Re: [OpenSIPS-Users] Opensips with MediaProxy

[Dan Pascu]

On Tuesday 07 April 2009, Stefano Favaro wrote:
> Hi,
>
>  I'm trying to use Opensips 1.5 together with mediaproxy 2.0 in the
> following scenario:
>
>  Opensips + Mediarelay and dispatcher on the same machine.
>  The server has 2 network interfaces: the first interface has a public
> ip, the second one has a private ip. My internal sip systems (switch,
> gateways, softphones etc.) connect to the private ip. External
> softphones are registered on the public ip.
>  Mediaproxy is binding on both ip addresses.

How exactly do you do that? Mediaproxy can only bind to a single IP.

>  RTP seems to work correctly, but i have some problems with the
> protocol: I've found these errors on opensips:
>  ERROR:core:udp_send: sendto(sock,0x81b0c30,474,0,0xbf898d78,16):
> Operation not permitted(1)
>
>  After the call is connected the 200 OK message is sent continously
> from the remote party and after 30 seconds the call terminates.

Looks like the ACK is not routed properly.

>
>  Do you think that this solution can be the right one or have you got
> better suggestions? Can you help me? Thanks.

I think things will get much simpler if you only use the public IP and 
allow forwarding from the private LAN to the public LAN for the devices 
on the private LAN to be able to use the public addresses of the proxy 
and the relay.

-- 
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Re: [OpenSIPS-Users] Modifying INVITE header to add phone-context

[Julian Yap]

I pretty much solved the issue.

This is what I used:
subst_uri('/^sip:([0-9]+)@(.*)$/sip:\...@\2;phone-context=sip.server.com/i


On Tue, Apr 7, 2009 at 4:25 PM, Julian Yap  wrote:
> I have a PSTN gateway which requires a Phone-Context value in the
> outgoing SIP INVITE message to further apply ISDN NPI/TON details.
>
> Here's an example of what I currently have going out to the PSTN gateway:
> INVITE sip:1...@sip.server.com:5060;user=phone SIP/2.0.
>
> This is what I require:
> INVITE sip:1...@sip.server.com:5060;phone-context=sip.server.com;user=phone
> SIP/2.0.
>
> Any clues on how to add the Phone-Context value?
>
> Thanks,
> Julian
>

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[OpenSIPS-Users] New MediaProxy release 2.3.3

[Adrian Georgescu]

Hello,

There is a new release of MediaProxy available, it contains various  
bug fixes. To upgrade your debian installation:


apt-get update
apt-get install mediaproxy-dispatcher mediaproxy-relay mediaproxy-web- 
sessions


Or download the tar file from:

http://download.ag-projects.com/MediaProxy/

The changelog since 2.3.2 is below:

mediaproxy (2.3.3) unstable; urgency=low

  * Re-raise the exception on failing to read RADIUS config file so  
we get a

full traceback
  * Have dispatcher close TLS connection cleanly when relay has  
duplicate IP

  * Added log_level to both Relay and Dispatcher configuration sections
  * Improved reconnection behaviour in relay to dispatcher
When the connection from the relay to the dispatcher is lost,  
first retry
in 1 second, then retry in 10 second on subsequent attempts if it  
loses

the connection again.
  * Fix bug where relay connects needlessly to previously removed  
dispatcher

  * Implemented a keepalive mechanism from relay to dispatcher
  * On relay reconnect don't have dispatcher query expired sessions
  * Removed superfluous datatype declaration
  * Only allow positive integers for time intervals and delays
  * Updated version dependency for python-application
  * In dispatcher, replace old connection from relay with new one  
instead of

giving an error
  * In the dispatcher, check if the reported expired session belongs  
to the

relay that reported it
  * Improved log messages when a relay reconnects to the dispatcher
  * In dispatcher, break the connection to a relay if a request times  
out
  * In dispatcher check if we know about the session that expired at  
relay

  * Use a more robust strategy to disconnect an unresponsive relay


Kind regards,
Adrian Georgescu


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[OpenSIPS-Users] OpenXcap

[opensips]

Hello,

IMSCore and opensips run on my local machine, now i install openxcap, it run 
also but the communication with the opensips and openxcap dont work.

you see a small tutorial on the button of this page:
http://openxcap.org/wiki/Installation

can my someone explain how to configure the opensips.cfg with the files from 
the page.
when i read the page right i think i must include the main but then i get 
errors when i start opensips (etc/init.d/opensips start)

my opensips.cfg:

### Global Parameters #

debug=3
log_stderror=yes
log_facility=LOG_LOCAL0

fork=yes
children=4

/* uncomment the following lines to enable debugging */
#debug=6
#fork=no
#log_stderror=yes

/* uncomment the next line to disable TCP (default on) */
#disable_tcp=yes

/* uncomment the next line to enable the auto temporary blacklisting of 
   not available destinations (default disabled) */
#disable_dns_blacklist=no

/* uncomment the next line to enable IPv6 lookup after IPv4 dns 
   lookup failures (default disabled) */
#dns_try_ipv6=yes

/* uncomment the next line to disable the auto discovery of local aliases
   based on revers DNS on IPs (default on) */
#auto_aliases=no

/* uncomment the following lines to enable TLS support  (default off) */
#disable_tls = no
#listen = tls:your_IP:5061
#tls_verify_server = 1
#tls_verify_client = 1
#tls_require_client_certificate = 0
#tls_method = TLSv1
#tls_certificate = "/etc/opensips/tls/user/user-cert.pem"
#tls_private_key = "/etc/opensips/tls/user/user-privkey.pem"
#tls_ca_list = "/etc/opensips/tls/user/user-calist.pem"


port=5065

/* uncomment and configure the following line if you want opensips to 
   bind on a specific interface/port/proto (default bind on all available) */
#listen=udp:192.168.2.5:5065
alias=presence.open-ims.test:5065
#alias=open-ims.test:5065
#alias=scscf.open-ims.test:5065
#alias=presence-server.open-ims.test:5065

### Modules Section 

#set module path
mpath="/usr/lib/opensips/modules/"

/* uncomment next line for MySQL DB support */
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_db.so"
loadmodule "uri.so"
loadmodule "xlog.so"
loadmodule "acc.so"

#loadmodule "mi_datagram.so"
#loadmodule "mysql.so"
#loadmodule "presence_mwi.so"
#loadmodule "presence_xcapdiff.so"
#loadmodule "pua.so"
#loadmodule "pua_mi.so"
#loadmodule "rls.so"



/* uncomment next lines for MySQL based authentication support 
   NOTE: a DB (like db_mysql) module must be also loaded */
loadmodule "auth.so"
loadmodule "auth_db.so"
/* uncomment next line for aliases support
   NOTE: a DB (like db_mysql) module must be also loaded */
loadmodule "alias_db.so"
/* uncomment next line for multi-domain support
   NOTE: a DB (like db_mysql) module must be also loaded
   NOTE: be sure and enable multi-domain support in all used modules
 (see "multi-module params" section ) */
loadmodule "domain.so"
/* uncomment the next two lines for presence server support
   NOTE: a DB (like db_mysql) module must be also loaded */
loadmodule "presence.so"
loadmodule "presence_xml.so"


# - setting module-specific parameters ---


# - mi_fifo params -
modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")


# - rr params -
# add value to ;lr param to cope with most of the UAs
modparam("rr", "enable_full_lr", 1)
# do not append from tag to the RR (no need for this script)
modparam("rr", "append_fromtag", 0)


# - registrar params -
modparam("registrar", "method_filtering", 1)
/* uncomment the next line to disable parallel forking via location */
# modparam("registrar", "append_branches", 0)
/* uncomment the next line not to allow more than 10 contacts per AOR */
#modparam("registrar", "max_contacts", 10)


# - usrloc params -
#modparam("usrloc", "db_mode",   0)
/* uncomment the following lines if you want to enable DB persistency
   for location entries */
modparam("usrloc", "db_mode",   2)
modparam("usrloc", "db_url",
"mysql://opensips:opensip...@localhost/opensips")


# - uri_db params -
/* by default we disable the DB support in the module as we do not need it
   in this configuration */
modparam("uri_db", "use_uri_table", 0)
modparam("uri_db", "db_url", "")


# - acc params -
/* what sepcial events should be accounted ? */
modparam("acc", "early_media", 1)
modparam("acc", "report_ack", 1)
modparam("acc", "report_cancels", 1)
/* by default ww do not adjust the direct of the sequential requests.
   if you enable this parameter, be sure the enable "append_fromtag"
   in "rr" module */
modparam("acc", "detect_direction", 0)
/* account triggers (flags) */
modparam("acc", "failed_transaction_flag", 3)
modparam("acc", "log_flag", 1)
modparam("acc", "log_missed_flag", 2)
/* uncomment the following lines to enable DB accoun

[OpenSIPS-Users] generate key

[Michael Ciupka]

Hello,

I will generate a certificate and a private key for my server (openxcap)
- tls/server.crt
- tls/server.key

i dont know how to generate this files.


regards
michael

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Re: [OpenSIPS-Users] generate key

[Uwe Kastens]

Hi Michael,

Try searching for openssl.

http://sial.org/howto/openssl/self-signed/

BR

Uwe
> Hello,
> 
> I will generate a certificate and a private key for my server (openxcap)
> - tls/server.crt
> - tls/server.key
> 
> i dont know how to generate this files.
> 
> 
> regards
> michael
> 
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> 


-- 

kiste lat: 54.322684, lon: 10.13586

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[OpenSIPS-Users] (no subject)

[opensips]

hi all,

i have some problems with the configuration from opensips for the openxcap.
the route from the file opensips.cfg must be configure
my problem is with the "main request routing logic"
I paste my main from the opensips.cfg
now i must configure it
with this help "http://openxcap.org/wiki/Installation";

when i copy and paste the main from the homepage and adjust the IP i get errors 
when i compile the file (/etc/init.d/ opensips start).

can somewhere help me to configure the main or can somewhere paste his/her main 
which works with opensips and opnexcap?

regards
michael


#
# $Id: opensips.cfg 5503 2009-03-22 16:22:32Z bogdan_iancu $
#
# OpenSIPS basic configuration script
# by Anca Vamanu 
#
# Please refer to the Core CookBook at:
#  http://www.opensips.org/index.php?n=Resources.DocsCookbooks
# for a explanation of possible statements, functions and parameters.
#


### Global Parameters #

debug=6
log_stderror=yes
log_facility=LOG_LOCAL0

fork=yes
children=4

/* uncomment the following lines to enable debugging */
#debug=6
#fork=no
#log_stderror=yes

/* uncomment the next line to disable TCP (default on) */
#disable_tcp=yes

/* uncomment the next line to enable the auto temporary blacklisting of
   not available destinations (default disabled) */
#disable_dns_blacklist=no

/* uncomment the next line to enable IPv6 lookup after IPv4 dns
   lookup failures (default disabled) */
#dns_try_ipv6=yes

/* uncomment the next line to disable the auto discovery of local aliases
   based on revers DNS on IPs (default on) */
#auto_aliases=no

/* uncomment the following lines to enable TLS support  (default off) */
#disable_tls = no
#listen = tls:your_IP:5061
#tls_verify_server = 1
#tls_verify_client = 1
#tls_require_client_certificate = 0
#tls_method = TLSv1
#tls_certificate = "/etc/opensips/tls/user/user-cert.pem"
#tls_private_key = "/etc/opensips/tls/user/user-privkey.pem"
#tls_ca_list = "/etc/opensips/tls/user/user-calist.pem"


port=5065

/* uncomment and configure the following line if you want opensips to
   bind on a specific interface/port/proto (default bind on all available) */
#listen=udp:192.168.2.5:5065
alias=presence.open-ims.test:5065
#alias=open-ims.test:5065
#alias=scscf.open-ims.test:5065
#alias=presence-server.open-ims.test:5065

### Modules Section 

#set module path
mpath="/usr/lib/opensips/modules/"

/* uncomment next line for MySQL DB support */
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_db.so"
loadmodule "uri.so"
loadmodule "xlog.so"
loadmodule "acc.so"

#loadmodule "mi_datagram.so"
#loadmodule "mysql.so"
#loadmodule "presence_mwi.so"
#loadmodule "presence_xcapdiff.so"
#loadmodule "pua.so"
#loadmodule "pua_mi.so"
#loadmodule "rls.so"



/* uncomment next lines for MySQL based authentication support
   NOTE: a DB (like db_mysql) module must be also loaded */
loadmodule "auth.so"
loadmodule "auth_db.so"
/* uncomment next line for aliases support
   NOTE: a DB (like db_mysql) module must be also loaded */
loadmodule "alias_db.so"
/* uncomment next line for multi-domain support
   NOTE: a DB (like db_mysql) module must be also loaded
   NOTE: be sure and enable multi-domain support in all used modules
 (see "multi-module params" section ) */
loadmodule "domain.so"
/* uncomment the next two lines for presence server support
   NOTE: a DB (like db_mysql) module must be also loaded */
loadmodule "presence.so"
loadmodule "presence_xml.so"


# - setting module-specific parameters ---


# - mi_fifo params -
modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")


# - rr params -
# add value to ;lr param to cope with most of the UAs
modparam("rr", "enable_full_lr", 1)
# do not append from tag to the RR (no need for this script)
modparam("rr", "append_fromtag", 0)


# - registrar params -
modparam("registrar", "method_filtering", 1)
/* uncomment the next line to disable parallel forking via location */
# modparam("registrar", "append_branches", 0)
/* uncomment the next line not to allow more than 10 contacts per AOR */
#modparam("registrar", "max_contacts", 10)


# - usrloc params -
#modparam("usrloc", "db_mode",   0)
/* uncomment the following lines if you want to enable DB persistency
   for location entries */
modparam("usrloc", "db_mode",   2)
modparam("usrloc", "db_url",
"mysql://opensips:opensip...@localhost/opensips")


# - uri_db params -
/* by default we disable the DB support in the module as we do not need it
   in this configuration */
modparam("uri_db", "use_uri_table", 0)
modparam("uri_db", "db_url", "")


# - acc params -
/* what sepcial events should be accounted ? */
modparam("acc", "early_media", 1)
modparam("acc", "report_ack", 1)
modparam("acc", "report_cancels",

Re: [OpenSIPS-Users] mysql problem on 1.5

[Bogdan-Andrei Iancu]

Hi Brett,

thanks to your logs, I spoted the problem. The fix is available on SVN.

Thanks and regards,
Bogdan

Brett Nemeroff wrote:
> Bogdan,
> For what it's worth, I've updated to latest 1_5 tonight (about 20 
> minutes ago) and I still am having problems. Full out crashes as well.
>
> I rewrote my queries so I'd have a bunch of little (select * from acc 
> where callid=X) kinds of queries. Of course, there is a lot of DB 
> activity while this happens. Crashes start to happen within seconds of 
> the DB activity ramping up.
>
> For grins, I slowed my queries down to ensure I only did one query per 
> second (in my database, not opensips).. after about 15-20 queries 
> (different each time really) opensips would just crash.
>
> I have acc and sip_trace loaded up, sip_trace isn't active for these 
> calls. Also potentially relevant, my acc table is an InnoDB table.
>
> Now if I slowed my call volume to 1CPS and keep the queries at 1 QPS, 
> it seemed to be happier, but still crashes eventually.
>
> -Brett
>
>
>
> On Mon, Apr 6, 2009 at 11:27 AM, Bogdan-Andrei Iancu 
> mailto:bog...@voice-system.ro>> wrote:
>
> Hi Brett,
>
> it looks like the DB connections are dropped and reconnect is
> taking place (this are the errors about). But to find out the real
> cause, I can enable some more logs to spot the reason for
> re-connect...
>
> I will do it later as right now I'm in the middle of some DB
> debugging and I'm afraid of mixing different patches and what goes
> on SVN :)
>
> Regards,
> Bogdan
>
> Brett Nemeroff wrote:
>
> Hi All,
> So I'm doing some load testing with sipp on my opensips 1.5
> system. I just checked out (like 2 hours ago, the 1.5 branch
> from SVN).  Everything works just fine, until I run some
> rating scripts on my database (perl scripts accessing the
> mysql db directly). While my scripts are running, I see the
> UAS in sipp retransmitting the 200 OKs and the following gets
> printed to the syslog:
> http://www.pastebin.ca/1381169
>
> As soon as my perl script is done, the 200OKs stop
> retransmitting...
> My PERL script isn't doing anything terribly unusual, however,
> it is performing the queries inside of a transaction,
> including a "SELECT/DELETE * FROM acc WHERE " kind of clause.
>
> Any ideas as to what is causing this? I'm afraid I may be
> losing call records..
>
> -Brett
>
> 
> 
>
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>  
>
>
>


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Re: [OpenSIPS-Users] opensips 1.5 with load_balancing

[Bogdan-Andrei Iancu]

Hi Uwe,

But there is not ERROR (as you mentioned) in the log you sent.

Regards,
Bogdan

Uwe Kastens wrote:
> Hi Bogdan,
>
> Here we go.
>
> BR
>
> Uwe
>
>
> Bogdan-Andrei Iancu schrieb:
>   
>> HI Uwe,
>>
>> can you post a debug=6 log of the entire call?
>>
>> Thanks and regards,
>> Bogdan
>>
>> Uwe Kastens wrote:
>> 
>>> Hi,
>>>
>>> I configured load_balancing following the tutorial.
>>>
>>> The call is relayed via t_relay to the 1st pstn gw. After that I will
>>> receive the following error: "ERROR:load_balancer:do_load_balance:
>>> failed to create dialog" and it looks like, that I am missing some
>>> answers.
>>>
>>> BR
>>>
>>> uwe
>>>
>>>
>>>   
>>>   
>> 
>
>
>   


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Re: [OpenSIPS-Users] mysql problem on 1.5

[Brett Nemeroff]

Is that on the 1_5 branch or trunk?

On Wed, Apr 8, 2009 at 7:45 AM, Bogdan-Andrei Iancu
wrote:

> Hi Brett,
>
> thanks to your logs, I spoted the problem. The fix is available on SVN.
>
> Thanks and regards,
> Bogdan
>
> Brett Nemeroff wrote:
>
>> Bogdan,
>> For what it's worth, I've updated to latest 1_5 tonight (about 20 minutes
>> ago) and I still am having problems. Full out crashes as well.
>>
>> I rewrote my queries so I'd have a bunch of little (select * from acc
>> where callid=X) kinds of queries. Of course, there is a lot of DB activity
>> while this happens. Crashes start to happen within seconds of the DB
>> activity ramping up.
>>
>> For grins, I slowed my queries down to ensure I only did one query per
>> second (in my database, not opensips).. after about 15-20 queries (different
>> each time really) opensips would just crash.
>>
>> I have acc and sip_trace loaded up, sip_trace isn't active for these
>> calls. Also potentially relevant, my acc table is an InnoDB table.
>>
>> Now if I slowed my call volume to 1CPS and keep the queries at 1 QPS, it
>> seemed to be happier, but still crashes eventually.
>>
>> -Brett
>>
>>
>>
>> On Mon, Apr 6, 2009 at 11:27 AM, Bogdan-Andrei Iancu <
>> bog...@voice-system.ro > wrote:
>>
>>Hi Brett,
>>
>>it looks like the DB connections are dropped and reconnect is
>>taking place (this are the errors about). But to find out the real
>>cause, I can enable some more logs to spot the reason for
>>re-connect...
>>
>>I will do it later as right now I'm in the middle of some DB
>>debugging and I'm afraid of mixing different patches and what goes
>>on SVN :)
>>
>>Regards,
>>Bogdan
>>
>>Brett Nemeroff wrote:
>>
>>Hi All,
>>So I'm doing some load testing with sipp on my opensips 1.5
>>system. I just checked out (like 2 hours ago, the 1.5 branch
>>from SVN).  Everything works just fine, until I run some
>>rating scripts on my database (perl scripts accessing the
>>mysql db directly). While my scripts are running, I see the
>>UAS in sipp retransmitting the 200 OKs and the following gets
>>printed to the syslog:
>>http://www.pastebin.ca/1381169
>>
>>As soon as my perl script is done, the 200OKs stop
>>retransmitting...
>>My PERL script isn't doing anything terribly unusual, however,
>>it is performing the queries inside of a transaction,
>>including a "SELECT/DELETE * FROM acc WHERE " kind of clause.
>>
>>Any ideas as to what is causing this? I'm afraid I may be
>>losing call records..
>>
>>-Brett
>>
>>
>>  
>>
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>>
>>
>>
>>
>
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Re: [OpenSIPS-Users] no audio from caller when using nathelper

[Bogdan-Andrei Iancu]

Hi Gabriel,

Gabriel Bermudez wrote:
> Thanks for your answer Bogdan
>
> Bogdan-Andrei Iancu escribió:
>> Hi Gabriel,
>>
>> So you are not using rtpproxy, but rely on the fact that * is all the 
>> time public. In this case, audio from caller to * should work all the 
>> time as the destination is public (of course, if the caller does send 
>> RTP). For the other way around, you can be sure * sends RTP to the 
>> public IP of the NAT (of the client) by doing fix_nated_sdp("1") for 
>> the invite - this will force the COMEDIA support in *.
> Yes, some phones set their contact header with the correct public IP 
> address, that's when rptproxy is not used.  In this case they use * 
> directly (but using opensips as a proxy).  I do use 
> fix_nated_sdp("1"), but only when the nat_uac_test("3") gets passed.
maybe the test "3" is not enough to detect all the NAT casestry to 
use more tests to see if makes a difference.
>
>>
>> Anyhow, for RTP nat traversal to work, it is mandatory for the party 
>> behind the nat to start sending RTP (to open the NAT). If the natted 
>> party will send no RTP, there will be no audio at all.
> And that's exactly what wasn't happening, *sometimes* (the sometimes 
> was the one bugging me really).  It seemed not a opensips nat issue 
> but a asterisk nat issue.  So I setted up asterisk's realtime with the 
> following view
>
> CREATE OR REPLACE VIEW sipfriends AS
> SELECT subscriber.username AS name, 'friend'::character varying AS 
> "type", subscriber.username, subscriber."password" AS secret, 
> 'dynamic'::character varying AS host, 'rfc2833'::character varying AS 
> dtmfmode, 'all'::character varying AS disallow, 'g729'::character 
> varying AS allow, 'no'::character varying AS canreinvite, 
> 'yes'::character varying AS nat, 'from-ser'::character varying AS 
> context, ''::character varying AS regserver, 0 AS regseconds
>   FROM subscriber;
>
> As you can see, nat=yes always.  Seems that this solved the problem, 
> I'll do some more testing tomorrow ;)

Cool :)

Regards,
Bogdan
> Thanks for your help.
>
> Regards,
>
>>
>> Regard,
>> Bogdan
>>
>> Gabriel Bermudez wrote:
>>> Hi everyone,
>>>
>>> I'm using the nathelper and dispatcher module to send calls to an 
>>> Asterisk server.  I'm using the Asterisk as a SIP to H.323 converter 
>>> because our PSTN gateway only speaks H.323
>>> For some reason *sometimes* the caller does not send RTP traffic to 
>>> the opensips (one way audio).  The caller's UA is behind a NAT, but 
>>> it doesn't gets detected as a nated UA, so the RTP flow is between 
>>> the client's public IP and the Asterisk public IP (rtpproxy is not 
>>> used).  I'm not sure if this problems happens also with UAs that get 
>>> NAT detected (not seen it happen).  I used tshark to capture the 
>>> invite from an undetected NAT UA (changed the UA ip with 
>>> *uac_public_ip* and opensip's ip with *opensips_public_ip*)


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Re: [OpenSIPS-Users] mysql problem on 1.5

[Bogdan-Andrei Iancu]

Both.

Brett Nemeroff wrote:
> Is that on the 1_5 branch or trunk?
>
>
> On Wed, Apr 8, 2009 at 7:45 AM, Bogdan-Andrei Iancu 
> mailto:bog...@voice-system.ro>> wrote:
>
> Hi Brett,
>
> thanks to your logs, I spoted the problem. The fix is available on
> SVN.
>
>
> Thanks and regards,
> Bogdan
>
> Brett Nemeroff wrote:
>
> Bogdan,
> For what it's worth, I've updated to latest 1_5 tonight (about
> 20 minutes ago) and I still am having problems. Full out
> crashes as well.
>
> I rewrote my queries so I'd have a bunch of little (select *
> from acc where callid=X) kinds of queries. Of course, there is
> a lot of DB activity while this happens. Crashes start to
> happen within seconds of the DB activity ramping up.
>
> For grins, I slowed my queries down to ensure I only did one
> query per second (in my database, not opensips).. after about
> 15-20 queries (different each time really) opensips would just
> crash.
>
> I have acc and sip_trace loaded up, sip_trace isn't active for
> these calls. Also potentially relevant, my acc table is an
> InnoDB table.
>
> Now if I slowed my call volume to 1CPS and keep the queries at
> 1 QPS, it seemed to be happier, but still crashes eventually.
>
> -Brett
>
>
>
> On Mon, Apr 6, 2009 at 11:27 AM, Bogdan-Andrei Iancu
> mailto:bog...@voice-system.ro>
>  >> wrote:
>
>Hi Brett,
>
>it looks like the DB connections are dropped and reconnect is
>taking place (this are the errors about). But to find out
> the real
>cause, I can enable some more logs to spot the reason for
>re-connect...
>
>I will do it later as right now I'm in the middle of some DB
>debugging and I'm afraid of mixing different patches and
> what goes
>on SVN :)
>
>Regards,
>Bogdan
>
>Brett Nemeroff wrote:
>
>Hi All,
>So I'm doing some load testing with sipp on my opensips 1.5
>system. I just checked out (like 2 hours ago, the 1.5
> branch
>from SVN).  Everything works just fine, until I run some
>rating scripts on my database (perl scripts accessing the
>mysql db directly). While my scripts are running, I see the
>UAS in sipp retransmitting the 200 OKs and the
> following gets
>printed to the syslog:
>http://www.pastebin.ca/1381169
>
>As soon as my perl script is done, the 200OKs stop
>retransmitting...
>My PERL script isn't doing anything terribly unusual,
> however,
>it is performing the queries inside of a transaction,
>including a "SELECT/DELETE * FROM acc WHERE " kind of
> clause.
>
>Any ideas as to what is causing this? I'm afraid I may be
>losing call records..
>
>-Brett
>
>  
>  
> 
>
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>
>
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Re: [OpenSIPS-Users] Stop time is not Updated by the internal bye in Local_route

[Bogdan-Andrei Iancu]

Hi Ashwini,

if the flag 3 is for RADIUS accounting, it will not work, as there is 
support for automatic acc in local_route (yet). So, use 
acc_rad_request() from script directly.

Regards,
Bogdan

ASHWINI NAIDU wrote:
> hi bogdan,
>
> I am referring to the stop time generated by radius which is 
> updated in radacct. Anyways i Have resolved the issue.
>
>
>
> On Mon, Apr 6, 2009 at 10:11 PM, Bogdan-Andrei Iancu 
> mailto:bog...@voice-system.ro>> wrote:
>
> Hi Ashwini,
>
> What "stop" time are you refering to?
>
> Regards,
> Bogdan
>
> ASHWINI NAIDU wrote:
>
> hi,
>  When the call controller sends a dlg_end_dlg  to the 2
> end-points after the balance tends to nil. The internal bye is
> entering the local_route but it is not updating the stop time.
>
> Below is the piece of code in local_route
>
> local_route {
>
>if ( method == "BYE") {
>log(1,"\nInternal bye was generated\n");
>setflag(3);
>unforce_rtp_proxy();
>acc_db_request("Internally generated BYE", "acc");
>};
> }
>
>
> -- 
> Thanking You,
> Ashwini BR Naidu
> 
> 
>
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>
>
>
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> Ashwini BR Naidu


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Re: [OpenSIPS-Users] Stop time is not Updated by the internal bye in Local_route

[ASHWINI NAIDU]

Hi Bogdan,

Thank you. I have fixed the issue in the same manner. thanks a lot for
the reply


On Wed, Apr 8, 2009 at 6:29 PM, Bogdan-Andrei Iancu
wrote:

> Hi Ashwini,
>
> if the flag 3 is for RADIUS accounting, it will not work, as there is
> support for automatic acc in local_route (yet). So, use acc_rad_request()
> from script directly.
>
> Regards,
> Bogdan
>
> ASHWINI NAIDU wrote:
>
>> hi bogdan,
>>
>>I am referring to the stop time generated by radius which is updated in
>> radacct. Anyways i Have resolved the issue.
>>
>>
>>
>> On Mon, Apr 6, 2009 at 10:11 PM, Bogdan-Andrei Iancu <
>> bog...@voice-system.ro > wrote:
>>
>>Hi Ashwini,
>>
>>What "stop" time are you refering to?
>>
>>Regards,
>>Bogdan
>>
>>ASHWINI NAIDU wrote:
>>
>>hi,
>> When the call controller sends a dlg_end_dlg  to the 2
>>end-points after the balance tends to nil. The internal bye is
>>entering the local_route but it is not updating the stop time.
>>
>>Below is the piece of code in local_route
>>
>>local_route {
>>
>>   if ( method == "BYE") {
>>   log(1,"\nInternal bye was generated\n");
>>   setflag(3);
>>   unforce_rtp_proxy();
>>   acc_db_request("Internally generated BYE", "acc");
>>   };
>>}
>>
>>
>>--Thanking You,
>>Ashwini BR Naidu
>>
>>  
>>
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>>
>>
>>
>>
>> --
>> Thanking You,
>> Ashwini BR Naidu
>>
>
>


-- 
Thanking You,
Ashwini BR Naidu
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[OpenSIPS-Users] New MediaProxy release 2.3.3 (amd64)

[Adrian Georgescu]

Hello,

amd64 packages have been uploaded to the repository.

To upgrade your debian installation for 64 bit architectures:

apt-get update
apt-get install mediaproxy-dispatcher mediaproxy-relay mediaproxy-web- 
sessions


Kind regards,
Adrian Georgescu



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Re: [OpenSIPS-Users] ACC Problem. to_did blown away.

[Bogdan-Andrei Iancu]

Hi Brett,

The acc module will log the last RURI of the failed branch...so what was 
the RURI before sending the INVITE outbecause if I look at the ACK, 
I guess the INVITE also has "mod_sofia", right ?
In failure route you cannot change the branch that was just completed

If you want something custom to be logged, use the extra_accounting options.

Regards,
Bogdan

Brett Nemeroff wrote:
> Hey All,
> I'm sure I'm doing someting stupid here.. In general, I set all the 
> acc_db flags at the top of my script so everything gets logs. I'm 
> getting 486 Busy from the far end and nice, pretty to_did from the 
> original RURI is being blown away with 'sip:mod_sofia'
>
> Question is.. what do I need to do to get the 486 logged, but to have 
> the right did in the record.. right now I use $oU. Maybe a revert_uri? 
> (am I making that up?)
>
>
> Here's the 486 from my upstream:
>
>
> U 5.6.239.142:5060  -> 1.2.204.8:5060 
> 
>
> SIP/2.0 486 Busy Here.
>
> Via: SIP/2.0/UDP 1.2.204.8;branch=z9hG4bK5a3.473e0af1.0.
>
> Via: SIP/2.0/UDP 
> 2.3.72.138:5060;received=2.3.72.138;branch=z9hG4bK0bf20a91;rport=5060.
>
> From: "custdomain.com "  @2.3.72.138 >;tag=as0e9ef59a.
>
> To:  >;tag=BeZcHaUmX1D8Q.
>
> Call-ID: 29728c50765bdc3275a6bd375a650...@2.3.72.138 
> .
>
> CSeq: 103 INVITE.
>
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12911.
>
> Accept: application/sdp.
>
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, 
> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH.
>
> Supported: timer, precondition, path, replaces.
>
> Content-Length: 0.
>
> .
>
>
>
> U 2.3.72.138:5060  -> 1.2.204.8:5060 
> 
>
> ACK sip:mod_so...@63.211.239.143:5060;transport=udp SIP/2.0.
>
> Via: SIP/2.0/UDP 2.3.72.138:5060;branch=z9hG4bK0bf20a91;rport.
>
> Route: 
> ,.
>
> Max-Forwards: 70.
>
> From: "custdomain.com "  @2.3.72.138 >;tag=as0e9ef59a.
>
> To:  >;tag=BeZcHaUmX1D8Q.
>
> Contact: http://custdomain.com>@2.3.72.138 
> >.
>
> Call-ID: 29728c50765bdc3275a6bd375a650...@2.3.72.138 
> .
>
> CSeq: 103 ACK.
>
> User-Agent: SS.
>
> Content-Length: 0.
>
>
>
>
> 
>
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Re: [OpenSIPS-Users] Modifying INVITE header to add phone-context

[Bogdan-Andrei Iancu]

Hi Julian,

a much nicer option is the add_uri_param() function from URI module:
http://www.opensips.org/html/docs/modules/devel/uri.html#id228164

Regards,
Bogdan

Julian Yap wrote:
> I pretty much solved the issue.
>
> This is what I used:
> subst_uri('/^sip:([0-9]+)@(.*)$/sip:\...@\2;phone-context=sip.server.com/i
>
>
> On Tue, Apr 7, 2009 at 4:25 PM, Julian Yap  wrote:
>   
>> I have a PSTN gateway which requires a Phone-Context value in the
>> outgoing SIP INVITE message to further apply ISDN NPI/TON details.
>>
>> Here's an example of what I currently have going out to the PSTN gateway:
>> INVITE sip:1...@sip.server.com:5060;user=phone SIP/2.0.
>>
>> This is what I require:
>> INVITE sip:1...@sip.server.com:5060;phone-context=sip.server.com;user=phone
>> SIP/2.0.
>>
>> Any clues on how to add the Phone-Context value?
>>
>> Thanks,
>> Julian
>>
>> 
>
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Re: [OpenSIPS-Users] ACC Problem. to_did blown away.

[Brett Nemeroff]

No, actually, the original invite, had a perfectly valid RURI in it.. but
the contact had sip:mod_sofia in it..
I am using extra_accounting and I'm looking $oU, which is why I'm confused
since the actual $oU is valid and not sip:mod_sofia



On Wed, Apr 8, 2009 at 8:40 AM, Bogdan-Andrei Iancu
wrote:

> Hi Brett,
>
> The acc module will log the last RURI of the failed branch...so what was
> the RURI before sending the INVITE outbecause if I look at the ACK, I
> guess the INVITE also has "mod_sofia", right ?
> In failure route you cannot change the branch that was just completed
>
> If you want something custom to be logged, use the extra_accounting
> options.
>
> Regards,
> Bogdan
>
> Brett Nemeroff wrote:
>
>> Hey All,
>> I'm sure I'm doing someting stupid here.. In general, I set all the acc_db
>> flags at the top of my script so everything gets logs. I'm getting 486 Busy
>> from the far end and nice, pretty to_did from the original RURI is being
>> blown away with 'sip:mod_sofia'
>>
>> Question is.. what do I need to do to get the 486 logged, but to have the
>> right did in the record.. right now I use $oU. Maybe a revert_uri? (am I
>> making that up?)
>>
>>
>> Here's the 486 from my upstream:
>>
>>
>> U 5.6.239.142:5060  -> 1.2.204.8:5060 <
>> http://1.2.204.8:5060>
>>
>> SIP/2.0 486 Busy Here.
>>
>> Via: SIP/2.0/UDP 1.2.204.8;branch=z9hG4bK5a3.473e0af1.0.
>>
>> Via: SIP/2.0/UDP 2.3.72.138:5060
>> ;received=2.3.72.138;branch=z9hG4bK0bf20a91;rport=5060.
>>
>> From: "custdomain.com " > http://custdomain.com>@2.3.72.138 >;tag=as0e9ef59a.
>>
>> To:  > sip%3a5212324399...@1.2.204.8 
>> >>;tag=BeZcHaUmX1D8Q.
>>
>> Call-ID: 29728c50765bdc3275a6bd375a650...@2.3.72.138 > 29728c50765bdc3275a6bd375a650...@2.3.72.138>.
>>
>> CSeq: 103 INVITE.
>>
>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12911.
>>
>> Accept: application/sdp.
>>
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
>> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH.
>>
>> Supported: timer, precondition, path, replaces.
>>
>> Content-Length: 0.
>>
>> .
>>
>>
>>
>> U 2.3.72.138:5060  -> 1.2.204.8:5060 <
>> http://1.2.204.8:5060>
>>
>> ACK sip:mod_so...@63.211.239.143:5060;transport=udp SIP/2.0.
>>
>> Via: SIP/2.0/UDP 2.3.72.138:5060;branch=z9hG4bK0bf20a91;rport.
>>
>> Route:
>> ,.
>>
>> Max-Forwards: 70.
>>
>> From: "custdomain.com " > http://custdomain.com>@2.3.72.138 >;tag=as0e9ef59a.
>>
>> To:  > sip%3a5212324399...@1.2.204.8 
>> >>;tag=BeZcHaUmX1D8Q.
>>
>> Contact: http://custdomain.com>@2.3.72.138 <
>> http://2.3.72.138>>.
>>
>> Call-ID: 29728c50765bdc3275a6bd375a650...@2.3.72.138 > 29728c50765bdc3275a6bd375a650...@2.3.72.138>.
>>
>> CSeq: 103 ACK.
>>
>> User-Agent: SS.
>>
>> Content-Length: 0.
>>
>>
>>
>>
>> 
>>
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Re: [OpenSIPS-Users] dispatcher and attended transfers

[Bogdan-Andrei Iancu]

Hi Stan,

if I got it right, you want to have a kind of dispatching to guarantee 
that all in or out calls for user A are going through the same PBX. Correct?

And the problem is when you to a REFERyou have A talking to PBX1 and 
it wants to do a transfer ? Or?

Regards,
Bogdan


Stanisław Pitucha wrote:
> 2009/4/7 Adrian Georgescu :
>   
>> You cannot do this reliable the way you propose. The only reliable way is to
>> sit behind a PBX/B2BUA that your control and behaves in a consistent and
>> reliable way. Otherwise you are at the mercy at the combinations of the SIP
>> User Agents that are involved in the call transfer operation.
>> 
>
> There is only one specific scenario I want to support:
> - phone has a dialog already open to a PBX
> - phone sends an new call INVITE  to a PBX
> - phone joins the call legs with a REFER
>
> I think, this is the PBX/B2BUA situation you're talking about?
>
> I'm not sure what you mean by "the combinations of the SIP User Agents
> that are involved". I didn't have any problems with this setup as long
> as the same phone always uses the same pbx.
>
>   
>> If you will try to fix incrementally every problem your discover in the SIP
>> Proxy for call transfer you will be busy forever solving this because is
>> end-point implementation dependent.
>> 
>
> I'm only trying to solve failover + distribution over PBXes in the
> proxy. Transfers are properly handled by N asterisk hosts.
> To be specific - my network looks like this:
> UAs <-> openser (with dispatcher) <-> N identical asterisk boxes
> All calls go through one of the asterisk boxes.
>
> Stan
>
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Re: [OpenSIPS-Users] generate key

[duane . larson]

Be sure to read through the links that you are given so that you have a  
good understanding, but here are the steps I always take



Now we need to create the TLS certifications and Keys  
(http://www.imacat.idv.tw/tech/sslcerts.html Read Create a Server  
Certificate)
openssl genrsa -des3 -out /etc/ssl/private/openxcap.key 2048 <-- Set  
the password to whatever you want

chmod og-rwx /etc/ssl/private/openxcap.key
openssl req -new -key /etc/ssl/private/openxcap.key -out /tmp/openxcap.req
US
State
City
Home
Home
openxcap01.blahblah.com CA

openssl x509 -req -days 7305 -sha1 \
-extfile /etc/ssl/openssl.cnf -extensions v3_ca \
-signkey /etc/ssl/private/openxcap.key \
-in /tmp/openxcap.req -out /etc/ssl/certs/openxcap.crt

rm -f /tmp/openxcap.req


openssl genrsa -out /etc/openxcap/tls/openxcapserver.key 2048
chmod og-rwx /etc/openxcap/tls/openxcapserver.key
openssl req -new -key /etc/openxcap/tls/openxcapserver.key -out  
/tmp/openxcapserver.req BE SURE NOT TO SET A PASSWORD**

US
State
City
Home
Home
openxcap01.blahblah.com

openssl x509 -req -days 3650 -sha1 \
-extfile /etc/ssl/openssl.cnf -extensions v3_req \
-CA /etc/ssl/certs/openxcap.crt -CAkey /etc/ssl/private/openxcap.key \
-CAserial /etc/ssl/openxcap.srl -CAcreateserial \
-in /tmp/openxcapserver.req -out /etc/openxcap/tls/openxcapserver.crt


openxcap.crt is the key that needs to be given out to the clients (Bria) -  
Copy it to the desktop, open IE and click on Tools -> Internet Options ->  
Content Tab -> Certifications Button -> Import -> And select "Automatically  
select the certificate store based on the type of certificate"

Then configure Bria with the following
Presence Tab - Mode = Presence Agent
Storage Tab - Storage Method = XCAP
Root URL: https://openxcap01.blahblah.com/xcap-root/


Good Luck


On Apr 8, 2009 4:28am, Uwe Kastens  wrote:

Hi Michael,







Try searching for openssl.







http://sial.org/howto/openssl/self-signed/







BR







Uwe




> Hello,




>




> I will generate a certificate and a private key for my server (openxcap)




> - tls/server.crt




> - tls/server.key




>




> i dont know how to generate this files.




>




>




> regards




> michael




>




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[OpenSIPS-Users] dialog profiles, DBs and restarts

[Jeff Pyle]

Hello,

It appears that while dialogs themselves survive opensips restarts (with
db_mode=1), the profile/value associations do not.  Is this configurable?
If not, is there a formal process to submit a feature request?


Thanks,
Jeff


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Re: [OpenSIPS-Users] New MediaProxy release 2.3.3 (amd64)

[Gavin Henry]

2009/4/8 Adrian Georgescu :
> Hello,
>
> amd64 packages have been uploaded to the repository.
>
> To upgrade your debian installation for 64 bit architectures:
>
> apt-get update
> apt-get install mediaproxy-dispatcher mediaproxy-relay mediaproxy-web-
> sessions
>

pbx:/etc/mediaproxy/tls# apt-get install mediaproxy-dispatcher
mediaproxy-relay mediaproxy-web-sessions
Reading package lists... Done
Building dependency tree
Reading state information... Done
The following extra packages will be installed:
  mediaproxy-common
The following NEW packages will be installed
  mediaproxy-common mediaproxy-dispatcher mediaproxy-relay
mediaproxy-web-sessions
0 upgraded, 4 newly installed, 0 to remove and 0 not upgraded.
Need to get 124kB/200kB of archives.
After this operation, 942kB of additional disk space will be used.
Do you want to continue [Y/n]? y
Get: 1 http://ag-projects.com unstable/main mediaproxy-dispatcher 2.3.3 [15.6kB]
Get: 2 http://ag-projects.com unstable/main mediaproxy-relay 2.3.3 [15.7kB]
Get: 3 http://ag-projects.com unstable/main mediaproxy-web-sessions
2.3.3 [92.7kB]
Fetched 124kB in 1s (97.3kB/s)
Selecting previously deselected package mediaproxy-common.
(Reading database ... 26719 files and directories currently installed.)
Unpacking mediaproxy-common (from .../mediaproxy-common_2.3.3_amd64.deb) ...
Selecting previously deselected package mediaproxy-dispatcher.
Unpacking mediaproxy-dispatcher (from
.../mediaproxy-dispatcher_2.3.3_all.deb) ...
dpkg: error processing
/var/cache/apt/archives/mediaproxy-dispatcher_2.3.3_all.deb
(--unpack):
 trying to overwrite `/usr/bin/media-dispatcher', which is also in
package mediaproxy-common
Selecting previously deselected package mediaproxy-relay.
Unpacking mediaproxy-relay (from .../mediaproxy-relay_2.3.3_all.deb) ...
dpkg: error processing
/var/cache/apt/archives/mediaproxy-relay_2.3.3_all.deb (--unpack):
 trying to overwrite `/usr/bin/media-relay', which is also in package
mediaproxy-common
Selecting previously deselected package mediaproxy-web-sessions.
Unpacking mediaproxy-web-sessions (from
.../mediaproxy-web-sessions_2.3.3_all.deb) ...
Errors were encountered while processing:
 /var/cache/apt/archives/mediaproxy-dispatcher_2.3.3_all.deb
 /var/cache/apt/archives/mediaproxy-relay_2.3.3_all.deb
E: Sub-process /usr/bin/dpkg returned an error code (1)

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Re: [OpenSIPS-Users] dialog profiles, DBs and restarts

[Bogdan-Andrei Iancu]

Hi Jeff,

It is a know limitation that I what to address in the next version - 
please open a feature request for it.

Regards,
Bogdan

Jeff Pyle wrote:
> Hello,
>
> It appears that while dialogs themselves survive opensips restarts (with
> db_mode=1), the profile/value associations do not.  Is this configurable?
> If not, is there a formal process to submit a feature request?
>
>
> Thanks,
> Jeff
>
>
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>
>   


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Re: [OpenSIPS-Users] mysql problem on 1.5

[Brett Nemeroff]

Bogdan,I no longer get crashes. However the opensips process hangs pretty
badly while the DB operations are going on. I've tried to rewrite my queries
to do more small queries rather than longer slow ones.

So what I'm doing, I'm using sipp performing calls at 30CPS lasting 10
seconds (to generate a lot of call records).

While this is running, I run my rating script, which gathers unique callid.
smashes records together into a cdr record.

My database engine is InnoDB and I'm using transactions. I'm not actually
getting to a commit in any of this.

So while my script is running. I see on the UAS side of sipp, it
stops receiving calls, and starts performing retransmissions. I've verified
with tshark that packets are hitting opensips, but not getting a reply.

I have 20 children running. Am I doing something wrong?

Thanks for your help,
Brett


On Wed, Apr 8, 2009 at 7:54 AM, Bogdan-Andrei Iancu
wrote:

> Both.
>
> Brett Nemeroff wrote:
>
>> Is that on the 1_5 branch or trunk?
>>
>>
>> On Wed, Apr 8, 2009 at 7:45 AM, Bogdan-Andrei Iancu <
>> bog...@voice-system.ro > wrote:
>>
>>Hi Brett,
>>
>>thanks to your logs, I spoted the problem. The fix is available on
>>SVN.
>>
>>
>>Thanks and regards,
>>Bogdan
>>
>>Brett Nemeroff wrote:
>>
>>Bogdan,
>>For what it's worth, I've updated to latest 1_5 tonight (about
>>20 minutes ago) and I still am having problems. Full out
>>crashes as well.
>>
>>I rewrote my queries so I'd have a bunch of little (select *
>>from acc where callid=X) kinds of queries. Of course, there is
>>a lot of DB activity while this happens. Crashes start to
>>happen within seconds of the DB activity ramping up.
>>
>>For grins, I slowed my queries down to ensure I only did one
>>query per second (in my database, not opensips).. after about
>>15-20 queries (different each time really) opensips would just
>>crash.
>>
>>I have acc and sip_trace loaded up, sip_trace isn't active for
>>these calls. Also potentially relevant, my acc table is an
>>InnoDB table.
>>
>>Now if I slowed my call volume to 1CPS and keep the queries at
>>1 QPS, it seemed to be happier, but still crashes eventually.
>>
>>-Brett
>>
>>
>>
>>On Mon, Apr 6, 2009 at 11:27 AM, Bogdan-Andrei Iancu
>>mailto:bog...@voice-system.ro>
>>>>> wrote:
>>
>>   Hi Brett,
>>
>>   it looks like the DB connections are dropped and reconnect is
>>   taking place (this are the errors about). But to find out
>>the real
>>   cause, I can enable some more logs to spot the reason for
>>   re-connect...
>>
>>   I will do it later as right now I'm in the middle of some DB
>>   debugging and I'm afraid of mixing different patches and
>>what goes
>>   on SVN :)
>>
>>   Regards,
>>   Bogdan
>>
>>   Brett Nemeroff wrote:
>>
>>   Hi All,
>>   So I'm doing some load testing with sipp on my opensips 1.5
>>   system. I just checked out (like 2 hours ago, the 1.5
>>branch
>>   from SVN).  Everything works just fine, until I run some
>>   rating scripts on my database (perl scripts accessing the
>>   mysql db directly). While my scripts are running, I see the
>>   UAS in sipp retransmitting the 200 OKs and the
>>following gets
>>   printed to the syslog:
>>   http://www.pastebin.ca/1381169
>>
>>   As soon as my perl script is done, the 200OKs stop
>>   retransmitting...
>>   My PERL script isn't doing anything terribly unusual,
>>however,
>>   it is performing the queries inside of a transaction,
>>   including a "SELECT/DELETE * FROM acc WHERE " kind of
>>clause.
>>
>>   Any ideas as to what is causing this? I'm afraid I may be
>>   losing call records..
>>
>>   -Brett
>>
>>
>> 
>>
>>   ___
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>>   Users@lists.opensips.org
>>
>>>>
>>
>>   http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
>>
>>
>
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Re: [OpenSIPS-Users] dialog profiles, DBs and restarts

[Jeff Pyle]

Bogdan,

No problem.  Where does one do that?


- Jeff



On 4/8/09 10:26 AM, "Bogdan-Andrei Iancu"  wrote:

> Hi Jeff,
> 
> It is a know limitation that I what to address in the next version -
> please open a feature request for it.
> 
> Regards,
> Bogdan
> 
> Jeff Pyle wrote:
>> Hello,
>> 
>> It appears that while dialogs themselves survive opensips restarts (with
>> db_mode=1), the profile/value associations do not.  Is this configurable?
>> If not, is there a formal process to submit a feature request?
>> 
>> 
>> Thanks,
>> Jeff
>> 
>> 
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>> 
>>   
> 


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Re: [OpenSIPS-Users] dialog profiles, DBs and restarts

[Bogdan-Andrei Iancu]

Jeff, see http://www.opensips.org/index.php?n=Development.Tracker

Regards,
Bogdan

Jeff Pyle wrote:
> Bogdan,
>
> No problem.  Where does one do that?
>
>
> - Jeff
>
>
>
> On 4/8/09 10:26 AM, "Bogdan-Andrei Iancu"  wrote:
>
>   
>> Hi Jeff,
>>
>> It is a know limitation that I what to address in the next version -
>> please open a feature request for it.
>>
>> Regards,
>> Bogdan
>>
>> Jeff Pyle wrote:
>> 
>>> Hello,
>>>
>>> It appears that while dialogs themselves survive opensips restarts (with
>>> db_mode=1), the profile/value associations do not.  Is this configurable?
>>> If not, is there a formal process to submit a feature request?
>>>
>>>
>>> Thanks,
>>> Jeff
>>>
>>>
>>> ___
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>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>   
>>>   
>
>
>   


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Re: [OpenSIPS-Users] dispatcher and attended transfers

[Stanisław Pitucha]

2009/4/8 Bogdan-Andrei Iancu :
> if I got it right, you want to have a kind of dispatching to guarantee that
> all in or out calls for user A are going through the same PBX. Correct?

In short:
Yes. But internal calls should use only one PBX in the cluster, not two.

Long explanation:
Actually, we don't care where the first call is exactly (can be
randomised / load balanced). Just that if some user is already talking
to someone on PBX X, we want his next call to route via X too.

We've got that already working via a hack - we're getting the caller /
callee contact from the dialog table into an avp. Then if
$avp(s:contact){uri.host} matches any of our pbxes, we use it for
routing; otherwise we use a dispatcher. That works relatively well,
but breaks when one of the PBXes dies for example. Dialog stays in the
database forever and that user can't dial out anyone, because he
always gets routed to the dead IP.

We can't simply route all the calls to/from user A to the same PBX,
because we'd like to use only one PBX when doing internal calls. If we
want internal calls to use only one PBX, hashing just on "From:" and
"To:" headers is not enough, because we have 2 transfer scenarios:

A calls B, B calls C, transfers
- (two different destinations - B, C and sources - A, B)

A calls B, A calls C, transfers
- (two different destinations - B, C, same sources)

Good enough solution would be, to set the $avp(s:contact){uri.host} as
the first destination, with failover positions filled with the result
of ds_select_domain().

Ok... I hope that explained the situation :)

Thanks,
Stan

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Re: [OpenSIPS-Users] New MediaProxy release 2.3.3

[Jeff Pyle]

Hello,

I attempted an upgrade from 2.3.2, recompiling from source.  Everything
seemed to build and install okay.  But, when running media-dispatcher on
2.3.3, I receive the following error:

Traceback (most recent call last):
  File "/usr/bin/media-dispatcher", line 32, in ?
log.level.current = config_file.get_option("Dispatcher", 'log_level',
default=log.level.DEBUG, type=datatypes.LogLevel)
AttributeError: 'module' object has no attribute 'level'


I reinstalled 2.3.2 and the error went away.


- Jeff



On 4/8/09 5:08 AM, "Adrian Georgescu"  wrote:

> Hello,
> 
> There is a new release of MediaProxy available, it contains various bug fixes.
> To upgrade your debian installation:
> 
> apt-get update
> apt-get install mediaproxy-dispatcher mediaproxy-relay mediaproxy-web-sessions
> 
> Or download the tar file from:
> 
> http://download.ag-projects.com/MediaProxy/
> 
> The changelog since 2.3.2 is below:
> 
> mediaproxy (2.3.3) unstable; urgency=low
> 
>   * Re-raise the exception on failing to read RADIUS config file so we get a
> full traceback
>   * Have dispatcher close TLS connection cleanly when relay has duplicate IP
>   * Added log_level to both Relay and Dispatcher configuration sections
>   * Improved reconnection behaviour in relay to dispatcher
> When the connection from the relay to the dispatcher is lost, first retry
> in 1 second, then retry in 10 second on subsequent attempts if it loses
> the connection again.
>   * Fix bug where relay connects needlessly to previously removed dispatcher
>   * Implemented a keepalive mechanism from relay to dispatcher
>   * On relay reconnect don't have dispatcher query expired sessions
>   * Removed superfluous datatype declaration
>   * Only allow positive integers for time intervals and delays
>   * Updated version dependency for python-application
>   * In dispatcher, replace old connection from relay with new one instead of
> giving an error
>   * In the dispatcher, check if the reported expired session belongs to the
> relay that reported it
>   * Improved log messages when a relay reconnects to the dispatcher
>   * In dispatcher, break the connection to a relay if a request times out
>   * In dispatcher check if we know about the session that expired at relay
>   * Use a more robust strategy to disconnect an unresponsive relay
> 
> 
> Kind regards,
> Adrian Georgescu
> 
> 
> 
> 
> ___
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Re: [OpenSIPS-Users] New MediaProxy release 2.3.3 (amd64)

[Jeff Pyle]

Similarly,

...
Processing triggers for man-db ...
Errors were encountered while processing:
 /var/cache/apt/archives/mediaproxy-common_2.3.3_amd64.deb
E: Sub-process /usr/bin/dpkg returned an error code (1)


- Jeff



On 4/8/09 10:23 AM, "Gavin Henry"  wrote:

> 2009/4/8 Adrian Georgescu :
>> Hello,
>> 
>> amd64 packages have been uploaded to the repository.
>> 
>> To upgrade your debian installation for 64 bit architectures:
>> 
>> apt-get update
>> apt-get install mediaproxy-dispatcher mediaproxy-relay mediaproxy-web-
>> sessions
>> 
> 
> pbx:/etc/mediaproxy/tls# apt-get install mediaproxy-dispatcher
> mediaproxy-relay mediaproxy-web-sessions
> Reading package lists... Done
> Building dependency tree
> Reading state information... Done
> The following extra packages will be installed:
>   mediaproxy-common
> The following NEW packages will be installed
>   mediaproxy-common mediaproxy-dispatcher mediaproxy-relay
> mediaproxy-web-sessions
> 0 upgraded, 4 newly installed, 0 to remove and 0 not upgraded.
> Need to get 124kB/200kB of archives.
> After this operation, 942kB of additional disk space will be used.
> Do you want to continue [Y/n]? y
> Get: 1 http://ag-projects.com unstable/main mediaproxy-dispatcher 2.3.3
> [15.6kB]
> Get: 2 http://ag-projects.com unstable/main mediaproxy-relay 2.3.3 [15.7kB]
> Get: 3 http://ag-projects.com unstable/main mediaproxy-web-sessions
> 2.3.3 [92.7kB]
> Fetched 124kB in 1s (97.3kB/s)
> Selecting previously deselected package mediaproxy-common.
> (Reading database ... 26719 files and directories currently installed.)
> Unpacking mediaproxy-common (from .../mediaproxy-common_2.3.3_amd64.deb) ...
> Selecting previously deselected package mediaproxy-dispatcher.
> Unpacking mediaproxy-dispatcher (from
> .../mediaproxy-dispatcher_2.3.3_all.deb) ...
> dpkg: error processing
> /var/cache/apt/archives/mediaproxy-dispatcher_2.3.3_all.deb
> (--unpack):
>  trying to overwrite `/usr/bin/media-dispatcher', which is also in
> package mediaproxy-common
> Selecting previously deselected package mediaproxy-relay.
> Unpacking mediaproxy-relay (from .../mediaproxy-relay_2.3.3_all.deb) ...
> dpkg: error processing
> /var/cache/apt/archives/mediaproxy-relay_2.3.3_all.deb (--unpack):
>  trying to overwrite `/usr/bin/media-relay', which is also in package
> mediaproxy-common
> Selecting previously deselected package mediaproxy-web-sessions.
> Unpacking mediaproxy-web-sessions (from
> .../mediaproxy-web-sessions_2.3.3_all.deb) ...
> Errors were encountered while processing:
>  /var/cache/apt/archives/mediaproxy-dispatcher_2.3.3_all.deb
>  /var/cache/apt/archives/mediaproxy-relay_2.3.3_all.deb
> E: Sub-process /usr/bin/dpkg returned an error code (1)
> 
> ___
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Re: [OpenSIPS-Users] New MediaProxy release 2.3.3 (amd64)

[Jeff Pyle]

And trying to create the package from source gives the following error:

Now signing changes and any dsc files...
 signfile mediaproxy_2.3.3.dsc Dan Pascu 
gpg: skipped "Dan Pascu ": secret key not available
gpg: [stdin]: clearsign failed: secret key not available
debsign: gpg error occurred!  Aborting
debuild: fatal error at line 1250:
running debsign failed


- Jeff



On 4/8/09 10:57 AM, "Jeff Pyle"  wrote:

> Similarly,
> 
> ...
> Processing triggers for man-db ...
> Errors were encountered while processing:
>  /var/cache/apt/archives/mediaproxy-common_2.3.3_amd64.deb
> E: Sub-process /usr/bin/dpkg returned an error code (1)
> 
> 
> - Jeff
> 
> 
> 
> On 4/8/09 10:23 AM, "Gavin Henry"  wrote:
> 
>> 2009/4/8 Adrian Georgescu :
>>> Hello,
>>> 
>>> amd64 packages have been uploaded to the repository.
>>> 
>>> To upgrade your debian installation for 64 bit architectures:
>>> 
>>> apt-get update
>>> apt-get install mediaproxy-dispatcher mediaproxy-relay mediaproxy-web-
>>> sessions
>>> 
>> 
>> pbx:/etc/mediaproxy/tls# apt-get install mediaproxy-dispatcher
>> mediaproxy-relay mediaproxy-web-sessions
>> Reading package lists... Done
>> Building dependency tree
>> Reading state information... Done
>> The following extra packages will be installed:
>>   mediaproxy-common
>> The following NEW packages will be installed
>>   mediaproxy-common mediaproxy-dispatcher mediaproxy-relay
>> mediaproxy-web-sessions
>> 0 upgraded, 4 newly installed, 0 to remove and 0 not upgraded.
>> Need to get 124kB/200kB of archives.
>> After this operation, 942kB of additional disk space will be used.
>> Do you want to continue [Y/n]? y
>> Get: 1 http://ag-projects.com unstable/main mediaproxy-dispatcher 2.3.3
>> [15.6kB]
>> Get: 2 http://ag-projects.com unstable/main mediaproxy-relay 2.3.3 [15.7kB]
>> Get: 3 http://ag-projects.com unstable/main mediaproxy-web-sessions
>> 2.3.3 [92.7kB]
>> Fetched 124kB in 1s (97.3kB/s)
>> Selecting previously deselected package mediaproxy-common.
>> (Reading database ... 26719 files and directories currently installed.)
>> Unpacking mediaproxy-common (from .../mediaproxy-common_2.3.3_amd64.deb) ...
>> Selecting previously deselected package mediaproxy-dispatcher.
>> Unpacking mediaproxy-dispatcher (from
>> .../mediaproxy-dispatcher_2.3.3_all.deb) ...
>> dpkg: error processing
>> /var/cache/apt/archives/mediaproxy-dispatcher_2.3.3_all.deb
>> (--unpack):
>>  trying to overwrite `/usr/bin/media-dispatcher', which is also in
>> package mediaproxy-common
>> Selecting previously deselected package mediaproxy-relay.
>> Unpacking mediaproxy-relay (from .../mediaproxy-relay_2.3.3_all.deb) ...
>> dpkg: error processing
>> /var/cache/apt/archives/mediaproxy-relay_2.3.3_all.deb (--unpack):
>>  trying to overwrite `/usr/bin/media-relay', which is also in package
>> mediaproxy-common
>> Selecting previously deselected package mediaproxy-web-sessions.
>> Unpacking mediaproxy-web-sessions (from
>> .../mediaproxy-web-sessions_2.3.3_all.deb) ...
>> Errors were encountered while processing:
>>  /var/cache/apt/archives/mediaproxy-dispatcher_2.3.3_all.deb
>>  /var/cache/apt/archives/mediaproxy-relay_2.3.3_all.deb
>> E: Sub-process /usr/bin/dpkg returned an error code (1)
>> 
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>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 
> 
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Re: [OpenSIPS-Users] New MediaProxy release 2.3.3 (amd64)

[Gavin Henry]

2009/4/8 Jeff Pyle :
> And trying to create the package from source gives the following error:
>
> Now signing changes and any dsc files...
>  signfile mediaproxy_2.3.3.dsc Dan Pascu 
> gpg: skipped "Dan Pascu ": secret key not available
> gpg: [stdin]: clearsign failed: secret key not available
> debsign: gpg error occurred!  Aborting
> debuild: fatal error at line 1250:
> running debsign failed

The INSTALL guide mentions this and that it's ok due to you not having
the signing key. You'll find the deb in the ../ directory.

Cheers.

-- 
http://www.suretecsystems.com/services/openldap/

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Re: [OpenSIPS-Users] New MediaProxy release 2.3.3

[Adrian Georgescu]

I personally do not know what you are doing wrong. If you recompile  
from sources it depends on many factors, what system you have and what  
dependencies your have also installed. So is hard to understand what  
the source of your problem might be from the few information you have  
provided.


Maybe other people can share their experience to see if your problem  
surfaces someplace else.


Regards,
Adrian

On Apr 8, 2009, at 4:54 PM, Jeff Pyle wrote:


Hello,

I attempted an upgrade from 2.3.2, recompiling from source.   
Everything seemed to build and install okay.  But, when running  
media-dispatcher on 2.3.3, I receive the following error:


Traceback (most recent call last):
  File "/usr/bin/media-dispatcher", line 32, in ?
log.level.current = config_file.get_option("Dispatcher",  
'log_level', default=log.level.DEBUG, type=datatypes.LogLevel)

AttributeError: 'module' object has no attribute 'level'


I reinstalled 2.3.2 and the error went away.


- Jeff



On 4/8/09 5:08 AM, "Adrian Georgescu"  wrote:

> Hello,
>
> There is a new release of MediaProxy available, it contains  
various bug fixes.

> To upgrade your debian installation:
>
> apt-get update
> apt-get install mediaproxy-dispatcher mediaproxy-relay mediaproxy- 
web-sessions

>
> Or download the tar file from:
>
> http://download.ag-projects.com/MediaProxy/
>
> The changelog since 2.3.2 is below:
>
> mediaproxy (2.3.3) unstable; urgency=low
>
>   * Re-raise the exception on failing to read RADIUS config file  
so we get a

> full traceback
>   * Have dispatcher close TLS connection cleanly when relay has  
duplicate IP
>   * Added log_level to both Relay and Dispatcher configuration  
sections

>   * Improved reconnection behaviour in relay to dispatcher
> When the connection from the relay to the dispatcher is lost,  
first retry
> in 1 second, then retry in 10 second on subsequent attempts if  
it loses

> the connection again.
>   * Fix bug where relay connects needlessly to previously removed  
dispatcher

>   * Implemented a keepalive mechanism from relay to dispatcher
>   * On relay reconnect don't have dispatcher query expired sessions
>   * Removed superfluous datatype declaration
>   * Only allow positive integers for time intervals and delays
>   * Updated version dependency for python-application
>   * In dispatcher, replace old connection from relay with new one  
instead of

> giving an error
>   * In the dispatcher, check if the reported expired session  
belongs to the

> relay that reported it
>   * Improved log messages when a relay reconnects to the dispatcher
>   * In dispatcher, break the connection to a relay if a request  
times out
>   * In dispatcher check if we know about the session that expired  
at relay

>   * Use a more robust strategy to disconnect an unresponsive relay
>
>
> Kind regards,
> Adrian Georgescu
>
>
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users


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Re: [OpenSIPS-Users] New MediaProxy release 2.3.3 (amd64)

[Adrian Georgescu]

You obviously do not have the key of the developer who made the  
package but you still have the package built. This is not an error per  
se.


Adrian

On Apr 8, 2009, at 5:06 PM, Jeff Pyle wrote:

And trying to create the package from source gives the following  
error:


Now signing changes and any dsc files...
signfile mediaproxy_2.3.3.dsc Dan Pascu 
gpg: skipped "Dan Pascu ": secret key not  
available

gpg: [stdin]: clearsign failed: secret key not available
debsign: gpg error occurred!  Aborting
debuild: fatal error at line 1250:
running debsign failed


- Jeff



On 4/8/09 10:57 AM, "Jeff Pyle"  wrote:


Similarly,

...
Processing triggers for man-db ...
Errors were encountered while processing:
/var/cache/apt/archives/mediaproxy-common_2.3.3_amd64.deb
E: Sub-process /usr/bin/dpkg returned an error code (1)


- Jeff



On 4/8/09 10:23 AM, "Gavin Henry"  wrote:


2009/4/8 Adrian Georgescu :

Hello,

amd64 packages have been uploaded to the repository.

To upgrade your debian installation for 64 bit architectures:

apt-get update
apt-get install mediaproxy-dispatcher mediaproxy-relay mediaproxy- 
web-

sessions



pbx:/etc/mediaproxy/tls# apt-get install mediaproxy-dispatcher
mediaproxy-relay mediaproxy-web-sessions
Reading package lists... Done
Building dependency tree
Reading state information... Done
The following extra packages will be installed:
 mediaproxy-common
The following NEW packages will be installed
 mediaproxy-common mediaproxy-dispatcher mediaproxy-relay
mediaproxy-web-sessions
0 upgraded, 4 newly installed, 0 to remove and 0 not upgraded.
Need to get 124kB/200kB of archives.
After this operation, 942kB of additional disk space will be used.
Do you want to continue [Y/n]? y
Get: 1 http://ag-projects.com unstable/main mediaproxy-dispatcher  
2.3.3

[15.6kB]
Get: 2 http://ag-projects.com unstable/main mediaproxy-relay 2.3.3  
[15.7kB]

Get: 3 http://ag-projects.com unstable/main mediaproxy-web-sessions
2.3.3 [92.7kB]
Fetched 124kB in 1s (97.3kB/s)
Selecting previously deselected package mediaproxy-common.
(Reading database ... 26719 files and directories currently  
installed.)
Unpacking mediaproxy-common (from .../mediaproxy- 
common_2.3.3_amd64.deb) ...

Selecting previously deselected package mediaproxy-dispatcher.
Unpacking mediaproxy-dispatcher (from
.../mediaproxy-dispatcher_2.3.3_all.deb) ...
dpkg: error processing
/var/cache/apt/archives/mediaproxy-dispatcher_2.3.3_all.deb
(--unpack):
trying to overwrite `/usr/bin/media-dispatcher', which is also in
package mediaproxy-common
Selecting previously deselected package mediaproxy-relay.
Unpacking mediaproxy-relay (from .../mediaproxy- 
relay_2.3.3_all.deb) ...

dpkg: error processing
/var/cache/apt/archives/mediaproxy-relay_2.3.3_all.deb (--unpack):
trying to overwrite `/usr/bin/media-relay', which is also in package
mediaproxy-common
Selecting previously deselected package mediaproxy-web-sessions.
Unpacking mediaproxy-web-sessions (from
.../mediaproxy-web-sessions_2.3.3_all.deb) ...
Errors were encountered while processing:
/var/cache/apt/archives/mediaproxy-dispatcher_2.3.3_all.deb
/var/cache/apt/archives/mediaproxy-relay_2.3.3_all.deb
E: Sub-process /usr/bin/dpkg returned an error code (1)

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Re: [OpenSIPS-Users] New MediaProxy release 2.3.3 (amd64)

[Jeff Pyle]

Understood.  I¹m rather new to the deb package process, and I did discover
the deb packages one level up.  Is it possible to run a 2.3.2 dispatcher and
a 2.3.3 relay, or do the versions needs to match exactly?


- Jeff



On 4/8/09 11:11 AM, "Adrian Georgescu"  wrote:

> You obviously do not have the key of the developer who made the package but
> you still have the package built. This is not an error per se.
> 
> Adrian
> 
> On Apr 8, 2009, at 5:06 PM, Jeff Pyle wrote:
> 
>> And trying to create the package from source gives the following error:
>> 
>> Now signing changes and any dsc files...
>>  signfile mediaproxy_2.3.3.dsc Dan Pascu 
>> gpg: skipped "Dan Pascu ": secret key not available
>> gpg: [stdin]: clearsign failed: secret key not available
>> debsign: gpg error occurred!  Aborting
>> debuild: fatal error at line 1250:
>> running debsign failed
>> 
>> 
>> - Jeff
>> 
>> 
>> 
>> On 4/8/09 10:57 AM, "Jeff Pyle"  wrote:
>> 
>>> Similarly,
>>> 
>>> ...
>>> Processing triggers for man-db ...
>>> Errors were encountered while processing:
>>>  /var/cache/apt/archives/mediaproxy-common_2.3.3_amd64.deb
>>> E: Sub-process /usr/bin/dpkg returned an error code (1)
>>> 
>>> 
>>> - Jeff
>>> 
>>> 
>>> 
>>> On 4/8/09 10:23 AM, "Gavin Henry"  wrote:
>>> 
 2009/4/8 Adrian Georgescu :
> Hello,
> 
> amd64 packages have been uploaded to the repository.
> 
> To upgrade your debian installation for 64 bit architectures:
> 
> apt-get update
> apt-get install mediaproxy-dispatcher mediaproxy-relay mediaproxy-web-
> sessions
> 
 
 pbx:/etc/mediaproxy/tls# apt-get install mediaproxy-dispatcher
 mediaproxy-relay mediaproxy-web-sessions
 Reading package lists... Done
 Building dependency tree
 Reading state information... Done
 The following extra packages will be installed:
   mediaproxy-common
 The following NEW packages will be installed
   mediaproxy-common mediaproxy-dispatcher mediaproxy-relay
 mediaproxy-web-sessions
 0 upgraded, 4 newly installed, 0 to remove and 0 not upgraded.
 Need to get 124kB/200kB of archives.
 After this operation, 942kB of additional disk space will be used.
 Do you want to continue [Y/n]? y
 Get: 1 http://ag-projects.com unstable/main mediaproxy-dispatcher 2.3.3
 [15.6kB]
 Get: 2 http://ag-projects.com unstable/main mediaproxy-relay 2.3.3 [15.7kB]
 Get: 3 http://ag-projects.com unstable/main mediaproxy-web-sessions
 2.3.3 [92.7kB]
 Fetched 124kB in 1s (97.3kB/s)
 Selecting previously deselected package mediaproxy-common.
 (Reading database ... 26719 files and directories currently installed.)
 Unpacking mediaproxy-common (from .../mediaproxy-common_2.3.3_amd64.deb)
 ...
 Selecting previously deselected package mediaproxy-dispatcher.
 Unpacking mediaproxy-dispatcher (from
 .../mediaproxy-dispatcher_2.3.3_all.deb) ...
 dpkg: error processing
 /var/cache/apt/archives/mediaproxy-dispatcher_2.3.3_all.deb
 (--unpack):
  trying to overwrite `/usr/bin/media-dispatcher', which is also in
 package mediaproxy-common
 Selecting previously deselected package mediaproxy-relay.
 Unpacking mediaproxy-relay (from .../mediaproxy-relay_2.3.3_all.deb) ...
 dpkg: error processing
 /var/cache/apt/archives/mediaproxy-relay_2.3.3_all.deb (--unpack):
  trying to overwrite `/usr/bin/media-relay', which is also in package
 mediaproxy-common
 Selecting previously deselected package mediaproxy-web-sessions.
 Unpacking mediaproxy-web-sessions (from
 .../mediaproxy-web-sessions_2.3.3_all.deb) ...
 Errors were encountered while processing:
  /var/cache/apt/archives/mediaproxy-dispatcher_2.3.3_all.deb
  /var/cache/apt/archives/mediaproxy-relay_2.3.3_all.deb
 E: Sub-process /usr/bin/dpkg returned an error code (1)
 
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[OpenSIPS-Users] CDRTool - ShowPrice -> No match for gateway parameter

[Dan-Cristian Bogos]

Guys,

some strange thing I noticed in the last versions of CDRTool related to
usage of the Gateway parameter in ShowPrice. Based on logs it looks like
the gateway parameter is somehow faked (or perhaps wrongly converted).

1. On ShowPrice commands:

 * Using default dataset, I have replaced the default entry (gateway,
domain, subscriber empty) with (gateway=10.0.0.1 , subscriber and domain
empty). Reloaded the cdrtool from console and executed:
ShowPrice From=sip:1...@example2.com
To=sip:0031650222...@example.com Gateway=10.0.0.1 Duration=59

The answer was: 
0

In the syslog I could find: 
 Apr  8 17:12:54 DellLaptop cdrtool[11081]: ShowPrice
From=sip:1...@example2.com To=sip:0031650222...@example.com
Gateway=10.0.0.1 Duration=59
Apr  8 17:12:54 DellLaptop cdrtool[11081]: Error: no customer found in
billing_customers table for billing party=...@example2.com,
domain=example2.com, gateway=0.0.0.0

In the mysql table I have:

mysql> select * from billing_customers;
++--+-+---+---+---+---+---+--+---+--+--+
| id | gateway  | domain  | subscriber| profile_name1 |
profile_name1_alt | profile_name2 | profile_name2_alt | timezone
| increment | min_duration | country_code |
++--+-+---+---+---+---+---+--+---+--+--+
|  4 | 10.0.0.1 | |   | 441   |
| 442   |   | Europe/Amsterdam | 0 |
0 |  | 
|  5 |  | example.com |   | 441   |
| 442   |   | Europe/Amsterdam | 0 |
0 |  | 
|  6 |  | | al...@example.com | 441   |
| 442   |   | Europe/Amsterdam | 0 |
0 |  | 
++--+-+---+---+---+---+---+--+---+--+--+
3 rows in set (0.01 sec)


Ta,
DanB


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[OpenSIPS-Users] ACK timout OpenSIPS 1.5

[Khan]

Hi everyone,

I'm rookie in SIP technology, strugling with several issues. I am
having problem with UAC's outside network. I have 3 UAC registered
within the network (SJ Phone, Xlite) they are working fine, I can talk
within the network but the problem arrise when I use the UAC outside
my network. I am seeing two different things from two different UAC's.

