Re: [OpenSIPS-Users] record_route and dialog module confusion

[Denis Putyato]

Please show a piece of opensips.cfg where you calling record_route() and SIP 
debug of such call 

-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Jesse Cloutier
Sent: Wednesday, July 20, 2011 7:08 PM
To: users@lists.opensips.org
Subject: [OpenSIPS-Users] record_route and dialog module confusion

Hello,

I am a little confused as to how our opensips proxy is fitting into our 
topology. I have opensips setup as a proxy for dynamic routing and I 
believe I have it set up for state full routing.

Our asterisk server calls the opensips proxy which calls one of our 
providers based on the dr routing. What confuses me is that all 
transactions after the initial invite go directly between our provider 
and the asterisk server. Bypassing the opensips proxy. I am calling 
record_route() on the calls so shouldnt all the transactions go through 
the proxy as well?

I have also setup the dialog module and my dialogs never get destroyed 
because opensips never gets the bye.

Is calling record_route enough?

Thanks,
Jesse Cloutier
Network Administrator

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Re: [OpenSIPS-Users] aaa_radius and b2b_logic modules will not compile

[Denis Putyato]

I think there were more error strings in log during compile process. Show all 
error strings.

-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Mark Holloway
Sent: Wednesday, July 06, 2011 11:23 AM
To: OpenSIPS users mailling list
Subject: [OpenSIPS-Users] aaa_radius and b2b_logic modules will not compile

When editing the Makefile to enable aaa_radius or b2b_logic and then executing 
make all I continue to receive the following error:

make[1]: *** [aaa_radius.o] Error 1
make[1]: Leaving directory `/usr/src/opensips-1.6.0-tls/modules/aaa_radius'
make: *** [modules] Error 2

make[1]: *** [b2b_logic.o] Error 1
make[1]: Leaving directory `/usr/src/opensips-1.6.0-tls/modules/b2b_logic'
make: *** [modules] Error 2


Does anyone know what causes this? 
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Re: [OpenSIPS-Users] aaa_radius and b2b_logic modules will not compile

[Denis Putyato]

: error: ‘b2b_scenario_t’ has no member named ‘doc’
b2b_logic.c:420: error: ‘b2b_scenario_t’ has no member named ‘request_rules’
b2b_logic.c:424: error: ‘b2b_rule_t’ has no member named ‘next’
b2b_logic.c:429: error: ‘b2b_scenario_t’ has no member named ‘reply_rules’
b2b_logic.c:433: error: ‘b2b_rule_t’ has no member named ‘next’
b2b_logic.c: In function ‘fixup_b2b_logic’:
b2b_logic.c:495: error: ‘b2b_scenario_t’ has no member named ‘next’
b2b_logic.c: In function ‘mi_trigger_scenario’:
b2b_logic.c:517: error: ‘xmlNodePtr’ undeclared (first use in this function)
b2b_logic.c:517: error: expected ‘;’ before ‘xml_node’
b2b_logic.c:544: error: ‘b2b_scenario_t’ has no member named ‘next’
b2b_logic.c:595: error: ‘xml_node’ undeclared (first use in this function)
b2b_logic.c:595: error: ‘b2b_scenario_t’ has no member named ‘init_node’
b2b_logic.c:598: warning: implicit declaration of function ‘xmlNodeGetContent’
b2b_logic.c:617: error: ‘b2b_scenario_t’ has no member named ‘init_node’
b2b_logic.c:624: error: too many arguments to function ‘process_bridge_action’
make[1]: *** [b2b_logic.o] Error 1
make[1]: Leaving directory `/usr/src/opensips-1.6.0-tls/modules/b2b_logic'
make: *** [modules] Error 2





On Jul 6, 2011, at 12:25 AM, Denis Putyato wrote:

 I think there were more error strings in log during compile process. Show all 
 error strings.
 
 -Original Message-
 From: users-boun...@lists.opensips.org 
 [mailto:users-boun...@lists.opensips.org] On Behalf Of Mark Holloway
 Sent: Wednesday, July 06, 2011 11:23 AM
 To: OpenSIPS users mailling list
 Subject: [OpenSIPS-Users] aaa_radius and b2b_logic modules will not compile
 
 When editing the Makefile to enable aaa_radius or b2b_logic and then 
 executing make all I continue to receive the following error:
 
 make[1]: *** [aaa_radius.o] Error 1
 make[1]: Leaving directory `/usr/src/opensips-1.6.0-tls/modules/aaa_radius'
 make: *** [modules] Error 2
 
 make[1]: *** [b2b_logic.o] Error 1
 make[1]: Leaving directory `/usr/src/opensips-1.6.0-tls/modules/b2b_logic'
 make: *** [modules] Error 2
 
 
 Does anyone know what causes this? 
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Re: [OpenSIPS-Users] B2B module and CANCEL

[Denis Putyato]

Hello Anca

 

I opened bug report on bug tracker 

Thank you for help

 

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Anca Vamanu
Sent: Tuesday, June 14, 2011 4:25 PM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] B2B module and CANCEL

 

Hi Denis,

Sorry, I have to make a correction - I forgot that the fix that is committed on 
svn is partial and works only if body lumps are applied. So your case will 
still not work
 and needs another fix. I suggest you to open a bug report in svn. 

Regards,
Anca

On 06/14/2011 02:35 PM, Anca Vamanu wrote: 

Hi Denis,

We hit this problem also some time ago, it was indeed a bug when applying lumps 
in local_route. We were just waiting for the fix to get enough testing. It is 
stable now. I have just committed the fix in tm module in both trunk and 1.6. 
Please upgrade and check.

Regards,
Anca

On 06/14/2011 06:57 AM, Denis Putyato wrote: 

Hello

 

I found that this problem appears when I use append_hf() to add some header in 
local route of the proxy1 before sending INVITE to proxy2. Without this adding 
problem disappears.

 

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Denis Putyato
Sent: Friday, June 10, 2011 5:52 PM
To: 'OpenSIPS users mailling list'
Subject: [OpenSIPS-Users] B2B module and CANCEL

 

Hello!

 

I have a such problem opensips1.6.4-2

 

There are two proxies of version 1.6.4.-2 which has been installed on the same 
server.

 

One proxy (proxy1) using B2B “top hiding” and located in /usr/local/sbc and 
using one signaling port 

Another proxy (proxy2) is just SIP proxy and located in 
/usr/local/opensips1.6.4/ and using another signaling port

 

Both proxies using the same ip address of the server

 

Call flow:

 

some UA – proxy1 – proxy2 – some gateway

 

When UA generate CANCEL then proxy1 does some strange things with FROM or TO 
uri headers (you can see it in attachment).  Because of this proxy2 cannot 
parse CANCEL properly and transaction in proxy2 cannot be canceled.

 

Thank you for any help.

 

 

 

 

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Re: [OpenSIPS-Users] B2B module and CANCEL

[Denis Putyato]

Hello

 

I found that this problem appears when I use append_hf() to add some header in 
local route of the proxy1 before sending INVITE to proxy2. Without this adding 
problem disappears.

 

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Denis Putyato
Sent: Friday, June 10, 2011 5:52 PM
To: 'OpenSIPS users mailling list'
Subject: [OpenSIPS-Users] B2B module and CANCEL

 

Hello!

 

I have a such problem opensips1.6.4-2

 

There are two proxies of version 1.6.4.-2 which has been installed on the same 
server.

 

One proxy (proxy1) using B2B “top hiding” and located in /usr/local/sbc and 
using one signaling port 

Another proxy (proxy2) is just SIP proxy and located in 
/usr/local/opensips1.6.4/ and using another signaling port

 

Both proxies using the same ip address of the server

 

Call flow:

 

some UA – proxy1 – proxy2 – some gateway

 

When UA generate CANCEL then proxy1 does some strange things with FROM or TO 
uri headers (you can see it in attachment).  Because of this proxy2 cannot 
parse CANCEL properly and transaction in proxy2 cannot be canceled.

 

Thank you for any help.

 

 

 

 

 

 

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[OpenSIPS-Users] B2B module and CANCEL

[Denis Putyato]

Hello!

 

I have a such problem opensips1.6.4-2

 

There are two proxies of version 1.6.4.-2 which has been installed on the same 
server.

 

One proxy (proxy1) using B2B “top hiding” and located in /usr/local/sbc and 
using one signaling port 

Another proxy (proxy2) is just SIP proxy and located in 
/usr/local/opensips1.6.4/ and using another signaling port

 

Both proxies using the same ip address of the server

 

Call flow:

 

some UA – proxy1 – proxy2 – some gateway

 

When UA generate CANCEL then proxy1 does some strange things with FROM or TO 
uri headers (you can see it in attachment).  Because of this proxy2 cannot 
parse CANCEL properly and transaction in proxy2 cannot be canceled.

 

Thank you for any help.

 

 

 

 

 

 

Jun 10 17:40:47 kam /usr/local/sbc/sbin/opensips[1814]: DBG:core:parse_msg: SIP 
Request:
Jun 10 17:40:47 kam /usr/local/sbc/sbin/opensips[1814]: DBG:core:parse_msg:  
method:  CANCEL
Jun 10 17:40:47 kam /usr/local/sbc/sbin/opensips[1814]: DBG:core:parse_msg:  
uri: sip:88123364021@1.1.1.1:5063;transport=UDP
Jun 10 17:40:47 kam /usr/local/sbc/sbin/opensips[1814]: DBG:core:parse_msg:  
version: SIP/2.0
Jun 10 17:40:47 kam /usr/local/sbc/sbin/opensips[1814]: DBG:core:parse_headers: 
flags=2
Jun 10 17:40:47 kam /usr/local/sbc/sbin/opensips[1814]: 
DBG:core:parse_via_param: found param type 232, branch = 
z9hG4bK-d8754z-816e76605de1b275-1---d8754z-; state=6
Jun 10 17:40:47 kam /usr/local/sbc/sbin/opensips[1814]: 
DBG:core:parse_via_param: found param type 235, rport = n/a; state=17
Jun 10 17:40:47 kam /usr/local/sbc/sbin/opensips[1814]: DBG:core:parse_via: end 
of header reached, state=5
Jun 10 17:40:47 kam /usr/local/sbc/sbin/opensips[1814]: DBG:core:parse_headers: 
via found, flags=2
Jun 10 17:40:47 kam /usr/local/sbc/sbin/opensips[1814]: DBG:core:parse_headers: 
this is the first via
Jun 10 17:40:47 kam /usr/local/sbc/sbin/opensips[1814]: DBG:core:receive_msg: 
After parse_msg...
Jun 10 17:40:47 kam /usr/local/sbc/sbin/opensips[1814]: DBG:core:receive_msg: 
preparing to run routing scripts...
Jun 10 17:40:47 kam /usr/local/sbc/sbin/opensips[1814]: DBG:core:parse_headers: 
flags=
Jun 10 17:40:47 kam /usr/local/sbc/sbin/opensips[1814]: DBG:core:parse_to: end 
of header reached, state=10
Jun 10 17:40:47 kam /usr/local/sbc/sbin/opensips[1814]: DBG:core:parse_to: 
display={}, ruri={sip:88123364021@1.1.1.1:5063;transport=UDP}
Jun 10 17:40:47 kam /usr/local/sbc/sbin/opensips[1814]: DBG:core:get_hdr_field: 
To [52]; uri=[sip:88123364021@1.1.1.1:5063;transport=UDP] 
Jun 10 17:40:47 kam /usr/local/sbc/sbin/opensips[1814]: DBG:core:get_hdr_field: 
to body [sip:88123364021@1.1.1.1:5063;transport=UDP#015#012]
Jun 10 17:40:47 kam /usr/local/sbc/sbin/opensips[1814]: DBG:core:get_hdr_field: 
cseq CSeq: 2 CANCEL
Jun 10 17:40:47 kam /usr/local/sbc/sbin/opensips[1814]: DBG:core:get_hdr_field: 
content_length=0
Jun 10 17:40:47 kam /usr/local/sbc/sbin/opensips[1814]: DBG:core:get_hdr_field: 
found end of header
Jun 10 17:40:47 kam /usr/local/sbc/sbin/opensips[1814]: 
DBG:b2b_entities:b2b_prescript_f: start - method = CANCEL
Jun 10 17:40:47 kam /usr/local/sbc/sbin/opensips[1814]: 
DBG:core:parse_to_param: tag=2e1b1846
Jun 10 17:40:47 kam /usr/local/sbc/sbin/opensips[1814]: DBG:core:parse_to: end 
of header reached, state=29
Jun 10 17:40:47 kam /usr/local/sbc/sbin/opensips[1814]: DBG:core:parse_to: 
display={}, ruri={sip:8123364079@1.1.1.1:5063;transport=UDP}
Jun 10 17:40:47 kam /usr/local/sbc/sbin/opensips[1814]: 
DBG:b2b_entities:b2b_parse_key: Does not have b2b_entities prefix
Jun 10 17:40:47 kam /usr/local/sbc/sbin/opensips[1814]: 
DBG:b2b_entities:b2bl_search_iteratively: Search for record with callid= 
MzBhYzkyODA2YjEzZGEyZTFhNjAxMzBhMjI1NWU3ZmU., tag= 2e1b1846
Jun 10 17:40:47 kam /usr/local/sbc/sbin/opensips[1814]: 
DBG:b2b_entities:b2bl_search_iteratively: Found callid= 
MzBhYzkyODA2YjEzZGEyZTFhNjAxMzBhMjI1NWU3ZmU., tag= 2e1b1846
Jun 10 17:40:47 kam /usr/local/sbc/sbin/opensips[1814]: DBG:tm:t_newtran: 
transaction on entrance=0x
Jun 10 17:40:47 kam /usr/local/sbc/sbin/opensips[1814]: DBG:core:parse_headers: 
flags=
Jun 10 17:40:47 kam /usr/local/sbc/sbin/opensips[1814]: DBG:core:parse_headers: 
flags=78
Jun 10 17:40:47 kam /usr/local/sbc/sbin/opensips[1814]: 
DBG:tm:t_lookup_request: start searching: hash=49978, isACK=0
Jun 10 17:40:47 kam /usr/local/sbc/sbin/opensips[1814]: DBG:tm:matching_3261: 
RFC3261 transaction matching failed
Jun 10 17:40:47 kam /usr/local/sbc/sbin/opensips[1814]: 
DBG:tm:t_lookup_request: no transaction found
Jun 10 17:40:47 kam /usr/local/sbc/sbin/opensips[1814]: DBG:core:parse_headers: 
flags=
Jun 10 17:40:47 kam /usr/local/sbc/sbin/opensips[1814]: 
DBG:core:check_ip_address: params 192.168.18.55, 192.168.18.55, 0
Jun 10 17:40:47 kam /usr/local/sbc/sbin/opensips[1814]: DBG:core:_shm_resize: 
resize(0) called
Jun 10 17:40:47 kam /usr/local/sbc/sbin/opensips[1814]: 
DBG:tm:cleanup_uac_timers: 

[OpenSIPS-Users] Cache module

[Denis Putyato]

Hello everybody!

 

Please give me some information about localcach module

 

Is it use shmem of Opensips for storing data?

 

Thank you

 

 

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Re: [OpenSIPS-Users] dialog and acc

[Denis Putyato]

Bogdan, one more question

You wrote “2) at ACK, before loose_route()”

Why before? Is this critical? 

I want to use $dlg_status variables to check if dialog exists (When received 
ACK), but in documentation said that this variables works only after 
loose_route() function

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Tuesday, May 10, 2011 12:43 PM
To: Denis Putyato
Cc: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] dialog and acc

 

Hi Denis,

I can tell it works for sure, as I'm using this kind of dirty trick to cope 
with some buggy clients.

Best regards,
Bogdan

On 05/10/2011 06:42 AM, Denis Putyato wrote: 

Hello!

 

Thank you Bogdan, I will try you decision

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Friday, May 06, 2011 9:25 PM
To: OpenSIPS users mailling list
Cc: Denis Putyato
Subject: Re: [OpenSIPS-Users] dialog and acc

 

Hi Denis,

From a proxy point of view, a 200OK means the dialog was establish. A proxy 
cannot interfere with the ACK part - the acknowledgment is done between end 
parties.

If the ACK is missing (in an established dialog), the callee party (according 
to RFC) must send a BYE (when finishing the 200 OK retransmission) to the 
caller.  This is something your callee doesn't do.

A simple work around is to use the dialog timeout in opensips:
1) at INVITE time, when dialog is created, set a 5 seconds timeout (dialog 
will be terminated by opensips, with BYE, in 5 secs after being established - 
do not forget to set the BYE_ON_TIMEOUT flag)
2) at ACK, before loose_route() set a new timeout to some long, long (3 
hours?) value.

So, if the ACK will mis, the 5 sec timeout will kick in and terminate the 
dialog; otherwise, opensips will prelong the dialog on ACK time.

Regards,
Bogdan

On 04/28/2011 03:06 PM, Denis Putyato wrote: 

Hello!

 

I noticed that cdr_flag in acc modules marks dialog for accounting as answered 
even there was no ACK on 200 OK.

As a result, I have acc record which has a big duration and status 200 OK.

 

Thank you for any help.

 

 

 

 
 
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Re: [OpenSIPS-Users] dialog and acc

[Denis Putyato]

Bogdan

 

I understand, thank you

 

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Tuesday, May 10, 2011 12:56 PM
To: Denis Putyato
Cc: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] dialog and acc

 

Hi Denis,

Yes, before the loose_route(), and it is critical as for sequential requests, 
the matching against existing dialogs is done somewhere inside loose_route() 
(via some callbacks). And when the dialog is matched, the dialog timeout is 
updated, so this is why you need to populate the AVP for dialog timeout before 
the loose_route.

Dialog related variables will be available available after the loose_route(), 
when the dialog was matched, so the dialog cotext is available.

Regards,
Bogdan

On 05/10/2011 11:51 AM, Denis Putyato wrote: 

Bogdan, one more question

You wrote “2) at ACK, before loose_route()”

Why before? Is this critical? 

I want to use $dlg_status variables to check if dialog exists (When received 
ACK), but in documentation said that this variables works only after 
loose_route() function

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Tuesday, May 10, 2011 12:43 PM
To: Denis Putyato
Cc: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] dialog and acc

 

Hi Denis,

I can tell it works for sure, as I'm using this kind of dirty trick to cope 
with some buggy clients.

Best regards,
Bogdan

On 05/10/2011 06:42 AM, Denis Putyato wrote: 

Hello!

 

Thank you Bogdan, I will try you decision

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Friday, May 06, 2011 9:25 PM
To: OpenSIPS users mailling list
Cc: Denis Putyato
Subject: Re: [OpenSIPS-Users] dialog and acc

 

Hi Denis,

From a proxy point of view, a 200OK means the dialog was establish. A proxy 
cannot interfere with the ACK part - the acknowledgment is done between end 
parties.

If the ACK is missing (in an established dialog), the callee party (according 
to RFC) must send a BYE (when finishing the 200 OK retransmission) to the 
caller.  This is something your callee doesn't do.

A simple work around is to use the dialog timeout in opensips:
1) at INVITE time, when dialog is created, set a 5 seconds timeout (dialog 
will be terminated by opensips, with BYE, in 5 secs after being established - 
do not forget to set the BYE_ON_TIMEOUT flag)
2) at ACK, before loose_route() set a new timeout to some long, long (3 
hours?) value.

So, if the ACK will mis, the 5 sec timeout will kick in and terminate the 
dialog; otherwise, opensips will prelong the dialog on ACK time.

Regards,
Bogdan

On 04/28/2011 03:06 PM, Denis Putyato wrote: 

Hello!

 

I noticed that cdr_flag in acc modules marks dialog for accounting as answered 
even there was no ACK on 200 OK.

As a result, I have acc record which has a big duration and status 200 OK.

 

Thank you for any help.

 

 

 

 
 
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OpenSIPS eBootcamp - 2nd of May 2011
OpenSIPS solutions and know-how







-- 
Bogdan-Andrei Iancu
OpenSIPS eBootcamp - 2nd of May 2011
OpenSIPS solutions and know-how






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Re: [OpenSIPS-Users] dialog and acc

[Denis Putyato]

Hello!

 

Thank you Bogdan, I will try you decision

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Friday, May 06, 2011 9:25 PM
To: OpenSIPS users mailling list
Cc: Denis Putyato
Subject: Re: [OpenSIPS-Users] dialog and acc

 

Hi Denis,

From a proxy point of view, a 200OK means the dialog was establish. A proxy 
cannot interfere with the ACK part - the acknowledgment is done between end 
parties.

