Hello,
I must admit that I have never set up an asterisk system with R2 signalling.
But from the config files
point of view, you stated TE_SIG_MODE in wanpipe1.conf as ccs which should be
cas, right ?
If this does not help, you need to connect an external E1 Monitor.
Regards,
Hans
Am
Le 07.02.19 à 12:59, Rui Mota a écrit :
> Hi.
> I am using an appliance as SIP-ISDN gateway that has a (not so nice) feature
> of hanging up a call if it detects more than 10 minutes of silence from
> the originating side.
> The problem is that the calls are always originated from a digital
Hello Tzafrir,
Am 2017-03-21 um 11:23 schrieb Tzafrir Cohen:
> On Tue, Mar 21, 2017 at 09:36:21AM +0100, Olivier wrote:
>
> I'm still having some questions:
>
> 1. I can't find any /etc/init.d/dahdi file in my newly built system so
> "service dahdi status" (or systemctl status dahdi) fails with:
On 2014-08-19 23:56, Jeff LaCoursiere wrote:
Hello,
I wrote earlier today about a new PRI installation in the Caribbean, where all outbound
calls are functioning fine *except* calls to Sprint phone numbers, which get rejected
immediately as busy.
The telco has been working with their switch
On 2014-08-08 21:54, Jerry Geis wrote:
On Thu, Aug 7, 2014 at 2:53 PM, Jerry Geis ge...@pagestation.com
mailto:ge...@pagestation.com wrote:
I am using a cyberdata sip paging adapter and with the
Dial(MulticastRTP/basic/IP:port) and with
tshark I see the RTP data, my device looks like
On 2014-03-07 17:31, Paul Belanger wrote:
On Thu, Mar 6, 2014 at 3:33 PM, Markus unive...@truemetal.org wrote:
Hi Thorolf,
Am 06.03.2014 16:21, schrieb Thorolf Godawa:
Using (para-)virtualization with Xen could be an other option, on
systems with low load this works reliable, but what
On 2014-02-28 14:04, Tahir Almas wrote:
1) We do not perform any transcoding whatsoever. All recordings, and
voice mail are in G729,
and allow=g729 for all peers and in sip.conf. Is there anything else
we need to perform g729 passthrough. More importantly are we still
liable?
On 2013-09-25 09:22, Endri Stefani wrote:
Hi
Greeting to all you out there.
I am new at asterisk, I have been working with PLMN platforms telecommunication
for 5 years with NSN and Huawei.
We have recently built an asterisk PBX with Trixbox and connected it to our MSS
using Digium E1
Maybe you should open 11955 on you fw as well. This could be the rtcp port.
Regards
Hans
On 2013-09-13 11:49, Jonas Kellens wrote:
Hello,
and when I define 11500 - 11954 it should use a random port in this range.
Where is it stated that you MUST use 1-2 ???
Someone else please ?
Hello,
I 'm looking for a way to pass the '302 moved temporarily' received from the
SIP device
back to the SIP provider.
Here is the setup:
Some SIP phones are connected to an Asterisk System version 1.8.
External connection to the public network is also done via SIP to a VoIP
provider.
Phone
You did not show how the Nortel side is configured, especially LD 17 ADAN
configuration.
Regards
Hans
On 2013-05-03 11:27, Danilo Dionisi wrote:
I'm sorry, the mail is automatically send :p
However, I am for the Asterisk, there are other external consultants for Nortel
... according to you
On 2013-03-03 18:41, Olivier wrote:
hello,
In a machine I've got :
CLI pri set debug off
No such command 'pri set debug off' (type 'core show help pri set' for other
possible commands)
CLI core show help pri
pri intense debug span no description available
pri service disable
Please check out the scripts located in contrib/scripts
Regards
Hans
On 2012-05-23 11:42, Danny Dias wrote:
Hi, thanks for your answers...
Can i delete like this:
rm -rf /var/spool/asterisk/voicemail/voicemailcontextcustomer/300/INBOX/*.*
Is that ok? will this break something?
A little
Hello,
is there a way to disable a span for maintenance purpose (i.e. send yellow
alarm) ?
