I've used these gateways and never experienced any of these problems. I
could imagine me missing the popping noise but I do know that MWI did
work just fine.
Steve Totaro wrote:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Stephen Bo
I've had mixed results with changing ulimit and not restarting asterisk.
Best bet is to stop and start asterisk so that it calls a new shell
Rilawich Ango wrote:
Thanks for your reply.
What I ready do is:
add ulimit -n 65535 in safe_asterisk
increase value to 203380 in /proc/sys/fs/file-max
Bo
don't know why, I just know that when my ulimit was set at 1024 I was
getting around 120 concurrent calls before getting the error.
Tzafrir Cohen wrote:
On Thu, Apr 26, 2007 at 08:43:17AM -0500, Jason Fuermann wrote:
1024 open files will get you around 120 concurrent calls.
8
1024 open files will get you around 120 concurrent calls. Rilawich,
putting the ulimit in safe asterisk doesn't always work (my experience,
and proven because your ulimit -n is still 1024). Add this line in
limits.conf "* - nofile 65535", its
located in /etc/securi
also I've seen that not having the correct version of sip.cfg and
phone1.cfg could cause weird problems. Make sure you are using the ones
that came with the firmware.
Mike wrote:
Exactly. It's a weird issue, and I can't imagine what the problem is,
except maybe for bad phones (but then again,
I have my system set up to check the cid of the calling number and if
the room number the user inputs matches the calling extension (the last
4 digits in my case) then the number is considered admin. This does have
the same downside that Dovid pointed out, the admin must be in the room
for user
we have this problem. In our case it was due to the voice mail app; it
was failing to unlink files in memory when creating mp3s. Not sure what
your specific problem might be
Giorgio Incantalupo wrote:
Hi,
my Asterisk 1.2.9.1 suddenly freezed ("stop now" did not work!!) . I
found the following
Your best bet is to use DUNDI
Azfhasterisk wrote:
Can someone point me to some documentation on how to configure an
Asterisk box to do Termination and Origination for a few other
Asterisk servers? We have a box with a T-1 in it and we want to share
it with some other companies that have Aste
your asterisk box has to do audio conversion, its getting bogged down
Yuan LIU wrote:
I'm greatly surprised when testing an Asterisk box with 802.11g.
Here's the topology:
VoIP caller --- 802.11g --- Asterisk --- 802.11g --- VoIP extension
|
our Polycoms reregister almost immediately. I think the problem your
running into is that when the softphone is registered the polycom gets
some kind of error from asterisk which prevents it from reregistering
Rob Schall wrote:
That's what I would have thought. I set the timeout to be 300 secs
We have a similar situation and we do a realtime lookup in an external
db, works like a champ
Steve Murphy wrote:
On Wed, 2007-02-07 at 22:21 -0500, Lee Jenkins wrote:
Eric Germann wrote:
We're beginning to test MultiTech's CallFinder CDMA Units, one for Sprint
PCS and one for Alltel.
We have done limited testing with the Citel gateways and they seem
pretty cool. We're fixing to deploy them as a replacement to a hotel
pbx, and after that use them as an interim solution until full VoIP
convergence in our campus environment. I would be interested to know
what other peoples exp
Its a problem in your database. something might have corrupted...be
prepared to load a backup
Gregory Duchatelet wrote:
Hi,
I have a working asterisk 1.4.0 with Mysql Realtime configuration, and
today I encountered this error.
Now, I have no acces to any information in mysql realtime
Yes it should, I'm not running bleeding edge 1.2 but it isn't an older
branch either.
Mark Johnson wrote:
Jason Fuermann wrote:
I'm not sure about the sippeer stuff, or where they get that
variable. We lookup our info in a database to set it. Also to use
sipcalledrpid you&#
I'm not sure about the sippeer stuff, or where they get that variable.
We lookup our info in a database to set it. Also to use sipcalledrpid
you'll probably need the patch at http://bugs2.digium.com/view.php?id=6643 .
Rob Schall wrote:
Here is what I have in my extensions.conf file now. Trustrc
try actually setting the rpid in the dialplan using
sipcalledrpid(name,number)
Rob Schall wrote:
I set both the trustrpid and sendrpid to "yes", but the calling phone
still doesn't show the caller id of the person they are calling.
Jason Fuermann wrote:
check out rpid
Mar
check out rpid
Mark Johnson wrote:
Rob Schall wrote:
This might sound like an odd question but here it is anyways...
We currently have Polycom 501 phones. We have Asterisk with
Realtime/MySQL running for our SIP/Voicemail/Extensions. When one phone
dials another, the receiving end does i
17 matches
Mail list logo
