>>> On Apr 27, 2016, at 12:12 PM, Mark Engelhardt
>>> ma...@intuitiveengineering.com wrote:
>>> 1) Old School ISDN BRI lines which I would connect to Asterisk with a
>>> OpenVOX B200P
I've never dealt with a BRI before, primarily PRI, but I'd go BRI instead of IP
if they're doing any
Hello,
I am installing Asterisk in a small office with just 4 lines and 8 Extensions.
I have two choices from my local telco (Fairpoint):
1) Old School ISDN BRI lines which I would connect to Asterisk with a OpenVOX
B200P
2) Telco supplied SIP trunks over a service called EDIA which is 1MB
Thanks Richard and Andres.
I had come to the same conclusion, however the provider was fairly snarky
in saying is was my equipment.
We were able to replace the Cisco 2800 with a Cisco 2900 series and the
problem appears to have been resolved.
Thanks again, I always appreciate another set of
We fairly recently switched service providers for our 4 PRI circuits. Since
that time, we started to notice some failed inbound calls. These calls
terminate with an ISDN cause code 47 'resource unavailble'. Most of the
time I see this error on the first or second channel on the second span in
a
On Mon, Jan 20, 2014 at 5:16 PM, Dale Noll dn...@wi.rr.com wrote:
We fairly recently switched service providers for our 4 PRI circuits.
Since that time, we started to notice some failed inbound calls. These
calls terminate with an ISDN cause code 47 'resource unavailble'. Most of
the time I
[Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04]
PRI Span: 4 TEI=0 Call Ref: len= 2 (reference 6918/0x1B06) (Sent
from originator)
[Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04]
PRI Span: 4 Message Type: RELEASE COMPLETE (90)
[Jan 14
: [asterisk-users] ISDN outgoing caller id
Date: Tue, 27 Aug 2013 21:28:36 +0200
Tuesday, August 27, 2013, 8:41:18 PM, Patrick wrote:
On 08/27/2013 08:04 PM, Gergo Csibra wrote:
Hi,
is anybody out there who can set the outgoing caller id on ISDN (CAPI
or misdn) channels? I've tryed everything
Hi,
is anybody out there who can set the outgoing caller id on ISDN (CAPI
or misdn) channels? I've tryed everything what I found in forums, os
voip-info.com but no luck. I use a fritz card with CAPI in my first
installation (1 BRI), and a hfc 4 port bri card with misdn on other.
The first
On 08/27/2013 08:04 PM, Gergo Csibra wrote:
Hi,
is anybody out there who can set the outgoing caller id on ISDN (CAPI
or misdn) channels? I've tryed everything what I found in forums, os
voip-info.com but no luck. I use a fritz card with CAPI in my first
installation (1 BRI), and a hfc 4 port
Tuesday, August 27, 2013, 8:41:18 PM, Patrick wrote:
On 08/27/2013 08:04 PM, Gergo Csibra wrote:
Hi,
is anybody out there who can set the outgoing caller id on ISDN (CAPI
or misdn) channels? I've tryed everything what I found in forums, os
voip-info.com but no luck. I use a fritz card with
On Tue, 27 Aug 2013 21:28:36 +0200
Gergo Csibra csi...@gmail.com wrote:
Tuesday, August 27, 2013, 8:41:18 PM, Patrick wrote:
On 08/27/2013 08:04 PM, Gergo Csibra wrote:
Hi,
is anybody out there who can set the outgoing caller id on ISDN
(CAPI or misdn) channels? I've tryed everything
Hello everyone.
I am looking for a E1 PRI card which supports network side signaling not
CPE. The main idea is to connect an plain old E1 compliant PBX which
doesn't have an VoIP module to the newly created VoIP infrastructure.
Could we use a Digium TE122P or something other to resolve this
Use pri_net as signalling mode inside chan_dadhi.conf in /etc/asterisk/
folder.
You can set this up using any pri card thats supported on Asterisk.
Mitul
On Mar 31, 2013 12:25 PM, Dimitar Dimitrov ddimit...@consult.bg wrote:
Hello everyone.
