date:20160422

Re: [OpenSIPS-Users] BYE generated locally going to private IP, not public IP

2016-04-22 Thread Daniel Zanutti

That makes a lot of sense.

Do you think is there anything I could do on my side to solve this? Maybe
manually fix the RR?

This client has some kind of link balancing, so public IP may change at any
time. This must be the reason he hasn't set it.

Thanks in advance!
Em 22/04/2016 7:40 PM, "Bogdan-Andrei Iancu"  escreveu:

> Hi Daniel,
>
> Actually, on the A side, you have another proxy ( see the Record Route
> with 172.20.17.11 in the INVITE). OpenSIPS tries to send the BYE to the RR
> header, but that is private. A SIP proxy, if sending traffic to public
> Internet, should not use at all private IPs.
>
> Bottom line, the broken link in your scenario is the 172.20.17.11 proxy
> before your opensips.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 23.04.2016 00:12, Daniel Zanutti wrote:
>
> Hi
>
> I'm facing an strange issue when my Opensips instance hangs up a call,
> generating BYE to both sides (timeout on dialog module or rtpproxy). The
> BYE is sent to both sides but A side is behind NAT and the BYE is sent to
> the local IP address and not to the public one.
>
> See trace bellow:
>
> Customer -> Opensips
>
> *U 200.200.200.200:27923  ->
> 199.199.199.199:5060 *
> INVITE sip:55113...@plat.test.com SIP/2.0
> Record-Route: 
> Via: SIP/2.0/UDP 172.20.17.11:5060;branch=z9hG4bKb99a.3c0442d4.0
> Via: SIP/2.0/UDP 172.28.0.12:57744
> ;received=172.28.0.12;branch=z9hG4bK-524287-1---08559406a5e9137b;rport=57744
> Max-Forwards: 68
> Contact: 
> 
> To: 
> From: ;tag=7db6f42e
> Call-ID: 79961NjFjMjY2MDhiYjhhNDYyN2UyMWZiMzFiNWM5NzBkNTM
> CSeq: 2 INVITE
> Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS,
> MESSAGE
> Content-Type: application/sdp
> Supported: replaces
> User-Agent: X-Lite release 4.9.3 stamp 79961
> Content-Length: 333
> P-hint: NAT
>
>
> BYE Opensips -> Customer
>
> *U 199.199.199.199:5060  -> 172.20.17.11:5060
> *
> BYE sip:100111@200.200.200.200:27923;rinstance=b4a1be0f56d73cfd SIP/2.0
> Via: SIP/2.0/UDP 199.199.199.199:5060;branch=z9hG4bKe99a.e1c423a3.0
> To: ;tag=7db6f42e
> From: ;tag=as4088ffc9
> CSeq: 1 BYE
> Call-ID: 79961NjFjMjY2MDhiYjhhNDYyN2UyMWZiMzFiNWM5NzBkNTM
> Route: 
> Max-Forwards: 70
> Content-Length: 0
> User-Agent: Softswitch
>
>
> On the initial invite, I fixed the Contact using fix_nated_contact() and
> signalling works fine between A and B sides, the problem is happening when
> Opensips hangup the call, because A side doesnt receive the BYE.
>
> Do you guys have an idea on how to fix this? Maybe is it a bug?
>
> Thanks
>
>
>
>
>
> ___
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> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
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Re: [OpenSIPS-Users] issues with latest head logging

2016-04-22 Thread Tito Cumpen

Bogdan,


Here are the first lines from my config:


  1 #

   2 # OpenSIPS residential configuration script

   3 # by OpenSIPS Solutions 

   4 #

   5 # Please refer to the Core CookBook at:

   6 #  http://www.opensips.org/Resources/DocsCookbooks

   7 # for a explanation of possible statements, functions and parameters.

