Hi All
I am trying to configure Openser as loadblancer for two more Kamailio. I wan
loardbalancer not stay in the middle once routed to other kamilio servers.
my first question is, dispathcer list always take IPs or I can configure
proxy names as well.
at the moment I am checking the to_uri and then redirecting the traffic to
my one of Kamilio on my network, it seems to work fine with Linksys but not
other phones e.g polycom xlite or even sjphone.
if(to_uri=~"sip:.+@sip.mydomain.ie <sip%3A.%2B(a)sip.mydomain.ie>") {
ds_select_domain("1", "4");
sl_send_reply("300","Redirect");
exit;
}
any suggestion.
Thanks in advance
Regards
Asim Riaz
Hi,
Need a little help with get_redirects(), using openser 1.2. I
am using the following in failure_route to capture contacts from a
"multiple choices" reply from cisco media gateway.
Following code sets the ruri to the last contact, however, the $ds
show more information about other contacts. Is there a way to reset
this so that only ruri information is used? Cisco response and xlog
information is also given below.
if(!get_redirects("*:*"))
{
xlog("L_ERROR", "Failed to fetch
contact '$ct' from 301/302 - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci
\n");
acc_db_request("480", "acc");
t_reply("480", "Temporarily
Unavailable");
exit;
}
# get last URI from destination-set and set it as R-URI
xlog("L_INFO", "Redirect from UAC intercepted 1 - M=
$rm RURI=$ru D=$ds B=$bR \nF=$fu T=$tu IP=$si ID=$ci\n");
avp_delete("$avp(s:tmp)/g");
$avp(s:tmp) = $ds;
avp_subst("$avp(s:tmp)", "/.*(sip:.+@[^:;>]+).*$/\1/");
avp_pushto("$ru", "$avp(s:tmp)");
setflag(29);
xlog("L_INFO", "Redirect from UAC intercepted 2 - M=
$rm RURI=$ru D=$ds B=$bR \ntmp=$avp(s:tmp) \nF=$fu T=$tu IP=$si ID=$ci
\n");
append_branch();
route(17); # process
exit;
}
Thanks in advance for your advice.
--
Zahid
The response from gateway:
U 2008/08/26 08:19:04.440411 10.10.0.32:5060 -> 10.10.0.98:5060
SIP/2.0 300 Multiple Choices.
Via: SIP/2.0/UDP 10.10.0.98;branch=z9hG4bK901b.335cb695.0,SIP/2.0/UDP
10.10.12.140;branch=z9hG4bK612a4f20B9293665.
From: "10521" <sip:10521@devproxy.myip.org>;tag=B395F9F2-4618AC3F.
To: <sip:40001@devproxy.myip.org;user=phone>;tag=195535C4-11AC.
Date: Tue, 26 Aug 2008 12:19:04 GMT.
Call-ID: f7932e23-d89cd014-e3ea20a9(a)10.10.12.140.
Server: Cisco-SIPGateway/IOS-12.x.
CSeq: 2 INVITE.
Allow-Events: telephone-event.
Diversion: <sip:40001@10.10.0.32>;reason=unconditional;counter=1.
Contact: <sip:10512@10.10.0.98>,<sip:10512@10.10.0.32>.
Content-Length: 0.
xlog entries:
Aug 26 08:19:04 mousse openser[15353]: Redirect from UAC intercepted 1 -
M=INVITE RURI=sip:40001@10.10.0.32:5060;transport=udp
D=Contact: sip:40001@10.10.0.32:5060;transport=udp, <sip:10512@10.10.0.32
>;q=0.01, <sip:10512@10.10.0.98>;q=0.01
B=<sip:10512@10.10.0.32>;q=0.01, <sip:10512@10.10.0.98>;q=0.01
F=sip:10521@devproxy.myip.org T=sip:40001@devproxy.myip.org;user=phone
IP=10.10.12.140 ID=f7932e23-d89cd014-e3ea20a9(a)10.10.12.140
Aug 26 08:19:04 mousse openser[15353]: Redirect from UAC intercepted 2 -
M=INVITE RURI=sip:10512@10.10.0.98
D=Contact: sip:10512@10.10.0.98, <sip:10512@10.10.0.32>;q=0.01, <sip:10512@10.10.0.98
>;q=0.01
B=<sip:10512@10.10.0.32>;q=0.01, <sip:10512@10.10.0.98>;q=0.01
tmp=sip:10512@10.10.0.98 F=sip:10521@devproxy.myip.org T=sip:40001@devproxy.myip.org;user=phone
IP=10.10.12.140 ID=f7932e23-d89cd014-e3ea20a9(a)10.10.12.140
Aug 26 08:19:04 mousse openser[15353]: Redirect from UAC intercepted -
M=INVITE RURI=sip:10512@10.10.0.98
D=Contact: sip:10512@10.10.0.98, <sip:10512@10.10.0.32>;q=0.01, <sip:10512@10.10.0.98
>;q=0.01, sip:10512@10.10.0.98
B=<sip:10512@10.10.0.32>;q=0.01, <sip:10512@10.10.0.98>;q=0.01, sip:10512@10.10.0.98
F=sip:10521@devproxy.myip.org T=sip:40001@devproxy.myip.org;user=phone
IP=10.10.12.140 ID=f7932e23-d89cd014-e3ea20a9(a)10.10.12.140
Hi all,
I'm using openser 1.4. Can anyone tell me how can i call t_relay() from
another module.
