2 UniCall/1  <- 3 on  [2/   2/Seize ack /Seize ack]
> >Apr  3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report:
MFC/R2 UniCall/1 1 on  ->  [2/   2/Seize ack /Seize ack]
> >Apr  3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report:
MFC/R2 UniCall/1  <- 3 off [2/   2/Group A   /DNIS request ]
> >Apr  3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report:
MFC/R2 UniCall/1 1 off ->  [2/   2/Group A   /DNIS request ]
> >Apr  3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report:
MFC/R2 UniCall/1  <- 3 on  [2/   2/Group A   /DNIS request ]
> >Apr  3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report:
MFC/R2 UniCall/1 1 on  ->  [2/   2/Group A   /DNIS request ]
> >Apr  3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report:
MFC/R2 UniCall/1  <- 3 off [2/   2/Group A   /DNIS request ]
> >Apr  3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report:
MFC/R2 UniCall/1 1 off ->  [2/   2/Group A   /DNIS request ]
> >Apr  3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_rep

Re: [Asterisk-Users] R2 protocol error

[Moises Silva]

a mirror to soft-switch can be found at:
http://zarzamora.com.mx/mirror/www.soft-switch.org/

regards

On 4/3/06, Steve Underwood <[EMAIL PROTECTED]> wrote:
> Hi Dennis,
>
> Update to libmfcr2-0.0.3 pre9. I made a slip in pre8. Sorry.
>
> Steve
>
>
> Dennis Nacino wrote:
>
> >Hi,
> >
> >I have three R2 installation on different carriers, all shows the same 
> >inconsistency at varying
> >degree. But, on most test calls we made, it reaches T3. The worst part of 
> >these, the carrier
> >claims that it's my R2 box that is not responding in time. Please, check the 
> >attached file and
> >take note of the timestamp, you'll find that in some call, it already 
> >contradict what the carrier
> >claims but they too have logs to counter my claim. So, I hope people, please 
> >give me a good
> >insight and direction to resolve this problem.
> >
> >I have the following for my R2 box:
> >unicall-0.0.3pre8
> > libmfcr2-0.0.3
> > libsupertone-0.0.2
> > libunicall-0.0.3
> >spandsp-0.0.2pre25
> >
> >asterisk-1.2.6
> >zaptel-1.2.5
> >wanpipe-2.3.3-2
> >
> >2.6.11-1.1369_FC4smp
> >sangoma A101
> >
> >in my zaptel.conf I got the following:
> >
> >span=1,0,0,cas,hdb3
> >loadzone = us
> >defaultzone=us
> >cas=1-15:1101
> >cas=17-31:1101
> >
> >in my unicall.conf I got these lines:
> >
> >[channels]
> >context=default
> >usecallerid=yes
> >hidecallerid=no
> >callwaitingcallerid=yes
> >threewaycalling=yes
> >transfer=yes
> >cancallforward=yes
> >callreturn=yes
> >echocancel=yes
> >echocancelwhenbridged=yes
> >rxgain=0.0
> >txgain=0.0
> >group=1
> >callgroup=1
> >pickupgroup=1
> >immediate=no
> >supertones=ph
> >loglevel=255
> >protocolclass=mfcr2
> >protocolvariant=ph,10,3,12
> >protocolend=co
> >group = 1
> >channel => 1-15
> >channel => 17-31
> >
> >
> >Thanks a lot.
> >
> >Dennis
> >
> >
> >
> >
> >
> >
> >__
> >Do You Yahoo!?
> >Tired of spam?  Yahoo! Mail has the best spam protection around
> >http://mail.yahoo.com
> >
> >
> >
> >Apr  3 11:34:54 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
> >UniCall/1  <- 0001  [1/   1/Idle  /Idle ]
> >Apr  3 11:34:54 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
> >UniCall/1 Detected
> >Apr  3 11:34:54 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
> >UniCall/1 Making a new call with CRN 32769
> >Apr  3 11:34:54 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
> >UniCall/1 1101  ->  [2/   2/Idle  /Idle ]
> >Apr  3 11:34:54 WARNING[17334]: chan_unicall.c:2644 handle_uc_event: 
> >Unicall/1 event Detected
> >Apr  3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
> >UniCall/1  <- 3 on  [2/   2/Seize ack /Seize ack]
> >Apr  3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
> >UniCall/1 1 on  ->  [2/   2/Seize ack /Seize ack]
> >Apr  3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
> >UniCall/1  <- 3 off [2/   2/Group A   /DNIS request ]
> >Apr  3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
> >UniCall/1 1 off ->  [2/   2/Group A   /DNIS request ]
> >Apr  3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
> >UniCall/1  <- 3 on  [2/   2/Group A   /DNIS request ]
> >Apr  3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
> >UniCall/1 1 on  ->  [2/   2/Group A   /DNIS request ]
> >Apr  3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
> >UniCall/1  <- 3 off [2/   2/Group A   /DNIS request ]
> >Apr  3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
> >UniCall/1 1 off ->  [2/   2/Group A   /DNIS request ]
> >Apr  3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
> >UniCall/1  <- 3 on  [2/   2/Group A   /DNIS request ]
> >Apr  3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
> >UniCall/1 5 on  ->  [2/   2/Group A   /DNIS request ]
> >Apr  3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
> >UniCall/1  <- 1 on  [2/   2/Group A   /Category req ]
> >Apr  3 11:35:20 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
> >UniCall/1  <- 1 off [2/   2/Group A   /ANI request  ]
> >Apr  3 11:35:20 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
> >UniCall/1 5 off ->  [2/   2/Group A   /ANI request  ]
> >Apr  3 11:35:20 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
> >UniCall/1 R2 prot. err. [2/   2/Group A   /ANI request  ] cause 
> >32771 - T3 timed out
> >Apr  3 11:35:20 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
> >UniCall/1 1001  ->  [1/   1/Idle  /Idle ]
> >Apr  3 11:35:20 WARNING[17334]: chan_unicall.c:2644 handle_uc_event: 
> >Unic

[Asterisk-Users] R2 protocol error

[Dennis Nacino]

Hi MM and Steve,

I still got the same problem when I changed the span configuration setting into

span=1,1,0,cas,hdb3

Where can I get the pre9? Is there something wrong with www.soft-switch.org 
site? It seems
unreachable.

Thanks again.


Dennis






__
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] R2 protocol error

[Steve Underwood]

Hi Dennis,

Update to libmfcr2-0.0.3 pre9. I made a slip in pre8. Sorry.

Steve


Dennis Nacino wrote:


Hi,

I have three R2 installation on different carriers, all shows the same 
inconsistency at varying
degree. But, on most test calls we made, it reaches T3. The worst part of 
these, the carrier
claims that it's my R2 box that is not responding in time. Please, check the 
attached file and
take note of the timestamp, you'll find that in some call, it already 
contradict what the carrier
claims but they too have logs to counter my claim. So, I hope people, please 
give me a good
insight and direction to resolve this problem.

I have the following for my R2 box:
unicall-0.0.3pre8
libmfcr2-0.0.3
libsupertone-0.0.2
libunicall-0.0.3
spandsp-0.0.2pre25

asterisk-1.2.6
zaptel-1.2.5
wanpipe-2.3.3-2

2.6.11-1.1369_FC4smp
sangoma A101

in my zaptel.conf I got the following:

span=1,0,0,cas,hdb3
loadzone = us
defaultzone=us
cas=1-15:1101
cas=17-31:1101

in my unicall.conf I got these lines:

[channels]
context=default
usecallerid=yes
hidecallerid=no
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
supertones=ph
loglevel=255
protocolclass=mfcr2
protocolvariant=ph,10,3,12
protocolend=co
group = 1
channel => 1-15
channel => 17-31


Thanks a lot.

Dennis






__
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 




Apr  3 11:34:54 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1  <- 0001  [1/   1/Idle  /Idle ]
Apr  3 11:34:54 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1 Detected
Apr  3 11:34:54 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1 Making a new call with CRN 32769
Apr  3 11:34:54 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1 1101  ->  [2/   2/Idle  /Idle ]
Apr  3 11:34:54 WARNING[17334]: chan_unicall.c:2644 handle_uc_event: Unicall/1 
event Detected
Apr  3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1  <- 3 on  [2/   2/Seize ack /Seize ack]
Apr  3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1 1 on  ->  [2/   2/Seize ack /Seize ack]
Apr  3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1  <- 3 off [2/   2/Group A   /DNIS request ]
Apr  3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1 1 off ->  [2/   2/Group A   /DNIS request ]
Apr  3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1  <- 3 on  [2/   2/Group A   /DNIS request ]
Apr  3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1 1 on  ->  [2/   2/Group A   /DNIS request ]
Apr  3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1  <- 3 off [2/   2/Group A   /DNIS request ]
Apr  3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1 1 off ->  [2/   2/Group A   /DNIS request ]
Apr  3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1  <- 3 on  [2/   2/Group A   /DNIS request ]
Apr  3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1 5 on  ->  [2/   2/Group A   /DNIS request ]
Apr  3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1  <- 1 on  [2/   2/Group A   /Category req ]
Apr  3 11:35:20 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1  <- 1 off [2/   2/Group A   /ANI request  ]
Apr  3 11:35:20 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1 5 off ->  [2/   2/Group A   /ANI request  ]
Apr  3 11:35:20 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1 R2 prot. err. [2/   2/Group A   /ANI request  ] cause 32771 - 
T3 timed out
Apr  3 11:35:20 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1 1001  ->  [1/   1/Idle  /Idle ]
Apr  3 11:35:20 WARNING[17334]: chan_unicall.c:2644 handle_uc_event: Unicall/1 
event Protocol failure
   -- Unicall/1 protocol error. Cause 32771
Apr  3 11:35:20 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1 Channel echo cancel
Apr  3 11:35:20 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1  <- 1001  [1/   1/Idle  /Idle ]
Apr  3 11:35:20 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1 1001  ->  [1/   1/Idle  /Idle ]

Re: [Asterisk-Users] R2 protocol error

[Melcon Moraes]

Why you are not using "span=1,1,0,cas,hdb3" ?

Since the other end is the Telco, they will provide you the source
timing sync.

Try upgrade your sangoma drivers and make sure the TE_CLOCK line inside
/etc/wanpipe/wanpipe1.conf says "NORMAL".

[]'s
MM

 -Original Message-
From:   Dennis Nacino <[EMAIL PROTECTED]>
To: asterisk-users@lists.digium.com
Cc: 
Sent:  Mon, 3 Apr 2006 02:10:10 -0700 (PDT)
Delivered:  Mon,  03 Apr 2006 03:13:17 
Subject:[Asterisk-Users] R2 protocol error

Hi,

I have three R2 installation on different carriers, all shows the same 
inconsistency at varying
degree. But, on most test calls we made, it reaches T3. The worst part of 
these, the carrier
claims that it's my R2 box that is not responding in time. Please, check the 
attached file and
take note of the timestamp, you'll find that in some call, it already 
contradict what the carrier
claims but they too have logs to counter my claim. So, I hope people, please 
give me a good
insight and direction to resolve this problem.

I have the following for my R2 box:
unicall-0.0.3pre8
 libmfcr2-0.0.3
 libsupertone-0.0.2
 libunicall-0.0.3
spandsp-0.0.2pre25

asterisk-1.2.6
zaptel-1.2.5
wanpipe-2.3.3-2

2.6.11-1.1369_FC4smp
sangoma A101

in my zaptel.conf I got the following:

span=1,0,0,cas,hdb3
loadzone = us
defaultzone=us
cas=1-15:1101
cas=17-31:1101

in my unicall.conf I got these lines:

[channels]
context=default
usecallerid=yes
hidecallerid=no
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
supertones=ph
loglevel=255
protocolclass=mfcr2
protocolvariant=ph,10,3,12
protocolend=co
group = 1
channel => 1-15
channel => 17-31


Thanks a lot.

