Re: [OpenSIPS-Users] ACK never reach UAS

2011-12-28 Thread Jose Solares 
I learned the hard way that sipp doesn't support record routing in their 
default scenarios, at first i thought the example for load balancing was wrong 
or something had changed from 1.5 to 1.7, sipp to asterisk directly worked 
fine, but sipp -> opensips -> 2 asterisk didn't, the acks were reaching 
opensips but opensips didn't send them on.

so i had to change it, first you dump the xml with `sipp -sd uac`, then you add 
rrs="true" in your recv tags
Like so:

 

You also need to change the ACK and BYE's like so, since you're using 
rrs="true" you have [next_url] and [routes] available to use:

ACK [next_url] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp ;tag=[pid]SIPpTag00[call_number]
To: sut [peer_tag_param]
[routes]
Call-ID: [call_id]
CSeq: 1 ACK
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0

This worked for the UAC part, not sure what would need to change in the UAS but 
it should be similar.

Regards,

On Dec 28, 2011, at 6:45 PM, M.Abdulaziz wrote:

> Thank you Bogdan for replying,
> 
> 
> UAC send the ACK to the correct port of opensips but after it reaches
> opensips, opensips doesn't know where to send this ACK
> 
> I tried to perform record routing in my cfg as in the performance test
> tutorial( http://www.opensips.org/Resources/StressTests#toc6 test_E.cfg ) 
> 
> the same problem happend, ACKs is never forwarded
> 
> I traced the packages exchanged using wireshark & I found out that ACK is
> looping inside opensips
> here is the wireshark trace of only one call
> http://opensips-open-sip-server.1449251.n2.nabble.com/file/n7134497/_221-6320%40192.168.1.69-Trace
> _221-6320%40192.168.1.69-Trace 
> 
> Attached the trace of wireshark
> 
> Note that port numbers were given as follow:
> UAC:5069
> opensips: 5060
> UAS: 5066
> 
> & all resides in the same machine (192.168.1.69)
> 
> here is my config file:
> 
> 
> ### Global Parameters #
> 
> debug=3
> log_stderror=no
> 
> fork=yes
> children=32
> 
> listen=udp:192.168.1.69:5060
> 
> ### Modules Section 
> 
> #set module path
> mpath="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.so"
> loadmodule "dialog.so"
> #loadmodule "acc.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")
> 
> modparam("dialog","db_mode",0)
> 
> 
> # - 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 -
> /* 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:opensipsrw@localhost/opensips")
> 
> 
> # - uri params -
> modparam("uri", "use_uri_table", 0)
> 
> modparam("tm", "wt_timer", 2)
> modparam("tm", "restart_fr_on_each_reply", 0)
> 
> # - 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 accounting also */
> #modparam("acc", "db_flag", 1)
> #modparam("acc

Re: [OpenSIPS-Users] Users Digest, Vol 41, Issue 75

2011-12-28 Thread auto-reply from antonio.spirande...@longwave.eu 
Sarò assente fino all'8 Gennaio compreso. Per urgenze 
rivolgersi direttamente ad assiste...@longwave.eu o 
chiamare lo 0522375500. Saluti

I will be out of office untill  Gen 8th 2012.

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

Re: [OpenSIPS-Users] ACK never reach UAS

2011-12-28 Thread M.Abdulaziz 
Thank you Bogdan for replying,


UAC send the ACK to the correct port of opensips but after it reaches
opensips, opensips doesn't know where to send this ACK

I tried to perform record routing in my cfg as in the performance test
tutorial( http://www.opensips.org/Resources/StressTests#toc6 test_E.cfg ) 

the same problem happend, ACKs is never forwarded

I traced the packages exchanged using wireshark & I found out that ACK is
looping inside opensips
here is the wireshark trace of only one call
http://opensips-open-sip-server.1449251.n2.nabble.com/file/n7134497/_221-6320%40192.168.1.69-Trace
_221-6320%40192.168.1.69-Trace 

Attached the trace of wireshark

Note that port numbers were given as follow:
UAC:5069
opensips: 5060
UAS: 5066

& all resides in the same machine (192.168.1.69)

here is my config file:


