Hi All,
I download the app_asyncgoto.c, compile the app_asyncgoto.so. Then according to
this page
http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO ;
when I dial ,there have this warning:
-- Executing AsyncGoto(SIP/111-086497c8,
SIP/113-08674628|dynamic-nway|111|1) in new
Dell 2950 doesn't come with Intel network chip. It comes with Broadcom
extreme and module is BNX2.
We are using these servers and using Gentoo linux. We are experiencing
issues when there are 30+ calls enabled for call recording and we assume
this is due to new SAS controller.
If you not
It comes with PCI express by default, but you can ask dell to provide
you the PCI raiser cards instead of PCI express.
Ahsan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Florell
Sent: 01 February 2007 20:07
To: Asterisk Users Mailing List -
I bought a Dell 2850 as a pbx server and it just sucks IMHO
The stupid thing has only 3 pci slots and even with only 3
pci slots Dell
managed to have a shared irq on every slot, 1 for the scsi
controller and
one for each nic
We're using a couple of Dell 1850's and I couldn't be
On 2/2/07, Robert DeVries [EMAIL PROTECTED] wrote:
I want to essentially transplant my existing Asterisk server to a new
machine, and take the old sever out of service.
Assuming I install Asterisk on the new machine, does anyone know what
files I would have to copy over? What comes to mind
On Thu, 1 Feb 2007, Christophorus Laube wrote:
We have a 2850 in a productive environment with a BNE1 performing well
(OpenSuSE 10) and a 2950 with BNE1 and BN8S0 also performing OK (on Ubuntu
Edgy). You only have to blacklist some hotplug kernel modules and yes, we do
have very long pings (1
Hi
Is there a way to control volume in VoIP calls just like the gain
parameters for ZAP lines?
Thanks,
François.
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Eric ManxPower Wieling wrote:
Leo Ann Boon wrote:
Eric ManxPower Wieling wrote:
You should not have quotes in Caller*ID info. MOST devices will
just ignore the quotes, but a few will refuse to accept Caller*ID
with quotes in it. At least one revision of SIP firmware for Cisco
phones does
On Friday 02 February 2007 12:29 am, Robert DeVries wrote:
Assuming I install Asterisk on the new machine, does anyone know what files
I would have to copy over? What comes to mind are the *.conf files in
/etc/asterisk, as well as the voicemail audio files. Anything else?
Asterisk is
Yuan LIU wrote:
From: Leo Ann Boon [EMAIL PROTECTED]
Yuan LIU wrote:
Related to callerid: I can't get text ID to work in an analog phone
on FXS. I tried the above format, it simply displays the entire
string in both numeric and text field (i.e., displays the same
string twice). Tried a few
Hello to all,
I post again (last time subject: Line drops strange problem(got event On
hook) because i have caught in debug a situation where i get a call and
the line drops and i get a call from the same caller and the line works
well and the call normally closes by both parties. The only
Hi,
I get this error while compiling install-misdn on a Debian box with
kernel 2.6.18:
*make[2]: ***
[/usr/src/install-misdn/mISDN/drivers/isdn/hardware/mISDN/hfc_multi.o]
Error 1*
*make[1]: ***
[_module_/usr/src/install-misdn/mISDN/drivers/isdn/hardware/mISDN] Error 2*
*make[1]: Leaving
Hi,
Is it possible to keep asterisk from logging exactly everything? I can
do the logger rotate and keep the files small enough, but I think it's
unneccesary to log exactly all data.
File grows by about 5 gb per month!
Thanks!
Regards,
Jan
___
On 2/2/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi,
Is it possible to keep asterisk from logging exactly everything? I can
do the logger rotate and keep the files small enough, but I think it's
unneccesary to log exactly all data.
File grows by about 5 gb per month!
Thanks!
Regards,
Jan
On Fri, Feb 02, 2007 at 12:29:34PM +0100, [EMAIL PROTECTED] wrote:
Hi,
Is it possible to keep asterisk from logging exactly everything? I can
do the logger rotate and keep the files small enough, but I think it's
unneccesary to log exactly all data.