Dennis






__
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 
E-mail classificado pelo Identificador de Spam Inteligente Terra.
Para alterar a categoria classificada, visite
http://mail.terra.com.br/protected_email/imail/imail.cgi?+_u=levelz&_l=1,1144055598.8638.2003.almora.terra.com.br,19124,20031127114101,20031127114101

 --Original Message Ends--

-- 
Melcon Moraes <[EMAIL PROTECTED]>

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] R2 protocol error

[Dennis Nacino]

Hi,

I have three R2 installation on different carriers, all shows the same 
inconsistency at varying
degree. But, on most test calls we made, it reaches T3. The worst part of 
these, the carrier
claims that it's my R2 box that is not responding in time. Please, check the 
attached file and
take note of the timestamp, you'll find that in some call, it already 
contradict what the carrier
claims but they too have logs to counter my claim. So, I hope people, please 
give me a good
insight and direction to resolve this problem.

I have the following for my R2 box:
unicall-0.0.3pre8
 libmfcr2-0.0.3
 libsupertone-0.0.2
 libunicall-0.0.3
spandsp-0.0.2pre25

asterisk-1.2.6
zaptel-1.2.5
wanpipe-2.3.3-2

2.6.11-1.1369_FC4smp
sangoma A101

in my zaptel.conf I got the following:

span=1,0,0,cas,hdb3
loadzone = us
defaultzone=us
cas=1-15:1101
cas=17-31:1101

in my unicall.conf I got these lines:

[channels]
context=default
usecallerid=yes
hidecallerid=no
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
supertones=ph
loglevel=255
protocolclass=mfcr2
protocolvariant=ph,10,3,12
protocolend=co
group = 1
channel => 1-15
channel => 17-31


Thanks a lot.

Dennis






__
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com Apr  3 11:34:54 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1  <- 0001  [1/   1/Idle  /Idle ]
Apr  3 11:34:54 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1 Detected
Apr  3 11:34:54 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1 Making a new call with CRN 32769
Apr  3 11:34:54 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1 1101  ->  [2/   2/Idle  /Idle ]
Apr  3 11:34:54 WARNING[17334]: chan_unicall.c:2644 handle_uc_event: Unicall/1 
event Detected
Apr  3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1  <- 3 on  [2/   2/Seize ack /Seize ack]
Apr  3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1 1 on  ->  [2/   2/Seize ack /Seize ack]
Apr  3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1  <- 3 off [2/   2/Group A   /DNIS request ]
Apr  3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1 1 off ->  [2/   2/Group A   /DNIS request ]
Apr  3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1  <- 3 on  [2/   2/Group A   /DNIS request ]
Apr  3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1 1 on  ->  [2/   2/Group A   /DNIS request ]
Apr  3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1  <- 3 off [2/   2/Group A   /DNIS request ]
Apr  3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1 1 off ->  [2/   2/Group A   /DNIS request ]
Apr  3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1  <- 3 on  [2/   2/Group A   /DNIS request ]
Apr  3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1 5 on  ->  [2/   2/Group A   /DNIS request ]
Apr  3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1  <- 1 on  [2/   2/Group A   /Category req ]
Apr  3 11:35:20 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1  <- 1 off [2/   2/Group A   /ANI request  ]
Apr  3 11:35:20 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1 5 off ->  [2/   2/Group A   /ANI request  ]
Apr  3 11:35:20 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1 R2 prot. err. [2/   2/Group A   /ANI request  ] cause 32771 - 
T3 timed out
Apr  3 11:35:20 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1 1001  ->  [1/   1/Idle  /Idle ]
Apr  3 11:35:20 WARNING[17334]: chan_unicall.c:2644 handle_uc_event: Unicall/1 
event Protocol failure
-- Unicall/1 protocol error. Cause 32771
Apr  3 11:35:20 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1 Channel echo cancel
Apr  3 11:35:20 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1  <- 1001  [1/   1/Idle  /Idle ]
Apr  3 11:35:20 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1 1001  ->  [1/   1/Idle  /Idle ]
==
Apr  3 11:41:49 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/2  <- 0001  [1/   1/Idle  /Idle ]
Apr  3 11:4

Re: [Asterisk-Users] R2 implementation problem

[Alberto Sagredo]

I hope this link will help you.

http://zarzamora.com.mx/asterisk/17

Regards


Manuel Marin Garcia escribió:

I have a TE110P connected to a Telmex E1 circuit with R2 signaling.

Asterisk version= 1.0.10
Zaptel= 1.0.1
Spandsp=0.0.3pre6
Unicall= 0.0.3pre8

*zaptel.conf
span=1,1,0,cas,hdb3
cas=1-15:1101
cas=17-31:1101
dchan=16
loadzone = us
defaultzone=us

*unicall.conf
immediate=no
loglevel=255
protocolclass=mfcr2
protocolvariant=mx,10,4
protocolend=cpe
group = 1
context = telmex
channel => 1-15
channel => 17-31
*
*chan_unicall is compiled without any problems and when asterisk 
starts I see all channels in idle state. The problem is that I am 
unable to make or receive calls. When there is an incoming call I see 
the following


Incoming Call (I receive 10 ANI digits and 4 DNIS digits) I suppose to 
receive 0875 for DNIS


Jan 31 10:08:48 WARNING[4293]: chan_unicall.c:704 unicall_report: 
MFC/R2 UniCall/25  <- 0001  [1/   1/Idle  /Idle ]
Jan 31 10:08:48 WARNING[4293]: chan_unicall.c:704 unicall_report: 
MFC/R2 UniCall/25 Detected
Jan 31 10:08:48 WARNING[4293]: chan_unicall.c:704 unicall_report: 
MFC/R2 UniCall/25 Making a new call with CRN 32769
Jan 31 10:08:48 WARNING[4293]: chan_unicall.c:704 unicall_report: 
MFC/R2 UniCall/25 1101  ->  [2/   2/Idle  /Idle ]
Jan 31 10:08:48 WARNING[4293]: chan_unicall.c:2865 handle_uc_event: 
Unicall/25 event Detected
Jan 31 10:08:48 WARNING[4293]: chan_unicall.c:704 unicall_report: 
MFC/R2 UniCall/25  <- 0 on  [2/   2/Seize ack /Seize ack]
Jan 31 10:08:48 WARNING[4293]: chan_unicall.c:704 unicall_report: 
MFC/R2 UniCall/25 1 on  ->  [2/   2/Seize ack /Seize ack]


There is a 9 or 10 seconds pause and the I receive the following 
message. There is a busy tone in the caller side after the message


Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:704 unicall_report: 
MFC/R2 UniCall/27  <- 1001  [2/   2/Group A   /DNIS request ]
Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:704 unicall_report: 
MFC/R2 UniCall/27 Far end disconnected(cause=Normal, unspecified cause 
[31]) - state 0x2
Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:704 unicall_report: 
MFC/R2 UniCall/27  <- 0 off [2/ 800/Clear fwd /DNIS request ]
Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:704 unicall_report: 
MFC/R2 UniCall/27 1 off ->  [2/ 800/Clear fwd /DNIS request ]
Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:704 unicall_report: 
MFC/R2 UniCall/27 R2 prot. err. [2/ 800/Clear fwd /DNIS 
request ] cause 32774 - Invalid state
Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:704 unicall_report: 
MFC/R2 UniCall/27 1001  ->  [1/   1/Idle  /Idle ]
Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:2865 handle_uc_event: 
Unicall/27 event Far end disconnected
Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:3198 handle_uc_event: 
CRN 32769 - far disconnected cause=Normal, unspecified cause [31]
Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:704 unicall_report: 
MFC/R2 UniCall/27 Call control(6)
Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:704 unicall_report: 
MFC/R2 UniCall/27 Drop call(cause=Normal Clearing [16])
Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:704 unicall_report: 
MFC/R2 UniCall/27 1101  ->  [1/   1/Idle  /Idle ]
Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:2865 handle_uc_event: 
Unicall/27 event Protocol failure

  -- Unicall/27 protocol error. Cause 32774
Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:704 unicall_report: 
MFC/R2 UniCall/28  <- 0001  [1/   1/Idle  /Idle ]
Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:704 unicall_report: 
MFC/R2 UniCall/28 Detected
Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:704 unicall_report: 
MFC/R2 UniCall/28 Making a new call with CRN 32769
Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:704 unicall_report: 
MFC/R2 UniCall/28 1101  ->  [2/   2/Idle  /Idle ]
Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:2865 handle_uc_event: 
Unicall/28 event Detected
Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:704 unicall_report: 
MFC/R2 UniCall/28  <- 0 on  [2/   2/Seize ack /Seize ack]
Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:704 unicall_report: 
MFC/R2 UniCall/28 1 on  ->  [2/   2/Seize ack /Seize ack]
Jan 31 10:10:35 WARNING[4311]: chan_unicall.c:704 unicall_report: 
MFC/R2 UniCall/28  <- 0 off [2/   2/Group A   /DNIS request ]
Jan 31 10:10:35 WARNING[4311]: chan_unicall.c:704 unicall_report: 
MFC/R2 UniCall/28 1 off ->  [2/   2/Group A   /DNIS request ]
Jan 31 10:10:35 WARNING[4311]: chan_unicall.c:704 unicall_report: 
MFC/R2 UniCall/28  <- 8 on  [2/   2/Group A   /DNIS request ]
Jan 31 10:10:35 WARNING[4311]: chan_unicall.c:704 unicall_report: 
MFC/R2 UniCall/28 1 on  ->  [2/   2/Group A   /DNIS request ]


*NOTE*:
I Already tried changing values for DNIS and ANIS and same prob

[Asterisk-Users] R2 implementation problem

[Manuel Marin Garcia]

I have a TE110P connected to a Telmex E1 circuit with R2 signaling.

Asterisk version= 1.0.10
Zaptel= 1.0.1
Spandsp=0.0.3pre6
Unicall= 0.0.3pre8

*zaptel.conf
span=1,1,0,cas,hdb3
cas=1-15:1101
cas=17-31:1101
dchan=16
loadzone = us
defaultzone=us

*unicall.conf
immediate=no
loglevel=255
protocolclass=mfcr2
protocolvariant=mx,10,4
protocolend=cpe
group = 1
context = telmex
channel => 1-15
channel => 17-31
*
*chan_unicall is compiled without any problems and when asterisk starts 
I see all channels in idle state. The problem is that I am unable to 
make or receive calls. When there is an incoming call I see the following


Incoming Call (I receive 10 ANI digits and 4 DNIS digits) I suppose to 
receive 0875 for DNIS


Jan 31 10:08:48 WARNING[4293]: chan_unicall.c:704 unicall_report: MFC/R2 
UniCall/25  <- 0001  [1/   1/Idle  /Idle ]
Jan 31 10:08:48 WARNING[4293]: chan_unicall.c:704 unicall_report: MFC/R2 
UniCall/25 Detected
Jan 31 10:08:48 WARNING[4293]: chan_unicall.c:704 unicall_report: MFC/R2 
UniCall/25 Making a new call with CRN 32769
Jan 31 10:08:48 WARNING[4293]: chan_unicall.c:704 unicall_report: MFC/R2 
UniCall/25 1101  ->  [2/   2/Idle  /Idle ]
Jan 31 10:08:48 WARNING[4293]: chan_unicall.c:2865 handle_uc_event: 
Unicall/25 event Detected
Jan 31 10:08:48 WARNING[4293]: chan_unicall.c:704 unicall_report: MFC/R2 
UniCall/25  <- 0 on  [2/   2/Seize ack /Seize ack]
Jan 31 10:08:48 WARNING[4293]: chan_unicall.c:704 unicall_report: MFC/R2 
UniCall/25 1 on  ->  [2/   2/Seize ack /Seize ack]


There is a 9 or 10 seconds pause and the I receive the following 
message. There is a busy tone in the caller side after the message


Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:704 unicall_report: MFC/R2 
UniCall/27  <- 1001  [2/   2/Group A   /DNIS request ]
Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:704 unicall_report: MFC/R2 
UniCall/27 Far end disconnected(cause=Normal, unspecified cause [31]) - 
state 0x2
Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:704 unicall_report: MFC/R2 
UniCall/27  <- 0 off [2/ 800/Clear fwd /DNIS request ]
Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:704 unicall_report: MFC/R2 
UniCall/27 1 off ->  [2/ 800/Clear fwd /DNIS request ]
Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:704 unicall_report: MFC/R2 
UniCall/27 R2 prot. err. [2/ 800/Clear fwd /DNIS request ] cause 
32774 - Invalid state
Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:704 unicall_report: MFC/R2 
UniCall/27 1001  ->  [1/   1/Idle  /Idle ]
Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:2865 handle_uc_event: 
Unicall/27 event Far end disconnected
Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:3198 handle_uc_event: CRN 
32769 - far disconnected cause=Normal, unspecified cause [31]
Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:704 unicall_report: MFC/R2 
UniCall/27 Call control(6)
Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:704 unicall_report: MFC/R2 
UniCall/27 Drop call(cause=Normal Clearing [16])
Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:704 unicall_report: MFC/R2 
UniCall/27 1101  ->  [1/   1/Idle  /Idle ]
Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:2865 handle_uc_event: 
Unicall/27 event Protocol failure

  -- Unicall/27 protocol error. Cause 32774
Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:704 unicall_report: MFC/R2 
UniCall/28  <- 0001  [1/   1/Idle  /Idle ]
Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:704 unicall_report: MFC/R2 
UniCall/28 Detected
Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:704 unicall_report: MFC/R2 
UniCall/28 Making a new call with CRN 32769
Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:704 unicall_report: MFC/R2 
UniCall/28 1101  ->  [2/   2/Idle  /Idle ]
Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:2865 handle_uc_event: 
Unicall/28 event Detected
Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:704 unicall_report: MFC/R2 
UniCall/28  <- 0 on  [2/   2/Seize ack /Seize ack]
Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:704 unicall_report: MFC/R2 
UniCall/28 1 on  ->  [2/   2/Seize ack /Seize ack]
Jan 31 10:10:35 WARNING[4311]: chan_unicall.c:704 unicall_report: MFC/R2 
UniCall/28  <- 0 off [2/   2/Group A   /DNIS request ]
Jan 31 10:10:35 WARNING[4311]: chan_unicall.c:704 unicall_report: MFC/R2 
UniCall/28 1 off ->  [2/   2/Group A   /DNIS request ]
Jan 31 10:10:35 WARNING[4311]: chan_unicall.c:704 unicall_report: MFC/R2 
UniCall/28  <- 8 on  [2/   2/Group A   /DNIS request ]
Jan 31 10:10:35 WARNING[4311]: chan_unicall.c:704 unicall_report: MFC/R2 
UniCall/28 1 on  ->  [2/   2/Group A   /DNIS request ]


*NOTE*:
I Already tried changing values for DNIS and ANIS and same problem
I tried with other versions of spandsp and unicall and same problem
The same occurs in Asterisk version 1.2.

Re: [Asterisk-Users] R2 variations by country

[Steve Underwood]

Noah Kamrat wrote:

Is anybody aware of a list or specifications of R2 signaling by 
country?  There are many variations and I am looking for documentation 
that will save us having to analyze the calls for each country to 
determine how the bits are sent for each variant.  We manufacture PRI 
hardware and are looking to add R2 as a feature.


 


Thanks,

Noah

The only centralised sources of such information I know of are your 
competitor's manuals. That's the main source of information I used when 
developing the R2 support for *. :-)


For more detailed information you need to look at each country's local 
specification. Some are easier to obtain than others.


Steve

___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] R2 variations by country

[Noah Kamrat]

Is anybody aware of a list or specifications of R2 signaling
by country?  There are many variations and I am looking for documentation that
will save us having to analyze the calls for each country to determine how the
bits are sent for each variant.  We manufacture PRI hardware and are looking to
add R2 as a feature. 

 

Thanks,

Noah

 

 






___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] R2-Digital (Q.421)

[Jesus Mogollon]

That's exactly how they explained it works. The DTMF is only to provide
DNIS and using the register signalling. When I run testcall, I get the
handshake but when testcall sends the first digit, the remote equipment
doesn't recognize it (because it's expecting a DTMF signal) and then it
times out. I don't want to buy the Cisco for this :( especially when I
wanted it to be a full open source solution. What would it take for you
to implement it? 2005/11/10, Julio Arruda <[EMAIL PROTECTED]>:
Just to clarify this in my head :-)..So...They are using E1/R2 (the R2 Digital)in fact, for all the line signaling  (nothing unusual)The register signaling, that I was under impression would be MF in each
timeslot (MFC5C in .br, not sure if the same in others), is in fact DTMFin this trunk, and only to provide DNIS ?(in Brazil R2, the register signaling has some collect call informationand etc).Steve Underwood wrote:
> Hi,>> I tried hunting for a little more info. I think all that happens with> this is they use the Q.421 spec for handling the ABCD bits, and then> simply send the DNIS through as DTMF after the seize if acknowledged.
> That means they loose some of the functionality of real R2 signalling -> e.g. no busy, NU, or congestion detection. It wouldn't take a lot of> work to implement that.>> Regards,> Steve
>>> Steve Underwood wrote:>>> Hi Jesus, The Cisco kit, and one or two other products, offer an R2 digital>> using DTMF mode, but this is the first time I have heard of it being
>> used. The spec for this is definitely not Q.421. That spec does not>> mention DTMF at all. R2 using DTMF doesn't appear to be in the ITU>> specs, as far as I can tell. Without a spec, or any equipment to play
>> with, there isn't a lot I can do right now. Steve>> Jesus Mogollon wrote:> Hi Steve:>>  Thanks for your help. I really appreciate it..
>>   My provider is CANTV in Venezuela. There's a venezuelan variant in>>> the code and I'm using that. Incoming works perfectly, outgoing is>>> not working. I'm being told that incoming is MFCR2 but outgoing is
>>> R2-Digital with DNIS DTMF. There is a Cisco router working and it's>>> using the following:>> r2-digital-dtmf-dnis R2 ITU Q421 DTMF tone signaling with DNIS>>>
>> What's the equivalent in libmfcr2 and Unicall?>> Again, thank you for your help and your code!>> Jesus Mogollon>>>
>>> 2005/11/5, Steve Underwood <[EMAIL PROTECTED]>>> [EMAIL PROTECTED]>>:>> Hi Jesus,
>> FX is not a variant of R2. It is a completely different signalling>>> protocol. This means your service provider is using R2 for some of>>> your>>> channels, and providing all your incoming calls on those channels.
>>> It is>>> use FX signalling for other channels, and you must make your>>> outgoing>>> calls there. Someone else told be about a similar configuration. I
>>> think>>> they were able to use chan_zap for the other channels, and make>>> use of>>> its FX signalling features. I am not sure how that works, as FX
>>> signalling over E1s is far from standardised.>> Regards,>>> Steve> Jesus Mogollon wrote:>>>
>>> > Steve:>>>  >   That's exactly what I'm using. Incoming calls work like a>>> charm but>>> > when I try calling I get a protocol error. My provider says
>>> that for>>> > outgoing I need to use fx signalling. I see that in unicall.conf>>> > there's such a thing as protocolvariant=fx but if I uncomment that>>> > line, unicall gives me an error. Any ideas? Thanks for your
>>> help...>>>  > 2005/11/4, Steve Underwood <[EMAIL PROTECTED]>>> 
[EMAIL PROTECTED] > [EMAIL PROTECTED] [EMAIL PROTECTED]>>>:>>> >
>>> > Jesus Mogollon wrote:>>>  > >Does anyone know how to make this work with Asterisk?>>> (R2-Digital>>>
> >(Q.421)) I have MFCR2 configured but
I'm told that outgoing>>> calls are>>>
> >to use Q421 R2 Digital signalling. Any
help is appreciated.>>> >  > >Jesus Mogollon>>> >  > 
> See http://www.soft-switch.org
 Steve>>> ___
>> --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list>> Asterisk-Users@lists.digium.com
>> http://lists.digium.com/mailman/listinfo/asterisk-users>> To UNSUBSCRIBE or update options visit:>>   
http://lists.digium.com/mailman/listinfo/asterisk-users ___> --Bandwidth and Colocation sponsored by Easynews.com
 -->> Asterisk-Users mailing list> Asterisk-Users@lists.digium.com> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:>   http://lists.digium.com/mailman/listinfo/asterisk-users___
--Bandwidth and Colocation sponsored by Easynews.com --Ast

Re: [Asterisk-Users] R2-Digital (Q.421)

[Steve Underwood]

This seems to be what Cisco have implemented as r2-digital-dtmf-dnis. 
Cisco have quite a few other combinations of strange R2 related options. 
I can't imagine they are all really used. It seems this one is, though, 
in Venezuela


Regards,
Steve


Julio Arruda wrote:


Just to clarify this in my head :-)..

So...
They are using E1/R2 (the R2 Digital)in fact, for all the line 
signaling  (nothing unusual)
The register signaling, that I was under impression would be MF in 
each timeslot (MFC5C in .br, not sure if the same in others), is in 
fact DTMF in this trunk, and only to provide DNIS ?
(in Brazil R2, the register signaling has some collect call 
information and etc).


Steve Underwood wrote:


Hi,

I tried hunting for a little more info. I think all that happens with 
this is they use the Q.421 spec for handling the ABCD bits, and then 
simply send the DNIS through as DTMF after the seize if acknowledged. 
That means they loose some of the functionality of real R2 signalling 
- e.g. no busy, NU, or congestion detection. It wouldn't take a lot 
of work to implement that.


Regards,
Steve


Steve Underwood wrote:


Hi Jesus,

The Cisco kit, and one or two other products, offer an R2 digital 
using DTMF mode, but this is the first time I have heard of it being 
used. The spec for this is definitely not Q.421. That spec does not 
mention DTMF at all. R2 using DTMF doesn't appear to be in the ITU 
specs, as far as I can tell. Without a spec, or any equipment to 
play with, there isn't a lot I can do right now.


Steve


Jesus Mogollon wrote:


Hi Steve:

 Thanks for your help. I really appreciate it..

  My provider is CANTV in Venezuela. There's a venezuelan variant 
in the code and I'm using that. Incoming works perfectly, outgoing 
is not working. I'm being told that incoming is MFCR2 but outgoing 
is R2-Digital with DNIS DTMF. There is a Cisco router working and 
it's using the following:


r2-digital-dtmf-dnis R2 ITU Q421 DTMF tone signaling with DNIS


What's the equivalent in libmfcr2 and Unicall?

Again, thank you for your help and your code!

Jesus Mogollon

2005/11/5, Steve Underwood <[EMAIL PROTECTED] 
>:


Hi Jesus,

FX is not a variant of R2. It is a completely different signalling
protocol. This means your service provider is using R2 for some of
your
channels, and providing all your incoming calls on those channels.
It is
use FX signalling for other channels, and you must make your 
outgoing

calls there. Someone else told be about a similar configuration. I
think
they were able to use chan_zap for the other channels, and make
use of
its FX signalling features. I am not sure how that works, as FX
signalling over E1s is far from standardised.

Regards,
Steve


Jesus Mogollon wrote:

> Steve:
>
>   That's exactly what I'm using. Incoming calls work like a
charm but
> when I try calling I get a protocol error. My provider says 
that for

> outgoing I need to use fx signalling. I see that in unicall.conf
> there's such a thing as protocolvariant=fx but if I uncomment 
that
> line, unicall gives me an error. Any ideas? Thanks for your 
help...

>
> 2005/11/4, Steve Underwood <[EMAIL PROTECTED]

> mailto:[EMAIL PROTECTED]>>>:
>
> Jesus Mogollon wrote:
>
> >Does anyone know how to make this work with Asterisk?
(R2-Digital
> >(Q.421)) I have MFCR2 configured but I'm told that outgoing
calls are
> >to use Q421 R2 Digital signalling. Any help is appreciated.
> >
> >Jesus Mogollon
> >
> >
> See http://www.soft-switch.org 
>
> Steve



___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] R2-Digital (Q.421)

[Julio Arruda]

Just to clarify this in my head :-)..

