Hello,
I need to write a CAC (Call Admission Control) module for an 802.11 AP
(Access Point).
The idea is to use a SIP Proxy to monitoring bandwith utilization, for
example according to codec, and allow or disallow new sessions,
depending on resources.
I'm thinking using OpenSER for this purpose.
On a first stage I want to install a SIP Proxy on the AP (embedded with
OS Linux) and on a second stage adding the CAC feature.
There is many modules in OpenSER and I'm not sure which one I should use
for a SIP Proxy.
I'm learning about SIP but my knowledge is limited (till last week I
never heard about it).
Thank you for advising.
Regards,
Michel.
Hi:
I keep getting the following error on a 0.9.x system. The latest 0.9.x
version of SER was just download and built on this server but I continue
to get this same error. I've check the mysql Makefile and it seems to be
correct. What else could cause this error?
ERROR: load_module: could not open module
</usr/local/lib/ser/modules/mysql.so>:
/usr/local/lib/ser/modules/mysql.so: undefined symbol: log
Thanks,Steve
Hello:
I would like to implement sequential hunting. Hunting is when a call
arrives for a given subscriber, no one answers the original number so
the proxy populates the username portion of the r-uri with another
number (from the database) and tries again. This process continues until
all numbers in the database are exhausted. I have the "failover" working
but the r-uri field is not updated on any interation though the
failure_route. Does anyone have any ideas how to make the following work?
Thanks,Steve
# hunting (serial forking)
if (avp_db_load("$ruri/username", "s:allow_hunt")) {
if (avp_check("s:allow_hunt", "eq/y/i")) {
avp_delete("s:allow_hunt");
xlog("L_INFO", "\n[SER]: [%Tf] [%ci] Call Hunting Enabled for
<%ru>\n");
avp_db_load("$ruri/username", "s:huntlist/sam");
xlog("L_INFO", "\n[SER]: [%Tf] [%ci] Call Hunting Started to
<%ru>\n");
t_on_failure("6");
t_relay();
break;
};
};
# Iterate through extension list for subscribers with call hunting enabled
failure_route[6] {
xlog("L_INFO", "\n[SER]: [%Tf] [%ci] Failure Block #6: HUNTING for
<%ru> from <%fu> at <%is>\n");
if (avp_pushto("$ruri/username", "s:huntlist/g"))
{
append_branch();
avp_delete("s:huntlist");
t_on_failure("6");
t_relay();
};
}
Hello,
I'm having a hard time getting RADIUS auth to work at all. I've reached
the point where I am debugging the network traffic with tshark.
I can see RADIUS packets generated from the openser box for accounting,
but no RADIUS authentication packets are sent. Also, it seems no
response comes back from the RADIUS server even for accounting requests.
radclient against my radius server returns me an Access-Reject at least
and the tshark dump contains auth request and response.
OpenSER 1.1.1 tells me:
Feb 6 17:27:04 openser openser[22596]: rc_send_server: no reply from
RADIUS server
I suspect the issues lie in the radiusclient-ng layer. I'm using
radiusclient-ng from Debian Etch (0.5.3-2), but I see many branches in
CVS, like RELENG_0_5_5 which seems newer.
thanks
HI All,
I have a problem with uac_replace_from() function not working for Linksys Unit.
This is what i have in the route[1],
if (search("From:.*<sip:900*")) {
uac_replace_from("anonymous","sip:anonymous@x.x.x.x");
}
I have 0900xx as internal sip account, then I have to remove the
0900xx callerid before i send the call to PSTN gateway. Because 0900xx
is not a valid number, Mobile phone providers block the call because
of that, as a result, I need to replace the callerid to anonymous.
I have tested with many other ATA and X-Lite without problem. The
problem only ocurrs when i use Linksys ata. I have monitored the SIP
message, the first request will change to anonymous then back to the
original username.
Anyone have an idea on how i can fix this ?
--
Howard Tang
ICQ : 259083
MSN : howard615(a)hotmail.com
Hi,
I can not use www_challenge(realm,qop)function in ser.cfg file even though
auth module is loaded. The readme file has documentation for it.
When I use the following in config file and start SER, the following error
is thrown
if (!radius_www_authorize("")) {
www_challenge("", "1"); // parse error
break;
};
parse error (197,49-50): unknown command, missing loadmodule?
I am using Ottendorf release. How to do digest authentication with new
release??** <http://ftp.iptel.org/pub/ser/daily-snapshots/unstable/>
-Venkat
Hi to all,
little question about populating the Route Header field.
I'm dealing with 2 CPE of different vendors (Patton and Netsynt) and an
Openser in the middle.
Patton in their SIP is not using angle bracket on SIP From,To and
Contact headers.
Netsynt is using strict routing method to build BYE messages.
This is causing that when Netsynt is creating the BYE it is putting the
proxy IP in the R-URI and the URI of Contact received from previuos
packets from patton in the Route header to let it be processed by the proxy.
In this way the Route header field constructed is without the <>
enclosing the uri. Netsynt keep it from previous Contact without any
manipulation on it.
This is causing an error on the proxy parsing the name-addr field of the
Route header field and the loose_route() fail. Seems that it is
mandatory for Openser to get angle bracket in Route header fields.
Now in the section 20 of RFC3261 is said that it is not a MUST that
From,To and Contact header have to be enclosed in angle bracket.
So i'm searching in the RFC where is said that if the Contact is without
the angle bracket the UAC have to check if the Contact have the <> and
if it don't have the <> it have to insert it before to out it on the
Route (i searched the RFC2543 too).
can someone give me clarification on this issue?
Where is written the Route header field is mandatory to have <>
enclosing the URI.
I'm searching all of this because i have to give a good reason to each
vendor (Netsynt and Patton) to modify their sip stack for this issue. I
don't know who of the 2 should fix the problem. If Patton to add <> in
each request or Netsynt to fix the Route field population.
Thanks,
Bye,
Marcello
Hi all.
could any one send me all the things that I should do for having only a call between two xlite phones with ser server?
I have installed ser and xlite on 10.10.10.138 and another xlite phone on 10.10.10.195. the xlite phone on 10.10.10.195 sends register massage to 10.10.10.138 but xlite phone that is on 10.10.10.138 doesn't send massage to ser.
I didn't change any thing in ser only fork=no,log-stderror=yes.children=4.
and I don't user sql.
thanks.
---------------------------------
Access over 1 million songs - Yahoo! Music Unlimited.
I'm seeing this *very* weird error where uac_restore_from is ending up with a corrupted 'From' header.
The weird thing is that it happens with only one end point (a Televantage SIP Trunk).
With virtually *everything* else - and I am talking about tens of thousands of end points, spanning all sorts of sip devices, we are having *no* problems.
The call is set up as follows -> the endpoint sends the INVITE, which is forwarded to a PSTN gateway
The ACKs and the 200 OK all have their From/To headers appropriately rewritten
The problem arises when the PSTN gateway initiates the BYE As you can see below, the 'To' header gets corrupted
The *only* thing I can think of is that it has something to do with the structure of the Televantage tags (the '+' signs?).
Any ideas?
FYI, Openser 1.1.0 (latest CVS checkout), and
modparam("uac","from_restore_mode","auto")
modparam("uac","rr_store_param","aptelavsf")
---INVITE--- endpoint->proxy
From: "Member service"<sip:user100.crunchhq@69.25.47.136>;tag=10.10.35.100+1+8d200003+32cc4d0e.
To: <sip:17038677006@69.25.47.136>;tag=277e3aa2408f2ad73a234f5bbb99133b.c13b.
---INVITE--- proxy->gateway
From: "Membersevice Queue"<sip:6462174165@flareon.aptela.com>;tag=10.10.35.100+1+8c020003+dd5437f3.
To: <sip:17038677006@69.25.47.136>
Record-Route: <sip:69.25.47.136;lr;ftag=10.10.35.100+1+8c020003+dd5437f3;nat=yes;aptelavsf=dGVzdDciIDR0YndrISJGfHtsaWBbPWQ7NiQ4LCJlISgv>
--- 183 Session Progress--- gateway->proxy
From: "Membersevice Queue"<sip:6462174165@flareon.aptela.com>;tag=10.10.35.100+1+8c020003+dd5437f3.
To: <sip:17038677006@69.25.47.136>;tag=1386496.
Record-Route: <sip:69.25.47.136;lr;ftag=10.10.35.100+1+8c020003+dd5437f3;nat=yes;aptelavsf=dGVzdDciIDR0YndrISJGfHtsaWBbPWQ7NiQ4LCJlISgv>
---183 Session Progress--- proxy->endpoint
From: "Member service"<sip:user100.crunchhq@69.25.47.136>;tag=10.10.35.100+1+8c020003+dd5437f3.
To: <sip:17038677006@69.25.47.136>;tag=1386496
---200 OK--- gateway->proxy
From: "Membersevice Queue"<sip:6462174165@flareon.aptela.com>;tag=10.10.35.100+1+8c020003+dd5437f3.
To: <sip:17038677006@69.25.47.136>;tag=1386496
Record-Route: <sip:69.25.47.136;lr;ftag=10.10.35.100+1+8c020003+dd5437f3;nat=yes;aptelavsf=dGVzdDciIDR0YndrISJGfHtsaWBbPWQ7NiQ4LCJlISgv>
---200 OK--- proxy->endpoint
From: "Member service"<sip:user100.crunchhq@69.25.47.136>;tag=10.10.35.100+1+8c020003+dd5437f3.
To: <sip:17038677006@69.25.47.136>;tag=1386496
Record-Route: <sip:69.25.47.136;lr;ftag=10.10.35.100+1+8c020003+dd5437f3;nat=yes;aptelavsf=dGVzdDciIDR0YndrISJGfHtsaWBbPWQ7NiQ4LCJlISgv>
---BYE--- gateway->proxy
To: "Membersevice Queue"<sip:6462174165@69.25.47.136>;tag=10.10.35.100+1+8c020003+dd5437f3.
From: <sip:17038677006@66.52.236.25>;tag=1386496.
Route: <sip:69.25.47.136;lr;ftag=10.10.35.100+1+8c020003+dd5437f3;nat=yes;aptelavsf=dGVzdDciIDR0YndrISJGfHtsaWBbPWQ7NiQ4LCJlISgv>
---BYE--- proxy->endpoint
To: "Membersevice Queue"<sip:user100.cru>6'(!.l asr}XV.R\[>;tag=10.10.35.100+1+8c020003+dd5437f3.
From: <sip:17038677006@66.52.236.25>;tag=1386496.
--
*******************************************
Mahesh Paolini-Subramanya (703) 386-1500 x9100
CTO mahesh(a)aptela.com
Aptela, Inc. http://www.aptela.com
"Aptela: How Business Answers The Call"
*******************************************