If the ACK is missing (in an established dialog), the callee party (according 
to RFC) must send a BYE (when finishing the 200 OK retransmission) to the 
caller.  This is something your callee doesn't do.

A simple work around is to use the dialog timeout in opensips:
1) at INVITE time, when dialog is created, set a 5 seconds timeout (dialog 
will be terminated by opensips, with BYE, in 5 secs after being established - 
do not forget to set the BYE_ON_TIMEOUT flag)
2) at ACK, before loose_route() set a new timeout to some long, long (3 
hours?) value.

So, if the ACK will mis, the 5 sec timeout will kick in and terminate the 
dialog; otherwise, opensips will prelong the dialog on ACK time.

Regards,
Bogdan

On 04/28/2011 03:06 PM, Denis Putyato wrote: 

Hello!

 

I noticed that cdr_flag in acc modules marks dialog for accounting as answered 
even there was no ACK on 200 OK.

As a result, I have acc record which has a big duration and status 200 OK.

 

Thank you for any help.

 

 

 

 
 
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Re: [OpenSIPS-Users] Validate To Uri

[Denis Putyato]

Hello

Try to use dialplan module
http://www.opensips.org/html/docs/modules/devel/dialplan.html#id249075

-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Jan D.
Sent: Wednesday, May 04, 2011 1:00 PM
To: users@lists.opensips.org
Subject: [OpenSIPS-Users] Validate To Uri

In an INVITE I want to validate the To Uri ($Tu). This should be a numeric
value (ie. 31201234567).

Is there a function I can use or should I validate against a regexp (if so,
do you have an example).

Regards,

Jan

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[OpenSIPS-Users] dialog and acc

[Denis Putyato]

Hello!

 

I noticed that cdr_flag in acc modules marks dialog for accounting as answered 
even there was no ACK on 200 OK.

As a result, I have acc record which has a big duration and status 200 OK.

 

Thank you for any help.

 

 

 

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Re: [OpenSIPS-Users] missed_calls doubt

[Denis Putyato]

Hello

 

Try to use failed_transaction_flag

http://www.opensips.org/html/docs/modules/devel/acc.html#id292642

 

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Wesley Volcov
Sent: Thursday, April 28, 2011 4:13 PM
To: OpenSIPS users mailling list
Subject: [OpenSIPS-Users] missed_calls doubt

 

Dear List,

I use my opensips, to make the routes based on prefix received and send the 
call to the termination carrier based on a lcr configured with dr_route module.
I have 2 route to each prefix. When I receive an error from the fist carrier, 
the opensips set the missed_call flag, and send the call to second carrier. If 
the second carrier send an error, the opensips set the missed_call flag again 
and relay this error to the user that made the call. In my missed_call table, I 
just see the second error, but I need to see both. How can I do this ?

Follow my failure route:

failure_route[1] {

xlog(FailureRoute entered);
setflag(10); #accounting missed calls
if (t_check_status((487)|(486))) {
xlogFailureRoute: $rm exited);
exit;
} else if(!next_routing()) {
xlog(FailureRoute: no more gateways available.);
}
route(1);
}


route[1] {

t_on_reply(ONREPLY);
if (!t_relay()) {
sl_reply_error();
};

exit;
}

Regards,

-- 
Wesley Volcov
Email: wesleyvol...@gmail.com
Messenger: vol...@live.com
Mobile: +55 11 9989-5348
Website: http://volcov.blogspot.com

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Re: [OpenSIPS-Users] missed_calls doubt

[Denis Putyato]

I am using and failed_transaction_flag and db_missed_flag 

for marking calls and I do not see any problem with sip_code while using 
next_routing()

 

From: Wesley Volcov [mailto:wesleyvol...@gmail.com] 
Sent: Thursday, April 28, 2011 4:33 PM
To: Denis Putyato
Cc: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] missed_calls doubt

 

Denis,

I already tried this, but sip_code field doens't have value.  I need this value.

Any ideas ?

Thanks

On 28 April 2011 09:17, Denis Putyato denis7...@mail.ru wrote:

Hello

 

Try to use failed_transaction_flag

http://www.opensips.org/html/docs/modules/devel/acc.html#id292642

 

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Wesley Volcov
Sent: Thursday, April 28, 2011 4:13 PM
To: OpenSIPS users mailling list
Subject: [OpenSIPS-Users] missed_calls doubt

 

Dear List,

I use my opensips, to make the routes based on prefix received and send the 
call to the termination carrier based on a lcr configured with dr_route module.
I have 2 route to each prefix. When I receive an error from the fist carrier, 
the opensips set the missed_call flag, and send the call to second carrier. If 
the second carrier send an error, the opensips set the missed_call flag again 
and relay this error to the user that made the call. In my missed_call table, I 
just see the second error, but I need to see both. How can I do this ?

Follow my failure route:

failure_route[1] {

xlog(FailureRoute entered);
setflag(10); #accounting missed calls
if (t_check_status((487)|(486))) {
xlogFailureRoute: $rm exited);
exit;
} else if(!next_routing()) {
xlog(FailureRoute: no more gateways available.);
}
route(1);
}


route[1] {

t_on_reply(ONREPLY);
if (!t_relay()) {
sl_reply_error();
};

exit;
}

Regards,

-- 
Wesley Volcov
Email: wesleyvol...@gmail.com
Messenger: vol...@live.com
Mobile: +55 11 9989-5348
Website: http://volcov.blogspot.com




-- 
Wesley Volcov
Email: wesleyvol...@gmail.com
Messenger: vol...@live.com
Mobile: +55 11 9989-5348
Website: http://volcov.blogspot.com

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[OpenSIPS-Users] memcach

[Denis Putyato]

Hello!

 

Sorry if my questions already appeared in mail list but

 

1)  If I don`t use timeout in cache_store func. then record in cache will 
live “forever” ?

2)  If I try to cache_store attribute which already has record in cache 
then this attribute will be  rewritten?

 

Thank you for help. 

 

 

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Re: [OpenSIPS-Users] memcach

[Denis Putyato]

Thank you very much!

 

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Kamen Petrov
Sent: Tuesday, April 19, 2011 5:09 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] memcach

 

Your questions are related more to the memcache rather than the opensips

anyway, yes on both of them :)





On 19 April 2011 15:12, Denis Putyato denis7...@mail.ru wrote:

Hello!

 

Sorry if my questions already appeared in mail list but

 

1)  If I don`t use timeout in cache_store func. then record in cache will 
live “forever” ?

2)  If I try to cache_store attribute which already has record in cache 
then this attribute will be  rewritten?

 

Thank you for help. 

 

 


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Re: [OpenSIPS-Users] dialog and CANCEL

[Denis Putyato]

Hello Bogdan

 

And if I use in onreply_route 

 

if (status=~200||18[0,3]  $rm==INVITE) {

if (t_was_cancelled()) {

   exit;

  }

}

 

This will help?

 

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Wednesday, April 06, 2011 6:35 PM
To: OpenSIPS users mailling list
Cc: Denis Putyato
Subject: Re: [OpenSIPS-Users] dialog and CANCEL

 

Hi Denis,

On 03/21/2011 01:55 PM, Denis Putyato wrote: 

Hello

 

There is such scheme of call

 

One gateway – 1.1.1.1

Opensips – 2.2.2.2

Another gateway – 3.3.3.3

Calls from 1.1.1.1 to 3.3.3.3 through 2.2.2.2

I use CDR_flag for accounting

 

As you can see in testlog file, 1.1.1.1 trying cancel initial request by 
sending CANCEL, this CANCEL Opensips forwarding to 3.3.3.3 but from 3.3.3.3 
Opensips receives 200 OK on INVITE. Because of this there is no CANCEL of the 
dialog on Opensips and after 1800 sec (see “default_timeout”) I have a CDR 
record in Opensips with duration of 1800 sec.

 

The question. Why does Opensips forward 200 OK from 3.3.3.3 to 1.1.1.1 when 
initial request was cancelled, and why Opensips makes accounting dialog when 
initial request was cancelled?  

RFC3261 says a proxy must forward all 2xx replies (disregarding the transaction 
state), just to solve the possible race between CANCEL and 2xx - such race 
must be handled by end point and not by proxy.

So, it your case, if caller sent a CANCEL but still receives a 200 (callee 
picked up before actually receiving the CANCEL from caller), the caller must 
sent a BYE and the callee should send a negative reply to the CANCEL.

So, it is a bug in the caller device.

Regards,
Bogdan




-- 
Bogdan-Andrei Iancu
OpenSIPS eBootcamp - 2nd of May 2011
OpenSIPS solutions and know-how
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[OpenSIPS-Users] dialog and accounting problem

[Denis Putyato]

Hello 

 

In SIP trace 

1.1.1.1– callee

2.2.2.2 – Opensips

3.3.3.3 – callee

 

I have Opensips 1.6.4-2.

….

modparam(dialog, hash_size, 4096)

modparam(dialog, log_profile_hash_size, 12)

modparam(dialog, default_timeout, 1800)

modparam(dialog, timeout_avp, $avp(i:995))

modparam(dialog, dlg_match_mode, 1)

modparam(dialog, db_mode, 1)

modparam(dialog, db_url, mysql://:@localhost/)

modparam(dialog, profiles_with_value, 
client;tgrp;tgrpin;tgrpout;answer;outdir;outdiranswer)

modparam(dialog, profiles_no_value, callin;callout)

….

modparam(acc, early_media, 0)

modparam(acc, report_ack, 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)

modparam(acc, db_url, mysql://:@localhost/)

modparam(acc, 
db_extra,src_in=$avp(i:600);src_user=$avp(i:500);src_domain=$si;

 
src_out=$avp(i:30);dst_in=$avp(s:dstin);dst_user=$avp(s:callee);dst_out=$avp(s:out);dst_domain=$avp(s:domain))

…..

 

route {

 

  if (is_method(BYE)) xlog(L_INFO, ….); 

  

  if (has_totag()) {

 if (is_method(BYE)) xlog(L_INFO, ….); 

 

   record_route();

   if (loose_route()) {

   

   if (is_method(BYE)) xlog(L_INFO, ….); 

   

if (!$DLG_status == NULL) {

 if (is_method(BYE)) {

 xlog(L_INFO, ….);

…

}

}

…

}

For accounting purposes I am using cdr_flag.

 

For the certain call, the SIP trace of which you can see in attachment, there 
is $avp(i:995) = . The call was successful, duration is about 50 s (if you 
see SIP trace). but in acc table I have a record with duration 10045. As you 
can see Opensips tries to finish the call  by sending BYE to both callee and 
caller after timeout of  $avp(i:995) expired although BYE from callee has been 
received before and has been successfully sent by Opensips to caller.  And as I 
suppose Opensips for some reason didn’t indicate the end of call when received 
first BYE.

All 4 xlog(L_INFO, ….); for the first BYE I can see in log file of Opensips.

 

Thank you for any help

 

U 2011/03/24 09:32:44.180633 1.1.1.1:59196 - 2.2.2.2:5060
INVITE sip:78124485322@2.2.2.2:5060 SIP/2.0.
Via: SIP/2.0/UDP 1.1.1.1:5060;x-route-tag=tgrp:TFOP;branch=z9hG4bK21E40ED7.
From: sip:8123215695@1.1.1.1;tag=9CBBDF4-D2B.
To: sip:78124485322@2.2.2.2.
Date: Thu, 24 Mar 2011 06:32:40 GMT.
Call-ID: 59DD6AAB-551711E0-9BFBA2C8-C63BD640@1.1.1.1.
Supported: 100rel,timer,resource-priority,replaces,sdp-anat.
Min-SE:  1800.
Cisco-Guid: 1507563059-1427575264-2679242786-2438473264.
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: 1300948360.
Contact: sip:8123215695@1.1.1.1:5060.
Expires: 60.
Allow-Events: telephone-event.
P-Asserted-Identity: sip:8123215695@1.1.1.1.
Content-Type: application/sdp.
Content-Disposition: session;handling=required.
Content-Length: 422.
.
v=0.
o=CiscoSystemsSIP-GW-UserAgent 9140 2676 IN IP4 1.1.1.1.
s=SIP Call.
c=IN IP4 1.1.1.1.
t=0 0.
m=audio 21598 RTP/AVP 8 0 18 4 98 3 101.
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:4 G723/8000.
a=fmtp:4 bitrate=6.3;annexa=yes.
a=rtpmap:98 G726-32/8000.
a=rtpmap:3 GSM/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.


U 2011/03/24 09:32:44.180909 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;x-route-tag=tgrp:TFOP;branch=z9hG4bK21E40ED7.
From: sip:8123215695@1.1.1.1;tag=9CBBDF4-D2B.
To: sip:78124485322@2.2.2.2.
Call-ID: 59DD6AAB-551711E0-9BFBA2C8-C63BD640@1.1.1.1.
CSeq: 101 INVITE.
Content-Length: 0.
.


U 2011/03/24 09:32:44.181410 2.2.2.2:5060 - 3.3.3.3:5060
INVITE sip:78124485322@3.3.3.3:5060 SIP/2.0.
Record-Route: sip:2.2.2.2;lr=on;ftag=9CBBDF4-D2B;did=9a3.8e2c5044.
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKb72a.96697e7.0.
Via: SIP/2.0/UDP 1.1.1.1:5060;x-route-tag=tgrp:TFOP;branch=z9hG4bK21E40ED7.
From: sip:8123215695@1.1.1.1;tag=9CBBDF4-D2B.
To: sip:78124485322@2.2.2.2.
Date: Thu, 24 Mar 2011 06:32:40 GMT.
Call-ID: 59DD6AAB-551711E0-9BFBA2C8-C63BD640@1.1.1.1.
Supported: 100rel,timer,resource-priority,replaces,sdp-anat.
Min-SE:  1800.
Cisco-Guid: 1507563059-1427575264-2679242786-2438473264.
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: 1300948360.
Contact: sip:8123215695@1.1.1.1:5060.
Remote-Party-ID:sip:8123215695@1.1.1.1;party=calling;screen=yes;privacy=off
Expires: 60.
Allow-Events: telephone-event.
P-Asserted-Identity: sip:8123215695@1.1.1.1.
Content-Type: application/sdp.
Content-Disposition: session;handling=required.
Content-Length: 422.

[OpenSIPS-Users] dialog and CANCEL

[Denis Putyato]

Hello

 

There is such scheme of call

 

One gateway – 1.1.1.1

Opensips – 2.2.2.2

Another gateway – 3.3.3.3

Calls from 1.1.1.1 to 3.3.3.3 through 2.2.2.2

I use CDR_flag for accounting

 

A piece of script config:

…

modparam(dialog, default_timeout, 1800)

…

…

onreply_route[1] {

if (t_was_cancelled()) {

   exit;

  }

if (status=~200||18[0,3]) {

if (isflagset(10)  has_body(application/sdp)) {

  rtpproxy_answer(con);

  }

  if (isflagset(21)  nat_uac_test(55)) fix_nated_contact();

   store_dlg_value(calleeip,$si);

   store_dlg_value(calleeport,$sp);

   store_dlg_value(calleecont,$ct.fields(uri));

   if (status=~200  $rm==INVITE  !isflagset(29)) { 

   set_dlg_profile(answer,$avp(i:71));

   set_dlg_profile(outdiranswer,$avp(i:3));

   }

  }

  return();

}

….

 

As you can see in testlog file, 1.1.1.1 trying cancel initial request by 
sending CANCEL, this CANCEL Opensips forwarding to 3.3.3.3 but from 3.3.3.3 
Opensips receives 200 OK on INVITE. Because of this there is no CANCEL of the 
dialog on Opensips and after 1800 sec (see “default_timeout”) I have a CDR 
record in Opensips with duration of 1800 sec.

 

The question. Why does Opensips forward 200 OK from 3.3.3.3 to 1.1.1.1 when 
initial request was cancelled, and why Opensips makes accounting dialog when 
initial request was cancelled?  

 

 

 

 



testlog
Description: Binary data
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[OpenSIPS-Users] dialog trouble

[Denis Putyato]

Hello

 

I attached an ngrep dump.

After this call I have accounting record with duration 1800 sec (I am using CDR 
flag accounting).

(…

modparam(dialog, default_timeout, 1800)

…)

 

Calls from 1.1.1.1 to 3.3.3.3 through 2.2.2.2 

2.2.2.2 – Opensips

1.1.1.1 and 3.3.3.3 – some gateways

 

Why this may happened?

 

P.S. Opensips 1.6.4-2

 

U 2011/03/21 11:19:17.193692 1.1.1.1:5060 - 2.2.2.2:5060
INVITE sip:84954462754@2.2.2.2 SIP/2.0.
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bKf74df06da419.
From: sip:826@1.1.1.1;tag=f74df06da4.
To: sip:84954462754@2.2.2.2.
Call-ID: f795724d-3ac1-f04e-936d-0002a404d1a6@1.1.1.1.
CSeq: 19 INVITE.
Supported: timer, replaces.
Min-SE: 1800.
Date: Sat, 05 Mar 2011 19:58:47 GMT.
User-Agent: AddPac SIP Gateway.
Contact: sip:826@1.1.1.1.
Accept: application/sdp.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO.
Content-Type: application/sdp.
Content-Length: 234.
Max-Forwards: 70.
.
v=0.
o=826 1299355127 1299355127 IN IP4 1.1.1.1.
s=AddPac Gateway SDP.
c=IN IP4 1.1.1.1.
t=1299355127 0.
m=audio 23680 RTP/AVP 4 18 0.
a=rtpmap:4 G723/8000.
a=rtpmap:18 G729/8000.
a=rtpmap:0 PCMU/8000.
a=ptime:30.


U 2011/03/21 11:19:17.194215 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=z9hG4bKf74df06da419.
From: sip:826@1.1.1.1;tag=f74df06da4.
To: sip:84954462754@2.2.2.2.
Call-ID: f795724d-3ac1-f04e-936d-0002a404d1a6@1.1.1.1.
CSeq: 19 INVITE.
Content-Length: 0.
.

U 2011/03/21 11:19:17.229835 2.2.2.2:5060 - 3.3.3.3:5060
INVITE sip:74954462754@3.3.3.3:5060 SIP/2.0.
Record-Route: sip:2.2.2.2;lr=on;ftag=f74df06da4;did=af3.abbf9015.
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK02bf.707ef165.2.
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bKf74df06da419.
From: sip:826@1.1.1.1;tag=f74df06da4.
To: sip:84954462754@2.2.2.2.
Call-ID: f795724d-3ac1-f04e-936d-0002a404d1a6@1.1.1.1.
CSeq: 19 INVITE.
Supported: timer, replaces.
Min-SE: 1800.
Date: Sat, 05 Mar 2011 19:58:47 GMT.
User-Agent: AddPac SIP Gateway.
Contact: sip:826@1.1.1.1.
P-Asserted-Identity: sip:78123364412@1.1.1.1
Remote-Party-ID: sip:78123364412@1.1.1.1;party=calling;screen=yes;privacy=full
Accept: application/sdp.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO.
Content-Type: application/sdp.
Content-Length: 252.
Max-Forwards: 70.
Privacy: user.
.
v=0.
o=826 1299355127 1299355127 IN IP4 2.2.2.2.
s=AddPac Gateway SDP.
c=IN IP4 2.2.2.2.
t=1299355127 0.
m=audio 64622 RTP/AVP 4 18 0.
a=rtpmap:4 G723/8000.
a=rtpmap:18 G729/8000.
a=rtpmap:0 PCMU/8000.
a=ptime:30.
a=nortpproxy:yes.


U 2011/03/21 11:19:17.257578 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=z9hG4bK02bf.707ef165.2.
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bKf74df06da419.
From: sip:826@1.1.1.1;tag=f74df06da4.
To: sip:84954462754@2.2.2.2.
Call-ID: f795724d-3ac1-f04e-936d-0002a404d1a6@1.1.1.1.
CSeq: 19 INVITE.
Content-Length: 0.
.


U 2011/03/21 11:19:17.469819 3.3.3.3:5060 - 2.2.2.2:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK02bf.707ef165.2.
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bKf74df06da419.
From: sip:826@1.1.1.1;tag=f74df06da4.
To: sip:84954462754@2.2.2.2;tag=ff464500020aff10ff24.
Call-ID: f795724d-3ac1-f04e-936d-0002a404d1a6@1.1.1.1.
CSeq: 19 INVITE.
Contact: sip:74954462754@3.3.3.3:5060;user=phone;transport=udp.
Server: MERA MSIP v.1.0.2.
Content-Length: 0.
Record-Route: sip:2.2.2.2;lr=on;ftag=f74df06da4;did=af3.abbf9015.
.