So...
They are using E1/R2 (the R2 Digital)in fact, for all the line signaling 
 (nothing unusual)
The register signaling, that I was under impression would be MF in each 
timeslot (MFC5C in .br, not sure if the same in others), is in fact DTMF 
in this trunk, and only to provide DNIS ?
(in Brazil R2, the register signaling has some collect call information 
and etc).


Steve Underwood wrote:

Hi,

I tried hunting for a little more info. I think all that happens with 
this is they use the Q.421 spec for handling the ABCD bits, and then 
simply send the DNIS through as DTMF after the seize if acknowledged. 
That means they loose some of the functionality of real R2 signalling - 
e.g. no busy, NU, or congestion detection. It wouldn't take a lot of 
work to implement that.


Regards,
Steve


Steve Underwood wrote:


Hi Jesus,

The Cisco kit, and one or two other products, offer an R2 digital 
using DTMF mode, but this is the first time I have heard of it being 
used. The spec for this is definitely not Q.421. That spec does not 
mention DTMF at all. R2 using DTMF doesn't appear to be in the ITU 
specs, as far as I can tell. Without a spec, or any equipment to play 
with, there isn't a lot I can do right now.


Steve


Jesus Mogollon wrote:


Hi Steve:

 Thanks for your help. I really appreciate it..

  My provider is CANTV in Venezuela. There's a venezuelan variant in 
the code and I'm using that. Incoming works perfectly, outgoing is 
not working. I'm being told that incoming is MFCR2 but outgoing is 
R2-Digital with DNIS DTMF. There is a Cisco router working and it's 
using the following:


r2-digital-dtmf-dnis R2 ITU Q421 DTMF tone signaling with DNIS


What's the equivalent in libmfcr2 and Unicall?

Again, thank you for your help and your code!

Jesus Mogollon

2005/11/5, Steve Underwood <[EMAIL PROTECTED] 
>:


Hi Jesus,

FX is not a variant of R2. It is a completely different signalling
protocol. This means your service provider is using R2 for some of
your
channels, and providing all your incoming calls on those channels.
It is
use FX signalling for other channels, and you must make your 
outgoing

calls there. Someone else told be about a similar configuration. I
think
they were able to use chan_zap for the other channels, and make
use of
its FX signalling features. I am not sure how that works, as FX
signalling over E1s is far from standardised.

Regards,
Steve


Jesus Mogollon wrote:

> Steve:
>
>   That's exactly what I'm using. Incoming calls work like a
charm but
> when I try calling I get a protocol error. My provider says 
that for

> outgoing I need to use fx signalling. I see that in unicall.conf
> there's such a thing as protocolvariant=fx but if I uncomment that
> line, unicall gives me an error. Any ideas? Thanks for your 
help...

>
> 2005/11/4, Steve Underwood <[EMAIL PROTECTED]

> mailto:[EMAIL PROTECTED]>>>:
>
> Jesus Mogollon wrote:
>
> >Does anyone know how to make this work with Asterisk?
(R2-Digital
> >(Q.421)) I have MFCR2 configured but I'm told that outgoing
calls are
> >to use Q421 R2 Digital signalling. Any help is appreciated.
> >
> >Jesus Mogollon
> >
> >
> See http://www.soft-switch.org 
>
> Steve



___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users



___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] R2-Digital (Q.421)

[Steve Underwood]

Hi,

I tried hunting for a little more info. I think all that happens with 
this is they use the Q.421 spec for handling the ABCD bits, and then 
simply send the DNIS through as DTMF after the seize if acknowledged. 
That means they loose some of the functionality of real R2 signalling - 
e.g. no busy, NU, or congestion detection. It wouldn't take a lot of 
work to implement that.


Regards,
Steve


Steve Underwood wrote:


Hi Jesus,

The Cisco kit, and one or two other products, offer an R2 digital 
using DTMF mode, but this is the first time I have heard of it being 
used. The spec for this is definitely not Q.421. That spec does not 
mention DTMF at all. R2 using DTMF doesn't appear to be in the ITU 
specs, as far as I can tell. Without a spec, or any equipment to play 
with, there isn't a lot I can do right now.


Steve


Jesus Mogollon wrote:


Hi Steve:

 Thanks for your help. I really appreciate it..

  My provider is CANTV in Venezuela. There's a venezuelan variant in 
the code and I'm using that. Incoming works perfectly, outgoing is 
not working. I'm being told that incoming is MFCR2 but outgoing is 
R2-Digital with DNIS DTMF. There is a Cisco router working and it's 
using the following:


r2-digital-dtmf-dnis R2 ITU Q421 DTMF tone signaling with DNIS


What's the equivalent in libmfcr2 and Unicall?

Again, thank you for your help and your code!

Jesus Mogollon

2005/11/5, Steve Underwood <[EMAIL PROTECTED] 
>:


Hi Jesus,

FX is not a variant of R2. It is a completely different signalling
protocol. This means your service provider is using R2 for some of
your
channels, and providing all your incoming calls on those channels.
It is
use FX signalling for other channels, and you must make your 
outgoing

calls there. Someone else told be about a similar configuration. I
think
they were able to use chan_zap for the other channels, and make
use of
its FX signalling features. I am not sure how that works, as FX
signalling over E1s is far from standardised.

Regards,
Steve


Jesus Mogollon wrote:

> Steve:
>
>   That's exactly what I'm using. Incoming calls work like a
charm but
> when I try calling I get a protocol error. My provider says 
that for

> outgoing I need to use fx signalling. I see that in unicall.conf
> there's such a thing as protocolvariant=fx but if I uncomment that
> line, unicall gives me an error. Any ideas? Thanks for your 
help...

>
> 2005/11/4, Steve Underwood <[EMAIL PROTECTED]

> mailto:[EMAIL PROTECTED]>>>:
>
> Jesus Mogollon wrote:
>
> >Does anyone know how to make this work with Asterisk?
(R2-Digital
> >(Q.421)) I have MFCR2 configured but I'm told that outgoing
calls are
> >to use Q421 R2 Digital signalling. Any help is appreciated.
> >
> >Jesus Mogollon
> >
> >
> See http://www.soft-switch.org 
>
> Steve



___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users



___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] R2-Digital (Q.421)

[Steve Underwood]

Hi Jesus,

The Cisco kit, and one or two other products, offer an R2 digital using 
DTMF mode, but this is the first time I have heard of it being used. The 
spec for this is definitely not Q.421. That spec does not mention DTMF 
at all. R2 using DTMF doesn't appear to be in the ITU specs, as far as I 
can tell. Without a spec, or any equipment to play with, there isn't a 
lot I can do right now.


Steve


Jesus Mogollon wrote:


Hi Steve:

 Thanks for your help. I really appreciate it..

  My provider is CANTV in Venezuela. There's a venezuelan variant in 
the code and I'm using that. Incoming works perfectly, outgoing is not 
working. I'm being told that incoming is MFCR2 but outgoing is 
R2-Digital with DNIS DTMF. There is a Cisco router working and it's 
using the following:


r2-digital-dtmf-dnis R2 ITU Q421 DTMF tone signaling with DNIS


What's the equivalent in libmfcr2 and Unicall?

Again, thank you for your help and your code!

Jesus Mogollon

2005/11/5, Steve Underwood <[EMAIL PROTECTED] 
>:


Hi Jesus,

FX is not a variant of R2. It is a completely different signalling
protocol. This means your service provider is using R2 for some of
your
channels, and providing all your incoming calls on those channels.
It is
use FX signalling for other channels, and you must make your outgoing
calls there. Someone else told be about a similar configuration. I
think
they were able to use chan_zap for the other channels, and make
use of
its FX signalling features. I am not sure how that works, as FX
signalling over E1s is far from standardised.

Regards,
Steve


Jesus Mogollon wrote:

> Steve:
>
>   That's exactly what I'm using. Incoming calls work like a
charm but
> when I try calling I get a protocol error. My provider says that for
> outgoing I need to use fx signalling. I see that in unicall.conf
> there's such a thing as protocolvariant=fx but if I uncomment that
> line, unicall gives me an error. Any ideas? Thanks for your help...
>
> 2005/11/4, Steve Underwood <[EMAIL PROTECTED]

> mailto:[EMAIL PROTECTED]>>>:
>
> Jesus Mogollon wrote:
>
> >Does anyone know how to make this work with Asterisk?
(R2-Digital
> >(Q.421)) I have MFCR2 configured but I'm told that outgoing
calls are
> >to use Q421 R2 Digital signalling. Any help is appreciated.
> >
> >Jesus Mogollon
> >
> >
> See http://www.soft-switch.org 
>
> Steve



___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] R2-Digital (Q.421)

[Julio Arruda]

Jesus Mogollon wrote:

Hi Steve:

Thanks for your help. I really appreciate it..

My provider is CANTV in Venezuela. There's a venezuelan variant in the code
and I'm using that. Incoming works perfectly, outgoing is not working. I'm
being told that incoming is MFCR2 but outgoing is R2-Digital with DNIS DTMF.
There is a Cisco router working and it's using the following:

r2-digital-dtmf-dnis R2 ITU Q421 DTMF tone signaling with DNIS


I may be a little lost here, but isn't MFC5C E1/R2 just plain R2-Digital ?
The fact that you have DNIS or not doesn't change it (in brazil was kind 
of weird, you had to have the proper number of digits or something like 
that), but from my memory:


E1/R2 - Line Signalling - TS 16, and ABCD bits
Register Signaling - Inband - with DTMF fwd and back (Compelled I thing 
was the term used).





What's the equivalent in libmfcr2 and Unicall?

Again, thank you for your help and your code!

Jesus Mogollon

2005/11/5, Steve Underwood <[EMAIL PROTECTED]>:


Hi Jesus,

FX is not a variant of R2. It is a completely different signalling
protocol. This means your service provider is using R2 for some of your
channels, and providing all your incoming calls on those channels. It is
use FX signalling for other channels, and you must make your outgoing
calls there. Someone else told be about a similar configuration. I think
they were able to use chan_zap for the other channels, and make use of
its FX signalling features. I am not sure how that works, as FX
signalling over E1s is far from standardised.

Regards,
Steve


Jesus Mogollon wrote:



Steve:

That's exactly what I'm using. Incoming calls work like a charm but
when I try calling I get a protocol error. My provider says that for
outgoing I need to use fx signalling. I see that in unicall.conf
there's such a thing as protocolvariant=fx but if I uncomment that
line, unicall gives me an error. Any ideas? Thanks for your help...

2005/11/4, Steve Underwood <[EMAIL PROTECTED]
>:

Jesus Mogollon wrote:



Does anyone know how to make this work with Asterisk? (R2-Digital
(Q.421)) I have MFCR2 configured but I'm told that outgoing calls are
to use Q421 R2 Digital signalling. Any help is appreciated.

Jesus Mogollon




See http://www.soft-switch.org

Steve



___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users







___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] R2-Digital (Q.421)

[Jesus Mogollon]

Hi Steve:

 Thanks for your help. I really appreciate it..

  My provider is CANTV in Venezuela. There's a venezuelan variant
in the code and I'm using that. Incoming works perfectly, outgoing is
not working. I'm being told that incoming is MFCR2 but outgoing is
R2-Digital with DNIS DTMF. There is a Cisco router working and it's
using the following:

r2-digital-dtmf-dnis R2 ITU Q421 DTMF tone signaling with DNIS 


What's the equivalent in libmfcr2 and Unicall?

Again, thank you for your help and your code!

