Hi All,
Is there a plan to write new book about OpenSIPS. For example something about
the internal design, some advanced use cases, examples on the new features, and
so on.
Regards.
___
Users mailing list
Users@lists.opensips.org
http://lists.open
Yes I think so.
//Binan
Från: Bogdan-Andrei Iancu
Till: Binan AL Halabi
Kopia: OpenSIPS users mailling list
Skickat: tisdag, 4 december 2012 11:21
Ämne: Re: [OpenSIPS-Devel] [HELP] Advising on GPL license issue
Hi Binan,
I found that page in the
Hi Bogdan,
So OpenSIPS module is not derivative.
I have found this page :
http://people.gnome.org/~markmc/openssl-and-the-gpl.html
and at the end : "all the copyright holders for the affected code agree"
and
Från: Bogdan-Andrei Iancu
Till: Binan
ån: Bogdan-Andrei Iancu
Till: Binan AL Halabi
Kopia: OpenSIPS users mailling list
Skickat: måndag, 3 december 2012 13:05
Ämne: Re: [OpenSIPS-Devel] [HELP] Advising on GPL license issue
Hi Binan,
And this exception clause (btw, found something similar here
http://lists.debian.org/de
Hi Bogdan.
Yes .
Special exception clauses allows this combination (linking). This is what i
know.
// Binan
Från: Bogdan-Andrei Iancu
Till: Binan AL Halabi
Kopia: OpenSIPS users mailling list
Skickat: fredag, 30 november 2012 17:57
Ämne: Re: [OpenSIPS
Hi Bogdan,
See this : "GPL-incompatible libraries with GPL software"
http://www.gnu.org/licenses/gpl-faq.html#GPLIncompatibleLibs
// Binan
Från: Bogdan-Andrei Iancu
Till: "users@lists.opensips.org" ;
"de...@lists.opensips.org"
Skickat: fredag, 30 november 20
Hi Miha,
Check if you have another OpenSIPS on your machine (maybe installed by your OS).
// Binan
Från: Miha
Till: Muhammad Shahzad
Kopia: OpenSIPS users mailling list
Skickat: tisdag, 20 november 2012 8:09
Ämne: Re: [OpenSIPS-Users] Log problem
Hi M
).
// Binan
Från: Ovidiu Sas
Till: Binan AL Halabi ; OpenSIPS users mailling list
Kopia: OpenSIPS mailling list
Skickat: måndag, 19 november 2012 20:50
Ämne: Re: [OpenSIPS-Users] OpenSIPS Provisioning
What are you looking for: to provision or to manage
Hi Christian,
load and configure "auth_db" and "auth" modules. Then use these function:
proxy_authorize() and www_authorize()
see these:
http://www.opensips.org/html/docs/modules/devel/auth_db.html#id250229
And chapter 5 of book "Building Telephony Systems with OpenSIPS 1.6"
// Binan.
___
Hi all,
Is there a way to provision OpenSIPS using JSON format instead of XML and REST
instead of XML-RPC ?
I think there should be new module named like MI_JSONREST or something like
that.
Thanks.
// Binan___
Users mailing list
Users@lists.opens
Hi Denis,The client must send BYE to end the dialog. but in case of timeout you can generate the local BYE to end the dialog:create_dialog("B"); The string "B" activate the BYE on timeout behavior. In OpenSIPS 1.6 use "bye_on_timeout_flag". // BinanFrån: dpa Till: 'OpenSIPS users mailling
Hi,
It could be the problem of the DHCP client on your devices (Lease to Expire,
Keeps Using IP Address)
IOS 4.1 -6.0.1 and Android 2.1 - 4.0.4 have this bug.
http://www.net.princeton.edu/android/android-stops-renewing-lease-keeps-using-IP-address-11236.html
http://www.net.princeton.edu/apple-i
Från: Dragomir Haralambiev
Till: Binan AL Halabi ; OpenSIPS users mailling list
Skickat: söndag, 11 november 2012 10:31
Ämne: Re: [OpenSIPS-Users] CRITICAL:dialog:log_next_state_dlg:
Hi ,
Thanks for your replay.