What would be the correct ioctl definition ? DAHDI_MAINT seems not to be the
right
candidate. Would DAHDI_SHUTDOWN send an alarm ?
Thanks
Hans
--
On 2012-03-13 18:38, Chris Bagnall wrote:
Greetings list,
I'm trying to source a very basic ISDN BRI - SIP gateway. Unfortunately,
everything I've seen seems to want to do lots of other things - registering
handsets, IVRs, voicemail, etc. I only want it to
present an ISDN BRI as a SIP account
, 2012 at 1:45 PM, Johann Steinwendtner
steinwendt...@gmx.net wrote:
On 2012-01-09 17:46, Alex Villacís Lasso wrote:
I am trying to collect information regarding a bug report for Elastix
(http://bugs.elastix.org/view.php?id=1146). In this bug, an user has
asterisk-1.8.7 and dahdi-2.4.1.2. He
On 2012-01-09 17:46, Alex Villacís Lasso wrote:
I am trying to collect information regarding a bug report for Elastix
(http://bugs.elastix.org/view.php?id=1146). In this bug, an user has
asterisk-1.8.7 and dahdi-2.4.1.2. He is trying to make an
outbound call through an ISDN trunk, by placing
Hello !
I 'm using a TE405P with a HW echocanceller module attached on it.
dahdi version is dahdi-linux-complete-2.2.0.2+2.2.0.
As far as I know, the fax tone detection is done on the FW board.
How can I verify that the echo canceller has been turned off ?
When I do a cat /proc/dahdi/1 for span
On 2010-06-22 12:36, Remco Bressers wrote:
Dear list,
I've got the following setup :
[FAX-ATA]--[PBX LAN]--[Firewall]--[PBX WAN]-[upstream SIP]
On the PBX's we run Asterisk 1.4.33 with t38pt_udptl=yes in [general].
The FAX ATA is a Teles VoIPBox with T.38 support (that works). On the
On 2010-06-22 15:16, Remco Bressers wrote:
On 06/22/2010 02:51 PM, Johann Steinwendtner wrote:
On 2010-06-22 12:36, Remco Bressers wrote:
Dear list,
I've got the following setup :
[FAX-ATA]--[PBX LAN]--[Firewall]--[PBX WAN]-[upstream SIP]
On the PBX's we run Asterisk 1.4.33
Zhang Shukun wrote:
hi, all
in my test,it shows Playback will answer the call automaticly, but i
don't want to so.
i will use answer function to answer the call. could you help me ?
core show application Playback
Regards
Hans
--
Magnus Benngård wrote:
Gentlemen,
I have 3 faxes attached to an Asterisk. Fax - SPA2102 - Asterisk.
0851711201 and 0851711290 is on our WAN, no NAT.
0197673581 is outside our WAN and needs to be NAT'ed.
Sending a fax from 0851711201 to 0851711290, no problem, switches to T38
and fax
Kevin P. Fleming wrote:
David Gibbons wrote:
snip
This doesn't work?
Dial(SIP/*31#ww061234123412)
/snip
When I was browsing the sip debugs, it seemed that the 'w' was not being
honored for one reason or another. My thought at the time was maybe it
didn't work at all over SIP.
Does the
Olivier schrieb:
2010/1/7 David Backeberg dbackeb...@gmail.com
mailto:dbackeb...@gmail.com
On Wed, Jan 6, 2010 at 6:23 PM, Olivier oza-4...@myamail.com
mailto:oza-4...@myamail.com wrote:
The second time I'm dialing an internal extension attached to the
same
randulo schrieb:
2009/10/9 Juan E. Rodríguez jerdg...@gmail.com:
Does any one know about a SIP hard phone capable of sending SMS messages
(Or a SIP MESSAGE) that could be read from Asterisk dial plan??
The Gigaset S675IP series of DECT/SIP phone has SMS capability, but
not sure it can work
Mindaugas Kezys schrieb:
Check this link: http://wiki.kolmisoft.com/index.php/RTPAUDIOQOS_Demystified
In the given example:
*ssrc=254186206;themssrc=2024901615;lp=0;rxjitter=0.020917;rxcount=150;txjitter=0.00;txcount=83;rlp=0;rtt=14818.715000*
How do I interprete the jitter value ? Is the
Hello !
I 'm having a machine running asterisk 1.6.0.10 with IAX and dahdi.
The calls are going in and out from IAX2 to dahdi (chan_dahdi + libpri)
and vice versa.
After a period of time, I got the following scenario:
NOTICE[860] chan_iax2.c: I should never be called!
WARNING[752] channel.c:
Hello !
Are there any plans at Digium to include also german voice prompts ?
Thanks
regards
Hans
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Lee Howard schrieb:
fred wrote:
That’s being said, before going through the T38 Gateway tests, I’ve
tried first the Fax2mail and Mail2fax solution with (Hylafax +
Iaxmodem + Asterisk), to make a well-tested Asterisk solution working
and I’m already facing some problems. Receiving faxes is
Hello !
In order to chase after a problem I implemented the following dialplan to have
an
answertime of exactly one minute:
exten = xxx,1,NoOp(Test wait)
exten = xxx,n,Answer
exten = xxx,n,NoOp(Current timestamp:
${STRFTIME(${EPOCH},,%C%y%m%d%H%M%S)})
exten =
Tony Mountifield wrote:
I have been asked by a potential customer whether we can connect an
Asterisk box to an Ericsson MD110 that has an E1 port with ISDN-USR.
They are unable or unwilling to upgrade their E1 port to QSIG.
Has anyone here had experience of successfully making such a
John Todd wrote:
Just a suggestion: have you tried more recent versions of Asterisk
with IAX2? I'm uncertain what version you're using, and if it's
1.2.4, that's getting to be quite old and the problems that you
reference may already be solved in more recent updates.
In addition, if
Hello !
I've upgraded our testsystem from asterisk 1.4.21 to asterisk 1.6.0.6.
We 've noticed that the log files are now in colour.
I could not find a note in the upgrade section about this.
Is this a feature or a bug ?
It might be usefull to have them not in colour.
best regards
Hans
Danny Nicholas wrote:
The log files themselves are not in color. It would be a style sheet change
on the GUI.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Johann
Steinwendtner
Sent: Friday, March 06
Wolfgang Pichler wrote:
Hi all,
we have the following setup
PSTN 3 PRI Lines --- Asterisk (1.4.22) --- Siemens HiCom
--- Bosch Integral
The Asterisk Machine does play the man in the middle - and adds some
extra functionality to the system (SIP users...) - the normal calls
Gordon Henderson wrote:
On Mon, 29 Sep 2008, Andres wrote:
In other words, I'd really appreciate feedback from voip administrators
(not from resellers) who have had experience testing their devices and are
happy with them.
I would recommend the Linksys SPA8000 (8 port ATA). It is as
Kevin P. Fleming wrote:
Benoit Plessis wrote:
Is it possible on a TE220p to deactivate the hardware echo canceler at
will ? (With a function in the dialpan for example)
example for fax SDA ,beeing able to shutdown the echo canceler could
give better results don't you think ?
All echo
Florian Hackenberger wrote:
On Tuesday 13 May 2008, Steve Totaro wrote:
You can be shot several times and not die. I would try
resetinterval=never just to be able to to say Not the problem
rather than Probably not the problem.
I'll do that, although I'm pretty sure that the setting is not
Christoph Fürstaller schrieb:
Can someone explain what the parameters pridialplan and prilocaldialplan
are? What do they do and do I need them?
I've connected an asterisk box via E1 (sangoma) to an alcatel 4200 pbx.
The pbx technican complains about the format of the nr asterisk sends.
Hello !
I 'm trying T.38 faxig with spandsp using rxfax/txfax as fax terminal.
As another endpoint I 'm using Grandstream HT 486 ATA and a fax machine.
Has anybody success with the HT486 as T.38 terminal ?
ATA as originator: I managed only onetimes a successfull T.38 fax
session. The other
Matthew Fredrickson schrieb:
On Oct 12, 2006, at 1:17 PM, Johann Steinwendtner wrote:
Hello !
