and Kamailio (OpenSER) - sr-users mailing
>> listsr-us...@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>>
>> --
>> Daniel-Constantin Mierla -
>> http://www.asipto.comhttp://twitter.com/#!/miconda -
>> http://www.linkedin
gt; Oracle is a registered trademark of Oracle Corporation and/or its
> affiliates. Other names may be trademarks of their respective
> owners.
>
> Type 'help;' or '\h' for help. Type '\c' to clear the current input
> statement.
>
> mysql>
>
> With
.0.0.1:5060"
> Feb 12 23:41:30 localhost /usr/local/sbin/kamailio[4283]: :
> [main.c:1625]: main_loop: Cannot fork
>
> With Regards
>
> N.Prakash
>
> -- Forwarded message --
>> From: SamyGo
>> Date: Tue, Feb 12, 2013 at 5:49 PM
>> Subject: Re
-
>> *
>> *# Dispatch requests*
>> *route[DISPATCH] {*
>> *if ( method=="INVITE" ) {*
>> *# dst_select( "GROUP", "HASH METHOD")*
>> * ds_select_dst("1","4");*
>> * sl_send_reply("100","Trying");*
Regards
>
> N.Prakash
>
>
>
> On Sat, Mar 2, 2013 at 6:30 PM, Muhammad Shahzad wrote:
>
>> Why are you forwarding instead of relaying the message to selected
>> destination? Forward is stateless and therefore likely to have NAT issues,
>> specially if destination
atcher
>> for Queue and IVR (One asterisk first and next Asterisk for second calls )
>>
>> But if try to calls extension it is landing both Asterisk server instead
>> landing one asterisk first and next Asterisk for second calls
>>
>> Please advice
>>
>&
gt; http://lists.sip-router.org/**cgi-bin/mailman/listinfo/sr-**dev<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev>
>
--
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network As
Kamailio trainings at http://www.asipto.com -
>
>
> __**_
> sr-dev mailing list
> sr-...@lists.sip-router.org
> http://lists.sip-router.org/**cgi-bin/mailman/listinfo/sr-**dev<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-d
Kamailio trainings at http://www.asipto.com -
>
>
> __**_
> sr-dev mailing list
> sr-...@lists.sip-router.org
> http://lists.sip-router.org/**cgi-bin/mailman/listinfo/sr-**dev<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-d
R) - sr-users mailing
> >>>>>> list
> >>>>>> sr-users@lists.sip-router.org
> >>>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
> >>>>>
> >>>>>
> >>>>> ___
tober/070458.html
>
> [2] http://lists.sip-router.org/pipermail/users/2008-November/020646.html
>
> ___
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users@lists.sip-router.org
> http://lists.sip-ro
y both
calls connects but then drop after a few seconds of ACK. Which indicate the
problem is likely to be on mediaproxy-ng end rather then kamailio..
Thank you.
--
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISC
nted on this topic:
> - http://www.slideshare.net/crocodilertc/webrtc-websockets
>
> Last part has also snippets of Kamailio config.
>
> Cheers,
> Daniel
>
>
> On 09/12/13 14:04, Muhammad Shahzad wrote:
>
> Hi,
>
> According to documentation, using kamailio's r
statics the is no packets travelling
> between them both.
>
> Thanks you all.
>
> ___
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users@lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/
detail ?
> ___
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users@lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
--
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO
t; Engineering for the Masses
>
> ___
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users@lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
--
Mit freundlichen Grüßen
Mu
(OpenSER) - sr-users mailing list
> sr-users@lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
--
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Netw
This is kamailio forum and as you mentioned all is working on kamailio
side. Any standard SIP client should work with kamailio without any
problem, regardless if its a mobile app or desktop softphone or even an ATA.
Regarding SIP client SDK for mobile apps, there are several available, both
commer
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
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
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
>>
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
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
> >
> > 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
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
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
;
> 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
_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
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
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
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 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
Nope, just WS handshake message,
INFO:
>
>
> On 06/06/14 11:55, Muhammad Shahzad wrote:
>
> Nope, just WS handshake message,
>
> INFO:
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 <
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
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
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
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
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
, 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
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
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
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
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
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
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,
>
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
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
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%
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
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
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
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
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.
>
> 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
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..
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
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
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
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,
>
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:
>
>
,
> 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
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
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.
&
;
> 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
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
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
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
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
&
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
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
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
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
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
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
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
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
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
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
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
:
> 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
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.
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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 :).
>
>
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
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
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