I am looking for a E1 PRI card which supports
In article caaogpgr4y2vyndtu3nsnlct_6t1vdojsgouh634ec+zfogp...@mail.gmail.com,
Mitul Limbani mi...@enterux.in wrote:
On Mar 31, 2013 12:25 PM, Dimitar Dimitrov ddimit...@consult.bg wrote:
Hello everyone.
I am looking for a E1 PRI card which supports network side signaling not
CPE. The
Thank you guys for the fast response.
I will try that.
Thanks.
Dimitar
On 03/31/2013 11:15 AM, Tony Mountifield wrote:
In article caaogpgr4y2vyndtu3nsnlct_6t1vdojsgouh634ec+zfogp...@mail.gmail.com,
Mitul Limbani mi...@enterux.in wrote:
On Mar 31, 2013 12:25 PM, Dimitar Dimitrov
Hi,
I have been trying to find solution to this problem by googleing and
experimenting but without success so you are my last chance.
I have asterisk 1.8.11 installation on Debian squeeze with HFC-S PCI
ISDN interface cards (dahdi channel, zaphfc kernel driver) , Huawei USB
3G stick,
Hi,
I've got the following setup:
PSTN/ISDN E1- Asterisk E1- Alcatel 4400 PBX
TDM phones
When a TDM phone is dialing out to a national number, it seems that
the PBX is using enbloc dialing.
When a TDM phone is dialing out to an international number (variable
length
...@yahoo.fr
Subject: [asterisk-users] ISDN, overlap and open dialing plans
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:
CAPeT9jjSovFSbyJyeHbSB1si0KNn_OcyUnmHm+=o-_tcrwu...@mail.gmail.com
Content-Type: text/plain; charset=ISO
Dear All;
I am afraid from IRQ misses: 1
The ISDN E1 was working fine on the machine, the electrical disconnected and
then the Red Allarm. I checked the dahdi and I found that I have to reinstall
dahdi again and I did. But still not becoming UP.
The output of the cat /proc/dahdi/1 is
On Tue, Jan 03, 2012 at 02:41:33PM -0800, bilal ghayyad wrote:
Dear All;
I am afraid from IRQ misses: 1
The ISDN E1 was working fine on the machine, the electrical
disconnected and then the Red Allarm. I checked the dahdi and I
found that I have to reinstall dahdi again and I did. But
Il 07/12/2011 23.45, Vieri ha scritto:
As far as I know, E1 usually use 16 as D channel. Anyway, I tried as you
suggested and set 1 as the D channel and 2-31 as B channels.
In the asterisk log I got these messages:
chan_dahdi.c: Channel 16 is reserved for D-channel.
chan_dahdi.c: Unable to
Il 07/12/2011 23.45, Vieri ha scritto:
As far as I know, E1 usually use 16 as D channel. Anyway, I tried as you
suggested and set 1 as the D channel and 2-31 as B channels.
In the asterisk log I got these messages:
chan_dahdi.c: Channel 16 is reserved for D-channel.
chan_dahdi.c: Unable to
On 12/07/2011 05:06 PM, Vieri wrote:
--- On Wed, 12/7/11, Kevin P. Flemingkpflem...@digium.com wrote:
Standard Ethernet cables do not always work for T-1/E-1
spans. They do work a rather large percentage of the time,
but not always. Distance between the NIU and the T-1/E-1
card can be a
In article 4ee0b0e2.3050...@digium.com,
Kevin P. Fleming kpflem...@digium.com wrote:
As I said before... an Ethernet cable will work nearly all the time, and
at a 5m length it's probably fine.
Kevin, under what circumstances would an Ethernet cable potentially not
work with T1/E1? And in
I am not Kevin, but I'll tell you that I will not EVER use an Ethernet
cable for T1 again. Kevin and I have discussed this at length, and the
should work plays out poorly in the real world, or at least mine. I've
had it be fine, and had major problems. I can't even find a pattern to it,
like
2011/12/8, Carlos Alvarez car...@televolve.com:
I am not Kevin, but I'll tell you that I will not EVER use an Ethernet
cable for T1 again. Kevin and I have discussed this at length, and the
should work plays out poorly in the real world, or at least mine. I've
had it be fine, and had major
A T1 cable according to this spec:
http://www.cisco.com/en/US/products/hw/routers/ps214/products_tech_note09186a00801f5d89.shtml
Crossing the 1/2 to 4/5 if needed.
