earliest as I
haven't worked on SNMP.
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#x27;opensipDialogTable'. How can I run this option using
snampwalk command to check current active
calls? Do I need to some more lines in OpenSIPs configuration?
Please advice.
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Anyone, please assist me out at earliest.
> Date: Fri, 1 Jun 2012 11:08:28 -0400
> From: Ahmed Munir
> Subject: [OpenSIPS-Users] OpenSIPs + SNMP
> To: OpenSIPs Users
> Message-ID:
> >
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi all,
&
t; what to do with it, for example, I set up on my OpenSIPS/Asterisk
> system a user freepbx1 so that OpenSIPS can authenticate it, on the
> FreePbx there are 50 users 100-150, FreePbx sends the INVITE as from
> 101@freepbx.
>
> When connecting the FreePbx directly to Asterisk i
'm getting
this message as listed below;
opensips: Unknown Object Identifier (Sub-id not found: (top) -> opensips)
Please assist me at earliest, to resolve this issue.
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ispatcher", "ds_probing_threshhold", 3)
modparam("dispatcher", "ds_probing_mode", 1)
modparam("dispatcher", "options_reply_codes", "501, 403")
modparam("dispatcher", "ds_ping_from", "sip:pr...@proxy.com")
Setting
into logs, unable to find the info what might causing the CPU
to spike.
Please advise what useful steps to take for narrowing down this issue.
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ainea
> OpenSIPS Solutions
> www.opensips-solutions.com
>
> On 01/11/2017 11:10 PM, Ahmed Munir wrote:
> > Hi,
> >
> > Our OpenSIPs service crashed with below error;
> >
> > Jan 11 12:16:19 QORBLPSIPROXY05 abrtd: Directory
> > 'ccpp-2017-01-11-12
;> Also, please let us know the version of OpenSIPS you are running.
>>
>> Best regards,
>>
>> Răzvan Crainea
>> OpenSIPS Solutions
>> www.opensips-solutions.com
>>
>>
>> On 01/11/2017 11:10 PM, Ahmed Munir wrote:
>> > Hi,
>>
loner () from
/usr/lib64/opensips/modules/tm.so
Missing separate debuginfos, use: debuginfo-install
opensips-1.6.3-notls.x86_64
Please advise what might be the reason causing opensips to crash.
On Thu, Jan 12, 2017 at 1:28 PM, Ahmed Munir
wrote:
>
> The version currently running is 1.
see if your problem is solved. Otherwise, upgrade to a
> newer, supported version, preferably the latest stable release, 2.2.2.
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Solutions
> www.opensips-solutions.com
>
> On 01/12/2017 11:55 PM, Ahmed Munir wrote:
> > Found c
(...). But anyways, this is something completely different,
> so please open a different topic for it.
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Solutions
> www.opensips-solutions.com
>
> On 01/14/2017 12:24 AM, Ahmed Munir wrote:
> > Hi,
> >
> > I
s.
Please advise the steps do I need to take to fix above issues.
BTW, declared avpops 'db_url' in module parameters.
modparam("dispatcher|avpops","db_url","mysql://opensips:opensipsrw@localhost
/opensips")
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Is there updates on this?
Date: Mon, 16 Jan 2017 09:51:13 -0500
> From: Ahmed Munir
> To: OpenSIPs Users
> Subject: Re: [OpenSIPS-Users] OpenSIPs crashed
> Message-ID:
> gmail.com>
> Content-Type: text/plain; charset="utf-8"
>
> See det
pe=SIP receiver udp:10.3.120.94:5060
Process:: ID=10 PID=3100 Type=SIP receiver udp:10.3.120.94:5060
Process:: ID=11 PID=3102 Type=SIP receiver udp:10.3.120.94:5060
Process:: ID=12 PID=3104 Type=Timer handler
I would like to know what changes required to fix this change? Please
advise.
--
Regard
lutions.com
>
> On 01/18/2017 11:55 PM, Ramachandran, Agalya (Contractor) wrote:
> >
> > Same with my case too.
> >
> > Regards,
> > Agalya
> >
> > *From:*Users [mailto:users-boun...@lists.opensips.org] *On Behalf Of
> > *Ahmed Munir
> > *Sent:*
s.com>
> On 01/18/2017 11:55 PM, Ramachandran, Agalya (Contractor) wrote:
> Same with my case too.
>
> Regards,
> Agalya
>
> From: Users [mailto:users-boun...@lists.opensips.org] On Behalf Of Ahmed
> Munir
> Sent: Wednesday, January 18, 2017 1:31 PM
> To: OpenSIPs Users <mailto:us
kid -> 117 | Return
Code: 0---
Is there is way to properly retain the $retcode/$rc in version 2.2.2? Seems
like using async return code(s) are not properly set or the avp variables
are not setting up correct using async statement.
Please advise, if the above async db statement is correct as shared in
sample above.
