Displaying 20 results from an estimated 2000 matches similar to: "looking for a link or pdf ot something about opensip/openser and load balancing"
2009 Jan 08
2
Problem incomming from openser
Hi
I have an asterisk 1.6 running, and our provider have an openser on their end.
When I get an incoming call I get this on my end
[Jan 8 14:51:56] NOTICE[16680]: chan_sip.c:16869 handle_request_invite: Call from '' to extension '0840303395' rejected because extension not found.
If I wait approx a minute and try again, the call will go trough.
We don't use REGISTER or
2008 Dec 13
3
SER, OpenSER, Kamailio, OpenSIPS -- what are you using?
One of the above is frequently used to front-end Asterisk.
I used OpenSER to front-end a farm of Asterisk servers and was very happy
with it. The ability to take a box out of service or to route a specific
DNIS to a box for testing rocks.
Since OpenSER has died (I don't care about the
politics/personalities/trademarks), Kamailio and OpenSIPS have risen from
the ashes. What are you using?
2008 Dec 19
0
OpenSer and MYSQL Lookup Queries!
Hi!
Can OpenSer perform some database lookup queries based on dialed number like we can do with Asterisk. Asterisk Can do it and there is MYSQL Function available which allow us to open connection and execute any query to get required results from database, Can we do same with OpenSer or OpenSIP etc.?
Thanks
Regards,
Muhammad Zulqarnain
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2009 Feb 24
1
Incoming call
Dera All,
I have the following scenario,
A customer dial a DID number...The call is routed to a PSTN GW that send the
call to asterisk...
On asterisk I created an AGI Script that send the call to an extension
registered on OpenSIPS server...
The extension is ringing successfully, but as soon as I accept the call on
OpenSIPS side the call is hangd up...
I checked rhe SIP debug and it seems that I
2009 Jan 12
1
bug(?) bandwidth problem
hi
i am using asterisk 1.4.22
ubuntu 8.4
i have two Ethernet one for ssh and other one only for voip calls
when i start a call using originate in the manager or the cli
in the voip Ethernet i get something like 4Mbits/sec of traffic only 1 G711
call.
if i start the call using a soft phone everything is normal.
any idea?
thanks
--
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2009 Mar 26
4
out of the box or do it your self?
hi
i want to ask for your opinion what is better for a call center 100 current
calls and other 200 current calls
make the server step by step or use a auto install cd like asterisk now,
druid elastix ....?
and why?
Thanks
David
--
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2009 Mar 19
3
Hardware suggestions
Hi,
I`m looking for reliable and redundant hardware for Asterisk. I`ve been
leaning towards buying one of these (HP 360 G5 with everything as redundant
as possible), which I know will be good enough for a few months before
needing to upgrade:
http://h10010.www1.hp.com/wwpc/us/en/en/WF05a/15351-15351-3328412-241644-241
475-1121486.html
Questions:
1) Any reason why I shouldn't? (bad
2009 Jan 25
5
soft phone
hi
wich soft phone do you recomend but i need this feature it must ask for user
name and password when it start.
i know xline and zoipper but they dont have that i can acomplish this whit
twinkle but i need it for Windows :-(
any ideas?
thanks
--
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2009 Mar 25
1
predictive dialer
hi
wich predicitive dialer are you using and wich one do you recomend?
a link to the project/product and a link to a how to will be VERY
apreciated.
Thanks
David
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2008 Dec 02
1
hi from argentina
hi
this is mi first email and just for say hello.
David
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2008 Dec 02
2
callcenter supervisor system
hi
i need an open source callcenter manager system like queuemetrics but
opensource any one know any?
i prefer to search before start a new one
thanks
David
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2008 Dec 09
1
about trasncoders
hi
where i should load the module for the trasncoder wctc4XX (lspci shows
TC400P)
thanks
David
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2009 Jan 15
1
R2
hi i am reading about new codecs and new stuff to be added to asterisk. (and
i say thanks to all the guys who are working to add all the new features).
will be R2 added to the main core of asterisk like ISDN?
Thanks
David
--
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2009 Feb 18
1
xorcom hardware good idea?
hi
i am thinking to use some xorcom hardware i only used digium hard...
what do you think?
it work ok? what about echo?
it is plug and play only reload or i need to restar asterisk ?
i understand the drivers are in the trunk of asterisk, is this rigth?
have any one a conf file of a xorcom device?
any other important comments are welcome.
Thanks
David
--
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(='.'=)This is
2009 Apr 11
2
[OFF TOPIC] wich virtualization solution to use?
hi
there are a lot of virtualization solution out there and every one "is the
best" and has some pro and some cons...
wich one do you recomend?
the idea to isolate diferents servers asterisk apache ... it is a good idea?
sorry for the off topic but here is a place where are a lot of linux gurus
Thanks
--
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2008 Dec 03
2
asterisk ooh323 avaya (URGENT!!!)
hi
sorry about the urgent but it is urgent
i have problems configuring a connection between asterisk and avaya using
H323.
the module i am usign is ooh323
what do you need to help me?
and any tip or hint?
thanks!!!
David
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An
2009 Jan 29
3
32 bit server is ok?
hi
i have a 32 bits server asterisk 1.6 will work ok in it? is just for a demo
server no more than 4 to 10 calls at the same time.
and a tdm board.
waht do you think?
thanks
David
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2009 Feb 26
1
incoming call problem
Dear All,
I have created an inbound context in SIP .conf that forward incoming call to
opensips server...The problem appears as soon as I enable t38pt_udptl =
yes...The Asterisk negotiate the SIP session with OpenSIPS without adding
voice codec to INVITE packet...It just contains T.38 protocol...When
t38pt_udptl is disabled everything looks OK and Ulaw is negotiated with
OpenSIPS and cal
2008 Dec 11
1
top posting again [was: Re: CDR Design]
Well, it seems this opened one large can of worms.
Anyway, just to repeat my previous plea - and to echo David's request - can we please stop all this 'top post' rubbish and move on with our lives?
Thanks and Merry Christmas
Andy
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of David fire
2009 Feb 28
2
No rtp activity
Hi all....
I'm using asterisk for making PSTN calls from extensions registered on
OpenSIPS...In peak hours ,number of calls Increase dramatically to a non
logic number..When checking the calls using asterisk CLI I saw a lot of
calls in ringing status and after 300s(rtphold timeout), asterisk release
all calls...I checked the log file and found..
[Feb 28 11:34:14] NOTICE[19197] chan_sip.c: