Hi!
after doing this i have some phones on different subnet's ie 255.255.255.248
or .192 or .252 and i am now unable to login to these phones from different
subnet's . I have one at home which is on a .248 ( Using an external IP for
the phone ) i can access this from my home network ( the
Hi!
It seems that if Asterisk recognises the DTMF digits, it will intercept them
and not send them to the ISDN card (either that, or the ISDN card isn't
regenerating them).
My guess: isdn4linux is the culprit.
Cheers, Philipp
___
Asterisk-Users
Hi!
Now it is pretty obvious that my setup is ok since it work half the
times.
Maybe, maybe when the IPs on dynamic servers change, * has different
information internally hence the transfer fails?
My feeling is that you have a firewall/NAT issue. Look at the qualify=
paramter and do some
Hi!
I can't seem to find the link to examples of asterisk installations for
different sized sites. I'm not after specific configuration of the conf
files, just an overview on what hardware/chassis cards people are
running and what channels - phones etc people are using.
Go to the WiKi
Hi!
I've just installed asterisk a few days ago and figured most of it
out, but I can't seem to forward calls. In the setup before I used
asterisk, when people called my (ISDN) number and I didn't answer the
phone in 4 rings, the call would be transferred to my cellphone. I
ofcourse made
Hi!
Is it possible to receiving incoming calls via IAX2 with a dynamic IP
address for the server?
For efficience reasons you could formulate your question shorter:
Is it possible?
;-
Cheers, Philipp
___
Asterisk-Users mailing list
[EMAIL
Hi!
Notice how Dial(IAX2/[EMAIL PROTECTED]) is being called, but the last line says
that IAX2[bendummy]/3 is ringing? Why is that? There is no call to bendummy
at that time!
I think it's just a messaging bug, as the other * server is authenticating
properly. Also iax2 show channels
Hi!
distorted, like it's over-driven. I'm running gentoo, and I emerged
mpg123 version 0.59s-r3
Try version 0.59r and see if that solves the problem.
office. But when I try to get the vm from the remote office, I can't
authenticate.
Check and maybe change your dtmfmode= settings and
Hi!
for subsequent reference. Another benefit with SMS is remote alarm handling
which is easier to guarantee because the messages are sent via a store and
forward system
BUT: There is no guarantee that a SMS message reaches the destination;
that's how the SMS network is designed.
Cheers,
Hi!
BUT: There is no guarantee that a SMS message reaches the destination;
that's how the SMS network is designed.
There's no guarantee that it will reach a recipient, however there's a
guarantee you'll know whether it got there or not and why.
Hm are you sure? User A is in Britain with
Internet telephony (VoIP): Regulators and industry debate 'irreversible'
trend
The Commission is weighing up its options on how to regulate internet
telephony. Major telecoms operators are already proposing services to
avoid being squeezed out of the market.
More at:
Hi!
is it just me, or are the VoIP providers for Germany more expensive than
going via call-by-call?
What about other countries? Same thing?!
The thing is that only in Germany Call-By-Call providers have *that* low
prices. Anywhere else in Europe - at least in the small and middle-sized
Hi!
Did you try out the new ring tones? One of them contains a regular ring,
followed by a voice announcing the caller id of the calling party. VERY
neat.
Hehe - you're right! And is my impression correct that Asterisk users
know that particular voice very well that is announcing You have
Hi!
Some fairly serious flaws in either asterisk or the GS iLBC code as
well. If I call BT to BT using iLBC with asterisk playing proxy etc,
often one end can't hear the other, although it sounds fine on the end
that can hear.
Interesting - I hadn't yet figured out that this was related
Hi!
Did you try out the new ring tones? One of them contains a regular ring,
followed by a voice announcing the caller id of the calling party. VERY
neat. It seems the ring tones can contain not only sound, but also
either code to be executed, or a flag to announce the caller id.
?? Which
Hi!
Our desktop phones were done as a package deal from the building owner
(who also runs the existing PBX) for almost nothing.
Then one option is to check if you can keep the PBX and the phones, and
just put Asterisk in between this PBX and the Telco. Compensate the costs
of the Asterisk
Hi!
BTW: And are you sure people wouldn't like to have voicemail? You'll need
to make them want that... ;- I guess you can even argue that voicemail
increases productivity.
Since we share phones (at least the developers/non customer facing
people) voicemail wouldn't work too well
Hi!