Yes, you right.
Why Opensips not close dialog
Hi,
It means OpenSIPS received BYE (Event 7) in early state(state 2 :dialog in
progress).
According to section 15 of RFC 3261 the caller is allowed to send BYE in early
state but calle must not do that.
I think OpenSIPS does not log this events as bogus for caller. check the
calle's side.
Hi Jorge,
In your config use dp_translate() function which searches the dialplan table to
find a pattern and
apply the translations needed. Also You can assign attribute to avp depending
on the match.
There is a dialplan example in chapter 7 of the book (Building Telephony
Systems with OpenSIP
Hi ALL,
According to 14.1 and 14.2 of RFC 3261 after receiving 491 the UAC will retry
again after 2.1-4.0 seconds if it owns the CALL-ID or 0.0.-2.0 if not.
So
If UAS have an ongoing in-dialog ping and receive another one let it send 491
and drop the request (the end point will retry it later)
Hi Jorge,
Use character code point instead:
\x{0023} matches #
\x{002A} matches *
// Binan.
Från: Jorge Medina
Till: users@lists.opensips.org; bog...@opensips.org
Skickat: lördag, 3 november 2012 0:16
Ämne: [OpenSIPS-Users] Dialplan with # or *
Hi,
Hi All,
If oversip uses Path extension OpenSIPS must support it.
1- Sending Path header values in 200 ok REGISTER response
2- Path header files syntax must confom to Route syntax
3- When look up the opensips must copies the stored path header fileds into
Route header fileds - preloaded route.
Hi Greg,
As Wget needs to know the size of data in
advance maybe your JSP page asks corresponding to its buffer size.
Use InputStreamReader when constructing BufferReader in your JSP page.
http://pipasoft.com/free-java-classes/download/wget.java
// Binan
F
gw. You
>>>need to create rule that call should not be routed to user B during 11h to
>>>14h. So drouting lookup will fail. Then you check in alias db which will
>>>tell you user c is alias to user b and you can
Attribute and Value(opensipsctl avp add ... ): Add them as
>>>two separate AVP.
>>>
>>>Then load them in your script as AVP (avp_db_load function).
>>>
>>>
>>>
>>>// Binan
>>>
>>>
>>>
>>>
Hallo,
See implementing call forwarding in opensips1.6 book.
// Binan
Från: Muhammad Shahzad
Till: Binan AL Halabi ; OpenSIPS users mailling list
Skickat: torsdag, 18 oktober 2012 11:30
Ämne: Re: [OpenSIPS-Users] Transfering a call by opensips
Binan
Hi,
Use change_reply_status(code, reason) in sipmsgops module
http://www.opensips.org/html/docs/modules/devel/sipmsgops.html#change_reply_status
// Binan
Från: spady
Till: users@lists.opensips.org
Skickat: torsdag, 18 oktober 2012 10:40
Ämne: Re: [OpenSIPS-Us
Hi,
Add them as Attribute and Value(opensipsctl avp add ... ): Add them as two
separate AVP.
Then load them in your script as AVP (avp_db_load function).
// Binan
Från: Engineer voip
Till: Binan AL Halabi ; OpenSIPS users mailling list
Skickat
Hi,
Store in usr_preferences table the time_from and time_to
then read them in your script using avpops module
Take the hour of the call and compare it to the range you got it from database.
http://www.opensips.org/html/docs/modules/1.7.x/avpops.html
// Binan
__
engage_media_proxy(), cannot
end_media_session() problem
It won't do anything if you used engage_media_proxy(), see
http://www.opensips.org/html/docs/modules/1.6.x/mediaproxy.html#id250230
2012/10/11 Binan AL Halabi :
> Hi,
>
> # BYE processing
> if (method==BYE) {
>
ing this problem? Do you have any other ideas?