On Thu, Dec 8, 2011 at 9:37 AM, Olivier oza_4...@yahoo.fr wrote:
2011/12/8, Carlos Alvarez car...@televolve.com:
I am not
Tony wrote:
Kevin P. Fleming kpflem...@digium.com wrote:
As I said before... an Ethernet cable will work nearly all the time, and
at a 5m length it's probably fine.
Kevin, under what circumstances would an Ethernet cable potentially not
work with T1/E1? And in those circumstances, what
2011/12/8, Carlos Alvarez car...@televolve.com:
A T1 cable according to this spec:
http://www.cisco.com/en/US/products/hw/routers/ps214/products_tech_note09186a00801f5d89.shtml
Crossing the 1/2 to 4/5 if needed.
In fact I was rather referring to the previous example in which a
cable did run
On Thu, Dec 8, 2011 at 9:37 AM, Olivier oza_4...@yahoo.fr wrote:
I usually make my own,
which type of cable are you then using ?
I just realized that I may have not answered the right question. Did you
mean what raw cable did I use to make T1 cables? Cat-3 or above is fine.
I use
On Thu, Dec 8, 2011 at 10:14 AM, Olivier oza_4...@yahoo.fr wrote:
In fact I was rather referring to the previous example in which a
cable did run OK for years and suddenly stopped to.
My THEORY is that the driver chips on either end were wearing out and no
longer able to send or receive as
Il 08/12/2011 18.17, Carlos Alvarez ha scritto:
If you use an ethernet cable, you are using a pair of wires that is
not twisted together, removing the electrical advantage of twisted-pair
cable.
This is wrong, in both T568A or T568B ethernet pins 1/2 and 4/5 runs on
a properly twisted pair.
On Thu, Dec 8, 2011 at 10:48 AM, giovanni.v i...@keybits.org wrote:
This is wrong, in both T568A or T568B ethernet pins 1/2 and 4/5 runs on a
properly twisted pair. Also 120 Ohm impedance is matching the ISDN pri
specification.
If a straight pri cable is needed then a straight ethernet cable
Try this instead:
http://www.ahk.com/t1_cable.html
That cisco link does not specify the cable itself, but only the pin
outs. True T1 cable has a foil shield around each pair, also called
ABAM cable in the telco world.
Ethernet cable is twisted pair without any shielding between pairs.
And
Interesting:
If you cannot obtain T1 specific cable, then use two runs of CAT 5. Use one
CAT5 cable for the Transmit (Tx) signal and one CAT5 cable for the Receive (Rx)
signal. It is necessary for the Tx and Rx signals to be in separate sheaths to
prevent cross talk interference
So pins 1
Hi,
A telco has recently installed a new line in our building and I need to connect
it to my Asterisk server with a Digium PRI card.
It's not the first time I set up and configure a PRI link but I'm failing to
make this one work.
The only information I got from the telco is:
Line Coding
On Wed, 7 Dec 2011, Vieri wrote:
A telco has recently installed a new line in our building and I need to
connect it to my Asterisk server with a Digium PRI card.
It's not the first time I set up and configure a PRI link but I'm
failing to make this one work.
chan_dahdi.c: No D-channels
On 12/07/2011 04:15 PM, Steve Edwards wrote:
On Wed, 7 Dec 2011, Vieri wrote:
A telco has recently installed a new line in our building and I need
to connect it to my Asterisk server with a Digium PRI card.
It's not the first time I set up and configure a PRI link but I'm
failing to make this
--- On Wed, 12/7/11, Steve Edwards asterisk@sedwards.com wrote:
A telco has recently installed a new line in our
building and I need to connect it to my Asterisk server with
a Digium PRI card.
It's not the first time I set up and configure a PRI
link but I'm failing to make this
--- On Wed, 12/7/11, Kevin P. Fleming kpflem...@digium.com wrote:
Vieri: You aren't even far enough along to worry about
D-channel assignments or anything like that. Your span is in
RED alarm; that means it can't see the far end at all. Until
you get that cured (layer 1 - physical layer)
On 12/07/2011 04:51 PM, Vieri wrote:
--- On Wed, 12/7/11, Kevin P. Flemingkpflem...@digium.com wrote:
Vieri: You aren't even far enough along to worry about
D-channel assignments or anything like that. Your span is in
RED alarm; that means it can't see the far end at all. Until
you get that
and maybe more but right now I don't recall any loopback device although I
won't be sure until I go to the site.
