Hi James,
we did sucessfully run two TE410P with 8xE1 in a decent Intel P4-3200
machine with quite heavy line usage. No codec conversion course.
I don't believe that there is a hard limit of E1s coded into Asterisk.
But the maximum lines you can squeeze out of your specific hardware
depends on
2010/3/25 Steve Edwards asterisk@sedwards.com:
On Thu, 25 Mar 2010, Tzafrir Cohen wrote:
[snipping a lot of interesting technical and historical details]
As you can see, there's actually a limit at the DAHDI level.
DAHDI_MAX_SPANS, which is 128. Likewise there's DAHDI_MAX_CHANS which is
2010/3/25 Zeeshan Zakaria zisha...@gmail.com:
Tzafrir, so you have actually worked with more than 192 concurrent zap
channels, which means more than 8 spans, on a single server, and can verify
that it actually works without freezing asterisk.
As I have written before - I did use 8 E1 in one
2010/3/23 Alejandro Cabrera Obed aco1...@gmail.com:
Dear all, I have an Asterisk SIP server in a LAN environment and I want your
opinion in order to decide the use of an audio codec:
What audio codec is better, G.711a or G.711u ??? Which suites to my LAN voip
calls ???
That depends most on
2010/3/19 tjoen tj...@dds.nl:
register = tjoen:mypas...@sip_proxy/1234
[sip_proxy]
type=peer
host=ekiga.net
I guess you need to register to the actual hostname, not the peers name.
register = tjoen:mypas...@ekiga.net/1234
Chris
--
2010/3/11 Eric Wheeler aster...@ew.ewheeler.org:
4. Does anyone have a couple TE2xx or TE4xx cards that can test such a
configuration? I would like to research their capability before
purchasing a couple $1200 cards.
Hi Eric,
I have four spare TE411P but never used bonded T1 or T1 for data
Yes, this machine will be enough for that task. Performance wise. The
other good thing is that it is not very likely that someone will steal
your PBX. As far as I remember it is a 7 rack unit box which weights
approx. one metric ton. ;-)
But remember - if anything dies in the box and you have to
2010/3/5 Danny Nicholas da...@debsinc.com:
Not possible. H exten is called by a hangup.
Well - sometimes not both parties hang up at the same time. ;-) If you
want to play something to the originating party after die Dial()ed
party hangs up use the option g in the Dial command to get more
2010/2/25 Zhang Shukun bit...@gmail.com:
next ,i want to dial from asterisk to PSTN now. i have see the sample
in the extensions.conf relevent to PSTN as follow:
; If you are freely delivering calls to the PSTN, list them here
;
;exten = _1256428,1,Dial(DAHDI/G2/${EXTEN:7}) ; Expose all
Not wit four - but two of them in a single core 3GHz machine worked
flawlessly doing only switching and IVR without codec conversion.
Many will suggest that you split your lines on two machines to to
prevent a total loss when a machine fails. This will add some work on
setup but maybe save you
Hi!
Having two TE410P with heavy load in a Pentium4 3,2GHz system running
Asterisk 1.2 was no problem. It did only IVR and bridging with no
transcoding though.
Chris
2009/12/14 das sandesh sandesh...@gmail.com:
Hi,
I was able to implement T122p one port PRI and was able to call out, but I
am
Hi!
Are you sure you are getting Astrisk out of the media path? I guess
reinvite must be allowed. Then it should work without transcoding
licenses.
Maybe you should take a look at the SIP DEBUG info to see what codec
Asterisk is trying to negotiate with the trunk. You could disallow
alaw and
2009/12/8 Ricardo Melendez rmelen...@utep.com.mx:
First I see at sangoma page that A101DE is PCI-Express (I think x1 for the
size of the connector)
Yes, it is PCIe x1. There is an A101D wich is PCI(-X).
for PCI Express
one x4 lane width
one x8 lane width
I can connect the card to any
2009/12/8 Joseph syscon...@gmail.com:
After pressing *1 console is not showing anything indicating that the call
is being recorded:
-- Executing [...@office-closed:1] Playback(SIP/479-1270-680060b0,
transfer) in new stack
-- SIP/479-1270-680060b0 Playing 'transfer' (language 'en')
mattias schrieb:
But are not pbx card and modem the same?
There are single FXO cards (to connect to a analogue line) that are
basically PCI modem with a special driver. But the chances that your
modem is compatible to this one specific type is very little.
