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--|
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
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
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
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
Am 30.04.2010 22:31, schrieb Jignesh Gandhi:
thank you for your reply.
Would you know of anything that can be used as a SIP Application Server with
capability to do REFER, RE-Invites and
Asterisk
parse
SIP-I payload passed to the server.
Not out of the box. But probably you could to
I have no idea ... :-(
Am 04.05.2010 15:41, schrieb Uriel Rozenbaum:
Thanks Guys, I'll be trying this. Do you know if I can use AVP or vars
as parameters for these functions?
|encode_contact(encoding_prefix)
||decode_contact()|
On Mon, May 3, 2010 at 5:47 AM, Klaus Darilion
klaus.mailingli
Jared Martin wrote:
Ok, this is probably an easy question:
Say I have my kamailio server set up and all of my voip clients are
gleefully calling each other... but now I want to connect them to the PSTN.
Can I set up kamailo to trunk calls using a few grandstream gateways? or
is Asterisk
Daniel-Constantin Mierla wrote:
Hello,
On 5/5/10 9:52 AM, Francisco José Méndez Cirera wrote:
Hello,
I've downloaded Kamailio 3.0.0 (the last release) and I´ve seen it´s
possible downloading a binary tar.gz or the source to compile
directly. I would like to know if the binary has enabled
Hi!
indiver wrote:
Hello Every one,
My network setup is one to one nating of public and private ip. I want to
run rtpproxy behind the NAT with in the local IP,but in vain. can't create
listener error appears when i run rtpproxy on local ip. Can we run rtpproxy
If there is an error message
Hi Pascal!
The TCP keepalive is not an application layer keep alive, but Kamailio
just sets the SO_KEEPALIVE socket option. The the TCP keep alive is
implemented and performed by the operating systems.
Thus, if you want to tweak the TCP keep alive, then you have to tweak
the kernel.
for
tcp_keepalive_intvl and similar (tcp_keepidle ...)
regards
klaus
Cheers
Pascal
On Mon, May 10, 2010 at 2:33 PM, Klaus Darilion
klaus.mailingli...@pernau.at mailto:klaus.mailingli...@pernau.at wrote:
Hi Pascal!
The TCP keepalive is not an application layer keep alive, but
Kamailio just
Are you using newest 3.0 ? (git checkout)
I remember I once also had a problem with PVs which had stored old
values, but I can not remember naymore which PV it was.
regards
klaus
Am 11.05.2010 14:59, schrieb Andreas Granig:
Hi,
Some interesting behavior with kam-3.0, which I haven't
Right now, I would say that lookup should reset any existing dst_uri if
received is null.
Sounds reasonable.
regards
klaus
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Am 12.05.2010 16:57, schrieb Alex Balashov:
Yep, it will,
Someone who has verified that should add it to the core cookbook:
http://sip-router.org/wiki/cookbooks/core-cookbook/devel
thanks
klaus
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Hi Jerome!
As your config and the log messages seems to be correct, you should
debug on the network and verify which SIP messages are sent to which
location. You can use tcpdump/wireshark or the simple (my preferred
choice) ngrep tool.
1. install ngrep
apt-get install ngrep
2. capture the
Am 18.05.2010 16:29, schrieb Nelson Pereira:
I downlaoded the source of 3.0.1 and untar’ed.
Then executed as per docs (INSTALL file):
make prefix=/
make prefix=/ install
I guess you are overwriting the default prefix (/usr/local/) with /.
regards
Klaus
PS: If you are using a Debian
Am 21.05.2010 14:33, schrieb Nelson Pereira:
The problem is that the Kamailio receives a 302 Moved Temporarily with a
contact field of CONTACT: sip:1...@10.98.6.5:5065;transport=TCP
I need to have Kamailio, use this contact field and re-send the invite.
How can this be done?
Use
Hi!
Please post the complete SIP trace of the scenario which does not work,
this means:
incoming INVITE, outgoing INVITE, incoming response, outgoing response
also you can increase loglevel (debug=3 for kamailio 3.0, debug=4 for
kamailio 1.5) and watch syslog.
regards
klaus
On 21.05.2010
I just tested with Asterisk trunk version and it does not have this bug.