   8 #

   9

  10

  11 ### Global Parameters #

  12

  13 #debug=3

  14 #log_level=3

  15 #log_stderror=yes

  16

  17 log_facility=LOG_LOCAL0

  18 log_level=3

  19 log_stderror=yes

  20 #memlog=1

  21

  22 debug_mode=no

  23 children=4

  24 auto_aliases=no

  25 #disable_tcp=no

  26 tcp_connection_lifetime=3600

  27 tcp_connect_timeout=3

  28 tcp_keepidle = 30

  29 tcp_keepinterval = 5

  30 tcp_keepalive = 1

  31 tcp_keepcount = 5

  32 #tcp_max_msg_chunks=8

  33 tcp_max_msg_time = 8

  34 tcp_children=6

On Fri, Apr 22, 2016 at 6:42 PM, Bogdan-Andrei Iancu 
wrote:

> Hi Tito,
>
> what are the values for the debug_mode and for log_stderror ? And in what
> order to they pop into your script ?
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 22.04.2016 21:50, Tito Cumpen wrote:
>
> Liviu,
>
>
> I don't see any messages regarding opensips in /var/log/messages. I don't
> see an issue finding opensips.cfg. Since opensips is starting with all the
> changes I make to the config being applied. Here is my definition in
> rsyslog.conf
>
> local0.*/var/log/opensips
>
> this is the log facility definition in opensips config
>
>   log_facility=LOG_LOCAL0
>
>
> Previous versions/pulls of opensips didn't appear to have this issue. It
> has made itself apparent within the last month.
>
>
> Thanks,
>
> Tito
>
>
> On Fri, Apr 22, 2016 at 4:47 AM, Liviu Chircu  wrote:
>
>> Hi Tito,
>>
>> Did you find anything into /var/log/{syslog,messages}? A common error is
>> that the config file is not found ("-f /bad/path/opensips.cfg"), and
>> OpenSIPS will naturally throw this error to syslog, rather than the log
>> file you expect.
>>
>> Liviu Chircu
>> OpenSIPS Developerhttp://www.opensips-solutions.com
>>
>> On 21.04.2016 01:06, Tito Cumpen wrote:
>>
>> Group and devs,
>>
>> I am having issues getting any sort of opensips logs with the latest head
>> pull git revision: 6c10501
>> . I am running the new param log_level = 4 but I am seeing opensips log
>> anything into the specified log facility or the file. Is anyone else seeing
>> this issue?
>>
>>
>> ___
>> Users mailing 
>> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
>> ___
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>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
>
> ___
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>
>
>
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Re: [OpenSIPS-Users] 487 for SIP forking

2016-04-22 Thread Bogdan-Andrei Iancu


Hi Chen-Che,

Usually you need to have in the very beginning of your failure route:
if (t_was_cancelled()) {
exit;
}

Regards,

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

On 21.04.2016 01:12, Chen-Che Huang wrote:

Dear all,

When a caller cancels a call, OpenSIPS will send the 487 response sent from
the callee to the caller. However, in the case of SIP forking, I find that
OpenSIPS will not send the 487 response to the caller after the caller
cancels the call. Do I need any particular configuration for the SIP forking
case? Any comment is appreciated.

Best regards,
Chen-Che



--
View this message in context: 
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Re: [OpenSIPS-Users] issues with latest head logging

2016-04-22 Thread Bogdan-Andrei Iancu


Hi Tito,

what are the values for the debug_mode and for log_stderror ? And in 
what order to they pop into your script ?


Regards,

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

On 22.04.2016 21:50, Tito Cumpen wrote:

Liviu,


I don't see any messages regarding opensips in /var/log/messages. I 
don't see an issue finding opensips.cfg. Since opensips is starting 
with all the changes I make to the config being applied. Here is my 
definition in rsyslog.conf


local0.* /var/log/opensips

this is the log facility definition in opensips config

  log_facility=LOG_LOCAL0


Previous versions/pulls of opensips didn't appear to have this issue. 
It has made itself apparent within the last month.



Thanks,

Tito



On Fri, Apr 22, 2016 at 4:47 AM, Liviu Chircu > wrote:


Hi Tito,

Did you find anything into /var/log/{syslog,messages}? A common
error is that the config file is not found ("-f
/bad/path/opensips.cfg"), and OpenSIPS will naturally throw this
error to syslog, rather than the log file you expect.

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

On 21.04.2016 01:06, Tito Cumpen wrote:

Group and devs,

I am having issues getting any sort of opensips logs with the
latest head pull git revision: 6c10501
. I am running the new param log_level = 4 but I am seeing
opensips log anything into the specified log facility or the
file. Is anyone else seeing this issue?


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Re: [OpenSIPS-Users] BYE generated locally going to private IP, not public IP

2016-04-22 Thread Bogdan-Andrei Iancu


Hi Daniel,

Actually, on the A side, you have another proxy ( see the Record Route 
with 172.20.17.11 in the INVITE). OpenSIPS tries to send the BYE to the 
RR header, but that is private. A SIP proxy, if sending traffic to 
public Internet, should not use at all private IPs.


Bottom line, the broken link in your scenario is the 172.20.17.11 proxy 
before your opensips.


Regards,

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

On 23.04.2016 00:12, Daniel Zanutti wrote:

Hi

I'm facing an strange issue when my Opensips instance hangs up a call, 
generating BYE to both sides (timeout on dialog module or rtpproxy). 
The BYE is sent to both sides but A side is behind NAT and the BYE is 
sent to the local IP address and not to the public one.


See trace bellow:

Customer -> Opensips

*U 200.200.200.200:27923  -> 
199.199.199.199:5060 *
INVITE sip:55113...@plat.test.com 
 SIP/2.0

Record-Route: 
Via: SIP/2.0/UDP 172.20.17.11:5060;branch=z9hG4bKb99a.3c0442d4.0
Via: SIP/2.0/UDP 
172.28.0.12:57744;received=172.28.0.12;branch=z9hG4bK-524287-1---08559406a5e9137b;rport=57744

Max-Forwards: 68
Contact: 
To: >
From: >;tag=7db6f42e

Call-ID: 79961NjFjMjY2MDhiYjhhNDYyN2UyMWZiMzFiNWM5NzBkNTM
CSeq: 2 INVITE
Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, 
OPTIONS, MESSAGE

Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite release 4.9.3 stamp 79961
Content-Length: 333
P-hint: NAT


BYE Opensips -> Customer

*U 199.199.199.199:5060  -> 
172.20.17.11:5060 *

BYE sip:100111@200.200.200.200:27923;rinstance=b4a1be0f56d73cfd SIP/2.0
Via: SIP/2.0/UDP 199.199.199.199:5060;branch=z9hG4bKe99a.e1c423a3.0
To: >;tag=7db6f42e
From: >;tag=as4088ffc9

CSeq: 1 BYE
Call-ID: 79961NjFjMjY2MDhiYjhhNDYyN2UyMWZiMzFiNWM5NzBkNTM
Route: 
Max-Forwards: 70
Content-Length: 0
User-Agent: Softswitch


On the initial invite, I fixed the Contact using fix_nated_contact() 
and signalling works fine between A and B sides, the problem is 
happening when Opensips hangup the call, because A side doesnt receive 
the BYE.


Do you guys have an idea on how to fix this? Maybe is it a bug?

Thanks





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Re: [OpenSIPS-Users] Startup event

2016-04-22 Thread Bogdan-Andrei Iancu


Hi Tito,

You can use the startup route and do any kind of processing there 
(including firing events):

 http://www.opensips.org/Documentation/Script-Routes-2-1#toc7

Regards,

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

On 22.04.2016 23:58, Tito Cumpen wrote:

Group,

Is there a way to launch or subscribe to an opensips event on startup 
or shutdown? I am looking to do this so I can add active opensips 
servers to a management list or remove them when they are down. I want 
to make sure they are up before trying to access the http_fifo 
interface. The down state is easier to deal with since monit can be 
used to verify is a particular socket is down.


