Hello List,
I am an Asterisk newbie, and I got a question about Asterisk Background
command's option noanswer:
What is required from the user agent, such as a SIP phone, to be able to
hear the playback without Answer()?
I'm asking this because when I used X-Lite, I could hear the the audio
Oh so once the call transfer it just frees
the line on the phone and asterisk makes a direct connection with the transferred
party.
Thats cool. Will try that then.
From: Tim Connolly
[mailto:[EMAIL PROTECTED]
Sent: Sunday, May 08, 2005 3:18 PM
To: 'Asterisk
Users Mailing List
Well only the receptionist and higher
level authorities will have the cisco 7960. For the rest I am probably thinking
of a Sipura or Snom phones to keep costs down.
From: Gregory Wiktor -
ADCom Corp. [mailto:[EMAIL PROTECTED]
Sent: Sunday, May 08, 2005 4:25 PM
To: Asterisk
Mark Wormgoor wrote:
Hi,
Is it possible to set a variable for a context for all extensions? I
haven't been able to find it.
Try looking up the application SetVar:
demo*CLI show application SetVar
demo*CLI
-= Info about application 'SetVar' =-
[Synopsis]:
Set variable to value
[Description]:
hello
i need help on PSTN Calls via quintum gateway. i have
a simple problem when i am try to send INVITE to PSTN
quintum gw. it is replying me 183 session progress and
call duration is starting at this point. after this he
is sending ringing then 200 OK. Billseconds are
incorrect in this case.
On Sat, 2005-05-07 at 17:08 +1000, Sophus wrote:
Hi there, I'm experiencing an echo problem and dammed If I can sort it out.
A dmesg also shows my card as a E/F REV.
Zapata Telephony Interface Registered on major 196
Registered Tormenta2 PCI
ACPI: PCI interrupt :05:08.0[A] - GSI 3
Hello,
Does anybody get any data in the 'Extension' column of the 'zap show
channels' output? I'm at a loss as to where it would be getting
any information to populate this column. I have an E1 (10 channels) and
channel 10 only shows the number in Extension column. Furthermore, that
number is
OK, I create the directory, then it works!
I found the tif file in this dir. But I also found the tif file is not complete.
I sent a page with 30 lines text and only received 6 lines. sometimes I received 5 lines.
That is, the channel hangup before it completely receive the fax.
-- Goto
Typo damn.
Should be read as NOT configured as an extension
Hello,
Does anybody get any data in the 'Extension' column of the 'zap show
channels' output? I'm at a loss as to where it would be getting
any information to populate this column. I have an E1 (10 channels) and
channel 10 only
I need to accept an incoming call, make a series of outgoing calls, and
once I find someone willing to accept the call, bridge the original
incoming call to the outgoing call.
Using Dial from an AGI script isn't enough because once the Dial'ed
number connects, the call is immediately bridged
These phones are mentioned in the Sip 1.5 manuals, anyone know what
the differences are?
Where are you getting SIP 1.5 from?
When I log into the Polycom download area, all I can find is 1.4.1.
They must have pulled it back.maybe some issues, like 1.3.0
-Matt
the interesting fact is, it works. I dunno why. but it works :o
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von bladerunner
Gesendet: Freitag, 6. Mai 2005 15:56
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: Re:
Hi!
Then
I phone other phones with my Zyxel 2000W (WI-FI) it just hang up when I answer
the phone I am ringing.
It
works fineif Icall the 2000W from other
phones.
I
have tried many sip settings. I use this
now:
[205]
type=friend
username=205
secret=passwd205
callerid="Zyxel"
205
Why not just keep it simple use dial with Macro argument
and this std macro-screen
like this
http://lists.digium.com/pipermail/asterisk-users/2005-March/098257.html
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Hi.
I'm using Asterisk oh323.
My gatekeeper needs h323 ID and e164 ID for registration.
But I don't know how to send e164 ID.
I configure register section in my oh323.conf file like below.
[general]
snip
gatekeeper=xxx.xxx.xxx.xxx
[register]
alias=abc
alias=777
snip
And the gatekeeper doesn't
I have no idea how to configure this gateway. Anybody who has links or tips?
Thanks,
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Hi!
I just noticed that there is a new firmware release, for those that are
interested:
http://www.grandstream.com/BETATEST/
2 quick notes, a quick test seem to indicate iLBC is broken (didn't try
any troubleshooting).
The release notes have a couple of comments on iLBC, you
Hi!
Well only the receptionist and higher level authorities will have the
cisco 7960. For the rest I am probably thinking of a Sipura or Snom
phones to keep costs down.
I would suggest that you take a good look at the SNOM 360 instead of
Cisco.
Cheers, Philipp
Hi, all
I have application with asterisk. 1 sip user and 2 or more pstn user start
a conference with meetme.
SIP user(PC)===SER=*=GWPSTN 1
MEETME ^
|---PSTN 2
SIP user start
Good day all
This is what i got off the net about queues and agents
Transfers of calls that are answered out of a queue must be done using
Asterisk '#' transfers (enabled with the 't' option above). SIP
transfers result in the Agent remaining affiliated with the call until
its eventual
Dear Asterisk users,
On Mon, 18 Apr 2005, Franz Knipp wrote:
thanks for this information. I've contacted my customer adviser at
Siemens, he'll try to organize me this version.
I've got back the phones with a new SIP firmware. The following
version informations are shown on the configuration
Title: Re: Sangoma A102 cards testing FIXED
Ok, I have tested with almost all versions both in
2.3.2--*-stable and 2.3.3-*-beta - I am getting same messages:
May 9 10:55:26 WARNING[3961]:
chan_zap.c:1925 pri_find_dchan: No D-channels available! Using Primary on
channel anyway 16!
and same
I have 20+ asterisk servers and need to network them together so a phone on
any of the servers can call a phone on any other server without any trouble.
I can think of IAX trunks between every server. So every server will have an
IAX trunk to every server and then prefix bases routing in the
As a test, I had a 7960 and a polycom, and made direct
calls between the two. I transferred the begeezez out of them back and
forth, sending to an incoming queue, etc. Asterisk performed very
well. Even over a vpn cable connection
(polycomroutercablemodeminternetcablemodem
On Sun, 2005-05-08 at 23:11 -0700, Manjit Riat wrote:
Well only the receptionist and higher level authorities will have the
cisco 7960. For the rest I am probably thinking of a Sipura or Snom
phones to keep costs down.
I'd suggest staying with the same provider for all handsets. It will
save
On Mon, 2005-05-09 at 12:25 +0200, Vikram Rangnekar wrote:
I have 20+ asterisk servers and need to network them together so a phone on
any of the servers can call a phone on any other server without any trouble.
I can think of IAX trunks between every server. So every server will have an
IAX
why not try using extensions from a database on all servers. That way
regardkess of the * server, the destination phone is the same.
http://www.voip-info.org/wiki-Asterisk+RealTime
Vikram Rangnekar wrote:
I have 20+ asterisk servers and need to network them together so a phone on
any of the
Hi there
i installed the asterisk stat v2 but on the page i
see an option for generating PDF files but when i click on the link it opens in
a blank page, does anyone know how to fix this or do i need to install some
additional package ?
Thanks
Sander Crombeen
On May 9, 2005 06:55 am, Sam Njenga wrote:
why not try using extensions from a database on all servers. That way
regardkess of the * server, the destination phone is the same.
Realtime isn't necessary for this; I agree that a database would be handy but
just regenerate the text config files
Hi,
Yes, I tried that, but it didn't work. I've tried this:
[contextname]
RING = r2
_RING = r2
__RING = r2
but nothing worked...
I did find that if I didn't set
RING =
under globals, the variable would actually never get reset between calls!
Kind regards,
Mark
trixter
Hi,
The problem is, that once I use:
exten = _.,1,SetGlobalVar(RING=r2)
it will never get to the extension anymore. The following line is never
reached, because now all extensions need to be _.
[ext-local]
exten = 1,1,dial(Zap/1R{RING})
The only workaround I can find is to set the RING for
After buying some additional lines from my telco, I
recently had my phone vendor wire the additional lines
from my phone box into an amphenol connector that's
plugged into my channel bank (Adit 600).
However, although I make the following changes in my
zaptel.conf and my zapata.conf files
The following works well with a Sipura spa3k:
exten = 3022,1,SetVar(_ALERT_INFO=bellcore-r3) ; selects Ringer
exten = 3022,2,Dial(SIP/3022,25,r) ; Sipura spa-3000 fxs port
Hi,
Yes, I tried that, but it didn't work. I've tried this:
[contextname]
RING = r2
_RING
Hello,
Have you tried the wanpipe-beta8c-2.3.3.tgz release in the custom/2.3.3 dir
on their FTP site? Also, have you contacted Sangoma for support? They are
very responsive.