- 5 x TE410P/TE405P quad T1 (try to not sharing IRQs on them)
It appears to be good practice to not have more than two of those cards
in one machine. That makes it 3 servers - putting all those cards into a
single box is most probably not the way to go.
- 20 x TA750 CB (60 FXO and 420
Hi!
How do you search? I don't see a search interface to the mailing list
archives. Did I miss something?
You are correct, there is note one, but there are several. ;-
http://www.voip-info.org/wiki-Asterisk+FAQ
Cheers, Philipp
___
Asterisk-Users
Hi!
This is possible? There are any best way to implement this?
Yes, look at asterisk -rx command
That command then can be sip show peers or database show sip.
Here is an example of a related CRON job that I use for restart:
# Restart Asterisk PBX once a day to prevent any problems from
Hi!
I have the rates that I currently pay my telco, and would like to
extract my CDR's and add an additional field displaying the actual price
paid for the call. I would like to do this based on destination phone
number, and outgoing channel.
Please look at this page for CDRtool
Hi!
The Stable cvs did have the iax2 timing fix applied as of May 21st (I
think that was the date). However, there seems to be a case that
channel_capi still needs attention.
And there is the no ring sound issue as well that would need to be
fixed in order to qualify for a 1.0 release.
Hi!
Thor, you are right here, the problem has crept back into CVS! :-(
In between it had been fixed, though.
Cheers, Philipp
However, on receiving calls from FWD, my Asterisk blocks the calls with
the following message:
May 30 20:19:24 NOTICE[180236]: chan_sip.c:6397 handle_request:
Hi!
Does anyone have a more clear beep tone for the voicemail?
Try Playtones():
http://www.voip-info.org/wiki-Asterisk+cmd+Playtones
Cheers, Philipp
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Hi!
Excellent!!
Philipp
See near the bottom for the interesting bit :-)
OK, while composing this post I decided to write a perl program to read
a uLaw stream on standard input and create a suitable header, writing
the result to an output file.
It can be found at
Hi!
I've made a couple of small contributions to the wiki but recently I
read the Terms of service, they are pretty draconian:
[...]
What worries me most is that the current terms seem crafted so as to
ensure that should the people who run voip-info ever decide to remove
content, or stop
Hi!
The failure has just been fixed as I saw in mantis:
http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0001738
Unfortunately that didn't solve my problem - however I am not sure
anymore that this is related, and maybe I just have a basic
misunderstanding concerning type=peer and
Hi!
The failure has just been fixed as I saw in mantis:
http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0001738
Unfortunately that didn't solve my problem - however I am not sure
anymore that this is related, and maybe I just have a basic
misunderstanding concerning type=peer
Hi!
Has anybody tried to use Freenet's Germany based iPhone Service with
Asterisk? Maybe even from behind a NAT? Freenet seems to use SER ... but I
can not get a connection to their SIP proxy from Asterisk going through a
NATed firewall.
Asterisk -SIP- Firewall with NAT -SIP- Freenet
Hi!
If that's the case, you have to know that you have to use one ISDN card
that supports NT mode (to emulate an NTBA) so that you can connect your
phone directly to it.
Question:
Do I need to insert the 100 (or 50) Ohm resistor if I want to connect an
old Auerswald PBX (ets2106) to the NT
Hi!
i upgraded to the actual CVS head from yesterday (27.5.) but can not get
incoming SIP calls from my provider (sipgate). If someone calls my
number, my asterisk responds with the following error:
May 27 21:30:21 NOTICE[1114606512]: chan_sip.c:6351 handle_request:
Failed to
Hi again!
i upgraded to the actual CVS head from yesterday (27.5.) but can not get
incoming SIP calls from my provider (sipgate). If someone calls my
number, my asterisk responds with the following error:
More details: SIP DEBUG on the called server reveals this:
Found peer
Hi!
my problem is to forwarding a call to a SIP phone and record the call at
the same time. How can I do?
This should help you to solve your problem:
http://www.voip-info.org/wiki-Monitor+setup+sample
Cheers, Philipp
___
Asterisk-Users mailing
Hi Petr!
I use about 300 IP phone combination Welltech LP101, welltech LP102
welltech3502-8, Cisco 7905 and Cisco 7960
Would you be able to write a short report about your findings with the
Welltech phones and post it to this list? I assume you are operating them
in SIP mode.
Cheers,
Hi!
I need to know if someone have an asterisk box with on, two, tree or fore T2
card, and which is the good materiel configuration to do that.
Have a look:
http://www.voip-info.org/wiki-Asterisk+hardware+recommendations
Cheers, Philipp
___
Hi!
Now, going the other way around is more difficult. #1 doesn't know
the IP address of #2. There is the concept of register= in
sip.conf, but that only registers _individual user-agents_ and does
not allow one server to know that another server is at a particular
IP address.
Hm, are
Hi!
The first thing we need to do is create the mailbox for Asterisk to use,
thankfully there is a little utility to do this:
/usr/src/asterisk/addmailbox
That's very old stuff, not needed anymore.
is this utility still in asterisk? or is this user guide in correct on this
part of the
Hi!
I'm tring to do some DB operations before and after a call. I see the 'g'
option in dial to continue in context if the destination hangs up, but
what if the originator hangs up?
You either need to run a CRON job for this clean up, or do that at the
beginning of the next call - whatever
Hi!
Below is my conf that i have now.Is there anything I need to configure in the
Dlink gateway for this to work with asterisk?
Here a few things you can try:
- upgrade to CVS-HEAD (not 0.9.0) and see if things are different
- issue a ngrep port 2727 to monitor what your dlink is sending
-
Hi!
On Sat, May 22, 2004 at 02:29:18PM +0200, Philipp von Klitzing wrote:
Note: The h extension is not reliable enough to solve your problem.
What is the problem with the hangup extension?
Not reliable - ask bkw for details, he can elaborate.
P
Hi!
P.S. You can really decode DTMF tones with your ear/brain?..:-)
Yes, sure, you easily remember recognize the melody.
Cheers, Philipp
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To
Hi!
exten = s,1,gotoif,$[${CALLERIDNAME} = \0]?2:3;
Do this:
GotoIf($[foo${CALLERIDNAME} = foo]?2:3)
Cheers, Philipp
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update
Hi!
On my SIP softphone, when I stop speaking the audio stops. So if im not
talking I cant hear the other person.
FAQ!
X-Lite: Menu -- Advanced settings -- Audio -- Silence
Set Transmit Silence to YES
P.
___
Asterisk-Users mailing list
[EMAIL
Hi!
| IF i use a sip softphone or a iax softphone with asterisk, i get a lag
| of about 1 second.
Would it get better if a straight codec were used? I.e., one that
does full 8KBps (I think that's either ALAW or ULAW). In other words,
bandwidth usage would obviously increase over a codec
Hi!
Ineed to pass the call duration and Bill Sec after a successfull
call to an AGI script. Is there a way to do this ?
- check out asterisk-addons from CVS
- enable the CFLAGS+=-DMYSQL_LOGUNIQUEID in the Makefile
- catch the unique ID of the call and pass it along to your AGI script
- let
Hi!
I am trying to find remote call forwarding feature in asterisk. I don't know
is it possible or any one had already done it.
The Wiki is your friend:
http://www.voip-info.org/wiki-Asterisk+call+forwarding
Cheers, Philipp
___
Asterisk-Users
Hi!
Should be
mailbox = [EMAIL PROTECTED]
Watch out - don't confuse an extension.conf context with a
voicemail.conf context! Go to /var/spool/asterisk/voicemail
and check the names of the directories (=voicemail contexts)
present.
Cheers, Philipp
Hi!
Should've been more clear, what I was referring to was the ability to select
the first or last message as a starting point when reviewing vm's.
You are referring to (4)(4) -- first msg and (6)(6) -- last msg.
However I don't think this plan made it into reality.
Cheers, Philipp
Hi!
I am trying to dial a mgcp extention from my sip phone and i am getting this
error message. anyone got any idea?
Do a mgcp show endpoints at the CLI and watch the output.
May 19 22:30:01 NOTICE[1251156800]: chan_mgcp.c:1104 find_subchannel: Gateway
'10.0.1.150' (and thus its endpoint
Hi!
A question: is there any way to get * to answer certain DTMF sequences
entered on an extension with a stutter tone?
Record the stutter tone in a .wav or .gsm file and use Playback() or
Background() to deliver it to the user.
See also:
http://www.voip-info.org/wiki-CLASS
Cheers, Philipp
Hi!
have managed to establish voicemail functionality using voicemail /
voicemailmain applications
the documentation on these applications from digium.com suggests that
voicemail greetings are customizable (as one would be expect), but am not
able to find any supporting documentation
Hi!
I am just getting started with AGI
so I wrote the following script as a simple test
but all that happens is silence before it times out and hangs up
can someone help to get me started?
Look here:
http://www.voip-info.org/wiki-Asterisk+AGI+php
php -q
?php
fputs(STDOUT 'SAY
Hi!
Can you use wildcards on the caller id? Eg certain area codes get a
certain ring type others get a different ring type?
Wildcards don't appear to be working (tried only * and ?), but partial
leading matches do. So 12 for ringtone 1 will match 1234 - unless you
have a more specific
taken from bug 881 (now resolved) :-(
--
markster - 05-19-2004 09:21 CDT
-- As
it turns out the 10S cannot conference on the device. From Jean-Francois
at
Hi!
caller ID already tells you who is calling). However, I can put anything
I want into the text boxes and nothing happens - I always get the
system ring tone.
No problem here, works just fine. If calls I get ringtone 2,
otherwise the default ringtone.
o System Ring Tone
o
Hi!
Grandstream v1.0.4.68 firmware
http://www.hellofone.com/downloads.html
Hehe, the ringtones are fun, but we'll need someone to reveal how to
upload our own samples...
Can I keep iLBC frame size at 20 ms or do I need to change this to 30 ms
for better (?) operation with Asterisk? It
It seems that entire days pass by before I
hang up... very odd, and very counter-productive to getting good
Asterisk work done.
Telephony is evil.
P.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Hi!
Well the idea is to run it on a 2 * Opteron 242 with 4 GB of memory and
4*73GB 15K-RPM disks RAID-1. However the systen should also be able to
accomodate up to 150 extentions with an approx. continous usage of 40%
(e.g. approx. 60 users simultanious using the phones), in such a setup
Hi there,
could anyone drop a short line on what cidinternalcontexts exactly does
in voicemail.conf? The Wiki explanation isn't sufficient - at least not
for me... :-
Also: How/where do I define an Operator extension?
Cheers, Philipp
___
Hi there,
today I made the German language prompts available for download:
http://www.karl.aegee.org/asterisk.nsf/HT/sound-de
Be aware: Asterisk doesn't yet fully support languages other than
English, there are still (smaller) issues with voicemail and date/time
announcements that require a
Hi there,
since a couple of days I can't seem to be able to compile CVS HEAD on
RH7.2. On a RH7.3 machine with bison-1.35-1 it appears to be fine
though... any advice?
Philipp
System: RH 7.2
bison-1.28-7
Related issue:
http://rpm.pbone.net/index.php3/stat/4/idpl/411535/com/bison-1.35-
Hm...
since a couple of days I can't seem to be able to compile CVS HEAD on
RH7.2. On a RH7.3 machine with bison-1.35-1 it appears to be fine
though... any advice?
Actually this doesn't seem to be related to bison - I can't even compile
my old CVS-HEAD-05/03/04-19:58:33 anymore, getting
Hi folks,
this does look like a bug to me: Asterisk replaces the @63.214.186.6 by
@context which obviously leads to a failure. Any comments, do I have a
configuration issue on my side that I missed?
Cheers, Philipp
-- Executing Dial(SIP/philipp-bd5f, SIP/[EMAIL PROTECTED]
out|90) in new
Hi!
302 Moved is not fully supported by chan_sip. Personally I like this
because the way Asterisk currently supports 302 Moved will prevent
calls from being forwarded outside of Asterisk's dialplan. I would
just create an exten = joesmith,1,GoTo(xxx,n) where xxx is the
extension you want
Hi!
Upgrade bison...i had the same problems until i upgraded bison.
Which means upgrading glibc ... :-((
In other words: Asterisk won't work with RH 7.2 (and the like) anymore,
basically. Still I wonder why I was once able to compile the March 5 CVS,
but can't do so anymore. Might be
Hi!
| exten = 999,1,SetGroup(moh)
| exten = 999,2,CheckGroup(1)
| exten = 999,3,Answer
| exten = 999,4,MusicOnHold(default)
|
| See?
| You can limit that to just 1 user at a time or what ever you wish :
|
| bkw
Great! So this is a means that can be used as an outgoing limit
Hi!
I don't want to re-invent the wheel if someone has already hacked a way
to do this.
One of my customers has a number of stores, and he wants to leave one
voicemail that would be delivered to all the managers at once. Each has
a voicemail account on his server.