Can a loopback device be bought seperately?
Sure, we use the below device all the time:
http://www.amazon.com/Cables-Unlimited-TST-LOOP-003-Smartronix-Superlooper/dp/B000V6Y7IY
Hi,am using an atcom isdn pbx its using a colegne HFC chip..It comes in TE
mode..I want to switch it to NT mode.Its not possible to select NT mode in the
GUI so I went to config filesmisdn-init.config and edited it there.But
everytime I do misdn show port 1 its alway in TE mode.Can anyone help
On Thu, Nov 18, 2010 at 10:54:53PM +0100, Thorolf Godawa wrote:
since some time I am looking for a current and reliable solution to send
and receive faxes (probably fax-2-mail and mail-2-fax) in conjunction
with Asterisk.
[snip]
What are you using? mISDN, CAPI4linux, HylaFAX, IAXmodem,
Hi everybody,
since some time I am looking for a current and reliable solution to send
and receive faxes (probably fax-2-mail and mail-2-fax) in conjunction
with Asterisk.
For testing I am using a HFC-ISDN passive PCI-card, in production a
Digium Dual T1/E1 PCI-card will be used.
I run CentOS
-users] ISDN-FAX with Asterisk
Hi everybody,
since some time I am looking for a current and reliable solution to
send
and receive faxes (probably fax-2-mail and mail-2-fax) in conjunction
with Asterisk.
For testing I am using a HFC-ISDN passive PCI-card, in production a
Digium Dual T1/E1
Paulo Santos wrote:
Hello,
Following my first mail about this issue [1], I think I know now what
the problem is.
When I have both lines being used and a third call comes in, the person
calling doesn't get a busy tone, he gets something like line unavailable.
I've been debugging mISDN
-users] ISDN SS7
On Sun, Oct 24, 2010 at 11:33:28AM -0500, Cary Fitch wrote:
SS7 is an inter-telco system using a separate network for all signaling.
You must have an SS7 network connection before anything will work.
Then the T1 Spans run 24 64k audio paths. The SS7 net exchanges
:21 AM
Subject: Re: [asterisk-users] ISDN SS7
On Sun, Oct 24, 2010 at 11:33:28AM -0500, Cary Fitch wrote:
SS7 is an inter-telco system using a separate network for all signaling.
You must have an SS7 network connection before anything will work.
Then the T1 Spans run 24 64k audio
Hi all,
I'm being requested to deploy an IVR service using SS7.
I've deployed Asterisk before using ISDN connection, but never with SS7.
Can anyone explain me the different between using ISDN and SS7 ? What need I do
now to change to use SS7 ?.
Many thanks,
Giang
--
-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sun, October 24, 2010 9:33:28 AM
Subject: Re: [asterisk-users] ISDN SS7
SS7 is an inter-telco system using a separate network for all signaling.
You must have an SS7 network connection before anything will work.
Then the T1 Spans run
List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ISDN SS7
Hi cary,
Can you recommend me what add-on vendors I should use ?
Can a open source solution such as chan_ss7 or libss7 support many
conncurrent calls (for example 240 calls) ?
Thanks
_
From: Cary Fitch
Still I am also facing the call disconnection when there is a third call. I
am using Netmod BRI router and the output of the BRI router lines are
connected to FXO ports in Asterisk.
Where in Asterisk I am facing the call disconnection when there is a third
call..
On Tue, Sep 28, 2010 at 4:22 PM,
Hello,
Gopalakrishnan A.N wrote:
I am also facing the call disconnection if there is a third call. I
tried disable call waiting in the BRI router, but now it has been
reduced, it means call disconnection is not permanent but seems to be
occasion, let say per day two times there is a call
I am also facing the call disconnection if there is a third call. I tried
disable call waiting in the BRI router, but now it has been reduced, it
means call disconnection is not permanent but seems to be occasion, let say
per day two times there is a call disconnection.
