ecified in the modparam(
> "proto_hep",
> "hep_id", ...).
>
> The only scenario when both addresses are the same is when both OpenSIPS
> and Homer are running on the same box, which I suppose is not the case.
>
> -Max
>
> On Tue, Jun 2, 2020 at 7:07
address, or possibly 0.0.0.0 if a
> particular source address does not matter.
>
> -Max
>
> -Max
>
> On Tue, Jun 2, 2020 at 6:07 AM Burhan Khan wrote:
>
>> Hi
>>
>> I am trying to send sip traces from opensips 3.0 to remote Homer server
>> but it is getti
Error is :::
*ERROR:core:udp_init_listener: bind(27, 0x7ff5729214c4, 16) on *1.2.3.4*:
Cannot assign requested address*
*ERROR:core:trans_init_all_listeners: failed to init listener [*1.2.3.4*],
proto hep_udp*
*ERROR:core:main: failed to init all SIP listeners, aborting*
*Regards*
*Burha
Hi
I have complied RTP Proxy and its working also I have added rtpproxy module
in opensips.cfg file but it is not working.
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; value: load_from_sql, using 'none'?
>
> It looks like you should be using "load-from-sql" (with dashes)
>
> Regards,
>
> Liviu Chircu
> OpenSIPS Developer
> http://www.opensips-solutions.com
>
> On 17.09.2019 12:
Hi All,
I am trying to save user location data in SQL database for restart
persistency. I am setting parameters as
modparam("usrloc", "db_url", "mysql://root:myuser@localhost/opensips")
modparam("usrloc", "use_domain", 1)
modparam("usrloc", "location_cluster", 1)
modparam("usrloc", "cluster_mode"
s longer than the static buffer used for building
> it. Nothing more, it is not a real error.
>
> Regards,
> Bogdan
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 24.06.2014 14:17, Juned Khan wrote:
>
> Hi Bogdan,
>
>
ges, etc ?
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 09.06.2014 08:42, Juned Khan wrote:
>
> nope I don't see such messages in Log, also I confirmed about scripts,
> but i don’t have any scripts wh
is not getting terminated (stopped) by mistake
> by other scripts on your server ? I see no indication of a crash so far.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 05.06.2014 14:59, Juned Khan wrote:
>
&g
when doing normal shutdown - do you have any other "signal" reported in the
> logs ?
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 05.06.2014 09:39, Juned Khan wrote:
>
>Hi All.
>
> Since
issue.
I am using
OpenSIPS 1.7.1
Debian squeeze 6.0.9
RAM : 16GB
Here is the opensips log
http://pastebin.com/5YnvF6ub
--
Thanks,
Juned Khan
<http://www.inextrix.com/>
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tion. Can
> you run 'which mysql' and see if you have it in your PATH?
>
> Best regards,
>
> Razvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solutions.**com <http://www.opensips-solutions.com>
>
>
> On 05/28/2013 03:20 PM, Sunny Khan wr
Hello,
Please help me in this I get the following error
opensipsdbctl create
/usr/local/lib64/opensips/opensipsctl/opensipsdbctl.mysql: line 25:
locate_tool: command not found
error: 'mysql' tool not found: set MYSQL variable to correct tool path
libmysqlclient-dev, and mysql-client already i
or it?
>
>
>
> With Best Regards,
>
> *Faisal Rehman*
>
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LinkedIn
Abid khan souhaite se connecter à vous sur LinkedIn :
--
Je vous invite à faire partie de mon réseau professionnel en ligne sur le site
LinkedIn.
Accepter l'invitation de Abid khan
http://www.linkedin.com/e/-i
e and then 504 (request time out)
in end. So any one can please help me in this.
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*
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Hello everyone,
I need help to configure openIMS core to real IP from
local host, plz suggest me some links or documentation.
Thanks in Advance.
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*
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http
Hello ,
I need some help regarding forwarding of call to to android phone
when user is busy or don't want to receive call on their system.How can i do
this using presence information. i have configured open ims core and
presence server.