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(). If you need it later
> in the script, better save it, as you do in the first example.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 01/20/2017 01:31 AM, Ahmed Munir wrote:
>
> Hi,
>
> Currently I&#x
artup.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 01/19/2017 06:01 PM, Ahmed Munir wrote:
>
> Hi Razvan,
>
> During starting up the opensips service, I see the first opensips child
> process (pid&qu
lhost:705 -Lsd -Lf /dev/null -smux -p
/var/run/snmpd.pid -c /etc/snmp/snmpd.conf
Please advise if I missed out any steps for executing objects
OPENSER-REG-MIB::openser &
OPENSER-SIP-COMMON-MIB::openserSIPCommonObjects using snmpbulkwalk.
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tcher", "ds_ping_method", "INFO")
modparam("dispatcher", "ds_ping_interval", 10)
modparam("dispatcher", "ds_probing_threshhold", 3)
modparam("dispatcher", "ds_probing_mode", 1)
Further added, as a note; I don'
figured OpenSips on my system as
well and its running and calls are made.
Kindly advise me solution how can I resolve this issue
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: text/plain; charset=ISO-8859-1; format=flowed
>
> Hi Ahmed,
>
> If you have the devices behind NAT and the server is public IP space,
> you need a NAT traversal solution for SIP . Either Near-End NAT
> traversal (via STUN) or via Far-Ed NAT traversal (in the opensips script).
&
ward();
route(1);
setflag(1); # do accounting
}
Further added I can even login to mysql using opensips credentials as well.
Kindly advise me how to resolve this issue.
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DB) must be a SIP URI, like sip:xx.xx.xx.xx:5060
>
> Regards,
> Bogdan
>
> Ahmed Munir wrote:
> > Hi,
> >
> > I'm getting error using Dispatcher module as listing below;
> >
> > Dec 11 07:22:57 newtest /usr/local/sbin/opensips[15487]:
> > ERROR:core:pars
5060
;received=203.215.176.22;received=203.215.176.22;branch=z9hG4bK-d87543-928337242-1--d87543-;rport=46183;rport=46183.
From: 322025;tag=9d7c6756.
To: 322025.
Call-ID: b115ce088a57d010.
CSeq: 8160 REGISTER.
User-Agent: Asterisk PBX.
Allow: INVI
me as I'm using for
OpenSIPs.
Kindly advise me to resolve this issue.
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k
> To: OpenSIPS users mailling list
> Message-ID:
>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Getting some ngrep traces will help some other to help you. Unauthorized
> message is for useraccount or for opensip.
> -Jai
>
>
> On Thu, Dec
Hi,
I've got 2 machines, i.e. machine A is OpenSIPS and machine B is Asterisk
which is registrar server. Can anybody tell me how can I configure machine A
as a Redirect Server?
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);
exit;
}
if (is_method("REGISTER")) {
log("REGISTER###");
ds_select_dst("1","4");
#t_replicate("77.66.2.136");
#sl_send_reply(&
do I require for doing it.
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Hi,
I want to forward an Access Number from OpenSIPS to Asterisk machine. Kindly
advise how can i do that? Which modules/functions are use to forward by
INVITE section?
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http
IP of Asterisk i.e. yy.yy.179.54 to
OpenSIPs, I'm getting error of Request Pending. Even same configuration is
set on yy.yy.179.137 Asterisk machine A as on xx.xx.2.136 Asterisk machine
B.
Kindly advise me how can I resolve this issue/which part I need to
configure.
--
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A
eq: 2 ACK.
Content-Length: 0.
.
Kindly help me to resolve this problem.
> Date: Fri, 1 Jan 2010 19:15:00 +0530
> From: ram
> Subject: Re: [OpenSIPS-Users] Need help for Call to another network
> To: OpenSIPS users mailling list
> Message-ID:
>
> Content-Type: text/pla
Hi,
I want to implement ACL on OpenSIPs to accept the call on behalf of source
URI + IP address. Can anyone tell me which modules and functions are
required for it?
Also kindly share some example template with it.
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Hi,
I want to implement ACL on OpenSIPs to accept the call on behalf of source
URI + IP address. Can anyone tell me which modules and functions are
required for it also which tables will involve in it?
Also kindly share some example template with it.
--
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Ahmed Munir
case. Further added, how can I enter domain name in 'ip' column
section of address table i.e. abc.com can't be used and gives me an error,
kindly advise this well.
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ing pvar. If the group_id argument is "0", the
> query can match any group in the cached address table"
>
>
> Secondly, as the name suggests, the ip column is reserved for IPs only.
> You cannot add domain name addresses to this column.
>
>
> Regards,
> Irina Stanescu
7 rose /usr/local/sbin/opensips[25988]:
ERROR:acc:acc_aaa_request: failed to add Source-IP, 13
And I also check table radacct in mysql database, no records are inserted
into it.
Kindly advise this issue.