Thanks,
Tito


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[OpenSIPS-Users] BYE generated locally going to private IP, not public IP

2016-04-22 Thread Daniel Zanutti

Hi

I'm facing an strange issue when my Opensips instance hangs up a call,
generating BYE to both sides (timeout on dialog module or rtpproxy). The
BYE is sent to both sides but A side is behind NAT and the BYE is sent to
the local IP address and not to the public one.

See trace bellow:

Customer -> Opensips

*U 200.200.200.200:27923  ->
199.199.199.199:5060 *
INVITE sip:55113...@plat.test.com SIP/2.0
Record-Route: 
Via: SIP/2.0/UDP 172.20.17.11:5060;branch=z9hG4bKb99a.3c0442d4.0
Via: SIP/2.0/UDP 172.28.0.12:57744
;received=172.28.0.12;branch=z9hG4bK-524287-1---08559406a5e9137b;rport=57744
Max-Forwards: 68
Contact: 
To: 
From: ;tag=7db6f42e
Call-ID: 79961NjFjMjY2MDhiYjhhNDYyN2UyMWZiMzFiNWM5NzBkNTM
CSeq: 2 INVITE
Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS,
MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite release 4.9.3 stamp 79961
Content-Length: 333
P-hint: NAT


BYE Opensips -> Customer

*U 199.199.199.199:5060  -> 172.20.17.11:5060
*
BYE sip:100111@200.200.200.200:27923;rinstance=b4a1be0f56d73cfd SIP/2.0
Via: SIP/2.0/UDP 199.199.199.199:5060;branch=z9hG4bKe99a.e1c423a3.0
To: ;tag=7db6f42e
From: ;tag=as4088ffc9
CSeq: 1 BYE
Call-ID: 79961NjFjMjY2MDhiYjhhNDYyN2UyMWZiMzFiNWM5NzBkNTM
Route: 
Max-Forwards: 70
Content-Length: 0
User-Agent: Softswitch


On the initial invite, I fixed the Contact using fix_nated_contact() and
signalling works fine between A and B sides, the problem is happening when
Opensips hangup the call, because A side doesnt receive the BYE.

Do you guys have an idea on how to fix this? Maybe is it a bug?

Thanks
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[OpenSIPS-Users] Startup event

2016-04-22 Thread Tito Cumpen

Group,

Is there a way to launch or subscribe to an opensips event on startup or
shutdown? I am looking to do this so I can add active opensips servers to a
management list or remove them when they are down. I want to make sure they
are up before trying to access the http_fifo interface. The down state is
easier to deal with since monit can be used to verify is a particular
socket is down.

Thanks,
Tito
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Re: [OpenSIPS-Users] issues with latest head logging

2016-04-22 Thread Tito Cumpen

Liviu,


I don't see any messages regarding opensips in /var/log/messages. I don't
see an issue finding opensips.cfg. Since opensips is starting with all the
changes I make to the config being applied. Here is my definition in
rsyslog.conf

local0.*/var/log/opensips

this is the log facility definition in opensips config

  log_facility=LOG_LOCAL0


Previous versions/pulls of opensips didn't appear to have this issue. It
has made itself apparent within the last month.


Thanks,

Tito


On Fri, Apr 22, 2016 at 4:47 AM, Liviu Chircu  wrote:

> Hi Tito,
>
> Did you find anything into /var/log/{syslog,messages}? A common error is
> that the config file is not found ("-f /bad/path/opensips.cfg"), and
> OpenSIPS will naturally throw this error to syslog, rather than the log
> file you expect.
>
> Liviu Chircu
> OpenSIPS Developerhttp://www.opensips-solutions.com
>
> On 21.04.2016 01:06, Tito Cumpen wrote:
>
> Group and devs,
>
> I am having issues getting any sort of opensips logs with the latest head
> pull git revision: 6c10501
> . I am running the new param log_level = 4 but I am seeing opensips log
> anything into the specified log facility or the file. Is anyone else seeing
> this issue?
>
>
> ___
> Users mailing 
> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> ___
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> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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Re: [OpenSIPS-Users] issues with latest head logging

2016-04-22 Thread Liviu Chircu


Hi Tito,

Did you find anything into /var/log/{syslog,messages}? A common error is 
that the config file is not found ("-f /bad/path/opensips.cfg"), and 
OpenSIPS will naturally throw this error to syslog, rather than the log 
file you expect.