Jesus Mogollon2005/11/5, Steve Underwood <[EMAIL PROTECTED]>:
Hi Jesus,FX is not a variant of R2. It is a completely different signallingprotocol. This means your service provider is using R2 for some of yourchannels, and providing all your incoming calls on those channels. It is
use FX signalling for other channels, and you must make your outgoingcalls there. Someone else told be about a similar configuration. I thinkthey were able to use chan_zap for the other channels, and make use of
its FX signalling features. I am not sure how that works, as FXsignalling over E1s is far from standardised.Regards,SteveJesus Mogollon wrote:> Steve:>>   That's exactly what I'm using. Incoming calls work like a charm but
> when I try calling I get a protocol error. My provider says that for> outgoing I need to use fx signalling. I see that in unicall.conf> there's such a thing as protocolvariant=fx but if I uncomment that
> line, unicall gives me an error. Any ideas? Thanks for your help...>> 2005/11/4, Steve Underwood <[EMAIL PROTECTED]> 
[EMAIL PROTECTED]>>:>> Jesus Mogollon wrote:>> >Does anyone know how to make this work with Asterisk? (R2-Digital> >(Q.421)) I have MFCR2 configured but I'm told that outgoing calls are
> >to use Q421 R2 Digital signalling. Any help is appreciated.> >> >Jesus Mogollon> >> >> See http://www.soft-switch.org
>> Steve>___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list
Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] R2-Digital (Q.421)

[Steve Underwood]

Hi Jesus,

FX is not a variant of R2. It is a completely different signalling 
protocol. This means your service provider is using R2 for some of your 
channels, and providing all your incoming calls on those channels. It is 
use FX signalling for other channels, and you must make your outgoing 
calls there. Someone else told be about a similar configuration. I think 
they were able to use chan_zap for the other channels, and make use of 
its FX signalling features. I am not sure how that works, as FX 
signalling over E1s is far from standardised.


Regards,
Steve


Jesus Mogollon wrote:


Steve:

  That's exactly what I'm using. Incoming calls work like a charm but 
when I try calling I get a protocol error. My provider says that for 
outgoing I need to use fx signalling. I see that in unicall.conf 
there's such a thing as protocolvariant=fx but if I uncomment that 
line, unicall gives me an error. Any ideas? Thanks for your help...


2005/11/4, Steve Underwood <[EMAIL PROTECTED] 
>:


Jesus Mogollon wrote:

>Does anyone know how to make this work with Asterisk? (R2-Digital
>(Q.421)) I have MFCR2 configured but I'm told that outgoing calls are
>to use Q421 R2 Digital signalling. Any help is appreciated.
>
>Jesus Mogollon
>
>
See http://www.soft-switch.org

Steve



___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] R2-Digital (Q.421)

[Jesus Mogollon]

Steve:

  That's exactly what I'm using. Incoming calls work like a charm
but when I try calling I get a protocol error. My provider says that
for outgoing I need to use fx signalling. I see that in unicall.conf
there's such a thing as protocolvariant=fx but if I uncomment that
line, unicall gives me an error. Any ideas? Thanks for your help...2005/11/4, Steve Underwood <[EMAIL PROTECTED]>:
Jesus Mogollon wrote:>Does anyone know how to make this work with Asterisk? (R2-Digital>(Q.421)) I have MFCR2 configured but I'm told that outgoing calls are>to use Q421 R2 Digital signalling. Any help is appreciated.
>>Jesus Mogollon>>See http://www.soft-switch.orgSteve___--Bandwidth and Colocation sponsored by 
Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] R2-Digital (Q.421)

[Steve Underwood]

Jesus Mogollon wrote:


Does anyone know how to make this work with Asterisk? (R2-Digital
(Q.421)) I have MFCR2 configured but I'm told that outgoing calls are
to use Q421 R2 Digital signalling. Any help is appreciated.

Jesus Mogollon
 


See http://www.soft-switch.org

Steve

___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] R2-Digital (Q.421)

[Jesus Mogollon]

Does anyone know how to make this work with Asterisk? (R2-Digital
(Q.421)) I have MFCR2 configured but I'm told that outgoing calls are
to use Q421 R2 Digital signalling. Any help is appreciated.

Jesus Mogollon
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] R2...channel type UniCall

[Steve Underwood]

Hi,

Philip Fleischer wrote:


I have done the mfc r2 setup as described in the wiki but
when I try a call

Oct 21 10:02:24 WARNING[14127]: channel.c:1913 ast_request: No channel 
type registered for 'UniCall'
Oct 21 10:02:24 NOTICE[14127]: app_dial.c:764 dial_exec: Unable to 
create channel of type 'UniCall'

 == Everyone is busy/congested at this time


I have a TE210p with an E1 crossover cable connecting the two ports.  
(Tests

out OK doing PRI ISDN or simple channelized E1 to E1.)
I have green lights on the card with ztcfg reporting all the channels.


In extensions.conf I use

exten => _981X.,1,Dial(UniCall/1/${EXTEN:3},20)
exten => _982X.,1,Dial(UniCall/35/${EXTEN:3},20)

and the unicall.conf contains standard stuff and:

loglevel=255
protocolclass=mfcr2
protocolvariant=cn,20,7
protocolend=co
group = 1
;  1st E1
channel => 1-15
channel => 17-31

protocolend=cpe
group = 2
; 2nd E1
channel => 31-46
channel => 48-62

How does the unicall channel get registered?

If you want to follow the instructions on the wiki, maybe you should ask 
the wiki to sort it out for you. :-)


Steve

___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] R2...channel type UniCall

[Philip Fleischer]

I have done the mfc r2 setup as described in the wiki but
when I try a call

Oct 21 10:02:24 WARNING[14127]: channel.c:1913 ast_request: No channel type 
registered for 'UniCall'
Oct 21 10:02:24 NOTICE[14127]: app_dial.c:764 dial_exec: Unable to create 
channel of type 'UniCall'

 == Everyone is busy/congested at this time


I have a TE210p with an E1 crossover cable connecting the two ports.  (Tests
out OK doing PRI ISDN or simple channelized E1 to E1.)
I have green lights on the card with ztcfg reporting all the channels.


In extensions.conf I use

exten => _981X.,1,Dial(UniCall/1/${EXTEN:3},20)
exten => _982X.,1,Dial(UniCall/35/${EXTEN:3},20)

and the unicall.conf contains standard stuff and:

loglevel=255
protocolclass=mfcr2
protocolvariant=cn,20,7
protocolend=co
group = 1
;  1st E1
channel => 1-15
channel => 17-31

protocolend=cpe
group = 2
; 2nd E1
channel => 31-46
channel => 48-62

How does the unicall channel get registered?


___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] R2 Digital

[Virmones P.T Miranda]

HI, plese I have try download files from www.opencall.org and
www.soft-switch.org and receive unknow host , somebody have others sites to
get files for asterisk to work with R2 Digital.


Thanks a lot.

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] r2 signalling in east europe

[Steve Underwood]

Hi Terje,
The only East European country my R2 software currently allows for is 
teh Czech Republic, since that is the only place I could find 
information for. If you have information about the protocol used in 
other countries, support should be easy to add.

Regards,
Steve
Terje Myhre wrote:
Hello,
We’re planning to use Digium cards for eastern european r2 signalling.
However, we would like to have a few references on the possibility to 
realise the signaling.

Please, can anyone tell me whether they have had any success in this, 
and if there are any special “hook-ups” to look out for ?

Br,
Terje Myhre
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] r2 signalling in east europe

[Terje Myhre]

Hello, 

 

We’re planning to use Digium cards for eastern
european r2 signalling.  

 

However, we would like to have a few references on the
possibility to realise the signaling. 

 

Please, can anyone tell me whether they have had any success
in this, and if there are any special “hook-ups” to look out for ? 

 

Br, 


Terje Myhre 






___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] R2 in Bolivia

[Jorge Verastegui Gallardo]

These are log of incoming calls

Jan 26 18:17:50 WARNING[8573]: chan_unicall.c:703 unicall_error: UniCall: 
2005/01/26 22:17:50 mfcr2 MFC/R2 call control(1)
Jan 26 18:17:50 WARNING[8573]: chan_unicall.c:703 unicall_error: UniCall: 
2005/01/26 22:17:50 mfcr2 MFC/R2 make call
Jan 26 18:17:50 WARNING[8573]: chan_unicall.c:703 unicall_error: UniCall: 
2005/01/26 22:17:50 mfcr2 Making a new call with CRN 32769
Jan 26 18:17:50 WARNING[8573]: chan_unicall.c:703 unicall_error: UniCall: 
2005/01/26 22:17:50 mfcr2 Tx bits 0x1   [1/   1/  0/  0]
-- Called g3/70513933
Jan 26 18:17:50 WARNING[8573]: chan_unicall.c:2848 handle_uc_event: UC event 
Dialing - 0x9841460
Jan 26 18:17:50 WARNING[8573]: chan_unicall.c:703 unicall_error: UniCall: 
2005/01/26 22:17:50 mfcr2 Rx bits 0xD   [1/  40/201/  0]
Jan 26 18:17:50 WARNING[8573]: chan_unicall.c:703 unicall_error: UniCall: 
2005/01/26 22:17:50 mfcr2 Tx tone 7 on  [2/  40/202/  0]
Jan 26 18:17:50 WARNING[8573]: chan_unicall.c:703 unicall_error: UniCall: 
2005/01/26 22:17:50 mfcr2 Rx tone 5 on  [2/  40/202/201]
Jan 26 18:17:50 WARNING[8573]: chan_unicall.c:703 unicall_error: UniCall: 
2005/01/26 22:17:50 mfcr2 Tx tone 7 off [2/  40/202/201]
Jan 26 18:17:50 WARNING[8573]: chan_unicall.c:703 unicall_error: UniCall: 
2005/01/26 22:17:50 mfcr2 Rx tone 5 off [2/  40/202/201]
Jan 26 18:17:50 WARNING[8573]: chan_unicall.c:703 unicall_error: UniCall: 
2005/01/26 22:17:50 mfcr2 Tx tone E on  [2/  40/202/201]
Jan 26 18:17:50 WARNING[8573]: chan_unicall.c:703 unicall_error: UniCall: 
2005/01/26 22:17:50 mfcr2 Rx tone 5 on  [2/  40/202/207]
Jan 26 18:17:50 WARNING[8573]: chan_unicall.c:703 unicall_error: UniCall: 
2005/01/26 22:17:50 mfcr2 Tx tone E off [2/  40/202/207]
Jan 26 18:17:51 WARNING[8573]: chan_unicall.c:703 unicall_error: UniCall: 
2005/01/26 22:17:51 mfcr2 Rx tone 5 off [2/  40/202/207]
Jan 26 18:17:51 WARNING[8573]: chan_unicall.c:703 unicall_error: UniCall: 
2005/01/26 22:17:51 mfcr2 Tx tone E on  [2/  40/202/207]
Jan 26 18:17:51 WARNING[8573]: chan_unicall.c:703 unicall_error: UniCall: 
2005/01/26 22:17:51 mfcr2 Rx tone 4 on  [2/  40/202/207]
Jan 26 18:17:51 WARNING[8573]: chan_unicall.c:703 unicall_error: UniCall: 
2005/01/26 22:17:51 mfcr2 Tx tone E off [2/  40/202/207]
Jan 26 18:17:51 WARNING[8573]: chan_unicall.c:703 unicall_error: UniCall: 
2005/01/26 22:17:51 mfcr2 Rx tone 4 off [2/  40/202/207]
Jan 26 18:17:51 WARNING[8573]: chan_unicall.c:703 unicall_error: UniCall: 
2005/01/26 22:17:51 mfcr2 Far end disconnected - state 0x40
Jan 26 18:17:51 WARNING[8573]: chan_unicall.c:2848 handle_uc_event: UC event 
Far end disconnected - 0x9841460
Jan 26 18:17:51 WARNING[8573]: chan_unicall.c:3181 handle_uc_event: CRN 32769 - 
far disconnect cause 42
-- Channel 0 got hangup
-- UniCall/1-1 is circuit-busy
Jan 26 18:17:51 WARNING[8573]: chan_unicall.c:703 unicall_error: UniCall: 
2005/01/26 22:17:51 mfcr2 MFC/R2 call control(6)
Jan 26 18:17:51 WARNING[8573]: chan_unicall.c:703 unicall_error: UniCall: 
2005/01/26 22:17:51 mfcr2 MFC/R2 drop call(cause=16)
Jan 26 18:17:51 WARNING[8573]: chan_unicall.c:703 unicall_error: UniCall: 
2005/01/26 22:17:51 mfcr2 Clearing fwd
Jan 26 18:17:51 WARNING[8573]: chan_unicall.c:703 unicall_error: UniCall: 
2005/01/26 22:17:51 mfcr2 Tx bits 0x9   [2/ 800/209/207]
-- Hungup 'UniCall/1-1'
  == Everyone is busy/congested at this time
Jan 26 18:17:51 WARNING[8573]: chan_unicall.c:703 unicall_error: UniCall: 
2005/01/26 22:17:51 mfcr2 Rx bits 0x9   [1/ 800/211/  0]
Jan 26 18:17:51 WARNING[8573]: chan_unicall.c:703 unicall_error: UniCall: 
2005/01/26 22:17:51 mfcr2 Call disconnected - state 0x800
Jan 26 18:17:51 WARNING[8573]: chan_unicall.c:2848 handle_uc_event: UC event 
Drop call - 0x9841460
Jan 26 18:17:51 WARNING[8573]: chan_unicall.c:703 unicall_error: UniCall: 
2005/01/26 22:17:51 mfcr2 MFC/R2 call control(7)
Jan 26 18:17:51 WARNING[8573]: chan_unicall.c:703 unicall_error: UniCall: 
2005/01/26 22:17:51 mfcr2 MFC/R2 release call
Jan 26 18:17:51 WARNING[8573]: chan_unicall.c:703 unicall_error: UniCall: 
2005/01/26 22:17:51 mfcr2 Destroying call with CRN 32769
Jan 26 18:17:51 WARNING[8573]: chan_unicall.c:2848 handle_uc_event: UC event 
Release call - 0x9841460
-- UC channel 1 released
-- H.323 call 'ip$200.87.125.195:30008/25331' cleared, reason 7 (Remote 
user stopped calling)



