Hello,
I'm getting the following error when compiling kamailio 4.1.
CC (gcc) [M gzcompress.so] gzcompress_mod.o
gzcompress_mod.c:42:18: fatal error: zlib.h: No such file or directory
compilation terminated.
make[1]: *** [gzcompress_mod.o] Error 1
make: *** [modules] Error 1
The
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] error compiling kamailio 4.1
Hello,
to compile, you have to install the devel package for gzip library.
Cheers,
Daniel
On 09/12/13 12:30, Grant Bagdasarian wrote:
Hello,
I'm getting the following error when compiling kamailio 4.1.
CC (gcc
Hello,
I'm seeing Kamailio generating 408 Request Timeout responses which are sent
back to the originator. And it also generates a CANCEL to the terminating
party. I assume this is normal behavior.
Which timer controls this timeout? Is the 408 generated only when no final
reply is received?
Hello,
I have my siptrace module configured with setflag(22) which make sure only the
forwarded messages are duplicated, and not the original messages.
This works great for requests, but for replies this doesn't work. Kamailio
duplicates both original and forwarded replies.
Is it possible to
Hello,
Using 4.0.4:
I'm getting the following error (removed the values part of the query):
3(2931) ERROR: db_unixodbc [dbase.c:133]: db_unixodbc_submit_query(): rv=-1.
Query= insert into Dialogs
at it.
Cheers,
Daniel
On 11/20/13 3:53 PM, Grant Bagdasarian wrote:
No, it has to be done in Kamailio. And since there is no way to append a
header to a CANCEL, I'm forced to do it another way.
I did notice after the CANCEL is sent by the UAC, the UAS generates a 487
Request Terminated. I'm
Hello,
Is it possible to have the sipcapture module write the duplicated messages to
multiple tables?
If not, is it possible to have the siptrace module duplicate the message to
multiple Sip Capture instances?
Regards,
Grant
___
SIP Express Router
That or replication would work, but I need it for redundancy. If the
triggered/replicated table would no longer be available due to an outage of the
database, then I would lose logging.
Would have been a cool feature to have the messages duplicated to multiple
capture servers or tables.
Hello,
I'm executing the following code for both the BYE and CANCEL in request_route:
if (is_method(BYE|CANCEL)) {
#finished
$var(finished) = $TS;
append_hf(X-Finished: $var(finished)\r\n);
}
For some reason, the X-Finished header is
Sent: Wednesday, November 20, 2013 3:09 PM
To: sr-users@lists.sip-router.org
Subject: Re: [SR-Users] append_hf to CANCEL
On 11/20/2013 09:05 AM, Grant Bagdasarian wrote:
Does append_hf not work for CANCEL requests?
Correct: CANCEL is a hop-by-hop request where the proxy is a distinct hop,
i.e
Sent: Wednesday, November 20, 2013 3:21 PM
To: sr-users@lists.sip-router.org
Subject: Re: [SR-Users] append_hf to CANCEL
If there is, I don't know it.
Out of curiosity, why would you want to do that?
On 11/20/2013 09:19 AM, Grant Bagdasarian wrote:
I see, so there is no way to append a header
.
-Original Message-
From: sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Alex Balashov
Sent: Wednesday, November 20, 2013 3:28 PM
To: sr-users@lists.sip-router.org
Subject: Re: [SR-Users] append_hf to CANCEL
On 11/20/2013 09:26 AM, Grant
Take a look at: http://www.kamailio.org/wiki/cookbooks/4.0.x/pseudovariables
It contains a list of variables you can use inside the kamailio.cfg script.
The modules page also provides a lot of information:
http://www.kamailio.org/docs/modules/4.0.x/
From: sr-users-boun...@lists.sip-router.org
Hello,
Any plans for building a module which generates a UUID or building it into the
core?
Daniel once told me to use $sruid, which basically returns a unique value. It
works good. I haven't had any duplicates yet, even under heavy load tests.
___
Hello,
Have these modules ever been tested under heavy load? For instance at 20+ CPS?
Any idea what the maximum performance is for these modules?
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
Hello,
I've setup two Kamailio machines, one which does all the processing and the
second one which always replies with a 500 Server Internal Error, to test my
Dispatcher fail-over.
When routing a call, the call is always routed to the second Kamailio first, to
test the fail-over.
What
@lists.sip-router.org
Subject: Re: [SR-Users] Apply changes made to sip reply in onreply_route
Am 08.10.2013 12:14, schrieb Grant Bagdasarian:
Hello,
I've setup two Kamailio machines, one which does all the processing and the
second one which always replies with a 500 Server Internal Error, to test
Hello,
Consider the following: the Dispatcher module has two destinations loaded.
