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
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=
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 mod
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 cal
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.
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 havi
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 f
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
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 A
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
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
(")_(")sig
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,
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 m
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
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
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
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 usual
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 simpl
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?
--
Telec
> 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..
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
> >
> >
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_CALLE
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) | -
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
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--Bandwidth and Colocation Provided b
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
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 option
n 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] r
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 7:0
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 pr
> 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
> a
?
> 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
&g
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-
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
&g
hannel
> 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
>
ll/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
Subj
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
>> 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
> 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
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 yo
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
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.
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
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 conf
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
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
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 sy
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
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__
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
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 thre
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
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 carri
ssage-
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 s
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 no
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
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
protocolcla
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
har
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 loo
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
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
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
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
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
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
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
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
c
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
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
___
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
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 ty
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 b
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.
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Asterisk-Users mailing list
Asterisk-Users@lists
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:
Hel
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
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 WARNI
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
[EM
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
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,c
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 qui
Now that I have your attention ;-)
Anyone know if a new release is planned, and if so when?
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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 als
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, sec
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
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: R
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 w
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 yo
; 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
>
>
; 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
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.
> >
&
ing 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 1
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 13/0
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
Sub
ot;'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 c
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 ca
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 ca
ROTECTED]>
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 c
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
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 Fedo
"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
>
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
Aster
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 supp
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