I need to utilize about 5 or more Asterisk servers to load balance some
traffic.
I need a device/server/something to load balance the SIP traffic among all
the asterisk servers, I'm thinking SER. Asterisk servers will be handling
the routing/billing/cdr's/everything (albeit they're utilizing a
On Sat, Jul 01, 2006 at 01:29:30PM +0200, Paul Hewlett wrote:
Maybe on other distros the sound files are being read from an IDe disk and
the interrupts generated are distorting the sound - on astlinux the
soundfiles are in a memory filesystem - no interrupts - no distortion ?
Bigger
Rich Adamson wrote:
Rushowr wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Christopher Corn wrote:
thanks for the reply. why are residential lines cheaper than businesses?
say for unlimited, it always costs more for residential.
*/Michael Graves [EMAIL PROTECTED]/* wrote:
I'd
Hi all,I am new in the asterisk company. I need to set up a small voip system for about 60 phones ( a small enterprise organization). The system must support voip calls (calls inside the enterprise) but must be able to send calls over isdn (24 channels). Thus the asterisk server must
http://www.voip-info.org/wiki/ here got alots of example but you need to find it. You can start with http://www.trixbox.org/ that install everything. Good luck!
On 9/11/06, Panagiotis Zikos [EMAIL PROTECTED] wrote:
Hi all,I am new in the asterisk company. I need to set up a small voip system
Dear
I have two contexts how could I isolate context A from
context B ,in other words I want to ban context A from calling context B
Regards
*
No employee or agent is authorized to conclude any binding agreement on behalf of
Hi all, I have tried to install freepbx and a2billing application. Now see both application is not integrated special on cdr part. Any idea how to integrated it?Confuse!-- Regards,
Sharon Lim *Good memories are to be folded neatly and tucked away into the back pocket *
You could disable dialing altogether unless they
press hash - that way they would learn about the hash key feature pretty quickly
:-)
Unfortunately I dont see an easy solution since a
dialplan covering all possibilities may be too complicated.
Cheers
Gerry
- Original Message
I have two contexts how could I isolate context A from context B ,in
other words I want to ban context A from calling context B
In sip.conf, define phones/extensions something like this:
[1000]
type=friend
other parameters as needed
context=cust-a
[1001]
type=friend
other parameters as
Rich Adamson wrote:
That's strange; how many people just responded with that worked?
None that I've seen! If they did, then they started with a subversion
directory, or they were responding to a different situation.
I suggest you start with a clean tarball and try it yourself. Or look in
the
Hello ppl,
Wanted to know your experiences, if you've worked with Asterisk Realtime
Architecture.
Which one do you prefer, static or realtime?
I personaly think, the static architecture is a better solution, cuz, in
the realtime config, to check the dialplan(n hence the sql database) for
Hi,If you set Asterisk to ring several extensions for an incoming call, it appears that the call will be added in every phone's missed calls logs though the call was picked by one extension.In the long run, this prevent users from using missed calls features as these logs would filled with many
Hi,Our setup is:Asterisk 1.0.7 running on Debian 2.4.27-2-386TE110P cardISDN 30 (UK E1 PRI)When making outgoing calls to the PSTN using call files I get the following problems:1. No hangup detection - have to wait for time-out2. No pickup detection - the dial-plan starts as soon as the line
don't know for conf size limitation (but i guess it won't be a problem with a well-sized machine).About asking on the fly vs writing on change: if your routing information varies very often, on the fly should make more sense. Otherwise, it's not useful to retrieve continually same data: better to
You missed my point completely. His original post was a reply was
hijacking a very long thread (the digest thread) that he did not trim.
Just trying to teach some netiquette so he will get more help. I have
found that top vs bottom posting is not a major issue to most, but few
make a big
Sharon Lim wrote:
Hi all,
I have tried to install freepbx and a2billing application. Now see
both application is not integrated special on cdr part.
Any idea how to integrated it?Confuse!
--
Regards,
Sharon Lim
*Good memories are to be folded neatly and tucked away into the back
pocket *
Hello
I have a problem with callerid in the manager interface.
I think that Asterisk has a strange way to handle callerid, until I
found out to set the o-switch in the DIAL statement, it did not work
the way I wanted, it still doesnt, but now it works ok in one
direction.
My extension is 311,
I always got this warning after I'm using IAX2 channels .
Sep 11 21:44:09 WARNING[30229]: chan_iax2.c:6536 socket_read: Received trunked
frame before first full voice frame
What's it mean?___
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Hi,For a small call center, we would like to change default behaviour.Current setup and behaviour are :- a bristuffed 1.2.10 Asterisk server with 4 BRI ports- 5 SIP hardphones (Snom 320) with BLF for line or extension monitoring
- incoming calls ring all phones and light BLF on- when a call is
I believe the TE412P is intended to supersede the TE411P. What, if any,
are the advantages of the TE412P over the TE411P? Apart from the
disadvantage that the hardware DTMF detection doesn't work on the TE412P
(will that be fixed in the future?)
My UK supplier can supply both cards, but the
I guess I will give it a try. The numbers are pretty much static in the
way they are routed.
I just did not know if Asterisk would choke on a conf file with a couple
thousand lines.
Thanks,
Steve
picciuX wrote:
don't know for conf size limitation (but i guess it won't be a problem
with a
use:exten = s,2,Read(fwdnum|audiofile-to-play|10)then you'll have the number entered in variable ${fwdnum}.Type show application read in asterisk console for further details...
Hope this helps...2006/9/9, James Williams [EMAIL PROTECTED]:
I'm currently trying to write a section into my dialplan
FRANCISCO PEREZ-LANDAETA wrote:
Hi, this message is for Steve.
Sorry for replying to the digest. It wasn't my intention.
I would appreciate if you can guide as to how make the tenor asm200
work with asterisk. I am using asterisk at home. I guess my problem is
configuring the tenor so that it
Benjamin Jacob wrote:
Hello ppl,
Wanted to know your experiences, if you've worked with Asterisk Realtime
Architecture.
Which one do you prefer, static or realtime?
I personaly think, the static architecture is a better solution, cuz, in
the realtime config, to check the dialplan(n hence
Hi,
How do the 302 redirects work in asterisk, and what is the
promiscredir directive doing? I am not getting the documentation on this.
I have following happening on my asterisk box:
-- Executing Dial(mISDN/1-1, SIP/[EMAIL PROTECTED]||Tt) in
new
Hello,
Here in Spain there is a VoIP provider (Telefonica) that only works
if when you make an outgoing call, the SIP headers are like this:
INVITE sip:phonenumber@telefonica.net SIP/2.0
But Asterisk is sending this:
INVITE sip:phonenumber@sbc.ngn.rima-tde.net SIP/2.0
Think of BerkeleyDB as abarebones embedded RDBMS. It carries the much the same functionality (such as ACID) as most Relational DBs but without overhead (such as SQL translation) - for many years it formed (possibly still forms - not sure of present) the core of MySQL.Steve Totaro [EMAIL PROTECTED]
Can Asterisk bind on multiple ports?
I wish I could in my sip.conf make Asterisk bind different ports per
different context, so that calls coming in udp port 5060 would fall in
one context and calls coming in port 5061 fall in other different
context. Is that possible? How can I edit my
Hi All
I use Grandsteam GXP2000 phones.
Is there anyway within the dialplan/indications etc to have a custom
ringtone based on who is calling the phone.
i.e if i have a call from an internal user i get one ringtone if its an
external call i get a different ringtone??
Many Thanks in Advance
On 11 Sep 2006, at 10:29, Roy Gardner wrote:
Hi,
Our setup is:
Asterisk 1.0.7 running on Debian 2.4.27-2-386
TE110P card
ISDN 30 (UK E1 PRI)
When making outgoing calls to the PSTN using call files I get the
following problems:
1. No hangup detection - have to wait for time-out
2. No
hello
If I want to use asterisk to hookup to a SIP account
I just use the register line in sip.conf with the
extension number at the end...
But how about if I want to use a SIP trunk from a
provider which gives me 10 DID numbers with the same account?
thanx in advance
rick
On 09/11/06 18:36 Paco Brufal said the following:
Hello,
Here in Spain there is a VoIP provider (Telefonica) that only works
if when you make an outgoing call, the SIP headers are like this:
INVITE sip:phonenumber@telefonica.net SIP/2.0
But Asterisk is sending this:
INVITE
Hi all,Currently i've made some tests with VoIP buster and everything is running ok for outbound calls.Now i've created new VoIPbuster account, and my goal is to allow VoIPbuster partners to dial into my dialplan IVRs for free.
