[OpenSIPS-Users] ACK not forwarding

[Schneur Rosenberg]

My ACK coming from this device does not get forwarded to its designated
destination, I see that the RURI is to my OpenSIPS server even though the
contact has been rewritten in the OK, is that the reason the OpenSIPS
thinks its for himself? if yes how can I workaround the issue, perhaps with
dialog module etc?

Below please find the OK and the ACK, it has a Record route and proper
contact.

U 104.131.18.123:5060 -> 80.64.118.39:64308
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 80.64.118.39:64308
;rport=64308;received=80.64.118.39;branch=z9hG4bK-1131176229.
Record-Route: .
From: SORSTestDoor1 ;tag=628175012.
To: ;tag=as3223454f.
Call-ID: 10007@192.168.1.8.
CSeq: 1018 INVITE.
Server: SIP Server 9.21/CS.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH.
Supported: replaces, timer.
Contact: .
Content-Type: application/sdp.
Content-Length: 236.


U 88.64.118.39:64308 -> 104.131.18.123:5060
ACK sip:2...@sip.sipserver.info SIP/2.0.
Via: SIP/2.0/UDP 80.64.118.39:64308;branch=z9hG4bK-608413784.
Route: .
From: SORSTestDoor1 ;tag=628175012.
To: ;tag=as3223454f.
Call-ID: 10007@192.168.1.8.
User-Agent: Valcom xp1.50.14-10-14-14.
CSeq: 1018 ACK.
Content-Length: 0.
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Re: [OpenSIPS-Users] ACK not forwarding

[Alex Balashov]

The RURI of an end-to-end ACK should always be equal to the remote dialog 
target, i.e. in this case the Contact URI in the 200 OK.

‎If the RURI of the ACK is not equal to the remote target, it's not a properly 
formed ACK. And certainly, it should not point to the OpenSIPS proxy itself. 
There's no reasonable way to work around it: the UA in question needs to be 
fixed.

--
Alex Balashov | Principal | Evariste Systems LLC
303 Perimeter Center North, Suite 300
Atlanta, GA 30346
United States

Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
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Sent from my BlackBerry.
  Original Message  
From: Schneur Rosenberg
Sent: Tuesday, February 2, 2016 13:22
To: OpenSIPS users mailling list
Reply To: OpenSIPS users mailling list
Subject: [OpenSIPS-Users] ACK not forwarding


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Re: [OpenSIPS-Users] Compile Error

[Bogdan-Andrei Iancu]

Hi Dragomir,
This because some latest changes in the code and how the make file works 
Basically, when refactoring some files, the makefile dependencies got broken, 
so you need to clean first. Go into the modules/nat_traversal and run locally a 
make proper. You may need to do the same for siptrace and sl modules.
Regards,Bogdan


Sent from my Samsung Galaxy smartphone. Original message From: 
Dragomir Haralambiev  Date: 2/2/2016  23:18  (GMT+02:00) 
To: OpenSIPS users mailling list  Subject: 
[OpenSIPS-Users] Compile Error 
Hello,I try to compile latest Opensips 2.2.
make cleanmake allCompiling nathelper.cCompiling 
nh_table.cLinking nathelper.somake[1]: Leaving directory 
`/root/opensips_head/modules/nathelper'

make: *** [modules] Error 2
Best regards,Dragomir
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[OpenSIPS-Users] Compile Error

[Dragomir Haralambiev]

Hello,
I try to compile latest Opensips 2.2.

make clean
make all

Compiling nathelper.c
Compiling nh_table.c
Linking nathelper.so
make[1]: Leaving directory `/root/opensips_head/modules/nathelper'


make: *** [modules] Error 2

Best regards,
Dragomir
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Re: [OpenSIPS-Users] rabbit event not passing params 2.2

[Tito Cumpen]

Razvan,


I spun up another server running

 opensips -V
version: opensips 2.2-dev (x86_64/linux)
flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC,
FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
git revision: b7db080
main.c compiled on 21:55:02 Apr 21 2015 with gcc 4.8.3



check out the trace and the body actually has a payload when using this
exchange. I a wrapper on the event that I was using and I am still seeing
the issue of no payload in the latest dev. I added another exchange that
subscribes to cdr