### Global Parameters #

debug=3
log_stderror=no

fork=yes
children=32

listen=udp:192.168.1.69:5060

### Modules Section 

#set module path
mpath="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.so"
loadmodule "dialog.so"
#loadmodule "acc.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")

modparam("dialog","db_mode",0)


# - 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 -
/* 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:opensipsrw@localhost/opensips")


# - uri params -
modparam("uri", "use_uri_table", 0)

modparam("tm", "wt_timer", 2)
modparam("tm", "restart_fr_on_each_reply", 0)

# - 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 accounting also */
#modparam("acc", "db_flag", 1)
#modparam("acc", "db_missed_flag", 2)


# - auth_db params -
/* uncomment the following lines if you want to enable the DB based
   authentication */
#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", "")


# - alias_db params -
/* uncomment the following lines if you want to enable the DB based
   aliases */
#modparam("alias_db", "db_url",
#   "mysql://opensips:opensipsrw@localhost/opensips")


# - domain params -
/* uncomment the following lines to enable multi-domain detection
   support */
#modparam("domain", "db_url",
#   "mysql://opensips:opensipsrw@localhost/opensips")
#modparam("domain", "db_mode", 1)   # Use caching


# - multi-module params -
/* uncomment the following line if you want to enable multi-domain support
   in the modules (dafault off) */
#modparam("auth_db|usrloc|uri", "use_domain", 1)


# - presence params -
/* uncomment the following lines if you want to enable presence */
#modparam("presence|presence_xml", "db_url",
#   "mysql://opensips:opensipsrw@localhost/opensips")
#modparam("presence_xml", "force_active", 1)
#modparam("presence", "server_address", "sip:1

[OpenSIPS-Users] Memory allocation failure

2011-12-28 Thread Wesley Volcov 
Hello Guys,

Someone know this error ?

I'm using opensips 1.6.1 with rtpproxy 1.2.1 and Centos 6.0.

Dec 28 14:01:09 veoscol /usr/local/sbin/opensips[27117]:
ERROR:sl:sl_send_reply_helper: response building failed
Dec 28 14:01:09 veoscol /usr/local/sbin/opensips[27116]:
ERROR:core:do_action: memory allocation failure
Dec 28 14:01:09 veoscol /usr/local/sbin/opensips[27116]:
ERROR:core:pv_set_ruri_user: do action failed
Dec 28 14:01:09 veoscol /usr/local/sbin/opensips[27116]:
ERROR:dialplan:dp_update: falied to set the output value!
Dec 28 14:01:09 veoscol /usr/local/sbin/opensips[27116]:
ERROR:dialplan:dp_translate_f: cannot set the output
Dec 28 14:01:09 veoscol /usr/local/sbin/opensips[27117]:
ERROR:core:build_res_buf_from_sip_req: out of pkg memory  ; needs 350
Dec 28 14:01:09 veoscol /usr/local/sbin/opensips[27117]:
ERROR:sl:sl_send_reply_helper: response building failed
Dec 28 14:01:10 veoscol /usr/local/sbin/opensips[27118]:
ERROR:core:build_res_buf_from_sip_req: out of pkg memory  ; needs 382
Dec 28 14:01:10 veoscol /usr/local/sbin/opensips[27118]:
ERROR:sl:sl_send_reply_helper: response building failed
Dec 28 14:01:10 veoscol /usr/local/sbin/opensips[27117]:
ERROR:core:received_builder: out of pkg memory
Dec 28 14:01:10 veoscol /usr/local/sbin/opensips[27117]:
ERROR:core:build_res_buf_from_sip_req: received_builder failed
Dec 28 14:01:10 veoscol /usr/local/sbin/opensips[27117]:
ERROR:sl:sl_send_reply_helper: response building failed
Dec 28 14:01:10 veoscol /usr/local/sbin/opensips[27118]:
ERROR:core:build_res_buf_from_sip_req: out of pkg memory  ; needs 363
Dec 28 14:01:10 veoscol /usr/local/sbin/opensips[27118]:
ERROR:sl:sl_send_reply_helper: response building failed
Dec 28 14:01:10 veoscol /usr/local/sbin/opensips[27117]:
ERROR:core:parse_headers: pkg memory allocation failed

Best regards!

-- 
Wesley Volcov
Email: wesleyvol...@gmail.com
Messenger: vol...@live.com
Mobile: +55 11 7999-7444
Website: http://volcov.blogspot.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Reboot phone with notify?

2011-12-28 Thread Bogdan-Andrei Iancu 
The bottom idea is that you need to send a custom NOTIFY to the phone, 
in order to restart (the ctl function has a builtin functionality for 
the cisco-like phone).


So, you can directly use the MI function from TM to fire a NOTIFY 
request  (via t_uac_dlg command) :

http://www.opensips.org/html/docs/modules/1.7.x/tm.html#id294568

Pass as parameters to the function:
NOTIFY
SIP_RURI_TO_POINT_TO PHONE
SIP_URI_TO_POINT_TO_OPENSIPS
.
"From:xxx \r\nTo: xx \r\nCustom-hdr: x \r\n"

You can use the "opensipsctl fifo " to fire a custom MI command

Regards,
Bogdan

On 12/28/2011 03:47 PM, Schneur Rosenberg wrote:

I understand, in asterisk I can modify the sip_notify.conf file to add
phones, question is how do I do it in opensips for example on linksys
in asterisk I do "sip notify linksys-cold-restart peername" I've seen
that opensips has " cisco_restart" is there something similar
for linsys/sipura phones?

On Wed, Dec 28, 2011 at 3:39 PM, Bogdan-Andrei Iancu
  wrote:

Hi Schneur,

That depends on the phone you are using - so you need to check the docs for
that.

Regards,
Bogdan


On 12/28/2011 03:09 PM, Schneur Rosenberg wrote:

Is there a way to reboot phones, similar to asterisk "sip notify" command?

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



--
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
OpenSIPS solutions and "know-how"




--
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
OpenSIPS solutions and "know-how"


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

Re: [OpenSIPS-Users] B2BUA + RTPproxy + Asterisk direct media

2011-12-28 Thread Lee Archer 
Thanks for the help Bogdan.  I will do some tests and see if I can figure it.

Regards

Lee

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
Sent: 28 December 2011 13:08
To: OpenSIPS users mailling list
Cc: Lee Archer
Subject: Re: [OpenSIPS-Users] B2BUA + RTPproxy + Asterisk direct media

Hello Lee,

Asterisk is doing the "direct media" by firing some re-INVITEs after the call 
is up in order to exchange the media IPs of the the end points.

So, if this does not work, most probably you do not correctly handle the 
re-INVITEs in opensips, like you are no forcing again rtpproxy for re-INVITEs.

Regards,
Bogdan

On 12/28/2011 12:26 PM, Lee Archer wrote:
Hi all, I wonder if someone can help me.  I have a system where I use the B2B 
module and RTPproxy for inbound calls but once answered the call might jump 
between Asterisk servers depending on what service is required.  I would like 
to use the Asterisk direct media option for SIP calls but when enabled the 
server is trying to talk to the SIP providers RTP gateway instead of my 
RTPproxy instance.  I've made changes to the RTPproxy configuration but I'm 
wondering if anyone else uses direct media with RTPproxy and can point me in 
the right direction config wise.

Thanks

Lee


thebigword Holdings Limited. Registered Office: Link Up House, Ring Road, Lower 
Wortley, Leeds, UK, LS12 6AB. Registered in England & Wales No. 05551907







___

Users mailing list

Users@lists.opensips.org

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




--

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

OpenSIPS solutions and "know-how"

thebigword Holdings Limited. Registered Office: Link Up House, Ring Road, Lower 
Wortley, Leeds, UK, LS12 6AB. Registered in England & Wales No. 05551907

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

Re: [OpenSIPS-Users] Users Digest, Vol 41, Issue 74

2011-12-28 Thread auto-reply from antonio.spirande...@longwave.eu 
Sarò assente fino all'8 Gennaio compreso. Per urgenze 
rivolgersi direttamente ad assiste...@longwave.eu o 
chiamare lo 0522375500. Saluti

I will be out of office untill  Gen 8th 2012.

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

Re: [OpenSIPS-Users] exec_avp in timer route

2011-12-28 Thread Bogdan-Andrei Iancu 

Hi guys,

I just opened  a ticket, to allow these functions in STARTUP route.

Regards,
Bogdan

On 12/20/2011 11:35 PM, ddgiants wrote:

Hmmm thats what I am doing using startup and timer routes except I don't use
avp query to get data I use exec_avp.

--
View this message in context: 
http://opensips-open-sip-server.1449251.n2.nabble.com/exec-avp-in-timer-route-tp7103327p7113390.html
Sent from the OpenSIPS - Users mailing list archive at Nabble.com.

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




--
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
OpenSIPS solutions and "know-how"


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

Re: [OpenSIPS-Users] Reboot phone with notify?

2011-12-28 Thread Bogdan-Andrei Iancu 

Hi Schneur,

That depends on the phone you are using - so you need to check the docs 
for that.


Regards,
Bogdan

On 12/28/2011 03:09 PM, Schneur Rosenberg wrote:

Is there a way to reboot phones, similar to asterisk "sip notify" command?

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




--
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
OpenSIPS solutions and "know-how"


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

Re: [OpenSIPS-Users] makeann problem

2011-12-28 Thread Bogdan-Andrei Iancu 

Hi Rodrigo,

To be honest never tried the makeann tool, but to be sure the conversion 
is correct, use "file ann_.wav" and "file ann_.wav.8" to 
check if the the properties (stereo, sampling, etc) are the same.


Regards,
Bogdan

On 12/20/2011 09:03 PM, Rodrigo Ferreira wrote:

Hi guys,
I'm trying to use rtpproxy_stream2uac, but everytime that I try to 
make the announcement file using the "makeann", the voice change, and 
the file turn into something slow.

I'm just using,
#makeann ann_.wav
then two files appears, ann_.wav.0 and ann_.wav.8
There's anything else that I have to do?
Eng.º Rodrigo Ferreira
Supervisor de Telecomunicações
Fone: (13) 4010-1037
Cel: (13) 9615-7774
VIPWay Serviços de Telecomunicações LTDA


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



--
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
OpenSIPS solutions and "know-how"

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

Re: [OpenSIPS-Users] drouting and free memory

2011-12-28 Thread dpa 
Ok, thank you, I will try

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Wednesday, December 28, 2011 5:05 PM
To: OpenSIPS users mailling list
Cc: dpa
Subject: Re: [OpenSIPS-Users] drouting and free memory

 

Hello,

DR module use private memory (pkg mem) in order to load info from DB. Usually 
OpenSIPS has 2M of pkg memory (which is also used by other module aside DR), so 
maybe that available amount of mem is not enough for loading a 40K rows at a 
time. 

Try to decrease the "fetch_rows" param to 1000 for example, to see if ok. This 
will have no side effect, it will simply load data via multiple chunks.

Regards,
Bogdan

On 12/28/2011 02:58 PM, dpa wrote: 

Hello!

 

There is such problem

 

Opensips 1.6.4-2

I am using drouting module to relay any calls.

 

modparam("drouting", "fetch_rows", 4)

 

When I am try using fifo dr_reload I see such error 

“Dec 28 16:38:24 opensips /usr/local/opensips1.6.4-2/sbin/opensips[27360]: 
ERROR:core:db_allocate_rows: no memory left

Dec 28 16:38:24 opensips /usr/local/opensips1.6.4-2/sbin/opensips[27360]: 
ERROR:db_mysql:db_mysql_fetch_result: no memory left

Dec 28 16:38:24 opensips /usr/local/opensips1.6.4-2/sbin/opensips[27360]: 
ERROR:drouting:dr_load_routing_info: Error fetching rows”

 

And I cannot see any problem with memory

“shmem:total_size = 104857600

shmem:used_size = 34574448

shmem:real_used_size = 38811912”

“pkmem:0-total_size = 5242880

pkmem:0-used_size = 591616

pkmem:0-real_used_size = 717016

pkmem:0-max_used_size = 919520

pkmem:0-free_size = 4525864

pkmem:0-fragments = 2”

Thank you for any help

 

 

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






-- 
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
OpenSIPS solutions and "know-how"
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

[OpenSIPS-Users] Reboot phone with notify?

