Re: [OpenSIPS-Users] mid_registrar

[volga629]

Hello Liviu,
Is contact header should be replaced with insert mode 1 ?
Configuration like this


 REGISTRAR module
loadmodule "mid_registrar.so"
modparam("mid_registrar", "mode", 1)
modparam("mid_registrar", "insertion_mode", 0)
modparam("mid_registrar", "received_avp", "$avp(rcv)")
modparam("mid_registrar", "max_contacts", 4)
modparam("mid_registrar", "tcp_persistent_flag", 10)
modparam("mid_registrar", "outgoing_expires", 900)
modparam("mid_registrar", "contact_match_param", "regid")

Because I see Contact get rewritten complete with opensips local ip and 
and port. Only parameters like transport are preserved which break rest 
routing.


=
Call-ID: 141351_rel51MTBiMTgzN2NmYjBhZmFkMzAzNTNjNDI4Yzk3YThmYmU
User: 4...@sip.company.tld
Contact: "volga629" 


Agent: Bria Android 3.9.2 build 96033
Status: Registered(TCP)(unknown) EXP(2017-04-26 22:31:32) EXPSECS(592)
Ping-Status:Reachable
Ping-Time:  0.00
Host: prod.com
IP: 10.18.130.27
Port: 5060
Auth-User: unknown
Auth-Realm: sip.company.tld
MWI-Account:4...@sip.company.tld



volga629


On Thu, 6 Apr, 2017 at 10:56 AM, volga...@networklab.ca wrote:

Hello Liviu,

Are planning open github issue about it ?

volga629

On Tue, 4 Apr, 2017 at 12:32 PM, volga...@networklab.ca wrote:

Hello Liviu,
Here are mod params

 REGISTRAR module
loadmodule "mid_registrar.so"
modparam("mid_registrar", "mode", 1)
modparam("mid_registrar", "insertion_mode", 0)
modparam("mid_registrar", "received_avp", "$avp(rcv)")
modparam("mid_registrar", "max_contacts", 4)
modparam("mid_registrar", "tcp_persistent_flag", 10)
modparam("mid_registrar", "contact_match_param", "regid")

volga629

On Tue, 4 Apr, 2017 at 12:25 PM, Liviu Chircu  
wrote:
Speaking of help, could you also dump all your mid_registrar 
modparams? This should speed up debugging a bit.

Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com

OpenSIPS Summit May 2017 Amsterdam
  http://www.opensips.org/events/Summit-2017Amsterdam.html
On 04.04.2017 18:22, volga...@networklab.ca wrote:

Hello Liviu,
Thank you for all help.

On Tue, 4 Apr, 2017 at 12:05 PM, Liviu Chircu  
wrote:
Ok, so it looks like there is a bug with Contact expirations, 
with some chunk of memory being freed twice. I will reply as soon 
as I have more info.

Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com

OpenSIPS Summit May 2017 Amsterdam
  http://www.opensips.org/events/Summit-2017Amsterdam.html
On 04.04.2017 17:40, volga...@networklab.ca wrote:

Hello Liviu,

I modified script and I see regid, but module crash again

Apr 4 10:35:57 casbc00 audit: ANOM_ABEND auid=4294967295 uid=992 
gid=992 ses=4294967295 pid=3154 comm="opensips" 
exe="/usr/sbin/opensips" sig=11
Apr 4 10:35:57 casbc00 kernel: opensips[3154]: segfault at 98 ip 
556b953bcdb0 sp 7ffe5dc47e40 error 4 in 
opensips[556b9538b000+219000]
Apr 4 10:35:57 casbc00 abrt-hook-ccpp: Process 3154 (opensips) 
of user 992 killed by SIGSEGV - dumping core
Apr 4 10:35:57 casbc00 audit: ANOM_ABEND auid=4294967295 uid=992 
gid=992 ses=4294967295 pid=3149 comm="opensips" 
exe="/usr/sbin/opensips" sig=11
Apr 4 10:35:57 casbc00 kernel: opensips[3149]: segfault at 98 ip 
556b953bcdb0 sp 7ffe5dc47e70 error 4 in 
opensips[556b9538b000+219000]
Apr 4 10:35:57 casbc00 abrt-hook-ccpp: Process 3149 (opensips) 
of user 992 killed by SIGSEGV - ignoring (repeated crash)
Apr 4 10:35:57 casbc00 systemd: opensips.service: Main process 
exited, code=dumped, status=11/SEGV



https://paste.fedoraproject.org/paste/GGfquSgHeN0ezjXHPqQzk15M1UNdIGYhyRLivL9gydE=

volga629


On Tue, 4 Apr, 2017 at 10:07 AM, Liviu Chircu 
 wrote:
Are you sure mid_registrar_save() is called at all? Please 
either provide some DBG logs of this scenario, or relevant part 
of the script.


Regards,
Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com

OpenSIPS Summit May 2017 Amsterdam
  http://www.opensips.org/events/Summit-2017Amsterdam.html
On 04.04.2017 15:37, volga...@networklab.ca wrote:

Hello Liviu,
Here are full trace of RIGISTRAR, but I don't see not in 
request not in reply regid.


https://paste.fedoraproject.org/paste/o2EcKdTbJcXTY9oX6lFfRV5M1UNdIGYhyRLivL9gydE=

Thank you.

volga629

On Tue, 4 Apr, 2017 at 5:13 AM, Liviu Chircu 
 wrote:

Hi, Volga!

The errors are pretty straightforward, and suggest that the 
"regid" Contact header field parameter has been stripped when 
the 200 OK reply contact set was constructed. To confirm or 
infirm this, however, we need a full SIP packet trace.


Best regards,
Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com

OpenSIPS Summit May 2017 Amsterdam
  http://www.opensips.org/events/Summit-2017Amsterdam.html
On 04.04.2017 04:24, volga...@networklab.ca wrote:

Hello Everyone,
Trying implement mid_registrar but module throwing error.


Apr  3 20:55:47 casbc00 /usr/sbin/opensips[1770]: 
ERROR:mid_registrar:fix_rpl_contact

Re: [OpenSIPS-Users] SIP SDP rfc1918 address fix

[Satish Patel]

Yes, whenever fix_nated_sdp() fiction run it produce that error which I 
mentioned in my previous email. Every single time. 

Sent from my iPhone

> On Apr 26, 2017, at 4:52 PM, Bogdan-Andrei Iancu  wrote:
> 
> So below is the SDP OpenSIPS receives (from network) and when doing 
> fix_nated_sdp() on that SDP leads to the "c=" errors ?
> 
> Regards,
> 
> Bogdan-Andrei Iancu
>  OpenSIPS Founder and Developer
>  http://www.opensips-solutions.com
> 
> OpenSIPS Summit May 2017 Amsterdam
>  http://www.opensips.org/events/Summit-2017Amsterdam.html
> 
>> On 04/26/2017 08:44 PM, Satish Patel wrote:
>> Here is my payload again we have custom application which is using SER
>> so some of them are custom values, This is the payload after i apply
>> fix_nated_sdp() function.
>> 
>> 
>> Max-Forwards: 16.
>> Content-Type: application/sdp.
>> Content-Length: 418.
>> Supported: path, 100rel.
>> P-hint: LOCAL.
>> P-hint: ALIASED OUTBOUND.
>> P-hint: DIRECT-RTP.
>> .
>> v=0.
>> o=user1 53655765 2353687637 IN IP4 192.168.1.8.
>> s=-.
>> c=IN IP4 173.71.121.4.
>> t=0 0.
>> m=audio 6000 RTP/AVP 0.
>> a=rtpmap:127 VANI/32000.
>> a=fmtp:127 ver=3;mode=3;sub-types=1,7;codecs=0x26.
>> a=rtpmap:111 SIREN14-3D/32000.
>> a=fmtp:111 bitrate=32000.
>> a=vx_payload_hdr_ver:2.
>> a=rtpmap:0 PCMU/8000.
>> a=vx_join_audio:1.
>> a=vx_join_text:0.
>> a=vx_jc:60.
>> a=setup:both.
>> a=vx_rtcp:0.
>> a=direction:active.
>> a=oldmediaip:192.168.1.8.
>> 
>> On Wed, Apr 26, 2017 at 6:18 AM, Bogdan-Andrei Iancu
>>  wrote:
>>> Hi Satish,
>>> 
>>> For the mime test, you can use the has_body() function:
>>> http://www.opensips.org/html/docs/modules/2.2.x/sipmsgops.html#idp3886992
>>> 
>>> About the error - could you post the actual SDP payload generating those
>>> errors ?
>>> 
>>> Regards,
>>> 
>>> Bogdan-Andrei Iancu
>>>   OpenSIPS Founder and Developer
>>>   http://www.opensips-solutions.com
>>> 
>>> OpenSIPS Summit May 2017 Amsterdam
>>>   http://www.opensips.org/events/Summit-2017Amsterdam.html
>>> 
>>> 
 On 04/25/2017 10:35 PM, Satish Patel wrote:
 We have some custome Voice solution and in-house media server so right
 now i don't care about PORT all i need correct IP address.
 
 I have tried following and it fixed issue but i am seeing following
 error in logs
 
 if (method=="INVITE") {
  if(search("^Content-Type:.*application/sdp")) {
  fix_nated_sdp("3");
  };
 };
 
 
 Error:
 
 ERROR: extract_mediaip: no `c=' in SDP
 ERROR: extract_mediaip: no `c=' in SDP
 
 Do you know what does that means and how to fix that issue?
 
 On Mon, Apr 24, 2017 at 11:41 PM, Alex Balashov
  wrote:
> The intent of my questions was to get what you think about what you
> actually want to accomplish. fix_nated_sdp() allows you to replace the
> IP with the received signalling IP:
> 
> http://www.opensips.org/html/docs/modules/2.1.x/nathelper.html#id293899
> 
> But what about the port?
> 
>> On Mon, Apr 24, 2017 at 11:39:14PM -0400, Satish Patel wrote:
>> 
>> after google found bunch of post where people suggesting use
>> fix_nated_sdp()  is that right approach ?
>> 
>> On Mon, Apr 24, 2017 at 11:25 PM, Alex Balashov
>>  wrote:
>>> Yes, but RTP can come from a different address than the signalling
>>> (SIP). What sense would there be in substituting the source of the SIP
>>> message in there?
>>> 
 On Mon, Apr 24, 2017 at 11:23:30PM -0400, Satish Patel wrote:
 
 I meant "origin public address of client"  if c line isn't public then
 media never work.
 
 c=IN IP4 192.168.1.8.
 
 It should be
 
 c=IN IP4 
 
 On Mon, Apr 24, 2017 at 11:04 PM, Alex Balashov
  wrote:
> Satish,
> 
> When you say "origin public address", do you mean the external source
> address and port of the SIP message, or the incoming RTP stream?
> 
>> On Mon, Apr 24, 2017 at 11:00:40PM -0400, Satish Patel wrote:
>> 
>> In my INVITE/SDP i am seeing sometime rfc1918 address which i want
>> fix
>> and replace it with origin public address. ex
>> 
>> I am seeing following info in INVITE
>> 
>> v=0.
>> o=amsip 0 0 IN IP4 192.168.1.8.
>> s= .
>> c=IN IP4 192.168.1.8.
>> t=0 0.
>> m=audio 22530 RTP/AVP 127 111 0 101.
>> 
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> --
> Alex Balashov | Principal | Evariste Systems LLC
> 
> Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
> Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
> 
> ___

Re: [OpenSIPS-Users] SIP SDP rfc1918 address fix

[Bogdan-Andrei Iancu]

So below is the SDP OpenSIPS receives (from network) and when doing 
fix_nated_sdp() on that SDP leads to the "c=" errors ?


Regards,

Bogdan-Andrei Iancu
  OpenSIPS Founder and Developer
  http://www.opensips-solutions.com

OpenSIPS Summit May 2017 Amsterdam
  http://www.opensips.org/events/Summit-2017Amsterdam.html

On 04/26/2017 08:44 PM, Satish Patel wrote:

Here is my payload again we have custom application which is using SER
so some of them are custom values, This is the payload after i apply
fix_nated_sdp() function.


Max-Forwards: 16.
Content-Type: application/sdp.
Content-Length: 418.
Supported: path, 100rel.
P-hint: LOCAL.
P-hint: ALIASED OUTBOUND.
P-hint: DIRECT-RTP.
.
v=0.
o=user1 53655765 2353687637 IN IP4 192.168.1.8.
s=-.
c=IN IP4 173.71.121.4.
t=0 0.
m=audio 6000 RTP/AVP 0.
a=rtpmap:127 VANI/32000.
a=fmtp:127 ver=3;mode=3;sub-types=1,7;codecs=0x26.
a=rtpmap:111 SIREN14-3D/32000.
a=fmtp:111 bitrate=32000.
a=vx_payload_hdr_ver:2.
a=rtpmap:0 PCMU/8000.
a=vx_join_audio:1.
a=vx_join_text:0.
a=vx_jc:60.
a=setup:both.
a=vx_rtcp:0.
a=direction:active.
a=oldmediaip:192.168.1.8.