I am using wanpipe-beta8a-2.3.3.tgz and it's been working great on my A104
for a week now.
MATT---
-Original
Hello,
Have you contacted Sangoma about this? What version of wanpipe and what
version of zaptel/asterisk are you using?
MATT---
-Original Message-
From: Nguyen Trung Tin [mailto:[EMAIL PROTECTED]
Sent: Monday, May 09, 2005 1:25 AM
To: asterisk-users@lists.digium.com
Subject:
I have an ISDN line with 10 numbers.
The line is then connected to * with one HFC-based card.
The format of the numbers is like below:
123456-0
123456-1
...
123456-9
Now I would like to connect those numbers to different telephones, i.e.
when someone dials 123456-0, he/she is connected to the
How well does the sangoma cards work with fedora core 3
Im doing the research on what hardware/os I need to use
Please help and advice
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Sangoma are very helpful, contact their tech support for assistance with the
card. Once you know the card is talking to the E1, you can work on Asterisk.
Chris Mason
www.anguillaguide.com
Tel: (305) 704-7249 Fax: (815)301-9759
i'm purchased sangoma card A-101. i connect to E1 with MF/R2
Has anyone come across the Cisco ata 186 not passing *70? WhenI press *70, the ata just goes back to a dial tone. The strange thing is, its only *70 and not the reset of the 70's. *71, *72, etcall go through fine. I've tried removing thethe dialplan all together from the ata to try and let it
Is there a way to enable call waiting by default in asterisk? Every timeI create an extension, it is disabled by default. Having to go to every phone is becoming quite annoying. I havent restarted the server yet, butI am afraid of all my extensions changing back to disabled again. Making it the
As I said before, I can not get help from junghanns, so I ask the list.
I installed * version 1.0.7 bristuffed latest version and this solves the
music on hold
problem. But this introduces a new problem that I did not have before.
Every 1 second pops up the message:
May 9 13:31:54 coscosmia
Nguyen Trung Tin wrote:
Hello All !
i'm purchased sangoma card A-101. i connect to E1 with MF/R2
signalling. but card don't work. negotiation with E1 fail.
please help me to correct it. i dont' know some parameters such as:
MTU, BAUDRATE
Are you running the voice drivers, or have you configure
Hi again,
Well - I didn't see beta8a-2.3.3 in custom dir. Will try.
Also I tried to contact Sangoma - they are very fast to answer but main
problem is time difference - it's 6 hours between Canada and Europe.
Br,
dmitry
Dmitry Zhukovski
System developer
ComX Networks A/S
Naverland 31, 2
Hi all
I have been searching through the mailing list and google, nothing of which i can findthat relates to my problem.
Here goes;
i need to create some kind of database on my E1/PRI that store all my 100 DID number, and when * pick up any zap line. It will random pick any one of the unused
Hi,
Yes, I know I can set the RING variable for each extension, but I would
really like to set it based on the initial context for the call.
Now I need to check for every extension that gets called if the call
originated inside or outside and set the ring accordingly.
Kind regards,
Mark
Rich
Ya well let me know when u solved this
We have the same thing
Do you have any other cards in with it
We have a diguim fxs/fxo card in so maybe its a error with working
together
Anyway
Let me know when you get a fix for it because no one seems to know(or
check their /var/log/messages)
This lets my
Hi All
Does anyone know how to set up two agents on one Snom 220, acting as
agents in a queue. When incoming call go to queue if both the accounts
on the snom 220 are busy the call stays in the queue and if one is not busy
the call will ring through.
I find that even though the second account in
Hi,
Is it possible to set a variable for a context for all extensions? I
haven't been able to find it.
Try looking up the application SetVar:
demo*CLI show application SetVar
demo*CLI
-= Info about application 'SetVar' =-
[Synopsis]:
Set variable to value
[Description]:
It should work even without answer, try putting in a wait(2) instead
of the answer.
On 5/9/05, Mark morris [EMAIL PROTECTED] wrote:
Hello List,
I am an Asterisk newbie, and I got a question about Asterisk Background
command's option noanswer:
What is required from the user agent, such as
mån 2005-05-09 klockan 15.28 skrev Mark Wormgoor:
Hi,
Is it possible to set a variable for a context for all extensions? I
haven't been able to find it.
Try looking up the application SetVar:
... snip ...
[Description]:
Setvar(#n=value): Sets channel specific variable n to
On Mon, May 09, 2005 at 03:28:22PM +0200, Mark Wormgoor said:
Hi,
Is it possible to set a variable for a context for all extensions? I
haven't been able to find it.
Try looking up the application SetVar:
demo*CLI show application SetVar
demo*CLI
-= Info about application
In article [EMAIL PROTECTED],
Mark Wormgoor [EMAIL PROTECTED] wrote:
Hi,
Is it possible to set a variable for a context for all extensions? I
haven't been able to find it. I want something like this in
extensions.conf:
[from-iaxfwd]
exten = .,1,RING=r3
exten =
Even if you use sip_buddies, you still have a sip.conf.
Go into your asterisk/configs/sip.conf.sample and read up on cache.
-Matthew
Callum McGillivray wrote:
Hi Matthew,
We no longer use sip.conf, the config has been moved to the
sip_buddies table in MySQL.
What do the rtcache settings
On Fri, May 06, 2005 at 10:20:55AM -0400, Nathan Goodwin wrote:
John Todd wrote:
Downsides:
1) They want you to sign an NDA before they'll discuss the methods
with you.
Industry typical. They should be reciprocal. Both parties have to
announce anything confidential is going to be
Oops!
I wrote:
In article [EMAIL PROTECTED],
Mark Wormgoor [EMAIL PROTECTED] wrote:
Hi,
Is it possible to set a variable for a context for all extensions? I
haven't been able to find it. I want something like this in
extensions.conf:
[from-iaxfwd]
exten = .,1,RING=r3
Vikram Rangnekar wrote:
I have 20+ asterisk servers and need to network them together so a
phone on any of the servers can call a phone on any other server
without any trouble.
If you use RealTime for registration of UA's, then every machine can contact
every UA regardless of which macine the
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On 5/9/05, Charles Wang [EMAIL PROTECTED] wrote:
Hi, ALL:
I use asterisk -r and sip debug to debug my sip channel.
And I build my sip proxy(5060) and asterisk(5065) on the same machine.
I make a call from 1011 to on sip proxy,
sip proxy forwards this call to 0939749001.
And
Try the random command
http://www.voip-info.org/wiki-Asterisk+cmd+random
just create the 100 lines of code with the dids in there, then have
random choose it for you.
I have never worked with random, but I think this will help you.
On 5/9/05, Lee Lee [EMAIL PROTECTED] wrote:
Hi all
I have
Hi,
Is it possible to set a variable for a context for all extensions? I
haven't been able to find it.
Try looking up the application SetVar:
... snip ...
[Description]:
Setvar(#n=value): Sets channel specific variable n to value
But how do I link SetVar() to all extensions in
Really??
call waiting disalbed? as far as I know there isn't even a function in
asterisk to enable or disable call waiting. There are lots of
workarounds but no function. So you have gone the extra mile to create
such a workaround that (you set it up that way) by default disables
call waiting,
exten = 1234560,1,Dial(phonea)
exten = 1234561,1,Dial(phoneb)
and so on...
It's called DID
On 5/9/05, Tomasz Chmielewski [EMAIL PROTECTED] wrote:
I have an ISDN line with 10 numbers.
The line is then connected to * with one HFC-based card.
The format of the numbers is like
-Original Message-
From: Vikram Rangnekar [mailto:[EMAIL PROTECTED]
Sent: 09 May 2005 11:26
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Connecting 20+ asterisk servers together
I have 20+ asterisk servers and need to network them together
so a phone on any of
Hi,
Running an X client in asterisk box might be ok to display it in a
remote X server.
afaik, the no-go is for X server running locally in same box as Asterisk.
regards
Time Bandit wrote:
You can run an X client on the station and display it in a remote X server.
Running X on an Asterisk
I have set up a "community type server" for Caller
Name storage and lookup.
IT IS NOT SS7 BASED(not yet)
Currently it supports five switches. This is
a proof of concept. If you are
receiving calling number, but not name, and would
like to be involved in this
project, please e-mail me
C F wrote:
exten = 1234560,1,Dial(phonea)
exten = 1234561,1,Dial(phoneb)
and so on...
It's called DID
OK, I had that, but it didn't work.
What I missed:
1) I have a zapata device - HFC-based ISDN card, so I needed to make:
immediate=no
in /etc/asterisk/zapata.conf (instead of
Actually, this is whats facing me right now. I think Dundi will resolve the
problem, but I've never really placed it to the test. Anyone tested Dundi?
