Hello Guys,
I'm trying to integrate ldap with opensips. For this purpose I configured
LDAP server and added 10 users there.
My ldap.cfg file is
[sipaccounts]
ldap_version = 2
ldap_server_url = "ldap://192.168.1.106:389";
ldap_bind_dn = "cn=Manager,dc=example,dc=net"
ldap_bind_password = "passwo
Hi Everyone,
I'm using opensips 1.5.3 version. Recently we purchased DID's and forwared
to the opensips IP. The problem is when we try to call from outside to our
DID Number, no audio on both sides of the server. I can successfully
register the DID numbers. When i debug the call flow signalling i
Hi Every One,
Please don't open any mails what ever you got today from 9:00pm (IST)
onwards , cause that some one hacked my email id.
I will get back to all with positive solution.
Thanks,
Nehru
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nfused in prepaid scenario. can any on explains why the above
message occurs. And what role does b2b_init_request parameters plays in the
b2b scenario. I'm aware of prepaid parameter but not of the remaining
parameters as
"sip:3...@sip.xx.com:5070","sip:3...@xx.com:5070".
Rega
Hi Bodgan,
Thanks for ur reply. I want to hear the live conversation between two
parties , not the completed call. Is there any chance of doing it in
opensips.
Regards,
Indiver
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Hi every one,
We got an requirement regarding call spying.I want to know whether call
spying is possible in opensips. I heard that in asterisk they have given
option for silent listening of conversations between two parties with out
knowing the two parties.
Regards,
Nehru.
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The issue is solved. we had some routing issues in the router. we fixed that
part. working fine.
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Hi Abdullah,
Thanks for your response. I tried the solution mentioned in ur post ,but no
luck. I even tried to listen the two interfaces on starting opensips by
using -l option. But nothing went right for me.
Regards,
Indiver
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IP [eth1], but in vain. Can any one suggests how to
configure opensips in order to listen on both interfaces.
Regards,
Indiver.
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Thanks for your response Daniel. I included the patch and it's started
working !. Now i'm testing the rtpproxy with different scenarios. Both
clients outside NAT , behind the NAT etc. I included recording option
also,but the concern part about the rtpproxy is that after every call we
have to manua
y one specify the method.
Regards,
Indiver.
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___
Hi Bodgan,
I tried the permission module. My opensips version is 1.5.3-notls. I'm
getting the error not found command check_address. I tried to replace with
check_source_address but same result. Does 1.5.3 doesn't support the above
functions. I added the below code in my config file
if (check_ad
HI Everyone,
I want to tighten up my server security. For that purpose i want to block
unwanted calls as well as unwanted invites from hackers, so that allowing
only trusted calls and ip's. Is there any module to acheive this. Thanks in
advance.
Regards,
Nehru.
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Hi Everyone,
I installed opensips with deb package. Every thing went right. Now i want to
add new modules like xmlrpc etc. We can modify make file when installed
from source but when installed from deb package how to add new
modules.Thanks in advance.
Regards,
Indiver.
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Hello Everyone,
I want to know the scalability of opensips 1.6.0. Such as its hardware
requirements and number of calls per second. Can any one provide this info.
Thanks in advance.
Regards,
Nehru.
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Hi Andrew,
Which proprietary tool using for mixing of RTP files. So that we contact
them and try to use that. Cause that we have facing problems while using sox
and rtpbreak interms of voice quality and other issues.
Regards,
Nehru.
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Hi Bodgan,
I forgot to mention that files are not storing by callee or caller number.
Moreover it is taking its own unique caller id. How to over come this in
order to modify the recording file name as callee<->caller and time stamp
format.
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Hi Bodgan,
Yes. These files are raw rtp files. When ever call is ended rtp proxy
storing the 2 raw rtp files in to specified destination folder. One for
callee and other for caller. The problem i faced is i have to merge these 2
raw rtp files of each call and convert into wav file to hear the
con
I found the solution and now i can hear my recorded files.
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Users m
Hi Everyone,
I'm trying to record calls using rtpproxy. i called call_recording() while
i get invite message and on onreply route.
as follows:
I) if (is_method("INVITE")){
force_rtp_proxy();
start_recording();
2) onreply_route[1] {
if ((isflagset(5) || i
Hi everyone,
i can successfully chat between to sip users using imc. But i can't
understand how to send chat room creation commands and join, add users
command to opensips server.(i.e #create chat-000 private,#create chat-000
private etc commands)
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Hi Every one,
I'm running opensips with mediaproxy. I want to record particular calls
running thru my server. In rtpproxy call recording feature is inbuilt. Is
there any option for call recording using mediaproxy. I tried to use orex
call recording tool but it has no debian 64 bit package. Is the
Hi Baz,
Thanks for you advice. I'm trying to sort out my issues.
Iñaki Baz Castillo wrote:
>
> El Martes, 29 de Diciembre de 2009, Indiver escribió:
>> I forgot to mention that when i'm trying to send message from sipclient
>> to
>> xmpp client i'm ge
Hi Everyone,
I created users for my opensips server and register the users for my server.
