Re: [OpenSIPS-Users] memory consumed by t_relay

2013-05-21 Thread microx
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



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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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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



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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] 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
 bog...@opensips.org 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
 mailto: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 http://202.152.203.195:5060
 listen=udp:202.152.203.195:6000 http://202.152.203.195:6000
 listen=udp:192.168.226.142:5060 http://192.168.226.142:5060
 listen=udp:192.168.226.142:6000 http://192.168.226.142:6000

 disable_tcp=no
 listen=tcp:202.152.203.195:5060 http://202.152.203.195:5060
 listen=tcp:202.152.203.195:6000 http://202.152.203.195:6000
 listen=tcp:192.168.226.142:5060 http://192.168.226.142:5060
 listen=tcp:192.168.226.142:6000 http://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
 bog...@opensips.org 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
 mailto: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
 mailto:qasimak...@gmail.com qasimak...@gmail.com
 mailto:qasimak...@gmail.com wrote:

 yes.

 Regards,
 Qasim


 On Thu, May 16, 2013 at 2:50 PM, Bogdan-Andrei Iancu
 bog...@opensips.org 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
 mailto:qasimak...@gmail.com wrote:
 Hi Bodgan,

 Yes i see the following route header in my packet.

 Route:
 
 sip:622190004002@202.152.203.195:6000;lr;ftag=3b710c25;did=e55.a77ff685


 And yes i am routing it through loose_route.

 Regards,
 Qasim


 On Wed, May 15, 2013 at 10:40 PM, Bogdan-Andrei
 Iancu bog...@opensips.org
 mailto:bog...@opensips.org 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
 mailto: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 in
 which both parties are on Public interface the
 INVITE gets relayed properly but and ACK of
 that invite gives the following error.

 

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 pa...@umbrela.org
 mailto:pa...@umbrela.org 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 mailto:shari_78...@hotmail.com
 Email: shaherya...@googlemail.com mailto:shaherya...@googlemail.com


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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
 mailto:4737d441-5fb15ea7-7142c0d8@192.168.2.11.
 Modified: Call-ID: 1fbe6fb90553da7c52d72b60076030f5@192.168.2.10:5060
 http://1fbe6fb90553da7c52d72b60076030f5@192.168.2.10:5060/.

 And the vanishing of RR: Record-Route:
 sip:192.168.2.5;lr;did=b82.180aabc6.
 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
 https://mail.google.com/mail/ca/?ui=2ik=e9f48992abview=attth=13eb42dafefa444eattid=0.1disp=inlinerealattid=f_hgttk2a11safe=1zw



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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 bog...@opensips.org 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
 mailto:4737d441-5fb15ea7-7142c0d8@192.168.2.11.
 Modified: Call-ID: 1fbe6fb90553da7c52d72b60076030f5@192.168.2.10:5060
 http://1fbe6fb90553da7c52d72b60076030f5@192.168.2.10:5060/.

 And the vanishing of RR: Record-Route:
 sip:192.168.2.5;lr;did=b82.180aabc6.
 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
 https://mail.google.com/mail/ca/?ui=2ik=e9f48992abview=attth=13eb42dafefa444eattid=0.1disp=inlinerealattid=f_hgttk2a11safe=1zw



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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  : null 

GW ATTRs are: null 

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

CR ATTRs are: , , , ,  

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

 

 


 
http://5.104.224.33/phpmyadmin/sql.php?db=opensipstable=dr_carrierssql_qu
ery=SELECT+%2A+FROM+%60dr_carriers%60goto=tbl_structure.phpdisplay_options
_form=1display_text=Ftoken=4c290d9fc747fadbbb216f2d571c00ef Description:
Full Texts

 
http://5.104.224.33/phpmyadmin/sql.php?db=opensipstable=dr_carrierssql_qu
ery=SELECT+%2A+FROM+%60dr_carriers%60+ORDER+BY+%60dr_carriers%60.%60id%60+AS
Ctoken=4c290d9fc747fadbbb216f2d571c00ef id

 
http://5.104.224.33/phpmyadmin/sql.php?db=opensipstable=dr_carrierssql_qu
ery=SELECT+%2A+FROM+%60dr_carriers%60+ORDER+BY+%60dr_carriers%60.%60carrieri
d%60+ASCtoken=4c290d9fc747fadbbb216f2d571c00ef carrierid

 
http://5.104.224.33/phpmyadmin/sql.php?db=opensipstable=dr_carrierssql_qu
ery=SELECT+%2A+FROM+%60dr_carriers%60+ORDER+BY+%60dr_carriers%60.%60gwlist%6
0+ASCtoken=4c290d9fc747fadbbb216f2d571c00ef gwlist

 
http://5.104.224.33/phpmyadmin/sql.php?db=opensipstable=dr_carrierssql_qu
ery=SELECT+%2A+FROM+%60dr_carriers%60+ORDER+BY+%60dr_carriers%60.%60flags%60
+ASCtoken=4c290d9fc747fadbbb216f2d571c00ef flags

 
http://5.104.224.33/phpmyadmin/sql.php?db=opensipstable=dr_carrierssql_qu
ery=SELECT+%2A+FROM+%60dr_carriers%60+ORDER+BY+%60dr_carriers%60.%60attrs%60
+ASCtoken=4c290d9fc747fadbbb216f2d571c00ef attrs

 
http://5.104.224.33/phpmyadmin/sql.php?db=opensipstable=dr_carrierssql_qu
ery=SELECT+%2A+FROM+%60dr_carriers%60+ORDER+BY+%60dr_carriers%60.%60descript
ion%60+ASCtoken=4c290d9fc747fadbbb216f2d571c00ef description


 
http://5.104.224.33/phpmyadmin/tbl_change.php?db=opensipstable=dr_carriers
where_clause=%60dr_carriers%60.%60id%60+%3D+14clause_is_unique=1sql_query
=SELECT+%2A+FROM+%60dr_carriers%60goto=sql.phptoken=4c290d9fc747fadbbb216f
2d571c00ef Description: Edit

 
http://5.104.224.33/phpmyadmin/sql.php?db=opensipstable=dr_carrierssql_qu
ery=DELETE+FROM+%60opensips%60.%60dr_carriers%60+WHERE+%60dr_carriers%60.%60
id%60+%3D+14zero_rows=The+row+has+been+deletedgoto=sql.php%3Fdb%3Dopensips
%26table%3Ddr_carriers%26sql_query%3DSELECT%2B%252A%2BFROM%2B%2560dr_carrier
s%2560%26zero_rows%3DThe%2Brow%2Bhas%2Bbeen%2Bdeleted%26goto%3Dtbl_structure
.php%26token%3D4c290d9fc747fadbbb216f2d571c00eftoken=4c290d9fc747fadbbb216f
2d571c00ef 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_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, 

[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=value optimized out, 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=value
optimized out, 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=value
optimized out, msg=0x0) at logic.c:293
#4  0x7fdaecd96882 in b2bl_new_client (to_uri=value optimized out,
from_uri=value optimized out, tuple=value optimized out,
ssid=0x7fdaeb026c00, msg=value optimized out) at logic.c:607
#5  0x7fdaecda3579 in process_bridge_200OK (msg=0x7fdaee8e8b30,
extra_headers=0x7fdaeb03d578, body=value optimized out,
tuple=0x7fdaeb01ada8, entity=value optimized out) at logic.c:816
#6  0x7fdaecda46c2 in b2b_logic_notify_reply (src=value optimized
out, msg=0x7fdaee8e8b30, key=value optimized out, 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=value optimized out) at dlg.c:2678
#9  0x7fdaee291441 in run_trans_callbacks (type=256,
trans=0x7fdaeb054118, req=value optimized out, rpl=value optimized out,
code=value optimized out) at t_hooks.c:212
#10 0x7fdaee29c0e2 in local_reply (t=0x7fdaeb054118, p_msg=value
optimized out, branch=value optimized out, msg_status=value optimized
out, cancel_bitmap=value optimized out) 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] 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=nkeeling(8)
May 21 23:39:15passwd=(0)
May 21 23:39:15host=akanvoice.com(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:core:parse_uri: 
   maddr=, val=
May 21 23:39:15   OpenSips[14397]: