Daniel-Constantin Mierla wrote:
On 4/13/10 11:49 AM, alexis heron wrote:
Iñaki Baz Castillo wrote:
2010/4/13 alexis heron alexis.he...@crihan.fr:
As a suggestion, if you are interested in matching just the RURI
username part then I strongly recommend you doing so as follows:
if ( $rU =~
Daniel-Constantin Mierla wrote:
On 4/13/10 11:59 AM, alexis heron wrote:
Daniel-Constantin Mierla wrote:
On 4/13/10 11:49 AM, alexis heron wrote:
Iñaki Baz Castillo wrote:
2010/4/13 alexis heron alexis.he...@crihan.fr:
As a suggestion, if you are interested in matching just the RURI
Laszlo wrote:
failure_route[FAIL_ONE] { is closed before your if ( $rU =~ ^9[0- .
2010/4/13 alexis heron alexis.he...@crihan.fr
mailto:alexis.he...@crihan.fr
Daniel-Constantin Mierla wrote:
On 4/13/10 11:59 AM, alexis heron wrote:
Daniel-Constantin Mierla
2010/4/13 alexis heron alexis.he...@crihan.fr:
help me please,
how to make a SIPtrunk??
Please, open a new thread for a new topic.
Anyhow, I recommend you reading the full documentation. There is not a
magic concept of a SIPtrunk.
--
Iñaki Baz Castillo
i...@aliax.net
Hello,
I have installed from source version 3.0.1 of Kamailio on an OpenSuSE 10
Linux system.
Currently I am attempting to enable persistence in my Kamailio with the
help of MySQL.
In the config file kamailio.cfg I have enabled MySQL as follows
#!define WITH_MYSQL
In the
2010/4/13 alexis heron alexis.he...@crihan.fr:
Please, open a new thread for a new topic.
Anyhow, I recommend you reading the full documentation. There is not a
magic concept of a SIPtrunk.
yes but with if(uri=~...) it's good, no??
There you are checking if the Request URI matches something.
Hi Alexis ou alors, salut Alexis,
Je vais le faire en français, cela t'aidera peut-être.
Tu a la chance d'avoir les conseils de Daniel et d'Inaki, suis les, fait
ce qu'ils te disent de faire.
Le harcèlement sur une mailing-list te mèneras juste à ne plus avoir de
réponses.
Un bon conseil, va
Iñaki Baz Castillo wrote:
2010/4/13 alexis heron alexis.he...@crihan.fr:
Hi,
I would like to make a SIPTrunk with kamailio to redirect to a 9xx numbers
in Cisco Call manager, can you help me please.
Here's what I did but it does not work.
Please specify the exact error you have. It
alexis heron wrote:
Iñaki Baz Castillo wrote:
2010/4/13 alexis heron alexis.he...@crihan.fr:
Hi,
I would like to make a SIPTrunk with kamailio to redirect to a 9xx
numbers
in Cisco Call manager, can you help me please.
Here's what I did but it does not work.
Please specify the
I'd like to build the latest version of sip-router from source but include
server modules from the modules_s directory. I thought the command 'make
include_modules=modules_s/permissions modules_s/pike modules_s/ratelimit
modules_s/sms modules_s/sanity ' would do the trick but it doesn't.
Hello all,
I need to have kamailio put its hostname on its via append instead of its IP
address. How can I do this or is it possible?
Thank you,Eric
_
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On 4/13/10 8:39 PM, Alex rsm wrote:
Hi,
My Kamailio server uses an external mysql database with perl script to
proxy traffic.
When mysql server is unreachable, Kamailio do not respond with 100
Trying and instead responses with 484 (Address Incomplete) to the
INVITE message.
Is this a
Hi
I need some guidelines to troubleshoot the following issue:
a) A is behind NAT
b) B is not behind NAT
c) A calls B, SIP INVITE is sent over TCP
d) A's firewall does NAT and changes the source port to let's say p1
e) B releases the call and sends BYE over UDP
f) Kamailio sends the BYE to
Klaus
I attached the ngrep you asked me to this email.
