Is it possible to automatically set up a call between two external
lines?
I would like Asterisk is call a cellphone number, wait for it to answer,
and then call another cellphone, when that answers connect the two
together.
I assume it is possible but can someone point me how to do it.
Thanks.
This comment comes up fairly regularly and is confusing people.
Why doesn't it say that it failed so we know?
The way it is now it kind of leaves you hanging there and you don't know
if the transfer happened or not.
And why was it even attempted if it is obvious that transfer is off?
(I know it
For work environments where you only get HTTP or HTTPS access, what is
the feasibility of doing IAX over HTTP?
I have heard of projects such as stunnel.
Has anyone tried something like this?
I did a quick search but didn't come up with much.
___
Subject: [Asterisk-Users] Re: IAX over HTTP
HTTP uses TCP. Too much overhead. Add SSL on to that and you have a huge
amount of overhead. The end result would be poor and choppy sound
quality.
Jason
On 21/07/05 21:58 +0200, Rob Scott wrote:
For work environments where you only get HTTP or HTTPS
:
[Asterisk-Users] Problem with DTFM and complex international
setup
Do you think this might have an impact on http://bugs.digium.com/view.php?id=4631?
Mark
On 7/3/05, Mohit
Muthanna [EMAIL PROTECTED]
wrote:
Right...
that's the one. My mistake.On 7/1/05, Rob Scott [EMAIL PROTECTED] wrote
We have some guys working in the US who can't always dial back to our
company in Europe easily (lots of clients require authorization to make
international calls), so I set up the following:
- ipkall.com number links to a FWD number
- office Asterisk box registers with FWD
Then I
Discussion
Subject: Re: [Asterisk-Users] Problem with DTFM and complex
international setup
Try compiling Asterisk with RELAX_DTMF (See Makefile).
Mohit.
On 7/1/05, Rob Scott [EMAIL PROTECTED] wrote:
We have some guys working in the US who can't always dial back to our
company in Europe easily (lots
I would also be interested.
I've tried several times unsuccessfully to set up H323 with
Asterisk.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adeel
-31Sent: 01 July 2005 23:32To:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] How to
Configure a H323 Phone
If you your board into an ISDN wall socket and it works then you are
acting as a terminal so you are in terminal mode.
Now, how are you connecting to the PBX?
If you are connecting to an ISDN extension on the PBX, then still you
have to match the kind of connection, whether it is point-2-point or
Looks normal to me.
What Dial with the '' means is that both lines ring, but the first one
to answer is connected on the call.
From you trace it looks like Zap/3-1 which is your number 206 answered
the call, so the other line goes to hangup.
The Dial with '' is used to implement call teams where
Around 250ms max. Over that and you will have the walkie-talkie effect
you are experiencing.
So with you 600ms delay you are way over the top.
There is also the delay on the call on the PSTN side you have to take
into account.
For example, I am in Europe and making a call to the UK via Voipjet is
For a start it should be
${EXTEN}
You have to realize that ALL variables look like that.
Dollar-open-curly-brackets-variablename-close-curly-brackets.
So it didn't see your text as a variable and it tried to call the number
$EXTEN on Zap/g2.
-Original Message-
From: [EMAIL
I am looking to roll out an Asterisk VoIP implementation to our 200
employees.
So far I have hooked up the Asterisk box to our Elmeg PBX via a PRI
interface card and have that working, plus about 30 test users on Xlite
softphones.
Up til now all the configuration has been done by hand editing
Short answer is no. You should always turn it off on any client you
have.
Longer answer is that is is being worked on and should be available any
day now (although that has been the case for some months).
Also someone is working on porting it to SIP as well as IAX2.
No idea if the new work will
over time
especially with answering calls, with the zaphfc drivers before.
A reboot usually cures it for a while, but it would be great if such a thing
wasn't needed.
-Original Message-
From: [EMAIL PROTECTED] on behalf of Rob Scott
Sent: Wed 3/23/2005 7:39 PM
To: Asterisk Users Mailing
I have the latest bristuff, a zaphfc card for external calls and a PRI
card for connecting to a PBX as a channel bank.
With a BRI I would expect to be able to have two incoming calls going at
the same time, but when I try it, one call connects and the other gives
the following console message, a
Looking at that list, the easiest way would be to disable all your USB
ports in your BIOS, reboot and see if the card has its own IRQ. Assuming
you don't need USB. In general, just turn off all the things you don't
need that use IRQs.
-Original Message-
From: [EMAIL PROTECTED]
You should have set up the two cards as zaptel as a different group in
the zapata.conf.
Then if you want to dial your pbx you are dialing out of Asterisk, so
you use the Dial command.
Assuming that the PBX PRI link is in group 2 in zapata.conf
Something like:
exten =
It seems to me silly to have a T1/E1 card to connect to a channel bank
when you could just have a 24/30 way FXS card in the slot in the first
place.
Does such a thing exist?
Wouldn't Digium have a lot of customers if they could produce one for
say $1000 retail?
Anyone have any experience with the Rhino T1 channel bank?
It looks very cost effective at around $1300 for 24 lines but I haven't
seen it mentioned on the Asterisk list yet.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
I have noticed the following:
- the PCI ID of the card seems to change over time which means that
loading the module does not always recognise the card, only way to reset
this is to power cycle the machine
- you cannot unload the module once it is loaded, it hangs the
machine, which also
I have an Asterisk box with TE110P PRI connected in net mode to a PBX.
Both are PRI EuroISDN.
The connection seems to work OK but when calling from Asterisk to the
PBX through an Xten, the Xten client does not get a ringing tone when
the PBX phone rings.
Is it possible to set this up?
Is there
OK I have to ask.
Why is it that Asterisk can't cope with silence suppression?
All the clients seem to be able to but not Asterisk.
What would be needed to get it to work with silence suppression?
What is the problem?
___
Asterisk-Users mailing list
I get a ton of these messages, a pair every 4 or 5 mins.
Is it a problem?
I am wondering where they come from and if they are important.
I have a zaphfc card running in TE mode connected to a PBX.
Feb 16 20:23:04 epbw202 kernel: zaphfc: b channel buffer underrun on
card 0
Feb 16
I am having problems loading the zaphfc from bristuff and wcte11xp
drivers at the same time.
If I load zaphfc then all works fine.
If I then load wcte11xp, the card using the zaphfc doesn't pick up calls
anymore.
I am using bristuff 0.2.0-RC5.
Anyone else seen this problem, know of a fix, or can
Turn of Silence Supression.
If you have already done that then I think you are having the usual
Xlite - Asterisk experience.
At least I have the same problems with it.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
I guess it is possible to have an HFC-S card and a Digium TE110P card
working at the same time?
The TE110P will work in E1 mode.
I think the zaptel.conf is probably right but the zapata.conf not (I
just tacked on another group at the end but I don't really know what I
am doing).
Can anyone help?
Any idea when that is likely to be ready?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of joachim
Sent: Tuesday, February 08, 2005 9:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] jitterbuffers - suggested
Apparantly the new one will do things like interpolation so that if
packets are lost it will generate new ones to fill the gap. The current
jitterbuffer doesn't do that so you get silence on packet loss. There
are a bunch of other features too that I don't remember, but that was
the most
I am also interested in sound quality with respect to the zaphfc
drivers.
What is your physical setup?
Where are you listening for the noise?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Corvin
Sent: Monday, February 07, 2005 7:54 PM
To: Asterisk
I have exactly the same problem.
It was also the same with RC3.
It seems that after a couple of days of working fine, at some point
incoming calls fail but outgoing calls still work (or I would hear user
complaints earlier).
For the lack of ring problem, I do the following in extensions.conf:
I allow them to us any codec except speex (which seems to crash Asterisk
when used from an Xlite).
But it would be good if the user could choose their preferred codec
because with a softphone on a laptop sometimes you are on a connection
with good bandwidth to Asterisk and sometimes somewhere
:)
Are these bugs known at Junghanns?
