Hello to all
I do not know what is causing choppy music on hold
when call comes in through E1 card (PRI).. but this channel info is somehow
strange.. We use Alaw over PRI (and I think its format number 8),
But why is WriteFormat at 2 ?
Thanks!
show channel Zap/1-1
--
Hi,
Anybody have some experience and did some testing
with ipVolution E1/T1 cards?
goran
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On Fri, 2005-10-21 at 09:39 -0700, Tielin Xu wrote:
Hi All:
I have a situation to be resolved.
Assume that one location call center with 150 agents.
I have two asterisk servers to serve those 150 sip phones. The servers
are connected to PSTN as 4 T1/PRI for each.
My question is why do you
. How big are
the files?
Thanks
BEN
Goran Skular wrote:
I changed my app_voicemail.c to work not with sendmail but with sendEmail
that connects to any SMTP and sends email with attachment...
It's dirty, but it works.
If you are interested I can upload app_voicemail.c and sendEmail package
On Tue, Oct 18, 2005 at 11:22:52PM -0500, Ben Brown wrote:
I have configured the voicemail.conf file as per the wiki to email
voicemails as an attachment. I cannot find any instructions/locations to
set the outgoing server login information. Furthermore, I can get no
emails from asterisk. Can
as much people, gathering
is to be expected next month (around 19th)
Send me an e-mail or even register on www.migo-systems.com. Further info will
be available later.
Looking forward for it,
Goran Skular
www.slsolucije.hr
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On Thu, Oct 20, 2005 at 08:58:01AM +0200, Goran Skular wrote:
I was playing with mta, but this is so complicated, specially if you are
on
dynamic ip address, so it is much easier to use smtp for sending mails..
Sending is never a problem. Recieving is a problem when you're on a
dynamic address
Hello,
I'm there with you, dude, haven't talked to you in some 5-6 years? :) I
know a couple of people that are working with Asterisk...
Cheers,
Vedran.
Nice surprise ! :)
Ok, you're the first participant along with me on this small gathering. I
sent you email, and let's ring on those guys you
Hi to all,
sorry for crossposting the -dev and -user lists, but I think this could
be quite interesting news for EuroISDN people, expecially BRI owners.
A new ISDN architecture, called vISDN, has been developed to fully
support EuroISDN protocol with HFC based cards: HFC-S PCI, HFC-4S and
HFC-8S
Always download programs directly from the homepage or from another
reliable source. Don't just grab programs and scripts from everywhere.
But why not just set mailcmd in voicemail.conf?
Also, quoting the homepage:
Why not use sendmail?
Sendmail is a large and complex mail server. Installing
people,
gathering is to be expected next month (around 19th)
Send me an e-mail or even register on www.migo-systems.com. Further info will
be available later.
Looking forward for it,
Goran Skular
www.slsolucije.hr
___
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I changed my app_voicemail.c to work not with sendmail but with sendEmail
that connects to any SMTP and sends email with attachment...
It's dirty, but it works.
If you are interested I can upload app_voicemail.c and sendEmail package
somewhere..
I have configured the voicemail.conf file as per
Here (Croatia) is also possible to get partial E1 (PRI/PRA)... from one of
our telcos (DT T-com) we can get PRA in 10 increments:
10B,
20B and
30B
We have a partial T1 (5B + D, iirc) from Allstream - there may be a
provider in your area that does something similar.
Regards,
--
Anthony Rodgers
They do not have NAT option.. and they do not have qualify...
Hi,
I have 3 SIP extensions, setup as follows:
# Device Location options
200 Sipura local
210 Sipura remote nat=yes qualify=yes
310 eyebeam remote nat=yes qualify=yes
This is the result of sip show peers:
Name/user Host
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 Austria. They deployed SS7 on Asterisk
Has anybody tried something like this:
http://www.arca-technologies.com/datasheets/arcaplexhorizon.pdf
It will be interesting to have ability to make systems like:
SCENARIO 1 (2 incoming BRI lines and 12 analog extensions with ability
to connect additional isdn devices to s0
For your information.. if someone get in
the same trouble.. problem is solved, but not with the software
We just changed our BRI NT device with a
different one.. from now on it works very well
We had Elcon NT1+2a/b and now it is
replaced with Santis ISDN NT1+2ab
Here is pri
Our Web is based on Mambo portal software and it is connected with our
Asterisk installation.
We wrote our own CDR rating engine and modules for Mambo. Also, you can
register for VoIP termination services inside mambo.. we wrote one component
and couple of modules.
So, when user create an
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 Austria. They deployed SS7 on Asterisk,
but not with Digium cards for one smaller telco.
___
We were using ilbc at first.. but it shows that it really needs a lot of
time for transcoding.. (CLI: show translation) resulting with hearable
delays.
So, where bandwidth is an issue, try to use g729. We are also using gsm, it
seems that it works very well.
OK .. what about ilbc ... could it be
What am I doing wrong here? Why is this happening?
libpri is version 1.0.7-1 (debian package) asterisk is version
1.0.7.dfsg.1-2 (debian package) zaptel is version 1.0.9.2
-- Executing Dial(SIP/739-5935, Zap/g1/0916000739) in new stack
-- Called g1/0916000739
-- Channel 0/1, span 1
Hi,
When we make an
outgoing call on ISDN (zaphfc) with overlap dialing we get immidiate hangup
after answer. But when we place a full number before dialing everything is ok.
Any help appriciated!! Thanks
here is info with
debug:
== Primary
D-Channel on span 1 up -- Executing
Hello all,
I have a problem with overlap dialing and don't know how to get rid of
it.
My setup is: 1 HFC card with bristuff - ZAP/g1 (2B + 1D channels),
SIP phones (I just removed TDM400P with 4 FXS)
I created test extension 222 which goes directly to g1. In Zapata.conf
My setup is: 1 HFC card with bristuff - ZAP/g1 (2B + 1D
channels), SIP phones (I just removed TDM400P with 4 FXS)
I created test extension 222 which goes directly to g1. In
Zapata.conf overlapdial is set to yes.
First I created this extension:
exten = 222,1,Dial(zap/g1,100,tc)
and channel
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