On Wed, Apr 26, 2017 at 6:18 AM, Bogdan-Andrei Iancu
 wrote:

Hi Satish,

For the mime test, you can use the has_body() function:
http://www.opensips.org/html/docs/modules/2.2.x/sipmsgops.html#idp3886992

About the error - could you post the actual SDP payload generating those
errors ?

Regards,

Bogdan-Andrei Iancu
   OpenSIPS Founder and Developer
   http://www.opensips-solutions.com

OpenSIPS Summit May 2017 Amsterdam
   http://www.opensips.org/events/Summit-2017Amsterdam.html


On 04/25/2017 10:35 PM, Satish Patel wrote:

We have some custome Voice solution and in-house media server so right
now i don't care about PORT all i need correct IP address.

I have tried following and it fixed issue but i am seeing following
error in logs

if (method=="INVITE") {
  if(search("^Content-Type:.*application/sdp")) {
  fix_nated_sdp("3");
  };
};


Error:

ERROR: extract_mediaip: no `c=' in SDP
ERROR: extract_mediaip: no `c=' in SDP

Do you know what does that means and how to fix that issue?

On Mon, Apr 24, 2017 at 11:41 PM, Alex Balashov
 wrote:

The intent of my questions was to get what you think about what you
actually want to accomplish. fix_nated_sdp() allows you to replace the
IP with the received signalling IP:

http://www.opensips.org/html/docs/modules/2.1.x/nathelper.html#id293899

But what about the port?

On Mon, Apr 24, 2017 at 11:39:14PM -0400, Satish Patel wrote:


after google found bunch of post where people suggesting use
fix_nated_sdp()  is that right approach ?

On Mon, Apr 24, 2017 at 11:25 PM, Alex Balashov
 wrote:

Yes, but RTP can come from a different address than the signalling
(SIP). What sense would there be in substituting the source of the SIP
message in there?

On Mon, Apr 24, 2017 at 11:23:30PM -0400, Satish Patel wrote:


I meant "origin public address of client"  if c line isn't public then
media never work.

c=IN IP4 192.168.1.8.

It should be

c=IN IP4 

On Mon, Apr 24, 2017 at 11:04 PM, Alex Balashov
 wrote:

Satish,

When you say "origin public address", do you mean the external source
address and port of the SIP message, or the incoming RTP stream?

On Mon, Apr 24, 2017 at 11:00:40PM -0400, Satish Patel wrote:


In my INVITE/SDP i am seeing sometime rfc1918 address which i want
fix
and replace it with origin public address. ex

I am seeing following info in INVITE

v=0.
o=amsip 0 0 IN IP4 192.168.1.8.
s= .
c=IN IP4 192.168.1.8.
t=0 0.
m=audio 22530 RTP/AVP 127 111 0 101.

___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

--
Alex Balashov | Principal | Evariste Systems LLC

Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

--
Alex Balashov | Principal | Evariste Systems LLC

Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

--
Alex Balashov | Principal | Evariste Systems LLC

Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

___
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Re: [OpenSIPS-Users] Strange entries in ACC

[Bogdan-Andrei Iancu]

Jim,

We will try to replicate the scenario to see if we can reproduce the 
error. If not, we will provide some patch for extra debugging, so you 
can run it on your end.


Thanks and regards,

Bogdan-Andrei Iancu
  OpenSIPS Founder and Developer
  http://www.opensips-solutions.com

OpenSIPS Summit May 2017 Amsterdam
  http://www.opensips.org/events/Summit-2017Amsterdam.html

On 04/26/2017 07:15 PM, Jim DeVito wrote:
Correct. Assuming all the destinations that the load balancer 
module send to gives me a negative reply I turn around and send the 
600 to my upstream.


On Wed, Apr 26, 2017 at 12:13 PM, Bogdan-Andrei Iancu 
mailto:bog...@opensips.org>> wrote:


So, your call is actually proxied out (to a switch), but the 600
reply is generated by you in failure route (upon a 408 timeout ?)

Bogdan-Andrei Iancu
   OpenSIPS Founder and Developer
   http://www.opensips-solutions.com 

OpenSIPS Summit May 2017 Amsterdam
   http://www.opensips.org/events/Summit-2017Amsterdam.html


On 04/26/2017 07:10 PM, Jim DeVito wrote:

Every time. Just routing the number to a non existent destination
in my class 5 switch so that it ends up hitting the failure route.
On Wed, Apr 26, 2017 at 12:08 PM, Bogdan-Andrei Iancu
mailto:bog...@opensips.org>> wrote:

And I suppose you can reproduce such funky records, right ?

Bogdan-Andrei Iancu
   OpenSIPS Founder and Developer
   http://www.opensips-solutions.com


OpenSIPS Summit May 2017 Amsterdam
   http://www.opensips.org/events/Summit-2017Amsterdam.html


On 04/26/2017 06:50 PM, Jim DeVito wrote:

Correct.
do_accounting("db", "cdr|failed|missed", "acc");
And I meant the username part of the R-URI so it has an 11
digit phone number in it.
On Wed, Apr 26, 2017 at 11:48 AM, Bogdan-Andrei Iancu
mailto:bog...@opensips.org>> wrote:

OK, thanks. I suppose you have the "failed" flag set in
do_accounting() ? And before ending the INVITE
processing, do you change the RURI (you mentioned it
originally has a username) ? Regards,

Bogdan-Andrei Iancu
   OpenSIPS Founder and Developer
   http://www.opensips-solutions.com


OpenSIPS Summit May 2017 Amsterdam
   http://www.opensips.org/events/Summit-2017Amsterdam.html


On 04/26/2017 06:33 PM, Jim DeVito wrote:

Yes. And correct the script is rejecting the call at
this point and returning the 600 to my upstream.
On Wed, Apr 26, 2017 at 11:28 AM, Bogdan-Andrei Iancu
mailto:bog...@opensips.org>> wrote:

Again, does the strange text correspond to the
to_tn extra value ? And the call is rejected by you
from script ? it not ever proxied further, right ?
Regards,

Bogdan-Andrei Iancu
   OpenSIPS Founder and Developer
   http://www.opensips-solutions.com


OpenSIPS Summit May 2017 Amsterdam
   http://www.opensips.org/events/Summit-2017Amsterdam.html


On 04/26/2017 06:25 PM, Jim DeVito wrote:

Hi Bogdan,
Sorry forgot the mention I am using 2.2.3 the
latest stable from the repo. I put a log line just
after send_reply("600","Busy Everywhere"); and $rU
looks good there. Is there another place I should
put a log line to see the value of $rU?
Thanks!!
On Wed, Apr 26, 2017 at 11:16 AM, Bogdan-Andrei
Iancu mailto:bog...@opensips.org>> wrote:

Hi Jim, What OpenSIPS version do you use ? Is
the x00 string corresponding to the to_tn
extra field ? If yes, is there any chance to
have the $rU null (no username in RURI) ?
Regards,

Bogdan-Andrei Iancu
   OpenSIPS Founder and Developer
   http://www.opensips-solutions.com


OpenSIPS Summit May 2017 Amsterdam
   http://www.opensips.org/events/Summit-2017Amsterdam.html


On 04/26/2017 05:43 PM, Jim DeVito wrote:

   

[OpenSIPS-Users] Opensips 2.3 ERROR:core:sr_load_module

[Dragomir Haralambiev]

Hello,

Opensips 2.2.3 working fine. When try to start new 2.3 I see follow ERROR:

Apr 26 23:10:15 dev opensips: ERROR:core:sr_load_module: could not open
module :
/usr/local/lib64/opensips/modules/aaa_radius.so: undefined symbol:
rc_dict_findvend


Best regards,
Dragomir
___
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Re: [OpenSIPS-Users] SIP SDP rfc1918 address fix

[Satish Patel]

Here is my payload again we have custom application which is using SER
so some of them are custom values, This is the payload after i apply
fix_nated_sdp() function.


Max-Forwards: 16.
Content-Type: application/sdp.
Content-Length: 418.
Supported: path, 100rel.
P-hint: LOCAL.
P-hint: ALIASED OUTBOUND.
P-hint: DIRECT-RTP.
.
v=0.
o=user1 53655765 2353687637 IN IP4 192.168.1.8.
s=-.
c=IN IP4 173.71.121.4.
t=0 0.
m=audio 6000 RTP/AVP 0.
a=rtpmap:127 VANI/32000.
a=fmtp:127 ver=3;mode=3;sub-types=1,7;codecs=0x26.
a=rtpmap:111 SIREN14-3D/32000.
a=fmtp:111 bitrate=32000.
a=vx_payload_hdr_ver:2.
a=rtpmap:0 PCMU/8000.
a=vx_join_audio:1.
a=vx_join_text:0.
a=vx_jc:60.
a=setup:both.
a=vx_rtcp:0.
a=direction:active.
a=oldmediaip:192.168.1.8.

On Wed, Apr 26, 2017 at 6:18 AM, Bogdan-Andrei Iancu
 wrote:
> Hi Satish,
>
> For the mime test, you can use the has_body() function:
> http://www.opensips.org/html/docs/modules/2.2.x/sipmsgops.html#idp3886992
>
> About the error - could you post the actual SDP payload generating those
> errors ?
>
> Regards,
>
> Bogdan-Andrei Iancu
>   OpenSIPS Founder and Developer
>   http://www.opensips-solutions.com
>
> OpenSIPS Summit May 2017 Amsterdam
>   http://www.opensips.org/events/Summit-2017Amsterdam.html
>
>
> On 04/25/2017 10:35 PM, Satish Patel wrote:
>>
>> We have some custome Voice solution and in-house media server so right
>> now i don't care about PORT all i need correct IP address.
>>
>> I have tried following and it fixed issue but i am seeing following
>> error in logs
>>
>> if (method=="INVITE") {
>>  if(search("^Content-Type:.*application/sdp")) {
>>  fix_nated_sdp("3");
>>  };
>> };
>>
>>
>> Error:
>>
>> ERROR: extract_mediaip: no `c=' in SDP
>> ERROR: extract_mediaip: no `c=' in SDP
>>
>> Do you know what does that means and how to fix that issue?
>>
>> On Mon, Apr 24, 2017 at 11:41 PM, Alex Balashov
>>  wrote:
>>>
>>> The intent of my questions was to get what you think about what you
>>> actually want to accomplish. fix_nated_sdp() allows you to replace the
>>> IP with the received signalling IP:
>>>
>>> http://www.opensips.org/html/docs/modules/2.1.x/nathelper.html#id293899
>>>
>>> But what about the port?
>>>
>>> On Mon, Apr 24, 2017 at 11:39:14PM -0400, Satish Patel wrote:
>>>
 after google found bunch of post where people suggesting use
 fix_nated_sdp()  is that right approach ?

 On Mon, Apr 24, 2017 at 11:25 PM, Alex Balashov
  wrote:
>
> Yes, but RTP can come from a different address than the signalling
> (SIP). What sense would there be in substituting the source of the SIP
> message in there?
>
> On Mon, Apr 24, 2017 at 11:23:30PM -0400, Satish Patel wrote:
>
>> I meant "origin public address of client"  if c line isn't public then
>> media never work.
>>
>> c=IN IP4 192.168.1.8.
>>
>> It should be
>>
>> c=IN IP4 
>>
>> On Mon, Apr 24, 2017 at 11:04 PM, Alex Balashov
>>  wrote:
>>>
>>> Satish,
>>>
>>> When you say "origin public address", do you mean the external source
>>> address and port of the SIP message, or the incoming RTP stream?
>>>
>>> On Mon, Apr 24, 2017 at 11:00:40PM -0400, Satish Patel wrote:
>>>
 In my INVITE/SDP i am seeing sometime rfc1918 address which i want
 fix
 and replace it with origin public address. ex

 I am seeing following info in INVITE

 v=0.
 o=amsip 0 0 IN IP4 192.168.1.8.
 s= .
 c=IN IP4 192.168.1.8.
 t=0 0.
 m=audio 22530 RTP/AVP 127 111 0 101.

 ___
 Users mailing list
 Users@lists.opensips.org
 http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>> --
>>> Alex Balashov | Principal | Evariste Systems LLC
>>>
>>> Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
>>> Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
>>>
>>> ___
>>> Users mailing list
>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
> --
> Alex Balashov | Principal | Evariste Systems LLC
>
> Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
> Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

 ___
 Users mailing list
 Users@lists.opensips.org
 http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>> --
>>> Alex Balashov | Pri

[OpenSIPS-Users] OpenSIPS 2.3 Stable: The Last Hurdle Before the Amsterdam Summit

[Liviu Chircu]

Great news for everyone in the VoIP community: we have just 
released OpenSIPS 2.3.0 stable!


This release is a follow-up of over a month full of testing and taking 
care of issues reported through the mailing lists, GitHub tracker and 
IRC. Over 150 fix-commits were backported into OpenSIPS 2.3, leading to 
what we now consider to be a stable, production-ready SIP server. We 
would like to take this opportunity to thank everyone who got involved 
in this phase and helped make OpenSIPS better!


