Hi Anca,
That sounds bad... :-(
Is there any other way to achieve what I'm trying to do?
Basically, as long as normal load balancing goes, I also need to select
specific destinations for certain numbers.
Moreover, the list is quite big. Moreover, the list is changing. Moreover,
destinations for ce
Hi,Bogdan
No,I want to received a INVITE request and a MESSAGE request, then I send
MESSAGE on public interface and send INVITE on private interface.How can I
accomplish this?
Regards,
CheeWii
2010/10/23 Bogdan-Andrei Iancu
> Hi CheeWii,
>
> Do you want to received the a INVITE request and to s
Bogan,
That resolved it. Thanks for the advice.
Ryan
On Fri, Oct 22, 2010 at 12:32 PM, thrillerbee wrote:
> Thank you!
> I'll give that a shot & report back.
>
> Ryan
>
>
> On Fri, Oct 22, 2010 at 12:07 PM, Bogdan-Andrei Iancu <
> bog...@voice-system.ro> wrote:
>
>> Hi,
>>
>> what group are y
Hi Bogdan,
I believe I found the problem. When sip_msg_cloner() within build_cell()
fails due to out-of-mem, and dangling pointer to the cell is left in the
global transaction pointer. Later on the post_cb() code attempts to clean
this up, and "resurrects" the now-free memory, and in particular p
Thank you!
I'll give that a shot & report back.
Ryan
On Fri, Oct 22, 2010 at 12:07 PM, Bogdan-Andrei Iancu <
bog...@voice-system.ro> wrote:
> Hi,
>
> what group are you using for your destination (in dispatcher) ? if "0",
> use another one :D...there is an issue there...
>
> Regards,
> Bogdan
>
Very true. You got me there ;)
On Oct 22, 2010 1:12 PM, "Bogdan-Andrei Iancu"
wrote:
Of course it did, but the subject was "static routing"not dynamic :)...
Regards,
Bogdan
Duane Larson wrote:
>
> Couldn't the dynamic routing module also be used depending on the
> prefix...
>> mailto:bo
OK, cool
James Mbuthia wrote:
> Hi Bogdan,
>
> I figured that out after I went through the SIP rfc in more detail.
> Thanks for your help though.
>
> james
>
>
> On Fri, Oct 22, 2010 at 7:04 PM, Bogdan-Andrei Iancu
> mailto:bog...@voice-system.ro>> wrote:
>
> Hi James,
>
> As I see, the
Of course it did, but the subject was "static routing"not dynamic :)...
Regards,
Bogdan
Duane Larson wrote:
>
> Couldn't the dynamic routing module also be used depending on the
> prefixes at each individal remote pbx system?
>
>> On Oct 22, 2010 12:29 PM, "Bogdan-Andrei Iancu"
>> mailto:bo
Hi Najib,
I guess you are using an opensips prior to 1.6 (by looking at the error
messages)...As the message says , the t_relay() has no valid new
destination where to send the request
Check for prior error - maybe the RURI was not valid, so the destination
you set was discarded -> no val
Hi Bogdan,
I figured that out after I went through the SIP rfc in more detail. Thanks
for your help though.
james
On Fri, Oct 22, 2010 at 7:04 PM, Bogdan-Andrei Iancu wrote:
> Hi James,
>
> As I see, the INVITE has in URI sip:ja...@198.168.0.1
>
>INVITE sip:ja...@198.162.0.1 SIP/2.0
Hi,
what group are you using for your destination (in dispatcher) ? if "0",
use another one :D...there is an issue there...
Regards,
Bogdan
thrillerbee wrote:
> I could still use some help on understanding what I'm missing that is
> preventing gws from transitioning back into the 'active' stat
Couldn't the dynamic routing module also be used depending on the prefixes
at each individal remote pbx system?
On Oct 22, 2010 12:29 PM, "Bogdan-Andrei Iancu"
wrote:
Hello Andrea,
If you check the default opensips.cfg, you can see that there is a step
where only the initial requests are gettin
Hi James,
As I see, the INVITE has in URI sip:ja...@198.168.0.1
INVITE sip:ja...@198.162.0.1 SIP/2.0
without any port indication, so the default 5060 is assumed. The proxy
cannot automatically discover what's the right port on the next hop if
not instructed by RURI or if not discover
Hi all,
I setup opensips with the tm module for call forward on timeout. It
works great but I would like to perform different actions depending on
if it is the "fr_timer" or the "fr_inv_timer", and I can find no way of
telling from the script which timer was hit.
How could I gain access to t
The call to opensipsdbctl create is asking me for the password on
every single table created. But the real problem is that after several
tables have been created the following error happens:
...
...
-e Creating core table: drouting
Password for user almira:
NOTICE: CREATE TABLE will create implic
Hi Leon,
maybe you should consider using OpenSIPS Control Panel and the CDR
procedure provided by this tool...
Of course, you can use only the CDR procedure (it is only at mysql
level). See:
http://opensips-cp.sourceforge.net/htmldoc/cdrviewer.html
http://opensips-cp.svn.sourcefo
Any chance with the backtraces ?
Regards,
Bogdan
Anca Vamanu wrote:
> Hi,
>
> You need to inspect them with gdb, run: gdb
> path_to_opensips_executable path_to_corefile, and then run 'bt full'
> and send the output.
>
> Regards,
> --
> Anca Vamanu
> www.voice-system.ro
>
>
> On 10/14/2010 10:1
Hi CheeWii,
Do you want to received the a INVITE request and to send it to two
destinations , one on private interface and one on public interface? Did
I get it right ?