On Wed, Sep 29, 2010 at
I'm resending this email to the list, apparently the first one didn't go
through. If it did, I apologize for the re-post.
Hello,
Following my first mail about this issue [1], I think I know now what
the problem is.
When I have both lines being used and a third call comes in, the person
calling
Hello,
Following my first mail about this issue [1], I think I know now what
the problem is.
When I have both lines being used and a third call comes in, the person
calling doesn't get a busy tone, he gets something like line unavailable.
I've been debugging mISDN and I think the reason is
Hi,
I have a Netmod ISDN BRI router and from the router I have connected the
analog port in Asterisk via FXO card. Two analog lines I have connected to
asterisk machine. When both the lines are established, after 31 minutes the
call is automatically disconnected.
While checking the log it shows
Friday, June 11, 2010, 12:27:08 AM, Tzafrir wrote:
On Fri, Jun 11, 2010 at 12:19:37AM +0200, Gergo Csibra wrote:
Okay. There's some problems with mISDN v2: I'm unable to compile
zaphfc, because there's no source for it. mISDN v2 works with hfcpci
too?
Certainly there is.
It's also part of
On 06/10/10 23:19, Philipp von Klitzing wrote:
Hi!
i am using asterisk 1.6.2 with mISDN and a passive AVM Fritz!card on
CentOS 5.5. The only thing, i want to do is a call-redirection from an
isdn-call to my mobile via sip-account.
Unless you are using mISDN v2: Do yourself a favour and
Hi!
But if i try to establish ISDN-SIP-Dialout, the redirection ist not
working.
Your logs are very sketchy and difficult to understand because you
stripped them of some details and cut out lines in between.
From: 5 sip:s...@sip;tag=as1ec770c5
This line does not make much sense.
i am using asterisk 1.6.2 with mISDN and a passive AVM Fritz!card on
CentOS 5.5. The only thing, i want to do is a call-redirection from an
isdn-call to my mobile via sip-account.
My extension conf is:
general]
static=yes
writeprotect=no
[globals]
OUT_PORT=1
[ISDN]
exten =
On Fri, Jun 11, 2010 at 12:19:37AM +0200, Gergo Csibra wrote:
Thursday, June 10, 2010, 11:19:16 PM, Philipp wrote:
i am using asterisk 1.6.2 with mISDN and a passive AVM Fritz!card on
CentOS 5.5. The only thing, i want to do is a call-redirection from an
isdn-call to my mobile via
Hi!
i am using asterisk 1.6.2 with mISDN and a passive AVM Fritz!card on
CentOS 5.5. The only thing, i want to do is a call-redirection from an
isdn-call to my mobile via sip-account.
Unless you are using mISDN v2: Do yourself a favour and switch to CAPI
with chan_capi and fcpci. mISDN v1 is
Thursday, June 10, 2010, 11:19:16 PM, Philipp wrote:
i am using asterisk 1.6.2 with mISDN and a passive AVM Fritz!card on
CentOS 5.5. The only thing, i want to do is a call-redirection from an
isdn-call to my mobile via sip-account.
Unless you are using mISDN v2: Do yourself a favour and
Quoting Tilghman Lesher tles...@digium.com:
http://www.digchip.com/datasheets/parts/datasheet/222/82V2088-pdf.php
See pages 17-18 of the associated PDF. While this is not the T1 framer chip
used, the values are identical, which leads me to believe that these values
are actually industry
Quoting Tilghman Lesher tles...@digium.com:
The value selected should almost always be zero. However, if the cable
is of a significant length, another value must be selected, but which
one? There are two columns: CSU and DSX-1. When is it appropriate to
use the one or the other to determine
On Sunday 16 May 2010 17:32:09 Jaap Winius wrote:
Quoting Tilghman Lesher tles...@digium.com:
The value selected should almost always be zero. However, if the cable
is of a significant length, another value must be selected, but which
one? There are two columns: CSU and DSX-1. When is it
Hi all,
When configuring Asterisk with an ISDN card, it will at one point
become necessary to select the LBO (Line Build-Out) value. This is an
integer (0-7) that is determined by the length of the cable and is
selected from the following table. Many of us are familiar with it:
On Saturday 15 May 2010 19:28:37 Jaap Winius wrote:
When configuring Asterisk with an ISDN card, it will at one point
become necessary to select the LBO (Line Build-Out) value. This is an
integer (0-7) that is determined by the length of the cable and is
selected from the following table. Many
All,
I have not found/seen a resolution to the issue where my TDM400P seems
to cause problems, as outlined in the mISDN (HFC-S) and TDM400P -
mISDN: ISAC XDU no TX_BUSY thread.
