On Thu, 11 Aug 2005, Joseph wrote:
In this case could somebody explain to me why run asterisk with ''-p
switch?
According to asterisk man explanation for -p is as follow:
If supported by the operating system (and executing as root), attempt
to
run with realtime priority for increased
- Original Message -
From: Ronald Wiplinger
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Friday, August 12, 2005 6:04 AM
Subject: [Asterisk-Users] list in asterisk cli is getting too long
How can I use something like|morein CLI ?
The lists are getting too
On Thu, 11 Aug 2005, Geoff Manning wrote:
We are having line noise issues in our Asterisk to legacy PBX integration.
All SIP calls originating from IP phones sound crystal clear. All calls that
originate from the legacy PBX (Isoetec 228) and route through the Asterisk
and out SIP have a lot
If you're using Cisco Switches:
Logon to the switch and go to config mode
int fa0/1
switchport access voice vlan untagged
sometheing in that direction configures the CDP to set the phone to
untagged frames.
Julian Lyndon-Smith wrote:
How ? Where ? I've been wanting to do this for ages, and
Hi
I got the 2nd Gen firmware upgraded on the TE405P. I recompiled after
putting in the upgraded board but did not change any conf, but the spans
become active but will not come up.
I guess I am missing something or are the any changes to the
zaptel/libpri software that is required. I
Some of my customers are using PSTN to call each other from the VoIP system.
I want to stop that, by setting up all internal numbers to be reachable
via VoIP first.
E.g. A calls B via VoipJet !!! But B is on our system.
I want now set it up so that if B is reachable via VoIP, than it
Is there an option txgain for SIP in Asterisk? My users all complain that
their other parties think that they are way too silent even though they all
have their mic volume all the way up and also enabled the 'mic boost' option.
This happens with all the clients that we're using and also with
Hello,
I've got a Junghanns ZapHFC E1 PRI Card (cwain) and this driver writes
very much messages into /var/log/messages like the following:
--- snip ---
Aug 2 17:58:02 asterisk1 kernel: cwain: card 1 RX [ 0x2 0x1 0x1 0x37
0x90 0xc3 ] 6 bytes
Aug 2 17:58:02 asterisk1 kernel: cwain: card 1 TX
Oh damn. I'm not using cisco switches, but a dell 3348 (I know, I know)
No way to turn it off on the phone, then ?
Julian.
Erik Versaevel - Infopact Netwerkdiensten BV wrote:
If you're using Cisco Switches:
Logon to the switch and go to config mode
int fa0/1
switchport access voice vlan
Hi,
we also got one V2 TE405P card. It works fine now. At the moment we use for
bridging the Pri to our old PBX.
You must use zaptel, libri and asterisk v 1.0.8 or higher. We use 1.0.9 at the
moment.
zaptel:
after make; make install i also executed make config. This copies the correct
startup
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
Hello,
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).
Best regards
Blaise
On Friday 12 August 2005 09:43, John Fawcett wrote:
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?
I don't know the exact specifics about the Billion card, but I have a setup
where I have an extra NTBA connected to the
Hello all,
can anybody how usable app_sms is? I want to use it in england (but not
with the BT) and in Germany. Is this possible with * 1.0.9 and either
with PRI lines or with simple ISDN and an AVM Fritz! Card??
Thx in advance,
Tobias Wolf
___
I have a * system at home with mainly 7960's. I run a trunk into each of the
phones, it is the only way to get cos bits set correctly from the phone into
the switch. I then translate the cos bits into DSCP for layer 3. I have
separate vlans for voice and data, * server has 2 nic's one in each
Is there a significant difference between the features set of [EMAIL PROTECTED]
and regular asterisk? (important missing features?)
About the cautions.. You're not kidding.. It blew out the hard drive
without so much as a you sure you want to do this?.
Unfortunately I figured this out too
The VoIP Connection a écrit :
Nicolas,
Just did some quick testing and the instructions are incorrect. You need to
press transfer to complete the transfer instead of the second flash.
This actually makes more sense.
Attended and regular transfer both work perfectly with the following
I'll second that. Make sure your script is in
/var/lib/asterisk/agi-bin and you have the right permissions on it. I
really just wanted to reply to your post though to congraduate you,
Dan Marino, on your recent induction into the Pro Football Hall of
Fame ;)
Sorry, wrong Dan Marino!
-Dan
On 8/10/05, Jean-Michel Hiver [EMAIL PROTECTED] wrote:
Dan Marino wrote:
I have installed the Perl library from
http://asterisk.gnuinter.net/asterisk-perl and am wondering how I
reference agi-test.agi from extensions.conf
I have added
exten = s,1,AGI,agi-test.agi
but that doesn't seem
Anybody knows, how to use the SIP_HEADER function?
Thanks
Tomas
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
On Thu, Aug 11, 2005 at 07:12:32PM +0100, Mark Thorpe wrote:
I have been trying to solve a problem wherby when I boot a cisco 7920 my
7940 seeks a new IP and the dhcpd log shows it released its existing IP. In
searching for the solution I notice there were 2 messages on this list in
Aug Sep
Michael Boger Jr wrote:
Sean,
What kind of hotel do you have? Some PMS vendors require the call accounting
and check-in interfaces to their system. I am not aware that asterisk
supports these serial interfaces.
No they have no call accounting etc as such everything is done manually.
I
Hi, all
I want voicemails to be delivered to recepients by e-mail. I have set up
voicenail, and I can see asterisk is using sendmail to send messages out.
Using Ethereal, I can see that messages are leaving my network, but
receipeint mail server never replies back. As a result, mail delivery
Default sendmail should work. Try to test sendmail from console. Some SMTP
maybe block the email.
run mail to see if your email is bounced back.
Kun
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Rudolf
Ladyzhenskii
Sent: Friday, August 12, 2005 5:10 PM
Thanks for reply.
I would expect it to work too, but it does not. I tried to send mail from
console -- same result. Messages are just sitting in teh queue. sendmail
times out sending them. Mail does not bounce.
Rudolf
- Original Message -
From: Wei Kun [EMAIL PROTECTED]
To:
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 think, I just sorted i out.
I have to run sendmail with optiosn -bm to be a mail sender. Without it, it
seems that sendmail is trying to use outside server for delivery. Without
valid username, this will not work...
Rudolf
- Original Message -
From: Rudolf Ladyzhenskii [EMAIL
how come you said mail is send out but still in the queue? Does it send out
or not?
Kun
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Rudolf
Ladyzhenskii
Sent: Friday, August 12, 2005 5:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
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
Old messages are in the queue.
I can see sendmail is trying to talk to the remote mail server, but never
gets a responce and times out. So message stays in the queue.
Rudolf
- Original Message -
From: Wei Kun [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial
On Friday 12 August 2005 11:46, Rudolf Ladyzhenskii wrote:
Old messages are in the queue.
I can see sendmail is trying to talk to the remote mail server, but never
gets a responce and times out. So message stays in the queue.
You should try and deliver them yourself. It sounds to me like
Hello,
Is it possible to use v92 ( a few chipsets version )
modem as FXO PCI modules ?
While googling I found some postings on the subject.
Thanks
Varun
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
l/asterisk/monitor/45/47-20050812-113631-in.wav"
"//var/spool/asterisk/monitor/45/47-20050812-113631-out.wav"
"//var/spool/asterisk/monitor/45/47-20050812-113631.wav" rm -f
"//var/spool/asterisk/monitor/45/47-20050812-113631-"* ) "and It doesn't wo
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?
regards
frank
___
On Friday 12 August 2005 06:42, [EMAIL PROTECTED] wrote:
Hello,
Is it possible to use v92 ( a few chipsets version )
modem as FXO PCI modules ?
Short answer: no.
Longer answer: perhaps, but you're on your own. Your googling efforts should
have shown you that. :-)
-A.
[EMAIL PROTECTED] wrote:
Hello,
Is it possible to use v92 ( a few chipsets version )
modem as FXO PCI modules ?
While googling I found some postings on the subject.
Thanks
Varun
___
Asterisk-Users mailing list
Are you using Redhat/Fedora? If I remember those init scripts is for
Redhat/Fedora. I am using gentoo.
Did you make any modifications to wct4xxp.c. or pass any parameters to
zaptel. I see there is a #define SUPPORT_GEN1 in to wct4xxp.c which I
commented out, but it made no difference. ztcfg
Christian Victor wrote:
The cards feeds no power to the s0. If the phone has its own power
supply normally it will work without external power supply. Otherwise
you will need a network terminator (NTBA) or a selfmade power injector.
Christian
Christoph Eicke wrote:
I don't know the exact
On Friday 12 August 2005 06:59, Frank Sautter wrote:
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?
Unfortunately not.
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. ;-)
John Fawcett wrote:
I have a doubt about how to connect the NTBA, since it
has a U interface and an S0 interface with two sockets. Would I connect
the Billion card to one of the S0 sockets on the NTBA (via a crossover
cable) and then the telephone to the other S0 socket. I assume that I
don't
Short answer: NO
Long answer: you have to send it to Digium for them to do an upgrade,
they don't have an official process for this yet and won't give you a
price, I have called and asked them many times. They also mention
upgrades from your 405/410 to a 406/411 are available too, but again
no
Easily doable. I've done it twice now. Problem is that your users will
never know they have messages waiting.
Install a T1/E1 card into the * box and then use a T1 cross-over cable
between the 2 boxes.
Create a dialplan on the Meridian that points calls to the VM out over
the new E1.
As
AAH uses the latest released version of *. This is somewhat different
to the CVS but can be considered stable for the purposes of production.
It also has the added benefits of a web front end to configure it as
well as some other nice webby features to make life easier when running it.
The
Yes, but your results may vary. Apparently some people have problems
with clone cards (aka regular modems), dropping calls, and having
echos. (Then again some people have reported no problems at all).
E-bay is a good source for these. You can also check out this list
with more information about
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.
___
Kevin P. Fleming wrote:
Geoff Manning wrote:
The TE110P card in the Asterisk server is set as the sync source:
span=1,1,0,d4,ami
em=1-24
That is incorrect. You have your span configured to recover timing
from the T1 and use that as the source for the card. If you want this
span to be
Anton Krall [EMAIL PROTECTED] wrote:
[EMAIL PROTECTED] intruder]# ps afx|more
PID TTY STAT TIME COMMAND
1 ?S 0:08 init
2 ?SW 0:00 [keventd]
3 ?SW 0:00 [kapmd]
4 ?SWN0:00 [ksoftirqd_CPU0]
9 ?SW 0:00
Peter Svensson wrote:
A blue alarm sounds really strange. That indicates that the remote end
(asterisk) in this case does not want to play at all. On a T1 it is
sent as a continous series of unframed 1:s. I am not sure if asterisk
ever sends a blue alarm (Alarm Indication Signal).
Receiving
I think it would help if you sent an excerpt from your maillog.
Cheers
Wayne
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
Only because I dont want to install yet another IM to
my existing Skype and MSN has anyone tried the new yahoo voice?
http://www.smh.com.au/news/technology/scramble-to-find-voice-on-the-web/2005/08/12/1123353481827.html
http://messenger.yahoo.com/feat_voice.php
Any thoughts?
Give me a handset with a ringer and hook that both work under Linux and
I will buy 30 pieces on the spot!
No kidding..
Bill McCready (PCPhoneline.com) wrote:
We are planning to develop versions of our USB based phone and gateway
products for Linux. The plan is to make them will work like
I got a good deal on one of these channel banks loaded with 24 FXS
ports. I know 24 seems pretty overkill for a home user, but I got this
shipped cheaper than I could have gotten a TDM400P w/ 1 FXS port. I've
read that these are compatible w/ asterisk, but can they be used w/o a
T1?? (I'm
I've got the latest zaptel and cvs asterisk software loaded on my phone
server running FC3. And yes, It's fully updated and udev is setup
correctly. I've got a TDM400P with one FXS and one FXO module
installed. When I load zaptel and wctdm and run ztcfg -vvv, I get this:
[EMAIL PROTECTED]
Chris,
Maybe you could write a generic config file and post it to the wiki?
MAC address =
Server IP = xxx.xxx.xxx.xxx
Username = user
password = pass,
Extension = 100
Just a thought
Tom
On Aug 11, 2005, at 10:15 PM, Chris Mason wrote:
Shaun Bolling wrote:
Jonathan, did
Hi Jeremy,
thanks for this, it looks like just what i was after. i've not finished
testing
it but it seems to do what i need.
Steve
On Tue, 9 Aug 2005, Jeremy Gault wrote:
Steve,
If I am understanding your situation correctly (i.e. you are using a SIP
client and then forcibly
Kevin P. Fleming wrote:
Geoff Manning wrote:
The TE110P card in the Asterisk server is set as the sync source:
span=1,1,0,d4,ami
em=1-24
That is incorrect. You have your span configured to recover timing
from the T1 and use that as the source for the card. If you want this
span to be
Wei Kun wrote:
mysql select * from extensions_table;
++--+---+--+---++
| id | context | exten | priority | app | appdata|
++--+---+--+---++
| 1 | from-sip | 2000 |
Hilton Williams wrote:
- Original Message - From: Ronald Wiplinger To: Asterisk Users
Mailing List - Non-Commercial Discussion Sent: Friday, August 12, 2005
6:04 AM
Subject: [Asterisk-Users] list in asterisk cli is getting too long
How can I use something like|morein CLI ?
[EMAIL PROTECTED] is not a feature limited version of asterisk designed for home
users. It is actually a collection of various software projects,
including Asterisk stable, the AMP Web interface, SugarCRM, a Cisco
phone configuration editor, plus ConfigEdit and a bunch of other
stuff, all
Matt Florell wrote:
Long answer: you have to send it to Digium for them to do an upgrade,
they don't have an official process for this yet and won't give you a
price, I have called and asked them many times. They also mention
upgrades from your 405/410 to a 406/411 are available too, but again
Andrew Kohlsmith wrote:
Unfortunately not. It's a configuration PROM, not EEPROM or FLASH. I am
pretty sure that Digium has an upgrade program in place though. It's best to
contact them directly for these types of inquiries instead of the list.
Actually, it is EEPROM, and will be
Well that would explain the choppy/stuttering sound we get on these
calls
since there is no audio during those error transmissions.
According to Kevin's reply I had my timing logic backwards. Should
I be
using any other timing settings on the Asterisk side?? The tech for
our
legacy PBX
RedHat also configures their sendmail to not accept connections from
any servers other than the localhost. Although I wouldn't expect that
to affect outgoing mail, I have found that it often does.
Google for redhat sendmail DAEMON_OPTIONS 127.0.0.1
Or check out this article:
Hi Matt,
Thanks for the information.
regards
Somesh
-Original Message-
From: Matt Riddell [mailto:[EMAIL PROTECTED]
Sent: Thursday, August 11, 2005 11:43 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [Asterisk-Users] help on receive text
On Friday 12 August 2005 09:30, Kevin P. Fleming wrote:
Andrew Kohlsmith wrote:
Unfortunately not. It's a configuration PROM, not EEPROM or FLASH. I am
pretty sure that Digium has an upgrade program in place though. It's
best to contact them directly for these types of inquiries instead
I've been using spandsp and rxfax to receive faxes for a while now and over
the past few days I've looked into the other side of things - txfax. I
can't seem to get it working properly. I've included debug logs below of
both the tx and rx side of things. I've tried three different servers,
It would be nice if there was a dialplan for each registration or
line, which would allow me to never press send for any of the systems I
register to
On my Sipura there isI haven't checked one of the polycoms but I
suspect they are no different.
-Original Message-
From: [EMAIL
On Fri, 12 Aug 2005, Geoff Manning wrote:
OK. So I changed it to:
span=1,0,0,d4,ami
And the Blue Alarms are still occurring but now in conjunction with Slip
errors. I feel like I am on the right track though.
Which side shows the slips?
I am not that familiar with T1, Are you sure the
Our vendor told us we can't buy the 841's anymoreanyone else have
this problem or have a vendor that is still selling them?
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex
Ternero
Sent: Wednesday, August 10, 2005 9:44 PM
To: 'Asterisk
On Fri, Aug 12, 2005 at 08:46:45AM -0400, Jeff Borders wrote:
I've got the latest zaptel and cvs asterisk software loaded on my phone
server running FC3. And yes, It's fully updated and udev is setup
correctly. I've got a TDM400P with one FXS and one FXO module
installed. When I load
I'm still having problems getting this to work. I cannot get anything to
display on my 7914 other than blank lines.
I have SIP/5920-5930 in [main] that I'd like to add to the 7914 and
indicate hook status. The 7960 is registering okay as SCCP/5000.
What exactly should my sccp.conf file look
On Wed, Aug 10, 2005 at 09:58:09PM -0600, Rich Adamson wrote:
That's a crack of crap sold by the marketing (not sales) people selling
firewalls. If you know what you're doing, one can very easily secure any
linux system to function on the Internet (etc) without a firewall. It all
depends on
yes, fedora 3 but without any changes at the sources
Master Abi wrote:
Are you using Redhat/Fedora? If I remember those init scripts is for
Redhat/Fedora. I am using gentoo.
Did you make any modifications to wct4xxp.c. or pass any parameters to
zaptel. I see there is a #define SUPPORT_GEN1
I'm attempting to use Festival with Asterisk on an x86_64 system. This
IVR application works ok on a P4 system.
I'm using the FC3 x86_64 distro on a single processor Opteron system.
Festival by itself (using the command line and speakers) seems to work
ok, and Asterisk without Festival works
Hi List,
I'm wondering if someone who uses VoipJet as their termination service
would do me a favor.
If I call the American Airlines reservation number (1-800-433-7300), the
call gets connected, but after 30 seconds asterisk drops the call
responding that no one answered.
I'm using
On Fri, Aug 12, 2005 at 07:09:45PM +1000, Rudolf Ladyzhenskii wrote:
I want voicemails to be delivered to recepients by e-mail. I have set up
voicenail, and I can see asterisk is using sendmail to send messages out.
[snip]
I got a book on sendmail and it looks quite complex.
Right.
Hej!
Jag är på semester vecka 33, åter igen på kontoret 22 aug
mvh
Gunnar / JMG
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
Peter Svensson wrote:
Which side shows the slips?
The slips are seen on the legacy PBX side (Isoetec 228)
I am not that familiar with T1, Are you sure the signalling between
the pbx and asterisk is set the same on both?
Unfortunately I am not aware of the signalling set on the Isoetec
We have a TDM400P installed here with four FXS modules. It works well
except for a couple of issues:
First, I have a Panasonic KX-TG2431 telephone (so others can reach me
when I am in o ther parts of the building) hooked up to one of the FXS
ports. When the other end hangs up, I get the
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 expeience / advice there was
Interesting. Something similar for me, except it comes back as busy
after about 30 seconds.
-- Called [EMAIL PROTECTED]/18004337300
-- Call accepted by 69.25.60.30 (format ulaw)
-- Format for call is ulaw
-- IAX2/voipjet-1 is making progress passing it to SIP/207-b8f3
--
I get the same problem @ home when I use it. I thought it was just me.
..o---o.
Brian Fertig
NOC/Network Engineer
Planet Telecom, Inc.
Tampa, FL Office
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
i have configured a sip phone to make calls through a
sip server but when i make call through the sip phone
to the sip server every thing goes well and the call
is done perfectly but on sip server it gives me these
messages(i have 2 pc with different ips one with a sip
phone and the another with
Jeremy Gault wrote:
We have a TDM400P installed here with four FXS modules. It works well
except for a couple of issues:
snip
Second, one of our ports has static on it. A tenant moved in and we
configured their fax for our fourth (until then unused) FXS port.
Confirm it is the
A user has their unavailable message played and once that message
is over the Comedian
message is played right after. Is there any way to prevent the
Comedian message being
played if the user's unavailable/busy message is being played.
Thanks,
Kurt
On Fri, 2005-08-12 at 08:26 +0200, Peter Svensson wrote:
Since Linux is not RTOS, why some folks are using this -p switch?
It has no effect on standard Linux box.
Linux is not a hard realtime os with guaranteed timing. What the -p
flag
does is to request the realtime scheduler. This
Give me a handset with a ringer and hook that both work under Linux and
I will buy 30 pieces on the spot!
No kidding..
RESPONSE: We are working on it and want to make sure we have the product
definition correct to best serve the Linux community. Please private
e-mail me to clarify if
I seem to remember (calling over isdn) that American Airlines doesn't
actually send back a Connect for quite awhile - there's just a
-Progress w/in-band info to cause voice cut-through. Or something
like that ;)
On 8/12/05, Brian C. Fertig [EMAIL PROTECTED] wrote:
I get the same problem @ home
We use Polycom 501s here and several users utilize the Forward
soft-button to forward their extension to another extension or outside
to a cell phone when they are out. My question is, how can I configure
the dial plan so that if they have forwarded their extension via the
phone, and the extension
Hello again,
I have a bunch of Polycom IP500 Phones with Boot 2.6.2
and SIP 1.4.1. I have defined seperate user and peer
settings for my extensions as per posts I have seen in
here. I can access voicemail...etc and the phone seem
work fine.
Question: when I do sip show registration there is
Hello All,
I just started to use asterisk with Digium card (4 fxo ports)
and I've met some problems ( I'm just new in asterisk so questions may
be stupid )
my environment:
Debian testing,
asterisk 1.0.9
zaptel-1.0.9
TDM04P
1) when asterisk receiving incoming call on TDM card all networking
Peter Svensson wrote:
I am not that familiar with T1, Are you sure the signalling between
the pbx and asterisk is set the same on both?
I have unearthed some documentation on the programming side of the legacy
PBX. I can set the following on the PBX for each line on the T1 card:
Line Type:
Geoff Manning wrote:
Peter Svensson wrote:
I am not that familiar with T1, Are you sure the signalling between
the pbx and asterisk is set the same on both?
I have unearthed some documentation on the programming side of the legacy
PBX. I can set the following on the PBX for each line on
Bruce Ferrell wrote:
You need to be looking at a lower level
Like hw/cabling errors?? If so, that's what I was afraid of for cost
reasons.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Geoff Manning wrote:
Bruce Ferrell wrote:
You need to be looking at a lower level
Like hw/cabling errors?? If so, that's what I was afraid of for cost
reasons.
Hardware, possible. Unlikely to be cabling. It's usually a timing setting.
___
Bruce Ferrell wrote:
You need to be looking at a lower level
Like hw/cabling errors?? If so, that's what I was afraid of for cost
reasons.
no - the stuff you found relates to configuring one 64k channel of the T1,
you need to find the settings to configure the overall t1.
- someone
Geoff Manning wrote:
Bruce Ferrell wrote:
You need to be looking at a lower level
Like hw/cabling errors?? If so, that's what I was afraid of for cost
reasons.
just out of curiousity - what are you paying for a T1 cable that you are
worried about cost ? you do realize any old ethernet
http://www.voip-info.org/tiki-index.php?page=Asterisk+CMD+voicemail
On Fri, 2005-08-12 at 11:37 -0400, kurt x wrote:
A user has their unavailable message played and once that message
is over the Comedian
message is played right after. Is there any way to prevent the
Comedian message being
Jon Pounder wrote:
Geoff Manning wrote:
Bruce Ferrell wrote:
You need to be looking at a lower level
Like hw/cabling errors?? If so, that's what I was afraid of for cost
reasons.
just out of curiousity - what are you paying for a T1 cable that you
are worried about cost ? you do
1 - 100 of 177 matches
Mail list logo
