login to mysql using opensips credentials as well.
Kindly advise me how to resolve this issue.
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-ID: b115ce088a57d010.
CSeq: 8160 REGISTER.
User-Agent: Asterisk PBX.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Supported: replaces.
Contact: sip:322...@zz.zz.zz.zz.zz.
Content-Length: 0.
.
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for
OpenSIPs.
Kindly advise me to resolve this issue.
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On Thu, Dec 17, 2009 at 10:25 PM, Ahmed Munir ahmedmunir...@gmail.com
wrote:
Hi,
I'm using OpenSIPs version 1.6, the module I'm using is dispatcher using
mysql. My question is how can I send SIP header 302 registration request
to
Asterisk? Because Asterisk is sending me unAuthorized
###);
ds_select_dst(1,4);
#t_replicate(77.66.2.136);
#sl_send_reply(200, ok);
forward();
log(#END);
#exit;
return;
}
}
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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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Message-ID:
b74751491001010545y73756771jc1ef52fb4d205...@mail.gmail.com
Content-Type: text/plain; charset=iso-8859-1
check any Firewall involved
or use ngrep to track the packet
Ram
On Fri, Jan 1, 2010 at 6:17 PM, Ahmed Munir ahmedmunir...@gmail.com
wrote
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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can't be used and gives me an error,
kindly advise this well.
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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
Ahmed Munir wrote:
Hi,
I'm using permission module's function check_source_address(), the
problem I'm facing is that I can
]:
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 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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; charset=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 add Source-IP, 13
And I also
=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 add Source-IP, 13
And I also check table
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.
Regards,
Bogdan
Ahmed Munir wrote:
Hi,
Thanks for your reply, I've copied dictionary.opensips to
/etc
,
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
freeradius service and make a call and still getting same error as I
mentioned on my previous mail.
Kindly
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
freeradius service and make a call and still getting same error as I
mentioned on my
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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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?
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)
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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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) on one
UAS(OpenSIPS) can call SIP-SIP, if any one phone is registered on UAS
and other is on PSTN, call will be re-routed
Ahmed,
Ahmed Munir wrote:
Hi Bogdan,
Thanks for reply. I forgot to mention earlier that for I'm using
OpenSIPS + FreeRadius, where radius is doing accounting and
authentication. I used aaa_does_uri_exist() function as well, but
seems not working or making mistake while implementing
Peers on OpenSIPS
To: OpenSIPS users mailling list users@lists.opensips.org
Message-ID: 4ba69927.2050...@voice-system.ro
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Hi Ahmed
Ahmed Munir wrote:
Hi Bogdan,
Thanks for your suggestion, few things I want to ask from you;
1
:]]*', ';nat=yes');
if (isbflagset(6)) {
fix_nated_contact();
}
exit;
}
Kindly state, how can I resolve this error in my above configuration. Please
advise.
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- 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
Regards,
Bogdan
Ahmed Munir wrote:
Hi,
I've configured OpenSIPs with Radius and now working
)) {
fix_nated_contact();
}
exit;
}
Kindly help me out with this problem, in which other section Natting is
required?(or am I missing something in the configuration?) Please assist me
on it.
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-Type: text/plain; charset=iso-8859-1
Ahmed,
Could you send an wireshark trace to the list? It will be easier to check
what's going wrong.
Besta regards,
Antonio Anderson M. Souza
Voice Technology
http://www.antonioams.com
Em 29/04/2010 11:47, Ahmed Munir ahmedmunir...@gmail.comescreveu
for your reply.
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/modules/1.6.x/nathelper.html#id229028
-Laszlo
2010/4/30 Ahmed Munir ahmedmunir...@gmail.com
Hi.
Thanks for your reply, the traces are metioned below;
U 203.215.176.22:55134 - 11.22.33.44:5060
.
.
..
U 81.201.82.45:5060 - 11.22.33.44:5060
INVITE sip:1234
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:
Hi,
Thanks for replying. Can you please check my
.
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May 17 05:59:48 newtest /usr/local/sbin/opensips[31303]: ERROR:core:main:
error while initializing modules
Kindly assist me how can I resolve this problem.
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at earliest as I
haven't worked on SNMP.
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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 ahmedmunir...@gmail.com
Subject: [OpenSIPS-Users] OpenSIPs + SNMP
To: OpenSIPs Users users@lists.opensips.org
Message-ID:
CAGMN=Jfaa2_YoAbo6T+ZUnbQD2euBUK2XYM9W4TU=y97tuo
to Asterisk it works fine, how
can I fix this?
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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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, 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't want to reload dispatcher once failed GW
in active state. Please advise.
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do with the avp_db_query(). 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:
>
>
heavy lifting
> during startup.
>
> 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 ch
Is there updates on this?
Date: Mon, 16 Jan 2017 09:51:13 -0500
> From: Ahmed Munir <ahmedmunir...@gmail.com>
> To: OpenSIPs Users <users@lists.opensips.org>
> Subject: Re: [OpenSIPS-Users] OpenSIPs crashed
> Message-ID:
>
ase 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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es) =
> avp_db_query(...). 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,
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.
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s 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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ww.opensips-solutions.com<http://www.opensips-solutions.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: We
ions
> www.opensips-solutions.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
> > *Ahm
ot;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 'Flag' value to 0 and '
; version and 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:
>
what might causing the CPU
to spike.
Please advise what useful steps to take for narrowing down this issue.
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IPS you are running.
>>
>> Best regards,
>>
>> Răzvan Crainea
>> OpenSIPS Solutions
>> www.opensips-solutions.com
>>
>>
>> On 01/11/2017 11:10 PM, Ahmed Munir wrote:
>> > Hi,
>> >
>> > Our OpenSIPs service crashed wit
gt;
> Răzvan Crainea
> 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
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