E_ACC_EVENT

 subscribe_event("E_ACC_EVENT", "rabbitmq:,myrabbitserver/cdr");but I till
getting nothing in the body check out the rabbitproblems2.pcap




Thanks,
Tito

On Tue, Feb 2, 2016 at 11:58 AM, Răzvan Crainea  wrote:

> Hi, Tito!
>
> Can you raise an event from a different application with the same
> routing-key (sip1dev) and the same exchange("")? Does it work? if it does,
> can you send me the trace for the working one?
>
> Best regards,
> Răzvan
>
>
> On 02/02/2016 02:57 AM, Tito Cumpen wrote:
>
> Razvan,
>
> Here is the trace. Please let me know if you need anything else. Also I
> remember there was mention of a user replication module coming out for 2.X
> that would allow servers to be aware of users registered on other opensips
> registrars.
>
>
> Thanks,
> Tito
>
> On Tue, Jan 26, 2016 at 3:42 AM, Răzvan Crainea 
> wrote:
>
>> Hi, Tito!
>>
>> Can you send me a trace?
>>
>> Thanks,
>> Răzvan
>>
>>
>> On 01/25/2016 10:41 PM, Tito Cumpen wrote:
>>
>> Hey Razvan,
>>
>> This is still an issue with the latest dev build. The event is entirely
>> empty when it is transmitted to the queue. I've tried
>> modparam("event_rabbitmq", "sync_mode", 1) but it does not make a
>> difference.
>>
>> THanks,
>> Tito
>>
>>
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>>
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>
>
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>


rabbitworks.pcap
Description: Binary data


rabbitproblems2.pcap
Description: Binary data
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Re: [OpenSIPS-Users] ACK not forwarding

[Schneur Rosenberg]

Thanks I will open a ticket with the hardware manufacturer, lets hope they
will fix it

On Tue, Feb 2, 2016 at 8:25 PM, Alex Balashov 
wrote:

> The RURI of an end-to-end ACK should always be equal to the remote dialog
> target, i.e. in this case the Contact URI in the 200 OK.
>
> ‎If the RURI of the ACK is not equal to the remote target, it's not a
> properly formed ACK. And certainly, it should not point to the OpenSIPS
> proxy itself. There's no reasonable way to work around it: the UA in
> question needs to be fixed.
>
> --
> Alex Balashov | Principal | Evariste Systems LLC
> 303 Perimeter Center North, Suite 300
> Atlanta, GA 30346
> United States
>
> Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
> Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
>
> Sent from my BlackBerry.
>   Original Message
> From: Schneur Rosenberg
> Sent: Tuesday, February 2, 2016 13:22
> To: OpenSIPS users mailling list
> Reply To: OpenSIPS users mailling list
> Subject: [OpenSIPS-Users] ACK not forwarding
>
>
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[OpenSIPS-Users] Fwd: Opensips as outbound proxy to Freeswitch

[David Wafula]

-- Forwarded message --
From: David Wafula 
Date: Tue, Feb 2, 2016 at 10:43 AM
Subject: Re: [OpenSIPS-Users] Opensips as outbound proxy to Freeswitch
To: Bogdan-Andrei Iancu 


Yes, the opensips.cfg was indeed  messed up. i was tinkering with it.
Anyway, i regenerated a clean one, put in the the alias, and happily got
registrations going through. And one way audio. Except that i think am
missing the big picture.
My goal is to have Opensips front Freeswitch:

Client  <->  OpenSips  <-> Freeswitch


To act as outbound proxy at this stage. As i learn more about it, i hope to
get into NATing, loadbalancing etc. Am afraid  i have not  succeeded in
achieving this goal unfortunately, yet, though i think am very close. This
is what i have done.

1. Added the alias as advised. The registrations fail. Then when i create
 users in opensips, registration work perfectly, and one of the users can
call the other. Except that in this case it appears all is happening in
Opensips and freeswitch is not in the loop here. Infact, i verified, that
obviously registrationds are done on Opensips and not freeswitch.

