[OpenSIPS-Users] Callpickup scenario help

2013-05-21 Thread Miha 

HI to all,

i need a little help regarding one scenaria (What would be the best way 
to implement this).


I am having opensips which is heandling registration and load balancing. 
Behind opensips I am having FS servers. What would be the best way to 
implement group call pickup. I need to know to which FS server send I 
call which will do call pickup. I must intercept only ringing phones. 
Call pickup future I have implemented on FS as would be also for PBX 
futures.


thanks!
miha

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Re: [OpenSIPS-Users] memory consumed by t_relay

2013-05-21 Thread microx 
Hi Bogdan-Andrei,

When you get the memory error, could you please do : "opensipsctl fifo 
get_statistics all" and send me the output (off list) please? 

When about 10,000 calls,  10kcalls.log

 
.
When about 20,000 calls,  20kcalls.log

  

Also, at step 4) , by stop, you mean you shutdown OpenSIPS or you stop 
the sipp load but OpenSIPS still runs ? 

I meant that I shutdown OpenSIPS.

Going back to my request on memory debugging, I was wondering why your 
logs do no show the shm dump (but only pkg) - at step 3) do the SIGUSR 
stuff and be sure you look for the Memory status for shm mem in your 
logs - it must be there. 

With killing one specific OpenSIPS process, I get the log as the attachment 
https://docs.google.com/file/d/0B-NXx5YS2KQZOW11YnhNY2hGUkk/edit?usp=sharing
(I set memdump=1, memlog=1, debug=6)

If any further information is required or any provided log has something in
lack, please feel free to let me know.
Many thanks for your kind support.

Best regards,
Chen-Che 



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Re: [OpenSIPS-Users] Registrar not saving received from Path header

2013-05-21 Thread Nathaniel L Keeling III 

Hello Bogdan,

Here is the output from the opensips log. I have also attached a snippet 
from the log file.


May 21 23:39:15   OpenSips[14397]: [ID 257313 local1.debug] 
DBG:registrar:save_aux: xXx - flags param is
May 21 23:39:15   OpenSips[14397]: [ID 154992 local1.debug] 
DBG:registrar:save_aux: xXx - flags bitmask is <0>


May 21 23:39:15   OpenSips[14397]: [ID 269964 local1.debug] 
DBG:registrar:pack_ci: xXx - flags are 0


Thanks

Nathaniel L Keeling

On 5/20/13 11:56 AM, Bogdan-Andrei Iancu wrote:

Hello Nathaniel,

See the attached patch - it logs more from the part where the params 
are handled .


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

On 05/18/2013 09:33 AM, Nathaniel L Keeling III wrote:

Hello Bogdan,

Here are snippets from my script. I only have one place where I 
execute the save function. Just wondering, could it be truncating a byte?


modparam("usrloc", "nat_bflag", 10)
modparam("usrloc", "use_domain", 1)
modparam("usrloc", "db_mode", 3)
modparam("usrloc", "db_url",
"postgres://opensips:opensip...@ama.akan.net/opensips181t")
modparam("registrar", "tcp_persistent_flag", 7)
modparam("registrar", "received_avp", "$avp(received_nh)")


xlog("SAVING THE SUBSCRIBER INTO THE LOCATION TABLE 
");

if (!save("location","p1"))
{
xlog("L_ERR", "ERR:callerid:$ci|end|System error trying to 
save Register's request location");

sl_reply_error();
}

xlog("L_NOTICE", "NOTICE:callerid:$ci|end|The subscriber has 
successfully registered with Akan Voice");

exit;

Thanks

Nathaniel

On 5/17/13 6:07 AM, Bogdan-Andrei Iancu wrote:

Hello Nathaniel,

That is odd.it's like you do not set the "p1" flag 

I tested and I with "p1" flag I get:
May 17 14:05:03 [7944] DBG:registrar:pack_ci: xXx - flags are 10

Are you sure your script gets to the right save() ??

