[asterisk-users] AEX800P on HP Prolaint ML115 - Error on Module Load -

[Alvaro Parres]

Hi list:

   I'm having problems, with a AEX800P card when plugin on HP ML115 G5
Server, when i load the mdule (wctdm24xxp), it loads with error con
dmesg, that say that is a kernel bug of invalid OPCODE 000 or
something like that.

   If i plug the card on SUN Server, i don't have problems with the
card, but the server's nvidia video card stop working :-S

   Any idea?

Thanks.


-- 
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 Tel: +52 (33) 35 63 6261 Ext. 112
  01 800  087 2260
 Cel: +52 (33) 33 68 1087
 alvaro.par...@xmarts.com.mx

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[asterisk-users] Common Inter-Queues Leastrecent Strategy

[Alvaro Parres]

Hi list:

   Is there any way, to set a common inter-queues leastrecent Strategy, i'm
searching a Behaviour like this:


 2 Queues Q1 and Q2
 2 Agentes A1 and A2

 Both agents are in both queues.

 First Call in the system is for Q1 and is answer by A1
 The next call in the system is for Q2, both agents are free,
the system deliver the call to A1... but we want that the call be answer by
the A2.

Any idea???


Thanks.




-- 
Alvaro I. Parres Peredo
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Tel: +52 (33) 35 63 6261 Ext. 112
01 800 087 2260
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Re: [asterisk-users] Help-ASTERISK-MFCR2

[Alvaro Parres]

Mariano:

   Could you send us please the log files, and the console output... so we
can help you.



On Mon, Jun 9, 2008 at 8:01 AM, Mariano Borgognone <
[EMAIL PROTECTED]> wrote:

> Moises, we've already set debug level at 255 on unicall.conf and at
> logger.conf we've enabled full log (notice,warning,error,debug,verbose).
>
> Has anyone experienced with a Siemens EWSD switch?
> Anyone knows about to change R2 timers at unicall.conf ?
>
> Please any comment is welcome, thank you..
> Mariano.-
>
>
> - Original Message -
> From: "Moises Silva" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> 
> Sent: Saturday, June 07, 2008 1:27 PM
> Subject: Re: [asterisk-users] Help-ASTERISK-MFCR2
>
>
> You need to enable loglevel=255 in unicall.conf and enable all the
> levels of logging in logger.conf, otherwise the logs you post don't
> say much.
>
> Moisés Silva
>
> On Fri, Jun 6, 2008 at 2:58 PM, Mariano Borgognone
> <[EMAIL PROTECTED]> wrote:
> >
> > Dears,
> > I have problem ASTERISK with PSTN SIEMENS EWSD (MFC R2), I don´t receive
> > call for PSTN, i don´t understand why. please i need your help 
> >
> > # MFC/R2 normalmente no usa CRC4
> > span=1,1,0,cas,hdb3
> > cas=1-15:1101
> > dchan=16
> > cas=17-31:1101
> > loadzone=us
> > defaultzone=us
> >
> >
> >  [channels]
> > usecallerid=yes
> > hidecallerid=no
> > callwaitingcallerid=yes
> > threewaycalling=yes
> > transfer=yes
> > cancallforward=yes
> > callreturn=yes
> > echocancel=yes
> > echocancelwhenbridged=yes
> > echotraining=yes
> > rxgain=0.0
> > txgain=0.0
> > group=1
> > callgroup=1
> > pickupgroup=1
> > immediate=no
> > musiconhold=default
> > protocolclass=mfcr2
> > protocolvariant=ar,10,10
> > protocolend=cpe
> > group = 1
> > context= e1-incoming
> > channel => 1-15
> > channel => 17-31
> > ;skip time slot 16
> >
> >
> >
> > Here is the LOGS when I try do make calls
> >
> > Jun  6 16:02:18 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2
> > UniCall/1  <- 0001  [1/   1/Idle  /Idle ]
> > Jun  6 16:02:18 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2
> > UniCall/1 Detected
> > Jun  6 16:02:18 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2
> > UniCall/1 Making a new call with CRN 32769
> > Jun  6 16:02:18 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2
> > UniCall/1 1101  ->  [2/   2/Idle  /Idle ]
> > Jun  6 16:02:18 WARNING[5060]: chan_unicall.c:2644 handle_uc_event:
> > Unicall/1 event Detected
> > Jun  6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2
> > UniCall/1  <- 1001  [2/   2/Seize ack /Seize ack]
> > Jun  6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2
> > UniCall/1 Far end disconnected(cause=Normal, unspecified cause [31]) -
> > state
> > 0x2
> > Jun  6 16:02:30 WARNING[5060]: chan_unicall.c:2644 handle_uc_event:
> > Unicall/1 event Far end disconnected
> > Jun  6 16:02:30 WARNING[5060]: chan_unicall.c:2930 handle_uc_event: CRN
> > 32769 - far disconnected cause=Normal, unspecified cause [31]
> > Jun  6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2
> > UniCall/1 Call control(6)
> > Jun  6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2
> > UniCall/1 Drop call(cause=Normal Clearing [16])
> > Jun  6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2
> > UniCall/1 Call disconnected(cause=Normal, unspecified cause [31]) - state
> > 0x800
> > Jun  6 16:02:30 WARNING[5060]: chan_unicall.c:2644 handle_uc_event:
> > Unicall/1 event Drop call
> > Jun  6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2
> > UniCall/1 Call control(7)
> > Jun  6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2
> > UniCall/1 Release call
> > Jun  6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2
> > UniCall/1 1001  ->  [1/1000/Clear fwd /Seize ack]
> > Jun  6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2
> > UniCall/1 Release guard expired
> > Jun  6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2
> > UniCall/1 Destroying call with CRN 32769
> > Jun  6 16:02:30 WARNING[5060]: chan_unicall.c:2644 handle_uc_event:
> > Unicall/1 event Release call
> > -- Unicall/1 released
> > Jun  6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2
> > UniCall/1 Channel echo cancel
> >
> > Best Regards,
> > Mariano Borgognone
> > ___
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>
>
>
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[asterisk-users] Problems with TE120, Kernel BUG

[Alvaro Parres]

Hi list

  I'm havining problems with a E1 Digium Card (TE120) here are the
description of the problem:

   Case 1:  Zaptel 1.4.8  Kernel 2.6.22

  The system start working correctly, but aleatory the asterisk
prosess give a kernel panic with the messages:
   Process asterisk (pid: 7321, ti=dfed6000 task=f7dbf030
task.ti=dfed6000)
   Stack ..

  Case 2:  Zaptel 1.2.20.1 Kernel 2.6.22
   The system between reboot some time load correcty de module
other times no, when the module is not correctly load
   when i try to unload it with a rmmod wcte12xp, give me a
segfault and at dmesg the next error:
 - DMESG OUTPUT ---

BUG: unable to handle kernel paging request at virtual address 20746f6c
 printing eip:
20746f6c
*pde = 
Oops:  [#1]
SMP
Modules linked in: rtc wcte12xp firmware_class zaptel crc_ccitt tg3 e1000
nfs lockd sunrpc jfs dm_mirror dm_mod scsi_wait_scan pdc_adma sata_mv
ata_piix ahci sata_qstor sata_vsc sata_uli sata_sis pata_sis sata_sx4
sata_nv sata_via sata_svw sata_sil24 sata_sil sata_promise libata sbp2
ohci1394 ieee1394 sl811_hcd usbhid ohci_hcd uhci_hcd usb_storage ehci_hcd
usbcore
CPU:0
EIP:0060:[<20746f6c>]Not tainted VLI
EFLAGS: 00010206   (2.6.22-gentoo-r9 #1)
eax: f7960270   ebx: f7960270   ecx: c03d0f10   edx: 20746f6c
esi: f88fab54   edi: c03d5300   ebp: dfed6000   esp: dfed7ee4
ds: 007b   es: 007b   fs: 00d8  gs: 0033  ss: 0068
Process rmmod (pid: 7321, ti=dfed6000 task=f7dbf030 task.ti=dfed6000)
Stack: f8e72804 f7960004 f796  dfc7f800 f88f32cc 378b6000
c035df8e
   c03af250 f796 f88fab54 c03d5300 f88f00c1 c03d5300 dfc7f800
c0261e88
   dfc7f848 c02aea83 dfc7f848 f88fab54 c02aee79 f88fab54 
c02ae65d
Call Trace:
 [] <0> [] <0> [] <0> [] <0>
[] <0> [] <0> [] <0> [] <0>
[] <0> [] <0> [] <0> [] <0>
[] <0> ===
Code:  Bad EIP value.
EIP: [<20746f6c>]  SS:ESP 0068:dfed7ee4

--- END DMESG OUTPUT
---

  The wire think is that the card was working with out problems, so any
help ?

   Those the card have a damage ? the kernel ? or what... ???

   We have al ready clean all the system and recompile the zaptel driver
a lot of times.

Thanks.


-- 
Alvaro I. Parres Peredo
Director de IT
Grupo Xmarts SA de CV
Tel: +52 (33) 35 63 6261 Ext. 112
 01 800  087 2260
Cel: +52 (33) 33 68 1087
[EMAIL PROTECTED]
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[asterisk-users] Asterisk 1.4.13 & Unicall

[Alvaro Parres]

Hello List:

   I have publish a tar.gz file with an Asterisk 1.4.13 correctly patched
for compile the chan_unicall and the apps rxfax and txfax. This tar.gz file
also contain all the necesary library for work.

  http://arabe.com.mx/blog/?p=10

  This tar was based on the one publish by Moy at his blog of astunicall (
http://www.moythreads.com/astunicall/).

-- 
Alvaro I. Parres Peredo
Director de IT
Grupo Xmarts SA de CV
Tel: +52 (33) 35 63 6261 Ext. 112
  01 800  087 2260
Cel: +52 (33) 33 68 1087
[EMAIL PROTECTED]
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Re: [asterisk-users] Asterisk 1.4, Unicall and Nextel...

[Alvaro Parres]

Moises:

Yes:

 On version 1.4 I use the next versions:

 libmfcr2-0.0.3_1.4
  libunicall-0.0.3_1.4

The ones at 1.2. i send tomorrow, becouse i don't remeber witch
versions i use (witch pre).


 I menssioned the Asterisk versions, becouse after some diff over the
libmfcr, chan_unicall, and libunicall i didn't find any change on protocol
managment.






On Mon, 2007-07-23 at 11:47 -0500, Moises Silva wrote:
> Alvaro,
>
> Naming Asterisk versions is of little help since Asterisk is not
> the one failing here. It would be more helpful know the libmfcr2 and
> spandsp versions that were used in the working/non working tests, is
> that possible? do you have the versions at hand?
>
I am using libmfcr2-0.0.3_1.4.tar.bz2 which I downloaded from your
site.  I am in big trouble because the only way I can get calls from
Nextel phones is to disable CallerID by setting ANI to 0.  Right now I
have to use:

protocolvariant=mx,0,4

I am having the same problem on E1 service from Telmex and
Avantel.
Obviously customers want CallerID on their phones.


> Thanks a lot.
>
> On 7/23/07, Alvaro Parres <[EMAIL PROTECTED]> wrote:
> > Steve:
> >
> > The problem occurs only in Asterisk 1.4, not at 1.2.
> >
> > The log is the same that Carlos post here. We recive an Unexpected
MF6
> > signal when a No caller ID or restriected Caller ID is recive.
> >
> >   This is for all the operatos.
> >
> >  If we add mfcr2->group_i_end_of_ANI_restricted =
> > R2_SIGI_12; to the Mexico Definition we can recive restricted Caller
ID
> > calls, but no Not Caller ID Calls.
> >
> >  I'm not shure why only on versions 1.4 and not on version 1.2.
> >
> > Thanks.
> >
> > On 7/22/07, Steve Underwood < [EMAIL PROTECTED]> wrote:
> > > Alvaro Parres wrote:
> > > > Search at mfcr2.c this:
> > > >
> > > > case MFCR2_PROT_MEXICO:
> > > >
> > > > And add the next line after that line:
> > > >
> > > >  mfcr2->group_i_end_of_ANI_restricted = R2_SIGI_12;
> > > >
> > > > This will help you on calls that have the restricted flag on the
ANI
> > > > only. (Nextel). But not on no caller id calls.
> > > >
> > > > I don't know if steve can help us whit the case where no caller id
is
> > > > send.
> > >
> > > Is that change appropriate for all Mexican operators, or just some?
For
> > > the time being I have added that to my latest source code.
> > >
> > > What happens when there is no ANI, and is it the same for all
operators
> > > in Mexico? Can you send a log of such a call with "loglevel=255" in
> > > unicall.conf, and I will try to sort this out. It seems strange I
> > > haven't had more feedback on this, since there seem to be quite a
few
> > > users in Mexico.
> > >
> > > Regards,
> > > Steve
> > >
> > >
> > > ___
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> > >
> > > asterisk-users mailing list
> > > To UNSUBSCRIBE or update options visit:
> > >
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> >
> >
> >
> > --
> > Alvaro I. Parres Peredo
> >  Director de IT
> >  Grupo Xmarts SA de CV
> >  Tel: +52 (33) 35 63 6261 Ext. 112
> >01 800  087 2260
> >  Cel: +52 (33) 33 68 1087
> >  [EMAIL PROTECTED]
> > ___
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> >
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> >
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> >
>
>
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001

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--
Alvaro I. Parres Peredo
Director de IT
Grupo Xmarts SA de CV
Tel: +52 (33) 35 63 6261 Ext. 112
 01 800  087 2260
Cel: +52 (33) 33 68 1087
[EMAIL PROTECTED]
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Re: [asterisk-users] Asterisk 1.4, Unicall and Nextel...

[Alvaro Parres]

Steve:

   The problem occurs only in Asterisk 1.4, not at 1.2.

   The log is the same that Carlos post here. We recive an Unexpected MF6
signal when a No caller ID or restriected Caller ID is recive.

This is for all the operatos.

If we add mfcr2->group_i_end_of_ANI_restricted = R2_SIGI_12; to the
Mexico Definition we can recive restricted Caller ID calls, but no Not
Caller ID Calls.

I'm not shure why only on versions 1.4 and not on version 1.2.

Thanks.

On 7/22/07, Steve Underwood <[EMAIL PROTECTED]> wrote:


Alvaro Parres wrote:
> Search at mfcr2.c this:
>
> case MFCR2_PROT_MEXICO:
>
> And add the next line after that line:
>
>  mfcr2->group_i_end_of_ANI_restricted = R2_SIGI_12;
>
> This will help you on calls that have the restricted flag on the ANI
> only. (Nextel). But not on no caller id calls.
>
> I don't know if steve can help us whit the case where no caller id is
> send.

Is that change appropriate for all Mexican operators, or just some? For
the time being I have added that to my latest source code.

What happens when there is no ANI, and is it the same for all operators
in Mexico? Can you send a log of such a call with "loglevel=255" in
unicall.conf, and I will try to sort this out. It seems strange I
haven't had more feedback on this, since there seem to be quite a few
users in Mexico.

Regards,
Steve


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--
Alvaro I. Parres Peredo
Director de IT
Grupo Xmarts SA de CV
Tel: +52 (33) 35 63 6261 Ext. 112
 01 800  087 2260
Cel: +52 (33) 33 68 1087
[EMAIL PROTECTED]
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Re: [asterisk-users] Asterisk 1.4, Unicall and Nextel...

[Alvaro Parres]

Yes Moises, i was looking for it.

The main problem is only on the files for version 1.4... it give that
error when no CallerID is recive or a private caller id is recive.

The change i made is to add to Mexico variant on mfcr2.c this line
mfcr2->group_i_end_of_ANI_restricted = R2_SIGI_12;

This works for nextel or phones that send private caller id.. But
doesn't work when no CallerID is recive.

I have al ready check diff files from 1.2 files and 1.4 files and i
didn't find any big difference between both version.





On 7/18/07, Moises Silva <[EMAIL PROTECTED]> wrote:


Alvaro, can you post the patch in a public place and post the URL
here? It might be a good idea to contact steve underwood to see what
he has to say about such a patch.