When i checked the user location table i can view my users register details.
But the problem is after some time the Details are deleted automatically
from usr loc table. Strangely my phones r working fine and
Hi Anca,
I forgot to mention that when i'm trying to send message from sipclient to
xmpp client i'm getting 503 message too big and from xmpp client to sip
client i'm getting 404 remote server not found. Can you suggest any changes
of above cfg regarding xmpp section. Thanks in a
allback: XMPP IM received
Dec 29 00:16:31 [12500] DBG:xmpp:stream_node_callback: received message
error stanza
Dec 29 00:16:31 [12499] DBG:tm:utimer_routine: timer
routine:4,tl=0x7ff4494404a0 next=(nil), timeout=41680
Dec 29 00:16:36 [12499] DBG:tm:timer_routine: timer
routine:2,tl=0x7ff449440300 next
error. Can u suggest me where i went wrong.Thanks in Advance!
Anca Vamanu-2 wrote:
>
> Hi Indiver,
>
> The 'destination address' refers to the address of the buddy in the
> contact list. So in the xmpp account you must add the SIP buddy with the
> address:
&g
Hi Every one,
I'm working on xmpp module of opensips. I configured opensips as component
mode. I installed jabberd2 and configured successfully. When i look up the
logs of opensips it showing the handshake successful message with my local
jabberd2 server installed in my system. This make foolish
Hi Everyone,
I'm planning to do web chat using opensips. The scenario i'm planning is
when ever customer enters to our site and clicks on live chat button or chat
button what ever, it must redirect to opensips. From opensips it must
redirect to the available user of our company. Is this scenario
oxes config and monit config both have identical
> passwords and usernames. hmmm. When i do a netstat -tal, I dont see
> anything
> running on port 2812 or 8080.
>
> On Thu, Dec 10, 2009 at 9:05 PM, Indiver wrote:
>
>>
>> Hi David,
>> I worked on Monit tool and
Hi David,
I worked on Monit tool and succeeded in working that. May be this procedure
i followed helps you:
Step 1:
I included xmlrpc module which is necessary for working of most of the tabs
in opensips control panel.
Step 2:
Downloaded and configured Monit tool.
Step 3:
Checked whether the s
gt; AN_fwdok,2,Set(DB(SIP/DND/${CALLERID(num)})=1)
exten => AN_fwdok,3,Playback(beep)
exten => AN_fwdok,4,Wait(2)
exten => AN_fwdok,5,Hangup
exten => AN_fwdko,1,Answer
exten => AN_fwdko,2,NoOp(${DB_DELETE(SIP/DND/${CALLERID(num)})})
exten => AN_fwdko,3,Playback(beep)
exten => AN_fw
follows
seturi("sip:an_fw...@asterisk_ip:ASTERISK_PORT");
t_relay();
exit;
Is my assumption is right?Thanks in advance
Bogdan-Andrei Iancu wrote:
>
> Hi Indiver,
>
> the Activation Codes service is pure scripting -
Hi Iulia,
Exactly what you mentioned. But it was not affecting the call flow as per my
observation. So i left it out and continued to work on drouting.
Iulia Bublea-2 wrote:
>
> Hi Indiver,
>
> Let me see if I understoodd which the problem is.
>
> It happens when you ad
voicemail
user dials 73 it adds prefix * and go to failure route for
callfwd .
Is this scenario right?. I'm new to dialplan scenario so ignore any
miscalculation.Thanks in Advance!.
Bogdan-Andrei Iancu wrote:
>
> Hi Indiver,
>
> the Activation Codes service is pure script
, Gateways in the
Gateway tab are in Activate State(i.e Green Color). But it is not affecting
call flow. So I Ignored this and continued working on dynamic routing.
Bogdan-Andrei Iancu wrote:
>
> Hi Indiver,
>
> so, using GWs or GW-lists in the rules-tab affects the GW-list tab?
};
};
if (isflagset(27) && t_check_status("408") && t_check_status("487")) {
if (avp_pushto("$ruri", "$avp(s:fwdnoanswer)")) {
avp_delete("$avp(s:fwdnoanswer)");
resetflag(27);
route(4);
exit;
Hai Every one,
I'm trying to implement callfwd,fwdnoanswer,fwdonbusy features. According to
that i tried some scripts and callfwd is working fine. The problem is when i
implement's fwdnoanswer,fwdonbusy i'm getting 513 message too big error. My
changes in cfg file as follows
if(avp_db_load("$ru
Hi Every one,
I'm working on dynamic routing module.I'm trying to add gateways,gateway
list from controlpanel. When im trying to add gatewaylist instead of
gateways in rules tab, the list of gateways in the gateways tab are
deactivating automatically. Again after revoking to gateways in rules ta
Hi Every One,
I registered in opensips voip services site. I found a tab regarding dialing
plan. such as *78 for enable dnd,*72 for setting to permanent redirect,*50
for voicemail inbox. I found no documentation in opensips regarding these
services. Is there a way for acheiving this thru opensips
Hi Every one,
I trying to integrate jabber module in my server in order to chat with
MSN,yahoo messengers. I followed the following steps.