U 2011/03/21 11:19:17.482978 2.2.2.2:5060 - 1.1.1.1:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bKf74df06da419.
From: sip:826@1.1.1.1;tag=f74df06da4.
To: sip:84954462754@2.2.2.2;tag=ff464500020aff10ff24.
Call-ID: f795724d-3ac1-f04e-936d-0002a404d1a6@1.1.1.1.
CSeq: 19 INVITE.
Contact: sip:74954462754@3.3.3.3:5060;user=phone;transport=udp.
Server: MERA MSIP v.1.0.2.
Content-Length: 0.
Record-Route: sip:2.2.2.2;lr=on;ftag=f74df06da4;did=af3.abbf9015.
.


U 2011/03/21 11:19:28.637401 3.3.3.3:5060 - 2.2.2.2:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK02bf.707ef165.2.
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bKf74df06da419.
From: sip:826@1.1.1.1;tag=f74df06da4.
To: sip:84954462754@2.2.2.2;tag=ff464500020aff10ff24.
Call-ID: f795724d-3ac1-f04e-936d-0002a404d1a6@1.1.1.1.
CSeq: 19 INVITE.
Contact: sip:74954462754@3.3.3.3:5060;user=phone;transport=udp.
Server: MERA MSIP v.1.0.2.
Content-Type: application/sdp.
Content-Length: 186.
Record-Route: sip:2.2.2.2;lr=on;ftag=f74df06da4;did=af3.abbf9015.
.
v=0.
o=- 1300695565 1300695565 IN IP4 3.3.3.3.
s=-.
c=IN IP4 3.3.3.3.
t=0 0.
m=audio 36904 RTP/AVP 4 18 0.
a=rtpmap:4 G723/8000.
a=rtpmap:18 G729/8000.
a=rtpmap:0 PCMU/8000.


U 2011/03/21 11:19:28.638364 2.2.2.2:5060 - 1.1.1.1:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bKf74df06da419.
From: sip:826@1.1.1.1;tag=f74df06da4.
To: sip:84954462754@2.2.2.2;tag=ff464500020aff10ff24.

Re: [OpenSIPS-Users] opensips 1.6.4 out of memory

[Denis Putyato]

Hello

 

I had the same problem on 1.6.4, you should use 1.6.4-2 version

 

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Iulian Macare
Sent: Friday, March 11, 2011 12:24 PM
To: users@lists.opensips.org
Subject: [OpenSIPS-Users] opensips 1.6.4 out of memory

 

2-3 times per day my opensips configuration with 300 channels and a load 
balancer fails ; It's opensips 1.6.4 on centos 5.5. 32 bit 

The erors I get are :

Any ideas?

Mar 11 11:15:08 opensipsh /usr/local/sbin/opensips[1718]: ERROR:tm:new_t: out 
of mem
Mar 11 11:15:08 opensipsh /usr/local/sbin/opensips[1701]: 
ERROR:tm:sip_msg_cloner: no more share memory
Mar 11 11:15:08 opensipsh /usr/local/sbin/opensips[1716]: 
ERROR:dialog:dlg_create_dialog: could not add further info to the dialog
Mar 11 11:15:08 opensipsh /usr/local/sbin/opensips[1706]: 
ERROR:dialog:dlg_create_dialog: could not add further info to the dialog
Mar 11 11:15:08 opensipsh /usr/local/sbin/opensips[1727]: ERROR:tm:t_newtran: 
new_t failed
Mar 11 11:15:08 opensipsh /usr/local/sbin/opensips[1704]: ERROR:tm:t_newtran: 
new_t failed
Mar 11 11:15:08 opensipsh /usr/local/sbin/opensips[1700]: 
ERROR:load_balancer:do_load_balance: failed to create dialog
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1698]: ERROR:tm:t_newtran: 
new_t failed
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1723]: ERROR:tm:t_newtran: 
new_t failed
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1725]: 
ERROR:core:print_rr_body: too many RR
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1708]: 
ERROR:dialog:dlg_add_leg_info: Failed to resize legs array
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1712]: 
ERROR:dialog:dlg_add_leg_info: Failed to resize legs array
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1714]: 
ERROR:dialog:dlg_create_dialog: could not add further info to the dialog
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1720]: ERROR:tm:t_newtran: 
new_t failed
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1721]: ERROR:tm:new_t: out 
of mem
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1710]: 
ERROR:tm:sip_msg_cloner: no more share memory
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1718]: 
ERROR:tm:sip_msg_cloner: no more share memory
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1701]: ERROR:tm:t_newtran: 
new_t failed
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1716]: 
ERROR:dialog:init_leg_info: dlg_add_leg_info failed
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1706]: 
ERROR:dialog:init_leg_info: dlg_add_leg_info failed
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1727]: ERROR:tm:new_t: out 
of mem
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1704]: ERROR:tm:new_t: out 
of mem
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1700]: 
ERROR:dialog:dlg_create_dialog: could not add further info to the dialog
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1698]: ERROR:tm:new_t: out 
of mem
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1723]: ERROR:tm:new_t: out 
of mem
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1708]: ERROR:tm:t_newtran: 
new_t failed
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1712]: ERROR:tm:t_newtran: 
new_t failed
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1701]: ERROR:tm:t_newtran: 
new_t failed
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1716]: ERROR:tm:new_t: out 
of mem
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1706]: ERROR:tm:new_t: out 
of mem
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1727]: 
ERROR:dialog:get_routing_info: failed to print route records
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1725]: 
ERROR:dialog:dlg_add_leg_info: Failed to resize legs array
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1704]: 
ERROR:dialog:get_routing_info: failed to print route records
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1700]: 
ERROR:dialog:get_routing_info: failed to print route records
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1698]: 
ERROR:dialog:get_routing_info: failed to print route records
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1723]: 
ERROR:dialog:dlg_add_leg_info: no more shm mem
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1708]: 
ERROR:dialog:dlg_add_leg_info: Failed to resize legs array
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1712]: 
ERROR:dialog:dlg_add_leg_info: Failed to resize legs array
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1714]: 
ERROR:dialog:build_new_dlg: no more shm mem (202)
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1720]: 
ERROR:tm:shm_clone_proxy: no more shm memory
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1721]: 
ERROR:tm:sip_msg_cloner: no more share memory
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1710]: 
ERROR:dialog:dlg_add_leg_info: Failed to resize legs array
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1718]: 
ERROR:core:build_req_buf_from_sip_req: out of 

Re: [OpenSIPS-Users] Script flag question

[Denis Putyato]

Hello Bogdan

Yes I am sure, because all exit offered after send_reply() function, i.e. 
call is unsuccessful.
But my test call pass through Opensips and goes to callee. 

-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
Sent: Friday, March 04, 2011 6:58 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Script flag question

Hi Denis,

are you sure that your script flow does not hit any of those exit 
statements before getting to the second xlog() ?

Regards,
Bogdan

Denis Putyato wrote:

 Hello everybody!

 There is a piece of script

 …

 …

 if (dp_translate(20005, $rU/$rU)) {

 xlog(L_INFO, RU after alias = $rU);

 $avp(i:200)=$rU;

 setsflag(1);

 if (issflagset(1)) xlog(L_INFO, FLAGS1 is set);

 }

 if (dp_translate(2, $rU/$var(ruri))) {

 $avp(i:502)=$avp(i:999);

 avp_subst($avp(i:502), /(.*)\*(.*)\*(.*)/\3/ig);

 if ($avp(i:502)==0) {

 $avp(i:500)=$rU;

 $rU=$var(ruri);

 $avp(i:5)=$avp(i:999);

 avp_subst($avp(i:5), /(.*)\*(.*)\*(.*)/\1/ig);

 $avp(i:5)=$(avp(i:5){s.int});

 $avp(i:1005)=1;

 $avp(s:callee)=$rU;

 $avp(i:92)=$avp(i:999);

 avp_subst($avp(i:92), /(.*)\*(.*)\*(.*)/\2/ig);

 $avp(i:92)=$(avp(i:92){s.int});

 get_profile_size(client,$avp(i:500),$avp(i:17));

 if ($avp(i:17) = $avp(i:92)) {

 acc_db_request(User busy, acc);

 send_reply(486, User busy);

 exit;

 }

 set_dlg_profile(client,$avp(i:500));

 setsflag(2);

 }

 }

 if (do_routing($avp(i:5))) setsflag(3);

 if (!issflagset(3)) {

 if (!$avp(i:6) == 0) {

 if (do_routing($avp(i:6))) setsflag(3);

 }

 }

 if (!issflagset(3)) {

 if (!$avp(i:7) == 0) {

 if (!do_routing($avp(i:7))) {

 acc_db_request(Not found, acc);

 send_reply(404, Not found);

 exit;

 }

 } else {

 acc_db_request(Not found, acc);

 send_reply(404, Not found);

 exit;

 }

 }

 if (issflagset(1)) xlog(L_INFO, FLAGS1 is set);

 …

 …

 I can see first FLAGS1 is set in log file but I don`t see second 
 FLAGS1 is set.

 What can be wrong?

 

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


-- 
Bogdan-Andrei Iancu
OpenSIPS eBootcamp - 28th February 2011
OpenSIPS solutions and know-how


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


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Re: [OpenSIPS-Users] Dialog module app callback hooks not saved to DB

[Denis Putyato]

Hello, Bogdan

When can I see information about the bug?

Thank you

-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
Sent: Friday, February 25, 2011 6:33 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Dialog module app callback hooks not saved to DB

Hi Dave,

it is a know issue, so please reported as a bug, to keep trace and fix it.

Best regards,
Bogdan

Dave Singer wrote:
 I tested setting up acc module to use the cdr_flag with dialog module.
 Works nicely until opensips is restarted while there is an open call.
 After an opensips restart, calls that were started before the restart 
 do not get an entry in the DB. Not even an old style BYE record.
 I verified that the dialog was matched using:
   xlog(L_NOTICE, CID-$ci, fU-$fU, orU-$oU: Loose Route $rm 
 Request - si=$si - next via $rd:$rp\n);
 if ( $DLG_status == NULL ) {
 xlog(L_WARN, CID-$ci, fU-$fU, orU-$oU: WARNING: BYE 
 without dialog. NO CDR, just old style rec for BYE in acc table.\n);
 in the loose routing logic that handles the bye.
 I get the first message but not the warning.
 I also ran into this problem previously with media_proxy_engage. It 
 cleans itself up because of no RTP after a bit though and so is not as 
 big of a deal.

 Am I missing something or is this a bug I need to report?

 Dave

 

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


-- 
Bogdan-Andrei Iancu
OpenSIPS eBootcamp - 28th February 2011
OpenSIPS solutions and know-how


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


___
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Users@lists.opensips.org
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Re: [OpenSIPS-Users] Dialog module app callback hooks not saved to DB

[Denis Putyato]

Thank you very much!

-Original Message-
From: sip.n...@gmail.com [mailto:sip.n...@gmail.com] On Behalf Of Ovidiu Sas
Sent: Thursday, March 03, 2011 4:14 PM
To: OpenSIPS users mailling list
Cc: Denis Putyato
Subject: Re: [OpenSIPS-Users] Dialog module app callback hooks not saved to DB

On the main opensips.org page there is a section called Development
(left bottom corner).
Click on Tracker and follow the links.

Regards,
Ovidiu Sas

On Thu, Mar 3, 2011 at 7:54 AM, Denis Putyato denis7...@mail.ru wrote:
 Hello, Bogdan

 When can I see information about the bug?

 Thank you

 -Original Message-
 From: users-boun...@lists.opensips.org 
 [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
 Sent: Friday, February 25, 2011 6:33 PM
 To: OpenSIPS users mailling list
 Subject: Re: [OpenSIPS-Users] Dialog module app callback hooks not saved to DB

 Hi Dave,

 it is a know issue, so please reported as a bug, to keep trace and fix it.

 Best regards,
 Bogdan

 Dave Singer wrote:
 I tested setting up acc module to use the cdr_flag with dialog module.
 Works nicely until opensips is restarted while there is an open call.
 After an opensips restart, calls that were started before the restart
 do not get an entry in the DB. Not even an old style BYE record.
 I verified that the dialog was matched using:
   xlog(L_NOTICE, CID-$ci, fU-$fU, orU-$oU: Loose Route $rm
 Request - si=$si - next via $rd:$rp\n);
 if ( $DLG_status == NULL ) {
 xlog(L_WARN, CID-$ci, fU-$fU, orU-$oU: WARNING: BYE
 without dialog. NO CDR, just old style rec for BYE in acc table.\n);
 in the loose routing logic that handles the bye.
 I get the first message but not the warning.
 I also ran into this problem previously with media_proxy_engage. It
 cleans itself up because of no RTP after a bit though and so is not as
 big of a deal.

 Am I missing something or is this a bug I need to report?

 Dave

 

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



 --
 Bogdan-Andrei Iancu
 OpenSIPS eBootcamp - 28th February 2011
 OpenSIPS solutions and know-how


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


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



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

Re: [OpenSIPS-Users] Dialog module app callback hooks not saved to DB

[Denis Putyato]

Thank you very much!

-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
Sent: Thursday, March 03, 2011 4:14 PM
To: OpenSIPS users mailling list; Razvan Crainea
Subject: Re: [OpenSIPS-Users] Dialog module app callback hooks not saved to DB

Hi Denis,

This is work on progress from Razvan - he will update you soon.

Regards,
Bogdan

Denis Putyato wrote:
 Hello, Bogdan

 When can I see information about the bug?

 Thank you

 -Original Message-
 From: users-boun...@lists.opensips.org 
 [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
 Sent: Friday, February 25, 2011 6:33 PM
 To: OpenSIPS users mailling list
 Subject: Re: [OpenSIPS-Users] Dialog module app callback hooks not saved to DB

 Hi Dave,

 it is a know issue, so please reported as a bug, to keep trace and fix it.

 Best regards,
 Bogdan

 Dave Singer wrote:
   
 I tested setting up acc module to use the cdr_flag with dialog module.
 Works nicely until opensips is restarted while there is an open call.
 After an opensips restart, calls that were started before the restart 
 do not get an entry in the DB. Not even an old style BYE record.
 I verified that the dialog was matched using:
   xlog(L_NOTICE, CID-$ci, fU-$fU, orU-$oU: Loose Route $rm 
 Request - si=$si - next via $rd:$rp\n);
 if ( $DLG_status == NULL ) {
 xlog(L_WARN, CID-$ci, fU-$fU, orU-$oU: WARNING: BYE 
 without dialog. NO CDR, just old style rec for BYE in acc table.\n);
 in the loose routing logic that handles the bye.
 I get the first message but not the warning.
 I also ran into this problem previously with media_proxy_engage. It 
 cleans itself up because of no RTP after a bit though and so is not as 
 big of a deal.

 Am I missing something or is this a bug I need to report?

 Dave

 

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


   


-- 
Bogdan-Andrei Iancu
OpenSIPS eBootcamp - 28th February 2011
OpenSIPS solutions and know-how


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


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

Re: [OpenSIPS-Users] Dialog module app callback hooks not saved to DB

[Denis Putyato]

Hello Razvan

During restart
Mar  3 18:08:38 opensips /usr/local/opensips1.6.4-2/sbin/opensips[22252]: 
ERROR:acc:acc_loaded_callback: cannot fetch flags string value

usr/local/opensips1.6.4-2/sbin/opensipsctl fifo ps
Process::  ID=0 PID=22252 Type=attendant
Process::  ID=1 PID=22254 Type=RTPP timeout receiver
Process::  ID=2 PID=22255 Type=MI FIFO
Process::  ID=3 PID=22256 Type=SIP receiver udp:213.170.100.150:5060 
Process::  ID=4 PID=22257 Type=SIP receiver udp:213.170.100.150:5068 
Process::  ID=5 PID=22258 Type=time_keeper
Process::  ID=6 PID=22259 Type=timer
Process::  ID=7 PID=22260 Type=timer
Process::  ID=8 PID=22261 Type=TCP receiver
Process::  ID=9 PID=22262 Type=TCP main

-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Razvan Crainea
Sent: Thursday, March 03, 2011 5:50 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Dialog module app callback hooks not saved to DB

Hello Denis,

I have just added a patch in trunk that fixes this issue. Please update 
your sources and let me know if the problem persists.

Regards,
Razvan

On 03/03/2011 03:21 PM, Denis Putyato wrote:
 Thank you very much!

 -Original Message-
 From: sip.n...@gmail.com [mailto:sip.n...@gmail.com] On Behalf Of Ovidiu Sas
 Sent: Thursday, March 03, 2011 4:14 PM
 To: OpenSIPS users mailling list
 Cc: Denis Putyato
 Subject: Re: [OpenSIPS-Users] Dialog module app callback hooks not saved to DB

 On the main opensips.org page there is a section called Development
 (left bottom corner).
 Click on Tracker and follow the links.

 Regards,
 Ovidiu Sas

 On Thu, Mar 3, 2011 at 7:54 AM, Denis Putyatodenis7...@mail.ru  wrote:
 Hello, Bogdan

 When can I see information about the bug?

 Thank you

 -Original Message-
 From: users-boun...@lists.opensips.org 
 [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
 Sent: Friday, February 25, 2011 6:33 PM
 To: OpenSIPS users mailling list
 Subject: Re: [OpenSIPS-Users] Dialog module app callback hooks not saved to 
 DB

 Hi Dave,

 it is a know issue, so please reported as a bug, to keep trace and fix it.

 Best regards,
 Bogdan

 Dave Singer wrote:
 I tested setting up acc module to use the cdr_flag with dialog module.
 Works nicely until opensips is restarted while there is an open call.
 After an opensips restart, calls that were started before the restart
 do not get an entry in the DB. Not even an old style BYE record.
 I verified that the dialog was matched using:
xlog(L_NOTICE, CID-$ci, fU-$fU, orU-$oU: Loose Route $rm
 Request - si=$si - next via $rd:$rp\n);
  if ( $DLG_status == NULL ) {
  xlog(L_WARN, CID-$ci, fU-$fU, orU-$oU: WARNING: BYE
 without dialog. NO CDR, just old style rec for BYE in acc table.\n);
 in the loose routing logic that handles the bye.
 I get the first message but not the warning.
 I also ran into this problem previously with media_proxy_engage. It
 cleans itself up because of no RTP after a bit though and so is not as
 big of a deal.

 Am I missing something or is this a bug I need to report?

 Dave

 

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


 --
 Bogdan-Andrei Iancu
 OpenSIPS eBootcamp - 28th February 2011
 OpenSIPS solutions and know-how


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


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


 ___
 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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Users mailing list
Users@lists.opensips.org
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Re: [OpenSIPS-Users] Dialog module app callback hooks not saved to DB

[Denis Putyato]

Razvan

I re-built them both and receive the ERROR


-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Razvan Crainea
Sent: Thursday, March 03, 2011 6:33 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Dialog module app callback hooks not saved to DB

Hello Denis,

The patch makes some changes to acc and dialog modules. Please make sure 
you re-built them both.

Regards,
Razvan

On 03/03/2011 05:11 PM, Denis Putyato wrote:
 Hello Razvan

 During restart
 Mar  3 18:08:38 opensips /usr/local/opensips1.6.4-2/sbin/opensips[22252]: 
 ERROR:acc:acc_loaded_callback: cannot fetch flags string value

 usr/local/opensips1.6.4-2/sbin/opensipsctl fifo ps
 Process::  ID=0 PID=22252 Type=attendant
 Process::  ID=1 PID=22254 Type=RTPP timeout receiver
 Process::  ID=2 PID=22255 Type=MI FIFO
 Process::  ID=3 PID=22256 Type=SIP receiver udp:213.170.100.150:5060
 Process::  ID=4 PID=22257 Type=SIP receiver udp:213.170.100.150:5068
 Process::  ID=5 PID=22258 Type=time_keeper
 Process::  ID=6 PID=22259 Type=timer
 Process::  ID=7 PID=22260 Type=timer
 Process::  ID=8 PID=22261 Type=TCP receiver
 Process::  ID=9 PID=22262 Type=TCP main

 -Original Message-
 From: users-boun...@lists.opensips.org 
 [mailto:users-boun...@lists.opensips.org] On Behalf Of Razvan Crainea
 Sent: Thursday, March 03, 2011 5:50 PM
 To: OpenSIPS users mailling list
 Subject: Re: [OpenSIPS-Users] Dialog module app callback hooks not saved to DB

 Hello Denis,

 I have just added a patch in trunk that fixes this issue. Please update
 your sources and let me know if the problem persists.

 Regards,
 Razvan

 On 03/03/2011 03:21 PM, Denis Putyato wrote:
 Thank you very much!

 -Original Message-
 From: sip.n...@gmail.com [mailto:sip.n...@gmail.com] On Behalf Of Ovidiu Sas
 Sent: Thursday, March 03, 2011 4:14 PM
 To: OpenSIPS users mailling list
 Cc: Denis Putyato
 Subject: Re: [OpenSIPS-Users] Dialog module app callback hooks not saved to 
 DB

 On the main opensips.org page there is a section called Development
 (left bottom corner).
 Click on Tracker and follow the links.

 Regards,
 Ovidiu Sas

 On Thu, Mar 3, 2011 at 7:54 AM, Denis Putyatodenis7...@mail.ru   wrote:
 Hello, Bogdan

 When can I see information about the bug?

 Thank you

 -Original Message-
 From: users-boun...@lists.opensips.org 
 [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
 Sent: Friday, February 25, 2011 6:33 PM
 To: OpenSIPS users mailling list
 Subject: Re: [OpenSIPS-Users] Dialog module app callback hooks not saved to 
 DB

 Hi Dave,

 it is a know issue, so please reported as a bug, to keep trace and fix it.

 Best regards,
 Bogdan

 Dave Singer wrote:
 I tested setting up acc module to use the cdr_flag with dialog module.
 Works nicely until opensips is restarted while there is an open call.
 After an opensips restart, calls that were started before the restart
 do not get an entry in the DB. Not even an old style BYE record.
 I verified that the dialog was matched using:
 xlog(L_NOTICE, CID-$ci, fU-$fU, orU-$oU: Loose Route $rm
 Request - si=$si - next via $rd:$rp\n);
   if ( $DLG_status == NULL ) {
   xlog(L_WARN, CID-$ci, fU-$fU, orU-$oU: WARNING: BYE
 without dialog. NO CDR, just old style rec for BYE in acc table.\n);
 in the loose routing logic that handles the bye.
 I get the first message but not the warning.
 I also ran into this problem previously with media_proxy_engage. It
 cleans itself up because of no RTP after a bit though and so is not as
 big of a deal.

 Am I missing something or is this a bug I need to report?

 Dave

 

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

 --
 Bogdan-Andrei Iancu
 OpenSIPS eBootcamp - 28th February 2011
 OpenSIPS solutions and know-how


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


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

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 http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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-- 
Razvan Crainea
OpenSIPS Developer


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Re: [OpenSIPS-Users] Dialog module app callback hooks not saved to DB

[Denis Putyato]

Hello Razvan

Yes, I restarted opensips during fresh dialog ongoing and there is no ERROR.
Besides, there is a CDR record in DB after opensips restart

Thank you very much! 

-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Razvan Crainea
Sent: Thursday, March 03, 2011 7:37 PM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] Dialog module app callback hooks not saved to DB

Hello Denis,

The acc module needs a new variable stored in the database. If you did 
your test with your old dialog records, it is normal to appear that 
error. Please try to restart OpenSIPS with a fresh dialog ongoing and 
let me know if the ERROR still appears.

Regards,
Razvan

On 03/03/2011 06:00 PM, Denis Putyato wrote:
 Razvan

 I re-built them both and receive the ERROR


 -Original Message-
 From: users-boun...@lists.opensips.org 
 [mailto:users-boun...@lists.opensips.org] On Behalf Of Razvan Crainea
 Sent: Thursday, March 03, 2011 6:33 PM
 To: OpenSIPS users mailling list
 Subject: Re: [OpenSIPS-Users] Dialog module app callback hooks not saved to DB

 Hello Denis,

 The patch makes some changes to acc and dialog modules. Please make sure
 you re-built them both.

 Regards,
 Razvan

 On 03/03/2011 05:11 PM, Denis Putyato wrote:
 Hello Razvan

 During restart
 Mar  3 18:08:38 opensips /usr/local/opensips1.6.4-2/sbin/opensips[22252]: 
 ERROR:acc:acc_loaded_callback: cannot fetch flags string value

 usr/local/opensips1.6.4-2/sbin/opensipsctl fifo ps
 Process::  ID=0 PID=22252 Type=attendant
 Process::  ID=1 PID=22254 Type=RTPP timeout receiver
 Process::  ID=2 PID=22255 Type=MI FIFO
 Process::  ID=3 PID=22256 Type=SIP receiver udp:213.170.100.150:5060
 Process::  ID=4 PID=22257 Type=SIP receiver udp:213.170.100.150:5068
 Process::  ID=5 PID=22258 Type=time_keeper
 Process::  ID=6 PID=22259 Type=timer
 Process::  ID=7 PID=22260 Type=timer
 Process::  ID=8 PID=22261 Type=TCP receiver
 Process::  ID=9 PID=22262 Type=TCP main

 -Original Message-
 From: users-boun...@lists.opensips.org 
 [mailto:users-boun...@lists.opensips.org] On Behalf Of Razvan Crainea
 Sent: Thursday, March 03, 2011 5:50 PM
 To: OpenSIPS users mailling list
 Subject: Re: [OpenSIPS-Users] Dialog module app callback hooks not saved to 
 DB

 Hello Denis,

 I have just added a patch in trunk that fixes this issue. Please update
 your sources and let me know if the problem persists.

 Regards,
 Razvan

 On 03/03/2011 03:21 PM, Denis Putyato wrote:
 Thank you very much!

 -Original Message-
 From: sip.n...@gmail.com [mailto:sip.n...@gmail.com] On Behalf Of Ovidiu Sas
 Sent: Thursday, March 03, 2011 4:14 PM
 To: OpenSIPS users mailling list
 Cc: Denis Putyato
 Subject: Re: [OpenSIPS-Users] Dialog module app callback hooks not saved to 
 DB

 On the main opensips.org page there is a section called Development
 (left bottom corner).
 Click on Tracker and follow the links.

 Regards,
 Ovidiu Sas

 On Thu, Mar 3, 2011 at 7:54 AM, Denis Putyatodenis7...@mail.ruwrote:
 Hello, Bogdan

 When can I see information about the bug?

 Thank you

 -Original Message-
 From: users-boun...@lists.opensips.org 
 [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
 Sent: Friday, February 25, 2011 6:33 PM
 To: OpenSIPS users mailling list
 Subject: Re: [OpenSIPS-Users] Dialog module app callback hooks not saved 
 to DB

 Hi Dave,

 it is a know issue, so please reported as a bug, to keep trace and fix it.

 Best regards,
 Bogdan

 Dave Singer wrote:
 I tested setting up acc module to use the cdr_flag with dialog module.
 Works nicely until opensips is restarted while there is an open call.
 After an opensips restart, calls that were started before the restart
 do not get an entry in the DB. Not even an old style BYE record.
 I verified that the dialog was matched using:
  xlog(L_NOTICE, CID-$ci, fU-$fU, orU-$oU: Loose Route $rm
 Request - si=$si - next via $rd:$rp\n);
if ( $DLG_status == NULL ) {
xlog(L_WARN, CID-$ci, fU-$fU, orU-$oU: WARNING: BYE
 without dialog. NO CDR, just old style rec for BYE in acc table.\n);
 in the loose routing logic that handles the bye.
 I get the first message but not the warning.
 I also ran into this problem previously with media_proxy_engage. It
 cleans itself up because of no RTP after a bit though and so is not as
 big of a deal.

 Am I missing something or is this a bug I need to report?

 Dave

 

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

 --
 Bogdan-Andrei Iancu
 OpenSIPS eBootcamp - 28th February 2011
 OpenSIPS solutions and know-how


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

Re: [OpenSIPS-Users] dialog statistic problem

[Denis Putyato]

Hello Bogdan

I applied patch and it seems that хХх shows real number of dialogs while 
active statistic shows incorrect number

-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
Sent: Monday, February 28, 2011 5:55 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] dialog statistic problem

Hi Denis,

Please apply this small patch that will provide some extra on startup (about 
the dialogs found in DB) - look for a line with marked with xXx 
and see if the printed values are correct.

Regards,
Bogdan

Denis Putyato wrote:
 Hello Bogdan

 - Server:: OpenSIPS (1.6.4-2-notls (x86_64/linux))
 - fork = yes (default)
 -
 listen=xxx.xxx.xxx.xxx:5060
 listen=xxx.xxx.xxx.xxx:5068
 - i do not use tcp
 -
 loadmodule db_mysql.so
 loadmodule sl.so
 loadmodule tm.so
 loadmodule signaling.so
 loadmodule auth.so
 loadmodule rr.so
 loadmodule maxfwd.so
 loadmodule textops.so
 loadmodule mi_fifo.so
 loadmodule uri.so
 loadmodule domain.so
 loadmodule drouting.so
 loadmodule siptrace.so
 loadmodule avpops.so
 loadmodule dialplan.so
 loadmodule dialog.so
 loadmodule permissions.so
 loadmodule usrloc.so
 loadmodule registrar.so
 loadmodule alias_db.so
 loadmodule auth_db.so
 loadmodule nathelper.so
 loadmodule acc.so
 loadmodule uac.so
 loadmodule aaa_radius.so

 -Original Message-
 From: users-boun...@lists.opensips.org 
 [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei 
 Iancu
 Sent: Monday, February 28, 2011 1:17 PM
 To: OpenSIPS users mailling list
 Subject: Re: [OpenSIPS-Users] dialog statistic problem

 Hi Denis,

 What revision number are you using ?

 Also, do you have fork enabled?
 Do you use multiple UDP interfaces?
 Do you have TCP enabled ?
 What are the modules you are using ?

 Regards,
 Bogdan

 Denis Putyato wrote:
   
 Hello!

 I found active dialog statistic problem

 For example in dialog::active_dialogs I have 130 active calls, after 
 restart Opensips I already have dialog::active_dialogs nearly 260, 
 i.e. the number of active dialogs increase by 2.

 Meanwhile, after restart, opensipsctl fifo dlg_list | grep dialog:: | 
 wc –l shows me that really number of calls greatly less.

 The opensipsctl fifo reset_statistics didn`t help. Only when I stop 
 all traffic via Opensips (but not Opensips) and then restart one, 
 dialog::active_dialogs become zero and begin increase when new calls 
 come to Opensips

 Thank you for any help

 -
 ---

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


   


--
Bogdan-Andrei Iancu
OpenSIPS eBootcamp - 28th February 2011
OpenSIPS solutions and know-how



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

Re: [OpenSIPS-Users] dialog statistic problem

[Denis Putyato]

No, during restart xXx appears twice

$cat /var/log/opensips | grep xXx
Mar  1 07:21:05 opensips /usr/local/opensips1.6.4-2/sbin/opensips[26589]: 
CRITICAL:dialog:child_init:  xXx - active dialogs=5 , early=0
Mar  1 07:21:05 opensips /usr/local/opensips1.6.4-2/sbin/opensips[26587]: 
CRITICAL:dialog:child_init:  xXx - active dialogs=5 , early=0
Mar  1 07:23:28 opensips /usr/local/opensips1.6.4-2/sbin/opensips[28228]: 
CRITICAL:dialog:child_init:  xXx - active dialogs=1 , early=0
Mar  1 07:23:28 opensips /usr/local/opensips1.6.4-2/sbin/opensips[28226]: 
CRITICAL:dialog:child_init:  xXx - active dialogs=1 , early=0
Mar  1 07:25:21 opensips /usr/local/opensips1.6.4-2/sbin/opensips[29507]: 
CRITICAL:dialog:child_init:  xXx - active dialogs=0 , early=0
Mar  1 07:25:21 opensips /usr/local/opensips1.6.4-2/sbin/opensips[29509]: 
CRITICAL:dialog:child_init:  xXx - active dialogs=0 , early=0
Mar  1 07:48:20 opensips /usr/local/opensips1.6.4-2/sbin/opensips[7610]: 
CRITICAL:dialog:child_init:  xXx - active dialogs=4 , early=0
Mar  1 07:48:20 opensips /usr/local/opensips1.6.4-2/sbin/opensips[7608]: 
CRITICAL:dialog:child_init:  xXx - active dialogs=4 , early=0
Mar  1 07:55:36 opensips /usr/local/opensips1.6.4-2/sbin/opensips[11382]: 
CRITICAL:dialog:child_init:  xXx - active dialogs=2 , early=0
Mar  1 07:55:36 opensips /usr/local/opensips1.6.4-2/sbin/opensips[11380]: 
CRITICAL:dialog:child_init:  xXx - active dialogs=2 , early=0

-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
Sent: Tuesday, March 01, 2011 11:47 AM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] dialog statistic problem

Denis,

this xXx marker appears only ones, right ?

Regards,
Bogdan

Denis Putyato wrote:
 Hello Bogdan

 I applied patch and it seems that ??? shows real number of dialogs while 
 active statistic shows incorrect number

 -Original Message-
 From: users-boun...@lists.opensips.org 
 [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
 Sent: Monday, February 28, 2011 5:55 PM
 To: OpenSIPS users mailling list
 Subject: Re: [OpenSIPS-Users] dialog statistic problem

 Hi Denis,

 Please apply this small patch that will provide some extra on startup (about 
 the dialogs found in DB) - look for a line with marked with xXx 
 and see if the printed values are correct.

 Regards,
 Bogdan

 Denis Putyato wrote:
   
 Hello Bogdan

 - Server:: OpenSIPS (1.6.4-2-notls (x86_64/linux))
 - fork = yes (default)
 -
 listen=xxx.xxx.xxx.xxx:5060
 listen=xxx.xxx.xxx.xxx:5068
 - i do not use tcp
 -
 loadmodule db_mysql.so
 loadmodule sl.so
 loadmodule tm.so
 loadmodule signaling.so
 loadmodule auth.so
 loadmodule rr.so
 loadmodule maxfwd.so
 loadmodule textops.so
 loadmodule mi_fifo.so
 loadmodule uri.so
 loadmodule domain.so
 loadmodule drouting.so
 loadmodule siptrace.so
 loadmodule avpops.so
 loadmodule dialplan.so
 loadmodule dialog.so
 loadmodule permissions.so
 loadmodule usrloc.so
 loadmodule registrar.so
 loadmodule alias_db.so
 loadmodule auth_db.so
 loadmodule nathelper.so
 loadmodule acc.so
 loadmodule uac.so
 loadmodule aaa_radius.so

 -Original Message-
 From: users-boun...@lists.opensips.org 
 [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei 
 Iancu
 Sent: Monday, February 28, 2011 1:17 PM
 To: OpenSIPS users mailling list
 Subject: Re: [OpenSIPS-Users] dialog statistic problem

 Hi Denis,

 What revision number are you using ?

 Also, do you have fork enabled?
 Do you use multiple UDP interfaces?
 Do you have TCP enabled ?
 What are the modules you are using ?

 Regards,
 Bogdan

 Denis Putyato wrote:
   
 
 Hello!

 I found active dialog statistic problem

 For example in dialog::active_dialogs I have 130 active calls, after 
 restart Opensips I already have dialog::active_dialogs nearly 260, 
 i.e. the number of active dialogs increase by 2.

 Meanwhile, after restart, opensipsctl fifo dlg_list | grep dialog:: | 
 wc –l shows me that really number of calls greatly less.

 The opensipsctl fifo reset_statistics didn`t help. Only when I stop 
 all traffic via Opensips (but not Opensips) and then restart one, 
 dialog::active_dialogs become zero and begin increase when new calls 
 come to Opensips

 Thank you for any help

 -
 ---

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


 --
 Bogdan-Andrei Iancu
 OpenSIPS eBootcamp - 28th February 2011
 OpenSIPS solutions and know-how



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


-- 
Bogdan-Andrei Iancu
OpenSIPS eBootcamp - 28th February 2011
OpenSIPS solutions and know-how

Re: [OpenSIPS-Users] dialog statistic problem

[Denis Putyato]

Bogdan, since Mar  1 07:55:36 I didn't restart Opensips so

 Mar  1 07:55:36 opensips /usr/local/opensips1.6.4-2/sbin/opensips[11382]: 
 CRITICAL:dialog:child_init:  xXx - active dialogs=2 , early=0
 Mar  1 07:55:36 opensips /usr/local/opensips1.6.4-2/sbin/opensips[11380]: 
 CRITICAL:dialog:child_init:  xXx - active dialogs=2 , early=0

/usr/local/opensips1.6.4-2/sbin/opensipsctl fifo ps
Process::  ID=0 PID=11378 Type=attendant
Process::  ID=1 PID=11380 Type=RTPP timeout receiver
Process::  ID=2 PID=11381 Type=MI FIFO
Process::  ID=3 PID=11382 Type=SIP receiver udp:213.170.100.150:5060 
Process::  ID=4 PID=11383 Type=SIP receiver udp:213.170.100.150:5068 
Process::  ID=5 PID=11384 Type=time_keeper
Process::  ID=6 PID=11385 Type=timer
Process::  ID=7 PID=11386 Type=timer
Process::  ID=8 PID=11387 Type=TCP receiver
Process::  ID=9 PID=11388 Type=TCP main

-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
Sent: Tuesday, March 01, 2011 12:27 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] dialog statistic problem

OK, that seams to be the problem - it should be triggered only once.

After a restart, please send me:
1) the two xXx logs (with the pid)
2) output of opensipsctl fifo ps

I want to identify what is the second process triggering the xXx

Thanks and regards,
Bogdan

Denis Putyato wrote:
 No, during restart xXx appears twice

 $cat /var/log/opensips | grep xXx
 Mar  1 07:21:05 opensips /usr/local/opensips1.6.4-2/sbin/opensips[26589]: 
 CRITICAL:dialog:child_init:  xXx - active dialogs=5 , early=0
 Mar  1 07:21:05 opensips /usr/local/opensips1.6.4-2/sbin/opensips[26587]: 
 CRITICAL:dialog:child_init:  xXx - active dialogs=5 , early=0
 Mar  1 07:23:28 opensips /usr/local/opensips1.6.4-2/sbin/opensips[28228]: 
 CRITICAL:dialog:child_init:  xXx - active dialogs=1 , early=0
 Mar  1 07:23:28 opensips /usr/local/opensips1.6.4-2/sbin/opensips[28226]: 
 CRITICAL:dialog:child_init:  xXx - active dialogs=1 , early=0
 Mar  1 07:25:21 opensips /usr/local/opensips1.6.4-2/sbin/opensips[29507]: 
 CRITICAL:dialog:child_init:  xXx - active dialogs=0 , early=0
 Mar  1 07:25:21 opensips /usr/local/opensips1.6.4-2/sbin/opensips[29509]: 
 CRITICAL:dialog:child_init:  xXx - active dialogs=0 , early=0
 Mar  1 07:48:20 opensips /usr/local/opensips1.6.4-2/sbin/opensips[7610]: 
 CRITICAL:dialog:child_init:  xXx - active dialogs=4 , early=0
 Mar  1 07:48:20 opensips /usr/local/opensips1.6.4-2/sbin/opensips[7608]: 
 CRITICAL:dialog:child_init:  xXx - active dialogs=4 , early=0
 Mar  1 07:55:36 opensips /usr/local/opensips1.6.4-2/sbin/opensips[11382]: 
 CRITICAL:dialog:child_init:  xXx - active dialogs=2 , early=0
 Mar  1 07:55:36 opensips /usr/local/opensips1.6.4-2/sbin/opensips[11380]: 
 CRITICAL:dialog:child_init:  xXx - active dialogs=2 , early=0

 -Original Message-
 From: users-boun...@lists.opensips.org 
 [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
 Sent: Tuesday, March 01, 2011 11:47 AM
 To: OpenSIPS users mailling list
 Subject: Re: [OpenSIPS-Users] dialog statistic problem

 Denis,

 this xXx marker appears only ones, right ?

 Regards,
 Bogdan

 Denis Putyato wrote:
   
 Hello Bogdan

 I applied patch and it seems that ??? shows real number of dialogs while 
 active statistic shows incorrect number

 -Original Message-
 From: users-boun...@lists.opensips.org 
 [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
 Sent: Monday, February 28, 2011 5:55 PM
 To: OpenSIPS users mailling list
 Subject: Re: [OpenSIPS-Users] dialog statistic problem

 Hi Denis,

 Please apply this small patch that will provide some extra on startup (about 
 the dialogs found in DB) - look for a line with marked with xXx 
 and see if the printed values are correct.

