Rajeev,
I do not use SEMS myself. We use Asterisk for voicemail -- but it can
do conferencing as far as I know. Perhaps you can visit
www.voip-info.org or www.asteriskdocs.org to find out how to configure
Asterisk for conferencing.
You can also check out www.onsip.org to get the Getting Started
document which shows how to do call forwarding with SER.
Regards,
Paul
On 10/5/05, rajeev.khetan(a)cem-solutions.net
<rajeev.khetan(a)cem-solutions.net> wrote:
> Hi Paul,
>
> Ok I downloded SEMS and installed it, but dont know how to use it,
> I think it is related to config file, can you mail me any sample config
> file to
> check conferencing , voice mail, callforwarding etc features.
>
> Regards,
> Rajeev
>
> Paul Hazlett wrote:
>
> >Rajeev,
> >
> >SER itself cannot do conferencing. In fact it doesn't understand
> >anything about RTP streams at all.
> >
> >If you want to do conferencing or voice mail or other sorts of things
> >then you need to look at integrating SER with other software
> >applications, such as Asterisk PBX or SEMS.
> >
> >Regards,
> >Paul
> >
> >On 10/4/05, rajeev.khetan(a)cem-solutions.net
> ><rajeev.khetan(a)cem-solutions.net> wrote:
> >
> >
> >>Hi,
> >>
> >>I want to know wheather ser proxy support conferencing ?.
> >>If it support plz. send the config file for it.
> >>Thanks in advance.
> >>
> >>Regards,
> >>Rajeev
> >>
> >>
> >>_______________________________________________
> >>Serdev mailing list
> >>serdev(a)lists.iptel.org
> >>http://lists.iptel.org/mailman/listinfo/serdev
> >>
> >>
> >>
> >
> >
> >
> >
>
>
>
Hello,
I wish to send an IM to some UAC from our custom UAS,so while framing the
MESSAGE request , in the Call Id field, whose Call id do i fill? Sender's or
receiver's?
Thanks and Regards.
Abhijit
I just wonder if this is a script error or if it is something else:
I am loading an INVITE timeout using avpops and the OpenSER terminates
the call. The callflow is as follows:
A -------------OpenSER------------B
---INVITE---->
<---100-------
---INVITE--->
<---100------
<---180------
<---180-------
<Inv Timeout delay>
<---408-------
---CANCEL--->
<---487------
---ACK------>
<---200 OK---
<----200 OK---
---ACK------->
---CANCEL--->
<---481------
---CANCEL--->
<---481------
Any comments?
I do not want anyone to fix my problem, just a direction :-)
--
mvh/best regards
Helge Waastad
System Engineer
Smartnet
Hi,
I am trying to use t_on_branch to identify if each branch is behind
the same NAT as the caller. A branch_route is defined and check
dst_uri's address. If any one branch is behind different NAT, a flag
is set. The flag is checked later to call rtp/mediaproxy functions.
But there is a problem. It looks like the branch_route is NOT called
until a t_relay is called. That's way after the flag is checked. If
that's the case, I don't see how branch_route can help any NAT
situation.
Thanks,
Richard
hello,
I use avpradius (table preferences) for call-forwarding.
When i call a subscriber, no problems, when i call an alias, no avp found
for the alias?
What do I wrong?
regards,
Olivier
Hi,
When i am trying to access serweb in http://host/ser/admin/index.php i
get the space to authenticate myself.. but independently from what i
type in username and password, when i press login nothing happens. any
idea?
Thanks in advance,
Jose Simoes
Hi,
I am looking for some input on how to program Click to Dial web application.
If someone has written or has some info on getting a sample to look at,
please direct me in the right direction.
Regards,
Viquar Syed
hi everybody,
I've just put my openser 0.9.5 in the production environment and it works well. For now, I jus use a query to extract the CDR as "call_date", "orig_number", "termination_number", "duration" and "src_ip".
but when the volume is very large I get the following problem. The scenario is as follows.
call_date | orig_number | term_number | duration | src_ip
2005-09-25 20:45:30 | 203@my_sip_server | 56924536@my_sip_server | 14:00:30 X.X.X.X
2005-09-25 20:45:30 |56924356@my_sip_server | 43234123@my_sip_server| 4:00:30 X.X.X.X
I get the same number as the origination number which was the destination number some moments before, when in reality it is not possible to call from that particular number.
But this only happens when the calling volume is really large. Does anyone have any idea regarding this.
If you want to see the query I can do that.
pls share if there is some better query to generate the CDR..
I have also downloaded the latest version 0.10.x from CVS head. Can somebody tell me how stable is it for the production environment.
thanx
jayesh
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Hi All
We came accross a typical problem, Let me describe you through
SIP IP Phone -----------> SIP Proxy --------------> MediaGateway -----------PSTN Network
IP address 'A' IP address 'B' IP address 'C' Phone No. 'xxxx'
Now when we make a call from SIP IP Phone to a PSTN network say xxxx no (an IVR no.). call gets establish, but in the mean while if the network of SIP IP Phone goes down, call is still going on.
Is there any method through which we can check if the IP phone is UP (on Network) or not, if not then call gets disconnected.
Ritesh Jalan
