date:20180116

Re: [OpenSIPS-Users] Send LUA logs via HEP to Homer server

2018-01-16 Thread Abdul Basit

Thank you for prompt reply.

I will look into it.


--
regards,

abdul basit


On 16 January 2018 at 23:08, Serge S. Yuriev  wrote:

> Hi
>
> I believe you should look at Pastash from Homer authors
>
> --
> Wbr, Serge via mobile
>
> 16.01.2018, 19:59, "Abdul Basit" :
>
> Hi team,
>
> I am processing incoming INVITE events in my lua script.
> To print the logs with kamailio syslog, i am using
>
>  sr.log("info", "-- Collecting data from kamailio -- ")
>
> I am searching for a way to send selective logs for an incoming call from
> lua script through HEP3 towards Homer server.
>
> How can I make it work? What utility or library is helpful.
>
> *PS:* captagent is configured on the same server for SIP traces.
>
> I know this should go to kamailio team, but how opensips will implement
> this lua integration with HEP.
>
> --
> regards,
>
> abdul basit
>
>
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Re: [OpenSIPS-Users] Send LUA logs via HEP to Homer server

2018-01-16 Thread Serge S . Yuriev

HiI believe you should look at Pastash from Homer authors-- Wbr, Serge via mobile16.01.2018, 19:59, "Abdul Basit" :Hi team,I am processing incoming INVITE events in my lua script.To print the logs with kamailio syslog, i am using     sr.log("info", "-- Collecting data from kamailio -- ")I am searching for a way to send selective logs for an incoming call from lua script through HEP3 towards Homer server. How can I make it work? What utility or library is helpful.PS: captagent is configured on the same server for SIP traces.I know this should go to kamailio team, but how opensips will implement this lua integration with HEP.--regards,abdul basit


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[OpenSIPS-Users] rtpproxy error messages

2018-01-16 Thread John Quick

Hello,

Can anyone help me understand what might be causing the following error
messages to be reported by OpenSIPS?
2018-01-16 16:42:47  ERROR:rtpproxy:engage_rtp_proxy4_f: error forcing rtp
proxy
2018-01-16 16:42:48  ERROR:rtpproxy:force_rtp_proxy_body: unhandled rtpproxy
error: 71
2018-01-16 16:42:48  ERROR:rtpproxy:force_rtp_proxy_body: unhandled rtpproxy
error: 71
2018-01-16 16:42:49  ERROR:rtpproxy:force_rtp_proxy_body: unhandled rtpproxy
error: 72

Thanks.

John Quick
Smartvox Limited
Web: www.smartvox.co.uk


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[OpenSIPS-Users] Send LUA logs via HEP to Homer server

2018-01-16 Thread Abdul Basit

Hi team,

I am processing incoming INVITE events in my lua script.
To print the logs with kamailio syslog, i am using

 sr.log("info", "-- Collecting data from kamailio -- ")

I am searching for a way to send selective logs for an incoming call from
lua script through HEP3 towards Homer server.

How can I make it work? What utility or library is helpful.

*PS:* captagent is configured on the same server for SIP traces.

I know this should go to kamailio team, but how opensips will implement
this lua integration with HEP.

--
regards,

abdul basit
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Re: [OpenSIPS-Users] SIP-I Calling Party number being changed

2018-01-16 Thread Aqs Younas

Thanks for the explanation. We changed + to 000 now things are pretty good.
Thanks.


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<#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>

On 15 January 2018 at 21:21, Vlad Patrascu  wrote:

> Hi,
>
> I'm not an expert on ITU-T standards but from what I know, the "+" sign is
> only an indication that an international prefix is required and thus cannot
> be set as a digit in the ISUP number related parameters. Being an
> unsupported character, the sip_i module wrote it as 0. Maybe a warning log
> message would be helpful.
>
> As such you should probably replace the plus sign with the international
> call prefix and only after that set the ISUP parameter.
> Regards,
>
> Vlad Patrascu
> OpenSIPS Developerhttp://www.opensips-solutions.com
>
> On 01/15/2018 12:39 PM, Aqs Younas wrote:
>
> Greetings list,
>
> I am using the sip-i module for conversion between sip to sip-i.  I want
> to set calling party number and called party number in e164 format. Below
> is how my code looks like.
> route{
> ...
> xlog("Here is Called party number: $rU");
> xlog("Here is Calling party number: $fU");
> do_accounting("log","missed");
> add_isup_part("Initial address");
> $isup_param(Nature of connection indicators) = "0x10";
> $isup_param(Forward Call Indicators) = "0x6001";
> $isup_param(Called party number|Nature of address indicator) = 4;
> $isup_param(Called party number|Numbering plan indicator) = 1;
> $isup_param(Called party number|Address signal) = $rU;
> $isup_param(Calling party number|Nature of address indicator) = 4;
> $isup_param(Calling party number|Numbering plan indicator) = 1;
> $isup_param(Calling party number|Screening indicator) = 4;
> $isup_param(Transmission Medium Requirement) = "speech";
> $isup_param(Calling party number|Address signal) = $fU;
> ...
> }
>
> branch_route[per_branch_ops] {
> xlog("NOA is: $isup_param_str(Called Party Number|Nature of
> address indicator)");
> xlog("Called party is: $isup_param_str(Called Party Number|Address
> signal)");
> xlog("NOA is: $isup_param_str(Calling Party Number|Nature of
> address indicator)");
> xlog("Calling is: $isup_param_str(Calling Party Number|Address
> signal)");
> }
>
> But I see Calling party and Called party number being changed from what i
> had set.
>
> This is what I set.
>
>  Jan 15 10:16:01 gridtele-ug-vpn ./opensips[6911]: Here is Called party
> number: +256723000422 <+256%20723%2422>
> Jan 15 10:16:01 gridtele-ug-vpn ./opensips[6911]: Here is Calling party
> number: +12064248192 <(206)%20424-8192>
>
> And what appears in ISUP body. You can notice *+ *changed to 0.
>
> Jan 15 10:16:01 gridtele-ug-vpn ./opensips[6911]: NOA is: international
> *Jan 15 10:16:01 gridtele-ug-vpn ./opensips[6911]: Called party is:
> 0256723000422*
> Jan 15 10:16:01 gridtele-ug-vpn ./opensips[6911]: NOA is: international
> *Jan 15 10:16:01 gridtele-ug-vpn ./opensips[6911]: Calling is:
> 012064248192*
>
> Any help/suggestion is much appreciated.
>
> Thanks.
>
>
>
> 
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> 
>
>
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Re: [OpenSIPS-Users] Not able to make Load Balancer work

2018-01-16 Thread Mayur Koshal

Hi Bogdan ,


I will try the script_trace and make a call.

Meanwhile I had put xlogs in the load balancer script to see the
destination and RURI.

The destination is coming as NULL when I try to make a call using sipp.
Also error message is seen in the LB logs when the call is executed  :

 ERROR:core:print_rr_body: too many RR
 ERROR:dialog:get_routing_info: failed to print route records


The logs and LB script route snippet is as below :



Logs_LB :
---
 stack-VirtualBox /usr/local/opensips_proxy/sbin/opensips[20729]: DEst
and RURI after record route sip:Mayur@192.168.100.19:5060/
Jan 16 16:35:41 stack-VirtualBox
/usr/local/opensips_proxy/sbin/opensips[20722]: Inside main route loop
LoadBalance
Jan 16 16:35:41 stack-VirtualBox
/usr/local/opensips_proxy/sbin/opensips[20722]: Destination available
before initial req are sip:Mayur@192.168.100.19:5060/
Jan 16 16:35:41 stack-VirtualBox
/usr/local/opensips_proxy/sbin/opensips[20722]: DEst and RURI before
record route sip:Mayur@192.168.100.19:5060/
Jan 16 16:35:41 stack-VirtualBox
/usr/local/opensips_proxy/sbin/opensips[20722]: DEst and RURI after
record route sip:Mayur@192.168.100.19:5060/
Jan 16 16:35:41 stack-VirtualBox
/usr/local/opensips_proxy/sbin/opensips[20722]:
ERROR:core:print_rr_body: too many RR
Jan 16 16:35:41 stack-VirtualBox
/usr/local/opensips_proxy/sbin/opensips[20722]:
ERROR:dialog:get_routing_info: failed to print route records
Jan 16 16:35:41 stack-VirtualBox
/usr/local/opensips_proxy/sbin/opensips[20721]: Inside main route loop
LoadBalance
Jan 16 16:35:41 stack-VirtualBox
/usr/local/opensips_proxy/sbin/opensips[20721]: Destination available
before initial req are sip:Mayur@192.168.100.19:5060/
Jan 16 16:35:41 stack-VirtualBox
/usr/local/opensips_proxy/sbin/opensips[20721]: DEst and RURI before
record route sip:Mayur@192.168.100.19:5060/
Jan 16 16:35:41 stack-VirtualBox
/usr/local/opensips_proxy/sbin/opensips[20721]: DEst and RURI after
record route sip:Mayur@192.168.100.19:5060/
Jan 16 16:35:41 stack-VirtualBox
/usr/local/opensips_proxy/sbin/opensips[20721]:
ERROR:core:print_rr_body: too many RR
Jan 16 16:35:41 stack-VirtualBox
/usr/local/opensips_proxy/sbin/opensips[20721]:
ERROR:dialog:get_routing_info: failed to print route records
Jan 16 16:35:41 stack-VirtualBox
/usr/local/opensips_proxy/sbin/opensips[20718]: Inside main route loop
LoadBalance
Jan 16 16:35:41 stack-VirtualBox
/usr/local/opensips_proxy/sbin/opensips[20718]: Destination available
before initial req are sip:Mayur@192.168.100.19:5060/
Jan 16 16:35:41 stack-VirtualBox
/usr/local/opensips_proxy/sbin/opensips[20718]: DEst and RURI before
record route sip:Mayur@192.168.100.19:5060/
Jan 16 16:35:41 stack-VirtualBox
/usr/local/opensips_proxy/sbin/opensips[20718]: DEst and RURI after
record route sip:Mayur@192.168.100.19:5060/
Jan 16 16:35:41 stack-VirtualBox
/usr/local/opensips_proxy/sbin/opensips[20718]:
ERROR:core:print_rr_body: too many RR
Jan 16 16:35:41 stack-VirtualBox
/usr/local/opensips_proxy/sbin/opensips[20718]:
ERROR:dialog:get_routing_info: failed to print route records
Jan 16 16:35:41 stack-VirtualBox
/usr/local/opensips_proxy/sbin/opensips[20739]: Inside failure_route
GW_FAILOVER
Jan 16 16:35:41 stack-VirtualBox
/usr/local/opensips_proxy/sbin/opensips[20728]: Inside main route loop
LoadBalance
Jan 16 16:35:41 stack-VirtualBox
/usr/local/opensips_proxy/sbin/opensips[20728]: Inside else of loose
route
---

