hello friends:
I crashed when i compile the SER source and use the tls.so!
I have install the openssl 0.9.8 in the default folder :/usr/local
and install the ser-2.0.0-rc1 in the default too.
run the command below:
make group_include="standard" include_modules="tls" all
make group_include="standard" include_modules="tls" installthe tls.so in the /usr/local/lib/ ser/modules folderthe I add the loadmodule "/usr/local/lib/ser/modules/tls.so" in the ser.cfg and start /usr/local/sbin/ serbut the ser start failed!I run the ser -c and found:ERROR: load_module: could not open module </usr/local/lib/ser/modules/tls.so>: /usr/local/lib/ser/modules/tls.so: undefined symbol: deflate
I don't know Why and ask help for you !
Thanks a lot !
qiulei212
2009-08-21
Hello,
I have 2 ip addresses being listened in my cfg file.
Is it possible to set which ip address will be used when relaying a request?
regards,
takeshi
Hi all,
I have a problem with
routing BYE messages (when called UAC sends BYE) when connection between two
Kamailio (1.5.1 with identical configurations) proxies is done by TLS/TCP.
WORKING:
UAC#1 <-- UDP/TCP/TLS --> Proxy#1 <--UDP--> Proxy#2 <-- UDP/TCP/TLS --> UAC#2
NOT WORKING:
UAC#1 <-- UDP/TCP/TLS --> Proxy#1 <-- TCP or TLS --> Proxy#2 <-- UDP/TCP/TLS --> UAC#2
By looking at ngrep
log I found out that when “UAC#1” sends INVITE, “Proxy#1” sends right “Contact”
header (Contact:<sip:bob@IP_ADDRESS_OF_UAC#1:6126;transport=udp>)
to “Proxy#2” , but “Proxy#2” sends wrong “Contact” header (Contact:<sip:bob@IP_ADDRESS_OF_PROXY#1:39879;transport=udp>)
to “UAC#2”.
When UAC#2 sends BYE,
it is sends message with this wrong header and when it arrives on “Proxy#1” proxy
doesn’t sends BYE message to UAC#1. Or when UACs with TCP, this are “Proxy#1”
errors:
DBG:core:tcp_send: no
open tcp connection found, opening new one
ERROR:core:tcp_blocking_connect:
poll error: flags 18
ERROR:core:tcp_blocking_connect:
failed to retrieve SO_ERROR (111) Connection refused
ERROR:core:tcpconn_connect:
tcp_blocking_connect failed
ERROR:core:tcp_send:
connect failed
ERROR:tm:msg_send:
tcp_send failed
ERROR:tm:t_forward_nonack:
sending request failed
DBG:tm:t_relay_to:
t_forward_nonack returned error
What bothers me is
that when I call the other UAC, he can hangup the call (even when he is CALLED UAC),
then I restart both proxies and it is other way around.
I do not know if I
described problem clearly but I am lost in where to look next so any help would
be appreciated. Why proxy change Contact header when I switch from UDP to TCP
between proxies? I'm not rewriting Contact header and only thing that I change in configuration when switching from UDP to TCP is this:
if($rd=="example.com")
{
t_relay("tcp: example.com:5063");
exit;
}
Thanks a lot
/dubravko
Hello,
First I would like to say thank you for the helpful members here, I have got
useful tips and suggestions here.
I am trying to implement some kind of "protection" for me, let me describe
what is my problem at the moment. I know I ask a bit much questions here .)
I would like to limit the number of incoming calls from my customers, but on
a customer basis. I have a subscriber with an Asterisk, so it will be a bad
idea to set a limit to 1 concurrent call for this customer, but limit 1 will
work for a customer with an IP phone (remember the 100rel thing what I
posted here? that was a workaround for this subscriber... I mean to remove
100rel from the supported header).
So basically I would like to limit every subscriber, on a per customer
basis.
The next big thing is the lcr gateways, so where I send the calls. I have 2
peers, one of then have a limit for me, it's 20 concurrent calls. They don't
limit me, so I can send out 30-40-50 calls to them, but they will charge me
a big amount of Lei (Romanian currency) for every calls above 20 at the same
time. They have freeswitch.... doesn't matter, I'm just saying they have
freeswitch :) But they have very very good price(for 20 calls what i
mentioned), so I would like to fill that "20 calls" space.