1. Xlite on a network behind NAT try to register, it registers
successfully after receiving 200 OK it starts senting SUBSCRIBE
requests, which results in 483 Erro (set up in my config) and when
call is placed on this it gives ACK time out, person on the other side
can hear me but i cant hear him.

2. SjPhone is on another network behind NAT, it regiesters fine, and
after registration it starts sending OPTIONS request, which I have
configured to respond as 200 OK. It continiously keep sending the
requst and my config respond to 200 OK.

My question is several parts, what am I doing wrong,
a) why don't I get ACK after 200 OK,
b) how do i handle SUBSCRIBE requests
c) how do i handle OPTIONS request

The sever is simply being used as SIP server for calls, no IM, Video,
or other applications are implemented yet. There are OpenSIPS, MySQL
server, and RTPProxy is running on the box.

Please respond to my request considering my skills in the SIP as
rookie, guide me on how to resolve problem...

Thanks,


Khan

Sorry for such a long email, I am frustrated :(

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Re: [OpenSIPS-Users] RtpRroxy sturtup (init.d) script for redhat

[Maxim Sobolev]

Bogdan-Andrei Iancu wrote:
> Hi Vladimir,
> 
> really nice, indeed - I did this manually all the time :)
> 
> Maybe Maxim can integrate this directly in the RTPproxy project

Yes, I will do it.

In fact we plan moving towards multi-threading design in the next 
release, which should make utilizing multi-core chips much easier.

Regards,
-- 
Maksym Sobolyev
Sippy Software, Inc.
Internet Telephony (VoIP) Experts
T/F: +1-646-651-1110
Web: http://www.sippysoft.com
MSN: sa...@sippysoft.com
Skype: SippySoft

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Re: [OpenSIPS-Users] RtpRroxy sturtup (init.d) script for redhat

[Maxim Sobolev]

Romanov Vladimir wrote:
> Hi!
> Could you please add command line option to change syslog FACILITY? Now I 
> simply modify this in source and recompile.

Vladimir,

Can you please send a patch?

Thanks!

Regards,
-- 
Maksym Sobolyev
Sippy Software, Inc.
Internet Telephony (VoIP) Experts
T/F: +1-646-651-1110
Web: http://www.sippysoft.com
MSN: sa...@sippysoft.com
Skype: SippySoft

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Re: [OpenSIPS-Users] sst min-se problem

[Jeff Pyle]

Hi Bogdan,

If the current code is operating contrary to the RFC, how might one such as
me request it be updated?


- Jeff



On 4/6/09 1:10 PM, "Bogdan-Andrei Iancu"  wrote:

> Looking in the code, the 422 is sent only if the proxy min-se (1800) is
> smaller than the min(received-min_se(90), received-se(300)) -> 1800 < 90
> -> false, no 422.
> 
> But reading the RFC 4028, I would say the condition is the other way
> around - if the local min-se is higher than min(received-min_se(90),
> received-se(300)) , the 422 should be sent out.
> 
> Regards,
> Bogdan
> 
> Jeff Pyle wrote:
>> Hi Bogdan,
>> 
>> Makes sense, but the why didn't the proxy reject the request with a 422
>> since the Session-Expires from the request is less than the proxy's Min-SE
>> of 1800?
>> 
>> 
>> - Jeff
>> 
>> 
>> 
>> On 4/6/09 12:56 PM, "Bogdan-Andrei Iancu"  wrote:
>> 
>>   
>>> Hi Jeff,
>>> 
>>> What you configure is the min-se of the proxy. (1800)
>>> 
>>> In
>>> DBG:sst:sst_check_min: Session-Expires: 300; MIN-SE: 90
>>> 
>>> are the values from received from request.
>>> 
>>> Regards,
>>> Bogdan
>>> 
>>> Jeff Pyle wrote:
>>> 
 Hello,
 
 I have the sst module configured as follows:
 
 loadmodule "sst.so"
 modparam("sst|dialog", "timeout_avp", "$avp(s:sst_timeout)")
 modparam("sst", "sst_flag", 6)
 modparam("sst", "enable_stats", 1)
 modparam("sst", "min_se", 1800)
 modparam("sst", "reject_to_small", 1)
 
 
 Opensips 1.5 receives an invite containing the following header:
 
   Session-Expires: 300
 
 sstCheckMin("1") at debug=6 shows this:
 
   DBG:sst:sst_check_min: No MIN-SE header found.
   DBG:sst:sst_check_min: Session-Expires: 300; MIN-SE: 90
   DBG:sst:sst_check_min: Done returning false (-1)
 
 
 Since the invite from my gateway didn't contain a MIN-SE, why doesn't it
 use
 the 1800 provided at the modparam?
 
 
 Thanks,
 Jeff
 
 
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>> 
>> 
>>   
> 


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Re: [OpenSIPS-Users] ACK timout OpenSIPS 1.5

[Uwe Kastens]

Hi Khan,

A easy way to debug this problem is to use a kind of network sniffer on
your opensips and directly after your UA. Try to debug this issue with a
softphone like xlite, so you can start your network dump on the client.

BR

Uwe

Khan schrieb:
> Hi everyone,
> 
> I'm rookie in SIP technology, strugling with several issues. I am
> having problem with UAC's outside network. I have 3 UAC registered
> within the network (SJ Phone, Xlite) they are working fine, I can talk
> within the network but the problem arrise when I use the UAC outside
> my network. I am seeing two different things from two different UAC's.
> 
> 1. Xlite on a network behind NAT try to register, it registers
> successfully after receiving 200 OK it starts senting SUBSCRIBE
> requests, which results in 483 Erro (set up in my config) and when
> call is placed on this it gives ACK time out, person on the other side
> can hear me but i cant hear him.
> 
> 2. SjPhone is on another network behind NAT, it regiesters fine, and
> after registration it starts sending OPTIONS request, which I have
> configured to respond as 200 OK. It continiously keep sending the
> requst and my config respond to 200 OK.
> 
> My question is several parts, what am I doing wrong,
> a) why don't I get ACK after 200 OK,
> b) how do i handle SUBSCRIBE requests
> c) how do i handle OPTIONS request
> 
> The sever is simply being used as SIP server for calls, no IM, Video,
> or other applications are implemented yet. There are OpenSIPS, MySQL
> server, and RTPProxy is running on the box.
> 
> Please respond to my request considering my skills in the SIP as
> rookie, guide me on how to resolve problem...
> 
> Thanks,
> 
> 
> Khan
> 
> Sorry for such a long email, I am frustrated :(
> 
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Re: [OpenSIPS-Users] opensips 1.5 with load_balancing

[Uwe Kastens]

Hi Bogdan,

Sorry, I need to clear up the configuration before trying use
loadbalancer. The behaviour was every time I made a call a little bit
strange - but every time in an different way.

I will setup some virtual servers and play around with the configuration.

Thanks

Uwe

Bogdan-Andrei Iancu schrieb:
> Hi Uwe,
> 
> But there is not ERROR (as you mentioned) in the log you sent.
> 
> Regards,
> Bogdan
> 
> Uwe Kastens wrote:
>> Hi Bogdan,
>>
>> Here we go.
>>
>> BR
>>
>> Uwe
>>
>>
>> Bogdan-Andrei Iancu schrieb:
>>  
>>> HI Uwe,
>>>
>>> can you post a debug=6 log of the entire call?
>>>
>>> Thanks and regards,
>>> Bogdan
>>>
>>> Uwe Kastens wrote:
>>>
 Hi,

 I configured load_balancing following the tutorial.

 The call is relayed via t_relay to the 1st pstn gw. After that I will
 receive the following error: "ERROR:load_balancer:do_load_balance:
 failed to create dialog" and it looks like, that I am missing some
 answers.

 BR

 uwe


 
>>> 
>>
>>
>>   
> 
> 


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Re: [OpenSIPS-Users] CDRTool - ShowPrice -> No match for gateway parameter

[Dan-Cristian Bogos]

More on the subject ...

Just to be sure that I am not doing any mistake, the log of mysql for
the same command (ShowPrice From=sip:1...@example2.com
To=sip:0031650222...@example.com Gateway=10.0.0.1 Duration=59) shows the
gateway parameter queried as suspected, faked:

090408 21:17:22 314 Init DB cdrtool
314 Query   select * from billing_customers
where subscriber = '1...@example2.com'
or domain= 'example2.com'
or gateway   = '0.0.0.0'
or (subscriber = '' and domain = '' and gateway = '')
order by subscriber desc, domain desc, gateway desc limit 1


Ta,
DanB

On Wed, 2009-04-08 at 17:34 +0200, Dan-Cristian Bogos wrote:
> Guys,
> 
> some strange thing I noticed in the last versions of CDRTool related to
> usage of the Gateway parameter in ShowPrice. Based on logs it looks like
> the gateway parameter is somehow faked (or perhaps wrongly converted).
> 
> 1. On ShowPrice commands:
> 
>  * Using default dataset, I have replaced the default entry (gateway,
> domain, subscriber empty) with (gateway=10.0.0.1 , subscriber and domain
> empty). Reloaded the cdrtool from console and executed:
> ShowPrice From=sip:1...@example2.com
> To=sip:0031650222...@example.com Gateway=10.0.0.1 Duration=59
> 
> The answer was: 
> 0
> 
> In the syslog I could find: 
>  Apr  8 17:12:54 DellLaptop cdrtool[11081]: ShowPrice
> From=sip:1...@example2.com To=sip:0031650222...@example.com
> Gateway=10.0.0.1 Duration=59
> Apr  8 17:12:54 DellLaptop cdrtool[11081]: Error: no customer found in
> billing_customers table for billing party=...@example2.com,
> domain=example2.com, gateway=0.0.0.0
> 
> In the mysql table I have:
> 
> mysql> select * from billing_customers;
> ++--+-+---+---+---+---+---+--+---+--+--+
> | id | gateway  | domain  | subscriber| profile_name1 |
> profile_name1_alt | profile_name2 | profile_name2_alt | timezone
> | increment | min_duration | country_code |
> ++--+-+---+---+---+---+---+--+---+--+--+
> |  4 | 10.0.0.1 | |   | 441   |
> | 442   |   | Europe/Amsterdam | 0 |
> 0 |  | 
> |  5 |  | example.com |   | 441   |
> | 442   |   | Europe/Amsterdam | 0 |
> 0 |  | 
> |  6 |  | | al...@example.com | 441   |
> | 442   |   | Europe/Amsterdam | 0 |
> 0 |  | 
> ++--+-+---+---+---+---+---+--+---+--+--+
> 3 rows in set (0.01 sec)
> 
> 
> Ta,
> DanB


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