Nevertheless, this release is only a sip of what's to come, as the 2017 
OpenSIPS Summit is almost here. Not only it will be the biggest one so 
far, but its true value lies in the plethora of VoIP areas that it will 
reach - this event is more than OpenSIPS and SIP.


The Summit will bring together experts from all VoIP related areas:

* WebRTC - Jitsi, Mediasoup, Janus
* SoftSwitches - OpenSIPS, Sippy, FreeSWITCH, Asterisk
* Capturing & RTP - Homer, RTPProxy
* Billing - CGRateS

See you next week,
the OpenSIPS team

--
Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com

OpenSIPS Summit May 2017 Amsterdam
  http://www.opensips.org/events/Summit-2017Amsterdam.html


___
Users mailing list
Users@lists.opensips.org
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[OpenSIPS-Users] busin...@lists.opensips.org

[Liviu Chircu]

Great news for everyone in the VoIP community: we have just 
released OpenSIPS 2.3.0 stable!


This release is a follow-up of over a month full of testing and taking 
care of issues reported through the mailing lists, GitHub tracker and 
IRC. Over 150 fix-commits were backported into OpenSIPS 2.3, leading to 
what we now consider to be a stable, production-ready SIP server. We 
would like to take this opportunity to thank everyone who got involved 
in this phase and helped make OpenSIPS better!


Nevertheless, this release is only a sip of what's to come, as the 2017 
OpenSIPS Summit is almost here. Not only it will be the biggest one so 
far, but its true value lies in the plethora of VoIP areas that it will 
reach - this event is more than OpenSIPS and SIP.


The Summit will bring together experts from all VoIP related areas:

* WebRTC - Jitsi, Mediasoup, Janus
* SoftSwitches - OpenSIPS, Sippy, FreeSWITCH, Asterisk
* Capturing & RTP - Homer, RTPProxy
* Billing - CGRateS

See you next week,
the OpenSIPS team

--
Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com

OpenSIPS Summit May 2017 Amsterdam
  http://www.opensips.org/events/Summit-2017Amsterdam.html


___
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Re: [OpenSIPS-Users] Strange entries in ACC

[Jim DeVito]

Correct. Assuming all the destinations that the load balancer module send
to gives me a negative reply I turn around and send the 600 to my upstream.

On Wed, Apr 26, 2017 at 12:13 PM, Bogdan-Andrei Iancu 
wrote:

> So, your call is actually proxied out (to a switch), but the 600 reply is
> generated by you in failure route (upon a 408 timeout ?)
>
> Bogdan-Andrei Iancu
>   OpenSIPS Founder and Developer
>   http://www.opensips-solutions.com
>
> OpenSIPS Summit May 2017 Amsterdam
>   http://www.opensips.org/events/Summit-2017Amsterdam.html
>
> On 04/26/2017 07:10 PM, Jim DeVito wrote:
>
> Every time. Just routing the number to a non existent destination in my
> class 5 switch so that it ends up hitting the failure route.
>
> On Wed, Apr 26, 2017 at 12:08 PM, Bogdan-Andrei Iancu <
> bog...@opensips.org> wrote:
>
>> And I suppose you can reproduce such funky records, right ?
>>
>> Bogdan-Andrei Iancu
>>   OpenSIPS Founder and Developer
>>   http://www.opensips-solutions.com
>>
>> OpenSIPS Summit May 2017 Amsterdam
>>   http://www.opensips.org/events/Summit-2017Amsterdam.html
>>
>> On 04/26/2017 06:50 PM, Jim DeVito wrote:
>>
>> Correct.
>> do_accounting("db", "cdr|failed|missed", "acc");
>> And I meant the username part of the R-URI so it has an 11 digit phone
>> number in it.
>> On Wed, Apr 26, 2017 at 11:48 AM, Bogdan-Andrei Iancu <
>> bog...@opensips.org> wrote:
>>>
>>> OK, thanks. I suppose you have the "failed" flag set in do_accounting()
>>> ? And before ending the INVITE processing, do you change the RURI (you
>>> mentioned it originally has a username) ? Regards,
>>>
>>> Bogdan-Andrei Iancu
>>>   OpenSIPS Founder and Developer
>>>   http://www.opensips-solutions.com
>>>
>>> OpenSIPS Summit May 2017 Amsterdam
>>>   http://www.opensips.org/events/Summit-2017Amsterdam.html
>>>
>>> On 04/26/2017 06:33 PM, Jim DeVito wrote:
>>>
>>> Yes. And correct the script is rejecting the call at this point and
>>> returning the 600 to my upstream.
>>> On Wed, Apr 26, 2017 at 11:28 AM, Bogdan-Andrei Iancu <
>>> bog...@opensips.org> wrote:

 Again, does the strange text correspond to the to_tn extra value ? And
 the call is rejected by you from script ? it not ever proxied further,
 right ? Regards,

 Bogdan-Andrei Iancu
   OpenSIPS Founder and Developer
   http://www.opensips-solutions.com

 OpenSIPS Summit May 2017 Amsterdam
   http://www.opensips.org/events/Summit-2017Amsterdam.html

 On 04/26/2017 06:25 PM, Jim DeVito wrote:

 Hi Bogdan,
 Sorry forgot the mention I am using 2.2.3 the latest stable from the
 repo. I put a log line just after send_reply("600","Busy Everywhere"); and
 $rU looks good there. Is there another place I should put a log line to see
 the value of $rU?
 Thanks!!
 On Wed, Apr 26, 2017 at 11:16 AM, Bogdan-Andrei Iancu <
 bog...@opensips.org> wrote:
>
> Hi Jim, What OpenSIPS version do you use ? Is the x00 string
> corresponding to the to_tn extra field ? If yes, is there any chance to
> have the $rU null (no username in RURI) ? Regards,
>
> Bogdan-Andrei Iancu
>   OpenSIPS Founder and Developer
>   http://www.opensips-solutions.com
>
> OpenSIPS Summit May 2017 Amsterdam
>   http://www.opensips.org/events/Summit-2017Amsterdam.html
>
> On 04/26/2017 05:43 PM, Jim DeVito wrote:
>
> Hi All,
> So I am seeing the below record in the ACC output that is causing me
> problems else where. Notice the \x00\x00\x00\x00\x00\x00\x00\x
> 00\x00\x00 part.
> INVITE|gK0c48d4aa||1464632400_16750753@REDACTED|600|Busy
> Everywhere|1493217263|\x00\x00\x00\x00\x00\x00\x00\x00\x00\x
> 00|+1REDACTED|sip-proxy03|from-PSTN|REDACTED||8877||
> In every other ACC entry the value is to be the user part of the
> request URI as described here...
> modparam("acc", "db_extra", "to_tn=$rU; (etc)
> Except when the call goes through the below failure route.
> failure_route[orig_load_balance_fail] {
> if (t_was_cancelled()) {
> exit();
> }
> if (t_check_status("[56][0-9][0-9]") ||
> t_local_replied("all")) {
> if (lb_next()) {
> t_on_failure("orig_load_balance_fail");
> t_relay();
> exit();
> } else {
> send_reply("600","Busy Everywhere");
> exit();
> }
> }
> }
>
> Thoughts?
> Thanks!!
> Jim D.
>
> ___
> Users mailing 
> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
> --
 -
 Jim DeVito
 Mobile 216.507.9497 <%28216%29%20507-9497>

 --
>>> -
>>> Jim DeVito
>>> Mobile 216.507.9497 <%28216%29%20507-9497>
>>>

Re: [OpenSIPS-Users] Strange entries in ACC

[Bogdan-Andrei Iancu]

So, your call is actually proxied out (to a switch), but the 600 reply 
is generated by you in failure route (upon a 408 timeout ?)


Bogdan-Andrei Iancu
  OpenSIPS Founder and Developer
  http://www.opensips-solutions.com

OpenSIPS Summit May 2017 Amsterdam
  http://www.opensips.org/events/Summit-2017Amsterdam.html

On 04/26/2017 07:10 PM, Jim DeVito wrote:
Every time. Just routing the number to a non existent destination in 
my class 5 switch so that it ends up hitting the failure route.


On Wed, Apr 26, 2017 at 12:08 PM, Bogdan-Andrei Iancu 
mailto:bog...@opensips.org>> wrote:


And I suppose you can reproduce such funky records, right ?

Bogdan-Andrei Iancu
   OpenSIPS Founder and Developer
   http://www.opensips-solutions.com 

OpenSIPS Summit May 2017 Amsterdam
   http://www.opensips.org/events/Summit-2017Amsterdam.html


On 04/26/2017 06:50 PM, Jim DeVito wrote:

Correct.
do_accounting("db", "cdr|failed|missed", "acc");
And I meant the username part of the R-URI so it has an 11 digit
phone number in it.
On Wed, Apr 26, 2017 at 11:48 AM, Bogdan-Andrei Iancu
mailto:bog...@opensips.org>> wrote:

OK, thanks. I suppose you have the "failed" flag set in
do_accounting() ? And before ending the INVITE processing, do
you change the RURI (you mentioned it originally has a
username) ? Regards,

Bogdan-Andrei Iancu
   OpenSIPS Founder and Developer
   http://www.opensips-solutions.com


OpenSIPS Summit May 2017 Amsterdam
   http://www.opensips.org/events/Summit-2017Amsterdam.html


On 04/26/2017 06:33 PM, Jim DeVito wrote:

Yes. And correct the script is rejecting the call at this
point and returning the 600 to my upstream.
On Wed, Apr 26, 2017 at 11:28 AM, Bogdan-Andrei Iancu
mailto:bog...@opensips.org>> wrote:

Again, does the strange text correspond to the to_tn
extra value ? And the call is rejected by you from
script ? it not ever proxied further, right ? Regards,

Bogdan-Andrei Iancu
   OpenSIPS Founder and Developer
   http://www.opensips-solutions.com


OpenSIPS Summit May 2017 Amsterdam
   http://www.opensips.org/events/Summit-2017Amsterdam.html


On 04/26/2017 06:25 PM, Jim DeVito wrote:

Hi Bogdan,
Sorry forgot the mention I am using 2.2.3 the latest
stable from the repo. I put a log line just after
send_reply("600","Busy Everywhere"); and $rU looks good
there. Is there another place I should put a log line
to see the value of $rU?
Thanks!!
On Wed, Apr 26, 2017 at 11:16 AM, Bogdan-Andrei Iancu
mailto:bog...@opensips.org>> wrote:

Hi Jim, What OpenSIPS version do you use ? Is the
x00 string corresponding to the to_tn extra field ?
If yes, is there any chance to have the $rU null
(no username in RURI) ? Regards,

Bogdan-Andrei Iancu
   OpenSIPS Founder and Developer
   http://www.opensips-solutions.com


OpenSIPS Summit May 2017 Amsterdam
   http://www.opensips.org/events/Summit-2017Amsterdam.html


On 04/26/2017 05:43 PM, Jim DeVito wrote:

Hi All,
So I am seeing the below record in the ACC output
that is causing me problems else where. Notice the
\x00\x00\x00\x00\x00\x00\x00\x00\x00\x00 part.
INVITE|gK0c48d4aa||1464632400_16750753@REDACTED|600|Busy

Everywhere|1493217263|\x00\x00\x00\x00\x00\x00\x00\x00\x00\x00|+1REDACTED|sip-proxy03|from-PSTN|REDACTED||8877||

In every other ACC entry the value is to be the
user part of the request URI as described here...
modparam("acc", "db_extra", "to_tn=$rU; (etc)
Except when the call goes through the below
failure route.
failure_route[orig_load_balance_fail] {
if (t_was_cancelled()) {
exit();
}
if (t_check_status("[56][0-9][0-9]") ||
t_local_replied("all")) {
if (lb_next()) {
t_on_failure("orig_load_balance_fail");
   

Re: [OpenSIPS-Users] Strange entries in ACC

[Jim DeVito]

Every time. Just routing the number to a non existent destination in my
class 5 switch so that it ends up hitting the failure route.

