Re: [OpenSIPS-Users] Call from Asterisk to Opensips

[ha do]

Hi Truong

first thing you should try to read the asterisk SIP TRUNK and here is the basic 
example and i think the problem is asterisk not opensips
http://www.panoramisk.com/90/sip-trunk-with-asterisk/en/

and make sure to check the debug from asterisk and opensips, i think you will 
get the clues :D


Ha`

--- On Thu, 5/5/11, Duong Manh Truong  wrote:

From: Duong Manh Truong 
Subject: [OpenSIPS-Users] Call from Asterisk to Opensips
To: "OpenSIPS users mailling list" 
Date: Thursday, May 5, 2011, 9:53 PM

Hi all, I've created sip trunk on Asterisk and defined asterisk server ip on 
address table of opensips
Then, from extension of Opensips , i can dial out to pstn through Asterisk

Now, i want to route PSTN call to the extension but when Asterisk receive the 
call from PSTN and dial Opensips through the Sip Trunki always got the message 
in the asterisk's console: 
 Called to-opensips/1001    -- SIP/to-opensips-0745 is circuit-busy  == 
Everyone is busy/congested at this time (1:0/1/0)
(1001 is the extension of Opensips) 
Then the call hangs up. 
Anyone got this problem ? please help me the way to deal with!
Thanks so much! 

-Inline Attachment Follows-

___
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

Re: [OpenSIPS-Users] Opensips account: Local calls => Proxy authentication required

[ha do]

Hi Truong
 
there are some parameters in your opensips.cfg
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "password_column", "password")
modparam("auth_db", "db_url",   
"mysql://opensips:opensipsrw@localhost/opensips")
modparam("auth_db", "load_credentials", "")
 
you use clear text password or encrypted password for your subscribers
take time to read and check it again in mysql database

Thank you
Ha`

--- On Thu, 5/5/11, ha do  wrote:


From: ha do 
Subject: Re: [OpenSIPS-Users] Opensips account: Local calls => Proxy 
authentication required
To: "OpenSIPS users mailling list" 
Date: Thursday, May 5, 2011, 2:54 AM






Hi Truong 

please use the pastebin.com to upload your opensips.conf
i can check it in details

Thanks
Ha`

--- On Wed, 5/4/11, Duong Manh Truong  wrote:


From: Duong Manh Truong 
Subject: Re: [OpenSIPS-Users] Opensips account: Local calls => Proxy 
authentication required
To: "OpenSIPS users mailling list" 
Date: Wednesday, May 4, 2011, 6:07 PM



Hi Ha`!


Thanks for your help. 


I read the book and followed steps written there


But now i can not figure out why local calls are not successful


Please help me in more details! 


Regards.







Hi Truong

try the 
ebook https://www.packtpub.com/building-telephony-systems-with-opensips-1-6/book

there are a lot of examples and explain how opensips work

hope this help
Ha`

--- On Wed, 5/4/11, Duong Manh Truong  wrote:

From: Duong Manh Truong 
Subject: [OpenSIPS-Users] Opensips account: Local calls => Proxy authentication 
required
To: "OpenSIPS users mailling list" 
Date: Wednesday, May 4, 2011, 4:05 AM

Hi all,?
After successfully installation of Opensips, my extensions can dial to each 
other ?(local call)?example: ext 1000 can dial to ext 1001?
Then i configure Opensips to go to PSTN through Asterisk (gateway) by using 
rewritehostport function and "address" table in "opensips" database
(for Asterisk server IP)
Now, i can call from Opensips to PSTN over Asterisk but.for local calls 
(1000 call 1001),?there're always the message "Proxy authentication required" 
in Xlite screen ???

I type "opensipsctl online" and find that both of 2 exts are online ???
Please tell me how to deal with this matter? ?( I attached my opensips.conf 
file in this email)?

-Inline Attachment Follows- 
-Inline Attachment Follows-


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

-Inline Attachment Follows-


___
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

Re: [OpenSIPS-Users] Opensips account: Local calls => Proxy authentication required

[ha do]

Hi Truong 

please use the pastebin.com to upload your opensips.conf
i can check it in details

Thanks
Ha`

--- On Wed, 5/4/11, Duong Manh Truong  wrote:

From: Duong Manh Truong 
Subject: Re: [OpenSIPS-Users] Opensips account: Local calls => Proxy 
authentication required
To: "OpenSIPS users mailling list" 
Date: Wednesday, May 4, 2011, 6:07 PM

Hi Ha`!

Thanks for your help. 

I read the book and followed steps written there

But now i can not figure out why local calls are not successful

Please help me in more details! 
Regards.





Hi Truong

try the 
ebook https://www.packtpub.com/building-telephony-systems-with-opensips-1-6/book


there are a lot of examples and explain how opensips work

hope this help
Ha`

--- On Wed, 5/4/11, Duong Manh Truong  wrote:


From: Duong Manh Truong 
Subject: [OpenSIPS-Users] Opensips account: Local calls => Proxy authentication 
required

To: "OpenSIPS users mailling list" 
Date: Wednesday, May 4, 2011, 4:05 AM

Hi all,?
After successfully installation of Opensips, my extensions can dial to each 
other ?(local call)?example: ext 1000 can dial to ext 1001?

Then i configure Opensips to go to PSTN through Asterisk (gateway) by using 
rewritehostport function and "address" table in "opensips" database
(for Asterisk server IP)
Now, i can call from Opensips to PSTN over Asterisk but.for local calls 
(1000 call 1001),?there're always the message "Proxy authentication required" 
in Xlite screen ???


I type "opensipsctl online" and find that both of 2 exts are online ???
Please tell me how to deal with this matter? ?( I attached my opensips.conf 
file in this email)?

-Inline Attachment Follows-

-Inline Attachment Follows-

___
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

Re: [OpenSIPS-Users] Opensips account: Local calls => Proxy authentication required

[ha do]

Hi Truong

try the ebook 
https://www.packtpub.com/building-telephony-systems-with-opensips-1-6/book

there are a lot of examples and explain how opensips work

hope this help
Ha`

--- On Wed, 5/4/11, Duong Manh Truong  wrote:

From: Duong Manh Truong 
Subject: [OpenSIPS-Users] Opensips account: Local calls => Proxy authentication 
required
To: "OpenSIPS users mailling list" 
Date: Wednesday, May 4, 2011, 4:05 AM

Hi all, 
After successfully installation of Opensips, my extensions can dial to each 
other  (local call) example: ext 1000 can dial to ext 1001 
Then i configure Opensips to go to PSTN through Asterisk (gateway) by using 
rewritehostport function and "address" table in "opensips" database
(for Asterisk server IP)
Now, i can call from Opensips to PSTN over Asterisk but.for local calls 
(1000 call 1001), there're always the message "Proxy authentication required" 
in Xlite screen ???

I type "opensipsctl online" and find that both of 2 exts are online ???
Please tell me how to deal with this matter?  ( I attached my opensips.conf 
file in this email) 

-Inline Attachment Follows-

___
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

Re: [OpenSIPS-Users] Opensips sometime resend Accouting-request to freeradius

[ha do]

Hi Dani

if you look at my opensips.cfg and there is only 1 time for INVITE and 1 time 
for BYE to do accouting

the problem here is sometimes freeradius is not send the Accouting-Response to 
radiusclient

so that is why radiusclient resend the Accouting-request 

opensips and freeradius run in the same server and freeradius  run at interface 
127.0.0.1


i use virtual box to test
i use the freeradius from Ag-Project

in my option, the problem is freeradius


Thank you
Ha`

--- On Thu, 3/24/11, Dani Popa  wrote:

> From: Dani Popa 
> Subject: Re: [OpenSIPS-Users] Opensips sometime resend Accouting-request to 
> freeradius
> To: users@lists.opensips.org
> Date: Thursday, March 24, 2011, 7:11 AM
> You have a problem with opensips
> script. You set acct start flag many 
> times and radius try each time to insert in mysql the same
> start query, 
> but you can not because you have key defined on your
> table.
> 
> 
> Dani
> On 03/23/11 10:06, ha do wrote:
> > Hi list
> >
> > i am test
> > opensips 1.6.4
> > freeradius from ag-projects
> > cdrtool version 8.0.17
> > callcontrol version 2.0.8
> >
> > there is sometimes opensips resend Accounting-Request
> to freeradius and make freeradius seems to reject message
> from opensips
> >
> > here is the pastebin link:
> > http://pastebin.com/d1k38n5a -  freeradius log
> > http://pastebin.com/ww2Dq3SC - opensips.cfg
> >
> > how to resolve problem
> >
> > Thank you
> >
> >
> >
> >
> > ___
> > 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
> 


  

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

Re: [OpenSIPS-Users] Opensips sometime resend Accouting-request to freeradius

[ha do]

Hi 
i found problem,

Thank you
Ha`

--- On Wed, 3/23/11, ha do  wrote:

> From: ha do 
> Subject: Opensips sometime resend Accouting-request to freeradius
> To: "OpenSIPS" 
> Date: Wednesday, March 23, 2011, 2:06 AM
> Hi list
> 
> i am test 
> opensips 1.6.4
> freeradius from ag-projects
> cdrtool version 8.0.17
> callcontrol version 2.0.8
> 
> there is sometimes opensips resend Accounting-Request to
> freeradius and make freeradius seems to reject message from
> opensips
> 
> here is the pastebin link:
> http://pastebin.com/d1k38n5a -  freeradius log
> http://pastebin.com/ww2Dq3SC - opensips.cfg
> 
> how to resolve problem
> 
> Thank you
> 
> 
>       
> 


  

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

[OpenSIPS-Users] Opensips sometime resend Accouting-request to freeradius

[ha do]

Hi list

i am test 
opensips 1.6.4
freeradius from ag-projects
cdrtool version 8.0.17
callcontrol version 2.0.8

there is sometimes opensips resend Accounting-Request to freeradius and make 
freeradius seems to reject message from opensips

here is the pastebin link:
http://pastebin.com/d1k38n5a -  freeradius log
http://pastebin.com/ww2Dq3SC - opensips.cfg

how to resolve problem

Thank you


  

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

Re: [OpenSIPS-Users] Munin monitoring plugin

[ha do]

Hi

interesting plug_in

do you have screen shot of pictures

:D

--- On Tue, 3/22/11, Henning Holtschneider  wrote:

> From: Henning Holtschneider 
> Subject: [OpenSIPS-Users] Munin monitoring plugin
> To: "OpenSIPS users mailling list" 
> Date: Tuesday, March 22, 2011, 2:01 AM
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
> 
> Hello everybody,
> 
> I wrote a Munin monitoring plugin which collects data via
> the 'opensipsctl fifo get_statistics' command the other day.
> The plugin is available at . If
> you have any questions or suggestions, feel free to contact
> me or just fork my code at Github and contribute!
> 
> Cheers,
> Henning Holtschneider
> - --
> LocaNet oHG - http://www.loca.net
> Lindemannstrasse 81, D-44137 Dortmund
> tel +49 231 91596-25, fax +49 231 91596-55
> sip 2...@voip.loca.net
> 
> Registergericht Amtsgericht Dortmund HRA 14208
> Geschäftsführer Sven Haufe, Henning Holtschneider
> 
> -BEGIN PGP SIGNATURE-
> Version: GnuPG v1.4.10 (Darwin)
> 
> iEYEARECAAYFAk2IV2EACgkQP9goCV2uudcm3gCfR/37oL7BtYGKdxvvGp9Y2qTS
> lMMAoO4PV6fO9+WOm9zCNjvdD62QZHdR
> =+tCa
> -END PGP SIGNATURE-
> 
> ___
> 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

Re: [OpenSIPS-Users] need help on connection CDRTool to opensips

[ha do]

Hi Tijmen

it works fine right now for callcontrol hangups a call in my last email
can i request to make the instruction on config.sample.ini of the OPENSIPS part
callcontrol uses the mi_datagram to make it clearer not confusing with mi_fifo
and the socket_path is the same socket_name in mi_datagram config of opensips
it should be good for someone, likes me

i still need help
> 2 opensips2 is a gateway
>endpoint-opensips1---opensips2internet-VOIP service 
provider
So basically opensips2 is just a gateway to opensips1?
[ha] yes,how to config cdrtool rating calls for opensips2


Thank you
Ha


- Original Message 
From: Tijmen de Mes 
To: OpenSIPS users mailling list 
Sent: Tue, November 30, 2010 3:44:12 PM
Subject: Re: [OpenSIPS-Users] need help on connection CDRTool to opensips

Hi,

On Nov 25, 2010, at 10:06 AM, ha do wrote:
> and opensips.cfg with fifo config
> # - mi_fifo params -
> modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")
> modparam("mi_fifo", "fifo_mode", 0600)
> modparam("mi_fifo", "fifo_group", 0)
> modparam("mi_fifo", "fifo_group", "root")
> modparam("mi_fifo", "fifo_user", 0)
> modparam("mi_fifo", "fifo_user", "root")
> 
> and the callcontrol config.ini
> [CallControl]
> group   = root
> [OpenSIPS]
> socket_path = /tmp
> ;max_connections = 10 -- does it disconnect 10 calls in 1 time, if i need 100
> terminated i need to increase to 100, right???
> 
> how to fix the issue
Try  using socket_path=/tmp/socket in config.ini. The socket path should match 
the "mi_datagram" ,"socket_name"
in the opensips config.

Tijmen de Mes
AG Projects


___
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

Re: [OpenSIPS-Users] need help on connection CDRTool to opensips

[ha do]

Hi Tijimen

[ha - solved] with help from bogdan_vs: i change the sock_path = 
/tmp/opensips.sock and use the mi_datagram in opensips.cfg
callcontrol hangup calls successfull. So the callcontrol use the mi_datagram, 
right??

i still need help
> 2 opensips2 is a gateway
>endpoint-opensips1---opensips2internet-VOIP service 
provider
So basically opensips2 is just a gateway to opensips1?
[ha] yes,

Thank you
Ha

- Original Message 
From: ha do 
To: OpenSIPS users mailling list 
Sent: Thu, November 25, 2010 4:06:14 PM
Subject: Re: [OpenSIPS-Users] need help on connection CDRTool to opensips

Hi all

can anyone help me on call-control, i already try the 
mi_datagrambut i get the same error, permission denied

Call id NTJjNzY1NzI1MzgwNmQ2MmU4NDMxMWIzZDgzNWFkOTI. of 1...@cucku.com to 
sip:00842...@cucku.com started for maximum 60 seconds
Call id NTJjNzY1NzI1MzgwNmQ2MmU4NDMxMWIzZDgzNWFkOTI. of 1...@cucku.com to 
sip:00842...@cucku.com connected for maximum 60 seconds
callcontrol.opensips.UNIXSocketProtocol starting on 
'/var/run/callcontrol/opensips_01.sock'
error: cannot write request to /tmp: Connection refused
error: failed to end dialog: Cannot send request to OpenSIPS
Call id NTJjNzY1NzI1MzgwNmQ2MmU4NDMxMWIzZDgzNWFkOTI. of 1...@cucku.com to 
sip:00842...@cucku.com disconnected by call control after 60 seconds, call 
price 

is 0.0998

prepaid account 100 has only 60s to connect the call, and after 60s callcontrol 
cannot disconnected call

the mi_fifo located in /tmp

and opensips.cfg with fifo config
# - mi_fifo params -
modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")
modparam("mi_fifo", "fifo_mode", 0600)
modparam("mi_fifo", "fifo_group", 0)
modparam("mi_fifo", "fifo_group", "root")
modparam("mi_fifo", "fifo_user", 0)
modparam("mi_fifo", "fifo_user", "root")

and the callcontrol config.ini
[CallControl]
group   = root
[OpenSIPS]
socket_path = /tmp
;max_connections = 10 -- does it disconnect 10 calls in 1 time, if i need 100 
terminated i need to increase to 100, right???

how to fix the issue

thank you
Ha



- Original Message 
From: ha do 
To: OpenSIPS users mailling list 
Sent: Wed, November 17, 2010 10:07:16 AM
Subject: Re: [OpenSIPS-Users] need help on connection CDRTool to opensips

Hi Tijimen


sorry for late reply


> 2 opensips2 is a gateway
>endpoint-opensips1---opensips2internet-VOIP service 
provider
So basically opensips2 is just a gateway to opensips1?
[ha] yes,



> i just try to see what happen in cdrtool, call from subscriber to subscriber
> 
> these are the logs
> Nov 11 12:39:46 opensips cdrtool[2302]: MaxSessionTime Duration=36000
> callid=1289463574-3016-hiep...@192.168.1.36 From=sip:843...@192.168.1.39
> Gateway=192.168.1.36 To=sip:00842...@192.168.1.39
> Nov 11 12:39:46 opensips cdrtool[2302]: MaxSessionTime=unlimited Type=postpaid
> callid=1289463574-3016-hiep...@192.168.1.36 billingparty=843...@192.168.1.39
> 
> log of call-control
> Call id 1289463575-3016-hiep...@192.168.1.36 of 843...@192.168.1.39 to
> sip:00842...@192.168.1.39 is postpaid not limited
> 
> it always return postpaid
You need to specify that a sip account is prepaid if you want it to act like a 
prepaid account.