Regards,

On 7/18/07, Alvaro Parres <[EMAIL PROTECTED]> wrote:
> Carlos:
>
>Only for check do this change:
>
> protocolvariant=mx,10,4
>
>for
>
> protocolvariant=mx,0,4
>
>If it's works, contact me and i will send you a patch for libmfcr.c
>
>
> Thanks.
>
>
> Carlos:
>
> Has el cambio que te pido arriva, para revisar si es lo del caller
ID.
> Casi estoy seguro nosotros en labortario y en el extrangero tuvimos esos
> problemas, si es asi te paso un parche solo que lo encuentre para la
> libmfcr.c Donde le digas como manejar la señal de private al recibir el
ANI.
>
> Saludos.
>
>
>
> On 7/18/07, Carlos Chavez < [EMAIL PROTECTED]> wrote:
> >
> > On Wed, 2007-07-18 at 08:10 -0500, Alvaro Parres wrote:
> > > Could you send please your unicall.conf file
> > >
> > > Thanks.
> > >
> > > It appers to be a problem with de ANI digits you want to recive.
> > >
> > >
> > Also Nextel never sends CallerID.  When someone calls me from
a
> Nextel
> > phone to my cell or to my Asterisk server I always get "Private Call".
> >
> > >
> > --
> > Telecomunicaciones Abiertas de México S.A. de C.V.
> > Carlos Chávez Prats
> > Director de Tecnología
> > +52-55-91169161 ext 2001
> >
> > ___
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> >
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> > To UNSUBSCRIBE or update options visit:
> >
> http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
>
>
>
> --
> Alvaro I. Parres Peredo
>  Director de IT
>  Grupo Xmarts SA de CV
>  Tel: +52 (33) 35 63 6261 Ext. 112
>   01 800  087 2260
>  Cel: +52 (33) 33 68 1087
>  [EMAIL PROTECTED]
> ___
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>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>
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>


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Tel: +52 (33) 35 63 6261 Ext. 112
 01 800  087 2260
Cel: +52 (33) 33 68 1087
[EMAIL PROTECTED]
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Re: [asterisk-users] Unicall/R2 for Asterisk 1.4 Available for TESTING

[Alvaro Parres]

Only to continue on this thread (becouse this is start in other meail).

   The 1.4.X. unicall patch is working well,  only with one problem: There
is a problem hen reciving calls with no Caller ID.

Thanks.


On 6/9/07, Moises Silva <[EMAIL PROTECTED]> wrote:


Alvaro...

Hum..., I never have tried RxFax... let me know if you need any extra
help with that. Sounds interesting

On 6/8/07, Alvaro Parres <[EMAIL PROTECTED]> wrote:
> Moy:
>
> I have working an Asterisk 1.4.4 with Unicall rn MFR2. The only
problem
> i have is the RxFAX application, that broke every time... With and error
in
> the linking to the spandsp library.
>
> If i have time this weekend i will review to fix the app,
>
> Thanks.
>
>
> On 6/4/07, Tobias Wolf <[EMAIL PROTECTED]> wrote:
> > Humberto Figuera schrieb:
> > > HI Tobias,
> > >
> > > look in www.soft-switch.org/unicall/unicall/index.html
> ;p
> > >
> > Thank you. Not very complete but it has given me an idea what to think
> > of unicall.
> >
> >
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Re: [asterisk-users] Asterisk 1.4, Unicall and Nextel...

[Alvaro Parres]

Search at mfcr2.c this:

   case MFCR2_PROT_MEXICO:

And add the next line after that line:

mfcr2->group_i_end_of_ANI_restricted = R2_SIGI_12;

This will help you on calls that have the restricted flag on the ANI only.
(Nextel). But not on no caller id calls.

I don't know if steve can help us whit the case where no caller id is send.


On 7/19/07, Carlos Chavez <[EMAIL PROTECTED]> wrote:


 *On Thu, 19 Jul 2007 12:14:53 -0500, Alvaro Parres wrote*
> Yes Moises, i was looking for it.
>
>  The main problem is only on the files for version 1.4... it give
that error when no CallerID is recive or a private caller id is recive.
>
>   The change i made is to add to Mexico variant on mfcr2.c this line
mfcr2->group_i_end_of_ANI_restricted = R2_SIGI_12;
>
>  This works for nextel or phones that send private caller id.. But
doesn't work when no CallerID is recive.
>
>  I have al ready check diff files from 1.2 files and 1.4 files and i
didn't find any big difference between both version.
>
Ok, I did the change you specified and now we can receive calls from
Nextel phones but get no callerid on any call.  How do I apply the patch to
libmfcr2?

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Re: [asterisk-users] Asterisk 1.4, Unicall and Nextel...

[Alvaro Parres]

Carlos:

  Only for check do this change:

protocolvariant=mx,10,4

  for

protocolvariant=mx,0,4

  If it's works, contact me and i will send you a patch for libmfcr.c


Thanks.


Carlos:

   Has el cambio que te pido arriva, para revisar si es lo del caller ID.
Casi estoy seguro nosotros en labortario y en el extrangero tuvimos esos
problemas, si es asi te paso un parche solo que lo encuentre para la
libmfcr.c Donde le digas como manejar la señal de private al recibir el ANI.

Saludos.


On 7/18/07, Carlos Chavez <[EMAIL PROTECTED]> wrote:


On Wed, 2007-07-18 at 08:10 -0500, Alvaro Parres wrote:
> Could you send please your unicall.conf file
>
> Thanks.
>
> It appers to be a problem with de ANI digits you want to recive.
>
>
Also Nextel never sends CallerID.  When someone calls me from a
Nextel
phone to my cell or to my Asterisk server I always get "Private Call".

>
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Re: [asterisk-users] Asterisk 1.4, Unicall and Nextel...

[Alvaro Parres]

Could you send please your unicall.conf file

Thanks.

It appers to be a problem with de ANI digits you want to recive.




On 7/17/07, Carlos Chavez <[EMAIL PROTECTED]> wrote:


On Tue, 2007-07-17 at 19:30 -0500, Moises Silva wrote:
> In order to help you I need testcall traces, with max level of
> logging, of incoming Nextel calls.
>
   Here is the log file from a couple of calls from a Nextel phone:

[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56
<- 0001  [1/   1/Idle  /Idle ]
[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56
Detected
[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56 Making
a new call with CRN 32769
[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56 1101
->  [2/   2/Idle  /Idle ]
[Jul 17 21:46:08] NOTICE[4771] chan_unicall.c: Unicall/56 event Detected
[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56
<- 9 on  [2/   2/Seize ack /Seize ack]
[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56 1 on
->  [2/   2/Seize ack /Seize ack]
[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56
<- 9 off [2/   2/Group A   /DNIS request ]
[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56 1 off
->  [2/   2/Group A   /DNIS request ]
[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56
<- 9 on  [2/   2/Group A   /DNIS request ]
[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56 1 on
->  [2/   2/Group A   /DNIS request ]
[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56
<- 9 off [2/   2/Group A   /DNIS request ]
[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56 1 off
->  [2/   2/Group A   /DNIS request ]
[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56
<- 9 on  [2/   2/Group A   /DNIS request ]
[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56 1 on
->  [2/   2/Group A   /DNIS request ]
[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56
<- 9 off [2/   2/Group A   /DNIS request ]
[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56 1 off
->  [2/   2/Group A   /DNIS request ]
[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56
<- 8 on  [2/   2/Group A   /DNIS request ]
[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56 6 on
->  [2/   2/Group A   /DNIS request ]
[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56
<- 8 off [2/   2/Group C   /Category req ]
[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56 6 off
->  [2/   2/Group C   /Category req ]
[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56
<- 9 on  [2/   2/Group C   /Category req ]
[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56 1 on
->  [2/   2/Group C   /Category req ]
[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56
<- 9 off [2/   2/Group C   /ANI request  ]
[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56 1 off
->  [2/   2/Group C   /ANI request  ]
[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56
<- F on  [2/   2/Group C   /ANI request  ]
[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56 R2
prot. err. [2/   2/Group C   /ANI request  ] cause 32772 -
Unexpected MF6 signal
[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56 1001
->  [1/   1/Idle  /Idle ]
[Jul 17 21:46:08] NOTICE[4771] chan_unicall.c: Unicall/56 event Protocol
failure
[Jul 17 21:46:08] ERROR[4771] chan_unicall.c: Unicall/56 protocol error.
Cause 32772
[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56
Channel echo cancel
[Jul 17 21:46:09] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/57
<- 0001  [1/   1/Idle  /Idle ]
[Jul 17 21:46:09] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/57
Detected
[Jul 17 21:46:09] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/57 Making
a new call with CRN 32769
[Jul 17 21:46:09] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/57 1101
->  [2/   2/Idle  /Idle ]
[Jul 17 21:46:09] NOTICE[4771] chan_unicall.c: Unicall/57 event Detected
[Jul 17 21:46:09] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/57
<- 9 on  [2/   2/Seize ack /Seize ack]
[Jul 17 21:46:09] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/57 1 on
->  [2/   2/Seize ack /Seize ack]
[Jul 17 21:46:09] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/57
<- 9 off [2/   2/Group A   /DNIS request ]
[Jul 17 21:46:09] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/57 1 off
->  [2/   2/Group A   /DNIS request ]
[Jul 17 21:46:09] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/57
<- 9 on  [2/   2/Group A   /DNIS 

[asterisk-users] Help doing one modification to libmfcr2.c of Unicall

[Alvaro Parres]

Hi list:

  I want to modify the libmfcr2. But i can't find where is define the end
DNIS signal is define. Actually the libmfcr2 send a ONE (1) at the end of
sending all the DNIS numbers. I need to send a TWO (2), this becouse in
Mexico the normal is to send a 2 at the end, not a ONE.

 In the next example i dialed

un  6 19:12:17 WARNING[719] chan_unicall.c: MFC/R2 UniCall/5 Call
control(1)
Jun  6 19:12:17 WARNING[719] chan_unicall.c: MFC/R2 UniCall/5 Make call
Jun  6 19:12:17 WARNING[719] chan_unicall.c: MFC/R2 UniCall/5 Making a
new call with CRN 32770
Jun  6 19:12:17 WARNING[719] chan_unicall.c: MFC/R2 UniCall/5 0001  ->
   [1/   1/Idle  /Idle ]
Jun  6 19:12:17 WARNING[719] chan_unicall.c: Unicall/5 event Dialing
Jun  6 19:12:17 WARNING[719] chan_unicall.c: MFC/R2 UniCall/5  <-
1101  [1/  40/Seize /Idle ]
Jun  6 19:12:17 WARNING[719] chan_unicall.c: MFC/R2 UniCall/5 3 on  ->
   [2/  40/Group I   /Idle ]
Jun  6 19:12:17 WARNING[719] chan_unicall.c: MFC/R2 UniCall/5  <- 1
on  [2/  40/Group I   /DNIS ]
Jun  6 19:12:17 WARNING[719] chan_unicall.c: MFC/R2 UniCall/5 3 off ->
   [2/  40/Group I   /DNIS ]
Jun  6 19:12:17 WARNING[719] chan_unicall.c: MFC/R2 UniCall/5  <- 1
off [2/  40/Group I   /DNIS ]
Jun  6 19:12:17 WARNING[719] chan_unicall.c: MFC/R2 UniCall/5 6 on  ->
   [2/  40/Group I   /DNIS ]
Jun  6 19:12:17 WARNING[719] chan_unicall.c: MFC/R2 UniCall/5  <- 1
on  [2/  40/Group I   /DNIS ]
Jun  6 19:12:17 WARNING[719] chan_unicall.c: MFC/R2 UniCall/5 6 off ->
   [2/  40/Group I   /DNIS ]
Jun  6 19:12:17 WARNING[719] chan_unicall.c: MFC/R2 UniCall/5  <- 1
off [2/  40/Group I   /DNIS ]
Jun  6 19:12:17 WARNING[719] chan_unicall.c: MFC/R2 UniCall/5 3 on  ->
   [2/  40/Group I   /DNIS ]
Jun  6 19:12:18 WARNING[719] chan_unicall.c: MFC/R2 UniCall/5  <- 1
on  [2/  40/Group I   /DNIS ]
Jun  6 19:12:18 WARNING[719] chan_unicall.c: MFC/R2 UniCall/5 3 off ->
   [2/  40/Group I   /DNIS ]
Jun  6 19:12:18 WARNING[719] chan_unicall.c: MFC/R2 UniCall/5  <- 1
off [2/  40/Group I   /DNIS ]
Jun  6 19:12:18 WARNING[719] chan_unicall.c: MFC/R2 UniCall/5 2 on  ->
   [2/  40/Group I   /DNIS ]
Jun  6 19:12:18 WARNING[719] chan_unicall.c: MFC/R2 UniCall/5  <- 1
on  [2/  40/Group I   /DNIS ]
Jun  6 19:12:18 WARNING[719] chan_unicall.c: MFC/R2 UniCall/5 2 off ->
   [2/  40/Group I   /DNIS ]
Jun  6 19:12:18 WARNING[719] chan_unicall.c: MFC/R2 UniCall/5  <- 1
off [2/  40/Group I   /DNIS ]
Jun  6 19:12:18 WARNING[719] chan_unicall.c: MFC/R2 UniCall/5 7 on  ->
   [2/  40/Group I   /DNIS ]
Jun  6 19:12:18 WARNING[719] chan_unicall.c: MFC/R2 UniCall/5  <- 1
on  [2/  40/Group I   /DNIS ]
Jun  6 19:12:18 WARNING[719] chan_unicall.c: MFC/R2 UniCall/5 7 off ->
   [2/  40/Group I   /DNIS ]
Jun  6 19:12:18 WARNING[719] chan_unicall.c: MFC/R2 UniCall/5  <- 1
off [2/  40/Group I   /DNIS ]
Jun  6 19:12:18 WARNING[719] chan_unicall.c: MFC/R2 UniCall/5 5 on  ->
   [2/  40/Group I   /DNIS ]
Jun  6 19:12:18 WARNING[719] chan_unicall.c: MFC/R2 UniCall/5  <- 1
on  [2/  40/Group I   /DNIS ]
Jun  6 19:12:18 WARNING[719] chan_unicall.c: MFC/R2 UniCall/5 5 off ->
   [2/  40/Group I   /DNIS ]
Jun  6 19:12:18 WARNING[719] chan_unicall.c: MFC/R2 UniCall/5  <- 1
off [2/  40/Group I   /DNIS ]
Jun  6 19:12:18 WARNING[719] chan_unicall.c: MFC/R2 UniCall/5 3 on  ->
   [2/  40/Group I   /DNIS ]
Jun  6 19:12:18 WARNING[719] chan_unicall.c: MFC/R2 UniCall/5  <- 1
on  [2/  40/Group I   /DNIS ]
Jun  6 19:12:18 WARNING[719] chan_unicall.c: MFC/R2 UniCall/5 3 off ->
   [2/  40/Group I   /DNIS ]
Jun  6 19:12:18 WARNING[719] chan_unicall.c: MFC/R2 UniCall/5  <- 1
off [2/  40/Group I   /DNIS ]
Jun  6 19:12:18 WARNING[719] chan_unicall.c: MFC/R2 UniCall/5 6 on  ->
   [2/  40/Group I   /DNIS ]
Jun  6 19:12:20 WARNING[719] chan_unicall.c: MFC/R2 UniCall/5  <- 3
on  [2/  40/Group I   /DNIS ]
Jun  6 19:12:20 WARNING[719] chan_unicall.c: MFC/R2 UniCall/5 6 off ->
   [2/  40/Group I   /DNIS ]
Jun  6 19:12:20 WARNING[719] chan_unicall.c: MFC/R2 UniCall/5  <- 3
off [2/  40/Group I   /DNIS ]

*** THIS ONE IS WHAT I MEEN ***

Jun  6 19:12:20 WARNING[719] chan_unicall.c: MFC/R2 UniCall/5 1 on  ->
   [2/  40/Group I   /DNIS ]
Jun  6 19:12:20 WARNING[719] chan_unicall.c: MFC/R2 UniCall/5  <- 4
on  [2/  40/Group II  /Category ]
Jun  6 19:12:20 WARNING[719] chan_unicall.c: MFC/R2 UniCall/5 1 off ->
   [2/  40/Group II   

Re: [asterisk-users] Asterisk 1.4 with Unicall

[Alvaro Parres]

Carlos:

I think you problem is on the queue systems, we have the same problem
on version 1.2.x and 1.4.x
one one of our call centers.

Try to change you agents to be dynamic, and also to change the login
method from AgentLogin to AgentCallBackLogin

Alvaro

On 6/8/07, Carlos Chavez <[EMAIL PROTECTED]> wrote:


I have a small call center running with Asterisk 1.4.4 and
Unicall.
Everything seems to be working but twice now we had to reset the server
because all lines stopped working.  You can see users dialing in and
reaching the queue but the agents never get the call and the lines are
not released.