Regards,
Pascal
On Wed, Apr 14, 2010 at 12:21 PM, Klaus Darilion
klaus.mailingli...@pernau.at wrote:
The contact after fix_nated_contact() should also contain ;transport=tcp.
Thus, Kamailio should relay the BYE with TCP.
Can you
alexis heron wrote:
Hi,
I have a problem to routing with kamailio. When I restart kamailio I
have this error message :
0(710) : core [cfg.y:3328]: parse error in config file
/usr/local/etc/kamailio/kamailio.cfg, line 640, column 2-3: syntax error
0(710) : core [cfg.y:3328]: parse error in
On Wednesday 14 April 2010, Pratab Ali wrote:
I read the reply from Daniel and as he suggested I got kamailio 3.0.0
using GIT this morning. However, I don't know if I actually got (Commit:
3a25f8327c on 3.0). I am unfamiliar with GIT as I use CVS.
I will recheck to see if I got the branch
Pratab Ali wrote:
Hi,
Thanks, I just followed your git instructions and it shows that I have
Daniel's change.
I will try a clean rebuild and run again.
Thanks.
pratab
If a clean build doesn't solve can you add to modules/db_mysql/Makefile
a -lm (link the math library) to the LIBS
On Wednesday 14 April 2010, Pratab Ali wrote:
Your suggestion with regards to appending -lm after -lz worked. I no
longer get the undefined symbol: log error. Thanks!
However, now I have kamailio unable to connect to mysql because I've not
told it to use the correct port. This I can fix from
i call drop() in branch route and it is the only branch left. the
branch gets correctly dropped, i.e., the request is not send out, but
the reply to UAC is strange:
U 2010/04/14 21:03:43.750712 192.98.102.10:5060 - 192.98.103.2:5074
SIP/2.0 477 Unfortunately error on sending to next hop occurred
2010/4/14 Juha Heinanen j...@tutpro.com:
i call drop() in branch route and it is the only branch left. the
branch gets correctly dropped, i.e., the request is not send out, but
the reply to UAC is strange:
U 2010/04/14 21:03:43.750712 192.98.102.10:5060 - 192.98.103.2:5074
SIP/2.0 477
Daniel-Constantin Mierla writes:
drop in branch_route was missing, as Andreas Granig reported few days
ago. I just committed the patch, can you test again?
daniel,
ok, i'll try. if failure route is set when i call drop(), will it be
executed and if so with which reply code?
-- juha
Hi Juha,
On 4/15/10 11:31 AM, Juha Heinanen wrote:
Daniel-Constantin Mierla writes:
drop in branch_route was missing, as Andreas Granig reported few days
ago. I just committed the patch, can you test again?
daniel,
ok, i'll try. if failure route is set when i call drop(), will it be
2010/4/15 Panagiotis Skoulikaritis psk...@algonet.gr
Dear list members
I'm using kamailio 3
and the problem that I'm facing is that when a call is rejected by global
blacklist, kamailio does not generate an accounting record.
Is this normal behavior or I need to configure something
Hello,
I started the page that should collect notes about how to upgrade from
version 3.0.x to 3.1.0:
http://sip-router.org/wiki/install/3.0.x-to-3.1.x
3.1.0 is not yet released, target date is September 2010, right now is
the code on master branch in GIT repository. However, this page will
#!define BLAH
means that BLAH is enabled, in the sense that #!ifdef BLAH check will
succeed.
hi daniel,
thank you for the reply
but how do i enable the #!define?
please givee me the commanted line and the uncommented line
thanks
Le 17/04/2010 10:26, Daniel-Constantin Mierla a écrit :
Den 15.04.2010 13:27, skrev Klaus Darilion:
How would you have written it?
I prefer this logic:
...
if (loose_route()) {
if (!has_totag()) {
xlog(L_WARN,$ci loose_route request without to-tag, 403...\n);
sl_send_reply(403, out-of-dialog loose_route not allowed);
exit;
}
...
Maybe the solution here is to generate a synthetic custom request with
$uac_req() and uac_send_req(), like with method SYNTHETIC_BYE?