On Wed, 2005-01-26 at 07:53 +0800, Steve Underwood wrote:

> Hi Jorge,
> 
> You might be the first person to try the Bolivian variant. I need more 
> information to make any sense of the problem. In 
> /etc/asterisk/unicall.conf add the line:
> 
> loglevel = 1023
> 
> and try again. You should get a much more detailed log of what happens. 
> Send that to me.
> 
> Regards,
> Steve
> 
> [EMAIL PROTECTED] wrote:
> 
> >Hi
> >I made some tests with new MFC/R2 an unicall support for asterisk
> >and now have dialing out problem using UniCall / R2.
> >This is the error report in cli>
> >
> >

Re: [Asterisk-Users] R2 in Bolivia

[Steve Underwood]

Hi Jorge,
You might be the first person to try the Bolivian variant. I need more 
information to make any sense of the problem. In 
/etc/asterisk/unicall.conf add the line:

loglevel = 1023
and try again. You should get a much more detailed log of what happens. 
Send that to me.

Regards,
Steve
[EMAIL PROTECTED] wrote:
Hi
I made some tests with new MFC/R2 an unicall support for asterisk
and now have dialing out problem using UniCall / R2.
This is the error report in cli>
UC channel 30 protocol error. Cause 32772
I hope this helps.
Thanks in advance,
Jorge
PD: de conf files
zaptel.conf
  span=1,1,1,cas,hdb3
  cas=1-15:1010
  cas=17-31:1010protocolclass=mfcr2
unicall.conf
protocolvariant=bo,20,4
protocolend=cpe
group = 3
channel => 1-15
;skip time slot 16
channel=17-31
 

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] R2 in Bolivia

[Miguel Cavazos]

What version are you using for chan_unicall?
On 25/01/2005, at 1:57 PM, [EMAIL PROTECTED] wrote:
Hi
I made some tests with new MFC/R2 an unicall support for asterisk
and now have dialing out problem using UniCall / R2.
This is the error report in cli>
 UC channel 30 protocol error. Cause 32772
I hope this helps.
Thanks in advance,
Jorge
PD: de conf files
zaptel.conf
   span=1,1,1,cas,hdb3
   cas=1-15:1010
   cas=17-31:1010protocolclass=mfcr2
unicall.conf
protocolvariant=bo,20,4
protocolend=cpe
group = 3
channel => 1-15
;skip time slot 16
channel=17-31
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

--
Saludos,
Miguel Cavazos
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] R2 in Bolivia

[jorge]

Hi
I made some tests with new MFC/R2 an unicall support for asterisk
and now have dialing out problem using UniCall / R2.
This is the error report in cli>

 UC channel 30 protocol error. Cause 32772


I hope this helps.

Thanks in advance,

Jorge


PD: de conf files
zaptel.conf

   span=1,1,1,cas,hdb3
   cas=1-15:1010
   cas=17-31:1010protocolclass=mfcr2

unicall.conf

protocolvariant=bo,20,4
protocolend=cpe
group = 3
channel => 1-15
;skip time slot 16
channel=17-31

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] R2 - Stable Asterisk

[Steve Underwood]

John Middleton wrote:
Now that I have your attention ;-)
Anyone know if a new release is planned, and if so when?
 

Release of what? R2 or a stable Asterisk? The latest update to R2 was 
15th Jan (unicall-0.0.2pre4). This version fixes a number of issues in 
previous versions. It is being used quite heavily by a couple of people. 
It still has a couple of bugs, but you may or may not see then, 
depending how you use it.

Steve
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] R2 - Stable Asterisk

[John Middleton]

Now that I have your attention ;-)
Anyone know if a new release is planned, and if so when?
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] R2/MFC Mexico FREE calls to test chan_unicall

[Gonzalo Gasca Meza]

Miguel,
Congrats, i was testing your R2/MFC link, and I was able to made lots of calls, all of them worked fine.Thanks for setting up this link.
When i hang up, there were no dead air, music on hold worked fine, when I called to a conference worked fine also, busy line Telmex recording worked also fine. Please let me know if there is anything I can help you with or if you want to test something.
Thanks again!
 
 
		Do you Yahoo!? 
Yahoo! Mail - Easier than ever with enhanced search. Learn more.___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] R2/MFC Mexico FREE calls to test chan_unicall

[Miguel Cavazos]

Yes, thats why i will do it for a very short time to do testing with 
real traffic.
On 13/01/2005, at 4:03 PM, Nathan Goodwin wrote:

Wouldn't that make routing free calls illegal as well, your still 
bypassing?

Miguel Cavazos wrote:
Thanx but that is consider in Mexico bypass and its illegal, second 
we are just doing a test with real traffic to get feedback of any 
weird thing going on. Testing Chan_unicall stability is our goal. If 
you can send alot of traffic while we are doing test i would thank 
you for that.

Till now we have only got 4 channels at most busy and we need to see 
if it will handle a full E1 to test then with 2,3 and 4 E1's

On 13/01/2005, at 12:49 PM, Nathan Goodwin wrote:
I tried to contact you off list, but your system rejected my e-mail, 
I was wondering ig you planed on selling minutes for routes into 
Mexico once you where done testing, if so, could you please contact 
me off list with your rates for Mexico City, or anyplace else in 
Mexico you service, thank you.

Nathan Goodwin
Diamondleaf LLC
Miguel Cavazos wrote:
Hi guys, I have one E1 with 30 channels in Mexico City, I guess 
that if i can fill this 30 channels with REAL traffic for 2 or 3 
days I can find new bugs on chan_unicall or I can see how stable it 
can be. Im using R2/MFC with chan_unicall the patch that Steve 
Underwood wrote.

I will let anyone make FREE LOCAL calls to Mexico City till 
saturday or maybe until monday to see how stable this can be with 
REAL traffic. Add this to your extensions.conf only gsm as a codec 
is going to be permitted.

exten => 
_,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED],90,Tt)

--
Saludos,
Miguel Cavazos
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

--
Saludos,
Miguel Cavazos
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

--
Saludos,
Miguel Cavazos
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] R2/MFC Mexico FREE calls to test chan_unicall

[Nathan Goodwin]

Wouldn't that make routing free calls illegal as well, your still bypassing?
Miguel Cavazos wrote:
Thanx but that is consider in Mexico bypass and its illegal, second we 
are just doing a test with real traffic to get feedback of any weird 
thing going on. Testing Chan_unicall stability is our goal. If you can 
send alot of traffic while we are doing test i would thank you for that.

Till now we have only got 4 channels at most busy and we need to see 
if it will handle a full E1 to test then with 2,3 and 4 E1's

On 13/01/2005, at 12:49 PM, Nathan Goodwin wrote:
I tried to contact you off list, but your system rejected my e-mail, 
I was wondering ig you planed on selling minutes for routes into 
Mexico once you where done testing, if so, could you please contact 
me off list with your rates for Mexico City, or anyplace else in 
Mexico you service, thank you.

Nathan Goodwin
Diamondleaf LLC
Miguel Cavazos wrote:
Hi guys, I have one E1 with 30 channels in Mexico City, I guess that 
if i can fill this 30 channels with REAL traffic for 2 or 3 days I 
can find new bugs on chan_unicall or I can see how stable it can be. 
Im using R2/MFC with chan_unicall the patch that Steve Underwood wrote.

I will let anyone make FREE LOCAL calls to Mexico City till saturday 
or maybe until monday to see how stable this can be with REAL 
traffic. Add this to your extensions.conf only gsm as a codec is 
going to be permitted.

exten => 
_,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED],90,Tt)

--
Saludos,
Miguel Cavazos
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

--
Saludos,
Miguel Cavazos
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] R2/MFC Mexico FREE calls to test chan_unicall

[Rene Kluwen]

I am also interested.
Pls. contact me at [EMAIL PROTECTED]

Rene Kluwen
Chimit

- Original Message -
From: "Nathan Goodwin" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Thursday, January 13, 2005 7:49 PM
Subject: Re: [Asterisk-Users] R2/MFC Mexico FREE calls to test chan_unicall


> I tried to contact you off list, but your system rejected my e-mail, I
> was wondering ig you planed on selling minutes for routes into Mexico
> once you where done testing, if so, could you please contact me off list
> with your rates for Mexico City, or anyplace else in Mexico you service,
> thank you.
>
> Nathan Goodwin
> Diamondleaf LLC
>
> Miguel Cavazos wrote:
>
> > Hi guys, I have one E1 with 30 channels in Mexico City, I guess that
> > if i can fill this 30 channels with REAL traffic for 2 or 3 days I can
> > find new bugs on chan_unicall or I can see how stable it can be. Im
> > using R2/MFC with chan_unicall the patch that Steve Underwood wrote.
> >
> > I will let anyone make FREE LOCAL calls to Mexico City till saturday
> > or maybe until monday to see how stable this can be with REAL traffic.
> > Add this to your extensions.conf only gsm as a codec is going to be
> > permitted.
> >
> > exten =>
_,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED],90,Tt)
> >
> > --
> > Saludos,
> > Miguel Cavazos
> >
> > ___
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
> ___
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] R2/MFC Mexico FREE calls to test chan_unicall

[Miguel Cavazos]

Thanx but that is consider in Mexico bypass and its illegal, second we 
are just doing a test with real traffic to get feedback of any weird 
thing going on. Testing Chan_unicall stability is our goal. If you can 
send alot of traffic while we are doing test i would thank you for 
that.

Till now we have only got 4 channels at most busy and we need to see if 
it will handle a full E1 to test then with 2,3 and 4 E1's

On 13/01/2005, at 12:49 PM, Nathan Goodwin wrote:
I tried to contact you off list, but your system rejected my e-mail, I 
was wondering ig you planed on selling minutes for routes into Mexico 
once you where done testing, if so, could you please contact me off 
list with your rates for Mexico City, or anyplace else in Mexico you 
service, thank you.

Nathan Goodwin
Diamondleaf LLC
Miguel Cavazos wrote:
Hi guys, I have one E1 with 30 channels in Mexico City, I guess that 
if i can fill this 30 channels with REAL traffic for 2 or 3 days I 
can find new bugs on chan_unicall or I can see how stable it can be. 
Im using R2/MFC with chan_unicall the patch that Steve Underwood 
wrote.