Load Balancer Script :

### Routing Logic 

# main request routing logic

route{

if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
exit;
}


  xlog("Inside main route loop LoadBalance\n");

if (has_totag()) {
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route()) {
  xlog("Inside loose_route loop LoadBalance\n");
# validate the sequential request against dialog
if ( $DLG_status!=NULL && !validate_dialog() ) {
xlog("In-Dialog $rm from $si (callid=$ci) is not valid according to dialog\n");
## exit;
}
if (is_method("BYE")) {
setflag(1); # do accounting ...
setflag(3); # ... even if the transaction fails
} else if (is_method("INVITE")) {
# even if in most of the cases is useless, do RR for
# re-INVITEs alos, as some buggy clients do change route set
# during the dialog.
   xlog("Inside else if is_method INVITE loop
LoadBalance\n");
record_route();
}

# route it out to whatever destination was set by loose_route()
# in $du (destination URI).
route(RELAY);
} else {
xlog("Inside else of loose route \n");
if ( is_method("ACK") ) {
if ( t_check_trans() ) {
# non loose-route, but stateful ACK; must be an ACK after

Re: [OpenSIPS-Users] nortpproxy_str usage

2018-01-16 Thread Bogdan-Andrei Iancu


IF you do not want to chain, yes, just ignore that attribute.

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  http://www.opensips-solutions.com
OpenSIPS Summit 2018
  http://www.opensips.org/events/Summit-2018Amsterdam

On 01/15/2018 06:42 PM, William Simon wrote:


Thanks Bogdan-Andrei. If I don't intend to chain multiple RTPProxy, is 
there any reason to keep the nortpproxy_str attribute, or is it safe 
to disable it by setting it to empty string?



*From:* Bogdan-Andrei Iancu 
*Sent:* Monday, January 15, 2018 11:08:41 AM
*To:* OpenSIPS users mailling list; William Simon
*Subject:* Re: [OpenSIPS-Users] nortpproxy_str usage
Hi William,

IF you end up chaining multiple RTPproxy'es, you need to be sure you 
avoid the "learning" deadlock between them - as rtpproxy is normally 
waiting to receive RTP in order to learn the end-points IP+port (as 
the IP coordinates from SDP are private, so not usable). So, if the IP 
in the received SDP in public, use the "r" flag when triggering 
rtpproxy (not to wait, but to trust the IP+port in SDP) - see 
http://www.opensips.org/html/docs/modules/2.3.x/rtpproxy.html#idp5556688


Best regards,
Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   http://www.opensips-solutions.com
OpenSIPS Summit 2018
   http://www.opensips.org/events/Summit-2018Amsterdam
On 01/12/2018 08:48 PM, William Simon wrote:
I have a provider that is reflecting the a=nortpproxy:yes attribute 
back to me in the SDP of its reply. I know it's being reflected back 
because when I customize that header using the nortpproxy_str 
setting, I get my own value back in the provider's replies.


The result is that the answer side of rtpproxy isn't engaged and 
there's no audio.


From an old thread 
(http://lists.opensips.org/pipermail/users/2012-November/023444.html) 
it seems like the best option for me is to simply disable this 
a=nortpproxy:yes attribute and let the opensips logic engage rtpproxy 
all the time.


