Thanks to Flavio E. Goncalves, a new edition of Building Telephony
Systems with OpenSIPS is now available.
It covers the latest stable release, the OpenSIPS 1.6, updating existing
topics (NAT traversal, accounting, etc), but also approaching new 1.6
specific technologies (data caching, dialog
I have just finished installing OpenSIPs about an hour ago. I have gone
through the cookbook available but didn't find any elaboration on Source IP
rewriting. Before even I start to confuse myself let me explain exactly
what I want to achieve with OpenSIPs. I have Asterisk and A2Billing setup
El Jueves, 21 de Enero de 2010, Slot Zero escribió:
I have just finished installing OpenSIPs about an hour ago.
Perhaps you mean 6 hours ago? XD
--
Iñaki Baz Castillo i...@aliax.net
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I'm facing the same task now - limit the number of concurrent calls per
group of accounts rather than a single number. I'm thinking of using the
group module to organize numbers into groups with group module, then
using get_user_group() to get group id and comparing the profile size
with
Inaki shutup
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Hi Andrei,
Hopefully this is it (with FASTLOCK)
#0 0xb77d5424 in __kernel_vsyscall ()
#1 0xb772babb in poll () from /lib/i686/cmov/libc.so.6
#2 0xb77ba83a in ?? () from /lib/i686/cmov/libresolv.so.2
#3 0xb77b8946 in __libc_res_nquery () from /lib/i686/cmov/libresolv.so.2
#4 0xb77b8fdb in ??
What is this, middle school?
In any case, unfortunately for you, you could not be more wrong.
Iñaki is one of the most leading experts on SIP protocol mechanics in
the open-source community, and his answers are unfailingly precise and
detailed.
Often, they also contain an element of
Well at least he shouldn't have been the one to start it. as far as personality
trait well you got two of us; the only difference is that I need loud mouthed
wisecracking trigger. So I am still better as far as personality goes. If he is
such an expert he could have at least sent a decent reply
Hi,
Another one.. It hangs for a number of seconds (but it's enough to cause to SIP
timeouts - MSG queue jumps to 260K), it's hard to make a bt at the right moment.
This one looks better because there's sched_yield() there :)
(gdb) bt
#0 0xb77d5424 in __kernel_vsyscall ()
#1 0xb771041c in
First of all: are you able to use a decent threads capable mail client? all
your cool mails appear as a new thread (new conversation) because they don't
contain the In-Reply-To header. In order to ask in a maillist you should use
a mail client respecting mail threads.
El Jueves, 21 de Enero
I think Iñaki was trying to tell you politely that your question, was
you formulated it, is somewhat naive. This is mainly so because you
are looking for a quick answer (as though it could be devised from
reading the cookbook) to a potentially rather complicated, larger
architectural issues.
Some more,
(gdb) bt
#0 0xb78dc424 in __kernel_vsyscall ()
#1 0xb781741c in sched_yield () from /lib/i686/cmov/libc.so.6
#2 0xb73d77fd in build_new_dlg () from /usr/lib/opensips/modules/dialog.so
#3 0xb73d4b81 in dlg_create_dialog () from /usr/lib/opensips/modules/dialog.so
#4 0xb73c9c9e in
Rather than have this thread continue its downward spiral, I would like to
suggest the following book:
snip from bogdan's post
Thanks to Flavio E. Goncalves, a new edition of Building Telephony
Systems with OpenSIPS is now available.
It covers the latest stable release, the OpenSIPS 1.6,
Dear All,
I guess none of you have a decent reply as far as my request is concerned. Oh
yeah its idiots like Iñaki that prove this place to be a waste of time
especially when someone who is trying to get things working. As far as paying
consultants dude I am just starting out on my own ok
And one more
(gdb) bt
#0 0xb78dc424 in __kernel_vsyscall ()
#1 0xb781741c in sched_yield () from /lib/i686/cmov/libc.so.6
#2 0x080c3705 in _shm_resize ()
#3 0xb7746069 in relay_reply () from /usr/lib/opensips/modules/tm.so
#4 0xb7746d74 in reply_received () from
Hi,
Since all the backtraces are in allocation routines my guess is that the
shared memory lock might be causing a problem.