[ha] it is done and work perfectly except 1 thing

Call id NTJjNzY1NzI1MzgwNmQ2MmU4NDMxMWIzZDgzNWFkOTI. of 1...@cucku.com to 
sip:00842...@cucku.com started for maximum 60 seconds
Call id NTJjNzY1NzI1MzgwNmQ2MmU4NDMxMWIzZDgzNWFkOTI. of 1...@cucku.com to 
sip:00842...@cucku.com connected for maximum 60 seconds
callcontrol.opensips.UNIXSocketProtocol starting on 
'/var/run/callcontrol/opensips_01.sock'
error: cannot write request to /tmp: Connection refused
error: failed to end dialog: Cannot send request to OpenSIPS
Call id NTJjNzY1NzI1MzgwNmQ2MmU4NDMxMWIzZDgzNWFkOTI. of 1...@cucku.com to 
sip:00842...@cucku.com disconnected by call control after 60 seconds, call 
price 


is 0.0998

prepaid account 100 has only 60s to connect the call, and after 60s callcontrol 
cannot disconnected call

the mi_fifo located in /tmp

and opensips.cfg with fifo config
# - mi_fifo params -
modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")
modparam("mi_fifo", "fifo_mode", 0600)
modparam("mi_fifo", "fifo_group", 0)
modparam("mi_fifo", "fifo_group", "root")
modparam("mi_fifo", "fifo_user", 0)
modparam("mi_fifo", "fifo_user", "root")

and the callcontrol config.ini
[CallControl]
group   = root
[OpenSIPS]
socket_path = /tmp
;max_connections = 10 -- does it disconnect 10 calls in 1 time, if i need 100 
terminated i need to increase to 100, right???

how to fix the 

Re: [OpenSIPS-Users] need help on connection CDRTool to opensips

[ha do]

Hi all

can anyone help me on call-control, i already try the 
mi_datagrambut i get the same error, permission denied

Call id NTJjNzY1NzI1MzgwNmQ2MmU4NDMxMWIzZDgzNWFkOTI. of 1...@cucku.com to 
sip:00842...@cucku.com started for maximum 60 seconds
Call id NTJjNzY1NzI1MzgwNmQ2MmU4NDMxMWIzZDgzNWFkOTI. of 1...@cucku.com to 
sip:00842...@cucku.com connected for maximum 60 seconds
callcontrol.opensips.UNIXSocketProtocol starting on 
'/var/run/callcontrol/opensips_01.sock'
error: cannot write request to /tmp: Connection refused
error: failed to end dialog: Cannot send request to OpenSIPS
Call id NTJjNzY1NzI1MzgwNmQ2MmU4NDMxMWIzZDgzNWFkOTI. of 1...@cucku.com to 
sip:00842...@cucku.com disconnected by call control after 60 seconds, call 
price 
is 0.0998

prepaid account 100 has only 60s to connect the call, and after 60s callcontrol 
cannot disconnected call

the mi_fifo located in /tmp

and opensips.cfg with fifo config
# - mi_fifo params -
modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")
modparam("mi_fifo", "fifo_mode", 0600)
modparam("mi_fifo", "fifo_group", 0)
modparam("mi_fifo", "fifo_group", "root")
modparam("mi_fifo", "fifo_user", 0)
modparam("mi_fifo", "fifo_user", "root")

and the callcontrol config.ini
[CallControl]
group   = root
[OpenSIPS]
socket_path = /tmp
;max_connections = 10 -- does it disconnect 10 calls in 1 time, if i need 100 
terminated i need to increase to 100, right???

how to fix the issue

thank you
Ha



- Original Message 
From: ha do 
To: OpenSIPS users mailling list 
Sent: Wed, November 17, 2010 10:07:16 AM
Subject: Re: [OpenSIPS-Users] need help on connection CDRTool to opensips

Hi Tijimen


sorry for late reply


> 2 opensips2 is a gateway
>endpoint-opensips1---opensips2internet-VOIP service 
provider
So basically opensips2 is just a gateway to opensips1?
[ha] yes,



> i just try to see what happen in cdrtool, call from subscriber to subscriber
> 
> these are the logs
> Nov 11 12:39:46 opensips cdrtool[2302]: MaxSessionTime Duration=36000
> callid=1289463574-3016-hiep...@192.168.1.36 From=sip:843...@192.168.1.39
> Gateway=192.168.1.36 To=sip:00842...@192.168.1.39
> Nov 11 12:39:46 opensips cdrtool[2302]: MaxSessionTime=unlimited Type=postpaid
> callid=1289463574-3016-hiep...@192.168.1.36 billingparty=843...@192.168.1.39
> 
> log of call-control
> Call id 1289463575-3016-hiep...@192.168.1.36 of 843...@192.168.1.39 to
> sip:00842...@192.168.1.39 is postpaid not limited
> 
> it always return postpaid
You need to specify that a sip account is prepaid if you want it to act like a 
prepaid account.

[ha] it is done and work perfectly except 1 thing

Call id NTJjNzY1NzI1MzgwNmQ2MmU4NDMxMWIzZDgzNWFkOTI. of 1...@cucku.com to 
sip:00842...@cucku.com started for maximum 60 seconds
Call id NTJjNzY1NzI1MzgwNmQ2MmU4NDMxMWIzZDgzNWFkOTI. of 1...@cucku.com to 
sip:00842...@cucku.com connected for maximum 60 seconds
callcontrol.opensips.UNIXSocketProtocol starting on 
'/var/run/callcontrol/opensips_01.sock'
error: cannot write request to /tmp: Connection refused
error: failed to end dialog: Cannot send request to OpenSIPS
Call id NTJjNzY1NzI1MzgwNmQ2MmU4NDMxMWIzZDgzNWFkOTI. of 1...@cucku.com to 
sip:00842...@cucku.com disconnected by call control after 60 seconds, call 
price 

is 0.0998

prepaid account 100 has only 60s to connect the call, and after 60s callcontrol 
cannot disconnected call

the mi_fifo located in /tmp

and opensips.cfg with fifo config
# - mi_fifo params -
modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")
modparam("mi_fifo", "fifo_mode", 0600)
modparam("mi_fifo", "fifo_group", 0)
modparam("mi_fifo", "fifo_group", "root")
modparam("mi_fifo", "fifo_user", 0)
modparam("mi_fifo", "fifo_user", "root")

and the callcontrol config.ini
[CallControl]
group   = root
[OpenSIPS]
socket_path = /tmp
;max_connections = 10 -- does it disconnect 10 calls in 1 time, if i need 100 
terminated i need to increase to 100, right???

how to fix the issue


Thank you
Ha`

- Original Message 
From: Tijmen de Mes 
To: OpenSIPS users mailling list 
Sent: Mon, November 15, 2010 4:11:20 PM
Subject: Re: [OpenSIPS-Users] need help on connection CDRTool to opensips

Hi,

> 
> The following steps are performed to rate a CDR:
> 
> 1. Determination of the billing party
> a. SIP account u...@domain
> b. SIP domain of the SIP account
> c. Source IP of the session
>d. Default (when none of the above matches)
> if there is no sip account u...@domain in cdrtool the cdrtool try next
> option(sip domain) if sip domain does not exist it try next

Re: [OpenSIPS-Users] how to add new domain in CDRTool

[ha do]

Hi Adrian

got it


Thank you



- Original Message 
From: Adrian Georgescu 
To: OpenSIPS users mailling list 
Sent: Thu, November 18, 2010 3:42:35 AM
Subject: Re: [OpenSIPS-Users] how to add new domain in CDRTool

An example for a valid domain name is 'example.com'. You have tried to add an 
IP 
address, which is not the same as a domain name.

Adrian

On Nov 10, 2010, at 6:46 PM, ha do wrote:

> hi all
> 
> how can i add new domain in CDRTool on Rating menu
> because when i add new domain the CDRTool always said error
> 
> Error: value '192.168.1.41' for field 'Domain' must be of format 
> 'example.com' 

> 
> 
> Thank you
> 
> 
> 
> 
> 
> ___
> 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



  

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

Re: [OpenSIPS-Users] need help on connection CDRTool to opensips

[ha do]

Hi Tijimen


sorry for late reply


> 2 opensips2 is a gateway
>endpoint-opensips1---opensips2internet-VOIP service 
provider
So basically opensips2 is just a gateway to opensips1?
[ha] yes,



> i just try to see what happen in cdrtool, call from subscriber to subscriber
> 
> these are the logs
> Nov 11 12:39:46 opensips cdrtool[2302]: MaxSessionTime Duration=36000
> callid=1289463574-3016-hiep...@192.168.1.36 From=sip:843...@192.168.1.39
> Gateway=192.168.1.36 To=sip:00842...@192.168.1.39
> Nov 11 12:39:46 opensips cdrtool[2302]: MaxSessionTime=unlimited Type=postpaid
> callid=1289463574-3016-hiep...@192.168.1.36 billingparty=843...@192.168.1.39
> 
> log of call-control
> Call id 1289463575-3016-hiep...@192.168.1.36 of 843...@192.168.1.39 to
> sip:00842...@192.168.1.39 is postpaid not limited
> 
> it always return postpaid
You need to specify that a sip account is prepaid if you want it to act like a 
prepaid account.

[ha] it is done and work perfectly except 1 thing

Call id NTJjNzY1NzI1MzgwNmQ2MmU4NDMxMWIzZDgzNWFkOTI. of 1...@cucku.com to 
sip:00842...@cucku.com started for maximum 60 seconds
Call id NTJjNzY1NzI1MzgwNmQ2MmU4NDMxMWIzZDgzNWFkOTI. of 1...@cucku.com to 
sip:00842...@cucku.com connected for maximum 60 seconds
callcontrol.opensips.UNIXSocketProtocol starting on 
'/var/run/callcontrol/opensips_01.sock'
error: cannot write request to /tmp: Connection refused
error: failed to end dialog: Cannot send request to OpenSIPS
Call id NTJjNzY1NzI1MzgwNmQ2MmU4NDMxMWIzZDgzNWFkOTI. of 1...@cucku.com to 
sip:00842...@cucku.com disconnected by call control after 60 seconds, call 
price 
is 0.0998

prepaid account 100 has only 60s to connect the call, and after 60s callcontrol 
cannot disconnected call

the mi_fifo located in /tmp

and opensips.cfg with fifo config
# - mi_fifo params -
modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")
modparam("mi_fifo", "fifo_mode", 0600)
modparam("mi_fifo", "fifo_group", 0)
modparam("mi_fifo", "fifo_group", "root")
modparam("mi_fifo", "fifo_user", 0)
modparam("mi_fifo", "fifo_user", "root")

and the callcontrol config.ini
[CallControl]
group   = root
[OpenSIPS]
socket_path = /tmp
;max_connections = 10 -- does it disconnect 10 calls in 1 time, if i need 100 
terminated i need to increase to 100, right???

how to fix the issue


Thank you
Ha`

- Original Message 
From: Tijmen de Mes 
To: OpenSIPS users mailling list 
Sent: Mon, November 15, 2010 4:11:20 PM
Subject: Re: [OpenSIPS-Users] need help on connection CDRTool to opensips

Hi,

> 
> The following steps are performed to rate a CDR:
> 
> 1. Determination of the billing party
> a. SIP account u...@domain
> b. SIP domain of the SIP account
> c. Source IP of the session
>d. Default (when none of the above matches)
> if there is no sip account u...@domain in cdrtool the cdrtool try next
> option(sip domain) if sip domain does not exist it try next option(source 
>ip)and
> then the default option
> please let me know if i am wrong
This is correct.

> 2. Determination of the destination id
> a. CanonicalURI (the destination after all lookups inside the SIP Proxy)
> b. SipTranslatedRequestURI (the Request URI as presented by the SIP UA)
> c. CalledStationId (the content of the To header, used as a last resort)
> if the CanonicalURI does not exist it try to next option(Request URI) ...
> please let me know if i am wrong
This is also correct.

> 3. Determination of the costs
> 2 opensips2 is a gateway
>endpoint-opensips1---opensips2internet-VOIP service 
provider
So basically opensips2 is just a gateway to opensips1?

> 
> how can i config cdrtool rating all calls from opensips1 and opensips2
> could you please guide me some steps to do
Did you load the sample data so you can see if a call gets rated?

> 
> i just try to see what happen in cdrtool, call from subscriber to subscriber
> 
> these are the logs
> Nov 11 12:39:46 opensips cdrtool[2302]: MaxSessionTime Duration=36000
> callid=1289463574-3016-hiep...@192.168.1.36 From=sip:843...@192.168.1.39
> Gateway=192.168.1.36 To=sip:00842...@192.168.1.39
> Nov 11 12:39:46 opensips cdrtool[2302]: MaxSessionTime=unlimited Type=postpaid
> callid=1289463574-3016-hiep...@192.168.1.36 billingparty=843...@192.168.1.39
> 
> log of call-control
> Call id 1289463575-3016-hiep...@192.168.1.36 of 843...@192.168.1.39 to
> sip:00842...@192.168.1.39 is postpaid not limited
> 
> it always return postpaid
You need to specify that a sip account is prepaid if you want it to act like a 
prepaid account.

--
Tijmen de Mes
AG Projects


___
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

Re: [OpenSIPS-Users] need help on connection CDRTool to opensips

[ha do]

Hi Tijmen

Maybe you forgot to remove the bind interfaces directive in the mysql config.
that is useful information, i edit the my.cnf and now it works fine now

i read the RATING.txt, PREPAID.txt times to times but i cannot understand it


The following steps are performed to rate a CDR:

1. Determination of the billing party
a. SIP account u...@domain 
b. SIP domain of the SIP account
c. Source IP of the session
d. Default (when none of the above matches)
if there is no sip account u...@domain in cdrtool the cdrtool try next 
option(sip domain) if sip domain does not exist it try next option(source 
ip)and 
then the default option
please let me know if i am wrong

2. Determination of the destination id
a. CanonicalURI (the destination after all lookups inside the SIP Proxy) 
b. SipTranslatedRequestURI (the Request URI as presented by the SIP UA)
c. CalledStationId (the content of the To header, used as a last resort)
if the CanonicalURI does not exist it try to next option(Request URI) ...
please let me know if i am wrong

3. Determination of the costs


i try to test 
1. opensips1 is the registrar server and try to do prepaid for every 
subscribers 
of opensips
subscriber registers to opensips1, opensips1 domain 192.168.1.39

2 opensips2 is a gateway
endpoint-opensips1---opensips2internet-VOIP service provider

how can i config cdrtool rating all calls from opensips1 and opensips2
could you please guide me some steps to do

i did check the global.inc and cdr_generic.php 
-global.inc is a copy of global.inc.simple.sample and i change 
# Normalize engine settings
$CDRTool['normalize']['defaultCountryCode']  = "84";
-cdr-generic.php does not change


i config  in web

the destination
Ops,Reseller,Trusted 
peer,Domain,Subscriber,Destination,Region,Description,Incr,Min Dur,Max Dur,Max 
Price
2,084,,vietnam,6,6,0,

rate
Ops,Reseller,Rate,Destination,App,Connect,Duration,Conn In,Duration In
2,0,84,84,audio,0,998,,

profile
Ops,Reseller,Profile,Rate 1,00-H1,Rate 2,H1-H2,Rate 3,H2-H3,Rate 4,H3-24
2,0,84,84,24,0,0,,0,,0


i just try to see what happen in cdrtool, call from subscriber to subscriber

these are the logs
Nov 11 12:39:46 opensips cdrtool[2302]: MaxSessionTime Duration=36000 
callid=1289463574-3016-hiep...@192.168.1.36 From=sip:843...@192.168.1.39 
Gateway=192.168.1.36 To=sip:00842...@192.168.1.39
Nov 11 12:39:46 opensips cdrtool[2302]: MaxSessionTime=unlimited Type=postpaid 
callid=1289463574-3016-hiep...@192.168.1.36 billingparty=843...@192.168.1.39

log of call-control
Call id 1289463575-3016-hiep...@192.168.1.36 of 843...@192.168.1.39 to 
sip:00842...@192.168.1.39 is postpaid not limited

it always return postpaid

Thank you




- Original Message 
From: Tijmen de Mes 
To: OpenSIPS users mailling list 
Sent: Thu, November 11, 2010 4:42:49 PM
Subject: Re: [OpenSIPS-Users] need help on connection CDRTool to opensips

Hi,

You should verify by hand that you can connect to the database by
executing a mysql -h 192.168.1.39 -u opensips -psips  opensips.

Maybe you forgot to remove the bind interfaces directive in the mysql config.

Best regards,

Tijmen de Mes
AG Projects

On Nov 9, 2010, at 4:11 AM, ha do wrote:

> Hi all
> 
> i installed opensips and mediaproxy in server 1, ip address : 192.168.1.39
> and cdrtool and freeradius in server 2, ip address : 192.168.1.42
> 
> the opensips, mediaproxy, freeradius starting are fine
> 
> 
> i use the link below as a guide
> http://cdrtool.ag-projects.com/browser/doc/INSTALL.txt
> 
> the problem i got only with cdrtool
> Nov  9 04:56:22 cdrtool cdrtool[3219]: Read 12291 PSTN destinations from cache
> in 0 seconds
> Nov  9 04:56:22 cdrtool cdrtool[3219]: Error in connect(): pconnect as 
opensips
> to 192.168.1.39 failed
> Nov  9 04:56:22 cdrtool cdrtool[3219]: Database DB_opensips error:  (0) 
> select 
>*
> from domain
> Nov  9 04:56:22 cdrtool cdrtool[3219]: Error in connect(): pconnect as 
opensips
> to 192.168.1.39 failed
> Nov  9 04:56:22 cdrtool cdrtool[3219]: Database DB_opensips error:  (0) 
> select 
>*
> from trusted
> Nov  9 04:56:22 cdrtool cdrtool[3219]: PSTN destinations cache size: 0.64 MB
> Nov  9 04:56:22 cdrtool cdrtool[3219]: SIP destinations cache size: 0.00 MB
> Nov  9 04:56:22 cdrtool cdrtool[3219]: Cached 12291 total, 12291 default, 0
> gateway, 0 domain, 0 subscriber destinations
> Nov  9 04:56:22 cdrtool cdrtool[3219]: Cached 0 SIP destinations
> 
> 
> the cdrtool is fine with almost component except the connection to opensips
> the global.inc with define opensips server
> class DB_opensips extends DB_Sql {
> var $Host = "192.168.1.39";
> var $Database = "opensips";
> var $User = "opensips";
> var $Password = "sips";
> var $Halt_On_Error = "no";
> }
> 
> and issue the c

[OpenSIPS-Users] need help on CDRTool

[ha do]

hi 

i try to learn CDRTool, 

i config the global.inc  with 
# Normalize engine settings
$CDRTool['normalize']['defaultCountryCode']  = "84";

the cdr_generic.php 
class CDRS {

var $CDR_class   = 'CDR';
var $intAccessCode   = '00';
var $natAccessCode   = '0';
var $maxrowsperpage  = 15;
var $status  = array();
var $normalizedField = 'Normalized';
var $DestinationIdField  = 'DestinationId';
var $BillingIdField  = 'UserName';

i config  in web

the destination
Ops,Reseller,Trusted 
peer,Domain,Subscriber,Destination,Region,Description,Incr,Min Dur,Max Dur,Max 
Price
2,084,,vietnam,6,6,0,

rate
Ops,Reseller,Rate,Destination,App,Connect,Duration,Conn In,Duration In
2,0,84,84,audio,0,998,,

profile
Ops,Reseller,Profile,Rate 1,00-H1,Rate 2,H1-H2,Rate 3,H2-H3,Rate 4,H3-24
2,0,84,84,24,0,0,,0,,0


when i make call and get the syslog of cdrtool
Nov 11 06:20:57 opensips cdrtool[2312]: MaxSessionTime Duration=36000 
callid=1289440810-2424-hiep...@192.168.1.36 From=sip:843...@192.168.1.41 
Gateway=192.168.1.36 To=sip:00842...@192.168.1.41
Nov 11 06:20:57 opensips cdrtool[2312]: MaxSessionTime=unlimited Type=postpaid 
callid=1289440810-2424-hiep...@192.168.1.36 billingparty=843...@192.168.1.41

get the log from callcontrol
Call id 1289440810-2424-hiep...@192.168.1.36 of 843...@192.168.1.41 to 
sip:00842...@192.168.1.41 is postpaid not limited


what do i miss thing to make CDRTool rating a call, please help

Thank you
Ha`



  

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

[OpenSIPS-Users] how to add new domain in CDRTool

[ha do]

hi all

how can i add new domain in CDRTool on Rating menu
because when i add new domain the CDRTool always said error

Error: value '192.168.1.41' for field 'Domain' must be of format 'example.com' 


Thank you



  

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

[OpenSIPS-Users] need help on connection CDRTool to opensips

[ha do]

Hi all

i installed opensips and mediaproxy in server 1, ip address : 192.168.1.39
and cdrtool and freeradius in server 2, ip address : 192.168.1.42

the opensips, mediaproxy, freeradius starting are fine


i use the link below as a guide
http://cdrtool.ag-projects.com/browser/doc/INSTALL.txt

the problem i got only with cdrtool
Nov  9 04:56:22 cdrtool cdrtool[3219]: Read 12291 PSTN destinations from cache 
in 0 seconds
Nov  9 04:56:22 cdrtool cdrtool[3219]: Error in connect(): pconnect as opensips 
to 192.168.1.39 failed
Nov  9 04:56:22 cdrtool cdrtool[3219]: Database DB_opensips error:  (0) select 
* 
from domain
Nov  9 04:56:22 cdrtool cdrtool[3219]: Error in connect(): pconnect as opensips 
to 192.168.1.39 failed
Nov  9 04:56:22 cdrtool cdrtool[3219]: Database DB_opensips error:  (0) select 
* 
from trusted
Nov  9 04:56:22 cdrtool cdrtool[3219]: PSTN destinations cache size: 0.64 MB
Nov  9 04:56:22 cdrtool cdrtool[3219]: SIP destinations cache size: 0.00 MB
Nov  9 04:56:22 cdrtool cdrtool[3219]: Cached 12291 total, 12291 default, 0 
gateway, 0 domain, 0 subscriber destinations
Nov  9 04:56:22 cdrtool cdrtool[3219]: Cached 0 SIP destinations


the cdrtool is fine with almost component except the connection to opensips
the global.inc with define opensips server
class DB_opensips extends DB_Sql {
  var $Host = "192.168.1.39";
  var $Database = "opensips";
  var $User = "opensips";
  var $Password = "sips";
  var $Halt_On_Error = "no";
}

and issue the command  grant all on opensips.* to opensips@'192.168.1.%' 
identified by 'sips'; in opensips server

please help

Ha`



  

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

Re: [OpenSIPS-Users] cdrtool install problem

[ha do]

HI Tijmen

i insert 2 lines in source.list and it is fine

deb http://ftp.us.debian.org/debian/ testing main
deb-src http://ftp.us.debian.org/debian/ testing main


Thank you
Ha


- Original Message 
From: ha do 
To: OpenSIPS users mailling list 
Sent: Wed, November 3, 2010 8:09:42 AM
Subject: Re: [OpenSIPS-Users] cdrtool install problem

Hi Tijmen

here is the source.list

cdrtool:~# uname -a
Linux cdrtool 2.6.26-2-686 #1 SMP Thu Sep 16 19:35:51 UTC 2010 i686 GNU/Linux

cdrtool:~# cat /etc/apt/sources.list
#
# deb cdrom:[Debian GNU/Linux 5.0.6 _Lenny_ - Official i386 CD Binary-1 
20100904-18:58]/ lenny main

deb cdrom:[Debian GNU/Linux 5.0.6 _Lenny_ - Official i386 CD Binary-1 
20100904-18:58]/ lenny main


deb http://ftp.us.debian.org/debian/ lenny main
deb-src http://ftp.us.debian.org/debian/ lenny main

deb http://ag-projects.com/debian unstable main
deb-src http://ag-projects.com/debian unstable main

deb http://security.debian.org/ lenny/updates main
deb-src http://security.debian.org/ lenny/updates main

deb http://volatile.debian.org/debian-volatile lenny/volatile main
deb-src http://volatile.debian.org/debian-volatile lenny/volatile main




let me try on Debian unstable

Thank you
Ha


- Original Message 
From: Tijmen de Mes 
To: OpenSIPS users mailling list 
Sent: Wed, November 3, 2010 2:38:10 AM
Subject: Re: [OpenSIPS-Users] cdrtool install problem

Hi,

I am unable to reproduce this,. I can install php5-geoip, geoip- 
database and cdrtool fine on Debian unstable.
Which version of Debian are you using?