I have also not found/seen a simple 'how to' on patching DAHDi with
ZAPHFC as outlined in the HFC-S card thread.
Do I
Hi,
I've set up an Asterisk as voip gatway:
VOIP - Asterisk - hfc-s card - NTBA - Siemens Gigaset Dect ISDN pbx.
Outgoing calls from dect handset to the world are working. Incoming calls don't
even ring the handset.
I'm using the dahdi driver with the zaphfc kernel module. The hfc-s card is
Overlap receiving timeout, plus dialplan latency, causes network to retry
SETUP
https://issues.asterisk.org/view.php?id=16789
This patch removes the requirement that some may have found that you need to
insert a Proceeding() statement very early in your dialplan, otherwise an
inbound overlap
I am wanting to use the ISDN cause code on an Asterisk 1.6 server to
determine the status of a call attempt, where the call might not actually
connect. Reason is I am checking for valid telephone numbers from a list of
numbers, and I would like to know if the call has answered and cleared which
I
On 12/14/2009 09:31 PM, Tzafrir Cohen wrote:
On Mon, Dec 14, 2009 at 05:52:40PM +0100, Christian Theune wrote:
Hi there,
I just upgraded a relatively old Asterisk installation (1.2) in our
office to a relatively new version (1.6svn from last wednesday) which
runs a Junghans QuadBRI card [1].
Hi,
(posting again as my previous log attachments were too large. Sorry if
this should end up as a double posting.)
On 12/14/2009 09:31 PM, Tzafrir Cohen wrote:
On Mon, Dec 14, 2009 at 05:52:40PM +0100, Christian Theune wrote:
Hi there,
I just upgraded a relatively old Asterisk
Hi there,
i'm using dahdi to manage a B400P openvox BRI card.
All works as expected, i would like to know if there ia a way to put the
call in REMOTE HOLD, like pressing R button on ISDN phone.
This can be done by CAPI using the proper application ,
It is implemented on DAHDI ?
Regards Andrea
Hi there,
I just upgraded a relatively old Asterisk installation (1.2) in our
office to a relatively new version (1.6svn from last wednesday) which
runs a Junghans QuadBRI card [1].
To get this flying I got dahdi-linux, dahdi-tools and libpri from SVN as
well.
After a while of juggling it
2009/12/14 Christian Theune c...@gocept.com
Hi there,
I just upgraded a relatively old Asterisk installation (1.2) in our
office to a relatively new version (1.6svn from last wednesday) which
runs a Junghans QuadBRI card [1].
To get this flying I got dahdi-linux, dahdi-tools and libpri
On 12/14/2009 06:45 PM, Olivier wrote:
2009/12/14 Christian Theune c...@gocept.com mailto:c...@gocept.com
Hi there,
I just upgraded a relatively old Asterisk installation (1.2) in our
office to a relatively new version (1.6svn from last wednesday) which
runs a Junghans
On Mon, Dec 14, 2009 at 05:52:40PM +0100, Christian Theune wrote:
Hi there,
I just upgraded a relatively old Asterisk installation (1.2) in our
office to a relatively new version (1.6svn from last wednesday) which
runs a Junghans QuadBRI card [1].
To get this flying I got dahdi-linux,
Since 2004 asterisk/libpri have been able to receive the Calling Sub Address
in the ISDN setup message, and the dialplan was able to use it if required.
It's support is limited to only NSAP, not BCD or user formatted.
At the time 25/06/04 the questioned was asked, wouldn't it be a good idea to
Hello All,
Just got one general question on ISDN error code 42. As per Cisco docs
and Wiki
(http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause) it
says Switch equipment Congestion with explanation The destination
cannot be reached because the network switching equipment is
Come on! Anyone? How about anyone doing Asterisk in Macau (China)?
Si Tai Fan wrote:
Hi
Has anyone successfully connected to Macau CTM using the E1 TE110P
card? They are using the R2 signaling for their IDAP connection.
Thanks,
Si
Hi
Has anyone successfully connected to Macau CTM using the E1 TE110P card?
They are using the R2 signaling for their IDAP connection.
Thanks,
Si
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
Martin escreveu:
Hi,
You're right. I wasn't aware of this patch getting into the code.
In the version you're running the code is already present.
The only problem I see is that some other timer kicks in here and the
T309 cannot be scheduled.
q931.c has this ...
/* For a call in Active
Martin escreveu:
What is the specification for T309 ? I'm too lazy to look it up.
The default behaviour when the alarm of layer 1 (electrical T1/E1) is
detected is to assume
all calls dropped on both sides and that's what Asterisk does.
The timer is simply deactivated since all the calls
Based on the Asterisk logs you posted the Asterisk doesn't have it
implemented per:
The implementation of timer T309 in the user side is optional
Martin
On Mon, Apr 6, 2009 at 6:22 AM, Afonso Zimmermann afo...@disc-os.org wrote:
Martin escreveu:
What is the specification for T309 ? I'm too
Martin escreveu:
Based on the Asterisk logs you posted the Asterisk doesn't have it
implemented per:
"The implementation of timer T309 in the user side is optional"
Martin
On Mon, Apr 6, 2009 at 6:22 AM, Afonso Zimmermann afo...@disc-os.org wrote:
Martin escreveu:
What is the
Hi,
You're right. I wasn't aware of this patch getting into the code.
In the version you're running the code is already present.
The only problem I see is that some other timer kicks in here and the
T309 cannot be scheduled.
q931.c has this ...
/* For a call in Active state, activate T309 only
Hi everione,
I'm make some test with pri timer T309. I'm using asterisk 1.4.23.1,
libpri 1.4.9, dahdi-linux 2.1.0.4 and dahdi-tools 2.1.0.2. But in my
tests, the timer fail with a telco link in this scenario:
Telco Phone -- Telco --- Asterisk
Sip Phone
When i
What is the specification for T309 ? I'm too lazy to look it up.
The default behaviour when the alarm of layer 1 (electrical T1/E1) is
detected is to assume
all calls dropped on both sides and that's what Asterisk does.
The timer is simply deactivated since all the calls are supposed to
drop. I
Hello!
Sorry for not being able to phrase the problem in one line. My phone
situation is this:
The calls go over analog line (or NGN/vip) I don't really get to see it. I
have got a router with a lot of jacks. One or two of them are for ISDN phones
or other ISDN capable devices. Can I use
Gondar Monn wrote:
Hi there!
Does anyone deal with Telus in BC ? We have some PRI lines that were
used for dialup, would like to convert them for pbx system, talked
with some technicians @ Telus, but the information given was not
clear, kind of: try this see if it works Does anyone
Thanks a lot, will let you know how things are going when I get them to turn
on two way dialing .
By the way any pointers on how to connect to Portmaster ?
Looks like we are going to have to share some PRIs lines with portmaster
(dialup)
FYI, I am sticking with Asterisk 1.4 for now ...
Thanks!
Hi there!
Does anyone deal with Telus in BC ? We have some PRI lines that were used
for dialup, would like to convert them for pbx system, talked with some
technicians @ Telus, but the information given was not clear, kind of: try
this see if it works Does anyone here have the settings
Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ISDN Cause codes
I have found that the messages are not played as the hangup
cause clears
down the channel and passed hangup to the other end
should I have progress() before the dial command?
Robb
Martin Smith
Subject: Re: [asterisk-users] ISDN Cause codes
Thanks for the reply
Could you be a little more specific?
Thanks
Robb
Martin Smith wrote:
Hi Robert,
I'd suggest tweaking the Dial() arguments so that you (1)
allow early
audio, (2) don't force it play ringing
Of
Robert Boardman
Sent: Thursday, November 20, 2008 5:56 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] ISDN Cause codes
Hi All
Just been looking at stats for one of my sites, and I'm
conserned about
the number of error cause codes being
Ext. 221
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Robert Boardman
Sent: Thursday, November 20, 2008 5:56 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] ISDN Cause codes
Hi All
Just been
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