I will let anyone make FREE LOCAL calls to Mexico City till saturday 
or maybe until monday to see how stable this can be with REAL 
traffic. Add this to your extensions.conf only gsm as a codec is 
going to be permitted.

exten => 
_,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED],90,Tt)

--
Saludos,
Miguel Cavazos
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

--
Saludos,
Miguel Cavazos
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] R2/MFC Mexico FREE calls to test chan_unicall

[Nathan Goodwin]

I tried to contact you off list, but your system rejected my e-mail, I 
was wondering ig you planed on selling minutes for routes into Mexico 
once you where done testing, if so, could you please contact me off list 
with your rates for Mexico City, or anyplace else in Mexico you service, 
thank you.

Nathan Goodwin
Diamondleaf LLC
Miguel Cavazos wrote:
Hi guys, I have one E1 with 30 channels in Mexico City, I guess that 
if i can fill this 30 channels with REAL traffic for 2 or 3 days I can 
find new bugs on chan_unicall or I can see how stable it can be. Im 
using R2/MFC with chan_unicall the patch that Steve Underwood wrote.

I will let anyone make FREE LOCAL calls to Mexico City till saturday 
or maybe until monday to see how stable this can be with REAL traffic. 
Add this to your extensions.conf only gsm as a codec is going to be 
permitted.

exten => _,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED],90,Tt)
--
Saludos,
Miguel Cavazos
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] R2/MFC Mexico FREE calls to test chan_unicall

[Miguel Cavazos]

For those who doesnt have an asterisk setup or cant make it work you 
can use any iaxsoftphone and use the user guest with no password using 
codec gsm and start dialing as if you are in mexico city. We need to 
have alot of calls going! The ip for the server is 200.53.121.233

On 13/01/2005, at 11:05 AM, Greg Blakely wrote:
Works for me, too.  But I found that the Benito Juarez International 
airport was reachable by 9-011-52-5-571-3600.

 
To get this from my PBX-like setup, I have the following in 
extensions.conf:

 
exten => _901152.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:6},90,Tt)
 
and the following in iax.conf
 
disallow=all   
 allow=GSM
allow=ULAW
allow=ALAW
allow=G726
allow=ILBC
allow=LPC10
allow=SPEEX
 
(Obviously, anything below "allow=GSM" isn't necessary for this 
particular connection.)

 
 
 
 
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Don Dawson
> Sent: Thursday, January 13, 2005 9:58 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] R2/MFC Mexico FREE calls to test
> chan_unicall
>
 > I changed to line to :
> exten => _,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN},90,Tt
>
 > and it works fine.
>
 > > > On 13/01/2005, at 9:22 AM, Gary Carr wrote:
> > >
> > >> I tried to call the mexico city airport and got the following
> > >>
> > >>
> > >> -- Executing Dial("SIP/9104044010-541d",
> > >> "IAX2/[EMAIL PROTECTED]/57644910
> > >> @guest|90.Tf") in new stack
> > >>    -- Called [EMAIL PROTECTED]/57644910 @guest
> > >> Jan 13 10:20:59 WARNING[1142135600]: chan_iax2.c:5339 
socket_read:

> > >> Call rejected
> > >> by 200.53.121.233: No such context/extension
> > >>    -- Hungup 'IAX2/200.53.121.233:4569/4'
> > >>  == No one is available to answer at this time
> > >>
> > >>
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
--
Saludos,
Miguel Cavazos
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] R2/MFC Mexico FREE calls to test chan_unicall

[Greg Blakely]

Works for me, too.  But I found that the Benito Juarez
International airport was reachable by 9-011-52-5-571-3600.

 

To get this from my PBX-like setup, I have the following in
extensions.conf:

 

exten => _901152.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:6},90,Tt)

 

and the following in iax.conf

 

disallow=all   


allow=GSM

allow=ULAW

allow=ALAW

allow=G726

allow=ILBC

allow=LPC10

allow=SPEEX

 

(Obviously, anything below "allow=GSM" isn't necessary for
this particular connection.)

 

 

 

 

> -Original Message-

> From: [EMAIL PROTECTED]
[mailto:asterisk-users-

> [EMAIL PROTECTED] On Behalf Of Don Dawson

> Sent: Thursday, January 13, 2005 9:58 AM

> To: Asterisk Users Mailing List - Non-Commercial Discussion

> Subject: Re: [Asterisk-Users] R2/MFC Mexico FREE calls to test

> chan_unicall

> 

> I changed to line to :

> exten =>
_,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN},90,Tt

> 

> and it works fine.

> 

> > > On 13/01/2005, at 9:22 AM, Gary Carr wrote:

> > >

> > >> I tried to call the mexico city airport and got the following

> > >>

> > >>

> > >> -- Executing Dial("SIP/9104044010-541d",

> > >> "IAX2/[EMAIL PROTECTED]/57644910

> > >> @guest|90.Tf") in new stack

> > >>    -- Called
[EMAIL PROTECTED]/57644910 @guest

> > >> Jan 13 10:20:59 WARNING[1142135600]:
chan_iax2.c:5339 socket_read:

> > >> Call rejected

> > >> by 200.53.121.233: No such context/extension

> > >>    -- Hungup
'IAX2/200.53.121.233:4569/4'

> > >>  == No one is available to answer at this time

> > >>

> > >>






___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] R2/MFC Mexico FREE calls to test chan_unicall

[Don Dawson]

I changed to line to :
exten => _,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN},90,Tt

and it works fine.

- Original Message -
From: "Miguel Cavazos" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Thursday, January 13, 2005 9:36 AM
Subject: Re: [Asterisk-Users] R2/MFC Mexico FREE calls to test chan_unicall


>
> On 13/01/2005, at 9:35 AM, Miguel Cavazos wrote:
>
> > Really weird calls are still getting in and i just called the same
> > number as you did. I will investigate.
> >
> > here is the context on extensions.conf
> >
> > [guest]
> > exten => _,1,Dial(Unicall/g1/${EXTEN},90,Tt)
> >
> > On 13/01/2005, at 9:22 AM, Gary Carr wrote:
> >
> >> I tried to call the mexico city airport and got the following
> >>
> >>
> >> -- Executing Dial("SIP/9104044010-541d",
> >> "IAX2/[EMAIL PROTECTED]/57644910
> >> @guest|90.Tf") in new stack
> >>-- Called [EMAIL PROTECTED]/57644910 @guest
> >> Jan 13 10:20:59 WARNING[1142135600]: chan_iax2.c:5339 socket_read:
> >> Call rejected
> >> by 200.53.121.233: No such context/extension
> >>-- Hungup 'IAX2/200.53.121.233:4569/4'
> >>  == No one is available to answer at this time
> >>
> >>
> >> Regards,
> >>
> >>
> >> Gary
> >>
> >> - Original Message - From: "Miguel Cavazos"
> >> <[EMAIL PROTECTED]>
> >> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> >> 
> >> Sent: Thursday, January 13, 2005 10:13 AM
> >> Subject: Re: [Asterisk-Users] R2/MFC Mexico FREE calls to test
> >> chan_unicall
> >>
> >>
> >>> any feedback would be awsome, the idea is to fill in the 30 channels
> >>> of the E1 all at the same time and see how stable it can be
> >>>
> >>>
> >>> On 13/01/2005, at 8:28 AM, Don Dawson wrote:
> >>>
> >>>> I have an asterisk system down here in Oaxaca. I don't know anyone
> >>>> there to
> >>>> call but I can call some hotels
> >>>> in the area for possible reservations and perhaps ticket
> >>>> information for the
> >>>> theater.
> >>>>
> >>>>
> >>>> - Original Message -
> >>>> From: "Miguel Cavazos" <[EMAIL PROTECTED]>
> >>>> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> >>>> 
> >>>> Sent: Wednesday, January 12, 2005 4:22 PM
> >>>> Subject: [Asterisk-Users] R2/MFC Mexico FREE calls to test
> >>>> chan_unicall
> >>>>
> >>>>
> >>>>> Hi guys, I have one E1 with 30 channels in Mexico City, I guess
> >>>>> that if
> >>>>> i can fill this 30 channels with REAL traffic for 2 or 3 days I can
> >>>>> find new bugs on chan_unicall or I can see how stable it can be. Im
> >>>>> using R2/MFC with chan_unicall the patch that Steve Underwood
> >>>>> wrote.
> >>>>>
> >>>>> I will let anyone make FREE LOCAL calls to Mexico City till
> >>>>> saturday or
> >>>>> maybe until monday to see how stable this can be with REAL
> >>>>> traffic. Add
> >>>>> this to your extensions.conf only gsm as a codec is going to be
> >>>>> permitted.
> >>>>>
> >>>>> exten =>
> >>>>> _,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED],90,Tt)
> >>>>>
> >>>>> --
> >>>>> Saludos,
> >>>>> Miguel Cavazos
> >>>>>
> >>>>> ___
> >>>>> Asterisk-Users mailing list
> >>>>> Asterisk-Users@lists.digium.com
> >>>>> http://lists.digium.com/mailman/listinfo/asterisk-users
> >>>>> To UNSUBSCRIBE or update options visit:
> >>>>>http://lists.digium.com/mailman/listinfo/asterisk-users
> >>>>
> >>>> ___
> >>>> Asterisk-Users mailing list
> >>>> Asterisk-Users@lists.digium.com
> >>>> http://lists.digium.com/mailman/listinfo/asterisk-users
> >>>> To UNSUBSCRIBE or update options visit:
> >>>>http://lists.digium.com/mailman/listinfo/asterisk-users
> >>>>
> >>>>
> >>> --
> >>> Saludos,
> >>> Miguel Cavazos
> >>>
> >>> ___
> >>> Asterisk-Users mailing list
> >>> Asterisk-Users@lists.digium.com
> >>> http://lists.digium.com/mailman/listinfo/asterisk-users
> >>> To UNSUBSCRIBE or update options visit:
> >>>   http://lists.digium.com/mailman/listinfo/asterisk-users
> >>
> >>
> > --
> > Saludos,
> > Miguel Cavazos
> >
> >
> --
> Saludos,
> Miguel Cavazos
>
> ___
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] R2/MFC Mexico FREE calls to test chan_unicall

[Miguel Cavazos]

On 13/01/2005, at 9:35 AM, Miguel Cavazos wrote:
Really weird calls are still getting in and i just called the same 
number as you did. I will investigate.

here is the context on extensions.conf
[guest]
exten => _,1,Dial(Unicall/g1/${EXTEN},90,Tt)
On 13/01/2005, at 9:22 AM, Gary Carr wrote:
I tried to call the mexico city airport and got the following
-- Executing Dial("SIP/9104044010-541d", 
"IAX2/[EMAIL PROTECTED]/57644910
@guest|90.Tf") in new stack
   -- Called [EMAIL PROTECTED]/57644910 @guest
Jan 13 10:20:59 WARNING[1142135600]: chan_iax2.c:5339 socket_read: 
Call rejected
by 200.53.121.233: No such context/extension
   -- Hungup 'IAX2/200.53.121.233:4569/4'
 == No one is available to answer at this time

Regards,
Gary
- Original Message - From: "Miguel Cavazos" 
<[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Thursday, January 13, 2005 10:13 AM
Subject: Re: [Asterisk-Users] R2/MFC Mexico FREE calls to test 
chan_unicall


any feedback would be awsome, the idea is to fill in the 30 channels 
of the E1 all at the same time and see how stable it can be

On 13/01/2005, at 8:28 AM, Don Dawson wrote:
I have an asterisk system down here in Oaxaca. I don't know anyone 
there to
call but I can call some hotels
in the area for possible reservations and perhaps ticket 
information for the
theater.

- Original Message -
From: "Miguel Cavazos" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"

Sent: Wednesday, January 12, 2005 4:22 PM
Subject: [Asterisk-Users] R2/MFC Mexico FREE calls to test 
chan_unicall


Hi guys, I have one E1 with 30 channels in Mexico City, I guess 
that if
i can fill this 30 channels with REAL traffic for 2 or 3 days I can
find new bugs on chan_unicall or I can see how stable it can be. Im
using R2/MFC with chan_unicall the patch that Steve Underwood 
wrote.