Best regards,

Tijmen de Mes
AG Projects

On Nov 2, 2010, at 4:03 PM, ha do wrote:

> Hi
>
> i try to install cdrtool in debian OS
> but i got stuck when install cdrtool
>
> cdrtool:/mnt# apt-get install php5-geoip
> Reading package lists... Done
> Building dependency tree
> Reading state information... Done
> The following extra packages will be installed:
>  libgeoip1
> Suggested packages:
>  geoip-bin
> The following packages will be REMOVED:
>  geoip-database
> The following NEW packages will be installed:
>  libgeoip1 php5-geoip
> 0 upgraded, 2 newly installed, 1 to remove and 0 not upgraded.
> Need to get 0B/660kB of archives.
> After this operation, 4403kB disk space will be freed.
> Do you want to continue [Y/n]?
>
> cdrtool requires php5-geoip + geoip-database
> but the php5-geoip  dependent on libgeoip1. when i install libgeoip1  
> it remove
> geoip-database
>
> Please help
>
> Thank you
> Ha`
>
>
>
>
> ___
> 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



  

___
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

Re: [OpenSIPS-Users] cdrtool install problem

[ha do]

Hi Tijmen

here is the source.list

cdrtool:~# uname -a
Linux cdrtool 2.6.26-2-686 #1 SMP Thu Sep 16 19:35:51 UTC 2010 i686 GNU/Linux

cdrtool:~# cat /etc/apt/sources.list
#
# deb cdrom:[Debian GNU/Linux 5.0.6 _Lenny_ - Official i386 CD Binary-1 
20100904-18:58]/ lenny main

deb cdrom:[Debian GNU/Linux 5.0.6 _Lenny_ - Official i386 CD Binary-1 
20100904-18:58]/ lenny main


deb http://ftp.us.debian.org/debian/ lenny main
deb-src http://ftp.us.debian.org/debian/ lenny main

deb http://ag-projects.com/debian unstable main
deb-src http://ag-projects.com/debian unstable main

deb http://security.debian.org/ lenny/updates main
deb-src http://security.debian.org/ lenny/updates main

deb http://volatile.debian.org/debian-volatile lenny/volatile main
deb-src http://volatile.debian.org/debian-volatile lenny/volatile main




let me try on Debian unstable

Thank you
Ha


- Original Message 
From: Tijmen de Mes 
To: OpenSIPS users mailling list 
Sent: Wed, November 3, 2010 2:38:10 AM
Subject: Re: [OpenSIPS-Users] cdrtool install problem

Hi,

I am unable to reproduce this,. I can install php5-geoip, geoip- 
database and cdrtool fine on Debian unstable.
Which version of Debian are you using?

Best regards,

Tijmen de Mes
AG Projects

On Nov 2, 2010, at 4:03 PM, ha do wrote:

> Hi
>
> i try to install cdrtool in debian OS
> but i got stuck when install cdrtool
>
> cdrtool:/mnt# apt-get install php5-geoip
> Reading package lists... Done
> Building dependency tree
> Reading state information... Done
> The following extra packages will be installed:
>  libgeoip1
> Suggested packages:
>  geoip-bin
> The following packages will be REMOVED:
>  geoip-database
> The following NEW packages will be installed:
>  libgeoip1 php5-geoip
> 0 upgraded, 2 newly installed, 1 to remove and 0 not upgraded.
> Need to get 0B/660kB of archives.
> After this operation, 4403kB disk space will be freed.
> Do you want to continue [Y/n]?
>
> cdrtool requires php5-geoip + geoip-database
> but the php5-geoip  dependent on libgeoip1. when i install libgeoip1  
> it remove
> geoip-database
>
> Please help
>
> Thank you
> Ha`
>
>
>
>
> ___
> 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



  

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

[OpenSIPS-Users] cdrtool install problem

[ha do]

Hi 

i try to install cdrtool in debian OS
but i got stuck when install cdrtool

cdrtool:/mnt# apt-get install php5-geoip
Reading package lists... Done
Building dependency tree
Reading state information... Done
The following extra packages will be installed:
  libgeoip1
Suggested packages:
  geoip-bin
The following packages will be REMOVED:
  geoip-database
The following NEW packages will be installed:
  libgeoip1 php5-geoip
0 upgraded, 2 newly installed, 1 to remove and 0 not upgraded.
Need to get 0B/660kB of archives.
After this operation, 4403kB disk space will be freed.
Do you want to continue [Y/n]?

cdrtool requires php5-geoip + geoip-database
but the php5-geoip  dependent on libgeoip1. when i install libgeoip1 it remove 
geoip-database

Please help

Thank you
Ha`


  

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

Re: [OpenSIPS-Users] get error with opensip-cp on centos 5

[ha do]

if someone else got the same problem with me, to solve the issue please follow 
the link 

http://lists.opensips.org/pipermail/users/2010-June/013139.html

Thank you
Ha`



- Original Message 
From: ha do 
To: OpenSIPS 
Sent: Tue, September 14, 2010 11:12:47 AM
Subject: [OpenSIPS-Users] get error  with opensip-cp on centos 5

Hi all

i try to use the opensips-cp 4:
i login the the web page http://192.168.1.100/cp successfull 

the menu Admin + User are working properly
but the system menu does not work
i get the error on the webpage: 

sorry -- cannot open write fifo

and this is the error in debug message : 
Sep 14 00:10:42 localhost kernel: type=1400 audit(1284437442.630:49): avc:  
denied  { getattr } for  pid=2093 comm="httpd" path="/tmp/opensips_fifo" 
dev=dm-0 ino=884744 scontext=root:system_r:httpd_t:s0 
tcontext=root:object_r:tmp_t:s0 tclass=fifo_file


i did set fifo_mode, 0666 in opensips.cfg

please help,what should i do to resolve problem

Thank you
Ha`



  

___
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

Re: [OpenSIPS-Users] problem with opensips startup on Centos 5

[ha do]

Hi all
i think the script file in packaging/fedora/opensips.init is not create the 
/var/run/opensips.pid
it is the same problem with 

Chrispy? — 01 September 2010, 01:46on webpage 
http://www.opensips.org/Resources/DocsTsStart

for workaround :
service opensips stop

then issue : opensipsctl start -> /var/run/opensips.pid should create 
successfull
but when server restart i cannot use the munin service

could someone help me

Thank you
Ha`




- Original Message 
From: ha do 
To: OpenSIPS 
Sent: Tue, September 14, 2010 10:05:20 AM
Subject: [OpenSIPS-Users] problem with opensips startup on Centos 5

Hello all

i use Centos 5:
[r...@localhost ~]# uname -a
Linux localhost.localdomain 2.6.18-194.el5 #1 SMP Fri Apr 2 14:58:35 EDT 2010 
i686 i686 i386 GNU/Linux

i am testing the opensips 1.6.2 and i installed successfully opensips

the step to get problem:
  1. Install opensips successfull
   2. copy file packaging/fedora/opensips.init to /etc/init.d/opensips
  3. chmod 755 /etc/init.d/opensips
   4. chkconfig --add /etc/init.d/opensips
   5. restart Server 
   6. check the opensips status:
[r...@localhost ~]# ps -ef | grep opensips
root  1539 1  0 22:56 ?00:00:00 /usr/local/sbin/opensips
root  1543  1539  0 22:56 ?00:00:00 /usr/local/sbin/opensips
root  1544  1539  0 22:56 ?00:00:00 /usr/local/sbin/opensips
root  1545  1539  0 22:56 ?00:00:00 /usr/local/sbin/opensips
root  1546  1539  0 22:56 ?00:00:00 /usr/local/sbin/opensips
root  1547  1539  0 22:56 ?00:00:00 /usr/local/sbin/opensips
root  1548  1539  0 22:56 ?00:00:00 /usr/local/sbin/opensips
root  1574  1539  0 22:56 ?00:00:00 /usr/local/sbin/opensips
root  1575  1539  0 22:56 ?00:00:00 /usr/local/sbin/opensips
root  1576  1539  0 22:56 ?00:00:00 /usr/local/sbin/opensips
root  1577  1539  0 22:56 ?00:00:00 /usr/local/sbin/opensips
root  1578  1539  0 22:56 ?00:00:00 /usr/local/sbin/opensips
root  1579  1539  0 22:56 ?00:00:00 /usr/local/sbin/opensips
root  1599  1539  0 22:56 ?00:00:00 /usr/local/sbin/opensips
root  1600  1539  0 22:56 ?00:00:00 /usr/local/sbin/opensips
root  1601  1539  0 22:56 ?00:00:00 /usr/local/sbin/opensips
root  1602  1539  0 22:56 ?00:00:00 /usr/local/sbin/opensips
root  1650  1627  0 22:57 pts/000:00:00 grep opensips

7. i use the opensipsctl restart and got the problem :
[r...@localhost ~]# opensipsctl restart

INFO: Restarting OpenSIPS :

ERROR: No PID file found (/var/run/opensips.pid)! OpenSIPS probably not running

INFO: Starting OpenSIPS :

ERROR: PID file /var/run/opensips.pid does not exist -- OpenSIPS start failed


could someone help me, i want opensips startup along with Server and the 
opensipsctl command should work properly

Thank you
Ha`



  

___
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

[OpenSIPS-Users] get error with opensip-cp on centos 5

[ha do]

Hi all

i try to use the opensips-cp 4:
i login the the web page http://192.168.1.100/cp successfull 

the menu Admin + User are working properly
but the system menu does not work
i get the error on the webpage: 

sorry -- cannot open write fifo

and this is the error in debug message : 
Sep 14 00:10:42 localhost kernel: type=1400 audit(1284437442.630:49): avc:  
denied  { getattr } for  pid=2093 comm="httpd" path="/tmp/opensips_fifo" 
dev=dm-0 ino=884744 scontext=root:system_r:httpd_t:s0 
tcontext=root:object_r:tmp_t:s0 tclass=fifo_file


i did set fifo_mode, 0666 in opensips.cfg

please help,what should i do to resolve problem

Thank you
Ha`



  

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

[OpenSIPS-Users] problem with opensips startup on Centos 5

[ha do]

Hello all

i use Centos 5:
[r...@localhost ~]# uname -a
Linux localhost.localdomain 2.6.18-194.el5 #1 SMP Fri Apr 2 14:58:35 EDT 2010 
i686 i686 i386 GNU/Linux

i am testing the opensips 1.6.2 and i installed successfully opensips

the step to get problem:
  1. Install opensips successfull
   2. copy file packaging/fedora/opensips.init to /etc/init.d/opensips
  3. chmod 755 /etc/init.d/opensips
   4. chkconfig --add /etc/init.d/opensips
   5. restart Server 
   6. check the opensips status:
[r...@localhost ~]# ps -ef | grep opensips
root  1539 1  0 22:56 ?00:00:00 /usr/local/sbin/opensips
root  1543  1539  0 22:56 ?00:00:00 /usr/local/sbin/opensips
root  1544  1539  0 22:56 ?00:00:00 /usr/local/sbin/opensips
root  1545  1539  0 22:56 ?00:00:00 /usr/local/sbin/opensips
root  1546  1539  0 22:56 ?00:00:00 /usr/local/sbin/opensips
root  1547  1539  0 22:56 ?00:00:00 /usr/local/sbin/opensips
root  1548  1539  0 22:56 ?00:00:00 /usr/local/sbin/opensips
root  1574  1539  0 22:56 ?00:00:00 /usr/local/sbin/opensips
root  1575  1539  0 22:56 ?00:00:00 /usr/local/sbin/opensips
root  1576  1539  0 22:56 ?00:00:00 /usr/local/sbin/opensips
root  1577  1539  0 22:56 ?00:00:00 /usr/local/sbin/opensips
root  1578  1539  0 22:56 ?00:00:00 /usr/local/sbin/opensips
root  1579  1539  0 22:56 ?00:00:00 /usr/local/sbin/opensips
root  1599  1539  0 22:56 ?00:00:00 /usr/local/sbin/opensips
root  1600  1539  0 22:56 ?00:00:00 /usr/local/sbin/opensips
root  1601  1539  0 22:56 ?00:00:00 /usr/local/sbin/opensips
root  1602  1539  0 22:56 ?00:00:00 /usr/local/sbin/opensips
root  1650  1627  0 22:57 pts/000:00:00 grep opensips

7. i use the opensipsctl restart and got the problem :
[r...@localhost ~]# opensipsctl restart

INFO: Restarting OpenSIPS :

ERROR: No PID file found (/var/run/opensips.pid)! OpenSIPS probably not running

INFO: Starting OpenSIPS :

ERROR: PID file /var/run/opensips.pid does not exist -- OpenSIPS start failed


could someone help me, i want opensips startup along with Server and the 
opensipsctl command should work properly

Thank you
Ha`



  

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

Re: [OpenSIPS-Users] Problem with OpenSIPS Control Panel 4.0

[ha do]

Hi Alex

i try new opensips-cp and it work fine now

but i have problem with add new admin in opensips-cp webpage
step1: add new admin with name : cucku
step2: add successfull
step3: login with username cucku -- the issue is here, there is no available 
tools for new admin which is added via webpage

the available tool for new admin is absent :(,where do i need to add available 
tool for admin with name : cucku

the query from mysql :
mysql> select * from  ocp_admin_privileges\G;
*** 1. row ***
 id: 55
 first_name:
  last_name:
   username: admin
   password: admin
    ha1: d2abaa37a7c3db1137d385e1d8c15fd2
available_tools: all
    permissions: all
*** 2. row ***
 id: 56
 first_name: cucku
  last_name: cucku
   username: cucku
   password:
    ha1: 91730b420f600d8da7817d96c20a59b9
available_tools:
    permissions: NULL
2 rows in set (0.00 sec)

ERROR:
No query specified

Thank you
Ha`

--- On Thu, 3/18/10, Alex Ionescu  wrote:

From: Alex Ionescu 
Subject: Re: [OpenSIPS-Users] Problem with OpenSIPS Control Panel 4.0
To: "OpenSIPS users mailling list" 
Date: Thursday, March 18, 2010, 5:46 AM





  
Hi,



Actually there were a few issues that were found and solved the last
few days. I hope you won't have anymore problems if you download the
new 4.0 tarball. 



So, give it a try and let me know.



Regards,

Alex







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

Re: [OpenSIPS-Users] Problem with OpenSIPS Control Panel 4.0

[ha do]

HI Alex
did you find new issue on opensips cp
Thank youHa`

--- On Sat, 3/13/10, Alex Ionescu  wrote:

From: Alex Ionescu 
Subject: Re: [OpenSIPS-Users] Problem with OpenSIPS Control Panel 4.0
To: "OpenSIPS users mailling list" 
Date: Saturday, March 13, 2010, 9:21 AM





  

 
Hi,



I will investigate the problem and will let you know if I find any
issue.



Regards,

Alex



On 3/13/2010 06:39, Do Nguyen Ha wrote:

  

  


Hi Alex



here is mysql query:

mysql> select * from ocp_admin_privileges\G;

*** 1. row ***

 id: 58

 first_name:

  last_name:

   username: admin

   password: admin

    ha1: d2abaa37a7c3db1137d385e1d8c15fd2

available_tools: all

    permissions: all



so i am still getting the invalid username on opensips-cp version 4

the i remove the folder opensips-cp version 4 and replace with
opensips-cp version 3 and now it works without problem with opensips-cp
version 3



i dont change anything and step i do :

step 1. cd /var/www

step 2. delete folder opensips-cp

step 3. download the opensips-cp_3.0.tar.gz

step 4. untar the file opensips-cp_3.0

step 5. chown www-data:www-data opensips-cp -R



Thank you for your help

Ha



--- On Fri, 3/12/10, Alex Ionescu 
wrote:



From: Alex Ionescu 

Subject: Re: [OpenSIPS-Users] Problem with OpenSIPS Control Panel 4.0

To: "OpenSIPS users mailling list" 

Date: Friday, March 12, 2010, 9:02 AM

  

   Sorry,

  

My mistake ... the MD5 should be applied on the value not on the field
name ... So, it is : 

  

INSERT INTO ocp_admin_privileges
(username,password,ha1,available_tools,permissions) values
('admin','admin',md5('admin:admin'),'all','all');

  

On 3/12/2010 13:06, Alex Ionescu wrote:
  

Hi,



The problem seems to be your "step 11" :



  step 11. INSERT INTO ocp_admin_privileges
(username,password,ha1,available_tools,permissions) values
('admin','admin','admin:admin','all','all');



You must encode the ha1 field. So, I believe that the correct
query would be : 



INSERT INTO ocp_admin_privileges
(username,password,md5(ha1),available_tools,permissions) values
('admin','admin','admin:admin','all','all');



Anyway, you can always check the INSTALL file (I think the query is
located on line 102 or 103).



Regards,



Alex



On 3/12/2010 12:41, Do Nguyen Ha wrote:
  step 11. INSERT INTO
ocp_admin_privileges
(username,password,ha1,available_tools,permissions) values
('admin','admin','admin:admin','all','all');




-- 
Alex Ionescu
www.voice-system.ro 

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

  

  -- 
Alex Ionescu
www.voice-system.ro 
  
  

-Inline Attachment Follows-

  

  ___

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 Ionescu
www.voice-system.ro 
 


-Inline Attachment Follows-

___
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

Re: [OpenSIPS-Users] need help on dialplan module

[ha do]

Hi Bogdan

i refer : String translation (regexp detection, subst translation) function

the repl_exp = a_value\1 the dialplan will use the a_value + subst_exp as the 
output if the match_exp=true

the repl_exp = a_value\2 the dialplan will use the columm a_value + (input 
string - subst_exp) as the output if the match_exp=true

it is right?

Thank you
Ha`



--- On Fri, 1/22/10, Bogdan-Andrei Iancu  wrote:

From: Bogdan-Andrei Iancu 
Subject: Re: [OpenSIPS-Users] need help on dialplan module
To: "OpenSIPS users mailling list" 
Date: Friday, January 22, 2010, 9:55 AM

Hi Ha,

The modules user PERL like substitution. A fast google gives some docs 
on this:
    http://www.anaesthetist.com/mnm/perl/Findex.htm#regex.htm

Regards,
Bogdan

ha do wrote:
> Hi all
>
> could you please need me to understand the translation on dialplan module;
> mysql> select * from dialplan;
> ++--++--+---+---++--+---+
> | id | dpid | pr | match_op | match_exp | match_len | subst_exp  | 
> repl_exp | attrs |
> ++--++--+---+---++--+---+
> | 73 |   15 |  0 |        1 | ^000      |         0 | ^(0)(.+)   | 
> \2       |       |
> | 78 |   16 |  0 |        1 | 000       |         0 | (000)(.+)  | 
> 8\2      |       |
> | 76 |   14 |  0 |        1 | ^000      |         0 | ^(000)(.+) | 
> 8\2      |       |
> | 75 |   15 |  0 |        1 | ^55       |         0 | ^(55)(.+)  | 
> \2       |       |
> ++--++--+---+---++--+---+
>
> [r...@localhost ~]# opensipsctl fifo dp_translate 14 00055980007
> Output:: 855980007
> [r...@localhost ~]# opensipsctl fifo dp_translate 15 0007
> Output:: 007
> [r...@localhost ~]# opensipsctl fifo dp_translate 15 55980007
> Output:: 980007
> [r...@localhost ~]# opensipsctl fifo dp_translate 16 55980007
> Output:: 87
>
> repl_exp : sometimes has value \2 or \1 - what does it mean?? does it 
> have other value?
> what does the ^ mean??
> is there more special character??
>
> where do i find more docs for translation rule
>
> Thank you
> Ha`
>
>
> 
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>   


-- 
Bogdan-Andrei Iancu
www.voice-system.ro


___
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

[OpenSIPS-Users] need help on dialplan module

[ha do]

Hi all

could you please need me to understand the translation on dialplan module;
mysql> select * from dialplan;
++--++--+---+---++--+---+
| id | dpid | pr | match_op | match_exp | match_len | subst_exp  | repl_exp | 
attrs |
++--++--+---+---++--+---+
| 73 |   15 |  0 |    1 | ^000  | 0 | ^(0)(.+)   | \2   
|   |
| 78 |   16 |  0 |    1 | 000   | 0 | (000)(.+)  | 8\2  
|   |
| 76 |   14 |  0 |    1 | ^000  | 0 | ^(000)(.+) | 8\2  
|   |
| 75 |   15 |  0 |    1 | ^55   | 0 | ^(55)(.+)  | \2   
|   |
++--++--+---+---++--+---+

[r...@localhost ~]# opensipsctl fifo dp_translate 14 00055980007
Output:: 855980007
[r...@localhost ~]# opensipsctl fifo dp_translate 15 0007
Output:: 007
[r...@localhost ~]# opensipsctl fifo dp_translate 15 55980007
Output:: 980007
[r...@localhost ~]# opensipsctl fifo dp_translate 16 55980007
Output:: 87

repl_exp : sometimes has value \2 or \1 - what does it mean?? does it have 
other value?
what does the ^ mean??
is there more special character??

where do i find more docs for translation rule

Thank you
Ha`



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

Re: [OpenSIPS-Users] need help on mediaproxy ports

[ha do]

Thank you for the link
Ha`

--- On Tue, 1/19/10, Duane Larson  wrote:

From: Duane Larson 
Subject: Re: [OpenSIPS-Users] need help on mediaproxy ports
To: "OpenSIPS users mailling list" 
Date: Tuesday, January 19, 2010, 8:11 AM

I believe you will always have (x ports * 2) since RTP is always accompanied by 
an odd numbered RTCP port.  So you do have only 2 RTP ports listed above, but 
those RTP ports also require a RTCP port.  Look at this
http://en.wikipedia.org/wiki/RTP_Control_Protocol


On Tue, Jan 19, 2010 at 1:56 AM, ha do  wrote:





Hi all

i debug on mediaproxy and see, mediaproxy create 4 ports but the real ports to 
relay media are only 2 ports: 118.69.239.140:50012 <-> 118.69.239.140:50014


is it normal, can i config mediaproxy create only 2 ports

Thank you
Ha`

mediaproxy.mediacontrol.StreamListenerProtocol starting on 50012
mediaproxy.mediacontrol.StreamListenerProtocol starting on 50013

mediaproxy.mediacontrol.StreamListenerProtocol starting on 50014
mediaproxy.mediacontrol.StreamListenerProtocol starting on 50015
debug: Added new stream: (audio) 192.168.1.4:45746 (RTP: Unknown, RTCP: 
Unknown) <-> 118.69.239.140:50012 <-> 118.69.239.140:50014 <-> Unknown (RTP: 
Unknown, RTCP: Unknown)

debug: created new session 7810dcdd0ceb7...@192.168.1.4: 8...@118.69.239.140 
(93846fc44ae3fe24) --> 9...@118.69.239.140

debug: updating existing session 7810dcdd0ceb7...@192.168.1.4: 
8...@118.69.239.140 (93846fc44ae3fe24) --> 9...@118.69.239.140

debug: Received updated SDP answer
debug: Got initial answer from callee for stream: (audio) 192.168.1.4:45746 
(RTP: Unknown, RTCP: Unknown) <-> 118.69.239.140:50012 <-> 118.69.239.140:50014 
<-> 192.168.1.6:48000 (RTP: Unknown, RTCP: Unknown)

debug: Got traffic information for stream: (audio) 192.168.1.4:45746 (RTP: 
210.245.35.150:45746, RTCP: Unknown) <-> 118.69.239.140:50012 <-> 
118.69.239.140:50014 <-> 192.168.1.6:48000 (RTP: Unknown, RTCP: Unknown)

debug: Got traffic information for stream: (audio) 192.168.1.4:45746 (RTP: 
210.245.35.150:45746, RTCP: Unknown) <-> 118.69.239.140:50012 <-> 
118.69.239.140:50014 <-> 192.168.1.6:48000 (RTP: 210.245.35.150:48000, RTCP: 
Unknown)

debug: removing session 7810dcdd0ceb7...@192.168.1.4: 8...@118.69.239.140 
(93846fc44ae3fe24) --> 9...@118.69.239.140

(Port 50012 Closed)
(Port 50013 Closed)
(Port 50014 Closed)
(Port 50015 Closed)



___
Users mailing list
Users@lists.opensips.org

http://lists.opensips.org/cgi-bin/mailman/listinfo/users




-- 
--
*--*--*--*--*--*

Duane
*--*--*--*--*--*
--


-Inline Attachment Follows-

___
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

[OpenSIPS-Users] need help on mediaproxy ports

[ha do]

Hi all

i debug on mediaproxy and see, mediaproxy create 4 ports but the real ports to 
relay media are only 2 ports: 118.69.239.140:50012 <-> 118.69.239.140:50014

is it normal, can i config mediaproxy create only 2 ports

Thank you
Ha`

mediaproxy.mediacontrol.StreamListenerProtocol starting on 50012
mediaproxy.mediacontrol.StreamListenerProtocol starting on 50013
mediaproxy.mediacontrol.StreamListenerProtocol starting on 50014
mediaproxy.mediacontrol.StreamListenerProtocol starting on 50015
debug: Added new stream: (audio) 192.168.1.4:45746 (RTP: Unknown, RTCP: 
Unknown) <-> 118.69.239.140:50012 <-> 118.69.239.140:50014 <-> Unknown (RTP: 
Unknown, RTCP: Unknown)
debug: created new session 7810dcdd0ceb7...@192.168.1.4: 8...@118.69.239.140 
(93846fc44ae3fe24) --> 9...@118.69.239.140
debug: updating existing session 7810dcdd0ceb7...@192.168.1.4: 
8...@118.69.239.140 (93846fc44ae3fe24) --> 9...@118.69.239.140
debug: Received updated SDP answer
debug: Got initial answer from callee for stream: (audio) 192.168.1.4:45746 
(RTP: Unknown, RTCP: Unknown) <-> 118.69.239.140:50012 <-> 118.69.239.140:50014 
<-> 192.168.1.6:48000 (RTP: Unknown, RTCP: Unknown)
debug: Got traffic information for stream: (audio) 192.168.1.4:45746 (RTP: 
210.245.35.150:45746, RTCP: Unknown) <-> 118.69.239.140:50012 <-> 
118.69.239.140:50014 <-> 192.168.1.6:48000 (RTP: Unknown, RTCP: Unknown)
debug: Got traffic information for stream: (audio) 192.168.1.4:45746 (RTP: 
210.245.35.150:45746, RTCP: Unknown) <-> 118.69.239.140:50012 <-> 
118.69.239.140:50014 <-> 192.168.1.6:48000 (RTP: 210.245.35.150:48000, RTCP: 
Unknown)
debug: removing session 7810dcdd0ceb7...@192.168.1.4: 8...@118.69.239.140 
(93846fc44ae3fe24) --> 9...@118.69.239.140
(Port 50012 Closed)
(Port 50013 Closed)
(Port 50014 Closed)
(Port 50015 Closed)




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

Re: [OpenSIPS-Users] need help on mediaproxy that cannot start

[ha do]

HI Richard and Dan
 
i am able to run media proxy
mediaproxy version 2.3.10
just follow the link and change 2 things:
use python-application 1.1.5 + python-gnutls 1.1.8
 
Thank you all 
Ha`


--- On Mon, 1/11/10, Richard Revels  wrote:


From: Richard Revels 
Subject: Re: [OpenSIPS-Users] need help on mediaproxy that cannot start
To: "OpenSIPS users mailling list" 
Date: Monday, January 11, 2010, 7:57 AM


My bad.  Got an error running the Dispatcher and downgraded sqlobject.  Later, 
when I realized the issue was that I didn't have a postgres module installed I 
never went back and updated my notes or tried running with the newer sqlobject.


On Jan 9, 2010, at 10:41 AM, Dan Pascu wrote:

> 
> On 8 Jan 2010, at 22:17, Richard Revels wrote:
> 
>> I've been meaning to come back to this.  I'll try to get something  
>> coherent together this weekend for CentOS users.  The process  
>> outlined at the link below is somewhat overkill as most of the  
>> python stuff can be done via easy_install with just a couple tweaks  
>> to the readme.  The gnutls must be compiled and installed from  
>> source but a standard ./configure; make; make install is all that is  
>> needed there as well.
>> 
>> I had a problem with asking for specific versions of both sqlobject  
>> and python-application because easy_install is not able to find them  
>> wherever it looks by default.  This is a drag since it worked fine  
>> for sqlobject just a couple of weeks ago.  Here are the command  
>> lines I am using to get around that:
>> 
>> #get recent version of setup tools
>> easy_install -U setuptools
>> 
>> #let easy_install download and install from svn for specific version  
>> (notice space between url and package identifier)
>> easy_install http://svn.colorstudy.com/SQLObject/tags/0.9.0/  
>> sqlobject==0.9.0
> 
> Why do you need sqlobject==0.9.0? I'm running with 0.11.0 just fine  
> and I expect it to run with 0.12.0 as well as there were no changes in  
> it to affect the code.
> 
> --
> Dan
> 
> 
> 
> 
> ___
> 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



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

[OpenSIPS-Users] need help on mediaproxy that cannot start

[ha do]

Hi all


i follow the instruction : 
http://www.smartvox.co.uk/serfaq_install_mediaproxy2.htm

my centos version:
[r...@centos-cucku application]# uname -a
Linux CentOS-Cucku 2.6.18-128.el5PAE #1 SMP Wed Jan 21 11:19:46 EST 2009 i686 
i686 i386 GNU/Linux

i get the error when start the mediaproxy on Centos :
[r...@centos-cucku application]# /usr/bin/media-dispatcher
Traceback (most recent call last):
  File "/usr/bin/media-dispatcher", line 12, in ?
    from application.process import process, ProcessError
  File "/usr/lib/python2.4/site-packages/application/process.py", line 12, in ?
    from application import log
  File "/usr/lib/python2.4/site-packages/application/log/__init__.py", line 12, 
in ?
    from application.log.extensions import twisted
  File 
"/usr/lib/python2.4/site-packages/application/log/extensions/twisted/__init__.py",
 line 4
    from __future__ import absolute_import
SyntaxError: future feature absolute_import is not defined


please help

thank you
Ha`



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

Re: [OpenSIPS-Users] Need help on flag in usrloc

[ha do]

Hi Bogdan

it is clear now :)

Thank you
Ha`

--- On Wed, 1/6/10, Bogdan-Andrei Iancu  wrote:

From: Bogdan-Andrei Iancu 
Subject: Re: [OpenSIPS-Users] Need help on flag in usrloc
To: "OpenSIPS users mailling list" 
Date: Wednesday, January 6, 2010, 3:51 AM

Hi Ha,

the two flags are different and may have different values - one is used 
as NAT marker, the other one is used as SIP-based pinging marker.

so, you can use different flags and both of them will be saved in cflag 
mask.

Regards,
Bogdan

ha do wrote:
> Hi Bogdan
>
> got it :)
>
> 1 more question about the flag
> modparam("nathelper", "sipping_bflag", 7)
> modparam("usrloc", "nat_bflag", 7)
>
> the modem ADSL will close the port after 3 mins(some minutes), so 
> Opensips should send OPTION message(sip ping) to modem to keep port 
> that should open for UA
>
> the sipping_blag of nathelper module should be the same value as 
> nat_bflag of usrloc ? or the cflag of usrloc just  has a value??
>
>
> Thank you
> Ha`
>
> --- On *Tue, 1/5/10, Bogdan-Andrei Iancu //* 
> wrote:
>
>
>     From: Bogdan-Andrei Iancu 
>     Subject: Re: [OpenSIPS-Users] Need help on flag in usrloc
>     To: "OpenSIPS users mailling list" 
>     Date: Tuesday, January 5, 2010, 7:20 AM
>
>     Hi Ha,
>
>     the NAT branch flag you use is 7 (nat_bflag) and in usrloc you
>     find the
>     branch flags in the cflags (contact flags) field. The cflags is a
>     mask
>     with all the branch flags: 192 = 128 (2^7) + 64 (2^6)
>
>     Regards,
>     Bogdan
>
>
>     ha do wrote:
>     > Hi all
>     >
>     > i am successfull to check the UA behind NAT but i dont know what
>     value
>     > of the flag will be stored in the usrloc
>     >
>     > Could someone please let me know the value of Nated UA flag,
>     that is
>     > stored in usrloc
>     >
>     >
>     > my config :
>     > modparam("nathelper", "natping_interval",180)
>     > modparam("nathelper", "ping_nated_only", 1)
>     > modparam("nathelper", "sipping_bflag", 7)
>     > modparam("nathelper", "sipping_from", "sip:cu...@kamailio.org
>     ")
>     > modparam("registrar|nathelper", "received_avp", "$avp(i:80)")
>     > modparam("usrloc", "nat_bflag", 7)
>     >
>     >
>     > route{
>     >
>     > 
>     > route("4");
>     > if (method=="REGISTER")
>     >         {
>     >                 if (isflagset(5)) {
>     >                         setbflag(6);
>     >                         setbflag(7);
>     >                 }
>     >                 if (!save("location"))
>     >                         sl_reply_error();
>     >                 exit;
>     >         }
>     > }
>     > route[4]{
>     >         force_rport();
>     >         if (nat_uac_test("19")) {
>     >                 if (method=="REGISTER") {
>     >                         fix_nated_register();
>     >                 } else {
>     >                         fix_nated_contact();
>     >                 }
>     >                 setflag(5);
>     >         }
>     >         return;
>     > }
>     >
>     > mysql> select * from location\G
>     > *** 1. row ***
>     >            id: 12
>     >      username: 1000
>     >        domain: NULL
>     >       contact: sip:1...@192.168.1.2:12280;rinstance=a4752f45bdc3ddd3
>     >      received: sip:210.245.35.150:12280
>     >          path: NULL
>     >       expires: 2010-01-05 17:48:56
>     >             q: -1.00
>     >        callid: YjE1NWJmYWYyYWMzZDg2ZDc5MjY0NDQyMGE5NDEwM2E.
>     >          cseq: 2
>     > last_modified: 2010-01-05 16:48:56
>     >          flags: 0
>     >         cflags: 192
>     > user_agent: eyeBeam release 1004p stamp 31962
>     >       socket: udp:118.69.193.198:5060
>     >       methods: 5951
>     >
>     >
>     > Thank you
>     > Ha`
>     >
>     >
>     >
>     >
>     
>     >
>     > ___
>     > Users mailing list
>     > Users@lists.opensips.org 
>     > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>     >   
>
>
>     -- 
>     Bogdan-Andrei Iancu
>     www.voice-system.ro
>
>
>     ___
>     Users mailing list
>     us...@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
>   


-- 
Bogdan-Andrei Iancu
www.voice-system.ro


___
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

Re: [OpenSIPS-Users] need advice on B2b

[ha do]

HI Anca

it is clear now :)
i am trying to learn the opensips,

Thank you
Ha`

--- On Wed, 1/6/10, Anca Vamanu  wrote:

From: Anca Vamanu 
Subject: Re: [OpenSIPS-Users] need advice on B2b
To: "OpenSIPS users mailling list" 
Date: Wednesday, January 6, 2010, 3:14 AM




  
  
Hi Ha`,



The excerpt from your script shows that you don't have a good
understanding of the opensips scripting logic. First, the route
block will only be called for SIP Requests. 

Calling this is not right:

 if(status=="200")

    route("b2b_reply");



The replies will go into reply route blocks. You can have a default
reply route ( one without an index or with index 0), or you can specify
a certain reply route for a request by calling  t_on_reply.



Second, you don't understand what happens with b2b request and replies.
It is explained in the documentation:



The requests and replies that are received by the B2BUA server,
belonging to the dialogs it is handling will not go into the script as
normal request do. The reason for this is that this are not normal
requests where the server is a proxy, but the server is an endpoint in
the dialog and therefore they should not go through the same routes.
However, it is normal for this request to be seen from the script and
allow the script writer to do the processing it desires based on them.
For this, it is possible to define two special B2B routes - one for
requests and one for replies. The routes are of type route
and have their name defined in the modules parameters script_req_route
and script_reply_route.




In other words, there are two important things:

1. the B2B requests/replies will not go into the default request/reply
route block. 

2. the b2b_request/b2_reply route will be called automatically for
every request/reply targeted to the b2b agent



So, for your script, you don't need this lines:



 if(is_method("INVITE")) {

    route("b2b_request");

    t_on_reply("2");

    }

    else

    if(status=="200")

    route("b2b_reply");



Hope this made things a bit clearer.



Regards,

-- 
Anca Vamanu
www.voice-system.ro


ha do wrote:

  

  
HI Anca



i am trying to use the b2b_request + b2b_reply



route{

...

if(is_method("INVITE") &&  !(src_ip == 192.168.1.249 &&
src_port ==5060)) 

    {

    if (! t_newtran()){

    sl_reply_error();

    exit;

    };



    b2b_init_request("top hiding");

    exit;

    };



route("1");

}

route[1] {

    if(is_method("INVITE")) {

    route("b2b_request");

    t_on_reply("2");

    }

    else

    if(status=="200")

    route("b2b_reply");

    if (!t_relay()) {

    sl_reply_error();

    };

    exit;

}

route[b2b_request] {

  xlog("b2b_request cucku ($ci)\n");

    force_rtp_proxy();

}

route[b2b_reply] {

  xlog("b2b_reply cucku ($ci)\n");

    force_rtp_proxy();

}



i get the errors :  

ERROR:nathelper:force_rtp_proxy: Unable to parse body 

and

 DBG:tm:utimer_routine: timer routine:4,tl=0xb615e9a8 next=(nil),
timeout=2900

 DBG:tm:timer_routine: timer routine:3,tl=0xb615c6f4 next=(nil),
timeout=29

 DBG:tm:delete_handler: removing 0xb615c690

 DBG:tm:delete_cell: delete_cell 0xb615c690: can't delete -- still
reffed (1)





===

 DBG:core:get_hdr_field: cseq : <2> 

 DBG:core:parse_headers: flags=8

 DBG:tm:t_reply_matching: hash 21530 label 76806763 branch 0

 DBG:tm:t_reply_matching: REF_UNSAFE: after is 2

 DBG:tm:t_reply_matching: reply matched (T=0xb615e85c)!

 DBG:tm:t_check: end=0xb615e85c

 DBG:tm:reply_received: org. status uas=0, uac[0]=100 local=2
is_invite=1)

 DBG:tm:t_should_relay_response: T_code=0, new_code=180

 DBG:tm:local_reply: branch=0, save=0, winner=0

 DBG:tm:local_reply: Passing provisional reply 180 to FIFO application

 DBG:tm:run_trans_callbacks: trans=0xb615e85c, callback type 1024, id 0
entered

 DBG:b2b_entities:b2b_parse_key: hash_index = [111]  - local_index= [0]

 DBG:core:parse_headers: flags=

 DBG:core:get_hdr_field: content_length=0

 DBG:core:get_hdr_field: found end of header

 DBG:b2b_entities:b2b_tm_cback: Received a reply with statuscode = 180

 DBG:core:parse_headers: flags=

 DBG:b2b_entities:b2b_new_dlg: 'To' header ALREADY PARSED:


 DBG:b2b_entities:b2b_new_dlg: Not an initial request

 DBG:core:parse_to_param: tag=bfad35cdb22f09f741816636d344f54b-19f0

 DBG:core:parse_to: end of header reached, state=29

 DBG:core:parse_to: display={

Re: [OpenSIPS-Users] Need help on flag in usrloc

[ha do]

Hi Bogdan

got it :)

1 more question about the flag 
modparam("nathelper", "sipping_bflag", 7)
 modparam("usrloc", "nat_bflag", 7)

the modem ADSL will close the port after 3 mins(some minutes), so Opensips 
should send OPTION message(sip ping) to modem to keep port that should open for 
UA 

the sipping_blag of nathelper module should be the same value as nat_bflag of 
usrloc ? or the cflag of usrloc just  has a value??


Thank you
Ha`

--- On Tue, 1/5/10, Bogdan-Andrei Iancu  wrote:

From: Bogdan-Andrei Iancu 
Subject: Re: [OpenSIPS-Users] Need help on flag in usrloc
To: "OpenSIPS users mailling list" 
Date: Tuesday, January 5, 2010, 7:20 AM

Hi Ha,

the NAT branch flag you use is 7 (nat_bflag) and in usrloc you find the 
branch flags in the cflags (contact flags) field. The cflags is a mask 
with all the branch flags: 192 = 128 (2^7) + 64 (2^6)

Regards,
Bogdan


ha do wrote:
> Hi all
>
> i am successfull to check the UA behind NAT but i dont know what value 
> of the flag will be stored in the usrloc
>
> Could someone please let me know the value of Nated UA flag, that is 
> stored in usrloc
>
>
> my config :
> modparam("nathelper", "natping_interval",180)
> modparam("nathelper", "ping_nated_only", 1)
> modparam("nathelper", "sipping_bflag", 7)
> modparam("nathelper", "sipping_from", "sip:cu...@kamailio.org")
> modparam("registrar|nathelper", "received_avp", "$avp(i:80)")
> modparam("usrloc", "nat_bflag", 7)
>
>
> route{
>
> 
> route("4");
> if (method=="REGISTER")
>         {
>                 if (isflagset(5)) {
>                         setbflag(6);
>                         setbflag(7);
>                 }
>                 if (!save("location"))
>                         sl_reply_error();
>                 exit;
>         }
> }
> route[4]{
>         force_rport();
>         if (nat_uac_test("19")) {
>                 if (method=="REGISTER") {
>                         fix_nated_register();
>                 } else {
>                         fix_nated_contact();
>                 }
>                 setflag(5);
>         }
>         return;
> }
>
> mysql> select * from location\G
> *** 1. row ***
>            id: 12
>      username: 1000
>        domain: NULL
>       contact: sip:1...@192.168.1.2:12280;rinstance=a4752f45bdc3ddd3
>      received: sip:210.245.35.150:12280
>          path: NULL
>       expires: 2010-01-05 17:48:56
>             q: -1.00
>        callid: YjE1NWJmYWYyYWMzZDg2ZDc5MjY0NDQyMGE5NDEwM2E.
>          cseq: 2
> last_modified: 2010-01-05 16:48:56
>          flags: 0
>         cflags: 192
> user_agent: eyeBeam release 1004p stamp 31962
>       socket: udp:118.69.193.198:5060
>       methods: 5951
>
>
> Thank you
> Ha`
>
>
>
> 
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>   


-- 
Bogdan-Andrei Iancu
www.voice-system.ro


___
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

Re: [OpenSIPS-Users] need advice on B2b

[ha do]

HI Anca

i am trying to use the b2b_request + b2b_reply

route{
...
if(is_method("INVITE") &&  !(src_ip == 192.168.1.249 && src_port ==5060)) 
    {
    if (! t_newtran()){
    sl_reply_error();
    exit;
    };

    b2b_init_request("top hiding");
    exit;
    };

route("1");
}
route[1] {
    if(is_method("INVITE")) {
    route("b2b_request");
    t_on_reply("2");
    }
    else
    if(status=="200")
    route("b2b_reply");
    if (!t_relay()) {
    sl_reply_error();
    };
    exit;
}
route[b2b_request] {
  xlog("b2b_request cucku ($ci)\n");
    force_rtp_proxy();
}
route[b2b_reply] {
  xlog("b2b_reply cucku ($ci)\n");
    force_rtp_proxy();
}

i get the errors :  
ERROR:nathelper:force_rtp_proxy: Unable to parse body 
and
 DBG:tm:utimer_routine: timer routine:4,tl=0xb615e9a8 next=(nil), 
timeout=2900
 DBG:tm:timer_routine: timer routine:3,tl=0xb615c6f4 next=(nil), timeout=29
 DBG:tm:delete_handler: removing 0xb615c690
 DBG:tm:delete_cell: delete_cell 0xb615c690: can't delete -- still reffed (1)


===
 DBG:core:get_hdr_field: cseq : <2> 
 DBG:core:parse_headers: flags=8
 DBG:tm:t_reply_matching: hash 21530 label 76806763 branch 0
 DBG:tm:t_reply_matching: REF_UNSAFE: after is 2
 DBG:tm:t_reply_matching: reply matched (T=0xb615e85c)!
 DBG:tm:t_check: end=0xb615e85c
 DBG:tm:reply_received: org. status uas=0, uac[0]=100 local=2 is_invite=1)
 DBG:tm:t_should_relay_response: T_code=0, new_code=180
 DBG:tm:local_reply: branch=0, save=0, winner=0
 DBG:tm:local_reply: Passing provisional reply 180 to FIFO application
 DBG:tm:run_trans_callbacks: trans=0xb615e85c, callback type 1024, id 0 entered
 DBG:b2b_entities:b2b_parse_key: hash_index = [111]  - local_index= [0]
 DBG:core:parse_headers: flags=
 DBG:core:get_hdr_field: content_length=0
 DBG:core:get_hdr_field: found end of header
 DBG:b2b_entities:b2b_tm_cback: Received a reply with statuscode = 180
 DBG:core:parse_headers: flags=
 DBG:b2b_entities:b2b_new_dlg: 'To' header ALREADY PARSED: 

 DBG:b2b_entities:b2b_new_dlg: Not an initial request
 DBG:core:parse_to_param: tag=bfad35cdb22f09f741816636d344f54b-19f0
 DBG:core:parse_to: end of header reached, state=29
 DBG:core:parse_to: display={}, ruri={sip:0873000...@192.168.1.249;user=phone}
 DBG:core:print_rr_body: current rr is 
 DBG:core:print_rr_body: out rr []
 DBG:core:print_rr_body: we have 1 records
 DBG:b2b_entities:b2b_tm_cback: Created new dialog structure 0xb61618c0
 DBG:core:print_rr_body: current rr is 
 DBG:core:print_rr_body: out rr []
 DBG:core:print_rr_body: we have 1 records
 DBG:b2b_logic:b2bl_parse_key: hash_index = [623]  - local_index= [0]
 DBG:core:parse_headers: flags=
 DBG:b2b_entities:b2b_parse_key: hash_index = [346]  - local_index= [0]
 DBG:core:parse_headers: flags=
 DBG:core:check_ip_address: params 192.168.1.4, 192.168.1.4, 0
 DBG:tm:t_reply_with_body: buffer computed
 DBG:tm:_reply_light: reply sent out. buf=0x81c70b8: SIP/2.0 1..., 
shmem=0xb615e534: SIP/2.0 1
 DBG:tm:_reply_light: finished
 b2b_reply cucku (B2B.111.0.1262765386)
 DBG:core:parse_headers: flags=
 DBG:core:parse_headers: flags=1000
 DBG:core:parse_content_type_hdr: missing Content-Type header
 ERROR:nathelper:force_rtp_proxy: Unable to parse body

Thank you
Ha`
--- On Mon, 1/4/10, Anca Vamanu  wrote:

From: Anca Vamanu 
Subject: Re: [OpenSIPS-Users] need advice on B2b
To: "OpenSIPS users mailling list" 
Date: Monday, January 4, 2010, 3:04 AM

Hi Ha`,

There is a very simple example in the documentation:

route[b2b_request] {
  xlog("b2b_request ($ci)\n");
}


route[b2b_reply] {
  xlog("b2b_reply ($ci)\n");
}


You can call in these routes any function that you call in a request route.

Regards,

-- 
Anca Vamanu
www.voice-system.ro



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

[OpenSIPS-Users] Need help on flag in usrloc

[ha do]

Hi all

i am successfull to check the UA behind NAT but i dont know what value of the 
flag will be stored in the usrloc 

Could someone please let me know the value of Nated UA flag, that is stored in 
usrloc 


my config :
modparam("nathelper", "natping_interval",180)
modparam("nathelper", "ping_nated_only", 1)
modparam("nathelper", "sipping_bflag", 7)
modparam("nathelper", "sipping_from", "sip:cu...@kamailio.org")
modparam("registrar|nathelper", "received_avp", "$avp(i:80)")
modparam("usrloc", "nat_bflag", 7)


route{


route("4");
if (method=="REGISTER")
    {
    if (isflagset(5)) {
    setbflag(6);
    setbflag(7);
    }
    if (!save("location"))
    sl_reply_error();
    exit;
    }
}
route[4]{
    force_rport();
    if (nat_uac_test("19")) {
    if (method=="REGISTER") {
    fix_nated_register();
    } else {
    fix_nated_contact();
    }
    setflag(5);
    }
    return;
}

mysql> select * from location\G
*** 1. row ***
   id: 12
 username: 1000
   domain: NULL
  contact: sip:1...@192.168.1.2:12280;rinstance=a4752f45bdc3ddd3
 received: sip:210.245.35.150:12280
 path: NULL
  expires: 2010-01-05 17:48:56
    q: -1.00
   callid: YjE1NWJmYWYyYWMzZDg2ZDc5MjY0NDQyMGE5NDEwM2E.
 cseq: 2
last_modified: 2010-01-05 16:48:56
     flags: 0
    cflags: 192
user_agent: eyeBeam release 1004p stamp 31962
  socket: udp:118.69.193.198:5060
  methods: 5951


Thank you
Ha`





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

Re: [OpenSIPS-Users] something on stun documment

[ha do]

Hi Bogdan

it is strange because the doc say:
Set
alternate_ip parameter
, but it is the port ;(

so you can check the 1.3.3.alternate_ip (str), 1.3.4.alternate_port (str)
they have the same value:
modparam("stun","alternate_port","3479")
 
Thank you
Ha`

--- On Mon, 1/4/10, Bogdan-Andrei Iancu  wrote:

From: Bogdan-Andrei Iancu 
Subject: Re: [OpenSIPS-Users] something on stun documment
To: "OpenSIPS users mailling list" 
Date: Monday, January 4, 2010, 6:59 AM

Hi Ha,

the alternate port is whatever port you want to use as secondary port by 
the STUN server. 3479 is the default alternate port in STUN, so it is ok.

Regards,
Bogdan

ha do wrote:
> Hi admin
>
> http://www.opensips.org/html/docs/modules/devel/stun.html#id227269
>
>
>       1.3.3.  |alternate_ip| (str)
>
> Another ip from another interface.
>
> *Example 1.3. Set |alternate_ip| parameter *
>
> ...
>
> modparam("stun","alternate_port","3479") --> this is right?
>
>
> Thank you
> Ha`
>
>
>         
>
>
> 
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>   


-- 
Bogdan-Andrei Iancu
www.voice-system.ro


___
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

[OpenSIPS-Users] something on stun documment

[ha do]

Hi admin

http://www.opensips.org/html/docs/modules/devel/stun.html#id227269
1.3.3. 
alternate_ip (str)

Another ip from another interface.
Example 1.3. Set
alternate_ip parameter
...

modparam("stun","alternate_port","3479") --> this is right?


Thank you
Ha`






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

[OpenSIPS-Users] need advice on B2b

[ha do]

Hi all

modparam("b2b_entities", "script_req_route", "b2b_request")
modparam("b2b_entities", "script_reply_route", "b2b_reply")

Could you please let me know how it work
is there the example of b2b_request and b2b_reply

the initial request can be seen as b2b_request?
the onreply_route is as same as b2b_reply?

Thank you
Ha`





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

Re: [OpenSIPS-Users] Need help Nathelper + rtpproxy

[ha do]

Hi Bogdan

Thank you for the tip :)


--- On Tue, 12/29/09, Bogdan-Andrei Iancu  wrote:

From: Bogdan-Andrei Iancu 
Subject: Re: [OpenSIPS-Users] Need help Nathelper + rtpproxy
To: "OpenSIPS users mailling list" 
Date: Tuesday, December 29, 2009, 7:42 AM

Hi Ha,

You need to call unforce_rtp_proxy() when BYE is received.

Regards,
Bogdan


> # -  request routing logic ---
>
> # main routing logic
> route{
>         # initial sanity checks -- messages with
>         # max_forwards==0, or excessively long requests
>         if (!mf_process_maxfwd_header("10")) {
>                 sl_send_reply("483","Too Many Hops");
>                 exit;
>         };
>
>         if (msg:len >=  2048 ) {
>                 sl_send_reply("513", "Message too big");
>                 exit;
>         };
>
>         # we record-route all messages -- to make sure that
>         # subsequent messages will go through our proxy; that's
>         # particularly good if upstream and downstream entities
>         # use different transport protocol
>         if (!method=="REGISTER")
>             record_route();
>         # subsequent messages withing a dialog should take the
>         # path determined by record-routing
>         if (loose_route()) {
>             # mark routing logic in request
>             append_hf("P-hint: rr-enforced\r\n");
>             route(1);
>         };
>
>         if (!uri==myself) {
>             # mark routing logic in request
>             append_hf("P-hint: outbound\r\n");
>             route(1);
>         };
>
>         # if the request is for other domain use UsrLoc
>         # (in case, it does not work, use the following command
>         # with proper names and addresses in it)
>         if (uri==myself) {
>                 if (method=="REGISTER") {
>                     save("location");
>                     exit;
>                 };
>         }
>         # native SIP destinations are handled using our USRLOC DB
>         if(method=="INVITE"){
>             if (dst_ip == 192.168.1.248)
>                 force_rtp_proxy("oei");
>             if (dst_ip == 172.26.0.2)
>                 force_rtp_proxy("oie");
>             t_on_reply("1");
>         };
>        if (is_method("BYE"))
>                     unforce_rtp_proxy();
>       
>         if (!lookup("location","m")) {
>             switch ($retcode) {
>                 case -1:
>                 case -3:
>                     t_newtran();
>                     t_on_failure("1");
>                     t_reply("404", "Not Found");
>                     exit;
>                 case -2:
>                     sl_send_reply("405", "Method Not Allowed");
>                     exit;
>                 }
>             }
>         route(1);
> }
> route[1] {
>         # send it out now; use stateful forwarding as it works
>         # reliably even for UDP2TCP
>         failure_route[1];
>         if (!t_relay()) {
>                 sl_reply_error();
>         };
>         exit;
> }
> onreply_route[1]{
>     if (status=="200"){
>         if(dst_ip == 172.26.0.2)
>                 force_rtp_proxy("oie");
>         if(dst_ip == 192.168.1.248)
>                 force_rtp_proxy("oei");
>     }
> }
>
> failure_route[1]{
>         unforce_rtp_proxy();
> }
>
>
>
> when i make call and check on rtpproxy debug and see the rtpproxy debug :
>
> DBUG:handle_command: received command "18781_4 
> UIEc0,18,4,97,9,2,15,8,101 09d614a45c92f...@172.26.0.100 172.26.0.100 
> 2908 824bcd8bb5ba14fa;1"
> INFO:handle_command: new session 09d614a45c92f...@172.26.0.100, tag 
> 824bcd8bb5ba14fa;1 requested, type strong
> INFO:handle_command: new session on a port 48190 created, tag 
> 824bcd8bb5ba14fa;1
> INFO:handle_command: pre-filling caller's address with 172.26.0.100:2908
> DBUG:doreply: sending reply "18781_4 48190 192.168.1.248
> "
> DBUG:handle_command: received command "18780_4 LEIc0,101 
> 09d614a45c92f...@172.26.0.100 192.168.1.6 17206 824bcd8bb5ba14fa;1 
> 49ee0e488eccead5;1"
> INFO:handle_command: lookup on ports 48190/42508, session timer restarted
> INFO:handle_command: pre-filling callee's address with 192.168.1.6:17206
> DBUG:doreply: sending reply "18780_4 42508 172.26.0.2
> "
> INFO:process_rtp: session timeout
> INFO:remove_session: RTP stats: 238 in from callee, 323 in from 
> caller, 561 relayed, 0 dropped
> INFO:remove_session: RTCP stats: 1 in from callee, 0 in from caller, 1 
> relayed, 0 dropped
> INFO:remove_session: session on ports 48190/42508 is cleaned up
>
>
>
> 
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>   


-- 
Bogdan-Andrei Iancu
www.voice-system.ro


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



  

[OpenSIPS-Users] Need help Nathelper + rtpproxy

[ha do]

Hi all

i set up rtpproxy run in same machine with opensips

my network topology:
ip phone1 (192.168.1.6) 
(192.168.1.248)opensips(172.26.0.2)---(172.26.0.100)ip phone 2

media :
ip phone1 (192.168.1.6) 
(192.168.1.248)rtpproxy(172.26.0.2)---(172.26.0.100)ip phone 2

i start rtpproxy : 
rtpproxy -l 172.26.0.2/192.168.1.248 -f -F -s udp:127.0.0.1:2 -d 
DBUG:LOG_LOCAL7

the IP Phone 2 call IP Phone 1 and i did successfull on signaling + media 
when i disconnect the call i didnt see the command tear down the media session 
on rtpproxy
 
it is normal or i mis-config  the opensips.cfg, please help


Thank you
Ha

here is my opensips.cfg:

# --- global configuration parameters 

debug=9    # debug level (cmd line: -dd)

fork=yes

log_facility=LOG_LOCAL7

log_stderror=no    # (cmd line: -E)

children=4

port=5060



# -- module loading --

#set module path

mpath="/usr/local/lib/opensips/modules/"

loadmodule "db_mysql.so"

loadmodule "signaling.so"

loadmodule "sl.so"

loadmodule "tm.so"

loadmodule "rr.so"

loadmodule "maxfwd.so"

loadmodule "usrloc.so"

loadmodule "registrar.so"

loadmodule "textops.so"

loadmodule "mi_fifo.so"

loadmodule "uri.so"

loadmodule "xlog.so"

loadmodule "nathelper.so"

#loadmodule "snmpstats.so"



# - setting module-specific parameters ---

# -- mi_fifo params --

modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")

# -- usrloc params --

#modparam("usrloc", "db_mode", 0)

# Uncomment this if you want to use SQL database

# for persistent storage and comment the previous line

modparam("usrloc", "db_url", "mysql://opensips:opensip...@localhost/opensips")

modparam("usrloc", "db_mode", 2)



# -- rr params --

# add value to ;lr param to make some broken UAs happy

modparam("rr", "enable_full_lr", 1)

modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:2")

modparam("nathelper", "nortpproxy_str", "")

# -  request routing logic ---



# main routing logic

route{

    # initial sanity checks -- messages with

    # max_forwards==0, or excessively long requests

    if (!mf_process_maxfwd_header("10")) {

    sl_send_reply("483","Too Many Hops");

    exit;

    };



    if (msg:len >=  2048 ) {

    sl_send_reply("513", "Message too big");

    exit;

    };



    # we record-route all messages -- to make sure that

    # subsequent messages will go through our proxy; that's

    # particularly good if upstream and downstream entities

    # use different transport protocol

    if (!method=="REGISTER")

    record_route();

    # subsequent messages withing a dialog should take the

    # path determined by record-routing

    if (loose_route()) {

    # mark routing logic in request

    append_hf("P-hint: rr-enforced\r\n");

    route(1);

    };



    if (!uri==myself) {

    # mark routing logic in request

    append_hf("P-hint: outbound\r\n");

    route(1);

    };



    # if the request is for other domain use UsrLoc

    # (in case, it does not work, use the following command

    # with proper names and addresses in it)

    if (uri==myself) {

    if (method=="REGISTER") {

                    save("location");

    exit;

    };

        }

    # native SIP destinations are handled using our USRLOC DB

    if(method=="INVITE"){

            if (dst_ip == 192.168.1.248)

                force_rtp_proxy("oei");

    if (dst_ip == 172.26.0.2)

                force_rtp_proxy("oie");

    t_on_reply("1");

        };

       if (is_method("BYE"))

                    unforce_rtp_proxy();

   

        if (!lookup("location","m")) {

            switch ($retcode) {

                case -1:

    case -3:

                    t_newtran();

    t_on_failure("1");

    t_reply("404", "Not Found");

                    exit;

    case -2:

                    sl_send_reply("405", "Method Not Allowed");

    exit;

    }

            }

    route(1);

}

route[1] {

    # send it out now; use stateful forwarding as it works

    # reliably even for UDP2TCP

        failure_route[1];

    if (!t_relay()) {

    sl_reply_error();

    };

    exit;

}

onreply_route[1]{

    if (status=="200"){

    if(dst_ip == 172.26.0.2)

    force_rtp_proxy("oie");

    if(dst_ip == 192.168.1.248)

    force_rtp_proxy("oei");

    }

}



failure_route[1]{

    unforce_rtp_proxy();

}



when i make call and check on rtpproxy debug and see the rtpproxy debug :

DBUG:handle_command: receiv

Re: [OpenSIPS-Users] Need advice rtpproxy + nathelper on opensips

[ha do]

Hi Bogdan

please ignore :
and i get message :
Dec 23 16:35:37 localhost /usr/local/sbin/opensips[5459]: DBG:tm:t_newtran:
 transaction on entrance=0x
Dec 23 16:35:37 localhost rtpproxy[5126]: DBUG:handle_command: received command 
"5459_4 D 5772c9303b3d7...@192.168.1.9 2adb6fd4a0d3fcb2 517a7cba18e3441a"
Dec 23 16:35:37 localhost /usr/local/sbin/opensips[5459]: 
DBG:core:parse_headers: flags=
Dec 23 16:35:37 localhost rtpproxy[5126]: INFO:handle_command: delete request 
failed: session 5772c9303b3d7...@192.168.1.9, tags 
2adb6fd4a0d3fcb2/517a7cba18e3441a not found
Dec 23 16:35:37 localhost /usr/local/sbin/opensips[5459]: 
DBG:core:parse_headers: flags=78
Dec 23 16:35:37 localhost rtpproxy[5126]: DBUG:doreply: sending reply "5459_4 
E8 " 
because i change :
if (loose_route()) {
    if (is_method("BYE")) {
    unforce_rtp_proxy();

and i still need help on media + the flag

Thank you
Ha

--- On Tue, 12/22/09, ha do  wrote:

From: ha do 
Subject: Re: [OpenSIPS-Users] Need advice rtpproxy + nathelper on opensips
To: "OpenSIPS users mailling list" 
Date: Tuesday, December 22, 2009, 8:38 PM

Hi Bogdan
 
i just make a test to see what happen with opensips + rtpproxy
ip phone 1(192.168.1.9), ip phone 2(192.168.1.5),  opensips + 
rtpproxy(192.168.1.248)
 
i use the default opensips.cfg and edit some lines:
loadmodule "nathelper.so"
 modparam("nathelper", "rtpproxy_sock", "udp:localhost:2")route {.
if(has_totag()){
  if (is_method("BYE")){..
  }else if (is_method("INVITE")){
  force_rtp_proxy();
  record_route();
   }
}
.
}

when i make call call from IP phone 1 to IP phone 2, and media go directly from 
ip phone 1 to ip phone 2
Media is not go through the rtpproxy

what should i do to force media go through the rtpproxy(just test)

1 more question on the flag:
from the alg.cfg
force_rtp_proxy("FAII"), force_rtp_proxy("FAIE"), 
force_rtp_proxy("FAEI"), force_rtp_proxy("FAEE"), 
i read on the nathelper module : the flag is Lower case - does it still or i 
have to change to the flag to lower case 

i run
 rtpproxy :
rtpproxy -l 192.168.1.248 -s udp:192.168.1.248:2 -d DBUG:LOG_LOCAL7 -F


[r...@localhost run]# ll
total 108
drwxr-xr-x 2 root  root  4096 May 25  2008 console
drwxr-xr-x 2 root  root  4096 Dec 22 17:35 dbus
-rw-r--r-- 1 root  root 5 Dec 22 17:35 haldaemon.pid
-rw--- 1 root  root 5 Dec 22 17:35 klogd.pid
-rw-r--r-- 1 root  root 5 Dec 22 17:35 messagebus.pid
drwxr-xr-x 2 mysql mysql 4096 Dec 22 20:55 mysqld
drwxrwxr-x 2 root  root  4096 Jun 15  2008 netreport
drwxr-xr-x 2 root  root  4096 May 28  2008 pm
-rw-r--r-- 1 root  root 5 Dec 23 17:30 rtpproxy.pid
drwxr-xr-x 2 root  root  4096 Dec 18 16:35 setrans
-rw-r--r-- 1 root  root 5 Dec 22 17:35 sshd.pid
-rw--- 1 root  root 5 Dec 22 17:35 syslogd.pid
-rw-rw-r-- 1 root  utmp  4992 Dec 23 15:56 utmp

and i get message :
Dec 23 16:35:37 localhost /usr/local/sbin/opensips[5459]: DBG:tm:t_newtran:
 transaction on entrance=0x
Dec 23 16:35:37 localhost rtpproxy[5126]: DBUG:handle_command: received command 
"5459_4 D 5772c9303b3d7...@192.168.1.9 2adb6fd4a0d3fcb2 517a7cba18e3441a"
Dec 23 16:35:37 localhost /usr/local/sbin/opensips[5459]: 
DBG:core:parse_headers: flags=
Dec 23 16:35:37 localhost rtpproxy[5126]: INFO:handle_command: delete request 
failed: session 5772c9303b3d7...@192.168.1.9, tags 
2adb6fd4a0d3fcb2/517a7cba18e3441a not found
Dec 23 16:35:37 localhost /usr/local/sbin/opensips[5459]: 
DBG:core:parse_headers: flags=78
Dec 23 16:35:37 localhost rtpproxy[5126]: DBUG:doreply: sending reply "5459_4 
E8 " 

Thank you
Ha

--- On Tue, 12/22/09, Bogdan-Andrei Iancu  wrote:

From: Bogdan-Andrei Iancu 
Subject: Re:
 [OpenSIPS-Users] Need advice rtpproxy + nathelper on opensips
To: "OpenSIPS users mailling list" 
Date: Tuesday, December 22, 2009, 3:50 AM

Hi Ha,

well, depends - do you want your opensips to talk to the outside world? 
if no, you do not need a WAN address.

Regards,
Bogdan

ha do wrote:
> Hi all
>
> i install rtpproxy in same Opensips machine
> Do i really need at least 1 Wan IP address for Opensips + rtpproxy work
>
> Thank you
> Ha`
>
>
> 
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>   


-- 
Bogdan-Andrei Iancu
www.voice-system.ro


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

Re: [OpenSIPS-Users] Need advice rtpproxy + nathelper on opensips

[ha do]

Hi Bogdan
 
i just make a test to see what happen with opensips + rtpproxy
ip phone 1(192.168.1.9), ip phone 2(192.168.1.5),  opensips + 
rtpproxy(192.168.1.248)
 
i use the default opensips.cfg and edit some lines:
loadmodule "nathelper.so"
 modparam("nathelper", "rtpproxy_sock", "udp:localhost:2")route {.
if(has_totag()){
  if (is_method("BYE")){..
  }else if (is_method("INVITE")){
  force_rtp_proxy();
  record_route();
   }
}
.
}

when i make call call from IP phone 1 to IP phone 2, and media go directly from 
ip phone 1 to ip phone 2
Media is not go through the rtpproxy

what should i do to force media go through the rtpproxy(just test)

1 more question on the flag:
from the alg.cfg
force_rtp_proxy("FAII"), force_rtp_proxy("FAIE"), 
force_rtp_proxy("FAEI"), force_rtp_proxy("FAEE"), 
i read on the nathelper module : the flag is Lower case - does it still or i 
have to change to the flag to lower case 

i run rtpproxy :
rtpproxy -l 192.168.1.248 -s udp:192.168.1.248:2 -d DBUG:LOG_LOCAL7 -F


[r...@localhost run]# ll
total 108
drwxr-xr-x 2 root  root  4096 May 25  2008 console
drwxr-xr-x 2 root  root  4096 Dec 22 17:35 dbus
-rw-r--r-- 1 root  root 5 Dec 22 17:35 haldaemon.pid
-rw--- 1 root  root 5 Dec 22 17:35 klogd.pid
-rw-r--r-- 1 root  root 5 Dec 22 17:35 messagebus.pid
drwxr-xr-x 2 mysql mysql 4096 Dec 22 20:55 mysqld
drwxrwxr-x 2 root  root  4096 Jun 15  2008 netreport
drwxr-xr-x 2 root  root  4096 May 28  2008 pm
-rw-r--r-- 1 root  root 5 Dec 23 17:30 rtpproxy.pid
drwxr-xr-x 2 root  root  4096 Dec 18 16:35 setrans
-rw-r--r-- 1 root  root 5 Dec 22 17:35 sshd.pid
-rw--- 1 root  root 5 Dec 22 17:35 syslogd.pid
-rw-rw-r-- 1 root  utmp  4992 Dec 23 15:56 utmp

and i get message :
Dec 23 16:35:37 localhost /usr/local/sbin/opensips[5459]: DBG:tm:t_newtran: 
transaction on entrance=0x
Dec 23 16:35:37 localhost rtpproxy[5126]: DBUG:handle_command: received command 
"5459_4 D 5772c9303b3d7...@192.168.1.9 2adb6fd4a0d3fcb2 517a7cba18e3441a"
Dec 23 16:35:37 localhost /usr/local/sbin/opensips[5459]: 
DBG:core:parse_headers: flags=
Dec 23 16:35:37 localhost rtpproxy[5126]: INFO:handle_command: delete request 
failed: session 5772c9303b3d7...@192.168.1.9, tags 
2adb6fd4a0d3fcb2/517a7cba18e3441a not found
Dec 23 16:35:37 localhost /usr/local/sbin/opensips[5459]: 
DBG:core:parse_headers: flags=78
Dec 23 16:35:37 localhost rtpproxy[5126]: DBUG:doreply: sending reply "5459_4 
E8 " 

Thank you
Ha

--- On Tue, 12/22/09, Bogdan-Andrei Iancu  wrote:

From: Bogdan-Andrei Iancu 
Subject: Re: [OpenSIPS-Users] Need advice rtpproxy + nathelper on opensips
To: "OpenSIPS users mailling list" 
Date: Tuesday, December 22, 2009, 3:50 AM

Hi Ha,

well, depends - do you want your opensips to talk to the outside world? 
if no, you do not need a WAN address.

Regards,
Bogdan

ha do wrote:
> Hi all
>
> i install rtpproxy in same Opensips machine
> Do i really need at least 1 Wan IP address for Opensips + rtpproxy work
>
> Thank you
> Ha`
>
>
> 
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>   


-- 
Bogdan-Andrei Iancu
www.voice-system.ro


___
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

Re: [OpenSIPS-Users] Need advice rtpproxy + nathelper on opensips

[ha do]

Hi Bogdan
 
i just make a test to see what happen with opensips + rtpproxy
ip phone 1(192.168.1.5), ip phone 2(192.168.1.9) opensips + 
rtpproxy(192.168.1.248)
 
i use the default opensips.cfg and edit some lines:
loadmodule "nathelper.so"
 modparam("nathelper", "rtpproxy_sock", "udp:localhost:2") route {. 


--- On Tue, 12/22/09, Bogdan-Andrei Iancu  wrote:


From: Bogdan-Andrei Iancu 
Subject: Re: [OpenSIPS-Users] Need advice rtpproxy + nathelper on opensips
To: "OpenSIPS users mailling list" 
Date: Tuesday, December 22, 2009, 3:50 AM


Hi Ha,

well, depends - do you want your opensips to talk to the outside world? 
if no, you do not need a WAN address.

Regards,
Bogdan

ha do wrote:
> Hi all
>
> i install rtpproxy in same Opensips machine
> Do i really need at least 1 Wan IP address for Opensips + rtpproxy work
>
> Thank you
> Ha`
>
>
> 
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>   


-- 
Bogdan-Andrei Iancu
www.voice-system.ro


___
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

[OpenSIPS-Users] Need advice rtpproxy + nathelper on opensips

[ha do]

Hi all

i install rtpproxy in same Opensips machine
Do i really need at least 1 Wan IP address for Opensips + rtpproxy work

Thank you
Ha`




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

Re: [OpenSIPS-Users] question on B2B module

[ha do]

Hi Anca


let me describe the topology:
VOIP Service Provider SIP trunk port 5070Opensips1 --sip trunk port 
5060Opensips2Ip phone

the Opnesips1 has 2 interface :
  interface 1 : IP address 192.168.1.2 -- trunking with opensips2
  interface 2 : IP address 115.22.22.3 -- trunking with VOIP service 
provider
  VOIP provider IP address : 115.1.1.2 
IP address of Opensips2 : 192.168.1.3

when i do top hiding 
 
if(is_method("INVITE") && src_ip=="115.1.1.2")
   $du=192.168.1.3//is it right ?
   b2b_init_request("top hiding");
what is value i set :
modparam("b2b_entities", "server_address", "which IP address Private or 
Public")Thank you
Ha`

--- On Wed, 12/16/09, Anca Vamanu  wrote:

From: Anca Vamanu 
Subject: Re: [OpenSIPS-Users] question on B2B module
To: "OpenSIPS users mailling list" 
Date: Wednesday, December 16, 2009, 2:30 AM

Hi Ha,

Yes, such a configuration is possible. You have to set the $du to the 
address of OpenSIPS2 before doing b2b_init on OpenSIPS1.

Regards,

-- 
Anca Vamanu
www.voice-system.ro



ha do wrote:
> Hi all
>
> network topology:
> VOIP Service Provider SIP trunk port 5070Opensips1 --sip 
> trunk port 5060Opensips2Ip phone
>
> Media :
> VOIP Service Provider Opensips1(rtpproxy)Ip phone
>
> can i use the B2B module on opensips1 to do topology hiding and use 
> the rtpproxy on opensips1 to force media from IP phone to VOIP SP
>
> please advice.
>
> Thank you
> Ha
>
>
> 
>
> ___
> 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



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

Re: [OpenSIPS-Users] Help with sip trace errors please

[ha do]

Hi Bogdan

the scripts/dbtext/opensips/sip_trace works without creating new sip_trace when 
opensips stop and start again :)

the option modparam("db_text", "db_mode", 1) doesnt help on writing :(

the db_flatstore does store real time message but the INVITE, RINGING, 200ok, 
ACK are stored in difference files(sip_trace_9.log, 
sip_trace_10.log,sip_trace_11.log,sip_trace_12.log)

"For each table there will be several
files, one file for every OpenSIPS process that wrote some data 
into
that table"

Thank you very much
Ha`

--- On Thu, 12/17/09, Bogdan-Andrei Iancu  wrote:

From: Bogdan-Andrei Iancu 
Subject: Re: [OpenSIPS-Users] Help with sip trace errors please
To: "OpenSIPS users mailling list" 
Date: Thursday, December 17, 2009, 2:44 AM

Hi Ha,

ha do wrote:
> Hi Bogdan
>
> you need to take the new sip_trace file that comes with 6439 - the table
> format was changed, not the code in opensips.
> >>>i cannt what to do, i did new complie the source and new install 
> the opensips, i do
> |svn co https://opensips.svn.sourceforge.net/svnroot/opensips/branches/1.6 
> opensips_1_6 on new machine
> then do make , then do install
> |
>
[bogdan]

It is not about recompiling sources - you need to use the new sip_trace 
file with the definition of the table (from 
scripts/dbtext/opensips/sip_trace)


> |
> my question :|do you have a plan to make enhancement of sip_trace that logs 
> all message without stop/start opensips how to admin opensips without log :((
> example : our opensips works fine for everything, but after some days someone 
> calls me and ask he cannot make call at 14:00PM yesterday and there is no log 
> to check 
> so ask him to make a call for debugging but the call is fine at debugging time
>
> if the siptrace module works without stop
>  and start opensips, i will open the log to check what happen with him at 
>this time :)
>         
>
[bogdan]
probably you mean that, when using dbtext, you do not see the content of 
the table untill a restart of opensips, right ? if so may try using 
db_mode = 1 for db_text module 
(http://www.opensips.org/html/docs/modules/devel/db_text.html#id228234), 
but I'm not sure it this non-caching mode will affect the "write" ops 
also (and not only the "read" ops).

Another solution is to use the db_flatstore driver (instead of db_text) 
-> this one does realtime writing on file.


Regards,
Bogdan

> Thank you
> Ha`
>         
> --- On *Tue, 12/15/09, Bogdan-Andrei Iancu //* 
> wrote:
>
>
>     From: Bogdan-Andrei Iancu 
>     Subject: Re: [OpenSIPS-Users] Help with sip trace errors please
>     To: "OpenSIPS users mailling list" 
>     Date: Tuesday, December 15, 2009, 12:47 AM
>
>     Hi Ha,
>
>     ha do wrote:
>     > Hi Bogdan
>     >
>     > i use the Checked out revision 6439.
>     > it is still issue when the sip_trace has content
>     >
>     you need to take the new sip_trace file that comes with 6439 - the
>     table
>     format was changed, not the code in opensips.
>     >
>     > do you have a plan to make enhancement of sip_trace that logs all
>     > message without stop/start opensips
>     > how to admin opensips without log :((
>     >
>     I do not understand your question...could you rephrase ?
>
>     Regards,
>     Bogdan
>
>
> 
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>   


-- 
Bogdan-Andrei Iancu
www.voice-system.ro


___
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

Re: [OpenSIPS-Users] Help with sip trace errors please

[ha do]

Hi Bogdan

you need to take the new sip_trace file that comes with 6439 - the table 
format was changed, not the code in opensips.
>>>i cannt what to do, i did new complie the source and new install the 
>>>opensips, i do 
svn co https://opensips.svn.sourceforge.net/svnroot/opensips/branches/1.6 
opensips_1_6 on new machine
then do make , then do install

my question :do you have a plan to make enhancement of sip_trace that logs all 
message without stop/start opensips how to admin opensips without log :((
example : our opensips works fine for everything, but after some days someone 
calls me and ask he cannot make call at 14:00PM yesterday and there is no log 
to check 
so ask him to make a call for debugging but the call is fine at debugging time

if the siptrace module works without stop and start opensips, i will open the 
log to check what happen with him at this time :)
Thank you
Ha`
--- On Tue, 12/15/09, Bogdan-Andrei Iancu  wrote:

From: Bogdan-Andrei Iancu 
Subject: Re: [OpenSIPS-Users] Help with sip trace errors please
To: "OpenSIPS users mailling list" 
Date: Tuesday, December 15, 2009, 12:47 AM

Hi Ha,

ha do wrote:
> Hi Bogdan
>
> i use the Checked out revision 6439.
> it is still issue when the sip_trace has content
>
you need to take the new sip_trace file that comes with 6439 - the table 
format was changed, not the code in opensips.
>
> do you have a plan to make enhancement of sip_trace that logs all 
> message without stop/start opensips
> how to admin opensips without log :((
>
I do not understand your question...could you rephrase ?

Regards,
Bogdan




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

Re: [OpenSIPS-Users] Help with sip trace errors please

[ha do]

Hi Bogdan

i use the Checked out revision 6439.
it is still issue when the sip_trace has content

do you have a plan to make enhancement of sip_trace that logs all message 
without stop/start opensips 
how to admin opensips without log :((

Thank you
Ha`

--- On Mon, 12/14/09, Bogdan-Andrei Iancu  wrote:

From: Bogdan-Andrei Iancu 
Subject: Re: [OpenSIPS-Users] Help with sip trace errors please
To: "OpenSIPS users mailling list" 
Date: Monday, December 14, 2009, 5:38 AM

Hi Ha,

found the bug - there was an issue when loading the sip_trace file with 
content (some columns being NULL).

I fixed the problem on SVN, so if you do an update and re-compile, you 
should not need to delete the file during restarts.

Thanks and regards,
Bogdan

ha do wrote:
> HI Bogdan
>
> it is clear now :)
>
> f the file has content (aside the first line), the siptrace module 
> does not start ?
> no, the sip_trace module starts fine, i only need to replace new 
> sip_trace file then the error is gone :)
>
> is there a module with the same function as sip_trace but no need to 
> shutdown opensips
>
>
> Thank you
> Ha`
>
> --- On *Fri, 12/11/09, Bogdan-Andrei Iancu //* 
> wrote:
>
>
>     From: Bogdan-Andrei Iancu 
>     Subject: Re: [OpenSIPS-Users] Help with sip trace errors please
>     To: "OpenSIPS users mailling list" 
>     Date: Friday, December 11, 2009, 11:18 AM
>
>     Hi Ha,
>
>     Note that the DB content is updated only at shutdown.
>
>     So, to synthesize - if the file has content (aside the first
>     line), the
>     siptrace module does not start ?
>
>     Regards,
>     Bogdan
>
>     ha do wrote:
>     > HI Brian
>     >
>     > [r...@localhost opensipsdb]# ls -ld /tmp/opensipsdb/
>     > drwxr-xr-x 2 root root 4096 Dec 11 14:56 /tmp/opensipsdb/
>     > [r...@localhost opensipsdb]# ls -ld /tmp/opensipsdb/sip_trace
>     > -rw-r--r-- 1 root root 168 Dec 11 14:56 /tmp/opensipsdb/sip_trace
>     > [r...@localhost opensipsdb]# ps axu | grep opensips
>     > root      2014  0.0  1.1  38072  2928 ?        S    14:56   0:00
>     > /usr/local/sbin/opensips
>     > root      2016  0.0  0.3  38072   824 ?        S    14:56   0:00
>     > /usr/local/sbin/opensips
>     > root      2017  0.0  0.3  38072   796 ?        S    14:56   0:00
>     > /usr/local/sbin/opensips
>     > root      2018  0.0  0.3  38072   796 ?        S    14:56   0:00
>     > /usr/local/sbin/opensips
>     > root      2019  0.0  0.3  38072   796 ?        S    14:56   0:00
>     > /usr/local/sbin/opensips
>     > root      2020  0.0  0.6  38072  1712 ?        S    14:56   0:00
>     > /usr/local/sbin/opensips
>     > root      2021  0.0  0.6  38072  1696 ?        S    14:56   0:00
>     > /usr/local/sbin/opensips
>     > root      2022  0.0  0.6  38072  1724 ?        S    14:56   0:00
>     > /usr/local/sbin/opensips
>     > root      2023  0.0  0.6  38072  1588 ?        S    14:56   0:00
>     > /usr/local/sbin/opensips
>     > root      2024  0.0  0.2  38072   632 ?        S    14:56   0:00
>     > /usr/local/sbin/opensips
>     > root      2025  0.0  0.3  38072   948 ?        S    14:56   0:00
>     > /usr/local/sbin/opensips
>     > root      2026  0.0  0.3  38076   824 ?        S    14:56   0:00
>     > /usr/local/sbin/opensips
>     > root      2027  0.0  0.3  38072   892 ?        R    14:56   0:00
>     > /usr/local/sbin/opensips
>     > root      2028  0.0  0.3  38072   892 ?        S    14:56   0:00
>     > /usr/local/sbin/opensips
>     > root      2029  0.0  0.3  38072   892 ?        S    14:56   0:00
>     > /usr/local/sbin/opensips
>     > root      2030  0.0  0.3  38072   892 ?        S    14:56   0:00
>     > /usr/local/sbin/opensips
>     > root      2031  0.0  0.3  38072   880 ?        S    14:56   0:00
>     > /usr/local/sbin/opensips
>     > root      2044  0.0  0.2   3904   672 pts/0    R+   15:00   0:00
>     grep
>     > opensips
>     >
>     > let me describe my steps:
>     > step1 : create file sip_trace then insert with 1 line :
>     > id(int,auto) time_stamp(int) callid(string) traced_user(string)
>     > msg(string) method(string) status(string) fromip(string)
>     toip(string)
>     > fromtag(string) direction(string)
>     >
>     > step 2 : start opensips with sip_trace enable :
>     /etc/init.d/opensips
>     > start
>     >
>     > step 3 : use xlite softphone(2000)  + eyebeam(8000) register and
>     > register successfull
>     >
>     > step 4 : make call from 2000 to 8000 then connected fine then hangup
>     >
>     >

[OpenSIPS-Users] question on B2B module

[ha do]

Hi all

network topology:
VOIP Service Provider SIP trunk port 5070Opensips1 --sip trunk port 
5060Opensips2Ip phone

Media :
VOIP Service Provider Opensips1(rtpproxy)Ip phone

can i use the B2B module on opensips1 to do topology hiding and use the 
rtpproxy on opensips1 to force media from IP phone to VOIP SP 

please advice.

Thank you
Ha



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

Re: [OpenSIPS-Users] Help with sip trace errors please

[ha do]

HI Bogdan
it is clear now :)
f the file has content (aside the first line), the siptrace module does not 
start ?no, the sip_trace module starts fine, i only need to replace new 
sip_trace file then the error is gone :)
is there a module with the same function as sip_trace but no need to shutdown 
opensips

Thank youHa`

--- On Fri, 12/11/09, Bogdan-Andrei Iancu  wrote:

From: Bogdan-Andrei Iancu 
Subject: Re: [OpenSIPS-Users] Help with sip trace errors please
To: "OpenSIPS users mailling list" 
Date: Friday, December 11, 2009, 11:18 AM

Hi Ha,

Note that the DB content is updated only at shutdown.

So, to synthesize - if the file has content (aside the first line), the 
siptrace module does not start ?

Regards,
Bogdan

ha do wrote:
> HI Brian
>
> [r...@localhost opensipsdb]# ls -ld /tmp/opensipsdb/
> drwxr-xr-x 2 root root 4096 Dec 11 14:56 /tmp/opensipsdb/
> [r...@localhost opensipsdb]# ls -ld /tmp/opensipsdb/sip_trace
> -rw-r--r-- 1 root root 168 Dec 11 14:56 /tmp/opensipsdb/sip_trace
> [r...@localhost opensipsdb]# ps axu | grep opensips
> root      2014  0.0  1.1  38072  2928 ?        S    14:56   0:00 
> /usr/local/sbin/opensips
> root      2016  0.0  0.3  38072   824 ?        S    14:56   0:00 
> /usr/local/sbin/opensips
> root      2017  0.0  0.3  38072   796 ?        S    14:56   0:00 
> /usr/local/sbin/opensips
> root      2018  0.0  0.3  38072   796 ?        S    14:56   0:00 
> /usr/local/sbin/opensips
> root      2019  0.0  0.3  38072   796 ?        S    14:56   0:00 
> /usr/local/sbin/opensips
> root      2020  0.0  0.6  38072  1712 ?        S    14:56   0:00 
> /usr/local/sbin/opensips
> root      2021  0.0  0.6  38072  1696 ?        S    14:56   0:00 
> /usr/local/sbin/opensips
> root      2022  0.0  0.6  38072  1724 ?        S    14:56   0:00 
> /usr/local/sbin/opensips
> root      2023  0.0  0.6  38072  1588 ?        S    14:56   0:00 
> /usr/local/sbin/opensips
> root      2024  0.0  0.2  38072   632 ?        S    14:56   0:00 
> /usr/local/sbin/opensips
> root      2025  0.0  0.3  38072   948 ?        S    14:56   0:00 
> /usr/local/sbin/opensips
> root      2026  0.0  0.3  38076   824 ?        S    14:56   0:00 
> /usr/local/sbin/opensips
> root      2027  0.0  0.3  38072   892 ?        R    14:56   0:00 
> /usr/local/sbin/opensips
> root      2028  0.0  0.3  38072   892 ?        S    14:56   0:00 
> /usr/local/sbin/opensips
> root      2029  0.0  0.3  38072   892 ?        S    14:56   0:00 
> /usr/local/sbin/opensips
> root      2030  0.0  0.3  38072   892 ?        S    14:56   0:00 
> /usr/local/sbin/opensips
> root      2031  0.0  0.3  38072   880 ?        S    14:56   0:00 
> /usr/local/sbin/opensips
> root      2044  0.0  0.2   3904   672 pts/0    R+   15:00   0:00 grep 
> opensips
>
> let me describe my steps:
> step1 : create file sip_trace then insert with 1 line :
> id(int,auto) time_stamp(int) callid(string) traced_user(string) 
> msg(string) method(string) status(string) fromip(string) toip(string) 
> fromtag(string) direction(string)
>
> step 2 : start opensips with sip_trace enable : /etc/init.d/opensips 
> start
>
> step 3 : use xlite softphone(2000)  + eyebeam(8000) register and 
> register successfull
>
> step 4 : make call from 2000 to 8000 then connected fine then hangup
>
> step5 : check the /tmp/opensipsdb/sip_trace : cat 
> /tmp/opensipsdb/sip_trace and result
> [r...@localhost opensipsdb]# cat /tmp/opensipsdb/sip_trace
> id(int,auto) time_stamp(int) callid(string) traced_user(string) 
> msg(string) method(string) status(string) fromip(string) toip(string) 
> fromtag(string) direction(string)
>
> step 6: wait about 3 mins
>
> step 7: check the sip_trace file :
> [r...@localhost opensipsdb]# cat /tmp/opensipsdb/sip_trace
> id(int,auto) time_stamp(int) callid(string) traced_user(string) 
> msg(string) method(string) status(string) fromip(string) toip(string) 
> fromtag(string) direction(string)
>
>
> step 8 : stop opensips : /etc/init.d/opensips stop
>
> step 9: check the sip_trace file : at this time all logs are now in 
> the /tmp/opensipsdb/sip_trace
> [r...@localhost opensipsdb]# cat /tmp/opensipsdb/sip_trace
> id(int,auto) time_stamp(int) callid(string) traced_user(string) 
> msg(string) method(string) status(string) fromip(string) toip(string) 
> fromtag(string) direction(string)
> 39:1260518390:MGRjYzNhYzQ0YTc2YTU0MWQyYzI4MTEyY2Q5NjcwYjI.::SIP/2.0 
> 200 OK\r\nVia\: SIP/2.0/UDP 
> 192.168.237.1\:5592;received=192.168.237.1;branch=z9hG4bK-d8754z-5d41606f451fca7c-1---d8754z-;rport=5592\r\nContact\:
>  
> \r\nTo\: 
> ;tag=5378ee52\r\nFrom\: 
> "2000";tag=aa323d0e\r\nCall-ID\: 
> MGRjYzNhYzQ0YTc2YTU0MWQyYzI4MTEyY2Q5NjcwYjI.\r\nCSeq\: 2 
> BYE\r\nUser-Agent\: eyeBeam release

Re: [OpenSIPS-Users] Help with sip trace errors please

[ha do]

--- On Fri, 12/11/09, Bogdan-Andrei Iancu  wrote:

From: Bogdan-Andrei Iancu 
Subject: Re: [OpenSIPS-Users] Help with sip trace errors please
To: "OpenSIPS users mailling list" 
Date: Friday, December 11, 2009, 11:18 AM

Hi Ha,

Note that the DB content is updated only at shutdown.

So, to synthesize - if the file has content (aside the first line), the 
siptrace module does not start ?

Regards,
Bogdan

ha do wrote:
> HI Brian
>
> [r...@localhost opensipsdb]# ls -ld /tmp/opensipsdb/
> drwxr-xr-x 2 root root 4096 Dec 11 14:56 /tmp/opensipsdb/
> [r...@localhost opensipsdb]# ls -ld /tmp/opensipsdb/sip_trace
> -rw-r--r-- 1 root root 168 Dec 11 14:56 /tmp/opensipsdb/sip_trace
> [r...@localhost opensipsdb]# ps axu | grep opensips
> root      2014  0.0  1.1  38072  2928 ?        S    14:56   0:00 
> /usr/local/sbin/opensips
> root      2016  0.0  0.3  38072   824 ?        S    14:56   0:00 
> /usr/local/sbin/opensips
> root      2017  0.0  0.3  38072   796 ?        S    14:56   0:00 
> /usr/local/sbin/opensips
> root      2018  0.0  0.3  38072   796 ?        S    14:56   0:00 
> /usr/local/sbin/opensips
> root      2019  0.0  0.3  38072   796 ?        S    14:56   0:00 
> /usr/local/sbin/opensips
> root      2020  0.0  0.6  38072  1712 ?        S    14:56   0:00 
> /usr/local/sbin/opensips
> root      2021  0.0  0.6  38072  1696 ?        S    14:56   0:00 
> /usr/local/sbin/opensips
> root      2022  0.0  0.6  38072  1724 ?        S    14:56   0:00 
> /usr/local/sbin/opensips
> root      2023  0.0  0.6  38072  1588 ?        S    14:56   0:00 
> /usr/local/sbin/opensips
> root      2024  0.0  0.2  38072   632 ?        S    14:56   0:00 
> /usr/local/sbin/opensips
> root      2025  0.0  0.3  38072   948 ?        S    14:56   0:00 
> /usr/local/sbin/opensips
> root      2026  0.0  0.3  38076   824 ?        S    14:56   0:00 
> /usr/local/sbin/opensips
> root      2027  0.0  0.3  38072   892 ?        R    14:56   0:00 
> /usr/local/sbin/opensips
> root      2028  0.0  0.3  38072   892 ?        S    14:56   0:00 
> /usr/local/sbin/opensips
> root      2029  0.0  0.3  38072   892 ?        S    14:56   0:00 
> /usr/local/sbin/opensips
> root      2030  0.0  0.3  38072   892 ?        S    14:56   0:00 
> /usr/local/sbin/opensips
> root      2031  0.0  0.3  38072   880 ?        S    14:56   0:00 
> /usr/local/sbin/opensips
> root      2044  0.0  0.2   3904   672 pts/0    R+   15:00   0:00 grep 
> opensips
>
> let me describe my steps:
> step1 : create file sip_trace then insert with 1 line :
> id(int,auto) time_stamp(int) callid(string) traced_user(string) 
> msg(string) method(string) status(string) fromip(string) toip(string) 
> fromtag(string) direction(string)
>
> step 2 : start opensips with sip_trace enable : /etc/init.d/opensips 
> start
>
> step 3 : use xlite softphone(2000)  + eyebeam(8000) register and 
> register successfull
>
> step 4 : make call from 2000 to 8000 then connected fine then hangup
>
> step5 : check the /tmp/opensipsdb/sip_trace : cat 
> /tmp/opensipsdb/sip_trace and result
> [r...@localhost opensipsdb]# cat /tmp/opensipsdb/sip_trace
> id(int,auto) time_stamp(int) callid(string) traced_user(string) 
> msg(string) method(string) status(string) fromip(string) toip(string) 
> fromtag(string) direction(string)
>
> step 6: wait about 3 mins
>
> step 7: check the sip_trace file :
> [r...@localhost opensipsdb]# cat /tmp/opensipsdb/sip_trace
> id(int,auto) time_stamp(int) callid(string) traced_user(string) 
> msg(string) method(string) status(string) fromip(string) toip(string) 
> fromtag(string) direction(string)
>
>
> step 8 : stop opensips : /etc/init.d/opensips stop
>
> step 9: check the sip_trace file : at this time all logs are now in 
> the /tmp/opensipsdb/sip_trace
> [r...@localhost opensipsdb]# cat /tmp/opensipsdb/sip_trace
> id(int,auto) time_stamp(int) callid(string) traced_user(string) 
> msg(string) method(string) status(string) fromip(string) toip(string) 
> fromtag(string) direction(string)
> 39:1260518390:MGRjYzNhYzQ0YTc2YTU0MWQyYzI4MTEyY2Q5NjcwYjI.::SIP/2.0 
> 200 OK\r\nVia\: SIP/2.0/UDP 
> 192.168.237.1\:5592;received=192.168.237.1;branch=z9hG4bK-d8754z-5d41606f451fca7c-1---d8754z-;rport=5592\r\nContact\:
>  
> \r\nTo\: 
> ;tag=5378ee52\r\nFrom\: 
> "2000";tag=aa323d0e\r\nCall-ID\: 
> MGRjYzNhYzQ0YTc2YTU0MWQyYzI4MTEyY2Q5NjcwYjI.\r\nCSeq\: 2 
> BYE\r\nUser-Agent\: eyeBeam release 1004p stamp 
> 31962\r\nContent-Length\: 
> 0\r\n\r\n:BYE:200:udp\:192.168.237.131\:5060:udp\:192.168.237.1\:5592:aa323d0e:out
>
> step 10 : start opensips again : /etc/initd/opensips start - start 
> successfull without problem
>
> step 11 : register 2000 + 8000 again
>
> step 12: get er

