On Sat, Apr 22, 2006 at 11:59:21AM -0400, Alexander Lopez wrote:
Can't anyone stop self-promotion and tell the poor guy what he needs.
A T1/E1 X-over cable using an RJ-45 (8-cond.) is pinned out as follows:
1 - 4
2 - 5
3 - NU
4 - 1
5 - 2
6 - NU
7 - NU
8 - NU
NU = Not Used
I
AM I think that you are speaking about the sound-addon right?
AM Did you use the make install option on the sounds?
AM Alex
Yup - it appears to populate all foders with sounds
except /digits and /priv-caller-intros.
Evidently, soemthing else is creating the /digits folder
because I don't see it
On Sat, 22 Apr 2006, [EMAIL PROTECTED] wrote:
1. Add the IP's into your sip.conf and set qualify=yes.
Ick, that's quite horrible :)
And rather defeats the point of DNS - I'd need to know if any of the IPs
were changed.
I'm wondering if there are any bad side effects to hacking
Hello to all,
Is there an how-to for asterisk and setting up a t38
fax gateway (SIP) ?
I look at http://bugs.digium.com/view.php?id=5090 to
patch asterisk chan_sip.c file.
What are the next steps to get a t38 fax gateway with
asterisk ?
Regards
Harry
PS:
I use hylafax server.
Hello All,
I always used a compiled version for a x86 system
From http://kvin.lv/pub/Linux/Asterisk/ , and
Works fine.
At this time , I'm using a Pentium IV Dual core,
Running Centos 4.3. I tried to install the 64 bits
Compiled version but has a translation time 20ms.
Is it correct ?
I also
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 not possible with
the CAPI driver, but found no clear answer about the mISDN driver.
Thanks for your
Get a licensed codec.
Jefferson Carvalho wrote:
Hello All,
I always used a compiled version for a x86 system
From http://kvin.lv/pub/Linux/Asterisk/ , and
Works fine.
At this time , I'm using a Pentium IV Dual core,
Running Centos 4.3. I tried to install the 64 bits
Compiled version but has a
Hello,
I read
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SIPredirect
I wish to configure asterisk as a redirect server.
I have badly understood this command .
ASTERISK
|
sip agents nated ==SER
When sip agents send INVITE to the
I can't seem to see any documentation on how to access Asterisk functions
(rather than applications) using AGI?
--
- Steve
xmpp:[EMAIL PROTECTED] sip:[EMAIL PROTECTED] http://www.nexusuk.org/
Servatis a periculum, servatis a maleficum - Whisper, Evanescence
While re-setting up my Asterisk system I decided that implementing call
routing through ENUM lookups would be a good idea. However, I've hit a
bit of a problem. Given DNS records like:
*.4.4 IN NAPTR blah
9.8.7.6.5.4.3.2.1.4.4 IN NAPTR blah
The first
2006/4/21, Armin Schindler [EMAIL PROTECTED]:
But if you want to forward a call (which was already accepted by Asterisk)to another CAPI application, it is not possible. (Well, Eicon has a specialdriver which can do a lot of CAPI extensions, but I did not try this yet).
So if you want to do that, I
What do you mean specifically?
-Original Message-
From: Steve Hill [mailto:[EMAIL PROTECTED]
Sent: Sun 4/23/2006 6:58 AM
To: asterisk-users@lists.digium.com
Cc:
Subject: [Asterisk-Users] Accessing functions from AGI
Hi
On Apr 22, 2006, at 9:02 PM, Leo Ann Boon wrote:
An example:
- assuming a hunting pstn number 2341000
- 4 lines in a the group: 2341001, 2341002, 2341003, 2341004
- The 4 lines, are connected to your TDM card as Zap/1, Zap/2, Zap/
3, Zap/4
If you want to find out which line was actually
On Sun, 23 Apr 2006, Olivier Krief wrote:
2006/4/21, Armin Schindler [EMAIL PROTECTED]:
But if you want to forward a call (which was already accepted by Asterisk)
to another CAPI application, it is not possible. (Well, Eicon has a
special
driver which can do a lot of CAPI extensions,
On Sat, 22 Apr 2006, Klaus Darilion wrote:
On Fri, April 21, 2006 15:19, Armin Schindler said:
Hi Klaus,
Thanks for the detailed answers and isdn for Linux basics. I will take
the opportunity to ask some more questions :-)
On Fri, April 21, 2006 12:24, Armin Schindler said:
On
Hi Louis-David,
Check without crc4
Best Regards,
Francois BERGERET.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Louis-David
Mitterrand
Envoyé : dimanche 23 avril 2006 09:39
À : asterisk-users@lists.digium.com
Objet : [Asterisk-Users] TE410P card
Hello,
The pinout is like 1,2,4,5 you need to use LED and check the signal, connect the telco send with your receive, your receive wth the telco send.
