Re: [OpenSIPS-Users] Opensips 1.11 permission module problem

2015-11-18 Thread dpa
2314

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Wednesday, November 18, 2015 1:17 PM
To: dpa; 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] Opensips 1.11 permission module problem

 

How many records do you have in the address table ?

Regards,



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

On 18.11.2015 12:14, dpa wrote:

Hello Bogdan,

 

Yes, I see errors

“Oct 28 15:55:45 opensips-main /usr/local/opensips1.11/sbin/opensips[7715]: 
WARNING:core:fm_malloc: Not enough free memory, will attempt defragmentation

Oct 28 15:55:45 opensips-main /usr/local/opensips1.11/sbin/opensips[7715]: 
ERROR:core:db_allocate_rows: no memory left

Oct 28 15:55:45 opensips-main /usr/local/opensips1.11/sbin/opensips[7715]: 
ERROR:db_mysql:db_mysql_convert_rows: no private memory left

Oct 28 15:55:45 opensips-main /usr/local/opensips1.11/sbin/opensips[7715]: 
ERROR:db_mysql:db_mysql_convert_result: error while converting rows

Oct 28 15:55:45 opensips-main /usr/local/opensips1.11/sbin/opensips[7715]: 
ERROR:db_mysql:db_mysql_store_result: error while converting result

Oct 28 15:55:45 opensips-main /usr/local/opensips1.11/sbin/opensips[7715]: 
ERROR:core:db_do_query: error while storing result for query [select 
ip,grp,mask,port,proto,pattern,context_info,id from ast_address ]

Oct 28 15:55:45 opensips-main /usr/local/opensips1.11/sbin/opensips[7715]: 
ERROR:permissions:reload_address_table: failed to query database”

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Wednesday, November 18, 2015 1:04 PM
To: OpenSIPS users mailling list; denis7...@mail.ru
Subject: Re: [OpenSIPS-Users] Opensips 1.11 permission module problem

 

Hi Denis,

Do you see any errors in logs when the reload fails ?

Regards,




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

On 18.11.2015 07:29, dpa wrote:

Hello Richard

 

It is about 1 – 1,5 week.

No, I do not use subnets, only IP address. 

 

“There was a package memory leak that was fixed in the development tree”

Where can I read about it? Thank you.

 

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Richard Revels
Sent: Tuesday, November 17, 2015 8:55 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Opensips 1.11 permission module problem

 

Are you using subnets in your address table?  How often do you reload the 
address table?

 

There was a package memory leak that was fixed in the development tree that 
might explain what you are seeing.

 

On Wed, Nov 11, 2015 at 4:26 AM, dpa <denis7...@mail.ru> wrote:



Hello!

 

Is there any assumption about problem?

 

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of dpa
Sent: Wednesday, October 28, 2015 4:12 PM
To: 'OpenSIPS users mailling list'
Subject: [OpenSIPS-Users] Opensips 1.11 permission module problem

 

Hello!

 

OpenSIPS (1.11.5-notls (x86_64/linux))

 

I have a periodic problem with permissions module.

I could not make opensipsctl address reload (“400 Trusted table reload failed” 
received). In the time I have no problem with dialplan or drouting modules.

Opensips reload solve the problem.

 

 

In attachment log from opensips and statistic.

 

Thank you for any help

 

 

 


___
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

 

 

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Re: [OpenSIPS-Users] Opensips 1.11 permission module problem

2015-11-18 Thread dpa
Hello Bogdan,

 

Yes, I see errors

“Oct 28 15:55:45 opensips-main /usr/local/opensips1.11/sbin/opensips[7715]: 
WARNING:core:fm_malloc: Not enough free memory, will attempt defragmentation

Oct 28 15:55:45 opensips-main /usr/local/opensips1.11/sbin/opensips[7715]: 
ERROR:core:db_allocate_rows: no memory left

Oct 28 15:55:45 opensips-main /usr/local/opensips1.11/sbin/opensips[7715]: 
ERROR:db_mysql:db_mysql_convert_rows: no private memory left

Oct 28 15:55:45 opensips-main /usr/local/opensips1.11/sbin/opensips[7715]: 
ERROR:db_mysql:db_mysql_convert_result: error while converting rows

Oct 28 15:55:45 opensips-main /usr/local/opensips1.11/sbin/opensips[7715]: 
ERROR:db_mysql:db_mysql_store_result: error while converting result

Oct 28 15:55:45 opensips-main /usr/local/opensips1.11/sbin/opensips[7715]: 
ERROR:core:db_do_query: error while storing result for query [select 
ip,grp,mask,port,proto,pattern,context_info,id from ast_address ]

Oct 28 15:55:45 opensips-main /usr/local/opensips1.11/sbin/opensips[7715]: 
ERROR:permissions:reload_address_table: failed to query database”

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Wednesday, November 18, 2015 1:04 PM
To: OpenSIPS users mailling list; denis7...@mail.ru
Subject: Re: [OpenSIPS-Users] Opensips 1.11 permission module problem

 

Hi Denis,

Do you see any errors in logs when the reload fails ?

Regards,



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

On 18.11.2015 07:29, dpa wrote:

Hello Richard

 

It is about 1 – 1,5 week.

No, I do not use subnets, only IP address. 

 

“There was a package memory leak that was fixed in the development tree”

Where can I read about it? Thank you.

 

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Richard Revels
Sent: Tuesday, November 17, 2015 8:55 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Opensips 1.11 permission module problem

 

Are you using subnets in your address table?  How often do you reload the 
address table?

 

There was a package memory leak that was fixed in the development tree that 
might explain what you are seeing.

 

On Wed, Nov 11, 2015 at 4:26 AM, dpa <denis7...@mail.ru> wrote:



Hello!

 

Is there any assumption about problem?

 

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of dpa
Sent: Wednesday, October 28, 2015 4:12 PM
To: 'OpenSIPS users mailling list'
Subject: [OpenSIPS-Users] Opensips 1.11 permission module problem

 

Hello!

 

OpenSIPS (1.11.5-notls (x86_64/linux))

 

I have a periodic problem with permissions module.

I could not make opensipsctl address reload (“400 Trusted table reload failed” 
received). In the time I have no problem with dialplan or drouting modules.

Opensips reload solve the problem.

 

 

In attachment log from opensips and statistic.

 

Thank you for any help

 

 

 


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

 






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

 

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


Re: [OpenSIPS-Users] Opensips 1.11 permission module problem

2015-11-18 Thread dpa
32M

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Wednesday, November 18, 2015 5:24 PM
To: dpa; 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] Opensips 1.11 permission module problem

 

How much pkg memory have you configured in your opensips ?

Regards,



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

On 18.11.2015 12:24, dpa wrote:

2314

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Wednesday, November 18, 2015 1:17 PM
To: dpa; 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] Opensips 1.11 permission module problem

 

How many records do you have in the address table ?

Regards,




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

On 18.11.2015 12:14, dpa wrote:

Hello Bogdan,

 

Yes, I see errors

“Oct 28 15:55:45 opensips-main /usr/local/opensips1.11/sbin/opensips[7715]: 
WARNING:core:fm_malloc: Not enough free memory, will attempt defragmentation

Oct 28 15:55:45 opensips-main /usr/local/opensips1.11/sbin/opensips[7715]: 
ERROR:core:db_allocate_rows: no memory left

Oct 28 15:55:45 opensips-main /usr/local/opensips1.11/sbin/opensips[7715]: 
ERROR:db_mysql:db_mysql_convert_rows: no private memory left

Oct 28 15:55:45 opensips-main /usr/local/opensips1.11/sbin/opensips[7715]: 
ERROR:db_mysql:db_mysql_convert_result: error while converting rows

Oct 28 15:55:45 opensips-main /usr/local/opensips1.11/sbin/opensips[7715]: 
ERROR:db_mysql:db_mysql_store_result: error while converting result

Oct 28 15:55:45 opensips-main /usr/local/opensips1.11/sbin/opensips[7715]: 
ERROR:core:db_do_query: error while storing result for query [select 
ip,grp,mask,port,proto,pattern,context_info,id from ast_address ]

Oct 28 15:55:45 opensips-main /usr/local/opensips1.11/sbin/opensips[7715]: 
ERROR:permissions:reload_address_table: failed to query database”

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Wednesday, November 18, 2015 1:04 PM
To: OpenSIPS users mailling list; denis7...@mail.ru
Subject: Re: [OpenSIPS-Users] Opensips 1.11 permission module problem

 

Hi Denis,

Do you see any errors in logs when the reload fails ?

Regards,





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

On 18.11.2015 07:29, dpa wrote:

Hello Richard

 

It is about 1 – 1,5 week.

No, I do not use subnets, only IP address. 

 

“There was a package memory leak that was fixed in the development tree”

Where can I read about it? Thank you.

 

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Richard Revels
Sent: Tuesday, November 17, 2015 8:55 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Opensips 1.11 permission module problem

 

Are you using subnets in your address table?  How often do you reload the 
address table?

 

There was a package memory leak that was fixed in the development tree that 
might explain what you are seeing.

 

On Wed, Nov 11, 2015 at 4:26 AM, dpa <denis7...@mail.ru> wrote:



Hello!

 

Is there any assumption about problem?

 

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of dpa
Sent: Wednesday, October 28, 2015 4:12 PM
To: 'OpenSIPS users mailling list'
Subject: [OpenSIPS-Users] Opensips 1.11 permission module problem

 

Hello!

 

OpenSIPS (1.11.5-notls (x86_64/linux))

 

I have a periodic problem with permissions module.

I could not make opensipsctl address reload (“400 Trusted table reload failed” 
received). In the time I have no problem with dialplan or drouting modules.

Opensips reload solve the problem.

 

 

In attachment log from opensips and statistic.

 

Thank you for any help

 

 

 


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

 








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

 

 

 

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


Re: [OpenSIPS-Users] Opensips 1.11 permission module problem

2015-11-17 Thread dpa
Hello Richard

 

It is about 1 – 1,5 week.

No, I do not use subnets, only IP address. 

 

“There was a package memory leak that was fixed in the development tree”

Where can I read about it? Thank you.

 

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Richard Revels
Sent: Tuesday, November 17, 2015 8:55 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Opensips 1.11 permission module problem

 

Are you using subnets in your address table?  How often do you reload the 
address table?

 

There was a package memory leak that was fixed in the development tree that 
might explain what you are seeing.

 

On Wed, Nov 11, 2015 at 4:26 AM, dpa <denis7...@mail.ru> wrote:



Hello!

 

Is there any assumption about problem?

 

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of dpa
Sent: Wednesday, October 28, 2015 4:12 PM
To: 'OpenSIPS users mailling list'
Subject: [OpenSIPS-Users] Opensips 1.11 permission module problem

 

Hello!

 

OpenSIPS (1.11.5-notls (x86_64/linux))

 

I have a periodic problem with permissions module.

I could not make opensipsctl address reload (“400 Trusted table reload failed” 
received). In the time I have no problem with dialplan or drouting modules.

Opensips reload solve the problem.

 

 

In attachment log from opensips and statistic.

 

Thank you for any help

 

 

 


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

 

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


Re: [OpenSIPS-Users] Opensips 1.11 permission module problem

2015-11-11 Thread dpa


Hello!

 

Is there any assumption about problem?

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of dpa
Sent: Wednesday, October 28, 2015 4:12 PM
To: 'OpenSIPS users mailling list'
Subject: [OpenSIPS-Users] Opensips 1.11 permission module problem

 

Hello!

 

OpenSIPS (1.11.5-notls (x86_64/linux))

 

I have a periodic problem with permissions module.

I could not make opensipsctl address reload ("400 Trusted table reload
failed" received). In the time I have no problem with dialplan or drouting
modules.

Opensips reload solve the problem.

 

 

In attachment log from opensips and statistic.

 

Thank you for any help

 

 

 

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


[OpenSIPS-Users] Opensips and TLS

2015-11-02 Thread dpa


Hello!

 

I want to ask an advice for making one scheme.

 

There is one softswitch (rather old) which does not support TLS protocol. 

Can I use Opensips as TLS intermediary between some SIP UA and old
softswitch?

 

Scheme:

SIP UA ßà Opensips (with TLS) ßà old softswitch.

 

Old softswitch must know SIP UA status (register/unregister) and must make
incoming call to SIP UA.

 

Thank you for any help

 

 

 

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


[OpenSIPS-Users] Opensips 1.11 permission module problem

2015-10-28 Thread dpa


Hello!

 

OpenSIPS (1.11.5-notls (x86_64/linux))

 

I have a periodic problem with permissions module.

I could not make opensipsctl address reload ("400 Trusted table reload
failed" received). In the time I have no problem with dialplan or drouting
modules.

Opensips reload solve the problem.

 

 

In attachment log from opensips and statistic.

 

Thank you for any help

 

 

 

Log from opensips

Oct 28 15:55:45 opensips-main /usr/local/opensips1.11/sbin/opensips[7715]: 
WARNING:core:fm_malloc: Not enough free memory, will attempt defragmentation
Oct 28 15:55:45 opensips-main /usr/local/opensips1.11/sbin/opensips[7715]: 
ERROR:core:db_allocate_rows: no memory left
Oct 28 15:55:45 opensips-main /usr/local/opensips1.11/sbin/opensips[7715]: 
ERROR:db_mysql:db_mysql_convert_rows: no private memory left
Oct 28 15:55:45 opensips-main /usr/local/opensips1.11/sbin/opensips[7715]: 
ERROR:db_mysql:db_mysql_convert_result: error while converting rows
Oct 28 15:55:45 opensips-main /usr/local/opensips1.11/sbin/opensips[7715]: 
ERROR:db_mysql:db_mysql_store_result: error while converting result
Oct 28 15:55:45 opensips-main /usr/local/opensips1.11/sbin/opensips[7715]: 
ERROR:core:db_do_query: error while storing result for query [select 
ip,grp,mask,port,proto,pattern,context_info,id from ast_address ]
Oct 28 15:55:45 opensips-main /usr/local/opensips1.11/sbin/opensips[7715]: 
ERROR:permissions:reload_address_table: failed to query database


Statistic

shmem:total_size:: 536870912
shmem:used_size:: 98591856
shmem:real_used_size:: 163253328
shmem:max_used_size:: 327067912
shmem:free_size:: 373617584
shmem:fragments:: 1246896
pkmem:0-total_size:: 0
pkmem:0-used_size:: 0
pkmem:0-real_used_size:: 0
pkmem:0-max_used_size:: 0
pkmem:0-free_size:: 0
pkmem:0-fragments:: 0
pkmem:1-total_size:: 33554432
pkmem:1-used_size:: 931760
pkmem:1-real_used_size:: 1154672
pkmem:1-max_used_size:: 4667952
pkmem:1-free_size:: 32399760
pkmem:1-fragments:: 32
pkmem:2-total_size:: 33554432
pkmem:2-used_size:: 989496
pkmem:2-real_used_size:: 1208520
pkmem:2-max_used_size:: 4667952
pkmem:2-free_size:: 32345912
pkmem:2-fragments:: 5
pkmem:3-total_size:: 33554432
pkmem:3-used_size:: 23821480
pkmem:3-real_used_size:: 30034288
pkmem:3-max_used_size:: 30460160
pkmem:3-free_size:: 3520144
pkmem:3-fragments:: 1102
pkmem:4-total_size:: 33554432
pkmem:4-used_size:: 950048
pkmem:4-real_used_size:: 1245200
pkmem:4-max_used_size:: 4720208
pkmem:4-free_size:: 32309232
pkmem:4-fragments:: 3171
pkmem:5-total_size:: 33554432
pkmem:5-used_size:: 950352
pkmem:5-real_used_size:: 1245672
pkmem:5-max_used_size:: 4667952
pkmem:5-free_size:: 32308760
pkmem:5-fragments:: 3178
pkmem:6-total_size:: 33554432
pkmem:6-used_size:: 949392
pkmem:6-real_used_size:: 1246152
pkmem:6-max_used_size:: 4667952
pkmem:6-free_size:: 32308280
pkmem:6-fragments:: 3238
pkmem:7-total_size:: 33554432
pkmem:7-used_size:: 950592
pkmem:7-real_used_size:: 1245840
pkmem:7-max_used_size:: 4667952
pkmem:7-free_size:: 32308592
pkmem:7-fragments:: 3175
pkmem:8-total_size:: 33554432
pkmem:8-used_size:: 949192
pkmem:8-real_used_size:: 1245736
pkmem:8-max_used_size:: 4667952
pkmem:8-free_size:: 32308696
pkmem:8-fragments:: 3229
pkmem:9-total_size:: 33554432
pkmem:9-used_size:: 949368
pkmem:9-real_used_size:: 1244328
pkmem:9-max_used_size:: 4667952
pkmem:9-free_size:: 32310104
pkmem:9-fragments:: 3163
pkmem:10-total_size:: 33554432
pkmem:10-used_size:: 949672
pkmem:10-real_used_size:: 1244056
pkmem:10-max_used_size:: 4667952
pkmem:10-free_size:: 32310376
pkmem:10-fragments:: 3139
pkmem:11-total_size:: 33554432
pkmem:11-used_size:: 951784
pkmem:11-real_used_size:: 1247896
pkmem:11-max_used_size:: 4667952
pkmem:11-free_size:: 32306536
pkmem:11-fragments:: 3211
pkmem:12-total_size:: 33554432
pkmem:12-used_size:: 954664
pkmem:12-real_used_size:: 1246744
pkmem:12-max_used_size:: 4667952
pkmem:12-free_size:: 32307688
pkmem:12-fragments:: 3044
pkmem:13-total_size:: 33554432
pkmem:13-used_size:: 953584
pkmem:13-real_used_size:: 1246264
pkmem:13-max_used_size:: 4667952
pkmem:13-free_size:: 32308168
pkmem:13-fragments:: 3069
pkmem:14-total_size:: 33554432
pkmem:14-used_size:: 954288
pkmem:14-real_used_size:: 1246008
pkmem:14-max_used_size:: 4667952
pkmem:14-free_size:: 32308424
pkmem:14-fragments:: 3029
pkmem:15-total_size:: 33554432
pkmem:15-used_size:: 954304
pkmem:15-real_used_size:: 1246096
pkmem:15-max_used_size:: 4667952
pkmem:15-free_size:: 32308336
pkmem:15-fragments:: 3032
pkmem:16-total_size:: 33554432
pkmem:16-used_size:: 954712
pkmem:16-real_used_size:: 1245832
pkmem:16-max_used_size:: 4667952
pkmem:16-free_size:: 32308600
pkmem:16-fragments:: 3004
pkmem:17-total_size:: 33554432
pkmem:17-used_size:: 953728
pkmem:17-real_used_size:: 1244944
pkmem:17-max_used_size:: 4667952
pkmem:17-free_size:: 32309488
pkmem:17-fragments:: 3008
pkmem:18-total_size:: 33554432
pkmem:18-used_size:: 954488
pkmem:18-real_used_size:: 1245800
pkmem:18-max_used_size:: 4667952
pkmem:18-free_size:: 32308632

Re: [OpenSIPS-Users] opensips1.11 dr_routing

2015-08-24 Thread dpa
Hello Bogdan!

 

Yes, you are right, I want use only carriers in routing.

 

For example.

I have for providers

#14, #34, #35, #2881.

 

Let`s  say that the goal of some prefix is

Prefix 9087: #14 - 1$, #34-2$, #35 - 2$, #2881 - 3$

Prefix 8765: #34-1$, #35-2$, #14-3$, #2881 - 3$

 

In ideal I want to see in routing table:

9087 #14,#34=50,#35=50,#2881

8765 #34,#35,#14=50,#2881=50

 

First Opensips make serial routing and then parallel.

 

In method you gave I must create carriers:

34=50, 35=50

34=33, 35=33, 2881=33 an so on.

And all these carriers I must remember in another system (which forms
routing table) and using it during forming routing table. 

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Friday, August 21, 2015 4:35 PM
To: dpa; 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] opensips1.11 dr_routing

 

Denis,

So what you want to do is to be able to serial and parallel (combined)
having carriers as elements in the list (and not GWs).

With the solution I gave you cannot do it as you cannot use carriers in the
definition of another carrier (when doing the parallel group), right ?

Regards,



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

On 21.08.2015 16:14, dpa wrote:

No, Bogdan, everything fine. You wright understand my question.

I wanted do a combination between serial and parallel selection but method
you suggest doesn`t fit to me.

The main reason I have many providers and using carrier for sharing
doesn`t convenient, because I must have all possible combination of carriers
and using (remember) it for making routing table (contents of routing table
formed by another system, and I do not want complicate it).

 

Thank you.

 

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Friday, August 21, 2015 3:52 PM
To: dpa; 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] opensips1.11 dr_routing

 

Denis,

I had the impression that I haven't managed to answer to your question.
Maybe because I do not fully understand the question. Could you detail why
the solution I gave does not fit you?

Regards,




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

On 21.08.2015 15:35, dpa wrote:

Hello Bogdan!

 

I understand and I thought about it before. But it doesn`t fit to me,
because I have many prefix which can sharing different carriers.

Anyway, thank you for help.

 

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Friday, August 21, 2015 3:29 PM
To: OpenSIPS users mailling list; denis7...@mail.ru
Subject: Re: [OpenSIPS-Users] opensips1.11 dr_routing

 

Hi Denis,

If you use weights alg and the GW has no weight, a 0 weight will be assumed,
so it will never match (unless after a failover).

Using weights, the selection is over all the GW in the set. 

If you want do a combination between serial and parallel selection , you
need to use carriers:

Make carrier X with wights flags and set for it destinations 34=50,35=50
and in the rule have the destination set 14,#X (and do not use the W flag
anymore)

Regards,





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

On 21.08.2015 12:16, dpa wrote:



Hello!

 

There is one question about dr_routing module.

 

Example 1.

I have such gw_list #14,#34=50,#35=50,#2881

Example 2.

I have such gw_list #14,#34,#35,#2881

 

I am using d_routing() with W flag.

 

In Example 1 a call first had been sent to sharing between #34 and #35 and
only then to #14.

In Example 2 a call first had been sent to #14, then #34, then #35 and
#2881.

 

Is there any way to tell Opensips in Example 1 first sends calls to #14 and
only then to sharing between #34 and #35?

 

Thank you for any help.

 








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

 

 

 

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


Re: [OpenSIPS-Users] opensips1.11 dr_routing

2015-08-24 Thread dpa
Second point seems interesting. I will test it.

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Monday, August 24, 2015 12:49 PM
To: dpa; 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] opensips1.11 dr_routing

 

Hi Denis,

Currently you cannot do this, but I see 2 options:

1) allow a carrier definition to contain other carriers (max 3 levels)
2) have a more complex syntax in destination set definition, with brackets,
like: 
#14,(#34=50,#35=50),#2881

Regards,



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

On 24.08.2015 10:42, dpa wrote:

Hello Bogdan!

 

Yes, you are right, I want use only carriers in routing.

 

For example.

I have for providers

#14, #34, #35, #2881.

 

Let`s  say that the goal of some prefix is

Prefix 9087: #14 - 1$, #34-2$, #35 - 2$, #2881 - 3$

Prefix 8765: #34-1$, #35-2$, #14-3$, #2881 - 3$

 

In ideal I want to see in routing table:

9087 #14,#34=50,#35=50,#2881

8765 #34,#35,#14=50,#2881=50

 

First Opensips make serial routing and then parallel.

 

In method you gave I must create carriers:

34=50, 35=50

34=33, 35=33, 2881=33 an so on.

And all these carriers I must remember in another system (which forms
routing table) and using it during forming routing table. 

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Friday, August 21, 2015 4:35 PM
To: dpa; 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] opensips1.11 dr_routing

 

Denis,

So what you want to do is to be able to serial and parallel (combined)
having carriers as elements in the list (and not GWs).

With the solution I gave you cannot do it as you cannot use carriers in the
definition of another carrier (when doing the parallel group), right ?

Regards,




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

On 21.08.2015 16:14, dpa wrote:

No, Bogdan, everything fine. You wright understand my question.

I wanted do a combination between serial and parallel selection but method
you suggest doesn`t fit to me.

The main reason I have many providers and using carrier for sharing
doesn`t convenient, because I must have all possible combination of carriers
and using (remember) it for making routing table (contents of routing table
formed by another system, and I do not want complicate it).

 

Thank you.

 

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Friday, August 21, 2015 3:52 PM
To: dpa; 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] opensips1.11 dr_routing

 

Denis,

I had the impression that I haven't managed to answer to your question.
Maybe because I do not fully understand the question. Could you detail why
the solution I gave does not fit you?

Regards,





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

On 21.08.2015 15:35, dpa wrote:

Hello Bogdan!

 

I understand and I thought about it before. But it doesn`t fit to me,
because I have many prefix which can sharing different carriers.

Anyway, thank you for help.

 

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Friday, August 21, 2015 3:29 PM
To: OpenSIPS users mailling list; denis7...@mail.ru
Subject: Re: [OpenSIPS-Users] opensips1.11 dr_routing

 

Hi Denis,

If you use weights alg and the GW has no weight, a 0 weight will be assumed,
so it will never match (unless after a failover).

Using weights, the selection is over all the GW in the set. 

If you want do a combination between serial and parallel selection , you
need to use carriers:

Make carrier X with wights flags and set for it destinations 34=50,35=50
and in the rule have the destination set 14,#X (and do not use the W flag
anymore)

Regards,






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

On 21.08.2015 12:16, dpa wrote:



Hello!

 

There is one question about dr_routing module.

 

Example 1.

I have such gw_list #14,#34=50,#35=50,#2881

Example 2.

I have such gw_list #14,#34,#35,#2881

 

I am using d_routing() with W flag.

 

In Example 1 a call first had been sent to sharing between #34 and #35 and
only then to #14.

In Example 2 a call first had been sent to #14, then #34, then #35 and
#2881.

 

Is there any way to tell Opensips in Example 1 first sends calls to #14 and
only then to sharing between #34 and #35?

 

Thank you for any help.

 









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

 

 

 

 

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


Re: [OpenSIPS-Users] opensips1.11 dr_routing

2015-08-24 Thread dpa
Ooops and I was very happy some minutes ago:)))

Ok, will wait an options.

 

Thank you!

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Monday, August 24, 2015 12:58 PM
To: dpa; 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] opensips1.11 dr_routing

 

Hold on ! none of the options are available now . I suggested them as
options for a fix ;)

Regards,



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

On 24.08.2015 12:55, dpa wrote:

Second point seems interesting. I will test it.

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Monday, August 24, 2015 12:49 PM
To: dpa; 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] opensips1.11 dr_routing

 

Hi Denis,

Currently you cannot do this, but I see 2 options:

1) allow a carrier definition to contain other carriers (max 3 levels)
2) have a more complex syntax in destination set definition, with brackets,
like: 
#14,(#34=50,#35=50),#2881

Regards,




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

On 24.08.2015 10:42, dpa wrote:

Hello Bogdan!

 

Yes, you are right, I want use only carriers in routing.

 

For example.

I have for providers

#14, #34, #35, #2881.

 

Let`s  say that the goal of some prefix is

Prefix 9087: #14 - 1$, #34-2$, #35 - 2$, #2881 - 3$

Prefix 8765: #34-1$, #35-2$, #14-3$, #2881 - 3$

 

In ideal I want to see in routing table:

9087 #14,#34=50,#35=50,#2881

8765 #34,#35,#14=50,#2881=50

 

First Opensips make serial routing and then parallel.

 

In method you gave I must create carriers:

34=50, 35=50

34=33, 35=33, 2881=33 an so on.

And all these carriers I must remember in another system (which forms
routing table) and using it during forming routing table. 

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Friday, August 21, 2015 4:35 PM
To: dpa; 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] opensips1.11 dr_routing

 

Denis,

So what you want to do is to be able to serial and parallel (combined)
having carriers as elements in the list (and not GWs).

With the solution I gave you cannot do it as you cannot use carriers in the
definition of another carrier (when doing the parallel group), right ?

Regards,





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

On 21.08.2015 16:14, dpa wrote:

No, Bogdan, everything fine. You wright understand my question.

I wanted do a combination between serial and parallel selection but method
you suggest doesn`t fit to me.

The main reason I have many providers and using carrier for sharing
doesn`t convenient, because I must have all possible combination of carriers
and using (remember) it for making routing table (contents of routing table
formed by another system, and I do not want complicate it).

 

Thank you.

 

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Friday, August 21, 2015 3:52 PM
To: dpa; 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] opensips1.11 dr_routing

 

Denis,

I had the impression that I haven't managed to answer to your question.
Maybe because I do not fully understand the question. Could you detail why
the solution I gave does not fit you?

Regards,






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

On 21.08.2015 15:35, dpa wrote:

Hello Bogdan!

 

I understand and I thought about it before. But it doesn`t fit to me,
because I have many prefix which can sharing different carriers.

Anyway, thank you for help.

 

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Friday, August 21, 2015 3:29 PM
To: OpenSIPS users mailling list; denis7...@mail.ru
Subject: Re: [OpenSIPS-Users] opensips1.11 dr_routing

 

Hi Denis,

If you use weights alg and the GW has no weight, a 0 weight will be assumed,
so it will never match (unless after a failover).

Using weights, the selection is over all the GW in the set. 

If you want do a combination between serial and parallel selection , you
need to use carriers:

Make carrier X with wights flags and set for it destinations 34=50,35=50
and in the rule have the destination set 14,#X (and do not use the W flag
anymore)

Regards,







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

On 21.08.2015 12:16, dpa wrote:



Hello!

 

There is one question about dr_routing module.

 

Example 1.

I have such gw_list #14,#34=50,#35=50,#2881

Example 2.

I have such gw_list #14,#34,#35,#2881

 

I am using d_routing() with W flag.

 

In Example 1 a call first had been sent to sharing between #34 and #35 and
only then to #14.

In Example 2 a call first had been sent to #14, then #34, then #35 and
#2881.

 

Is there any way to tell Opensips in Example 1 first sends calls to #14 and
only then to sharing between #34 and #35?

 

Thank you for any help

Re: [OpenSIPS-Users] opensips1.11 dr_routing

2015-08-21 Thread dpa
Hello Bogdan!

 

I understand and I thought about it before. But it doesn`t fit to me,
because I have many prefix which can sharing different carriers.

Anyway, thank you for help.

 

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Friday, August 21, 2015 3:29 PM
To: OpenSIPS users mailling list; denis7...@mail.ru
Subject: Re: [OpenSIPS-Users] opensips1.11 dr_routing

 

Hi Denis,

If you use weights alg and the GW has no weight, a 0 weight will be assumed,
so it will never match (unless after a failover).

Using weights, the selection is over all the GW in the set. 

If you want do a combination between serial and parallel selection , you
need to use carriers:

Make carrier X with wights flags and set for it destinations 34=50,35=50
and in the rule have the destination set 14,#X (and do not use the W flag
anymore)

Regards,



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

On 21.08.2015 12:16, dpa wrote:



Hello!

 

There is one question about dr_routing module.

 

Example 1.

I have such gw_list #14,#34=50,#35=50,#2881

Example 2.

I have such gw_list #14,#34,#35,#2881

 

I am using d_routing() with W flag.

 

In Example 1 a call first had been sent to sharing between #34 and #35 and
only then to #14.

In Example 2 a call first had been sent to #14, then #34, then #35 and
#2881.

 

Is there any way to tell Opensips in Example 1 first sends calls to #14 and
only then to sharing between #34 and #35?

 

Thank you for any help.

 






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

 

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


Re: [OpenSIPS-Users] opensips1.11 dr_routing

2015-08-21 Thread dpa
No, Bogdan, everything fine. You wright understand my question.

I wanted do a combination between serial and parallel selection but method
you suggest doesn`t fit to me.

The main reason I have many providers and using carrier for sharing
doesn`t convenient, because I must have all possible combination of carriers
and using (remember) it for making routing table (contents of routing table
formed by another system, and I do not want complicate it).

 

Thank you.

 

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Friday, August 21, 2015 3:52 PM
To: dpa; 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] opensips1.11 dr_routing

 

Denis,

I had the impression that I haven't managed to answer to your question.
Maybe because I do not fully understand the question. Could you detail why
the solution I gave does not fit you?

Regards,



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

On 21.08.2015 15:35, dpa wrote:

Hello Bogdan!

 

I understand and I thought about it before. But it doesn`t fit to me,
because I have many prefix which can sharing different carriers.

Anyway, thank you for help.

 

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Friday, August 21, 2015 3:29 PM
To: OpenSIPS users mailling list; denis7...@mail.ru
Subject: Re: [OpenSIPS-Users] opensips1.11 dr_routing

 

Hi Denis,

If you use weights alg and the GW has no weight, a 0 weight will be assumed,
so it will never match (unless after a failover).

Using weights, the selection is over all the GW in the set. 

If you want do a combination between serial and parallel selection , you
need to use carriers:

Make carrier X with wights flags and set for it destinations 34=50,35=50
and in the rule have the destination set 14,#X (and do not use the W flag
anymore)

Regards,




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

On 21.08.2015 12:16, dpa wrote:



Hello!

 

There is one question about dr_routing module.

 

Example 1.

I have such gw_list #14,#34=50,#35=50,#2881

Example 2.

I have such gw_list #14,#34,#35,#2881

 

I am using d_routing() with W flag.

 

In Example 1 a call first had been sent to sharing between #34 and #35 and
only then to #14.

In Example 2 a call first had been sent to #14, then #34, then #35 and
#2881.

 

Is there any way to tell Opensips in Example 1 first sends calls to #14 and
only then to sharing between #34 and #35?

 

Thank you for any help.

 







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

 

 

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


[OpenSIPS-Users] opensips1.11 dr_routing

2015-08-21 Thread dpa


Hello!

 

There is one question about dr_routing module.

 

Example 1.

I have such gw_list #14,#34=50,#35=50,#2881

Example 2.

I have such gw_list #14,#34,#35,#2881

 

I am using d_routing() with W flag.

 

In Example 1 a call first had been sent to sharing between #34 and #35 and
only then to #14.

In Example 2 a call first had been sent to #14, then #34, then #35 and
#2881.

 

Is there any way to tell Opensips in Example 1 first sends calls to #14 and
only then to sharing between #34 and #35?

 

Thank you for any help.

 

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


[OpenSIPS-Users] FW: Opensips 1.10 crash

2015-08-10 Thread dpa


Hello!

 

Is there any ideas about problem?

 

Thank you.

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of dpa
Sent: Wednesday, July 15, 2015 4:11 PM
To: 'OpenSIPS users mailling list'
Subject: [OpenSIPS-Users] Opensips 1.10 crash

 

Hello!

 

Opensips1.10.1. Working with rtpproxy2.0.

Today Opensips has crashed. Core file was generated. 

Information from core file is in attachment.

 

The question is why Opensips has crashed?

 

Thank you for any help.

 

 

 

gdb /usr/local/opensips1.10.1/sbin/opensips /opensipscore/core
GNU gdb (Ubuntu/Linaro 7.4-2012.04-0ubuntu2.1) 7.4-2012.04
Copyright (C) 2012 Free Software Foundation, Inc.
License GPLv3+: GNU GPL version 3 or later http://gnu.org/licenses/gpl.html
This is free software: you are free to change and redistribute it.
There is NO WARRANTY, to the extent permitted by law.  Type show copying
and show warranty for details.
This GDB was configured as x86_64-linux-gnu.
For bug reporting instructions, please see:
http://bugs.launchpad.net/gdb-linaro/...
Reading symbols from /usr/local/opensips1.10.1/sbin/opensips...done.
[New LWP 9791]

warning: Can't read pathname for load map: Îøèáêà ââîäà/âûâîäà.
[Thread debugging using libthread_db enabled]
Using host libthread_db library /lib/x86_64-linux-gnu/libthread_db.so.1.

warning: no loadable sections found in added symbol-file system-supplied DSO at 
0x7fff78bfe000
Core was generated by `/usr/local/opensips1.10.1/sbin/opensips -P 
/sock/opensips.pid -u opensips -w /o'.
Program terminated with signal 11, Segmentation fault.
#0  fm_remove_free (n=0x7fcf529e6840, qm=0x7fcf44a9b000) at mem/f_malloc.c:175
175 n-u.nxt_free-prev = pf;
(gdb) bt
#0  fm_remove_free (n=0x7fcf529e6840, qm=0x7fcf44a9b000) at mem/f_malloc.c:175
#1  fm_malloc (qm=0x7fcf44a9b000, size=704) at mem/f_malloc.c:383
#2  0x7fcf685860a5 in shm_malloc_unsafe (size=optimized out) at 
../../mem/shm_mem.h:248
#3  shm_malloc (size=optimized out) at ../../mem/shm_mem.h:258
#4  add_rt_info (pn=0x7fcf4e465d18, r=0x7fcf4f6bc9f8, rgid=127) at routing.c:367
#5  0x7fcf68581f3c in add_prefix (ptree=0x7fcf4e465cf8, prefix=optimized 
out, r=0x7fcf4f6bc9f8, rg=127) at prefix_tree.c:260
#6  0x7fcf685715a4 in add_rule (rule=optimized out, prefix=optimized 
out, grplst=optimized out, rdata=optimized out) at dr_load.c:188
#7  dr_load_routing_info (dr_dbf=0x7fcf687941a0, db_hdl=0x7fcf6ac053b0, 
drd_table=optimized out, drc_table=optimized out, drr_table=0x7fcf687939f0) 
at dr_load.c:512
#8  0x7fcf685811c0 in dr_reload_data () at drouting.c:425
#9  dr_reload_cmd (cmd_tree=optimized out, param=optimized out) at 
drouting.c:813
#10 0x7fcf68baf504 in run_mi_cmd (param=0x25e5aa0, f=optimized out, 
t=0x0, cmd=optimized out) at ../../mi/mi.h:109
#11 mi_fifo_server (fifo_stream=0x25e8e30) at fifo_fnc.c:490
#12 0x7fcf68bb0bef in fifo_process (rank=optimized out) at mi_fifo.c:213
#13 0x004b9de5 in start_module_procs () at sr_module.c:586
#14 0x0041475a in main_loop () at main.c:840
#15 main (argc=optimized out, argv=optimized out) at main.c:1598
(gdb) bt full
#0  fm_remove_free (n=0x7fcf529e6840, qm=0x7fcf44a9b000) at mem/f_malloc.c:175
pf = 0x7fcf44a9b640
hash = 2096
#1  fm_malloc (qm=0x7fcf44a9b000, size=704) at mem/f_malloc.c:383
frag = 0x7fcf529e6840
n = optimized out
hash = optimized out
__FUNCTION__ = fm_malloc
#2  0x7fcf685860a5 in shm_malloc_unsafe (size=optimized out) at 
../../mem/shm_mem.h:248
p = optimized out
#3  shm_malloc (size=optimized out) at ../../mem/shm_mem.h:258
p = 0x2c0
#4  add_rt_info (pn=0x7fcf4e465d18, r=0x7fcf4f6bc9f8, rgid=127) at routing.c:367
trg = optimized out
rtl_wrp = optimized out
rtlw = 0x0
i = optimized out
__FUNCTION__ = add_rt_info
#5  0x7fcf68581f3c in add_prefix (ptree=0x7fcf4e465cf8, prefix=optimized 
out, r=0x7fcf4f6bc9f8, rg=127) at prefix_tree.c:260
tmp = optimized out
res = 0
__FUNCTION__ = add_prefix
#6  0x7fcf685715a4 in add_rule (rule=optimized out, prefix=optimized 
out, grplst=optimized out, rdata=optimized out) at dr_load.c:188
tmp = optimized out
t = optimized out
ep = 0x2aa79cc ,130,139,142,145,148
n = optimized out
#7  dr_load_routing_info (dr_dbf=0x7fcf687941a0, db_hdl=0x7fcf6ac053b0, 
drd_table=optimized out, drc_table=optimized out, drr_table=0x7fcf687939f0) 
at dr_load.c:512
int_vals = {1522874, optimized out, 0, optimized out}
str_vals = {0x2aa7948 
19,20,23,25,27,31,33,35,37,29,40,44,47,50,12,53,56,59,62,65,66,69,72,75,78,81,84,97,88,91,94,100,103,106,109,112,115,118,121,124,127,130,139,142,145,148,
  optimized out, optimized out, 0x2aa79ed 13, optimized out, 
optimized out}
tmp = {s = 0x2aa79e1 81061881, len = 8}
columns = {0x7fcf68793400, 0x7fcf68793410, 0x7fcf68793420, 
0x7fcf68793430

[OpenSIPS-Users] changing $rU number

2015-07-23 Thread dpa


Hello!

 

Opensips 1.10.

 

I am using DROUTING module from making routing.

But some SIP UA sends to Opensips tel. number with  some unnecessary
characters, such as %20 and -.

 

The question is how can I delete these characters from request user?

 

Thank you for any help.

 

 

 

___
Users mailing list
Users@lists.opensips.org
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Re: [OpenSIPS-Users] changing $rU number

2015-07-23 Thread dpa
Thank you.

But what can I use for do it?

 

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Schneur Rosenberg
Sent: Thursday, July 23, 2015 10:40 AM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] changing $rU number

 

$rU is read/write so you can use regexp and just rewrite the variable

 

On Thu, Jul 23, 2015 at 10:33 AM, dpa denis7...@mail.ru wrote:



Hello!

 

Opensips 1.10.

 

I am using DROUTING module from making routing.

But some SIP UA sends to Opensips tel. number with  some unnecessary 
characters, such as “%20” and “-”.

 

The question is how can I delete these characters from request user?

 

Thank you for any help.

 

 

 


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

 

___
Users mailing list
Users@lists.opensips.org
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Re: [OpenSIPS-Users] changing $rU number

2015-07-23 Thread dpa
I understand, thank you.
I will try.

-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Trevor Steyn
Sent: Thursday, July 23, 2015 12:46 PM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] changing $rU number

Documentation can be found here

http://www.opensips.org/html/docs/modules/1.10.x/dialplan.html

For egsample to remove all - characters in $rU do the following

Add the following into the dialplan table

mysql select * from dialplan\G;
*** 1. row ***
 id: 6
   dpid: 1
 pr: 98
   match_op: 0
  match_exp: -
match_flags: 0
  subst_exp:
   repl_exp:
timerec:
   disabled: 0
  attrs:
*** 2. row ***


Then call dp_translate(1,$rU/$rU); in your script.

Regards
Trevor Steyn


On 23/07/2015 10:51, dpa wrote:

 And how dialplan helps me to do it, if ,for example, one time I have 
 such characters 8%2089-0987-09, and in another time I have 987-89%20908-1?

  

 *From:*users-boun...@lists.opensips.org
 [mailto:users-boun...@lists.opensips.org] *On Behalf Of *Trevor Steyn
 *Sent:* Thursday, July 23, 2015 11:27 AM
 *To:* users@lists.opensips.org
 *Subject:* Re: [OpenSIPS-Users] changing $rU number

  

 Hi Denis

 You can use the dialplan module to rewrite variables as below

 dp_translate(1,$rU/$rU);

 Then insert you regexp into the dialplan tables, read the dialplan 
 module documentation for more info.

 Regards
 Trevor

 On 23/07/2015 09:49, dpa wrote:

 Thank you.

 But what can I use for do it?

  

 *From:*users-boun...@lists.opensips.org
 mailto:users-boun...@lists.opensips.org
 [mailto:users-boun...@lists.opensips.org] *On Behalf Of *Schneur
 Rosenberg
 *Sent:* Thursday, July 23, 2015 10:40 AM
 *To:* OpenSIPS users mailling list
 *Subject:* Re: [OpenSIPS-Users] changing $rU number

  

 $rU is read/write so you can use regexp and just rewrite the 
 variable

  

 On Thu, Jul 23, 2015 at 10:33 AM, dpa denis7...@mail.ru
 mailto:denis7...@mail.ru wrote:

 Hello!

  

 Opensips 1.10.

  

 I am using DROUTING module from making routing.

 But some SIP UA sends to Opensips tel. number with  some
 unnecessary characters, such as %20 and -.

  

 The question is how can I delete these characters from request user?

  

 Thank you for any help.

  

  

  


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Re: [OpenSIPS-Users] changing $rU number

2015-07-23 Thread dpa
And how dialplan helps me to do it, if ,for example, one time I have such
characters 8%2089-0987-09, and in another time I have 987-89%20908-1?

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Trevor Steyn
Sent: Thursday, July 23, 2015 11:27 AM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] changing $rU number

 

Hi Denis

You can use the dialplan module to rewrite variables as below

dp_translate(1,$rU/$rU);

Then insert you regexp into the dialplan tables, read the dialplan module
documentation for more info.

Regards
Trevor

On 23/07/2015 09:49, dpa wrote:

Thank you.

But what can I use for do it?

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Schneur Rosenberg
Sent: Thursday, July 23, 2015 10:40 AM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] changing $rU number

 

$rU is read/write so you can use regexp and just rewrite the variable

 

On Thu, Jul 23, 2015 at 10:33 AM, dpa denis7...@mail.ru wrote:



Hello!

 

Opensips 1.10.

 

I am using DROUTING module from making routing.

But some SIP UA sends to Opensips tel. number with  some unnecessary
characters, such as %20 and -.

 

The question is how can I delete these characters from request user?

 

Thank you for any help.

 

 

 


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Re: [OpenSIPS-Users] changing $rU number

2015-07-23 Thread dpa
As I understand regex module just matches string against regexp. but not makes 
any manipulation.

 

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Schneur Rosenberg
Sent: Thursday, July 23, 2015 11:24 AM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] changing $rU number

 

http://www.opensips.org/html/docs/modules/1.10.x/regex.html

 

On Thu, Jul 23, 2015 at 10:49 AM, dpa denis7...@mail.ru wrote:

Thank you.

But what can I use for do it?

 

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Schneur Rosenberg
Sent: Thursday, July 23, 2015 10:40 AM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] changing $rU number

 

$rU is read/write so you can use regexp and just rewrite the variable

 

On Thu, Jul 23, 2015 at 10:33 AM, dpa denis7...@mail.ru wrote:



Hello!

 

Opensips 1.10.

 

I am using DROUTING module from making routing.

But some SIP UA sends to Opensips tel. number with  some unnecessary 
characters, such as “%20” and “-”.

 

The question is how can I delete these characters from request user?

 

Thank you for any help.

 

 

 


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Re: [OpenSIPS-Users] Ref: Opensips is not terminating call by “BYE” instead it is sending 404 error

2015-07-20 Thread dpa
Hello

 

Opensips does not responding to BYE (if you specially didn`t “ask” to do it). 
It just retransmit BYE to another SIP UA and wait a response from it. Once 
response had been got Opensips retransmit this response to the first SIP UA 
(witch initiated BYE).

If Opensips doesn`t retransmit BYE than something wrong with BYE message or in 
opensips.cfg script.

 

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of AMPTEL PTY LTD | RuvixTel
Sent: Sunday, July 19, 2015 4:01 PM
To: users@lists.opensips.org
Cc: rajat kapoor
Subject: [OpenSIPS-Users] Ref: Opensips is not terminating call by “BYE” 
instead it is sending 404 error

 

Hi, 

 

Please help us to resolve below issue will be appreciated. 

 

Our Opensips server (version1.6) is not responding to “Bye sent from 
termination server.  This is causing the calls to not end properly as it 
should. By checking sample SIP traces , it shows SIP server is always 
responding with a 404 not found respond of termination server's Bye message 
and keeps on sending re-invites request. 

 

--
Best Regards +

Anup ​





AMPtel PTY LTD, ( RuvixTel )

ABN: 77 162 081 905 
http://abr.business.gov.au/SearchByAbn.aspx?SearchText=77+162+081+905 

Ph:  +61 413 777 075 (Anup)

Ph: +61 433 232 177 (Piyali)

E-mail: i...@ruvixtel.net

 

  http://thedaneshproject.com/wp-content/uploads/2010/04/Green_footers_8.gif 

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[OpenSIPS-Users] Opensips 1.10 crash

2015-07-15 Thread dpa


Hello!

 

Opensips1.10.1. Working with rtpproxy2.0.

Today Opensips has crashed. Core file was generated. 

Information from core file is in attachment.

 

The question is why Opensips has crashed?

 

Thank you for any help.