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

On 05/17/2013 09:37 AM, Nathaniel L Keeling III wrote:

Hello Bogdan,

I added the patch and here is what I found: "OpenSips[4378]: [ID 
269964 local1.debug] DBG:registrar:pack_ci: xXx - flags are 0". I 
have also included the log file.


Thanks

Nathaniel Keeling

On 5/16/13 3:47 AM, Bogdan-Andrei Iancu wrote:

Hello Nathaniel,

Attached is an extended patch - remove the old one and apply this 
one. Again look for any xXx logs .


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

On 05/14/2013 02:47 PM, Nathaniel L Keeling III wrote:

Hello Bogdan,

here is the output from opensips's og file of the save() with the 
patch and the code snippet from the opensips.cfg. I did not see 
any ant logs with "xXx". Also,I have usrloc's db_mode set to 3.


xlog("SAVING THE SUBSCRIBER INTO THE LOCATION TABLE 
");

if (!save("location","p1"))
{
xlog("L_ERR", "ERR:callerid:$ci|end|System error trying 
to save Register's request location");

sl_reply_error();
}
xlog("L_NOTICE", "NOTICE:callerid:$ci|end|The subscriber has 
successfully registered with Akan Voice");
exit; 



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May 21 23:39:15   OpenSips[14397]: [ID 197553 local1.error] SAVING THE 
SUBSCRIBER INTO THE LOCATION TABLE 

May 21 23:39:15   OpenSips[14397]: [ID 257313 local1.debug] 
DBG:registrar:save_aux: xXx - flags param is  
May 21 23:39:15   OpenSips[14397]: [ID 154992 local1.debug] 
DBG:registrar:save_aux: xXx - flags bitmask is <0> 
May 21 23:39:15   OpenSips[14397]: [ID 497291 local1.debug] 
DBG:core:parse_headers: flags=
May 21 23:39:15   OpenSips[14397]: [ID 421386 local1.debug] DBG:core:parse_uri: 
parsed uri:
May 21 23:39:15type=1 user=(8)
May 21 23:39:15passwd=<>(0)
May 21 23:39:15host=(13)
May 21 23:39:15port=<>(0): 0
May 21 23:39:15params=<>(0)
May 21 23:39:15headers=<>(0)
May 21 23:39:15   OpenSips[14397]: [ID 883594 local1.debug] DBG:core:parse_uri: 
 uri params:
May 21 23:39:15  transport=<>, val=<>, proto=0
May 21 23:39:15   OpenSips[14397]: [ID 996803 local1.debug] DBG:core:parse_uri: 
   user-param=<>, val=<>
May 21 23:39:15   OpenSips[14397]: [ID 784403 local1.debug] DBG:core:parse_uri: 
   method=<>, val=<>
May 21 23:39:15   OpenSips[14397]: [ID 185236 local1.debug] DBG:core:parse_uri: 
   ttl=<>, val=<>
May 21 23:39:15   OpenSips[14397]: [ID 628781 local1.debug] DBG

[OpenSIPS-Users] B2BUA Segfault

2013-05-21 Thread Tolga Tarhan 
Hello,

While using the B2BUA module in OpenSIPS 1.9.0, we've encountered a
consistent segfault. We are using a refer scenario just like the one in the
B2BUA sample docs, and after several REFERs for the same call (to different
destinations), OpenSIPS crashes with a segfault. The core file indicates
the following backtrace:

