The reason people are yelling on the internet Freeswitch is much better than
asterisk is pure frustration.
They have used asterisk for years, were faced with crashes and since they are
using freeswitch they don't see those crashes anymore (apart from the reason of
those crashes).
No wonder
On 12/09/2010 08:18 PM, Duane Larson wrote:
Anca,
It failed quickly. But I am not able to do the Prints like I did last
time because _h doesn't exist
(gdb) backtrace
#0 0x7ff75074f165 in raise () from /lib/libc.so.6
#1 0x7ff750751f70 in abort () from /lib/libc.so.6
#2
Hi Russell,
I see from the trace that the BYE is forwarded by OpenSIPS to the other
end here:
E..:@.@.12
.B.O}j.. .BYE sip:0702923002334...@public.address SIP/2.0
Max-Forwards: 67
To: 0702923002334053
sip:0702923002334...@customer.address;tag=as6fbb0b2b
From:
Hello.
It seems that when I call do_route() the route[1] was automatically called..
As I understand I should manually call relay functions after do_routing().
Any ideas why that is happening? :)
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812
On 12/10/2010 01:47 PM, Anton Zagorskiy wrote:
Hello.
It seems that when I call do_route() the route[1] was automatically called..
As I understand I should manually call relay functions after do_routing().
Have you set the routeid field in in the dr_rules table record to 1 :) ?
If you set
Yes, you were right! Thank you. But now openSIPS goes to
crash...
[6176]: *** route[invite]: call do_routing()
[6176]: DBG:core:db_new_result: allocate 28 bytes for
result set at 0x81db470
[6176]: DBG:db_mysql:db_mysql_get_columns: 1 columns
returned from the query
[6176]:
Hi,
Please investigate the crash. First you have to get a core. Do:
ulimit -c unlimited
and when starting opensips also use a -w path argument ( to set the
working directory).
Then after you get a core ( you will see the message core was generated
) investigate the core with gdb, run 'gdb
That SIP message is actually OpenSIPS receiving the BYE from the customer, the
problem is OpenSIPS is not forwarding/relaying that BYE message to the Asterisk
server that is handling the call, so Asterisk doesn't know the call ended, and
it's still letting the phone ring (and possibly be
I haven't seen many posts from frustrated peole, majority of them come from
people either selling fs based services or part of fs development team.
From my experience with fs 1.0.4 it was crashing every 2 months, 1.0.6 is
better, I already posted crashing rate for our use case.
I haven't
Guys,
Point taken. Personally I prefer Coke over Pepsi.
- Opensips user Jeff
On 12/10/10 10:04 AM, paul.gor...@gmail.com paul.gor...@gmail.com
wrote:
I haven't seen many posts from frustrated peole, majority of them come
from people either selling fs based services or part of fs development
Paul,
I do not quite understand what is find me doing with NAT
Thanks
Lloyd
On Fri, Dec 10, 2010 at 10:11 AM, Jeff Pyle jp...@fidelityvoice.com wrote:
Guys,
Point taken. Personally I prefer Coke over Pepsi.
- Opensips user Jeff
On 12/10/10 10:04 AM, paul.gor...@gmail.com
Hmm, it's like Ferrari owners talking about which one is better: Volkswagen
or Toyota :)
2010/12/10 Aloysius Lloyd lloyd.aloys...@gmail.com
Paul,
I do not quite understand what is find me doing with NAT
Thanks
Lloyd
On Fri, Dec 10, 2010 at 10:11 AM, Jeff Pyle
Not a problem. Here you go
(gdb) frame 2
#2 0x7ff74cc69ac1 in msg_watchers_clean (ticks=value optimized out,
param=value optimized out) at subscribe.c:485
485 abort();
(gdb) print pa_db
$1 = (db_con_t *) 0x83e530
(gdb) print pa_db-curr_ps
$6 = (db_ps_t *)
That did it! I found and fixed the bug, please update the fix from svn.
Thanks, Duane!
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That simple?!! I will compile the new code and update. Thanks!!