 Regards,
 Bogdan

 Denis Putyato wrote:
   
 
 Hello Bogdan

 - Server:: OpenSIPS (1.6.4-2-notls (x86_64/linux))
 - fork = yes (default)
 -
 listen=xxx.xxx.xxx.xxx:5060
 listen=xxx.xxx.xxx.xxx:5068
 - i do not use tcp
 -
 loadmodule db_mysql.so
 loadmodule sl.so
 loadmodule tm.so
 loadmodule signaling.so
 loadmodule auth.so
 loadmodule rr.so
 loadmodule maxfwd.so
 loadmodule textops.so
 loadmodule mi_fifo.so
 loadmodule uri.so
 loadmodule domain.so
 loadmodule drouting.so
 loadmodule siptrace.so
 loadmodule avpops.so
 loadmodule dialplan.so
 loadmodule dialog.so
 loadmodule permissions.so
 loadmodule usrloc.so
 loadmodule registrar.so
 loadmodule alias_db.so
 loadmodule auth_db.so
 loadmodule nathelper.so
 loadmodule acc.so
 loadmodule uac.so
 loadmodule aaa_radius.so

 -Original Message-
 From: users-boun...@lists.opensips.org 
 [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei 
 Iancu
 Sent: Monday, February 28, 2011 1:17 PM
 To: OpenSIPS users mailling list
 Subject: Re: [OpenSIPS-Users] dialog statistic problem

 Hi Denis,

 What revision number are you using ?

 Also, do

Re: [OpenSIPS-Users] dialog statistic problem

[Denis Putyato]

Bogdan, sorry, but as I understand everything I need is download 
timeout_process.c and recompile nathelper module? 


-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
Sent: Tuesday, March 01, 2011 1:16 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] dialog statistic problem

Hi Denis,

Problem found and fixed - something related to the nathelper module and 
the RTP timeout notification - see revisions 7763 and 7764.

Thanks a lot for the help in troubleshooting this.

Regards,
Bogdan

Denis Putyato wrote:
 Bogdan, since Mar  1 07:55:36 I didn't restart Opensips so

   
 Mar  1 07:55:36 opensips /usr/local/opensips1.6.4-2/sbin/opensips[11382]: 
 CRITICAL:dialog:child_init:  xXx - active dialogs=2 , early=0
 Mar  1 07:55:36 opensips /usr/local/opensips1.6.4-2/sbin/opensips[11380]: 
 CRITICAL:dialog:child_init:  xXx - active dialogs=2 , early=0
 

 /usr/local/opensips1.6.4-2/sbin/opensipsctl fifo ps
 Process::  ID=0 PID=11378 Type=attendant
 Process::  ID=1 PID=11380 Type=RTPP timeout receiver
 Process::  ID=2 PID=11381 Type=MI FIFO
 Process::  ID=3 PID=11382 Type=SIP receiver udp:213.170.100.150:5060 
 Process::  ID=4 PID=11383 Type=SIP receiver udp:213.170.100.150:5068 
 Process::  ID=5 PID=11384 Type=time_keeper
 Process::  ID=6 PID=11385 Type=timer
 Process::  ID=7 PID=11386 Type=timer
 Process::  ID=8 PID=11387 Type=TCP receiver
 Process::  ID=9 PID=11388 Type=TCP main

 -Original Message-
 From: users-boun...@lists.opensips.org 
 [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
 Sent: Tuesday, March 01, 2011 12:27 PM
 To: OpenSIPS users mailling list
 Subject: Re: [OpenSIPS-Users] dialog statistic problem

 OK, that seams to be the problem - it should be triggered only once.

 After a restart, please send me:
 1) the two xXx logs (with the pid)
 2) output of opensipsctl fifo ps

 I want to identify what is the second process triggering the xXx

 Thanks and regards,
 Bogdan

 Denis Putyato wrote:
   
 No, during restart xXx appears twice

 $cat /var/log/opensips | grep xXx
 Mar  1 07:21:05 opensips /usr/local/opensips1.6.4-2/sbin/opensips[26589]: 
 CRITICAL:dialog:child_init:  xXx - active dialogs=5 , early=0
 Mar  1 07:21:05 opensips /usr/local/opensips1.6.4-2/sbin/opensips[26587]: 
 CRITICAL:dialog:child_init:  xXx - active dialogs=5 , early=0
 Mar  1 07:23:28 opensips /usr/local/opensips1.6.4-2/sbin/opensips[28228]: 
 CRITICAL:dialog:child_init:  xXx - active dialogs=1 , early=0
 Mar  1 07:23:28 opensips /usr/local/opensips1.6.4-2/sbin/opensips[28226]: 
 CRITICAL:dialog:child_init:  xXx - active dialogs=1 , early=0
 Mar  1 07:25:21 opensips /usr/local/opensips1.6.4-2/sbin/opensips[29507]: 
 CRITICAL:dialog:child_init:  xXx - active dialogs=0 , early=0
 Mar  1 07:25:21 opensips /usr/local/opensips1.6.4-2/sbin/opensips[29509]: 
 CRITICAL:dialog:child_init:  xXx - active dialogs=0 , early=0
 Mar  1 07:48:20 opensips /usr/local/opensips1.6.4-2/sbin/opensips[7610]: 
 CRITICAL:dialog:child_init:  xXx - active dialogs=4 , early=0
 Mar  1 07:48:20 opensips /usr/local/opensips1.6.4-2/sbin/opensips[7608]: 
 CRITICAL:dialog:child_init:  xXx - active dialogs=4 , early=0
 Mar  1 07:55:36 opensips /usr/local/opensips1.6.4-2/sbin/opensips[11382]: 
 CRITICAL:dialog:child_init:  xXx - active dialogs=2 , early=0
 Mar  1 07:55:36 opensips /usr/local/opensips1.6.4-2/sbin/opensips[11380]: 
 CRITICAL:dialog:child_init:  xXx - active dialogs=2 , early=0

 -Original Message-
 From: users-boun...@lists.opensips.org 
 [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
 Sent: Tuesday, March 01, 2011 11:47 AM
 To: OpenSIPS users mailling list
 Subject: Re: [OpenSIPS-Users] dialog statistic problem

 Denis,

 this xXx marker appears only ones, right ?

 Regards,
 Bogdan

 Denis Putyato wrote:
   
 
 Hello Bogdan

 I applied patch and it seems that ??? shows real number of dialogs while 
 active statistic shows incorrect number

 -Original Message-
 From: users-boun...@lists.opensips.org 
 [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
 Sent: Monday, February 28, 2011 5:55 PM
 To: OpenSIPS users mailling list
 Subject: Re: [OpenSIPS-Users] dialog statistic problem

 Hi Denis,

 Please apply this small patch that will provide some extra on startup 
 (about the dialogs found in DB) - look for a line with marked with xXx 
 and see if the printed values are correct.

 Regards,
 Bogdan

 Denis Putyato wrote:
   
 
   
 Hello Bogdan

 - Server:: OpenSIPS (1.6.4-2-notls (x86_64/linux))
 - fork = yes (default)
 -
 listen=xxx.xxx.xxx.xxx:5060
 listen=xxx.xxx.xxx.xxx:5068
 - i do not use tcp
 -
 loadmodule db_mysql.so
 loadmodule sl.so
 loadmodule tm.so
 loadmodule signaling.so
 loadmodule auth.so
 loadmodule rr.so
 loadmodule maxfwd.so
 loadmodule textops.so
 loadmodule mi_fifo.so
 loadmodule uri.so
 loadmodule

Re: [OpenSIPS-Users] dialog statistic problem

[Denis Putyato]

Hello Bogdan

- Server:: OpenSIPS (1.6.4-2-notls (x86_64/linux))
- fork = yes (default)
- 
listen=xxx.xxx.xxx.xxx:5060
listen=xxx.xxx.xxx.xxx:5068
- i do not use tcp
- 
loadmodule db_mysql.so
loadmodule sl.so
loadmodule tm.so
loadmodule signaling.so
loadmodule auth.so
loadmodule rr.so
loadmodule maxfwd.so
loadmodule textops.so
loadmodule mi_fifo.so
loadmodule uri.so
loadmodule domain.so
loadmodule drouting.so
loadmodule siptrace.so
loadmodule avpops.so
loadmodule dialplan.so
loadmodule dialog.so
loadmodule permissions.so
loadmodule usrloc.so
loadmodule registrar.so
loadmodule alias_db.so
loadmodule auth_db.so
loadmodule nathelper.so
loadmodule acc.so
loadmodule uac.so
loadmodule aaa_radius.so

-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
Sent: Monday, February 28, 2011 1:17 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] dialog statistic problem

Hi Denis,

What revision number are you using ?

Also, do you have fork enabled?
Do you use multiple UDP interfaces?
Do you have TCP enabled ?
What are the modules you are using ?

Regards,
Bogdan

Denis Putyato wrote:

 Hello!

 I found active dialog statistic problem

 For example in dialog::active_dialogs I have 130 active calls, after 
 restart Opensips I already have dialog::active_dialogs nearly 260, 
 i.e. the number of active dialogs increase by 2.

 Meanwhile, after restart, opensipsctl fifo dlg_list | grep dialog:: | 
 wc –l shows me that really number of calls greatly less.

 The opensipsctl fifo reset_statistics didn`t help. Only when I stop 
 all traffic via Opensips (but not Opensips) and then restart one, 
 dialog::active_dialogs become zero and begin increase when new calls 
 come to Opensips

 Thank you for any help

 

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OpenSIPS eBootcamp - 28th February 2011
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[OpenSIPS-Users] dialog statistic problem

[Denis Putyato]

 

Hello!

 

I found active dialog statistic problem

 

For example in dialog::active_dialogs I have 130 active calls, after restart 
Opensips I already have dialog::active_dialogs nearly 260, i.e. the number of 
active dialogs increase by 2.

Meanwhile, after restart,  opensipsctl fifo dlg_list | grep dialog:: | wc –l 
shows me that really number of calls greatly less.

 

The opensipsctl fifo reset_statistics didn`t help. Only when I stop all traffic 
via Opensips (but not Opensips) and then restart one, dialog::active_dialogs 
become zero and begin increase when new calls come to Opensips

 

Thank you for any help

 

 

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[OpenSIPS-Users] SQL query

[Denis Putyato]

Hello!

Opensips 1.6.4-2, MySQL installed on the same server as opensips.

 

 

Please can somebody explain why such message can appear in syslog? This happens 
when I make “opensipsctl fifo dp_reload” after long period of time nothing to 
do with opensips.

During processing calls opensips make some SQL queries (there is no problem 
with it).

 

“Feb  3 16:17:33 opensips /usr/local/opensips1.6.4-2/sbin/opensips[1355]: 
INFO:db_mysql:switch_state_to_disconnected: disconnect event for 0x8078e0

Feb  3 16:17:33 opensips /usr/local/opensips1.6.4-2/sbin/opensips[1355]: 
INFO:db_mysql:reset_all_statements: reseting all statements on connection: 
(0x808bf0) 0x8078e0

Feb  3 16:17:33 opensips /usr/local/opensips1.6.4-2/sbin/opensips[1355]: 
INFO:db_mysql:connect_with_retry: re-connected successful for 0x8078e0”

 

Thank you for any help

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Re: [OpenSIPS-Users] BYE request for proper signalling

[Denis Putyato]

Hello Bogdan

 because of some NAT presence, right ?

No, I need use IP address when there is more than one SIP proxy in call path. 

-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
Sent: Wednesday, February 02, 2011 3:36 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] BYE request for proper signalling

Hi Denis,

 From SIP point of view, the BYE must be sent to the contact URIs . I 
guess your contact is different than the layer3 IP because of some NAT 
presence, right ? if so, use fix_nated_contact() for INVITE and 200 OK, 
so that the received contact will be fixed with the layer3 IP, so the 
dialog module will use the contact with a useful info.

Regards,
Bogdan

Denis Putyato wrote:

 Hello!

  

 I am using dialog module for control of call duration.

 When timeout of dialog expires I need Opensips send BYE not to caller 
 and callee contact (which is stored during creation of dialog) but to 
 IP address and port from which INVITE (caller) and 200 OK (callee) had 
 been received.

  

 Thank you for any help

  

  

 

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-- 
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OpenSIPS Event - expo, conf, social, bootcamp
2 - 4 February 2011, ITExpo, Miami,  USA
OpenSIPS solutions and know-how


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[OpenSIPS-Users] BYE request for proper signalling

[Denis Putyato]

Hello!

 

I am using dialog module for control of call duration.

When timeout of dialog expires I need Opensips send BYE not to caller and 
callee contact (which is stored during creation of dialog) but to IP address 
and port from which INVITE (caller) and 200 OK (callee) had been received.

 

Thank you for any help

 

 

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Re: [OpenSIPS-Users] Problem with the configuration of permissions module

[Denis Putyato]

Hello

Try to modify your code 
if(!check_source_address(0)){

sl_send_reply(403, Forbidden);
 exit;

-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Alejandro Recarey
Sent: Monday, January 17, 2011 5:10 AM
To: OpenSIPS users mailling list
Subject: [OpenSIPS-Users] Problem with the configuration of permissions module

Hi all,

I have been checking the SIP security of my configuration and am shocked
to find out that my configuration is currently not working correctly.

I am using OpenSIPS 1.6.2 and the check_source_address function to only
allow calls from my own domain but it seems that no matter what I write
to the address table, I can always call! That means that any IP address
has access to my OpenSIPS server for outbound calls.

Now, I am certain that it must be a problem with my configuration of the
opensips.cfg file, so any help would be appreciated to find out the
problem.

I am using the permissions module with the following configuration:

mysql select * from address;
++-+---+--+--+---+--+--+
| id | grp | ip| mask | port | proto | pattern  | context_i|
++-+---+--+--+---+--+--+
|  1 |   0 | 130.117.93.0  |   25 | 5060 | any   | ^sip:.*$ |  |
++-+---+--+--+---+--+--+

My route table is as follows:

route{

if (!mf_process_maxfwd_header(10)) {
sl_send_reply(483,Too Many Hops);
exit;
}

if (has_totag()) {
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route()) {
if (is_method(BYE)) {
setflag(1); # do accounting ...
setflag(3); # ... even if the transaction fails
} else if (is_method(INVITE)) {
# even if in most of the cases is useless, do RR for
# re-INVITEs alos, as some buggy clients do change route set
# during the dialog.
record_route();
}
# route it out to whatever destination was set by loose_route()
# in $du (destination URI).
route(1);
} else {
/* uncomment the following lines if you want to enable presence */
##if (is_method(SUBSCRIBE)  $rd == your.server.ip.address) {
### in-dialog subscribe requests
##route(2);
##exit;
##}
if ( is_method(ACK) ) {
if ( t_check_trans() ) {
# non loose-route, but stateful ACK; must be an ACK after
# a 487 or e.g. 404 from upstream server
t_relay();
exit;
} else {
# ACK without matching transaction -
# ignore and discard
exit;
}
}
sl_send_reply(404,Not here);
}
exit;
}

#initial requests

# CANCEL processing
if (is_method(CANCEL))
{
if (t_check_trans())
t_relay();
exit;
}

t_check_trans();


if (!(method==REGISTER)  from_uri==myself) /*no multidomain version*/
{
# Here is where I check that the INVITE comes from my servers
if(!check_source_address(0)){

sl_send_reply(403, Forbidden);

if (!proxy_authorize(, subscriber)) {
proxy_challenge(, 0);
exit;
}
if (!db_check_from()) {
sl_send_reply(403,Forbidden auth ID);
exit;
}

consume_credentials();
# caller authenticated
}
}

# preloaded route checking
if (loose_route()) {
xlog(L_ERR,
Attempt to route with preloaded Route's [$fu/$tu/$ru/$ci]);
if (!is_method(ACK))
sl_send_reply(403,Preload Route denied);
exit;
}

# record routing
if (!is_method(REGISTER|MESSAGE))
record_route();

# account only INVITEs
if (is_method(INVITE)) {
setflag(1); # do accounting
}

if (!uri==myself)
{
append_hf(P-hint: outbound\r\n);

# Do not act as an open relay
#   only allow requests from handled domains
if(from_uri==myself){
route(1);
}
else {
sl_send_reply(403, Not here);
}
}

# requests for my domain

if (is_method(PUBLISH))
{
sl_send_reply(503, Service Unavailable);
exit;
}


if (is_method(REGISTER))
{
# authenticate the REGISTER requests (uncomment to enable auth)
if (!www_authorize(, subscriber))
{
www_challenge(, 0);
exit;
}

if (!db_check_to())

Re: [OpenSIPS-Users] Pacth rtpproxy

[Denis Putyato]

Razvan, I got rtpproxy from  http://opensips.org/pub/rtpproxy/ 
http://opensips.org/pub/rtpproxy/ as you wrote.

I started it using such cli command “/usr/local/rtpproxy1/bin/rtpproxy -u 
opensips -l 1.1.1.1 -s /var/run/rtpproxy.sock -T 80 -i -n /var/run/timer.sock  
-d INFO” and made test call.

Callee has been ringing during about 2 minutes and nothing happens at all. What 
I did wrong? 

 

P.S. I use such function in my script for rtp proxy “rtpproxy_offer(con);”

 

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Razvan Crainea
Sent: Wednesday, January 12, 2011 3:14 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Pacth rtpproxy

 

Hello Denis,

In the official release of RTPProxy, the timeout parameter (-T) controls both 
session establishment and rtp timeout. This is a problem since we would like to 
have a long period for call establishment, but a fast media timeout detection.
In the patched version of RTPProxy, the -W parameter allows you to specify a 
longer period for call establishment. If not set, it has the default value of 
60 seconds. 
If you decide not to use patched version of RTPProxy, the timeout notification 
will work, but you will have the same timeout in both situations.

Regards,
Razvan


On 01/12/2011 07:38 AM, Denis Putyato wrote: 

Hello Razvan,

 

“OpenSIPS shouldn't even try to terminate the call because it isn't established 
yet”

As I understand I just do not need to use –W key when starting rtpproxy, it 
does not work at all?

 

 

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Razvan Crainea
Sent: Tuesday, January 11, 2011 6:49 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Pacth rtpproxy

 

Hello Denis,

You are right, OpenSIPS shouldn't even try to terminate the call because it 
isn't established yet. I just added a small fix to solve this problem. Please 
update your code from svn to use this fix.
The RTPProxy patch was done against commit 
600c80493793bafd2d69427bc22fcb43faad98c5. You can either get the RTPProxy 
from git, change it's branch and then apply the patch, or you can download an 
already patched version from http://opensips.org/pub/rtpproxy/.

Regards,
Razvan 

On 1/11/2011 2:19 PM, Denis Putyato wrote: 

Hello!

 

I try patch rtpproxy gotten from git. And there is such error during patching

 

patch  rtpproxy_timeout_notification.patch

patching file main.c

Hunk #1 succeeded at 70 (offset 2 lines).

Hunk #2 FAILED at 120.

Hunk #3 succeeded at 132 with fuzz 1 (offset 4 lines).

Hunk #4 succeeded at 211 with fuzz 2 (offset 4 lines).

Hunk #5 succeeded at 276 (offset 4 lines).

Hunk #6 succeeded at 742 with fuzz 2 (offset -26 lines).

Hunk #7 succeeded at 758 with fuzz 2 (offset -26 lines).

1 out of 7 hunks FAILED -- saving rejects to file main.c.rej

patching file rtpp_command.c

Hunk #1 FAILED at 795.

Hunk #2 FAILED at 888.

2 out of 2 hunks FAILED -- saving rejects to file rtpp_command.c.rej

patching file rtpp_defines.h

Hunk #1 FAILED at 95.

1 out of 1 hunk FAILED -- saving rejects to file rtpp_defines.h.rej

patching file rtpp_notify.c

 

rtpproxy_timeout_notification.patch is a patch for timeout notification which  
divide rtp timeout and session initiation timeout notification as said in

http://www.opensips.org/html/docs/modules/devel/nathelper.html#id249142

 

This patch I got from SVN version of latest Opensips. 

 
 
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Re: [OpenSIPS-Users] Pacth rtpproxy

[Denis Putyato]

Hello, Razvan

 

“This is a problem since we would like to have a long period for call 
establishment” and what does it mean “call establishment” in such context?

 

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Razvan Crainea
Sent: Wednesday, January 12, 2011 5:18 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Pacth rtpproxy

 

Hello Denis,

RTPProxy is only used to detect the media timeout. If OpenSIPS receives a 
timeout notification on an unestablished call, it simply ignores it.
If you want to terminate the call when the callee doesn't answer you can use 
the tm module and set the fr_inv_timer parameter. You can get more details 
from:
http://www.opensips.org/html/docs/modules/devel/tm.html#id250344

Regards,
Razvan

On 01/12/2011 02:38 PM, Denis Putyato wrote: 

Razvan, I got rtpproxy from  http://opensips.org/pub/rtpproxy/ 
http://opensips.org/pub/rtpproxy/ as you wrote.

I started it using such cli command “/usr/local/rtpproxy1/bin/rtpproxy -u 
opensips -l 1.1.1.1 -s /var/run/rtpproxy.sock -T 80 -i -n /var/run/timer.sock  
-d INFO” and made test call.