I have googled
Hi!
Upgrade bison...i had the same problems until i upgraded bison.
Which means upgrading glibc ... :-((
Ok ok, I got it - compiled bison from source and disregarded those good-
looking tail-shaking RPMs. ;- Works fine now.
Cheers, Philipp
___
Hi!
able to support intercom/paging. Having searched the archives, it
appears that this question was asked about 6 months ago, and the answer
was that the Cisco phones support this using SCCP and having one line
set to auto-answer, but at the time this was not supported in the SIP
Hi!
I have 2 grandstream budgetone 100 series. I can call allright, but I
can™t do call transfer, park and call conference. (all features works
with tdm devices) the
1. Check if Asterisk is always in the media path, i.e. you need the t or
T option (or something similar) in your Dial
Hi there,
I have serious problems with MGCP and Swissvoice ip10s, and it appears
that recent CVS also introduced trouble for other MGCP users. Please
check and add comments in the bugtracker so that we can get a clearer
picture - thanks! Also comment if things are working fine for you.
Hi!
What I would like to do is set up an If Then Else type statement along the
following lines: -
If extension 7957 Then
Dialout on Capi msn 383590
Create a macro in extensions.conf:
exten = s,1,AbsoluteTimeout(${TIMEOUTABS})
exten = s,2,NoOp
exten = s,3,GotoIf($[$[${CALLERIDNUM} = 103] |
Hi!
I'm trying to get Asterisk to talk SIP to Vocal and so far have only
managed to get it partially working. Calls in from Vocal are working
fine but outbound calls aren't.
I haven't looked at your settings, but two days ago I upgraded to latest
CVS and since then I am unable to place
Hi!
Boy after really digging into this, I have discovered that there is more
information about each of these topics than I previously realized.
Strangely, searching the wiki on iax returns exactly nothing. But
searching on iax2 does start to dig up some good stuff.
Unfortunately the Wiki
Hi!
Are there anybody that have tested the application? Is it working
correctly for you ?
In fact, I didn't received any comments/feedbacks!
Give me some news, even bad, I spent long time to make it, sni ;(
It surely makes an excellent impression. I am sorry to say thought that I
Hi!
after some configchanges the CDR/Master still logs s insteed
of the called number?
,MYNUMBER,s,pstn-out,
Can someone tell me, what I have done wrong?
Don't use macros, or if use them make sure you get back to a realy
extension before the dialplan processing is completed.
Hi!
Can anyone suggest a way in which all users could dial the prefix 8 and *
would automatically associate the correct FWD account for the outbound call?
Try using GoToIf [show application gotoif] in combination with
${CALLERIDNUM} [asterisk/doc/README.variables]
I prefer a slightly
Hi!
I see this very same effect rather often in the following setup:
SIP (GS101) -- *1 -- IAX2 -- *2 -- MGCP (ip10)
In fact I think I've seen it also with SIP instead of MGCP at the end.
The first client is behind NAT, by the way.
That must be it. I have seen this happening with
Hi!
Only Grandstream phones appear to be affected. All phones affected
have been behind a coned NAT, running firmware 1.0.4.39 with STUN
enabled. The hangup only occurs in dialogs with CSeq set to '0'.
Ok, I'll watch for that as well since I upgraded my desk's Grandstream to
1.0.4.54 an
Hi!
I have set up a new SwissVoice phone and it can receive calls but I
cannot make calls out from it. The setup is simple for now, 2 phones:
SwissVoice is ext 7726 and Cisco 7960 (SIP) is ext 7999.
Which verasion of Asterisk are you using?
Please do check the known MGCP bugs, especially
Hi!
Registration only works if you have set host=dynamic for the client! In
case of a static host registration makes no sense, anyway! The only
purpose of registration is to tell the server at which IP address the
phone can be found.
Cheers, Philipp
Hi!
I would like to know if chan_capi is prepared to receive faxes. I have a
eicon deiva server 4bri with chan_capi and Grandstream HandyTone connected
to a Fax, but this fax can't receive faxes.
Good question - my attempts at getting this to work with a passive AVM
Fritz! failed - at
Hi!
Lately, I have been experiencing unexpected hangups just when the a call
has been established. This effects a small percentage of all calls
coming from sip phone which are terminated on a zap pri channel.
I see this very same effect rather often in the following setup:
SIP (GS101) -- *
Hi!
On Tue, 13 Apr 2004, Dmitry Mishchenko waxed:
In other words can I receive information which we are usually getting in CDRs
during the time when the call is still active?
Yes, via the manager interface. Check manager.conf, it
lets * talk on port 5038.
The other option is that
Hi!
4) Installed the Flash Operator Panel
5) Installed a modified version of Monastery to show me which agents were
[...]
I suspect the problem to be either caused by 4 or 5, in which case they
will be very easy to rectify. I would however like to know if anyone else
has had a) the same
Hi!
Again, could anybody say the concurrent calls limit for one Asterisk Box?
Look here:
http://www.voip-info.org/wiki-Asterisk+dimensioning
Cheers, Philipp
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Hi!
does anybody successfully managed to get swissvoice ip10s with h323
firmware work with asterisk ? mgcp firmware works fine, but with h323
i'm still getting one way audio.
Never tried, no clue. But I can tell you that newer ip10 firmware and
latest head CVS (yesterday) don't play
Hi!
If you put mp3 files into your mohmp3 directory and these files have ID3v2
tags, mpg123 will throw an error message Found new ID3 Header,
regardless of the -q flag.
This, in turn, will cause Asterisk to crash (yes), although it's a soft
crash (exits cleanly).
It took me forever to
Hi!
I am interested to learn if I need to have ztdummy installed if I do not
have any zaptel hardware in my machine?
No, not necessarily. You'll only need it if you want to use MeetMe
conferencing. Look here:
http://www.voip-info.org/tiki-index.php?page=Asterisk+timer
I have found a lot of
Hi!
is there any howto available?
1. Where to get firmware-files (cant find anything at grandstream.com)
2. How to configure tftpd (with snom200 my tftp works fine)
3. How to place (naming,config) the files in tftpdir
http://www.voip-info.org/wiki-Asterisk+phone+grandstream+budgetone
Hi!
What I want to look at having a Small Home Office setup were I can use
the 1 BRI for both DID and Internet at the same time.
Is it possible to use something like a FRITZ! ISDN BRI card to have full
time Internet on one of the B channels and have the other B channel for
* for both
Hi!
Initial thoughts are to use a counter, increment on call presentation,
decrement on call tear down, and give the inbound call busy or congestion
treatment if the counter is above a certain value when the call is
presented?
I guess you need to protect your rather thin Internet uplink so
Hi!
I'm still running 1.0.3.81 because I read that once you move up to
1.0.4.x you can't go back again, and my experience isn't *that* crappy.
You probably want to start with 1.0.4.26 although also 1.0.4.17 seems to
have been relatively stable. Then there is also 1.0.4.39 but it seems to
be
Hi there,
see subject - any suggestions? I'd like to be able to get a local number
in Rome or Milano or Torino...
Cheers, Philipp
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To
Hi!
What is the relationship between when CDR recording occurs and the
hangup extension is executed. Normally CDR happens before the h
extension is executed.
In short: Do not rely on h for CDR purposes.
I use the h extension to clean up for routines, but sometimes it gets
called to
How about 120? Look here:
http://www.voip-info.org/tiki-
index.php?page=Asterisk+setup+medium+office+100
I've set up 75 extensions... I'm 100. Sorry.
Would anyone care to share some experience with big installs, ie.
multiple PRI's and excess of 100-200 extensions.
Thanks
Rob
Hi!
Even though it was 100, I'm also keen to hear about large installs,
what kind of experience did you have setting it up, and what hardware
for the * server did you use?
This might help if you are interested in no. of concurrent calls
instead of number of extensions/phones:
Hi!
Can build a switchboard with TDM400P + X100P?
I need a receptionist to pick up the incoming calls and transfer them to
appropriate employee.
You might want to read the handbook draft:
http://www.digium.com/handbook-draft.pdf
Do I need those Nortel telephones for this or Panasonic KXTD
Hi!
good day i just install successfully asterisk and when i try iax client
to connect to my asterisk server im getting a Call reject by Remote
register = test:[EMAIL PROTECTED]
host=10.1.1.2
Registration makes only sense - and only works - if you have
host=dynamic. The sole purpose of
Hi!
has anybody managed to call a (old fashioned) phone using Sipgate.de and
asterisk? (yes I have money on my account :-) )
extension.conf:
exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)
Try this instead:
exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)
Philipp
401 - 500 of 760 matches
Mail list logo
