Re: [OpenSIPS-Users] How to Solve 'Discarding fwd for a 6xx Transaction'?

2020-09-30 Thread Diptesh Patel
Hello,

Thank you Liviu and Ben for your response.

It's working fine for me. I know it is not recommended by RFC :)

Thanks & Regards
*Diptesh Patel*
Software Developer
Ecosmob Technologies Ltd,
Ahmedabad
Mo:*+919898962659*


On Fri, Sep 25, 2020 at 6:56 PM Ben Newlin  wrote:

> https://opensips.org/docs/modules/3.1.x/tm.html#param_disable_6xx_block
>
>
>
> Ben Newlin
>
>
>
> *From: *Users 
> *Date: *Friday, September 25, 2020 at 5:58 AM
> *To: *OpenSIPS users mailling list 
> *Subject: *[OpenSIPS-Users] How to Solve 'Discarding fwd for a 6xx
> Transaction'?
>
> Hello,
>
>
>
> I want to fail-over my call on 6xx response and I use failure route for
> that but t_relay() gives below errors and reply with '500 No destination
> available (18/SL)'.
>
>
>
> Sep 25 05:47:10 OLD-OVW-SBCA01 /usr/local/sbin/opensips[28174]:
> INFO:tm:t_forward_nonack: discarding fwd for a 6xx transaction
> Sep 25 05:47:10 OLD-OVW-SBCA01 /usr/local/sbin/opensips[28174]:
> ERROR:tm:w_t_relay: t_forward_nonack failed
>
>
>
> Anyone have an idea how I can resolve this?
>
>
>
>
>
> Thanks & Regards
>
> *Diptesh Patel*
>
> Software Developer
>
> Ecosmob Technologies Ltd,
>
> Ahmedabad
>
> Mo:*+919898962659*
>
>
>
> *Disclaimer*
>
> In addition to generic Disclaimer which you have agreed on our website,
> any views or opinions presented in this email are solely those of the
> originator and do not necessarily represent those of the Company or its
> sister concerns. Any liability (in negligence, contract or otherwise)
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[OpenSIPS-Users] OpenSIPs Crash on reporting delay.

2020-09-30 Thread Diptesh Patel
Hello,

My OpenSIPS crashed once in heavy load. I don't know the exact reason and
it does not usually happen. can you please suggest what is wrong with it?

Please find gdb backtrace in attachment. Let me know If you need anything
else.

version: opensips 2.4.1 (x86_64/linux)
flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC,
FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535
poll method support: poll, epoll, sigio_rt, select.
git revision: 5d042cffc
main.c compiled on 02:09:27 Feb 24 2020 with gcc 4.8.5

Thanks & Regards
*Diptesh Patel*
Software Developer
Ecosmob Technologies Ltd,
Ahmedabad
Mo:*+919898962659*

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This communication, including any 
attachments, may not be free of viruses, trojans, similar or new 
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GNU gdb (GDB) Red Hat Enterprise Linux 7.6.1-115.el7
Copyright (C) 2013 Free Software Foundation, Inc.
License GPLv3+: GNU GPL version 3 or later <http://gnu.org/licenses/gpl.html>
This is free software: you are free to change and redistribute it.
There is NO WARRANTY, to the extent permitted by law.  Type "show copying"
and "show warranty" for details.
This GDB was configured as "x86_64-redhat-linux-gnu".
For bug reporting instructions, please see:
<http://www.gnu.org/software/gdb/bugs/>...
Reading symbols from /usr/local/sbin/opensips...done.
[New LWP 13167]
[Thread debugging using libthread_db enabled]
Using host libthread_db library "/lib64/libthread_db.so.1".
Core was generated by `/usr/local/sbin/opensips -P 
/var/run/opensips/opensips.pid -f /usr/local/etc/op'.
Program terminated with signal 11, Segmentation fault.
#0  0x00462bbf in log_expiry (func_info=0x60fc5a "msg processing", 
tcp=0, dbg_len=1056, 
extra_dbg=0x89e5a0  "INVITE 
sip:6042352637@216.123.235.115:5060;user=phone SIP/2.0\r\nVia: SIP/2.0/UDP 
209.91.94.6:5060;branch=z9hG4bKketago30306suciqkm90.1\r\nFrom: \"PRABHJOT 
KAUR\", 
time_diff=60635)
at ut.h:1085
1085LM_WARN("#%i is a module action 
: %s - %dus - line %d\n",i+1,
Missing separate debuginfos, use: debuginfo-install 
MariaDB-shared-10.1.43-1.el7.centos.x86_64 glibc-2.17-292.el7.x86_64 
gmp-6.0.0-15.el7.x86_64 gnutls-3.3.29-9.el7_6.x86_64 
keyutils-libs-1.5.8-3.el7.x86_64 krb5-libs-1.15.1-37.el7_7.2.x86_64 
libcom_err-1.42.9-16.el7.x86_64 libffi-3.0.13-18.el7.x86_64 
libgcc-4.8.5-39.el7.x86_64 libgcrypt-1.5.3-14.el7.x86_64 
libgpg-error-1.12-3.el7.x86_64 libmicrohttpd-0.9.33-2.el7.x86_64 
librabbitmq-0.8.0-2.el7.x86_64 libselinux-2.5-14.1.el7.x86_64 
libstdc++-4.8.5-39.el7.x86_64 libtasn1-4.10-1.el7.x86_64 
libxml2-2.9.1-6.el7_2.3.x86_64 nettle-2.7.1-8.el7.x86_64 
openssl-libs-1.0.2k-19.el7.x86_64 p11-kit-0.23.5-3.el7.x86_64 
pcre-8.32-17.el7.x86_64 xz-libs-5.2.2-1.el7.x86_64 zlib-1.2.7-18.el7.x86_64
(gdb) bt full
#0  0x00462bbf in log_expiry (func_info=0x60fc5a "msg processing", 
tcp=0, dbg_len=1056, 
extra_dbg=0x89e5a0  "INVITE 
sip:6042352637@216.123.235.115:5060;user=phone SIP/2.0\r\nVia: SIP/2.0/UDP 
209.91.94.6:5060;branch=z9hG4bKketago30306suciqkm90.1\r\nFrom: \"PRABHJOT 
KAUR\", 
time_diff=60635)
at ut.h:1085
param = {s = 0x5f639328 , len = 
100387}
list = 
i = 0
func_str = {s = 0x60fca1 "source", len = 6}
time_str = {s = 0x63935d "time", len = 4}
extra_str = {s = 0x61a8a6 "extra", len = 5}
#1  receive_msg (
buf=0x89e5a0  "INVITE 
sip:6042352637@216.123.235.115:5060;user=phone SIP/2.0\r\nVi

[OpenSIPS-Users] How to Solve 'Discarding fwd for a 6xx Transaction'?

2020-09-25 Thread Diptesh Patel
Hello,

I want to fail-over my call on 6xx response and I use failure route for
that but t_relay() gives below errors and reply with '500 No destination
available (18/SL)'.

Sep 25 05:47:10 OLD-OVW-SBCA01 /usr/local/sbin/opensips[28174]:
INFO:tm:t_forward_nonack: discarding fwd for a 6xx transaction
Sep 25 05:47:10 OLD-OVW-SBCA01 /usr/local/sbin/opensips[28174]:
ERROR:tm:w_t_relay: t_forward_nonack failed

Anyone have an idea how I can resolve this?


Thanks & Regards
*Diptesh Patel*
Software Developer
Ecosmob Technologies Ltd,
Ahmedabad
Mo:*+919898962659*

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Re: [OpenSIPS-Users] siprec module does response re-invite from a siprec server.

2020-07-10 Thread Diptesh Patel
Hello Razvan,

Thanks for your prompt response.

I opened a feature request.

https://github.com/OpenSIPS/opensips/issues/2167

Thanks & Regards
*Diptesh Patel*
Software Developer
Ecosmob Technologies Ltd,
Ahmedabad
Mo:*+919898962659*


On Fri, Jul 10, 2020 at 9:05 PM Răzvan Crainea  wrote:

> Hi, Diptesh!
>
> Unfortunately reinvites from the SIPREC servers are not supported by the
> module right now. Please open a feature request for this.
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solutions.com
>
> On 7/10/20 4:28 PM, Diptesh Patel wrote:
> > Hello,
> >
> > I am using opensips 2.4.7 and opensips is working as a siprec client
> > with rtpproxy. It is working fine but it does not respond to any
> > re-invite from siprec server as a refresher after a few seconds and the
> > siprec server rejects the call with BYE due to timeout and recording
> > stops there. Is there any configuration required for that?
> >
> > Thanks & Regards
> > *Diptesh Patel*
> > Software Developer
> > Ecosmob Technologies Ltd,
> > Ahmedabad
> > Mo:*+919898962659*
> >
> > *Disclaimer*
> > In addition to generic Disclaimer which you have agreed on our website,
> > any views or opinions presented in this email are solely those of the
> > originator and do not necessarily represent those of the Company or its
> > sister concerns. Any liability (in negligence, contract or otherwise)
> > arising from any third party taking any action, or refraining from
> > taking any action on the basis of any of the information contained in
> > this email is hereby excluded.
> >
> > *Confidentiality*
> > This communication (including any attachment/s) is intended only for the
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> > CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or
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> > you have received it in error.
> >
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> > viruses, trojans, similar or new contaminants/malware, interceptions or
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> > including its sister concerns shall not be liable for any damage that
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[OpenSIPS-Users] siprec module does response re-invite from a siprec server.

2020-07-10 Thread Diptesh Patel
Hello,

I am using opensips 2.4.7 and opensips is working as a siprec client with
rtpproxy. It is working fine but it does not respond to any re-invite from
siprec server as a refresher after a few seconds and the siprec server
rejects the call with BYE due to timeout and recording stops there. Is
there any configuration required for that?

Thanks & Regards
*Diptesh Patel*
Software Developer
Ecosmob Technologies Ltd,
Ahmedabad
Mo:*+919898962659*

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Re: [OpenSIPS-Users] changing callerids

2020-07-09 Thread Diptesh Patel
Hello Vic,

Can you tell me from which route are you trying this function? It only
allows on request route, branch route and failure route. I think you
misunderstood how to use the exported function, let me guide you. It will
not update only the user part of the 'from' uri but works on uri. So you
need to create a custom string like below to fulfill your requirements and
the first argument is for display part not for uri part.

$avp(fromuri) = "sip:" + *$avp(650)* + "@" + $fd;

It should be called only once at initial requests and the
subsequent requests(if you loaded the dialog module and created module or
used forced_dialog in the uri module) and responses(using tm module) will
be automatically updated your 'from uri'.

Refer the below example which helps you to guide how you can use this
function in different use cases.

*$avp(display) = "Jolin";*
*$avp(fromuri) = "sip:+18736352...@myopensipsdomain.com
"*

*REPLACE BOTH display name and uri*
*uac_replace_from("$avp(display)","$avp(fromuri)");*
*e.g. received From: "Vic" >;tag=ca4a1348*
*   sent   From: "Jolin" >;tag=ca4a1348*

REPLACE ONLY display name
*uac_replace_from("$avp(display)","");*
*e.g. received From: "Vic" >;tag=ca4a1348*
*   sent   From: "Jolin" ;tag=ca4a1348*

*REPLACE ONLY uri WITHOUT TOUCHING display name*
*uac_replace_from( ,"$avp(fromuri)");*
*e.g. received From: "Vic" >;tag=ca4a1348*
*   sent   From: "Vic" >;tag=ca4a1348*

REMOVE display and REPLACE uri
*uac_replace_from("","$avp(fromuri)");*
*e.g. received From: "Vic" >;tag=ca4a1348*
*   sent   From: >;tag=ca4a1348*


For more information you can read the uri module documentation here.
<https://opensips.org/html/docs/modules/devel/uac.html#func_uac_replace_from>

*NOTE:* If you're using *3.X.X* version then avoid double quotes for avp
variables.
*e.g. **uac_replace_from($avp(display),$avp(fromuri));*


Thanks & Regards
*Diptesh Patel*
Software Developer
Ecosmob Technologies Ltd,
Ahmedabad
Mo:*+919898962659*


On Fri, Jul 10, 2020 at 12:29 AM Vic Jolin  wrote:

> Hi,
>
> Are there other ways of "changing" callerids for a certain call aside from
> uac_replace_from?
>
> And what is the correct way of doing this?
>
> I have uac_replace_from("$avp(650)", "");
>
> I find it giving errors when I try to update the uri with
>
> sip:$avp(650)@ipadddress:poprt
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Re: [OpenSIPS-Users] adding and reading a SIP header to a 302 Redirect

2020-06-11 Thread Diptesh Patel
Hello,

The next destination(s) should be decided based on Contact Header(s) of 3xx
Responses. You can configure one or more destinations based on priority and
fail-over enabled to multiple destinations. But in your case I found,
Contact Header is malformed i.e. *Contact: . *

Thanks & Regards
*Diptesh Patel*
Software Developer
Ecosmob Technologies Ltd,
Ahmedabad
Mo:*+919898962659*


On Sat, May 30, 2020 at 5:47 PM Saint Michael  wrote:

> I need to add a new SIP header to the response below.
>
>
> if ($rm=="INVITE") {
> /* add the redirect destinations as branches */
> $branch = "sip:bat...@gotham.com";
> $branch = $avp(my_custom_uri);
> /* sending a 3xx reply will automatically push all
>  * existing branches as Contact URIs */
> send_reply("302","Moved Temporarily");
> exit;
> }
> the final result should be:
>
> SIP/2.0 302 Moved Temporarily
> Via: SIP/2.0/UDP 172.16.7.254:52169
> ;rport=52169;received=47.205.172.89;branch=z9hG4bK-524287-1---129f4244aaba9f04
> Call-ID: 102650Mzg4NmFiNTQzOGY5NDJmNjM3OTYzNmE5MzNlZDIwZmI
> From: "X" ;tag=81a25c36
> To:  .124.224.87>;tag=9e198dc4-7ce8-433d-ae23-05b9bc14d55a
>
> Identity:eyJhbGciOiJFUzI1NiIsInR5cCI6InBhc3Nwb3J0IiwicHB0Ijoic2hha2VuIiwieDV1IjoiaHR0cHM6Ly9jZXJ0LmV4YW1wbGUub3JnL3Bhc3Nwb3J0LmNlciJ9.eyJhdHRlc3QiOiJBIiwiZGVzdCI6eyJ0biI6WyI2MzE3OTE4Mzc4Il19LCJpYXQiOjE1OTA4MTEyMzgsIm9yaWciOnsidG4iOiI3Mjc0NDMzMDE5In0sIm9yaWdpZCI6IjEyM2U0NTY3LWU4OWItMTJkMy1hNDU2LTQyNjY1NTQ0MDAwMCJ9.AKViDWA3uonP6tt5cKBh0FUPY5zBuJnwZLQNTrp9LCWJ-vLY1Xx5i3_oXGh1ERL4tnD-KK5wsP3FdByDa_cjGw;info=<
> https://cert.example.org/passport.cer>;alg=ES256;ppt=shaken
> CSeq: 1 INVITE
> Server: Asterisk PBX 16.10.0
> Contact: 
> Reason: Q.850;cause=0
> Content-Length:  0
>
> also if I get that packet at my end, how do I extract the Identity header
> and apply it to the next INVITE?
> Is this even doable?
>
>
>
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[OpenSIPS-Users] SIPREC server does not send BYE while capturing Early-Media.

2020-06-09 Thread Diptesh Patel
Hello,

I am using the SIPREC module to send the request to SRS with fail-over.
While recording early media and after Originator Cancel OpenSIPS does not
send a BYE request to the SRS.  Please look at the script snippet of how I
implemented that.

if (is_method ("INVITE"))
{
  if (siprec_start_recording ("sip:XX.XX.XXX.X00:5060"))
{
 xlog ("L_INFO", "SIP RECORDING THROUGH XX.XX.XXX.X00");
}
  else if (siprec_start_recording ("sip:XX.XX.XXX.X04:5060"))
{
 xlog ("L_INFO", "SIP RECORDING THROUGH XX.XX.XXX.X04");
}
  else
sl_send_reply ("404", "SIPREC Server NOT FOUND ");
}
}

local_route {
if(is_method("INVITE")) {
xlog("L_INFO","LOCAL ROUTE");

subst('/^Contact:(.*)(.*)$/Contact:\1;+sip.src\3/ig');
}
}

Thanks & Regards
*Diptesh Patel*
Software Developer
Ecosmob Technologies Ltd,
Ahmedabad
Mo:*+919898962659*

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Re: [OpenSIPS-Users] 502 Bad Gateway events leads to calls being rejected with 480 Temporarily Unavailable

2020-06-08 Thread Diptesh Patel
Hello Solarmon,

I think, The * ds_mark_dst("p");* put your destinations on Probing and
after a few seconds you will get the reply for OPTIONS and now your
destinations are Active. Are you making a second call immediately? If yes
then it is clear. Please remove the ds_mark_dst("p"), OpenSIPS
automatically change the destination's state using OPTIONS.

Thanks & Regards
*Diptesh Patel*
Software Developer
Ecosmob Technologies Ltd,
Ahmedabad
Mo:*+919898962659*


On Mon, Jun 8, 2020 at 4:27 PM solarmon  wrote:

> Hi Diptesh ,
>
> Thanks for your reply.
>
> Apologies, I'm using the term 'blacklist' to generally mean that the
> endpoints are not available.
>
> Also, the 502 Bad Gateway is response to an INVITE, not SIP OPTIONS,
> returned by the far end and the ITSP is just passing that back to us,
> because the call has failed. For such call failures, I'm not expecting for
> the dispatcher endpoints to be marked as unavailable for routing.
>
> I am not using, or have not set, the '*ds_define_blacklist (str)*' option
> in my dispatcher module config.
>
> My probing mode is:
>
> modparam("dispatcher", "ds_probing_mode", 1)
>
> I'm not seeing anything in the logs regarding the dispatcher nodes going
> into Probing mode - should there be logs for that, or can it be enabled to
> be logged?
>
> When I check the endpoints with ' opensipsctl dispatcher dump' they always
> seem to be '*Active*' - so it is either they are like that, or they
> may have only been in '*Probing*' mode very briefly. Again, I was hoping
> to see mode/state change in the historical logs.
>
> In my *opensips.cfg* (which was created for me) I can see the following
> code, which looks like this is where it is introducing this behaviour in
> question:
>
> failure_route[call_failover]
> {
> xlog("[$ci] call failed to established with $T_reply_code code\n");
>
> rtpproxy_unforce("$avp(rtpp_set)");
>
> if (t_was_cancelled()) {
> t_reply("487","Request cancelled");
> exit;
> }
>
> # any failure indication ?
> if ( t_check_status("[56][0-9][0-9]")
> || (t_check_status("408") && t_local_replied("all"))
> ) {
> xlog("[$ci] destination $rd failed  with $T_reply_code ->
> retry\n ");
>
>* ds_mark_dst("p");*
>
> if ( ds_next_domain() ) {
> xlog("[$ci] using new destination <$rd>\n ");
>
> # send it out again
> t_on_failure("call_failover");
> t_relay();
> exit;
> } else {
> xlog("[$ci] no other destination to retry\n ");
>         t_reply("503","Service not available");
> exit;
> }
> }
>
> # if call failure, allow the reply to propagate to caller
> exit;
> }
>
>
>
> Thank you for the tip about the  'modparam("dispatcher",
> "options_reply_codes", "502")' option. I will try that if it is not
> recommend to change the above code.
>
> Thank you.
>
>
> On Mon, 8 Jun 2020 at 10:47, Diptesh Patel 
> wrote:
>
>> Hello Solarmon,
>>
>> Need some clarification on term blacklisting, Are you using the blacklist
>> for Probing Mode of destination? or Are you using the *'ds_define_blacklist
>> (str)' *parameter. If you are not using the blacklist parameter then
>> below information help you. It is great if you share your script snippet
>> and output of *'opensipsctl dispatcher dump'* which shows you the
>> current status of your destinations.
>>
>> If you are getting success(200 OK) response on OPTIONS then it is not
>> possible that you got a negative response from a destination and it will
>> not be blacklisted(probing mode) in dispatcher until you blacklist(probing
>> mode) from the script using 'ds_mark_dst()' exported function. I doubt that
>> you are also getting '502 Bad Gateway' on OPTIONS which is sending to the
>> destination to check the availability.  If It is right and you want to add
>> the 502 response as a good response for OPTIONS. you can add the 502 as 
>> *'modparam("dispatcher",
>> "options_reply_codes", "502")'.*
>>
>> Thanks &am

Re: [OpenSIPS-Users] 502 Bad Gateway events leads to calls being rejected with 480 Temporarily Unavailable

2020-06-08 Thread Diptesh Patel
Hello Solarmon,

Need some clarification on term blacklisting, Are you using the blacklist
for Probing Mode of destination? or Are you using the *'ds_define_blacklist
(str)' *parameter. If you are not using the blacklist parameter then below
information help you. It is great if you share your script snippet and
output of *'opensipsctl dispatcher dump'* which shows you the current
status of your destinations.

If you are getting success(200 OK) response on OPTIONS then it is not
possible that you got a negative response from a destination and it will
not be blacklisted(probing mode) in dispatcher until you blacklist(probing
mode) from the script using 'ds_mark_dst()' exported function. I doubt that
you are also getting '502 Bad Gateway' on OPTIONS which is sending to the
destination to check the availability.  If It is right and you want to add
the 502 response as a good response for OPTIONS. you can add the 502
as *'modparam("dispatcher",
"options_reply_codes", "502")'.*

Thanks & Regards
*Diptesh Patel*
Software Developer
Ecosmob Technologies Ltd,
Ahmedabad
Mo:*+919898962659*


On Mon, Jun 8, 2020 at 2:24 PM solarmon  wrote:

> Hi,
>
> I'm trying to understand whether this is the correct or expected behaviour.
>
> We have two destinations configured in Dispatcher.
>
> What I am noticing is that when we receive 502 Bad Gateway messages
> (logged as ("call failed to established with 502 code") from both
> endpoints. After both endpoints have returned 502 Bad Gateway, opensips
> pass back 503 Service Unavailable back to the originating endpoint of the
> call. However, subsequent calls are being immediately rejected with 480
> Temporarily Unavailable (logged as "failed to find an available
> destination, rejecting") for a period of time.
>
> It seems that opensips is blacklisting the Dispatcher endpoints because of
> receiving the 502 Bad Gateway messages. Is this the correct/expected
> behaviour? I would have thought the blacklisting should be based on the SIP
> OPTIONS sent to the Dispatcher endpoints.
>
> I do not currently see any issues with SIP OPTIONS to these endpoints so
> I'm confused as to why they are seemingly blacklisted.
>
> If this is the correct/expected behaviour, can it be changed to only
> blacklist based on the SIP OPTIONs pings?
>
> Thank you.
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>

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sister concerns. Any liability (in negligence, contract or otherwise) 
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Re: [OpenSIPS-Users] ACK Routing Issue

2020-06-05 Thread Diptesh Patel
Hello Mark,

You need to modify your script. Let me explain to you how topology
hiding works. If you're using topology hiding then we are hiding the
information from the other node right? So the topology hiding stores your
next hop information. When you call topology_hiding_match() then it will
replace the request uri with the next hop. So you need to use
topology_hiding_match() on every within dialog request(sequential
requests), otherwise it will create looping.

you can check the following snippet which i modified. It helps you !!

if (has_totag()) {
xlog("CUSTOM_LOG: in-dialog $rm has message flags: $mf and branch
flags: $bf");
#Set correct SIP User-Agent Header
if (remove_hf("User-Agent")) {
xlog("CUSTOM_LOG: Setting SIP User-Agent on In-Dialog Request");
insert_hf("User-Agent: OpenSIPS\r\n");
}
# handle hop-by-hop ACK (no routing required)
if ( is_method("ACK") && t_check_trans() ) {
xlog("CUSTOM_LOG: ACK detected with valid transaction - t_relay");
t_relay();
exit;
}
if (topology_hiding_match()) {
  ##TH match and relay all the within dialog packets.
t_relay();
exit;
} else {
xlog("CUSTOM_LOG: cannot match request to a dialog \n");
    send_reply(404,"Not found");
exit;
}
}


Thanks & Regards
*Diptesh Patel*
Software Developer
Ecosmob Technologies Ltd,
Ahmedabad
Mo:*+919898962659*


On Fri, Jun 5, 2020 at 6:37 PM Mark Farmer  wrote:

> Thanks Diptesh
>
> I'm using topology hiding, so now I have this:
>
> if (has_totag()) {
> xlog("CUSTOM_LOG: in-dialog $rm has message flags: $mf and
> branch flags: $bf");
>
> #Set correct SIP User-Agent Header
> if (remove_hf("User-Agent")) {
> xlog("CUSTOM_LOG: Setting SIP User-Agent on In-Dialog
> Request");
> insert_hf("User-Agent: OpenSIPS\r\n");
> }
>
> if (!topology_hiding_match() ) {
> xlog("CUSTOM_LOG: cannot match request to a dialog
> \n");
> send_reply(404,"Not found");
> }
>
> # handle hop-by-hop ACK (no routing required)
> if ( is_method("ACK") && t_check_trans() ) {
> xlog("CUSTOM_LOG: ACK detected with valid
> transaction - t_relay");
> t_relay();
> exit;
> }
>
> I don't see a 404 going out so I think topology_hiding_match is working.
> But it tries to send the ACK to itself on it's private interface (I have
> mhomed=1).
>
> ACK sip:+44XX@10.150.50.72;did=e07.595f3776 SIP/2.0
> Via: SIP/2.0/UDP PUB.LIC.IP.ADDR:5060;branch=z9hG4bKc219.d1f5b08.2
> From: ;tag=gK0c801c8d
> To: ;tag=3800350621-1224267434
> Call-ID: 543691539-3800350621-1514620...@sbc-uk-bs13b.uk.sdin.bt.net
> CSeq: 202841 ACK
> Max-Forwards: 69
> Content-Length: 0
>
> Best regards
> Mark.
>
>
> On Fri, 5 Jun 2020 at 13:03, Diptesh Patel 
> wrote:
>
>> Hello Mark,
>>
>> Are you using Topology Hiding or Loose Routing?
>>
>> If you are using Topology Hiding then you need to match the topology
>> using *topology_hiding_match()* first.
>>
>> It is great if you can share SIP packets.
>>
>> Thanks & Regards
>> *Diptesh Patel*
>> Software Developer
>> Ecosmob Technologies Ltd,
>> Ahmedabad
>> Mo:*+919898962659*
>>
>>
>> On Fri, Jun 5, 2020 at 5:00 PM Mark Farmer  wrote:
>>
>>> Hi everyone
>>>
>>> I've upgraded an OpenSIPS box to 3.1 and am now seeing an issue with
>>> ACK's trying to route to an incorrect IP - in this case our own advertised
>>> IP.
>>>
>>> I think I'm right in saying that PRACK's & ACK's are treated equally and
>>> should route in the same manner? However, PRACK's are routing correctly.
>>>
>>> I have this:
>>>
>>> if (has_totag()) {
>>> ---
>>>
>>> # handle hop-by-hop ACK (no routing required)
>>> #if ( is_method("ACK") && t_check_trans() ) {
>>> if (is_method("ACK")) {
>>> t_relay();
>>> exit;
>>> }
>>> ---
>>>
>>> Thanks for any ideas!
>>> Mark.
>>>
>>>
>>> _

Re: [OpenSIPS-Users] ACK Routing Issue

2020-06-05 Thread Diptesh Patel
Hello Mark,

Are you using Topology Hiding or Loose Routing?

If you are using Topology Hiding then you need to match the topology using
*topology_hiding_match()* first.

It is great if you can share SIP packets.

Thanks & Regards
*Diptesh Patel*
Software Developer
Ecosmob Technologies Ltd,
Ahmedabad
Mo:*+919898962659*


On Fri, Jun 5, 2020 at 5:00 PM Mark Farmer  wrote:

> Hi everyone
>
> I've upgraded an OpenSIPS box to 3.1 and am now seeing an issue with ACK's
> trying to route to an incorrect IP - in this case our own advertised IP.
>
> I think I'm right in saying that PRACK's & ACK's are treated equally and
> should route in the same manner? However, PRACK's are routing correctly.
>
> I have this:
>
> if (has_totag()) {
> ---
>
> # handle hop-by-hop ACK (no routing required)
> #if ( is_method("ACK") && t_check_trans() ) {
> if (is_method("ACK")) {
> t_relay();
> exit;
> }
> ---
>
> Thanks for any ideas!
> Mark.
>
>
> ___
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> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>

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*Disclaimer*
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the originator and do not necessarily represent those of the Company or its 
sister concerns. Any liability (in negligence, contract or otherwise) 
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