Regards,
Bogdan
CheeWii wrote:
> Hi,
> My OpenSIPS server has two network cards. One is public ip
> address such as 202
Yes to both. Bt is below. I'll recompile with with optimization and
reproduce the problem today.
(gdb) where
#0 0x7f8b8356c2c2 in insert_timer_unsafe (new_tl=0x7f8b7a54e310,
list_id=WT_TIMER_LIST, ext_timeout=) at
timer.c:731
#1 set_1timer (new_tl=0x7f8b7a54e310, list_id=WT_TIMER_LIST,
bla, emailing via "subject" :DI wonder what the email body is good
for in this case :P
Check the dialplan rules - your opensips is not recognizing the local
subscribers - the rules in DP must match the local subscribers
Regards,
Bogdan
Marcella wrote:
>
> KINGSSQUEEN
>
>
>
>
>
> __
put more listening definitions:
listen=udp:ip:port1
listen=udp:ip:port2
Regards,
Bogdan
Marcella wrote:
>
> KINGSSQUEEN
>
>
>
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
Hello Andrea,
If you check the default opensips.cfg, you can see that there is a step
where only the initial requests are getting there - starting from that
point you can implement your static routing using "if" statements,
checking the $rU (request username ) and setting new destination
(writ
Hi Maciej
Maciej Bylica wrote:
> Hi,
>
> Have anyone tried to use usr_preferences, AVPops to determine the
> service to be fetched by the script?
>
That is the the proper module for handling generic attribute. Uisng
AVPops module you can load from db, for a certain user, a certain
attribute (
Hi Marcio,
first of all be sure you are using the latest SVN check out from 1.6 branch.
What is really interesting in your case I do not see any err / warning
message...and the code is generating err/warn messages before destroying
the rule.
What is happening is that during the DB load, when a
Hi Dave,
the crash refers to an invalid dst_uri and I see in your failure route
you are not using any $du related op Also, I see you add a single
new branch (via RURI) (no parallel forking)
Also, in branch_route, you do not do any dst_uri or ruri ops - you work
only on headers ?
Could you
Hi Alexandr,
The second parameter of load_balance() function can not be a
pseudovariable, but only string.
Regards,
--
Anca Vamanu
www.voice-system.ro
On 10/22/2010 03:53 PM, Alexandr A. Alexandrov wrote:
> Hi!
>
> I have a strange problem with trying to use avps in load_balance function.
Hi Dave,
The core files do not help as they need to be investigated with your
binaries. Please run gdb path_to_executable path_to_corefile and then
run 'bt full' and send the output.
Regards,
--
Anca Vamanu
www.voice-system.ro
On 10/21/2010 10:08 PM, Dave Singer wrote:
On my production se
Installing postgresql-client package and configuring a ./pgpass file
with the following format fixed the problem:
hostname:port:database:username:password
Regards,
David
On Fri, Oct 22, 2010 at 3:26 PM, David Santiago
wrote:
>
> Hello,
>
> I have successfully compiled and installed OpenSIPS 1.
Hi Anton,
No, this is not normal and the user agent that you use has a bad SIP
implementation - a CANCEL cancels the request with the same Via Branch id.
RFC 3261 - section 9.1
"A CANCEL constructed by a
client MUST have only a single Via header field value matching the
top Via value
Hi Maciej,
default value for restore type is "auto", which means you do not have to
do anything for proper fixing of all messages in the dialog.
Some hints:
1) are you using the rr module for doing record_route and
loose_route ? This is essential for auto restore to work
2) in the o
Hi Jeff,
I have just tried like this with 1.6 and it works:
$var(text)="";
$avp(s:call_carname) = $(var(text){param.value,carrier});
xlog("carrier = $avp(s:call_carname)\n");
=> carrier = AT&T
But if you take it from a header, you don't have the quotes escaped. Do
you see any error in the log
Hi Kennard,
I suppose the bt is the same ? do you still have the core file ?
Regards,
Bogdan
kennard_wh...@logitech.com wrote:
>
> Hi Bodgen,
>
> I replicated the error. Unfortunately the entire insert_timer_unsafe
> and been in-lined and little is available:
>
> Program terminated with signal
I could still use some help on understanding what I'm missing that is
preventing gws from transitioning back into the 'active' state from
'probing'. Currently, I have to babysit this OpenSIPS instance.
Again, to summarize, when dispatcher detects a failure, it puts the gw into
'probing' state & b
Hello,
I have successfully compiled and installed OpenSIPS 1.6.3 (no tls) with the
PostgreSQL module for an OpenSIPS Registrar testing installation.
I have edited the *opensipsctlrc* file to specify the connection details for
a remote PostgreSQL database server I want to use as the persistence st
Hi!
I have a strange problem with trying to use avps in load_balance function.
I'm trying to do balancing like this:
avp_db_query("select phone, resource from phone_resource where
phone like '%$fU%'", "$avp(i:111);$avp(i:112)");
avp_print();
xlog("L_INFO","$fu = $avp(
Hi,
My setup:
- 11.22.33.44 : openSIPS 1.6.3
- 11.22.33.45 : one of the Asterisk 1.6.2.13 servers
- 88.77.66.55 : my public ip-address
- 192.168.1.10 : my local ip-address (NAT)
All is working well except Session Timers where the Re-Invite originates from
Asterisk.
I have a SIP trace ( http://p
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