2. I removed the alias, deleted the users from opensips. This time the
registrations goes through successfully onto freeswitch. Very happy, as
this is what i want. But now obviously INVITES fail with 403   Rely
forbidden.

It appears to me, there is more configuration that i must do, beyond
setting alias, to get Opensips to act as outboundproxy to freeswitch. I
need  registrations and calls to happen on Freeswitch, but via Opensips.

So after again searching the internet, i can  see that  may be something
like dynamic routing  can be done (
http://www.taitclarridge.com/techlog/2012/02/opensips-dynamic-routing.html),
or i can see whole lot of other  way of doing it (
http://www.opensips.org/Documentation/Tutorials-OpenSIPSFreeSwitchIntegration
).

What is the recommended, possibly simplest way to do this: to get opensips
proxy registrations, invites , messaging to freeswitch.

Many thanks.







On Mon, Feb 1, 2016 at 5:30 PM, Bogdan-Andrei Iancu 
wrote:

> David,
>
> Your opensips cfg is bogus - the test for the presence of username in RURI
> should not be done for a REGISTER requests (as they do not have a username
> part).
>
> Where have you taken the script from ? have you changed it by yourself ?
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 01.02.2016 15:06, David Wafula wrote:
>
> Thank you Bogdan. Ok  i made the alias change, and now am getting "Address
> Incomplete"  reply.  Please see the log below, am not sure what could be
> now causing this:
>
> REQUEST
> 
> 02-01 14:58:53.144: I/System.out(14726): REGISTER sip:192.168.0.46 SIP/2.0
> 02-01 14:58:53.144: I/System.out(14726): Via: SIP/2.0/UDP 10.0.1.175:30802
> ;rport;branch=z9hG4bKPjGfzHRCN4S.QyGFhBBv9njI9Amj2oU7ko
> 02-01 14:58:53.144: I/System.out(14726): Route: 
> 02-01 14:58:53.144: I/System.out(14726): Max-Forwards: 70
> 02-01 14:58:53.144: I/System.out(14726): From:  >;tag=eNVm.45KTH3OkXAjSikua1cwsZ.XdeRE
> 02-01 14:58:53.144: I/System.out(14726): To: 
> 02-01 14:58:53.144: I/System.out(14726): Call-ID: 9
> TVBTDFtUPz05axr9qvyynhFWhJetb6m
> 02-01 14:58:53.144: I/System.out(14726): CSeq: 64838 REGISTER
> 02-01 14:58:53.144: I/System.out(14726): User-Agent: Pjsua2 Android 2.4.5
> 02-01 14:58:53.144: I/System.out(14726): Contact:  34708;ob>
> 02-01 14:58:53.144: I/System.out(14726): Expires: 300
> 02-01 14:58:53.144: I/System.out(14726): Allow: PRACK, INVITE, ACK, BYE,
> CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
> 02-01 14:58:53.144: I/System.out(14726): Content-Length:  0
> 02-01 14:58:53.144: I/System.out(14726):
> 02-01 14:58:53.144: I/System.out(14726): --end msg--
>
> REPLY
> =
>
>
> 02-01 14:58:53.585: I/System.out(14726): SIP/2.0 484 Address Incomplete
> 02-01 14:58:53.585: I/System.out(14726): Via: SIP/2.0/UDP 10.0.1.175:30802
> ;received=10.0.1.175;rport=30802
> ;branch=z9hG4bKPjGfzHRCN4S.QyGFhBBv9njI9Amj2oU7ko
> 02-01 14:58:53.585: I/System.out(14726): From:  >;tag=eNVm.45KTH3OkXAjSikua1cwsZ.XdeRE
> 02-01 14:58:53.585: I/System.out(14726): To:  >;tag=a0a925d2eca49498ea7382b7b1f63f38.d365
> 02-01 14:58:53.585: I/System.out(14726): Call-ID: 9
> TVBTDFtUPz05axr9qvyynhFWhJetb6m
> 02-01 14:58:53.585: I/System.out(14726): CSeq: 64838 REGISTER
> 02-01 14:58:53.585: I/System.out(14726): Server: OpenSIPS (2.1.2
> (x86_64/linux))
> 02-01 14:58:53.585: I/System.out(14726): Content-Length: 0
>
> On Mon, Feb 1, 2016 at 1:07 PM, Bogdan-Andrei Iancu <
> bog...@opensips.org> wrote:
>
>> Hi David,
>>
>> I see. The problem is all the SIP traffic contains references to this 
>> 192.168.0.46,
>> but opensips has no idea 