I saw that there is a new Zaptel driver which fixes a racing
condition
with a TE110P card which is what we are using.  Could this be the
problem?  I also keep getting the following messages:

[Jun  8 18:36:02] NOTICE[16202]: chan_unicall.c:2401 unicall_indicate:
unicall_indicate 16
[Jun  8 18:36:02] NOTICE[16350]: chan_unicall.c:2401 unicall_indicate:
unicall_indicate -1
[Jun  8 18:36:02] WARNING[16202]: chan_unicall.c:2449 unicall_indicate:
Don't know how to set condition 16 on channel UniCall/1-1
[Jun  8 18:36:02] NOTICE[16202]: chan_unicall.c:2401 unicall_indicate:
unicall_indicate 16
[Jun  8 18:36:02] WARNING[16202]: chan_unicall.c:2449 unicall_indicate:
Don't know how to set condition 16 on channel UniCall/1-1
[Jun  8 18:36:02] NOTICE[16350]: chan_unicall.c:2401 unicall_indicate:
unicall_indicate 17
[Jun  8 18:36:02] WARNING[16350]: chan_unicall.c:2449 unicall_indicate:
Don't know how to set condition 17 on channel UniCall/14-1

What do they mean?  I've never seen them under Asterisk 1.2

--
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Director de Tecnología
+52-55-91169161 ext 2001

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Director de IT
Grupo Xmarts SA de CV
Tel: +52 (33) 35 63 6261 Ext. 112
 01 800  087 2260
Cel: +52 (33) 33 68 1087
[EMAIL PROTECTED]
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Re: [asterisk-users] Problems with ChanIsAvail always return status 0

[Alvaro Parres]

Jared:

  As you see i have the s option. That works fine on Version 1.2. Let me
see config the call limit con sip channels it works.

Thanks.


On 6/25/07, Jared Smith <[EMAIL PROTECTED]> wrote:


On 6/25/07, Alvaro Parres <[EMAIL PROTECTED]> wrote:
> I'm having the next problem, it appear that the application
ChanIsAvail
> is not working on Asterisk 1.4.5 always return me 0 in AVAILSTATUS.
> I add my dialplan and the output to the cli.

This isn't really a problem with ChanIsAvail... it's more of a
misunderstanding of what's going on.  In your case, it appears that
your SIP device will accept multiple calls at the same time from
Asterisk.  So even if your phone is on a call, Asterisk will come
along, try to make another call to it, and the phone says "Hey, go
ahead! I don't mind!"

You've got quite a few options to solve your problem.  While none of
them are exactly perfect, it's good to have lots of options:

o  Try using the 's' option to ChanIsAvail().  (You might have to
turn on call limits in sip.conf to get this to work correctly.  Last
time I played with this, it seems that the limitonpeers setting had to
be set to yes as well.)
o  Use the GROUP() dialplan function to assign calls to call groups,
and then use the GROUP_COUNT() function to check to see if that phone
is already on any calls.
o  Turn off call waiting on your IP phone, so that it'll only accept
one call at a time
o  Simply get call limits in sip.conf working correctly.  (This is
probably the hardest to do, unfortunately.)

Hopefully, one of those options will help you out.  (I've placed them
in the order I'd try... but your mileage may vary.)

-Jared

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Director de IT
Grupo Xmarts SA de CV
Tel: +52 (33) 35 63 6261 Ext. 112
 01 800  087 2260
Cel: +52 (33) 33 68 1087
[EMAIL PROTECTED]
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[asterisk-users] Problems with ChanIsAvail always return status 0

[Alvaro Parres]

Hi list:

   I'm having the next problem, it appear that the application ChanIsAvail
is not working on Asterisk 1.4.5 always return me 0 in AVAILSTATUS.
I add my dialplan and the output to the cli.

THanks.


   In the example i'm dialing from extension SIP/112


My DialPlan Secction:

[macro-callonlyiffree]
exten => s,1,ChanIsAvail(${ARG1}|s)
exten => s,n,NoOp(${AVAILCHAN})
exten => s,n,NoOp(${AVAILORIGCHAN})
exten => s,n,NoOp(${AVAILSTATUS})
exten => s,n,GoToIf($[${AVAILSTATUS} < 1]?autoanswer:fail)
exten => s,n,NoOp()
exten => s,n(autoanswer),Dial(${ARG1}||)
exten => s,102(fail),Hangup

[pruebas]
exten => *99,1,Dial(Local/[EMAIL PROTECTED]&Local/[EMAIL PROTECTED]||r)

[inpuerta]
exten => _1XX,1,Macro(callonlyiffree,SIP/${EXTEN})


The Log:

   -- Executing [EMAIL PROTECTED]:1] Dial("SIP/112-08236be8",
"Local/[EMAIL PROTECTED]&Local/[EMAIL PROTECTED]||r")
in new stack
   -- Called [EMAIL PROTECTED]
   -- Called [EMAIL PROTECTED]
   -- Executing [EMAIL PROTECTED]:1] Macro("Local/[EMAIL PROTECTED],2",
"callonlyiffree|SIP/111") in new stack
   -- Executing [EMAIL PROTECTED]:1] ChanIsAvail("
Local/[EMAIL PROTECTED],2", "SIP/111|s") in new stack
   -- Executing [EMAIL PROTECTED]:1] Macro("Local/[EMAIL PROTECTED],2",
"callonlyiffree|SIP/112") in new stack
   -- Executing [EMAIL PROTECTED]:1] ChanIsAvail("
Local/[EMAIL PROTECTED],2", "SIP/112|s") in new stack
   -- Executing [EMAIL PROTECTED]:2] NoOp("Local/[EMAIL PROTECTED],2",
"SIP/111-081f7d18") in new stack
   -- Executing [EMAIL PROTECTED]:3] NoOp("Local/[EMAIL PROTECTED],2",
"SIP/111") in new stack
   -- Executing [EMAIL PROTECTED]:4] NoOp("Local/[EMAIL PROTECTED],2",
"0") in new stack
   -- Executing [EMAIL PROTECTED]:5] GotoIf("Local/[EMAIL PROTECTED],2",
"1?autoanswer:fail") in new stack
   -- Goto (macro-callonlyiffree,s,7)
   -- Executing [EMAIL PROTECTED]:7] Dial("Local/[EMAIL PROTECTED],2",
"SIP/111||") in new stack
   -- Called 111
   -- Executing [EMAIL PROTECTED]:2] NoOp("Local/[EMAIL PROTECTED],2",
"SIP/112-0822a4b8") in new stack
   -- Executing [EMAIL PROTECTED]:3] NoOp("Local/[EMAIL PROTECTED],2",
"SIP/112") in new stack
   -- Executing [EMAIL PROTECTED]:4] NoOp("Local/[EMAIL PROTECTED],2",
"0") in new stack
   -- Executing [EMAIL PROTECTED]:5] GotoIf("Local/[EMAIL PROTECTED],2",
"1?autoanswer:fail") in new stack
   -- Goto (macro-callonlyiffree,s,7)
   -- Executing [EMAIL PROTECTED]:7] Dial("Local/[EMAIL PROTECTED],2",
"SIP/112||") in new stack
   -- Called 112
   -- SIP/111-08342ec0 is ringing
   -- Local/[EMAIL PROTECTED],1 is ringing
   -- SIP/112-08346e28 is ringing
   -- Local/[EMAIL PROTECTED],1 is ringing





--
Alvaro I. Parres Peredo
Director de IT
Grupo Xmarts SA de CV
Tel: +52 (33) 35 63 6261 Ext. 112
 01 800  087 2260
Cel: +52 (33) 33 68 1087
[EMAIL PROTECTED]
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[asterisk-users] Set a global queue policy

[Alvaro Parres]

Hi list:

  Is there any way or an idea of how to made a global queue policy. I need
to have a Global Policy or a common policy to many queues.

  What i need is:

  I have 20 agents they are members of 5 queues, i have a last recent
strategy for all the queues, the problem is that the strategy is for each
queue. So one agent recive a call from all the queues, then another, and so
on... So we have agent that don't work for more than 20 minutes. And
anothers that have long brakes.

  So what i need is that 5 queues share the strategy, so the agent who
get the call the agent that have more idle time in all five queues.


So any idea ?








--
Alvaro I. Parres Peredo
Director de IT
Grupo Xmarts SA de CV
Tel: +52 (33) 35 63 6261 Ext. 112
 01 800  087 2260
Cel: +52 (33) 33 68 1087
[EMAIL PROTECTED]
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Re: [asterisk-users] Unicall/R2 for Asterisk 1.4 Available for TESTING

[Alvaro Parres]

Moy:

   I have working an Asterisk 1.4.4 with Unicall rn MFR2. The only problem
i have is the RxFAX application, that broke every time... With and error in
the linking to the spandsp library.

   If i have time this weekend i will review to fix the app,

Thanks.


On 6/4/07, Tobias Wolf <[EMAIL PROTECTED]> wrote:


Humberto Figuera schrieb:
> HI Tobias,
>
> look in www.soft-switch.org/unicall/unicall/index.html ;p
>
Thank you. Not very complete but it has given me an idea what to think
of unicall.


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Re: [asterisk-users] Unicall/R2 for Asterisk 1.4 Available for TESTING

[Alvaro Parres]

Grate job Moy... i will test it on my PBX tomorrow...

Thanks.


On 4/20/07, Moises Silva <[EMAIL PROTECTED]> wrote:


Thanks a lot for the fix Humberto.

On 4/18/07, Humberto Figuera <[EMAIL PROTECTED]> wrote:
> Hi Moises,
>
> the Asterisk SVN-branch-1.4-r60989 make a change in the
> ast_channel_alloc function:
>
> "This is a big improvement over the current CDR fixes. It may still
> need refinement, but this won't have as many folks bothered."
>
> here the patch for chan_unicall.c ;p
>
> --- chan_unicall.c.orig 2007-04-18 03:32:17.0 -0400
> +++ chan_unicall.c  2007-04-18 03:32:26.0 -0400
> @@ -2485,7 +2485,7 @@
>  }
>  while (x < 3);
>
> -if ( ( tmp = ast_channel_alloc(0, state, 0, 0, chan_name) ) ==
NULL)
> +if ( ( tmp = ast_channel_alloc(0, state, 0, 0, i->accountcode,
> i->exten, i->context, i->amaflags, chan_name) ) == NULL)
>  {
>  ast_log(LOG_WARNING, "Unable to allocate channel structure\n");
>  return  NULL;
>
> --
> Humberto Figuera - Using Linux 2.6.20
> Usuario GNU/Linux 369709
> Caracas - Venezuela
> GPG Key Fingerprint = 5AAC DF0C 00F4 2834 28BA  37AD 3364 01D1 74CA 0603
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--
"Su nombre es GNU/Linux, no solamente Linux, mas info en
http://www.gnu.org";
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Re: [asterisk-users] PickUp a call with feature pickup (*8) from a IAX2channel

[Alvaro Parres]

I had set it

On 3/21/07, LKS GMAIL <[EMAIL PROTECTED]> wrote:


 Try to set the callgroup and pickupgroup up in the IAX conf.



Saludos, Lukassky.
  --

*De:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *En nombre de *Alvaro Parres
*Enviado el:* miércoles, 21 de marzo de 2007 16:55
*Para:* Asterisk Users Mailing List - Non-Commercial Discussion
*Asunto:* [asterisk-users] PickUp a call with feature pickup (*8) from a
IAX2channel



Hi list, i'm trying to do that iax channels can acces the pickup
feature(normaly *8 dialing).

But always the iax channel when dial *8, search for the extensión *8 on
its context.

I know i can program the *8 extension with the pickup applicatión. But its
doesn't works for me, becouse i need to pickup some calls comming from IVR's
o Queues.
And there de exten is no the same as the channel, etc.

Any idea or help ?

Thaks.

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[asterisk-users] PickUp a call with feature pickup (*8) from a IAX2 channel

[Alvaro Parres]

Hi list, i'm trying to do that iax channels can acces the pickup
feature(normaly *8 dialing).

But always the iax channel when dial *8, search for the extensión *8 on its
context.

I know i can program the *8 extension with the pickup applicatión. But its
doesn't works for me, becouse i need to pickup some calls comming from IVR's
o Queues.
And there de exten is no the same as the channel, etc.

Any idea or help ?

Thaks.
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Re: [asterisk-users] G729 Licence Consumption Problem

[Alvaro Parres]

Yes i'm recording...On 10/8/06, Thomas Kenyon <[EMAIL PROTECTED]> wrote:
Alvaro Parres wrote:> Hi List:>> I have the next diagram:> GSM   G729> G729>  IdeFisk -- Asterisk A -> [INTERNET] Asterisk B - PSTN ( Via Unicall / Zap )
>>The user at IdeFisk Login as Agents on Asterisk B> at this moment we have the next Licence Use:>   A) 1/1>   B) 1/0>>When a Call from the QUEUE on Asterisk B is Bridge to the Agent I
> have the next Use:>>  A) 1/1>  B) 1/3>> Any one can explain me this ?, why the incress of licence consumptions.>> Thanks.>Are you recording the call? There may be a separate process decoding the
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[asterisk-users] G729 Licence Consumption Problem

[Alvaro Parres]

Hi List:    I have the next diagram:    GSM   G729    G729     IdeFisk -- Asterisk A - [INTERNET] Asterisk B - PSTN ( Via Unicall / Zap )
   The user at IdeFisk Login as Agents on Asterisk B at this moment we have the next Licence Use:  A) 1/1  B) 1/0   When a Call from the QUEUE on Asterisk B is Bridge to the Agent I have the next Use:
 A) 1/1     B) 1/3Any one can explain me this ?, why the incress of licence consumptions.Thanks.
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Re: [Asterisk-Users] grandstream GXV-3000

[Alvaro Parres]

I think here are 2 mixed subject  One Substitution of the GXP 3000 Video Phone Phone with a great speaker phone. For the second Subject a think Polycom are the greatest.
On 6/13/06, Steve Underwood <[EMAIL PROTECTED]> wrote:
Mike Fedyk wrote:> Or any polycom phone that has speakerphone like the IP501 and IP430.>> Time Bandit wrote: Can you, or anyone else comment on the speakerphone ability of the
>>> GVX-3000>>> ?   We run the GXP-2000's and for the most part are happy with them,>>> but for>>> upper management we're looking at phones with better speakerphone.
>>> These>>> would be ideal if the speakerphone isn't as terrible as the GXP-2000. I never tried the GVX-3000, but I can recommend with confidence a>> Cisco 7940 or 7960 for the quality of the speakerphone.
>And how good do you find the video on the Cisco 7940, 7960 and PolycomIP501, IP430? Personally, I've never seen them do video, but if you aresuggesting them as a substitute for a GVX-3000 I guess I must be wrong. :-)
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Re: [Asterisk-Users] Voicemail limit?

[Alvaro Parres]

I have more than 100 users with out problemand i'm using the file no db.On 3/23/06, Antonio Rabena <
[EMAIL PROTECTED]> wrote:You can try using asterisk-addons
http://www.voip-info.org/wiki/view/Asterisk+voicemail+database orasterisk realtimehttp://www.voip-info.org/wiki/view/Asterisk+RealTime+Voicemail
At 04:51 AM 3/23/2006, you wrote:>Hi,> Is there a howto to do this? I'm using voicemail.conf and> sip.conf for my voicemail users. Does it really has a limit?>>Thanks,
>Ryan>>At 08:23 PM 3/23/2006, Antonio Rabena wrote:>>How about moving your voicemail users into db?At 03:50 AM 3/23/2006, you wrote:>>>Hi,>>> Is there an account limit for voicemail? I have 80+ users
>>> in the voicemail and I can only reach the 70-ieth user. If there>>> is a limit how can I increase it to hundred for example?>>Thanks,>>>Ryan
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Re: [Asterisk-Users] Page about 70 users crash my Asterisk

[Alvaro Parres]

I have here de backtrace resultUsing host libthread_db library "/lib/libthread_db.so.1".Core was generated by `asterisk -g'.Program terminated with signal 11, Segmentation fault.

#0  0xb7ece142 in ?? ()As I see it was in the libthread library.. So can it confirmmy theory that is a memory problem ?On 3/23/06, 
BJ Weschke <[EMAIL PROTECTED]> wrote:

On 3/23/06, Alvaro Parres <[EMAIL PROTECTED]> wrote:> Hi list, i have and asterisk into a Pentium IV Server with 1GB of RAM
> about 75 Polycom Phones, one E1 for incoming calls.
>> We have program a page system with the page command and the auto answer> funtion> of polycom.>> We have detect via diaplan if the phone isn't in call we place the call. All

> this via Macro.>> But in the our that they are not many calls. So much user that can be page..> The Asterisk> crash.  We think it is a RAM Memory problem..>> Do you have any idea for this ?
> It's nearly impossible to tell without a core dump or backtrace ofthe core file, but there have been a few key fixes to 1.2.X and /trunkrecently in app_meetme that may solve a problem you're having here.
Since app_page depends on app_meetme to function, you may want toupgrade to the latest version that's appropriate for you and thenretest.--Bird's The Word Technologies, Inc.

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Re: [Asterisk-Users] Page about 70 users crash my Asterisk

[Alvaro Parres]

I have here de backtrace resultUsing host libthread_db library "/lib/libthread_db.so.1".Core was generated by `asterisk -g'.Program terminated with signal 11, Segmentation fault.
#0  0xb7ece142 in ?? ()As I see it was in the libthread library.. So can it confirmmy theory that is a memory problem ?On 3/23/06, 
BJ Weschke <[EMAIL PROTECTED]> wrote:
On 3/23/06, Alvaro Parres <[EMAIL PROTECTED]> wrote:> Hi list, i have and asterisk into a Pentium IV Server with 1GB of RAM> about 75 Polycom Phones, one E1 for incoming calls.
>> We have program a page system with the page command and the auto answer> funtion> of polycom.>> We have detect via diaplan if the phone isn't in call we place the call. All
> this via Macro.>> But in the our that they are not many calls. So much user that can be page..> The Asterisk> crash.  We think it is a RAM Memory problem..>> Do you have any idea for this ?
> It's nearly impossible to tell without a core dump or backtrace ofthe core file, but there have been a few key fixes to 1.2.X and /trunkrecently in app_meetme that may solve a problem you're having here.
Since app_page depends on app_meetme to function, you may want toupgrade to the latest version that's appropriate for you and thenretest.--Bird's The Word Technologies, Inc.
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[Asterisk-Users] Page about 70 users crash my Asterisk

[Alvaro Parres]

Hi list, i have and asterisk into a Pentium IV Server with 1GB of RAMabout 75 Polycom Phones, one E1 for incoming calls.We have program a page system with the page command and the auto answer funtionof polycom.
We have detect via diaplan if the phone isn't in call we place the call. All this via Macro.But in the our that they are not many calls. So much user that can be page.. The Asteriskcrash.  We think it is a RAM Memory problem..
Do you have any idea for this ?Thanks.
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Re: [Asterisk-Users] Unicall E1 Error in Mexico

[Alvaro Parres]

Jorge nosotros tenemos problemas de fax solo con lineas de telemex con
lineas de axtel no. Y no hemos podido tampoco detectar el problemas.

SI logras resolverlas hasnolo saber porfabor.
On 12/27/05, Martinez Felix <[EMAIL PROTECTED]> wrote:
no podria decirte, porqe tengo problemas con los scripts de email2fax y
Asterfax...espero resolverlos pronto y verificar el correcto envio de
faxes...On 12/21/05, Jorge Cisneros <
[EMAIL PROTECTED]> wrote:
gracias Felix por el tip, ya lo hice y si funciono todo bien. tengo
otro problema no puedo enviar fax a traves de las lineas de unicall
creo que el problema esta en la cancelacion de echo. Tu has tenido este
problema

GRacias 
On 12/21/05, Martinez Felix <[EMAIL PROTECTED]
> wrote:
Es un timeout...necesitas incrementarlo...en la libreria de unicall existe un archivo qe se llama mfcr2.c...

#define BLOCKING_RELEASE_TIME   450
#define
ANSWER_GUARD_TIME  
100#define
DEFAULT_T1 
5000  <-Dale una valor mas alto...2 por ejemplo
#define
DEFAULT_T1A
150
#define
DEFAULT_T1B
6
#define
DEFAULT_T2 
5000
#define
DEFAULT_T3 
15000

vuelves a compilar y a instalar y listo...
On 12/20/05, Jorge Cisneros <


[EMAIL PROTECTED]> wrote:

Hi 
 
  I have a weired problem. when i make a call with some numer the unicall show me a error. 
 
For example when i dial 30003300 in mexico city then log show 
 
MFC/R2 UniCall/3 R2 prot. err. [2/ 
40/Group I  
/DNIS ] cause 32769 -
T1 timed outDec 21 00:22:46 WARNING[17649]: MFC/R2 UniCall/3 8 off
-> 
[1/  
1/Idle 
/Idle ]Dec 21
00:22:46 WARNING[17649]: MFC/R2 UniCall/3 1001 
-> 
[1/  
1/Idle 
/Idle ]
Dec 21 00:22:46 WARNING[17649]: Unicall/3 event Protocol failureDec 21 00:22:46 VERBOSE[17649]: -- Unicall/3 protocol error. Cause 32769 
 
But with other number work fine. The problem is only with a few number.
 
thanks
 
 

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[Asterisk-Users] A company that sells Toll Free Number in USA

[Alvaro Parres]

Hi any one can recommend me a company in the USA that can sell me a Toll Free Number
and send me the call via IP. 

Thanks.


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[Asterisk-Users] Snom 360 and 320 AutoAnswer

[Alvaro Parres]

Hi list..
 
   I want to make Snom 360 and 30 to autoanswer so i can have a paging sistem.
 
I tried tu send intercom=true with the little patch to chan_sip.c and it didm't work
 
any one have and idea of ow to do this.
 
 
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Re: [Asterisk-Users] Connecting 2 Asterisk using SIP

[Alvaro Parres]

Why using SIP instead of IAX2 ???
 
Only a question becouse i always use IAX
 
 
On 12/6/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote:
Well... not so perfectly.What I'm experiencing is that during certain call volumes, many calls
go thru from box1 to box2. However, there are some cases where I getthis message:Dec  6 11:11:19 WARNING[203]: chan_sip.c:9525 handle_response_invite:Forbidden - wrong password on authentication for INVITE to
'"5095551212" ;tag=as3e387d65'and the caller gets busy signal. However, other callers calling thesame number get thru with no problems. Why is this?
Thanks,WaldoOn Dec 5, 2005, at 10:30 AM, Waldo Rubinstein wrote:> This worked perfectly.>> Thanks,> Waldo>> On Dec 5, 2005, at 4:32 AM, xcel wrote:>
 Try this ___>> 1st Machine sip.conf [box2]>> username=box1>> type=friend>> host=
10.0.0.2>> secret=* in extensions.conf exten => _XX,1,Dial(SIP/box2/${EXTEN}) __
>> 2nd Machine sip.conf [box1]>> username=box2>> type=friend>> host=10.0.0.1>> secret=* in 
extensions.conf>> exten => _X,1,Dial(SIP/box1/${EXTEN}) --xce>> *** REPLY SEPARATOR  *** On 12/5/2005 at 12:11 AM Waldo Rubinstein wrote:
> I have 2 Asterisk servers running 1.2.0. One of them is a PSTN>>> gateway. Currently they are connected using IAX2. I wanted to play>>> with SIP.>> I setup a sip entry (type=friend) in the PSTN gateway box and a sip
>>> entry (type=user) in the second box in order to send calls using SIP>>> to the second box. This works fine. However, when I setup the second>>> box as type=friend in order for it to be able to send calls back to
>>> the gateway box, then calls no longer work from gateway box to the>>> second box. The reported error is:>> Dec  5 00:07:14 NOTICE[203]: chan_sip.c:9514 handle_response_invite:
>>> Failed to authenticate on INVITE to '"2125551212" >>> [EMAIL PROTECTED]>;tag=as0698b1b9'>> In the gateway box, my 
sip.conf looks like this:>> [general]>>> allowguest=yes>>> autocreatepeer=no>> [secondbox]>>> type=friend>>> host=
10.0.0.2>>> secret=mysecret>> In the second box, my sip.conf looks like this:>> [general]>>> allowguest=yes
>>> autocreatepeer=no>> [secondbox]>>> type=user>>> host=10.0.0.1>>> secret=mysecret>> Any ideas on how to correctly set this up?
>> Thanks,>>> Waldo>>> ___>>> --Bandwidth and Colocation provided by Easynews.com
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Re: [Asterisk-Users] Is a BUG ? Hints and incominglimit

[Alvaro Parres]

which version of Asterisk do you have ?, Becouse when i change the function to your code, every time that one phone with call-limit the Asterisk crash.
 
I have 1.2.0 
On 12/3/05, Paradise Dove <[EMAIL PROTECTED]> wrote:
hi,This is the new update_call_counter() which works fine for me:/*! \brief  update_call_counter: Handle call_limit for SIP users
* Note: This is going to be replaced by app_groupcount* Thought: For realtime, we should propably update storage with inusecounter... */static int update_call_counter(struct sip_pvt *fup, int event){
   char name[256];   int *inuse, *call_limit;   int outgoing = ast_test_flag(fup, SIP_OUTGOING);   struct sip_user *u = NULL;   struct sip_peer *p = NULL;   if (option_debug > 2)   ast_log(LOG_DEBUG, "Updating call counter for %s call\n",
outgoing ? "outgoing" : "incoming");   /* Test if we need to check call limits, in order to avoid  realtime lookups if we do not need it */   if (!ast_test_flag(fup, SIP_CALL_LIMIT))
   return 0;   ast_copy_string(name, fup->username, sizeof(name));   /* Check the list of users */   // paradise dove   p = find_peer(name, NULL, 1);   if (p) {   inuse = &p->inUse;
   call_limit = &p->call_limit;   } else if (!u) {   /* Try to find user */   u = find_user(name, 1);   if (u) { inuse = &u->inUse; call_limit = &u->call_limit;
   } else {   if (option_debug > 1)   ast_log(LOG_DEBUG, "%s is not a local user, no calllimit\n", name);   return 0;   }   }   switch(event) {
   /* incoming and outgoing affects the inUse counter */   case DEC_CALL_LIMIT:   if ( *inuse > 0 ) {   (*inuse)--;   } else {   *inuse = 0;   }
   if (option_debug > 1 || sipdebug) {   ast_log(LOG_DEBUG, "Call %s %s '%s' removed from calllimit %d\n", outgoing ? "to" : "from", u ? "user":"peer"
   }   break;   case INC_CALL_LIMIT:   if (*call_limit > 0 ) {   if (*inuse >= *call_limit) {   ast_log(LOG_ERROR, "Call %s %s '%s' rejected due
to usage limit of %d\n", outgoing ? "to" : "from", u ? "u   // paradise dove   if (p)   ASTOBJ_UNREF(p,sip_destroy_peer);
   else if (u)   ASTOBJ_UNREF(u,sip_destroy_user);   return -1;   }   }   (*inuse)++;   if (option_debug > 1 || sipdebug) {
   ast_log(LOG_DEBUG, "Call %s %s '%s' is %d out of%d\n", outgoing ? "to" : "from", u ? "user":"peer", name, *in   }   break;
   default:   ast_log(LOG_ERROR, "update_call_counter(%s, %d) calledwith no event!\n", name, event);   }   // paradise dove   if (p)   ASTOBJ_UNREF(p,sip_destroy_peer);
   else if (u)   ASTOBJ_UNREF(u,sip_destroy_user);   return 0;}Paradise DoveOn 12/2/05, Alvaro Parres <[EMAIL PROTECTED]> wrote:> Could you send it patch please.
>>>>> On 11/30/05, Paradise Dove <[EMAIL PROTECTED]> wrote:> >> > btw, i've patched this part of code and now its working fine for me.
> > i'm going to upload it.> >> > Paradise Dove> >> > On 11/30/05, Kevin Hanson <[EMAIL PROTECTED]> wrote:> > > Paradise Dove wrote:
> > >> > > >>Yes with version 1.2. I have tried already with call-limit and the> same.> > > >>> > > >>> > > >i agree with you, it seems to be a bug which i've submited before (bug
> > > >#5281) but it's now closed by bug marshals!> > > >> > > >> > > >> > > It's not closed.  It's suspended waiting input from you:> > >
> > > "Closing until the appropriate debug/trace output can be provided."> > >> > > On 10/30 you said you were still trying to get the debug output.> > >> > > Cheers,
> > > Kevin> > > ___> > > --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> > >
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[Asterisk-Users] Re: DIAXY to DIAXY problems

[Alvaro Parres]

This is the output at the CLI when this happend.
 
    -- Accepted AUTHENTICATED TBD call from 201.128.234.38    -- Accepting DIAL from 201.128.234.38, formats = 0x4    -- Executing Dial("
IAX2/[EMAIL PROTECTED]/5", "IAX2/111|120|tT") in new stack    -- Called 111    -- Call accepted by 201.153.202.214 (format ulaw)    -- Format for call is ulaw
    -- IAX2/111/7 is ringing    -- IAX2/111/7 answered IAX2/[EMAIL PROTECTED]/5    -- Attempting native bridge of IAX2/[EMAIL PROTECTED]/5 and IAX2/111/7
    -- Hungup 'IAX2/111/7'  == Spawn extension (home, 111, 1) exited non-zero on 'IAX2/[EMAIL PROTECTED]/5'    -- Hungup 'IAX2/[EMAIL PROTECTED]/5' 

 
Any IDEA ??
 
 
 
On 12/2/05, Alvaro Parres <[EMAIL PROTECTED]> wrote:

Hi list:
 
   I'm having problem with some DIAXY ATA FROM DIGIUM, I have 3 of them in different points, all of them register
to a central asterisk server. If i call from any of the ATA's to  Asterisk or Asterisk's to ATAs. But when any ATA's want to talk

to another ATA's.. The ATA's rings, but when the call is establish it hangups... And at the CLI i don't have any error.
 
Any Idea ???
 
Thanks
 
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[Asterisk-Users] DIAXY to DIAXY problems

[Alvaro Parres]

Hi list:
 
   I'm having problem with some DIAXY ATA FROM DIGIUM, I have 3 of them in different points, all of them register
to a central asterisk server. If i call from any of the ATA's to  Asterisk or Asterisk's to ATAs. But when any ATA's want to talk
to another ATA's.. The ATA's rings, but when the call is establish it hangups... And at the CLI i don't have any error.
 
Any Idea ???
 
Thanks
 
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Re: [Asterisk-Users] Is a BUG ? Hints and incominglimit

[Alvaro Parres]

Could you send it patch please.
 
 
On 11/30/05, Paradise Dove <[EMAIL PROTECTED]> wrote:
btw, i've patched this part of code and now its working fine for me.i'm going to upload it.Paradise Dove
On 11/30/05, Kevin Hanson <[EMAIL PROTECTED]> wrote:> Paradise Dove wrote:>> >>Yes with version 1.2. I have tried already with call-limit and the same.
> >>> >>> >i agree with you, it seems to be a bug which i've submited before (bug> >#5281) but it's now closed by bug marshals!> >> >> >> It's not closed.  It's suspended waiting input from you:
>> "Closing until the appropriate debug/trace output can be provided.">> On 10/30 you said you were still trying to get the debug output.>> Cheers,> Kevin> ___
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[Asterisk-Users] Help with a Company or Site for a DEMO. AYUDA con una empresa para una DEMO

[Alvaro Parres]

(ENGLISH VERSION AT THE END)
Hola lista:
    
 Requiero saber si alguien tiene un cliente o empresa donde se encuentren montado algun
Asterisk como PBX de tamaño mediano (al menos unas 50 extensiones).  Esto para dar una
demostracion a un cliente mio que esta interesado en invertir en Asterisk.
 
    Les pido que todos los que conoscan algo asi, me contacten ya sea a este correo fuera de la
lista o a mi correo arabe AT xmarts.com.mx
 
   Gracias.
 
 
Hi List:
   
 I want to now if any one have a client or company with a asterisk with more or less 50 extensiones,
This for a demostration that i need to give to a client who want to invert on Asterisk.
 
    I ask to all if have some one like this contact me out of the list, to this email or to arabe AT xmarts.com.mx
 
Thanks.
 
 
 
 
 
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Re: [Asterisk-Users] Is a BUG ? Hints and incominglimit

[Alvaro Parres]

Yes with version 1.2. I have tried already with call-limit and the same.
 
On 11/28/05, Kevin Hanson <[EMAIL PROTECTED]> wrote:
Alvaro Parres wrote:> Hi list...>>  I have been testing the hint extension. And i detect
> that when i have in the sip.fg of the extension the> incominiglimit=X (any number) the hint doesn't work all the> time show the extesion as idle.>>>  If this is a bug or not ??
>> Thanks.>>>>>___>>What version of Asterisk?  
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[Asterisk-Users] Is a BUG ? Hints and incominglimit

[Alvaro Parres]

Hi list...