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harset=ISO-8859-1; format=flowed
>
> On 26.02.2010 14:33, Ahmed Munir wrote:
> > Feb 26 14:47:47 rose /usr/local/sbin/opensips[25988]:
> > ERROR:aaa_radius:rad_avp_add: failure
> > Feb 26 14:47:47 rose /usr/local/sbin/opensips[25988]:
> > ERROR:acc:acc_aaa_request: failed to
radacct in mysql database, no records are inserted
> into it.
I think this means an incorrect RADIUS dictionary. You should verify
that the extra attributes you have defined are present there.
--
Sincerely,
Andrew Pogrebennyk
--
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Ahmed Munir
__
s.dictionary, then it no longer needs to be in
> my local.dictionary.
>
> Regards,
> Norm
>
> Bogdan-Andrei Iancu wrote:
> > Hi Ahmed,
> >
> > Do you see in any RADIUS dictionary the Source-IP AVP ? if yes, please
> > post here its definition.
&g
reeRadius
> To: OpenSIPS users mailling list
> Message-ID: <4b912d24.40...@goes.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> Please post the current error that you're receiving so that someone on
> the list might be able to point you in the correct dir
able to point you in the correct direction.
>
> Regards,
> Norm
>
> Ahmed Munir wrote:
> > Hi,
> >
> > Thanks for replying, Norman I've added the line in dictionary.opensips
> > i.e. ATTRIBUTE Source-IP 214 string, and start
> > freeradiu
I can authenticate and
register my softphone? Like in tables radreply, radgroupcheck, radgroureply,
realms, etc.
Kindly put the light on it and assist me with some sample data.
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a call to User B who is not
registered on UAS but located on PSTN, SIP-PSTN.
In summary I need to know how can I configure SIP-SIP, SIP-PSTN and PSTN-SIP
peers and how can I distribute their routes? Further added, which modules,
modparam and function requires for it?
--
R
ation")
function works and I need to know how can I forward call to SIP-PSTN ?
Kindly advise me the method/ function can used for it.
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unction:
> http://www.opensips.org/html/docs/modules/1.6.x/uri.html#id271131
>
> Regards,
> Bogdan
>
> Ahmed Munir wrote:
> > Hi,
> >
> > I want to know how can I check the peers of source and destination
> > phones? Like if both phones are located (registered)
;location"))
sl_reply_error();
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;
}
}
Kindly assist
3 +0200
> From: Bogdan-Andrei Iancu
> Subject: Re: [OpenSIPS-Users] Check Live Peers on OpenSIPS
> To: OpenSIPS users mailling list
> Message-ID: <4ba69927.2050...@voice-system.ro>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> Hi Ahmed
>
> Ahmed M
_register();
} else {
fix_nated_contact();
};
setflag(5);
};
}
route[4]
{
log("#### FUNCTION ROUTE 4 RTP PROXY
");
if (is_method("BYE"))
ny rtpproxy to be used - the
> rtpproxy_sock parameter is empty:
> modparam("nathelper","rtpproxy_sock","")
>
> You need to set a valid link to a running rtpproxy :
>
> http://www.opensips.org/html/docs/modules/1.6.x/nathelper.html#id228332
>
;
setflag(5);
};
}
route[4]
{
log(" FUNCTION ROUTE 4 RTP PROXY
");
if (is_method("BYE")) {
unforce_rtp_proxy();
} else if (is_method("INVITE")){
force_rtp_proxy();
#t_on_failure("
OpenSIPs) -> Asterisk
> --------> UACtwo way voice is establised
> PSTN--> UAS(OpenSIPs)
> -> UAC one way
> voice is establised
> (hears the dest voice)
if (is_method("BYE")) {
unforce_rtp_proxy();
} else if (is_method("INVITE")){
force_rtp_proxy();
#t_on_failure("2");
t_on_failure("3");
};
if (isflagset(5))
NAT Problem using Nat helper
> To: OpenSIPS users mailling list
> Message-ID:
>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi Ahmed,
>
> As you can see, the other party gets local ip in SDP
>
> c=IN IP4 192.168.0.168.
>
>
bflag will will
> automatically set if the callee location is nated -> you can use that
> flag to detect the nated callee and to do the nat fixups -> force rtpp
> and fix the 200 ok from the callee (SDP and contact).
>
> Regards,
> Bogdan
>
> Ahmed Munir wrote:
>
} else {
fix_nated_contact();
};
setflag(5);
};
}
route[4]
{
log(" FUNCTION ROUTE 4 RTP PROXY
");
if (is_method("BYE")) {
unforce_r
test /usr/local/sbin/opensips[31303]:
DBG:stun:stun_mod_init: grep_sock_in()1 failed
May 17 05:59:48 newtest /usr/local/sbin/opensips[31303]:
ERROR:core:init_mod: failed to initialize module stun
May 17 05:59:48 newtest /usr/local/sbin/opensips[31303]: ERROR:core:main:
error while initializi
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