Re: [OpenSIPS-Users] Help with sip trace errors please

[ha do]

HI Brian

[r...@localhost opensipsdb]# ls -ld /tmp/opensipsdb/
drwxr-xr-x 2 root root 4096 Dec 11 14:56 /tmp/opensipsdb/
[r...@localhost opensipsdb]# ls -ld /tmp/opensipsdb/sip_trace
-rw-r--r-- 1 root root 168 Dec 11 14:56 /tmp/opensipsdb/sip_trace
[r...@localhost opensipsdb]# ps axu | grep opensips
root  2014  0.0  1.1  38072  2928 ?    S    14:56   0:00 
/usr/local/sbin/opensips
root  2016  0.0  0.3  38072   824 ?    S    14:56   0:00 
/usr/local/sbin/opensips
root  2017  0.0  0.3  38072   796 ?    S    14:56   0:00 
/usr/local/sbin/opensips
root  2018  0.0  0.3  38072   796 ?    S    14:56   0:00 
/usr/local/sbin/opensips
root  2019  0.0  0.3  38072   796 ?    S    14:56   0:00 
/usr/local/sbin/opensips
root  2020  0.0  0.6  38072  1712 ?    S    14:56   0:00 
/usr/local/sbin/opensips
root  2021  0.0  0.6  38072  1696 ?    S    14:56   0:00 
/usr/local/sbin/opensips
root  2022  0.0  0.6  38072  1724 ?    S    14:56   0:00 
/usr/local/sbin/opensips
root  2023  0.0  0.6  38072  1588 ?    S    14:56   0:00 
/usr/local/sbin/opensips
root  2024  0.0  0.2  38072   632 ?    S    14:56   0:00 
/usr/local/sbin/opensips
root  2025  0.0  0.3  38072   948 ?    S    14:56   0:00 
/usr/local/sbin/opensips
root  2026  0.0  0.3  38076   824 ?    S    14:56   0:00 
/usr/local/sbin/opensips
root  2027  0.0  0.3  38072   892 ?    R    14:56   0:00 
/usr/local/sbin/opensips
root  2028  0.0  0.3  38072   892 ?    S    14:56   0:00 
/usr/local/sbin/opensips
root  2029  0.0  0.3  38072   892 ?    S    14:56   0:00 
/usr/local/sbin/opensips
root  2030  0.0  0.3  38072   892 ?    S    14:56   0:00 
/usr/local/sbin/opensips
root  2031  0.0  0.3  38072   880 ?    S    14:56   0:00 
/usr/local/sbin/opensips
root  2044  0.0  0.2   3904   672 pts/0    R+   15:00   0:00 grep opensips

let me describe my steps:
step1 : create file sip_trace then insert with 1 line :
id(int,auto) time_stamp(int) callid(string) traced_user(string) msg(string) 
method(string) status(string) fromip(string) toip(string) fromtag(string) 
direction(string)

step 2 : start opensips with sip_trace enable : /etc/init.d/opensips start 

step 3 : use xlite softphone(2000)  + eyebeam(8000) register and register 
successfull

step 4 : make call from 2000 to 8000 then connected fine then hangup

step5 : check the /tmp/opensipsdb/sip_trace : cat /tmp/opensipsdb/sip_trace and 
result
[r...@localhost opensipsdb]# cat /tmp/opensipsdb/sip_trace
id(int,auto) time_stamp(int) callid(string) traced_user(string) msg(string) 
method(string) status(string) fromip(string) toip(string) fromtag(string) 
direction(string)

step 6: wait about 3 mins

step 7: check the sip_trace file : 
[r...@localhost opensipsdb]# cat /tmp/opensipsdb/sip_trace

id(int,auto) time_stamp(int) callid(string) traced_user(string)
msg(string) method(string) status(string) fromip(string) toip(string)
fromtag(string) direction(string)



step 8 : stop opensips : /etc/init.d/opensips stop

step 9: check the sip_trace file : at this time all logs are now in the 
/tmp/opensipsdb/sip_trace
[r...@localhost opensipsdb]# cat /tmp/opensipsdb/sip_trace
id(int,auto) time_stamp(int) callid(string) traced_user(string) msg(string) 
method(string) status(string) fromip(string) toip(string) fromtag(string) 
direction(string)
39:1260518390:MGRjYzNhYzQ0YTc2YTU0MWQyYzI4MTEyY2Q5NjcwYjI.::SIP/2.0 200 
OK\r\nVia\: SIP/2.0/UDP 
192.168.237.1\:5592;received=192.168.237.1;branch=z9hG4bK-d8754z-5d41606f451fca7c-1---d8754z-;rport=5592\r\nContact\:
 \r\nTo\: 
;tag=5378ee52\r\nFrom\: 
"2000";tag=aa323d0e\r\nCall-ID\: 
MGRjYzNhYzQ0YTc2YTU0MWQyYzI4MTEyY2Q5NjcwYjI.\r\nCSeq\: 2 BYE\r\nUser-Agent\: 
eyeBeam release 1004p stamp 31962\r\nContent-Length\: 
0\r\n\r\n:BYE:200:udp\:192.168.237.131\:5060:udp\:192.168.237.1\:5592:aa323d0e:out

step 10 : start opensips again : /etc/initd/opensips start - start successfull 
without problem

step 11 : register 2000 + 8000 again 

step 12: get error from debug :
Dec 11 15:03:49 localhost /usr/local/sbin/opensips[2120]: 
ERROR:db_text:dbt_insert: table does not exist!
Dec 11 15:03:49 localhost /usr/local/sbin/opensips[2120]: 
ERROR:siptrace:sip_trace: error storing trace


what happen in your lab, Brian
does the log of sip_trace write down to /tmp/opensipsdb/sip_trace file right 
away or you need to wait some times


Thank you
Ha
Are you saying that when no /tmp/opensipsdb/sip_trace exists, and
you start opensips fresh, then no error appears in the log and the
sip_trace file is created normally?

And are you saying that when a /tmp/opensipsdb/sip_trace exists,
and you start opensips fresh, then a error appears in the log and
the sip_trace file is not written to with new transactions? Please
show the errors in the log file if this is the case.

Also, please tell us the results of:

  [r...@localhost ~]# ls -ld /tmp/opensipsdb
  [r...@localhost ~]# ls -ld /tmp/opensip

Re: [OpenSIPS-Users] Help with sip trace errors please

[ha do]

Hi all

i found the problem here
when we using the db_text, i dont know how the opensips will write the log to 
the text file(sip_trace)
i just stop then start opensips again and then i see all logs in the text 
file(sip_trace)

the problem here when i start the second time -- opensips doesnt accept the old 
sip_trace text file, i need to create new sip_trace text file

my question how long opensips will write the log to text file ??
why opensips doesnt accept the old sip_trace text file? is there any 
enhancement to use the old text file

Thank you
Ha`

--- On Thu, 12/10/09, ha do  wrote:

From: ha do 
Subject: Re: [OpenSIPS-Users] Help with sip trace errors please
To: "OpenSIPS users mailling list" 
Date: Thursday, December 10, 2009, 3:29 AM

Hi Brian

i follow your suggestion and it is successfully writing the log to text file 
sip_trace

the opensips.cfg

loadmodule "db_text.so"
modparam("siptrace", "trace_flag", 22)
modparam("siptrace", "db_url", "text:///tmp/opensipsdb")
modparam("siptrace", "trace_on", 1)
modparam("siptrace", "enable_ack_trace", 1)
modparam("siptrace", "table", "sip_trace")

but when you enable debug=9 in the opensips.cfg

it say the error:
Dec 10 23:53:44 localhost /usr/local/sbin/opensips[1923]: 
INFO:db_text:dbt_init: using database at: /tmp/opensipsdb

Dec 10 23:54:01 localhost /usr/local/sbin/opensips[1915]: 


  


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

Re: [OpenSIPS-Users] Help with sip trace errors please

[ha do]

Hi Brian

i follow your suggestion and it is successfully writing the log to text file 
sip_trace

the opensips.cfg

loadmodule "db_text.so"
modparam("siptrace", "trace_flag", 22)
modparam("siptrace", "db_url", "text:///tmp/opensipsdb")
modparam("siptrace", "trace_on", 1)
modparam("siptrace", "enable_ack_trace", 1)
modparam("siptrace", "table", "sip_trace")

but when you enable debug=9 in the opensips.cfg

it say the error:
Dec 10 23:53:44 localhost /usr/local/sbin/opensips[1923]: 
INFO:db_text:dbt_init: using database at: /tmp/opensipsdb

Dec 10 23:54:01 localhost /usr/local/sbin/opensips[1915]: 
ERROR:db_text:dbt_insert: table does not exist!
Dec 10 23:54:01 localhost /usr/local/sbin/opensips[1915]: 
ERROR:siptrace:sip_trace: error storing trace
Dec 10 23:54:01 localhost /usr/local/sbin/opensips[1915]: 
ERROR:db_text:dbt_insert: table does not exist!
Dec 10 23:54:01 localhost /usr/local/sbin/opensips[1915]: 
ERROR:siptrace:trace_onreply_out: error storing trace
Dec 10 23:54:01 localhost /usr/local/sbin/opensips[1915]: 
ERROR:db_text:dbt_insert: table does not exist!
Dec 10 23:54:01 localhost /usr/local/sbin/opensips[1915]: 
ERROR:siptrace:sip_trace: error storing trace


[r...@localhost ~]# cat /tmp/opensipsdb/sip_trace
id(int,auto) time_stamp(int) callid(string) traced_user(string) msg(string) 
method(string) status(string) fromip(string) toip(string) fromtag(string) 
direction(string)
36:1260463989:YzJhOTE1OWFiNWVkMzM1ZTJlM2E5ZjI1NTc2NWQyM2Y.::SIP/2.0 200 
OK\r\nVia\: SIP/2.0/UDP 
192.168.237.1\:53352;received=192.168.237.1;branch=z9hG4bK-d87543-a60eba2316543c40-1--d87543-;rport=53352\r\nContact\:
 \r\nTo\: 
"2000";tag=6e5a5b08\r\nFrom\: 
;tag=415aae68\r\nCall-ID\: 
YzJhOTE1OWFiNWVkMzM1ZTJlM2E5ZjI1NTc2NWQyM2Y.\r\nCSeq\: 2 BYE\r\nUser-Agent\: 
X-Lite release 1103k stamp 53621\r\nContent-Length\: 
0\r\n\r\n:BYE:200:udp\:192.168.237.131\:5060:udp\:192.168.237.1\:53352:415aae68:out
35:1260463989:YzJhOTE1OWFiNWVkMzM1ZTJlM2E5ZjI1NTc2NWQyM2Y.::SIP/2.0 200 
OK\r\nVia\: SIP/2.0/UDP 192.168.237.131;branch=z9hG4bKde3e.722ffa76.0\r\nVia\: 
SIP/2.0/UDP 
192.168.237.1\:53352;received=192.168.237.1;branch=z9hG4bK-d87543-a60eba2316543c40-1--d87543-;rport=53352\r\nContact\:
 \r\nTo\: 
"2000";tag=6e5a5b08\r\nFrom\: 
;tag=415aae68\r\nCall-ID\: 
YzJhOTE1OWFiNWVkMzM1ZTJlM2E5ZjI1NTc2NWQyM2Y.\r\nCSeq\: 2 BYE\r\nUser-Agent\: 
X-Lite release 1103k stamp 53621\r\nContent-Length\: 
0\r\n\r\n:BYE:200:udp\:192.168.237.1\:62174:udp\:192.168.237.131\:5060:415aae68:in


--- On Wed, 12/9/09, opensipsl...@encambio.com  
wrote:

From: opensipsl...@encambio.com 
Subject: Re: [OpenSIPS-Users] Help with sip trace errors please
To: users@lists.opensips.org
Date: Wednesday, December 9, 2009, 11:16 AM


Hello Bogdan,

On wed., dec 09, 2009, Bogdan-Andrei Iancu wrote:
>Field 0 is the "msg" field (used as DB_BLOB by siptrace) should be
>"STR" to work, but I already see it is set at string in your table
>definition... so I looked for a bug and found one in the db_text
>module - if you update from SVN, it should be ok.
>
That is great news, but sadly I might have been able to find and
correct the problem myself if the variable naming was not so cryptic.

But thanks to you indeed, the problem is solved. I've tested your
revision 6382:

Index: modules/db_text/dbt_lib.c
diff -Nau modules/db_text/dbt_lib.c.orig modules/db_text/dbt_lib.c
--- modules/db_text/dbt_lib.c.orig  2009-10-16 02:35:15.0 +0200
+++ modules/db_text/dbt_lib.c   2009-12-09 18:14:52.707887067 +0100
@@ -446,7 +446,7 @@
            if(_t0==DB_STRING || _t0==DB_BLOB)
                return 0;
        case DB_BLOB:
-           if(_t0==DB_STR)
+           if(_t0==DB_STR || _t0==DB_STRING)
                return 0;
        case DB_BITMAP:
            if (_t0==DB_INT)

...and it works.

Regards,
Brian

___
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

Re: [OpenSIPS-Users] need help on registrar Module

[ha do]

HI Alex

Thank you for your answer
i think it is not fair for opensips because i check on the kamailio website
http://kamailio.org/docs/modules/1.5.x/registrar.html#id2530309

there is a flag to archive my goal : 0x04
0x04 - store and maintain one
contact per AoR. If there are other 
contact addresses for
AoR not matching current registration, 
remove them.
This mode ensures one contact per AoR 
(user).

why do i ask about the thing because if my WAN connection is down an up again  
then i will get new WAN IP address, 

maybe next release opensips will have :)

Thank you
Ha`

--- On Wed, 12/9/09, Alex Balashov  wrote:

From: Alex Balashov 
Subject: Re: [OpenSIPS-Users] need help on registrar Module
To: users@lists.opensips.org
Date: Wednesday, December 9, 2009, 11:20 PM

One way is to have the xlite "unregister" (AKA register with an 
expiration value of 0 (immediate)), if it has that capability.

On 12/09/2009 10:39 PM, ha do wrote:

> HI all
>
> i using the registrar module and set
>
> modparam("registrar", "max_contacts", 1)
>
> i using xlite and then register successfull the time out for registration = 
> 3600
>
> then i shutdown the xlite then reopen xlite and get error
> Dec 10 06:07:11 Opensips Cucku[1937]: INFO:registrar:test_max_contacts: too 
> many contacts for AOR<2000>, max=1
>
> is there any way to save the latest AOR and then delete the old AOR
>
>
>
>
> Thank you
> Ha`
>
>
>
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users


-- 
Alex Balashov - Principal
Evariste Systems
Web     : http://www.evaristesys.com/
Tel     : (+1) (678) 954-0670
Direct  : (+1) (678) 954-0671

___
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

[OpenSIPS-Users] need help on registrar Module

[ha do]

HI all

i using the registrar module and set 
modparam("registrar", "max_contacts", 1)

i using xlite and then register successfull the time out for registration = 3600

then i shutdown the xlite then reopen xlite and get error
Dec 10 06:07:11 Opensips Cucku[1937]: INFO:registrar:test_max_contacts: too 
many contacts for AOR <2000>, max=1

is there any way to save the latest AOR and then delete the old AOR




Thank you
Ha`




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

[OpenSIPS-Users] Need help on sip_trace module write the log to a file

[ha do]

Hi admin

i do config to use the siptrace module write the log to text file but i got the 
error:

from the debug=9, log_facility=LOG_LOCAL7:
Dec  7 15:14:37 Opensips Cucku[2168]: DBG:siptrace:trace_sl_onreply_out: dest 
[udp:192.168.237.1:7226]
Dec  7 15:14:37 Opensips Cucku[2168]: DBG:siptrace:trace_sl_onreply_out: 
storing info...
Dec  7 15:14:37 Opensips Cucku[2168]: DBG:db_text:dbt_db_get_table: cache or 
mtime succeeded for [sip_trace]
Dec  7 15:14:37 Opensips Cucku[2168]: ERROR:db_text:dbt_insert: incompatible 
types v[6] - c[1]!
Dec  7 15:14:37 Opensips Cucku[2168]: ERROR:siptrace:trace_sl_onreply_out: 
error storing trace
Dec  7 15:14:37 Opensips Cucku[2168]: DBG:core:destroy_avp_list: destroying 
list (nil)
Dec  7 15:14:37 Opensips Cucku[2168]: DBG:core:receive_msg: cleaning up

from
 the tail -f /var/log/message :
Dec  7 15:14:37 Opensips Cucku[2168]: ERROR:db_text:dbt_insert: incompatible 
types v[6] - c[1]!
Dec  7 15:14:37 Opensips Cucku[2168]: ERROR:siptrace:sip_trace: error storing 
trace
Dec  7 15:14:37 Opensips Cucku[2168]: ERROR:db_text:dbt_insert: incompatible 
types v[6] - c[1]!
Dec  7 15:14:37 Opensips Cucku[2168]: ERROR:siptrace:trace_sl_onreply_out: 
error storing trace

the config:
loadmodule "siptrace.so"
#modparam("siptrace", "db_url","mysql://opensips:opensip...@localhost/opensips")
modparam("siptrace", "db_url","text:///tmp/opensipsdb")
modparam("siptrace", "enable_ack_trace", 1)
modparam("siptrace", "trace_on", 0)
modparam("siptrace", "trace_flag",2)


but i am successfull when i write the log to mysql 

i create the file name : sip_trace in the directory : /tmp/opensipsdb
the structure of the sip_trace file :
id(int) time_stamp(str)
 callid(str) traced_user(str) msg(str) method(str) status(str) fromip(str) 
toip(str) fromtag(str) direction(str)


please help me to create a right format sip_trace file

Thank you
Ha`



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