Re: [OpenSIPS-Users] opensips transparent technology

[Bogdan-Andrei Iancu]

Hi Michael,

$ru is the destination, the called number. No nothing from it will be 
displayed on the callee side - the callee will see (as caller ID) info 
from FROM URI ($fu).


Regards,

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

On 02.02.2016 06:33, MichaelLeung wrote:

yes , $ru

does the carrier will display the "ringing from Name " base on the 
string $ru ? i doubt it b because string $ru include domain of sip 
server string, or string $rU  will also be sent into carrier and will 
be accepted as the display number when ringing the phone only when we 
change it to a real phone number.


On 02/01/2016 05:50 PM, Bogdan-Andrei Iancu wrote:

Hi,

Assuming you were talking about $ru (and not $ur), the answer is yes, 
the new $ru will be set into the SIP request and sent out to the next 
SIP hop (your carrier). Note that $ru will push changes only in the 
Request URI.


Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 01.02.2016 10:44, MichaelLeung wrote:
does $ur willl be transmitted to carriar when i change the uri to a 
real number  ?


On 01/29/2016 01:35 PM, MichaelLeung wrote:

ok, thanks .


On 01/29/2016 12:05 PM, Pavel Eremin wrote:
Opensips will not change CLI or CAllerID (real number from 
carrier) if you don't tell him to do it. So, It's like ask what 
name for "painting white pages" - there is no name it's just blank 
pages.. I think



2016-01-29 8:08 GMT+05:00 MichaelLeung :

any reply ?


On 01/26/2016 04:42 PM, MichaelLeung wrote:

can uac_replace_from read real phone number from databases?

On 01/25/2016 01:03 PM, MichaelLeung wrote:

thanks for reply
no , it is just a asking , i don't have real phone number
database, or should i have one ?
can you tell me what is the name of this technology ?

On 01/24/2016 07:33 PM, Stefano Pisani wrote:

Where is their real phone number?
Do you have it in a database?
You can change the From header to show the real phone number.



Il 24/01/2016 12.22, MichaelLeung ha scritto:

Hi all

i was trying to make my opensips users to sent their real
phone number when they call .

what is the name of this technology ? transmit
transparently ?

i search google find nothing, and where can i read
document of this technology ?

thanks.


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Re: [OpenSIPS-Users] Linphone and OpenSIPS over TCP

[Bogdan-Andrei Iancu]

Hi Nabeel,

Your OpenSIPS tries to connect via TCP to the destination and to avoid 
blocking it is doing it async:



Feb  1 19:22:13 server1 opensips: Feb  1 19:22:13 [20587] 
DBG:core:print_ip: tcpconn_new: new tcp connection to: 188.29.165.162
Feb  1 19:22:13 server1 opensips: Feb  1 19:22:13 [20587] 
DBG:core:tcpconn_new: on port 58985, proto 2
Feb  1 19:22:13 server1 opensips: Feb  1 19:22:13 [20587] 
DBG:core:proto_tcp_send: Successfully started async connection


After that, in 5 seconds, the final timer hits (as timeout for no 
reply), while the TCP connect still haven;t finished (so there is no 
actual packet sent out).


Regards,

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

On 01.02.2016 21:35, Nabeel wrote:

Hi Bogdan,

Below is the requested log for TCP call attempt.  User  is trying 
to call user  via OpenSIPS server 162.248.6.120 :


http://pastebin.com/UQ9mEemd

On 1 February 2016 at 09:41, Bogdan-Andrei Iancu > wrote:


Hi,

I strongly suggest to look into the opensips logs and see what
opensips try to do with the call. Based on your saying (that you
see a timeout), I suspect your OpenSIPS tries to deliver the call
over TCP to a destination which does not listen on TCP.
If you do not know hoe to interpret the logs, run opensips in
debug=4 mode, upload the logs corresponding the INVITE execution
and provide the link.

Regards,

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

On 31.01.2016 16:28, Nabeel wrote:

Without using alias=domain.com , TCP still
does not work.  My initial request for someone to test this using
Linphone remains. Please test and let me know if you can call
using TCP with OpenSIPS listening on an IP address.

On 31 January 2016 at 09:28, Nabeel > wrote:

On further testing, using the IP address instead of the
domain name in the URI setting of Linphone works with TCP, so
I think this might be to do with SRV/NAPTR records associated
with the domain.

On 31 January 2016 at 08:29, Nabeel > wrote:

Hello,

There seems to be a problem with calls over TCP using
Linphone, and since Linphone is a popular open source
application, I would like someone to please verify this
problem. Calls work fine with Linphone over UDP, but
after registering with TCP using the same credentials,
calls do not connect at all and lead to a request
timeout.  A request timeout does not say much about the
cause, but in this case I suspect there is something
wrong with TCP on the server side. I would like someone
to please install Linphone on your phone and connect to
your OpenSIPS server using UDP and TCP. Please report
here if the calls work over both transports.





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Re: [OpenSIPS-Users] Linphone and OpenSIPS over TCP

[Nabeel]

Hi Bogdan,

Even if I remove the 5 second timeout by removing #modparam("tm",
"fr_inv_timeout", 30), the timeout occurs after about 20 seconds. What do
you suggest is the solution?
Hi Nabeel,

Your OpenSIPS tries to connect via TCP to the destination and to avoid
blocking it is doing it async:


Feb  1 19:22:13 server1 opensips: Feb  1 19:22:13 [20587]
DBG:core:print_ip: tcpconn_new: new tcp connection to: 188.29.165.162
Feb  1 19:22:13 server1 opensips: Feb  1 19:22:13 [20587]
DBG:core:tcpconn_new: on port 58985, proto 2
Feb  1 19:22:13 server1 opensips: Feb  1 19:22:13 [20587]
DBG:core:proto_tcp_send: Successfully started async connection

After that, in 5 seconds, the final timer hits (as timeout for no reply),
while the TCP connect still haven;t finished (so there is no actual packet
sent out).

Regards,

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

On 01.02.2016 21:35, Nabeel wrote:

Hi Bogdan,

Below is the requested log for TCP call attempt.  User  is trying to
call user  via OpenSIPS server 162.248.6.120 :

http://pastebin.com/UQ9mEemd

On 1 February 2016 at 09:41, Bogdan-Andrei Iancu 
wrote:

> Hi,
>
> I strongly suggest to look into the opensips logs and see what opensips
> try to do with the call. Based on your saying (that you see a timeout), I
> suspect your OpenSIPS tries to deliver the call over TCP to a destination
> which does not listen on TCP.
> If you do not know hoe to interpret the logs, run opensips in debug=4
> mode, upload the logs corresponding the INVITE execution and provide the
> link.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 31.01.2016 16:28, Nabeel wrote:
>
> Without using alias=domain.com, TCP still does not work.  My initial
> request for someone to test this using Linphone remains. Please test and
> let me know if you can call using TCP with OpenSIPS listening on an IP
> address.
>
> On 31 January 2016 at 09:28, Nabeel  wrote:
>
>> On further testing, using the IP address instead of the domain name in
>> the URI setting of Linphone works with TCP, so I think this might be to do
>> with SRV/NAPTR records associated with the domain.
>>
>> On 31 January 2016 at 08:29, Nabeel  wrote:
>>
>>> Hello,
>>>
>>> There seems to be a problem with calls over TCP using Linphone, and
>>> since Linphone is a popular open source application, I would like someone
>>> to please verify this problem. Calls work fine with Linphone over UDP, but
>>> after registering with TCP using the same credentials, calls do not connect
>>> at all and lead to a request timeout.  A request timeout does not say much
>>> about the cause, but in this case I suspect there is something wrong with
>>> TCP on the server side. I would like someone to please install Linphone on
>>> your phone and connect to your OpenSIPS server using UDP and TCP.  Please
>>> report here if the calls work over both transports.
>>>
>>
>>
>
>
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Re: [OpenSIPS-Users] Interpreting the debug memory dump.