Chris
2009/11/2 Doug Lytle supp...@drdos.info
Dan Journo wrote:
I need to get it up and running before we can put in the order to
transfer the fixed line number over to SIP.
Faxing over SIP is never a good idea.
And why would that be? I think that faxing over SIP using T.38 is a
fantastic
2009/11/2 Doug Lytle supp...@drdos.info
Christian Victor wrote:
2009/11/2 Doug Lytle supp...@drdos.info mailto:supp...@drdos.info
Faxing over SIP is never a good idea.
And why would that be? I think that faxing over SIP using T.38 is a
fantastic idea.
As far as I know, T.38
Olivier schrieb:
2009/9/21 Vijay Gandhi vi...@gandhiinfotech.com
There are FTC’s available,
What is it (a FTC) ? a cable ?
Any pointer to that (Google is helpless)? ?
My guess would be fixed to cell or FX to cell adapter.
Chris
___
--
2009/9/11 ABBAS SHAKEEL shakeel.abbas@gmail.com
Thanks you very much Kevin.I will try it by connecting one end of
Ethernet cable to one slot and other to second slot . Configuring one
as pri_net and the other as pri_cpe.
I will provide you feed on monday either i succed or not
2009/8/6 Alex Balashov abalas...@evaristesys.com
Sure it is. Just get a media gateway that does T.38 - and does it
relatively well.
Wich the Pattons do quite well afaik.
Chris
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tom schrieb:
hi
just donwloaded the 1.6.1 branch and made configure install. so far
so good. after staerting asterisk with:
asterisk -cr
Could not load features.conf
== Registered application 'ParkedCall'
== Registered application 'Park'
== Manager registered action
Philipp Kempgen schrieb:
Elliot Murdock schrieb:
I am wondering how the Asterisk community has been working on
solutions to deal with the asymmetric quality of ADSL. Voip is
becoming popular and a bottleneck does exists on the ADSL upload side.
One participant's upload is the
Jeff LaCoursiere schrieb:
I have a question in to them about how that floating licensing works,
though. Does that mean that with every call a license check must be made?
I don't see how it would work otherwise, and that means my whole business
- every call - is dependant on their license
Danny Nicholas schrieb:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Miguel Molina
Sent: Thursday, June 04, 2009 11:01 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Right! Whatever somebody likes more! I just say that the Snoms look
better at the side of my Mac. Wich is of course by far the superiour
system. ;-)
Chris
John Novack schrieb:
Hasn't this religious argument/discussion gone on long enough??
zoach...@securax.org wrote:
I personally find
Manoj Panicker - FOES schrieb:
However I can always call any one pre-configured PSTN number using the
call forwarding feature, however I should be able to use my sogtphone
and dial a PSTN number using the integration which is not happening
today.
As far as I know the FritzBox only supports
2009/5/24 Manoj Panicker - FOES manoj.panic...@emirates.com
Kare,
Thanks much appreciated. It connected as soon as I created a SIP
account. However I must try and figure out as how to get this box use IAX2.
Are you sure the FritzBox actually supports IAX2? As far as I remember it
does
Duuh guys - it's so easy. Ever thought of simply compressing the compressed
data AGAIN???
Do that the necessary amount of times and - tadaa - it's done.
Chris
2009/4/1 Brent Davidson br...@texascountrytitle.com
Cary Fitch wrote:
It uses proprietary EDC. (Extreme Data Compression) The 140
2009/3/30 Peer Oliver Schmidt po...@theinternet.de
The Horst-Box Professional has a lot of problems in the ADSL area
(like stopping transfers after a dozen or so megabytes for example),
and I have had lots of needs to hard-reboot the box, after enabling
VoIP functionality.
Well - I never
Here in germany D-Link sells a device called the Horst-Box
Professional wich is a ADSL modem/router with WiFi and an integrated
embedded asterisk platform with 1xBRI in, 1xBRI out and 3xFXS if my mind
serves me right. Size is about 180x250x50mm. Its been around for some
years so maybe it is
Andreas-Johann Ulvestad schrieb:
When inserting the cable going into TE122 into an ISDN phone, the phone
works perfectly.
That should not happen with an E1 line as your phone normally has a BRI
(S0) connector with only two b-channels.
Seems that your line is configured ar BRI and not PRI.
Hi!
A customer of mine wants to connect an asterisk system with 240 to 480 lines
to a PSTN switch. To save the costs for E1 cards and the corresponding E1
mainlines he wants to connect the system to the switch by a SIP trunk.
Phones will be connected to the server through the same SIP trunk as
2009/3/24 Cary Fitch ca...@usawide.net
First Issue to be addressed is how many simultaneous calls and bandwidth
availability.
Number of “lines” (numbers) is not a limitation in it self unless they are
all in use.
Sorry for being a bit unclear in this point. What I meant was 240 to 480
2009/3/24 Danny Nicholas da...@debsinc.com
Here are a few “look outs”; Using conference rooms will increase your
bandwidth requirements. On board Network controllers will affect
performance in this “high-use” scenario. 250 simultaneous calls will use
about 7.5Mb of bandwidth depending on
2009/3/24 Grygoriy Dobrovolskyy megaho...@gmail.com
If the switch is fine why not ? But i wander why kind if switch is that
240-480 fxo ? ;)
Sounds like a big overkill.
And i dont see a problem with asterisk, if not too much transcoding
involved and with the right hardware.
It's an ISDN
2009/3/24 Danny Nicholas da...@debsinc.com
I use a Dell with the 1Gb Ethernet on board, but had to clock it down to
100 Mhz because * has an issue with Dell on board Ethernet.
Ah - good to know. I think we will use SUN machines. But I'll keep that in
mind.
Chris
2009/3/24 Steve Gladden aster...@michiganbroadband.com
I REALLY like the Snom M3 DECT SIP base.
Yeah - it's such a pitty that you always have to buy it bundled with one of
these crappy handsets. Or is there a way to get only the base that I don't
know?
Chris
Maybe the Siemens DE380 IP R could help you. It's a brand new IP phone with
an integrated router.
Chris
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grandstream gxp-2000 works fine for that.
depending on firmware rev its two ports are either a switch or router.
Grandstream removed this functionality in recent softwware upgrades - I
guess they needed the code space for other things.
Why would you want a router in the phone and not let
2009/3/11 Håkan Källberg h...@simulina.se
Hello!
Does anyone of you have Caller Presentation working in the other
direction?? My mv370 is working well, execpt the Caller ID on outgoing
GSM calls. This works with the SIM card/Provider I am using if I put
the SIM card in a telephone, but not
2009/3/10 Sasa s...@shoponweb.it
Hi, I have modified in Mobile/Setting the parameter SIP From from
tel/user to tel/tel and now I view the correct incoming number.
Thanks.
Glad I could help. It took me nearly a month to figure that out. ;-)
Chris
2009/3/9 Sasa s...@shoponweb.it
Hi, I have a problem with Asterisk-1.4.22 (with TB 2.6.2) Portech MV-370,
my problem is that when arrived an external call I don't view (on my
internal phone) the phone number but I have the number extension that is
...
..now what parameter can I modify
2009/3/4 Atis Lezdins a...@iq-labs.net
Bottle of Riga Black Balsam (45%), just have to figure out a way to send it
:)
Balsam??? By mail? Doesn't that count as liquid explosive? ;-)
Chris
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2009/2/27 Bill Michaelson b...@cosi.com
Is there a convenient way to limit the number of call files (outgoing
directory) that are processed concurrently?
Afaik only by limiting the number of call files in the directory.
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2009/2/25 Alejandro Cabrera Obed aco1...@gmail.com
But in my case, I don't need trascoding because every chanel is in GSM
and voicemail has gsm sound files.
And for the moment, my Asterisk is not connected to the PSTN, so there
is no trascoding gsm-to-PCM or to analog.
So I think gsm is a
2009/2/2 Singer XJ Wang w...@pythian.com
[snipped]
You can do that by using fans other than the tiny, whiney, 40mm fans
that vibrate at 6000 to 18,000 Hz. A couple of 80 or 120 mm muffin
fans at the back or front, pushing air in (hence the deep
dimensions), but the top and bottom would need
2009/1/1 Olivier oza-4...@myamail.com
To attack DECT equipments, a ComOnAir module was used.
This module is a PCMCIA addon which provides DECT connectivity.
I don't think this module is available or manufactured anymore.
So it seems difficult for anyone to reproduce this DECT attack.
Or
Steven Howes schrieb:
Have created a system that involves using call files in the outgoing
spool folder. On some occasions it retries which is fine is there
any way to view calls waiting retries from the CLI? Using 1.4 btw.
Have googled to no avail (although it is near the end of the
Hi Ken,
we are quite satisfied with Linksys SRW248G4P. 48 port PoE, 4 GB uplinks
and 2 GBIC slots. VLAN, QoS and all the like is on board. Around US$600
I guess.
Only drawback in my opinion is that they are loud like a starting
airplane. You definately don't want them next to your desk. ;-)
Hi Asterisk users!
I have a little problem with an Asterisk 1.4.22 installation for a
customer. The PBX is connected to an E1 line and we have a few snom 300
attached to it.
The goal is to emulate traditional german PBX behaviour wich is the play
a stuttered internal dialtone after pickup and
Tom Moore schrieb:
What are your suggestions to people who have pbx systems that interface with
the world over pri and want to convert them to sip interfaces so that they
can use sip trunking?
I'd go for a Patton SmartNode. See www.patton.com - they have SIP
gateways up to 4 T1/E1.
Christian
2008/8/31 Olivier [EMAIL PROTECTED]
What happens if the PC supporting this card is powered off ?
It is powered over USB from the main (internal USB) and backup (external
USB) server. If one of the power fails it will switch to the other server.
If both servers power fail you have a problem
2008/9/1 Karl Fife [EMAIL PROTECTED]
It is powered over USB from the main (internal USB) and backup (external
USB) server. If one of the power fails it will switch to the other
server.
If both servers power fail you have a problem anyway. ;-)
This is incorrect. According to Jim Rhodes
I'm setting up my IVR system, how can I register in a mysql database the
IVR menus accessed by the clients ?
Just use the MYSQL-Functions in the dialplan to write the menues name
(and datetime maybe) in a table.
To access MYSQL from the dialplan you need to have the asterisk-addons.
Anton schrieb:
Does anyone tried BRI with asterisk for DATA transfer? My
customer
wants BRI connection, but he wants it for the data, and I
have to
bring connection to his office, so I see the connection as
follows:
E1-(CORE_ASTERISK)-(IAX2)-(EDGE_ASTERISK)-BRI - so
Why would
I don't know if there is something like that prebuilt. But is seems to be
quite easy. Push the call events in the database, let a cron run ever minute
and create a .call file for evry call thet is due.
The alternative is to not use a database and create a .call file with a
future date/time. Afaik
Hi!
I just upgraded my Asterisk server from 1.4.6 to 1.4.21 and now I experience
some strange behaviour.
1) The Asterisk CLI (asterisk -r) stops responding after some minutes. I
cant CTRL-C or exit the CLI anymore and no activity is shown. Just like if
the connection is interrupted.
2) When I
Hello!
I just encountered a strange thing in my mysql cdr records. From a
certain date on Asterisk (1.4.6) stopped to populate the CLIR and SCR
flieds in the cdr table. As far as I know no changes happened to the
system on that date and until then CLIR are recorded properly.
The CLIR is still
I have a standard E1 line, but want to receive only 10 calls
simultaneously. I want to give engaged tone to the 11th caller
onwards. Can I configure E1 to do this?
Yes - that can be done on the carrier side. Lines can be configured to be
outgoing or incoming only.
Christian
Gergo Csibra schrieb:
Well, using more than one TDM card in your PC is not a good idea,
because of interrupts. If you have to have 16 FXO you can more
options:
1. Using TDM2400P with 4 FXO modules ($1775)
2. Using Xorcom's Astribank (external) ($1170)
3. Using some T1/E1 card with Channel
fateme fatah schrieb:
Please every that work with A102d say how about is it?Is it really difficult
to install card for me new in asterisk?
Best regards.
It is not more difficult to install than any other E1 card for Asterisk.
In fact in my opinion it's one of the easier to install.
Christian
I have the same problem. My mail sent yesterday around 20:00h and it still
not arrived at the list. Sent from germany by the way.
Christian
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Hi!
Just ashort question - obviously I am too stupid too find the answer on
the net. :-)
I want to upgrade a running Asterisk 1.4.2 to 1.4.6 - what will I have
to do? Just install it over the existing version? Do I need to backup
the configuration? Will I need to reconfigure the source or
Did you try using the card in an other PCI slot?Chris
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Crazy Boy schrieb:
We have implemented Asterisk in our organization. There are 150 members in
our organization. At present all are using softphones. Now, I want to buy
hardphones for our staff. Can anybody suggest me that what is the best
hardphone for Asterisk with low-cost?
I would say a
Christian Stredicke schrieb:
snom 300 :-)
Could be a bit hard to get 150 of them at one time imho. ;-)
Chris
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Okay, same for the Eicon cards. I was just wondering about the almost.
Almost like in not in broken PCI slots or PCIe or 97,2% specification
slots ;-)
Christian
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Tristan schrieb:
Does someone knows about digium and sangoma ?
A lot ;-) What do you want to know?
Christian
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Tristan schrieb:
What would you recommend ?
Digium TE411P, Sangoma A104D, Eicon Diva Cards ?
Ah - I should have read this bevor my last answer. ;-)
I personally prefer the Sangoma E1 cards. The work in almost every PCI
system and the echo cancel - if you really need it - is far better than
Regarding echo cancel. Is there someone with hands-on experience
regarding the echo canceller performance of the Junghanns E1 cards
compared to for example the Sangoma ones?
Well - the Junghanns does the echocancel in software and the Sangoma
A104d does it in hardware.
So on the Sangoma echo
Hi Armin,
I personally prefer the Sangoma E1 cards. The work in almost every PCI
sorry to ask that, but what does almost every PCI system mean?
First of all compared to the Digium TE4xx the Sangomas work in 3,3V and
5V PCI slots. That means they run in every PCI slot but PCIexpress.
In
Well - the Junghanns does the echocancel in software and the Sangoma
A104d does it in hardware.
Alright, i just had a look at their product lineup. It seems as not only
the A104d but also the low end of their E1 cards (i.e. A101) comes with
this onbard echo canceller (EDAC), right?
No -
In fact the Sangoma has 128ms echo cancel per channel. As far as I know
neither the Digum harware echo cancel nor the available software
solutions offers this.
I thought the Eicon cards were the only ones with 128ms echo-cancel ;-)
I am happy that I could enlighten you a bit in this point.
Christian Victor schrieb:
I personally prefer the Sangoma E1 cards. The work in almost every PCI
system
Okay guys - I have to add something to this so it will not be misunderstood:
I never had problems with a Sangoma card in any mainboard and just added
the word almost to prevent somebody
Callum McGillivray schrieb:
We experienced this issue some time ago on our 2850.
Question - Are you using queues ?
In my case we did not use queues and we had the problems with a custom
made machine with a Intel Torrey Pines Mainboard.
Chris
___
Joshua Colp schrieb:
Edu wrote:
Hi!
We have an Asterisk Bussines Edition ABE-A.1-6 on a PowerEdge 2850
with 4Gb RAM. It was working 24/7 without any for a month, but for not
related causes I rebooted it a week ago. Yesterday the machine
suddenly stop working, with a kernel panic. We was
Well - a Sangoma Card won't bring you your money back. At least not
immidiately. ;-) And a expensive highend echo cancelling card is not a
good replacement for a relatively cheap TE405. So let's try to bring
your existing investion to work.
I presume you checked that your machine is working again
James Harper schrieb:
Has anyone every attempted to set up a mini PC to achieve much the same
functionality as the fonebridge box?
The sort of thing I'm imagining is a micro itx board case in a
completely solid state configuration (flash disk, maybe a psu fan but
only if really required),
John Jensen schrieb:
I'm looking for an interface card for termination of Euro-ISDN2 (BRI)
lines.
That is ISDN lines from the telco into my Asterisk box.
Any recommendations, good/bad expiriences ?
At present I'm looking at cards from BeroNet and Junghanns.
How many lines do you want to
I've just compiled asterisk-addon1.2.1 after installing MySQL and
MySQl-devel packages. and adjust my cdr_mysql.conf to use the defined
database using username and password. But as soon as starting asterisk
i get error messages informing me of error, error message is as
follows :
I'd recommend the Digium dual port cards - generation 2 card are
excellent and the support we receive superb.
And it supports Digiums support and development of Asterisk - Sangomas
contribution is token if any.
Unfortunately I cant speak for the 2nd gen. cards as we haven't used
them. The 1st
Matt Burleigh schrieb:
Thanks for the responses. I guess the next step is to get a Digium
TE210P. Are there any other 2 port PRI cards anyone would recommend for
*?
Yes - there are 2xPRI cards based on the CologneChip HFC-E1 chip and the
A102u cards from the Canada based manufacturer SANGOMA:
Chris Gamble schrieb:
I have 2 TDM04b cards currently running in an asterisk at home box that I am
ready to replace with the CVS version of asterisk. What I am looking for is
thoughts / recommendations. I want to move this to a small form factor (
shuttle ) machine and was wandering what
when using in NT mode does the card require additional power or is it
able to supply enough power by itself to the S0 bus?