Now let's say we have the following two scenarios:
Scenario 1:
Kamailio tries to relay the INVITE to the first destination selected by the
Dispatcher module, but this destination does not reply for some reason.
work. Same applies
to the subst functions, but I'm assuming I'm doing something wrong there.
Regards,
Grant
-Original Message-
From: Klaus Darilion [mailto:klaus.mailingli...@pernau.at]
Sent: Wednesday, October 2, 2013 9:13 PM
To: Kamailio (SER) - Users Mailing List
Cc: Grant Bagdasarian
in failure_route.
Could someone explain to me what the branch_routes are used for? Does it have
to do with forking?
-Original Message-
From: sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Grant Bagdasarian
Sent: Thursday, October 3
Of Olle E. Johansson
Sent: Thursday, October 3, 2013 9:44 AM
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] Replace header value in failure_route
3 okt 2013 kl. 09:38 skrev Grant Bagdasarian g...@cm.nl:
Could someone explain to me what the branch_routes are used for? Does
Hello,
Is it possible to append a new header to a reply generated by Kamailio and also
have it present when duplicating the message to a capture server?
At the moment the 603 Reply is duplicated to my capture server, but I don't
know how to append a new header, because the kamailio script stops
are added to the INVITE and 603 (Via only).
Why does this happen?
From: sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Grant Bagdasarian
Sent: Thursday, October 3, 2013 11:39 AM
To: sr-users@lists.sip-router.org
Subject: [SR-Users] append_hf to reply
the 483 Too Many Hops response?
From: sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Grant Bagdasarian
Sent: Thursday, October 3, 2013 12:31 PM
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] append_hf to reply generated by kamailio
Hello,
I'm trying to replace the value of a custom header in the failure_route, using
the subst function.
subst('/^X-Dispatcher:(.+)$/X-Dispatcher:
\$(avp(dsattrs){param.value,dispatcher})/i');
I also tried it with the subst_hf, but that didn't work either.
I'm getting the following message:
September 2013 08:01, Grant Bagdasarian g...@cm.nlmailto:g...@cm.nl
wrote:
Hello,
I've loaded the mi_rpc module and configured the rpc_url param, but I can't
issue the kamcmd mi ds_reload command from the web interface. Executing the
command in the console works just fine, but only when I leave
Hello,
I've loaded the mi_rpc module and configured the rpc_url param, but I can't
issue the kamcmd mi ds_reload command from the web interface. Executing the
command in the console works just fine, but only when I leave the binrpc
modparam as default.
I'm probably doing something wrong.
#
to reload table:
http://www.kamailio.org/docs/modules/stable/modules/dispatcher.html#idp16998512
Regards,
Charles
On 25 September 2013 11:40, Grant Bagdasarian g...@cm.nlmailto:g...@cm.nl
wrote:
Hello,
When adding rows to the dispatcher table, the table in memory needs to be
refreshed. Since we
alphanumeric values, so I can't specify a
full URL.
From: sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Grant Bagdasarian
Sent: Friday, September 27, 2013 10:04 AM
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] Communicate
TCP.
Regards,
Charles
On 27 September 2013 12:27, Grant Bagdasarian g...@cm.nlmailto:g...@cm.nl
wrote:
What am I doing wrong here?
# - xhttp params --
modparam(xhttp, url_match, null)
# - xhttp_rpc params --
modparam(xhttp_rpc, xhttp_rpc_root, http_rpc)
### Routing Logic
Of Grant Bagdasarian
Sent: Friday, September 27, 2013 1:39 PM
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] Communicate with Kamailio through external application
Right! Totally forgot about that.
Thanks, Kamailio is now receiving the requests!
From:
sr-users-boun...@lists.sip
up the MI and RPC commands.
Try:
kamctl fifo ds_reload
or
kamcmd dispatcher.reload
(note the difference between kamctl and kamcmd)
Regards,
Charles
On 27 September 2013 12:57, Grant Bagdasarian g...@cm.nlmailto:g...@cm.nl
wrote:
What kind of DB handler do I need for the dispatcher module
Hello,
When adding rows to the dispatcher table, the table in memory needs to be
refreshed. Since we have multiple Kamailio instances running on different
machines, I'd like to automate the process of reloading the table.