Maybe you should update your Asterisk server.
regards
klaus
Am 27.05.2010 11:50, schrieb Ernest Mavrel:
Internet Protocol, Src: 80.81.82.83 (80.81.82.83), Dst: 123.124.125.126
(123.124.125.126)
User Datagram Protocol,
Am 27.05.2010 17:09, schrieb Ricardo Coelho:
I have 2 machines running openser and some phones are registered on openserA
and some on openserB.
Why are some phones registered at openserA and others at openserB?
- Do they have different SIP domains where one SIP domain point to A and
other
. But you still have not answered my question. How do you achieve that
some phones are registered to proxyA and others to proxyB?
regards
klaus
Thanks
On May 27, 2010, at 5:05 PM, Klaus Darilion wrote:
Am 27.05.2010 17:09, schrieb Ricardo Coelho:
I have 2 machines running openser
Am 01.06.2010 19:02, schrieb Iñaki Baz Castillo:
2010/6/1 Klaus Darilionklaus.mailingli...@pernau.at:
Shouldn't it use strncmp()?
Probably it should.
regards
Klaus
btw: how is if ($au!=$tU) compared?
The problem is that more SIP username related stuff is also case
insensitive in
Hi!
Does somebody know any SIP clients (hard/soft) which support
out-of-dialog REFERs?
thanks
klaus
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FYI - the summary from SIPit
regards
Klaus
Original-Nachricht
Betreff: [RAI] SIPit26 summary
Datum: Fri, 4 Jun 2010 13:50:50 -0500
Von: Robert Sparks rjspa...@nostrum.com
An: r...@ietf.org
SIPit 26 was hosted by Edvina and TANDBERG in Kista, Sweden
the week of May 17-21,
On Tue, 2010-06-08 at 08:45 +0200, Klaus Darilion wrote:
You have to install and configure a media relay, e.g. rtpproxy or
mediaproxy.
Just take a look at the default script, it contains configuration for
rtpproxy.
regards
Klaus
Am 08.06.2010 07:38, schrieb Information:
hi,
I have configured
routing decisions and t_relay(), e.g.
route {
...sanity checks
...authentication
...NAT traversal
...routing decisions
if ($rd = 1.2.3.4) { #ip address of gateway
$rd = domain.com;
}
t_relay();
exit;
}
regards
Klaus
//Anders
On Wed, Jun 9, 2010 at 12:28 PM, Klaus Darilion
Am 10.06.2010 03:12, schrieb JinKevin:
AppSer01:root@/usr/local/kamailio-3.0/sbin$ kamctl add 2000 2000
usage: tail [+/-[n][lbc][f]] [file]
tail [+/-[n][l][r|f]] [file]
INFO: user '2000' already exists
between your kamctl command and the error message (user '2000' already
exists)
If you use the default configuration phone A just have to dial
sip:us...@ip.address.of.serverb
and it works out of the box.
regards
Klaus
Am 10.06.2010 16:57, schrieb SR-USER:
Hi everyone.
I have two machines running OpenSER, one machines runs OpenSER A and the other
runs OpenSER B. Phone
Am 17.06.2010 15:51, schrieb Ján ONDREJ (SAL):
WWW-Authenticate: DIGEST
realm=BroadWorks,qop=auth,algorithm=MD5,nonce=BroadWorksXgajlo4shTqsrkgdBW.
looks like the header is buggy, it should be Digest:
challenge = (Digest LWS digest-cln *(COMMA digest-cln))
In
Just the theory.
The proxy - if stateful forwarding is used, e.g. by using t_relay() -
must not forward the CANCEL unless a provisional reply is received from
the cisco phone.
regards
klaus
Am 17.06.2010 21:49, schrieb David:
Hey,
I am using a Cisco WIP310 wifi phone. Seeing as wifi is
Hi!