Are you compiling with -DF_MALLOC?
What version of OpenSIPS are you using?
What is the total shared memory pool you are allocating?
What amount of memory are you using? (
Hey Kiddo,
Someday, when you grow up, you may realise that someone was just
trying to tell you - in a rather sagacious, terse manner - that the
question you were asking was not really of a manageable or addressable
scope, nor especially coherent.
You're actually much better off that way,
Hello Bogdan,
An mer., janv 20, 2010, Bogdan-Andrei Iancu schrieb:
opensipsl...@encambio.com wrote:
Here's a record I see when I run 'opensipsctl ul show':
AOR:: mylogin-osips
Contact::
sip:mylogin-os...@192.168.0.31:2310;transport=tls;line=2acy67zm Q=1
Hello Bogdan,
An mer., janv 20, 2010, Bogdan-Andrei Iancu schrieb:
opensipsl...@encambio.com wrote:
Because nat_traversal is newer and seems to be where most of the NAT
logic will be developed in the future, it seems the natural choice.
If I'm right about that,
That's not really true.
Hi!
Yes, with -DF_MALLOC.
1.6.1 from sources, I build deb package.
I use 128M of shared and 10*1024*1024 private memory (can increase - no
problem).
H, opensipsctl fifo get_statistics all crashes/stops the opensips.
'fifo uptime' or 'fifo debug' are OK.
strace while 'fifo get_statistics
Hi,
Right now I am using OpenSIPS 1.5.3 no-tls version in production.
Suddenly OpenSIPS got crashed.
I did check coredump but can not understand it.
Please help me out to interpret it.
Here is the pastebin link which contains output of bt.
http://pastebin.com/m49520853
Thanks in advance!!!
--
My guess is that there is not enough shared memory. When an allocation
failes OpenSIPS tries to defragment memory to make room which takes a
lot of time and must be done under lock.
Please try to increase the shared memory size and tell me if it persists.
Alex Massover wrote:
Hi!
Yes, with
On 09/12/09 18:09, Bogdan-Andrei Iancu wrote:
Hi Lorenzo,
check with opensipsctl ul show how your phone is registered - what is
important are the contact and received fields - which is present and
which contains private IPs.
Hi Bogdan!
first of all, thanks for the very late reply, been
yes, with same user-name and passwd i am able to login from opensips box
to remote mysql server.
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Sent from the OpenSIPS - Users mailing list archive at
On 16/11/09 15:49, Bogdan-Andrei Iancu wrote:
Hi Lorenzo,
Check the followings:
(1) your REGISTERS hits opensips (run an ngrep on opensips's machine)
(2) check where the reply from opensips is sent to (maybe it is sent to
wrong destination)
hi bogdan!
again, sorry for the late reply!
I'm having a problem that I think is the same as this discussion. When a call
from a natted user comes in to my opensips proxy, my config does auth and then
immediately fires up the nat ping and media proxy ( engage-mediaproxy() ) to
provide far end nat traversal. Now I'm trying to add top
Hello again,
An mer., janv 20, 2010, Bogdan-Andrei Iancu schrieb:
opensipsl...@encambio.com wrote:
Here's a record I see when I run 'opensipsctl ul show':
AOR:: mylogin-osips
Contact::
sip:mylogin-os...@192.168.0.31:2310;transport=tls;line=2acy67zm Q=1
Hello,
The next figure describes my problem.
openSIPS
+--+
-- INVITE(1) --| |--- INVITE(1) --
| |
-- INVITE(2) --| |--- INVITE(2) --
-- 180(2) --| |-- 180(2) --
-- 200(2) --| |-- 200(2)
I'd just like to say that I for one appreciate both Inaki's participation in
the group and his wit. If you can't take it, then perhaps you'd do better
off learning on your own. We are all happy to help (after all, we
participate because we enjoy it... well I do anyway), but we arn't going to
do
Hi Yannick,
indeed the issue is tricky as the media relay uses as a key (for the
streams) the SIP callid+from_tag+to_tag(optional) . so, it an early
stage, when to_tag is not yet know, the media relay cannot make
difference between the two branches of the same call.
My advice - do not do stop
This is an interesting question. The requests should have different tags
.I'm not sure how OpenSIPs handles the individual requests..