 

 

 

gdb /usr/local/opensips1.10.1/sbin/opensips /opensipscore/core
GNU gdb (Ubuntu/Linaro 7.4-2012.04-0ubuntu2.1) 7.4-2012.04
Copyright (C) 2012 Free Software Foundation, Inc.
License GPLv3+: GNU GPL version 3 or later http://gnu.org/licenses/gpl.html
This is free software: you are free to change and redistribute it.
There is NO WARRANTY, to the extent permitted by law.  Type show copying
and show warranty for details.
This GDB was configured as x86_64-linux-gnu.
For bug reporting instructions, please see:
http://bugs.launchpad.net/gdb-linaro/...
Reading symbols from /usr/local/opensips1.10.1/sbin/opensips...done.
[New LWP 9791]

warning: Can't read pathname for load map: Îøèáêà ââîäà/âûâîäà.
[Thread debugging using libthread_db enabled]
Using host libthread_db library /lib/x86_64-linux-gnu/libthread_db.so.1.

warning: no loadable sections found in added symbol-file system-supplied DSO at 
0x7fff78bfe000
Core was generated by `/usr/local/opensips1.10.1/sbin/opensips -P 
/sock/opensips.pid -u opensips -w /o'.
Program terminated with signal 11, Segmentation fault.
#0  fm_remove_free (n=0x7fcf529e6840, qm=0x7fcf44a9b000) at mem/f_malloc.c:175
175 n-u.nxt_free-prev = pf;
(gdb) bt
#0  fm_remove_free (n=0x7fcf529e6840, qm=0x7fcf44a9b000) at mem/f_malloc.c:175
#1  fm_malloc (qm=0x7fcf44a9b000, size=704) at mem/f_malloc.c:383
#2  0x7fcf685860a5 in shm_malloc_unsafe (size=optimized out) at 
../../mem/shm_mem.h:248
#3  shm_malloc (size=optimized out) at ../../mem/shm_mem.h:258
#4  add_rt_info (pn=0x7fcf4e465d18, r=0x7fcf4f6bc9f8, rgid=127) at routing.c:367
#5  0x7fcf68581f3c in add_prefix (ptree=0x7fcf4e465cf8, prefix=optimized 
out, r=0x7fcf4f6bc9f8, rg=127) at prefix_tree.c:260
#6  0x7fcf685715a4 in add_rule (rule=optimized out, prefix=optimized 
out, grplst=optimized out, rdata=optimized out) at dr_load.c:188
#7  dr_load_routing_info (dr_dbf=0x7fcf687941a0, db_hdl=0x7fcf6ac053b0, 
drd_table=optimized out, drc_table=optimized out, drr_table=0x7fcf687939f0) 
at dr_load.c:512
#8  0x7fcf685811c0 in dr_reload_data () at drouting.c:425
#9  dr_reload_cmd (cmd_tree=optimized out, param=optimized out) at 
drouting.c:813
#10 0x7fcf68baf504 in run_mi_cmd (param=0x25e5aa0, f=optimized out, 
t=0x0, cmd=optimized out) at ../../mi/mi.h:109
#11 mi_fifo_server (fifo_stream=0x25e8e30) at fifo_fnc.c:490
#12 0x7fcf68bb0bef in fifo_process (rank=optimized out) at mi_fifo.c:213
#13 0x004b9de5 in start_module_procs () at sr_module.c:586
#14 0x0041475a in main_loop () at main.c:840
#15 main (argc=optimized out, argv=optimized out) at main.c:1598
(gdb) bt full
#0  fm_remove_free (n=0x7fcf529e6840, qm=0x7fcf44a9b000) at mem/f_malloc.c:175
pf = 0x7fcf44a9b640
hash = 2096
#1  fm_malloc (qm=0x7fcf44a9b000, size=704) at mem/f_malloc.c:383
frag = 0x7fcf529e6840
n = optimized out
hash = optimized out
__FUNCTION__ = fm_malloc
#2  0x7fcf685860a5 in shm_malloc_unsafe (size=optimized out) at 
../../mem/shm_mem.h:248
p = optimized out
#3  shm_malloc (size=optimized out) at ../../mem/shm_mem.h:258
p = 0x2c0
#4  add_rt_info (pn=0x7fcf4e465d18, r=0x7fcf4f6bc9f8, rgid=127) at routing.c:367
trg = optimized out
rtl_wrp = optimized out
rtlw = 0x0
i = optimized out
__FUNCTION__ = add_rt_info
#5  0x7fcf68581f3c in add_prefix (ptree=0x7fcf4e465cf8, prefix=optimized 
out, r=0x7fcf4f6bc9f8, rg=127) at prefix_tree.c:260
tmp = optimized out
res = 0
__FUNCTION__ = add_prefix
#6  0x7fcf685715a4 in add_rule (rule=optimized out, prefix=optimized 
out, grplst=optimized out, rdata=optimized out) at dr_load.c:188
tmp = optimized out
t = optimized out
ep = 0x2aa79cc ,130,139,142,145,148
n = optimized out
#7  dr_load_routing_info (dr_dbf=0x7fcf687941a0, db_hdl=0x7fcf6ac053b0, 
drd_table=optimized out, drc_table=optimized out, drr_table=0x7fcf687939f0) 
at dr_load.c:512
int_vals = {1522874, optimized out, 0, optimized out}
str_vals = {0x2aa7948 
19,20,23,25,27,31,33,35,37,29,40,44,47,50,12,53,56,59,62,65,66,69,72,75,78,81,84,97,88,91,94,100,103,106,109,112,115,118,121,124,127,130,139,142,145,148,
  optimized out, optimized out, 0x2aa79ed 13, optimized out, 
optimized out}
tmp = {s = 0x2aa79e1 81061881, len = 8}
columns = {0x7fcf68793400, 0x7fcf68793410, 0x7fcf68793420, 
0x7fcf68793430, 0x7fcf68793440, 0x7fcf68793450, 0x7fcf68793460, 0x7fcf68793390}
res = 0x7fcf6c24dd48
row = optimized out
ri = 0x7fcf4f6bc9f8
rdata = 0x7fcf4955e898
time_rec = optimized out
i = 4018
n = 29697
no_rows = optimized out

Re: [OpenSIPS-Users] Opensips 1.10.1 and radius problem

2014-06-06 Thread dpa
Hello Bogdan

 

What I can see in logs during test call 

ERROR:aaa_radius:send_auth_func: radius authentication message failed with
ERROR

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Thursday, June 05, 2014 6:20 PM
To: dpa; 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] Opensips 1.10.1 and radius problem

 

Hi,

-1 ret code means some internal error - check the opensips logs for any
error.

Regards,




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

On 03.06.2014 07:01, dpa wrote:

Hello Bogdan

 

Sorry for long answer.

 

I changed opensips.cfg just a little

 

..

radius_send_auth(set1,set2);

 switch($rc){

 

 case -2:

 xlog (Case -2 in radius route detected);

 break;

 

 case -1:

 xlog (Case -1 in radius route detected);

 break;

   }

..

 

In log file I see 

Jun  3 07:19:13 opensips-mirror
/usr/local/opensipsdev/sbin/opensips[14956]: Case -1 in radius route
detected

 

Meanwhile tcpdump shows that radius server answers Rejected on the
request.  

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Thursday, May 29, 2014 11:26 AM
To: OpenSIPS users mailling list; Denis Putyato
Subject: Re: [OpenSIPS-Users] Opensips 1.10.1 and radius problem

 

Hi,

On each case, please an xlog to print something from the script - in
this way you can double check which way your script went. Or, you can use
the script_trace() function to do that . See:
http://www.opensips.org/Documentation/Script-CoreFunctions-1-11#toc42

Regards,




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

On 29.05.2014 09:22, dpa wrote:



Hello!

 

I have such configuration in opensips.cfg

..

radius_send_auth(set1,set2);

 switch($rc){

 

 case -2:

 exec_msg(echo '$avp(500) $rU $time(%c) rejected' 
/opensips/alarmradius.txt);

 break;

 

 case -1:

 exec_msg(echo '$avp(500) $rU $time(%c) radius' 
/opensips/alarmradius.txt);

 break;

 

   }

.

In attachment you can see communication of Opensips with radius server
during making some test call (1.1.1.1 - Opensips, 2.2.2.2 - radius server)

 

After making a test call i see in alarmradius.txt record with radius word.

The question is why do I see radius word (and it means that Opensips
detected retcode -1) but not rejected (retcode -2) word? 

 

Thank you for any help.

 







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Re: [OpenSIPS-Users] Opensips 1.10.1 and radius problem

2014-06-06 Thread dpa
Hello Bogdan

 

Yes, I see reject.

 

I will try the patch and let you know about the result

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Friday, June 06, 2014 1:21 PM
To: dpa; 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] Opensips 1.10.1 and radius problem

 

Hi,

The libradius returns the error via the rc_auth() function. I notices a bit
of an inconsistency in the handling of the retcode of rc_auth() - old stuff
:).

You say you see a REJECT and the RADIUS level, right ?

Could you please try the attached patch and see if the failures are properly
reported to the script.

Regards,



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

On 06.06.2014 11:27, dpa wrote:

Hello Bogdan

 

What I can see in logs during test call 

ERROR:aaa_radius:send_auth_func: radius authentication message failed with
ERROR

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Thursday, June 05, 2014 6:20 PM
To: dpa; 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] Opensips 1.10.1 and radius problem

 

Hi,

-1 ret code means some internal error - check the opensips logs for any
error.

Regards,





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

On 03.06.2014 07:01, dpa wrote:

Hello Bogdan

 

Sorry for long answer.

 

I changed opensips.cfg just a little

 

..

radius_send_auth(set1,set2);

 switch($rc){

 

 case -2:

 xlog (Case -2 in radius route detected);

 break;

 

 case -1:

 xlog (Case -1 in radius route detected);

 break;

   }

 

 

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Re: [OpenSIPS-Users] Opensips 1.10.1 and radius problem

2014-06-06 Thread dpa
Unfortunately it doesn`t help.

 

I tried the patch (make command I gave from module/aaa_radius directory).

A test call shows 

Jun  6 13:57:42 opensips-mirror /usr/local/opensipsdev/sbin/opensips[4297]:
ERROR:aaa_radius:send_auth_func: radius authentication message failed with
ERROR

Jun  6 13:57:42 opensips-mirror /usr/local/opensipsdev/sbin/opensips[4297]:
Case -1 in radius route detected 

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Friday, June 06, 2014 1:21 PM
To: dpa; 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] Opensips 1.10.1 and radius problem

 

Hi,

The libradius returns the error via the rc_auth() function. I notices a bit
of an inconsistency in the handling of the retcode of rc_auth() - old stuff
:).

You say you see a REJECT and the RADIUS level, right ?

Could you please try the attached patch and see if the failures are properly
reported to the script.

Regards,



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

On 06.06.2014 11:27, dpa wrote:

Hello Bogdan

 

What I can see in logs during test call 

ERROR:aaa_radius:send_auth_func: radius authentication message failed with
ERROR

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Thursday, June 05, 2014 6:20 PM
To: dpa; 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] Opensips 1.10.1 and radius problem

 

Hi,

-1 ret code means some internal error - check the opensips logs for any
error.

Regards,





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

On 03.06.2014 07:01, dpa wrote:

Hello Bogdan

 

Sorry for long answer.

 

I changed opensips.cfg just a little

 

..

radius_send_auth(set1,set2);

 switch($rc){

 

 case -2:

 xlog (Case -2 in radius route detected);

 break;

 

 case -1:

 xlog (Case -1 in radius route detected);

 break;

   }

 

 

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Re: [OpenSIPS-Users] Opensips 1.10.1 and radius problem

2014-06-06 Thread dpa
Sorry, Bogdan

 

I missed one moment during starting opensips (I used wrong directory for
opensips modules)

The patch fixed the main problem. 

Now I see retcode -2.

 

Thank you very much!

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Friday, June 06, 2014 3:38 PM
To: dpa; 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] Opensips 1.10.1 and radius problem

 

Strange...
Could you directly add in your modules/aaa_radius/aaa_radius.c file, after
line 331 which is:
res = rc_auth(rh, SIP_PORT, send, recv, mess);
and extra log:
LM_ERR(rc_auth returned %d\n,res);

Recompile and run again - let's see exactly what the libradius has to return
:).

Thanks,



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

On 06.06.2014 13:00, dpa wrote:

Unfortunately it doesn`t help.

 

I tried the patch (make command I gave from module/aaa_radius directory).

A test call shows 

Jun  6 13:57:42 opensips-mirror /usr/local/opensipsdev/sbin/opensips[4297]:
ERROR:aaa_radius:send_auth_func: radius authentication message failed with
ERROR

Jun  6 13:57:42 opensips-mirror /usr/local/opensipsdev/sbin/opensips[4297]:
Case -1 in radius route detected 

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Friday, June 06, 2014 1:21 PM
To: dpa; 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] Opensips 1.10.1 and radius problem

 

Hi,

The libradius returns the error via the rc_auth() function. I notices a bit
of an inconsistency in the handling of the retcode of rc_auth() - old stuff
:).

You say you see a REJECT and the RADIUS level, right ?

Could you please try the attached patch and see if the failures are properly
reported to the script.

Regards,




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

On 06.06.2014 11:27, dpa wrote:

Hello Bogdan

 

What I can see in logs during test call 

ERROR:aaa_radius:send_auth_func: radius authentication message failed with
ERROR

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Thursday, June 05, 2014 6:20 PM
To: dpa; 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] Opensips 1.10.1 and radius problem

 

Hi,

-1 ret code means some internal error - check the opensips logs for any
error.

Regards,






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

On 03.06.2014 07:01, dpa wrote:

Hello Bogdan

 

Sorry for long answer.

 

I changed opensips.cfg just a little

 

..

radius_send_auth(set1,set2);

 switch($rc){

 

 case -2:

 xlog (Case -2 in radius route detected);

 break;

 

 case -1:

 xlog (Case -1 in radius route detected);

 break;

   }

 

 

 

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Re: [OpenSIPS-Users] Opensips 1.10.1 and radius problem

2014-06-02 Thread dpa
Hello Bogdan

 

Sorry for long answer.

 

I changed opensips.cfg just a little

 

..

radius_send_auth(set1,set2);

 switch($rc){

 

 case -2:

 xlog (Case -2 in radius route detected);

 break;

 

 case -1:

 xlog (Case -1 in radius route detected);

 break;

   }

..

 

In log file I see 

Jun  3 07:19:13 opensips-mirror
/usr/local/opensipsdev/sbin/opensips[14956]: Case -1 in radius route
detected

 

Meanwhile tcpdump shows that radius server answers Rejected on the
request.  

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Thursday, May 29, 2014 11:26 AM
To: OpenSIPS users mailling list; Denis Putyato
Subject: Re: [OpenSIPS-Users] Opensips 1.10.1 and radius problem

 

Hi,

On each case, please an xlog to print something from the script - in
this way you can double check which way your script went. Or, you can use
the script_trace() function to do that . See:
http://www.opensips.org/Documentation/Script-CoreFunctions-1-11#toc42

Regards,



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

On 29.05.2014 09:22, dpa wrote:



Hello!

 

I have such configuration in opensips.cfg

..

radius_send_auth(set1,set2);

 switch($rc){

 

 case -2:

 exec_msg(echo '$avp(500) $rU $time(%c) rejected' 
/opensips/alarmradius.txt);

 break;

 

 case -1:

 exec_msg(echo '$avp(500) $rU $time(%c) radius' 
/opensips/alarmradius.txt);

 break;

 

   }

.

In attachment you can see communication of Opensips with radius server
during making some test call (1.1.1.1 - Opensips, 2.2.2.2 - radius server)

 

After making a test call i see in alarmradius.txt record with radius word.

The question is why do I see radius word (and it means that Opensips
detected retcode -1) but not rejected (retcode -2) word? 

 

Thank you for any help.

 






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[OpenSIPS-Users] Opensips 1.10.1 and radius problem

2014-05-29 Thread dpa


Hello!

 

I have such configuration in opensips.cfg

:.

radius_send_auth(set1,set2);

 switch($rc){

 

 case -2:

 exec_msg(echo '$avp(500) $rU $time(%c) rejected' 
/opensips/alarmradius.txt);

 break;

 

 case -1:

 exec_msg(echo '$avp(500) $rU $time(%c) radius' 
/opensips/alarmradius.txt);

 break;

 

   }

:

In attachment you can see communication of Opensips with radius server
during making some test call (1.1.1.1 - Opensips, 2.2.2.2 - radius server)

 

After making a test call i see in alarmradius.txt record with radius word.

The question is why do I see radius word (and it means that Opensips
detected retcode -1) but not rejected (retcode -2) word? 

 

Thank you for any help.

 

No. TimeSourceDestination   Protocol Length 
Info
  3 3.2684971.1.1.12.2.2.2   RADIUS   113   
 Access-Request(1) (id=50, l=69)

Frame 3: 113 bytes on wire (904 bits), 113 bytes captured (904 bits)
Linux cooked capture
Internet Protocol Version 4, Src: 1.1.1.1 (1.1.1.1), Dst: 2.2.2.2 (2.2.2.2)
User Datagram Protocol, Src Port: 42580 (42580), Dst Port: radius (1812)
Radius Protocol
Code: Access-Request (1)
Packet identifier: 0x32 (50)
Length: 69
Authenticator: 9283df0bc7566891b5103fdcb78ae86d
[The response to this request is in frame 4]
Attribute Value Pairs
AVP: l=13  t=Called-Station-Id(30): 89213039240
AVP: l=12  t=Calling-Station-Id(31): 8612033848
AVP: l=12  t=User-Name(1): 8612033848
AVP: l=6  t=NAS-Port(5): 5060
AVP: l=6  t=NAS-IP-Address(4): 1.1.1.1

No. TimeSourceDestination   Protocol Length 
Info
  4 4.2755562.2.2.2 1.1.1.1  RADIUS   64
 Access-Reject(3) (id=50, l=20)

Frame 4: 64 bytes on wire (512 bits), 64 bytes captured (512 bits)
Linux cooked capture
Internet Protocol Version 4, Src: 2.2.2.2 (2.2.2.2), Dst: 1.1.1.1 (1.1.1.1)
User Datagram Protocol, Src Port: radius (1812), Dst Port: 42580 (42580)
Radius Protocol
Code: Access-Reject (3)
Packet identifier: 0x32 (50)
Length: 20
Authenticator: a41bae3fb27b20d439161eb531fd74a9
[This is a response to a request in frame 3]
[Time from request: 1.007059000 seconds]


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[OpenSIPS-Users] Opensips 1.10.1 stopped to process SIP messages

2014-04-08 Thread dpa


Hello!

 

Today there was one problem.

Opensips stopped to process SIP messages during about 2 minutes.

 

Log analyses of problem`s time period shows some unusual messages

 

ERROR:core:udp_send: sendto(sock,0x7fd068403ad0,35904,0,0x7fff63d1ddd0,16):
No buffer space available(105)

ERROR:aaa_radius:send_auth_func: radius authentication message failed with
TIMEOUT

rc_send_server: no reply from RADIUS server unknown:1812

rc_ip_hostname: couldn't look up host by addr: D5AA5098

 

 

Opensips during call processing interacts with radius server (and there was
some problem with radius server during problem`s time period). It was done
for prepaid scheme (although not all clients use prepaid)

 

More

During problem`s time period there were a lot of messages

BYE has been sent because of dialog lifetime = 10

It`s my own. Xlog (function located in local route of the opensips.cfg)
generates this message if Opensips terminates dialog for some reason. 

lifetime = 10 means that modparam(dialog, default_timeout, 10)
expired and there were no ACK on 200 OK. 

And ngrep shows that 200 OK had not translated to caller by Opensips during
problem`s time period.

 

After radius server UP ( and rc_send_server: no reply from RADIUS server
unknown:1812 and rc_ip_hostname: couldn't look up host by addr: D5AA5098
disappeared) Opensips became work normally. 

 

What can it be? And how can I avoid such problem in future?

 

Thank you for any help.

 

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Re: [OpenSIPS-Users] Opensips 1.9.1 crash

2014-04-01 Thread dpa
Hello Bogdan

 

OK, I will try upgrading to 1.10

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Tuesday, April 01, 2014 9:42 PM
To: OpenSIPS users mailling list; Denis Putyato
Subject: Re: [OpenSIPS-Users] Opensips 1.9.1 crash

 

Hello,

1.9 is at the end of its life span - it became unmaintained starting with
the 1.11 release (see http://www.opensips.org/About/AvailableVersions).

I strongly suggest upgrading to 1.10 , as you may experience problems which
were already fixed in the maintained versions. Is this an option for you ?

Regards,



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

On 01.04.2014 07:30, dpa wrote:



Hello!

 

I had two times Opensips crashed during one week.

 

On this link http://files.mail.ru/56128EED55BD49219EC1362E7161747A

you can find backtrace from core file for the first time.

 

On this link http://files.mail.ru/793A146CF453427893E2A66DF096A41E

you can find information from the log file. Unfortunately in this case no
core file had been generated.

 

The problem is very critical.

Thank you for any help!

 

 






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[OpenSIPS-Users] Opensips 1.9.1 crash

2014-03-31 Thread dpa


Hello!

 

I had two times Opensips crashed during one week.

 

On this link http://files.mail.ru/56128EED55BD49219EC1362E7161747A

you can find backtrace from core file for the first time.

 

On this link http://files.mail.ru/793A146CF453427893E2A66DF096A41E

you can find information from the log file. Unfortunately in this case no
core file had been generated.

 

The problem is very critical.

Thank you for any help!

 

 

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Re: [OpenSIPS-Users] Opensips 1.9.1 problem

2014-03-27 Thread dpa
Hello Bogdan

 

Unfortunaly  there were no more messages in log. Only what I sent.

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Thursday, March 27, 2014 12:34 PM
To: OpenSIPS users mailling list; Denis Putyato
Subject: Re: [OpenSIPS-Users] Opensips 1.9.1 problem

 

Hello,

Looks like you get HTTP traffic on your SIP/TCP ports - and opensips gives a
pars error (as expecting SIP, not HTTP syntax).

The critical error from the tm-timer indicates an huge amount of time spent
in the TM timer (process 32563) - but I see not previous errors from this
process to indicate what is going on there.

Regards,



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

On 25.03.2014 15:02, dpa wrote:



Hello!

 

In attachment there is some messages from log file.

 

What does it mean? These messages appeared in log after Opensips cannot
process SIP messages during about 1 minute.

 

Thank you

 

 






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Re: [OpenSIPS-Users] An Error in Opensips 1.9.1

2014-03-27 Thread dpa
Hello Bogdan

 

It will be super, because I spent a much time for finding this:)))

 

P.S. Of course not ($avp(63==1)) but ($avp(63)==1) 

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Thursday, March 27, 2014 12:37 PM
To: OpenSIPS users mailling list; Denis Putyato
Subject: Re: [OpenSIPS-Users] An Error in Opensips 1.9.1

 

Hello Denis,

Indeed, we need to improve a bit that error message, to make it more human
readable and less cryptic :). I will take care of that.

Regards, 



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

On 25.03.2014 14:04, dpa wrote:

No problem

In opensips.cfg there were one string

If ($avp(63)==1) {..

 

So after I changed the string to 

f ($avp(63==1)) {... errors disappear from the log

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Schneur Rosenberg
Sent: Tuesday, March 25, 2014 3:55 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] An Error in Opensips 1.9.1

 

It would be nice if you give us the problem and the solution.

On Mar 25, 2014 1:49 PM, dpa denis7...@mail.ru wrote:

It seems I found the problem myself.

Thank you

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of dpa
Sent: Tuesday, March 25, 2014 7:49 AM
To: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] An Error in Opensips 1.9.1

 

Hello

 

I am ready. But what type of information do I need provide?

To tell the truth I rarely look into opensips log file. Only when some
critical case happens (opensips crash for example).

Yesterday there was such case. 

 

Thank you.

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of qasimak...@gmail.com
Sent: Monday, March 24, 2014 8:29 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] An Error in Opensips 1.9.1

 

I think a little more information than this would be required if you need
help :).

Regards,
Qasim

 

On Mon, Mar 24, 2014 at 4:36 PM, dpa denis7...@mail.ru wrote:



Hello!

 

1.   In log file I see many errors

CRITICAL:core:comp_scriptvar: invalid operation 20/3/4!!

WARNING:core:do_action: error in expression (l=662)

 

As I understand there is a problem somewhere in opensips.cfg 

How can I understand where problem is?

 

2.   I have modparam(dialog, db_mode, 1) in opensips.cfg. 

In normal operation of opensips I can see in statistics, for example, 800
active dialogs, but after restart I see much more active dialogs. Why can it
be? 

 

Thank you for any help.

 

 


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[OpenSIPS-Users] Opensips 1.9.1 accounting

2014-03-26 Thread dpa


Hello!

 

In attachment you can find ordinary call (not successful)

1.1.1.1- SIP UA

1.1.1.2- Opensips

1.1.1.3- SIP UA

 

 

In opensips.cfg

modparam(acc, early_media, 0)

modparam(acc, report_cancels, 1)

modparam(acc, detect_direction, 1)

modparam(acc, db_flag, 15)

modparam(acc, db_missed_flag, 16)

modparam(acc, failed_transaction_flag, 17)

modparam(acc, db_table_acc, acc)

modparam(acc, db_table_missed_calls, acc)

modparam(acc, cdr_flag, 22)

 

and 

 

before INVITE will be translated to callee SIP UA

  setflag(15);

  setflag(16);

  setflag(17);

  setflag(22);

 

I see that Opensips tried to insert several entries into acc. 