[Bogdan-Andrei Iancu]

Hi Surya,

Doesn't look to be any leak.

You can check anytime with "opensipsctl fifo get_statistics shmem:"

Regards,

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

On 01.02.2016 22:08, surya wrote:

Hi Bogdan,

Basically, I am looking for performance bottleneck in the modified presence
server. I suspect because of memory leaks may be the application is not
performing well. So, I am looking if my changed code is leaking any memory.

Here is the link of whole dump. DumpAtShutdown

.

Thanks,
Surya



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Re: [OpenSIPS-Users] Linphone and OpenSIPS over TCP

[Bogdan-Andrei Iancu]

The problem is not the 5 seconds timeout at sip level, is the fact that 
the end point you are trying to connect to via TCP does not accept the 
connection. You can check that via tcpdump.


Regards,

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

On 02.02.2016 11:17, Nabeel wrote:


Hi Bogdan,

Even if I remove the 5 second timeout by removing #modparam("tm", 
"fr_inv_timeout", 30), the timeout occurs after about 20 seconds. What 
do you suggest is the solution?


Hi Nabeel,

Your OpenSIPS tries to connect via TCP to the destination and to avoid 
blocking it is doing it async:



Feb  1 19:22:13 server1 opensips: Feb  1 19:22:13 [20587] 
DBG:core:print_ip: tcpconn_new: new tcp connection to: 188.29.165.162
Feb  1 19:22:13 server1 opensips: Feb  1 19:22:13 [20587] 
DBG:core:tcpconn_new: on port 58985, proto 2
Feb  1 19:22:13 server1 opensips: Feb  1 19:22:13 [20587] 
DBG:core:proto_tcp_send: Successfully started async connection


After that, in 5 seconds, the final timer hits (as timeout for no 
reply), while the TCP connect still haven;t finished (so there is no 
actual packet sent out).


Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 01.02.2016 21:35, Nabeel wrote:

Hi Bogdan,

Below is the requested log for TCP call attempt. User  is trying 
to call user  via OpenSIPS server 162.248.6.120 :


http://pastebin.com/UQ9mEemd

On 1 February 2016 at 09:41, Bogdan-Andrei Iancu > wrote:


Hi,

I strongly suggest to look into the opensips logs and see what
opensips try to do with the call. Based on your saying (that you
see a timeout), I suspect your OpenSIPS tries to deliver the call
over TCP to a destination which does not listen on TCP.
If you do not know hoe to interpret the logs, run opensips in
debug=4 mode, upload the logs corresponding the INVITE execution
and provide the link.

Regards,

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

On 31.01.2016 16:28, Nabeel wrote:

Without using alias=domain.com , TCP still
does not work.  My initial request for someone to test this
using Linphone remains. Please test and let me know if you can
call using TCP with OpenSIPS listening on an IP address.

On 31 January 2016 at 09:28, Nabeel > wrote:

On further testing, using the IP address instead of the
domain name in the URI setting of Linphone works with TCP,
so I think this might be to do with SRV/NAPTR records
associated with the domain.

On 31 January 2016 at 08:29, Nabeel > wrote:

Hello,

There seems to be a problem with calls over TCP using
Linphone, and since Linphone is a popular open source
application, I would like someone to please verify this
problem. Calls work fine with Linphone over UDP, but
after registering with TCP using the same credentials,
calls do not connect at all and lead to a request
timeout.  A request timeout does not say much about the
cause, but in this case I suspect there is something
wrong with TCP on the server side. I would like someone
to please install Linphone on your phone and connect to
your OpenSIPS server using UDP and TCP.  Please report
here if the calls work over both transports.





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Re: [OpenSIPS-Users] rabbit event not passing params 2.2

[Răzvan Crainea]

Hi, Tito!