But I always get my VoIPbuster account (currently registred with my
Hi to all,
I read
http://www.voip-info.org/wiki/view/MS+LCS+2005+%252F+SER+%252F+Asterisk+Integration
Is it possible to use ser as a presence server instead
of LCS 2005 ?
Harry
___
Tony, the VPM450 is far better than the TE411P's VPM400. One main thing is that is has full 128ms tails on all spans whereas the VPM400 shared 128ms as you used more spans, and second is the Octasic chip makes the sound real crisp and clear.
For the moment, if you need FAX tone detection, you will
Rushowr wrote:
Benjamin Jacob wrote:
Hello ppl,
Wanted to know your experiences, if you've worked with Asterisk Realtime
Architecture.
Which one do you prefer, static or realtime?
I personaly think, the static architecture is a better solution, cuz, in
the realtime config, to check the
Hello ppl,
Any idea how do I write in include lines(for contexts or include files)
in the database, in the ARA static config?
thanks in advance.
Ben.
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To
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Benjamin Jacob wrote:
Rushowr wrote:
Benjamin Jacob wrote:
Hello ppl,
Wanted to know your experiences, if you've worked with Asterisk Realtime
Architecture.
Which one do you prefer, static or realtime?
I personaly think, the static
Hi There,
We've done this before. We just used TenorAX as a gatwaye for IP-PBX with
160 extensions. There is no big problem just minor tricks in the Tenor and
Asterisk configs. Just let me knopw what is your problem exactly.
Regards,
M. Shokuie Nia.
Richard Klingler wrote:
hello
If I want to use asterisk to hookup to a SIP account
I just use the register line in sip.conf with the
extension number at the end...
But how about if I want to use a SIP trunk from a
provider which gives me 10 DID numbers with the same account?
I'd
Thanks for all the help from Collin, I figured out the postgres database creation. I now have a different problem trying download files. I get this error. Maybe wget does not work in Fedora.
[EMAIL PROTECTED] a2billing]# wget http://www.areski.net/Open_A2Billing_version_Racoon.tar.gz--05:52:17--
I figured it out, I had old install manual.
-- Original message -- From: [EMAIL PROTECTED]
Thanks for all the help from Collin, I figured out the postgres database creation. I now have a different problem trying download files. I get this error. Maybe wget does not work
I have two machines connected with IAX presently.
I attempted to switch that to SIP.
Basically CUT the entries in iax.conf that apply
and pasted into sip.conf on both machines.
stopped asterisk, edited my extensions.conf
changed any IAX2 channels to SIP channels. restarted asterisk
and when I
I was successful in getting the tarball for a2billing
[EMAIL PROTECTED] a2billing]# ls -alltotal 4872drwxr-xr-x 2 root root 4096 Sep 11 06:22 .drwxr-xr-x 20 root root 4096 Sep 10 21:28 ..-rw-r--r-- 1 root root 165 Sep 11 06:16
And now for something completely different (o;
Is there a way out of a problem when registering
2 times with different account with same host?
I've setup 2 seperate peers using seperate
context in sip.conf...but as soon I change one
extension in one context it influences the other
as well and
[EMAIL PROTECTED] wrote:
I was successful in getting the tarball for a2billing
[EMAIL PROTECTED] a2billing]# ls -all
total 4872
drwxr-xr-x 2 root root4096 Sep 11 06:22 .
drwxr-xr-x 20 root root4096 Sep 10 21:28 ..
-rw-r--r-- 1 root root 165 Sep 11 06:16
Yes, I put the filename "Asterisk2Billing_release_Chameleon_v1_2_3.tar.gz" and got that error.
-- Original message -- From: "Jamin W. Collins" [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I was successful in getting the tarball for a2billing [EMAIL PROTECTED]
You're right. How did I miss that?
-- Original message -- From: [EMAIL PROTECTED]
Yes, I put the filename "Asterisk2Billing_release_Chameleon_v1_2_3.tar.gz" and got that error.
-- Original message -- From: "Jamin W. Collins" [EMAIL PROTECTED]
On 11/sep/2006, Dinesh Nair wrote:
try adding fromdomain=telefonica.net in the config for that peer.
I have it, but doesn't works... :? I have a sniffer and I see:
INVITE sip:telephonenumber@sbc.ngn.rima-tde.net SIP/2.0
but with the softphone that the telco distributes, I see:
Folks,
Anyone have any News about partner among Digium and Intel Dialogic for
the Asterisk Business Edition offers support for the Dialogic Boards?
Regards,
Cleviton Mendes de Araujo
Telecommunications Engineer
CAIXA ECONOMICA FEDERAL of Brazil
+55 11 3685-6641.
Everything was going well, I got the tarball, unpacked the tarballs, created the postgre user and password, database is created and checked ownership and even got a list of database users. I even imported the data schema into the new database. My problem now is verification of database
Hi,
I installed an Asterisk box with a sangoma A102 PRI card. Sometimes Asterisk
drops calls...there is nothing inside logs but these warnings:
Sep 11 15:00:18 WARNING[3503] chan_zap.c: Ring requested on channel 0/6 already
in use on span 1. Hanging up owner.
Sep 11 15:00:22 WARNING[3503]
I have the same problem and so do many others. I filed a bug report but it
was cancelled because the developers did not feel it was a problem with
Asterisk. Perhaps they just don't want to support native sounds. If they
did it would technically be considered a problem with Asterisk. So
Hi,I'm trying to connect my cell phone (motorola V3) to asterisk, using this guide:
http://www.thetechguide.com/howto/asterisk/chanbluetooth.html
Everything has worked ok, but when I actually want to start
asterisk, my phone doesn't connect all the way. All I'm getting in the
asterisk CLI is
Hi,
I want to set chan_h323. If you think this is not the best then please tell
me the setup information of the best one.
Thanks for you reply.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Saturday, September 09, 2006 7:40 AM
To: [EMAIL PROTECTED];
I have some problems with this option.Could you help me?Thanks
___
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I have tested Grandstream Budgetone 102 and Grandstream Budgetone
200 and with both, if they are called from a caller that is an
alphanumeric user, their display shows a unintelligible name impossible
to figure out who is calling!! If the caller is a numeric one, in both
phones their display
Tzafrir Cohen wrote:
On Sat, Jul 01, 2006 at 01:29:30PM +0200, Paul Hewlett wrote:
Maybe on other distros the sound files are being read from an IDe disk and
the interrupts generated are distorting the sound - on astlinux the
soundfiles are in a memory filesystem - no interrupts - no
They only do numeric callerid.On 9/11/06, Ricardo Carvalho [EMAIL PROTECTED] wrote:
I have tested Grandstream Budgetone 102 and Grandstream Budgetone200 and with both, if they are called from a caller that is analphanumeric user, their display shows a unintelligible name impossible
to figure out
Ricardo Carvalho wrote:
I've updated their firmwares to the latest ones and that problem
persists...
Does anybody also experienced this?
These phones aren't capable of alphanumeric entries, only numeric.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase
Thanks Tom,
That's too bad... right now that I was thinking about buying them to a
mass deployment environment...
Regards,
Ricardo.
Tom Vile wrote:
They only do numeric callerid.
On 9/11/06, *Ricardo Carvalho* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
I have tested
You are seeing the difference between a resale product and a wholesale
product.
Origination and termination and telecom terms used to describe which way the
call is going. Time costs - no matter what. Even if the provider pays a
flat rate for their PRIs, the capacity multiplied by the number
There is a different mpg123 that is included with asterisk. It seems to work
a lot better than the other version that gets installed from a port or
package. I'm not sure why, but try removing your existing one, and run make
mpg123 install from the unpacked directory.
On September 10, 2006
Any experience using a Dlink DVC-2000 VideoPhone (H.323) with Asterisk?
Any references to this that I can read. Google was not my friend, in this
case.
___
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asterisk-users mailing list
To
vitel isn't bad. They have a nice asterisk-friendly interface, and their
rates are good. Quality has been fine.
-Tim
On September 10, 2006 17:51, Christopher Corn wrote:
ok maybe thats asking for too much. how about a voip provider that provides
729 codec support ? :)
Christopher Corn
i found this out also, after purchsing the phones.Tom Vile [EMAIL PROTECTED] wrote: They only do numeric callerid. On 9/11/06, Ricardo Carvalho [EMAIL PROTECTED] wrote: I have tested "Grandstream Budgetone 102" and "Grandstream Budgetone200" and with both, if they are called from a caller that
I have an extension.conf composite (including all the included files) that is
over 2000 lines. I do all my rating in the dialplan and it seems to work
just fine. I produced these from spreadsheets containing cost vs. number
information for overseas calls, so it has to pattern match every call
They do this because business customers tend to use more minutes than
residential customers, and an unlimited plan is always an ESTIMATION of
usage. It costs them for every minute you use, so they try to sell
residential customers a block of time, and call it unlimited, which it really
isn't
You could probably build one easily.