Again, thanks a lot for your help.
Regards
Diego
On Tue, Oct 9, 2012 at 7:20 PM, Binan AL Halabi wrote:
>
>Do the following to increase TCP performance:
>
>
>1- Increase the number of available local ports:
>
>
Hi,
# BYE processing
if (method==BYE) {
end_media_session();
}
// Binan
Från: ㄨ冷se灬 <291490...@qq.com>
Till: users
Skickat: torsdag, 11 oktober 2012 11:14
Ämne: [OpenSIPS-Users] opensips use engage_media_proxy(), cannot
end_media_session() problem
You have changed && to ||
//Binan
Från: spady
Till: users@lists.opensips.org
Skickat: onsdag, 10 oktober 2012 16:20
Ämne: Re: [OpenSIPS-Users] Need Hint for Opensips as UDP/TCP Proxy
I talked to early !! :-(
It's happening a very strange thing.
Seems that
Hi,
Do it like this:
if (is_avp_set("$avp(trace_id)"){
# Do tracing
}
or like this :
If (!avp_check("$avp(trace_id)","re/^$/")){
# Do tracing
}
where "/^$/" regex pattern for empty string.
// Binan
Från: Dragomir Haralambiev
Till: OpenSIPS users mailling li
Hi,
Do it like this:
if (is_avp_set("$avp(trace_id)"){
# Do tracing
}
or like this :
If (!avp_check("$avp(trace_id)", "re/^$/")){
# Do tracing
}
where "/^$/" regex pattern matches an empty string
// Binan
Från: Dragomir Haralambiev
Till: OpenSIPS users mai
great.
Från: spady
Till: users@lists.opensips.org
Skickat: onsdag, 10 oktober 2012 12:09
Ämne: Re: [OpenSIPS-Users] Need Hint for Opensips as UDP/TCP Proxy
SOLVED!!
i was not reseted flag!!! :-(
This is the right code:
branch_route[2] {
if (is_method(
Hi spady,
If is_audio_on_hold() function returned true, this means the received message
has an SDP body and at least one audio stream on hold.
http://www.opensips.org/html/docs/modules/devel/sipmsgops.html#id292843
// Binan
Från: spady
Till: users@lists.op
;
>I tried upgrading to 1.8.1 but I'm having some issues I'm duscussin with
>Bogdan on the following thread:
>
>http://lists.opensips.org/pipermail/users/2012-August/022764.html
>
>Thanks
>Diego
>
>
>
>On Mon, Sep 10, 2012 at 4:47 PM, Binan AL Halabi
>
Hi Spady,
You need something like true or false.
Use a dialog flag in this way:
flag idx is set ---> The call is hold on <> a=Inactive
If (is_dlg_flag_set(idx)) && (status==200)
{
# Rewrite SDP
}
http://www.opensips.org/html/docs/modules/devel/dialog.html
Dont forget to
Hi Duane,
Try to add SDP body manually if it is not added:
- Save the original SDP ($rb) somewhere before calling b2b_init_request()
function.
- Add it to reInvite in local_route.
//Binan
Från: Duane Larson
Till: OpenSIPS users mailling list
Skickat: månd
Hi,
If you have opensips 1.8, you can group the GWs in carrier
carrier is a set of gateways.
Example : c1 = g1=50,g2=50
http://www.opensips.org/html/docs/modules/1.8.x/drouting.html#id293148
//Binan
Från: Engineer voip
Till: OpenSIPS users mailling list
Hi Jorge,
Modify the IPTABLES parameter.
Read section "Loading iptables Modules to Particular VPSs" in OpenVZ user guide:
http://download.openvz.org/doc/OpenVZ-Users-Guide.pdf
// Binan
Från: Jorge Ortea
Till: Binan AL Halabi
Skickat: fredag,
Hi,
In Centus 6 The iptables libraries have been changed, the old are no longer
supported - iptable_nat !!
use last version of Media proxy with centus 6
or
Mediaproxy 2.4.4 with centos 5
Iptables should be installed by default on all CentOS 5.x and 6.x installations.
Enable iptables : syst
_
Från: Hanie Maghsoudy
Till: Binan AL Halabi
Kopia: OpenSIPS users mailling list
Skickat: torsdag, 4 oktober 2012 7:09
Ämne: Re: [OpenSIPS-Users] Registration via RADIUS
Free Radius users:
DEFAULT Framed-Protocol == PPP
Framed-Protocol = PPP,
Framed-Compression
Binan
Från: Hanie Maghsoudy
Till: Binan AL Halabi ; OpenSIPS users mailling list
Skickat: onsdag, 3 oktober 2012 14:10
Ämne: Re: [OpenSIPS-Users] Registration via RADIUS
Thanks Binan for the reply.
I tested both and none of them works.
On Wed, Oct 3, 2012 at 2:48 PM, Binan AL Halabi wrote:
Post your Radius configuration.
//Binan
Från: Hanie Maghsoudy
Till: Binan AL Halabi ; OpenSIPS users mailling list
Skickat: onsdag, 3 oktober 2012 14:10
Ämne: Re: [OpenSIPS-Users] Registration via RADIUS
Thanks Binan for the reply.
I tested both and
look for the type of password you want to use whether plaintext or HA
modparam("auth", "calculate_ha1", 1) # plaintext password
modparam("auth", "calculate_ha1", 0) # pre-calculated HA1
//Binan
From: Hanie Maghsoudy
To: users@lists.opensips.org
Sent: Wednes
hi,
Use probing (at SIP level) so opensips does not need to send it many times to
realize that the gateway is down.
//Binan
From: Engineer voip
To: OpenSIPS users mailling list ; users-request
Sent: Wednesday, October 3, 2012 12:02 PM
Subject: [OpenSIP
Hi,
Add setflag(2) before t_relay() in failure_route[2].
//Binan
From: Alex
To: users@lists.opensips.org
Sent: Tuesday, October 2, 2012 9:35 PM
Subject: [OpenSIPS-Users] no acc after failure_route in 1.8
Hello,
I am using dialog to store cdr into mysql D
Hi,
See cachedb_redis instead.
cache_store and cache_fetch
//Binan
From: Marwan El-Sadek
To: users@lists.opensips.org
Cc: Marwan El-Sadek
Sent: Tuesday, October 2, 2012 12:59 PM
Subject: Re: [OpenSIPS-Users] Store values of the REGISTER and Retrieve the
Hi,
use B2BUA , write simple xml scinareo for that.
I can imagine client and servers B2BUA entities to receive and send SIP
messages between sipA and sipB
//Binan
From: Nick Chang
To: 'OpenSIPS users mailling list'
Sent: Tuesday, October 2, 2012 11:59
SELinux disabled !
//Binan
- Forwarded Message -
From: Binan AL Halabi
To: OpenSIPS users mailling list
Sent: Thursday, September 27, 2012 4:31 PM
Subject: Re: [OpenSIPS-Users] Rtpproxy connection
hej spady,
I see rtpproxy is root
Is opensips is root in your system ?
//Binan
hej spady,
I see rtpproxy is root
Is opensips is root in your system ?
//Binan
From: spady
To: users@lists.opensips.org
Sent: Thursday, September 27, 2012 2:26 PM
Subject: Re: [OpenSIPS-Users] Rtpproxy connection
I have
Now i will suggest stupid thing,
write simple script with only nathelper (old fasion) and without rtpproxy module
modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:7890")
instead of
modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7890")
And enable the notification socket.
see what you
You have this in your previous logs :
ERROR:rtpproxy:send_rtpp_command: proxy
and in config file
modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7890") # CUSTOMIZE ME
and your are running the proxy like
rtpproxy -F -s udp:127.0.0.1:10177 -l 10.9.23.41/151.x.x.201 -u root -d
DBUG:LOG_LOCA
hej,
You need rtpproxy-client
//Binan
From: spady
To: users@lists.opensips.org
Sent: Thursday, September 27, 2012 12:35 AM
Subject: Re: [OpenSIPS-Users] Rtpproxy connection
Hi Binan, thanks for your reply.