#0  0x0049a334 in fm_malloc ()
#1  0x7fdaecd96230 in shm_malloc_unsafe (type=B2B_CLIENT,
entity_id=0x7fdaee8ec750, to_uri=0x7fff2d346360, from_uri=0x7fff2d346320,
from_dname=0x0, ssid=, msg=0x0) at
../../mem/shm_mem.h:248
#2  shm_malloc (type=B2B_CLIENT, entity_id=0x7fdaee8ec750,
to_uri=0x7fff2d346360, from_uri=0x7fff2d346320, from_dname=0x0, ssid=, msg=0x0) at ../../mem/shm_mem.h:258
#3  b2bl_create_new_entity (type=B2B_CLIENT, entity_id=0x7fdaee8ec750,
to_uri=0x7fff2d346360, from_uri=0x7fff2d346320, from_dname=0x0, ssid=, msg=0x0) at logic.c:293
#4  0x7fdaecd96882 in b2bl_new_client (to_uri=,
from_uri=, tuple=,
ssid=0x7fdaeb026c00, msg=) at logic.c:607
#5  0x7fdaecda3579 in process_bridge_200OK (msg=0x7fdaee8e8b30,
extra_headers=0x7fdaeb03d578, body=,
tuple=0x7fdaeb01ada8, entity=) at logic.c:816
#6  0x7fdaecda46c2 in b2b_logic_notify_reply (src=, msg=0x7fdaee8e8b30, key=, body=0x7fff2d3468b0,
extra_headers=0x7fff2d3468a0, b2bl_key=0x7fff2d3476d0, hash_index=649,
local_index=0)
at logic.c:1133
#7  0x7fdaecda6081 in b2b_logic_notify (src=1, msg=0x7fdaee8e8b30,
key=0x7fdaeb03d500, type=1, param=0x7fff2d3476d0) at logic.c:2040
#8  0x7fdaecfca7ad in b2b_tm_cback (t=0x7fdaeb054118,
htable=0x7fdaeb014630, ps=) at dlg.c:2678
#9  0x7fdaee291441 in run_trans_callbacks (type=256,
trans=0x7fdaeb054118, req=, rpl=,
code=) at t_hooks.c:212
#10 0x7fdaee29c0e2 in local_reply (t=0x7fdaeb054118, p_msg=, branch=, msg_status=, cancel_bitmap=) at t_reply.c:1391
#11 0x7fdaee29d31d in reply_received (p_msg=0x7fdaee8e8b30) at
t_reply.c:1540
#12 0x0042625a in forward_reply ()
#13 0x00451c28 in receive_msg ()
#14 0x00494e45 in udp_rcv_loop ()
#15 0x0042d1a3 in main ()

I'm not really sure how to diagnose this one. Any hints/fixes/suggestions
would be very appreciated.

Thanks,
Tolga
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Re: [OpenSIPS-Users] Do_Routing problem

2013-05-21 Thread M.Khaled W Chehab 
Hi,

 

I set in dr_rules #lebanon   as a carrier route to terminate the  rule
that match 777961 ,

Please find below the output for your script 

 

loadmodule "drouting.so"

modparam("drouting", "db_url", "virtual://set1") # CUSTOMIZE ME

modparam("drouting", "use_domain", 1)

modparam("drouting", "drd_table", "dr_gateways")

modparam("drouting", "drr_table", "dr_rules")

modparam("drouting", "drg_table", "dr_groups")

modparam("drouting", "drc_table", "dr_carriers")

modparam("drouting", "drg_user_col", "username")

modparam("drouting", "drg_domain_col", "domain")

modparam("drouting", "drg_grpid_col", "groupid")

modparam("drouting", "force_dns", 1)

#frequency of probing per paramete in seconds

modparam("drouting", "probing_interval", 120)

modparam("drouting", "probing_method", "OPTIONS")

modparam("drouting", "probing_from", "sip:pinger@opensips")

modparam("drouting", "probing_reply_codes", "487")

#501, 403,404,

modparam("drouting", "use_domain", 1)

modparam("drouting", "gw_attrs_avp", '$avp(gw_attrs)')

modparam("drouting", "gw_id_avp", '$avp(gw_id)')

modparam("drouting", "rule_attrs_avp", '$avp(rule_attrs)')

 

modparam("drouting","ruri_avp", "$avp(dr_ruri)")

modparam("drouting","rule_id_avp", "$avp(dr_rule)")

modparam("drouting","rule_attrs_avp", "$avp(dr_rule_attrs)")

modparam("drouting","rule_prefix_avp", "$avp(dr_rule_prefix)")

modparam("drouting","carrier_id_avp", "$avp(dr_cr)")

modparam("drouting","carrier_attrs_avp", "$avp(dr_cr_attrs)")