You will need an additional power (for example
http://shop.beronet.com/product_info.php/products_id/48).
This is for the 4xBRI or 8xBRI cards from Beronet. The Billion
I'm looking at experimenting with asterisk with an ISDN BRI and ISDN
phones (since I have these already).
I saw that the Billion card was cheap and could be used in either TE or
NT modes.
I have the following question which I couldn't answer by reading through
the manual. Maybe someone has
Frank Sautter schrieb:
hi,
after stumbling over the compile time flag in zaptel and after reading
the new features of the 2nd generation firmware of the TE405P/TE410P, i
was wondering if the cards are capable of upgrading the firmware in field?
It is said so - but I don't believe it. ;-)
Hello!
Sorry for cross posting this message but I am in urgent need for E1
cards from Sangoma.
My company develops a telephony network for a worlwide operating
company. The hardware - including the E1 cards - will be set up
delivered by local service companies for the different national
Zeeshan schrieb:
hi,
I am going to open up a call center starting with 5 and expanding to 20
seats in 3 months. I have decided to use asterisk. I don't think I need
FXO or any other card from digium.
If you have any document regarding setting up a call center with
asterisk then please let me
Joao Pereira schrieb:
Im writing my dial plan, in witch every SIP phone begins with 74 and has
more 3 numbers (like 74XXX).
So, I want to route all 74XXX calls to my sip channel. For this I wrote
this line:
exten = s,1,Dial(SIP/[EMAIL PROTECTED],30,r)
but this way all calls go to [EMAIL
' in context 'default', but no invalid handler
I think it needs the s... but how do I put the s and route the call
to [EMAIL PROTECTED]
Thanks
Joao
Christian Victor wrote:
Joao Pereira schrieb:
Im writing my dial plan, in witch every SIP phone begins with 74 and
has more 3 numbers (like
Matt schrieb:
is euroisdn DSS1 protocol working with asterisk?
Jep.
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Hi!
- could the T1 channelbanks be connected to a TE405P with two channels
in E1 mode (telco and hicom pbx) and two channels to the channel banks
(i think yes, but just to be shure)?
Yes - no problem.
- will the faxmachines work (56kpbs-64kbps)? is asterisk translating
this (btw. how do
I'm trying to configurate my first asterisk system.
My test plant has an E100P Card and a 4 FXS TDM 400P card.
I've an E1 configured for 15 channels.
Telco says that crc4 is configurated so my zaptel.conf is:
loadzone = it
defaultzone=it
span=1,0,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
Hi Rgis,
Were going to build an IVR system with a TE405P and 4 E1. Were sure
that the 120 channels will be filled by 120 simultaneous calls during
peak, so we want to have the good server to manage this.
We wonder a lot of things and maybe you could help us.
- Are you ever build a similar
Christian Victor schrieb:
I am trying to suppres the transmission of my CallerID when I place a
call using a .call file in /var/spool/asterisk/outgoing
Okay - now I have a little Progress. :-) Suppressing CallerID on a PRI
is done by setting the CallingPres parameter. But unfortunately
What about the CallerID parameter in the .call file?
http://www.voip-info.org/wiki-Asterisk+auto-dial+out
Yes - that was what I thought too. But unfortunately leaving out the
parameter or setting it to '' will cause transmission of the default
number (usually subscriber number + 0)
On a PRI you
Hi!
What about the CallerID parameter in the .call file?
http://www.voip-info.org/wiki-Asterisk+auto-dial+out
I'm not the original poster, but I think this will not work. Just changing
the Calling Number (where the callerid field ends up in the isdn setup
message) to nothing will most of the
Henry Devito schrieb:
Most LEC's CLEC's, at least in our area, require sending a number (CSID)
before the call is completed. This is do to E911 features and ANI. If you
do not send a number the call will fail. If you truly want to block caller
ID I would contact your carrier and they should be
Hi!
At the moment I am trying to use Asterisks /var/spool/asterisk/outgoing
folder to dial larger amounts of calls at specific times to realise a
wakeup call/reminder solution.
The problem is that when I dump more .call files in the spool directory
the free lines I get many Unable to request
Hi!
I frequently get errors like Call failed to go through, reason 0 in
/var/log/asterisk/messages
Are the reasons (0,3 and 5 in my case) explained anywhere? I did not
find any info in the wiki.
Thanks,
Christian
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