Thanks in Advance
--
*Abid
since last week.
Flow of packets and opensips.cfg file is attached.
--
*Abid khan
NUST-SEECS BSIT
*
text
Description: Binary data
opensips.cfg
Description: Binary data
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wtran();
t_reply("404", "Not Found");
exit;
case -2:
sl_send_reply("405", "Method Not Allowed");
exit;
}
}
# when routing via usrloc, log the missed calls also
setflag(2);
route(1);
}
route[1] {
# for INVITEs enable some additional helper routes
if (is_method("INVITE")) {
t_on_branch("2");
t_on_reply("2");
t_on_failure("1");
}
if (!t_relay()) {
sl_reply_error();
};
exit;
}
# Presence route
/* uncomment the whole following route for enabling presence
NOTE: do not forget to enable the call of this route from the main
route */
route[2]
{
if (!t_newtran())
{
sl_reply_error();
exit;
};
if(is_method("PUBLISH"))
{
handle_publish();
}
else
if( is_method("SUBSCRIBE"))
{
handle_subscribe();
}
exit;
}
branch_route[2] {
xlog("new branch at $ru\n");
}
onreply_route[2] {
xlog("incoming reply\n");
}
failure_route[1] {
if (t_was_cancelled()) {
exit;
}
# uncomment the following lines if you want to block client
# redirect based on 3xx replies.
##if (t_check_status("3[0-9][0-9]")) {
##t_reply("404","Not found");
##exit;
##}
# uncomment the following lines if you want to redirect the failed
# calls to a different new destination
##if (t_check_status("486|408")) {
##sethostport("192.168.2.100:5060");
### do not set the missed call flag again
##t_relay();
##}
}
--
*Abid khan
NUST-SEECS BSIT
*
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7.0.0.1;orig-ioi="open-ims.test"
P-hint: I-CSCF rr-enforced
P-hint: outbound
P-hint: I-CSCF rr-enforced
P-hint: outbound
P-hint: I-CSCF rr-enforced
P-hint: outbound
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ppose you configured your SIP client so that it sends Subscribe and
> Publish with a Route header ( a preloaded route) and OpenSIPS does not allow
> this. You should watch the network trace to see exactly what happens.
>
> Regards,
>
> --
> Anca Vamanu
> OpenSIPS Developer
>
&
e denied (for SUBSCRIBE)
14:49:44> NOTIFY
14:49:44> OK (NOTIFY)
--
*Abid khan
NUST-SEECS BSIT
Final Year Student*
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Ricky,
If you expand on the problem where you are stuck may be someone can help
you...
Please be specific that what is the nature of the problem where you are
stuck...
Khan
On Tue, Feb 1, 2011 at 12:21 PM, Robin Malhotra wrote:
>
> Guys I a newbie to OpenSIPS
>
> I have instal
Please let me know when you setup the blogspot i'm very interested in seeing
how you did it, or if you could provide me your configuration that will be
really great.
Thanks in advance...
Khan
On Wed, Oct 28, 2009 at 7:34 AM, Iñaki Baz Castillo wrote:
> 2009/10/28 Peter den Hartog :
&
Mani,
There is a complete working configuration posted at the following blog link:
http://voiprookie.blogspot.com/2009/04/rtpproxy-12x-installation.html
You might need to tuneup a little bit based on your needs but last time i
have tried it and it was functional.
--
Khan
VoIP Rookie
Every
ding as it works reliably
# even for UDP2TCP
if (!t_relay()) {
sl_reply_error();
};
exit;
}
# !! Nathelper
onreply_route[1] {
# NATed transaction ?
if (isbflagset(6) && status =~ "(183)|2[0-9][0-9]") {
fix_nated_contact();
force_rtp_proxy();
# otherwise, is it a transaction behind a NAT and we did not
# know at time of request processing ? (RFC1918 contacts)
} else if (nat_uac_test("1")) {
fix_nated_contact();
};
}
onreply_route[2] {
xlog("incoming reply\n");
}
failure_route[1] {
if (t_was_cancelled()) {
exit;
}
}
*
The output capture from WireShark is at the following link.
http://pastebin.com/m1c17484d
Please help me figure out this problem, I appreciate your time.
Thank you,
Khan
VoIP Rookie
Every beginning has an end regardless we believe it or not...
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ips.org/cgi-bin/mailman/listinfo/users
>> >
>> >
>>
>
>
> ___
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>
--
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VoIP Rookie
Every beginning has an end regardless we believe it or not...
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shark trace...
--
Khan
VoIP Rookie
Every beginning has an end regardless we believe it or not...
080909
Description: Binary data
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documentations and applying the knowledge.
I will keep in mind and stick with specific question, even though it
has been asked again and again...
Thanks again,
Khan
On Tue, May 12, 2009 at 11:04 AM, Adrian Georgescu wrote:
>> Khan,
>
>
> The quality of an answer seldom exceeds the
i
dont believe i am slow learner either...
Frustrated
Khan
VoIP Rookie
Every beginning has an end regardless we believe it or not...
On Mon, May 11, 2009 at 5:54 AM, Ruud Klaver wrote:
> Hi,
>
> On 09 May 2009, at 04:59, Khan wrote:
>
>> Sorry forgot the attachment of WS...
t; #911 is handled by E911 service provider
> else {
> sl_send_reply("404", "Invalid destination");
> exit;
> }
>
> # Set the callerid for the user from an AVP
> #if (
Sorry forgot the attachment of WS...
Please Help...
On Fri, May 8, 2009 at 9:56 PM, Khan wrote:
Hi everyone,
I am having trouble running OpenSIPS, while back I had same problem,
this problem is surfaced back again, I am sure there is something I am
missing in my configuration.
Currently
f
other party an error of network failure.
I have produced WS trace, please look at it and guide me what is wrong
with this situation, also my configuration is on the following link as
of today... http://pastebin.com/m3cf2769e
Thanks for all your he
ind it try following command
find / -name db_mysql.so
Drop a line if it doesnt solve your issue.
Khan
2009/5/2 khánh nguyễn :
> hello
> i'm very excited with ser project.
> when i try to compile and install i found a pob with "make
> prefix=/usr/local install include_mod
es/accounting/radius.py",
line 53, in __init__
self.radius = RadiusAccounting()
File
"/etc/mediaproxy/mediaproxy-2.3.4/mediaproxy/interfaces/accounting/radius.py",
line 74, in __init__
secrets = dict(line.rstrip("\n").split(None, 1) for line in
open(config["server
11246a
I tried to figure out the problem, the highlighted parts might be the
problem area if you could please give a quick look at see where in
configuration script i went wrong?
I know i am asking for too much but please help me, I really
appreciate your help !
Khan
On Fri, Apr 10, 2009 at 7:44
me and helping me out to
learn and implement this project.
Thank you all,
Khan
On Fri, Apr 10, 2009 at 7:49 AM, Bogdan-Andrei Iancu
wrote:
> Hi Khan,
>
> What version and what revision are you using (use opensips -V) ?
>
> Also, try to get a core file (set "ulimit -c un
ove changes when i start my opensips it
crashes, the detailed out put is at this link
http://pastebin.com/m4adecddb
Can someone please help me or guide me if i need to update my opensips
1.5 because i see a lot of discussion going on about crashes and bug
fixes.
Also can someone explain why am i getting ACK timed out and how to
resolve auth failure or its suppose to happen???
Please help
Khan
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the right track
Thanks for all your time and efforts...
Khan
On Thu, Apr 9, 2009 at 11:00 AM, Khan wrote:
> On Thu, Apr 9, 2009 at 2:13 AM, Uwe Kastens wrote:
>> Khan,
>>
>> Would it be possible to add a tcpdump/wirshark on the opensips and on
>> the client in
. There are OpenSIPS, MySQL
server, and RTPProxy is running on the box.
Please respond to my request considering my skills in the SIP as
rookie, guide me on how to resolve problem...
Thanks,
Khan
Sorry for such a long email, I am frustrated :(
__
to screen output / opensips.cfg file :
http://pastebin.com/m738bb3f8
The highlighted portion is the error part, I can not pin point the
problem area, so please help me resolve this issue.
I am new to this environment so i need all the help i can get.
Thank you in advance for your time...
Khan
ext/opensips/`basename "$FILE"` ; \
fi ;\
done ;\
fi
ERROR: module modules/”db_mysql”/”db_mysql”.so not compiled
make[1]: Entering directory `/usr/src/opensips-1.5.0-tls/modules/acc'
make[1]: Nothing to be done for `install_module_custom'.
make[1]: Leaving directory `/usr/src/opensips-1.5.0-tls/modules/acc'
make[1]: Entering directory `/usr/src/opensips-1.5.0-tls/modules/alias_db'
make[1]: Nothing to be done for `install_module_custom'.
Thanks in advance...
Khan
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?
Khan
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rom R0 ... NAT test - M=SUBSCRIBE
RURI=sip:10...@mydomain.com F=sip:10...@mydomain.com
T=sip:10...@mydomain.c
om IP=
My request... M=SUBSCRIBE sip:10...@mydomain.com
Requested Service Unavailable PUBLISH/SUBSCRIBE/NOTIFY - M=SUBSCRIBE
Khan
___
Use
e to open any special port tcp/udp for rtpproxy? )
I'm also attaching the SIP trace and the command line dump for review. I'm
unable to understand what did i do wrong. I'm rookie to VoIP so please help
me grow in this technology.
Any suggestion is appreciated, I'm out of ideas
Hi everyone,
I have installed OpenSIPS 1.4.4 recently on a ubuntu 8.04, I have 3 UAC
configured. Two of them are SJphone which are within the same network. One
of them (Xlite) is outside the network and behind NAT. I am facing two
problems.
1. The SJphone within network works fine but user hear h
ve to create new database with the name
opensips_1_4?
Please help.
Khan
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Hello All,
I am in need of help, I did not receive any response from anyone. Please
take a look at this problem and guide me, seriously I am frustrated :(
May be there is a secret handshake for this list to get responses :)
Khan
On Mon, Jan 5, 2009 at 9:19 PM, Khan Friend wrote:
> He
get_hdr_field: found end of header
Jan 5 20:38:15myosips[5021]: ERROR:core:forward_reply: no 2nd via found in
reply
Jan 5 20:38:15myosips[5021]: DBG:core:destroy_avp_list: destroying list
(nil)
Jan 5 20:38:15myosips[5021]: DBG:core:receive_msg: cleaning up
###
I would be interested in tutorial too, please keep me updated.
--
Thank you,
Khan
On Mon, Jan 5, 2009 at 5:22 AM, Asim Riaz wrote:
> Hi Bogdan,
> could you please explain me what is involve in the migration from openser
> 1.2 to opensips or send me link.
>
> Thanks,
> Asim
t not much luck
understanding what part needs fixing.
Please help me identify the root of the problem and how to fix. How do i
find SIP replies, what do i do to see them and capture them.
Thanks in advance,
Khan
On Sun, Dec 28, 2008 at 4:33 AM, Bogdan-Andrei Iancu wrote:
> Hi Khan,
exit;
}
# route to Asterisk Media Server
rewritehostport("192.168.1.11:5060");
route(1);
}
failure_route[1] {
if (t_was_cancelled()) {
xdbg("transaction was cancelled by UAC\n");
return;
}
# restore initial uri
avp_pushto("$ru","$avp(i:10)");
#avp_pushto("$ru", "i:10");
prefix("1");
# route to Asterisk Media Server
rewritehostport("192.168.1.11:5060");
resetflag(2);
route(1);
}
Thank you,
Khan
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I have OpenSIPS Server - Installed and functional on Debian, Problem presist
in NAT traversal area (RTPProxy, Asterisk) needs fine tunning...
Authentication is done with MySQL (digest authentication with qop=auth),
Managing multiple domains, alternative routes, inbound-to-outbond,
outbond-to-inbou
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