 

 

 

GW IDs are  :  

GW ATTRs are:  

CR IDs are  : Lebanon, Lebanon, Lebanon, Lebanon,  

CR ATTRs are: , , , ,  

RULE: id=104, , attrs=0, , prefix=777961,  

 

 


 
 Description:
Full Texts

 
 id

 
 carrierid

 
 gwlist

 
 flags

 
 attrs

 
 description


 
 Description: Edit

 
 Description: Delete

14

Lebanon

Dahdah_JVS,Bics,IDT,Snell_93

0

 

 



 

And the problem still exists, please advice

Regards

 

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Tuesday, May 21, 2013 8:00 PM
To: OpenSIPS users mailling list
Cc: M.Khaled W Chehab
Subject: Re: [OpenSIPS-Users] Do_Routing problem

 

Hello,

You can check what do_routing() set (as GWs to be used) by adding this in
your script (just after do_routing):

xlog("Destinations are: $ru $(avp(dr_ruri)[*]) \n");
xlog("GW IDs are  : $(avp(dr_gw)[*]) \n");
xlog("GW ATTRs are: $(avp(dr_gw_attrs)[*]) \n");
xlog("CR IDs are  : $(avp(dr_cr)[*]) \n");
xlog("CR ATTRs are: $(avp(dr_cr_attrs)[*]) \n");
xlog("RULE: id=$(avp(dr_rule)[*]), attrs=$(avp(dr_rule_attrs)[*]),
prefix=$(avp(dr_rule_prefix)[*]) \n");

Be sure you have these params set:

modparam("drouting","ruri_avp", "$avp(dr_ruri)")
modparam("drouting","gw_id_avp", "$avp(dr_g

Re: [OpenSIPS-Users] Slight problem routing 100s and 183s

2013-05-21 Thread Nick Khamis 
Bogdan I am so sorry!!! 192.168.2.11 is actually a UAC polycom phone.
The only asterisk box that is being used in the scenario right now is
192.168.2.10, as seen in the traces. Please forgive me! :)

N.

On 5/21/13, Bogdan-Andrei Iancu  wrote:
> Hello Nick,
>
> To be honest, I'm a bit confused - looking at the trace, I see the
> INVITE comes from .11 (an aterisks), goes to .5 (opensipsIn) and then to
> .10 (another asterisk)This does not match the network diagram ..
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
>
> On 05/17/2013 11:30 PM, Nick Khamis wrote:
>> Bogdan,
>>
>> I see how busy you are with OpenSIPS so I will make it count.
>> Yes OpenSIP-Out is the new box that we have put in place to:
>>
>> Bellow is a quick network diagram. The issue we are experiencing is
>> that the 100s, 183s and 200s
>> that come back from the carrier do not get processed or even responded
>> to by OpenSIPS-In.
>> The complete sip trace for OpenSIPS-In can be found at
>> "http://pastebin.com/iGeWsc40";.
>> I did not include anything for "OUT" since it is performing as expected.
>>
>> Some things to notice are the changed CallID. This is done by asterisk
>> (192.168.2.10):
>>
>> Initial: Call-ID: 4737d441-5fb15ea7-7142c0d8@192.168.2.11
>> .
>> Modified: Call-ID: 1fbe6fb90553da7c52d72b60076030f5@192.168.2.10:5060
>> .
>>
>> And the vanishing of RR: Record-Route:
>> .
>> This is also due to asterisk's recreation of the initial INVITE.
>>
>> When it comes to network appliances, this is the last piece of the pie.
>> From now on it's mainly business logic, which should be less of a
>> learning
>> curve for us!!!
>>
>> I decided to post my problem online with example values, so it would
>> hopefully help someone
>> in the future.
>>
>> Kind Regards,
>>
>> Nick.
>>
>> network.jpg
>> 
>>
>>
>>
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>

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Re: [OpenSIPS-Users] Slight problem routing 100s and 183s

2013-05-21 Thread Bogdan-Andrei Iancu 
Hello Nick,

To be honest, I'm a bit confused - looking at the trace, I see the
INVITE comes from .11 (an aterisks), goes to .5 (opensipsIn) and then to
.10 (another asterisk)This does not match the network diagram ..