TM only handles retransmissions of single transaction.
The ACK is a new transaction, thus there won't be any retransmissions by
tm. tm will forward the retransmitted 200 OK and the client should
retransmit ACK.
Maybe you have a problem with NATs and the ACK is routing falsely. Take
a
That's true.
Openser/Kamailio does not decode URI automatically - it has to be done
manually in script.
In your case you need to replace check_from and check_to with manual
checks, which compare unescaped To/From URIs with unescaepd
authentication user, see:
IIRC the problem is that the string is interpreted as pattern, thus
pua matches pua_dialoginfo and pua.
So you have to specify pua separately.
regards
Klaus
Am 25.06.2010 15:28, schrieb David:
Hey,
Is this format still valid in Kamailio 3.0 ?
modparam(module1|module2|module3, db_url,
Am 25.06.2010 17:11, schrieb David:
Hey,
Is it possible that Kamailio 1.5.x silently discarded modparams that
were not valid and Kamailio 3.0 doesn't ?
I do not think so. At least I had similar problem several years ago.
regards
klaus
David
On 2010-06-25 10:11, Klaus Darilion wrote
IIRC force_send_socket operates on branch[0]. Now it depends how the
the new branch is added, e.g. if send--socket properties are copied into
new branch or not. IRRC there were some changes either in 1.5 or 3.0.
You can also access a branch's aprameter directly:
The log does not tell us anything.
If you want to know from us if your config is wrong or your
clients/server is buggy, then we need an ngrep dump recorded at the SIP
proxy:
ngrep -Wbyline -t -q port 5060
regards
Klaus
Am 29.06.2010 12:37, schrieb Ole Kaas:
Hello,
See transmission below.
Am 01.07.2010 15:14, schrieb Ján ONDREJ (SAL):
But I con't find any response in ACK packet send to our provider:
The ACK is just to finalize the rejected INVITE. Kamailio should send
another INVITE request with proper credentials.
regards
Klaus
Am 01.07.2010 17:40, schrieb Claudio Furrer:
Am 01.07.2010 14:38, schrieb Claudio Furrer:
Hi, I have a similar issue,
It is not possible to debug this issue without full SIP trace!
ngrep -Wbyline -t -d any port 5060
I find it unpleasant to read such a trace. Please really use ngrep to
Am 02.07.2010 15:54, schrieb Claudio Furrer:
Hi, I have a similar issue,
It is not possible to debug this issue without full SIP trace!
ngrep -Wbyline -t -d any port 5060
I find it unpleasant to read such a trace. Please really use ngrep to get a
nice formated SIP trace, or supply the
If you allways route the requests from the broken caller to the other
gateway, you can use something like that:
...
if (loose_route() {
if (src__ip=10.10.10.128) {
$rd=10.20.20.153;
$du=sip:10.20.20.153:5060;
}
... normal loose-route processing
t_relay();
exit;
}
...
Obiously the device is buggy.
But there is also on strange thing at Kamailio: it retransmits the
CANCEL although it has already received 481 response on the CANCEL request.
regards
klaus
Am 06.07.2010 00:06, schrieb David:
Hello,
Here are the corrected attachments, I sent the email from
Am 06.07.2010 17:04, schrieb Andrei Pelinescu-Onciul:
On Jul 06, 2010 at 16:55, Klaus Darilionklaus.mailingli...@pernau.at wrote:
Am 06.07.2010 16:36, schrieb Andrei Pelinescu-Onciul:
On Jul 06, 2010 at 14:06, Klaus Darilionklaus.mailingli...@pernau.at wrote:
Obiously the device is
, but only after 101-199 response.
regards
Klaus
David
On 2010-07-06 11:39, Klaus Darilion wrote:
Am 06.07.2010 17:04, schrieb Andrei Pelinescu-Onciul:
On Jul 06, 2010 at 16:55, Klaus
Darilionklaus.mailingli...@pernau.at wrote:
Am 06.07.2010 16:36, schrieb Andrei Pelinescu-Onciul:
On Jul
there is no difference between
sip-router 3.0 branch and Kamailio 3.0. The only difference is that
Kamailio has releases - that means there is a tarball and debian
packages available too.
regards
Klaus
Le mercredi 07 juillet 2010 à 09:17 +0200, Klaus Darilion a écrit :
Am 06.07.2010 20:03, schrieb inge
Am 08.07.2010 00:53, schrieb David:
Hey,
I do not do anything IP level forwarding. All my forwarding is done
using Kamailio. It looks like what I am doing is called hairpin routing.