I read links you posted me but i cannot understan
k the
log
//Binan
________
From: Binan AL Halabi
To: OpenSIPS users mailling list
Sent: Wednesday, September 26, 2012 9:27 PM
Subject: Re: [OpenSIPS-Users] Rtpproxy connection
Hi Spady,
see these links:
RTPProxy Control Protocol:
http://www.b2bua.org/wiki/RTPproxy/Protocol#RTPproxyControlPr
Hi Spady,
see these links:
RTPProxy Control Protocol:
http://www.b2bua.org/wiki/RTPproxy/Protocol#RTPproxyControlProtocol
and RTPProxy Howitworks section:
http://linux.die.net/man/8/rtpproxy
Try to send command to rtpproxy like creating new session and see what it gets,
according to that you ca
Re: [OpenSIPS-Users] Request Timeout (for PUBLISH and SUBSCRIBE)
in IMS+opensips Presence Server
Binan AL Halabi writes:
>
> Hi,If you have this issue http://lists.opensips.org/pipermail/users/2011-
February/016935.htmland the same configuration file for IMS application serv
hej Alexandre,
somthing like:
1- Add this to failure_route (this for 408 locally generated)
if(t_check_status("408") && (t_local_replied("all"))
//cache here
2- To get something like global variable , use cache_store from localcache
module
//Binan
From
Hi,
If you have this issue
http://lists.opensips.org/pipermail/users/2011-February/016935.html
and the same configuration file for IMS application server (port 5065)
1- check if the domain is in the database or add it as an alias in the config
file
2- check if opensips execute handle_subscribe
hi Engineer VOIP,
You can use what is called "database replication"
You have 4 database servers : one must be the master and 3 are slaves.
opensips-cp update the master database, and then the master database will be
replicated to the rest.
If you are using mysql see this link:
http://dev.mysql.co
hej Sajjad,
If you mean the "username" in Authorization header is invalid in step-3 you can
do:
adjust your script to send "404 Not Found" since the proxy_authorize and www_
authorize return value is number and if this number is equal to -1 (invalid
user) - authentication user does
hej Sajjad,
If you mean the "username" in Authorization header is invalid in step-3 you can
do:
adjust
your script to send "404 Not Found" since the proxy_authorize and www_
authorize return value is number and if this number is equal to -1 (invalid
user) - authentication user does
hi Diego,
1- As you dont have state in database so you dont need to check that.
2- Why you dont upgrade to opensips 1.8.1 ? since it contains TCP fix.
Regards.
//Binan
--- On Mon, 9/10/12, Binan AL Halabi wrote:
From: Binan AL Halabi
Subject: Re: [OpenSIPS-Users] FW: Opensips 1.6.4 doesn&#
do you say with "play around TCP connection lifetime" however
I don't get the part you say "check the database". I don't have any database,
opensips is working without any DB.
Which database are you talking about?
Thanks
Diego
On Mon, Sep 10, 2012 at 1:37 PM, Bi
Hi Diego,
play around TCP connection lifetime, you could find something.
one thing more check the database during the busy hour and see if something
expired.
//Binan
--- On Mon, 9/10/12, Diego Barberio wrote:
From: Diego Barberio
Subject: Re: [OpenSIPS-Users] FW: Opensips 1.6.4 doesn't send
x27;, ';nat=yes'); }
exit;} ThanksNick From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Binan AL Halabi
Sent: Monday, September 10, 2012 5:12 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] RTPPRoxy behind
ut, My problem still it.It’s not work for this setting. Or Do you
have any suggest? ThanksNick From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Binan AL Halabi
Sent: Monday, September 10, 2012 4:57 PM
To: OpenSIPS users mailling list
Subject:
ips 1.8.1. It is old function.