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

On 21.04.2016 01:06, Tito Cumpen wrote:

Group and devs,

I am having issues getting any sort of opensips logs with the latest 
head pull git revision: 6c10501
. I am running the new param log_level = 4 but I am seeing opensips 
log anything into the specified log facility or the file. Is anyone 
else seeing this issue?



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Re: [OpenSIPS-Users] event for expiring sessions

2016-04-22 Thread Bogdan-Andrei Iancu


Hi Tito,

There is no dedicated event in dialog module for expired calls. What you 
can do is to use the local_route to capture the local BYE requests (when 
dialog module terminates an expired dialog) - if is a BYE and direction 
is UPSTREAM (to get only one BYE per call), you can fire an event from 
the script ; this event may be captured by an external app to do 
whatever job.


Regards,

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

On 22.04.2016 00:55, Tito Cumpen wrote:

Group,

Is there anyway to raise an event for sessions that have expired. 
Meaning if a call/sessions expires an event is raised. I am trying to 
deal with the possibility of removing active sessions from a remote 
database in the event that a bye was never transmitted due to a client 
crashing or losing network connectivity.


Thanks,
Tito


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Re: [OpenSIPS-Users] Rely forbidden

2016-04-22 Thread Jeff Wilkie

Do you have a link to provide your config and confirm what modules are
compiled and loaded from install?  Receiving calls from outside to your
OPENSIPS should be straight forward.  I'm not sure how your RURI even
routes to you if the upstream provider is a real provider and not in a test
lab.  sip:opensips@localhost:5060 is very non specific and local.  I'm
assuming your trying to call to sip:0972550428@10.7.1.68.

Jeff Wilkie
Chief Technology Officer
US IP Communications
919.297.1057


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On Fri, Apr 15, 2016 at 6:03 AM, Francjos <35...@heb.be> wrote:

> Here is the output of :ngrep -d eth0 -t -W byline "$1" port 5060 -q
> Can you help me to understand the problem and how to resolve it?
>
> ^Croot@front-2:~# ngrep -d eth0 -t -W byline "$1" port 5060 -q
> interface: eth0 (10.0.0.0/255.255.255.0)
> filter: (ip or ip6) and ( port 5060 )
>
> U 2016/04/15 09:59:46.445723 91.121.129.159:5060 -> 10.0.0.5:5060
> INVITE sip:opensips@localhost:5060;transport=udp SIP/2.0.
> Call-ID: 02023-ii-1b75f5ca-75e715...@sip3.ovh.fr.
> Contact: .
> Content-Type: application/sdp.
> CSeq: 426731554 INVITE.
> From: "003228800555"
> ;tag=02023-LB-1b75f5cb-77f5c7f17.
> Max-Forwards: 27.
> Record-Route: .
> To: .
> Via: SIP/2.0/UDP 91.121.129.159:5060
> ;branch=z9hG4bK-UXTN-2667384e-065369b8.
> Allow:
>
> REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK.
> User-Agent: Cirpack/v4.70 (gw_sip).
> Content-Length: 315.
> .
> v=0.
> o=cp10 146071438652 146071438652 IN IP4 10.7.16.156.
> s=SIP Call.
> c=IN IP4 91.121.129.144.
> t=0 0.
> m=audio 31298 RTP/AVP 18 0 8 101.
> b=AS:21.
> a=rtpmap:18 G729/8000/1.
> a=fmtp:18 annexb=no.
> a=rtpmap:0 PCMU/8000/1.
> a=rtpmap:8 PCMA/8000/1.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
> a=ptime:30.
> a=sendrecv.
>
>
> U 2016/04/15 09:59:46.446223 10.0.0.5:5060 -> 91.121.129.159:5060
> SIP/2.0 500 Service full.
> Call-ID: 02023-ii-1b75f5ca-75e715...@sip3.ovh.fr.
> CSeq: 426731554 INVITE.
> From: "003228800555"
> ;tag=02023-LB-1b75f5cb-77f5c7f17.
> To:
>  ;user=phone>;tag=c97b4d1cb1f3d0da549e06a8d482ef63.57bb.
> Via: SIP/2.0/UDP 91.121.129.159:5060
> ;branch=z9hG4bK-UXTN-2667384e-065369b8.
> Server: OpenSIPS (2.1.2 (x86_64/linux)).
> Content-Length: 0.
> .
>
>
> U 2016/04/15 09:59:46.459264 91.121.129.159:5060 -> 10.0.0.5:5060
> ACK sip:opensips@localhost:5060;transport=udp SIP/2.0.
> Call-ID: 02023-ii-1b75f5ca-75e715...@sip3.ovh.fr.
> CSeq: 426731554 ACK.
> From: "003228800555"
> ;tag=02023-LB-1b75f5cb-77f5c7f17.
> Max-Forwards: 27.
> To:
>  ;user=phone>;tag=c97b4d1cb1f3d0da549e06a8d482ef63.57bb.
> Via: SIP/2.0/UDP 91.121.129.159:5060
> ;branch=z9hG4bK-UXTN-2667384e-065369b8.
> Content-Length: 0.
> .
>
>
> U 2016/04/15 09:59:48.514025 91.121.129.159:5060 -> 10.0.0.5:5060
> INVITE sip:opensips@localhost:5060;transport=udp SIP/2.0.
> Call-ID: 25873-ax-1b75f716-0d45b7...@sip3.ovh.fr.
> Contact: .
> Content-Type: application/sdp.
> CSeq: 426731844 INVITE.
> From: "003228800555"
> ;tag=25873-BW-1b75f717-40cd751b1.
> Max-Forwards: 27.
> Record-Route: .
> To: .
> Via: SIP/2.0/UDP 91.121.129.159:5060
> ;branch=z9hG4bK-ZRXL-26673a19-1004fd6f.
> Allow:
>
> REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK.
> User-Agent: Cirpack/v4.70 (gw_sip).
> Content-Length: 315.
> .
> v=0.
> o=cp10 146071438849 146071438849 IN IP4 10.7.16.156.
> s=SIP Call.
> c=IN IP4 91.121.129.145.
> t=0 0.
> m=audio 34728 RTP/AVP 18 0 8 101.
> b=AS:21.
> a=rtpmap:18 G729/8000/1.
> a=fmtp:18 annexb=no.
> a=rtpmap:0 PCMU/8000/1.
> a=rtpmap:8 PCMA/8000/1.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
> a=ptime:30.
> a=sendrecv.
>
>
> U 2016/04/15 09:59:48.514288 10.0.0.5:5060 -> 91.121.129.159:5060
> SIP/2.0 500 Service full.
> Call-ID: 25873-ax-1b75f716-0d45b7...@sip3.ovh.fr.
> CSeq: 426731844 INVITE.
> From: "003228800555"
> ;tag=25873-BW-1b75f717-40cd751b1.
> To:
>  ;user=phone>;tag=c97b4d1cb1f3d0da549e06a8d482ef63.9640.
> Via: SIP/2.0/UDP 91.121.129.159:5060
> ;branch=z9hG4bK-ZRXL-26673a19-1004fd6f.
> Server: OpenSIPS (2.1.2 (x86_64/linux)).
> Content-Length: 0.
> .
>
>
> U 2016/04/15 09:59:48.527361 91.121.129.159:5060 -> 10.0.0.5:5060
> ACK sip:opensips@localhost:5060;transport=udp S