Can you raise an event from a different application with the same 
routing-key (sip1dev) and the same exchange("")? Does it work? if it 
does, can you send me the trace for the working one?


Best regards,
Răzvan

On 02/02/2016 02:57 AM, Tito Cumpen wrote:

Razvan,

Here is the trace. Please let me know if you need anything else. Also 
I remember there was mention of a user replication module coming out 
for 2.X that would allow servers to be aware of users registered on 
other opensips registrars.



Thanks,
Tito

On Tue, Jan 26, 2016 at 3:42 AM, Răzvan Crainea > wrote:


Hi, Tito!

Can you send me a trace?

Thanks,
Răzvan


On 01/25/2016 10:41 PM, Tito Cumpen wrote:

Hey Razvan,

This is still an issue with the latest dev build. The event is
entirely empty when it is transmitted to the queue. I've tried
modparam("event_rabbitmq", "sync_mode", 1) but it does not make a
difference.

THanks,
Tito


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Re: [OpenSIPS-Users] Interpreting the debug memory dump.

[surya]

Hi Bogdan,

Thanks for the confirmation.

I request you to please share with me (if possible in email.) how did you
identify the leak is not there and if it's there how it looks in the log.
This way you won't be going thru the same question again and will be fast
for me too :)

I remember doing this "opensipsctl fifo get_statistics shmem:" sometime back
but I got some errors running the command. I'll try again but I guess this
will just show the stats not where is the leak. Am I correct?

Another request, can you please suggest what or how to look for the
performance bottleneck. I am really struggling to find that.

Thanks,
Surya



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Re: [OpenSIPS-Users] opensips transparent technology

[Bogdan-Andrei Iancu]

If you want o push a different CallerID (to carrier) rather than the one 
you have in FROM hdr, consider using the Remote-Party-ID or 
P-Asserted-Identity headers.


Regards,

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

On 02.02.2016 11:44, MichaelLeung wrote:

yes, Pavel had explaned this to me .

and , you can tell me my next questin ?

so , we don't  need to do anything to $fu if we sent the call to carrier.

but what if $fu isn't start with a regluar phone number ?

---
On 02/02/2016 04:22 PM, Bogdan-Andrei Iancu wrote:

Hi Michael,

$ru is the destination, the called number. No nothing from it will be 
displayed on the callee side - the callee will see (as caller ID) 
info from FROM URI ($fu).


Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 02.02.2016 06:33, MichaelLeung wrote:

yes , $ru

does the carrier will display the "ringing from Name " base on the 
string $ru ? i doubt it b because string $ru include domain of sip 
server string, or string $rU  will also be sent into carrier and 
will be accepted as the display number when ringing the phone only 
when we change it to a real phone number.


On 02/01/2016 05:50 PM, Bogdan-Andrei Iancu wrote:

Hi,

Assuming you were talking about $ru (and not $ur), the answer is 
yes, the new $ru will be set into the SIP request and sent out to 
the next SIP hop (your carrier). Note that $ru will push changes 
only in the Request URI.


Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 01.02.2016 10:44, MichaelLeung wrote:
does $ur willl be transmitted to carriar when i change the uri to 
a real number  ?


On 01/29/2016 01:35 PM, MichaelLeung wrote:

ok, thanks .


On 01/29/2016 12:05 PM, Pavel Eremin wrote:
Opensips will not change CLI or CAllerID (real number from 
carrier) if you don't tell him to do it. So, It's like ask what 
name for "painting white pages" - there is no name it's just 
blank pages.. I think



2016-01-29 8:08 GMT+05:00 MichaelLeung :

any reply ?


On 01/26/2016 04:42 PM, MichaelLeung wrote:

can uac_replace_from read real phone number from databases?

On 01/25/2016 01:03 PM, MichaelLeung wrote:

thanks for reply
no , it is just a asking , i don't have real phone number
database, or should i have one ?
can you tell me what is the name of this technology ?

On 01/24/2016 07:33 PM, Stefano Pisani wrote:

Where is their real phone number?
Do you have it in a database?
You can change the From header to show the real phone number.