The problem is that I cannot populate it with the From URI, Call-ID,
etc. because that information is not available in the
timeout-triggered route to begin
On 04/19/2010 03:28 AM, Iñaki Baz Castillo wrote:
2010/4/19 Alex Balashovabalas...@evaristesys.com:
2) The spoofed BYEs that are generated by the proxy with this setting do not
show up as sequential requests, nor are catchable in script at all. As a
result, BYE events are not written to CDRs
On 04/19/2010 03:50 AM, Alex Balashov wrote:
On 04/19/2010 03:36 AM, Iñaki Baz Castillo wrote:
AFAIR the main purpose of local_route was to handle the locallly
generated BYE for accounting purposes.
I defined an event_route[tm:local-request] but it does not appear to
fire in this scenario.
Juha,
what was the conclusion regarding this? did the problem go away when
you called nat traversal functions both on 180 ringing and 183 session
progress?
Yes. The problem did go away once we started forcing rtpproxy on 180
Ringing also.
Thanks and Regards,
Vikram.
Juha-
Vikram had posted a while back on the thread where he explained a bit more
about his resolution and the fact the issue
was occurring with VoipSwitch but not asterisk. Let us know if you want a link
to that.
Thanks for following up.
-Jeff
Original Message
Alex Balashov writes:
However, we all have to make pragmatic concessions to
the realities of real-world operation, which I assume is the
motivation for dialog timeouts, dlg_bye(), and other perversions from
the point of view of a purist. :-)
I welcome your thoughts and
On 04/21/2010 05:43 AM, Juha Heinanen wrote:
just route your calls via sems as a b2bua and make it send the byes, if
media stops flowing. if something is missing in order to allow that to
be easily implemented by dsm or ivr plugins, i'm sure stephan and
raphael will take care of it.
The
Hi,
Alex Balashov wrote:
1) Inline B2BUA in the signaling path of all calls;
1a) Make it do SSTs; or
1b) Make it relay media, too, and hang up the call (bidirectional BYE)
on RTP receive timeout;
[...]
Needless to say, I am interested in the option that requires the least
work but
Hello,
On 4/21/10 12:23 PM, Kelvin Chua wrote:
hi daniel,
i'm not using git version. so maybe i'm missing some patches. can you
confirm if what i am
experiencing is the same problem and the fix is indeed available from
the git version? thanks
I recommend using at least 3.0.1, as a matter
2010/4/21 Daniel-Constantin Mierla mico...@gmail.com:
Hello,
are you using the latest git version of branch kamailio_3.0? It was a fix to
dialog after the 3.0.0 release, adding some sanity checks for broken
messages:
(gdb) bt
#0 0x2ab61b62779a in update_dialog_dbinfo (cell=0x2ab61c9100f8) at
dlg_db_handler.c:501
#1 0x2ab61b628ea8 in dlg_onreply (t=0x7d5228, type=value optimized
out, param=value optimized out) at dlg_handlers.c:361
#2 0x2ab617965505 in run_trans_callbacks_internal
Hello,
yesterday I committed to git new feature that allow write access to
branch attributes. It is about branches kept in core before TM takes
over and creates the branches for transaction. These branches are filled
when you do location lookup, alias db lookup or append_branch in route
I'm definitely favouring the B2BUA+SST approach here.
I really, really don't want to proxy media unless I have to; it makes
the bandwidth bill much higher, it lengthens many users' network path to
reach the carriers, and for no benefit whatsoever except in the case of
far-end NAT traversal,
Yes, it is complex if the B2BUA does not have a media relay ( in which
case providing the SDP (at least the port information) itself).
Am 22.04.2010 16:01, schrieb Stefan Sayer:
or, other possibility, empty reinvite:
A b2b B
|---INVITE / SDPa--|
Hello,
thanks, so the Contact header is missing, that makes the 200ok invalid
for invitate - afaik contact is mandatory. Anyhow, crash should not
happen, but I wonder what happens with such call, practically the caller
does not know where to send the BYE. Or maybe 18x reply has contact hdr
Hi Rajnikant,
you may also have a look at Siremis web management interface for
Kamailio, a light-weight application that you can use for accounting
purposes too.
