Displaying 20 results from an estimated 2000 matches similar to: "Another * crash"
2003 Nov 06
2
Asterisk and SIP Proxy on same machine?
Hi
Is it possible (or recommended) to run both Asterisk and
say SER on the same physical machine? How about port conflicts?
Maybe the easiest way is to change the default SIP port on Asterisk?
But how will that work if I register some SIP accounts directly
from asterisk (like my SIP provider) but then wanna dial outbound
pure SIP calls via my SER... Has anyone got a functional system like
this
2003 Oct 22
1
Placing SIP calls to other SIP domains?
Hi!
Does * do DNS-lookups when outgoing calls are placed to a different
SIP domain? Can I call from <sip:1000@mydomain.com> to <sip:2000@remote_domain.com>?
Can * work as a regular SIP proxy in that aspect?
Can * handle SIP URI:s that are complete SIP URI:s (sip:user@domain) instead
of numbers only? Or should I run a SIP proxy on a different machine to handle
pure SIP requests and let
2003 Oct 25
6
cdr_mysql.so
Can anyone give me presise instructions on how to compile cdr_mysql.so?
When I initially installed asterisk on the system, I didn't have mysql
installed. Since then I have installed mysql, created the database and
table structure for cdr_mysql and placed the appropriate settings in the
cdr_mysql.conf file. However when I do a show modules at the CLI I cannot
find cdr_mysql.so.
2004 Jan 13
5
linux journal article on asterisk
For anybody who didn't know there is an article on asterisk in February's
Linux Journal.
AJ
2003 Aug 17
4
Grandstream Budgetone
Does anyone know what the Grandstream Budgetone is going for $$$ in the
US? I didn't immediately see pricing on the phones page.
AJ
2003 Nov 25
1
SIMPLE support in Asterisk?
Hi
Is there any work being done on implementing IM/SIMPLE support
for SIP on Asterisk? Like a presence server?
rdgs,
/Staffan Kerker
2003 Oct 27
3
OT Vonage soft phone
In taking a cursory browse at Vonage's site today, I noticed they are now
offering a soft phone. Has anyone had any experience using this? And does
this possibly open new opportunities for using Vonage with Asterisk? Just
thinking outloud on the list, soliciting thoughts and experiences from
others.
AJ
2003 Jun 15
3
Reminder paging for voicemail (?)
Is there a way to configure voicemail to do reminder paging? I would like
to configure some voicemail boxes to send an e-mail message to a pager
every 10 minutes until the message is retrieved.
2003 Jun 07
4
Another PRI based question
In speaking to the representative at Verizon, we came to the conclusion
that DID numbers were not the correct solution; however we were told by
Verizon that they could do something called assign individual numbers to
the PRI. What this would in effect do is give us an additional phone
number that we would like to route to a specific extension; however unlike
the DID number, it would not be
2003 Jun 21
2
PRI & BRI question
Greetings all,
As most of you probably know from my previous questions on the list, I'm
still in the newbie category. My question today is pretty brief, as I
told you all a few weeks ago I ordered a PRI from Verizon. I understand
that there is a "B" channel that comes with this. The question is just
what can I use this "B" channel for and how???
Thanks
AJ
2003 Dec 29
4
asterisk crash
Hello all
I just checked out the latest
zaptel/zapata/libpri/asterisk/asterisk-addons from the cvs and ran through
the entire make procedures. Everything seemed to go fine however now when
I attempt to start asterisk, it says ok but it seems to be immediately
crashing. The following messages are displayed in my
/var/log/asterisk/messages file for the time right around the crash:
Dec 29
2003 Jun 08
2
zapata.conf and zaptel.conf
Can anyone explain to me the difference in zaptel.conf and zapata.conf?
I'm trying to get a real clear understanding of them but its getting a
little murky in places. I will be setting up a PBX running asterisk with
2 T100P cards. I will be bringing a 23 channel PRI into one card and
connecting the other card to a Nortell 24 channel FXS channel bank. As I
understand it zapata.conf is
2003 Jul 23
3
how do I do s extensions with PRI
I would like to know how to define the s extension when I have an incoming
PRI line? Currently I have 5 incoming DID numbers. Four of these DID
numbers I have going to specific extensions, the fifth number which is the
main number I wish to go to a background sound where callers can hear
message, get directory, dial extension, whatever. I see that the way to
normally do this would be to
2003 Oct 22
0
SV: Running Asterisk and NAT on the same box?
Hi
I'm running exactly the same setup. Asterisk is running on my FW/NAT/Router
with two interfaces. My local phones are situated behind the NAT and connects
to the outer interface of the */FW/NAT/Router. * is then connected to my
SIP providers (since I'm only using the SIP-part of *, PSTN connection through
my SIP-provider). Works fine!
rgds,
/staffan kerker
sweden
-----Ursprungligt
2003 Oct 31
0
One more QoS question for RH9
Hi
I know this is a bit off topic, but still pretty interesting.
I'm running Asterisk on my Linux router/NAT/FW connected via
cable (1mbit/200kbit) to the internet.
Now, I wanna do local QoS implementation. Just very simple to
give RTP (UDP) highest priority on my outbound interface. So,
whenever I got an ongoing call, the RTP traffic should be handled
first and other data (file transfers
2003 Jun 15
7
VoicemailMain
Hello guys
Is there anyway for me to change the sounds that are presented in
VoicemailMain? For instance, instead of it saying "mailbox", I would like
it to say something like "please enter your mailbox number now". Is there
a way for me to do this?
I also noticed that when in some of the menus, even if I select one of the
announced options it simply repeats the same menu
2003 Nov 15
10
MeetMe problem
Hi guys,
Having a bit of a problem trying to get conference bridges working. In my
meetme.conf file I have the following line
[rooms]
conf => 6000
In my extensions.conf file I have:
exten => 1000,1,MeetMe,6000
My problem is that when I dial into extension 1000 it is telling me "this
is not a valid conference number". Can anybody telling me what I'm doing
wrong here?
2003 Dec 15
2
iaxclients missing calls
Hello All
When I open up iaxcomm, it registers fine with the asterisk server. If I
call into it, iaxcomm will ring; however if I leave iaxcomm sitting idle
for awhile (I haven't figured out exactly how long) it seems to miss
calls. I can see the calls coming in on the asterisk server but they
never ring through on iaxcomm. If I close it and reopen it, it takes
calls again fine. I thought I
2003 Nov 05
2
asterisk nightmare from hell!
Ok for those of you all up in a tizzy over my subject line, please don't
take it literally because I'm certainly not saying that asterisk is the
problem here. I just got a little nightmare problem that I need a bit of
help figuring out. I installed an asterisk system a few months ago for a
client, it has run almost flawlessly with the exception of a few small
glitches. However, I got a
2003 May 15
3
Linux SIP/IX clients
DOes anyone have any good suggestions as to good SIP or IAX clients for
linux? I have set up and am currently testing asterisk in a controlled
environment. I have gnophone running on one of my boxes but the gnophone
site has been down. So I can't seem to fing the IAX and Ix-devel rpms or
the gsm and gsm-devel rpms. So that prevents me from setting up gnophone
on another box. I am