I'd tend to expect a 491 Request Pending or a 500 Requests Merged. But maybe
that's B2BUA behavior.
Since opensips isn't replying to the second INVITE, it's
Just for grins, can you try adding an entry in your /etc/hosts file for your
mysql server address?
The error specifically says it can't reverse resolve it:
Jan 11 07:01:15 localhost opensips: WARNING:core:fix_socket_list: could not
rev. resolve 200.200.100.11
BTW, is that really the IP you are
Hi Richard, the solution I am using is working, but the header was useless, the
only problem is that a second media_proxy session is still opened, but this is
not an issue
#--- Force media proxy
if(search_body(c=IN IP4 1.1.1.1) method == INVITE){
xlog(L_NOTICE,
Hi,
Is there any feature in drouting or lcr module to stop selecting a
failing gw for a particular amount of time? It would suffice to have a
ping mechanism or alternatively, mark the gateway as defunct in the
failure_route. I was thinking pinging mechanism is present at least in
lcr, but
Hi Andrew,
None of them do support pinging to GW, but I guess it will be a nice
feature for DR..
Regards,
Bogdan
Andrew Pogrebennyk wrote:
Hi,
Is there any feature in drouting or lcr module to stop selecting a
failing gw for a particular amount of time? It would suffice to have a
ping
Hello,
I'd like to implement the function Music on hold on Opensips.
Thank's a lot.
Mehdi
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OpenSIPS is a SIP proxy, not a media endpoint. So, it doesn't do that.
On 01/21/2010 12:00 PM, Mehdi Bouchefra wrote:
Hello,
I'd like to implement the function Music on hold on Opensips.
Thank’s a lot.
Mehdi
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Hi,
Now shared memory is 1G (-m 1024), and all memory is dedicated to the virtual
machine (it was shared till now).
But it still happens, just not so often.
I originate the calls for this stress test in Asterisk with the same resources
and looks like Asterisk performs much better than
Hello,
An jeu., janv 21, 2010, Alex Balashov schrieb:
On 01/21/2010 12:00 PM, Mehdi Bouchefra wrote:
I'd like to implement the function Music on hold on Opensips.
OpenSIPS is a SIP proxy, not a media endpoint. So, it doesn't do
that.
If while OpenSIPS routes a call a person presses 'hold' on
On Thursday 21 January 2010 17:30:36 opensipsl...@encambio.com wrote:
Hello,
An jeu., janv 21, 2010, Alex Balashov schrieb:
On 01/21/2010 12:00 PM, Mehdi Bouchefra wrote:
I'd like to implement the function Music on hold on Opensips.
OpenSIPS is a SIP proxy, not a media endpoint. So, it
Hello list,
An jeu., janv 21, 2010, opensipsl...@encambio.com schrieb:
An mer., janv 20, 2010, Bogdan-Andrei Iancu schrieb:
opensipsl...@encambio.com wrote:
Here's a record I see when I run 'opensipsctl ul show':
AOR:: mylogin-osips
Contact::
El Jueves, 21 de Enero de 2010, opensipsl...@encambio.com escribió:
If while OpenSIPS routes a call a person presses 'hold' on a
telephone, often it will send a INVITE (I think it's called a
REINVITE.) What about detecting this 'hold' REINVITE in the
route script and redirecting the message to
I actually was looking into this last night and found this post from Bogdan
http://n2.nabble.com/Request-for-Brain-storming-New-types-of-routes-in-config-tt2693231.html#a2693231
Bogdan mentions in the subject that he is just brainstorming about new types
of routing.
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Hi,
I'm running a statefull proxy that in most cases need to relay the calls to
a PSTN gateway.
After the migration to the Opensips 1.6.1, there is a problem with
compatibility / RR module and the gateway (Cisco AS5300).
Opensips does not relay 'correctly' (in my case) the ACK messages. The Cisco
Hi Everyone,
I'm trying to record calls using rtpproxy. i called call_recording() while
i get invite message and on onreply route.
as follows:
I) if (is_method(INVITE)){
force_rtp_proxy();
start_recording();
2) onreply_route[1] {
if ((isflagset(5) ||
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