 I have been testing the hint extension. And i detect
that when i have in the sip.cfg of the extension the
incominiglimit=X (any number) the hint doesn't work all the
time show the extesion as idle.


 If this is a bug or not ??

Thanks.


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Re: [Asterisk-Users] SNOM and 1.0.9

[Alvaro Parres]

Josheph:

   I had have that problem, and it get solve when i take out the incominglimit from my sip.cfg

   Also if you send you sip.cfg and extensions.cfg will be easier to help you

Tray it.

Alvaro Parres
On 11/28/05, BJ Weschke <[EMAIL PROTECTED]> wrote:
On 11/28/05, Kevin Hanson <[EMAIL PROTECTED]> wrote:> Joseph Rothstein wrote:>> >Greetings to all,> >> >I am trying to get the line lights on a SNOM 320 to work using 'hint' in
> >extensions.conf. Unfortunately I have not been able to get it to work> >properly.> >> >Does anyone know for sure if the hint function works properly in 1.0.9?> >> >If anyone has gotten this to work properly under 
1.0.9 please post a sample.> > This is definitely a 1.2 only feature. It is not in 1.0.9.--Bird's The Word Technologies, Inc.http://www.btwtech.com/
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Re: [Asterisk-Users] Polycom IP50X Park Softkey

[Alvaro Parres]

Serach in the list, about 1 o 2 weeks ago.. there is a guide for how to setup the key with asterisk
On 11/25/05, Mojo with Horan & Company, LLC <[EMAIL PROTECTED]> wrote:
But more importantly, what would you do with it if you found it?  Hasanybody made this softkey interface with Asterisk's parking functionality?Anyway, is this what you want?  in ipmid.cfg,HTH!MojoBob Knight wrote:> I am now running sip 1.6.2 with a 2.6.1 bootrom.> After moving from a 1.5 I now only see 2 softkeys at the main window:
> New Call and Forward.>> How do I get a Park softkey?--Mojo <[EMAIL PROTECTED]>Office Manger, Horan & Company, LLC(907) 747- x112
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Re: [Asterisk-Users] "Local Directory" feature on Polycom Soundpo int 501s

[Alvaro Parres]

we have TFTP and also those files are created upgrade automatic.. And
also we create manually the file  for the new phones so they have
the minimal addres book of the company.

On 11/25/05, Watkins, Bradley <[EMAIL PROTECTED]> wrote:
Hrmmm... I'm not sure how much more help I can be on this exactly.  For allof my users, we use FTP and the files get created and updated automatically.I should note that this is with all IP600s/601s but this should be the same
even for the 501s.- Brad-Original Message-From: [EMAIL PROTECTED][mailto:
[EMAIL PROTECTED]] On Behalf Of hugolivudeSent: Friday, November 25, 2005 12:44 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] "Local Directory" feature on Polycom Soundpo
int 501sI tried changing the name to the MAC address format, but still no luck.  Nocontacts appear after re-boot and I still can't add them manually either.No particular reason for using TFTP over FTP.  I'm a hack so I just followed
the instructions at:http://www.voip-info.org/wiki/index.php?page=Polycom+Soundpoint+IP+501Thanks,HughOn 11/25/05, Watkins, Bradley <
[EMAIL PROTECTED]> wrote:> The filename needs to be -directory.xml, not > Address>-directory.xml.  Grab the MAC off the back of the phone.
>> This is the same as for the provisioning files if you are using your> TFTP server to do that.  Also, is there a reason that you aren't using> FTP?  It's much more robust, and does not require that you
> pre-configure the directory file (unless you want to fill in specific> values).>> - Brad>> -Original Message-> From: 
[EMAIL PROTECTED]> [mailto:[EMAIL PROTECTED]] On Behalf Of> hugolivude> Sent: Friday, November 25, 2005 11:55 AM
> To: asterisk-users@lists.digium.com> Subject: [Asterisk-Users] "Local Directory" feature on Polycom Soundpoint> 501s>>
> I cannot seem to get the "Local Directory" feature to work.  I've> consulted section 3.1.17 of the Administrator Guide.  It says to put a> file > address>-directory.xml (where  is the IP address of
> address>the> phone) into the TFTP directory.  Polycom provides a template.>> The IP address for one of my phones is 192.168.0.113, so I placed the> file 
192168000113-directory.xml (shown below) into my TFTP directory,> but the local directory on the phone was not updated when it rebooted.> I think the TFTP is configured correctly because the phone has no
> problem loading the firmware etc.>> I thought it might have something to do DHCP, so I gave the phone a> static IP address, still no luck.  Next I tried entering the info> manually right on the phone.  I went into the "Contact Directory"
> pressed "Add" and entered the infor for a contact.  I pressed "Save"> and the phone came back with "No Records".  Occasionally I see a> message "Busy.  Please try again..."
>> Anyone have better success?>> Thanks,> Hugh>>> >  > > >
Smith>
Fred>
301>
1>
3>
>
0>
0>
0>
0> > >
Jones>
Bob>
302>
2>
3>
>
0>
0>
0>
0> > >
Ng>
Vin>
333>
3>
3>
>
0>
0>
0>
0> > > > ___> --Bandwidth and Colocation sponsored by 
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Re: [Asterisk-Users] Call parking on Polycom IP501

[Alvaro Parres]

Hi... I have the polycom 301 with firmware 1.6.3

When i Press Park, i get a dialog to enter a extension.

A dial 700 ther

and the call get parked, and i recive a call announceme where the calls was parked.

is this normal ???
On 11/24/05, Alvaro Parres <[EMAIL PROTECTED]> wrote:
i have the 1.6.3 firmware and also when i press park i need to dial another extension..

On 11/24/05, Adam Goryachev <
[EMAIL PROTECTED]> wrote:
What firmware version did you use for the polycom phone ??I just tried it on my IP600, and when I press the park button, it waitsfor me to dial an extension number, then I press park again, and it justhangs up the call.
Thanks,AdamOn Tue, 2005-11-22 at 13:56 -0800, Anthony Rodgers wrote:> Hi there,>> Instead of asking a question, I thought I'd post an answer. I got the> Polycom IP501 'Park' softkey working with * by doing the following:
>> features.conf:>> [general]> parkext => 1000> parkpos => 1001-1009> context => parkedcalls> parkingtime => 120> transferdigittimeout => 3

> courtesytone = beep>> Nothing unusual there. Here's the neat bit:>> extensions.conf:>> [internal] ; or whatever the relevant context is for you - it's usually> wherever your Polycom lives
> include => parkedcalls> exten =>> callpark,1,ParkAndAnnounce(pbx-transfer:PARKED|120|SIP/> ${DIALEDPEERNUMBER}|internal,${DIALEDPEERNUMBER},1)>> By using SIP DEBUG, I discovered that the Polycom attempts to re-invite
> the call to an extension called callpark. I couldn't get Park() to work> (it announces the stall number to the parked caller, instead of the> parker, for some reason), but using ParkAndAnnouce puts the parked call
> on hold, hangs up the parker and then immediately calls them back with> an announcement of the stall number.>> Hope this helps someone out..>> Regards,___
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Re: [Asterisk-Users] Call parking on Polycom IP501

[Alvaro Parres]

i have the 1.6.3 firmware and also when i press park i need to dial another extension..

On 11/24/05, Adam Goryachev <[EMAIL PROTECTED]> wrote:
What firmware version did you use for the polycom phone ??I just tried it on my IP600, and when I press the park button, it waitsfor me to dial an extension number, then I press park again, and it justhangs up the call.
Thanks,AdamOn Tue, 2005-11-22 at 13:56 -0800, Anthony Rodgers wrote:> Hi there,>> Instead of asking a question, I thought I'd post an answer. I got the> Polycom IP501 'Park' softkey working with * by doing the following:
>> features.conf:>> [general]> parkext => 1000> parkpos => 1001-1009> context => parkedcalls> parkingtime => 120> transferdigittimeout => 3
> courtesytone = beep>> Nothing unusual there. Here's the neat bit:>> extensions.conf:>> [internal] ; or whatever the relevant context is for you - it's usually> wherever your Polycom lives
> include => parkedcalls> exten =>> callpark,1,ParkAndAnnounce(pbx-transfer:PARKED|120|SIP/> ${DIALEDPEERNUMBER}|internal,${DIALEDPEERNUMBER},1)>> By using SIP DEBUG, I discovered that the Polycom attempts to re-invite
> the call to an extension called callpark. I couldn't get Park() to work> (it announces the stall number to the parked caller, instead of the> parker, for some reason), but using ParkAndAnnouce puts the parked call
> on hold, hangs up the parker and then immediately calls them back with> an announcement of the stall number.>> Hope this helps someone out..>> Regards,___
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Re: [Asterisk-Users] Re: Call parking on Polycom IP501

[Alvaro Parres]

How do you enable the parked soft key 
 
 
On 11/22/05, Noah Miller <[EMAIL PROTECTED]> wrote:
Hi Anthony -> Instead of asking a question, I thought I'd post an answer. I got the> Polycom IP501 'Park' softkey working with * by doing the following:
You are my favorite person today!  This rocks, and solves an annoyingproblem I've been trying to figure out for a while (single button parking).I didn't even realize there was a park softkey, because I don't have that
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Re: [Asterisk-Users] Infinitum bloquenado SIP ???? / Is Infinitum

[Alvaro Parres]

Esto lo hemos estado detectando en la ciudad de Guadalajara
 
On 11/15/05, Servers-R-Us <[EMAIL PROTECTED]> wrote:
Hola Alvaro,En qué parte de México tienes * y desde qué partes de México te conectas?Nosotros tenemos clientes en BCS, Oaxaca, Puebla, Morelos y el DF que se
conecan con Infinitum y no hemos tenido problemas recientemente. A principiodel año tuvimos muchos problemas en BCS, por dos meses o más, perosúbitamente desaparecieron.Saludos,Jorge Gonzalez
High Sierra Networks, Inc.www.highsierranetworks.com+1-775-236-5841___--Bandwidth and Colocation sponsored by 
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[Asterisk-Users] Infinitum bloquenado SIP ???? / Is Infinitum blocling SIP ????

[Alvaro Parres]

Hola lista:
 
 Para todos los de Mexico, desde hace aproximidamente 2 meses, estoy teniendo problemas con 
la trasmicion de voz via protocolo SIP. Entre Infinitums o desde Infinitum hacia otros provedores.
 
    Alguien mas esta experimentando esto ??? par poder corroborar que Telmex este bloqueando el 
protocolo SIP ?...
 
    Por que el protocolo IAX2 funciona a la perfeccion.
 
   Nota: Si se firman los equipos mas no viaja la voz despues de eso.
 
 
Espero respuesta
 
 
Hi list
   
    I'm having problems here in Mexico, connecting SIP devices via Infinitum ISP, as i see the ISP is
blocking the SIP traffic. 
 
   I want to knwo if any one is having this problem also. So i can confirm that Infinitum is doing this.
 
   Note: The devices get register but they can't send any more data as dialed number or voice.
 
Thanks
 
 
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[Asterisk-Users] HNT PROBLEM

[Alvaro Parres]

Hi list, i have the next problem: 

  I have conifgured hint for all my extension ( SIP and ZAP) but at the console
i send show hints and always all the channels are idle.. 

 My config files:

 at extension.conf

...

[sip-test]
exten => 101,hint,ZAP/35
exten => 101,1,Dial(ZAP/35)
exten => 102,hint,ZAP/35
exten => 102,1,Dial(ZAP/23)
exten => 111,hint,SIP/111
exten => 111,1,Dial(SIP/111)
exten => 112,hint,SIP/112
exten => 112,1,Dial(SIP/112)
exten => 113,hint,SIP/113
exten => 113,1,Dial(SIP/113)
exten => 121,hint,SIP/121
exten => 121,1,Dial(SIP/121)
exten => 122,hint,SIP/122
exten => 122,1,Dial(SIP/122)
exten => 132,hint,ZAP/36
exten => 132,1,Dial(ZAP/36)
exten => 141,hint,SIP/311
exten => 141,1,Dial(SIP/141)
...

all the phones have as context = home (where are more extension).
 
and only the SIP/116 (and snom phone) have the context and subscription context as sip-test

thanks

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[Asterisk-Users] sip.ld for a SoundStation IP 4000

[Alvaro Parres]

Hi does any one have the sip.ld file of a SoundStatios IP 4000

Thanks.

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[Asterisk-Users] Errors With Hint

[Alvaro Parres]

Hi list, i have the next problem:

  I create 3 hints.. (111 (SIP/111), 112 (SIP/112), and 102 (ZAP/35) )

   the SIP/111 is a GrandStream ATA

   the SIP/112 is a Polycom 301

   the ZAP/35 is a Analogic Phone.



The SIP/112 hints works great. But the other 2 no.



The ZAP/35 is say is always in USE and as you see en the

next console output is not in use. any Idea



asterisk*CLI>
    -= Registered Asterisk Dial Plan Hints =-

  
111
:
SIP/111  
State:Idle   
Watchers  4

  
102
:
ZAP/35   
State:InUse  
Watchers  5

  
112
:
SIP/112  
State:InUse  
Watchers  2



- 3 hints registered

asterisk*CLI> show cha

channel   channels  channeltypes

asterisk*CLI> show channels

Channel 
Location
State   Application(Data)

Zap/34-1
[EMAIL PROTECTED]:1   
Up  Bridged Call(SIP/112-1f3d)

SIP/112-1f3d
[EMAIL PROTECTED]: Up 
Dial(ZAP/34/3338182842|120|Tt)

2 active channels

1 active call



And also the SIP/111 is always in Idle any idea of why ???



thanks


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[Asterisk-Users] Re: SoundStation IP 4000 App and Cfg files.

[Alvaro Parres]

I think i only need the sip.ld file.. 

Becouse when the SS is booting it said that the sip.ld is not for that phone.

I have actually the sip.ld 1.6.x version.

thanks.On 11/10/05, Alvaro Parres <[EMAIL PROTECTED]> wrote:
Hi does any one have the Polycom SoundStation IP 4000 files for FTP

becouse my SS IP 4000 take the files of my IP SoundPoint 301 and 501...

and it apper to be no the sames.

Thanks..




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[Asterisk-Users] SoundStation IP 4000 App and Cfg files.

[Alvaro Parres]

Hi does any one have the Polycom SoundStation IP 4000 files for FTP

becouse my SS IP 4000 take the files of my IP SoundPoint 301 and 501...

and it apper to be no the sames.

Thanks..


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Re: [Asterisk-Users] Play message and dial extensions simultaneously

[Alvaro Parres]

But M play Hold Music. 

And what we need, as the other to users ask, is to play a especific file while the phone is rinning.On 11/10/05, 
[EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
You can play music instead of providing a ringtone. ( I think it's the Moption for the dial command)We used this for a reception solution so that the caller would not know thatthey were not being ignored.
PaulH- Original Message -From: "Hugh Jackman" <[EMAIL PROTECTED]>To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Thursday, November 10, 2005 7:04 PMSubject: Re: [Asterisk-Users] Play message and dial extensionssimultaneously
> Hi,> I've been looking for such solution without lucks. The prompts (either> by Playback or  Background or app_queues) will have to complete before> the Dial cmd kicks in, which takes a lot of time.
> Please let me know if you become aware of any solutions for this> apparently obvious problem.> Regards,> H.>> On 11/10/05, C F <[EMAIL PROTECTED]
> wrote:> > For what purpose?> > Have you tried:> > exten => s,1,Dial(SIP/,15&Local/[EMAIL PROTECTED])> >> > exten => 123,1,Background(custom/msg1)> >
> > It might not work, I have never tried something like this, but it mightwork.> >> > On 11/8/05, Mike Clark <[EMAIL PROTECTED]> wrote:
> > > Ok, this has to be simple and I'm just not seeing it. On and inbound> > > call, I want to play a specific message while simultaneously ringing> > > extensions. Its basically like music on hold and queues, but I need
the> > > message to always start from the beginning, not just play from wherethe> > > MOH process happens to be at that time. I tried Googling, but no luck.> > >> > > I did try
> > >> > > exten => 1,1,Answer> > > exten => 1,2,Wait(1)> > > exten => 1,3,BackGround(custom/msg1)> > > exten => 1,4,Dial(SIP/,15)> > >
> > > but it played the entire message before dialing.> > >> > > Thanks,> > >> > > Mike Clark> > > ___
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Re: [Asterisk-Users] Problems with HINT

[Alvaro Parres]

Thanks with the upgrade they work... Now i only have one problem.

I create 3 hints.. (111 (SIP/111), 112 (SIP/112), and 102 (ZAP/35) )
   the SIP/111 is a GrandStream ATA
   the SIP/112 is a Polycom 301
   the ZAP/35 is a Analogic Phone.

The SIP/112 hints works great. But the other 2 no.

The ZAP/35 is say is always in USE and as you see en the
next console output is not in use. any Idea

asterisk*CLI>
    -= Registered Asterisk Dial Plan Hints =-
  
111
:
SIP/111  
State:Idle   
Watchers  4
  
102
:
ZAP/35   
State:InUse  
Watchers  5
  
112
:
SIP/112  
State:InUse  
Watchers  2

- 3 hints registered
asterisk*CLI> show cha
channel   channels  channeltypes
asterisk*CLI> show channels
Channel 
Location
State   Application(Data)
Zap/34-1
[EMAIL PROTECTED]:1   
Up  Bridged Call(SIP/112-1f3d)
SIP/112-1f3d
[EMAIL PROTECTED]: Up 
Dial(ZAP/34/3338182842|120|Tt)
2 active channels
1 active call

And also the SIP/111 is always in Idle any idea of why ???

thanks
On 11/9/05, Peter Dean <[EMAIL PROTECTED]> wrote:
Upgrade to asterisk-1.2.0-rc1 and ensure that your sip file containssubscribecontext=sip-text.___--Bandwidth and Colocation sponsored by 
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[Asterisk-Users] Problems with HINT

[Alvaro Parres]

Hi list:

   I'm configuring in a CVS-HEAD asterisk ( about 2 weeks ago), some hint extension. But they are not working:


   /etc/asterisk/extensions.conf
   
   [sip-test]
   exten => 116,1,dial(SIP/116)
   exten => 112,hint,SIP/112
   exten => 112,1,dial(SIP/112)
   

   /etc/asterisk/sip.conf
   ...
 [112]

type=friend
; either "friend" (peer+user), "peer" or "user"
 context=sip-test

host=dynamic   
; we have a static but private IP address

nat=no 
; there is not NAT between phone and Asterisk

canreinvite=yes
; allow RTP voice traffic to bypass Asterisk 

dtmfmode=RFC2833   
; either RFC2833 or INFO for the BudgeTone

incominglimit=2
; permit only 1 outgoing call at a time

callgroup=2
; callgroup

pickupgroup=2  
; pickupgroup

[EMAIL PROTECTED]
; mailbox 1234 in voicemail context "default"  

allow=all  
; need to disallow=all before we can use allow=
 language=es
 progressinband=no
 qualify=yes
  

    when at the console i send a show hints i get this:

 -= Registered Asterisk Dial Plan Hints =-
  
112
:
SIP/112  
State:Unknown
Watchers  1
----
- 1 hints registered


 any idea ???

Thanks.

ALvaro Parres.


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Re: [Asterisk-Users] PRI E1 Problem only chan 17-31

[Alvaro Parres]

Also the people of Nortell told me that is posibble that mismatch that my channel 26 is they channel 25 ?
how can i fix that on asterisk ?? or on the nortell ?

Thanks.
On 11/3/05, Johann Steinwendtner <[EMAIL PROTECTED]> wrote:
Another problem could be that there is a B-channel mismatch.e.g. Asterisk uses channel 26 and Nortel uses channel 25. Thiscan be modified on at least QSIG trunks. But on EuroISDN thereshould not be a problem.
Hans[EMAIL PROTECTED] schrieb:>> On Wed, 2 Nov 2005, Alvaro Parres wrote:>>>>Hi list>>  I have a problem on a PRI E1 card.
>> The connection diagram is:>>  [ASTERISK] -- PRI-NET -- PRI-CPE -- [NORTELL]>> The problem is:>>  When i made a call in channels 17 to 31, there is no voice in any way...
>>but on channels 1 to 15 i have no problems...>>> The Nortel end only has 15 channels enabled?; probably they want you to> pay more to enable more?>> Steve>> ___
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Re: [Asterisk-Users] PRI E1 Problem only chan 17-31

[Alvaro Parres]

Hi all

I have already Try bchan=1-15,17-31

The nortell people have all the 30 channels enbled.

We are using ess5 (Lucent) ...

Any other suggest ?


On 11/3/05, Johann Steinwendtner <[EMAIL PROTECTED]> wrote:
Another problem could be that there is a B-channel mismatch.e.g. Asterisk uses channel 26 and Nortel uses channel 25. Thiscan be modified on at least QSIG trunks. But on EuroISDN thereshould not be a problem.
Hans[EMAIL PROTECTED] schrieb:>> On Wed, 2 Nov 2005, Alvaro Parres wrote:>>>>Hi list>>  I have a problem on a PRI E1 card.
>> The connection diagram is:>>  [ASTERISK] -- PRI-NET -- PRI-CPE -- [NORTELL]>> The problem is:>>  When i made a call in channels 17 to 31, there is no voice in any way...
>>but on channels 1 to 15 i have no problems...>>> The Nortel end only has 15 channels enabled?; probably they want you to> pay more to enable more?>> Steve>> ___
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[Asterisk-Users] E1 PRI card 17:31 channels problems

[Alvaro Parres]

Hi list
 
    I have a  problem on a PRI E1 card.
 
The connection diagram  is:
 
  [ASTERISK] -- PRI-NET -- PRI-CPE -- [NORTELL]
 
The problem is:
 
    When i made a call in channels 17 to 31, there is no voice in any way...
but on channels 1 to 15 i have no problems... 
 
 
Any suggestion ???
 
 
my config files
/etc/zaptel.confspan=1,0,0,ccs,hdb3bchan=1-15dchan=16bchan=17-31

loadzone = usdefaultzone=us
   
/etc/asterisk/zapata.conf[channels]
language=encontext=homeswitchtype=5ess
signalling=pri_netusecallerid=yeshidecallerid=nocallwaiting=yescallwaitingcallerid=yesthreewaycalling=yestransfer=yescancallforward=yescallreturn=yesechocancel=yesechocancelwhenbridged=yes

rxgain=0.0txgain=-5.0
group=1callgroup=1pickupgroup=1channel => 1-15channel => 17-31
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[Asterisk-Users] Problems with some channels on PRI E1 card

[Alvaro Parres]

Hi list
 
    I have a  problem on a PRI E1 card.
 
The connection diagram  is:
 
  [ASTERISK] -- PRI-NET -- PRI-CPE -- [NORTELL]
 
The problem is:
 
    When i made a call in channels 17 to 31, there is no voice in any way...
but on channels 1 to 15 i have no problems... 
 
 
Any suggestion ???
 
 
my config files
/etc/zaptel.confspan=1,0,0,ccs,hdb3bchan=1-15dchan=16bchan=17-31

loadzone = usdefaultzone=us
   
/etc/asterisk/zapata.conf[channels]
language=encontext=homeswitchtype=5ess
signalling=pri_netusecallerid=yeshidecallerid=nocallwaiting=yescallwaitingcallerid=yesthreewaycalling=yestransfer=yescancallforward=yescallreturn=yesechocancel=yesechocancelwhenbridged=yes

rxgain=0.0txgain=-5.0
group=1callgroup=1pickupgroup=1channel => 1-15channel => 17-31
 
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[Asterisk-Users] PRI E1 Problem only chan 17-31

[Alvaro Parres]

Hi list
 
    I have a  problem on a PRI E1 card.
 
The connection diagram  is:
 
  [ASTERISK] -- PRI-NET -- PRI-CPE -- [NORTELL]
 
The problem is:
 
    When i made a call in channels 17 to 31, there is no voice in any way...
but on channels 1 to 15 i have no problems... 
 
 
Any suggestion ???
 
 
my config files
/etc/zaptel.confspan=1,0,0,ccs,hdb3bchan=1-15dchan=16bchan=17-31

loadzone = usdefaultzone=us
   
/etc/asterisk/zapata.conf[channels]
language=encontext=homeswitchtype=5ess
signalling=pri_netusecallerid=yeshidecallerid=nocallwaiting=yescallwaitingcallerid=yesthreewaycalling=yestransfer=yescancallforward=yescallreturn=yesechocancel=yesechocancelwhenbridged=yes

rxgain=0.0txgain=-5.0
group=1callgroup=1pickupgroup=1channel => 1-15channel => 17-31
 
 
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[Asterisk-Users] HT-486 Voice Nat Problem

[Alvaro Parres]

Hi list, i have the next situation:
 
    B
    [HT486] -- (NAT/ROUTER)  INTERNET  - [* server] 
 |
 |
 |
  A|
 [HT486] -
 
Both HT486 register to * server, with no problem, but when they call each other
the voice only goes from B to A but not from A to B.
 
My configs
 
sip.conf
 
[B]type=friendcontext=ATAscallerid=<113>host=dynamic    ; we have a static but private IP addressnat=yesqualify=no ; there is not NAT between phone and
canreinvite=no ; allow RTP voice traffic to bypass Asteriskdtmfmode=inband ; either RFC2833 or INFO for the BudgeTone;incominglimit=2 ; permit only 1 outgoing call at a time
disallow=all    ; need to disallow=all before we can useallow=ulaw  ; Note: In user sections the order of codecsallow=alawsecret=xxgroup=1
 

[A]type=friendcontext=ATAscallerid=<113>host=dynamic    ; we have a static but private IP addressnat=yesqualify=no ; there is not NAT between phone and
canreinvite=no ; allow RTP voice traffic to bypass Asteriskdtmfmode=inband ; either RFC2833 or INFO for the BudgeTone;incominglimit=2 ; permit only 1 outgoing call at a time
disallow=all    ; need to disallow=all before we can useallow=ulaw  ; Note: In user sections the order of codecsallow=alawsecret=xxgroup=1
 
At the HT486 i have the next:
 
SIP SERVER=xxx.xxx.xxx.xxx.
NAT Traversal: = yes  IP:stun.xten.org
 
any idea or help ??
 
thanks
 
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[Asterisk-Users] PROBLEM WITH A PRI INCOMING CALLS

[Alvaro Parres]

Hi list, i have the next situation

I've a asterisk connect with a Nortel Meridian Op 11, via a PRI CARD with 5ess switch

[Asterisk] -- PRI - NET ---  PRI- CPE --[Nortell]

I can call from Asterisk to Nortell with no problem, but when Nortell
place a call to me, i have the channel bridge but no audio can hear y
any way..

Any idea ???

thanks.




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Re: [Asterisk-Users] Wildcard TE110P in Mexico

[Alvaro Parres]

Yes it works, the only thing is that you need to patch you asterisk for support R2 
 
 
On 9/23/05, Alex Kauffmann <[EMAIL PROTECTED]> wrote:


We have several in operation but with isdn and not R2.  I know I've seen emails from people that use them with Telmex and have them operating, albeit with some difficulty.


 
-Original Message-From: 
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
] On Behalf Of Jorge CisnerosSent: Friday, September 23, 2005 5:57 PMTo: 
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Wildcard TE110P in Mexico
 
Hi    I have one question, somebody  can tell me if the card TE110P work in mexico, and maybe can tell me the config. 
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[Asterisk-Users] Ask for config files of Nortell Meridian Op 11 & Asterisk for PRI

[Alvaro Parres]

Hi list, any one can let me his config files for interconecting a Meridian Op 11 and Asterisk
via a E1 PRI CARD. 
 
Actually i need the nortell config part, becouse my client nortell provider doesn't know
how to config the PRI card at his part.
 
Thanks all.
 
 
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Re: [Asterisk-Users] Grandstream GXP-2000 Poor sound Quality

[Alvaro Parres]

I have one GXP-2000 and i prefer the SPA 841 of SIPURA, (look their are same price) y the price is not a problem POLYCOM or SNOM
 
 
On 9/1/05, Joe McConnaughey <[EMAIL PROTECTED]> wrote:

Check out the Aastra 9133i.  Fantastic phone for about $179.  I have two of them and will be adding more.  More bang for the buck than the Polycoms but slightly less feature set.  These phones are more of an office standard like the Nortel or Avaya phones.

 
 
Message: 9Date: Thu, 1 Sep 2005 13:28:54 -0400From: Aaron W <
[EMAIL PROTECTED]>Subject: Re: [Asterisk-Users] Grandstream GXP-
2000 Poor sound QualityTo: Asterisk Users Mailing List - Non-Commercial Discussion<
asterisk-users@lists.digium.com>Message-ID: <
[EMAIL PROTECTED]>Content-Type: text/plain; charset="iso-8859-1"Thanks..I am begining to agree with you about these phones. Which poylcoms 
do you have? I have been looking at the polycom soundpoint IP501. It seems like a good phone for just under 200USD.Thanks again,Aaron ___
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Re: [Asterisk-Users] Asterisk and a Meridian Nortell Release 11

[Alvaro Parres]

In witch part... at Nortell or at Asterisk ??

and how to do this ?
On 8/29/05, Jerry Geis <[EMAIL PROTECTED]> wrote:
Sir,Can you turn off Multipart SDP headers?That was the problem I had.jerry___--Bandwidth and Colocation sponsored by 
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Re: [Asterisk-Users] Asterisk and a Meridian Nortell Release 11

[Alvaro Parres]

Actually we can make calls from the * to the nortel, with no problems. 

The problems begins when we want to make a call from the nortel to the *

On 8/29/05, Damon Estep <[EMAIL PROTECTED]> wrote:
Where does the CNAM originate, is it sent to the Nortel from the PSTNand then passed on to *, or does it originate on the Nortel?There may be another way, but without more info I do not wan to peak outof context.
> -Original Message-> From: [EMAIL PROTECTED] [mailto:asterisk-users-> 
[EMAIL PROTECTED]] On Behalf Of Anthony Rodgers> Sent: Monday, August 29, 2005 10:54 AM> To: Asterisk Users Mailing List - Non-Commercial Discussion> Subject: Re: [Asterisk-Users] Asterisk and a Meridian Nortell Release
11>> Hi there,>> We are using * with an Option 11C - we tried all of the various> protocols and the only one we could get to work satisfactorily was> 5ESS, with the * as CO and the Nortel as remote. The one drawback of
> this approach is getting name information for caller ID - because the> Nortel sees the * as CO, it won't send the name information.>> /etc/zaptel.conf:>> span=1,1,0,esf,b8zs
> bchan=1-23> dchan=24> #clear=1-24> loadzone = us> defaultzone=us>> /etc/asterisk/zapata.conf:>> [trunkgroups]>> [channels]>> context=incoming
> switchtype=5ess> usecallingpres=yes> echocancel=128> usecallerid=yes> echocancelwhenbridged=yes> echotraining=yes> echotraining=800>> rxgain=-4.0> txgain=-
6.0> group=1> callgroup=1> pickupgroup=1> signalling = pri_net> channel => 1-23>> musiconhold=default>> Any use?>> Regards,> --> Anthony Rodgers
> Business Systems Analyst> District of North Vancouver> Web: http://www.dnv.org> RSS Feed: http://www.dnv.org/rss.asp
>>> On 27-Aug-05, at 7:20 AM, Alvaro Parres wrote:>> > Hi, i have one Asterisk with a Digium E1 card, and a Meridian Nortel> > Release 11.> > I need to connect both of them. We are using MFC/R2 for this..
> >> > The Diagram:> >> > [ NORTEL ] ( AMI ) > > (DIGIUM) [ ASTERISK]> >> > we have green light at the digium card, and at asterisk we see all
31> > channels as idle.> >> > But when i want to recive a call from the Nortel to the Asterisk iget> > at the Nortel only a empty sound, and after about 15 o 20 sec it's
> > hangup.> >> > Any suggestion ?> >> > The log at Asterisk is:> >> > Aug 27 09:16:20 WARNING[6496]: chan_unicall.c:704 unicall_report:>
> MFC/R2 UniCall/30  <-
0001  [1/  
1/Idle  /Idle> >  ]> > Aug 27 09:16:20 WARNING[6496]: chan_unicall.c:704 unicall_report:> > MFC/R2 UniCall/30 Detected> > Aug 27 09:16:20 WARNING[6496]: chan_unicall.c:704 unicall_report:
> > MFC/R2 UniCall/30 Making a new call with CRN 32769> > Aug 27 09:16:20 WARNING[6496]: chan_unicall.c:704 unicall_report:>
> MFC/R2 UniCall/30
1101  ->  [2/  
2/Idle  /Idle> >  ]> > Aug 27 09:16:20 WARNING[6496]: chan_unicall.c:2865 handle_uc_event:> > Unicall/30 event Detected> > Aug 27 09:16:20 WARNING[6496]: chan_unicall.c:704 unicall_report:
>
> MFC/R2 UniCall/30  <-
0001  [1/  
1/Idle  /Idle> >  ]> > Aug 27 09:16:20 WARNING[6496]: chan_unicall.c:704 unicall_report:> > MFC/R2 UniCall/30 Detected> > Aug 27 09:16:20 WARNING[6496]: chan_unicall.c:704 unicall_report:
> > MFC/R2 UniCall/30 Making a new call with CRN 32769> > Aug 27 09:16:20 WARNING[6496]: chan_unicall.c:704 unicall_report:>
> MFC/R2 UniCall/30
1101  ->  [2/  
2/Idle  /Idle> >  ]> > Aug 27 09:16:20 WARNING[6496]: chan_unicall.c:2865 handle_uc_event:> > Unicall/30 event Detected> > tel2*CLI> Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704
>
> unicall_report: MFC/R2
UniCall/30  <-
1001  [2/   2/Seizeack> >/Seize ack]> > Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report:> > MFC/R2 UniCall/30 Far end disconnected(cause=Normal, unspecified
cause> > [31]) - state 0x2> > Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:2865 handle_uc_event:> > Unicall/30 event Far end disconnected> > Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:3198 handle_uc_event:
> > CRN 32769 - far disconnected cause=Normal, unspecified cause [31]> > Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report:> > MFC/R2 UniCall/30 Call control(6)> > Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report:
> > MFC/R2 UniCall/30 Drop call(cause=Normal Clearing [16])> > Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report:> >

[Asterisk-Users] Nortell Release 11 and Asterisk E1

[Alvaro Parres]

Hi, i have one Asterisk with a Digium E1 card, and a Meridian Nortel
Release 11.
I need to connect both of them. We are using MFC/R2 for this..