 

The question is, why did Opensips try to insert into acc several entries due
to one call? Is this because of db_flag and failed_transaction_flag?

 

Thank you for any help. 

 

 

image001.gifU 2014/03/26 10:49:08.556019 1.1.1.1:53876 - 1.1.1.2:5060
INVITE sip:3364399@1.1.1.2:5060 SIP/2.0.
Via: SIP/2.0/UDP 1.1.1.1:5060;x-route-tag=tgrp:TFOP;branch=z9hG4bK448B21124B.
From: sip:8123364283@1.1.1.1;tag=32C4640-18B3.
To: sip:3364399@1.1.1.2.
Date: Wed, 26 Mar 2014 06:49:08 GMT.
Call-ID: 90A25AC0-B3E911E3-B077DBBA-3DF918A8@1.1.1.1.
Supported: 100rel,timer,resource-priority,replaces,sdp-anat.
Min-SE:  1800.
Cisco-Guid: 2426479216-3018396131-3064463394-2438468884.
User-Agent: Cisco-SIPGateway/IOS-12.x.
Accept-Language: ru.
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, 
NOTIFY, INFO, REGISTER.
CSeq: 101 INVITE.
Max-Forwards: 15.
Timestamp: 1395816548.
Contact: sip:8123364283@1.1.1.1:5060.
Expires: 100.
Allow-Events: telephone-event.
P-Asserted-Identity: sip:8123364283@1.1.1.1.
Content-Type: application/sdp.
Content-Disposition: session;handling=required.
Content-Length: 310.
.
v=0.
o=CiscoSystemsSIP-GW-UserAgent 4030 1274 IN IP4 1.1.1.1.
s=SIP Call.
c=IN IP4 1.1.1.1.
t=0 0.
m=audio 18038 RTP/AVP 8 0 18 101.
c=IN IP4 1.1.1.1.
a=rtpmap:8 PCMA/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=yes.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.


U 2014/03/26 10:49:08.557516 1.1.1.2:5060 - 1.1.1.1:5060
SIP/2.0 100 Giving a try.
Via: SIP/2.0/UDP 1.1.1.1:5060;x-route-tag=tgrp:TFOP;branch=z9hG4bK448B21124B.
From: sip:8123364283@1.1.1.1;tag=32C4640-18B3.
To: sip:3364399@1.1.1.2.
Call-ID: 90A25AC0-B3E911E3-B077DBBA-3DF918A8@1.1.1.1.
CSeq: 101 INVITE.
Content-Length: 0.
.


U 2014/03/26 10:49:08.566102 1.1.1.2:5060 - 1.1.1.3:5060
INVITE sip:78123364399@1.1.1.3:5060 SIP/2.0.
Record-Route: sip:1.1.1.2;lr;ftag=32C4640-18B3;did=8de.cf20f431.
Via: SIP/2.0/UDP 1.1.1.2:5060;branch=z9hG4bKcad3.e39f27b7.0.
Via: SIP/2.0/UDP 1.1.1.1:5060;x-route-tag=tgrp:TFOP;branch=z9hG4bK448B21124B.
From: sip:8123364283@1.1.1.1;tag=32C4640-18B3.
To: sip:3364399@1.1.1.2.
Date: Wed, 26 Mar 2014 06:49:08 GMT.
Call-ID: 90A25AC0-B3E911E3-B077DBBA-3DF918A8@1.1.1.1.
Supported: 100rel,timer,resource-priority,replaces,sdp-anat.
Min-SE:  1800.
Cisco-Guid: 2426479216-3018396131-3064463394-2438468884.
User-Agent: Cisco-SIPGateway/IOS-12.x.
Accept-Language: ru.
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, 
NOTIFY, INFO, REGISTER.
CSeq: 101 INVITE.
Max-Forwards: 15.
Timestamp: 1395816548.
Contact: sip:8123364283@1.1.1.1:5060.
Remote-Party-ID:sip:8123364283@1.1.1.1;party=calling;screen=yes;privacy=off.
Expires: 100.
Allow-Events: telephone-event.
P-Asserted-Identity: sip:8123364283@1.1.1.1.
Content-Type: application/sdp.
Content-Disposition: session;handling=required.
Content-Length: 310.
.
v=0.
o=CiscoSystemsSIP-GW-UserAgent 4030 1274 IN IP4 1.1.1.1.
s=SIP Call.
c=IN IP4 1.1.1.1.
t=0 0.
m=audio 18038 RTP/AVP 8 0 18 101.
c=IN IP4 1.1.1.1.
a=rtpmap:8 PCMA/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=yes.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.


U 2014/03/26 10:49:08.573798 1.1.1.3:5060 - 1.1.1.2:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 1.1.1.2:5060;branch=z9hG4bKcad3.e39f27b7.0.
Via: SIP/2.0/UDP 1.1.1.1:5060;x-route-tag=tgrp:TFOP;branch=z9hG4bK448B21124B.
From: sip:8123364283@1.1.1.1;tag=32C4640-18B3.
To: sip:3364399@1.1.1.2.
Call-ID: 90A25AC0-B3E911E3-B077DBBA-3DF918A8@1.1.1.1.
CSeq: 101 INVITE.
Server: Telphin SoftSwitch.
Content-Length: 0.
.


U 2014/03/26 10:49:08.576295 1.1.1.3:5060 - 1.1.1.2:5060
SIP/2.0 603 Declined.
Via: SIP/2.0/UDP 1.1.1.2:5060;branch=z9hG4bKcad3.e39f27b7.0.
Via: SIP/2.0/UDP 1.1.1.1:5060;x-route-tag=tgrp:TFOP;branch=z9hG4bK448B21124B.
From: sip:8123364283@1.1.1.1;tag=32C4640-18B3.
To: sip:3364399@1.1.1.2;tag=as26700a4a.
Call-ID: 90A25AC0-B3E911E3-B077DBBA-3DF918A8@1.1.1.1.
CSeq: 101 INVITE.
Server: Telphin MediaServer.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
Supported: replaces.
Content-Length: 0.
.


U 2014/03/26 10:49:08.576787 1.1.1.2:5060 - 1.1.1.3:5060
ACK sip:78123364399@1.1.1.3:5060 SIP/2.0.
Via: SIP/2.0/UDP 1.1.1.2:5060;branch=z9hG4bKcad3.e39f27b7.0.
From: sip:8123364283@1.1.1.1;tag=32C4640-18B3.
Call-ID: 

Re: [OpenSIPS-Users] An Error in Opensips 1.9.1

2014-03-25 Thread dpa
It seems I found the problem myself.

Thank you

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of dpa
Sent: Tuesday, March 25, 2014 7:49 AM
To: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] An Error in Opensips 1.9.1

 

Hello

 

I am ready. But what type of information do I need provide?

To tell the truth I rarely look into opensips log file. Only when some
critical case happens (opensips crash for example).

Yesterday there was such case. 

 

Thank you.

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of qasimak...@gmail.com
Sent: Monday, March 24, 2014 8:29 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] An Error in Opensips 1.9.1

 

I think a little more information than this would be required if you need
help :).

Regards,
Qasim

 

On Mon, Mar 24, 2014 at 4:36 PM, dpa denis7...@mail.ru wrote:



Hello!

 

1.   In log file I see many errors

CRITICAL:core:comp_scriptvar: invalid operation 20/3/4!!

WARNING:core:do_action: error in expression (l=662)

 

As I understand there is a problem somewhere in opensips.cfg 

How can I understand where problem is?

 

2.   I have modparam(dialog, db_mode, 1) in opensips.cfg. 

In normal operation of opensips I can see in statistics, for example, 800
active dialogs, but after restart I see much more active dialogs. Why can it
be? 

 

Thank you for any help.

 

 


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Re: [OpenSIPS-Users] An Error in Opensips 1.9.1

2014-03-25 Thread dpa
No problem

In opensips.cfg there were one string

If ($avp(63)==1) {..

 

So after I changed the string to 

f ($avp(63==1)) {... errors disappear from the log

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Schneur Rosenberg
Sent: Tuesday, March 25, 2014 3:55 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] An Error in Opensips 1.9.1

 

It would be nice if you give us the problem and the solution.

On Mar 25, 2014 1:49 PM, dpa denis7...@mail.ru wrote:

It seems I found the problem myself.

Thank you

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of dpa
Sent: Tuesday, March 25, 2014 7:49 AM
To: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] An Error in Opensips 1.9.1

 

Hello

 

I am ready. But what type of information do I need provide?

To tell the truth I rarely look into opensips log file. Only when some
critical case happens (opensips crash for example).

Yesterday there was such case. 

 

Thank you.

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of qasimak...@gmail.com
Sent: Monday, March 24, 2014 8:29 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] An Error in Opensips 1.9.1

 

I think a little more information than this would be required if you need
help :).

Regards,
Qasim

 

On Mon, Mar 24, 2014 at 4:36 PM, dpa denis7...@mail.ru wrote:



Hello!

 

1.   In log file I see many errors

CRITICAL:core:comp_scriptvar: invalid operation 20/3/4!!

WARNING:core:do_action: error in expression (l=662)

 

As I understand there is a problem somewhere in opensips.cfg 

How can I understand where problem is?

 

2.   I have modparam(dialog, db_mode, 1) in opensips.cfg. 

In normal operation of opensips I can see in statistics, for example, 800
active dialogs, but after restart I see much more active dialogs. Why can it
be? 

 

Thank you for any help.

 

 


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[OpenSIPS-Users] An Error in Opensips 1.9.1

2014-03-24 Thread dpa


Hello!

 

1.   In log file I see many errors

CRITICAL:core:comp_scriptvar: invalid operation 20/3/4!!

WARNING:core:do_action: error in expression (l=662)

 

As I understand there is a problem somewhere in opensips.cfg 

How can I understand where problem is?

 

2.   I have modparam(dialog, db_mode, 1) in opensips.cfg. 

In normal operation of opensips I can see in statistics, for example, 800
active dialogs, but after restart I see much more active dialogs. Why can it
be? 

 

Thank you for any help.

 

 

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Re: [OpenSIPS-Users] An Error in Opensips 1.9.1

2014-03-24 Thread dpa
Hello

 

I am ready. But what type of information do I need provide?

To tell the truth I rarely look into opensips log file. Only when some
critical case happens (opensips crash for example).

Yesterday there was such case. 

 

Thank you.

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of qasimak...@gmail.com
Sent: Monday, March 24, 2014 8:29 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] An Error in Opensips 1.9.1

 

I think a little more information than this would be required if you need
help :).

Regards,
Qasim

 

On Mon, Mar 24, 2014 at 4:36 PM, dpa denis7...@mail.ru wrote:



Hello!

 

1.   In log file I see many errors

CRITICAL:core:comp_scriptvar: invalid operation 20/3/4!!

WARNING:core:do_action: error in expression (l=662)

 

As I understand there is a problem somewhere in opensips.cfg 

How can I understand where problem is?

 

2.   I have modparam(dialog, db_mode, 1) in opensips.cfg. 

In normal operation of opensips I can see in statistics, for example, 800
active dialogs, but after restart I see much more active dialogs. Why can it
be? 

 

Thank you for any help.

 

 


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Re: [OpenSIPS-Users] Opensips 1.9.1 and REGISTER process

2014-03-05 Thread dpa
Hello Bogdan,

 

Yes E flags is working. But without in doesn`t

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Wednesday, March 05, 2014 9:56 PM
To: OpenSIPS users mailling list; Denis Putyato
Subject: Re: [OpenSIPS-Users] Opensips 1.9.1 and REGISTER process

 

Hello Denis,

Have you tried to use the E option in the save() function? see:
http://www.opensips.org/html/docs/modules/1.9.x/registrar.html#id250454

It should have the same effect (setting the max expire), but is per REGISTER
bases. Just to see if this works for you.

Regards,



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

On 03.03.2014 15:39, dpa wrote:



Hello!

 

There is one question.

 

A little part of opensips.cfg

..

modparam(registrar, default_expires, 60)

modparam(registrar, max_expires, 60)

modparam(registrar, min_expires, 0)

...

 

If I enter register timeout on my SIP UA to 1600, for example, Opensips will
return to SIP UA 1600 timeout.

In 1.6.4-2 there were no problem with it. If I enter 1600 timeout Opensips
returned 60 and after 60 s there was another attempt to register to
Opensips.

 

What did I miss? 

 

Thank you for any help.

 

 






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[OpenSIPS-Users] Opensips 1.9.1 and REGISTER process

2014-03-03 Thread dpa


Hello!

 

There is one question.

 

A little part of opensips.cfg

..

modparam(registrar, default_expires, 60)

modparam(registrar, max_expires, 60)

modparam(registrar, min_expires, 0)

...

 

If I enter register timeout on my SIP UA to 1600, for example, Opensips will
return to SIP UA 1600 timeout.

In 1.6.4-2 there were no problem with it. If I enter 1600 timeout Opensips
returned 60 and after 60 s there was another attempt to register to
Opensips.

 

What did I miss? 

 

Thank you for any help.

 

 

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[OpenSIPS-Users] Opensips 1.9.1 and dialog process

2014-01-10 Thread dpa


Hello

 

Early I used opensips1.6.4.-2. There were some fields (hash_entry and hash
id) in DB scheme of dialog module. I used this field to end dialog.

Now, in 1.9.1, the DB scheme of dialog module has been changed and there are
no fields hash_entry and hash id in шею But there is the dlg_id  field.

 

The question is, How can I bind dlg_id in mysql with hash entry and hash_id
for dialog ending.

 

Thank you for any help.

 

 

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Re: [OpenSIPS-Users] Opensips 1.9.1 and opensips_fifo file

2013-12-22 Thread dpa
Unfortunately I cannot tell which commands leave these entries. Anyway thank
you for help. 

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Liviu Chircu
Sent: Friday, December 20, 2013 6:03 PM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] Opensips 1.9.1 and opensips_fifo file

 

Are you able to tell which opensipsctl commands leave undeleted entries in
your /tmp directory?

In any case, OpenSIPS will work if you manually delete them, although this
is not normal :)

On 12/20/2013 03:54 PM, dpa wrote:

Hello

 

Sorry, but I don`t quite understand to replicate this issue?

Where to replicate?

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Liviu Chircu
Sent: Friday, December 20, 2013 5:17 PM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] Opensips 1.9.1 and opensips_fifo file

 

Hello,

That is actually a named pipe and it is opened by opensipsctl itself with
each command, in order to receive a reply back from OpenSIPS. Normally, they
should be automatically cleaned up.

Are you able to replicate this issue?

Best regards,




Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com

On 12/20/2013 06:17 AM, dpa wrote:



Hello

 

I made a decision to locate all files associated with Opensips to one
directory (say /sock/). This package of files includes opensps_fifo and
opensips.pid files.

I made an appropriate correction in opensips.cfg and  opensipsctlrc.

 

After restart Opensips everything works fine. Fifo commands is working.

 

But I found that in /tmp/ directory became appear some files. The name of
this files: opensips_receiver_some nubers. These files have 0 size.

The question is what is it? If I delete them, Will Opensips still work?

 

Thank you.

 

 







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[OpenSIPS-Users] Opensips 1.9.1 and opensips_fifo file

2013-12-19 Thread dpa


Hello

 

I made a decision to locate all files associated with Opensips to one
directory (say /sock/). This package of files includes opensps_fifo and
opensips.pid files.

I made an appropriate correction in opensips.cfg and  opensipsctlrc.

 

After restart Opensips everything works fine. Fifo commands is working.

 

But I found that in /tmp/ directory became appear some files. The name of
this files: opensips_receiver_some nubers. These files have 0 size.

The question is what is it? If I delete them, Will Opensips still work?

 

Thank you.

 

 

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Re: [OpenSIPS-Users] Opensips 1.9.1 and NAT

2013-11-25 Thread dpa
Hello

 

I understand but in onreply route I make a test: nat_uac_test(55) and only
if it successful I make “fix_nated_contact()”.

In my case nat_uac_test(55) must be fail after checking 183 ringing and
200 OK.

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Ali Pey
Sent: Friday, November 22, 2013 5:37 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Opensips 1.9.1 and NAT

 

Hello,

 

The question is not quite clear. In your opensips.cfg you call
fix_nated_contact() on both route and reply route. that's why it changes
the route.

 

There is no such a thing as direction in opensips unless it's implemented in
your logic. A message goes through your route or reply route scripts when it
reaches opensips. 

 

Regards,

Ali

 

On Fri, Nov 22, 2013 at 1:10 AM, dpa denis7...@mail.ru wrote:



Hello

 

I have a one question about nat processing of Opensips.

 

There is such scheme

 

UAC1 (softphone behind nat) à Opensips - UAC2 (another softswitch), i.e.
UAC1 initiates a call to Opensips and a signaling port = 5068.

 

So UAC2 becomes “ringing” by sending 183 message.

 

In attachment 183 message from UAC2 and some parts of opensips.cfg

 

After 183 processing by Opensips port in Contact header of 183 (and later
200 OK) messages become 5060, i.e. Opensips detects NAT and changes Contact
header.

 

So my question is, Why does Opensips changes Contact header?

Once Opensips detects nat transaction (setting setflag(21)) is it check all
reply messages (and doesn`t matter from which UAC they have been received)
or Opensips can detect direction and makes decision about nat process?

 

Thank you  for any help.

 

 

 

 

 


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Re: [OpenSIPS-Users] Opensips 1.9.1 and NAT

2013-11-25 Thread dpa
OK, but in the current case another question.

Is nat_bflag appear in transaction when call goes TO uac behind nat (by
using lookup(), for example), or it appears  in transaction when call goes
FROM uac behind nat?

 

Thank you.

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Ali Pey
Sent: Monday, November 25, 2013 8:38 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Opensips 1.9.1 and NAT

 

You also examining isbflagset(10). Isn't that set?

 

Regards,

Ali Pey

 

On Mon, Nov 25, 2013 at 3:09 AM, dpa denis7...@mail.ru wrote:

Hello

 

I understand but in onreply route I make a test: nat_uac_test(55) and only
if it successful I make “fix_nated_contact()”.

In my case nat_uac_test(55) must be fail after checking 183 ringing and
200 OK.

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Ali Pey
Sent: Friday, November 22, 2013 5:37 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Opensips 1.9.1 and NAT

 

Hello,

 

The question is not quite clear. In your opensips.cfg you call
fix_nated_contact() on both route and reply route. that's why it changes
the route.

 

There is no such a thing as direction in opensips unless it's implemented in
your logic. A message goes through your route or reply route scripts when it
reaches opensips. 

 

Regards,

Ali

 

On Fri, Nov 22, 2013 at 1:10 AM, dpa denis7...@mail.ru wrote:



Hello

 

I have a one question about nat processing of Opensips.

 

There is such scheme

 

UAC1 (softphone behind nat) à Opensips - UAC2 (another softswitch), i.e.
UAC1 initiates a call to Opensips and a signaling port = 5068.

 

So UAC2 becomes “ringing” by sending 183 message.

 

In attachment 183 message from UAC2 and some parts of opensips.cfg

 

After 183 processing by Opensips port in Contact header of 183 (and later
200 OK) messages become 5060, i.e. Opensips detects NAT and changes Contact
header.

 

So my question is, Why does Opensips changes Contact header?

Once Opensips detects nat transaction (setting setflag(21)) is it check all
reply messages (and doesn`t matter from which UAC they have been received)
or Opensips can detect direction and makes decision about nat process?

 

Thank you  for any help.

 

 

 

 

 


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Re: [OpenSIPS-Users] Opensips 1.9.1 and NAT

2013-11-25 Thread dpa
“What is it that you are trying to do?”

 

I try to check if message came from behind NAT or not and make some actions
depending on result of checking.

To speak the truth I tried to include in opensips.cfg logic about direction
of the call.

 

What I thought. I thought that nat_bflag activated only after lookup()
function processing, i.e. when call goes TO UAC behind NAT and NOT activated
when call came FROM UAC behind NAT. So in onreply route I make the test
about this flag and waiting (in my current case) that the test of natb_flag
will be failed.

 

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Ali Pey
Sent: Tuesday, November 26, 2013 8:16 AM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Opensips 1.9.1 and NAT

 

As I said earlier, there is no such a thing as FROM or TO (i.e. direction)
unless it's implemented in your logic. When a SIP request reaches the IP
address of your opensips server, it goes through your routing script in
opensips.cfg and then it's routed out. What you have there, it would apply
to the request message. Put yourself in opensips shoes and follow the logic
- that's how we debugged code while at uni :)

 

If you have a concept of direction and you need different behavior based on
your concept of direction, you need to implement it in your logic. For
instance you can examine the source IP or subnet to decide if this message
is from internal or external and then apply different logic to it - or
whatever else that is specific to your environment. 

 

What is it that you are trying to do?

 

Regards,

Ali Pey

 

 

On Mon, Nov 25, 2013 at 10:52 PM, dpa denis7...@mail.ru wrote:

OK, but in the current case another question.

Is nat_bflag appear in transaction when call goes TO uac behind nat (by
using lookup(), for example), or it appears  in transaction when call goes
FROM uac behind nat?

 

Thank you.

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Ali Pey
Sent: Monday, November 25, 2013 8:38 PM


To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Opensips 1.9.1 and NAT

 

You also examining isbflagset(10). Isn't that set?

 

Regards,

Ali Pey

 

On Mon, Nov 25, 2013 at 3:09 AM, dpa denis7...@mail.ru wrote:

Hello

 

I understand but in onreply route I make a test: nat_uac_test(55) and only
if it successful I make “fix_nated_contact()”.

In my case nat_uac_test(55) must be fail after checking 183 ringing and
200 OK.

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Ali Pey
Sent: Friday, November 22, 2013 5:37 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Opensips 1.9.1 and NAT

 

Hello,

 

The question is not quite clear. In your opensips.cfg you call
fix_nated_contact() on both route and reply route. that's why it changes
the route.

 

There is no such a thing as direction in opensips unless it's implemented in
your logic. A message goes through your route or reply route scripts when it
reaches opensips. 

 

Regards,

Ali

 

On Fri, Nov 22, 2013 at 1:10 AM, dpa denis7...@mail.ru wrote:



Hello

 

I have a one question about nat processing of Opensips.

 

There is such scheme

 

UAC1 (softphone behind nat) à Opensips - UAC2 (another softswitch), i.e.
UAC1 initiates a call to Opensips and a signaling port = 5068.

 

So UAC2 becomes “ringing” by sending 183 message.

 

In attachment 183 message from UAC2 and some parts of opensips.cfg

 

After 183 processing by Opensips port in Contact header of 183 (and later
200 OK) messages become 5060, i.e. Opensips detects NAT and changes Contact
header.

 

So my question is, Why does Opensips changes Contact header?

Once Opensips detects nat transaction (setting setflag(21)) is it check all
reply messages (and doesn`t matter from which UAC they have been received)
or Opensips can detect direction and makes decision about nat process?

 

Thank you  for any help.

 

 

 

 

 


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[OpenSIPS-Users] Opensips 1.9.1 and NAT

2013-11-21 Thread dpa


Hello

 

I have a one question about nat processing of Opensips.

 

There is such scheme

 

UAC1 (softphone behind nat) à Opensips - UAC2 (another softswitch), i.e.
UAC1 initiates a call to Opensips and a signaling port = 5068.

 

So UAC2 becomes “ringing” by sending 183 message.

 

In attachment 183 message from UAC2 and some parts of opensips.cfg

 

After 183 processing by Opensips port in Contact header of 183 (and later
200 OK) messages become 5060, i.e. Opensips detects NAT and changes Contact
header.

 

So my question is, Why does Opensips changes Contact header?

Once Opensips detects nat transaction (setting setflag(21)) is it check all
reply messages (and doesn`t matter from which UAC they have been received)
or Opensips can detect direction and makes decision about nat process?

 

Thank you  for any help.

 

 

 

 

 

image001.gifU 2013/11/22 08:56:58.606648 2.2.2.2:5060 - 1.1.1.1:5068
SIP/2.0 183 Progress.
Via: SIP/2.0/UDP 1.1.1.1:5068;branch=z9hG4bK8b6b.d794d636.0.
Via: SIP/2.0/UDP 
192.168.18.150:41314;rport=43238;received=3.3.3.3;branch=z9hG4bK-d8754z-37977264babb1a1c-1---d8754z-.
Record-Route: sip:AQEAEBU0TSMfDQMFY3/lbBsPg80DAAQUm+/P@2.2.2.2;lr.
Record-Route: sip:AQEAEL8TN9Pcghi+IDj0gLMRnv4DAAR5OlYR@2.2.2.2;lr.
Record-Route: sip:1.1.1.1:5068;lr;ftag=9b44a312;did=486.b9e72ff6.
From: sip:8172264014@1.1.1.1:5068;transport=UDP;tag=9b44a312.
To: 
sip:89213039240@1.1.1.1:5068;transport=UDP;tag=8maN3eG8u*ECE8XhvLvxAhPEyZHydAAq.
Call-ID: YWZlNGYwMTcwNmMxZjE3MGYwNGUxNGQ3NzY4MTEwZTU..
CSeq: 2 INVITE.
Contact: sip:79213039240@2.2.2.2:5061;transport=UDP.
Server: TS-v4.6.0-09d.
Content-Length: 0.