You can saw this page http://www.opensips.org/Resources/DocsMigration163to164
Now, I setting “rtpproxy_answer("cowf","202.55.233.194")” ThanksNick From:
users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On
Behalf Of Bi
Hi Nick,
use force_rtp_proxy
here is the link:
http://www.opensips.org/html/docs/modules/1.4.x/nathelper.html#id271191
//Binan
--- On Mon, 9/10/12, Nick Chang wrote:
From: Nick Chang
Subject: [OpenSIPS-Users] RTPPRoxy behind NAT
To: "'OpenSIPS users mailling list'"
Date: Monday, September
?? I input "patch -p0 < rtpproxy_nat_address_fix.txt" in
rtpproxy-1.2.1 But, It's error.patch unexpectedly ends in middle of line
patch: Only garbage was found in the patch input.
ThanksNick
2012/9/7 Binan AL Halabi
Hi,
use patch command line
//Binan
--- On Fri, 9/7/1
sn't have a real world
practice.
Thanks,Ali
On Fri, Sep 7, 2012 at 5:09 AM, Binan AL Halabi wrote:
Hi,
$si - reference to IP source address of the message
if ($si==GW1-IP-ADDRESS) {
# Incoming call
# Increment $avp(IC_GW1)
} //Binan
--- On Thu, 9/6/12, SamyGo wrote:
Hi,
use patch command line
//Binan
--- On Fri, 9/7/12, Nick Chang wrote:
From: Nick Chang
Subject: [OpenSIPS-Users] How to patch for rtpproxy
To: "'OpenSIPS users mailling list'"
Date: Friday, September 7, 2012, 3:19 AM
Hello I found this patch. It fixed nat problem.
http://opensips-open-
Hi,
$si - reference to IP source address of the message
if ($si==GW1-IP-ADDRESS) {
# Incoming call
# Increment $avp(IC_GW1)
} //Binan
--- On Thu, 9/6/12, SamyGo wrote:
From: SamyGo
Subject: Re: [OpenSIPS-Users] Can load balancer show total number of call for a
gateway
To: "OpenS
Hi again,
You can also use B2BUA to play an announcement
see this tutorial http://www.opensips.org/Resources/B2buaTutorial
//Binan
--- On Wed, 9/5/12, Binan AL Halabi wrote:
From: Binan AL Halabi
Subject: Re: [OpenSIPS-Users] Announcement before cdr
To: "OpenSIPS users mailling list&q
lling list"
Date: Tuesday, September 4, 2012, 2:19 PM
And what about the “rtpproxy_stream2uas” to use opensips to play
audio?
From: Binan AL Halabi
Sent: Tuesday, September 04, 2012 3:31 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Announcement b
: "OpenSIPS users mailling list"
Date: Tuesday, September 4, 2012, 10:09 AM
Ali,
the devices would be IP Phones, because I only have Opensips in my
network.
Binan,
Do I have to use asterisk? Because I was looking for something pure
Opensips.
Thanks.
From: Binan
Hi,
You can inject announcement based on early media(183) before the call is
completed.
use media server like Asterisk. Opensips will redirect the call to asterisk
If you want this see how you can integrate Asterisk and Opensips:
http://www.opensips.org/Resources/DocsTutAsterisk
//Binan
---
Hi Bogdan,
and the sip domain but not the customer sip domain which points to ip address
of the device
right ?
thank you very much.
--- On Fri, 8/31/12, Binan AL Halabi wrote:
From: Binan AL Halabi
Subject: Re: [OpenSIPS-Users] [Re: Routing problem with Record-Route]
To: users
: Hubert Mickael
Subject: Re: [OpenSIPS-Users] [Re: Routing problem with Record-Route]
To: "OpenSIPS users mailling list"
Cc: "Binan AL Halabi"
Date: Friday, August 31, 2012, 12:18 PM
Hi,
Yes, CUSTOMER_DEVICE_SIP_DOMAIN is insert to "domain" table in
opensips&#
Hi,
Is CUSTOMER_DEVICE_SIP_DOMAIN defined in opensips ?
//Binan
--- On Fri, 8/31/12, Kevin Mathy wrote:
From: Kevin Mathy
Subject: Re: [OpenSIPS-Users] [Re: Routing problem with Record-Route]
To: "Bogdan-Andrei Iancu" , "OpenSIPS users mailling list"
Date: Friday, August 31, 2012, 5:53 AM
H
Is CUSTOMER_DEVICE_SIP_DOMAIN defined in opensips?
//Binan
--- On Fri, 8/31/12, Kevin Mathy wrote:
From: Kevin Mathy
Subject: Re: [OpenSIPS-Users] [Re: Routing problem with Record-Route]
To: "Bogdan-Andrei Iancu" , "OpenSIPS users mailling list"
Date: Friday, August 31, 2012, 5:53 AM
Hi Bog
-Route]
To: "OpenSIPS users mailling list"
Cc: "Binan AL Halabi"
Date: Tuesday, August 28, 2012, 7:29 AM
Hi,
That is not true - OpenSIPS acts all the time as a loose router.
The param you mentioned simply changes on how the "
Hi,
lets say you have 2 branches 0 towards user in location and 1 towards PSTN.
I suggest to use branch flags which are used to activate some functions in
branch level
when creating the branches in main route use this variable $branch() to set
branch flags to each indivisual branch depending
Hi Bogdan,
i see in the opensips Docs the definition of loose_route() function:
" The function performs routing of SIP requests which contain a route
set. The name is a little bit confusing, as this function also
routes
requests which are in the “strict router”
Hi ,
the statment *modparam("rr", "enable_full_lr", 1)* in the script lets
opensips uses lr=on instead of just ;lr to work as loose router, so it
behaves as strict router where it should be loose router in fail
case, since it detects only ;lr in messages.
--- On Mon, 8/27/12, Bogdan-And
drouting module exports
the "dr_reload" command.
//Binan
--- On Tue, 8/28/12, Vlad Paiu wrote:
From: Vlad Paiu
Subject: Re: [OpenSIPS-Users] restarting opensips after modification of db
To: users@lists.opensips.org
Date: Tuesday, August 28, 2012, 2:03 AM
Hello,
read this :
http://www.opensips.org/index.php?n=Resources.DocsTutLoadbalancing
//Binan
--- On Mon, 8/27/12, Engineer voip wrote:
From: Engineer voip
Subject: [OpenSIPS-Users] loadbalancer in opensips 1.8
To: "OpenSIPS users mailling list" , "users-request"
Date: Monday, August 27, 2012, 5:04
Hi,
loose route parameter lr , which can be present in sip or sips Record-Route and
Route URIs to indicate that the proxy server identified by the URI supports
loose routing.
RFC 3261 explains the "lr" parameter as just ";lr", not lr=on. This brokes some
UAs which add =on to the "lr". opens
ng address range?
On Sat, Aug 25, 2012 at 5:43 AM, Binan AL Halabi
wrote:
Indeed, the function check_address() takes the groupid as one of its arguments
In opensips script you can write a condition to use specific group in specific
case.
http://www.opensips.org/html/docs/modules/1.6.x/permis
Hi Mickael,
Do you have this in you script:
modparam("rr", "enable_full_lr", 1)
BR.