On Wed, Apr 26, 2017 at 12:08 PM, Bogdan-Andrei Iancu 
wrote:

> And I suppose you can reproduce such funky records, right ?
>
> Bogdan-Andrei Iancu
>   OpenSIPS Founder and Developer
>   http://www.opensips-solutions.com
>
> OpenSIPS Summit May 2017 Amsterdam
>   http://www.opensips.org/events/Summit-2017Amsterdam.html
>
> On 04/26/2017 06:50 PM, Jim DeVito wrote:
>
> Correct.
>
> do_accounting("db", "cdr|failed|missed", "acc");
>
> And I meant the username part of the R-URI so it has an 11 digit phone
> number in it.
>
> On Wed, Apr 26, 2017 at 11:48 AM, Bogdan-Andrei Iancu <
> bog...@opensips.org> wrote:
>
>> OK, thanks.
>>
>> I suppose you have the "failed" flag set in do_accounting() ?
>>
>> And before ending the INVITE processing, do you change the RURI (you
>> mentioned it originally has a username) ?
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>>   OpenSIPS Founder and Developer
>>   http://www.opensips-solutions.com
>>
>> OpenSIPS Summit May 2017 Amsterdam
>>   http://www.opensips.org/events/Summit-2017Amsterdam.html
>>
>> On 04/26/2017 06:33 PM, Jim DeVito wrote:
>>
>> Yes. And correct the script is rejecting the call at this point and
>> returning the 600 to my upstream.
>> On Wed, Apr 26, 2017 at 11:28 AM, Bogdan-Andrei Iancu <
>> bog...@opensips.org> wrote:
>>>
>>> Again, does the strange text correspond to the to_tn extra value ? And
>>> the call is rejected by you from script ? it not ever proxied further,
>>> right ? Regards,
>>>
>>> Bogdan-Andrei Iancu
>>>   OpenSIPS Founder and Developer
>>>   http://www.opensips-solutions.com
>>>
>>> OpenSIPS Summit May 2017 Amsterdam
>>>   http://www.opensips.org/events/Summit-2017Amsterdam.html
>>>
>>> On 04/26/2017 06:25 PM, Jim DeVito wrote:
>>>
>>> Hi Bogdan,
>>> Sorry forgot the mention I am using 2.2.3 the latest stable from the
>>> repo. I put a log line just after send_reply("600","Busy Everywhere"); and
>>> $rU looks good there. Is there another place I should put a log line to see
>>> the value of $rU?
>>> Thanks!!
>>> On Wed, Apr 26, 2017 at 11:16 AM, Bogdan-Andrei Iancu <
>>> bog...@opensips.org> wrote:

 Hi Jim, What OpenSIPS version do you use ? Is the x00 string
 corresponding to the to_tn extra field ? If yes, is there any chance to
 have the $rU null (no username in RURI) ? Regards,

 Bogdan-Andrei Iancu
   OpenSIPS Founder and Developer
   http://www.opensips-solutions.com

 OpenSIPS Summit May 2017 Amsterdam
   http://www.opensips.org/events/Summit-2017Amsterdam.html

 On 04/26/2017 05:43 PM, Jim DeVito wrote:

 Hi All,
 So I am seeing the below record in the ACC output that is causing me
 problems else where. Notice the \x00\x00\x00\x00\x00\x00\x00\x
 00\x00\x00 part.
 INVITE|gK0c48d4aa||1464632400_16750753@REDACTED|600|Busy
 Everywhere|1493217263|\x00\x00\x00\x00\x00\x00\x00\x00\x00\x
 00|+1REDACTED|sip-proxy03|from-PSTN|REDACTED||8877||
 In every other ACC entry the value is to be the user part of the
 request URI as described here...
 modparam("acc", "db_extra", "to_tn=$rU; (etc)
 Except when the call goes through the below failure route.
 failure_route[orig_load_balance_fail] {
 if (t_was_cancelled()) {
 exit();
 }
 if (t_check_status("[56][0-9][0-9]") ||
 t_local_replied("all")) {
 if (lb_next()) {
 t_on_failure("orig_load_balance_fail");
 t_relay();
 exit();
 } else {
 send_reply("600","Busy Everywhere");
 exit();
 }
 }
 }

 Thoughts?
 Thanks!!
 Jim D.

 ___
 Users mailing 
 listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users

 --
>>> -
>>> Jim DeVito
>>> Mobile 216.507.9497 <%28216%29%20507-9497>
>>>
>>> --
>> -
>> Jim DeVito
>> Mobile 216.507.9497 <%28216%29%20507-9497>
>>
>> --
> -
> Jim DeVito
> Mobile 216.507.9497 <(216)%20507-9497>
>
>


-- 
-
Jim DeVito
Mobile 216.507.9497
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Strange entries in ACC

[Bogdan-Andrei Iancu]

And I suppose you can reproduce such funky records, right ?

Bogdan-Andrei Iancu
  OpenSIPS Founder and Developer
  http://www.opensips-solutions.com

OpenSIPS Summit May 2017 Amsterdam
  http://www.opensips.org/events/Summit-2017Amsterdam.html

On 04/26/2017 06:50 PM, Jim DeVito wrote:

Correct.

do_accounting("db", "cdr|failed|missed", "acc");

And I meant the username part of the R-URI so it has an 11 digit phone 
number in it.


On Wed, Apr 26, 2017 at 11:48 AM, Bogdan-Andrei Iancu 
mailto:bog...@opensips.org>> wrote:


OK, thanks.

I suppose you have the "failed" flag set in do_accounting() ?

And before ending the INVITE processing, do you change the RURI
(you mentioned it originally has a username) ?

Regards,

Bogdan-Andrei Iancu
   OpenSIPS Founder and Developer
   http://www.opensips-solutions.com 

OpenSIPS Summit May 2017 Amsterdam
   http://www.opensips.org/events/Summit-2017Amsterdam.html


On 04/26/2017 06:33 PM, Jim DeVito wrote:

Yes. And correct the script is rejecting the call at this point
and returning the 600 to my upstream.
On Wed, Apr 26, 2017 at 11:28 AM, Bogdan-Andrei Iancu
mailto:bog...@opensips.org>> wrote:

Again, does the strange text correspond to the to_tn extra
value ? And the call is rejected by you from script ? it not
ever proxied further, right ? Regards,

Bogdan-Andrei Iancu
   OpenSIPS Founder and Developer
   http://www.opensips-solutions.com


OpenSIPS Summit May 2017 Amsterdam
   http://www.opensips.org/events/Summit-2017Amsterdam.html


On 04/26/2017 06:25 PM, Jim DeVito wrote:

Hi Bogdan,
Sorry forgot the mention I am using 2.2.3 the latest stable
from the repo. I put a log line just after
send_reply("600","Busy Everywhere"); and $rU looks good
there. Is there another place I should put a log line to see
the value of $rU?
Thanks!!
On Wed, Apr 26, 2017 at 11:16 AM, Bogdan-Andrei Iancu
mailto:bog...@opensips.org>> wrote:

Hi Jim, What OpenSIPS version do you use ? Is the x00
string corresponding to the to_tn extra field ? If yes,
is there any chance to have the $rU null (no username in
RURI) ? Regards,

Bogdan-Andrei Iancu
   OpenSIPS Founder and Developer
   http://www.opensips-solutions.com


OpenSIPS Summit May 2017 Amsterdam
   http://www.opensips.org/events/Summit-2017Amsterdam.html


On 04/26/2017 05:43 PM, Jim DeVito wrote:

Hi All,
So I am seeing the below record in the ACC output that
is causing me problems else where. Notice the
\x00\x00\x00\x00\x00\x00\x00\x00\x00\x00 part.
INVITE|gK0c48d4aa||1464632400_16750753@REDACTED|600|Busy

Everywhere|1493217263|\x00\x00\x00\x00\x00\x00\x00\x00\x00\x00|+1REDACTED|sip-proxy03|from-PSTN|REDACTED||8877||

In every other ACC entry the value is to be the user
part of the request URI as described here...
modparam("acc", "db_extra", "to_tn=$rU; (etc)
Except when the call goes through the below failure route.
failure_route[orig_load_balance_fail] {
if (t_was_cancelled()) {
exit();
}
if (t_check_status("[56][0-9][0-9]") ||
t_local_replied("all")) {
if (lb_next()) {
t_on_failure("orig_load_balance_fail");
t_relay();
exit();
} else {
send_reply("600","Busy
Everywhere");
exit();
}
}
}
Thoughts?
Thanks!!
Jim D.

___
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-- 
-

Jim DeVito
Mobile 216.507.9497 


-- 
-

Jim DeVito
Mobile 216.507.9497 


--
-
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Re: [OpenSIPS-Users] Strange entries in ACC

[Jim DeVito]

Also it should be noted I am using the db_flatstore module with the acc
module.

On Wed, Apr 26, 2017 at 11:48 AM, Bogdan-Andrei Iancu 
wrote:

> OK, thanks.
>
> I suppose you have the "failed" flag set in do_accounting() ?
>
> And before ending the INVITE processing, do you change the RURI (you
> mentioned it originally has a username) ?
>
> Regards,
>
> Bogdan-Andrei Iancu
>   OpenSIPS Founder and Developer
>   http://www.opensips-solutions.com
>
> OpenSIPS Summit May 2017 Amsterdam
>   http://www.opensips.org/events/Summit-2017Amsterdam.html
>
> On 04/26/2017 06:33 PM, Jim DeVito wrote:
>
> Yes. And correct the script is rejecting the call at this point and
> returning the 600 to my upstream.
>
> On Wed, Apr 26, 2017 at 11:28 AM, Bogdan-Andrei Iancu <
> bog...@opensips.org> wrote:
>
>> Again, does the strange text correspond to the to_tn extra value ?
>>
>> And the call is rejected by you from script ? it not ever proxied
>> further, right ?
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>>   OpenSIPS Founder and Developer
>>   http://www.opensips-solutions.com
>>
>> OpenSIPS Summit May 2017 Amsterdam
>>   http://www.opensips.org/events/Summit-2017Amsterdam.html
>>
>> On 04/26/2017 06:25 PM, Jim DeVito wrote:
>>
>> Hi Bogdan,
>> Sorry forgot the mention I am using 2.2.3 the latest stable from the
>> repo. I put a log line just after send_reply("600","Busy Everywhere"); and
>> $rU looks good there. Is there another place I should put a log line to see
>> the value of $rU?
>> Thanks!!
>> On Wed, Apr 26, 2017 at 11:16 AM, Bogdan-Andrei Iancu <
>> bog...@opensips.org> wrote:
>>>
>>> Hi Jim, What OpenSIPS version do you use ? Is the x00 string
>>> corresponding to the to_tn extra field ? If yes, is there any chance to
>>> have the $rU null (no username in RURI) ? Regards,
>>>
>>> Bogdan-Andrei Iancu
>>>   OpenSIPS Founder and Developer
>>>   http://www.opensips-solutions.com
>>>
>>> OpenSIPS Summit May 2017 Amsterdam
>>>   http://www.opensips.org/events/Summit-2017Amsterdam.html
>>>
>>> On 04/26/2017 05:43 PM, Jim DeVito wrote:
>>>
>>> Hi All,
>>> So I am seeing the below record in the ACC output that is causing me
>>> problems else where. Notice the \x00\x00\x00\x00\x00\x00\x00\x
>>> 00\x00\x00 part.
>>> INVITE|gK0c48d4aa||1464632400_16750753@REDACTED|600|Busy
>>> Everywhere|1493217263|\x00\x00\x00\x00\x00\x00\x00\x00\x00\x
>>> 00|+1REDACTED|sip-proxy03|from-PSTN|REDACTED||8877||
>>> In every other ACC entry the value is to be the user part of the request
>>> URI as described here...
>>> modparam("acc", "db_extra", "to_tn=$rU; (etc)
>>> Except when the call goes through the below failure route.
>>> failure_route[orig_load_balance_fail] {
>>> if (t_was_cancelled()) {
>>> exit();
>>> }
>>> if (t_check_status("[56][0-9][0-9]") || t_local_replied("all"))
>>> {
>>> if (lb_next()) {
>>> t_on_failure("orig_load_balance_fail");
>>> t_relay();
>>> exit();
>>> } else {
>>> send_reply("600","Busy Everywhere");
>>> exit();
>>> }
>>> }
>>> }
>>>
>>> Thoughts?
>>> Thanks!!
>>> Jim D.
>>>
>>> ___
>>> Users mailing 
>>> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>> --
>> -
>> Jim DeVito
>> Mobile 216.507.9497 <%28216%29%20507-9497>
>>
>> --
> -
> Jim DeVito
> Mobile 216.507.9497 <(216)%20507-9497>
>
>


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Re: [OpenSIPS-Users] Strange entries in ACC

[Jim DeVito]

Correct.

do_accounting("db", "cdr|failed|missed", "acc");

And I meant the username part of the R-URI so it has an 11 digit phone
number in it.

On Wed, Apr 26, 2017 at 11:48 AM, Bogdan-Andrei Iancu 
wrote:

> OK, thanks.
>
> I suppose you have the "failed" flag set in do_accounting() ?
>
> And before ending the INVITE processing, do you change the RURI (you
> mentioned it originally has a username) ?
>
> Regards,
>
> Bogdan-Andrei Iancu
>   OpenSIPS Founder and Developer
>   http://www.opensips-solutions.com
>
> OpenSIPS Summit May 2017 Amsterdam
>   http://www.opensips.org/events/Summit-2017Amsterdam.html
>
> On 04/26/2017 06:33 PM, Jim DeVito wrote:
>
> Yes. And correct the script is rejecting the call at this point and
> returning the 600 to my upstream.
>
> On Wed, Apr 26, 2017 at 11:28 AM, Bogdan-Andrei Iancu <
> bog...@opensips.org> wrote:
>
>> Again, does the strange text correspond to the to_tn extra value ?
>>
>> And the call is rejected by you from script ? it not ever proxied
>> further, right ?
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>>   OpenSIPS Founder and Developer
>>   http://www.opensips-solutions.com
>>
>> OpenSIPS Summit May 2017 Amsterdam
>>   http://www.opensips.org/events/Summit-2017Amsterdam.html
>>
>> On 04/26/2017 06:25 PM, Jim DeVito wrote:
>>
>> Hi Bogdan,
>> Sorry forgot the mention I am using 2.2.3 the latest stable from the
>> repo. I put a log line just after send_reply("600","Busy Everywhere"); and
>> $rU looks good there. Is there another place I should put a log line to see
>> the value of $rU?
>> Thanks!!
>> On Wed, Apr 26, 2017 at 11:16 AM, Bogdan-Andrei Iancu <
>> bog...@opensips.org> wrote:
>>>
>>> Hi Jim, What OpenSIPS version do you use ? Is the x00 string
>>> corresponding to the to_tn extra field ? If yes, is there any chance to
>>> have the $rU null (no username in RURI) ? Regards,
>>>
>>> Bogdan-Andrei Iancu
>>>   OpenSIPS Founder and Developer
>>>   http://www.opensips-solutions.com
>>>
>>> OpenSIPS Summit May 2017 Amsterdam
>>>   http://www.opensips.org/events/Summit-2017Amsterdam.html
>>>
>>> On 04/26/2017 05:43 PM, Jim DeVito wrote:
>>>
>>> Hi All,
>>> So I am seeing the below record in the ACC output that is causing me
>>> problems else where. Notice the \x00\x00\x00\x00\x00\x00\x00\x
>>> 00\x00\x00 part.
>>> INVITE|gK0c48d4aa||1464632400_16750753@REDACTED|600|Busy
>>> Everywhere|1493217263|\x00\x00\x00\x00\x00\x00\x00\x00\x00\x
>>> 00|+1REDACTED|sip-proxy03|from-PSTN|REDACTED||8877||
>>> In every other ACC entry the value is to be the user part of the request
>>> URI as described here...
>>> modparam("acc", "db_extra", "to_tn=$rU; (etc)
>>> Except when the call goes through the below failure route.
>>> failure_route[orig_load_balance_fail] {
>>> if (t_was_cancelled()) {
>>> exit();
>>> }
>>> if (t_check_status("[56][0-9][0-9]") || t_local_replied("all"))
>>> {
>>> if (lb_next()) {
>>> t_on_failure("orig_load_balance_fail");
>>> t_relay();
>>> exit();
>>> } else {
>>> send_reply("600","Busy Everywhere");
>>> exit();
>>> }
>>> }
>>> }
>>>
>>> Thoughts?
>>> Thanks!!
>>> Jim D.
>>>
>>> ___
>>> Users mailing 
>>> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>> --
>> -
>> Jim DeVito
>> Mobile 216.507.9497 <%28216%29%20507-9497>
>>
>> --
> -
> Jim DeVito
> Mobile 216.507.9497 <(216)%20507-9497>
>
>


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Re: [OpenSIPS-Users] Strange entries in ACC

[Bogdan-Andrei Iancu]

OK, thanks.

I suppose you have the "failed" flag set in do_accounting() ?

And before ending the INVITE processing, do you change the RURI (you 
mentioned it originally has a username) ?


Regards,

Bogdan-Andrei Iancu
  OpenSIPS Founder and Developer
  http://www.opensips-solutions.com

OpenSIPS Summit May 2017 Amsterdam
  http://www.opensips.org/events/Summit-2017Amsterdam.html

On 04/26/2017 06:33 PM, Jim DeVito wrote:
Yes. And correct the script is rejecting the call at this point and 
returning the 600 to my upstream.


On Wed, Apr 26, 2017 at 11:28 AM, Bogdan-Andrei Iancu 
mailto:bog...@opensips.org>> wrote:


Again, does the strange text correspond to the to_tn extra value ?

And the call is rejected by you from script ? it not ever proxied
further, right ?

Regards,

Bogdan-Andrei Iancu
   OpenSIPS Founder and Developer
   http://www.opensips-solutions.com 

OpenSIPS Summit May 2017 Amsterdam
   http://www.opensips.org/events/Summit-2017Amsterdam.html


On 04/26/2017 06:25 PM, Jim DeVito wrote:

Hi Bogdan,
Sorry forgot the mention I am using 2.2.3 the latest stable from
the repo. I put a log line just after send_reply("600","Busy
Everywhere"); and $rU looks good there. Is there another place I
should put a log line to see the value of $rU?
Thanks!!
On Wed, Apr 26, 2017 at 11:16 AM, Bogdan-Andrei Iancu
mailto:bog...@opensips.org>> wrote:

Hi Jim, What OpenSIPS version do you use ? Is the x00 string
corresponding to the to_tn extra field ? If yes, is there any
chance to have the $rU null (no username in RURI) ? Regards,

Bogdan-Andrei Iancu
   OpenSIPS Founder and Developer
   http://www.opensips-solutions.com


OpenSIPS Summit May 2017 Amsterdam
   http://www.opensips.org/events/Summit-2017Amsterdam.html


On 04/26/2017 05:43 PM, Jim DeVito wrote:

Hi All,
So I am seeing the below record in the ACC output that is
causing me problems else where. Notice the
\x00\x00\x00\x00\x00\x00\x00\x00\x00\x00 part.
INVITE|gK0c48d4aa||1464632400_16750753@REDACTED|600|Busy

Everywhere|1493217263|\x00\x00\x00\x00\x00\x00\x00\x00\x00\x00|+1REDACTED|sip-proxy03|from-PSTN|REDACTED||8877||

In every other ACC entry the value is to be the user part of
the request URI as described here...
modparam("acc", "db_extra", "to_tn=$rU; (etc)
Except when the call goes through the below failure route.
failure_route[orig_load_balance_fail] {
if (t_was_cancelled()) {
exit();
}
if (t_check_status("[56][0-9][0-9]") ||
t_local_replied("all")) {
if (lb_next()) {
t_on_failure("orig_load_balance_fail");
t_relay();
exit();
} else {
send_reply("600","Busy Everywhere");
exit();
}
}
}
Thoughts?
Thanks!!
Jim D.

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Mobile 216.507.9497 


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Re: [OpenSIPS-Users] Strange entries in ACC

[Jim DeVito]

Yes. And correct the script is rejecting the call at this point and
returning the 600 to my upstream.

On Wed, Apr 26, 2017 at 11:28 AM, Bogdan-Andrei Iancu 
wrote:

> Again, does the strange text correspond to the to_tn extra value ?
>
> And the call is rejected by you from script ? it not ever proxied further,
> right ?
>
> Regards,
>
> Bogdan-Andrei Iancu
>   OpenSIPS Founder and Developer
>   http://www.opensips-solutions.com
>
> OpenSIPS Summit May 2017 Amsterdam
>   http://www.opensips.org/events/Summit-2017Amsterdam.html
>
> On 04/26/2017 06:25 PM, Jim DeVito wrote:
>
> Hi Bogdan,
>
> Sorry forgot the mention I am using 2.2.3 the latest stable from the repo.
> I put a log line just after send_reply("600","Busy Everywhere"); and $rU
> looks good there. Is there another place I should put a log line to see the
> value of $rU?
>
> Thanks!!
>
>
>
> On Wed, Apr 26, 2017 at 11:16 AM, Bogdan-Andrei Iancu <
> bog...@opensips.org> wrote:
>
>> Hi Jim,
>>
>> What OpenSIPS version do you use ?
>>
>> Is the x00 string corresponding to the to_tn extra field ? If yes, is
>> there any chance to have the $rU null (no username in RURI) ?
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>>   OpenSIPS Founder and Developer
>>   http://www.opensips-solutions.com
>>
>> OpenSIPS Summit May 2017 Amsterdam
>>   http://www.opensips.org/events/Summit-2017Amsterdam.html
>>
>> On 04/26/2017 05:43 PM, Jim DeVito wrote:
>>
>> Hi All,
>> So I am seeing the below record in the ACC output that is causing me
>> problems else where. Notice the \x00\x00\x00\x00\x00\x00\x00\x
>> 00\x00\x00 part.
>> INVITE|gK0c48d4aa||1464632400_16750753@REDACTED|600|Busy
>> Everywhere|1493217263|\x00\x00\x00\x00\x00\x00\x00\x00\x00\
>> x00|+1REDACTED|sip-proxy03|from-PSTN|REDACTED||8877||
>> In every other ACC entry the value is to be the user part of the request
>> URI as described here...
>> modparam("acc", "db_extra", "to_tn=$rU; (etc)
>> Except when the call goes through the below failure route.
>> failure_route[orig_load_balance_fail] {
>> if (t_was_cancelled()) {
>> exit();
>> }
>> if (t_check_status("[56][0-9][0-9]") || t_local_replied("all")) {
>> if (lb_next()) {
>> t_on_failure("orig_load_balance_fail");
>> t_relay();
>> exit();
>> } else {
>> send_reply("600","Busy Everywhere");
>> exit();
>> }
>> }
>> }
>>
>> Thoughts?
>> Thanks!!
>> Jim D.
>>
>> ___
>> Users mailing 
>> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>> --
> -
> Jim DeVito
> Mobile 216.507.9497 <(216)%20507-9497>
>
>


-- 
-
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Mobile 216.507.9497
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Re: [OpenSIPS-Users] Strange entries in ACC

[Bogdan-Andrei Iancu]

Again, does the strange text correspond to the to_tn extra value ?

And the call is rejected by you from script ? it not ever proxied 
further, right ?


Regards,

Bogdan-Andrei Iancu
  OpenSIPS Founder and Developer
  http://www.opensips-solutions.com

OpenSIPS Summit May 2017 Amsterdam
  http://www.opensips.org/events/Summit-2017Amsterdam.html

On 04/26/2017 06:25 PM, Jim DeVito wrote:

Hi Bogdan,

Sorry forgot the mention I am using 2.2.3 the latest stable from the 
repo. I put a log line just after send_reply("600","Busy Everywhere"); 
and $rU looks good there. Is there another place I should put a log 
line to see the value of $rU?


Thanks!!



On Wed, Apr 26, 2017 at 11:16 AM, Bogdan-Andrei Iancu 
mailto:bog...@opensips.org>> wrote:


Hi Jim,

What OpenSIPS version do you use ?

Is the x00 string corresponding to the to_tn extra field ? If yes,
is there any chance to have the $rU null (no username in RURI) ?

Regards,

Bogdan-Andrei Iancu
   OpenSIPS Founder and Developer
   http://www.opensips-solutions.com 

OpenSIPS Summit May 2017 Amsterdam
   http://www.opensips.org/events/Summit-2017Amsterdam.html


On 04/26/2017 05:43 PM, Jim DeVito wrote:

Hi All,
So I am seeing the below record in the ACC output that is causing
me problems else where. Notice the
\x00\x00\x00\x00\x00\x00\x00\x00\x00\x00 part.
INVITE|gK0c48d4aa||1464632400_16750753@REDACTED|600|Busy

Everywhere|1493217263|\x00\x00\x00\x00\x00\x00\x00\x00\x00\x00|+1REDACTED|sip-proxy03|from-PSTN|REDACTED||8877||

In every other ACC entry the value is to be the user part of the
request URI as described here...
modparam("acc", "db_extra", "to_tn=$rU; (etc)
Except when the call goes through the below failure route.
failure_route[orig_load_balance_fail] {
if (t_was_cancelled()) {
exit();
}
if (t_check_status("[56][0-9][0-9]") ||
t_local_replied("all")) {
if (lb_next()) {
t_on_failure("orig_load_balance_fail");
t_relay();
exit();
} else {
send_reply("600","Busy Everywhere");
exit();
}
}
}
Thoughts?
Thanks!!
Jim D.

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Re: [OpenSIPS-Users] Strange entries in ACC

[Jim DeVito]

Hi Bogdan,

Sorry forgot the mention I am using 2.2.3 the latest stable from the repo.
I put a log line just after send_reply("600","Busy Everywhere"); and $rU
looks good there. Is there another place I should put a log line to see the
value of $rU?

Thanks!!



On Wed, Apr 26, 2017 at 11:16 AM, Bogdan-Andrei Iancu 
wrote:

> Hi Jim,
>
> What OpenSIPS version do you use ?
>
> Is the x00 string corresponding to the to_tn extra field ? If yes, is
> there any chance to have the $rU null (no username in RURI) ?
>
> Regards,
>
> Bogdan-Andrei Iancu
>   OpenSIPS Founder and Developer
>   http://www.opensips-solutions.com
>
> OpenSIPS Summit May 2017 Amsterdam
>   http://www.opensips.org/events/Summit-2017Amsterdam.html
>
> On 04/26/2017 05:43 PM, Jim DeVito wrote:
>
> Hi All,
>
> So I am seeing the below record in the ACC output that is causing me
> problems else where. Notice the \x00\x00\x00\x00\x00\x00\x00\
> x00\x00\x00 part.
>
> INVITE|gK0c48d4aa||1464632400_16750753@REDACTED|600|Busy
> Everywhere|1493217263|\x00\x00\x00\x00\x00\x00\x00\x00\
> x00\x00|+1REDACTED|sip-proxy03|from-PSTN|REDACTED||8877||
>
> In every other ACC entry the value is to be the user part of the request
> URI as described here...
>
> modparam("acc", "db_extra", "to_tn=$rU; (etc)
>
> Except when the call goes through the below failure route.
>
>
> failure_route[orig_load_balance_fail] {
>
> if (t_was_cancelled()) {
> exit();
> }
>
> if (t_check_status("[56][0-9][0-9]") || t_local_replied("all")) {
>
> if (lb_next()) {
> t_on_failure("orig_load_balance_fail");
> t_relay();
> exit();
> } else {
> send_reply("600","Busy Everywhere");
> exit();
> }
> }
> }
>
>
> Thoughts?
>
> Thanks!!
>
> Jim D.
>
>
> ___
> Users mailing 
> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>


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Re: [OpenSIPS-Users] Strange entries in ACC

[Bogdan-Andrei Iancu]

Hi Jim,

What OpenSIPS version do you use ?

Is the x00 string corresponding to the to_tn extra field ? If yes, is 
there any chance to have the $rU null (no username in RURI) ?


Regards,

Bogdan-Andrei Iancu
  OpenSIPS Founder and Developer
  http://www.opensips-solutions.com

OpenSIPS Summit May 2017 Amsterdam
  http://www.opensips.org/events/Summit-2017Amsterdam.html

On 04/26/2017 05:43 PM, Jim DeVito wrote:

Hi All,

So I am seeing the below record in the ACC output that is causing me 
problems else where. Notice the 
\x00\x00\x00\x00\x00\x00\x00\x00\x00\x00 part.


INVITE|gK0c48d4aa||1464632400_16750753@REDACTED|600|Busy 
Everywhere|1493217263|\x00\x00\x00\x00\x00\x00\x00\x00\x00\x00|+1REDACTED|sip-proxy03|from-PSTN|REDACTED||8877||


In every other ACC entry the value is to be the user part of the 
request URI as described here...


modparam("acc", "db_extra", "to_tn=$rU; (etc)

Except when the call goes through the below failure route.


failure_route[orig_load_balance_fail] {

if (t_was_cancelled()) {
exit();
}

if (t_check_status("[56][0-9][0-9]") || t_local_replied("all")) {

if (lb_next()) {
t_on_failure("orig_load_balance_fail");
t_relay();
exit();
} else {
send_reply("600","Busy Everywhere");
exit();
}
}
}

Thoughts?

Thanks!!

Jim D.


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[OpenSIPS-Users] Strange entries in ACC

[Jim DeVito]

Hi All,

So I am seeing the below record in the ACC output that is causing me
problems else where. Notice the
\x00\x00\x00\x00\x00\x00\x00\x00\x00\x00 part.

INVITE|gK0c48d4aa||1464632400_16750753@REDACTED|600|Busy
Everywhere|1493217263|\x00\x00\x00\x00\x00\x00\x00\x00\x00\x00|+1REDACTED|sip-proxy03|from-PSTN|REDACTED||8877||

In every other ACC entry the value is to be the user part of the request
URI as described here...

modparam("acc", "db_extra", "to_tn=$rU; (etc)

Except when the call goes through the below failure route.


failure_route[orig_load_balance_fail] {

if (t_was_cancelled()) {
exit();
}

if (t_check_status("[56][0-9][0-9]") || t_local_replied("all")) {

if (lb_next()) {
t_on_failure("orig_load_balance_fail");
t_relay();
exit();
} else {
send_reply("600","Busy Everywhere");
exit();
}
}
}


Thoughts?

Thanks!!

Jim D.
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Re: [OpenSIPS-Users] BLF with only one domain?

[maatohewetbi]

I think I didn't describe it correctly. Let me explain my scenario:

AsteriskBox1(IP:192.168.0.100)<-SIP->Opensips1.11(192.168.0.110)

Asterisk sends Invites to Opensips to an IP, eg. 192.168.0.110. Opensips
listens on this IP, but it has also a few domains:
sip1.com
sip2.com
...
There are users, that uses different domain names to register to OpenSips.
If one user registers to domain sip1.com, and he wants to know BLF states of
other user which registers to sip2.com, it can't be done, because asterisk
sends invite with an IP and SIP Presentity doesn't match. So I wanted to
store every subscription with and IP(eg.: 123456789@192.168.0.110 instead of
123456...@sip1.com in active_watchers), even a user uses domain name. Ok, I
can do it to tell a user to register using an IP, but I want them to use
domain names instead of IP. So I wanted to use  modparam("presence",
"bla_presentity_spec", "$var(bla_pres)")  to change it, but I might don't
understand what it should do.



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Re: [OpenSIPS-Users] Fwd: websocket recover after opensips restart

[Mikhail Sidorov]

Hi

You could find opensips logs here (log_level=3)
https://pastebin.com/Gmq3BLkt

And here I tried to extract last BYE proceed from logs in debug mode
(log_level=4)
https://pastebin.com/exyVuqj0


I am using load balancer to send requests to one of 2 freeswitches.
Both users (USER_1 and USER_2) are webrtc users.
After opensips restart both clients reconnects and re-register, you could
find it in logs.
First routing of BYE from webrtc to freeswitch is successful, but secondary
from freeswitch to another client fails.


Below is a part of my routing for totag.
It works fine till I restart opensips.
I expect to see in log "t_relay error" message, but it is not there.

if (has_totag())
{
xlog("L_ERR", "Following ToTag: $tt $proto");
if (loose_route())
{
xlog("L_ERR", "loose_route success\n");
if (!t_relay())
{
xlog("L_ERR", "t_relay error [$du]");
send_reply("500","Internal Error");
}
}
else
{
xlog("L_ERR", "loose_route fail\n");
if ( is_method("ACK") )
{
if ( t_check_trans() )
{
t_relay();
exit;
}
else
{
exit;
}
}
sl_send_reply("404","Not here");
}
exit;
}



2017-04-26 16:05 GMT+07:00 Răzvan Crainea :

> Are you sure t_relay() is not returning an error? Can you post somewhere
> the entire log for this scenario?
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Solutionswww.opensips-solutions.com
>
> On 04/26/2017 12:00 PM, Mikhail Sidorov wrote:
>
> I am sorry for spamming you, I missed subscription verification email.
>
> Regarding my question, I understand that opensips could not recover
> connection and my client already reconnect.
> All I need is ability to catch this error to execute lookup and find
> reconnected client.
> And I could not find how to detect this kind of failure, because t_relay
> does not return error.
> Do you have any suggestions?
>
> ср, 26 апр. 2017 г. в 15:49, Răzvan Crainea :
>
>> Hi, Mikhail!
>>
>> I've already replied to your question[1]. Please register to the opensips
>> list if you want to get further emails.
>>
>> [1] http://lists.opensips.org/pipermail/users/2017-April/037179.html
>>
>> Best regards,
>>
>> Răzvan Crainea
>> OpenSIPS Solutionswww.opensips-solutions.com
>>
>> On 04/26/2017 11:05 AM, Mikhail Sidorov wrote:
>>
>> Hi
>>
>> I am using opensips 2.2.3 to handle websocket connections from web client
>> (sip.js).
>> I just started testing recover after opensips restart and find that I
>> could not handle t_relay() failure.
>>
>> Assume I have a call between 2 webrtc clients and I restart opensips
>> during this call. Media handled by freeswitch, so call continue without
>> problem.
>>
>> Then, one client send BYE. I use loose_route to find another party and
>> send him message.
>> But here I got error that I could not handle.
>> t_on_failure trigger does not fire
>> t_relay return success.
>>
>> Are there any special flags or triggers to handle this case?
>>
>> Opensips log:
>>
>> loose_route success
>> INFO:core:probe_max_sock_buff: using snd buffer of 416 kb
>> INFO:core:init_sock_keepalive: TCP keepalive enabled on socket 16
>> ERROR:core:tcp_connect_blocking: poll error: flags 28 - 4 8 16 32
>> ERROR:core:tcp_connect_blocking: failed to retrieve SO_ERROR [server=
>> 172.19.235.225:62594] (111) Connection refused
>> ERROR:proto_wss:ws_sync_connect: tcp_blocking_connect failed
>> ERROR:proto_wss:ws_connect: connect failed
>> ERROR:proto_wss:proto_wss_send: connect failed
>> ERROR:tm:msg_send: send() for proto 6 failed
>> ERROR:tm:t_forward_nonack: sending request failed
>>
>> My configuration:
>>
>> if (has_totag())
>> {
>> xlog("L_ERR", "Following ToTag: $tt $proto");
>> # sequential requests within a dialog should
>> # take the path determined by record-routing
>> if (loose_route())
>> {
>> #if (is_method("INVITE"))
>> #{
>> # record_route();
>> #}
>> xlog("L_ERR", "loose_route success\n");
>> route(relay);
>> }
>> 
>> route[relay]
>> {
>> t_on_failure("loose_route");
>> if (!t_relay())
>> {
>> xlog("L_ERR", "t_relay error]");
>> send_reply("500","Internal Error");
>> }
>> exit;
>> }
>>
>> failure_route[loose_route]
>> {
>> xlog( "L_ERR", "loose_route failed" );
>> ...
>> }
>>
>>
>>
>>
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>>
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Re: [OpenSIPS-Users] ERROR topology_hiding_match() and WSS

[Bogdan-Andrei Iancu]

Thank you Dragomir,

So, there is an existing connection to your browser.

Please redo the list_tcp_conns and send me the capture of the BYE 
request received by OpenSIPS.


Best regards,

Bogdan-Andrei Iancu
  OpenSIPS Founder and Developer
  http://www.opensips-solutions.com

OpenSIPS Summit May 2017 Amsterdam
  http://www.opensips.org/events/Summit-2017Amsterdam.html

On 04/25/2017 04:35 PM, Dragomir Haralambiev wrote:

Here is connection list before send BYE

[root@dev opensips]# opensipsctl fifo list_tcp_conns
Connection::  ID=1 Type=wss State=0 Source=0:59562 
Destination=:10062 Lifetime=2017-04-25 16:31:14



2017-04-25 11:37 GMT+03:00 Bogdan-Andrei Iancu >:


Hi Dragomir,

So, the problem is about the BYE. Just before sending BYE from
Zoiper, please run on your opensips:
opensipsctl fifo list_tcp_conns

Lets see if your SIP.JS still has a connection to OpenSIPS at that
point.

Best regards,

Bogdan-Andrei Iancu
   OpenSIPS Founder and Developer
   http://www.opensips-solutions.com 

OpenSIPS Summit May 2017 Amsterdam
   http://www.opensips.org/events/Summit-2017Amsterdam.html


On 04/25/2017 12:14 AM, Dragomir Haralambiev wrote:

Hi Bogdan,
Thanks for the detailed and comprehensive answer.
Zoiper talking (connection is established) with SIP.JS
Zoiper ---(over UDP)--> Opensips --(over WSS)---> SIP.JS
Why not possible Opensips send BYE to SIP.JS over established
connection?
Why all is OK when SIP.JS send BYE?
I need help to setup Opensips to solve this problem.
Best regards,
Dragomir
2017-04-24 17:47 GMT+03:00 Bogdan-Andrei Iancu
mailto:bog...@opensips.org>>:

Hi, Same question, same answer - see
http://lists.opensips.org/pipermail/users/2017-April/036925.html
Regards,

Bogdan-Andrei Iancu
   OpenSIPS Founder and Developer
   http://www.opensips-solutions.com


OpenSIPS Summit May 2017 Amsterdam
   http://www.opensips.org/events/Summit-2017Amsterdam.html


On 04/21/2017 07:24 PM, Dragomir Haralambiev wrote:

Hello,
I have problem with "topology_hiding_match()" and WSS.
Zoiper ---(send BYE)--> Opensips --(can not relay to)--->
SIP.JS
Here part ot script:
if (has_totag()) {
if (topology_hiding_match()) {
t_relay();
exit;
}
...
Opensips receive BYE. When execute "t_relay()" give follow
ERRORS:
INFO:core:probe_max_sock_buff: using snd buffer of 416 kb
INFO:core:init_sock_keepalive: TCP keepalive enabled on
socket 23
ERROR:core:tcp_connect_blocking: timeout 99195 ms elapsed
from 10 s
ERROR:proto_ws:ws_sync_connect: tcp_blocking_connect failed
ERROR:proto_ws:ws_connect: connect failed
ERROR:proto_ws:proto_ws_send: connect failed
ERROR:tm:msg_send: send() for proto 5 failed
ERROR:tm:t_forward_nonack: sending request failedscheduled
Where is problem?