File grows by about 5 gb per month!
Bruno Castelo Branco wrote:
Hi
Try that one http://www.counterpath.com/index.php?menu=Productssmenu=xlite
Bruno C. Branco
Do they still have the web-based one available, formerly X-Web?
--
Chris Hills | Tel: +44 (0)1527 572754
IT Services |
Hi Brian!
Actually i play a message to the caller, something like Hello and
welcome at ..., someone will take your call in a few seconds - so a
random musiclist is not an option. Even if i would play music only, it
doesn't sound very cool if the music is changed within 5 secs. I would
suggest to
*ping*
I am interested in this too if anyone has any clues? I am looking to
do this on a Cisco 7941/7961.
Thanks,
Steve
On 1/26/07, Naija Man [EMAIL PROTECTED] wrote:
Hello,
We have an asterisk system with about 40 cisco 7940/7960 phones and a few
linksys SPA941. I recently analyzed our
ER == Eric Rousse [EMAIL PROTECTED] writes:
ER Hi, I was planning on getting a Dell PowerEdge 2950 for our new
ER Asterisk configuration. But while searching for documentation
ER about it and/or reported issues, I found this:
ER http://www.voip-info.org/wiki/view/Asterisk+hardware WARNING -
ER
Leo Ann Boon wrote:
Eric ManxPower Wieling wrote:
Leo Ann Boon wrote:
Eric ManxPower Wieling wrote:
You should not have quotes in Caller*ID info. MOST devices will
just ignore the quotes, but a few will refuse to accept Caller*ID
with quotes in it. At least one revision of SIP firmware for
Yes. This is a function of the VoIP endpoint devices, not Asterisk.
François Delawarde wrote:
Hi
Is there a way to control volume in VoIP calls just like the gain
parameters for ZAP lines?
Thanks,
François.
___
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You're looking at only the logfiles, which don't reflect the problem at
the other side, the switch which sees the incoming request abort before
it can complete the connection, and before the 45s timeout. What you're
missing is my reports of that difference on either side of the network,
Hi,
An incoming call is redirected to another number by our asterisk
server. In the incoming call the caller name is present but when
redirect the call, the end receiver is not able to see the callerid
name. The caller id number is visible.
our related changes to extensions conf is below.
On Tue, 30 Jan 2007 17:50:53 +0100
Benko [EMAIL PROTECTED] wrote:
Hello!
I've upgraded from 1.2.9 to 1.2.14 recently but experience an
unexpected behaviour with musiconhold: While in 1.2.9 musiconhold was
playing continuous on sequential extensions after a
timeout, it is restarted for
Hi,
Is it easy to show calls elapsed duration within a FOP button ?
FOP documentation mentions timer but I couldn't find any example or clue
proving it's possible to do what I'm looking for.
Anyway, would you recommend another software to customize Asterisk call
display ?
Regards
I have the following setup:
UA1 (SPA2000) -- Nat1 -- Asterisk (public internet) -- Nat 1 -- UA2 (X-Lite)
Relevant parts of sip.conf
[general]
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0; Address to bind to (all addresses on machine)
externip = 60.234.100.100
Hello all
I asked similar question some time ago but didn't get answer... Maybe
this should asked in asterisk-dev or bugs.digium.com?
For example, I have 3 sip phones defined in sip.conf - 101, 103, 109 and
this simple dialplan:
[local-ext]
exten = 101,1,Dial(SIP/101,,t)
exten =
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Michael Collins
Sent: Thursday, February 01, 2007 12:38 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Dial option G - Passing parameters?
Has anyone used the G
Hi,
I have an queue stored in relatime and defined members called through
LOCAL/
I found out that if I call the members through the LOCAL think the queue
statistics is not updated.