On Fri, Dec 10, 2010 at 10:16 AM, Anca Vamanu a...@opensips.org wrote:
That did it! I found and fixed the bug, please update the fix from svn.
Thanks, Duane!
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Users mailing
When I try to recompile I get this error
opens...@proxy01:~/opensips-1.6.3-tls$ sudo make prefix=/usr/local install
include_modules=presence
make[1]: Entering directory
`/home/opensips/opensips-1.6.3-tls/modules/presence'
Compiling subscribe.c
gcc -fPIC -DPIC -g -O9 -funroll-loops
What code is this? 1.6 svn has no problem..
On 12/10/2010 06:28 PM, duane.lar...@gmail.com wrote:
When I try to recompile I get this error
opens...@proxy01:~/opensips-1.6.3-tls$ sudo make prefix=/usr/local
install include_modules=presence
make[1]: Entering directory
I have 5 Asterisk servers with an average of 6000 calls/day. All work at
least eight months without stopping, somene with stops planned for
maintenance (upgrade, patch, etc...).
Mounting the operating system properly and taking care of the major faults I
see no problem with Asterisk.
The two
I am not sure if I did this right.
At first I just replaced the subscriber.c file and tried to compile. That
was my last email.
Now I did the following
svn co
https://opensips.svn.sourceforge.net/svnroot/opensips/branches/1.6opensips_1_6
Then I did the following within the opensips svn
I did update the presentity table to be 5 instead of 4. Perhaps i didn't
clean up the previous install. Let me try that.
On Dec 10, 2010 11:46 AM, Ovidiu Sas o...@voipembedded.com wrote:
Do you see any errors in the logs?
Make sure that you clean up your previous install (remove all the
Have you also updated the layout of the presentity table?
Check for the extra_hdrs field.
Regards,
Ovidiu Sas
On Fri, Dec 10, 2010 at 12:52 PM, Duane Larson duane.lar...@gmail.com wrote:
I did update the presentity table to be 5 instead of 4. Perhaps i didn't
clean up the previous install.
I did a total removal of the older OpenSIPS install and dropped and readded
the Presence tables into MySQL. That really helped thanks. I still had
some core dumps though not sure if it is going to happen more or not (will
have to keep my eye on it), but this is what the core dump says
(gdb)
This is a crash in the pua module (different than the previous one).
send_publish.c:496 = free(body-s);
Can you print the value and content of the body?
Thanks,
Ovidiu
On Fri, Dec 10, 2010 at 3:28 PM, Duane Larson duane.lar...@gmail.com wrote:
I did a total removal of the older OpenSIPS
Thanks for working with me on this Ovidiu. I think this is what you are
wanting (sorry not good with debugging)???
(gdb) frame 4
#4 0x7f1e935bc0b8 in send_publish_int (presentity=value optimized
out,
publ=value optimized out, ev=0x7f1e88d57518, hash_index=332) at
send_publish.c:496
496
You are using mi to send a publish. The body of that publish is taken
from the mi command directly.
I think the body should be copied into the shared memory and not used
directly from the mi structure.
Or maybe if it's used from the mi structure, the free is not necessary
(as it may be done by
Thanks. I won't do anything for now since it hasn't core dumped again.
Will wait to see what the OpenSIPS team thinks. Hopefully though the first
issue is fixed though. Should know over the weekend if OpenSIPS hasn't done
a core dump with the same error.
Appreciate the help.
On Fri, Dec 10,
I guys,
I'm try to configure opensips 1.6.3 + rtpproxy on a dual homed server
with one interface on private network and one on public network.
I want to make possibile the calls from inside to outside and viceversa.
There is a smart way to do that?
I'm going to write a cfg to manually change
I think the b2b modules would help with topolopy hiding
On Dec 10, 2010 5:31 PM, Stefano Pisani stefano.pis...@omnianet.it
wrote:
I guys,
I'm try to configure opensips 1.6.3 + rtpproxy on a dual homed server with
one interface on private network and one on public network.
I want to make
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