Hi all,
I am new to OpenSER/SER and I am trying to save acc data in mysql database
but I keep getting an error whenever I dial the 976 prefix number. What do
I need to do, to configure the database as there is a table for acc, and i
am not quite sure of what to do now.
Here is a snippet of my openser.cfg file relating to the area in question:
if (uri=~"^sip:919(00|76)[0-9]{7}@") {
acc_db_request("403 - 900/976 Disabled", "acc");
xlog("L_INFO", "Cancel attempt to call 9xx number (not 911),
db recorded\n");
exit;
};
Here is the .cfg error displayed on the terminal when calling 9-1 and then
the 976 prefix:
2(12906) ERROR:acc:w_acc_db_request: DB support not configured
2(12906) Cancel attempt to call 9xx number (not 911), db recorded
3(12908) ERROR:acc:w_acc_db_request: DB support not configured
Any help greatly appreciated.
Thanks,
Tracy
_________________________________________________________________
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Hi,
Sorry for my ignorance. I tried to stop SER, which is installed at RedHat
Linux 9, however there is no ser.pid file(in /proc/ or /var/run/
directories). Can anybody suggest me a good way to stop SER so that I don't
need to restart the machine to stop it?
thanks!
Hi,
In my test setup, I have Openser as proxy and SEMS as announcement server.
In Openser config I'm sending forking invite to SEMS using append_branch for
every call made. For example when User1 dials User2, Openser send INVITES to
both SEMS and User2, while the User2 phone rings, announcement is played
from SEMS to User1 using the early media 183 session progress message. Once
the User2 answers the call, a CANCEL message is sent to SEMS and normal
conversation happens between User1 and User2 which is expected.
The problem which I'm facing is that when User2 doesn't answer the call or
if rejects the call, Openser never sends CANCEL to SEMS, so the annoucement
is being played continously to User1 and User1 is unaware of the remote
party status. How can I configure the Openser script to handle such
unanswered or call reject cases and send CANCEL to SEMS and inform User1
about the unanswered or cancel status?
Does it cancel 183 progress only in case of 200 OK from other invite
session? how can we handle others like 408, 486 etc?
I'm not pasting my Openser.cfg here because it is the basic sample script
with just append_branch("SEMS") in case of INVITE.
Did anybody try out such scenarios? Any help would be appreciated.
Thanks,
~Vamsi
> Hello i am using the ser 0.8.12, it works fine, right now i run the linux
> comand ps -aux , to see the list of proces that are runing and i see that
> i have SER proces like the next, can you said me if it is correct or i
> have a problem with my ser or is a problem with the version of the SER,
> because sometimes the SER needs to reset because don't respond, if you see
> the last SER reset tht i need to made was August 25.
>
>
> USER PID %CPU %MEM VSZ RSS TTY STAT START
> TIME COMMAND
> root 24416 32.4 0.7 38668 7456 ? R Aug25 5547:04
> /usr/sbin/ser
> root 24421 32.4 0.7 38684 7468 ? R Aug25 5537:54
> /usr/sbin/ser
> root 24426 32.4 0.7 38664 7484 ? R Aug25 5541:14
> /usr/sbin/ser
> root 24431 32.4 0.7 38652 7468 ? R Aug25 5543:54
> /usr/sbin/ser
> root 24437 0.0 0.8 38676 9056 ? S Aug25 5:56
> /usr/sbin/ser
> root 24439 0.0 0.8 38756 9064 ? S Aug25 6:10
> /usr/sbin/ser
> root 24444 32.4 0.7 38684 7456 ? R Aug25 5551:00
> /usr/sbin/ser
> root 24450 32.4 0.7 38664 7472 ? R Aug25 5539:39
> /usr/sbin/ser
>
> I ran the vmstat 1 10 too and i believe is not normall the processing,
> can you help me?.
>
> r b swpd free buff cache si so bi bo in cs us sy
> id wa
> 6 0 27440 17824 168972 649488 0 0 1 2 0 2 0 0
> 1 0
> 6 0 27440 17844 168972 649488 0 0 0 40 130 480364 12 87
> 1 0
> 6 0 27440 17844 168972 649488 0 0 0 0 118 475900 11 89
> 0 0
> 6 0 27440 17840 168972 649488 0 0 0 0 120 481403 11 87
> 2 0
> 6 0 27440 17840 168972 649488 0 0 0 0 114 483715 12 86
> 2 0
> 6 0 27440 17836 168972 649488 0 0 0 36 134 476488 12 87
> 1 0
> 6 0 27440 17840 168972 649488 0 0 0 64 137 482903 14 84
> 1 0
> 6 0 27440 17840 168972 649488 0 0 0 68 148 473984 14 86
> 0 0
> 6 0 27440 17840 168972 649488 0 0 0 64 143 476887 12 88
> 0 0
> 6 0 27440 17840 168972 649488 0 0 0 52 147 476060 12 88
> 0 0
>
>
Hello everyone,
I have multiple Asterisk systems registered to an OpenSER server. Each
of them have several different SIP accessible extensions. Using
alias_db (or something), I need to be able to create contacts that are
reachable from the OpenSER server to individual extensions on each
Asterisk system. For instance:
SIP Phone -> Asterisk System -> OpenSER
The SIP phone is registered to the Asterisk system and has an extension
(100).
The Asterisk System is registered to OpenSER and has a usrloc entry
(asterisk).
I would like to setup an alias like this:
asterisk-100
This alias should be able to dynamically get the location of the
Asterisk system from usrloc and reach that extension directly, basically
forming a SIP uri like this:
sip://100@[asterisk server ip address - loaded dynamically from usrloc]
That way, if I were to do this:
External SIP Phone -> OpenSER -> Asterisk System -> SIP Phone (ext 100)
I could "dial" asterisk-100 from that External SIP phone and get
connected to the Asterisk System at extension 100.
This would be similar to the following (in Asterisk land):
[sip-in]
exten => asterisk-100,1,Dial(SIP/asterisk/100,20)
exten => asterisk-100,2,Congestion(10)
The most obvious way to do this is create multiple subscriber entries
for each Asterisk system and one register => line for each entry in
Asterisk. I would rather not have to do that!!!
Is this clear enough? What do you think?
Thanks!
--
Kristian Kielhofner
How can I prevent a dialled username (without dialling the domain) to
ring a phone registered with that username?
I pretend to use a multi-domain architecture where apart from aliases of
users, only dialling with username@domain will ring registered phones.
Thanks,
Ricardo.
I know, not *directly* openser related, but hey, it is tangentially related
I've got a scenario where some of my users want to 'hard-wire' their fones so that when their fone goes off-hook, it automatically sends an INVITE to a different (pre-specified) fone.
And yes, they don't want to have to press a key on their fone (don't ask. i *know* its dumb. but hey, its what they want...)
Has anyone seen *anything* along these lines?
Any fones that do something similar?
cheers
--
*******************************************
Mahesh Paolini-Subramanya (703) 386-1500 x9100
CTO mahesh(a)aptela.com
Aptela, Inc. http://www.aptela.com
"Aptela: How Business Answers The Call"
*******************************************
Hi Bogdan,
I think its a very good idea, there are many people using OpenSER for their SIP solutions around the world including us
here in Brazil.
Unfortunately won't be possible for me to get VoN Berlin but I'm going to VON Boston on next 9-11 (uhhhh), perhaps I
could meet the OpenSER members that are visiting or passing by Boston there or in another place.
Does some one is going to VoN Boston and wants to have a beer with others OpenSER members?
Best regards,
--
============================================
Rodrigo P. Telles <telles(a)devel.it>
IT Manager
Devel-IT - http://www.devel.it
IVOZ # 1029
Ourinhos: +55 14 3324-1200
São Paulo: +55 11 2124-7474
============================================
>
> Hello everybody,
>
> It's more than one year since the OpenSER project started. A large community of users and developers has been formed
around OpenSER. It's a very active community and one of the most important goals of the OpenSER project is to keep the
users community as synchronized as possible with the developers and management groups.
>
> During past time we met a lot of people that were happy to see us or we were happy to see them. So, we came to the
idea of organizing a OpenSER Summit where OpenSER fans (all types of fans) should meet to share their experience, to
discuss about future ways of developing the project or just to personally meet people from the mailing lists. The idea
is to have people talking about how they are using OpenSER in different contexts and which were/might be their needs,
development people talking about possible solutions and about their ideas of enhancing OpenSER, etc...
>
> We were thinking of taking advantage of the upcoming VoN in Berlin in the beginning of November and to organize the
Summit within the same time frame - it will be more convenient for people to travel for two events at the same time.
>
> We would like to have some feedback from the community - how many people are interested in such a meeting, what will
be their primary interests.
>
> Any other OpenSER-related ideas are most welcome!
>
> Best regards,
> Bogdan
Hello All,
I am new to OpenSER/Ser and have learned a lot and received a lot of
assistance from all of you. Anyway, I would like to express my gratitude by
extending Free Dynamic DNS services for anyone interested in utilizing a
dynamic IP address in a static capacity while learning and using your
OpenSER/SER sip router and/or Asterisk server. If interested go to:
http://www.t4tm.net.
While visiting the site review the details regarding the FREE Dynamic DNS
Service, signup, add my name servers where you registered your domain name,
download the dynamic IP updater client and that's it.
I hope you all enjoy this service.
Tracy
_________________________________________________________________
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Hello to all
can someone recommend me a nice SIP client with video for windows??
I tried X-Lite 3.0 but it's a lousy piece of software.....
Does someone knows about a better software?
Regards
Joao Pereira