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Re: [OpenSIPS-Users] opensips 2.3 crashed on user reg - mid_registrar

[Liviu Chircu]

Hi Kirill,

Are you running the latest 2.3 code? (verify with "opensips -V"). There 
have been lots of backported mid_registrar fixes over the last week 
(such as this one [1]).


Regards,

[1]: https://github.com/OpenSIPS/opensips/commit/ed5b3900878

Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com

OpenSIPS Summit May 2017 Amsterdam
  http://www.opensips.org/events/Summit-2017Amsterdam.html

On 26.04.2017 12:39, Kirill Galinurov wrote:

Hi I also have the same problem. Our config is:

loadmodule "mid_registrar.so"
modparam("mid_registrar", "mode", 1) /* 0 = mirror / 1 = ct / 2 = AoR */
modparam("mid_registrar", "outgoing_expires", 3600)
modparam("mid_registrar", "insertion_mode", 0) /* 0 = contact; 1 = path */
modparam("mid_registrar", "max_contacts", 1)
modparam("mid_registrar", "retry_after", 30)
modparam("mid_registrar", "contact_match_param", "rid")

### Routing Logic 

# main request routing logic

route{

if (method=="OPTIONS")
{
sl_send_reply("200", "OK");
exit;
};
if (method=="PUBLISH")
{
exit;
};
if (method=="SUBSCRIBE")
{
sl_send_reply("404", "Not Found");
exit;
};

if (!mf_process_maxfwd_header("40")) {
sl_send_reply("483","Too Many Hops");
exit;
  }


if (is_method("REGISTER")) {
  mid_registrar_save("location");
  switch ($retcode) {
  case 1:
xlog("L_INFO", "forwarding REGISTER to main registrar...\n");
$ru = "sip:192.168.10.201:5070 ";
if (!t_relay()) {
  send_reply("500", "Server Internal Error 1");
}

break;
  case 2:
xlog("L_INFO", "REGISTER has been absorbed!\n");
break;
  default:
xlog("L_ERR", "mid-registrar error!\n");
send_reply("500", "Server Internal Error 2");
  }

  exit;
}






if (is_method("INVITE|MESSAGE") && $si == "192.168.10.201" && $sp == 
5070) {

   xlog("looking up $ru!\n");
  if (!mid_registrar_lookup("location")) {
t_reply("404", "Not Found");
exit;
}

   t_relay();

exit;
}

}

Apr 26 12:58:11 [7524] DBG:core:parse_msg:  method:  
Apr 26 12:58:11 [7524] DBG:core:parse_msg:  uri: 


Apr 26 12:58:11 [7524] DBG:core:parse_msg:  version: 
Apr 26 12:58:11 [7524] DBG:core:parse_headers: flags=2
Apr 26 12:58:11 [7524] DBG:core:parse_via_param: found param type 232, 
 = ; state=16

Apr 26 12:58:11 [7524] DBG:core:parse_via: end of header reached, state=5
Apr 26 12:58:11 [7524] DBG:core:parse_headers: via found, flags=2
Apr 26 12:58:11 [7524] DBG:core:parse_headers: this is the first via
Apr 26 12:58:11 [7524] DBG:core:receive_msg: After parse_msg...
Apr 26 12:58:11 [7524] DBG:core:receive_msg: preparing to run routing 
scripts...

Apr 26 12:58:11 [7524] DBG:core:parse_headers: flags=100
Apr 26 12:58:11 [7524] DBG:maxfwd:is_maxfwd_present: value = 70
Apr 26 12:58:11 [7524] DBG:mid_registrar:mid_reg_save: saving to 
location...

Apr 26 12:58:11 [7517] DBG:core:handle_sigs: status = 139
Apr 26 12:58:11 [7517] INFO:core:handle_sigs: child process 7524 
exited by a signal 11

Apr 26 12:58:11 [7517] INFO:core:handle_sigs: core was generated
Apr 26 12:58:11 [7517] INFO:core:handle_sigs: terminating due to SIGCHLD
Apr 26 12:58:11 [7518] INFO:core:sig_usr: signal 15 received
Apr 26 12:58:11 [7519] INFO:core:sig_usr: signal 15 received
Apr 26 12:58:11 [7520] INFO:core:sig_usr: signal 15 received
Apr 26 12:58:11 [7521] INFO:core:sig_usr: signal 15 received
Apr 26 12:58:11 [7522] INFO:core:sig_usr: signal 15 received
Apr 26 12:58:11 [7523] INFO:core:sig_usr: signal 15 received
Apr 26 12:58:11 [7525] INFO:core:sig_usr: signal 15 received
Apr 26 12:58:11 [7517] INFO:core:cleanup: cleanup
Apr 26 12:58:11 [7517] DBG:core:pool_remove: removing connection from 
the pool
Apr 26 12:58:11 [7517] DBG:db_postgres:db_postgres_free_connection: 
PQfinish(0x1bd8980)
Apr 26 12:58:11 [7517] DBG:db_postgres:db_postgres_free_connection: 
pkg_free(0x7fbfc037fb68)

Apr 26 12:58:11 [7517] DBG:uac_auth:mod_destroy: done
Apr 26 12:58:11 [7517] DBG:tm:tm_shutdown: tm_shutdown : start
Apr 26 12:58:11 [7517] DBG:tm:unlink_timer_lists: emptying DELETE list 
for set 0

Apr 26 12:58:11 [7517] DBG:tm:tm_shutdown: emptying hash table
Apr 26 12:58:11 [7517] DBG:tm:tm_shutdown: releasing timers
Apr 26 12:58:11 [7517] DBG:tm:tm_shutdown: removing semaphores
Apr 26 12:58:11 [7517] DBG:tm:tm_shutdown: destroying callback lists
Apr 26 12:58:11 [7517] DBG:tm:tm_shutdown: tm_shutdown : done
Apr 26 12:58:11 [7517] DBG:core:shm_mem_destroy: destroying the shared 
memory lock

Apr 26 12:58:11 [7517] DBG:core:handle_sigs: terminating due to SIGCHLD

gdb dump file:
https://gist.github.com/anonymous/351c2feb6907c697554ede46f18b0f55


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Re: [OpenSIPS-Users] BLF with only one domain?

[Bogdan-Andrei Iancu]

Hi,

That param will not help you  (if I correctly understand your scenario). 
Even if you leverage all the presentities at subscription time, what 
about NOTIFY time. Here is the my case:

* A@dom1 subscribers to B@dom1
* in opensips, in SUBSCRIBE, you change B@dom1 into B@dom2
* when you have a call involving B@dom2, a NOTIFY with B@dom2 will 
be fired
* A@dom1 receives a NOTIFY for B@dom2, for which A never subscribed 
?!?!


Regards,

Bogdan-Andrei Iancu
  OpenSIPS Founder and Developer
  http://www.opensips-solutions.com

OpenSIPS Summit May 2017 Amsterdam
  http://www.opensips.org/events/Summit-2017Amsterdam.html

On 04/25/2017 12:02 PM, maatohewetbi wrote:

I've realized that bla_presentity_spec (str) should do the trick:

modparam("presence", "bla_presentity_spec", "$var(bla_pres)")

I could set this module, and set a variable ($var(bla_pres)) to
$var(bla_pres) = "sip:" + $au + "@" + "192.168.0.111";
burt active_watchers table still show auto-generated values. Why is that?
Did I miss something?



--
View this message in context: 
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Re: [OpenSIPS-Users] SIP SDP rfc1918 address fix

[Bogdan-Andrei Iancu]

Hi Satish,

For the mime test, you can use the has_body() function:
http://www.opensips.org/html/docs/modules/2.2.x/sipmsgops.html#idp3886992

About the error - could you post the actual SDP payload generating those 
errors ?


Regards,

Bogdan-Andrei Iancu
  OpenSIPS Founder and Developer
  http://www.opensips-solutions.com

OpenSIPS Summit May 2017 Amsterdam
  http://www.opensips.org/events/Summit-2017Amsterdam.html

On 04/25/2017 10:35 PM, Satish Patel wrote:

We have some custome Voice solution and in-house media server so right
now i don't care about PORT all i need correct IP address.

I have tried following and it fixed issue but i am seeing following
error in logs

if (method=="INVITE") {
 if(search("^Content-Type:.*application/sdp")) {
 fix_nated_sdp("3");
 };
};


Error:

ERROR: extract_mediaip: no `c=' in SDP
ERROR: extract_mediaip: no `c=' in SDP

Do you know what does that means and how to fix that issue?

On Mon, Apr 24, 2017 at 11:41 PM, Alex Balashov
 wrote:

The intent of my questions was to get what you think about what you
actually want to accomplish. fix_nated_sdp() allows you to replace the
IP with the received signalling IP:

http://www.opensips.org/html/docs/modules/2.1.x/nathelper.html#id293899

But what about the port?

On Mon, Apr 24, 2017 at 11:39:14PM -0400, Satish Patel wrote:


after google found bunch of post where people suggesting use
fix_nated_sdp()  is that right approach ?

On Mon, Apr 24, 2017 at 11:25 PM, Alex Balashov
 wrote:

Yes, but RTP can come from a different address than the signalling
(SIP). What sense would there be in substituting the source of the SIP
message in there?

On Mon, Apr 24, 2017 at 11:23:30PM -0400, Satish Patel wrote:


I meant "origin public address of client"  if c line isn't public then
media never work.

c=IN IP4 192.168.1.8.

It should be

c=IN IP4 

On Mon, Apr 24, 2017 at 11:04 PM, Alex Balashov
 wrote:

Satish,

When you say "origin public address", do you mean the external source
address and port of the SIP message, or the incoming RTP stream?

On Mon, Apr 24, 2017 at 11:00:40PM -0400, Satish Patel wrote:


In my INVITE/SDP i am seeing sometime rfc1918 address which i want fix
and replace it with origin public address. ex

I am seeing following info in INVITE

v=0.
o=amsip 0 0 IN IP4 192.168.1.8.
s= .
c=IN IP4 192.168.1.8.
t=0 0.
m=audio 22530 RTP/AVP 127 111 0 101.

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Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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Re: [OpenSIPS-Users] opensips 2.3 crashed on user reg - mid_registrar

[Kirill Galinurov]

Hi I also have the same problem. Our config is:

loadmodule "mid_registrar.so"
modparam("mid_registrar", "mode", 1) /* 0 = mirror / 1 = ct / 2 = AoR */
modparam("mid_registrar", "outgoing_expires", 3600)
modparam("mid_registrar", "insertion_mode", 0) /* 0 = contact; 1 = path */
modparam("mid_registrar", "max_contacts", 1)
modparam("mid_registrar", "retry_after", 30)
modparam("mid_registrar", "contact_match_param", "rid")

### Routing Logic 

# main request routing logic

route{

if (method=="OPTIONS")
{
sl_send_reply("200", "OK");
exit;
};
if (method=="PUBLISH")
{
exit;
};
if (method=="SUBSCRIBE")
{
sl_send_reply("404", "Not Found");
exit;
};

if (!mf_process_maxfwd_header("40")) {
sl_send_reply("483","Too Many Hops");
exit;
  }


if (is_method("REGISTER")) {
  mid_registrar_save("location");
  switch ($retcode) {
  case 1:
xlog("L_INFO", "forwarding REGISTER to main registrar...\n");
$ru = "sip:192.168.10.201:5070";
if (!t_relay()) {
  send_reply("500", "Server Internal Error 1");
}

break;
  case 2:
xlog("L_INFO", "REGISTER has been absorbed!\n");
break;
  default:
xlog("L_ERR", "mid-registrar error!\n");
send_reply("500", "Server Internal Error 2");
  }

  exit;
}






if (is_method("INVITE|MESSAGE") && $si == "192.168.10.201" && $sp == 5070) {
   xlog("looking up $ru!\n");
  if (!mid_registrar_lookup("location")) {
t_reply("404", "Not Found");
exit;
}

   t_relay();

exit;
}

}

Apr 26 12:58:11 [7524] DBG:core:parse_msg:  method:  
Apr 26 12:58:11 [7524] DBG:core:parse_msg:  uri:

Apr 26 12:58:11 [7524] DBG:core:parse_msg:  version: 
Apr 26 12:58:11 [7524] DBG:core:parse_headers: flags=2
Apr 26 12:58:11 [7524] DBG:core:parse_via_param: found param type 232,
 = ; state=16
Apr 26 12:58:11 [7524] DBG:core:parse_via: end of header reached, state=5
Apr 26 12:58:11 [7524] DBG:core:parse_headers: via found, flags=2
Apr 26 12:58:11 [7524] DBG:core:parse_headers: this is the first via
Apr 26 12:58:11 [7524] DBG:core:receive_msg: After parse_msg...
Apr 26 12:58:11 [7524] DBG:core:receive_msg: preparing to run routing
scripts...
Apr 26 12:58:11 [7524] DBG:core:parse_headers: flags=100
Apr 26 12:58:11 [7524] DBG:maxfwd:is_maxfwd_present: value = 70
Apr 26 12:58:11 [7524] DBG:mid_registrar:mid_reg_save: saving to location...
Apr 26 12:58:11 [7517] DBG:core:handle_sigs: status = 139
Apr 26 12:58:11 [7517] INFO:core:handle_sigs: child process 7524 exited by
a signal 11
Apr 26 12:58:11 [7517] INFO:core:handle_sigs: core was generated
Apr 26 12:58:11 [7517] INFO:core:handle_sigs: terminating due to SIGCHLD
Apr 26 12:58:11 [7518] INFO:core:sig_usr: signal 15 received
Apr 26 12:58:11 [7519] INFO:core:sig_usr: signal 15 received
Apr 26 12:58:11 [7520] INFO:core:sig_usr: signal 15 received
Apr 26 12:58:11 [7521] INFO:core:sig_usr: signal 15 received
Apr 26 12:58:11 [7522] INFO:core:sig_usr: signal 15 received
Apr 26 12:58:11 [7523] INFO:core:sig_usr: signal 15 received
Apr 26 12:58:11 [7525] INFO:core:sig_usr: signal 15 received
Apr 26 12:58:11 [7517] INFO:core:cleanup: cleanup
Apr 26 12:58:11 [7517] DBG:core:pool_remove: removing connection from the
pool
Apr 26 12:58:11 [7517] DBG:db_postgres:db_postgres_free_connection:
PQfinish(0x1bd8980)
Apr 26 12:58:11 [7517] DBG:db_postgres:db_postgres_free_connection:
pkg_free(0x7fbfc037fb68)
Apr 26 12:58:11 [7517] DBG:uac_auth:mod_destroy: done
Apr 26 12:58:11 [7517] DBG:tm:tm_shutdown: tm_shutdown : start
Apr 26 12:58:11 [7517] DBG:tm:unlink_timer_lists: emptying DELETE list for
set 0
Apr 26 12:58:11 [7517] DBG:tm:tm_shutdown: emptying hash table
Apr 26 12:58:11 [7517] DBG:tm:tm_shutdown: releasing timers
Apr 26 12:58:11 [7517] DBG:tm:tm_shutdown: removing semaphores
Apr 26 12:58:11 [7517] DBG:tm:tm_shutdown: destroying callback lists
Apr 26 12:58:11 [7517] DBG:tm:tm_shutdown: tm_shutdown : done
Apr 26 12:58:11 [7517] DBG:core:shm_mem_destroy: destroying the shared
memory lock
Apr 26 12:58:11 [7517] DBG:core:handle_sigs: terminating due to SIGCHLD

gdb dump file:
https://gist.github.com/anonymous/351c2feb6907c697554ede46f18b0f55
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Re: [OpenSIPS-Users] Fwd: websocket recover after opensips restart

[Răzvan Crainea]

Are you sure t_relay() is not returning an error? Can you post somewhere 
the entire log for this scenario?


Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 04/26/2017 12:00 PM, Mikhail Sidorov wrote:

I am sorry for spamming you, I missed subscription verification email.

Regarding my question, I understand that opensips could not recover 
connection and my client already reconnect.
All I need is ability to catch this error to execute lookup and find 
reconnected client.
And I could not find how to detect this kind of failure, because 
t_relay does not return error.

Do you have any suggestions?

ср, 26 апр. 2017 г. в 15:49, Răzvan Crainea >:


Hi, Mikhail!

I've already replied to your question[1]. Please register to the
opensips list if you want to get further emails.

[1] http://lists.opensips.org/pipermail/users/2017-April/037179.html

Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com 

On 04/26/2017 11:05 AM, Mikhail Sidorov wrote:

Hi

I am using opensips 2.2.3 to handle websocket connections from
web client (sip.js).
I just started testing recover after opensips restart and find
that I could not handle t_relay() failure.

Assume I have a call between 2 webrtc clients and I restart
opensips during this call. Media handled by freeswitch, so call
continue without problem.

Then, one client send BYE. I use loose_route to find another
party and send him message.
But here I got error that I could not handle.
t_on_failure trigger does not fire
t_relay return success.

Are there any special flags or triggers to handle this case?

Opensips log:

loose_route success
INFO:core:probe_max_sock_buff: using snd buffer of 416 kb
INFO:core:init_sock_keepalive: TCP keepalive enabled on socket 16
ERROR:core:tcp_connect_blocking: poll error: flags 28 - 4 8 16 32
ERROR:core:tcp_connect_blocking: failed to retrieve SO_ERROR
[server=172.19.235.225:62594 ] (111)
Connection refused
ERROR:proto_wss:ws_sync_connect: tcp_blocking_connect failed
ERROR:proto_wss:ws_connect: connect failed
ERROR:proto_wss:proto_wss_send: connect failed
ERROR:tm:msg_send: send() for proto 6 failed
ERROR:tm:t_forward_nonack: sending request failed

My configuration:

if (has_totag())
{
xlog("L_ERR", "Following ToTag: $tt $proto");
# sequential requests within a dialog should
# take the path determined by record-routing
if (loose_route())
{
#if (is_method("INVITE"))
#{
#record_route();
#}
xlog("L_ERR", "loose_route success\n");
route(relay);
}

route[relay]
{
t_on_failure("loose_route");
if (!t_relay())
{
xlog("L_ERR", "t_relay error]");
send_reply("500","Internal Error");
}
exit;
}

failure_route[loose_route]
{
xlog( "L_ERR", "loose_route failed" );
...
}




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Re: [OpenSIPS-Users] Fwd: websocket recover after opensips restart

[Mikhail Sidorov]

I am sorry for spamming you, I missed subscription verification email.

Regarding my question, I understand that opensips could not recover
connection and my client already reconnect.
All I need is ability to catch this error to execute lookup and find
reconnected client.
And I could not find how to detect this kind of failure, because t_relay
does not return error.
Do you have any suggestions?

ср, 26 апр. 2017 г. в 15:49, Răzvan Crainea :

> Hi, Mikhail!
>
> I've already replied to your question[1]. Please register to the opensips
> list if you want to get further emails.
>
> [1] http://lists.opensips.org/pipermail/users/2017-April/037179.html
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Solutionswww.opensips-solutions.com
>
> On 04/26/2017 11:05 AM, Mikhail Sidorov wrote:
>
> Hi
>
> I am using opensips 2.2.3 to handle websocket connections from web client
> (sip.js).
> I just started testing recover after opensips restart and find that I
> could not handle t_relay() failure.
>
> Assume I have a call between 2 webrtc clients and I restart opensips
> during this call. Media handled by freeswitch, so call continue without
> problem.
>
> Then, one client send BYE. I use loose_route to find another party and
> send him message.
> But here I got error that I could not handle.
> t_on_failure trigger does not fire
> t_relay return success.
>
> Are there any special flags or triggers to handle this case?
>
> Opensips log:
>
> loose_route success
> INFO:core:probe_max_sock_buff: using snd buffer of 416 kb
> INFO:core:init_sock_keepalive: TCP keepalive enabled on socket 16
> ERROR:core:tcp_connect_blocking: poll error: flags 28 - 4 8 16 32
> ERROR:core:tcp_connect_blocking: failed to retrieve SO_ERROR [server=
> 172.19.235.225:62594] (111) Connection refused
> ERROR:proto_wss:ws_sync_connect: tcp_blocking_connect failed
> ERROR:proto_wss:ws_connect: connect failed
> ERROR:proto_wss:proto_wss_send: connect failed
> ERROR:tm:msg_send: send() for proto 6 failed
> ERROR:tm:t_forward_nonack: sending request failed
>
> My configuration:
>
> if (has_totag())
> {
> xlog("L_ERR", "Following ToTag: $tt $proto");
> # sequential requests within a dialog should
> # take the path determined by record-routing
> if (loose_route())
> {
> #if (is_method("INVITE"))
> #{
> # record_route();
> #}
> xlog("L_ERR", "loose_route success\n");
> route(relay);
> }
> 
> route[relay]
> {
> t_on_failure("loose_route");
> if (!t_relay())
> {
> xlog("L_ERR", "t_relay error]");
> send_reply("500","Internal Error");
> }
> exit;
> }
>
> failure_route[loose_route]
> {
> xlog( "L_ERR", "loose_route failed" );
> ...
> }
>
>
>
>
> ___
> Users mailing 
> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
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Re: [OpenSIPS-Users] Fwd: websocket recover after opensips restart

[Răzvan Crainea]

Hi, Mikhail!

I've already replied to your question[1]. Please register to the 
opensips list if you want to get further emails.


[1] http://lists.opensips.org/pipermail/users/2017-April/037179.html

Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 04/26/2017 11:05 AM, Mikhail Sidorov wrote:

Hi

I am using opensips 2.2.3 to handle websocket connections from web 
client (sip.js).
I just started testing recover after opensips restart and find that I 
could not handle t_relay() failure.


Assume I have a call between 2 webrtc clients and I restart opensips 
during this call. Media handled by freeswitch, so call continue 
without problem.


Then, one client send BYE. I use loose_route to find another party and 
send him message.

But here I got error that I could not handle.
t_on_failure trigger does not fire
t_relay return success.

Are there any special flags or triggers to handle this case?

Opensips log:

loose_route success
INFO:core:probe_max_sock_buff: using snd buffer of 416 kb
INFO:core:init_sock_keepalive: TCP keepalive enabled on socket 16
ERROR:core:tcp_connect_blocking: poll error: flags 28 - 4 8 16 32
ERROR:core:tcp_connect_blocking: failed to retrieve SO_ERROR 
[server=172.19.235.225:62594 ] (111) 
Connection refused

ERROR:proto_wss:ws_sync_connect: tcp_blocking_connect failed
ERROR:proto_wss:ws_connect: connect failed
ERROR:proto_wss:proto_wss_send: connect failed
ERROR:tm:msg_send: send() for proto 6 failed
ERROR:tm:t_forward_nonack: sending request failed

My configuration:

if (has_totag())
{
xlog("L_ERR", "Following ToTag: $tt $proto");
# sequential requests within a dialog should
# take the path determined by record-routing
if (loose_route())
{
#if (is_method("INVITE"))
#{
#record_route();
#}
xlog("L_ERR", "loose_route success\n");
route(relay);
}

route[relay]
{
t_on_failure("loose_route");
if (!t_relay())
{
xlog("L_ERR", "t_relay error]");
send_reply("500","Internal Error");
}
exit;
}

failure_route[loose_route]
{
xlog( "L_ERR", "loose_route failed" );
...
}




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[OpenSIPS-Users] Fwd: websocket recover after opensips restart

[Mikhail Sidorov]

Hi

I am using opensips 2.2.3 to handle websocket connections from web client
(sip.js).
I just started testing recover after opensips restart and find that I could
not handle t_relay() failure.

Assume I have a call between 2 webrtc clients and I restart opensips during
this call. Media handled by freeswitch, so call continue without problem.

Then, one client send BYE. I use loose_route to find another party and send
him message.
But here I got error that I could not handle.
t_on_failure trigger does not fire
t_relay return success.

Are there any special flags or triggers to handle this case?

Opensips log:

loose_route success
INFO:core:probe_max_sock_buff: using snd buffer of 416 kb
INFO:core:init_sock_keepalive: TCP keepalive enabled on socket 16
ERROR:core:tcp_connect_blocking: poll error: flags 28 - 4 8 16 32
ERROR:core:tcp_connect_blocking: failed to retrieve SO_ERROR [server=
172.19.235.225:62594] (111) Connection refused
ERROR:proto_wss:ws_sync_connect: tcp_blocking_connect failed
ERROR:proto_wss:ws_connect: connect failed
ERROR:proto_wss:proto_wss_send: connect failed
ERROR:tm:msg_send: send() for proto 6 failed
ERROR:tm:t_forward_nonack: sending request failed

My configuration:

if (has_totag())
{
xlog("L_ERR", "Following ToTag: $tt $proto");
# sequential requests within a dialog should
# take the path determined by record-routing
if (loose_route())
{
#if (is_method("INVITE"))
#{
# record_route();
#}
xlog("L_ERR", "loose_route success\n");
route(relay);
}

route[relay]
{
t_on_failure("loose_route");
if (!t_relay())
{
xlog("L_ERR", "t_relay error]");
send_reply("500","Internal Error");
}
exit;
}

failure_route[loose_route]
{
xlog( "L_ERR", "loose_route failed" );
...
}
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Re: [OpenSIPS-Users] NOTIFY in dialog

[Răzvan Crainea]

Hi, Volga!

If the NOTIFY is in dialog, you should route it using loose_route(). The 
default script should already do this.


Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 04/26/2017 07:58 AM, volga...@networklab.ca wrote:

Hello Everyone,
How to handle NOTIFY in dialog  when pbx return 202 OK from SUBSCRIBE 
and NOTIFY is not routed back to client.

Please see attached trace or fpaste link.

https://paste.fedoraproject.org/paste/zMzLoYIl6bFlSCBuw~6u3l5M1UNdIGYhyRLivL9gydE= 




I tried

if(is_method("NOTIFY")) {
remove_hf("Route");
remove_hf("Record-Route");
$du = $ru;
}


volga629


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