2011-12-28 Thread Schneur Rosenberg 
Is there a way to reboot phones, similar to asterisk "sip notify" command?

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

Re: [OpenSIPS-Users] B2BUA + RTPproxy + Asterisk direct media

2011-12-28 Thread Bogdan-Andrei Iancu 

Hello Lee,

Asterisk is doing the "direct media" by firing some re-INVITEs after the 
call is up in order to exchange the media IPs of the the end points.


So, if this does not work, most probably you do not correctly handle the 
re-INVITEs in opensips, like you are no forcing again rtpproxy for 
re-INVITEs.


Regards,
Bogdan

On 12/28/2011 12:26 PM, Lee Archer wrote:
Hi all, I wonder if someone can help me.  I have a system where I use 
the B2B module and RTPproxy for inbound calls but once answered the 
call might jump between Asterisk servers depending on what service is 
required.  I would like to use the Asterisk direct media option for 
SIP calls but when enabled the server is trying to talk to the SIP 
providers RTP gateway instead of my RTPproxy instance.  I've made 
changes to the RTPproxy configuration but I'm wondering if anyone else 
uses direct media with RTPproxy and can point me in the right 
direction config wise.

Thanks
Lee
thebigword Holdings Limited. Registered Office: Link Up House, Ring Road, Lower 
Wortley, Leeds, UK, LS12 6AB. Registered in England&  Wales No. 05551907



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



--
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
OpenSIPS solutions and "know-how"

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

Re: [OpenSIPS-Users] Users Digest, Vol 41, Issue 73

2011-12-28 Thread auto-reply from antonio.spirande...@longwave.eu 
Sarò assente fino all'8 Gennaio compreso. Per urgenze 
rivolgersi direttamente ad assiste...@longwave.eu o 
chiamare lo 0522375500. Saluti

I will be out of office untill  Gen 8th 2012.

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

Re: [OpenSIPS-Users] drouting and free memory

2011-12-28 Thread Bogdan-Andrei Iancu 

Hello,

DR module use private memory (pkg mem) in order to load info from DB. 
Usually OpenSIPS has 2M of pkg memory (which is also used by other 
module aside DR), so maybe that available amount of mem is not enough 
for loading a 40K rows at a time.


Try to decrease the "fetch_rows" param to 1000 for example, to see if 
ok. This will have no side effect, it will simply load data via multiple 
chunks.


Regards,
Bogdan

On 12/28/2011 02:58 PM, dpa wrote:


Hello!

There is such problem

Opensips 1.6.4-2

I am using drouting module to relay any calls.

modparam("drouting", "fetch_rows", 4)

When I am try using fifo dr_reload I see such error

“Dec 28 16:38:24 opensips 
/usr/local/opensips1.6.4-2/sbin/opensips[27360]: 
ERROR:core:db_allocate_rows: no memory left


Dec 28 16:38:24 opensips 
/usr/local/opensips1.6.4-2/sbin/opensips[27360]: 
ERROR:db_mysql:db_mysql_fetch_result: no memory left


Dec 28 16:38:24 opensips 
/usr/local/opensips1.6.4-2/sbin/opensips[27360]: 
ERROR:drouting:dr_load_routing_info: Error fetching rows”


And I cannot see any problem with memory

“shmem:total_size = 104857600

shmem:used_size = 34574448

shmem:real_used_size = 38811912”

“pkmem:0-total_size = 5242880

pkmem:0-used_size = 591616

pkmem:0-real_used_size = 717016

pkmem:0-max_used_size = 919520

pkmem:0-free_size = 4525864

pkmem:0-fragments = 2”

Thank you for any help


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



--
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
OpenSIPS solutions and "know-how"

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

Re: [OpenSIPS-Users] all sip body headers regarding video removed

2011-12-28 Thread Bogdan-Andrei Iancu 

Hi Cindy,

What you describe as bug #2 is not really a bug - First of all opensips 
works at codec level - you are able to remove /change order of codecs, 
but you cannot remove streams or sessions from SDP.  So whatever info is 
associated to the sessions/streams will stay there. Secondly, opensips 
does not have the fully SDP "wisdom" to be able to remove all attributes 
related to a codec.