I have asked them to give some limits for me, so refuse calls above 20 (so
this way lcr can send out to the another gateway), but they don't want to do
it (well, they need money I think)
I would like to store values for these limits in mysql (column "limit" in
subscriber table, and column "calllimit" in gw table, or I can store it in a
separate table).
Right now I store these values in a mysql table called "iplimits"
I am not perfectly sure about how I can select these values from the table,
and how can I "play" with these values. Even don't know how to retrieve the
current number of concurrent calls per gateway or per subscriber.
And what to do with these selected values? Store it in an avp and then do
some checks against it?
I am reading the module docs, especially sqlops, but I think I can't see
the wood for the trees.
I hope somebody can give me suggestions (or an example) about this.
Kind regards,
Dmitri
Sorry, I have accidentally sent this message to opensips mailing list, I am
sending it now to the right place
Hello,
I would like to implement some kind of failover for my Asterisk, let me
describe how I would like to see it.
I registered 2 sip users with my Kamailio, 1020(a)domain.com and
1030(a)domain.com.
I have added aliases to 1020 (aliases from 1021@domain com to
1029(a)domain.com).
Right now calls to 1020-1029 goes well to 1020, it works fine.
But I would like to do the following:
If 1020 isn't registered with Kamailio (let's say if registration for 1020
is down in Kamailio, so AOR not found for 1020), it is possible to route
calls to 1030?
So route calls to 1030 only when registration for 1020 isn't active in
Kamailio.
I have tried manipulating with faillure_route, but without any luck.
I hope it is clear what I would like to do :)
Any help would be much appreciated.
Kind regards,
Dmitri
On Mittwoch, 19. August 2009, Jason Penton wrote:
> Hey Henning,
>
> no, none of my log messages from the cfg file come into the log file.
> However I do get the following:
>
> Aug 19 17:35:38 ug2s02-zone2 kamailio[12273]: [ID 797031 local0.debug]
> DBG:core:fm_malloc_init: F_OPTIMIZE=16384, /ROUNDTO=2048
> Aug 19 17:35:38 ug2s02-zone2 kamailio[12273]: [ID 596959 local0.debug]
> DBG:core:fm_malloc_init: F_HASH_SIZE=2067, fm_block size=16560
> Aug 19 17:35:38 ug2s02-zone2 kamailio[12273]: [ID 362601 local0.debug]
> DBG:core:fm_malloc_init: params (fbe00000, 33554432), start=fbe00000
> Aug 19 17:35:38 ug2s02-zone2 kamailio[12273]: [ID 551434 local0.debug]
> DBG:core:shm_mem_init_mallocs: success
> Aug 19 17:35:38 ug2s02-zone2 kamailio[12273]: [ID 123545 local0.info]
> INFO:core:init_tcp: using /dev/poll as the TCP io watch method (auto
> detected)
>
> weird that I get some messages but not the others. I have alot of LM_INFO
> messages in my code that I dont see too?????
Hi Jason,
so it seems that you indeed get only DEBUG messages in the log file, so try
the suggestion Alex gave to you - local0.* This should catch all.
Cheers,
Henning
Hi All,
I am having some problems logging Kamailio in debug mode using syslog. All
is fine if I log to stderr. However, when I log to syslog I get a few dbug
messages and then it stops -- I dont get any of my routing messages or
anything????
i am suing solaris?
Any suggestions
Cheers
Jason
Hi all,
I am new to SIP. I've downloaded Netbeans 6.7 and installed related plugins.
When I tried to create an SIP Project, I couldn't find any SIP Categories in
NB 6.7.
Is there any basic guidance for SIP in Netbeans ?
thank you very much.
Sangmane
--
View this message in context: http://www.nabble.com/SIP-Project-for-Beginner-tp25010377p25010377.html
Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.
Hello,
Is there a way to remove 100rel from Supported header (only this, and keep
the others) by Kamailio?
I am using Kamailio 1.5.1 notls.
Kind regards,
Dmitri