Regards,

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


On 05/17/2013 11:30 PM, Nick Khamis wrote:
> Bogdan,
>
> I see how busy you are with OpenSIPS so I will make it count.
> Yes OpenSIP-Out is the new box that we have put in place to:
>
> Bellow is a quick network diagram. The issue we are experiencing is
> that the 100s, 183s and 200s
> that come back from the carrier do not get processed or even responded
> to by OpenSIPS-In.
> The complete sip trace for OpenSIPS-In can be found at
> "http://pastebin.com/iGeWsc40";.
> I did not include anything for "OUT" since it is performing as expected.
>
> Some things to notice are the changed CallID. This is done by asterisk
> (192.168.2.10):
>
> Initial: Call-ID: 4737d441-5fb15ea7-7142c0d8@192.168.2.11
> .
> Modified: Call-ID: 1fbe6fb90553da7c52d72b60076030f5@192.168.2.10:5060
> .
>
> And the vanishing of RR: Record-Route:
> .
> This is also due to asterisk's recreation of the initial INVITE.
>
> When it comes to network appliances, this is the last piece of the pie.
> From now on it's mainly business logic, which should be less of a learning
> curve for us!!!
>
> I decided to post my problem online with example values, so it would
> hopefully help someone
> in the future.
>
> Kind Regards,
>
> Nick.
>
> network.jpg
> 
>
>
>
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Re: [OpenSIPS-Users] firewall configuration

2013-05-21 Thread Bogdan-Andrei Iancu 
AFAIR, the default range starts from 20 000 or 30 000 ..

Regards,

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


On 05/20/2013 09:47 AM, Muhammad Shahzad wrote:
> The default port range media proxy uses is UDP 5 - 6. You can
> modify it in its config.ini file. So this port range must be open in
> firewall.
>
> Thank you.
>
>
>
> On Sun, May 19, 2013 at 8:35 PM, Nicholas Papadakos  > wrote:
>
>
> Hello,
>
> I have the following setup :
>
> Internet --->BSD firewall>   Opensips
> 78.xx.xx.xx 78.xx.xx.xx
>
>
>
> Both the firewall and the opensips machine have live internet ips.
>
> How should I configure opensips with mediaproxy ?
> The firewall is blocking all ports except the ones I explicitly allow.
>
> Kind Regards,
>
> Nicholas Papadakos
>
>
>
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>
>
>
> -- 
> Mit freundlichen Grüßen
> Muhammad Shahzad
> ---
> CISCO Rich Media Communication Specialist (CRMCS)
> CISCO Certified Network Associate (CCNA)
> Cell: +49 176 99 83 10 85
> MSN: shari_78...@hotmail.com 
> Email: shaherya...@googlemail.com 
>
>
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Re: [OpenSIPS-Users] VIA relay error using mhomed=1

2013-05-21 Thread Bogdan-Andrei Iancu 
Hi Qasim,

Looking at the ACK related logs, I see you get the script log
 Sequencial 'ACK' request from caller '622190004001' for call from .

twice - also the logs from the loose_route() function - I suspect you
loop somehow in your script and a route is triggered twice (the route
doing loose_route)