Any idea, or isnt possible to call members with LOCAL channel.
best regards
Thomas
On 1/25/07, Wayne Jensen [EMAIL PROTECTED] wrote:
I'm having various issues that may or may not be related to each other (I'm
pretty sure they are). We've had this system for a year now (quad T1 card,
right now we have 1 T1 coming in, 2 going out to channel banks) and we've
had intermittent
Don't you think it could be an interesting feature in Asterisk? It
already does transcoding, why not gain when voice flow passes through it?
François.
Eric ManxPower Wieling wrote:
Yes. This is a function of the VoIP endpoint devices, not Asterisk.
François Delawarde wrote:
Hi
Is there a
I'm having trouble building res_snmp (under Asterisk 1.4.0) on a box
running Debian Sarge. res_snmp says its dependencies are netsnmp but
Debian doesn't seem to have a netsnmp package. I've tried installing
pretty much every package available related to snmp and no luck. I'm
just wondering if
Perhaps you can write the functionality? I'm sure you can do a quick
hack of you modify app_voicechangedial.
On 2/2/07, François Delawarde [EMAIL PROTECTED] wrote:
Don't you think it could be an interesting feature in Asterisk? It
already does transcoding, why not gain when voice flow passes
I have been having a very similar problem. Has anyone here gotten a
DIALSTATUS for calls started with originate?
I did some research and saw some posts that local channels are the solution
to this problem. However, I could not find examples of how to use local
channels with originate. I could
McGhee, Stefano wrote:
Hello all,
Please don't reply to an existing message to start a new topic. It
screws up the message threading.
Thanks,
-Stephen-
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To
Porier, Jeremy M. wrote:
Are there any scripts out there that would help me stress test two boxes
that are setup back to back with 4 PRI connections? We're having
problems with Sangoma cards w/ PCI-e on HP DL385 G2 hardware and I'm
tired of testing them in a production environment. As
On Fri, Feb 02, 2007 at 12:49:26PM -0600, Jeremiah Millay wrote:
I'm having trouble building res_snmp (under Asterisk 1.4.0) on a box
running Debian Sarge. res_snmp says its dependencies are netsnmp but
Debian doesn't seem to have a netsnmp package. I've tried installing
pretty much every
Leo Ann Boon wrote:
Alessio Focardi wrote:
Hi,
I'm looking for an hardware platform for an * installation that should
have at least 3 PCI slot with no irq sharing whatsoever.
Use an industrial PC with a backplane bus. You can easily get 3-4 usable
slots in a 2U and 10-14 slots if you
Hi:
Tzafrir Cohen wrote:
On Wed, Jan 31, 2007 at 07:57:54PM -0700, Stephen Bosch wrote:
Hi, folks:
Can the loop drop detection threshold (normally defined in milliseconds)
be set on the Digium TDM-400 cards? Most PBXs let you set this value.
What exactly do you need it for?
On the FXO
Hi Asterisk Gurus,
I am new to Asterisk server. We are trying to use Asterisk for testing one
of our new products. I was wondering if anyone can tell me if it is RFC
compliant or how can i use Asterisk to test it for some basic RFC
compliance.
Thanks in Advance,
hi,
did you think to tx rx params you can consult asterisktutorials.com,
i m not sure of this but may be it will work
2007/2/2, Stephen Bosch [EMAIL PROTECTED]:
Porier, Jeremy M. wrote:
Are there any scripts out there that would help me stress test two boxes
that are setup back to back with
no you can look for tyan machines they aren't expencive
2007/2/2, Stephen Bosch [EMAIL PROTECTED]:
Leo Ann Boon wrote:
Alessio Focardi wrote:
Hi,
I'm looking for an hardware platform for an * installation that should
have at least 3 PCI slot with no irq sharing whatsoever.
Use an
A S wrote:
Hi Asterisk Gurus,
I am new to Asterisk server. We are trying to use Asterisk for testing
one of our new products. I was wondering if anyone can tell me if it is
RFC compliant or how can i use Asterisk to test it for some basic RFC
compliance.
Thanks in Advance,
That's an
Voipjet locks $1.2 per running call and unlocks when call ends .. so $12 =
10 simultaneous calls ( if rate is 1.2 cents ) .
On 02/02/07, Robert DeVries [EMAIL PROTECTED] wrote:
I have found that if you don't have the minimum balance required for the
voipjet premium server, you get the circuits
On 1 Feb 2007, at 16:34, Porier, Jeremy M. wrote:
Are there any scripts out there that would help me stress test two
boxes
that are setup back to back with 4 PRI connections? We're having
problems with Sangoma cards w/ PCI-e on HP DL385 G2 hardware and I'm
tired of testing them in a
I have a 6 - 7912's.
I have a TDM2402E echo cancel card.
Asterisk 1.2.12.1
Some times extension to extension the audio is only heard one way.
Some times extension to TDM2402 calls audio is only heard one way.
I have turned off the echo suppression on the 7912's config.
Any idea why it would be
I have been having a very similar problem. Has anyone here gotten a
DIALSTATUS for calls started with originate?
I did some research and saw some posts that local channels are the
solution
to this problem. However, I could not find examples of how to use
local
channels with originate. I
On Fri, 2 Feb 2007 16:56:59 +0100
Benko [EMAIL PROTECTED] wrote:
I've filed a bugreport in the meanwhile, hope there'll be a resolution
- http://bugs.digium.com/view.php?id=8969
issue was resolved in 1.2 rev 53084 and 1.4 rev 53088 (see
http://bugs.digium.com/view.php?id=8672) but there's
On 2/2/07, Thomas Winter [EMAIL PROTECTED] wrote:
Hi,
I have an queue stored in relatime and defined members called through
LOCAL/
I found out that if I call the members through the LOCAL think the queue
statistics is not updated.
Any idea, or isnt possible to call members with LOCAL
The problem can be reproduced in the same way by putting a caller on hold,
unholding, and holding again. The MOH restarts from the beginning of
whichever file it was playing last. (I have random enabled, so it randomly
picks a please wait for the next blah blah blah file). (I'm using 1.4
release).
have a Grandstream and SJPhone SIP phones going to asterisk.
with SJPhone (on Linux) getting. any ideas??
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
192.168.2.100;branch=z9hG4bKc0a80264001045c3c2c52331d4920678;received=24.10.123.39;rport=60754
From: sip:[EMAIL
Stephen Bosch wrote:
snip
...and have zillions of dollars :)
Industrial PCs are pretty expensive.
Over here, they're actually quite reasonably priced. A 2U rackmount P4
D930 3.0GHz, 1GB RAM system with 4 PCI (32bit) slots starts around US$1K.
Leo
Good question. Anyone knows if the TDM-400 actually detect loop drops?
Well, that's really what kewlstart (and loopstart) means. If it
couldn't, then Asterisk wouldn't know that the call had been hung up,
and hog the channel.
For loopstart lines, I don't think Asterisk detects loop
Shivram u wrote:
Hi,
An incoming call is redirected to another number by our asterisk
server. In the incoming call the caller name is present but when
redirect the call, the end receiver is not able to see the callerid
name. The caller id number is visible.
If you're calling PSTN, caller id
Problem: *Call* *waiting* comes in, I press flash to answer it, and the
first caller gets disconnected after 3 seconds. This is all ZAP - no VOIP.
System:
Analog stations and trunks running on a pair of TDM400's. It does NOT have *
call* *waiting* from the phone company, and I have enabled it in
From: Leo Ann Boon [EMAIL PROTECTED]
Works fine with my GE29393GE2-A. I think you need the right syntax, in
your .conf it should look like
callerid=John Doe 1234
Note the quotes around the name.
Leo
Ain't working. 27935GE3-B simply says unknown or displays a blank if
the string contains
From: Andrew Joakimsen [EMAIL PROTECTED]
Perhaps you can write the functionality? I'm sure you can do a quick
hack of you modify app_voicechangedial.
Not sure if this is a good idea. How do you handle situations where no
transcoding is required? You don't want unnecessary heavy lifting.
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