Good day all
Can hylafax work with asterisk..and how
I'm trying to find a way to send a fax over my E1 connection
Please Help
Altus
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I also have x100p cards but ... Shouldn't signalling in the zapata.conf file
be
signalling=fks_ks instead of signalling=fxo_ks?
I keep forgetting if you have to set fxo on zapata or zaptel...
One thing that I see when I do a reload on my asterisc box is this always:
Feb 23 02:36:47
Hello,
I am trying to come
up with a good way to enforce a limit on the number of simultaneous calls that
can occupy an IAX trunk at any given time. I have searched around and so
far can't locate a config option that would directly label a IAX trunk with a
specific number to obey (is there
Each release of BRIstuff is made for a specific * version. BRIstuff
installer automatically downloads the correct version, patches and
installs it. You should just run the install.sh and it will replace your
current * installation. Your existing configuration (extensions.conf
etc.) will not be
I have a server setup that runs sip no problem. I want to try a cisco phone.
how do I
a) Tell if I have skinny support loaded
b) Load it onto a debian system
How about 'show modules' from the cli?
Might look at contents of /etc/asterisk/modules.conf as well.
Hey Guys,
I need to create an automated system that will allow people to call in,
enter their code and be able to access all the various features (such as
pre-recorded messages with news, sports...etc). Also need to be able to
perform tasks based on buttons pressed on the phone. For example,
Guys.. Im doing a simple IVR system with some menus but I was wondering,
maybe it already does but does asterisk keep track of themenu hoices that
each call did? for example, is a caller calls in and then hits 1,3,2,6 does
that stay on some log file?
Would it be nice if it could also say stuff in realtime... Like using
festival or something??? Anybody using festival in text to speech apps
using php or something... Or even better, anybodu using festival with
spanish???
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
On Wed, 2005-02-23 at 01:07 -0800, Gabriel Afana wrote:
Hey Guys,
I need to create an automated system that will allow people to call in,
enter their code and be able to access all the various features (such as
pre-recorded messages with news, sports...etc). Also need to be able to
Guys.. I just saw this for the first time... I did some google and wiki
without any luck.. what does a red or yellow alarm mean in zaptel?
Feb 23 02:54:16 WARNING[16890]: chan_zap.c:5865 handle_init_event: Detected
alarm on channel 2: Red Alarm
Feb 23 02:54:24 NOTICE[16890]: chan_zap.c:5860
I know AGI is a method for languages like PHP to interact with
asterisk, but can it do this kind of stuff?
Should be quite straight forward to implement your requirement.
An AGI script using your favourite programming language is a way to go.
We have implement all bits you mentioned so
Hi,
I would like to use the asterisk box with zapta card to enable some
conferencing. I would like to use only TDM connections without VoIP. I'd
like also use the Meetme app. I have some questions:
1. Does any one use it for a few conference rooms at ones ?
2. Is it possible to restrict the
sure, but what about using asterisk?
On Feb 22, 2005, at 12:39, googleplex wrote:
google for inalp isdn sip gateway
On Tue, 22 Feb 2005 12:23:39 +0100, Roy Sigurd Karlsbakk
[EMAIL PROTECTED] wrote:
Hi
Is it, or could it be possible to gateway from ISDN videophones to IP
videophoning with asterisk
I have a * box running * version 1.0.3 with two X100P line cards in it and
Cisco 7960 IP
phones. Everything seems to work pretty well with the exeption that the system
hangs up on
phone calls for no apparent reason. It does this on both incoming and outgoing
calls through
the POTS line
It means for some reason you lost your CO line for 10 Seconds.
Either someone pulled the plug out by mistake or the Exchange line went away
for 10 seconds.
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Anton Krall
Sent: 23 February 2005 09:35
To:
Hi Guys,
Subject says it all I guess.
If so, can you post working config for
me
Thanks
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Title: cdr_odbc logging insane integer values
I'm having a problem with * (tried both HEAD and STABLE). When logging with cdr_odbc through unixODBC to MySQL, I get insane integer values in the duration, billsec, disposition and amaflags fields. I have enabled MySQL logging, and that's the
As soon as a call hits the asterisk a menu is
played "Press 1 for ... and 2 for..."