you need to be sure about some parametres, the line code (HDB3 or other, the CRC4 is on or off, the impedance 75 or 120 if you are using CAT5 you
On Sun, 23 Apr 2006, Steve Totaro wrote:
What do you mean specifically?
I've actually just worked it out - I was trying to access the ENUMLOOKUP
function. It seems I can just execute it by treating it as if it were a
variable.
--
- Steve
xmpp:[EMAIL PROTECTED] sip:[EMAIL
On Apr 22, 2006, at 9:02 PM, Leo Ann Boon wrote:
Jim,
You might want to be a little more specific:
a. You want to find out which line the call came in on, OR
b. The actual PSTN number that was dialed
As I think about this more, and considering the solution you provide,
I think there is still
When I had enabled crc4 and it was not supported by the carrier, my LED
flashed from green to red very quickly.
[EMAIL PROTECTED] wrote:
Hi Louis-David,
Check without crc4
Best Regards,
Francois BERGERET.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la
Hi
I have can get my phones to register with SER and dialout for PSTN via
my Asterisk box over a SIP channel to my VoIP provider. If the phone
requests hangup then the bridged channel on Asterisk gets destroyed
however if the called party hangups the channel stays up and the phone
connected.
Set up an IVR and prompt the callers to dial the extension they wish to reach.
On 4/23/06, Jim Freeze [EMAIL PROTECTED] wrote:
On Apr 22, 2006, at 9:02 PM, Leo Ann Boon wrote:
Jim,
You might want to be a little more specific:
a. You want to find out which line the call came in on, OR
Have you thought about making them agents, they would both be reachable by
dialing there agent number then, and I know only one agent can be logged in at
once. Just a thought.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent:
You will not be able to determine what number was DIALED unless you have
DID service from you phone company. CF's suggestion is your best bet,
unless you move over to DID service.
Alex
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of C
Jefferson Carvalho wrote:
I always used a compiled version for a x86 system
From [...]
Someone could help me on this?
Yes, the folks at Digium will be more than happy to help you.
Visit http://www.digium.com/en/products/voice/g729codec.php and get a
licensed codec.
Please:
1 Follow the suggestions that are sent out on EVERY Digest and
edit your subject line.
2 Trim your posts to only include the relevant information, the
list is quite large and brevity is a plus as smaller message distribute
faster than larger ones.
3 Make sure your
Hello
I
have an E1 connection with 32 channels.
To
access to our IVR, you use 8898 phone number or
8899.
If
you call the 8898 you listen to the first
application
If
you call the 8899 you listen to the second
application
How
can I configure asterisk to do this
Thank you
issam
Read the wiki
issam wrote:
Hello
I have an E1 connection with 32 channels.
To access to our IVR, you use 8898 phone number or 8899.
If you call the 8898 you listen to the first application
If you call the 8899 you listen to the second application
How can I configure asterisk to do this
I still have the problem, im using DTMF not pulses, and the problem isnt happening with the tdm400, does anyone has any clue??Thank you
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I can't seem to register properly with broadvoice servers. Looking at
tcpdump and log files I see registrations attemtps and traffic to
broadvoice, but no traffic or error messages of any kind from broadvoice.
Do my rules look ok ?
ACCEPT all -- anywhere anywhere
ACCEPT all
issam napisał(a):
Hello
I have an E1 connection with 32 channels.
I've never seen telco that gives more than 30 B channels per E1 connection.
Just a clue: some time slots are used for sygnalization, you really want
to read some wiki and tel-co faq of any kind.
kd,
--
Krzysztof Drewicz
Krzysztof Drewicz wrote:
issam napisał(a):
Hello
I have an E1 connection with 32 channels.
I've never seen telco that gives more than 30 B channels per E1
connection.
Just a clue: some time slots are used for sygnalization, you really
want to read some wiki and tel-co faq of any kind.
Hi everyoneI am having problems with my call queueWe currently run a customer care call center which has attendants login during the daytime. Customers who call the 'customer care line (a specific number) always get routed to the cutomer care queue (called 124). After hours, staffs of the Network
that there is no longer a T38
CFLAG in the Makefile. T38 support is compiled by default. Nevertheless, you can
configure it on a per peer/user basis.
Untarring will create asterisk-1.2.7.1-t38-20060423/. Copy the files therein to
your out-of-the-box asterisk-1.2.7.1; don't use a checkout with T38
I have encountered the following problem with the latest Asterisk source
(as of 4/23/2006):
Someone calls me on my PSTN line, it then dials my analog extension (I
have both SIP and analog phones where all analog phones are a shared
extension.) After a while, I get a busy signal. How can I
Jim Freeze wrote:
I have a TDM card with 4 lines on a hunt group coming in.
The latest version of FreePBX (2.1 Beta 1 - currently in SVN, but should
be released soon, I'm told) allows you to create inbound routes based on
Zap Channel, which I believe is what you're look for.
You may want
Thanks for the suggestion ,
But I post a message to get a FREE codec (OPEN) , and not a PURCHASED.
If I was interested in get a licensed one , believe that I never had
Post a message on this list.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Jim Freeze wrote:
snip
The published number that everyone calls is 234-1000.
I want all calls to that number to be sent to the front desk.
That means, that even if the 1000 line is busy, and the next
open line found is 567-1002, I still wan the call to go to the
front desk.
But, if someone
Louis-David Mitterrand wrote:snip
Should I use a T1 cross cable to connect the telco's socket to the
TE410P card?
When I tried straight cat5 cables, both leds remained red at each end.
However this E1 socket works fine with the Matra PBX, so it must be a
cable problem or TE410P
I hav been trying to debug this for a while and I am drawing a blank. I
have an Asterisk PBX with a Adtran 750 and a Digium T1 card.
Sip to Sip calls work great.
Incoming PSTN calls through the CB have one way to audio when sent to a
Sip phone on the local network. What could cause that?
Are using g.729, by any chance?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Chris Mason (Lists)
Sent: Sunday, April 23, 2006 8:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Zap -
Hi,
What I would suggest doing, since we have a similar setup (where our 24
support contracts can enter a pin number to be routed to an on call
tech) is I use the function QUEUEAGENTCOUNT(queue name). Since you said
that the calls should only be routed after the last support person logs
out,
When I set asterisk to to sequence the lines as Ground Start the system is
not starting. It is giving the following error Invalid Argument 22
Do you have any ideas about this.
Any help or assistance appreciated.
Thanks!!
___
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Hi All I want todo features as belows. user --- call ( from telco) -- asterisk --- IVR -- SIP 1. after that, SIP1 transfer to SIP2 (unattendant or attendant transfer). i want to SIP1 hear stream sound data of call conversation between user and SIP 2 (don't used call conference) SIP3 want to hear
Hi,
I need help getting my SPA3000 to work with the line settings in
Singapore. Has anyone gotten it to work in Singapore?
Regards,
KokMeng Loh
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Robert,
I have the same problem, and I discover that when you use de flash or hang up you need a time toasterisk detect that you not do a flash function. A suggest is put de little time umflash.
[]'
Leonardo Silva
2006/4/23, Robert La Ferla [EMAIL PROTECTED]:
I have encountered the following
Does anyone on-list use Clearpath for termination or DIDs? I had an 800 number
from them until late last week when their server began rejecting my
registration attempts. thus far they have not
respondd to my email inquiry.
Michael Graves
[EMAIL PROTECTED]
is there any way to make the Flash Operator Panel show which agents are
logged in in a specific queue? (both static and dynamic agents)
I've played around with the queue / queue agents settings from the Flash
Panel documentation (http://www.asternic.org). The way it is described
there, I
Davi-Ann wrote:
When I set asterisk to to sequence the lines as Ground Start the system
is not starting. It is giving the following error Invalid Argument 22
Do you have any ideas about this.
Any help or assistance appreciated.
I don't think Digium's analog cards support Ground Start..
--
On Mon, Apr 24, 2006 at 07:21:18AM +0800, Leo Ann Boon wrote:
Louis-David Mitterrand wrote:snip
Should I use a T1 cross cable to connect the telco's socket to the
TE410P card?
When I tried straight cat5 cables, both leds remained red at each end.
However this E1 socket works fine with
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