Create a context in extensions.conf that answers the call, asks them the time,
saves the digits as a variable, then passes these variables as arguments to a
shell script using the System() application.
Create a shell script that takes it's arguments and
That is the default behavior. If you don't include the contexts into each
other, they can't call each other.
On September 11, 2006 03:35, Khaled Chehab wrote:
Dear
I have two contexts how could I isolate context A from context B ,in other
words I want to ban context A from calling
I guess this functionality will be in the future added to new firmware
releases don't you people think so?
Ricardo.
Doug Lytle wrote:
These phones aren't capable of alphanumeric entries, only numeric.
Doug
Tom Vile wrote:
They only do numeric callerid.
This isn't really possible at the asterisk level.
The phones log missed calls as calls that ring, but are not answered.
It's not possible to have a call ring, and not be answered, but still ring at
the phone.
On September 11, 2006 05:07, Olivier wrote:
Hi,
If you set Asterisk to ring
Make a context called DID or something like that, and set your peer entry in
sip.conf to have your provider's calls go tho this context. The incoming SIP
invites will be directed to the DID [EMAIL PROTECTED] server.
Use Goto to direct the calls where you want them to end up.
ie.
[DID]
exten =
I set up this but the musiconhold does not work.
I have no idea why, but it works for static queues. Maybe a problem
with odbc and postgres?
Here goes my queues.conf:
[general]
;
; Global settings for call queues
;
; Persistent Members
;Store each dynamic agent in each queue in the astdb
I am sure I am missing something obvious here, but I have searched this list,
and googled around with no real luck.
I have quite a few SIP and IAX2 connections working like a champ into my
Asterisk setup, and saw I could setup a free inbound line from TRXtel, so
figured what the heck.
As I
Thanks for your tips, I already made it work, but your additional
input gives me a broader understanding of how Asterisk works... and
hey, chill out, I thought I used a descriptive subject (you know,
instead of just Please help), but I'll try harder next time.
Thanks again.
On 9/9/06, Tzafrir
I have two boxes on the net that support local phones
in two offices.
I am not using any VOIP providers. Just local TDM04B cards then IAX
between offices.
I experience between offices drop outs, half way conversations things
like that.
Is that normal for asterisk to asterisk?
I have two 3
Ricardo,From what I know its a physical limitation of the display Grandstream chose on that phone, Grandstream recommends purchasing the GXP-2000 phone instead if you're looking for this feature.Grandstream has no plans from what I am aware of of making this change to the BudgetTone series
When are Digium going to upload a corrected 1.2.9 zaptel tarball that
compiles?
I know it's correct in svn, but the public ftp servers still hold the
incorrect version.
Steve
--
NetTek Ltd UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Jerry Geis wrote:
I have two boxes on the net that support local phones in two offices.
I am not using any VOIP providers. Just local TDM04B cards then IAX
between offices.
I experience between offices drop outs, half way conversations things
like that.
Is that normal for asterisk to
Hi Group:
I have 3 asterisk boxes in different
countries which are interconnected using IAX2 trunks. The outbound routing makes
the link to the various 4 digit extensions transparent.
Users would like to be able to dial into
their local box viaa PSTN connection (landline or cell) and
There was activity in late 2005 concerning PRI channel lockups. The
telco sends a call to channel n, but Asterisk thinks channel n is
busy and rejects the call. There was an entry in the bug tracker and
chatter on the list.
Has this problem been resolved? I have two accounts experiencing
Hi Friends,I have Fritz ISDN2 card and want to configure with Trixbox and Asterisk. I tried to findout for tutorials and installation procedures to install this card. But, I am unable to find. Can anybody please give me a good link or tutorial to install this? Looking forwrad to your response.
Have nobody any idea or tipps for me ?