Best Regards,
==
David Choo
Sales Engineer
Business Technology Division
Engineered for Changing Businesses
Espore
Greetings,
Does anyone know if there is a cost effective way to interface an
older ATT Spirit system into Asterisk.
I'm only interested in A) being able to offer voicemail and B)
possibly an AAT to callers.
I've thought about just stringing the FXO cards into the line1/2 slots
that go into the
Which version of * and bristuff did you install, I had bristuff-0.2.0-RC8a
and now I am trying bristuff-0.2.0-RC8c
- Original Message -
From: Altus Snyman [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, May 09,
Gregory Wiktor - ADCom Corp. wrote:
For example, I usa a Panasonic dbs at one of
my offices, because the multi-line abilities work well with the people I
have there. If they had to learn how to transfer a call, forget it.
Then they will never work for a company with a real PBX (not a key
Any ideas?
--- Deon [EMAIL PROTECTED] wrote:
Anybody have any updates to this posting? Why won't it route calls from
209.237.227.185 to the [termination-test] context? I've tried
type=user,
type=peer and type=friend
The funny thing is in the sip.conf, I have an entry, type=peer and it
Vikram,
Instead of trying to be over-ambitious and try to connect 20 Asterisk
boxes together, why don't you try top connect three (3) of them together
first.
There may lie a plausible solution for you. If this is done, you may go
and string four of them together and so on and so forth.
Take the
In article [EMAIL PROTECTED],
Mark Wormgoor [EMAIL PROTECTED] wrote:
Not certain about this but couldn't this work:
exten = _.,1,SetVar()
exten = _.,2,Goto(${EXTEN},1)
exten = 1234,1,Dial()
That's what I wanted... Should have thought of that myself.
Thanks!
It won't work unless
Hi list,
I'm currently running a site with Asterisk CVS-HEAD-03/29/05-14:49:04 in
a small call center environment and it works great. 3 queues, 25
agents, all using Cisco 79XX phones.
Periodically we have a problem where asterisk grinds to a halt and just
stops working. The problem starts to
Dear All,
Here the version 2.2 a new version of your dear CallingCard Software !!!
http://www.areski.net/areskicc-doc-v2/
Many new features have been added and several enhancements made!
Newest features :
- A new re-build rate-engine
- LCR LCD management (HHH
On 5/2/05, Gary Stimson [EMAIL PROTECTED] wrote:
On Friday 29 April 2005 23:20, Paul Tyreman wrote:
What are you using instead of SIPGATE in the UK ?
I also have this problem with DTMF tones not being passed to Asterisk from
a PSTN line and my e-mails are being ignored too !
Both are
This is too funny. I have 2 different visa check card numbers connected
to the company bank account. Only one is used for any online purchasing.
I do this so that in the event of unauthorized or fraudulent
transactions, I only have to cancel one card and still have one to use
for gasoline and
On Mon, 2005-05-09 at 09:50 -0500, Tom Chandler wrote:
I have set up a community type server for Caller Name storage and
lookup.
IT IS NOT SS7 BASED(not yet)
So what is it based on? Where are you drawing your data from?
Currently it supports five switches. This is a proof of
Hey all -
Attached is quick perl script i whipped together this morning to dial from
Kontact (well, KAddressBook). You'll need Net::Telnet (perl -MCPAN -e
'install Net::Telnet') for it to work. You'll also need to make sure you've
got a valid user/secret in /etc/asterisk/manager.conf on your *
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Disgruntled
Asterisk Luser
Sent: Sunday, May 08, 2005 9:06 PM
To: Luki; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] RE: Asterisk at home with Broadvoice?
Don't worry about these subtle
AstriCon Europe 2005 will be held at the Auditorium Hotel Madrid in
Madrid, Spain in less than six weeks. This is the first ever
all-Asterisk conference to be held in Europe and we expect it to be a
blast. The first AstriCon, held this last fall in Atlanta, drew in
nearly 500 Asterisk users.
We have just posted
a review of the Sipura SPA-3000 with a complete setup guide for using the device
as a trunk to Asterisk. It is easy to setup and works as good as any Zaptel
setup and includes an ATA. It really is one cool device!
Outward dialing is a no brainer. VoipJet is the best outbound call
provider I have come across. Period.
It always works for me and the call quality is always very very good.
So far that seems to be the norm for them.
I am still working on getting my inward DIDs solidified so no opinion
there...
