Hi,
I am successful installing the SER 0.8.14 and the latest radiusclient 0.4.3
When I run the SER, its giving the following erorr message:
0(32009) ERROR: acc: can't get code for the Failed attribute value
0(32009) init_mod(): Error while initializing module acc
ERROR: error while initializing modules
I have also went through the documentation and appended the required SIP attributes in the dictionary.
Does any one has faced similar issue or am I mising something here?
suraj
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Hi,
When I have one phone behind NAT, rtpproxy works well.
However if a call goes to two phones (one behind NAT
and one not) and ser (or more precisely cpl-c) does
parallel forking, the call can go through but there is
no voice on either direction. I have a
modparam("cpl-c", "proxy_route", 2) and route(2) calls
force_rtp_proxy().
Has anyone tried this before? Is rtpproxy capable to
handle multiple forked calls?
Thanks,
Richard
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Hi,
I'm having problem install ser-0.8.11_linux_i386.tar.gz , after I run
tar -xvzf <filename> then i try tu run "make all" but the system not
allow to do that with error msg
" make: *** No rule to make target 'all' . Stop. " Can you pls help me
how to make it work?
br
zahari
Hi,
I gone through some previous message of SEMS implementer. I found every one used two SER instance for voicemail integration.
I have used only one SER instace with SEMS for voice integration, but I am getting some trouble with some sip based clients at the time of retrieving greeting message from SEMS.
I would like to know Why the two instance of SER is necessary to integrate voicemail.
regards
koyama
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Hi All,
There is a problem in SER module auth_radius. It authenticate each call
serially not parallely. So SER is blocked to receive the response from the
radius server and can't do other thing at the same time.
How to resolve this problem? Of is there someone who have the best answer?
Michael
Hey ML,
The role of Ser and * are very different. * has been built as a PBX
replacement first and foremost and so handles VOIP --> PSTN integration,
has a VM/conference/meetme etc server (although SEMS is becoming more
and more advanced all the time) and performs transcoding etc. However,
it's SIP implementation is difficult to work with for a high volume of
users, it doesn't handle presence, doesn't support multiple logons by
the same client etc. SER is a fantastically performing high volume SIP
proxy/redirect/registration agent etc with very flexible call processing
and route processing logic. Many people on this list use Ser as the
point of contact for all SIP UA's, and then forward to * or a
commercial product to do PSTN gateway relays.
Hope this helps
Dave
________________________________
From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On
Behalf Of muralikrishnan lakshmanan
Sent: 04 August 2004 05:07
To: serusers(a)lists.iptel.org
Subject: [Serusers] how to use ser
Hello all,
Im new in this group. Im using asterisk. I don't know where I have to
use "ser". What is the part of "ser" in asterisk ? How to use "ser"? if
anyone knows please mail me.
Thanks in advance
Muralikrishnan L
Software Engineer Trainee
Calsoft-Elnet Software City
Taramani
hello friends,
iam spending too much time
if any body got this
please help me
with regards
serdiehard
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Hello all,
Im new in this group. Im using asterisk. I don't know where I have to use "ser". What is the part of "ser" in asterisk ? How to use "ser"? if anyone knows please mail me.
Thanks in advance
Muralikrishnan L
Software Engineer Trainee
Calsoft-Elnet Software City
Taramani
Thank you Jamie for the great tip! It did work very well :-) The sip gateway dropped the first 183 for unknown Call-ID. sip gateway didn't send BYE msg back after second 183 msg, call was successfully connected.
SER is really powerful enough that can almost do everything you imagined ^_^
Regards,
----- Original Message -----
From: Jamie Yukes <jyukes(a)gmail.com>
Date: Tue, 3 Aug 2004 09:00:42 -0400
To: ksabc(a)lycos.com, serusers(a)lists.iptel.org
Subject: Re: [Serusers] How to drop 183 msg
> As a total kludge, you may try changing the packet sufficiently such
> that the sip phone will drop it itself...
>
> replace_all( "Call-ID: .*@", "Call-ID: XXX@" );
>
>
> On Tue, 03 Aug 2004 14:46:15 +0200, Bogdan-Andrei IANCU
> <iancu(a)fokus.fraunhofer.de> wrote:
> > It is not possible to drop a reply from reply_route - you can change the
> > reply, but not to drop it entirely.
> > bogdan
> >
> >
> >
> > ks lf wrote:
> >
> > >SIP termination Gateway send two 183 "Session in Progress" msg back to SER, first one without SDP, second one with SDP ( for remote ring back tone purpose ), but this crashed the caller party sip phone somehow, how can we drop the first 183 msg in SER.CFG? We just need forward the second 183 msg to caller party, but don't forwarding the first one, how to porgram onreply_route part? just use break sentense doesn't help. Thanks your great help.
> > >
> > >Regards,
> > >
> > >
> > >
> > >
> >
> > _______________________________________________
> > Serusers mailing list
> > serusers(a)lists.iptel.org
> > http://lists.iptel.org/mailman/listinfo/serusers
> >
>
> Jamie Yukes
> Global Connect http://www.gc1.com/
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