Il 24/01/2016 12.22, MichaelLeung ha scritto:

Hi all

i was trying to make my opensips users to sent their
real phone number when they call .

what is the name of this technology ? transmit
transparently ?

i search google find nothing, and where can i read
document of this technology ?

thanks.


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Re: [OpenSIPS-Users] Opensips 1.11 permission module problem

[Liviu Chircu]

To fix that, replace *memlog *with *memdump*! Printing every malloc() by 
setting the *memlog *will just pollute the logs, and is not very useful 
anyway...


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

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Re: [OpenSIPS-Users] Linphone and OpenSIPS over TCP

[Nabeel]

Hi,

Please see the tcpdump trace below.  Can this be caused by incorrect use of
the TURN server? I am convinced that the problem relates to use of the TURN
server, but not sure if the problem is caused by OpenSIPS or the TURN
server itself.

http://pastebin.com/HPZ7nRYS
The problem is not the 5 seconds timeout at sip level, is the fact that the
end point you are trying to connect to via TCP does not accept the
connection. You can check that via tcpdump.

Regards,

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

On 02.02.2016 11:17, Nabeel wrote:

Hi Bogdan,

Even if I remove the 5 second timeout by removing #modparam("tm",
"fr_inv_timeout", 30), the timeout occurs after about 20 seconds. What do
you suggest is the solution?
Hi Nabeel,

Your OpenSIPS tries to connect via TCP to the destination and to avoid
blocking it is doing it async:


Feb  1 19:22:13 server1 opensips: Feb  1 19:22:13 [20587]
DBG:core:print_ip: tcpconn_new: new tcp connection to: 188.29.165.162
Feb  1 19:22:13 server1 opensips: Feb  1 19:22:13 [20587]
DBG:core:tcpconn_new: on port 58985, proto 2
Feb  1 19:22:13 server1 opensips: Feb  1 19:22:13 [20587]
DBG:core:proto_tcp_send: Successfully started async connection

After that, in 5 seconds, the final timer hits (as timeout for no reply),
while the TCP connect still haven;t finished (so there is no actual packet
sent out).

Regards,

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

On 01.02.2016 21:35, Nabeel wrote:

Hi Bogdan,

Below is the requested log for TCP call attempt.  User  is trying to
call user  via OpenSIPS server 162.248.6.120 :

http://pastebin.com/UQ9mEemd

On 1 February 2016 at 09:41, Bogdan-Andrei Iancu 
wrote:

> Hi,
>
> I strongly suggest to look into the opensips logs and see what opensips
> try to do with the call. Based on your saying (that you see a timeout), I
> suspect your OpenSIPS tries to deliver the call over TCP to a destination
> which does not listen on TCP.
> If you do not know hoe to interpret the logs, run opensips in debug=4
> mode, upload the logs corresponding the INVITE execution and provide the
> link.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 31.01.2016 16:28, Nabeel wrote:
>
> Without using alias=domain.com, TCP still does not work.  My initial
> request for someone to test this using Linphone remains. Please test and
> let me know if you can call using TCP with OpenSIPS listening on an IP
> address.
>
> On 31 January 2016 at 09:28, Nabeel  wrote:
>
>> On further testing, using the IP address instead of the domain name in
>> the URI setting of Linphone works with TCP, so I think this might be to do
>> with SRV/NAPTR records associated with the domain.
>>
>> On 31 January 2016 at 08:29, Nabeel  wrote:
>>
>>> Hello,
>>>
>>> There seems to be a problem with calls over TCP using Linphone, and
>>> since Linphone is a popular open source application, I would like someone
>>> to please verify this problem. Calls work fine with Linphone over UDP, but
>>> after registering with TCP using the same credentials, calls do not connect
>>> at all and lead to a request timeout.  A request timeout does not say much
>>> about the cause, but in this case I suspect there is something wrong with
>>> TCP on the server side. I would like someone to please install Linphone on
>>> your phone and connect to your OpenSIPS server using UDP and TCP.  Please
>>> report here if the calls work over both transports.
>>>
>>
>>
>
>
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>
>
>
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