I have got the speech in different mp3 file and the
music in different mp3 file. is there anyway to mix these two
files?
Kindest
Muhammad Muzzamil Luqman
___
is there
anyway to mix these two files?
Use the soxmix utility.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Muhammad Muzzamil Luqman
Sent: February 23, 2005 11:01 AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] mixing
sound files?
Hi everybody !
I had patched asterisk to install chanspy weeks ago and everything was ok.
With the current version of cvs i am having failures when i try to apply the
same patch and the url where i originally downloaded it seems no longer
active.
Is the patch any longer maintained or has it been
Hi Roy,
-Original Message-
sure, but what about using asterisk?
On Feb 22, 2005, at 12:39, googleplex wrote:
google for inalp isdn sip gateway
Asterisk currently doesn't understand the ISDN video side. If you use one of
those gateways you probably could get it to interop with
Sorry to have had to
post this, But I need urgent help with configuring one adit 600 I picked
up from e-bay.
Issues. I
cannot access the console port, I am using HyperTerminal with settings VT100,
9600, 8-N-1
I also do not have
any user-manual so I am kind of stuck. Any help in getting
Hi there, quick question...do digium make any BRI cards (ISDN2) or even
better a quad port BRI, maybe im going blind, but I cant see any on their
website
Cheers
Gary
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I don't know of any download source, sorry.
However, if you get it working, please let me know - we need this :)
Julian.
Mamadou Lamine KA wrote:
Hi everybody !
I had patched asterisk to install chanspy weeks ago and everything was ok.
With the current version of cvs i am having failures when i
Hi all,
I have seen the term E1 and PRI used interchangably when referring to
a voice service with 30B channels and 1 D channel. Are they just
different terms for the same thing or is there some technical
difference. Even Newton's telco dictonary seemed a bit fuzzy on this
topic. I have seen it
I have this error when i try to conect to my remote
mysql server:
Host xxx.xxx.xxx.xxx is not allow to connect to this
MySQL server.
can some bady tell me what i have to do???
thanks in advance
wert
__
Do you Yahoo!?
PRI comes in 2versions E1 European and T1 US
E1 30 channels T1 23 channels
On Wed, 2005-02-23 at 14:15, Eric Bishop wrote:
Hi all,
I have seen the term E1 and PRI used interchangably when referring to
a voice service with 30B channels and 1 D channel. Are they just
different terms for the
On February 23, 2005 06:57 am, Etoenyo Ntumi wrote:
Sorry to have had to post this, But I need urgent help with configuring
one adit 600 I picked up from e-bay.
Sorry we don't provide urgent help here. That we charge for. Don't leave
things to the absolute last minute and then ask for
Hi, All
This is the technical support from Welltech.
About the registration with the Asterisk, this is because the old sip stack
version.
In the new sip stack of our firmware, it could resolve this kind of problems
with Asterisk.
If you need the new testing firmware version for our CPE, please
On February 23, 2005 02:20 am, Jean-Michel Hiver wrote:
NB: I use this modified script I found on the voip-info wiki. It makes a
very noticeable difference...
That's my rc.tc script. The latest and greatest version is at
http://www.mixdown.ca/~andrew/dump/rc.tc. I really should put that at a
Guys.. Im doing a simple IVR system with some menus but I was
wondering,
maybe it already does but does asterisk keep track of themenu hoices
that
each call did? for example, is a caller calls in and then hits 1,3,2,6
does
that stay on some log file?
I doubt it, but you could always run off an
Eric Bishop wrote:
Hi all,
I have seen the term E1 and PRI used interchangably when referring to
a voice service with 30B channels and 1 D channel. Are they just
different terms for the same thing or is there some technical
difference. Even Newton's telco dictonary seemed a bit fuzzy on this
asterisk user wrote:
Hello
I am using asterisk 1.0.0, here i am facing one
problem that the email-aatchment setting for each
extesion is not working individually.
When globally attach=yes is set the voicemail will be
sent as attachment no matter for any extension if
attach=no is set for it.
Same
Title: RE: [Asterisk-Users] Help With Adit 600 Configuration
;) thanks Andrew, and I do take the reprimand in good faith. I am using the Usual Cisco type roll-over cable.
When you say regular do you mean straight through ?, and can you kindly confirm for me if the HyperTerminal settings I
hello
how to register with irc. i want to connect to
#asterisk through x-chat
thanks
kamran
__
Do you Yahoo!?
Yahoo! Mail - You care about security. So do we.
http://promotions.yahoo.com/new_mail
E1 is a European T1. T1/E1 is the transport. PRI is the protocol. PRI on an
T1 id 23B+D, PRI on an E1 is 30B+D.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Eric Wieling
Sent: Wednesday, February 23, 2005 7:50 AM
To: Eric Bishop; Asterisk Users
On Wed, 23 Feb 2005, Eric Bishop wrote:
I have seen the term E1 and PRI used interchangably when referring to
a voice service with 30B channels and 1 D channel. Are they just
different terms for the same thing or is there some technical
difference. Even Newton's telco dictonary seemed a bit
Hi,
-Original Message-
Hi there, quick question...do digium make any BRI cards
(ISDN2) or even
better a quad port BRI, maybe im going blind, but I cant see
any on their
website
They don't. If you are in need of a european ISDN2 type, see if
Eric,
E1 is a physical layer protocol, like ethernet. It defines a 2Mbps pipe,
which can be used for data, or can be split into 32 64Kbps telephone
channels, or a mixture. If used for telephone channels, 30 of these
channels can carry one telephone conversation each, and 2 carry
signalling and
Hello
I am using asterisk 1.0.0, here i am facing one
problem that the email-aatchment setting for each
extesion is not working individually.
When globally attach=yes is set the voicemail will be
sent as attachment no matter for any extension if
attach=no is set for it.
Same in
I´m just using the CDR user field to store all the IVR information
Guillermo
From: Roy Sigurd Karlsbakk [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
Title: RE: [Asterisk-Users] Help With Adit 600 Configuration
Thanks Guys,
Straight through cable works just fine.
bst rgds
Etoenyo Ntumi
-Original Message-
From: Etoenyo Ntumi
Sent: Wednesday, February 23, 2005 7:51 AM
To: 'Asterisk Users Mailing List - Non-Commercial
We have set up an HP DL380 with 3 4BRI cards, Fedora core 2 (kernel 2.6.10)
and asterisk (bristuff-0.2.0-RC7f with asterisk 1.0.5). 4 ports are
configured in TE mode and connected to the PSTN; the other 8 are in NT mode
and connected to isdn phones.
the other outbound calls to PSTN are fine,
To all new subscribers:
This is not a friendly list, and the members are mostly sick of giving
away knowledge they reckon someone should pay for.
Although you will find friendly replies here and there, you'll have to
wade through many obnoxious replies from the * elite.
You will have to help
On Wed, Feb 23, 2005 at 04:21:04AM -0800, R A said:
I have this error when i try to conect to my remote
mysql server:
Host xxx.xxx.xxx.xxx is not allow to connect to this
MySQL server.
can some bady tell me what i have to do???
This has nothing to do with Asterisk. The error message
When you say regular do you mean straight through ?, and can you kindly
confirm for me if the HyperTerminal settings I am using are OK?. I am
definitely not using a null-modem cable, and yes the port works cos I use it
every day on various Cisco gear
I have called Carrier Access, I am just
Thanks Florian, that's great, is this card (junghanns QuadBRI) really stable
with * ?
Do you or anybody else have any experiences with this card and also is it ok
to run multiple cards in one machine
cheers
-Original Message-
From: Florian Overkamp [mailto:[EMAIL PROTECTED]
Sent: 23
RIGHT ON!
Too bad you also didn't post in HTML as well
Perhaps this list needs to be split?
One for the folks who simply want to get it working, and another for
the self appointed list police who want to be rude and nasty and are
only interested in feeding their own egos, all the while
Hi,
Is it possible to send CDR to a database (cdr_mysql.so for example) and to files
(cdr_csv.so) ?
As soon as I activated CDR writes to mysql, Master.csv stopped to grow, and
since CDRs seems to be registered in a linked list in cdr.c I thought it was
possible...
TIA,
--
Ludovic DROLEZ
On February 23, 2005 07:51 am, Etoenyo Ntumi wrote:
;) thanks Andrew, and I do take the reprimand in good faith. I am using
the Usual Cisco type roll-over cable.
Nope this isn't Cisco gear. You want a regular 9-pin serial cable (straight
through).
When you say regular do you mean straight
ok, not that i'm such an expert myself, but
1. there's a big difference between newbies asking specific question
and the i want asterisk to run my life, make me coffee, and solve my
problems, does asterisk do that? questions that are appearing lately.
I'm not a member of the list police and they
On February 23, 2005 08:25 am, [EMAIL PROTECTED] wrote:
This is not a friendly list, and the members are mostly sick of giving
away knowledge they reckon someone should pay for.
Although you will find friendly replies here and there, you'll have to
wade through many obnoxious replies from the
could be that someone plugged out ur telephone line and plugged it back in.
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BTW what versions of libtiff spandsp u using, cuz i can't recieve
faxes at all.
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Is there a way for asterisk to notify you of this? Send an email? Send a
page? Call you?
-Matthew
- Original Message -
From: Giovanni Powell [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion asterisk-users@lists.digium.com
Sent: Wednesday,
Alistair-
Good writeup! Question regarding Q.SIG: Can it be used to solve the
problem of signaling a remote switch to take a call back and extend it
to another channel instead? This, as you know, is always a challenge
when using IVR in a call centre environment, when one wants to extend an
Hello,
Before putting any effort, I
would like to know if somebody has successfully run asterisk receiving FAXs in IP and sending them out in IP as well?
If yes using which components
please?
Any help is greatly
appreciated !
___
Hi Gary,
-Original Message-
Thanks Florian, that's great, is this card (junghanns
QuadBRI) really stable
with * ?
Do you or anybody else have any experiences with this card
and also is it ok
to run multiple cards in one machine
We have built and use systems with one QuadBRI
I'm was having a couple issues also, mainly callerid when turned on
was crashing asterisk, but its was my fault still.
But does any of the digium cards beside x100p offer redundancy.
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Yair wrote:
1. there's a big difference between newbies asking specific question
and the i want asterisk to run my life, make me coffee, and solve my
problems, does asterisk do that? questions that are appearing lately.
Who else is going to tell them that they need a wife? Do we have to be
Andrew wrote:
Please remove your head from your arse and reread what I wrote. THEN hit
the
reply button and post something relevant.
What the hell makes you think this post had anything to do with you?
Thanks for being polite!
Teddy
___
I'd say that would depend on the configuration you are considering. We
have a number of fax machines running off of sipura spa-2000's that
connect to a remote asterisk server and terminate to the pstn via voip
as well.
I'd say it's about 90% reliable at this point. However, we've noticed
Scott,
Do a search on Tromboning I have no idea if asterisk is capable of doing
this but I remember this was a feature introduce into Fujitsu Qsig stack
in or about 94-95 which solved a heap of customer problems at the time
so I remember it was a big deal.
Cheers,
Dean
-Original
Scott,
Yes, and this is one of the principal reasons people choose Q.SIG.
I've worked on quite a few large voicemail servers, and these tend to do
a lot of transfers for follow-me and operator features. Q.SIG support
can significantly reduce the number of telephony channels needed, as not
only
Neither can I. Check out my bug for my list of versions:
http://www.opencall.org/mantis/bug_view_page.php?bug_id=029
-Matthew
- Original Message -
From: Giovanni Powell [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
On February 23, 2005 09:43 am, [EMAIL PROTECTED] wrote:
And specifically in this case, recyclying a subject line that
has nothing to do with your email is just lazy and screws up threads.
Oh beg I your pardon, but I didn't have access to previous messages to
spot that this subject was
On Wed, 23 Feb 2005 [EMAIL PROTECTED] wrote:
2. many of the list police are active in the development process
well, so your remarkably clever comments about the lack of help are
uncalled for and untrue. People will help you, but they won't hold
your hand. If you want your hand held, then
On February 23, 2005 09:31 am, Hakem Taourchi wrote:
Hello,
Before putting any effort, I would like to know if somebody has
successfully run asterisk receiving FAXs in IP and sending them out in
IP as well?
No. Not without t.38. Googling for asterisk fax IP site:lists.digium.com
should
On February 23, 2005 10:03 am, Peter Svensson wrote:
New Asterisk users should try to help themselves first. Perhaps we should
create a list asterisk-newbies for thos who do not want to go through the
effort of reading up enough to ask well formulated questions.
That already exists.
Looked at the mISDN bits and pieces, which looked promising, so decided
to take the plung and go for Mandrake 10.1 (Kernel 2.6.8.1) and well -
What a disaster!
I should probably have mentioned I am using an EPIA 5000/classic
motherboard, which between Mandrake 10.1 and the motherboard come up
Thank you ver much for this help Steven
What I am planning is this:
1-) Receive fax on a DID that is being routed in IP to the asterisk
server;
2-) Based on the rule on that incoming fax, Asterisk needs to capture
it, store it as pdf file and e-mail it to a predefined destinoatin
(based on
Peter Svensson wrote:
New Asterisk users should try to help themselves first. Perhaps we should
create a list asterisk-newbies for thos who do not want to go through the
effort of reading up enough to ask well formulated questions.
Strikes me as a very good suggestion.
Perhaps all subscribers to
*spews coffee over keyboard*
- FUNNIEST - THREAD - EVER -
Also one of the most insightful.
Teddy, your gmail invite is on the way.
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to the list? I've never gotten a clear answer. It seems like a hell of a
lot of work to reply to a message, erase everything and start anew.
Oh I'm sorry. This is the first list I've joined where this is such a big
issue! Forgive me for not having your superior understanding of mail
clients,
Please remove your head from your arse and reread what I wrote. THEN hit
the reply button and post something relevant.
What the hell makes you think this post had anything to do with you?
Children! PLEASE! You'll ALL be going to bed with no tea at this rate. I
don't care WHO started it, you can
What is the best Asterisk manager to use, i do not
mind web based or GUI.
Thank You
Kanishka
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Hi
I want to
create 2 groups of extensions, for example group 1 cant make outgoing calls
they can only call other extensions and extensions of group 2. group 2 can call
any of theextensions + they can make out going calls using our SIP
server.
Please
let me know how to do this. I was
Hi,
We're running a very old version of Asterisk
(CVS-HEAD-08/03/04) and we're having some
problems with logging.
Our logger.conf has the following:
full = notice,warning,error,debug,verbose
After having started Asterisk, asterisk will hang in
/usr/sbin/asterisk -rx 'logger reload'
I don't believe this will work, but haven't tried myself. TAFM requires
spandsp. I'd do some investigation there as to whether spandsp can
function with g711 (the last i checked it didn't).
Good luck - if you make any progress please post to the list.
-Steve
Hakem Taourchi wrote:
Thank you ver
-Original Message-
From: Peter Svensson [mailto:[EMAIL PROTECTED]
Sent: 23 February 2005 15:04
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] List tips for new subscribers
This list is for discussions among users of
On Wed, 2005-02-23 at 15:25 +0200, [EMAIL PROTECTED] wrote:
To all new subscribers:
This is not a friendly list, and the members are mostly sick of giving
away knowledge they reckon someone should pay for.
Oh fun, we hit slashdot once again. Here comes a new wave of annoying,
lazy, and 3-10
in extensions.conf, create a context for your internal extensions. In the context for outgoing calls, add include = internalextensions. Then in zapata.conf, for each extension put context=internalextensions for people with no outgoing access, and put the others in the context of the outgoing
Hi,
I've got an * box with 4 E1 (TE405P) and a Class 4 Cirpack switch
(www.cirpack.com).
IP Network--*--Cirpack--Public PSTN Network
ISDN interco is EuroIsdn, ports are configured ccs, hdb3, crc4. Cirpack
is Network, * is Terminal/User.
As I encountered some pb with Sip to Zap transcoding
On Wed, 2005-02-23 at 15:19 +, Adrian Chapman wrote:
Peter Svensson wrote:
New Asterisk users should try to help themselves first. Perhaps we should
create a list asterisk-newbies for thos who do not want to go through the
effort of reading up enough to ask well formulated
G'Day All.
So I got the TFTP server all set up -thanks to much help from this list-
the 7960 found it and updated to SIP the first firmware P0S30200. What I
am now trying to do is upgrate through all the versions, as recommended,
to the latest version, P003-07-3-00.
I thought this would be
Razza wrote:
Trying the get back to position of a running * PBX, I tried to install
the zaptel drivers, using the following process -
CD zaptel.
Make linux26
Make install
When I modprobe zaptel I get the following errors -
[EMAIL PROTECTED] zaptel-1.0.4]# modprobe zaptel
FATAL: Error
On Wed, 2005-02-23 at 15:27 +, Kanishka Somaratne wrote:
Hi
I want to create 2 groups of extensions, for example group 1 cant
make outgoing calls they can only call other extensions and extensions
of group 2. group 2 can call any of the extensions + they can make out
going calls using
Hi,
I did recognice an rather strage behaviour on Music on Hold:
Situation
Caller C does call Person A
Person A puts C on hold to ask B
MOH is (correctly) activated for C
After talking to B A does hangup to transfer C to B
In this moment MOH is activated for
I am using an IAX connection via Free World Dial Up.
On a incoming call from any of the Michigan based numbers the DTMF for the
menus work just fine.
But when calling in from any number from
http://www.notaduck.com/My_Homepage_Files/Page1.html
DTMF does not work. Is this a problem on my end ?
This list is for discussions among users of Asterisk, not a getting
started hotline for beginners. Beginners learn by reading documentation
and examining the sample files included.
Mmm, I (respectfully) disagree. One of the unstated objectives of mechanisms
like this list is to evangelize the
From: Niksa Baldun [EMAIL PROTECTED]
Each release of BRIstuff is made for a specific * version. BRIstuff
installer automatically downloads the correct version, patches and
installs it. You should just run the install.sh and it will replace your
current * installation. Your existing
[EMAIL PROTECTED] wrote:
(side note: If you havent bought their hardware and are using
Asterisk for free them again you should expect even less
assistance imo)
Right, so I have to buy hardware I don't need?
--
Andreas SikkemaRits tele.com
Van Vollenhovenstraat 33016 BE
On February 23, 2005 10:21 am, [EMAIL PROTECTED] wrote:
Oh I'm sorry. This is the first list I've joined where this is such a big
issue! Forgive me for not having your superior understanding of mail
clients, and/or list servers!
You have a *servere* inferiority complex.
I asked a simple
I am a serious Asterisk newbie: just installed asterisk last week and it is
now running with our Voicetronix OpenLine4 hardware.
All is working as expected with one exception, in the following sequence
(extracted from my extensions.conf file):
[GetConfirmation]
exten = s,n,SetVar(TimeOut=0)
Colin wrote: A lot of good sensible stuff. Well done Colin.
Bill Seddon
Lyquidity Solutions Limited
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Colin
Anderson
Sent: February 23, 2005 3:54 PM
To: 'Asterisk Users Mailing List - Non-Commercial
How do I route all the outgoing calls
througha SIP gateway, this should send more than one outgoing call to the
sip gateway at once. please show me the sample configurations on how to do
this.
my SIP gatway can accecpt direct IP traffic or SIP
proxy traffc.
Thank You
Kanishka
Hi,
I'm trying to connect a PC with a TE410P to an E1/IP equipment.
Unfortunately I keep getting a yellow alarm from zaptel (in zttool)
and a Loss of Framing alarm on the remote equipment.
The E1/IP is connected on the other side to a PRI interface on a GSM
MSC.
I have configured the span as:
You have to change the image name in the OS79XX.txt and SIPDefault.cnf files
to match the name of BIN file you are trying to load ... With versions of
the firmware prior to 7.x, the name you put in the OS79XX.txt file and the
SIPDefault.cnf files are the same; simply the BIN file name less the
On Wed, 23 Feb 2005, Johan Bilien wrote:
I'm trying to connect a PC with a TE410P to an E1/IP equipment.
Unfortunately I keep getting a yellow alarm from zaptel (in zttool)
and a Loss of Framing alarm on the remote equipment.
The E1/IP is connected on the other side to a PRI interface on a
After downloading the firmware from a tftp server for
avaya 4602 ip phone, my phone console keeps getting
message saying contacting http server.
according to this link:
http://voip-info.org/wiki-Avaya+4602+configuration
My phone hangs at Contacting HTTP Server while
performing a firmware
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