http://siremis.asipto.com/install-accounting/
For a quick look at Siremis, here you can get a demo with
username=admin,
On 04/23/2010 05:18 AM, Raphael Coeffic wrote:
On 23.04.10 09:51, Alex Balashov wrote:
On 04/22/2010 10:01 AM, Stefan Sayer wrote:
]
clean? I am not so sure any more, trying to hack something together to
see where this gets. Is there a clean and simple method to do re-invite
from the b2bua,
On 04/23/2010 06:14 AM, Raphael Coeffic wrote:
On 23.04.10 11:58, Alex Balashov wrote:
On 04/23/2010 05:18 AM, Raphael Coeffic wrote:
On 23.04.10 09:51, Alex Balashov wrote:
On 04/22/2010 10:01 AM, Stefan Sayer wrote:
]
clean? I am not so sure any more, trying to hack something together to
On 04/23/2010 07:42 AM, Raphael Coeffic wrote:
Ok, so even the standard 60 minutes expire would be an improvement ;-)
Oh yes.
--
Alex Balashov - Principal
Evariste Systems LLC
1170 Peachtree Street
12th Floor, Suite 1200
Atlanta, GA 30309
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web:
Hi, I must deal with a B2BUA which keeps the original SDP unchanged
except the fact that it removes the a=nortpproxy:yes line added by
RtpProxy.
The B2BUA intercommunicates two Kamailio, both forcing its own RtpProxy server.
I've not tested it again but expect there will be no RTP as each
You can try use the flags r and f for this case.
These flags can force rtpproxy to send a RTP.
You can see:
http://www.kamailio.org/docs/modules/1.5.x/nathelper.html#id2468157
--- Em sex, 23/4/10, Iñaki Baz Castillo i...@aliax.net escreveu:
De: Iñaki Baz Castillo i...@aliax.net
Assunto:
Hi everyone,
In the page where a module is described, there is a list of dependencies that
have to be met in order to use the module, the list of exported parameters that
might need to be set in the configuration file and a list of exported functions
which can be used in the configuration
Alex-
On 04/23/2010 06:14 AM, Raphael Coeffic wrote:
On 23.04.10 11:58, Alex Balashov wrote:
On 04/23/2010 05:18 AM, Raphael Coeffic wrote:
On 23.04.10 09:51, Alex Balashov wrote:
On 04/22/2010 10:01 AM, Stefan Sayer wrote:
]
clean? I am not so sure any more, trying to hack something
On 04/23/2010 08:10 AM, Jeff Brower wrote:
But... is it being considered to add functionality to rtpproxy so it can send
something asynchronously to Kamailio which either sends BYEs or does something
to
cause the endpoints to do so? Currently, as far as I know, rtpproxy only
responds to
Alex-
But... is it being considered to add functionality to rtpproxy so it can
send
something asynchronously to Kamailio which either sends BYEs or does
something to
cause the endpoints to do so? Currently, as far as I know, rtpproxy only
responds to
commands from nathelper;
Hello,
On 4/23/10 2:03 PM, hector.or...@swisscom.com wrote:
Hi everyone,
In the page where a module is described, there is a list of
dependencies that have to be met in order to use the module, the list
of exported parameters that might need to be set in the configuration
file and a list
Am 23.04.2010 13:32, schrieb Juha Heinanen:
Iñaki Baz Castillo writes:
I've not tested it again but expect there will be no RTP as each
RtpProxy will wait for RTP coming from the other. Am I right?
hopefully you are not right. it must be possible to have a chain of
rtproxys for
Hi,
in r1821 you can find my first shot at this scenario, a b2bua which
enables SST on both sides, and does the SDP ping pong as below.
I am sure that there are many cases which are not handled properly,
for example, if we have started one INVITE ping-pong from the B2B,
then we should not
2010/4/23 Klaus Darilion klaus.mailingli...@pernau.at:
Am 23.04.2010 13:32, schrieb Juha Heinanen:
Iñaki Baz Castillo writes:
I've not tested it again but expect there will be no RTP as each
RtpProxy will wait for RTP coming from the other. Am I right?
hopefully you are not right.
2010/4/24 Iñaki Baz Castillo i...@aliax.net:
So let's suppose Kamailio-1 (1.1.1.1) using RtpProxy-1 (9.9.9.1) and
Kamailio-2 (2.2.2.2) using RtpProxy-2 (9.9.9.2), and also a
transparent SIP proxy between them (Proxy-X with IP 5.5.5.5).
- Kamailio-1 receives an INVITE from a client and forces
Iñaki,
I would be curious to know what practical circumstances demand such a
convoluted topology.
--
Alex Balashov - Principal
Evariste Systems LLC
1170 Peachtree Street
12th Floor, Suite 1200
Atlanta, GA 30309
Tel: +1-678-954-0670
Fax: +1-404-961-1892
On Apr 24, 2010, at 5:48 AM, Iñaki Baz
2010/4/24 Juha Heinanen j...@tutpro.com:
Iñaki Baz Castillo writes:
If these rtpproxy use the same IP as the Kamailio's SIP signalling IP,
then there is no problem and no need to use -r flag.
inaki,
does it work in the common case where there is two sip proxys and two
rtpproxies
2010/4/24 Alex Balashov abalas...@evaristesys.com:
I read that, but was curious about the larger picture; why do you need the
transparent proxy/B2BUA?
It's the accounting softswitch in my company. All the SIP calls to the
PSTN must go through it, even if the destination number is also a
local
On 04/24/2010 04:49 PM, Iñaki Baz Castillo wrote:
http://www.youtube.com/watch?v=4nKh8bg3eHo
That one is one of the best ones made.
But it'd be hilarious if someone really made one about Kamailio, LOL.
Mein Führer, OpenSIPS has crossed the river here, here, and here...
Joe Hart at my
Den 20.04.2010 10:52, skrev Klaus Darilion:
Hi Espen, can you provide the traces with proper line break?
E.g. I always use
ngrep -W byline -t -q -P any-pattery port 5060
It seems that Kamailio/RDP-proxy where not the root of my problems, the
new FFA from Digium partly solved the issue.
Am 24.04.2010 15:47, schrieb Iñaki Baz Castillo:
2010/4/24 Alex Balashovabalas...@evaristesys.com:
I read that, but was curious about the larger picture; why do you need the
transparent proxy/B2BUA?
It's the accounting softswitch in my company. All the SIP calls to the
PSTN must go through
Am 24.04.2010 11:25, schrieb Juha Heinanen:
Iñaki Baz Castillo writes:
If these rtpproxy use the same IP as the Kamailio's SIP signalling IP,
then there is no problem and no need to use -r flag.
inaki,
does it work in the common case where there is two sip proxys and two
rtpproxies
2010/4/26 Klaus Darilion klaus.mailingli...@pernau.at:
As I said, using -r is the solution. In fact, I thing that -r
should be the default value, as it makes more sense RtpProxy to
prefill the media address with the address in the SDP rather than the
SIP_RECEIVED_IP:SDP_PORT (which is mostly
The htable size is static; it doesn't dynamically resize itself up to
the next power of 2 after a certain utilisation threshold, right?
--
Alex Balashov - Principal
Evariste Systems LLC
1170 Peachtree Street
12th Floor, Suite 1200
Atlanta, GA 30309
Tel: +1-678-954-0670
Fax: +1-404-961-1892
On Monday 26 April 2010, Alex Balashov wrote:
The htable size is static; it doesn't dynamically resize itself up to
the next power of 2 after a certain utilisation threshold, right?
Hi Alex,
i think this is a static setting, as the memory is allocated on startup in
shared memory. Do you've
On 04/26/2010 07:31 AM, Henning Westerholt wrote:
On Monday 26 April 2010, Alex Balashov wrote:
The htable size is static; it doesn't dynamically resize itself up to
the next power of 2 after a certain utilisation threshold, right?
Hi Alex,
i think this is a static setting, as the memory is
Am 26.04.2010 12:19, schrieb Iñaki Baz Castillo:
2010/4/26 Klaus Darilionklaus.mailingli...@pernau.at:
As I said, using -r is the solution. In fact, I thing that -r
should be the default value, as it makes more sense RtpProxy to
prefill the media address with the address in the SDP rather
On 04/26/2010 07:48 AM, Henning Westerholt wrote:
On Monday 26 April 2010, Alex Balashov wrote:
i think this is a static setting, as the memory is allocated on startup
in shared memory. Do you've run into performance problems or memory size
constraints with the htable module?