I will let anyone make FREE LOCAL calls to Mexico City till 
saturday or
maybe until monday to see how stable this can be with REAL 
traffic. Add
this to your extensions.conf only gsm as a codec is going to be
permitted.

exten =>
_,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED],90,Tt)
--
Saludos,
Miguel Cavazos
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

--
Saludos,
Miguel Cavazos
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

--
Saludos,
Miguel Cavazos

--
Saludos,
Miguel Cavazos
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] R2/MFC Mexico FREE calls to test chan_unicall

[Gary Carr]

I tried to call the mexico city airport and got the following
-- Executing Dial("SIP/9104044010-541d", "IAX2/[EMAIL PROTECTED]/57644910
@guest|90.Tf") in new stack
   -- Called [EMAIL PROTECTED]/57644910 @guest
Jan 13 10:20:59 WARNING[1142135600]: chan_iax2.c:5339 socket_read: Call 
rejected
by 200.53.121.233: No such context/extension
   -- Hungup 'IAX2/200.53.121.233:4569/4'
 == No one is available to answer at this time

Regards,
Gary
- Original Message - 
From: "Miguel Cavazos" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Thursday, January 13, 2005 10:13 AM
Subject: Re: [Asterisk-Users] R2/MFC Mexico FREE calls to test chan_unicall


any feedback would be awsome, the idea is to fill in the 30 channels of 
the E1 all at the same time and see how stable it can be

On 13/01/2005, at 8:28 AM, Don Dawson wrote:
I have an asterisk system down here in Oaxaca. I don't know anyone there 
to
call but I can call some hotels
in the area for possible reservations and perhaps ticket information for 
the
theater.

- Original Message -
From: "Miguel Cavazos" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"

Sent: Wednesday, January 12, 2005 4:22 PM
Subject: [Asterisk-Users] R2/MFC Mexico FREE calls to test chan_unicall

Hi guys, I have one E1 with 30 channels in Mexico City, I guess that if
i can fill this 30 channels with REAL traffic for 2 or 3 days I can
find new bugs on chan_unicall or I can see how stable it can be. Im
using R2/MFC with chan_unicall the patch that Steve Underwood wrote.
I will let anyone make FREE LOCAL calls to Mexico City till saturday or
maybe until monday to see how stable this can be with REAL traffic. Add
this to your extensions.conf only gsm as a codec is going to be
permitted.
exten =>
_,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED],90,Tt)
--
Saludos,
Miguel Cavazos
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

--
Saludos,
Miguel Cavazos
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] R2/MFC Mexico FREE calls to test chan_unicall

[Miguel Cavazos]

any feedback would be awsome, the idea is to fill in the 30 channels of 
the E1 all at the same time and see how stable it can be

On 13/01/2005, at 8:28 AM, Don Dawson wrote:
I have an asterisk system down here in Oaxaca. I don't know anyone 
there to
call but I can call some hotels
in the area for possible reservations and perhaps ticket information 
for the
theater.

- Original Message -
From: "Miguel Cavazos" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"

Sent: Wednesday, January 12, 2005 4:22 PM
Subject: [Asterisk-Users] R2/MFC Mexico FREE calls to test chan_unicall

Hi guys, I have one E1 with 30 channels in Mexico City, I guess that 
if
i can fill this 30 channels with REAL traffic for 2 or 3 days I can
find new bugs on chan_unicall or I can see how stable it can be. Im
using R2/MFC with chan_unicall the patch that Steve Underwood wrote.

I will let anyone make FREE LOCAL calls to Mexico City till saturday 
or
maybe until monday to see how stable this can be with REAL traffic. 
Add
this to your extensions.conf only gsm as a codec is going to be
permitted.

exten =>
_,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED],90,Tt)
--
Saludos,
Miguel Cavazos
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

--
Saludos,
Miguel Cavazos
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] R2/MFC Mexico FREE calls to test chan_unicall

[Don Dawson]

I have an asterisk system down here in Oaxaca. I don't know anyone there to
call but I can call some hotels
in the area for possible reservations and perhaps ticket information for the
theater.


- Original Message -
From: "Miguel Cavazos" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"

Sent: Wednesday, January 12, 2005 4:22 PM
Subject: [Asterisk-Users] R2/MFC Mexico FREE calls to test chan_unicall


> Hi guys, I have one E1 with 30 channels in Mexico City, I guess that if
> i can fill this 30 channels with REAL traffic for 2 or 3 days I can
> find new bugs on chan_unicall or I can see how stable it can be. Im
> using R2/MFC with chan_unicall the patch that Steve Underwood wrote.
>
> I will let anyone make FREE LOCAL calls to Mexico City till saturday or
> maybe until monday to see how stable this can be with REAL traffic. Add
> this to your extensions.conf only gsm as a codec is going to be
> permitted.
>
> exten =>
> _,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED],90,Tt)
>
> --
> Saludos,
> Miguel Cavazos
>
> ___
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] R2/MFC Mexico FREE calls to test chan_unicall

[Brian Johnson]

Cool idea.

Unfortunately I don't know anyone in Mexico City to call



Miguel Cavazos ([EMAIL PROTECTED]) wrote:
>
> Hi guys, I have one E1 with 30 channels in Mexico City, I guess that if
> i can fill this 30 channels with REAL traffic for 2 or 3 days I can
> find new bugs on chan_unicall or I can see how stable it can be. Im
> using R2/MFC with chan_unicall the patch that Steve Underwood wrote.
>
> I will let anyone make FREE LOCAL calls to Mexico City till saturday or
> maybe until monday to see how stable this can be with REAL traffic. Add
> this to your extensions.conf only gsm as a codec is going to be
> permitted.
>
> exten =>
> _,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED],90,Tt)
>
>
> ___
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] R2/MFC Mexico FREE calls to test chan_unicall

[Miguel Cavazos]

Debian woody stable, nothing special most of the trouble are paths
On 13/01/2005, at 1:26 AM, Sam Njenga wrote:
Hi
Am setting up * with R2/MfC support but am 90% done. I seem to be 
missing
something in my setup. Can you tell me what Linux distribution and the
packages you have used to complete your setup to a working level ?

/Sam
- Original Message -
From: "Miguel Cavazos" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"

Sent: Thursday, January 13, 2005 1:22 AM
Subject: [Asterisk-Users] R2/MFC Mexico FREE calls to test chan_unicall

Hi guys, I have one E1 with 30 channels in Mexico City, I guess that 
if
i can fill this 30 channels with REAL traffic for 2 or 3 days I can
find new bugs on chan_unicall or I can see how stable it can be. Im
using R2/MFC with chan_unicall the patch that Steve Underwood wrote.

I will let anyone make FREE LOCAL calls to Mexico City till saturday 
or
maybe until monday to see how stable this can be with REAL traffic. 
Add
this to your extensions.conf only gsm as a codec is going to be
permitted.

exten =>
_,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED],90,Tt)
--
Saludos,
Miguel Cavazos
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

--
Saludos,
Miguel Cavazos
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] R2/MFC Mexico FREE calls to test chan_unicall

[Sam Njenga]

Hi
Am setting up * with R2/MfC support but am 90% done. I seem to be missing
something in my setup. Can you tell me what Linux distribution and the
packages you have used to complete your setup to a working level ?

/Sam

- Original Message - 
From: "Miguel Cavazos" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"

Sent: Thursday, January 13, 2005 1:22 AM
Subject: [Asterisk-Users] R2/MFC Mexico FREE calls to test chan_unicall


> Hi guys, I have one E1 with 30 channels in Mexico City, I guess that if
> i can fill this 30 channels with REAL traffic for 2 or 3 days I can
> find new bugs on chan_unicall or I can see how stable it can be. Im
> using R2/MFC with chan_unicall the patch that Steve Underwood wrote.
>
> I will let anyone make FREE LOCAL calls to Mexico City till saturday or
> maybe until monday to see how stable this can be with REAL traffic. Add
> this to your extensions.conf only gsm as a codec is going to be
> permitted.
>
> exten =>
> _,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED],90,Tt)
>
> --
> Saludos,
> Miguel Cavazos
>
> ___
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] R2/MFC Mexico FREE calls to test chan_unicall

[Miguel Cavazos]

Hi guys, I have one E1 with 30 channels in Mexico City, I guess that if 
i can fill this 30 channels with REAL traffic for 2 or 3 days I can 
find new bugs on chan_unicall or I can see how stable it can be. Im 
using R2/MFC with chan_unicall the patch that Steve Underwood wrote.

I will let anyone make FREE LOCAL calls to Mexico City till saturday or 
maybe until monday to see how stable this can be with REAL traffic. Add 
this to your extensions.conf only gsm as a codec is going to be 
permitted.

exten => 
_,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED],90,Tt)

--
Saludos,
Miguel Cavazos
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] R2 for Mexico?

[Carlos Chavez]

On Mon, 10 Jan 2005 22:24:33 +0300, Sam Njenga wrote
> What Linux distribution are you using ? I can help you if your using 
> redhat 9 as am 90% done with R2.( Thanks to Steve Underwood) You can 
> start here http://www.opencall.org/installing-mfcr2.html
> 
> /Sam
> 
 Right now I am using Fedora Core 1 but I plan to install new servers
using Fedora Core 3

--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] R2 for Mexico?

[Sam Njenga]

What Linux distribution are you using ? I can help you if your using redhat
9 as am 90% done with R2.( Thanks to Steve Underwood)
You can start here
http://www.opencall.org/installing-mfcr2.html

/Sam

- Original Message - 
From: "Carlos Chavez" <[EMAIL PROTECTED]>
To: "Asterisk" 
Sent: Monday, January 10, 2005 8:16 PM
Subject: [Asterisk-Users] R2 for Mexico?


>  Does anyone have a document on how to implement R2 for use in Mexico?
> What packages do I have to download and compile?
>
> --
> Carlos Chavez
> Director de Tecnología
> Telecomunicaciones Abiertas de México S.A. de C.V.
>
> ___
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] R2 for Mexico?

[Carlos Chavez]

 Does anyone have a document on how to implement R2 for use in Mexico? 
What packages do I have to download and compile?

--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] R2 support

[Bartosz Jozwiak]

Are there any signaling converters?
>From R2 to something which is supported in asterisk ?

Bart

- Original Message - 
From: "Steve Underwood" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, May 13, 2004 12:17 PM
Subject: Re: [Asterisk-Users] R2 support


> Hi,
>
> I make it compile. I just never finished it. :-)
>
> What I did to make R2 work properly was throw away the code that is now
> in CVS at Digium, and write a new implementation from scratch. I guess
> that is not quite the answer you wanted to hear. :-(
>
> Regards,
> Steve
>
>
> Jorge Verastegui wrote:
>
> >Hi
> > I know there is no support for R2, but I succcessfully compiled and ran
> >the libr2.  However, I am not able to initiate calls.
> >
> >The error I get is
> >
> >Couldn't call g3/71605538
> >-- Hungup 'Zap/32-1'
> >  == Everyone is busy at this time
> >
> >I understand that the idle signaling is not working right, any ideas on
> >what I can do to fix this problem?
> >
> >Looking forward to your responses.
> >
> >Regards,
> >
> >Jorge
> >
> >
> >On Fri, 2004-04-30 at 22:58, Steve Underwood wrote:
> >
> >
> >>jorge verastegui wrote:
> >>
> >>
> >>
> >>>Hi
> >>>
> >>>i have successfully downloaded and compiled libr2 from source.
> >>>
> >>>But i dont seem to find how to properly configure it. When i run it
> >>>(partcially unconfigured) the following error occurrs
> >>>
> >>>Signalling requested is R2 Signalling but line is in PRI Signalling
> >>>signalling
> >>>
> >>>
> >>>
> >>>
> >>This error is easy to fix by changing ccs to cas, and removing crc4, in
> >>your zaptel.conf file. However libr2 does not work. It is a partly
> >>implemented solution which I abandoned. It only gets you about 10% of
> >>the way to a working R2 system :-(
> >>
> >>My current R2 software is completely different.
> >>
> >>Regards,
> >>Steve
> >>
> >>
> ]
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users