On Mon, 2005-05-09 at 14:04 +0200, Mark Wormgoor wrote:
Hi,
The problem is, that once I use:
exten = _.,1,SetGlobalVar(RING=r2)
it will never get to the extension anymore. The following line is never
reached, because now all extensions need to be _.
perhaps a goto(${EXTEN}) after that?
really??
On 5/9/05, Alberto [EMAIL PROTECTED] wrote:
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Yes,
This is the solution that I am using and it works every time.
You can dial a number, put in a pin and it makes calls.
I've never received a bill, the minutes are free.
I can't understand how these people make any money.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
When broadvoice first started screwing up in DECEMBER, I put a
charge-back through with Amex, without a single hitch. BV never
complained, they just cancelled my service and I got the cash back from
Amex. Don't ever forgot, if you're paying for services by Credit Card,
you generally have better
Matt wrote:
Greetings,
Does anyone know if there is a cost effective way to interface an
older ATT Spirit system into Asterisk.
I'm only interested in A) being able to offer voicemail and B)
possibly an AAT to callers.
I've thought about just stringing the FXO cards into the line1/2 slots
that go
I seem to be having the exact same issue with the cisco ata 188.
Not sure, but looking at the cisco manuals there are alot of options in
hex format one can add, though 0x should cover all.
On Mon, 9 May 2005, Christopher Kenna wrote:
Has anyone come across the Cisco ata 186 not
As a general rule of thumb it would be good to make the distinction
between 'Credit Card' and 'Debit Card' too.
If possible, never ever use a debit card for online purchases.
It taps directly into your account and removes REAL money.
Credit cards are 'virtual' money in that they are credit and
Can anyone here help me understand what
I missing with this setup. I want to use Asterisk as a feature server only,
speaking only SIP (no IAX), and use SER for registration to minimize
necessary bandwidth.
SIP-phone --SER -- *
-- PSTN Provider -- Regular-phoneRegular-phone
-- PSTN Provider
Right.. and the plan is to eventually phase the Spirit system out...
however, if I can convince the persons there this is the best thing to
do right away is another thing all togethor :) We'll see..
On 5/9/05, Richard Lyman [EMAIL PROTECTED] wrote:
Matt wrote:
Greetings,
Does anyone know
+++ Alex Barnes [09/05/05 15:26 +0100]:
But this doesn't sound particularly elegant specially once you start
trying to scale it.
If you do get any other ideas I would be interested to know so that I
can start this
structure out properly.
Your right using dialplan extensions for every
Hello List,
Thank you CF for your reply, but it still doesn't work for my ATA. Is it
safe to say it is the ATA's problem?
C F shmaltz at gmail.com wrote:
It should work even without answer, try putting in a wait(2) instead
of the answer.
On 5/9/05, Mark morris mmorris36 at hotmail.com wrote:
Hi List!
As a child did you ever receive a present that required batteries only to
find all the shops were shut? That's the way I feel at the moment with my
cisco 7960 IP phone. Recently purchased off ebay was looking forward to
connecting up to my asterisk pbx ( installed a few days ago ).
+++ Kanuri, Seshu (Company IT) [09/05/05 11:25 -0400]:
Vikram,
Instead of trying to be over-ambitious and try to connect 20 Asterisk
boxes together, why don't you try top connect three (3) of them together
first.
There may lie a plausible solution for you. If this is done, you may go
and
+++ Sam Njenga [09/05/05 13:55 +0300]:
why not try using extensions from a database on all servers. That way
regardkess of the * server, the destination phone is the same.
http://www.voip-info.org/wiki-Asterisk+RealTime
I think you might be on track but again I dont know how stable
+++ Vikram Rangnekar [09/05/05 12:25 +0200]:
I have 20+ asterisk servers and need to network them together so a phone on
any of the servers can call a phone on any other server without any trouble.
I can think of IAX trunks between every server. So every server will have an
IAX trunk to
Hi,
This is my Zyxel P2000W config :
[2000]
type=friend
username=myusername
secret=mypassword
host=dynamic
context=from-sip
(whatever)
mailbox=100
disallow=all
allow=g729
dtmfmode=rfc2833
... it works fine for incoming and outgoing calls
Regards,
Alexander
From: [EMAIL PROTECTED]
I think Cisco charges about $150 for the license and the load.
SW-SM-UL-7960 SIP and MGCP license for single 7960 IP phone D $150
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Monday, May 09, 2005 1:43 PM
To: asterisk-users@lists.digium.com
Subject:
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