1. Create the database for jabber in which 4 tables(icq,yahoo,msn...)
created/
2. Installed local jabber server jabberd.
2.1. changed the configuration in
nds.Thanks in advance.
Schumann Sebastian wrote:
>
> Indiver,
>
> X-Lite might sent those commands not in plain text, but HTML embedded.
>
> The _very first_ character in the body of the message must be a sharp
> (i.e., #). As HTML starts <, it is not considered as a c
Hi Every One, I used the below to code to successfully chat between two sip
softphones.
if(uri=~"sip:q.*@")
{
# IMC - message
xdbg("script: message from [$fu] r-uri [$ru] msg [$rb]\n");
if(is_method("MESSAGE"))
{
log("MESSAGE received -> processing with imc\n");
sl_send_reply("200", "ok");
imc_
Hi Every One, I used the below to code to successfully chat between two sip
softphones.
if(uri=~"sip:q.*@")
{
# IMC - message
xdbg("script: message from [$fu] r-uri [$ru] msg [$rb]\n");
if(is_method("MESSAGE"))
{
log("MESSAGE received -> processing with imc\n");
sl_send_reply("200", "ok");
imc_
Hi Airton,
I worked on the drouting module. I hope this info helps you.
Gateways: you have to enter the enter the address of the gateway to route.
for instance 192.x.x.x
Address strip
prefix
192.168.x.x Number of digits
Hi Flavio,
Thanks for your response. I had to implement the similar scenario as
mentioned by you. I tried with different scenario by using forums,but in
vain. can you send the configuration implemented by your for local and pstn.
we have to authenticate our customers for certain destinations.
F
Hi Every one,
I'm new to the opensips. I dont know wheter it is a right question to post
here. I worked on the amost all modules of opensips. I had some queries
regarding dialplan module.
1)What is the exact usage of dialplan module other than calling
regexpression from database.
I tried some s
Hi Everyone,
I tried the dynamic routing with all types of scenarios(Prefix
based,caller,timer,Priority based routing), Which works fine .But i'm little
bit confused on script route triggering of dynamic routing module. I had
found no proper documentation regarding this. Can any on explain what i
Hi Bodgan,
Thanks for your quick response. I changed my script and drouting is working
now.
Bogdan-Andrei Iancu wrote:
>
> Hi Indiver,
>
> looking at the logs I can say your script is not right - the call does
> not hit do_routing() in the script, but rather goes to
&g
Hi Bodgan,
Sorry for the mistake. I meant that in the above when i dialed the local
number 1005 not the 1000
Bogdan-Andrei Iancu wrote:
>
> Hi Indiver,
>
> what exactly is not working?
>
> Can you post the output logs (during INVITE) with debug=6 ? Post also
> the in
@192.168.3.36]
Oct 20 08:14:19 [4760] DBG:core:get_hdr_field: to body
[]
Oct 20 08:14:19 [4760] DBG:sl:sl_filter_ACK: local ACK found -> dropping it!
Oct 20 08:14:19 [4760] DBG:core:destroy_avp_list: destroying list (nil)
Bogdan-Andrei Iancu wrote:
>
> Hi Indiver,
>
> what exactly is
Hi Every body,
I want to implement prefix based dynamic routing. I followed the following
procedure.
step 1: Enetered gateways, gateway lists, rules,groups in opensips-cp as
follows
dr_gateways table:
+--+--++---++---+-+
| gwid | type | ad
quot;1", "$ruri.user/$ruri.user")
>
> THEN your shown regex will work ($ruri.user is just the dialed number,
> does
> not contain the sip:)
> -Brett
>
> On Tue, Oct 13, 2009 at 3:05 AM, Indiver wrote:
>
>
HI Everyone,
I want to implement the dialplan module. But i was confused little bit in
the configuration part. my database table of dialplan module is
++--++--+---+---+---+--+---+
| id | dpid | pr | match_op | match_exp | match_len
t;-$var(tmp)\n");
}.
My actual intention is when the client dials the number 678... a prefix 1
should add to that numbers. Is the above configuration correct for my
architecture?.
Thanks,
Nehru.
On Mon, Oct 12, 2009 at 10:37 AM, indiver nehru wrote:
> Hi Bodgan,
>
ption for monit, in boxes.global.inc.php - maybe CP
> tried to uses https instead of http.
>
> Regards,
> Bogdan
>
> Indiver wrote:
> > Hi Everyone,
> >
> > I was unable to connect to monit tool of opensips control panel. I
> started
> > monit ser
Hi Everyone,
I was unable to connect to monit tool of opensips control panel. I started
monit server and loaded mi_xmlrpc module in opensips correctly. i had given
ip address, usename, password correctly in boxes.global.inc.php file. Still
When i tried to connect monit tool it displays i can't c
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