Can anyone help me? any suggestion?
Thanks for your help.
Regards
--
Sabino Frisardi
hi guys:
I get a sip message from my asterisk,and I want to route the message to the ip contained in the "record-route" header.How can i get the ip and transport the message?
2008-09-03
邱磊
hi guys:
I get a sip message from my asterisk,and I want to route the message to the ip contained in the "record-route" header.How can i get the ip and transport the message?
regard~
2008-09-03
邱磊
hi guys:
I get a sip message from my asterisk,and I want to route the message to the ip contained in the "record-route" header.How can i get the ip and transport the message?
regard~
2008-09-03
邱磊
Thanks David for your answer, but can you explain how radius can do that
since sending BYE messages is more at the signaling level. I dont quite
understand how radius can even communicate at the SIP signaling level.
From: David Villasmil
Sent: Sunday, August 31, 2008 4:33 PM
To: Matteo D'Amato
Subject: Re: [Kamailio-Users] CDRTool and Prepaid
You could use radius.
On Sun, Aug 31, 2008 at 10:09 PM, Matteo D'Amato > wrote:
Hello,
In "/doc/PREPAID.txt" it talks about "an external call control module
(not provided by CDRTool) that maintains call status and terminates calls by
sending BYEs to both SIP end-points"
What is this 3-party module and where can I get it?
--Matteo
_______________________________________________
Users mailing list
Users(a)lists.kamailio.org
http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
I'd like to ask if anyone if running a service using SER 0.9.7-pre3 software with greater than 2000 registrations? If so are any of you experiencing issues which seem to be related to timers not firing when expected. We set the inv_timeout to a number which approximates a certain number of rings. This worked well when our subscriber base was small however as the number of subscribers, and associated calling activity, grew the timers seem to take longer to fire. We are planning on upgrading to a newer version of SER but I'd like to hear about other real world experiences.
Thanks, Steve
I have downloades ser's debian packages and installed them in Ubuntu
Linux 6.0Ls Desktop edition
ser_0.9.7_i386.deb and the correspoding ser_mysql modules,
trying commands serctrl ps or ul show i get an error "Error opening
ser's fifo /tmp/ser_fifo", trying to add a user using serctl add .. I
get error SER/FIFO not accessible: 2
the are some instructions in the mailing list saying I should add
fifo_mode=0777 in the configuration file, I have done it but still the
errors exist.
can any one help.
--
Myalla J.C
Kibaha Education Centre
P.O Box 30054 Kibaha
Pwani
Tanzania
+255787680744
I have downloades ser's debian packages and installed them in Ubuntu
Linux 8.0Ls Desktop edition
ser_0.9.7_i386.deb and the correspoding ser_mysql modules,
trying commands serctrl ps or ul show i get an error "Error opening
ser's fifo /tmp/ser_fifo", trying to add a user using serctl add .. I
get error SER/FIFO not accessible: 2
the are some instructions in the mailing list saying I should add
fifo_mode=0777 in the configuration file, I have done it but still the
errors exist.
can any one help.
--
Myalla J.C
Kibaha Education Centre
P.O Box 30054 Kibaha
Pwani
Tanzania
+255787680744
--
Myalla J.C
Kibaha Education Centre
P.O Box 30054 Kibaha
Pwani
Tanzania
+255787680744
