hi,
I'm trying to set up 2 SER and 2 Asterisks boxes using a bigip switch to do
load-balancing. I'm using Asterisk as a voicemail application only and have
successfully integrated SER with Asterisk without the switch. But when I try
to use the switch as a load-balancer, I get lots of NAT problems. Does anyone
know how to setup the switch and SER/Asterisk properly?
Thanks,
Jack
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Hi,
I installed ser 0.9.6 on fc4
Compiled and installed with all modules but vm.so module is missing.
Looked at the source modules directory, there is no vm directory or file.
I installed sems from cvs
Sems is working
Also
Do i have to add down lines to ser.cfg
loadmodule "/home/srouter/sip_router/modules/vm/vm.so"
modparam("vm", "auth_db|domain|uri_db|usrloc", "db_url",
"mysql://ser:heslo@localhost/ser")
modparam("vm", "email_column", "email")
modparam("vm", "subscriber_table", "subscriber")
modparam("vm", "user_column", "username")
modparam("vm", "domain_column", "domain")
if(method=="ACK" || method=="INVITE" || method=="BYE"){
if (!t_newtran()) {
log("could not create new
transaction\n");
sl_send_reply("500","could not create new
transaction");
break;
};
t_reply("100","Trying -- just wait a minute !");
if(method=="INVITE"){
log("**************** vm start - begin
******************\n");
if(!vm("/tmp/am_fifo","voicemail")){
log("could not contact the
answer machine\n");
t_reply("500","could not
contact the answer machine");
};
log("**************** vm start - end
******************\n");
} else if(method=="BYE"){
log("**************** vm end - begin
******************\n");
if(!vm("/tmp/am_fifo","bye")){
log("could not contact the
answer machine\n");
t_reply("500","could not
contact the answer machine");
};
log("**************** vm end - end
******************\n");
};
break;
};
if (method=="CANCEL") {
sl_send_reply("200", "cancels are junked here");
break;
};
sl_send_reply("501", "method not understood here");
Toygun
Hi folks,
I wanted to route sip requests based on user id with
it's appropriate Gateway IP address which is stored in
Mysql database , is this possible?
Database structure:
userid,gateway
617978XXX 64.80.XXX.XXX
78189XXXX 64.100.XXX.XXXX
example: if someone call 617978XXX he will be routed
to it's approriate gateway
at present I am using static route like:
if (uri=~"^sip:617978XXX@") {
rewritehost("64.80.XXX.XXX");
}
if (uri=~"^sip:78189XXXX@") {
rewritehost("664.100.XXX.XXXX");
}
etc,....
I am afraid the config file gonna get larger if I add
more than 500 users(DIDs), I am looking for an
alternative to store DID numbers in database (Mysql or
text) and
route them accordingly with the appropriate Gateway
address.
Thank you so much and I will appreciate if you can
help me out!
Ben
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Hi all,
i want to introduce following in my configuration:
CallerA calls CallerB -> ringing
CallerC calls CallerB -> reject
during this transaction, without that CallerB picks up the phone,
CallerC should not be able to call CallerB during this transaction,
After this transactions, CallerC can call CallerB also.
Any ideas how to check if there is an ongoing transaction ?
regards,
Andreas M.
>
> Hi all,
> I have a problem with audio, when the destination of a call is natted and
> the other side is public. This problem is cause because the SDP info only is
> present in the 200 OK and in the ACK message.
>
> I'm using ser (8.14) with mediaproxy (1.2.1).
>
> I have an H323 to SIP translator (and SIP to H323 too) connected to the
> PSTN and to the SIP Proxy (SER).
> H323 is the origin of the call (public client), and SIP UAC is the
> destination (Natted client).
> My proxy is working in state full mode and has a public IP.
>
> The H323 side is using slow start, so when it starts the transaction with
> an INVITE, it will not send the SDP info (it will send it in the ACK
> message).
> The proxy forwards this message to the destination.
> SIP UAC, starts ringing. When we hooks off, the SIP UAC send an 200 OK
> with SDP message to the H323 side.
>
> This is the 200 OK / SDP that the proxy receives from the SIP side.
>
> No. Time Source Destination Protocol
> Info
> 488 28.488811 200.68.89.2 200.68.89.12 SIP/SDP
> Status: 200 OK, with session description
> Message body
> Session Description Protocol
> Session Description Protocol Version (v): 0
> Owner/Creator, Session Id (o): 700600 6015 6015 IN IP4
> 192.168.0.101
> Owner Username: 700600
> Session ID: 6015
> Session Version: 6015
> Owner Network Type: IN
> Owner Address Type: IP4
> Owner Address: 192.168.0.101
> Session Name (s): AddPac Gateway SDP
> Connection Information (c): IN IP4 192.168.0.101
> Connection Network Type: IN
> Connection Address Type: IP4
> Connection Address: 192.168.0.101
> Time Description, active time (t): 0 0
> Session Start Time: 0
> Session Stop Time: 0
> Media Description, name and address (m): audio 23018 RTP/AVP
> 18 8 0 101
> Media Type: audio
> Media Port: 23018
> Media Proto: RTP/AVP
> Media Format: ITU-T G.729
> Media Format: ITU-T G.711 PCMA
> Media Format: ITU-T G.711 PCMU
> Media Format: 101
>
> Following the ser.cfg, this 200 OK / SDP will make the proxy to use
> mediaproxy (use_media_proxy()), because the destination user is natted. So
> the mediaproxy module, will generate a lookup command to the
> proxydispatcher.py.
>
> proxydispatcher[30535]: command lookup 2281401749(a)200.68.89.10
> 192.168.0.101:23018:audio 200.68.89.2 200.68.89.10 remote 200.68.89.10unknown AddPac=20SIP=20Gateway info=
> from:1150316660@200.68.89.10,to:1152464490@200.68.89.10
> ,fromtag:3512844671,totag:7b07717a4
> proxydispatcher[30535]: warning: trying to lookup session with
> non-existent id: '2281401749(a)200.68.89.10'
>
> The proxydispatcher does not recognize that command because it has not
> generate the session before. So, the proxy forwards the 200 OK /SDP message
> without changing the SDP info. When the H323 sides receives that info, it
> thinks that it has to send RTP to the private IP.
>
> Then the H323 sends the ACK with SDP.
>
> Then, we cannot ear audio in both sides.
>
> Here is the problem, because, I didn't receive an INVITE with SDP that can
> create the session into the dispatcher.
>
>
> A is public, B is natted
>
> side A Proxy side B
>
> INVITE without SDP
> ---------->
> 100 Trying
> <----------
>
> INVITE without SDP
> ---------->
>
> 100 Trying
> <----------
> 180 Ringing
> <----------
> 180 Ringing
> <----------
>
> 200 OK SDP (B private IP)
> <----------
> *1*
> 200 OK SDP (B private IP)
> <----------
>
> ACK SDP (A public IP)
> ---------->
> *2*
> ACK SDP (mediaproxy public IP)
> ---------->
>
> ......................................................
> RTP from B
> <----------------------
> *3*
> ......................................................
>
>
> *1* The proxy must replace the private IP of the SDP, with the ip of the
> mediaproxy. The looku prequest does not work, because no session was found
> *2* the proxy generates the session in the mediaproxy. So B thinks that
> the RTP must go to the mediaproxy. But A never knows the mediaproxy address.
> *3* RTP from A goes to anywhere, because A does not know where it is the
> private adrress of B.
>
>
Do you know a solution to this? Is the sequence of the message all right?
Regards.
Carla
Hi.
As I had no answers to this question, here it is again.
Till now I was happy using SER handling communications between UAs using
nathelper and RPT proxy.
Now I changed the ser.cfg script to the one with PSTN support and none of
the UAs get registered.
I am getting a 403 Forbidden.
Analyzing the script I understood that this 403 was coming from the Call
Type Processing Section so I changed
if (!is_uri_host_local()) {
if (is_from_local() || allow_trusted()) { and so on
to
if (uri!=myself)
and now the UAs are registering again.
But of course I can't make calls.
So I would like to know how this< is_uri_host_local and is_from _local>
work and what should I do to make them allow my UAs to register.
I am attaching the ser.cfg that doesn't work.
Thanks in advance
Juan Ferrari
I am running a proxy pushing 2 million calls a day.
Every now and then, I drop 5 or 10 BYE messages.
I've attached the sip_scenario log file, F16,F17,F20 all send BYE
messages to 12.46.104.62. F21 is sending the BYE form 12.46.104.62 to
12.46.104.252. The OK comes back, but is never relayed.
BYEs continue to retry until 12.46.104.62 finally gives up and
sends a Request timeout 408 in frame F38. So, F38 is evidence
that the transaction started. Right?
Why are all of the BYEs ignored, and why isn't the OK relayed?
I have xlog output from 12.46.104.58 indicating each time a BYE
is loose-routed to 12.46.104.62. The syslog file shows each attempt
(F15,F18,F19,F23, etc).
I also have xlog output from 12.46.104.62, however, the syslog on that
machine never shows receiving or loose-routing a BYE with this callid.
It seems like receiving more than one BYE in a short period of
time is screwing up the t_relay()???
The calls are being generated with sipp, and 999,990 times out of
a million it works.
---greg
--
Greg Fausak
greg(a)thursday.com
Dear all,
I am a student and am trying to setup a demonstration with TLS and DTLS
support between a SIP client and a Proxy.
Has anyone some information regarding adding DTLS support for a SIP
client/Proxy or a more generic one using OpenSSL. I just need it for
demonstration purposes and so error handling's if any need not be complete.
Just a simple support and a demonstration of both would do fine. If anyone
has any information regarding this, may be a how to start or any
introductory article would be of great help, kindly let me know. This would
be of great help. Any suggestions regarding the same are most welcome.
kindly let me know and thank you very much.
regards,
Pjothi
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