Hi,
Lasting night i was doing some tests and identified a strange behavior from
dialog module. I am using Kamailio v4.3.0 rev. c6aa95.
I basically wanted to identify if an incoming sequential ACK (from caller
for an answered call) or a BYE (from callee who wants to hangup the call)
belong to an e
i somewhat agree with Daniel. The wstunnel should forward Connection and
Upgrade headers, instead of absorbing them. Otherwise kamailio or whatever
back-end server you may use will never know it is WS connection, thus would
treat the connection in simple HTTP context.
In my opinion, Nginx has a be
Not just in IMS but in general media proxies usually expect traffic from
both ends before they start relaying it from one endpoint to another. This
is basically how media proxies get aware of as from where the RTP would
come from and where it is suppose to go.
During the call setup, the media prox
e last one matching in
> tls.cfg is selected first time. If no server name is matching after SNI
> callback, the the 'default' server context is selected.
>
> I did just basic testing so far with SIP registration, therefore proper
> testing would be required on your side and feed
Call-ID +
> From-tag as key.
>
> --
> Sent from my BlackBerry. Please excuse errors and brevity.
> *From: *Muhammad Shahzad
> *Sent: *Monday, February 16, 2015 12:55 PM
> *To: *Kamailio (SER) - Users Mailing List
> *Reply To: *Kamailio (SER) - Users Mailing List
>
t; I misunderstood the notion of "transaction". I was thinking that it was
> the whole call-flow.
>
>
>
> Regards,
>
>
>
> Igor.
>
>
>
> *De :* sr-users [mailto:sr-users-boun...@lists.sip-router.org
> ] *De la part de* Muhammad Shahzad
> *Envoyé :* lundi 1
BTW, if nothing works, you can always use "network:msg" event route to find
/ replace any part of the SIP request and response, including media IP in
SDP. ;-)
http://kamailio.org/docs/modules/4.2.x/modules/corex.html#async.evr.network_io
Thank you.
On Mon, Feb 16, 2015 at 6:39 PM
I haven't done something like that myself but i think if you use RTPEngine
with "media-address" set correctly in offer and answer functions, you can
easily achieve this. Simply check if request/reply is coming from FS or the
end-user and adjust the media appropriately without even invoking
auto-bri
As far as i know AVPs are transaction specific only. So they will be
deleted as soon as transaction is over, i.e. 200 OK for INVITE is received
for example. They will not be available in in-dialog transactions such as
ACK, or BYE etc. What you need is to set dialog variable instead, see more
info h
Hi,
I want to deploy a kamailio v4.2.x setup with multiple domains, all resolve
to same IPv4 address kamailio is listening on. I am bit confused about how
to configure TLS certificates using tls config file as mentioned here,
http://kamailio.org/docs/modules/4.2.x/modules/tls.html#tls.p.config
T
Seems to me that your ATA has wrong DNS configuration, since with IP it
works fine, with domain name it doesn't.
If you are using public domain name (resolvable by public dns), then
perhaps you may use google DNS server in your ATA box, i.e. 8.8.8.8 and
8.8.4.4
If you are using private domain nam
I think i have similar problem last week with rtpengine deployment which
was about 1-2 weeks old. There was no audio although the logs say that STUN
bindings are successful from both side (SAVPF <-> AVP). One symptom of the
problem is this log message,
--
rtpengine[16455]: [kr8shv3uca0fnmg4ktd4 po
Thanks for explanation, this definitely makes more sense.
Thank you.
On Thu, Feb 5, 2015 at 9:29 AM, Olle E. Johansson wrote:
>
> On 05 Feb 2015, at 08:27, Muhammad Shahzad wrote:
>
> I post it on both since i was a bit confused about which list is
> appropriate fo
I post it on both since i was a bit confused about which list is
appropriate for this question.
Anyways, i will toss a coin next time. ;-)
Thank you.