None whatsoever,
On Monday 26 April 2010, Alex Balashov wrote:
There is probably a certain point on which further increase of the htable
size make not that much sense anymore. It seems the module only supports
from 256 to 16384 buckets in the htable. So depending on the distribution
of the hash function
We are facing a periodically problem with a server that is running Kamailio
1.5.2.
I think that it is not the case, but I will ask anyway. Some clients, sometimes,
send us messages with sintax errors. This machine needs to be rebooted every
week. It stops working completely. I don't think that
I'm sure this has been asked before, but I couldn't find the answer
anywhere when I googled it.
I have a temporary situation where I need to terminate calls to a
carrier with user/pwd authentication, and not through the IP
authentication. As I understand it this is not a normal setup
situation
Hello all,
the email address for old iptel.org mailing list
serus...@lists.iptel.org (and serus...@iptel.org) is now forwarded to
the new sip-router project mailing list sr-us...@lists.sip-router.org.
Pavel
iptel.org
___
SIP Express Router (SER)
Iñaki Baz Castillo writes:
However after checking the module code it seems that there are just
two cases:
- IP addresses (mask = 32).
- Subnets (mask != 32).
So first the source address is always matched against he address hash,
and if it doesn't match then it is matched against
Hello,
can you paste here the sip message and the result of the substitution?
It will help to troubleshoot if is something wrong there.
Cheers,
Daniel
On 4/27/10 2:35 AM, Brandon Armstead wrote:
Hello All,
Correction, it seems both the last supplied regex and xlog(L_INFO,
[$ci] m=audio
Hi!
I have a problem with Kamailio. Sometimes (can not catch the moment
and conditions of) situations arise when, calling from AAA to uri XXX
call comes to uri YYY. The call comes from PSTN to the gray subnet of
user-agents registered in kamailio. As the PSTN we using Audiocodes
mediant.
Daniel,
Here is the XLOG output, the top log is the unmodified version and the
bottom is the modified version.
xlog(L_INFO, [$ci] $rb) OUTPUTS:
Apr 29 20:17:23 sip-core02 /sbin/kamailio[23550]: [
6db72a2f-7e263...@192.168.1.75] v=0#015#012o=- 24986155 24986155 IN IP4
Hi to all.
I'm new here.
I'm getting crazy with presence_xml error.
I'm using kamailio 1.5.4 notls with xlite 3.0 (winXP client) softphone and
postgresql 8.4 as db backend.
Seems that presence works, but i have error in parsing xml:
those are the relevant log:
Apr 29 16:42:54 ocs
Hi, i have a problem about the handling of the cancel message.
The call flow is this:
A - (Invite)Proxy (P)
B
(100 Tryng) ---
-(Invite) B
(100 Tryng)
Hello all,
the email address for old iptel.org mailing list
serus...@lists.iptel.org (and serus...@iptel.org) is now forwarded to
the new sip-router project mailing list sr-us...@lists.sip-router.org.
Pavel
iptel.org
___
SIP Express Router (SER)
Am 30.04.2010 11:37, schrieb Iñaki Baz Castillo:
2010/4/30 Klaus Darilionklaus.mailingli...@pernau.at:
200 OK seems correct as long as the transaction is still in memory.
http://tools.ietf.org/html/rfc3261#section-9.2
I don't agree. As per RFC 3261 when a proxy receives a 200 for an
INVITE
2010/4/30 Klaus Darilion klaus.mailingli...@pernau.at:
I don't agree. As per RFC 3261 when a proxy receives a 200 for an
INVITE the transaction is terminated so a CANCEL after the 200 should
not match such transaction. Then the proxy should reply 481 to the
CANCEL rather than a 200.
If
2010/4/30 Iñaki Baz Castillo i...@aliax.net:
But anyway, if the INVITE has been replied with a 200 it makes no
sense at all to send a CANCEL so the proxy shouldn't reply 200 to the
CANCEL, but a 481.
Not exactly, sorry, the proxy should forward the CANCEL stateless:
16.10 CANCEL Processing
Iñaki Baz Castillo wrote:
2010/4/30 Klaus Darilion klaus.mailingli...@pernau.at:
200 OK seems correct as long as the transaction is still in memory.
http://tools.ietf.org/html/rfc3261#section-9.2
I don't agree. As per RFC 3261 when a proxy receives a 200 for an
INVITE the transaction is
Hello,
On 4/30/10 11:07 AM, Alex Rendour wrote:
Hi,
We need to add header on the failure route.
Exemple:
On failure 404 Not found, we need add header
Reason: Q.850; cause=1
We tried this in failure_route:
if(t_check_status(404)){
xlog(L_ERR, 404 Not found\n);
2010/4/30 Jiri Kuthan j...@iptel.org:
I don't agree. As per RFC 3261 when a proxy receives a 200 for an
INVITE the transaction is terminated so a CANCEL after the 200 should
not match such transaction.
That's a bug in the RFC and we shall not better projects RFC bugs in
implementations :) A
I'm attempting to run an existing SER config file under a recent build of
sip-router. I get a syntax error on the t_on_failure(noroute) call that
exists in my config file. The specific error is bad expression: type mismatch
(str instead of int). I presume this means sip-router does not
Hello,
On 4/30/10 7:58 PM, Steven C. Blair wrote:
I'm attempting to run an existing SER config file under a recent build
of sip-router. I get a syntax error on the t_on_failure(noroute)
call that exists in my config file. The specific error is bad
expression: type mismatch (str instead of
The error is produced within the following block per the config file syntax
check -c.
if (method==INVITE !...@to.tag) {
t_on_failure(FAILURE_ROUTE);
}
From: Daniel-Constantin Mierla [mailto:mico...@gmail.com]
Sent: Friday, April 30, 2010 2:09 PM
To: Steven C. Blair
Cc:
Hello All,
Can openser be used as a SIP Application Server in an IMS infrastructure
?
Should I use Kamailio instead ?
thanks,
--Jignesh
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
2010/4/30 Jignesh Gandhi jignesh.gan...@moviuscorp.com:
Hello All,
Can openser be used as a SIP Application Server in an IMS infrastructure ?
Should I use Kamailio instead ?
By definition a SIP proxy is not a SIP Application Server(which in
most cases are B2BUA's).
However if you mean a IMS
On 4/30/10 8:11 PM, Steven C. Blair wrote:
The error is produced within the following block per the config file
syntax check -c.
if (method==INVITE !...@to.tag) {
t_on_failure(FAILURE_ROUTE);
}
not using selects in this way but via PV framework, but I think
On 04/30/2010 02:12 PM, Jignesh Gandhi wrote:
Hello All,
Can openser be used as a SIP Application Server in an IMS infrastructure ?
Should I use Kamailio instead ?
Jignesh, note that these days when you are talking about Kamailio and
OpenSER, they are one and the same.
--
Alex Balashov -
Hi Brandon,
you are missing the multi-line matching flag, try:
xlog(L_INFO, [$ci] $(rb{re.subst,/^(.*)m=audio ([0-9]+)(.*)$/\2/s}))
Note the 's' after the last '/' in subst expression.
Cheers,
Daniel
On 4/29/10 10:21 PM, Brandon Armstead wrote:
Daniel,
Here is the XLOG output, the top log
Hello,
On 4/26/10 5:59 PM, Tristan Mahé wrote:
Hi guys,
I was debugging an install on kamailio 3.0, and was asked to add per
user custom outbound proxies.
To do so, I used load_credentials from auth_db to store the custom
route in an avp, as you can see in the relevant part below.
I
Daniel-
On 4/30/10 8:24 PM, Iñaki Baz Castillo wrote:
2010/4/30 Jignesh Gandhijignesh.gan...@moviuscorp.com:
Hello All,
Can openser be used as a SIP Application Server in an IMS infrastructure ?
Should I use Kamailio instead ?
By definition a SIP proxy is not a SIP Application
Hello,
a short note that the merging of SIP Express Routers (SER) users mailing
list into sr-users is completed from subscriber point of view:
- posts to serus...@lists.iptel.org are directed to
sr-users@lists.sip-router.org
- posts to sr-users@lists.sip-router.org are received by subscribers
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