Hi!
The log messages are stored by default in /var/log/messages, but I would like to know which way to obtain only
the log messages that I wrote in the configuration file, without all the debug messages. On the other hand, is
there a way to know exactly how the debugging mode acts (which level corresponds to each 'd' in the debug
option), when I make an instance of ser, and what relation it has with the debug option inside a configuration
file?
Thank you,
Blanca
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Hello,
We have the following setup. A proxy forwarding all
the calls to our SER and the SER replies to the proxy
with the appropriate location where the call should be
routed (statless). So the first call could go to A and
the next call could go to C and so on.
I want to be able to do some load balancing and smart
routing. By smart routing I mean that a program
running on the same box as the SER will keep track of
the health status of all the destination boxes like A,
B, C stated above. So when an invite comes from the
proxy, How can I call an external let's say "c"
program which will return the destination address the
SER should should return to the Proxy to use.
I have looked at exec_dst but the admin guide provides
only exec_msg examples. I have a feeling that I could
use it, but not sure.
I hope that I explained the setup properly. If anyone
has any questions please ask.
thanks in advance for your help.
Jignesh Gandhi
Software Engineer II
Jignesh.Gandhi(a)glenayre.com
Hello,
If I run the "serctl moni " as 'roo' , it executes not a problem.
But , if I run it as a not 'root'' , for example ,as "murat", I have
following errors!.
******************************************
Error opening ser's FIFO /tmp/ser_fifo
Make sure you have line fifo=/tmp/ser_fifo in your config
**************************************************
Even I have ' fifo="/tmp/ser_fifo" ' lines on my default 'ser.cfg'
file.
Question <1>: What is the problem about this?
Question <2>: I have some SIP sicripts, and how can I run it?.
Thank you.
Hello,
In the admin documentation of ser, it says that all messages greater than 300 are considered failures when using t_on_failure.
In our scenario, we would like the ability to forward to a voice mail on 4xx failure, but not on 302 failure. We are using XTEN Pro softphones and want to be able to allow users the functionality to forward to another SIP url (uses 302). But when ser is seeing 302 (and t_on_failure is active) it considers it a failure and immediately runs failure route block (to voicemail). So our users cannot using forwarding to URL.
An example would be ... When I'm out for lunch I would like to forward my line to my secretary first before going to voicemail.
We saw some mention of t_check_status ... but ser wont start with that in the cfg. We are using ser v.0.8.12 rpm. Is that in a unstable version only. If some could post how to accomplish this it would be appreciated.
Thanks
Kurt
Hi all,
I'm now trying to setup some SER servers with location based routing ability
which can handle the traffic according to the source. Before getting to that,
I want to know if the media proxy of SER can be interconnected together to
form a larger adaptive server? The scenario is as follow :
SIP phone <=> Media Proxy A <=> Media Proxy B <=> SIP phone
From the Media Server readme, I just found out that the dispatcher.py can
load balance the traffic by using the DNS SRV records. Any people got some
ideas? Thanks a lot!
Kenny Lam
SIP Application Engineer
Deltapath Commerce & Technology Limited
---------------------------------------
SIP By Deltapath!
www.deltapath.com
As you guys seem to have some experience with this area perhaps I will
re-ask my question this thread..
I have SER-->asterisk forwarding working fine now, and I have divert on
unavailable working. However, I have two outstanding problems:
(1) If a user is called with their alphanumeric ID instead of their
numerical alias, * does not pick up the call. This is as expected, as
the dial pattern in * is _[1-9] [0-9] [0-9] [0-9]. However, it must be
fairly common to call people with their email addresses for example...
so how can I make ser pass the alias to * instead of the alpha URI?
(2) If a user is offline, I get a 404 immediately, instead of anything
else - for example diverting immediately to vm. I don't quite
understand this at the moment.. as I have the t_on_failure set up before
the location lookups... does the t_on_failure not catch 404 failures?
Any thoughts would be very much appreciated.... anything I can provide,
please let me know...
Thanks again everyone,
Dave
-----Original Message-----
From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On
Behalf Of GR S
Sent: 28 July 2004 21:31
To: jon(a)bostontech.com
Cc: serusers(a)lists.iptel.org
Subject: Re: [Serusers] Asterisks to ser to asterisk (voicemail)
Hello,
--- jon(a)bostontech.com wrote:
> yes, i know that this will work, but the issue is that not every sip
user
> who is called has voicemail. I want SER to determine who should be
> rerouted or who shouldn't.
Still you dont need to worry. Let all un-attended calls come back to
Asterisk. It will drop the
calls if it can't find a mail box number. May not be the right method,
though.
> -Jon
>
>
>
>
>
> GR S <gr_sh2003(a)yahoo.com>
> 07/28/2004 04:02 PM
>
> To: jon(a)bostontech.com, serusers(a)lists.iptel.org
> cc: oej(a)edvina.net, andres(a)telesip.net
> Fax to:
> Subject: Re: [Serusers] Asterisks to ser to asterisk
> (voicemail)
>
>
> Hello,
>
> --- "Olle E. Johansson" <oej(a)edvina.net> wrote:
>
> > Andres wrote:
> >
> > >
> > >>
> > >> My question is, is there any way to have ser receive a call from
> > >> asterisk and then reroute it back to the same asterisk server
without
>
> > >> getting a "loop detected" error?
> > >>
> > > Aren't you seeing this "loop detected" on the Asterisk CLI?? If
so
> > > should post this in the Asterisk list instead. We know this
happens
> > > anytime you try to loop a call back to Asterisk, but its Asterisk
who
> > > complains. Not SER.
> > >
> > Answer from the Asterisk users list :-)
> >
> > No, there's not a way to do it, but maybe to issue a 302 redirect.
> > Haven't tried it, but that may work.
> >
> > The Loop Detected stuff is annoying, yes.
> >
> > /O
> >
>
> >From a great fan of Asterisk and SER :-)
>
> I am not sure about the exact problem, but there is another way to
acheive
> this. You dont need to
> re-route the calls back from SER to Asterisk. Set a timeout in the
> Asterisk Dial statement and
> forward the call to SER. If the callee attends the call, you can talk,
and
> if not, make Asterisk
> forward the call to voicemail when it hits the timeout. I have this
> feature enabled in a local
> system running SER on 5060 and Asterisk on 5070.
>
> Best Regards,
>
=====
Girish Gopinath <gr_sh2003(a)yahoo.com>
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(i'm resending this because I don't think it got posted right...)
I'm running into a little problem with forwarding.
This is what's happening:
I have a PBX connected into an asterisk box, allowing the PBX users to
make SIP calls from their regular phones.
I'm also running a SER box with my own sip domain. I have the SER box send
unavailable calls to voicemail on the Asterisk server.
Now, when someone uses the PBX phone to call a sip user from my sip domain
and they are unavailable, i'm getting a "loop detected" from ser.
PBX -> Asterisk -> Ser -> Asterisk
I can see why i am getting this (since from ser's perspective it's doing a
loop), even though once Ser forwarded the call to asterisk, it won't
receive it again, since the incoming call is in a different context.
My question is, is there any way to have ser receive a call from asterisk
and then reroute it back to the same asterisk server without getting a
"loop detected" error?
Thanks
-Jon
Its ok.. finally found out what it was!
D
________________________________
From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On
Behalf Of Dave Bath
Sent: 28 July 2004 20:47
To: serusers(a)lists.iptel.org
Subject: [Serusers] 0.8.14 headache....
Hey guys,
I am having a small headache with serweb and the latest version of ser..
tried to upgrade, and things went moderately smoothly... the main
problem was forgetting to use the "prefix=..." command when compiling
the source! However, whenever I have upgraded, the database always seems
to be a problem ... I don't know why this is, and the serweb database
fields never seem to be present in the ser_mysql.sh script... perhaps I
have done something very strange. However, I now seem to have a
functioning database and ser install again, and it has decided to start
loading my config again, so all is well. BUT! I seem to have lost
admin permissions, particularly for administering ACL's through
serweb... I can't figure out which combination of database entries to
use.
So far I have 'perms' field in 'subscriber' table set to 'admin'
And I can see I need to put something in the admin_privileges table, but
I'm not sure of the values..
Can anyone who has it working please have a look in their DB and let me
know the values?
Many thanks
Dave