Re: [OpenSIPS-Users] Rely forbidden

2016-04-22 Thread SamyGo

Did you modify something. Now the error from your opensips seem like 500
Service Full.
That usually is configured with load balancer. Means that it was unable to
find any active server to route the call to.
On Apr 15, 2016 06:04, "Francjos" <35...@heb.be> wrote:

> Here is the output of :ngrep -d eth0 -t -W byline "$1" port 5060 -q
> Can you help me to understand the problem and how to resolve it?
>
> ^Croot@front-2:~# ngrep -d eth0 -t -W byline "$1" port 5060 -q
> interface: eth0 (10.0.0.0/255.255.255.0)
> filter: (ip or ip6) and ( port 5060 )
>
> U 2016/04/15 09:59:46.445723 91.121.129.159:5060 -> 10.0.0.5:5060
> INVITE sip:opensips@localhost:5060;transport=udp SIP/2.0.
> Call-ID: 02023-ii-1b75f5ca-75e715...@sip3.ovh.fr.
> Contact: .
> Content-Type: application/sdp.
> CSeq: 426731554 INVITE.
> From: "003228800555"
> ;tag=02023-LB-1b75f5cb-77f5c7f17.
> Max-Forwards: 27.
> Record-Route: .
> To: .
> Via: SIP/2.0/UDP 91.121.129.159:5060
> ;branch=z9hG4bK-UXTN-2667384e-065369b8.
> Allow:
>
> REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK.
> User-Agent: Cirpack/v4.70 (gw_sip).
> Content-Length: 315.
> .
> v=0.
> o=cp10 146071438652 146071438652 IN IP4 10.7.16.156.
> s=SIP Call.
> c=IN IP4 91.121.129.144.
> t=0 0.
> m=audio 31298 RTP/AVP 18 0 8 101.
> b=AS:21.
> a=rtpmap:18 G729/8000/1.
> a=fmtp:18 annexb=no.
> a=rtpmap:0 PCMU/8000/1.
> a=rtpmap:8 PCMA/8000/1.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
> a=ptime:30.
> a=sendrecv.
>
>
> U 2016/04/15 09:59:46.446223 10.0.0.5:5060 -> 91.121.129.159:5060
> SIP/2.0 500 Service full.
> Call-ID: 02023-ii-1b75f5ca-75e715...@sip3.ovh.fr.
> CSeq: 426731554 INVITE.
> From: "003228800555"
> ;tag=02023-LB-1b75f5cb-77f5c7f17.
> To:
>  ;user=phone>;tag=c97b4d1cb1f3d0da549e06a8d482ef63.57bb.
> Via: SIP/2.0/UDP 91.121.129.159:5060
> ;branch=z9hG4bK-UXTN-2667384e-065369b8.
> Server: OpenSIPS (2.1.2 (x86_64/linux)).
> Content-Length: 0.
> .
>
>
> U 2016/04/15 09:59:46.459264 91.121.129.159:5060 -> 10.0.0.5:5060
> ACK sip:opensips@localhost:5060;transport=udp SIP/2.0.
> Call-ID: 02023-ii-1b75f5ca-75e715...@sip3.ovh.fr.
> CSeq: 426731554 ACK.
> From: "003228800555"
> ;tag=02023-LB-1b75f5cb-77f5c7f17.
> Max-Forwards: 27.
> To:
>  ;user=phone>;tag=c97b4d1cb1f3d0da549e06a8d482ef63.57bb.
> Via: SIP/2.0/UDP 91.121.129.159:5060
> ;branch=z9hG4bK-UXTN-2667384e-065369b8.
> Content-Length: 0.
> .
>
>
> U 2016/04/15 09:59:48.514025 91.121.129.159:5060 -> 10.0.0.5:5060