I 've some questions how bridging of ISDN calls is done.
Assume an asterisk system with a TE405 card equipped.
(PRI1 - PRI4)
An incoming ISDN call on PRI1 is transfered back to
PRI3. Unless
Matthew Fredrickson schrieb:
On Oct 12, 2006, at 1:17 PM, Johann Steinwendtner wrote:
Hello !
I 've some questions how bridging of ISDN calls is done.
Assume an asterisk system with a TE405 card equipped.
(PRI1 - PRI4)
An incoming ISDN call on PRI1 is transfered back to
PRI3. Unless
Hello !
I 've some questions how bridging of ISDN calls is done.
Assume an asterisk system with a TE405 card equipped.
(PRI1 - PRI4)
An incoming ISDN call on PRI1 is transfered back to
PRI3. Unless there is DTMF detection or other things
involved, the bridging is done without Asterisk. Does
this
May be you can build an application which controls the background
terminal of the Meridian. (This would be a serial connection to the M1)
This application sends background commands like: se mw 3000.
This could be a try.
Best regards
Hans
Andrew Kohlsmith schrieb:
Please keep responses to the
Araklidas schrieb:
Yeah is true.but we have to sincronize this console command with
Asterisk SIP MWI
Regards.
Cris.
From: Johann Steinwendtner [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: Asterisk Users Mailing
Sebastian,
This is possible and most likley the reason. To make sure, check the
location code of the cause IE in your ISDN disconnect message.
You have two options:
1) call your provider and describe your problem.
2) Change your provider
Best regards
Hans
Sebastian Reitenbach schrieb:
Hi,
The BT guy should check LD 73 block LPTI and prompt AFF.
If it is crc then you need crc4 as well.
Best regards
Hans
Steve Totaro schrieb:
Andy Kirby wrote:
I am new to the group but have searched the doc's FAQ's etc before
posting here.
We are attempting tie our asterisk server/service
Yes, it is possible. I'm using PtP and TE mode at home with chan_misdn.
Hans
Ralf Mueller schrieb:
Hello,
can someone on the list confirm, that it is possible to connect a FritzCard to an
Anlagenschluss, when I use the mISDN driver?
I have read a number of posting and articles, that this is
Did you try rtpholdtimeout in sip.conf ?
Hans
Marco Mouta schrieb:
How do I report a Bug to Digium? or asterisk project?
On 4/19/06, *Doug Lytle* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Marco Mouta wrote:
I've tested maxexpirey=120 and even with this, asterisk didn't
Hi !
What is now the difference between a:
reload - (cli command reload).
restart - (I assume the application asterisk is restarted. o.k starting
from new)
sip reload - (cli command sip reload). Is sip reload part of the
reload command ?
Please confirm:
Which is the correct command when
I can only guess, but I think I can remember that the creflen needs
to be 2 octets for qsig. Check what the Alcatel switch sends in the
setup message to *.
Anyway, why do use QSIG ? Does name display work on the * implementation ?
Best regards
Hans
P.S.: Schoene Gruesse an Kurt Krenn
Enable pass thru fax mode on the HT486, or enable ulaw in your SIP config.
Hans
Garth van Sittert schrieb:
Hi All
Is there any special configuration needed to send and receive faxes on
an ATA device?
I am using G711.a with a Grandstream Handytone 486. I can send faxes
from a fax machine on
ulaw was neccessary when pass through was disabled. What does a sip
debug tell you ?
Hans
Garth van Sittert schrieb:
I am using alaw and I have already enabled the pass through. Does alaw
and ulaw work?
I can fax out, but not receive faxes.
Garth
Johann Steinwendtner wrote:
Enable pass
Hello !
Asterisk 1.0.9 running on Linux 2.6.12.
I'm not able to call iax2 channels. There can be no translation path
found.
When I try to call from a ZAP PRI channel the following error occurs:
channel.c:1891 ast_request: No translator path exists for channel type
IAX2 (native 63488) to 72
Make sure that you compile misdnuser with gcc3.x, gcc4 did
not work for me.
Hans
Yoann Le Bihan schrieb:
Jose,
I met so many problems these last 8 days that I don't remember exactly
which config was mine at that time, so I can't testify the answer...