Regards,

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


On 05/20/2013 02:46 PM, qasimak...@gmail.com wrote:
> Hi Bodgan,
>
> Sorry for the late reply as i was traveling this weekend. Please find
> attached call logs with debug mode 4.
>
> Regards,
> Qasim
>
>
> On Fri, May 17, 2013 at 8:50 PM, Bogdan-Andrei Iancu
> mailto:bog...@opensips.org>> wrote:
>
> Funny, as I do not see anything wrong on a first look - while
> running in debug mode (4), please send me the logs corresponding
> to the ACK processing.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
>
> On 05/17/2013 02:34 PM, qasimak...@gmail.com
>  wrote:
>> Hi,
>>
>> Please find attached trace. This is server on Public IP that is
>> why i cannot send the trace on the list. I am listening to IP's
>> as follows
>>
>> listen=udp:202.152.203.195:5060 
>> listen=udp:202.152.203.195:6000 
>> listen=udp:192.168.226.142:5060 
>> listen=udp:192.168.226.142:6000 
>>
>> disable_tcp=no
>> listen=tcp:202.152.203.195:5060 
>> listen=tcp:202.152.203.195:6000 
>> listen=tcp:192.168.226.142:5060 
>> listen=tcp:192.168.226.142:6000 
>>
>> If you need anything else i would be happy to provide it to you.
>>
>> Regards,
>> Qasim
>>
>>
>>
>> On Fri, May 17, 2013 at 3:50 PM, Bogdan-Andrei Iancu
>> mailto:bog...@opensips.org>> wrote:
>>
>> Hello Qasim,
>>
>> So you have multiple interfaces in OpenSIPS - are all of them
>> the same protocol ?
>>
>> Please try to post a SIP capture of the full call, to see how
>> the RR part is done.
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developer
>> http://www.opensips-solutions.com
>>
>>
>> On 05/16/2013 01:07 PM, qasimak...@gmail.com
>>  wrote:
>>> On further investigation i see that i only face this issue
>>> when both caller and callee are on the same network. If both
>>> are on separate network it works fine.
>>>
>>> Regards,
>>> Qasim
>>>
>>>
>>> On Thu, May 16, 2013 at 3:05 PM, qasimak...@gmail.com
>>>  >> > wrote:
>>>
>>> yes.
>>>
>>> Regards,
>>> Qasim
>>>
>>>
>>> On Thu, May 16, 2013 at 2:50 PM, Bogdan-Andrei Iancu
>>> mailto:bog...@opensips.org>> wrote:
>>>
>>> And do you have UDP 202.152.203.195 port 6000 as
>>> listener defined in OpenSIPS ??
>>>
>>> Regards,
>>>
>>> Bogdan-Andrei Iancu
>>> OpenSIPS Founder and Developer
>>> http://www.opensips-solutions.com
>>>
>>>
>>> On 05/16/2013 12:32 PM, qasimak...@gmail.com
>>>  wrote:
 Hi Bodgan,

 Yes i see the following route header in my packet.

 Route:
 
 


 And yes i am routing it through loose_route.

 Regards,
 Qasim


 On Wed, May 15, 2013 at 10:40 PM, Bogdan-Andrei
 Iancu >>> > wrote:

 Hello Qasim,

 The ACK should be routed via loose_route()
 based on the "Route" headers from it. Could you
 check if the Route hdrs (from the ACK) are
 correctly reflecting your opensips interfaces ?

 Best regards,

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


 On 05/14/2013 07:55 AM, qasimak...@gmail.com
  wrote:
> Hi,
>
> I am using OpenSIPs in Public<->Private
> bridging mode and have enabled mhomed=1. But
> the problem is that when we have a call

Re: [OpenSIPS-Users] Do_Routing problem

2013-05-21 Thread Bogdan-Andrei Iancu 
Hello,

You can check what do_routing() set (as GWs to be used) by adding this
in your script (just after do_routing):

xlog("Destinations are: $ru $(avp(dr_ruri)[*]) \n");
xlog("GW IDs are  : $(avp(dr_gw)[*]) \n");
xlog("GW ATTRs are: $(avp(dr_gw_attrs)[*]) \n");
xlog("CR IDs are  : $(avp(dr_cr)[*]) \n");
xlog("CR ATTRs are: $(avp(dr_cr_attrs)[*]) \n");
xlog("RULE: id=$(avp(dr_rule)[*]),
attrs=$(avp(dr_rule_attrs)[*]), prefix=$(avp(dr_rule_prefix)[*]) \n");