- Original Message -
From:
Dominik Weber
To: asterisk-users@lists.digium.com
Sent: Saturday, September 09, 2006 8:34
AM
Subject: [asterisk-users] Receive Fax
with rxfax on asterisk with debian
Hello,
my name
Hi at all,
I have make a IAX2 trunk over openvpn between [EMAIL PROTECTED] and trixbox.
[EMAIL PROTECTED] have extension 200 to 299 and Trixbox 300 to 399
In my 2 box I set in outbound routing that if I call 7|XXX I want to use
the IAX trunk.
The call from Trixbox (ext 301) to [EMAIL PROTECTED]
Hi,
Does anyone know how to do a re-map of a key on the Polycom
to make it dial a number.
I know how to remap a key to a certain function, but I dont
know how to make it dial a number.
Im wanting to re-map the Service key to
dial *8 for a group pickup.
Any help is greatly
Disa is what you want -- just make sure the context is the same as a
local extension so they can dial anything a local extension can.
on Monday 09/11/2006 Tony Di Bona([EMAIL PROTECTED]) wrote
Hi Group:
I have 3 asterisk boxes in different countries which are interconnected
using IAX2
On Mon, 2006-09-11 at 11:41 -0700, Crazy Boy wrote:
Hi Friends,
I have Fritz ISDN2 card and want to configure with Trixbox and
Asterisk. I tried to findout for tutorials and installation procedures
to install this card. But, I am unable to find. Can anybody please
give me a good link or
Hi list!
I was using bristuff-0.3.0-PRE-1s with florz patch but where normally the
TEI check request message were I was getting errors.
Concerned about that I switched to plain vanilla bristuff.
Now everything *seems* to be working without errors but I regulary get
reports from people trying
Hi All,
I have a Cable Systems ICS-G302 but cannot seem to find any info on
this unit. The company is none responsive to my request for a
user/admin guide and I cannot find the CD lying around.
If you have a user/admin guide for this unit, please forward it to me.
Thanks in advance.
JR
--
I just ran into this situation 15 mins ago
and I installed NvFaxDetect and it works great so farI tested it out
with a few one page and a couple of multi page faxes and all worked.
http://www.voip-info.org/wiki-NVFaxDetect
Bill
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Hello everyone,
I am trying to build zaptel 1.2.9 for AstLinux. I have already done an
svn export of the 1.2.9 tag, so I am not experiencing the missing
octastic issue.
However, I am having a funny problem. The zaptel.log that I have
attached tells the full story, but I'll give you a
positive? negative? yay? nay?___
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Anyone could help to use Static RealTime with SIP.CONF. I use Dynamic Realtime successfully. In fact, I want to know how to compos the correct DB(postgres or mysql) fields (I think
STATIC configuration is different from DYNAMIC).
Regards,
Hugo
___
Has anyone got a clue about this?I need to know which operator to
send a message to, prior to the queue command ringing him, (just after
he is assigned)
Anyone knows if I can get to know the operator ACD choosed to send the
call by using Realtime Queue, or maybe via the manager?
On 14:55, Tue 12 Sep 06, MF wrote:
Has anyone got a clue about this?I need to know which operator to
send a message to, prior to the queue command ringing him, (just after
he is assigned)
Anyone knows if I can get to know the operator ACD choosed to send the
call by using Realtime
On Mon, 2006-09-11 at 11:41 -0700, Crazy Boy wrote:
Hi Friends,
I have Fritz ISDN2 card and want to configure with Trixbox and
Asterisk. I tried to findout for tutorials and installation procedures
to install this card. But, I am unable to find. Can anybody please
give me a good link or
Hugo wrote:
Anyone could help to use Static RealTime with SIP.CONF. I use Dynamic
Realtime successfully. In fact, I want to know how to compos the correct
DB(postgres or mysql) fields (I think STATIC configuration is different
from DYNAMIC).
Regards,
Hugo
Hello,
Is it possible to configure asterisk in order to send
domains not handled or allowed to a specific context ?
Harry
___
Découvrez un nouveau moyen de poser toutes vos
Hoping someone can point me in the right direction. I
have the following setup:
Trixbox latest (asterisk 1.2.11)
DID thru IAX trunk (EXGN/Vitelity)
Termination thru IAX trunk (EXGN/Vitelity)
3 GXP-2000 phones with firmware (1.1.0.16)
6 Digium g729 licenses
I have no codecs specified per trunk or
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