//Binan
--- On Fri, 8/24/12, mick...@winlux.fr wrote:
From: mick...@winlux.fr
Subject: Re: [OpenSIPS-Users] [Re: Routing problem with Record-Route]
To: "OpenSIPS users mailling list"
Date: Friday, August 24
3:51 AM
Hi,
Thank you for the response but it seems this solution is for to do
siptrace of a spefiy user that we mut create in usr_preferences table.
but i want to have siptrace of all the users do the calls by my opensips!
have you an idea to do that?
2012/8/25, Binan AL Halabi :
> add the foll
Indeed, the function check_address() takes the groupid as one of its arguments
In opensips script you can write a condition to use specific group in specific
case.
http://www.opensips.org/html/docs/modules/1.6.x/permissions.html#id293674
//Binan.
On Fri, Aug 24, 2012 at 10:4
add the following instructions to your script:
loadmodule "siptrace.so"
modparam("siptrace", "db_url",
"mysql://opensips:fedora13579@localhost/opensips")
modparam("siptrace", "trace_flag", 22)
modparam("siptrace","trace_on",1)
modparam("siptrace", "traced_user_avp", "$avp(traceuser)")
modparam("si
nnection user
>> >> $config->db_user = "root";
>> >>
>> >> //database connection password
>> >> $config->db_pass = "1234";
>> >>
>> >> //database name
>> >> $config->db
Hi,
why lr has different values in two cases?
--- On Fri, 8/24/12, mick...@winlux.fr wrote:
From: mick...@winlux.fr
Subject: [OpenSIPS-Users] Routing problem with Record-Route
To: users@lists.opensips.org
Date: Friday, August 24, 2012, 1:09 AM
Hi list,
I have a routing problem with my Opensip
param("dialog", "from_tag_column", "from_tag")
> modparam("dialog", "to_uri_column", "to_uri")
> modparam("dialog", "to_tag_column", "to_tag")
> modparam("dialog", "from_cseq_column&quo
--- On Thu, 8/23/12, Engineer voip wrote:
From: Engineer voip
Subject: Re: [OpenSIPS-Users] dr_groups opensips 1.8
To: "OpenSIPS users mailling list"
Date: Thursday, August 23, 2012, 3:21 PM
HI,
So we must specify the username and domain for any group???
thanks
2012/8/23, Binan
Hi Sindrek,
look to this example from opensips documentation :# replace the uri in to: with
the value of avp
sip_address subst('/^To:(.*)sip:[^@]*@[a-zA-Z0-9.]+(.*)$/t:\1$avp(sip_address)\2/ig')
regards//Binan
--- On Mon, 8/20/12, sindrek wrote:
From: sindrek
Subject: [OpenSIPS-Users] textop
Hi,
Table dr_groups is used to associate routes to specific users.
//Binan
--- On Thu, 8/23/12, Engineer voip wrote:
From: Engineer voip
Subject: [OpenSIPS-Users] dr_groups opensips 1.8
To: "OpenSIPS users mailling list"
Date: Thursday, August 23, 2012, 8:51 AM
Hello All,
Why we can't ad
st"
Date: Thursday, August 23, 2012, 6:15 AM
hello,
when i click "dialog" i get the message "
ERROR:mi_fifo:mi_fifo_server: command dlg_list is not available "
thank you for help
2012/8/23, Binan AL Halabi :
> Hi,
> what you get when you click "dialog"
Hi,
what you get when you click "dialog" under "system" list on opensips-cp ?
//Binan
--- On Thu, 8/23/12, Engineer voip wrote:
From: Engineer voip
Subject: [OpenSIPS-Users] cdr and Dialog on opensips-cp
To: "OpenSIPS users mailling list"
Date: Thursday, August 23, 2012, 3:06 AM
Hello All,
Hi all ,
can any one tell me how the AVP s:country takes its value,
$var(sdpid)=$avp(s:country);
$var(dpid)=$(var(sdpid){s.int});
xlog("$var(dpid)");
if(!dp_translate("$var(dpid)","$ruri.user/$ruri.user")){
send_reply("420", "Invalid Destination");
}
one thing more : what is the result of exe
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