I think the correct term is spiral - at least the RFC uses this term.
How do you forward the request -
Am 06.07.2010 20:48, schrieb Lucas Alvarez:
Hi, I new with kamailio, I've been able to integrate kamailio 3.02
with asterisk 1.6. The only issue I'm having is if I have to restart
asterisk( for some config update) I loose all the sip registration in
asterisk, is there any way of fixing this?
Am 08.07.2010 18:10, schrieb Daniel-Constantin Mierla:
Hello,
On 7/8/10 5:59 PM, Matteo Campana wrote:
Hi all,
I'm using kamailio 1.5 with TLS module.
I need to make ENUM query and get NAPTR record.
From NAPTR lookup, I'd like to relay my SIP Invite with tls protocol.
How can I tell
that?
Regards,
Daniel
Il 08/07/2010 18.45, Matteo Campana ha scritto:
Messaggio originale
Oggetto:Re: [SR-Users] Kamailio and NAPTR lookup with TLS
Data: Thu, 08 Jul 2010 18:44:27 +0200
Mittente: Klaus Darilion klaus.mailingli...@pernau.at
A: Daniel-Constantin
great, don't forget to update the wiki and mention the limitations ;-)
Am 13.08.2010 08:20, schrieb Daniel-Constantin Mierla:
Hello,
I committed in master branch (to be 3.1.0) code that allows to set uri,
username, domain and display name for To and From headers using
assignments to their
Hi!
For your scenario the domainpolicy module is not needed, and actually it
is obsolete and not maintained anymore.
regards
klaus
truong ngoc THANH wrote:
hi all,
I try to configure domainpolicy but it does not work.
===
my scenarios is :
1: i configure multi domain with kamailio,
I think you have multiple issues:
(not sure if I am right as I have to guess your network setup)
1. If Kamailio is an Application Layer Gateway between the public and
the internal network, then of course Asterisk should listen only on the
internal interface. Thus in sip.conf add:
increase the log level and verify if force_rtp_proxy is called during
response processing or not. If yes - there is a problem in
force_rtp_proxy. If no - you have to debug your configuration by adding
more xlog() messages, e.g. to log the value of the relevant flags and
the status.
regards
kamailio = 3.0 is based on ser's core which has implemented its own
caching resolver library. I do not know if there is a way to disable it
complete and use the system's stub resolver. You could try the
use_dns_cache option:
Hi!
I tried to summarize the options at
http://sip-router.org/wiki/cookbooks/core-cookbook/devel#dns_parameters
Please review and fix if something is wrong.
btw: are the tuning options like dns_try_ipv6, dns_retr_time,
dns_retr_no, dns_use_search_list also used when the internal
Am 02.09.2010 17:46, schrieb Andrei Pelinescu-Onciul:
ipv6_only and
prefer_ipv6 are obeyed only if the cache is used.
they do not even exists on the Wiki :-(
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The REGISTER request contains the pre-loaded route set. Pre-loaded
route sets are for security reasons disabled in the default config
(to-tag check):
if (has_totag()) {
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route()) {
Am 06.09.2010 11:00, schrieb Olle E. Johansson:
Route sets and REGISTER doesn't work well together. From RFC3261 section 10.3:
Registrars MUST ignore the Record-Route header field if it is included in a
REGISTER request. Registrars
MUST NOT include a Record-Route header field in any response
Am 06.09.2010 11:19, schrieb peter_green lion:
i have the same problem when add user-privkey.pem in SIP client, I use
3CX soft phone.
You have to import the self-signed certificate of the root CA which
signed the server certificate. Maybe cakey.pem ?
Probably you have to read some
Hi!
I have some problems with 3.0 dispatcher module:
module configuration:
# - dispatcher -
/* enable failover support */
modparam(dispatcher, flags, 2)
modparam(dispatcher, dst_avp, $avp(i:271))
modparam(dispatcher, grp_avp, $avp(i:272))
modparam(dispatcher, cnt_avp, $avp(i:273))
If
Am 07.09.2010 20:58, schrieb Daniel-Constantin Mierla:
Hello,
you have to set the ping interval parameter:
http://kamailio.org/docs/modules/stable/modules_k/dispatcher.html#id2590104
I see that documentation says its default value is 10, but in sources is
0, which means this feature is
Hi!
I use acc module with extra accounting.
modparam(acc, db_extra, direction=$avp(direction))
I set $avp(direction) during request processing (in main request route).
The value stored with INVITE and BYE is fine, but ACK requests do not
have the assigned value, but the same value as the
Hi!
Is DB accounting (mysql) asynchronous? I.e. if non-local DB is slow -
does this affect Kamailio's message routing?
thanks
Klaus
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Am 09.09.2010 10:17, schrieb peter_green lion:
hi all,
i have configure tls support as this link:
http://www.kamailio.org/docs/tls-devel.html#id2451496
and i add certificate to 3CX sip phone is cacert.pem but when i
register sip phone, the log file in kamailio server is :
Sep 9 15:13:36
I always set the fr_timer to a smaller value as I want to detect gone
UASs much faster. This also solves the overlapping OPTIONS issue as tm
gives up after 2 seconds.
# - tm params -
/* we want fast failover in case a destination does not answer
Note: fr_timer takes milliseconds */
Am 13.09.2010 11:10, schrieb peter_green lion:
enable_tls=1
tcp_async=no
listen=tls:192.168.1.81:5060
The default is for TLS is port 5061.
modparam(tls, tls_method, TLSv1)
modparam(tls, tls_method, SSLv23)
You can not use TLS and SSL - only on e or the other. SIP is
standardized with
Am 13.09.2010 17:27, schrieb peter_green lion:
and when caller call callee, i use ngrep to catch package through port
5060/5061 I see the TLS package, it is not a sip message. so I thing the
problem is futher.and only 3CX phone can register with my server.
Instead of ngrep you should use
Try putting force_rtp_proxy() into 2 different branch_routes. Then
activate the first branch_route during normal route processing, and
activate the second branch_route in failure_route.
regards
klaus
Am 15.09.2010 08:32, schrieb Ernest Mavrel:
Hi
I have problem with force_rtp_proxy.
Am 20.09.2010 21:21, schrieb Santiago Soares:
Hello,
We use Kamailio as a stateless load balancer.
It runs on a VM, with 512 MB RAM, and it seems to have some kind of
memory leak.
The kamailio processes consume all available memory, until Kamailio
crashes and restart.
What do you mean with
Am 21.09.2010 17:12, schrieb Ovidiu Sas:
Hello all,
I would like to propose the removal of force_rtp_proxy function from
the rtpproxy (k) module.
Instead, the rtpproxy_offer/rtpproxy_answer should be used.
Why?
- force_rtp_proxy no longer support 200ok/ACK SDP negotiation ('s'
flag was
Am 22.09.2010 15:31, schrieb Santiago Soares:
Thank's for the answers.
Based on your explanation, I think that the problem is an external library.
The kamailio process is taking all the system memory.
When the system runs out of memory, it kills kamailio.
But how can I find out what is the
Am 28.09.2010 10:15, schrieb Rouskol Andrey:
Hello,
I have a problem running kamailio with tls support.
I can successfully register only for the first time, all further registration
fails with the following message in the log:
Sep 28 11:53:55 siptest /usr/sbin/kamailio[16963]: ERROR: tls
You are right.
Thanks for fixing my bugs :-)
Klaus
Am 30.09.2010 17:27, schrieb Juha Heinanen:
now that 3.1 has async tls support, i decided (first time ever) to try
to test tls. things went quite smoothly when i followed Create
Certificates to be used with Kamailio document
Am 30.09.2010 17:27, schrieb Juha Heinanen:
now that 3.1 has async tls support, i decided (first time ever) to try
to test tls. things went quite smoothly when i followed Create
Certificates to be used with Kamailio document
Thanks, great!
regards
Klaus
Am 12.10.2010 09:36, schrieb Jon Bonilla (Manwe):
Starting today, Kamailio's debian repository offers nightly development
versions of master and 3.1 branches for Ubuntu and Debian distributions.
Brief summary of the repository status at the moment:
* Flavours:
Am 15.10.2010 02:59, schrieb Joe Uelk:
Hello all,
I'm looking to implement the following scenario:
Step 1
SIP Server A sends INVITE to port 5060 over IPv6 to Kamailio:
2001::1 --udp/tcp-- 2001::2:5060
Step 2
Kamailio SIP NATs the INVITE and sends it out IPv4 to SIP Server B on
port 6000
On 10/27/2010 12:43 PM, Daniel-Constantin Mierla wrote:
Now, if it is easier for you to count hex codes by hart in a combined
content, then it is fine. For me is easier to get the body in text
format and use wc tool to count.
Except it is hard to say if there is CR, CRLF, LF (whatever) at the
On 10/29/2010 11:55 AM, Sergey Okhapkin wrote:
Kamailio 3.1 log is filled with lines like
Oct 29 03:32:46 west /usr/local/sbin/kamailio[632]: INFO:script: incoming
reply from udp:188.62.4.186:65333 SIP/2.0 404 Not Found Via: SIP/2.0/UDP
204.74.213.5:5060;branch=0
Am 01.11.2010 19:00, schrieb Jijo:
Hi all.
What changes i have to make in the build to use tls from modules_s
instead from core modules.
Why do you want to do that?
Thanks
Jijo
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Am 04.11.2010 12:04, schrieb Mino Haluz:
Hi,
I tried to add RPID headers at the beggining of SIP options on the
client side, and I was able to change the caller number simply this way.
How can I avoid this ? I'm searching for some function in Kamailio which
would delete all RPID headers so
Am 05.11.2010 09:10, schrieb MÉSZÁROS Mihály:
Hello all!
Is there any known problem to run multiple kamailio/sip-router instance
in one host.
No. Just start kamailio several times and provide a different config
file for each:
kamilio -f /etc/kamailio/kamailio-foobar.cfg
kamilio -f
Am 22.11.2010 14:28, schrieb Komáromi Péter:
Hello!
Daniel thanks for your help! I'd need a little bit more now, because
I have to make the concrete plan for this IPv4-IPv6 SIP communication
before I could ask for any modification on the server. There is not
Anyway, you should not install
Am 23.11.2010 18:25, schrieb Johny Kadarisman Kwan:
Hi there,
Is there a way to relay call to secure gateway or ITSP.
ie, invite being challange to provide authorization (username/password)
Not sure what you mean: Kamailio shoudl authorize the incoming INVITE
and then forward the INVITE to
the second invite to gateway/itsp,
the gateway itself is another kamailio that want to authorize that
invite. so sort of
client kamailioA -- kamailio B
So kamailioA, act as a client for kamailioB
On Tue, Nov 23, 2010 at 1:11 PM, Klaus Darilion
klaus.mailingli...@pernau.at wrote:
Am
On 23.11.2010 21:22, Joe Hart wrote:
I'm trying to figure out if there is some way to identify a call for
which an rtpproxy session has been set up when the call is cancelled.
I can't check the flag I set for the call, nor can I check the route
param I set (which is what I'm doing to figure
On 23.11.2010 19:58, Uriel Rozenbaum wrote:
Hey Guys,
I was wondering if anyone knows some Windows Based softphone to do
some tests on my Kamailio 1.5.4 deployment that will allow insertion
of custom headers or at least Remote-Party-ID and/or
P-Asserted-Identity.
Not the answer you want to
You can try todays release of QjSimple 0.6.5:
http://www.ipcom.at/telefonie/qjsimple/
It allows you to specify 2 arbitrary headers to be added to the INVITE.
regards
Klaus
Am 23.11.2010 19:58, schrieb Uriel Rozenbaum:
Hey Guys,
I was wondering if anyone knows some Windows Based softphone to
Am 29.11.2010 01:24, schrieb Eric Hiller:
Hello all, I have seen some other complaining about this issue, but the
posts were over 4 years ago so I would think xlite would be fixed at
this point. I wanted to ensure my formatting was correct before I
started trying other phones.
All I am trying
Am 29.11.2010 15:04, schrieb Komáromi Péter:
So it listens only on IPv4 address socket. There is no IPv6 socket. Is it the
normal working, if the interface has both IPv4 and IPv6 address?
If yes, how can I manage the registering from the IPv6 agents?
Have you added an IPv6 address to the
If you do not want to authenticate the requests then disable
authentication kamailio.cfg
regards
Klaus
Am 29.11.2010 12:53, schrieb Andrés S. García Ruiz:
Hi everybody,
I'm trying to deploy an IMS network with OpenIMSCore and Kamailio. Since
OpenIMSCore has been already tested along with
Am 29.11.2010 15:27, schrieb Andrés S. García Ruiz:
Thanks for your comment,
This is my configuration, could you please tell me how to disable
authentication?
Open it in a text editor and search for authent
I am sure you will find the respective route. Depending on your actual
setup
Am 30.11.2010 13:04, schrieb Komáromi Péter:
Hi again,
My kamailio 1.5 is working well and I'm able to create the session
between an IPv4 and an IPv6 UA, but there is no RTP session. That's
the point where the rtpproxy comes in. I installed it, and made a
bridging (I hope it is):
Am 30.11.2010 21:52, schrieb Daniel-Constantin Mierla:
On 11/30/10 1:07 AM, Eric Hiller wrote:
No, I do not have an outbound proxy set.
Maybe there is some other hidden setting. AFAIK, there is a special code
that you can dial in X-Lite to get advanced settings window -- just
google it.
Maybe you have to call t_on_failure() in failure route again?
klaus
Am 01.12.2010 16:22, schrieb Anders:
Hi there,
Kamailio 1.5:
I cannot find an answer to this in the documentation: Is there a limit
to how many fail-overs Kamailio does? I have lcr setup for a
destination where it is
Am 02.12.2010 13:06, schrieb Komáromi Péter:
Hi!
So if you say it is possible to solve the problem with the only
location table, the location_inet4 and location_inet6 is not
certainly necessary... do I _have to_ use the 4to6.cfg file from the
source of kamailio, or not?
When you call
Am 07.12.2010 15:20, schrieb Bernhard Suttner:
Hi,
I am using Kamailio with usrloc, nathelper and register module (and
some other). All the data will be stored within a MySQL database. The
contact address will be stored within the database. Is it somehow
possible to store the SOURCE-IP of a
Am 09.12.2010 18:55, schrieb Jijo:
Hi Andrei,
I'm observing TCP Recv buffer getting full when we are doing a load run
(30cps) on Proxy with TCP. Basically the congestion is happening on the
TCP connection from SIP Server to Proxy. I beleive kamailio is not
processing the message fast enough,
rls is extra_excluded in debian rules file:
# extra modules to skip, because they are not compilable now
# - regardless if they go to the main kamailio package or to some module
package,
# they will be excluded from compile and install of all
EXTRA_EXCLUDED_MODULES=bdb dbtext oracle pa rls
Am 14.12.2010 21:11, schrieb Daniel-Constantin Mierla:
On 12/14/10 8:17 PM, Klaus Darilion wrote:
rls is extra_excluded in debian rules file:
# extra modules to skip, because they are not compilable now
# - regardless if they go to the main kamailio package or to some
module package
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