Regards,
Bogdan

On 12/21/2011 11:03 PM, Cindy Leung wrote:

Thanks for the reply.

It sounds like the ticket only addresses issue #1.  Do you know if it will also 
fix #2?
"The m= and a= lines for "video" and "application" are still there while the payload 
types are all deleted."



On Dec 16, 2011, at 8:50 AM, Vlad Paiu wrote:


Hello,

What you're describing is a bug, and your issue is very much related to this 
bug report [1].
We'll try to fix it soon so keep an eye on it.

[1] 
http://sourceforge.net/tracker/?func=detail&aid=3436522&group_id=232389&atid=1086410

Regards,

Vlad Paiu
OpenSIPS Developer


On 12/16/2011 12:36 AM, Cindy Leung wrote:

Hello,

I have to delete codec for similar reasons.  I'm trying to use 
codec_delete_except_re().

Here's the incoming SDP:
m=audio 49152 RTP/AVP 123 122 125 121 124 9 0 8 113 100
a=sendrecv
a=ptime:20
a=rtpmap:123 G7221/32000
a=fmtp:123 bitrate=48000
a=rtpmap:122 G7221/32000
a=fmtp:122 bitrate=32000
a=rtpmap:125 G7221/16000
a=fmtp:125 bitrate=32000
a=rtpmap:121 G7221/32000
a=fmtp:121 bitrate=24000
a=rtpmap:124 G7221/16000
a=fmtp:124 bitrate=24000
a=rtpmap:113 MP4A-LATM/32000
a=rtpmap:100 telephone-event/8000
m=video 49154 RTP/AVP 98 34 96 97
a=sendrecv
a=rtpmap:98 H264/9
a=fmtp:98 profile-level-id=42801f; max-fs=21600
a=rtpmap:34 H263/9
a=fmtp:34 CIF=3;QCIF=2
a=rtpmap:96 H263-1998/9
a=fmtp:96 CIF4=1;CIF=1;QCIF=1;F=1;I=1;J=1;T=1
a=rtpmap:97 H263-2000/9
a=fmtp:97 profile=1; level=42
a=content:main
a=rtcp-fb:* nack pli
a=rtcp-fb:* ccm fir
m=application 49158 RTP/AVP 110
a=sendrecv
a=rtpmap:110 H224/4800

... which then goes through
codec_delete_except_re("PCMU|G729|telephone-event");

... and I end up getting
m=audio 28964 RTP/AVP 9 0 8 100
a=sendrecv
a=ptime:20
a=rtpmap:100 telephone-event/8000
m=video 28734 RTP/AVP
a=sendrecv
a=content:main
a=rtcp-fb:* nack pli
a=rtcp-fb:* ccm fir
m=application 27676 RTP/AVP
a=sendrecv

Two issues I'm seeing here:
1.  Static audio payloads are not being removed (in this case G722 and G711a).  
However, if the a= lines are provided, then they get removed as expected.
2.  The m= and a= lines for "video" and "application" are still there while the 
payload types are all deleted.

Thanks.


Cinthia



On Aug 4, 2011, at 11:41 AM, Dani Popa wrote:


Thanks,

I already did that.

Dani

On 08/04/11 18:19, Razvan Crainea wrote:

Hi Dani,

You can try by deleting the most common video codecs (like H261, H263, H264).
You can do that using the codec_delete[1] functions from the textops module.
I think you should also replace the video port with 0.

[1] http://www.opensips.org/html/docs/modules/devel/textops.html#id293910

Regards,

Razvan Crainea
OpenSIPS Developer


On 04.08.2011 18:03, Dani Popa wrote:

Hi,

In fact, i have some problems with one of my pstn gw's that send "400 Incorrect 
content length", i think, because of too long sip packet. So, because it is pstn, i 
want to remove video capability(many lines in first invite packet).

Dani

On 08/04/11 17:02, Razvan Crainea wrote:

Hi Dani,

Why would you do that? If you don't want to allow video, you can simply replace the video 
port in the "m=" line with 0.

Regards,

Razvan Crainea
OpenSIPS Developer


On 04.08.2011 16:58, Dani Popa wrote:

Hi all,

How can i remove all sip video body headers regardin video. Should i remove any line from 
body after "m=video", or how. Please give me a hint, if you have.

Thanks,
Dani




--
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
OpenSIPS solutions and "know-how"


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

[OpenSIPS-Users] drouting and free memory

2011-12-28 Thread dpa 
Hello!

 

There is such problem

 

Opensips 1.6.4-2

I am using drouting module to relay any calls.

 

modparam("drouting", "fetch_rows", 4)

 

When I am try using fifo dr_reload I see such error 

“Dec 28 16:38:24 opensips /usr/local/opensips1.6.4-2/sbin/opensips[27360]: 
ERROR:core:db_allocate_rows: no memory left

Dec 28 16:38:24 opensips /usr/local/opensips1.6.4-2/sbin/opensips[27360]: 
ERROR:db_mysql:db_mysql_fetch_result: no memory left

Dec 28 16:38:24 opensips /usr/local/opensips1.6.4-2/sbin/opensips[27360]: 
ERROR:drouting:dr_load_routing_info: Error fetching rows”

 

And I cannot see any problem with memory

“shmem:total_size = 104857600

shmem:used_size = 34574448

shmem:real_used_size = 38811912”

“pkmem:0-total_size = 5242880

pkmem:0-used_size = 591616

pkmem:0-real_used_size = 717016

pkmem:0-max_used_size = 919520

pkmem:0-free_size = 4525864

pkmem:0-fragments = 2”

Thank you for any help

 

 

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

Re: [OpenSIPS-Users] ACK never reach UAS

2011-12-28 Thread Bogdan-Andrei Iancu 

Hello,

Your opensips script does not do Record-Routing, so the sequential 
requests (in-dialog requests, like ACK) will not go through opensips, 
but rather directly between UAC and UAS, based on the Contact IPs.


Check on the UAC side, where the ACK request is send to.

Regards,
Bogdan

On 12/28/2011 01:35 AM, M.Abdulaziz wrote:

Hello All,

I'm doing the follwing scenario:

UAC (192.168.1.65:5097)->opensips
(192.168.1.65:5060)->UAS(192.168.1.65:5099)

UAS command:
sipp -sn uas -rsa 192.168.1.65:5060 -i 192.168.1.65 -p 5099

UAC command:
sipp -sn uac -s uas_user 192.168.1.65:5060 -i 192.168.1.65 -p 5097

Note: uas_usr is already registred in UAS's domain 192.168.1.65:5099

Everything goes fine but ACK requests from UAC
http://opensips-open-sip-server.1449251.n2.nabble.com/file/n7131340/03.png
03.png  never reaches UAS
http://opensips-open-sip-server.1449251.n2.nabble.com/file/n7131340/52.png
52.png  which produces dead calls(I attached a screen shot of the scenario)


Can anyone kindly help how to handle this problem ASAP?

Thank you in advance,

Here is my opensips.cfg

# --- global configuration parameters 

debug=6# debug level (cmd line: -dd)
fork=yes
log_stderror=yes# (cmd line: -E)
children=32
log_facility=LOG_LOCAL0
log_name="OPENSIPS-proxy"
disable_tcp=yes
disable_dns_blacklist=yes
disable_dns_failover=yes



listen=udp:192.168.1.65:5060

# -- module loading --
mpath="/usr/local/lib64/opensips/modules/"

loadmodule "tm.so"
loadmodule "sl.so"
loadmodule "textops.so"
loadmodule "statistics.so"
loadmodule "db_mysql.so"
loadmodule "dialog.so"
loadmodule "mi_fifo.so"
loadmodule "usrloc.so"
loadmodule "signaling.so"
loadmodule "maxfwd.so"
loadmodule "rr.so"
loadmodule "registrar.so"


modparam("tm", "wt_timer", 2)
modparam("statistics","variable","INVs")
modparam("statistics","variable","nonINVs")
modparam("usrloc", "db_mode",2)
modparam("usrloc",
"db_url","mysql://opensips:opensipsrw@localhost/opensips")
modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")


# -  request routing logic ---

# main routing logic

route{

if(is_method("INVITE"))
{
update_stat("INVs","+1");

}
else
update_stat("nonINVs","+1");


t_relay();


}


--
View this message in context: 
http://opensips-open-sip-server.1449251.n2.nabble.com/ACK-never-reach-UAS-tp7131340p7131340.html
Sent from the OpenSIPS - Users mailing list archive at Nabble.com.

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




--
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
OpenSIPS solutions and "know-how"


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

Re: [OpenSIPS-Users] Call pickup

2011-12-28 Thread Bogdan-Andrei Iancu 

Hi Dmitriy,

The docs on the MI functions (names and what they are doing) can be 
found as part of the module online docs (like 
http://www.opensips.org/html/docs/modules/1.7.x/tm.html#id294662).


MI commands can be triggered via FIFO files, via XMLRPC or via simple 
UDP package.


For same examples, check "examples/web_im/" directory.

Regards,
Bogdan

On 12/28/2011 10:46 AM, Dmitriy Abramov wrote:

Hello, Bogdan.
Thank you for answer
Where i can read more about MI and how to use it. maybe examples.

Regards,
Dmitriy Abramov.
/
/
On Dec 27, 2011, at 2:15 PM, Bogdan-Andrei Iancu wrote:


Hello Dmitriy,

A call pickup scenario is related to call setup, so it is more 
appropriate to handle it at transaction level (while b2bua is at call 
level).


On the other hand, as far I understand the call pickup scenario, you 
can try using the MI commands to cancel/terminate the current ringing 
call and from failure route to create a new branch to the new device.


Regards,
Bogdan

On 12/26/2011 09:21 AM, Dmitriy Abramov wrote:

Hi.
Anyone has example files how make call pickup with opensips?
I read about b2bua, but i can't understand how to use it for call 
pickup.

/
/
/Best regards, /
/Dmitriy Abramov/


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



--
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
OpenSIPS solutions and "know-how"
___
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



--
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
OpenSIPS solutions and "know-how"

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

Re: [OpenSIPS-Users] Users Digest, Vol 41, Issue 72

2011-12-28 Thread auto-reply from antonio.spirande...@longwave.eu 
Sarò assente fino all'8 Gennaio compreso. Per urgenze 
rivolgersi direttamente ad assiste...@longwave.eu o 
chiamare lo 0522375500. Saluti

I will be out of office untill  Gen 8th 2012.

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

[OpenSIPS-Users] B2BUA + RTPproxy + Asterisk direct media

2011-12-28 Thread Lee Archer 
Hi all, I wonder if someone can help me.  I have a system where I use the B2B 
module and RTPproxy for inbound calls but once answered the call might jump 
between Asterisk servers depending on what service is required.  I would like 
to use the Asterisk direct media option for SIP calls but when enabled the 
server is trying to talk to the SIP providers RTP gateway instead of my 
RTPproxy instance.  I've made changes to the RTPproxy configuration but I'm 
wondering if anyone else uses direct media with RTPproxy and can point me in 
the right direction config wise.

Thanks

Lee


thebigword Holdings Limited. Registered Office: Link Up House, Ring Road, Lower 
Wortley, Leeds, UK, LS12 6AB. Registered in England & Wales No. 05551907

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

Re: [OpenSIPS-Users] Call pickup

2011-12-28 Thread Dmitriy Abramov 
Hello, Bogdan.
Thank you for answer
Where i can read more about MI and how to use it. maybe examples.

Regards,
Dmitriy Abramov.

On Dec 27, 2011, at 2:15 PM, Bogdan-Andrei Iancu wrote:

> Hello Dmitriy,
> 
> A call pickup scenario is related to call setup, so it is more appropriate to 
> handle it at transaction level (while b2bua is at call level).
> 
> On the other hand, as far I understand the call pickup scenario, you can try 
> using the MI commands to cancel/terminate the current ringing call and from 
> failure route to create a new branch to the new device.
> 
> Regards,
> Bogdan
> 
> On 12/26/2011 09:21 AM, Dmitriy Abramov wrote:
>> 
>> Hi.
>> Anyone has example files how make call pickup with opensips? 
>> I read about b2bua, but i can't understand how to use it for call pickup. 
>> 
>> Best regards, 
>> Dmitriy Abramov
>> 
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 
> 
> -- 
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> OpenSIPS solutions and "know-how"
> ___
> 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