Be sure you have these params set:

modparam("drouting","ruri_avp", "$avp(dr_ruri)")
modparam("drouting","gw_id_avp", "$avp(dr_gw)")
modparam("drouting","gw_attrs_avp", "$avp(dr_gw_attrs)")
modparam("drouting","rule_id_avp", "$avp(dr_rule)")
modparam("drouting","rule_attrs_avp", "$avp(dr_rule_attrs)")
modparam("drouting","rule_prefix_avp", "$avp(dr_rule_prefix)")
modparam("drouting","carrier_id_avp", "$avp(dr_cr)")
modparam("drouting","carrier_attrs_avp", "$avp(dr_cr_attrs)")

Regards,

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


On 05/21/2013 06:17 PM, M.Khaled W Chehab wrote:
>
> Hi,
>
>  
>
> I am using opensips 1.8.3
>
> I set three gateways  in carrier route ,  the call always go to  the
> 1st and jump to  third gateway  and ignores the second
>
>  
>
> Moreover  I can see in debug level 4 ,
>
> DBG:drouting:push_gw_for_usage: adding gw [gw1] in order 0 adding gw
> [gw2]  in order 1 adding gw [gw3] in order 3
>
>  
>
> But using wireshark I can just see the invite to the first and the third
>
>  
>
> Please advice
>
>  
>
>
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Re: [OpenSIPS-Users] memory consumed by t_relay

2013-05-21 Thread Bogdan-Andrei Iancu 
Hello ,

When you get the memory error, could you please do : "opensipsctl fifo
get_statistics all" and send me the output (off list) please?

Also, at step 4) , by stop, you mean you shutdown OpenSIPS or you stop
the sipp load but OpenSIPS still runs ?

Going back to my request on memory debugging, I was wondering why your
logs do no show the shm dump (but only pkg) - at step 3) do the SIGUSR
stuff and be sure you look for the Memory status for shm mem in your
logs - it must be there.

Regards,

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


On 05/21/2013 09:07 AM, microx wrote:
> Hi Bogdan-Andrei, 
>
> With SIPp keeps creating and terminating dialogs round after round, the only
> things at the outbound proxy I know are
> 1) each SIP process occupies more and more memory.
>
> 2) "opensipsctl fifo get_statistics shmem:" shows the shmem:used_size and
> shmem:real_used_size both keep increasing (but tm:inuse_transactions is
> nearly constant)
>
> 3) After processing 60,000 dialogs (call per second: 100; within less than
> an hour), 
> "WARNING:core:fm_malloc: Not enough free memory, will atempt defragmenation 
> ERROR:tm:new_t: out of mem 
> ERROR:tm:t_newtran: new_t failed" are output to the log and no new INVITEs
> could be forwarded from outbound proxy to internal SIP server.
>
> 4) if OpenSIPS at the outbound proxy stops, the occupied memory seems to be
> freed normally.
>
> It is possible that there exists a flaw in my configuration file. However, I
> am not capable of identifying the problem. 
> Is there anything I can provide to help address this issue?
>
> Best regards,
> Chen-Che
>
>
>
> --
> View this message in context: 
> http://opensips-open-sip-server.1449251.n2.nabble.com/memory-consumed-by-t-relay-tp7586016p7586430.html
> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>
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> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>

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[OpenSIPS-Users] Do_Routing problem

2013-05-21 Thread M.Khaled W Chehab 
Hi,

 

I am using opensips 1.8.3

I set three gateways  in carrier route ,  the call always go to  the 1st and
jump to  third gateway  and ignores the second

 

Moreover  I can see in debug level 4 , 

DBG:drouting:push_gw_for_usage: adding gw [gw1] in order 0 adding gw [gw2]
in order 1 adding gw [gw3] in order 3

 

But using wireshark I can just see the invite to the first and the third 

 

Please advice

 

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