What kind of negative side effects would I face by disabling this and 
letting rtpproxy engage every time? Is this flag only really useful 
when you are working with a hierarchy of SIP proxies in your own 
environment? My topology has two equal-priority equal-weight opensips 
proxies that serve as clustered load balancer as well as 
outbound-proxy for internal freeswitch servers.


Thanks
W Simon


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Re: [OpenSIPS-Users] Not able to make Load Balancer work

2018-01-16 Thread Bogdan-Andrei Iancu


Hi Mayur,

The simplest way to debug this is by using the script_trace() function - 
see:

http://www.opensips.org/Documentation/Script-CoreFunctions-2-3#toc43

Simply add in the very begging of your script:
script_trace( 1, "$rm from $si, ruri=$ru/$du", "LB");

and make a call - see the execution trace through the script for your INVITE.

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  http://www.opensips-solutions.com
OpenSIPS Summit 2018
  http://www.opensips.org/events/Summit-2018Amsterdam

On 01/15/2018 07:23 PM, Mayur Koshal wrote:

Hi Bogdan,

Thanks for your time and looking into this.

I am using the Load balancer script which is generated from the 
menuconfig as it is. The script was generated after I compiled and 
build opensips . Did not made much changes to it.


Need to find out the reason why it is looping inside LB and not 
forwarding it to Proxy.



Regards
Mayur



On Mon, Jan 15, 2018 at 10:43 PM, Bogdan-Andrei Iancu 
> wrote:


Hi MAyur,

Your trance does not show the INVITE being route outside the LB
(.19). Even more there is a "Too many hops reply" back, suggesting
that a loop is created on the LB and the INVITE is never sent to
the destination (but keep re-routed back to LB).

When you do t_relay() in your script, place before it an xlog to
see the destinations:
xlog("sending out request to $ru/$du\n");

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
http://www.opensips-solutions.com 
OpenSIPS Summit 2018
http://www.opensips.org/events/Summit-2018Amsterdam


On 01/12/2018 07:36 PM, Mayur Koshal wrote:

Hi Bogdan,

Thanks for the Reply.

Yes The opensips load balancer is able to reach destinations
which is
configured in the load balancer table as I can see in the
wireshark
logs and opensips logs of the SIP Proxy server.




I have removed one SIP proxy in the table and the load
distribution
table looks like :

mysql> select * from load_balancer;
| id | group_id | dst_uri  | resources |
probe_mode |
description |

++--+--+---++-+
|  2 |1 | sip:192.168.100.187:5060
 | sip=50|  0 |
serv1   |

++--+--+---++-+


Now I am using the setup with only one sip proxy to make it
simpler as :


UAC(192.168.100.19:5065)...>Load_Balancer(192.168.100.19:5060)..>Proxy(192.168.100.187:5060)..>
  UAS(192.168.187:5070)


I have added a user in Load Balancer:

sip:bharti@192.168.100.187:5070




Database changed
mysql> select * from location;

+-+--++-+--+--+-+--+-+--+-+---++++-+--+
| id  | username | domain | contact  | received
| path | expires | q| callid
| cseq | last_modified   | flags | cflags | user_agent   
 |

socket | methods | sip_instance |

+-+--++-+--+--+-+--+-+--+-+---++++-+--+
| 272 | bharti   || sip:bharti@192.168.100.187:5070
 | NULL
| NULL | 1970-01-01 05:30:00 | 1.00 |
dfjrewr12386fd6-...@opensips.mi
|1 | 1970-01-01 05:30:00 | 0 |  0 | OpenSIPS MI
Server |
NULL   |NULL | NULL |

+-+--++-+--+--+-+--+-+--+-+---++++-+--+
1 row in set (0.01 sec)




I am using commands in sipp:

UAC ---./sipp -sn uac 192.168.100.19:5060
 -s bharti -i 192.168.100.19
-p 5065 -m 10 -r 1 -rp 1000

UAS -- ./sipp -sn uas 192.168.100.19:5060
 -i 192.168.100.187 -p 5070


If I remove the load balancer and execute similar sipp
commands and
run a single

Re: [OpenSIPS-Users] Send LUA logs via HEP to Homer server

Re: [OpenSIPS-Users] Send LUA logs via HEP to Homer server

[OpenSIPS-Users] rtpproxy error messages

[OpenSIPS-Users] Send LUA logs via HEP to Homer server

Re: [OpenSIPS-Users] SIP-I Calling Party number being changed

Re: [OpenSIPS-Users] Not able to make Load Balancer work

Re: [OpenSIPS-Users] nortpproxy_str usage

Re: [OpenSIPS-Users] Not able to make Load Balancer work

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