Is there an easy way of doing this with an already existing module, which
] On Behalf Of Daniel-Constantin
Mierla
Sent: Thursday, September 19, 2013 9:11 PM
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] dispatcher: set $du in failure_route
On 9/18/13 4:53 PM, Grant Bagdasarian wrote:
My Dispatcher table has been extended with more columns and also
from the
$avp(dsattrs) avp, so I can set the $du in failure_route.
From: sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Grant Bagdasarian
Sent: Friday, September 20, 2013 10:03 AM
To: mico...@gmail.com; Kamailio (SER) - Users Mailing List
=' + TerminatingSBCAddress + ';' + Attributes) AS
Attributes)
Note the commas been replaced by semicolons.
I hope this will help others with similar issues in the future.
From: sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Grant Bagdasarian
Sent: Friday
Hello,
I compiled and installed Kamailio 4.0.3 like usual on a new test machine, but
changed the CFGFILE variable in /etc/init.d/kamailio to
/usr/local/etc/kamailio/kamailio.cfg. Kamailio started normally.
When I reset CFGFILE to CFGFILE=/etc/kamailio/kamailio.cfg and start Kamailio
again,
: Thursday, September 19, 2013 10:24 AM
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] Modifying path to kamailio.cfg
Hello,
what operating system are you using? How do you start/stop kamailio?
Also, provide the logs here with the errors.
Cheers,
Daniel
On 9/19/13 9:24 AM, Grant
Hello,
Yes, that worked. Thanks!
Will this be pushed into the next (minor) release of Kamailio?
Regards,
Grant
From: Daniel-Constantin Mierla [mailto:mico...@gmail.com]
Sent: Thursday, September 19, 2013 11:57 AM
To: Grant Bagdasarian; Kamailio (SER) - Users Mailing List
Subject: Re: [SR
Hello,
Is it possible to store the weight attribute in the dispatcher table? If so,
which column must I use?
I see it is possible when using the list file.
The attrs_avp is the closest thing, but I think this value has to be loaded
from the database also?
Regards,
Grant
weight - some notes are in the
readme, but I guess can be made more clear:
-http://kamailio.org/docs/modules/stable/modules/dispatcher.html#idp16940048
As example, the attrs can be:
weight=20;abc=xyz;
Cheers,
Daniel
On 9/18/13 10:29 AM, Grant Bagdasarian wrote:
Hi,
Do you mean using
Hello,
Can the $du value also be set in a failure_route? For instance in the case the
first destination fails.
I'm currently setting the $du value before I'm calling the ds_next_domain
function in request_route.
We have multiple carriers which are connected to different SBCs.
In case $du is
.
Grant Bagdasarian g...@cm.nlmailto:g...@cm.nl wrote:
Hello,
Can the $du value also be set in a failure_route? For instance in the case the
first destination fails.
I’m currently setting the $du value before I’m calling the ds_next_domain
function in request_route.
We have multiple carriers which
: sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Alex Balashov
Sent: Wednesday, September 18, 2013 4:23 PM
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] dispatcher: set $du in failure_route
Of course.
Grant Bagdasarian g
Hello,
Is this page still up-to-date:
http://www.kamailio.org/dokuwiki/doku.php/development:write-module ?
I'm researching what it takes to create our own custom Kamailio module.
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
/2013 06:41 AM, Grant Bagdasarian wrote:
Hello,
Is this page still up-to-date:
http://www.kamailio.org/dokuwiki/doku.php/development:write-module ?
I'm researching what it takes to create our own custom Kamailio module.
1. Skilled C programming.
2. What do you want to accomplish? Are you sure
/020977.html
Cheers,
Daniel
On 8/20/13 1:33 PM, Grant Bagdasarian wrote:
It's going to be a custom routing module, with our own business logic.
It should load the routes from a database and keep refreshing the data every
x interval.
For SIP-to-PSTN calls (outgoing to carrier) we need to be able
the
onreply_route route, but there is no command which relays this to its
destination, or is this done by the core behind the scenes?
Regards,
Grant
-Original Message-
From: sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Grant Bagdasarian
instances.
Cheers,
Daniel
On 8/8/13 4:43 PM, Grant Bagdasarian wrote:
That's too bad. Any idea if this already on the backlog for future Kamailio
versions?
I guess the only option left is to use the EXEC module to run the uuidgen
command to get a new UUID?
-Original Message-
From
for a dynamic GUID, not a constant UID for an
instance.
Grant Bagdasarian g...@cm.nlmailto:g...@cm.nl wrote:
Nice!
Thanks!
-Original Message-
From:
sr-users-boun...@lists.sip-router.orgmailto:sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org
Hello,
I'm adding some additional SIP headers to the INVITE received, but when I use
the siptrace module to duplicate the messages to a capture server, the INVITE
does not contain the additional headers.
I'm using a softphone to test my configuration, and it looks like Kamailio only
sends the
] Duplicate relayed INVITE message using siptrace not
working
Grant,
Have you tried msg_apply_changes() after the header append to see if that helps?
Grant Bagdasarian g...@cm.nlmailto:g...@cm.nl wrote:
Hello,
I’m adding some additional SIP headers to the INVITE received, but when I use
not
working
Try: setflag(22);
Grant Bagdasarian g...@cm.nlmailto:g...@cm.nl wrote:
I just tried it and it did the trick. Thanks!
I was also looking at the documentation of the siptrace module and found this:
by setting the flag equal with the value of 'trace_flag' (e.g.,
setflag(__trace_flag__
-Users] Duplicate relayed INVITE message using siptrace not
working
On 08/09/2013 10:11 AM, Grant Bagdasarian wrote:
That does it also. Thanks!
Sure thing. Bear in mind that Kamailio is wildly inconsistent about when it
requires a numerical parameter (e.g. a flag) as a string literal (in double
Hello,
Is it possible to generate a UUID inside the kamailio script? Is there a module
available which does this already or do I need to call something external from
the script?
Regards,
Grant
___
SIP Express Router (SER) and Kamailio (OpenSER) -
AM, Grant Bagdasarian wrote:
Hello,
Is it possible to generate a UUID inside the kamailio script? Is there
a module available which does this already or do I need to call
something external from the script?
Regards,
Grant
___
SIP Express
-Original Message-
From: sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Grant Bagdasarian
Sent: Tuesday, August 6, 2013 3:49 PM
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] Workings of the kamailio dispatcher module
Ahh
dispatcher module
Grant,
On 08/07/2013 05:00 AM, Grant Bagdasarian wrote:
How does the dispatcher module handle the priorities in the dispatcher
table? For example:
ID;Set;Destination;Priority
1;1;192.168.16.10;1 2;1;192.168.16.11;2
Which one will be chosen first? I would assume, the higher
Hello,
Consider the following Kamailio script:
route {
route(DISPATCH);
route(RELAY);
}
route[DISPATCH] {
ds_select_domain(1, 8);
return;
}
Dispatcher Table
SetID
Destination
1
192.168.1.10
1
192.168.1.11
Algorithm 8 uses the first
dispatcher module
On 08/06/2013 09:00 AM, Grant Bagdasarian wrote:
Hello,
Consider the following Kamailio script:
route {
route(DISPATCH);
route(RELAY);
}
route[DISPATCH] {
ds_select_domain(1, 8);
return;
}
Dispatcher
Hello,
Which module can I use to have Kamailio generate a CANCEL request when it
receives a certain reply code? I want to cancel a dialog when Kamailio receives
a 181 Call Forwarded.
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
most requests -- are for user agents
to send, not proxies.
-- Alex
On 07/04/2013 03:30 AM, Grant Bagdasarian wrote:
Hello,
Which module can I use to have Kamailio generate a CANCEL request when
it receives a certain reply code? I want to cancel a dialog when
Kamailio receives a 181 Call
So if I use the t_cancel_callid function, Kamailio sends a CANCEL to the UAS,
but doesn't send anything to the UAC? Or do I need to handle that myself?
-Original Message-
From: sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Olle E.
Mailing List
Subject: Re: [SR-Users] Sending CANCEL
4 jul 2013 kl. 09:58 skrev Grant Bagdasarian g...@cm.nl:
So if I use the t_cancel_callid function, Kamailio sends a CANCEL to the UAS,
but doesn't send anything to the UAC? Or do I need to handle that myself?
The UAS that receives the CANCEL
Hello,
I don't know if this has already been reported, but I've downloaded kamailio
4.0.2 and tried to compile it with db_unixodbc. I got the following error:
CC (gcc) [M db_unixodbc.so] _con.o
_con.c:34:17: fatal error: con.h: No such file or directory
compilation terminated.
Hello,
I need to query a database for every SIP request coming into Kamailio, but I
want this to be handled as fast as possible, so I was thinking of loading the
data I need in memory using the HTABLE or MTREE modules.
When the SIP request is coming from one of our carriers, the called number
Hello,
The description of this module says it hides the SIP routing headers, does this
also include the Via headers?
Grant
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
Hello,
I understand that machines processing RTP are recommended to be physical
machines. Does the same apply for machines just proxy'ing RTP?
Also, after how many RTP session would you start noticing issues with RTP when
using rtpproxy on a virtual VMware machine?
Regard,
Grant
/2013 07:44 AM, Grant Bagdasarian wrote:
Hello,
I understand that machines processing RTP are recommended to be
physical machines. Does the same apply for machines just proxy'ing RTP?
Also, after how many RTP session would you start noticing issues with
RTP when using rtpproxy on a virtual
-Users] Topoh module also masking Via?
It does indeed.
On 05/01/2013 06:57 AM, Grant Bagdasarian wrote:
Hello,
The description of this module says it hides the SIP routing headers,
does this also include the Via headers?
Grant
___
SIP
: Wednesday, May 1, 2013 3:12 PM
To: sr-users@lists.sip-router.org
Subject: Re: [SR-Users] Topoh module also masking Via?
On 05/01/2013 08:54 AM, Grant Bagdasarian wrote:
Let's say we are using SEMS in SBC mode and letting the traffic pass
through it, so no new call leg is created.
It doesn't work like
Hello,
Is it possible to have Kamailio rewrite the Request URI of the INVITE message
but sent the INVITE to another address first?
For example
- Kamailio (10.0.0.1) receives an INVITE
- Kamailio rewrites the RURI to 10.0.0.3
- Kamailio sends the INVITE to 10.0.0.2
On 04/29/2013 04:32 AM, Grant Bagdasarian wrote:
Hello,
Is it possible to have Kamailio rewrite the Request URI of the INVITE
message but sent the INVITE to another address first?
For example
-Kamailio (10.0.0.1) receives an INVITE
-Kamailio rewrites the RURI to 10.0.0.3
-Kamailio
using two phones
and only one made it through.
Is this a bug?
Regards,
Grant
From: sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Grant Bagdasarian
Sent: Monday, April 22, 2013 10:55 AM
To: sr-users@lists.sip-router.org
Subject: [SR-Users
agent fails to add the parameter when it sends the subsequent requests,
the dialog will not be properly tracked and you could end with a bunch of
active unfinished dialogs in memory.
Regards,
On Thu, Apr 25, 2013 at 8:13 AM, Grant Bagdasarian
g...@cm.nlmailto:g...@cm.nl wrote:
Hello,
Another
(SER) - Users Mailing List
Subject: Re: [SR-Users] Kamailio dialog termination
Hello,
On 4/25/13 3:07 PM, Grant Bagdasarian wrote:
Hello,
I've set the default_timeout to 3600 seconds, which is 1 hour.
In the database the dialog record shows:
StartTimeTimeout
1366886241 1366929441
-router.org
[mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Grant Bagdasarian
Sent: Thursday, April 04, 2013 4:58 PM
To: sr-users@lists.sip-router.org
Subject: [SR-Users] sip_capture columns
Hello,
I couldn’t find any documentation on what the table columns of the sip_capture
module
Hello,
I couldn't find any documentation on what the table columns of the sip_capture
module are used for. Most of them are straightforward, but I still don't know
what the following columns mean and what values to expect:
- PID_user
- Proto
- Family
-
Hello,
Is there a maximum length/size defined for the From and To Tag?
The RFC states: it MUST be globally unique and cryptographically random with
at least 32 bits of randomness.
Though I can't find anything in the RFC about the maximum.
The sip_capture table stores the to_tag and from_tag in
:42 AM
To: Kamailio (SER) - Users Mailing List
Cc: Grant Bagdasarian
Subject: Re: [SR-Users] Tag maximum size/length
On 3/28/13 11:21 AM, Grant Bagdasarian wrote:
Hello,
Is there a maximum length/size defined for the From and To Tag?
The RFC states: “it MUST be globally unique
@lists.sip-router.org
Subject: Re: [SR-Users] Need help understanding/separating signaling from media
Hi Grant,
On 01/28/2013 11:03 AM, Grant Bagdasarian wrote:
I think that the RTP Proxy module does precisely this, but what I
still don't understand is how the rtp stream is passed between
can try via modparam, setting 'authorization_column' to [authorization]. If not, rename that column and set this modparam to the new column name.
Cheers,
Daniel
On 2/28/13 3:02 PM, Grant Bagdasarian wrote:
I meant the columns, not the values.
Insert into sip_capture([id],….,[authorization
Hello,
We are using SQL Server 2008 as our database. When the sip_capture module tries
to insert a row, the following error is given:
db_unixodbc [con.c:220]: unixodbc:SQLExecDirect=42000:1:156:[FreeTDS][SQL
Server]Incorrect syntax near the keyword 'authorization'.
Using Kamailio 3.3.4.
Is
I meant the columns, not the values.
Insert into sip_capture([id],,[authorization])
Values(1,.auth goes here)
From: sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Grant Bagdasarian
Sent: Thursday, February 28, 2013 2:59 PM
To: sr-users
it was running? Can it be eventually reproduced?
Also, to be sure is not an issue already fixed, upgrade to latest 3.3.x, you
don't have to change anything in database or config file.
Cheers,
Daniel
On 2/19/13 2:38 PM, Grant Bagdasarian wrote:
Forgot to mention the version:
version: kamailio
Hello,
Last Thursday our kamailio process crashed. There are a lot of memory related
errors in there. See attachment for details.
Did the machine run out of memory? How do I prevent this from happening again?
The machine has 4GB of memory.
Regards,
Grant
2013-02-14T12:31:24+01:00 kamailio01
From: sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Grant Bagdasarian
Sent: Tuesday, February 19, 2013 2:36 PM
To: sr-users@lists.sip-router.org
Subject: [SR-Users] Errors leading to kamailio proc crash
Hello,
Last Thursday our kamailio process
Could someone please take a look at my previous question below?
-Original Message-
From: Grant Bagdasarian
Sent: Tuesday, February 5, 2013 1:12 PM
To: sr-users@lists.sip-router.org
Subject: RE: [SR-Users] Need help understanding/separating signaling from media
Hello,
Consider the below
signaling from media
Hi Grant,
On 01/28/2013 11:03 AM, Grant Bagdasarian wrote:
I think that the RTP Proxy module does precisely this, but what I
still don't understand is how the rtp stream is passed between these
two agents for each call.
The Kamailio 'rtpproxy' module goes out
media
Hi Grant,
On 01/28/2013 11:03 AM, Grant Bagdasarian wrote:
I think that the RTP Proxy module does precisely this, but what I
still don’t understand is how the rtp stream is passed between these
two agents for each call.
The Kamailio 'rtpproxy' module goes out to the rtpproxy service[1
numbers of concurrent calls, on a non-oversubscribed
hypervisor.
Still, it is not recommended from a best practical point of view, definitely
not.
Grant Bagdasarian g...@cm.nl wrote:
Hello Alex,
Thank you for the explanation. It's clear to me now.
One more thing, I've read that it's not recommended
Hello,
A while ago I came across the siptrace and sipcapture modules and thought this
would be a good way to generate logging out of all the SIP messages that are
passed through Kamailio and stored in the siptrace/sipcapture table.
My idea was to extract certain values from SIP messages and
software is suitable for these
requirements? We are currently using Asterisk, which works great, but perhaps
there are others available which are more suitable for these requirements.
Thanks in advance!
Regards,
Grant Bagdasarian
___
SIP
Hello,
We have a cluster of three Asterisk machines. Each machine answers an incoming
call and transfers it to unique remote destination.
So, Asterisk01 transfers to Destination01, Asterisk02 to Destination02,
Asterisk03 to Destination03.
The Asterisk machines are allowed to receive and answer
cluster
12 dec 2012 kl. 10:18 skrev Grant Bagdasarian g...@cm.nlmailto:g...@cm.nl:
Hello,
We have a cluster of three Asterisk machines. Each machine answers an incoming
call and transfers it to unique remote destination.
So, Asterisk01 transfers to Destination01, Asterisk02 to Destination02
Hello,
I've been searching the internet to find an explanation on how SIP transfer
works using Re-INVITE and/or UPDATE, but I can't seem to find a good source.
From what I understand(and this is the way we do it), the following happens:
Bob=Caller
Alice=Called
John=Transfer party
1)Bob
a=25 n=dlg_manage
Debug is currently set to 2: debug=2
Should I set this to 4?
Grant
From: Daniel-Constantin Mierla [mailto:mico...@gmail.com]
Sent: maandag 19 november 2012 9:59
To: Grant Bagdasarian
Cc: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users
Mailing List
Subject: Re
Hello Daniel,
Yes, dlg_manage() is being called in the BYE, yet it sometimes still occurs
that dialogs aren't being cleared.
It is critical for us the dialogs are being cleared as we have an application
which heavily relies on this.
Kind regards,
Grant Bagdasarian
From: Daniel-Constantin
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