(just for fun : my linux box is having 3
John,
why don't you migrate slowly to asterisk ? If you want to keep
most of your analog phone hardware, leave it on your Meridian 1.
The M1 is doing a good job on features on analog phone sets.
Also, your users are familiar with the call handling of the
M1. Install VoIP phones on Asterisk and
Another problem could be that there is a B-channel mismatch.
e.g. Asterisk uses channel 26 and Nortel uses channel 25. This
can be modified on at least QSIG trunks. But on EuroISDN there
should not be a problem.
Hans
[EMAIL PROTECTED] schrieb:
On Wed, 2 Nov 2005, Alvaro Parres wrote:
Hi
Goran,
which company ist this ? Do they use the www.ss7box.com
approach ?
Thanks and best regards
Hans
Goran Skular schrieb:
anyone running SS7 with Asterisk ? Please help me out.
I need to know the hardware used for SS7 with Digium E1 cards...
I can point you to one company in
I had the same problem. It seems that a fix into bristuff for .at
does not work very well.
I 've patched chan_zap.c
Best regards
Hans
old:
} else {
if (pri-nodetype == BRI_CPE) {
/* fix for .at p2p bri lines */
Tirpák Miklós schrieb:
Yes. 34 is required by the Nortel to send the call to an alternative
destination.
Cause 38 or 42 triggers the rerouting also for both options.
Hans
___
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Hi !
Asterisk sends a RELASE COMPLETE with cause code 34. It seems that
Nortel expects a RELEASE message in this state. The conversion
is done in the protocol engine of the MSDL.
Why would you want the cause code 34 to be sent ? Do you need a
special rerouting on the Nortel side ?
Would it be a
Christian Wengel schrieb:
Hi!
I tried install-misdn.tgz from http://www.beronet.com/download/ , some
minutes ago. Also I switched to an older kernel (2.6.8), but I get the
same error.
I think that I made the correct changes in the Makefiles, but I will
attach them to this e-mail, maybe you
This error would occur if you compile chan_misdn against Asterisk
stable and not specifying it in the makefile of chan_misdn.
Check the makefile of chan_misdn.
Hans
Christian Wengel schrieb:
Hi all!
I'm getting an error when I try to start asterisk with chan_misdn.
I patched my kernel
This chan_misdn version is old, use a newer one. It seems that
TypeOfNumber interpretation has not been integrated in this verison.
Best regards
Hans
Christian Peter schrieb:
Hi all,
I have a cologne chip card which is connected directly to the ntba.
Outgoing and incoming calls work fine,
Isn't it possible to turn on MWI via background terminal ? In
that case an application needs to do this via serial interface.
best regards
Hans
Users will have to get into the habit of calling the VM to check if
there's messages.
___
There is no called party ie but sending complete ie included in the
setup message. Hence, it tries to terminate.
Best regards
Hans
Paul Belanger schrieb:
Where are your calls coming from? Are you connected to the Telco or PBX?
PB
Panitaxx wrote:
Hi,
thanks for your response. here is the
Hello !
I 'd like to connect Cisco IP phones to *. (7940 7960)
Shall I use SIP or SCCP. Which approach provides better support
of features of the Cisco IP phones ?
Thanks !
Johann
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Hello !
Following scenario:
Party A: SIP Analog Terminal Adapter Grandstream HT486 (analog phone)
Party B: any other external PSTN set
Asterisk 1.0.9
Party A calls external party. Call is established. Party A presses the
flash key and goes on hook.
The external Party still gets Music on Hold.
Hello !
I got a dual E1 card from Dialogic (D300/SC-2E1 old card with ISA)
at my desk.
Is there a channel driver available for this kind of card ?
Best regards
Johann
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Hello !
I would like to get working a Fritz PCI card using chan_misdn
operating in ptp mode.
Afer compiling mISDN into the kernel and building chan_misdn
Asterisk stops loading with :
[chan_misdn.so] = (Channel driver for mISDN Support (Bri/Pri))
== Parsing '/etc/asterisk/misdn.conf': Found
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