> INVITE sip:opensips@localhost:5060;transport=udp SIP/2.0.
> Call-ID: 25873-ax-1b75f716-0d45b7...@sip3.ovh.fr.
> Contact: .
> Content-Type: application/sdp.
> CSeq: 426731844 INVITE.
> From: "003228800555"
> ;tag=25873-BW-1b75f717-40cd751b1.
> Max-Forwards: 27.
> Record-Route: .
> To: .
> Via: SIP/2.0/UDP 91.121.129.159:5060
> ;branch=z9hG4bK-ZRXL-26673a19-1004fd6f.
> Allow:
>
> REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK.
> User-Agent: Cirpack/v4.70 (gw_sip).
> Content-Length: 315.
> .
> v=0.
> o=cp10 146071438849 146071438849 IN IP4 10.7.16.156.
> s=SIP Call.
> c=IN IP4 91.121.129.145.
> t=0 0.
> m=audio 34728 RTP/AVP 18 0 8 101.
> b=AS:21.
> a=rtpmap:18 G729/8000/1.
> a=fmtp:18 annexb=no.
> a=rtpmap:0 PCMU/8000/1.
> a=rtpmap:8 PCMA/8000/1.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
> a=ptime:30.
> a=sendrecv.
>
>
> U 2016/04/15 09:59:48.514288 10.0.0.5:5060 -> 91.121.129.159:5060
> SIP/2.0 500 Service full.
> Call-ID: 25873-ax-1b75f716-0d45b7...@sip3.ovh.fr.
> CSeq: 426731844 INVITE.
> From: "003228800555"
> ;tag=25873-BW-1b75f717-40cd751b1.
> To:
>  ;user=phone>;tag=c97b4d1cb1f3d0da549e06a8d482ef63.9640.
> Via: SIP/2.0/UDP 91.121.129.159:5060
> ;branch=z9hG4bK-ZRXL-26673a19-1004fd6f.
> Server: OpenSIPS (2.1.2 (x86_64/linux)).
> Content-Length: 0.
> .
>
>
> U 2016/04/15 09:59:48.527361 91.121.129.159:5060 -> 10.0.0.5:5060
> ACK sip:opensips@localhost:5060;transport=udp SIP/2.0.
> Call-ID: 25873-ax-1b75f716-0d45b7...@sip3.ovh.fr.
> CSeq: 426731844 ACK.
> From: "003228800555"
> ;tag=25873-BW-1b75f717-40cd751b1.
> Max-Forwards: 27.
> To:
>  ;user=phone>;tag=c97b4d1cb1f3d0da549e06a8d482ef63.9640.
> Via: SIP/2.0/UDP 91.121.129.159:5060
> ;branch=z9hG4bK-ZRXL-26673a19-1004fd6f.
> Content-Length: 0.
> .
>
>
> U 2016/04/15 09:59:50.821619 91.121.129.159:5060 -> 10.0.0.5:5060
> INVITE sip:opensips@localhost:5060;transport=udp SIP/2.0.
> Call-ID: 30481-qo-1b75f83d-7dfab0...@sip3.ovh.fr.
> Contact: .
> Content-Type: application/sdp.
> CSeq: 426732128 INVITE.
> From: "003228800555"
> ;tag=30481-QM-1b75f83e-2775f2177.
> Max-Forwards: 27.
> Record-Route: .
> To: .
> Via: SIP/2.0/UDP 91.121.129.159:5060
> ;branch=z9hG4bK-BGVQ-26673bc9-1067a939.
> Allow:
>
> REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK.
> User-Agent: Cirpack/v4.70 (gw_sip).
> Content-Length: 445.
> .
> v=0.
> o=cp10 146071439056 146071439056 IN IP4 10.7.1.129.
> s=SIP Call.
> c=IN IP4 91.121.129.144.
> t=0 0.
> m=audio 30962 RTP/AVP 18 4 0

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