2.2.2.2 - UAC2
1.1.1.1 - Opensips
3.3.3.3 - NAT

 Opensips.cfg ***

modparam(usrloc, nat_bflag, 10)

Route [2] {
…
  if ($Rp==5068) {
 if (nat_uac_test(55)) {
 fix_nated_contact();
 setflag(21);
 }
  }
…
}

onreply_route[1] {
….
if (isbflagset(10) || (isflagset(21)  nat_uac_test(55))) 
fix_nated_contact();
….
}
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[OpenSIPS-Users] rtpproxy logging shut off

2013-09-25 Thread dpa


Hello!

 

I am using rtpproxy with Opensips for SIP connections.

 

Now rtpproxy makes a log to syslog during call establishment.

 

The question is may I shut off this process for rtpproxy working without
logging?

 

Thank you from any help.

 

 

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Re: [OpenSIPS-Users] rtpproxy logging shut off

2013-09-25 Thread dpa
I understand, thank you

-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of kamika
Sent: Wednesday, September 25, 2013 3:48 PM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] rtpproxy logging shut off

you have to change options that you start RTPProxy with. See -d flag

e.g.: -d CRIT:LOG_LOCAL5 





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Re: [OpenSIPS-Users] rtpproxy logging shut off

2013-09-25 Thread dpa
Hello

I made the change (-d INFO:LOG_LOCAL5) but log from rtpproxy is still in
syslog.1 file.
In config file of syslog there is no mentioned about local7 log_facility.


-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of kamika
Sent: Wednesday, September 25, 2013 3:48 PM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] rtpproxy logging shut off

you have to change options that you start RTPProxy with. See -d flag

e.g.: -d CRIT:LOG_LOCAL5 





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View this message in context:
http://opensips-open-sip-server.1449251.n2.nabble.com/rtpproxy-logging-shut-
off-tp7587893p7587894.html
Sent from the OpenSIPS - Users mailing list archive at Nabble.com.

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[OpenSIPS-Users] Problem with DB migration from 1.8.2 to 1.9.1

2013-08-13 Thread dpa


Hello!

 

Now I am using Opensips 1.6.4-2 but decide to upgrade ones to 1.9.1

First of all I try to migrate MySQL DB.

Migrate from 1.6.4.2 to 1.7.0 - OK

Migrate from 1.7.0 to 1.8.2 - OK

Migrate from 1.8.2 to 1.9.1 - OK except one table dr_carriers

 

During migration process from 1.8 to 1.9  i see such error

ERROR: failed to migrate opensips8.dr_carriers to opensips9.dr_carriers!!!

Skip it and continue (y/n)? 

 

Other tables had been migrated successfully

 

Thank you for any help

 

 

 

 

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Re: [OpenSIPS-Users] Database Record

2013-05-08 Thread dpa
acc_db_request inserts information in a table that you mentioned in second
parameter of the function with some comments mentioned in first parameter.

For example, acc_db_request(some information, acc).

The function works only in request and failure route. 

If you want put some information in acc table before call hang up then you
should make acc_db_request(some information, acc) in any request route
where INVITE processed.

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of M.Khaled W Chehab
Sent: Wednesday, May 08, 2013 2:27 PM
To: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] Database Record

 

version: opensips 1.8.2-notls (x86_64/linux)

flags: STATS: Off, EXTRA_DEBUG, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST,
SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT

ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535

poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.

svnrevision: 2:9954M

@(#) $Id: main.c 9791 2013-02-15 10:15:25Z bogdan_iancu $

main.c compiled on 15:29:28 Apr 18 2013 with gcc 4.4.5

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of dpa
Sent: Wednesday, May 08, 2013 7:48 AM
To: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] Database Record

 

The version of Opensips?

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of M.Khaled W Chehab
Sent: Tuesday, May 07, 2013 4:52 PM
To: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] Database Record

 

Hello,

 

Please can you make it more clear or give me an example  for the function
acc_log_request(Some comment, Some table);

 

onreply_route[2] {

   

   if (is_method(INVITE)  t_check_status(200) ) {

   

acc_log_request(coonect time ,acc); ???

  }

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of dpa
Sent: Tuesday, May 07, 2013 1:06 PM
To: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] Database Record

 

Hello

 

Try http://www.opensips.org/html/docs/modules/1.6.x/acc.html#id293991

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of M.Khaled W Chehab
Sent: Tuesday, May 07, 2013 2:01 PM
To: users@lists.opensips.org
Subject: [OpenSIPS-Users] Database Record

 

Dears,

 

I am using acc module and I am inserting two rows for each call (INVITE,BYE)
,

I have a problem such I want to update the  record in acc table before the
calls ends which have a method = INVITE, and now that cant be done now
since opensips 

Does not insert the invite record before the calls ends 

 

How to let the invite  record inserted in database before calls ends

 

 

Please advice 

 

 

Khaled Chehab

Senior NGN Engineer

Description: icucall

Operations Office - Lebanon

Office: +961 1 515155 ext 300

Mobile  : +961 3 045212

E-mail: kche...@icucall.com

MSN ID :khalidche...@hotmail.com 

Skype: k_chehab 

Web Site: http://www.icucall.com

 http://www.allohi.com

 

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Re: [OpenSIPS-Users] Database Record

2013-05-07 Thread dpa
The version of Opensips?

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of M.Khaled W Chehab
Sent: Tuesday, May 07, 2013 4:52 PM
To: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] Database Record

 

Hello,

 

Please can you make it more clear or give me an example  for the function
acc_log_request(Some comment, Some table);

 

onreply_route[2] {

   

   if (is_method(INVITE)  t_check_status(200) ) {

   

acc_log_request(coonect time ,acc); ???

  }

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of dpa
Sent: Tuesday, May 07, 2013 1:06 PM
To: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] Database Record

 

Hello

 

Try http://www.opensips.org/html/docs/modules/1.6.x/acc.html#id293991

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of M.Khaled W Chehab
Sent: Tuesday, May 07, 2013 2:01 PM
To: users@lists.opensips.org
Subject: [OpenSIPS-Users] Database Record

 

Dears,

 

I am using acc module and I am inserting two rows for each call (INVITE,BYE)
,

I have a problem such I want to update the  record in acc table before the
calls ends which have a method = INVITE, and now that cant be done now
since opensips 

Does not insert the invite record before the calls ends 

 

How to let the invite  record inserted in database before calls ends

 

 

Please advice 

 

 

Khaled Chehab

Senior NGN Engineer

Description: icucall

Operations Office - Lebanon

Office: +961 1 515155 ext 300

Mobile  : +961 3 045212

E-mail: kche...@icucall.com

MSN ID :khalidche...@hotmail.com 

Skype: k_chehab 

Web Site: http://www.icucall.com

 http://www.allohi.com

 

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[OpenSIPS-Users] Opensips 1.6.4-2 call handling problem

2013-05-06 Thread dpa


Hello! 

 

There is  Opensips 1.6.4-2.

Today I found a problem that Opensips  hasn`t been responded to any SIP
message that has been sent to it for a sometime.

 

I began to learn log file and found such messages: 

rc_ip_hostname: couldn't look up host by addr: xx

May  6 09:50:17 opensips-main
/usr/local/opensips1.6.4-2/sbin/opensips[2146]: rc_send_server: no reply
from RADIUS server unknown:1812

May  6 09:50:17 opensips-main
/usr/local/opensips1.6.4-2/sbin/opensips[2146]:
ERROR:aaa_radius:send_auth_func: radius authentication message failed with
TIMEOUT

 

These messages appeared in log file somewhere about at that time while
Opensips hasn`t been responded to SIP messages. 

The fact is I am using radius for prepaid scheme which I apply to some
subscribers (but not for all). 

 

May be it is just coincidence but my question is.  May radius problem affect
normal Opensips working?

 

Thank you for any help.

 

 

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Re: [OpenSIPS-Users] opensips 1.6.4-2 and dlg_end_dlg

2012-12-20 Thread dpa
Hello

 

Yes the problem was there. In local route I had exec function to do
dlg_end_dlg. Thank you very much.

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Thursday, December 20, 2012 10:40 AM
To: dpa
Cc: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] opensips 1.6.4-2 and dlg_end_dlg

 

yes, it is, as when you do dlg_end_dlg, opensips will generate and send 2
BYEs (one to caller, one to callee); and each time opensips sends a locally
generated request, local route is triggered.

Regards,



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


On 12/20/2012 10:34 AM, dpa wrote: 

Bogdan, as I understand when I am doing fifo dlg_end_dlg through FIFO
command from console the config of  local route is activated? 

 

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Thursday, December 20, 2012 9:57 AM
To: dpa; users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] opensips 1.6.4-2 and dlg_end_dlg

 

Hi,

You are doing something wrong there - based on the script, it seems you do
dlg_end_dlg, which triggers sending BYE - local route - you do an exec
there to do again dlg_end_dlg. So you try to do nested FIFO commands which
does not work - FIFO engine will not read a new command until the current is
not finished.

Regards,





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


On 12/20/2012 07:36 AM, dpa wrote: 

Hello Bogdan

 

The gdb  information in attachment.

Unfortunately for some unknown reason I cannot see logs. I will check the
problem and write later.

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Wednesday, December 19, 2012 11:17 AM
To: dpa
Cc: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] opensips 1.6.4-2 and dlg_end_dlg

 

Try to set debug=4 and see the logs from the MI fifo process when running
the dlg_end_dlg command.

Also, try the get a backtrace from the MO FIFO proc after blocking. For
that, before the dlg command, do opensipsctl fifo ps to find out the PID
of the FIFO process.
After running the dlg command and blocking, use gdb /path/to/opensips
FIFO_PID and run bt inside - these will give some ideas on where the
process blocked.

Regards,






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


On 12/19/2012 10:07 AM, dpa wrote: 

 So only the MI fifo part blocks?

As I understand, yes.

 

do you see any errors in the logs?

 

I see only INFO:core:buf_init: initializing... in log file

 

also, does it recover after some time (like minutes) ?

To speak the truth I didn`t wait recovering. I used Ctrl+Z to stop process
of dlg_end_dlg and tried another fifo command. ps -A show me that
opensipsctl process appears.

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Wednesday, December 19, 2012 9:44 AM
To: dpa
Cc: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] opensips 1.6.4-2 and dlg_end_dlg

 

So only the MI fifo part blocks ? do you see any errors in the logs? also,
does it recover after some time (like minutes) ?

Regards,






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


On 12/19/2012 09:42 AM, dpa wrote: 

Hello Bogdan

 

it simply blocks and does not answer

I can make calls. But no fifo command available.

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Wednesday, December 19, 2012 9:39 AM
To: OpenSIPS users mailling list
Cc: dpa
Subject: Re: [OpenSIPS-Users] opensips 1.6.4-2 and dlg_end_dlg

 

Hi,

Does your opensips actually crashes when you run the dlg_end_dlg command ?
or it simply blocks and does not answer ?

Regards,







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


On 12/19/2012 07:37 AM, dpa wrote: 



Hello!

 

I discovered a problem with using dlg_end_dlg fifo command.

When I try to make it, dialog hasn't been ended and I cannot see in console
that command has been finished success. In /tmp directory (where
opensips_fifo is located) I found some new files opensips_receiver_x,
where x - some numbers. Ls -la shows that files were generated when I
make dlg_end_dlg command.

Moreover after this I cannot make any fifo command. 

After opensips restart I can make any fifo command but until dlg_end_dlg.

 

There were no problems with it but recently I decided to make opensips
generates core file if it crashes.

I use this documentation http://www.opensips.org/Resources/DocsTsCrash. 

I used  -w option and make changes in /proc/sys/kernel/core_pattern. May
be some problems with it?

 

Thank you for any help! 

 

 

 
 
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Re: [OpenSIPS-Users] opensips 1.6.4-2 and dlg_end_dlg

2012-12-19 Thread dpa
Bogdan, as I understand when I am doing fifo dlg_end_dlg through FIFO
command from console the config of  local route is activated? 

 

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Thursday, December 20, 2012 9:57 AM
To: dpa; users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] opensips 1.6.4-2 and dlg_end_dlg

 

Hi,

You are doing something wrong there - based on the script, it seems you do
dlg_end_dlg, which triggers sending BYE - local route - you do an exec
there to do again dlg_end_dlg. So you try to do nested FIFO commands which
does not work - FIFO engine will not read a new command until the current is
not finished.

Regards,




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


On 12/20/2012 07:36 AM, dpa wrote: 

Hello Bogdan

 

The gdb  information in attachment.

Unfortunately for some unknown reason I cannot see logs. I will check the
problem and write later.

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Wednesday, December 19, 2012 11:17 AM
To: dpa
Cc: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] opensips 1.6.4-2 and dlg_end_dlg

 

Try to set debug=4 and see the logs from the MI fifo process when running
the dlg_end_dlg command.

Also, try the get a backtrace from the MO FIFO proc after blocking. For
that, before the dlg command, do opensipsctl fifo ps to find out the PID
of the FIFO process.
After running the dlg command and blocking, use gdb /path/to/opensips
FIFO_PID and run bt inside - these will give some ideas on where the
process blocked.

Regards,





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


On 12/19/2012 10:07 AM, dpa wrote: 

 So only the MI fifo part blocks?

As I understand, yes.

 

do you see any errors in the logs?

 

I see only INFO:core:buf_init: initializing... in log file

 

also, does it recover after some time (like minutes) ?

To speak the truth I didn`t wait recovering. I used Ctrl+Z to stop process
of dlg_end_dlg and tried another fifo command. ps -A show me that
opensipsctl process appears.

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Wednesday, December 19, 2012 9:44 AM
To: dpa
Cc: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] opensips 1.6.4-2 and dlg_end_dlg

 

So only the MI fifo part blocks ? do you see any errors in the logs? also,
does it recover after some time (like minutes) ?

Regards,





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


On 12/19/2012 09:42 AM, dpa wrote: 

Hello Bogdan

 

it simply blocks and does not answer

I can make calls. But no fifo command available.

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Wednesday, December 19, 2012 9:39 AM
To: OpenSIPS users mailling list
Cc: dpa
Subject: Re: [OpenSIPS-Users] opensips 1.6.4-2 and dlg_end_dlg

 

Hi,

Does your opensips actually crashes when you run the dlg_end_dlg command ?
or it simply blocks and does not answer ?

Regards,






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


On 12/19/2012 07:37 AM, dpa wrote: 



Hello!

 

I discovered a problem with using dlg_end_dlg fifo command.

When I try to make it, dialog hasn't been ended and I cannot see in console
that command has been finished success. In /tmp directory (where
opensips_fifo is located) I found some new files opensips_receiver_x,
where x - some numbers. Ls -la shows that files were generated when I
make dlg_end_dlg command.

Moreover after this I cannot make any fifo command. 

After opensips restart I can make any fifo command but until dlg_end_dlg.

 

There were no problems with it but recently I decided to make opensips
generates core file if it crashes.

I use this documentation http://www.opensips.org/Resources/DocsTsCrash. 

I used  -w option and make changes in /proc/sys/kernel/core_pattern. May
be some problems with it?

 

Thank you for any help! 

 

 

 
 
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[OpenSIPS-Users] opensips 1.6.4-2 and dlg_end_dlg

2012-12-18 Thread dpa


Hello!

 

I discovered a problem with using dlg_end_dlg fifo command.

When I try to make it, dialog hasn't been ended and I cannot see in console
that command has been finished success. In /tmp directory (where
opensips_fifo is located) I found some new files opensips_receiver_x,
where x - some numbers. Ls -la shows that files were generated when I
make dlg_end_dlg command.

Moreover after this I cannot make any fifo command. 

After opensips restart I can make any fifo command but until dlg_end_dlg.

 

There were no problems with it but recently I decided to make opensips
generates core file if it crashes.

I use this documentation http://www.opensips.org/Resources/DocsTsCrash. 

I used  -w option and make changes in /proc/sys/kernel/core_pattern. May
be some problems with it?

 

Thank you for any help! 

 

 

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Re: [OpenSIPS-Users] opensips 1.6.4-2 and dlg_end_dlg

2012-12-18 Thread dpa


And one more, in log file I can see INFO:core:buf_init: initializing...
while I make dlg_end_dlg command

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of dpa
Sent: Wednesday, December 19, 2012 7:38 AM
To: 'OpenSIPS users mailling list'
Subject: [OpenSIPS-Users] opensips 1.6.4-2 and dlg_end_dlg

 

Hello!

 

I discovered a problem with using dlg_end_dlg fifo command.

When I try to make it, dialog hasn't been ended and I cannot see in console
that command has been finished success. In /tmp directory (where
opensips_fifo is located) I found some new files opensips_receiver_x,
where x - some numbers. Ls -la shows that files were generated when I
make dlg_end_dlg command.

Moreover after this I cannot make any fifo command. 

After opensips restart I can make any fifo command but until dlg_end_dlg.

 

There were no problems with it but recently I decided to make opensips
generates core file if it crashes.

I use this documentation http://www.opensips.org/Resources/DocsTsCrash. 

I used  -w option and make changes in /proc/sys/kernel/core_pattern. May
be some problems with it?

 

Thank you for any help! 

 

 

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Re: [OpenSIPS-Users] opensips 1.6.4-2 and dlg_end_dlg

2012-12-18 Thread dpa
Hello Bogdan

 

it simply blocks and does not answer

I can make calls. But no fifo command available.

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Wednesday, December 19, 2012 9:39 AM
To: OpenSIPS users mailling list
Cc: dpa
Subject: Re: [OpenSIPS-Users] opensips 1.6.4-2 and dlg_end_dlg

 

Hi,

Does your opensips actually crashes when you run the dlg_end_dlg command ?
or it simply blocks and does not answer ?

Regards,



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


On 12/19/2012 07:37 AM, dpa wrote: 



Hello!

 

I discovered a problem with using dlg_end_dlg fifo command.

When I try to make it, dialog hasn't been ended and I cannot see in console
that command has been finished success. In /tmp directory (where
opensips_fifo is located) I found some new files opensips_receiver_x,
where x - some numbers. Ls -la shows that files were generated when I
make dlg_end_dlg command.

Moreover after this I cannot make any fifo command. 

After opensips restart I can make any fifo command but until dlg_end_dlg.

 

There were no problems with it but recently I decided to make opensips
generates core file if it crashes.

I use this documentation http://www.opensips.org/Resources/DocsTsCrash. 

I used  -w option and make changes in /proc/sys/kernel/core_pattern. May
be some problems with it?

 

Thank you for any help! 

 

 

 
 
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Re: [OpenSIPS-Users] opensips 1.6.4-2 and dlg_end_dlg

2012-12-18 Thread dpa
 So only the MI fifo part blocks?

As I understand, yes.

 

do you see any errors in the logs?

 

I see only INFO:core:buf_init: initializing... in log file

 

also, does it recover after some time (like minutes) ?

To speak the truth I didn`t wait recovering. I used Ctrl+Z to stop process
of dlg_end_dlg and tried another fifo command. ps -A show me that
opensipsctl process appears.

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Wednesday, December 19, 2012 9:44 AM
To: dpa
Cc: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] opensips 1.6.4-2 and dlg_end_dlg

 

So only the MI fifo part blocks ? do you see any errors in the logs? also,
does it recover after some time (like minutes) ?

Regards,



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


On 12/19/2012 09:42 AM, dpa wrote: 

Hello Bogdan

 

it simply blocks and does not answer

I can make calls. But no fifo command available.

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Wednesday, December 19, 2012 9:39 AM
To: OpenSIPS users mailling list
Cc: dpa
Subject: Re: [OpenSIPS-Users] opensips 1.6.4-2 and dlg_end_dlg

 

Hi,

Does your opensips actually crashes when you run the dlg_end_dlg command ?
or it simply blocks and does not answer ?

Regards,




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


On 12/19/2012 07:37 AM, dpa wrote: 



Hello!

 

I discovered a problem with using dlg_end_dlg fifo command.

When I try to make it, dialog hasn't been ended and I cannot see in console
that command has been finished success. In /tmp directory (where
opensips_fifo is located) I found some new files opensips_receiver_x,
where x - some numbers. Ls -la shows that files were generated when I
make dlg_end_dlg command.

Moreover after this I cannot make any fifo command. 

After opensips restart I can make any fifo command but until dlg_end_dlg.

 

There were no problems with it but recently I decided to make opensips
generates core file if it crashes.

I use this documentation http://www.opensips.org/Resources/DocsTsCrash. 

I used  -w option and make changes in /proc/sys/kernel/core_pattern. May
be some problems with it?

 

Thank you for any help! 

 

 

 
 
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Re: [OpenSIPS-Users] opensips 1.6.4-2 and dlg_end_dlg

2012-12-18 Thread dpa
ОК, i will try tomorrow morning.

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Wednesday, December 19, 2012 11:17 AM
To: dpa
Cc: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] opensips 1.6.4-2 and dlg_end_dlg

 

Try to set debug=4 and see the logs from the MI fifo process when running
the dlg_end_dlg command.

Also, try the get a backtrace from the MO FIFO proc after blocking. For
that, before the dlg command, do opensipsctl fifo ps to find out the PID
of the FIFO process.
After running the dlg command and blocking, use gdb /path/to/opensips
FIFO_PID and run bt inside - these will give some ideas on where the
process blocked.

Regards,




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


On 12/19/2012 10:07 AM, dpa wrote: 

 So only the MI fifo part blocks?

As I understand, yes.

 

do you see any errors in the logs?

 

I see only INFO:core:buf_init: initializing... in log file

 

also, does it recover after some time (like minutes) ?

To speak the truth I didn`t wait recovering. I used Ctrl+Z to stop process
of dlg_end_dlg and tried another fifo command. ps -A show me that
opensipsctl process appears.

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Wednesday, December 19, 2012 9:44 AM
To: dpa
Cc: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] opensips 1.6.4-2 and dlg_end_dlg

 

So only the MI fifo part blocks ? do you see any errors in the logs? also,
does it recover after some time (like minutes) ?

Regards,




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


On 12/19/2012 09:42 AM, dpa wrote: 

Hello Bogdan

 

it simply blocks and does not answer

I can make calls. But no fifo command available.

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Wednesday, December 19, 2012 9:39 AM
To: OpenSIPS users mailling list
Cc: dpa
Subject: Re: [OpenSIPS-Users] opensips 1.6.4-2 and dlg_end_dlg

 

Hi,

Does your opensips actually crashes when you run the dlg_end_dlg command ?
or it simply blocks and does not answer ?

Regards,





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


On 12/19/2012 07:37 AM, dpa wrote: 



Hello!

 

I discovered a problem with using dlg_end_dlg fifo command.

When I try to make it, dialog hasn't been ended and I cannot see in console
that command has been finished success. In /tmp directory (where
opensips_fifo is located) I found some new files opensips_receiver_x,
where x - some numbers. Ls -la shows that files were generated when I
make dlg_end_dlg command.

Moreover after this I cannot make any fifo command. 

After opensips restart I can make any fifo command but until dlg_end_dlg.

 

There were no problems with it but recently I decided to make opensips
generates core file if it crashes.

I use this documentation http://www.opensips.org/Resources/DocsTsCrash. 

I used  -w option and make changes in /proc/sys/kernel/core_pattern. May
be some problems with it?

 

Thank you for any help! 

 

 

 
 
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Re: [OpenSIPS-Users] Problem with timeout_avp.

2012-12-13 Thread dpa
The version of opensips?

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Dragomir Haralambiev
Sent: Thursday, December 13, 2012 12:46 PM
To: OpenSIPS users mailling list
Subject: [OpenSIPS-Users] Problem with timeout_avp.

 

Hello,

 

I try to use timeout_avp. Here is part of my opensips.cfg:

 

modparam(dialog, timeout_avp, $avp(maxtime))

...

  create_dialog();

  $avp(maxtime)=10;

 

The call is connected and I see record in dialog tabble.

After 10 sek opensips delete record from dialog table but not stop the call.

 

Where is problem?

 

Thanks in advance

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Re: [OpenSIPS-Users] Problem with timeout_avp.

2012-12-13 Thread dpa
Make change to opensips.cfg as Federico wrote.

create_dialog(B);

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Dragomir Haralambiev
Sent: Thursday, December 13, 2012 12:51 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Problem with timeout_avp.