The Diagram:

[ NORTEL ] ( AMI ) 
(DIGIUM) [ ASTERISK]

we have green light at the digium card, and at asterisk we see all 31
channels as idle.

But when i want to recive a call from the Nortel to the Asterisk i get
at the Nortel only a empty sound, and after about 15 o 20 sec it's
hangup.

Any suggestion ?

The log at Asterisk is:

Aug 27 09:16:20 WARNING[6496]: chan_unicall.c:704 unicall_report:
MFC/R2 UniCall/30  <- 0001  [1/   1/Idle  /Idle
 ]
Aug 27 09:16:20 WARNING[6496]: chan_unicall.c:704 unicall_report:
MFC/R2 UniCall/30 Detected
Aug 27 09:16:20 WARNING[6496]: chan_unicall.c:704 unicall_report:
MFC/R2 UniCall/30 Making a new call with CRN 32769
Aug 27 09:16:20 WARNING[6496]: chan_unicall.c:704 unicall_report:
MFC/R2 UniCall/30 1101  ->  [2/   2/Idle  /Idle
 ]
Aug 27 09:16:20 WARNING[6496]: chan_unicall.c:2865 handle_uc_event:
Unicall/30 event Detected
Aug 27 09:16:20 WARNING[6496]: chan_unicall.c:704 unicall_report:
MFC/R2 UniCall/30  <- 0001  [1/   1/Idle  /Idle
 ]
Aug 27 09:16:20 WARNING[6496]: chan_unicall.c:704 unicall_report:
MFC/R2 UniCall/30 Detected
Aug 27 09:16:20 WARNING[6496]: chan_unicall.c:704 unicall_report:
MFC/R2 UniCall/30 Making a new call with CRN 32769
Aug 27 09:16:20 WARNING[6496]: chan_unicall.c:704 unicall_report:
MFC/R2 UniCall/30 1101  ->  [2/   2/Idle  /Idle
 ]
Aug 27 09:16:20 WARNING[6496]: chan_unicall.c:2865 handle_uc_event:
Unicall/30 event Detected
tel2*CLI> Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704
unicall_report: MFC/R2 UniCall/30  <- 1001  [2/   2/Seize ack
  /Seize ack]
Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report:
MFC/R2 UniCall/30 Far end disconnected(cause=Normal, unspecified cause
[31]) - state 0x2
Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:2865 handle_uc_event:
Unicall/30 event Far end disconnected
Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:3198 handle_uc_event:
CRN 32769 - far disconnected cause=Normal, unspecified cause [31]
Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report:
MFC/R2 UniCall/30 Call control(6)
Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report:
MFC/R2 UniCall/30 Drop call(cause=Normal Clearing [16])
Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report:
MFC/R2 UniCall/30 Call disconnected(cause=Normal, unspecified cause
[31]) - state 0x800
Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:2865 handle_uc_event:
Unicall/30 event Drop call
Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report:
MFC/R2 UniCall/30 Call control(7)
Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report:
MFC/R2 UniCall/30 Release call
Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report:
MFC/R2 UniCall/30 1001  ->  [1/1000/Clear fwd /Seize ack
 ]
Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report:
MFC/R2 UniCall/30  <- 1001  [2/   2/Seize ack /Seize ack
 ]
Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report:
MFC/R2 UniCall/30 Far end disconnected(cause=Normal, unspecified cause
[31]) - state 0x2
Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:2865 handle_uc_event:
Unicall/30 event Far end disconnected
Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:3198 handle_uc_event:
CRN 32769 - far disconnected cause=Normal, unspecified cause [31]
Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report:
MFC/R2 UniCall/30 Call control(6)
Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report:
MFC/R2 UniCall/30 Drop call(cause=Normal Clearing [16])
Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report:
MFC/R2 UniCall/30 Call disconnected(cause=Normal, unspecified cause
[31]) - state 0x800
Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:2865 handle_uc_event:
Unicall/30 event Drop call
Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report:
MFC/R2 UniCall/30 Call control(7)
Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report:
MFC/R2 UniCall/30 Release call
Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report:
MFC/R2 UniCall/30 1001  ->  [1/1000/Clear fwd /Seize ack
 ]
tel2*CLI> Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704
unicall_report: MFC/R2 UniCall/30 Release guard expired
Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report:
MFC/R2 UniCall/30 Destroying call with CRN 32769
Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report:
MFC/R2 UniCall/30 Release guard expired
Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report:
MFC/R2 UniCall/30 Destroying call with CRN 32769
tel2*CLI> Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:2865
handle_uc_event: Unicall/30 event Release call
   -- Unicall/30 released
Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:2865 handle

[Asterisk-Users] Asterisk and a Meridian Nortell Release 11

[Alvaro Parres]

Hi, i have one Asterisk with a Digium E1 card, and a Meridian Nortel
Release 11.
I need to connect both of them. We are using MFC/R2 for this..

The Diagram:

[ NORTEL ] ( AMI ) 
(DIGIUM) [ ASTERISK]

we have green light at the digium card, and at asterisk we see all 31
channels as idle.

But when i want to recive a call from the Nortel to the Asterisk i get
at the Nortel only a empty sound, and after about 15 o 20 sec it's
hangup.

Any suggestion ?

The log at Asterisk is:

Aug 27 09:16:20 WARNING[6496]: chan_unicall.c:704 unicall_report:
MFC/R2 UniCall/30  <- 0001  [1/   1/Idle  /Idle   
 ]
Aug 27 09:16:20 WARNING[6496]: chan_unicall.c:704 unicall_report:
MFC/R2 UniCall/30 Detected
Aug 27 09:16:20 WARNING[6496]: chan_unicall.c:704 unicall_report:
MFC/R2 UniCall/30 Making a new call with CRN 32769
Aug 27 09:16:20 WARNING[6496]: chan_unicall.c:704 unicall_report:
MFC/R2 UniCall/30 1101  ->  [2/   2/Idle  /Idle   
 ]
Aug 27 09:16:20 WARNING[6496]: chan_unicall.c:2865 handle_uc_event:
Unicall/30 event Detected
Aug 27 09:16:20 WARNING[6496]: chan_unicall.c:704 unicall_report:
MFC/R2 UniCall/30  <- 0001  [1/   1/Idle  /Idle   
 ]
Aug 27 09:16:20 WARNING[6496]: chan_unicall.c:704 unicall_report:
MFC/R2 UniCall/30 Detected
Aug 27 09:16:20 WARNING[6496]: chan_unicall.c:704 unicall_report:
MFC/R2 UniCall/30 Making a new call with CRN 32769
Aug 27 09:16:20 WARNING[6496]: chan_unicall.c:704 unicall_report:
MFC/R2 UniCall/30 1101  ->  [2/   2/Idle  /Idle   
 ]
Aug 27 09:16:20 WARNING[6496]: chan_unicall.c:2865 handle_uc_event:
Unicall/30 event Detected
tel2*CLI> Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704
unicall_report: MFC/R2 UniCall/30  <- 1001  [2/   2/Seize ack 
   /Seize ack]
Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report:
MFC/R2 UniCall/30 Far end disconnected(cause=Normal, unspecified cause
[31]) - state 0x2
Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:2865 handle_uc_event:
Unicall/30 event Far end disconnected
Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:3198 handle_uc_event:
CRN 32769 - far disconnected cause=Normal, unspecified cause [31]
Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report:
MFC/R2 UniCall/30 Call control(6)
Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report:
MFC/R2 UniCall/30 Drop call(cause=Normal Clearing [16])
Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report:
MFC/R2 UniCall/30 Call disconnected(cause=Normal, unspecified cause
[31]) - state 0x800
Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:2865 handle_uc_event:
Unicall/30 event Drop call
Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report:
MFC/R2 UniCall/30 Call control(7)
Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report:
MFC/R2 UniCall/30 Release call
Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report:
MFC/R2 UniCall/30 1001  ->  [1/1000/Clear fwd /Seize ack  
 ]
Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report:
MFC/R2 UniCall/30  <- 1001  [2/   2/Seize ack /Seize ack  
 ]
Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report:
MFC/R2 UniCall/30 Far end disconnected(cause=Normal, unspecified cause
[31]) - state 0x2
Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:2865 handle_uc_event:
Unicall/30 event Far end disconnected
Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:3198 handle_uc_event:
CRN 32769 - far disconnected cause=Normal, unspecified cause [31]
Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report:
MFC/R2 UniCall/30 Call control(6)
Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report:
MFC/R2 UniCall/30 Drop call(cause=Normal Clearing [16])
Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report:
MFC/R2 UniCall/30 Call disconnected(cause=Normal, unspecified cause
[31]) - state 0x800
Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:2865 handle_uc_event:
Unicall/30 event Drop call
Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report:
MFC/R2 UniCall/30 Call control(7)
Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report:
MFC/R2 UniCall/30 Release call
Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report:
MFC/R2 UniCall/30 1001  ->  [1/1000/Clear fwd /Seize ack  
 ]
tel2*CLI> Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704
unicall_report: MFC/R2 UniCall/30 Release guard expired
Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report:
MFC/R2 UniCall/30 Destroying call with CRN 32769
Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report:
MFC/R2 UniCall/30 Release guard expired
Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report:
MFC/R2 UniCall/30 Destroying call with CRN 32769
tel2*CLI> Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:2865
handle_uc_event: Unicall/30 event Release call
-- Unicall/30 released
Aug 27 09:16:36 WARN

Re: [Asterisk-Users] Asterisk Voice mail to nortel PBX Option 11

[Alvaro Parres]

Wich kind of E1 card do you use at the NORTEL ??

it was a PRI one??? witch model ???


On 8/12/05, Mark Phillips <[EMAIL PROTECTED]> wrote:
> Easily doable. I've done it twice now. Problem is that your users will
> never know they have messages waiting.
> 
> Install a T1/E1 card into the * box and then use a T1 cross-over cable
> between the 2 boxes.
> 
> Create a dialplan on the Meridian that points calls to the VM out over
> the new E1.
> 
> As for forwarding the calls when busy or no answer, that's a little more
> tricky. You'll have to come up with some rules and numbers to allow the
>   Meridian to decide what to do with those calls.
> 
> In my case I wrote a forward on no answer and a forward on busy rule for
> every phone that needed VM. When you called ext 200 the call was sent to
>mailbox 2200 on the *.
> 
> Users will have to get into the habit of calling the VM to check if
> there's messages.
> 
> Hope that helps.
> 
> Mark
> 
> craz sead wrote:
> > Hi all,
> >
> >
> > Could somebody help me, i wanna connect asterisk for
> > voice mail in the existing nortel pbx option 11 using
> > e1 card ?
> >
> > anyone have a clue ?  please help the conf. file
> >
> > thank all
> >
> > __
> > Do You Yahoo!?
> > Tired of spam?  Yahoo! Mail has the best spam protection around
> > http://mail.yahoo.com
> > ___
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
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> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> 
> --
> 
> Mark, G7LTT/KC2ENI
> Randolph, NJ
> http://www.g7ltt.com
> ___
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Re: [Asterisk-Users] Polycom IP301 and 501 with asterisk...

[Alvaro Parres]

Jonathan:

   Our provider continue selleing us SPA-841, if you want the contact,
mail me outside the list.

On 8/13/05, Chris Mason (Lists) <[EMAIL PROTECTED]> wrote:
> Tom Rymes wrote:
> 
> > Chris,
> >
> > Maybe you could write a generic config file and post it to the wiki?
> >
> I tried to post as a comment but the XML was excluded. How do I do that?
> 
> --
> Chris Mason
> NetConcepts
> (264) 497-5670 Fax: (264) 497-8463
> Int:  (305) 704-7249 Fax: (815)301-9759
> Cell: 264-235-5670
> Yahoo IM: [EMAIL PROTECTED]
> 
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Re: [Asterisk-Users] SPA 841 form SIPURA

[Alvaro Parres]

We have been using SIPURA and have no problem. With the last firmware
and silence supression off.



On 8/7/05, Paul Dugas <[EMAIL PROTECTED]> wrote:
> On Sun, August 7, 2005 1:15 pm, Thierry Wehr said:
> > This is not true
> > You have to switch to last firmware and/or disable silent suppression
> 
> I believe Thierry is not alone in having success with these units.  I
> cannot explain it but my guess is that there are some inconsistencies in
> their hardware or something in the myriad of configuration setting is
> awry.  I've been contacting Sipura for help with the problems but have
> received nothing other than "try this updated firmware".  I've been
> religiously checking for, and installing, any upgrades posted on their
> support site.  None have addressed the handset voice quality or inaudible
> speaker-phone problems I am getting.
> 
> I intend to keep one of the 7 I have as a test unit but will be ebay'ing
> the other 6; gently used, original packaging if anybody's interested ;)
> 
> Cheers!
> --
> Paul Dugas, Computer Engineer   Dugas Enterprises, LLC
> [EMAIL PROTECTED] phone: 404-932-1355   522 Black Canyon Park
> http://dugas.cc fax: 866-751-6494   Canton, GA 30114 USA
> ___
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Re: RE: [Asterisk-Users] ip phones

[Alvaro Parres]

I'm using SPA 841 form SIPURA and they work very nice, and are cheep
only 80 USD...
and all the options, (transfer, DND, conference, etc) work nice.


On 8/5/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> Hard phones.
> 
> Varun
> 
> - Original Message -
> From: Jason Walker <[EMAIL PROTECTED]>
> Date: Friday, August 5, 2005 10:35 am
> Subject: RE: [Asterisk-Users] ip phones
> 
> > Soft phones or hard phones?
> >
> > There are many free VOIP soft phones out there.
> >
> >
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [EMAIL PROTECTED] On Behalf Of
> > [EMAIL PROTECTED]
> > Sent: Thursday, August 04, 2005 9:57 PM
> > To: asterisk-users@lists.digium.com
> > Subject: [Asterisk-Users] ip phones
> >
> > Hello,
> >  I want to setup asterisk and do VOIP.
> >
> > Somebody from US has offered to get me ip phones.
> >
> > Can anybody suggest a few good and resonably priced phones
> > models.
> >
> > Thanks
> >
> > Varun
> >
> >
> > ___
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> > Asterisk-Users@lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
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> >
> 
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Re: [Asterisk-Users] Nortel Option 11 and TE110P of Digium

[Alvaro Parres]

Well the Actual Digramm of conecctions is:
E1   E1
PSTN --  NORTEL - ASTERISK
   Local Lines 

Actually the client the only think want is the solution working even
how much cost it.

The proyect was start by a employ and hi never can do this.. Thats why
i'm asking
to see if i take the proyect o not.

So with the PRI it's going to be easy all the work ??

Only one question the Nortel guys here, say that they need one more
clock to have a PRI card, is this correct 


On 8/5/05, Paul Belanger <[EMAIL PROTECTED]> wrote:
> Hello,
> 
> See comments inline
> 
> Alvaro Parres wrote:
> > Hi list:
> >
> > I have a client that needs to connect a Asterisk PBX with a TE110P
> > of Digium and one Nortel Option 11.
> >
> >Actually the Nortel Option 11 have a AMI E1 card. With it the have
> > problems of clock sync.
> Is the Nortel the CPE or Network side?
> >
> >They can change the AMI CARD to a PRI CARD, te questions are:
> >
> >  1) Which model of PRI is suggest for this ?
> NI2 or DMS100
> >  2) Some one have already do this ?
> Yup, have a site up with a TE405P.
> >  3) Is there form of correct de AMI problem ?
> Not sure what your asking.
> >
> >Well i hope that you will answered me.
> >
> > Alvaro Parres
> >
> > P.D. If any one from Mexica have done this before pleas contact me
> > (33) 35636261
> > ___
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> > Asterisk-Users@lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
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> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
>
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Re: Abwesenheitsnotiz: [Asterisk-Users] Nortel Option 11 and TE110P o f Digium

[Alvaro Parres]

??? i dont understand.



On 8/5/05, Siegel, Joerg <[EMAIL PROTECTED]> wrote:
>  
> 
> Ich bin am 9.8. wieder im Hause! 
> 
> Mit freundlichen Grüßen, 
> 
> Jörg Siegel.
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[Asterisk-Users] Nortel Option 11 and TE110P of Digium

[Alvaro Parres]

Hi list:
 
I have a client that needs to connect a Asterisk PBX with a TE110P
of Digium and one Nortel Option 11.

   Actually the Nortel Option 11 have a AMI E1 card. With it the have
problems of clock sync.

   They can change the AMI CARD to a PRI CARD, te questions are:

 1) Which model of PRI is suggest for this ?
 2) Some one have already do this ?
 3) Is there form of correct de AMI problem ?

   Well i hope that you will answered me. 

Alvaro Parres

P.D. If any one from Mexica have done this before pleas contact me
(33) 35636261
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[Asterisk-Users] Hardphones Console o Secretarial One

[Alvaro Parres]

Hi list.

I want to know if there is any console o secretarial hardphone that
works with asterisks.

I mean a phone in witch i can see the state of the extensions, the
phone lineas, etc. Can hold o transfer easly a call, etc.

Thanks

Alvaro Parres
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Re: [Asterisk-Users] HOW TO PROGRAM NEW MODULES

[Alvaro Parres]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Yes but i want to limit like user of ext 120 only have 20 min of calls
other one have 60 min of calls at moth thinks like that
brian wrote:

| app_groupcount.c (this is in cvs-head)
|
| exten => 999,1,SetGroup(moh) exten => 999,2,CheckGroup(1) exten =>
| 999,3,Answer exten => 999,4,MusicOnHold(default) exten =>
| 999,103,Busy
|
| See?
|
| You can limit that to just 1 user at a time or what ever you wish :
|
|
| bkw
|
|> -Original Message- From:
|> [EMAIL PROTECTED] [mailto:asterisk-users-
|> [EMAIL PROTECTED] On Behalf Of Alvaro Parres Sent:
|> Thursday, May 06, 2004 5:07 PM To:
|> [EMAIL PROTECTED] Subject: [Asterisk-Users] HOW TO
|> PROGRAM NEW MODULES
|>
| Hi
|
| ~  Some one know where i can find some documentation about how to
| programm some modules for asterisk.
|
| ~   Becouse i want to program a call limit per user.
|
|
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| ___ Asterisk-Users
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- --
Alvaro Ivan Parres Peredo
Director de IT
[EMAIL PROTECTED]
Tel: (33) 36301294
~ (33) 36309553
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oh48XAs2bYpFYCd6HryUsng=
=8MEK
-END PGP SIGNATURE-
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[Asterisk-Users] HOW TO PROGRAM NEW MODULES

[Alvaro Parres]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi

~  Some one know where i can find some documentation about how to
programm some
modules for asterisk.
~   Becouse i want to program a call limit per user.

- --
Alvaro Ivan Parres Peredo
Director de IT
[EMAIL PROTECTED]
Tel: (33) 36301294
~ (33) 36309553
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iD8DBQFAmrcZCDkd3lP6lKQRAlmoAJwJXst6RmmP26zFiq94Jz52gODgfACghR/7
fvV5TNyaNdaXNlk4kRL9fCg=
=BQYA
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[Asterisk-Users] A GOOD IP PHONE IAX OR SIP

[Alvaro Parres]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi some one can give me information about a good and ship ip phone IAX
or SIP
Thanks

- --
Alvaro Ivan Parres Peredo
Director de IT
[EMAIL PROTECTED]
Tel: (33) 36301294
~ (33) 36309553
-BEGIN PGP SIGNATURE-
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Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org
iD8DBQFAmByaCDkd3lP6lKQRAvWnAJ93b2/Yv6+YmAuGssLz2SuiQdu03QCfQyF0
2satIwN0367cmBzwxjqFOFE=
=OrTx
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[Asterisk-Users] FLASH TONE

[Alvaro Parres]

Hi list.

  I'm having the next problem. I Bought a new analog phone, it have 
flash button, but it send a tone not a cut on the line.  So the flash 
key is not working, a thing that was problem of the phone, but i connect 
another phone that have the same problem.

  I suppose that the flash key send a tone, becouse when i push it i 
lost the dial tone.

  Any idea how can i do, so * detect that tone as flash key ?

Alvaro Parres



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[Asterisk-Users] RxFax

[Alvaro Parres]

i have check at internet, that some one use RxFax application for
recive faxes...
Where i can get this application, becouse i have the cvs of today
and it does not have application???


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Re: [Asterisk-Users] Flash Problem

[Alvaro Parres]

Only this phone have the problem. I have other phones and they 
dosent have the problem

Brian West wrote:

Do any other phones work fine?  Or just this one?

bkw

On Tue, 16 Dec 2003, Alvaro Parres wrote:

 

Hi, i have the next problem:

I have a new Motorola cordless analog phone plug at FXS ports at my
*. But the * does not detect when i press flash?...
Any idea how can i solve this?

Is any way i can set another key to work as flash? maybe # or * ??

Thanks

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[Asterisk-Users] Flash Problem

[Alvaro Parres]

Hi, i have the next problem:

I have a new Motorola cordless analog phone plug at FXS ports at my 
*. But the * does not detect when i press flash?...

Any idea how can i solve this?

Is any way i can set another key to work as flash? maybe # or * ??

Thanks

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[Asterisk-Users] 2 T100P Problem. Broken Pipe

[Alvaro Parres]

Hi list.

  I'm having the next problem.

 I have a * with 1 TDM400P (4 ports) and
 one X100P, with a working configuration.
 Today i add one more X100P card, and i change
 the config files as next:
 zapatel.conf:
  fxsks=1-2
  fxoks=3-6
  loadzone = us
  defaultzone=us
   zapata.conf
  ...
  context=bell
  signalling=fxs_ks
  channel=1-2
  ...
  signalling=fxo_ks
  channel=3-6
   I load the modules in the next order:
wcfxo
wcfxs
   And when I start asterisk i get the next error:

ERROR[16384]: File chan_zap.c, Line 4921 (mkintf): Signalling requested 
is FXO Kewlstart but line i   is   in FXS Kewlstart signalling
ERROR[16384]: File chan_zap.c, Line 6663 (load_module): Unable to 
register channel '2'
WARNING[16384]: File loader.c, Line 301 (ast_load_resource): 
chan_zap.so: load_module failed, returning -1
WARNING[16384]: File loader.c, Line 396 (load_modules): Loading module 
chan_zap.so failed!
Ouch ... error while writing audio data: : Broken pipe
Ouch ... error while writing audio data: : Broken pipe
  




 

   



 



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Re: [Asterisk-Users] pick up ringing exten

[Alvaro Parres]

I have ZAP channels.. so i add the lines at zapata.conf
and it does not work. When i dial *8 it return me a busy tone.
my zapta.conf is..context=home
group=2
pickupgroup=2
signalling=fxo_ks
channel=2-3
callerid="FIJO" <200>
channel=3
callerid="INALAMBRICO" <100>
channel=2
Rich Adamson wrote:

Is it possible with Asterisk to pick up ringing extension from other extension?
So I do not have to run to other desk to pick up the phone.
   

Sure, just add
callgroup=2
pickupgroup=2
to each extension definition in sip.conf as an example. Dial *8 to
pick up that ringing extn.


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[Asterisk-Users] channel bank and sip phone

[Alvaro Parres]

Hi list...

  I have to questions. The first i have to mount a medium PBX so i'm 
looking
arround some channel banks at ebay.. So i have some questiones:

  ¿ Is ebay a good place for serch?

  I found one "T1 to 8 FXO and 16 FXS Channel Bank/Multiplexer from 
Carrier Access"
 (  
http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=3056375835&category=11908 
)
 I think i can connect this one to a T100P Card .. Is this 
possible, and correct?.

  And one more which one are good IP PHONE for *, where i can buy it?

Well Thanks.

Alvaro Parres



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[Asterisk-Users] Broken Pipe

[Alvaro Parres]

Hi.. i'm having the next problem with a Asterisks Box... like every 24 hours it
give the next error when i tray to connect to it.

[EMAIL PROTECTED]:~# asterisk -r
Asterisk CVS-09/29/03-17:13:53, Copyright (C) 1999-2001 Linux Support Services, Inc.
Written by Mark Spencer <[EMAIL PROTECTED]>
=
Broken pipe
[EMAIL PROTECTED]:~#


Any idea before re-installing that PC ???




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[Asterisk-Users] Zap Analog Line Hangup Problem

[Alvaro Parres]

Hi..

   I'm having the next problem... with the busy detect = yes...

   If i have it...  The * it hang up the calls when they are 
active... ( Incoming ant Outgoings calls).

   If i haven't it  * Doesn't detect when some one hang up and 
never close the channels...

  WHAT I CAN DO 





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[Asterisk-Users] HI.. .Any comments about VoicePulse

[Alvaro Parres]

Hi... some one can tell me his comments about VoicePulse Services...

for Pre-Paid long distance..

Alvaro parres





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[Asterisk-Users] HOW TO GET REGISTER WITH NUFONE??

[Alvaro Parres]

Hi all...
   How can i register wit nufone i was serching at its pages... and 
I never find how to get register...

Thanks.





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[Asterisk-Users] Busy detect and Hangup with VoiceMail Problem

[Alvaro Parres]

Hi:

 I have the next problem at VoiceMail application, when i call is 
recive, it does not detect when the user hangup the call..

 The first solution i think was to place the busydetect=yes at 
zapata.conf  but it cause that some calls get lost ( when some one is 
talking, the * hangup the call)...

 So i limit VoiceMail for 2 minutes each messages, it resolve the 
problem in a first moment, but i have the problem, that if some user who 
is chechink its mailbox from outside, and hangup the program does not 
detect it.

 In resume, my VoiceMail application does not detect when some one 
HangUp and if I use busydetect at zapata.conf it hangup active calls.

  Any suggestions??

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[Asterisk-Users] PROBLEMS WITH IAXATEL AND DIGIUM IAX

[Alvaro Parres]

Hi

 I'm having a extrange problem I cant register with Iaxtel or call to digium...

  But i cant make or recive IAX calls... ( I made some one with irc users )

 Any idea why?


At my logs i have this from iaxtel:

NOTICE[196621]: File chan_iax2.c, Line 2832 (register_verify): No registration
for peer 'xmarts' (from 192.168.0.11)
NOTICE[196621]: File chan_iax2.c, Line 4389 (socket_read): Registration of
'arabe' rejected: Registration Refused

And when i tray to call Digium i get:

-- Called [EMAIL PROTECTED]/[EMAIL PROTECTED]
NOTICE[196621]: File chan_iax2.c, Line 4024 (socket_read): Rejected connect
attempt from 192.168.0.11
WARNING[196621]: File chan_iax2.c, Line 4124 (socket_read): Call rejected by
216.207.245.8: No authority found



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[Asterisk-Users] SOME QUESTIONES (LOG, MySQL, Extensions)

[Alvaro Parres]

Hi all.

   I have some questions:

 1) Is there a way to get a full log of the calls (incoming and outgoing)
 
 2) How is the intregation of Mysql and Asterisk. At witch Aplicattions.

 3) And of the Extension 

 a) I have a Support Call Center. Almost all the time all the
extensions are busy, and some calls at hold. Is there a way that when some one
hang off, can take the next call in the hold. At this moment, i have that if all
are busy, send the call to hold. And in 1 minute tray again to call any extension. 

 b) Is ther any way.. that other person can take the ringing
call of other extension at his phone?... Example: There is a ringing call at
extension 100 which is alone and i'm at extension 150, So and i want to take it?
so i dial some number and i take it... How to do this?


   Well thanks... all





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[Asterisk-Users] PROBLEM RECIVING CALLS AT FXO

[Alvaro Parres]

Hi... 
 
  I have the next problem.. I have a FXO card with i can make calls but i cant 
recive calls.  
 
  At the consol, i get the next error: 
 
-- Zap/2-1 is ringing 
-- Zap/2-1 is ringing 
-- Zap/2-1 answered Zap/1-1 
-- Attempting native bridge of Zap/1-1 and Zap/2-1 
WARNING[262160]: File chan_zap.c, Line 2857 (zt_handle_event): Ring/Off-hook 
in strange state 6 on channel 1 
 
   My config files are: 
 
 zaptel.conf - 
fxsks=1 
fxoks=2-3 
loadzone = us 
defaultzone=us 
 
--- zapata.conf - 
[channels] 
relaxdtmf=yes 
busydetect=yes 
callprogress=yes 
callwaiting=yes 
callwaitingcallerid=yes 
threewaycalling=yes 
transfer=yes 
cancallforward=yes 
usecallerid=yes 
hidecallerid=no 
echocancel=yes 
echocancelwhenbridged=yes 
rxgain=0.0 
txgain=0.0 
group=1 
pickupgroup=1-2 
;immediate=no 
context=bell 
signalling=fxs_ks 
mailbox=yes 
;callerid=asrecive 
channel=1 
context=home 
group=2 
signalling=fxo_ks 
channel=2-3 
callerid="FIJO" <200> 
channel=3 
callerid="INALAMBRICO" <100> 
channel=2 
 
 extensions.conf  
 
[dialout] 
ignorepat => 9 
exten => _9.,1,Dial(${PSTN}/${EXTEN:1},120,T) 
exten => 9,1,Dial(Zap/g1/) 
exten => 9,2,Congestion 
 
Thanks. 
 
 
 
  
 
 
-- 
Alvaro I. Parres Peredo 
Director de IT 
Xmarts, Soluciones Inteligentes 
Bernardo de Balbuena #35 
Tel: 36301294 
http://www.xmarts.com 

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[Asterisk-Users] Some Question of extension.conf

[Alvaro Parres]

Hi...

  Some questions.

 ¿How do you make that some user who is in a menu, can dial any extension
  that is define in other context ? Example..

  [office]
   100,1..
   200,1..
   300,1..
  [menu]
s,1 <- When the user is here.. can dial 200 and it takes
1,1the 200 extension of office context.
2,1
3,1
 
 ¿If i have 2 Asterisk with IAX, the server A have some  
extension(100,110,120,130) and the server B have others (200,210,220,230). How
can i do that when a user of Server A dial 210 call the extension 210 that is in
the B server ?

Well thanks.

 
 
  



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[Asterisk-Users] Hardware for a Big PBX

[Alvaro Parres]

Hi list...

  I have already installed a small PBX ( 1 FXO (E100P) and 4 FXS (TDM400P) ) 
but now i want to know how to build a bigger one... maybe 8 FXO and 24 FXS 
something like that o bigger.  But i dont know witch hardware i need. 

  And also any of you have installed Asterisk only for VoIP and connect this 
with a Normal PBX (Nortell), so that the actual telephone system can dial to 
the Asterisk and this system can recive call form Asterisks.


Alvaro Parres




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[Asterisk-Users] GASTMAN AUTH QUESTION

[Alvaro Parres]

Hi,

  Any of you know where to define the user and password for gastman.???

PLEAS HELP ME!

Alvaro Parres

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[Asterisk-Users] a web admin

[Alvaro Parres]

Hi:

   Any of you know a good web admin for asterisk???

   Thanks 


Alvaro Parres




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