[OpenSIPS-Users] UDP Fragmentation

2022-11-20 Thread Antonis Psaras 
Dear Team



I have a strange issue and I need you thoughts.



We have an installation with 1700 devices registered on an OpenSIPs 2.4 latest 
running on Centos 7 on a XenServer Virtualization. For some reason, some 
replies from OpenSIPs to the devices is fragmented to 48b. Is that something 
that controlled by OpenSIPs or OS? Why could something like that be occurred?



Example of normal packet



09:43:12.375943 IP 172.21.111.10.5060 > 172.24.4.14.5070: SIP: SIP/2.0 200 OK

EY@.@.o

SIP/2.0 200 OK

Via: SIP/2.0/UDP 
172.24.4.14:5070;received=172.24.4.14;branch=z9hG4bKb65a2607229aaf9b5;rport=5070

From: "1110401" ;tag=199725f5b3

To: "1110401" 
;tag=4074.edeca69b8a0dee9664e1f1fc59897b8c

Call-ID: 450061df04238489

CSeq: 1782007243 REGISTER

Contact: ;q=1;expires=600

Content-Length: 0



And just after that other replies to other device fragmented



09:43:12.483483 IP 172.21.111.10.5060 > 10.122.2.71.5060: SIP

E.. ..@.@._...o



z.G..'.

09:43:12.483563 IP 172.21.111.10.5060 > 172.29.20.26.5070: SIP

E.. ..@.@.v...o

...t

09:43:12.483597 IP 172.21.111.10.5060 > 172.29.19.25.5080: SIP

E.. ..@.@..Y..o

...sONS sip:172.21.1

09:43:12.483632 IP 172.21.111.10.5060 > 10.122.7.44.48577: SIP

E.. ..@.@.o



z.,..,.2.0 200 OK

Via:





Thank you for your thoughts in advance



Antonis Psaras





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Re: [OpenSIPS-Users] can not get uac_redirect to work

2022-11-20 Thread Bogdan-Andrei Iancu 

Perfect, glad it solved the issue for you!

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
OpenSIPS Bootcamp 5-16 Dec 2022, online
  https://www.opensips.org/training/OpenSIPS_eBootcamp_2022/

On 11/18/22 10:50 PM, Babak Yakhchali wrote:

Oh!
Thanks man. That was the reason! I removed it and it is working as 
expected.



On Fri, Nov 18, 2022 at 1:56 PM Bogdan-Andrei Iancu 
mailto:bog...@opensips.org>> wrote:


Hi Babak,

Are you somehow doing any fix_nated_contact() for that 302 reply??
(maybe in the onreply_route)

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com  
OpenSIPS Bootcamp 5-16 Dec 2022, online
   https://www.opensips.org/training/OpenSIPS_eBootcamp_2022/  


On 11/15/22 3:02 PM, Babak Yakhchali wrote:

Hi
I'm tryin to handle 302 redirect replies to opensips, In my
failure route I have this:

if (t_check_status("30[12]") && $(hdr(X-xferByCharger))==
"true") {

      xlog("failure_route: redirect by charger to
$(hdr(Contact))");
      

      if (get_redirects(1,1)){
          xlog("failure_route: after redirect ru:$ru and du:$du");
           t_relay();
      }
        exit;
}

and the 302 msg is:
2022/11/15 15:29:25.253868 10.0.0.82:5060 
-> 10.0.0.192:5060 
SIP/2.0 302 Moved Temporarily
Via: SIP/2.0/UDP 10.0.0.192:5060;branch=z9hG4bK4897.63b48a64.0
Via: SIP/2.0/UDP
172.18.120.236:49276;received=80.191.36.252;rport=49276;branch=z9hG4bK14771
Max-Forwards: 69
From: "" mailto:sip%3a123...@my-domain.com>>;tag=5447
To: http://my-domain.com:5060>>;tag=6Dyya5c1yvpQK
Call-ID: 21726
CSeq: 20 INVITE
*Contact: "unknown" mailto:sip%3A123456@10.0.0.109>>*
User-Agent:
FreeSWITCH-mod_sofia/1.9.0+git~20190122T161705Z~5ac757ce54~64bit
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Reason: Q.850;cause=31;text="NORMAL_UNSPECIFIED"
Content-Length: 0

and opensips logs:
Nov 15 15:52:12 lta-opensips-stage /usr/sbin/opensips[40099]:
DBG:uac_redirect:get_redirect: resume branch=0
Nov 15 15:52:12 lta-opensips-stage /usr/sbin/opensips[40099]:
DBG:uac_redirect:get_redirect: checking branch=0 (added=0)
Nov 15 15:52:12 lta-opensips-stage /usr/sbin/opensips[40099]:
DBG:uac_redirect:get_redirect: branch=0 is a redirect (added=0)
Nov 15 15:52:12 lta-opensips-stage /usr/sbin/opensips[40099]:
DBG:core:parse_headers: flags=
*Nov 15 15:52:12 lta-opensips-stage /usr/sbin/opensips[40099]:
DBG:uac_redirect:sort_contacts: sort_contacts:
http://sip:123456@10.0.0.82:5060>> q=10*
Nov 15 15:52:12 lta-opensips-stage /usr/sbin/opensips[40099]:
DBG:uac_redirect:shmcontact2dset: 1 contacts remaining after
filtering and sorting
Nov 15 15:52:12 lta-opensips-stage /usr/sbin/opensips[40099]:
DBG:uac_redirect:shmcontact2dset: adding contact
http://sip:123456@10.0.0.82:5060>>
Nov 15 15:52:12 lta-opensips-stage /usr/sbin/opensips[40099]:
DBG:core:pv_get_dsturi: no destination URI
Nov 15 15:52:12 lta-opensips-stage /usr/sbin/opensips[40099]:
DBG:tracer:is_id_traced: trace=on dyn=off
Nov 15 15:52:12 lta-opensips-stage /usr/sbin/opensips[40099]:
DBG:tracer:is_id_traced: trace=on dyn=off
Nov 15 15:52:12 lta-opensips-stage /usr/sbin/opensips[40099]:
DBG:tracer:sip_context_trace_impl: name hep_tid, hash 1710534437,
type 0, traceable on
Nov 15 15:52:12 lta-opensips-stage /usr/sbin/opensips[40099]:
DBG:proto_hep:add_hep_chunk: Chunk with (id=17; vendor=0) not
found! Creating!
Nov 15 15:52:12 lta-opensips-stage /usr/sbin/opensips[40099]:
DBG:proto_hep:add_hep_chunk: Hep chunk with (id=17; vendor=0)
successfully built!
Nov 15 15:52:12 lta-opensips-stage /usr/sbin/opensips[40099]:
DBG:core:mk_proxy: doing DNS lookup...
Nov 15 15:52:12 lta-opensips-stage /usr/sbin/opensips[40099]:
failure_route: after redirect *ru:sip:123456@10.0.0.82:5060
 and du:*

Reading uac_redirect docs I expect that Contact header to be used
as a new destination but the original ru is used again!


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Re: [OpenSIPS-Users] relay invites UAC

2022-11-20 Thread Wadii ELMAJDI | Evenmedia 
I know i can route calls to any sip uri , i'm just trying to find the most 
suitable solution.

Its a usecase i never faced before, on which the sip trunk provider requires a 
registration and also challenging every outbound invite.

Do you confirm it wouldnt be a bad idea running a registration using 
uac_registrant module. which will allow me to receive calls from the sip 
provider endpoint.  At the same time add their host/ip as a gateway in order to 
route outbound calls to this same exact provider using DR module ?


De : Bogdan-Andrei Iancu 
Envoyé : vendredi 18 novembre 2022 11:27
À : OpenSIPS users mailling list ; Wadii ELMAJDI | 
Evenmedia 
Objet : Re: [OpenSIPS-Users] relay invites UAC

Hi Wadii,

You can route the call to any SIP URI you want :)

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
OpenSIPS Bootcamp 5-16 Dec 2022, online
  https://www.opensips.org/training/OpenSIPS_eBootcamp_2022/

On 11/15/22 10:14 PM, Wadii ELMAJDI | Evenmedia wrote:
Hello Everyone,

I am using opensips as sip proxy, receiving invites from my clients (mostly 
b2bua), that i should relay to some gateways.
I am using Dynamic route module for such purpose.
Some of those invites (depending on called DID or gateways probing) should be 
relayed to a sip trunk service provider.
My provider requires a registration, which i did using uac registrant module.
My question, is it possible to route calls to my sip provider knowing it is 
neither a Gateway, nor a subscriber ?

Thank you



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