 

Opensips 1.8.2 revision 9513.

2012/12/13 dpa denis7...@mail.ru

The version of opensips?

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Dragomir Haralambiev
Sent: Thursday, December 13, 2012 12:46 PM
To: OpenSIPS users mailling list
Subject: [OpenSIPS-Users] Problem with timeout_avp.

 

Hello,

 

I try to use timeout_avp. Here is part of my opensips.cfg:

 

modparam(dialog, timeout_avp, $avp(maxtime))

...

  create_dialog();

  $avp(maxtime)=10;

 

The call is connected and I see record in dialog tabble.

After 10 sek opensips delete record from dialog table but not stop the call.

 

Where is problem?

 

Thanks in advance


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

 

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Re: [OpenSIPS-Users] Timer process opensips1.6.4-2

2012-12-12 Thread dpa
Hello Bogdan

 

As I understand I should start opensips using opensips -w path to core
file. Now I am using opensipsctl script for starting opensips. Can -w
option be used with the script? And If cannot how can I stop opensips if
early I started with opensips -w path to core file? I tried to stop it
using opensipsctl stop but got error No PID file found.

 

Thank you for any help.

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Friday, December 07, 2012 4:15 PM
To: OpenSIPS users mailling list
Cc: dpa
Subject: Re: [OpenSIPS-Users] Timer process opensips1.6.4-2

 

See more on:
http://www.opensips.org/Resources/DocsTsCrash

Regards,



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


On 12/07/2012 12:24 PM, Bogdan-Andrei Iancu wrote: 

Hi,

maybe you do not have permissions to generate core. Be sure you do ulimits
-c unlimited before starting OpenSIPS - this will allow opensips to write a
core file if crashing.

Regards,



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


On 12/07/2012 05:45 AM, dpa wrote: 

Hello Bogdan

 

Sorry I was wrong at the beginning. In log during crash I see
INFO:core:handle_sigs: core was not generated

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Thursday, December 06, 2012 3:08 PM
To: dpa
Cc: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] Timer process opensips1.6.4-2

 

In your logs, when crashing, after the line with:
INFO:core:handle_sigs: child process 2405 exited by a signal 11
Do you have any line with 
core was generated 
or similar. ?





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


On 12/06/2012 12:57 PM, dpa wrote: 

ls -la /core* doesn't show any file

 

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Thursday, December 06, 2012 2:48 PM
To: dpa
Cc: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] Timer process opensips1.6.4-2

 

Hi, look into the root of you file system:
ls -la /core*
Regards,





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


On 12/06/2012 06:36 AM, dpa wrote: 

Hello

 

Thank you for consultation, but I cannot find any core* file in opensips
directory (/usr/local/opensips1.6.4-2/). Can you tell me how can I get it?

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Tuesday, December 04, 2012 6:37 PM
To: dpa
Cc: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] Timer process opensips1.6.4-2

 

Hi,

First locate the core file - usually in the working directory of OpenSIPS
(typically the / dir - try ls -la /core* ).

Second, start gdb :
gdb /path/to/opensips  /path/to/core

Run the bt command in gdb:
 bt
  and 
 bt full


Post the output here.

Regards,







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


On 12/04/2012 03:12 PM, dpa wrote: 

Hello 

Sorry Bogdan but can you tell me how can I use gdb? I have not made it
before.


I noticed crash while the radius server (which I use for call limit duration
request) has been failed. I use radius_send_auth() to get call limit
duration from Billing. When radius has been failing opensips didn't receive
any reply from it. Crashes have been stopped when I commented this function
in opensips.cfg.


 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Tuesday, December 04, 2012 4:49 PM
To: OpenSIPS users mailling list
Cc: dpa
Subject: Re: [OpenSIPS-Users] Timer process opensips1.6.4-2

 

Hi,

In opensips there are multiple timer processes - different modules may ask
for separate timer processes to avoid interferences with modules (in terms
of load on the timer routines). Nothing unusual here.

Regarding the crash, do you have a core file resulted ? if so, please use
gdb to extract a backtrace from it.

Thanks and Regards,







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


On 12/03/2012 08:35 AM, dpa wrote: 



Hello

 

When I use /usr/local/opensips1.6.4-2/sbin/opensipsctl fifo ps i see many
processes.

There are two of them have TYPE=timer. What does it mean?

 

Today I have a twice problem with this process.

INFO:core:handle_sigs: child process 2405 exited by a signal 11

Where 2405 was a timer process.

 

Thank you for any help.

 

 

 

 

 
 
___
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Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
 
 
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Re: [OpenSIPS-Users] Timer process opensips1.6.4-2

2012-12-12 Thread dpa
I understand, thank you.

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Wednesday, December 12, 2012 4:33 PM
To: dpa
Cc: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] Timer process opensips1.6.4-2

 

Hi,

If you use opensipsctl start/stop, you can add extra parameters to opensips
via the STARTOPTIONS variable in opensipsctlrc file.

Regards,



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


On 12/12/2012 02:04 PM, dpa wrote: 

Hello Bogdan

 

As I understand I should start opensips using opensips -w path to core
file. Now I am using opensipsctl script for starting opensips. Can -w
option be used with the script? And If cannot how can I stop opensips if
early I started with opensips -w path to core file? I tried to stop it
using opensipsctl stop but got error No PID file found.

 

Thank you for any help.

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Friday, December 07, 2012 4:15 PM
To: OpenSIPS users mailling list
Cc: dpa
Subject: Re: [OpenSIPS-Users] Timer process opensips1.6.4-2

 

See more on:
http://www.opensips.org/Resources/DocsTsCrash

Regards,




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


On 12/07/2012 12:24 PM, Bogdan-Andrei Iancu wrote: 

Hi,

maybe you do not have permissions to generate core. Be sure you do ulimits
-c unlimited before starting OpenSIPS - this will allow opensips to write a
core file if crashing.

Regards,




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


On 12/07/2012 05:45 AM, dpa wrote: 

Hello Bogdan

 

Sorry I was wrong at the beginning. In log during crash I see
INFO:core:handle_sigs: core was not generated

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Thursday, December 06, 2012 3:08 PM
To: dpa
Cc: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] Timer process opensips1.6.4-2

 

In your logs, when crashing, after the line with:
INFO:core:handle_sigs: child process 2405 exited by a signal 11
Do you have any line with 
core was generated 
or similar. ?






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


On 12/06/2012 12:57 PM, dpa wrote: 

ls -la /core* doesn't show any file

 

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Thursday, December 06, 2012 2:48 PM
To: dpa
Cc: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] Timer process opensips1.6.4-2

 

Hi, look into the root of you file system:
ls -la /core*
Regards,






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


On 12/06/2012 06:36 AM, dpa wrote: 

Hello

 

Thank you for consultation, but I cannot find any core* file in opensips
directory (/usr/local/opensips1.6.4-2/). Can you tell me how can I get it?

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Tuesday, December 04, 2012 6:37 PM
To: dpa
Cc: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] Timer process opensips1.6.4-2

 

Hi,

First locate the core file - usually in the working directory of OpenSIPS
(typically the / dir - try ls -la /core* ).

Second, start gdb :
gdb /path/to/opensips  /path/to/core

Run the bt command in gdb:
 bt
  and 
 bt full


Post the output here.

Regards,








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


On 12/04/2012 03:12 PM, dpa wrote: 

Hello 

Sorry Bogdan but can you tell me how can I use gdb? I have not made it
before.


I noticed crash while the radius server (which I use for call limit duration
request) has been failed. I use radius_send_auth() to get call limit
duration from Billing. When radius has been failing opensips didn't receive
any reply from it. Crashes have been stopped when I commented this function
in opensips.cfg.


 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Tuesday, December 04, 2012 4:49 PM
To: OpenSIPS users mailling list
Cc: dpa
Subject: Re: [OpenSIPS-Users] Timer process opensips1.6.4-2

 

Hi,

In opensips there are multiple timer processes - different modules may ask
for separate timer processes to avoid interferences with modules (in terms
of load on the timer routines). Nothing unusual here.

Regarding the crash, do you have a core file resulted ? if so, please use
gdb to extract a backtrace from it.

Thanks and Regards,








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


On 12/03/2012 08:35 AM, dpa wrote: 



Hello

 

When I use /usr/local/opensips1.6.4-2/sbin/opensipsctl fifo ps i see many
processes.

There are two of them have TYPE=timer. What does it mean?

 

Today I have a twice problem with this process.

INFO:core:handle_sigs: child process 2405 exited by a signal 11

Where 2405 was a timer process.

 

Thank you for any help.

 

 

 

 

 
 
___
Users mailing

Re: [OpenSIPS-Users] Timer process opensips1.6.4-2

2012-12-07 Thread dpa
Thank you Bogdan I will learn the Docs.

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Friday, December 07, 2012 4:15 PM
To: OpenSIPS users mailling list
Cc: dpa
Subject: Re: [OpenSIPS-Users] Timer process opensips1.6.4-2

 

See more on:
http://www.opensips.org/Resources/DocsTsCrash

Regards,



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


On 12/07/2012 12:24 PM, Bogdan-Andrei Iancu wrote: 

Hi,

maybe you do not have permissions to generate core. Be sure you do ulimits
-c unlimited before starting OpenSIPS - this will allow opensips to write a
core file if crashing.

Regards,



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


On 12/07/2012 05:45 AM, dpa wrote: 

Hello Bogdan

 

Sorry I was wrong at the beginning. In log during crash I see
INFO:core:handle_sigs: core was not generated

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Thursday, December 06, 2012 3:08 PM
To: dpa
Cc: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] Timer process opensips1.6.4-2

 

In your logs, when crashing, after the line with:
INFO:core:handle_sigs: child process 2405 exited by a signal 11
Do you have any line with 
core was generated 
or similar. ?





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


On 12/06/2012 12:57 PM, dpa wrote: 

ls -la /core* doesn't show any file

 

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Thursday, December 06, 2012 2:48 PM
To: dpa
Cc: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] Timer process opensips1.6.4-2

 

Hi, look into the root of you file system:
ls -la /core*
Regards,





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


On 12/06/2012 06:36 AM, dpa wrote: 

Hello

 

Thank you for consultation, but I cannot find any core* file in opensips
directory (/usr/local/opensips1.6.4-2/). Can you tell me how can I get it?

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Tuesday, December 04, 2012 6:37 PM
To: dpa
Cc: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] Timer process opensips1.6.4-2

 

Hi,

First locate the core file - usually in the working directory of OpenSIPS
(typically the / dir - try ls -la /core* ).

Second, start gdb :
gdb /path/to/opensips  /path/to/core

Run the bt command in gdb:
 bt
  and 
 bt full


Post the output here.

Regards,







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


On 12/04/2012 03:12 PM, dpa wrote: 

Hello 

Sorry Bogdan but can you tell me how can I use gdb? I have not made it
before.


I noticed crash while the radius server (which I use for call limit duration
request) has been failed. I use radius_send_auth() to get call limit
duration from Billing. When radius has been failing opensips didn't receive
any reply from it. Crashes have been stopped when I commented this function
in opensips.cfg.


 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Tuesday, December 04, 2012 4:49 PM
To: OpenSIPS users mailling list
Cc: dpa
Subject: Re: [OpenSIPS-Users] Timer process opensips1.6.4-2

 

Hi,

In opensips there are multiple timer processes - different modules may ask
for separate timer processes to avoid interferences with modules (in terms
of load on the timer routines). Nothing unusual here.

Regarding the crash, do you have a core file resulted ? if so, please use
gdb to extract a backtrace from it.

Thanks and Regards,







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


On 12/03/2012 08:35 AM, dpa wrote: 



Hello

 

When I use /usr/local/opensips1.6.4-2/sbin/opensipsctl fifo ps i see many
processes.

There are two of them have TYPE=timer. What does it mean?

 

Today I have a twice problem with this process.

INFO:core:handle_sigs: child process 2405 exited by a signal 11

Where 2405 was a timer process.

 

Thank you for any help.

 

 

 

 

 
 
___
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
image001.gif___
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Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users


Re: [OpenSIPS-Users] Timer process opensips1.6.4-2

2012-12-05 Thread dpa
Hello

 

Thank you for consultation, but I cannot find any core* file in opensips
directory (/usr/local/opensips1.6.4-2/). Can you tell me how can I get it?

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Tuesday, December 04, 2012 6:37 PM
To: dpa
Cc: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] Timer process opensips1.6.4-2

 

Hi,

First locate the core file - usually in the working directory of OpenSIPS
(typically the / dir - try ls -la /core* ).

Second, start gdb :
gdb /path/to/opensips  /path/to/core

Run the bt command in gdb:
 bt
  and 
 bt full


Post the output here.

Regards,




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


On 12/04/2012 03:12 PM, dpa wrote: 

Hello 

Sorry Bogdan but can you tell me how can I use gdb? I have not made it
before.


I noticed crash while the radius server (which I use for call limit duration
request) has been failed. I use radius_send_auth() to get call limit
duration from Billing. When radius has been failing opensips didn't receive
any reply from it. Crashes have been stopped when I commented this function
in opensips.cfg.


 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Tuesday, December 04, 2012 4:49 PM
To: OpenSIPS users mailling list
Cc: dpa
Subject: Re: [OpenSIPS-Users] Timer process opensips1.6.4-2

 

Hi,

In opensips there are multiple timer processes - different modules may ask
for separate timer processes to avoid interferences with modules (in terms
of load on the timer routines). Nothing unusual here.

Regarding the crash, do you have a core file resulted ? if so, please use
gdb to extract a backtrace from it.

Thanks and Regards,




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


On 12/03/2012 08:35 AM, dpa wrote: 



Hello

 

When I use /usr/local/opensips1.6.4-2/sbin/opensipsctl fifo ps i see many
processes.

There are two of them have TYPE=timer. What does it mean?

 

Today I have a twice problem with this process.

INFO:core:handle_sigs: child process 2405 exited by a signal 11

Where 2405 was a timer process.

 

Thank you for any help.

 

 

 

 

 
 
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
image001.gif___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users


Re: [OpenSIPS-Users] Install opensips 1.8.2

2012-12-04 Thread dpa
Thank you, I will check 2 point.  

 

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Muhammad Shahzad
Sent: Tuesday, December 04, 2012 4:48 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Install opensips 1.8.2

 

Sorry i think two separate issues are missed in this thread and I answered the 
second one instead of first one.

 

Anyhow, here is the answer for your original problem.

 

1. Make sure you have libmysqlclient-dev installed. If you have mysql libraries 
installed on some custom location then make sure to add them using CFLAGS and 
LDFLAGS directive.

2. Look at Makefile.conf in opensips sources and make sure db_mysql is NOT in 
exclude_modules list.

3. Do make and make install.

 

Thank you.

 

On Tue, Dec 4, 2012 at 5:18 AM, dpa denis7...@mail.ru wrote:



I can compile and install opensips with db_mysql  only make

“make prefix=/usr/local/opensips1.8.2/ all include_modules=db_mysql

make prefix=/usr/local/opensips1.8.2/ install include_modules=db_mysql”

 

 

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of dpa
Sent: Monday, December 03, 2012 1:23 PM
To: 'OpenSIPS users mailling list'
Subject: [OpenSIPS-Users] Install opensips 1.8.2

 

Hello

 

There is a desire to install opensips 1.8.2 from src.

I download it from web, change Makefile in “exclude_modules” area by excluding 
db_mysql from the list of “exclude_modules”.

Do “make prefix=/usr/local/opensips1.8.2/ all”. After that I look at 
modules/db_mysql directory and cannot see any db_mysql.so.

So  if I make “make prefix=/usr/local/opensips1.8.2/ install” i cannot see any 
db_mysql.so in “/usr/local/opensips1.8.2/lib/opensips/modules”

 

I tried to do “make prefix=/usr/local/opensips1.8.2/ install 
include_modules=modules/mysql” but problem is still exists.

The only way to compile db_mysql is to do “make” from “modules/db_mysql” 
directory.

 

My goal is to migrate DB from 1.7.0 to 1.8.2 and as I understand I need to 
compile db_mysql from general directory but not from module directory to make 
opensipsctl support mysql.

 

Thank you for any help.

 

 


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Re: [OpenSIPS-Users] Timer process opensips1.6.4-2

2012-12-04 Thread dpa
Hello 

Sorry Bogdan but can you tell me how can I use gdb? I have not made it
before.


I noticed crash while the radius server (which I use for call limit duration
request) has been failed. I use radius_send_auth() to get call limit
duration from Billing. When radius has been failing opensips didn't receive
any reply from it. Crashes have been stopped when I commented this function
in opensips.cfg.


 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Tuesday, December 04, 2012 4:49 PM
To: OpenSIPS users mailling list
Cc: dpa
Subject: Re: [OpenSIPS-Users] Timer process opensips1.6.4-2

 

Hi,

In opensips there are multiple timer processes - different modules may ask
for separate timer processes to avoid interferences with modules (in terms
of load on the timer routines). Nothing unusual here.

Regarding the crash, do you have a core file resulted ? if so, please use
gdb to extract a backtrace from it.

Thanks and Regards,



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


On 12/03/2012 08:35 AM, dpa wrote: 



Hello

 

When I use /usr/local/opensips1.6.4-2/sbin/opensipsctl fifo ps i see many
processes.

There are two of them have TYPE=timer. What does it mean?

 

Today I have a twice problem with this process.

INFO:core:handle_sigs: child process 2405 exited by a signal 11

Where 2405 was a timer process.

 

Thank you for any help.

 

 

 

 

 
 
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[OpenSIPS-Users] Install opensips 1.8.2

2012-12-03 Thread dpa


Hello

 

There is a desire to install opensips 1.8.2 from src.

I download it from web, change Makefile in exclude_modules area by
excluding db_mysql from the list of exclude_modules.

Do make prefix=/usr/local/opensips1.8.2/ all. After that I look at
modules/db_mysql directory and cannot see any db_mysql.so.

So  if I make make prefix=/usr/local/opensips1.8.2/ install i cannot see
any db_mysql.so in /usr/local/opensips1.8.2/lib/opensips/modules

 

I tried to do make prefix=/usr/local/opensips1.8.2/ install
include_modules=modules/mysql but problem is still exists.

The only way to compile db_mysql is to do make from modules/db_mysql
directory.

 

My goal is to migrate DB from 1.7.0 to 1.8.2 and as I understand I need to
compile db_mysql from general directory but not from module directory to
make opensipsctl support mysql.

 

Thank you for any help.

 

 

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Re: [OpenSIPS-Users] Install opensips 1.8.2

2012-12-03 Thread dpa


I can compile and install opensips with db_mysql  only make

make prefix=/usr/local/opensips1.8.2/ all include_modules=db_mysql

make prefix=/usr/local/opensips1.8.2/ install include_modules=db_mysql

 

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of dpa
Sent: Monday, December 03, 2012 1:23 PM
To: 'OpenSIPS users mailling list'
Subject: [OpenSIPS-Users] Install opensips 1.8.2

 

Hello

 

There is a desire to install opensips 1.8.2 from src.

I download it from web, change Makefile in exclude_modules area by
excluding db_mysql from the list of exclude_modules.

Do make prefix=/usr/local/opensips1.8.2/ all. After that I look at
modules/db_mysql directory and cannot see any db_mysql.so.

So  if I make make prefix=/usr/local/opensips1.8.2/ install i cannot see
any db_mysql.so in /usr/local/opensips1.8.2/lib/opensips/modules

 

I tried to do make prefix=/usr/local/opensips1.8.2/ install
include_modules=modules/mysql but problem is still exists.

The only way to compile db_mysql is to do make from modules/db_mysql
directory.

 

My goal is to migrate DB from 1.7.0 to 1.8.2 and as I understand I need to
compile db_mysql from general directory but not from module directory to
make opensipsctl support mysql.

 

Thank you for any help.

 

 

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[OpenSIPS-Users] Timer process opensips1.6.4-2

2012-12-02 Thread dpa


Hello

 

When I use /usr/local/opensips1.6.4-2/sbin/opensipsctl fifo ps i see many
processes.

There are two of them have TYPE=timer. What does it mean?

 

Today I have a twice problem with this process.

INFO:core:handle_sigs: child process 2405 exited by a signal 11

Where 2405 was a timer process.

 

Thank you for any help.

 

 

 

 

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Re: [OpenSIPS-Users] DB migration from 1.6.4 to 1.7.0

2012-11-30 Thread dpa
Yes, problem was there. Thank you very much.

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Friday, November 30, 2012 1:05 PM
To: dpa
Cc: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] DB migration from 1.6.4 to 1.7.0

 

Assuming that you have other files in that dir
(/usr/local/opensips1.7.0//lib/opensips/opensipsctl/), I suppose you did not
include mysql module in the list of modules to be compiled + installed -
the mysql support in opensipsctl was not installed too.

So do make install include_modules=modules/mysql. 

Regards,




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


On 11/30/2012 06:19 AM, dpa wrote: 

Hello, Bogdan

 

No, there is no opensipsdbctl.mysql file in
/usr/local/opensips1.7.0//lib/opensips/opensipsctl/ 

I should make it myself?

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Thursday, November 29, 2012 9:36 PM
To: OpenSIPS users mailling list
Cc: dpa
Subject: Re: [OpenSIPS-Users] DB migration from 1.6.4 to 1.7.0

 

Hi,

and have you checked if the file
/usr/local/opensips1.7.0//lib/opensips/opensipsctl/opensipsdbctl.mysql
really exists on your system ?

Regards,





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


On 11/28/2012 10:46 AM, dpa wrote: 



Hello!

 

I want to upgrade opensips  from 1.6.4-2 to latest version (1.8).

As I understand first I should migrate my DB from 1.6.4 to 1.7.0 and then
migrate 1.7.0 to 1.8

I have installed opensips 1.7.0. Make ./opensipsdbctl migrate opensips5
opensips51 
where opensips5 - existing DB.
and receive such errors
 
ERROR: could not load the script in
/usr/local/opensips1.7.0//lib/opensips/opensipsctl/opensipsdbctl.mysql for
database engine MYSQL
ERROR: database engine not loaded - tried 'MYSQL'
 
Thank you for any help.

 

 

 
 
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Re: [OpenSIPS-Users] DB migration from 1.6.4 to 1.7.0

2012-11-29 Thread dpa
Hello, Bogdan

 

No, there is no opensipsdbctl.mysql file in
/usr/local/opensips1.7.0//lib/opensips/opensipsctl/ 

I should make it myself?

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Thursday, November 29, 2012 9:36 PM
To: OpenSIPS users mailling list
Cc: dpa
Subject: Re: [OpenSIPS-Users] DB migration from 1.6.4 to 1.7.0

 

Hi,

and have you checked if the file
/usr/local/opensips1.7.0//lib/opensips/opensipsctl/opensipsdbctl.mysql
really exists on your system ?

Regards,




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


On 11/28/2012 10:46 AM, dpa wrote: 



Hello!

 

I want to upgrade opensips  from 1.6.4-2 to latest version (1.8).

As I understand first I should migrate my DB from 1.6.4 to 1.7.0 and then
migrate 1.7.0 to 1.8

I have installed opensips 1.7.0. Make ./opensipsdbctl migrate opensips5
opensips51 
where opensips5 - existing DB.
and receive such errors
 
ERROR: could not load the script in
/usr/local/opensips1.7.0//lib/opensips/opensipsctl/opensipsdbctl.mysql for
database engine MYSQL
ERROR: database engine not loaded - tried 'MYSQL'
 
Thank you for any help.

 

 

 
 
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[OpenSIPS-Users] DB migration from 1.6.4 to 1.7.0

2012-11-28 Thread dpa


Hello!

 

I want to upgrade opensips  from 1.6.4-2 to latest version (1.8).

As I understand first I should migrate my DB from 1.6.4 to 1.7.0 and then
migrate 1.7.0 to 1.8

I have installed opensips 1.7.0. Make ./opensipsdbctl migrate opensips5
opensips51 
where opensips5 - existing DB.
and receive such errors
 
ERROR: could not load the script in
/usr/local/opensips1.7.0//lib/opensips/opensipsctl/opensipsdbctl.mysql for
database engine MYSQL
ERROR: database engine not loaded - tried 'MYSQL'
 
Thank you for any help.

 

 

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Re: [OpenSIPS-Users] extra duration of a call in 1.6.4-2

2012-11-12 Thread dpa
Hello

 

I am using “bye_on_timeout_flag” for each transaction in Opensips. The default 
timeout – 10 s. “Real timeout” rewriting after loose_routing() when routing ACK 
request on 200 ОК and different for clients. The max timeout in Opensips for 
dialogs is  s.