On Thu, Feb 5, 2015 at 1:16 AM, Juha Heinanen wrote:
> Muhammad Shahzad writes:
>
> > I have latest stable release of RTPEngine d
Hi,
I have latest stable release of RTPEngine deployed in a virtual machine
(KVM) along with Kamailio v4.2. All is working fine except i see this
message in RTPEngine logs,
--
rtpengine[16455]: [82qrjq0hdtt45afbqo98 port 40960] Kernelizing media
stream
rtpengine[16455]: [82qrjq0hdtt45afbqo98 port
Thank you so much guys for this insight into the issue. Let me talk to the
SIP client developers and see what is their point-of-view on this.
Thank you
On Sun, Feb 1, 2015 at 10:36 PM, Richard Fuchs wrote:
> On 02/01/15 09:17, Muhammad Shahzad wrote:
> > Thanks for detailed reply an
andled.
>
> If you do a packet capture can you still see Browser sending Video to SIP
> Client after those initial 5-7 seconds. (Check Webrtc logs/packet capture)
>
> Some details about WebRTC handling packet loss.
>
> https://groups.google.com/forum/#!topic/discuss-webrtc/0Zbx
Hi,
This may be a bit out of focus topic for this forum but i am posting it
here anyway with hope that some guru would shed some light on it and point
me to right direction.
The problem is that i want to establish video call between a webrtc and a
sip client using kamailio (for signalling) and RT
our help and support.
On Tue, Jan 6, 2015 at 7:00 AM, Muhammad Shahzad
wrote:
> OK, finally back at office after holidays.
>
> I have done extensive testing of various kamailio revisions (backwards up
> to November) and it seems that problem is not related to any change in
> na
from mid-November, which
inserts all ACC event records) with current cfg file (which only inserts
BYE event records) and see if i can find that configuration changes that
are causing this behavior.
Thank you.
On Tue, Dec 30, 2014 at 2:56 AM, Muhammad Shahzad
wrote:
> OK, i will run some te
Happy new year, hope we will have auto-fit shoes, auto-clean clothes and
hover-boards as predicted in "Back To The Future", all before November 2015
... ;-)
On Thu, Jan 1, 2015 at 4:16 AM, Brandon Armstead wrote:
> Happy New Year!!!
>
> Many great new things to come. #2015 here we are :).
>
>
Many GMSCs nowadays support SIP interconnect (with or without an IMS
setups). So, it entirely depends on the
capability of your GMSC to allow interconnect of your SIP network to your
mobile network.
If you are looking to connect your SIP network to *any* mobile network,
then answer is NO, unless y
corner case
> situation...
>
> Cheers,
> Daniel
>
>
> On 24/12/14 15:23, Muhammad Shahzad wrote:
>
> After upgrade to version 4.2.1-a2aa22, result is same.
>
> Thank you.
>
>
>
> On Wed, Dec 24, 2014 at 1:32 PM, Muhammad Shahzad
> wrote:
>
>&g
Each authentication method in kamailio always gives some return values
which are very useful to help understand and debug authentication failures.
For example read return values of www_authenticate method here,
http://kamailio.org/docs/modules/4.2.x/modules/auth_db.html#auth_db.f.www_authenticate
I am not sure if i understand your question correctly, but if you want to
use any authentication source or encryption algorithm (for back-end
storage, e.g. for compliance with PCI DSS v2.0 and above) other then
standard db and ha1 hash then you may consider using pv_auth_check,
http://kamailio.org
Thanks to entire Kamailio community, especially to Daniel for excellent and
to-the-point help and support.
Happy holidays of birth of Jesus and Muhammad to everyone.
Thank you.
On Thu, Dec 25, 2014 at 1:58 AM, Will Ferrer
wrote:
> Hi Daniel
>
> Thanks so much and Happy Holidays to you and yo
After upgrade to version 4.2.1-a2aa22, result is same.
Thank you.
On Wed, Dec 24, 2014 at 1:32 PM, Muhammad Shahzad
wrote:
> Looking at log level 3 logs, i see when INVITE has been authenticated ACC
> module creates the dialog,
>
> --
> DEBUG: acc [acc_cdr.c:726]: cdr_on_c
unc(): acc callback called for
t(0xa591d840) event type 2, reply
code 200
--
Between these two log lines there is no log from acc module.
Thank you.
On Wed, Dec 24, 2014 at 11:04 AM, Muhammad Shahzad
wrote:
> See attached SIP trace.
>
> Note, i have obfuscated source and destination
See attached SIP trace.
Note, i have obfuscated source and destination number and IPs etc. due to
privacy reasons.
Thank you.
On Wed, Dec 24, 2014 at 10:36 AM, Muhammad Shahzad
wrote:
> OK, i will upgrade my staging server and do some testing.
>
> The acc module does not pos
n a file for later analysis. When you find a missing record,
> search in the file with the sip traffic and see if something is broken
> there.
>
> Cheers,
> Daniel
>
>
> On 23/12/14 21:45, Muhammad Shahzad wrote:
>
> Hi,
>
> About 3 weeks ago i upgraded one of m
Hi,
About 3 weeks ago i upgraded one of my production server with latest stable
kamailio version 4.2.1-fad00a. Now i am getting a lot of complaints about
missing CDR events in ACC table. I observe following problems,
1. There are only BYE records in acc table, no record for INVITE or ACK.
2. In k
The users in subscriber table are the actual users who are allowed to
register to your SIP service. This is where kamailio gets the
authentication information, e.g. username and password etc.
The location table is where kamailio stores currently registered i.e.
online users. Obviously the records
Don't do loose route in main route block. The WITHINDLG route will take
care of that.
Also you may need to do "handle_ruri_alias" just after loose route in
WITHINDLG route. See below link for more details,
http://www.kamailio.org/docs/modules/4.2.x/modules/nathelper.html#nathelper.f.handle_ruri_a
nents licensed under the GNU General
> Public
> License version 2 and other licenses; you are welcome to redistribute it
> under
> certain conditions. Type 'core show license' for details.
> =
> Connect
and it
> is how to tell Kamailio about the SIP users in the Asterisk DB ?!
>
> Best Regards,
>
>
> On Sun, Nov 16, 2014 at 3:01 PM, Muhammad Shahzad
> wrote:
>
>> This seems to be fine. The user MUST authenticate to Kamailio, only then
>> Kamailio will create
This seems to be fine. The user MUST authenticate to Kamailio, only then
Kamailio will create REGISTER request that is send to asterisk. That's the
key security feature behind the idea.
Look at the register architecture diagram,
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.
:
> Thanks Muhammad Shahzad,
>
> There is a step for P-CSCF install in above ng-voice site
>
> cd /etc/kamailio
> mv kamailio.cfg kamailio.cfg.dist
> ln -s pcscf.cfg kamailio.cfg
>
>
> First here, we are moving kamailio.cfg to kamailio.cfg.dist
>
> Then we are execut
Hi,
I am trying add WebRTC support to existing IMS using kamailio. The idea is
to let kamailio handle all webrtc calls alone with diameter backend to IMS
for AAA.
So, i thought of using auth_diameter and aac module for authentication and
accounting. However auth_diameter docs say that the module
This is bit old, but should still work,
http://nil.uniza.sk/ngnims/kamailio-ims/installing-base-kamailio-ims-platform-debian-squeeze-32bit
Thank you.
On Mon, Oct 27, 2014 at 11:08 AM, Kamal Palei wrote:
> Hi All
> Last few days we have tried to setup Kamailio for IMS server setup.
> We have
t; Adding transformations to "quote" and "unquote" can be useful indeed if
> json operation returns the full value.
>
> Cheers,
> Daniel
>
>
> On 23/10/14 21:02, Muhammad Shahzad wrote:
>
>Hi,
>
> Using method json_get_field i am able to extr
Hi,
Using method json_get_field i am able to extract json string. However, this
string comes with quotes which cause a hurdle in assigning to various
pseudo variables, for example,
--
json_get_field($redis(r=>value), "to_number", "$var(wim_to)");
json_get_field($redis(r=>val
The AVP stands for Attribute Value Pair. If you are familiar with any
programming language then it is like a variable, which can be assigned any
String or Integer value, you can later use that variable in your script
(kamailio.cfg) or pass it to any module function that accepts AVP in its
input arg
I did installed kamailio on ARMv6 platform (Raspberry Pi). It works really
good for me (5-10 encrypted calls with peer-to-peer media etc.).
However rtpengine kernel driver does not load due to module checksum error
(the stock kernel that comes with Raspbain does not allow kernel modules
compiled o
OK, i have done some testing in various possible scenarios and it seems to
work fine.
Can you merge this to 4.2 branch so i can upgrade my production servers
with stable release?
Thank you.
On Fri, Oct 17, 2014 at 12:00 PM, Muhammad Shahzad
wrote:
> Great. I am upgrading a couple of
Good stuff. I am upgrading a couple of kamailio dev servers now to test
this. I will get back to you if i find any problem later today.
I think max 31 branches are fine, at least for my needs.
Max branches per transaction would be really great. Per my own
requirements, I need higher number of max
s
>
> Should be easy to cherry-pick the two commits in branch 4.2, given there
> were no other changes meanwhile.
>
> Feedback would be very useful to see if it works properly.
>
> I will send a different email to present more about this.
>
> Cheers,
> Daniel
&g
Yes, making it configurable will be really cool.
Thank you so much for your help.
On Thu, Oct 16, 2014 at 10:42 AM, Olle E. Johansson wrote:
>
> On 16 Oct 2014, at 10:27, Daniel-Constantin Mierla
> wrote:
>
> > Hello,
> >
> > On 16/10/14 10:04, Muhammad Shahzad w
Hi,
After some testing it appears that kamailio can do maximum 12 serial forks
using transaction manager (tm module). Is it possible to make it
configurable in kamailio.cfg or at least increase it with static higher
value (e.g. 15)?
Thank you.
___
SIP E
xy with
> several user agents behind. To identify peers you should use the data
> from the transport: IP, port, protocol. That should be unique for a
> peer. For received messages it should be simple to extract them, for
> sending, the data should be available too (e.g. in DURI or some
&
receive the
> data. However, for WS transport this topmost VIA is useless static constant
> string. So VIA checking is pointless (all remote endpoints will or may have
> same top most VIA).
>
> So i was thinking if there is another way to do it? I thought of using
> GRUU, but it is no
IA).
So i was thinking if there is another way to do it? I thought of using
GRUU, but it is not always present, especially in SIP replies.
Thank you.
On Mon, Aug 25, 2014 at 3:24 PM, Vitaliy Aleksandrov wrote:
> On 22.08.14 03:26, Muhammad Shahzad wrote:
>
>> Sorry for putting thi
Sorry for putting this question on both dev and user mailing lists, as it
is a rather theoretical question and i hope some SIP guru on either mail
list will answer.
For non-WS endpoints which use TCP or UDP for SIP transport, each upstream
request has top most VIA header pointing to the previous h
;
> On 14/08/14 21:45, Muhammad Shahzad wrote:
>
> Fixed as discussed,
>
> webrtc.voip-demos.com/0001-added-support-for-network-io-intercept.patch
>
> Thank you.
>
>
>
>
> On Fri, Aug 15, 2014 at 12:15 AM, Muhammad Shahzad
> wrote:
>
>> oops, spoke too so
Fixed as discussed,
webrtc.voip-demos.com/0001-added-support-for-network-io-intercept.patch
Thank you.
On Fri, Aug 15, 2014 at 12:15 AM, Muhammad Shahzad
wrote:
> oops, spoke too soon. These are not declared static.
>
> Anyways, let me apply your suggestion.
>
> Thank you.
&
oops, spoke too soon. These are not declared static.
Anyways, let me apply your suggestion.
Thank you.
On Fri, Aug 15, 2014 at 12:13 AM, Muhammad Shahzad
wrote:
> Sorry for late relay, we have had internet blockage due to mass protests
> here.
>
> Anyways, i think the v
,
> Daniel
>
>
> On 13/08/14 15:05, Muhammad Shahzad wrote:
>
> OK, no problem.
>
> Thank you.
>
>
>
>
> On Wed, Aug 13, 2014 at 2:52 PM, Daniel-Constantin Mierla <
> mico...@gmail.com> wrote:
>
>> I will get back to it and push it if all ok -- g
OK, no problem.
Thank you.
On Wed, Aug 13, 2014 at 2:52 PM, Daniel-Constantin Mierla wrote:
> I will get back to it and push it if all ok -- got caught by some other
> stuff meanwhile.
>
> Cheers,
> Daniel
>
>
> On 08/08/14 15:08, Muhammad Shahzad wrote:
>
>
g->len;
> return pv_get_strval(msg, param, res, &s);
> }
>
> They are the same apart of variables, so no matter where they will be used
> (before or after event route processing), they point to the same buffer,
> therefore they will return the same.
>
> Cheers,
>
patched updated, as discussed.
http://webrtc.voip-demos.com/0001-added-support-for-network-io-intercept.patch
Thank you.
On Fri, Aug 8, 2014 at 4:37 PM, Muhammad Shahzad
wrote:
> humm, original function must have got lost while moving the code to corex.
> Anyways, lets just remove t
patched updated, as discussed.
http://webrtc.voip-demos.com/0001-added-support-for-network-io-intercept.patch
Thank you.
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.or
Aug 8, 2014 at 11:07 AM, Daniel-Constantin Mierla wrote:
> I see that the code introduces a new variable $raw_msg. It looks like
> being message buffer which is returned by $mb, thus redundant.
>
>
> Cheers,
> Daniel
>
> On 05/08/14 12:06, Muhammad Shahzad wrote:
>
&g
yup, i always download kamailio from official repo mentioned here,
http://www.kamailio.org/wiki/install/devel/git
Thank you.
On Tue, Aug 5, 2014 at 3:03 PM, Daniel-Constantin Mierla
wrote:
>
> On 05/08/14 11:55, Muhammad Shahzad wrote:
>
> OK, created the patch finally..
>
> git format-patch -1 HEAD
>
> Cheers,
> Daniel
>
>
> On 05/08/14 11:25, Muhammad Shahzad wrote:
>
> These commands do not seem to work for me. Can you please do the patch?
>
> 1. adding files work,
>
> git add modules/corex/corex_nio.c modules/c
changes. Practically use:
>
> - git add -- to add new files
> - git commit -- to commit changes
> - git format-patch -- to get the commit in a file
>
> Cheers,
> Daniel
>
>
> On 05/08/14 01:14, Muhammad Shahzad wrote:
>
> Done all changes as you suggested.
review and download at,
http://webrtc.voip-demos.com/corex.tbz2
Regarding the actual encryption / compression etc., i am planning to add
some example PERL / LUA scripts later on.
Thank you.
On Mon, Aug 4, 2014 at 8:19 PM, Muhammad Shahzad
wrote:
> Thank you for your valuable suggestions
r from what you listed (http ecapsulation is at least
> interesting, considering many allow port 80 and inspect for http).
>
> Of course, these are my opinions, so the discussion can go on for deciding
> on how to proceed.
>
> Cheers,
> Daniel
>
>
>
>
> On
See reply inline below,
Thank you.
On Mon, Aug 4, 2014 at 12:08 PM, aawaise wrote:
> I want to ask some queries regarding logging in kamailio.
>
> 1. First of all if I want to add LM_DBG or LM_ERR in lookup.c file of
> registrar module, where to add the command as on extraction of
> kamailio-3
Hi,
As already discussed in detail in following email thread,
https://www.mail-archive.com/sr-users@lists.sip-router.org/msg19922.html
The new Kamailio module obfuscate is ready for testing and can be
downloaded at,
http://webrtc.voip-demos.com/obfuscate.tbz2
It contains full code, with docume
Thanks for good insight in to this topic.
As mentioned in my first email, there are a number of reasons for trying
out custom encryption schemes. Low-end android devices is just one of them.
There is a huge market for low-end android devices in south and south east
Asia for example, where over 35%
uly 2014 06:37:31 Muhammad Shahzad wrote:
> > Humm, no reply so far, may be because my email was very long and no body
> > bothered to read it all. Anyways, here is the shorter more direct version
> > of it.
>
> I read it all and my only though was: use a VPN.
>
> If
hopping to release a free and open source implementation
using idoubs within next couple of months on Apple app store.
Thank you.
On Wed, Jul 30, 2014 at 12:22 PM, Daniel-Constantin Mierla <
mico...@gmail.com> wrote:
>
> On 30/07/14 06:37, Muhammad Shahzad wrote:
>
> Humm, no re
x27;s core, encrypts it and then send it out to actual
destination.
In case above is not possible. Can i do it in kamailio's native code? Any
hooks / example code for reference?
Many thanks and kind regards for your help.
On Mon, Jul 28, 2014 at 2:38 AM, Muhammad Shahzad
wrote:
> Hi,
>
Hi,
As the mobile voip is getting more and more popular these days, there has
been a strong opposition from GSM operators against mobile voip apps. They
often use tactics like blocking voip ports, or detect and block voip
traffic and in some cases restricting udp traffic altogether to very low
upl
Well, this
*if (from_uri!=myself && uri!=myself)*
Means neither source nor destination is our user. Which implies that if our
domain is A, then call from domain "B to C" is not possible. However, calls
from "B or C to A" and "A to B or C" are possible. That is way an
unauthorized user gets passed
OK, this seems to have solved the problem.
Many thanks and kind regards.
On Fri, Jul 11, 2014 at 1:27 AM, Richard Fuchs wrote:
> On 07/10/14 20:25, Muhammad Shahzad wrote:
> > Hi,
> >
> > I am trying to upgrade from mediaproxy-ng to rtpengine trunk version.
> > Th
Hi,
I am trying to upgrade from mediaproxy-ng to rtpengine trunk version. The
compilation steps go well and i have deb packages created. However when i
try to install them (on same machine where they compiled), i get this error
for every deb package,
ngcp-rtpengine-daemon pre-depends on ngcp-sys
at 4:44 AM, Muhammad Shahzad
wrote:
> After upgrade to latest revision of 4.1 branch, now i get this error log,
>
> --
> ERROR: auth [auth_mod.c:690]: pv_www_authenticate2(): failed to get method
> value from msg 0xa5813680 var 0xb67c13a0
> --
>
> Complete debug level
, Daniel-Constantin Mierla wrote:
> I added an enhancement to print the pointers involved in retrieving the
> method. Can you test with latest master or 4.1 branches from git?
>
> Cheers,
> Daniel
>
>
> On 11/06/14 18:35, Muhammad Shahzad wrote:
>
> Sent logs to pri
t;
>
> On 11/06/14 18:35, Muhammad Shahzad wrote:
>
> Sent logs to private email of yours. Now there don't seem to be any
> parsing error however, method pv_www_authenticate2 still fails with same
> error,
>
> ERROR: auth [auth_mod.c:690]: pv_www_authenticate2
antin Mierla wrote:
> Are those all the log messages? Previously there were parsing errors in
> the logs you sent to me.
>
> Get them with debug=3 in kamailio.cfg.
>
> Cheers, Daniel
>
>
> On 11/06/14 17:51, Muhammad Shahzad wrote:
>
> Many thanks for your time
s, Daniel
>
>
> On 11/06/14 17:51, Muhammad Shahzad wrote:
>
> Many thanks for your time and help.
>
> I just tried with msrp:// scheme, still get same result,
>
> --
> MSRP nv755d8c AUTH
> To-Path: msrp://ms11.xyz.com
> From-Path: msrp://xe4a9fqm.inv
Many thanks for your time and help.
I just tried with msrp:// scheme, still get same result,
--
MSRP nv755d8c AUTH
To-Path: msrp://ms11.xyz.com
From-Path: msrp://xe4a9fqm.invalid:2855/bcuf2gk7co;ws
---nv755d8c$
MSRP nv755d8c 401 Unauthorized
To-Path: msrp://xe4a9fqm.invalid:2855/bcuf2gk
Any update? Do you need any additional info?
Thank you.
On Fri, Jun 6, 2014 at 11:29 PM, Muhammad Shahzad
wrote:
> I have sent you logs to your private email separately, did you get them?
>
> Thank you.
>
>
>
>
> On Fri, Jun 6, 2014 at 3:48 PM, Muhammad Shahzad
I have sent you logs to your private email separately, did you get them?
Thank you.
On Fri, Jun 6, 2014 at 3:48 PM, Muhammad Shahzad
wrote:
> OK sure. I will provide it tonight.
>
> Thank you.
>
>
>
>
> On Fri, Jun 6, 2014 at 2:48 PM, Daniel-Constantin Mierla <
>
>
> On 06/06/14 11:55, Muhammad Shahzad wrote:
>
> Nope, just WS handshake message,
>
> INFO:
Nope, just WS handshake message,
INFO:
Hi,
I am trying to authentication MSRP connection using the example code of
msrp event route in module documentation here,
http://kamailio.org/docs/modules/4.1.x/modules/msrp.html#idp119248
--
...
} else if ($msrp(method)=="AUTH") {
...
if (!pv_www_authenticate("WEBRTC_SIP_REALM", "$va
Yes, but problem is that main server has pretty good billing integrated
with it, which is the key reason my client want to keep it.
Thank you.
On Mon, Apr 28, 2014 at 11:57 AM, Juha Heinanen wrote:
> Muhammad Shahzad writes:
>
> > The main server manages SIP register, calls, mes
Hi,
I have a complex setup consisting of two sip server, lets call them main
server and presence server.
The main server manages SIP register, calls, messages and so, however it
does not support presence at all. It returns SIP response "405 Method Not
Allowed" for any SIP PUBLISH, SUBSCRIBE or NO
You are right patch does not work for Async commands. I will try to fix it
per your guidelines. Once again many thanks for your insight into the
matter.
Thank you.
On Sat, Mar 29, 2014 at 3:53 AM, Muhammad Shahzad wrote:
> humm,
>
> When datagram server is initialized
_sock=0
> - try to bind to same socket in async handler, provided the initial socket
> is created with resuse addr/port
>
> Cheers,
> Daniel
>
>
> On 28/03/14 21:49, Muhammad Shahzad wrote:
>
> Please find attached updated patch as requested.
>
>
> On Fri, Mar 28
;
> On 28/03/14 01:29, Muhammad Shahzad wrote:
>
>> Hi,
>>
>> After wasting most of the day trying to make mi_datagram over UDP socket
>> work. I eventually realize that it does asymmetric UDP communication, which
>> creates a lot of trouble for writing a useful
Hi,
After wasting most of the day trying to make mi_datagram over UDP socket
work. I eventually realize that it does asymmetric UDP communication, which
creates a lot of trouble for writing a useful MI script using PERL or
Python etc.
Anyhow, i go through the module code and was able to write a p
What..!
Who gave you the idea that increasing memory buffers has ANYTHING to do
with jitter and latency? These are network problems and have nothing to do
with shared or package memory. Especially in case of kamailio + mediaproxy
which merely relay media from one end to another (no transcoding,
re
> >
> > On Wed, Feb 5, 2014 at 5:41 PM, Richard Fuchs > <mailto:rfu...@sipwise.com>> wrote:
> >
> > Hey,
> >
> > If you're trying to connect two WebRTC endpoints with each, you don't
> > need any of mediaproxy-ng's m
I think its dependent on db_update_period parameter. So any changes
happening to dialogs are likely not committed to db at max this time value.
kamailio.org/docs/modules/4.1.x/modules/dialog.html#idp125200
Thank you.
On Thu, Feb 6, 2014 at 5:35 AM, jay binks wrote:
> So I have tracked this
There are several problems that need to be addressed in your kamailio.cfg
but let me try to focus only on mediaprxoy-ng related ones.
First instead of engaging mediaproxy in failure route, engage it main route
or branch route. Why wait for failure when we know call will fail anyway if
you try to c
wrote:
> On 02/02/2014 08:29 PM, Muhammad Shahzad wrote:
>
> Yes, I can either move to static IP for router OR bind kamailio with
>> ddclient, so that whenever ddclient reports an IP change i restart
>> kamailio with new advertised_address. But these are out of the box
>>
Yes, I can either move to static IP for router OR bind kamailio with
ddclient, so that whenever ddclient reports an IP change i restart kamailio
with new advertised_address. But these are out of the box solutions. I
wonder if we have any "in-the-box" solution. :-)
Thank you.
On Mon, Feb 3, 201
Hi,
I am setting up Raspberry Pi to run Kamailio with mediaproxy-ng. This
machine is running on local LAN behind a DSL router. Using dynamic DNS and
DMZ services of router, i can access this box from the Internet.
However, i do not know how to define advertised_address parameter to public
IP of r
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