I've been trying to make my
scripts look a little neater.
I'm having problems with xlog specifically.
I have these lines that wrap 2 or 3 times which
are real ugly. For example:
if(loose_route())
{
xlog("L_NOTICE", "LOOSE: time_t=%Ts, call_id=%ci, cseq=%ci,
contact=%ct, from=%fu, fromtag=%ft, to=%tu, totag=%tt, method=%rm,
ruri=%ru, messageid=%mi\n");
t_relay();
};
I just tried this:
xlog("L_WARN",
"TOOMANYHOPS: %Ts call_id=%ci\
cseq=%ci\
contact=%ct\
from=%fu\
fromtag=%ft\
to=%tu\
totag=%tt\
method=%rm\
ruri=%ru\
messageid=%mi\n");
That looks better, but I get ^I (that is TWO characters)
in the log output .... presumably all of the
tabs.
Is there any 'HERE' document functionality, or a way
to make my long lines look neater?
---greg
Hi all,
Anyone have any suggestions for reasonably priced tools for monitoring SER
SIP server availability? Ideally I'm looking for something that can collect
traps, poll, and perform application monitoring of the SER SIP service
state.
Thanks,
Paul
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Hi, I have just finished installing ser and the related radius modules and clients.
I would like to know how to configure ser.cfg so I can use radius authentication and PSTN calling.
Thanks
Johnny Lum, Programmer
VoIP, ADSL, Wireless Hot Spots, 56K Roaming, Call Center, Server Hosting
www.aebc.com Sales: 604.288.1088 Support: 604.279.9078 Fax: 604.207.0155
First time caller, long time reader...
I'm looking to leverage * voicemail application. I see other posts but
never a firm answer. My question is:
How do I program SER to recognize "unavailable and/or busy" (similar to
how * does this)? Then, if this
state is recognized, I want SER to re-direct the call to *. If I can
figure this out, then I believe all I have to do with
* is set it up for no rings and do the normal "exten" commands. Seams
pretty simple...
TIA and happy new year,
Mike N.
Hello !
I have problem with registrar module
I have 2 different version of files registrar modules:
one - last from cvs and second from original 0.8.12 source
In version from cvs I don't see "use_domian" option declaration
It is correct ? Maybe this option was replaced by something another ??
Thanks
Andrzej
Hi All
Just installed Serweb, but i can't log in. I tried admin/heslo, ser/heslo as
username password combinations but no good. Both users exist in the
database...so i don't know what the issue is. Does it use different username
and password. Please help.
Thanks
Andy
Hi All
I've got two MS messenger client on 5.0, when one client is logged
in and 2nd one logs on...the 1st client doesn't update the status that the
2nd client has logged on and vice versa. I've seen some messages about this
that other people also had problems with it but i didn't see if any1 was
able to fix it.
Please help
Thanks
Andy Singh
Hi,
I am totally new to SER.
I have installed it and have the following set up:
pingtel phone --- SER ----- pingtel phone
I somehow dont see the sip headers in the tcpdump/trace.
Any suggestions.
Thanks,
-Darshan
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Hi,
I have a few questions about SIP and SER - feel free to blast me if
they've been asked and answered before - I have tried to find my answers.
I'm fairly new to SIP but not to VOIP (I'm sorry, most of my life is spent
with CCM). I've played with SIP a bit but not really indepth.
The Scenario:
I have a few SIP phones at home (mostly Cisco 7940s and ATAs etc) and I want
to build a residential gateway that will allow me to do the following in
terms of "external" stuff:
Route outgoing calls to particular proxies that require (usually) digest
authentication (eg. iconnecthere for international dialing, fwd, etc) - the
thing I don't get is the authentication side of things - ie. the phone won't
know it needs to authenticate so the SIP proxy must provide this ...
"Register" with external proxies so that (eg. FWD and iptel.org ) and so
that it knows to send calls to my FWD number to my proxy
server so that my proxy can route to my internal phones and/or send the call
on to whereever I may be.
I've read through the SER documentation and I can see how to rewrite URIs
but I was confused about how digest auth works with that.
Can I actually do the SER registering with other proxies? I realize this
really isn't the role of a SIP server. Is there a better way of doing this?
Have I really just missed the point somewhere along the line? (I'm quite
prepared to accept that I may have ..)
I've had a play with getting Asterisk to do this and got it to register etc,
but got a bit stuck with the whole media/codec side of things (and basically
gave up prematurely!) I just wanted to do some SIP things!
My idea with doing these things is to see if I can figure out a bit of a
"packaged" up version of SER for those of use who want to integrate things
like FWD/iptel.org etc into our "normal" telephony environment at home.
(Anyone know what the easiest way of getting an FXO port at home is?)
--
Matthew
--
Matthew(a)Moyle-Croft.com | mmc(a)mmc.com.au | mmc(a)206gti.net
http://www.Moyle-Croft.com | http://www.mmc.com.au | http://206gti.net