 

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Binan AL Halabi
Sent: Monday, November 12, 2012 1:31 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] extra duration of a call in 1.6.4-2

 



Hi Denis,

The client must send BYE to end the dialog. but in case of timeout you can 
generate the local BYE to end the dialog:
create_dialog(B);  The string B activate the BYE on timeout behavior. In 
OpenSIPS 1.6 use bye_on_timeout_flag.

 // Binan

 

 

  _  

Från: dpa denis7...@mail.ru
Till: 'OpenSIPS users mailling list' users@lists.opensips.org 
Skickat: måndag, 12 november 2012 7:30
Ämne: [OpenSIPS-Users] extra duration of a call in 1.6.4-2

 



Hello, 

 

I have detected one call  which has duration 28 seconds. I found this call 
in ngrep log (you can get it from here http://files.mail.ru/CU4GGS).

1.1.1.1– One UA

2.2.2.2 – Opensips 1.6.4-2

3.3.3.3 – Another UA

 

After analyze the log I cannot find a reason why this call has so big duration. 
In addition I control a duration of any calls and from this client the call 
shouldn`t be alive more than 1800 seconds. As you can see in log there is no 
local BYE for the session.

 

Thank you for any help

 

 


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[OpenSIPS-Users] extra duration of a call in 1.6.4-2

2012-11-11 Thread dpa


Hello, 

 

I have detected one call  which has duration 28 seconds. I found this
call in ngrep log (you can get it from here http://files.mail.ru/CU4GGS).

1.1.1.1- One UA

2.2.2.2 - Opensips 1.6.4-2

3.3.3.3 - Another UA

 

After analyze the log I cannot find a reason why this call has so big
duration. In addition I control a duration of any calls and from this client
the call shouldn`t be alive more than 1800 seconds. As you can see in log
there is no local BYE for the session.

 

Thank you for any help

 

 

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[OpenSIPS-Users] Opensips upgrade

2012-10-10 Thread dpa


Hello!

 

Sorry if it were but anyway.

I am using opensips 1.6.4-2 and have a wish to upgrade to 1.8

 

I have directory /usr/local/opensips1.6.4-2/ where opensips 1.6.4-2 has been
installed at one time.

If I will make tar archive of this directory and install to this directory
opensips 1.8? Would opensips 1.8  work after this?

(of course I will make change to opensips.cfg and sql database)

And if for some reason I will need change back opensips to 1.6.4-2? May I
make same procedure using tar archive which I made in previous step?

 

Thank you for any help.

 

P.S. It is important to using directory /usr/local/opensips1.6.4-2/ after
upgrade to 1.8.

 

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Re: [OpenSIPS-Users] Rtpproxy connection

2012-09-27 Thread dpa
You should first start opensips, it creates unix socket and then rtpproxy.


-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of spady
Sent: Thursday, September 27, 2012 4:27 PM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] Rtpproxy connection

I have some news but issue still present.
I used unix socket, instead of udp, but RTPPROXY crashes at each call.
Below traces of opensips side and rtpproxy side:

OPENSIPS:

*Sep 27 13:01:57 opensips /usr/local/opensips_proxy/sbin/opensips[26026]:
ERROR:rtpproxy:send_rtpp_command: can't read reply from a RTP proxy Sep 27
13:01:57 opensips /usr/local/opensips_proxy/sbin/opensips[26026]:
ERROR:rtpproxy:send_rtpp_command: proxy unix:/var/run/rtpproxy.sock does
not respond, disable it Sep 27 13:01:57 opensips
/usr/local/opensips_proxy/sbin/opensips[26026]:
ERROR:rtpproxy:force_rtp_proxy_body: no available proxies Sep 27 13:01:57
opensips /usr/local/opensips_proxy/sbin/opensips[26020]:
ERROR:rtpproxy:force_rtp_proxy: Unable to parse body Sep 27 13:01:57
opensips /usr/local/opensips_proxy/sbin/opensips[26021]:
ERROR:rtpproxy:force_rtp_proxy: Unable to parse body Sep 27 13:02:02
opensips /usr/local/opensips_proxy/sbin/opensips[26024]:
ERROR:rtpproxy:unforce_rtp_proxy_f: no available proxies Sep 27 13:02:02
opensips /usr/local/opensips_proxy/sbin/opensips[26022]:
ERROR:rtpproxy:force_rtp_proxy: Unable to parse body Sep 27 13:02:02
opensips /usr/local/opensips_proxy/sbin/opensips[26022]:
ERROR:rtpproxy:unforce_rtp_proxy_f: no available proxies Sep 27 13:02:02
opensips /usr/local/opensips_proxy/sbin/opensips[26022]:
ERROR:rtpproxy:engage_close_callback: cannot unforce rtp proxy Sep 27
13:02:02 opensips /usr/local/opensips_proxy/sbin/opensips[26025]:
ERROR:rtpproxy:unforce_rtp_proxy_f: no available proxies Sep 27 13:02:03
opensips /usr/local/opensips_proxy/sbin/opensips[26026]:
ERROR:rtpproxy:unforce_rtp_proxy_f: no available proxies Sep 27 13:02:05
opensips /usr/local/opensips_proxy/sbin/opensips[26024]:
ERROR:rtpproxy:unforce_rtp_proxy_f: no available proxies *


RTPPROXY:

*Sep 27 13:01:46 opensips rtpproxy[25993]: DBUG:handle_command: received
command VF 20071116
Sep 27 13:01:57 opensips rtpproxy[25993]: DBUG:handle_command: received
command UIER172.16.52.121c0,8,18,101
ZGY2Yjk5ZGJiNDQ0ZWI0MGVhZmMzODAxNDc3YzgwMjI 151.x.x.19 60242 671d9c3d;1

*

Any idea???




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Re: [OpenSIPS-Users] Rtpproxy connection

2012-09-26 Thread dpa
Hello

Instead of udp socket for control try to use unix socket. Maybe it will
help.

-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of spady
Sent: Thursday, September 27, 2012 2:41 AM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] Rtpproxy connection

- No!, after opensips module sends command to rtpproxy this one shut down (
i cannot see anymore with ps command )

- in rtpproxy logs i cannot see any specific error. Below example of code
that I get from rtpproxy:
*Sep 27 00:31:01 opensips rtpproxy[19991]: DBUG:handle_command: received
command 20126_5 UIER172.16.52.121c0,8,18,101
M2EzYzNmMmRmYzg0MGNiY2M1YjhjMTUzMWFkOWNiYTg 37.103.117.107 60234 cfd2be4f;1
Sep 27 00:31:01 opensips rtpproxy[19991]: INFO:handle_command: new session
M2EzYzNmMmRmYzg0MGNiY2M1YjhjMTUzMWFkOWNiYTg, tag cfd2be4f;1 requested, type
strong*

As you can see, no error!!!

P.S. I started rtpproxy like below:
*rtpproxy -F -s udp:127.0.0.1:10177 -l 10.9.23.41/151.x.x.201 -u root -d
DBUG:LOG_LOCAL2*



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[OpenSIPS-Users] Opensips and rtpproxy

2012-08-13 Thread dpa


Hello!

 

There is one question about Opensips and rtpproxy interaction.

Documentation says that I can use more than one rtpproxy set. And to choose
which rtpproxy to use I should write to opensips.cfg something like 

modparam(nathelper, rtpproxy_sock, 1 == /sock/rtpproxy.sock)

modparam(nathelper, rtpproxy_sock, 2 == /sock/rtpproxy1.sock)

 

And what about rtpp_notify_socket parameter? Can I write it to
opensips.cfg the same way or multiple rtpproxy can use similar unix socket
for timeout notification?

 

Opensips 1.6.4-2.

 

Thank you for any help

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Re: [OpenSIPS-Users] Opensips and rtpproxy

2012-08-13 Thread dpa
Hello!

 

And I cannot use one unix socket (timeout notification) for several rtpproxy 
set? 

 

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Razvan Crainea
Sent: Monday, August 13, 2012 4:46 PM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] Opensips and rtpproxy

 

Hello!

No, currently OpenSIPS doesn't support multiple notification sockets.

Best regards,



Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com

On 08/13/2012 01:25 PM, dpa wrote:



Hello!

 

There is one question about Opensips and rtpproxy interaction.

Documentation says that I can use more than one rtpproxy set. And to choose 
which rtpproxy to use I should write to opensips.cfg something like 

“modparam(nathelper, rtpproxy_sock, 1 == /sock/rtpproxy.sock)

modparam(nathelper, rtpproxy_sock, 2 == /sock/rtpproxy1.sock)”

 

And what about “rtpp_notify_socket” parameter? Can I write it to opensips.cfg 
the same way or multiple rtpproxy can use similar unix socket for timeout 
notification?

 

Opensips 1.6.4-2.

 

Thank you for any help






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Re: [OpenSIPS-Users] Opensips and rtpproxy

2012-08-13 Thread dpa
I understand, thank you.

 

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Razvan Crainea
Sent: Monday, August 13, 2012 5:06 PM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] Opensips and rtpproxy

 

Hello!

You can use for several sets, but only a single socket for all of them.

Regards,




Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com

On 08/13/2012 03:56 PM, dpa wrote:

Hello!

 

And I cannot use one unix socket (timeout notification) for several rtpproxy 
set? 

 

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Razvan Crainea
Sent: Monday, August 13, 2012 4:46 PM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] Opensips and rtpproxy

 

Hello!

No, currently OpenSIPS doesn't support multiple notification sockets.

Best regards,




Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com

On 08/13/2012 01:25 PM, dpa wrote:



Hello!

 

There is one question about Opensips and rtpproxy interaction.

Documentation says that I can use more than one rtpproxy set. And to choose 
which rtpproxy to use I should write to opensips.cfg something like 

“modparam(nathelper, rtpproxy_sock, 1 == /sock/rtpproxy.sock)

modparam(nathelper, rtpproxy_sock, 2 == /sock/rtpproxy1.sock)”

 

And what about “rtpp_notify_socket” parameter? Can I write it to opensips.cfg 
the same way or multiple rtpproxy can use similar unix socket for timeout 
notification?

 

Opensips 1.6.4-2.

 

Thank you for any help







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Re: [OpenSIPS-Users] rtpproxy

2012-07-25 Thread dpa
Hello Bogdan

 

http://files.mail.ru/7HLCXK - here you can get faxcall and rtpproxy
communication.

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Wednesday, July 25, 2012 8:40 AM
To: dpa
Cc: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] rtpproxy

 

Denis, if you are connecting to rtpproxy via an UDP connector, simply use
ngrep (on the configured port). If using a unixsock connector, switch to a
UDP one (rtpproxy_sock = udp:localhost:8899 and do a similar change in the
rtpproxy command line).

Regards,



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


On 07/23/2012 12:02 PM, dpa wrote: 

Hello

 

To speak the truth I do not know how to do it.  

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Monday, July 23, 2012 12:54 PM
To: dpa
Cc: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] rtpproxy

 

Hi Denis,

could you make a capture to see the comunication between opensips and
rtproxy ?

Regards,




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


On 07/20/2012 01:11 PM, dpa wrote: 

Hello Bogdan

 

I am very sorry, last time I gave wrong information about the problem. Only
with r flag there is no fax stream through rtpproxy (last email I said
that everything is OK, but I made wrong test).

Here http://files.mail.ru/XF4J0L 

Is tcpdump of unsuccessful call.

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Tuesday, July 17, 2012 11:44 AM
To: dpa
Cc: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] rtpproxy

 

Hi ,

Try only with r then.

Regards,





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


On 07/17/2012 06:26 AM, dpa wrote: 

Hello Bogdan

 

I applied r and a flags and problem with fax disappeared, but I got
another problem. After I applied ra clients behind nat cannot hear caller
or callee (never mind who makes call).

Rtpproxy is working, i.e. it changes SDP body of INVITE and replies, but
there is no rtp stream.

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Thursday, July 12, 2012 7:01 PM
To: OpenSIPS users mailling list
Cc: dpa
Subject: Re: [OpenSIPS-Users] rtpproxy

 

Hi,

According to your trace the RTPproxy is properly inserted during the
re-INVITE. Try to put the r and a flags when using RTPproxy from script.

Regards,







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


On 07/11/2012 04:32 PM, dpa wrote: 

Hello Bogdan

 

Thank you for answering. 

My case is 2).

First I have audio stream which is replaced later by another fax stream.

Here http://files.mail.ru/8BYVCQ

is a tcpdump of unsuccessful call where you can see that rtpproxy doesn`t
proxy fax stream.

Rtpproxy (as an Opensips) is located on 213.170.100.150

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
Sent: Wednesday, July 11, 2012 4:43 PM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] rtpproxy

 

Hi,

You may have 2 cases, not sure which is yours:

1) the 2 media sessions are in parallel in the same calls (you can have 2
-or more- RTP streams in the same SIP session). In this case, RTPproxy will
take care automatically of each RTP stream)

2) you have first one audio stream which is replaced by another FAX stream,
via a re-INVITE - in this case, if you the normal offer / answer for
RTPproxy also for the re-INVITE, it should be ok.

Regards,







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


On 07/11/2012 11:42 AM, dpa wrote: 

Описание: cid:part1.00030104.06050802@opensips.org

Hello

 

There is one question about rtpproxy working.

If some UA using 2 different UDP ports for sending RTP and t38 fax packets
during one call session, whether rtpproxy is working (proxy both UDP
streams) in such case? 

 

Thank you for any help

 

 
 
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
 
 
___
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Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
image001.gif___
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[OpenSIPS-Users] Opensips 1.6.4-2

2012-07-24 Thread dpa


Hello 

 

I have one question

I had such opensips.cfg

 

route {

..

create_dialog(); //INVITE processing

store_dlg_value(error,yes);

.

}

onreply_route[1] {

..

if (status=~200||18[0,3]  $rm==INVITE) {

..

store_dlg_value(error,no);

..

}

.

}

failure_route [1] {

.

fetch_dlg_value(error,$avp(i:42));

xlog(L_INFO, FAIL_route:$rs was received (IP=$si, CALLID=$ci,
FROMTAG=$ft, TOTAG=$tt, AVP(i:42)=$avp(i:42)));

..

}

 

In attachment there is log of a call. 

2.2.2.2 - Opensips. 1.1.1.1 and 3.3.3.3 - some UA. As you can see there is
no 200 or 183 replies in this call. 

 

But in log file I see

 Jul 24 15:52:20 opensips-mirror
/usr/local/opensips1.6.4-2/sbin/opensips[11522]: FAIL_route:null was
received (IP=1.1.1.1, CALLID=676e0e00-ac77-0f22-80c7-0002a4085a16@1.1.1.1,
FROMTAG=67000fc7a4, TOTAG=null, AVP(i:42)=no) 

 

The question is why does AVP(i:42) has meaning no?

 

Thank you for any help.

 

 

 

image001.gifU 2012/07/24 15:52:20.968054 1.1.1.1:5060 - 2.2.2.2:5060
INVITE sip:735603@2.2.2.2 SIP/2.0.
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK67000fc7a4216.
From: sip:4862442964@1.1.1.1;tag=67000fc7a4.
To: sip:735603@2.2.2.2.
Call-ID: 676e0e00-ac77-0f22-80c7-0002a4085a16@1.1.1.1.
CSeq: 216 INVITE.
Supported: replaces, timer, 100rel, early-session.
Min-SE: 1800.
Date: Sun, 11 Jan 2010 22:42:47 GMT.
Session-Expires: 1800.
User-Agent: AddPac SIP Gateway.
Contact: sip:4862442964@1.1.1.1.
Accept: application/sdp.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, PRACK, REFER, NOTIFY, INFO.
Content-Type: application/sdp.
Content-Length: 270.
Max-Forwards: 70.
.
v=0.
o=4862442964 945767 945767 IN IP4 1.1.1.1.
s=AddPac Gateway SDP.
c=IN IP4 1.1.1.1.
t=0 0.
m=audio 23002 RTP/AVP 18 8 0 101.
a=ptime:80.
a=rtpmap:18 G729/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.


U 2012/07/24 15:52:20.968307 2.2.2.2:5060 - 1.1.1.1:5060
SIP/2.0 100 Giving a try.
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK67000fc7a4216.
From: sip:4862442964@1.1.1.1;tag=67000fc7a4.
To: sip:735603@2.2.2.2.
Call-ID: 676e0e00-ac77-0f22-80c7-0002a4085a16@1.1.1.1.
CSeq: 216 INVITE.
Content-Length: 0.
.


U 2012/07/24 15:52:20.970247 2.2.2.2:5060 - 3.3.3.3:5060
INVITE sip:735603@3.3.3.3:5060 SIP/2.0.
Record-Route: sip:2.2.2.2;lr=on;ftag=67000fc7a4;did=4b6.ab160851.
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK8f52.5ec4773.0.
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK67000fc7a4216.
From: sip:4862442964@1.1.1.1;tag=67000fc7a4.
To: sip:735603@2.2.2.2.
Call-ID: 676e0e00-ac77-0f22-80c7-0002a4085a16@1.1.1.1.
CSeq: 216 INVITE.
Supported: replaces, timer, 100rel, early-session.
Min-SE: 1800.
Date: Sun, 11 Jan 2010 22:42:47 GMT.
Session-Expires: 1800.
User-Agent: AddPac SIP Gateway.
Contact: sip:4862442964@1.1.1.1.
P-Asserted-Identity: sip:4862442964@1.1.1.1.
Remote-Party-ID:sip:4862442964@1.1.1.1;party=calling;screen=yes;privacy=off.
Accept: application/sdp.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, PRACK, REFER, NOTIFY, INFO.
Content-Type: application/sdp.
Content-Length: 294.
Max-Forwards: 70.
.
v=0.
o=4862442964 945767 945767 IN IP4 2.2.2.2.
s=AddPac Gateway SDP.
c=IN IP4 2.2.2.2.
t=0 0.
m=audio 40834 RTP/AVP 18 8 0 101.
a=ptime:80.
a=rtpmap:18 G729/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=nortpproxy:yes.


U 2012/07/24 15:52:20.994730 3.3.3.3:5060 - 2.2.2.2:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK8f52.5ec4773.0,SIP/2.0/UDP 
1.1.1.1:5060;branch=z9hG4bK67000fc7a4216.
From: sip:4862442964@1.1.1.1;tag=67000fc7a4.
To: sip:735603@2.2.2.2.
Date: Tue, 24 Jul 2012 11:56:20 GMT.
Call-ID: 676e0e00-ac77-0f22-80c7-0002a4085a16@1.1.1.1.
CSeq: 216 INVITE.
Allow-Events: telephone-event.
Server: Cisco-SIPGateway/IOS-12.x.
Content-Length: 0.
.


U 2012/07/24 15:52:21.981394 3.3.3.3:5060 - 2.2.2.2:5060
SIP/2.0 486 Busy here.
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK8f52.5ec4773.0,SIP/2.0/UDP 
1.1.1.1:5060;branch=z9hG4bK67000fc7a4216.
From: sip:4862442964@1.1.1.1;tag=67000fc7a4.
To: sip:735603@2.2.2.2;tag=E9560618-197A.
Date: Tue, 24 Jul 2012 11:56:20 GMT.
Call-ID: 676e0e00-ac77-0f22-80c7-0002a4085a16@1.1.1.1.
CSeq: 216 INVITE.
Allow-Events: telephone-event.
Server: Cisco-SIPGateway/IOS-12.x.
Reason: Q.850;cause=17.
Content-Length: 0.
.


U 2012/07/24 15:52:21.981549 2.2.2.2:5060 - 3.3.3.3:5060
ACK sip:735603@3.3.3.3:5060 SIP/2.0.
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK8f52.5ec4773.0.
From: sip:4862442964@1.1.1.1;tag=67000fc7a4.
Call-ID: 676e0e00-ac77-0f22-80c7-0002a4085a16@1.1.1.1.
To: sip:735603@2.2.2.2;tag=E9560618-197A.
CSeq: 216 ACK.
Max-Forwards: 70.
Content-Length: 0.
.


U 2012/07/24 15:52:22.011214 2.2.2.2:5060 - 1.1.1.1:5060
SIP/2.0 486 Busy here.
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK67000fc7a4216.
From: sip:4862442964@1.1.1.1;tag=67000fc7a4.
To: sip:735603@2.2.2.2;tag=E9560618-197A.
Date: Tue, 24 Jul 2012 11:56:20 GMT.
Call-ID: 676e0e00-ac77-0f22-80c7-0002a4085a16@1.1.1.1.
CSeq: 216 INVITE.

Re: [OpenSIPS-Users] rtpproxy

2012-07-23 Thread dpa
Hello

 

To speak the truth I do not know how to do it.  

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Monday, July 23, 2012 12:54 PM
To: dpa
Cc: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] rtpproxy

 

Hi Denis,

could you make a capture to see the comunication between opensips and
rtproxy ?

Regards,



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


On 07/20/2012 01:11 PM, dpa wrote: 

Hello Bogdan

 

I am very sorry, last time I gave wrong information about the problem. Only
with r flag there is no fax stream through rtpproxy (last email I said
that everything is OK, but I made wrong test).

Here http://files.mail.ru/XF4J0L 

Is tcpdump of unsuccessful call.

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Tuesday, July 17, 2012 11:44 AM
To: dpa
Cc: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] rtpproxy

 

Hi ,

Try only with r then.

Regards,




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


On 07/17/2012 06:26 AM, dpa wrote: 

Hello Bogdan

 

I applied r and a flags and problem with fax disappeared, but I got
another problem. After I applied ra clients behind nat cannot hear caller
or callee (never mind who makes call).

Rtpproxy is working, i.e. it changes SDP body of INVITE and replies, but
there is no rtp stream.

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Thursday, July 12, 2012 7:01 PM
To: OpenSIPS users mailling list
Cc: dpa
Subject: Re: [OpenSIPS-Users] rtpproxy

 

Hi,

According to your trace the RTPproxy is properly inserted during the
re-INVITE. Try to put the r and a flags when using RTPproxy from script.

Regards,






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


On 07/11/2012 04:32 PM, dpa wrote: 

Hello Bogdan

 

Thank you for answering. 

My case is 2).

First I have audio stream which is replaced later by another fax stream.

Here http://files.mail.ru/8BYVCQ

is a tcpdump of unsuccessful call where you can see that rtpproxy doesn`t
proxy fax stream.

Rtpproxy (as an Opensips) is located on 213.170.100.150

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
Sent: Wednesday, July 11, 2012 4:43 PM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] rtpproxy

 

Hi,

You may have 2 cases, not sure which is yours:

1) the 2 media sessions are in parallel in the same calls (you can have 2
-or more- RTP streams in the same SIP session). In this case, RTPproxy will
take care automatically of each RTP stream)

2) you have first one audio stream which is replaced by another FAX stream,
via a re-INVITE - in this case, if you the normal offer / answer for
RTPproxy also for the re-INVITE, it should be ok.

Regards,






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


On 07/11/2012 11:42 AM, dpa wrote: 

Описание: cid:part1.00030104.06050802@opensips.org

Hello

 

There is one question about rtpproxy working.

If some UA using 2 different UDP ports for sending RTP and t38 fax packets
during one call session, whether rtpproxy is working (proxy both UDP
streams) in such case? 

 

Thank you for any help

 

 
 
___
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
image001.gif___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users


Re: [OpenSIPS-Users] rtpproxy

2012-07-20 Thread dpa
Hello Bogdan

 

I am very sorry, last time I gave wrong information about the problem. Only
with r flag there is no fax stream through rtpproxy (last email I said
that everything is OK, but I made wrong test).

Here http://files.mail.ru/XF4J0L 

Is tcpdump of unsuccessful call.

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Tuesday, July 17, 2012 11:44 AM
To: dpa
Cc: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] rtpproxy

 

Hi ,

Try only with r then.

Regards,



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


On 07/17/2012 06:26 AM, dpa wrote: 

Hello Bogdan

 

I applied r and a flags and problem with fax disappeared, but I got
another problem. After I applied ra clients behind nat cannot hear caller
or callee (never mind who makes call).

Rtpproxy is working, i.e. it changes SDP body of INVITE and replies, but
there is no rtp stream.

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Thursday, July 12, 2012 7:01 PM
To: OpenSIPS users mailling list
Cc: dpa
Subject: Re: [OpenSIPS-Users] rtpproxy

 

Hi,

According to your trace the RTPproxy is properly inserted during the
re-INVITE. Try to put the r and a flags when using RTPproxy from script.

Regards,





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


On 07/11/2012 04:32 PM, dpa wrote: 

Hello Bogdan

 

Thank you for answering. 

My case is 2).

First I have audio stream which is replaced later by another fax stream.

Here http://files.mail.ru/8BYVCQ

is a tcpdump of unsuccessful call where you can see that rtpproxy doesn`t
proxy fax stream.

Rtpproxy (as an Opensips) is located on 213.170.100.150

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
Sent: Wednesday, July 11, 2012 4:43 PM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] rtpproxy

 

Hi,

You may have 2 cases, not sure which is yours:

1) the 2 media sessions are in parallel in the same calls (you can have 2
-or more- RTP streams in the same SIP session). In this case, RTPproxy will
take care automatically of each RTP stream)

2) you have first one audio stream which is replaced by another FAX stream,
via a re-INVITE - in this case, if you the normal offer / answer for
RTPproxy also for the re-INVITE, it should be ok.

Regards,





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


On 07/11/2012 11:42 AM, dpa wrote: 

Описание: cid:part1.00030104.06050802@opensips.org

Hello

 

There is one question about rtpproxy working.

If some UA using 2 different UDP ports for sending RTP and t38 fax packets
during one call session, whether rtpproxy is working (proxy both UDP
streams) in such case? 

 

Thank you for any help

 

 
 
___
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
image001.gif___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users


Re: [OpenSIPS-Users] rtpproxy

2012-07-17 Thread dpa
Hello Bogdan

 

Now it's working properly, both fax and rtp. Thank you very much.

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Tuesday, July 17, 2012 11:44 AM
To: dpa
Cc: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] rtpproxy

 

Hi ,

Try only with r then.

Regards,



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


On 07/17/2012 06:26 AM, dpa wrote: 

Hello Bogdan

 

I applied r and a flags and problem with fax disappeared, but I got
another problem. After I applied ra clients behind nat cannot hear caller
or callee (never mind who makes call).

Rtpproxy is working, i.e. it changes SDP body of INVITE and replies, but
there is no rtp stream.

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Thursday, July 12, 2012 7:01 PM
To: OpenSIPS users mailling list
Cc: dpa
Subject: Re: [OpenSIPS-Users] rtpproxy

 

Hi,

According to your trace the RTPproxy is properly inserted during the
re-INVITE. Try to put the r and a flags when using RTPproxy from script.

Regards,





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


On 07/11/2012 04:32 PM, dpa wrote: 

Hello Bogdan

 

Thank you for answering. 

My case is 2).

First I have audio stream which is replaced later by another fax stream.

Here http://files.mail.ru/8BYVCQ

is a tcpdump of unsuccessful call where you can see that rtpproxy doesn`t
proxy fax stream.

Rtpproxy (as an Opensips) is located on 213.170.100.150

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
Sent: Wednesday, July 11, 2012 4:43 PM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] rtpproxy

 

Hi,

You may have 2 cases, not sure which is yours:

1) the 2 media sessions are in parallel in the same calls (you can have 2
-or more- RTP streams in the same SIP session). In this case, RTPproxy will
take care automatically of each RTP stream)

2) you have first one audio stream which is replaced by another FAX stream,
via a re-INVITE - in this case, if you the normal offer / answer for
RTPproxy also for the re-INVITE, it should be ok.

Regards,





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


On 07/11/2012 11:42 AM, dpa wrote: 

Описание: cid:part1.00030104.06050802@opensips.org

Hello

 

There is one question about rtpproxy working.

If some UA using 2 different UDP ports for sending RTP and t38 fax packets
during one call session, whether rtpproxy is working (proxy both UDP
streams) in such case? 

 

Thank you for any help

 

 
 
___
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
image001.gif___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users


Re: [OpenSIPS-Users] rtpproxy

2012-07-16 Thread dpa
Hello Bogdan

 

I applied r and a flags and problem with fax disappeared, but I got
another problem. After I applied ra clients behind nat cannot hear caller
or callee (never mind who makes call).

Rtpproxy is working, i.e. it changes SDP body of INVITE and replies, but
there is no rtp stream.

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Thursday, July 12, 2012 7:01 PM
To: OpenSIPS users mailling list
Cc: dpa
Subject: Re: [OpenSIPS-Users] rtpproxy

 

Hi,

According to your trace the RTPproxy is properly inserted during the
re-INVITE. Try to put the r and a flags when using RTPproxy from script.

Regards,




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


On 07/11/2012 04:32 PM, dpa wrote: 

Hello Bogdan

 

Thank you for answering. 

My case is 2).

First I have audio stream which is replaced later by another fax stream.

Here http://files.mail.ru/8BYVCQ

is a tcpdump of unsuccessful call where you can see that rtpproxy doesn`t
proxy fax stream.

Rtpproxy (as an Opensips) is located on 213.170.100.150

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
Sent: Wednesday, July 11, 2012 4:43 PM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] rtpproxy

 

Hi,

You may have 2 cases, not sure which is yours:

1) the 2 media sessions are in parallel in the same calls (you can have 2
-or more- RTP streams in the same SIP session). In this case, RTPproxy will
take care automatically of each RTP stream)

2) you have first one audio stream which is replaced by another FAX stream,
via a re-INVITE - in this case, if you the normal offer / answer for
RTPproxy also for the re-INVITE, it should be ok.

Regards,




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


On 07/11/2012 11:42 AM, dpa wrote: 

Описание: cid:part1.00030104.06050802@opensips.org

Hello

 

There is one question about rtpproxy working.

If some UA using 2 different UDP ports for sending RTP and t38 fax packets
during one call session, whether rtpproxy is working (proxy both UDP
streams) in such case? 

 

Thank you for any help

 

 
 
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[OpenSIPS-Users] rtpproxy

2012-07-11 Thread dpa


Hello

 

There is one question about rtpproxy working.

If some UA using 2 different UDP ports for sending RTP and t38 fax packets
during one call session, whether rtpproxy is working (proxy both UDP
streams) in such case? 

 

Thank you for any help

 

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Re: [OpenSIPS-Users] rtpproxy

2012-07-11 Thread dpa
Hello Bogdan

 

Thank you for answering. 

My case is 2).

First I have audio stream which is replaced later by another fax stream.

Here http://files.mail.ru/8BYVCQ

is a tcpdump of unsuccessful call where you can see that rtpproxy doesn`t
proxy fax stream.

Rtpproxy (as an Opensips) is located on 213.170.100.150

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
Sent: Wednesday, July 11, 2012 4:43 PM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] rtpproxy

 

Hi,

You may have 2 cases, not sure which is yours:

1) the 2 media sessions are in parallel in the same calls (you can have 2
-or more- RTP streams in the same SIP session). In this case, RTPproxy will
take care automatically of each RTP stream)

2) you have first one audio stream which is replaced by another FAX stream,
via a re-INVITE - in this case, if you the normal offer / answer for
RTPproxy also for the re-INVITE, it should be ok.

Regards,



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


On 07/11/2012 11:42 AM, dpa wrote: 

Описание: cid:part1.00030104.06050802@opensips.org

Hello

 

There is one question about rtpproxy working.

If some UA using 2 different UDP ports for sending RTP and t38 fax packets
during one call session, whether rtpproxy is working (proxy both UDP
streams) in such case? 

 

Thank you for any help

 

 
 
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[OpenSIPS-Users] FW: Opensips memory problem

2012-06-27 Thread dpa


Hello!

 

Opensips 1.6.4-2.

When doing dp_reload or dr_reload there are such errors:

 

Jun 27 17:12:53 opensips-mirror
/usr/local/opensips1.6.4-2/sbin/opensips[21801]: WARNING:core:fm_malloc: Not
enough free memory, will atempt defragmenation

Jun 27 17:12:53 opensips-mirror
/usr/local/opensips1.6.4-2/sbin/opensips[21801]:
ERROR:core:db_allocate_rows: no memory left

Jun 27 17:12:53 opensips-mirror
/usr/local/opensips1.6.4-2/sbin/opensips[21801]:
ERROR:db_mysql:db_mysql_fetch_result: no memory left

Jun 27 17:12:53 opensips-mirror
/usr/local/opensips1.6.4-2/sbin/opensips[21801]: ERROR:dialplan:dp_load_db:
failed to fetch

Jun 27 17:12:53 opensips-mirror
/usr/local/opensips1.6.4-2/sbin/opensips[21801]:
ERROR:dialplan:mi_reload_rules: failed to reload database data

Jun 27 17:12:53 opensips-mirror
/usr/local/opensips1.6.4-2/sbin/opensips[21801]:
ERROR:mi_fifo:mi_fifo_server: command () processing failed

 

Jun 27 17:14:43 opensips-mirror
/usr/local/opensips1.6.4-2/sbin/opensips[21801]:
INFO:drouting:dr_reload_cmd: dr_reload MI command received!

Jun 27 17:14:43 opensips-mirror
/usr/local/opensips1.6.4-2/sbin/opensips[21801]: WARNING:core:fm_malloc: Not
enough free memory, will atempt defragmenation

Jun 27 17:14:43 opensips-mirror
/usr/local/opensips1.6.4-2/sbin/opensips[21801]:
ERROR:core:db_allocate_rows: no memory left

Jun 27 17:14:43 opensips-mirror
/usr/local/opensips1.6.4-2/sbin/opensips[21801]:
ERROR:db_mysql:db_mysql_fetch_result: no memory left

Jun 27 17:14:43 opensips-mirror
/usr/local/opensips1.6.4-2/sbin/opensips[21801]:
ERROR:drouting:dr_load_routing_info: Error fetching rows

Jun 27 17:14:43 opensips-mirror
/usr/local/opensips1.6.4-2/sbin/opensips[21801]:
CRITICAL:drouting:dr_reload_data: failed to load routing info

Jun 27 17:14:43 opensips-mirror
/usr/local/opensips1.6.4-2/sbin/opensips[21801]:
CRITICAL:drouting:dr_reload_cmd: failed to load routing data

 

When, for example, doing address reload there is no problem.

There is no problem with call proceeding. 

 

opensips.cfg

...

modparam(drouting, fetch_rows, 1000)

..

modparam(dialplan, fetch_rows, 1000)

 

Opensips memory

 

shmem:total_size = 104857600

shmem:used_size = 41207560

shmem:real_used_size = 46998880

shmem:max_used_size = 53773824

shmem:free_size = 57858720

shmem:fragments = 54557

 

pkmem:0-total_size = 20971520

pkmem:0-used_size = 710496

pkmem:0-real_used_size = 857016

pkmem:0-max_used_size = 1036592

pkmem:0-free_size = 20114504

pkmem:0-fragments = 3

 

Other process has nearly same values.

 

Thank you for any help

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[OpenSIPS-Users] opensips and rtpproxy

2012-05-23 Thread dpa


Hello!

Opensips 1.6.4-2

 

There is one question. 

rtpp_notify_socket  may be only unix socket? Or I can use UPD or TCP too?

 

Thank you for any help

 

 

 

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Re: [OpenSIPS-Users] opensips and rtpproxy

2012-05-23 Thread dpa
Hello Răzvan

 

Thank you for replay. 

But one more question 

In man for rtpproxy I read -n timeout_socket.. The socket should be created
by another  application, preferably before starting rtpproxy..

So as I understand rtpp_notify_socket should be like
modparam(nathelper, rtpp_notify_socket, tcp:localhost:)

and after this rtpproxy should be started with -n tcp:1.1.1.1: where
1.1.1.1 - ip of opensips serever.

Am I right?

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Razvan Crainea
Sent: Wednesday, May 23, 2012 5:47 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] opensips and rtpproxy

 

Hi, Denis!

The socket can be UNIX or TCP according to the documentation[1].

[1] http://www.opensips.org/html/docs/modules/devel/rtpproxy.html#id250454 

Regards,



--
Răzvan Crainea
OpenSIPS Developer
http://www.opensips-solutions.com


On 05/23/2012 04:11 PM, dpa wrote: 



Hello!

Opensips 1.6.4-2

 

There is one question. 

rtpp_notify_socket  may be only unix socket? Or I can use UPD or TCP too?

 

Thank you for any help

 

 

 






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Re: [OpenSIPS-Users] opensips and rtpproxy

2012-05-23 Thread dpa
Result after timeout detection

 

May 23 18:30:10 kam rtpproxy[5925]: ERR:reconnect_timeout_handler: can't
connect to timeout socket: Connection refused

May 23 18:30:10 kam rtpproxy[5925]: ERR:do_timeout_notification: unable to
send timeout notification

 

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Razvan Crainea
Sent: Wednesday, May 23, 2012 6:23 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] opensips and rtpproxy

 

Hi, Denis!
 
Try to use the same socket for both rtpproxy module and application:
 
modparam(nathelper, rtpp_notify_socket, tcp:127.0.0.1:)
rtpproxy -n tcp:127.0.0.1:
 
Regards,
--
Răzvan Crainea
OpenSIPS Developer
http://www.opensips-solutions.com


On 05/23/2012 05:13 PM, dpa wrote: 

Hello Răzvan

 

Thank you for replay. 

But one more question 

In man for rtpproxy I read -n timeout_socket.. The socket should be created
by another  application, preferably before starting rtpproxy..

So as I understand rtpp_notify_socket should be like
modparam(nathelper, rtpp_notify_socket, tcp:localhost:)

and after this rtpproxy should be started with -n tcp:1.1.1.1: where
1.1.1.1 - ip of opensips serever.

Am I right?

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Razvan Crainea
Sent: Wednesday, May 23, 2012 5:47 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] opensips and rtpproxy

 

Hi, Denis!

The socket can be UNIX or TCP according to the documentation[1].

[1] http://www.opensips.org/html/docs/modules/devel/rtpproxy.html#id250454 

Regards,




--
Răzvan Crainea
OpenSIPS Developer
http://www.opensips-solutions.com


On 05/23/2012 04:11 PM, dpa wrote: 



Hello!

Opensips 1.6.4-2

 

There is one question. 

rtpp_notify_socket  may be only unix socket? Or I can use UPD or TCP too?

 

Thank you for any help

 

 

 







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Re: [OpenSIPS-Users] Mediaproxy and RTP timeout

2012-05-15 Thread dpa
I try to test it and after 180 s. (I didn`t change default value) dialog was
being in active state. 

-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Saul Ibarra Corretge
Sent: Monday, May 14, 2012 12:21 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Mediaproxy and RTP timeout


On May 14, 2012, at 9:23 AM, dpa wrote:

 I understand but how mediaproxy should notice dialog module to terminate
call? And can I configure timeout for RTP missing?
  

MediaProxy does that internally through the management interface. Look on
the sample config.ini file for directions on how to configure the timeout
(by default is 180 seconds).


Regards,

--
Saúl Ibarra Corretgé
AG Projects




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Re: [OpenSIPS-Users] Mediaproxy and RTP timeout

2012-05-14 Thread dpa
I understand but how mediaproxy should notice dialog module to terminate
call? And can I configure timeout for RTP missing? 

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Adrian Georgescu
Sent: Sunday, May 13, 2012 11:19 AM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Mediaproxy and RTP timeout

 

There is nothing to set in mediaproxy.  The dialog module takes care of
terminating the dialog.

 

Adrian

 

On May 12, 2012, at 8:30 AM, dpa wrote:





Thank you for your reply

What should I do to make it working? I have tested the feature by emulating
situation when there were no RTP streams neither from caller nor from callee
and there was nothing, dialog was in active state.

 

Thank toy

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Adrian Georgescu
Sent: Friday, May 11, 2012 9:38 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Mediaproxy and RTP timeout

 

Yes, sending BYEs on RTP timeout is supported.

 

Adrian

 

On May 11, 2012, at 12:04 PM, dpa wrote:






Hello!

 

In rtpproxy there is a feature which allows controlling absence of RTP
stream (in both directions) and signaling to Opensips to disconnect the
call. Is there such feature in mediaproxy?

 

Thank you for any help.

 

 

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Re: [OpenSIPS-Users] Mediaproxy and RTP timeout

2012-05-12 Thread dpa
Thank you for your reply

What should I do to make it working? I have tested the feature by emulating
situation when there were no RTP streams neither from caller nor from callee
and there was nothing, dialog was in active state.

 

Thank toy 

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Adrian Georgescu
Sent: Friday, May 11, 2012 9:38 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Mediaproxy and RTP timeout

 

Yes, sending BYEs on RTP timeout is supported.

 

Adrian

 

On May 11, 2012, at 12:04 PM, dpa wrote:





Hello!

 

In rtpproxy there is a feature which allows controlling absence of RTP
stream (in both directions) and signaling to Opensips to disconnect the
call. Is there such feature in mediaproxy?

 

Thank you for any help.

 

 

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[OpenSIPS-Users] Mediaproxy and RTP timeout

2012-05-11 Thread dpa


Hello!

 

In rtpproxy there is a feature which allows controlling absence of RTP
stream (in both directions) and signaling to Opensips to disconnect the
call. Is there such feature in mediaproxy?

 

Thank you for any help.

 

 

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[OpenSIPS-Users] Opensips 1.6.4.2 and some problem

2012-04-27 Thread dpa


Hello!

Once I found that Opensips does not response to any message. Rather it
response but with great delay (debug log is in attachment. Debug log has
been done using ngrep utility).

1.1.1.1- Some UAC

1.1.1.2- Opensips

1.1.1.3- Some UAS   

During this problem I found errors in opensips log (debug level = 3)

CRITICAL:tm:set_timer: set_timer for 1 list called on a detached timer --
ignoring: 0x7ffd073d5968 And the part 0x7ffd073d5968 is different in each
message.

And one time I saw ERROR:dialog:dlg_onroute: failed to update dialog
lifetime.

And many messages CRITICAL:dialog:log_next_state_dlg: bogus event 7 in
state 1 for dlg 0x7ffd08c2f1c8 [3221:2137264810] with clid
'21009543-8F6F11E1-A7309929-19A1016D' and tags '221D86F4-1EA3' 'NULL'

Besides, during problem I saw that Opensips disconnects some calls (see
attachment) using default dialog timeout (5 sec.)

The problem disappeared without anything action from me.

 

Thank you for any help.

 

 

 

image001.gifU 2012/04/27 10:59:18.690929 1.1.1.1:62160 - 1.1.1.2:5060
INVITE sip:74997033315@1.1.1.2:5060 SIP/2.0.
Via: SIP/2.0/UDP 1.1.1.1:5060;x-route-tag=tgrp:TFOP;branch=z9hG4bKC4DEF61640.
From: sip:929530@1.1.1.1;tag=4202CB20-9D.
To: sip:74997033315@1.1.1.2.
Date: Sun, 07 Sep 2003 03:33:12 GMT.
Call-ID: D8179B0E-E01A11D7-BE26E851-360FA078@1.1.1.1.
Supported: 100rel,timer,resource-priority,replaces,sdp-anat.
Min-SE:  1800.
Cisco-Guid: 3625385702-3759804887-3051749410-2438464902.
User-Agent: Cisco-SIPGateway/IOS-12.x.
Accept-Language: ru.
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, 
NOTIFY, INFO, REGISTER.
CSeq: 101 INVITE.
Max-Forwards: 15.
Timestamp: 1062905592.
Contact: sip:929530@1.1.1.1:5060.
Expires: 60.
Allow-Events: telephone-event.
P-Asserted-Identity: sip:929530@1.1.1.1.
Content-Type: application/sdp.
Content-Disposition: session;handling=required.
Content-Length: 310.
.
v=0.
o=CiscoSystemsSIP-GW-UserAgent 1053 1891 IN IP4 1.1.1.1.
s=SIP Call.
c=IN IP4 1.1.1.1.
t=0 0.
m=audio 17534 RTP/AVP 8 18 0 101.
c=IN IP4 1.1.1.1.
a=rtpmap:8 PCMA/8000.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=yes.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.


U 2012/04/27 10:59:19.188543 1.1.1.1:62160 - 1.1.1.2:5060
INVITE sip:74997033315@1.1.1.2:5060 SIP/2.0.
Via: SIP/2.0/UDP 1.1.1.1:5060;x-route-tag=tgrp:TFOP;branch=z9hG4bKC4DEF61640.
From: sip:929530@1.1.1.1;tag=4202CB20-9D.
To: sip:74997033315@1.1.1.2.
Date: Sun, 07 Sep 2003 03:33:13 GMT.
Call-ID: D8179B0E-E01A11D7-BE26E851-360FA078@1.1.1.1.
Supported: 100rel,timer,resource-priority,replaces,sdp-anat.
Min-SE:  1800.
Cisco-Guid: 3625385702-3759804887-3051749410-2438464902.
User-Agent: Cisco-SIPGateway/IOS-12.x.
Accept-Language: ru.
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, 
NOTIFY, INFO, REGISTER.
CSeq: 101 INVITE.
Max-Forwards: 15.
Timestamp: 1062905593.
Contact: sip:929530@1.1.1.1:5060.
Expires: 60.
Allow-Events: telephone-event.
P-Asserted-Identity: sip:929530@1.1.1.1.
Content-Type: application/sdp.
Content-Disposition: session;handling=required.
Content-Length: 310.
.
v=0.
o=CiscoSystemsSIP-GW-UserAgent 1053 1891 IN IP4 1.1.1.1.
s=SIP Call.
c=IN IP4 1.1.1.1.
t=0 0.
m=audio 17534 RTP/AVP 8 18 0 101.
c=IN IP4 1.1.1.1.
a=rtpmap:8 PCMA/8000.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=yes.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.


U 2012/04/27 10:59:20.188422 1.1.1.1:62160 - 1.1.1.2:5060
INVITE sip:74997033315@1.1.1.2:5060 SIP/2.0.
Via: SIP/2.0/UDP 1.1.1.1:5060;x-route-tag=tgrp:TFOP;branch=z9hG4bKC4DEF61640.
From: sip:929530@1.1.1.1;tag=4202CB20-9D.
To: sip:74997033315@1.1.1.2.
Date: Sun, 07 Sep 2003 03:33:14 GMT.
Call-ID: D8179B0E-E01A11D7-BE26E851-360FA078@1.1.1.1.
Supported: 100rel,timer,resource-priority,replaces,sdp-anat.
Min-SE:  1800.
Cisco-Guid: 3625385702-3759804887-3051749410-2438464902.
User-Agent: Cisco-SIPGateway/IOS-12.x.
Accept-Language: ru.
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, 
NOTIFY, INFO, REGISTER.
CSeq: 101 INVITE.
Max-Forwards: 15.
Timestamp: 1062905594.
Contact: sip:929530@1.1.1.1:5060.
Expires: 60.
Allow-Events: telephone-event.
P-Asserted-Identity: sip:929530@1.1.1.1.
Content-Type: application/sdp.
Content-Disposition: session;handling=required.
Content-Length: 310.
.
v=0.
o=CiscoSystemsSIP-GW-UserAgent 1053 1891 IN IP4 1.1.1.1.
s=SIP Call.
c=IN IP4 1.1.1.1.
t=0 0.
m=audio 17534 RTP/AVP 8 18 0 101.
c=IN IP4 1.1.1.1.
a=rtpmap:8 PCMA/8000.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=yes.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.


U 2012/04/27 10:59:26.188959 1.1.1.1:62160 - 1.1.1.2:5060
INVITE sip:74997033315@1.1.1.2:5060 SIP/2.0.
Via: SIP/2.0/UDP 1.1.1.1:5060;x-route-tag=tgrp:TFOP;branch=z9hG4bKC4DEF61640.
From: sip:929530@1.1.1.1;tag=4202CB20-9D.
To: sip:74997033315@1.1.1.2.
Date: Sun, 07 Sep 2003 03:33:20 GMT.
Call-ID: 

Re: [OpenSIPS-Users] [RELEASE] OpenSIPS 1.8.0 - a new major release is out

2012-03-23 Thread dpa
Pay attention on “curses.h:31:19: error: curses.h: No such file or directory”

 

I think You should install libncurses-dev package first

 

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of goup2010
Sent: Friday, March 23, 2012 11:52 AM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] [RELEASE] OpenSIPS 1.8.0 - a new major release is 
out

 

Hello,

I try to install 1.8.0. When run make install I receive follow errors:


make[1]: Entering directory `/root/opensips_1_8/menuconfig'
rm -f configure
rm -f cfg.o curses.o items.o commands.o menus.o parser.o main.o
make[1]: Leaving directory `/root/opensips_1_8/menuconfig'
make[1]: Entering directory `/root/opensips_1_8/menuconfig'
gcc -g -Wall 
-DMENUCONFIG_CFG_PATH=\/usr/local/share/opensips//menuconfig_templ ates/\ 
-DMENUCONFIG_GEN_PATH=\/etc/opensips/\ -DMENUCONFIG_HAVE_SOURCES=0 - c -o 
cfg.o cfg.c
In file included from main.h:33,
from cfg.c:30:
curses.h:31:19: error: curses.h: No such file or directory
In file included from cfg.c:30:
main.h:35: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘*’ 
to ken
main.h:39: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘*’ 
to ken
make[1]: *** [cfg.o] Error 1
make[1]: Leaving directory `/root/opensips_1_8/menuconfig'
make: *** [opensipsmc] Error 2

Where is problem?

Best regards,

PlayMen

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