On Fri, 4 Feb 2005, Rob Scott wrote:
I have exactly the same problem.
It was also the same with RC3.
It seems that after a couple of days of working fine, at some point
incoming calls fail but outgoing calls still work (or I would hear
user complaints earlier
I use
pritrustusercid = no
In zapata.conf and then it seems to work.
No idea if it is a bug or not or if this is a proper solution.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Remco
Barende
Sent: Tuesday, February 01, 2005 10:11 PM
To: Asterisk
Also just adding
callerid=asreceived
To zapata.conf also seems to work.
Works for local or national calls where I am.
I don't know about international calls.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Remco
Barende
Sent: Tuesday, February 01,
Hi,
I've got an Asterisk box with grandstream and xlite clients on it.
No here's the thing:
- I grey out all the codecs on the Xlite except for GSM
- I call the Grandstream from the Xlite, the Xlite uses the GSM codec
and the Grandstream uses ulaw, with Asterisk doing the conversion,
-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rob Scott
Sent: Thursday, February 03, 2005 7:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Odd behaviour between Grandstream and Xlite
Hi,
I've got an Asterisk box with grandstream
I've tried setting the QoS settings on the card and using the Microsoft
QoS packet scheduler, in all combinations, but no changes.
I don't think these applications use QoS anyway.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich
Adamson
Sent: Friday,
Also sound quality seems to be poor using the ULAW codec.
I am using:
- latest Firefly on Windows XP SP2
- Asterisk 1.0.5 patched coupled with Bristuff-0.2.0-RC5 with Florz
patch for zaphfc
- Linux kernel 2.6.9-1.681_FC3 Fedora Core 3 (obviously)
- connecting to FWD dialing 411 info
Anyone got any tips on improving sound quality on soft phones running
under Window XP SP2?
I have tried Xlite, SJPhone and Firefly.
They all seem to have significant sound quality problems. We have a
reasonable sized network of several hundred devices connected together
using Layer 2 switches,
Try commenting out the line
pritrustusercid = yes
Or set it to 'no'.
That worked for me.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jens
Sent: Friday, January 21, 2005 7:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
What are the advantages in using mISDN over other solutions?
If I knew why it was a good idea (like does it have better sound quality than
alternatives?) then I would put the time in to test it, and also improve the
Wiki.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL
I've been struggling with connection Asterisk to ISDN BRI lines for a
while.
I have it working with the latest bristuff and compatible Asterisk
version:
Asterisk 1.0.3-BRIstuffed-0.2.0-RC3a
I am using a cheap Centronics ISDN card and the zaphfc drivers.
It works but users complain that the
I am hoping someone is going to bite on the sound quality issue.
I have Asterisk connected via a Conceptronics HFC-S card to an Elmeg
ICT880 PBX internal extension line. Running Asterisk 1.0.3 and latest
Bristuff.
I have firefly and Xlite clients running on Windows XP.
Calls between Xlites
Asterisk seems to regard an unregistered phone to be busy.
Is that correct? Is not an unregistered phone unavailable?
It is odd to me that if someone dials an unregistered extension, then
the dialplan jumps to busy and voicemail kicks in saying that the person
is on the phone, when clearly they
channel in case it is too quiet.
Nothing so far has helped.
Calling directly through the PBX from a normal extension phone doesn't
seem to have any problems.
Anyone have any idea what I should look at?
Thanks
Rob Scott
EPAM Systems Ltd.
___
Asterisk-Users
At one point I did have this kit working but since upgrading to the latest
Asterisk, it no longer seems to.
I had the following problems after several reinstallations:
- USB adaptor had a proper dialtone, asterisk recognised the pickup,
but pressing keys on the handset had no effect
- USB
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Todd
Sent: 21 August 2003 21:01
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] RE:911, networks of * servers, etc. (was: VOIP
Dialtone?)
Yes, I'm familiar with the E911 platforms and their
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