Callee has been ringing during about 2 minutes and nothing happens at all. What 
I did wrong? 

 

P.S. I use such function in my script for rtp proxy “rtpproxy_offer(con);”

 

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Razvan Crainea
Sent: Wednesday, January 12, 2011 3:14 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Pacth rtpproxy

 

Hello Denis,

In the official release of RTPProxy, the timeout parameter (-T) controls both 
session establishment and rtp timeout. This is a problem since we would like to 
have a long period for call establishment, but a fast media timeout detection.
In the patched version of RTPProxy, the -W parameter allows you to specify a 
longer period for call establishment. If not set, it has the default value of 
60 seconds. 
If you decide not to use patched version of RTPProxy, the timeout notification 
will work, but you will have the same timeout in both situations.

Regards,
Razvan


On 01/12/2011 07:38 AM, Denis Putyato wrote: 

Hello Razvan,

 

“OpenSIPS shouldn't even try to terminate the call because it isn't established 
yet”

As I understand I just do not need to use –W key when starting rtpproxy, it 
does not work at all?

 

 

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Razvan Crainea
Sent: Tuesday, January 11, 2011 6:49 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Pacth rtpproxy

 

Hello Denis,

You are right, OpenSIPS shouldn't even try to terminate the call because it 
isn't established yet. I just added a small fix to solve this problem. Please 
update your code from svn to use this fix.
The RTPProxy patch was done against commit 
600c80493793bafd2d69427bc22fcb43faad98c5. You can either get the RTPProxy 
from git, change it's branch and then apply the patch, or you can download an 
already patched version from http://opensips.org/pub/rtpproxy/.

Regards,
Razvan 

On 1/11/2011 2:19 PM, Denis Putyato wrote: 

Hello!

 

I try patch rtpproxy gotten from git. And there is such error during patching

 

patch  rtpproxy_timeout_notification.patch

patching file main.c

Hunk #1 succeeded at 70 (offset 2 lines).

Hunk #2 FAILED at 120.

Hunk #3 succeeded at 132 with fuzz 1 (offset 4 lines).

Hunk #4 succeeded at 211 with fuzz 2 (offset 4 lines).

Hunk #5 succeeded at 276 (offset 4 lines).

Hunk #6 succeeded at 742 with fuzz 2 (offset -26 lines).

Hunk #7 succeeded at 758 with fuzz 2 (offset -26 lines).

1 out of 7 hunks FAILED -- saving rejects to file main.c.rej

patching file rtpp_command.c

Hunk #1 FAILED at 795.

Hunk #2 FAILED at 888.

2 out of 2 hunks FAILED -- saving rejects to file rtpp_command.c.rej

patching file rtpp_defines.h

Hunk #1 FAILED at 95.

1 out of 1 hunk FAILED -- saving rejects to file rtpp_defines.h.rej

patching file rtpp_notify.c

 

rtpproxy_timeout_notification.patch is a patch for timeout notification which  
divide rtp timeout and session initiation timeout notification as said in

http://www.opensips.org/html/docs/modules/devel/nathelper.html#id249142

 

This patch I got from SVN version of latest Opensips. 

 
 
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Re: [OpenSIPS-Users] RTPProxy timeout notifications

[Denis Putyato]

Hello Razvan!

Now it's working, thank you.

But I want to tell you that -W key in rtpproxy and Opensips doesn't work. When 
-W timer expires rtpproxy notifies about it Opensips but last one cannot drop 
call because there is no information about callee and caller contacts in 
dialog. 

-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Razvan Crainea
Sent: Monday, January 03, 2011 7:25 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] RTPProxy timeout notifications

Hello Denis,

The problem you are dealing with is that you are using a TCP socket to 
receive timeout notifications from RTPProxy.
When a timeout notification is received through TCP, the nathelper 
module searches for the sender in the rtpproxies specified in 
rtpproxy_sock module parameter. It cannot find it in that list, so it 
ignores the notification.
Because you are using a UNIX socket to communicate with RTPProxy, then 
you should also use a UNIX socket to receive timeout notifications from it.
Note that I posted today a bug fix for this behavior. You can watch the 
thread at:
http://lists.rtpproxy.org/pipermail/devel/2011-January/thread.html

Regards,
Razvan

On 12/28/2010 04:52 PM, Denis Putyato wrote:
 Hello Bogdan

 RTP Proxy is working but timeout notification does not.
 There is error /usr/local/opensips1.6.4/sbin/opensips[26496]:
 DBG:nathelper:timeout_listener_process: unknown rtpproxy – ignoring 
 -Original Message-
 From: users-boun...@lists.opensips.org 
 [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
 Sent: Tuesday, December 28, 2010 5:49 PM
 To: OpenSIPS users mailling list; Razvan Crainea
 Subject: Re: [OpenSIPS-Users] RTPProxy timeout notifications

 Hi Denis,


 Denis Putyato wrote:
 Hello Bogdan!

 1) There is no patch in official release of Opensips 1.6.4 which I can 
 download from web site (source tar). There is a patch only in SVN version of 
 Opensips 1.6.4

 Hmm..that's a packaging bug :(I will take care of this.
 2) The patch which I can use from SVN version I can apply only to rtpproxy 
 from git. If I use rtpproxy from web site I cannot apply patch to it (there 
 are some errors during process of patch).

 I will ask Razvan (the author of this work) to see if the patch can be
 ported to official rtpproxy release too (not as coding, but as
 functionality).
 In my case I use rtpproxy from git with applied patch from SVN version of 
 Opensips (when I start rtpproxy I use such command 
 /usr/local/rtpproxy/bin/rtpproxy -l 1.1.1.1 -s unix:/var/run/rtpproxy.sock 
 -F -i -n tcp:1.1.1.1:2 -T 20 -W 60. As I understand without patch -W 
 doesn`t work) and official release of Openspis 1.6.4 which I downloaded from 
 web site (not from SVN)

 And this works ?

 Regards,
 Bogdan

 -Original Message-
 From: users-boun...@lists.opensips.org 
 [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
 Sent: Tuesday, December 28, 2010 1:31 PM
 To: OpenSIPS users mailling list
 Subject: Re: [OpenSIPS-Users] RTPProxy timeout notifications

 Hi Denis,

 Silly question, but have you applied to the official RTPproxy the
 patches that comes with the nathelper module ?

 Regards,
 Bogdan

 Denis Putyato wrote:

 Hello!

 During tests of new feature in rtpproxy I received such problem:

 “Dec 27 11:42:42 opensips
 /usr/local/opensips1.6.4/sbin/opensips[26496]:
 DBG:nathelper:timeout_listener_process: unknown rtpproxy – ignoring”

 And log for rtpproxy

 “Dec 28 07:43:37 opensips rtpproxy[28223]: INFO:handle_command: new
 session 6fe1-00bf-8e08-8065-0002a405c...@172.31.255.250, tag
 6f008e65a4;1 requested, type strong

 Dec 28 07:43:37 opensips rtpproxy[28223]: INFO:handle_command: new
 session on a port 64922 created, tag 6f008e65a4;1

 Dec 28 07:43:37 opensips rtpproxy[28223]: INFO:handle_command: setting
 timeout handler

 Dec 28 07:43:37 opensips rtpproxy[28223]: INFO:handle_command:
 pre-filling caller's address with 3.3.3.3:23066

 Dec 28 07:43:37 opensips
 /usr/local/opensips1.6.4/sbin/opensips[28196]:
 ERROR:nathelper:force_rtp_proxy: Unable to parse body

 Dec 28 07:43:38 opensips rtpproxy[28223]: INFO:handle_command: lookup
 on ports 64922/4, session timer restarted

 Dec 28 07:43:38 opensips rtpproxy[28223]: INFO:handle_command:
 pre-filling callee's address with 2.2.2.2:18408

 Dec 28 07:43:39 opensips rtpproxy[28223]: INFO:handle_command: lookup
 on ports 64922/4, session timer restarted

 Dec 28 07:44:07 opensips rtpproxy[28223]: INFO:process_rtp: session
 timeout

 Dec 28 07:44:07 opensips rtpproxy[28223]: INFO:remove_session: RTP
 stats: 1449 in from callee, 421 in from caller, 1870 relayed, 0 dropped

 Dec 28 07:44:07 opensips rtpproxy[28223]: INFO:remove_session: RTCP
 stats: 7 in from callee, 2 in from caller, 9 relayed, 0 dropped

 Dec 28 07:44:07 opensips rtpproxy[28223]: INFO:remove_session: session
 on ports 64922/4 is cleaned up

 Dec 28 07:44

[OpenSIPS-Users] Pacth rtpproxy

[Denis Putyato]

Hello!

 

I try patch rtpproxy gotten from git. And there is such error during patching

 

patch  rtpproxy_timeout_notification.patch

patching file main.c

Hunk #1 succeeded at 70 (offset 2 lines).

Hunk #2 FAILED at 120.

Hunk #3 succeeded at 132 with fuzz 1 (offset 4 lines).

Hunk #4 succeeded at 211 with fuzz 2 (offset 4 lines).

Hunk #5 succeeded at 276 (offset 4 lines).

Hunk #6 succeeded at 742 with fuzz 2 (offset -26 lines).

Hunk #7 succeeded at 758 with fuzz 2 (offset -26 lines).

1 out of 7 hunks FAILED -- saving rejects to file main.c.rej

patching file rtpp_command.c

Hunk #1 FAILED at 795.

Hunk #2 FAILED at 888.

2 out of 2 hunks FAILED -- saving rejects to file rtpp_command.c.rej

patching file rtpp_defines.h

Hunk #1 FAILED at 95.

1 out of 1 hunk FAILED -- saving rejects to file rtpp_defines.h.rej

patching file rtpp_notify.c

 

rtpproxy_timeout_notification.patch is a patch for timeout notification which  
divide rtp timeout and session initiation timeout notification as said in

http://www.opensips.org/html/docs/modules/devel/nathelper.html#id249142

 

This patch I got from SVN version of latest Opensips. 

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Re: [OpenSIPS-Users] Pacth rtpproxy

[Denis Putyato]

Hello Razvan,

 

“OpenSIPS shouldn't even try to terminate the call because it isn't established 
yet”

As I understand I just do not need to use –W key when starting rtpproxy, it 
does not work at all?

 

 

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Razvan Crainea
Sent: Tuesday, January 11, 2011 6:49 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Pacth rtpproxy

 

Hello Denis,

You are right, OpenSIPS shouldn't even try to terminate the call because it 
isn't established yet. I just added a small fix to solve this problem. Please 
update your code from svn to use this fix.
The RTPProxy patch was done against commit 
600c80493793bafd2d69427bc22fcb43faad98c5. You can either get the RTPProxy 
from git, change it's branch and then apply the patch, or you can download an 
already patched version from http://opensips.org/pub/rtpproxy/.

Regards,
Razvan 

On 1/11/2011 2:19 PM, Denis Putyato wrote: 

Hello!

 

I try patch rtpproxy gotten from git. And there is such error during patching

 

patch  rtpproxy_timeout_notification.patch

patching file main.c

Hunk #1 succeeded at 70 (offset 2 lines).

Hunk #2 FAILED at 120.

Hunk #3 succeeded at 132 with fuzz 1 (offset 4 lines).

Hunk #4 succeeded at 211 with fuzz 2 (offset 4 lines).

Hunk #5 succeeded at 276 (offset 4 lines).

Hunk #6 succeeded at 742 with fuzz 2 (offset -26 lines).

Hunk #7 succeeded at 758 with fuzz 2 (offset -26 lines).

1 out of 7 hunks FAILED -- saving rejects to file main.c.rej

patching file rtpp_command.c

Hunk #1 FAILED at 795.

Hunk #2 FAILED at 888.

2 out of 2 hunks FAILED -- saving rejects to file rtpp_command.c.rej

patching file rtpp_defines.h

Hunk #1 FAILED at 95.

1 out of 1 hunk FAILED -- saving rejects to file rtpp_defines.h.rej

patching file rtpp_notify.c

 

rtpproxy_timeout_notification.patch is a patch for timeout notification which  
divide rtp timeout and session initiation timeout notification as said in

http://www.opensips.org/html/docs/modules/devel/nathelper.html#id249142

 

This patch I got from SVN version of latest Opensips. 

 
 
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Re: [OpenSIPS-Users] RTPProxy timeout notifications

[Denis Putyato]

Hello Bogdan!

1) There is no patch in official release of Opensips 1.6.4 which I can download 
from web site (source tar). There is a patch only in SVN version of Opensips 
1.6.4
2) The patch which I can use from SVN version I can apply only to rtpproxy from 
git. If I use rtpproxy from web site I cannot apply patch to it (there are some 
errors during process of patch).

In my case I use rtpproxy from git with applied patch from SVN version of 
Opensips (when I start rtpproxy I use such command 
/usr/local/rtpproxy/bin/rtpproxy -l 1.1.1.1 -s unix:/var/run/rtpproxy.sock -F 
-i -n tcp:1.1.1.1:2 -T 20 -W 60. As I understand without patch -W doesn`t 
work) and official release of Openspis 1.6.4 which I downloaded from web site 
(not from SVN)   

-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
Sent: Tuesday, December 28, 2010 1:31 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] RTPProxy timeout notifications

Hi Denis,

Silly question, but have you applied to the official RTPproxy the 
patches that comes with the nathelper module ?

Regards,
Bogdan

Denis Putyato wrote:

 Hello!

 During tests of new feature in rtpproxy I received such problem:

 “Dec 27 11:42:42 opensips 
 /usr/local/opensips1.6.4/sbin/opensips[26496]: 
 DBG:nathelper:timeout_listener_process: unknown rtpproxy – ignoring”

 And log for rtpproxy

 “Dec 28 07:43:37 opensips rtpproxy[28223]: INFO:handle_command: new 
 session 6fe1-00bf-8e08-8065-0002a405c...@172.31.255.250, tag 
 6f008e65a4;1 requested, type strong

 Dec 28 07:43:37 opensips rtpproxy[28223]: INFO:handle_command: new 
 session on a port 64922 created, tag 6f008e65a4;1

 Dec 28 07:43:37 opensips rtpproxy[28223]: INFO:handle_command: setting 
 timeout handler

 Dec 28 07:43:37 opensips rtpproxy[28223]: INFO:handle_command: 
 pre-filling caller's address with 3.3.3.3:23066

 Dec 28 07:43:37 opensips 
 /usr/local/opensips1.6.4/sbin/opensips[28196]: 
 ERROR:nathelper:force_rtp_proxy: Unable to parse body

 Dec 28 07:43:38 opensips rtpproxy[28223]: INFO:handle_command: lookup 
 on ports 64922/4, session timer restarted

 Dec 28 07:43:38 opensips rtpproxy[28223]: INFO:handle_command: 
 pre-filling callee's address with 2.2.2.2:18408

 Dec 28 07:43:39 opensips rtpproxy[28223]: INFO:handle_command: lookup 
 on ports 64922/4, session timer restarted

 Dec 28 07:44:07 opensips rtpproxy[28223]: INFO:process_rtp: session 
 timeout

 Dec 28 07:44:07 opensips rtpproxy[28223]: INFO:remove_session: RTP 
 stats: 1449 in from callee, 421 in from caller, 1870 relayed, 0 dropped

 Dec 28 07:44:07 opensips rtpproxy[28223]: INFO:remove_session: RTCP 
 stats: 7 in from callee, 2 in from caller, 9 relayed, 0 dropped

 Dec 28 07:44:07 opensips rtpproxy[28223]: INFO:remove_session: session 
 on ports 64922/4 is cleaned up

 Dec 28 07:44:07 opensips rtpproxy[28223]: ERR:do_timeout_notification: 
 failed to send timeout notification: Broken pipe”

 Opensips 1.6.4

 Latest rtpproxy from git with patch for RTPProxy timeout notifications

 The start string of rtpproxy:

 /usr/local/rtpproxy/bin/rtpproxy -l 1.1.1.1 -s 
 unix:/var/run/rtpproxy.sock -F -i -n tcp:1.1.1.1:2 -T 20 -W 60

 Opensips.cfg:

 …

 …

 modparam(nathelper, rtpproxy_sock, /var/run/rtpproxy.sock)

 modparam(nathelper, rtpp_notify_socket, tcp:1.1.1.1:2)

 …

 …

 rtpproxy_offer(con);

 ….

 rtpproxy_answer(con);

 …

 During voice session everything fine (bidirectional voice flow). Then 
 I emulate LAN problem and after 20 s expire I received such message. 
 Call is still active.

 Thank you for any help.

 

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


-- 
Bogdan-Andrei Iancu
OpenSIPS Event - expo, conf, social, bootcamp
2 - 4 February 2011, ITExpo, Miami,  USA
www.voice-system.ro


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Re: [OpenSIPS-Users] RTPProxy timeout notifications

[Denis Putyato]

Hello Bogdan

RTP Proxy is working but timeout notification does not.
There is error /usr/local/opensips1.6.4/sbin/opensips[26496]: 
 DBG:nathelper:timeout_listener_process: unknown rtpproxy – ignoring 

-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
Sent: Tuesday, December 28, 2010 5:49 PM
To: OpenSIPS users mailling list; Razvan Crainea
Subject: Re: [OpenSIPS-Users] RTPProxy timeout notifications

Hi Denis,


Denis Putyato wrote:
 Hello Bogdan!

 1) There is no patch in official release of Opensips 1.6.4 which I can 
 download from web site (source tar). There is a patch only in SVN version of 
 Opensips 1.6.4
   
Hmm..that's a packaging bug :(I will take care of this.
 2) The patch which I can use from SVN version I can apply only to rtpproxy 
 from git. If I use rtpproxy from web site I cannot apply patch to it (there 
 are some errors during process of patch).
   
I will ask Razvan (the author of this work) to see if the patch can be 
ported to official rtpproxy release too (not as coding, but as 
functionality).
 In my case I use rtpproxy from git with applied patch from SVN version of 
 Opensips (when I start rtpproxy I use such command 
 /usr/local/rtpproxy/bin/rtpproxy -l 1.1.1.1 -s unix:/var/run/rtpproxy.sock 
 -F -i -n tcp:1.1.1.1:2 -T 20 -W 60. As I understand without patch -W 
 doesn`t work) and official release of Openspis 1.6.4 which I downloaded from 
 web site (not from SVN)   
   
And this works ?

Regards,
Bogdan

 -Original Message-
 From: users-boun...@lists.opensips.org 
 [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
 Sent: Tuesday, December 28, 2010 1:31 PM
 To: OpenSIPS users mailling list
 Subject: Re: [OpenSIPS-Users] RTPProxy timeout notifications

 Hi Denis,

 Silly question, but have you applied to the official RTPproxy the 
 patches that comes with the nathelper module ?

 Regards,
 Bogdan

 Denis Putyato wrote:
   
 Hello!

 During tests of new feature in rtpproxy I received such problem:

 “Dec 27 11:42:42 opensips 
 /usr/local/opensips1.6.4/sbin/opensips[26496]: 
 DBG:nathelper:timeout_listener_process: unknown rtpproxy – ignoring”

 And log for rtpproxy

 “Dec 28 07:43:37 opensips rtpproxy[28223]: INFO:handle_command: new 
 session 6fe1-00bf-8e08-8065-0002a405c...@172.31.255.250, tag 
 6f008e65a4;1 requested, type strong

 Dec 28 07:43:37 opensips rtpproxy[28223]: INFO:handle_command: new 
 session on a port 64922 created, tag 6f008e65a4;1

 Dec 28 07:43:37 opensips rtpproxy[28223]: INFO:handle_command: setting 
 timeout handler

 Dec 28 07:43:37 opensips rtpproxy[28223]: INFO:handle_command: 
 pre-filling caller's address with 3.3.3.3:23066

 Dec 28 07:43:37 opensips 
 /usr/local/opensips1.6.4/sbin/opensips[28196]: 
 ERROR:nathelper:force_rtp_proxy: Unable to parse body

 Dec 28 07:43:38 opensips rtpproxy[28223]: INFO:handle_command: lookup 
 on ports 64922/4, session timer restarted

 Dec 28 07:43:38 opensips rtpproxy[28223]: INFO:handle_command: 
 pre-filling callee's address with 2.2.2.2:18408

 Dec 28 07:43:39 opensips rtpproxy[28223]: INFO:handle_command: lookup 
 on ports 64922/4, session timer restarted

 Dec 28 07:44:07 opensips rtpproxy[28223]: INFO:process_rtp: session 
 timeout

 Dec 28 07:44:07 opensips rtpproxy[28223]: INFO:remove_session: RTP 
 stats: 1449 in from callee, 421 in from caller, 1870 relayed, 0 dropped

 Dec 28 07:44:07 opensips rtpproxy[28223]: INFO:remove_session: RTCP 
 stats: 7 in from callee, 2 in from caller, 9 relayed, 0 dropped

 Dec 28 07:44:07 opensips rtpproxy[28223]: INFO:remove_session: session 
 on ports 64922/4 is cleaned up

 Dec 28 07:44:07 opensips rtpproxy[28223]: ERR:do_timeout_notification: 
 failed to send timeout notification: Broken pipe”

 Opensips 1.6.4

 Latest rtpproxy from git with patch for RTPProxy timeout notifications

 The start string of rtpproxy:

 /usr/local/rtpproxy/bin/rtpproxy -l 1.1.1.1 -s 
 unix:/var/run/rtpproxy.sock -F -i -n tcp:1.1.1.1:2 -T 20 -W 60

 Opensips.cfg:

 …

 …

 modparam(nathelper, rtpproxy_sock, /var/run/rtpproxy.sock)

 modparam(nathelper, rtpp_notify_socket, tcp:1.1.1.1:2)

 …

 …

 rtpproxy_offer(con);

 ….

 rtpproxy_answer(con);

 …

 During voice session everything fine (bidirectional voice flow). Then 
 I emulate LAN problem and after 20 s expire I received such message. 
 Call is still active.

 Thank you for any help.

 

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


   


-- 
Bogdan-Andrei Iancu
OpenSIPS Event - expo, conf, social, bootcamp
2 - 4 February 2011, ITExpo, Miami,  USA
www.voice-system.ro


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[OpenSIPS-Users] RTPProxy timeout notifications

[Denis Putyato]

Hello!

 

During tests of new feature in rtpproxy I received such problem:

 

“Dec 27 11:42:42 opensips /usr/local/opensips1.6.4/sbin/opensips[26496]: 
DBG:nathelper:timeout_listener_process: unknown rtpproxy – ignoring”

 

And log for rtpproxy 

“Dec 28 07:43:37 opensips rtpproxy[28223]: INFO:handle_command: new session 
6fe1-00bf-8e08-8065-0002a405c...@172.31.255.250, tag 6f008e65a4;1 
requested, type strong

Dec 28 07:43:37 opensips rtpproxy[28223]: INFO:handle_command: new session on a 
port 64922 created, tag 6f008e65a4;1

Dec 28 07:43:37 opensips rtpproxy[28223]: INFO:handle_command: setting timeout 
handler

Dec 28 07:43:37 opensips rtpproxy[28223]: INFO:handle_command: pre-filling 
caller's address with 3.3.3.3:23066

Dec 28 07:43:37 opensips /usr/local/opensips1.6.4/sbin/opensips[28196]: 
ERROR:nathelper:force_rtp_proxy: Unable to parse body

Dec 28 07:43:38 opensips rtpproxy[28223]: INFO:handle_command: lookup on ports 
64922/4, session timer restarted

Dec 28 07:43:38 opensips rtpproxy[28223]: INFO:handle_command: pre-filling 
callee's address with 2.2.2.2:18408

Dec 28 07:43:39 opensips rtpproxy[28223]: INFO:handle_command: lookup on ports 
64922/4, session timer restarted

Dec 28 07:44:07 opensips rtpproxy[28223]: INFO:process_rtp: session timeout

Dec 28 07:44:07 opensips rtpproxy[28223]: INFO:remove_session: RTP stats: 1449 
in from callee, 421 in from caller, 1870 relayed, 0 dropped

Dec 28 07:44:07 opensips rtpproxy[28223]: INFO:remove_session: RTCP stats: 7 in 
from callee, 2 in from caller, 9 relayed, 0 dropped

Dec 28 07:44:07 opensips rtpproxy[28223]: INFO:remove_session: session on ports 
64922/4 is cleaned up

Dec 28 07:44:07 opensips rtpproxy[28223]: ERR:do_timeout_notification: failed 
to send timeout notification: Broken pipe”

 

Opensips  1.6.4

Latest rtpproxy from git with patch for RTPProxy timeout notifications 

 

The start string of rtpproxy:

/usr/local/rtpproxy/bin/rtpproxy -l 1.1.1.1 -s unix:/var/run/rtpproxy.sock -F 
-i -n tcp:1.1.1.1:2 -T 20 -W 60

 

Opensips.cfg:

…

…

modparam(nathelper, rtpproxy_sock, /var/run/rtpproxy.sock)

modparam(nathelper, rtpp_notify_socket, tcp:1.1.1.1:2)

…

…

rtpproxy_offer(con);

….

rtpproxy_answer(con);

…

 

During voice session everything fine (bidirectional voice flow). Then I emulate 
LAN problem and after 20 s expire I received such message. Call is still active.

 

Thank you for any help.

 

 

 

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[OpenSIPS-Users] ul_add MI function

[Denis Putyato]

Hello!

 

Can anybody explain me how to use such command?

 

Do I need fill all parameters or I may fill just AOR and contact, for example?

Just when I try it I receive “400 Too few or too many arguments” or “400 Bad 
parameter” and cannot understand what is wronge

 

Thank you 

 

 

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[OpenSIPS-Users] receive port during call process

[Denis Putyato]

 

Hello!

 

I have a such problem.

 

Opensips using 2 ports 

One – 5068 for client which must register  on Opensips

Second – 5060 for all other clients.

 

1)  Client А registering on Opensips (socket: udp:2.2.2.2:5068). Client A 
is behind NAT.

2)  Client А receives incoming call (via lookup() function).

3)  Call has such way

Cisco (source port 1 sends  INVITE to port 5060 of Opensips) – Opensips 
(receives INVITE from Cisco to port 5060 and sends the INVITE to client from 
source port 5068 to some client`s port) – Client A. 

Everything work fine until client А answers. Then Client A sends to Opensips 
(port 5068) 200 OK, Opensips retransmit it to Cisco (from port 5060 to 1), 
Cisco sends to Opensips ACK (from port 1 to 5060) AND Opensips retransmit 
this ACK to client`s port FROM PORT 5060, BUT NOT 5068. This ACK didn`t reach 
client A because of NAT.

 

Thank you for any help.

 

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Re: [OpenSIPS-Users] TEXTOPS module

[Denis Putyato]

Thank you Bogdan, that is working

-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
Sent: Monday, December 20, 2010 5:27 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] TEXTOPS module

Hi Denis,

the best way to do it is via branch_route 
(http://www.opensips.org/Resources/DocsCoreRoutes16#toc2) . Whatever 
changes you do there will be applied only for that particular branch and 
not for all branches. When you do changes in the request route, the 
changes will be applied to all future branches !

So, do something like this:

- in request route (first time) put the RPID, PAI and FROM (new vals ) 
in 2 different AVPS (according to the first selected GW)
- arm a branch route and failure route
- do t_relay() - this will trigger the branch route and you can do the 
changes to the messages ( as you do it now)
- if you end up in failure route - set new values for the 3 AVPs, 
reflecting the new destination
- do t_relay() - triggers branch route, etc..

Regards,
Bogdan


Denis Putyato wrote:
 Thank you Bogdan for your answer. Now I understood that apply changes is a 
 bad idea.
 But during process a call I have to make some changes to INVITE message. 
 For example,
 I need to add Remote-Party-ID (RPI) and/or P-Asserted-ID (PAI) and make 
 uac_replace_from(). If I make it for the first time everything fine. 
 But if I need then change these fields (via subst or uac_replace_from() 
 again)(for example, some gateways fails and cannot accepts call, I use 
 use_next_gw() of d_routing module and MUST change callerid information) then 
 my tests show that during, for example, second time call of 
 uac_replace_from() there are two uri in From: header field (as you understand 
 that is wrong), or if I make subst() of RPI or PAI then second header RPI and 
 PAI appear in addition of first headers which I added (or subst) before. 


 And to avoid this I make signaling loop. New INVITE process as a new message 
 with modified early headers, so I can change it again.   

 -Original Message-
 From: users-boun...@lists.opensips.org 
 [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
 Sent: Monday, December 20, 2010 4:07 PM
 To: OpenSIPS users mailling list
 Subject: Re: [OpenSIPS-Users] TEXTOPS module

 Hello Denis,

 So far there is no good arguments for such a function, but there are a 
 lot of performance penalties while using such a function.

 Basically, to apply the change to a message, opensips/kamilio has to 1) 
 take the received buffer and the changes and to generate a new buffer 
 with the whole message (including the changes) and to 2) take the newly 
 generate buffer and to parse it as a SIP content in order to be able to 
 use internally it.
 Bottom line, each time you use such a function you double the processing 
 effort for parsing and generating SIP messages. And if you check the 
 code profiling we did (see 
 http://www.opensips.org/Resources/TestsProfiling), these operations are 
 ~50% from the total CPU usage (cumulate the PARSE and BUILD times).

 Now, in most of the cases (99% of the case) you do not really need to 
 apply changes in realtime - there are a lot of simple tricks to avoid 
 it. If you describe the problem you have, I can help you in putting some 
 extra logic in the script to avoid the need to apply changes.

 Using a smart approach is more efficient than a brute force approach - 
 the idea is that you are aware of the changes you do in script and you 
 remember (in script) these changes, so you can take them into account in 
 your later processing even if they are not actually applied on the SIPS 
 message.

 Regards,
 Bogdan

 Denis Putyato wrote:
   
 Hello!

 In kamailio project there is a function |msg_apply_changes() ||in 
 textops module for applying changes (for example add or subst some 
 header field) in SIP messages. Is there some way on opensips for doing 
 such operation? Now I need make signaling “loop” for change header 
 fields which I, for example, add during call process.|

 | |

 |Opensips 1.6.3|

 | |

 |Thank you || |

 

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2 - 4 February 2011, ITExpo, Miami,  USA
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Re: [OpenSIPS-Users] [OpenSIPS-Devel] [RELEASE] OpenSIPS 1.6.4 major release is out

[Denis Putyato]

Hello

In scripts/mysql_update_1_6_4.sh there is such string
 run_query - Adding new 'attrs' field in DR_GATEWAYS table ALTER TABLE 
ast_dr_gateways ADD COLUMN attrs CHAR(255) DEFAULT NULL

But there is already attrs fields in DR_GATEWAYS. May be in DR_RULES?

-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
Sent: Tuesday, December 21, 2010 1:10 AM
To: users@lists.opensips.org
Subject: [OpenSIPS-Users] [OpenSIPS-Devel] [RELEASE] OpenSIPS 1.6.4 major 
release is out

Hello all,

OpenSIPS 1.6.4 is a major release and it is the third release following 
the new release policy .

1.6.4 release brings both new features / enhancement and a lot of fixes. 
The most important additions :
- CDR support in ACC module
- Media timeout and call termination with nathelper and dialog module
- new dialog Presence Call Info
- B2BUA API

The listing with all additions and fixes is available under 
http://www.opensips.org/Main/Ver164.

Migration documentation (from 1.6.3 to 1.6.4) can be found under 
http://www.opensips.org/Resources/DocsMigration163to164

OpenSIPS 1.6.4 is now ready for download on project web site and SF 
download system.

The full ChangeLog is available under 
http://opensips.org/pub/opensips/1.6.4/src/ChangeLog

To get the OpenSIPS 1.6.4 version, see the Downloads page - 
http://www.opensips.org/Resources/Downloads


Enjoy,
Bogdan

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Re: [OpenSIPS-Users] receive port during call process

[Denis Putyato]

Hello, Bogdan!

modparam(rr, enable_double_rr, 1) helps me. Before this parameter had value 
0.

Thank you. 


-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
Sent: Tuesday, December 21, 2010 5:32 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] receive port during call process

Hi Denis,

The ACK is routed based on the RR headers that were collected by INVITE 
and returned in 200 OK. For such a scenario to work, opensips is doing 
double routing (adds for itself 2 RR headers, one with the inbound 
interface, one with the outbound interface).

Could you post the SIP capture of such a call to check if correct from 
SIP point of view?

Regards,
Bogdan

Denis Putyato wrote:

 Hello!

 I have a such problem.

 Opensips using 2 ports

 One – 5068 for client which must register on Opensips

 Second – 5060 for all other clients.

 1) Client А registering on Opensips (socket: udp:2.2.2.2:5068). Client 
 A is behind NAT.

 2) Client А receives incoming call (via lookup() function).

 3) Call has such way

 Cisco (source port 1 sends INVITE to port 5060 of Opensips) – 
 Opensips (receives INVITE from Cisco to port 5060 and sends the INVITE 
 to client from source port 5068 to some client`s port) – Client A.

 Everything work fine until client А answers. Then Client A sends to 
 Opensips (port 5068) 200 OK, Opensips retransmit it to Cisco (from 
 port 5060 to 1), Cisco sends to Opensips ACK (from port 1 to 
 5060) AND Opensips retransmit this ACK to client`s port FROM PORT 
 5060, BUT NOT 5068. This ACK didn`t reach client A because of NAT.

 Thank you for any help.

 

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2 - 4 February 2011, ITExpo, Miami,  USA
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[OpenSIPS-Users] TEXTOPS module

[Denis Putyato]

Hello!

 

In kamailio project there is a function msg_apply_changes() in textops module 
for applying changes (for example add or subst some header field) in SIP 
messages. Is there some way on opensips for doing such operation? Now I need 
make signaling “loop” for change header fields which I, for example, add during 
call process.

 

Opensips 1.6.3

 

Thank you   

 

 

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Re: [OpenSIPS-Users] TEXTOPS module

[Denis Putyato]

Thank you Bogdan for your answer. Now I understood that apply changes is a bad 
idea.
But during process a call I have to make some changes to INVITE message. 
For example,
I need to add Remote-Party-ID (RPI) and/or P-Asserted-ID (PAI) and make 
uac_replace_from(). If I make it for the first time everything fine. 
But if I need then change these fields (via subst or uac_replace_from() 
again)(for example, some gateways fails and cannot accepts call, I use 
use_next_gw() of d_routing module and MUST change callerid information) then my 
tests show that during, for example, second time call of uac_replace_from() 
there are two uri in From: header field (as you understand that is wrong), or 
if I make subst() of RPI or PAI then second header RPI and PAI appear in 
addition of first headers which I added (or subst) before. 

And to avoid this I make signaling loop. New INVITE process as a new message 
with modified early headers, so I can change it again.   

-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
Sent: Monday, December 20, 2010 4:07 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] TEXTOPS module

Hello Denis,

So far there is no good arguments for such a function, but there are a 
lot of performance penalties while using such a function.

Basically, to apply the change to a message, opensips/kamilio has to 1) 
take the received buffer and the changes and to generate a new buffer 
with the whole message (including the changes) and to 2) take the newly 
generate buffer and to parse it as a SIP content in order to be able to 
use internally it.
Bottom line, each time you use such a function you double the processing 
effort for parsing and generating SIP messages. And if you check the 
code profiling we did (see 
http://www.opensips.org/Resources/TestsProfiling), these operations are 
~50% from the total CPU usage (cumulate the PARSE and BUILD times).

Now, in most of the cases (99% of the case) you do not really need to 
apply changes in realtime - there are a lot of simple tricks to avoid 
it. If you describe the problem you have, I can help you in putting some 
extra logic in the script to avoid the need to apply changes.

Using a smart approach is more efficient than a brute force approach - 
the idea is that you are aware of the changes you do in script and you 
remember (in script) these changes, so you can take them into account in 
your later processing even if they are not actually applied on the SIPS 
message.

Regards,
Bogdan

Denis Putyato wrote:

 Hello!

 In kamailio project there is a function |msg_apply_changes() ||in 
 textops module for applying changes (for example add or subst some 
 header field) in SIP messages. Is there some way on opensips for doing 
 such operation? Now I need make signaling “loop” for change header 
 fields which I, for example, add during call process.|

 | |

 |Opensips 1.6.3|

 | |

 |Thank you || |

 

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


-- 
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OpenSIPS Event - expo, conf, social, bootcamp
2 - 4 February 2011, ITExpo, Miami,  USA
www.voice-system.ro


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Re: [OpenSIPS-Users] TEXTOPS module

[Denis Putyato]

Hello Stan 
I will try to do what you say.


-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Stanislaw Pitucha
Sent: Monday, December 20, 2010 5:10 PM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] TEXTOPS module

On 20/12/10 13:51, Denis Putyato wrote:
 Thank you Bogdan for your answer. Now I understood that apply changes is a 
 bad idea.
 But during process a call I have to make some changes to INVITE message. 
 For example,
 I need to add Remote-Party-ID (RPI) and/or P-Asserted-ID (PAI) and make 
 uac_replace_from(). If I make it for the first time everything fine. 
 But if I need then change these fields (via subst or uac_replace_from() 
 again)(for example, some gateways fails and cannot accepts call, I use 
 use_next_gw() of d_routing module and MUST change callerid information) then 
 my tests show that during, for example, second time call of 
 uac_replace_from() there are two uri in From: header field (as you understand 
 that is wrong), or if I make subst() of RPI or PAI then second header RPI and 
 PAI appear in addition of first headers which I added (or subst) before. 

 
 And to avoid this I make signaling loop. New INVITE process as a new message 
 with modified early headers, so I can change it again.   

I'm not 100% sure if that will work in your case (never used uac), but
you may be able to do all this by starting transaction early. If you
force t_newtran() early in the script (you probably should anyways), any
modification done to the message after that point should not be present
in the failure route.

So do t_newtran(), get the first gateway, modify as needed - if you
fail, just reapply the changes as if you were working on the original
message.

Unless someone shouts now that it doesn't interact properly with uac
that is ;)

Regards,
Stan

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Re: [OpenSIPS-Users] TEXTOPS module

[Denis Putyato]

Thank you Bogdan I will try and let you know

-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
Sent: Monday, December 20, 2010 5:27 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] TEXTOPS module

Hi Denis,

the best way to do it is via branch_route 
(http://www.opensips.org/Resources/DocsCoreRoutes16#toc2) . Whatever 
changes you do there will be applied only for that particular branch and 
not for all branches. When you do changes in the request route, the 
changes will be applied to all future branches !

So, do something like this:

- in request route (first time) put the RPID, PAI and FROM (new vals ) 
in 2 different AVPS (according to the first selected GW)
- arm a branch route and failure route
- do t_relay() - this will trigger the branch route and you can do the 
changes to the messages ( as you do it now)
- if you end up in failure route - set new values for the 3 AVPs, 
reflecting the new destination
- do t_relay() - triggers branch route, etc..

Regards,
Bogdan


Denis Putyato wrote:
 Thank you Bogdan for your answer. Now I understood that apply changes is a 
 bad idea.
 But during process a call I have to make some changes to INVITE message. 
 For example,
 I need to add Remote-Party-ID (RPI) and/or P-Asserted-ID (PAI) and make 
 uac_replace_from(). If I make it for the first time everything fine. 
 But if I need then change these fields (via subst or uac_replace_from() 
 again)(for example, some gateways fails and cannot accepts call, I use 
 use_next_gw() of d_routing module and MUST change callerid information) then 
 my tests show that during, for example, second time call of 
 uac_replace_from() there are two uri in From: header field (as you understand 
 that is wrong), or if I make subst() of RPI or PAI then second header RPI and 
 PAI appear in addition of first headers which I added (or subst) before. 


 And to avoid this I make signaling loop. New INVITE process as a new message 
 with modified early headers, so I can change it again.   

 -Original Message-
 From: users-boun...@lists.opensips.org 
 [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
 Sent: Monday, December 20, 2010 4:07 PM
 To: OpenSIPS users mailling list
 Subject: Re: [OpenSIPS-Users] TEXTOPS module

 Hello Denis,

 So far there is no good arguments for such a function, but there are a 
 lot of performance penalties while using such a function.

 Basically, to apply the change to a message, opensips/kamilio has to 1) 
 take the received buffer and the changes and to generate a new buffer 
 with the whole message (including the changes) and to 2) take the newly 
 generate buffer and to parse it as a SIP content in order to be able to 
 use internally it.
 Bottom line, each time you use such a function you double the processing 
 effort for parsing and generating SIP messages. And if you check the 
 code profiling we did (see 
 http://www.opensips.org/Resources/TestsProfiling), these operations are 
 ~50% from the total CPU usage (cumulate the PARSE and BUILD times).

 Now, in most of the cases (99% of the case) you do not really need to 
 apply changes in realtime - there are a lot of simple tricks to avoid 
 it. If you describe the problem you have, I can help you in putting some 
 extra logic in the script to avoid the need to apply changes.

 Using a smart approach is more efficient than a brute force approach - 
 the idea is that you are aware of the changes you do in script and you 
 remember (in script) these changes, so you can take them into account in 
 your later processing even if they are not actually applied on the SIPS 
 message.

 Regards,
 Bogdan

 Denis Putyato wrote:
   
 Hello!

 In kamailio project there is a function |msg_apply_changes() ||in 
 textops module for applying changes (for example add or subst some 
 header field) in SIP messages. Is there some way on opensips for doing 
 such operation? Now I need make signaling “loop” for change header 
 fields which I, for example, add during call process.|

 | |

 |Opensips 1.6.3|

 | |

 |Thank you || |

 

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


   


-- 
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OpenSIPS Event - expo, conf, social, bootcamp
2 - 4 February 2011, ITExpo, Miami,  USA
www.voice-system.ro


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[OpenSIPS-Users] ACC module

[Denis Putyato]

Hello everybody!

 

There was  a news in web site of Opensips about ACC module and CDR generation

http://lists.opensips.org/pipermail/news/2010-August/96.html

 

I use 1.6.3. Can I already install ACC for the Opensips with new feature? Or it 
will be accessed only in next release of Opensips?

 

Thank you.  

 

 

 

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Re: [OpenSIPS-Users] ACC module

[Denis Putyato]

Bogdan, i understand, thank you

-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
Sent: Monday, December 06, 2010 1:49 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] ACC module

Hi Denis.

The feature to directly generate CDRs in available in OpenSIPS trunk 
(devel) and  it will be available in opensips 1.6.4 stable (in mid December)

Regards,
Bogdan

Denis Putyato wrote:

 Hello everybody!

  

 There was  a news in web site of Opensips about ACC module and CDR 
 generation

 http://lists.opensips.org/pipermail/news/2010-August/96.html

  

 I use 1.6.3. Can I already install ACC for the Opensips with new 
 feature? Or it will be accessed only in next release of Opensips?

  

 Thank you.  

  

  

  

 

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Re: [OpenSIPS-Users] 1.6.4 ACC module

[Denis Putyato]

Hello Anton

And a reason of this? And what do you mean while ACC logging? 

-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Anton Zagorskiy
Sent: Monday, December 06, 2010 5:27 PM
To: 'OpenSIPS users mailling list'
Subject: [OpenSIPS-Users] 1.6.4 ACC module

Hello.

I'm using a dialog module and store values in dialogs. Is it possible to
fetch values from dialog while ACC logging?






WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.zagors...@oyster-telecom.ru
www.oyster-telecom.ru





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Re: [OpenSIPS-Users] $DLG_status values

[Denis Putyato]

Yes, you are right there is no information about status 2 in doc. But as I 
understand status 2 exists during time after create dialog and until final 
reply.
Then status can be 3,4 (if final reply received) or 5(if there is no final 
reply). 

-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Anton Zagorskiy
Sent: Monday, December 06, 2010 7:04 PM
To: 'OpenSIPS users mailling list'
Subject: [OpenSIPS-Users] $DLG_status values

When INVITE is sent and OK isn't received $DLG_status equals 2. But in the
documentation $DLG_status can be NULL, 3, 4, 5.
When does $DLG_status is 2? Can it has other values?






WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.zagors...@oyster-telecom.ru
www.oyster-telecom.ru





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[OpenSIPS-Users] DROUTING module

[Denis Putyato]

Hello!

 


Please tell me where in dr_rules table rule_attrs_avp (str) of DROUTING module 
store?


http://www.opensips.org/html/docs/modules/1.6.x/drouting.html#id294102


Thank you.

 

 

 

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Re: [OpenSIPS-Users] Timer Based Failover Question

[Denis Putyato]

And what about

http://www.opensips.org/html/docs/modules/1.6.x/tm.html#id250384

 

 

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Bruce Borrett
Sent: Tuesday, November 16, 2010 1:40 PM
To: Users@lists.opensips.org
Subject: [OpenSIPS-Users] Timer Based Failover Question

 

Hi All

I am having a problem where a SIP provider are sometimes sending us a 100, but 
then nothing afterwards. I would like to fail these calls over using a timer, 
but fr_timer wont work since we are receiving a 100, and fr_inv_timer requires 
a very lengthy duration which also will not work as I would like for the call 
to failover within 5 seconds maximum.

Does anyone have any other suggestion for me please?

Regards,
Bruce Borrett

 

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Re: [OpenSIPS-Users] acconting messages

[Denis Putyato]

I understand, thank you

-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Ovidiu Sas
Sent: Wednesday, November 17, 2010 8:23 AM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] acconting messages

It seems that you will need to stick to manual accounting.

Regards,
Ovidiu Sas

On Tue, Nov 16, 2010 at 1:02 AM, Denis Putyato denis7...@mail.ru wrote:
 Thank you for reply

 First variant is not quite flexible for me.
 The second variant more interesting, but it doesn't work

 A piece of code from opensips.cfg:

 modparam(tm, fr_timer, 10)
 modparam(tm, wt_timer, 30)
 modparam(tm, fr_inv_timer_avp, $avp(i:25))
 modparam(tm, T1_timer, 1000)
 ...
 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)
 ...

 if ($avp(i:200)==1) {
 t_newtran();
 setflag(16);
 setflag(17);
 t_flush_flags();
 t_reply(403, Forbidden_gw);
 exit;
 }

 And after this there is no records in ACC table.
 May be I do something wrong?


 -Original Message-
 From: users-boun...@lists.opensips.org 
 [mailto:users-boun...@lists.opensips.org] On Behalf Of Ovidiu Sas
 Sent: Tuesday, November 16, 2010 8:40 AM
 To: OpenSIPS users mailling list
 Subject: Re: [OpenSIPS-Users] acconting messages

 You can do manual accounting:
 http://www.opensips.org/html/docs/modules/1.6.x/acc.html#id294003

 Or, you can create a new transaction, flag it for acc and then
 terminate it t_reply:
 http://www.opensips.org/html/docs/modules/1.6.x/tm.html#id293687


 Regards,
 Ovidiu Sas

 On Tue, Nov 16, 2010 at 12:30 AM, Denis Putyato denis7...@mail.ru wrote:
 Hello!

 Is there any chancy for accounting calls which were finished by sending
 failure code using send_reply() func.?

 Thank you

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Re: [OpenSIPS-Users] acconting messages

[Denis Putyato]

Thank you for reply

First variant is not quite flexible for me.
The second variant more interesting, but it doesn't work

A piece of code from opensips.cfg:

modparam(tm, fr_timer, 10)
modparam(tm, wt_timer, 30)
modparam(tm, fr_inv_timer_avp, $avp(i:25))
modparam(tm, T1_timer, 1000)
...
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)
...

 if ($avp(i:200)==1) {
 t_newtran();
 setflag(16);
 setflag(17);
 t_flush_flags();
 t_reply(403, Forbidden_gw);
 exit;
 }

And after this there is no records in ACC table.
May be I do something wrong?


-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Ovidiu Sas
Sent: Tuesday, November 16, 2010 8:40 AM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] acconting messages

You can do manual accounting:
http://www.opensips.org/html/docs/modules/1.6.x/acc.html#id294003

Or, you can create a new transaction, flag it for acc and then
terminate it t_reply:
http://www.opensips.org/html/docs/modules/1.6.x/tm.html#id293687


Regards,
Ovidiu Sas

On Tue, Nov 16, 2010 at 12:30 AM, Denis Putyato denis7...@mail.ru wrote:
 Hello!

 Is there any chancy for accounting calls which were finished by sending
 failure code using send_reply() func.?

 Thank you

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[OpenSIPS-Users] mediaproxy and ip_conntrack_udp_timeout_stream

[Denis Putyato]

 

Hello !

 

Please can somebody tell me how “ip_conntrack_udp_timeout_stream” works with 
mediaproxy?

 

For example. 

SIP client A à Opensips with mediaproxy à SIP client B

 

Call established. RTP send/receive for both clients. Then there is some problem 
with ethernet for client A and he can`t send RTP. 

Should mediaproxy  stop relays RTP for both clients and indicates to Opensips 
that calls must be stopped or not (after “ip_conntrack_udp_timeout_stream” 
expires)?

 

Thank you   

 

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Re: [OpenSIPS-Users] mediaproxy and ip_conntrack_udp_timeout_stream

[Denis Putyato]

I understand 
Thank you

-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Saul Ibarra Corretge
Sent: Thursday, October 21, 2010 4:12 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] mediaproxy and ip_conntrack_udp_timeout_stream

On 10/21/2010 12:30 PM, Denis Putyato wrote:
 Hello !

 Please can somebody tell me how “ip_conntrack_udp_timeout_stream” works
 with mediaproxy?

 For example.

 SIP client A à Opensips with mediaproxy à SIP client B

 Call established. RTP send/receive for both clients. Then there is some
 problem with ethernet for client A and he can`t send RTP.

 Should mediaproxy  stop relays RTP for both clients and indicates to
 Opensips that calls must be stopped or not (after
 “ip_conntrack_udp_timeout_stream” expires)?


No, it shouldn't. A might not be sending RTP because it has Voice 
Activity Detection (VAD) enabled, so the call would always be torn down 
but it wouldn't be correct.

You must address that issue in the signaling plane, with session timers 
for example.


Regards,

-- 
Saúl Ibarra Corretgé
AG Projects

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Re: [OpenSIPS-Users] FW: Error when setting OpenSips with Radius

[Denis Putyato]

Try just User-Name attribute in aaa_radius

 

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Tung Tran
Sent: Tuesday, July 27, 2010 8:58 PM
To: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] FW: Error when setting OpenSips with Radius

 

Hello

 

Thanks for your reponse

I still have a question for you that do I need to set the whole set like this:

 

User-Name = 1...@x.x.x.x

Calling-Station-Id = 100@ x.x.x.x 

Called-Station-Id = sip: x.x.x.x

Digest-User-Name = 100

Digest-Realm =  x.x.x.x 

Digest-Nonce = 4c4f0b702b314d48a3fcc1aa986ee74362ddff63

Digest-Uri = sip: x.x.x.x 

Digest-Method = REGISTER

Digest-Response = ab287b28ee499cc733c27a0c198e066c

Service-Type = Sip-Session

Sip-Uri-User = 100

cisco-avpair = call-id=554f9e2e09304b13

NAS-Port = 5060

NAS-IP-Address x.x.x.x

 

Or just an User-Name attribute in aaa_radius module?

 

Above Radius message is what I am doing with Openser 1.2, but I don’t know how 
to do the same with Opensips 1.6.2 version. 

There is quite changes between 02 versions.

 

Thanks

T.T

 

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Denis Putyato
Sent: Tuesday, July 27, 2010 8:11 PM
To: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] FW: Error when setting OpenSips with Radius

 

Try to add “Sets” param. with User-Name attribute for acc_radius module in your 
opensips.cfg 

For example, 

modparam(aaa_radius,sets,set1 = (User-Name=$var(usr))”)

 

where $var(usr) is some PV of your callerid 

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Tung Tran
Sent: Tuesday, July 27, 2010 4:55 PM
To: 'OpenSIPS users mailling list'
Subject: [OpenSIPS-Users] FW: Error when setting OpenSips with Radius

 

Hello, 

 

Just wanna recall if someone can help me out.

 

Thanks

T.T

 

From: Tung Tran [mailto:tr.t...@gmail.com] 
Sent: Tuesday, July 27, 2010 9:33 AM
To: 'OpenSIPS users mailling list'
Subject: Error when setting OpenSips with Radius

 

Hi all

 

I am building the Opensips 1.6.2 to run with external Billing via Radius (using 
radiusclient-ng), but I get this error when trying to register. No radius 
message is sent to Radius server yet

 

Jul 27 08:12:03 SVR60 /usr/local/sbin/opensips[19907]: rc_avpair_new: unknown 
attribute 0

Jul 27 08:12:03 SVR60 /usr/local/sbin/opensips[19907]: 
ERROR:aaa_radius:rad_avp_add: failure 

Jul 27 08:12:03 SVR60 /usr/local/sbin/opensips[19907]: 
ERROR:auth_aaa:aaa_authorize_sterman: unable to add User-Name attribute

 

Here is my opensips.cfg

 

loadmodule auth.so

loadmodule auth_db.so

loadmodule aaa_radius.so

loadmodule auth_aaa.so

 

modparam(auth_aaa, aaa_url, 
radius:/usr/local/etc/radiusclient-ng/radiusclient.conf)

 

..

  if(!aaa_www_authorize())

   {

   xlog(L_INFO, *-*-* False in Radius_www_authorize , 
challenging M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci .. \n);

   www_challenge(, 1);

   exit;

   }

…

 

I am very appreciated if someone can point me a hint where is the problem

Thanks in advance

 

 

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Re: [OpenSIPS-Users] FW: Error when setting OpenSips with Radius

[Denis Putyato]

Try add these attributs in aaa_radius

 

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Tung Tran
Sent: Wednesday, July 28, 2010 10:52 AM
To: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] FW: Error when setting OpenSips with Radius

 

Just another question, do you know how can I insert the “Calling-station-id” 
and “Called-Station-id” in authorize message for INVITE?

Like this ( what I am doing with openser right now)

 


User-Name = 6528822...@x.x.x.x


Calling-Station-Id = 6528822724@ x.x.x.x 


Called-Station-Id = sip:0018323822177@ x.x.x.x 


Digest-User-Name = 6528822724


Digest-Realm =  x.x.x.x 


Digest-Nonce = 4a6763dbd5352b1bf9b8f0873f7bcf781068e516


Digest-Uri = sip:0018323822177@ x.x.x.x


Digest-Method = INVITE


Digest-Response = 0bb87b4cc20f3892c4d743a35cd1fb01


Service-Type = Sip-Session


Sip-Uri-User = 6528822724


cisco-avpair = call-id=b3b259f5-f6fe-1810-86e0-001a803f2...@192.168.1.2


NAS-Port = 5060


NAS-IP-Address = 10.84.0.21 

 

Thanks again.

 

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Denis Putyato
Sent: Wednesday, July 28, 2010 1:06 PM
To: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] FW: Error when setting OpenSips with Radius

 

Try just User-Name attribute in aaa_radius

 

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Tung Tran
Sent: Tuesday, July 27, 2010 8:58 PM
To: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] FW: Error when setting OpenSips with Radius

 

Hello

 

Thanks for your reponse

I still have a question for you that do I need to set the whole set like this:

 

User-Name = 1...@x.x.x.x

Calling-Station-Id = 100@ x.x.x.x 

Called-Station-Id = sip: x.x.x.x

Digest-User-Name = 100

Digest-Realm =  x.x.x.x 

Digest-Nonce = 4c4f0b702b314d48a3fcc1aa986ee74362ddff63

Digest-Uri = sip: x.x.x.x 

Digest-Method = REGISTER

Digest-Response = ab287b28ee499cc733c27a0c198e066c

Service-Type = Sip-Session

Sip-Uri-User = 100

cisco-avpair = call-id=554f9e2e09304b13

NAS-Port = 5060

NAS-IP-Address x.x.x.x

 

Or just an User-Name attribute in aaa_radius module?

 

Above Radius message is what I am doing with Openser 1.2, but I don’t know how 
to do the same with Opensips 1.6.2 version. 

There is quite changes between 02 versions.

 

Thanks

T.T

 

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Denis Putyato
Sent: Tuesday, July 27, 2010 8:11 PM
To: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] FW: Error when setting OpenSips with Radius

 

Try to add “Sets” param. with User-Name attribute for acc_radius module in your 
opensips.cfg 

For example, 

modparam(aaa_radius,sets,set1 = (User-Name=$var(usr))”)

 

where $var(usr) is some PV of your callerid 

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Tung Tran
Sent: Tuesday, July 27, 2010 4:55 PM
To: 'OpenSIPS users mailling list'
Subject: [OpenSIPS-Users] FW: Error when setting OpenSips with Radius

 

Hello, 

 

Just wanna recall if someone can help me out.

 

Thanks

T.T

 

From: Tung Tran [mailto:tr.t...@gmail.com] 
Sent: Tuesday, July 27, 2010 9:33 AM
To: 'OpenSIPS users mailling list'
Subject: Error when setting OpenSips with Radius

 

Hi all

 

I am building the Opensips 1.6.2 to run with external Billing via Radius (using 
radiusclient-ng), but I get this error when trying to register. No radius 
message is sent to Radius server yet

 

Jul 27 08:12:03 SVR60 /usr/local/sbin/opensips[19907]: rc_avpair_new: unknown 
attribute 0

Jul 27 08:12:03 SVR60 /usr/local/sbin/opensips[19907]: 
ERROR:aaa_radius:rad_avp_add: failure 

Jul 27 08:12:03 SVR60 /usr/local/sbin/opensips[19907]: 
ERROR:auth_aaa:aaa_authorize_sterman: unable to add User-Name attribute

 

Here is my opensips.cfg

 

loadmodule auth.so

loadmodule auth_db.so

loadmodule aaa_radius.so

loadmodule auth_aaa.so

 

modparam(auth_aaa, aaa_url, 
radius:/usr/local/etc/radiusclient-ng/radiusclient.conf)

 

..

  if(!aaa_www_authorize())

   {

   xlog(L_INFO, *-*-* False in Radius_www_authorize , 
challenging M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci .. \n);

   www_challenge(, 1);

   exit;

   }

…

 

I am very appreciated if someone can point me a hint where is the problem

Thanks in advance

 

 

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Re: [OpenSIPS-Users] FW: Error when setting OpenSips with Radius

[Denis Putyato]

Try to add “Sets” param. with User-Name attribute for acc_radius module in your 
opensips.cfg 

For example, 

modparam(aaa_radius,sets,set1 = (User-Name=$var(usr))”)

 

where $var(usr) is some PV of your callerid 

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Tung Tran
Sent: Tuesday, July 27, 2010 4:55 PM
To: 'OpenSIPS users mailling list'
Subject: [OpenSIPS-Users] FW: Error when setting OpenSips with Radius

 

Hello, 

 

Just wanna recall if someone can help me out.

 

Thanks

T.T

 

From: Tung Tran [mailto:tr.t...@gmail.com] 
Sent: Tuesday, July 27, 2010 9:33 AM
To: 'OpenSIPS users mailling list'
Subject: Error when setting OpenSips with Radius

 

Hi all

 

I am building the Opensips 1.6.2 to run with external Billing via Radius (using 
radiusclient-ng), but I get this error when trying to register. No radius 
message is sent to Radius server yet

 

Jul 27 08:12:03 SVR60 /usr/local/sbin/opensips[19907]: rc_avpair_new: unknown 
attribute 0

Jul 27 08:12:03 SVR60 /usr/local/sbin/opensips[19907]: 
ERROR:aaa_radius:rad_avp_add: failure 

Jul 27 08:12:03 SVR60 /usr/local/sbin/opensips[19907]: 
ERROR:auth_aaa:aaa_authorize_sterman: unable to add User-Name attribute

 

Here is my opensips.cfg

 

loadmodule auth.so

loadmodule auth_db.so

loadmodule aaa_radius.so

loadmodule auth_aaa.so

 

modparam(auth_aaa, aaa_url, 
radius:/usr/local/etc/radiusclient-ng/radiusclient.conf)

 

..

  if(!aaa_www_authorize())

   {

   xlog(L_INFO, *-*-* False in Radius_www_authorize , 
challenging M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci .. \n);

   www_challenge(, 1);

   exit;

   }

…

 

I am very appreciated if someone can point me a hint where is the problem

Thanks in advance

 

 

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Re: [OpenSIPS-Users] Dialplan module

[Denis Putyato]

Hello

Are you using regexp in  repl_exp ?

-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Антон Загорский
Sent: Tuesday, July 20, 2010 4:19 PM
To: users@lists.opensips.org
Subject: [OpenSIPS-Users] Dialplan module

Hello.

I'm using a head version of opensips. In case when in the column
subst_expr there is nothing, dp_translate() does not place to the output
the repl_exp column value. 
But it should accordingly the dialplan module documentation.



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Re: [OpenSIPS-Users] Dialplan module

[Denis Putyato]

Show the string in dialplan table wholly and what can you see in syslog while 
process a call? 

-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Антон Загорский
Sent: Tuesday, July 20, 2010 4:39 PM
To: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] Dialplan module

No, just an exact string like sip:1...@mydomain.com

 
 Hello
 
 Are you using regexp in  repl_exp ?
 
 -Original Message-
 From: users-boun...@lists.opensips.org [mailto:users-
 boun...@lists.opensips.org] On Behalf Of Антон Загорский
 Sent: Tuesday, July 20, 2010 4:19 PM
 To: users@lists.opensips.org
 Subject: [OpenSIPS-Users] Dialplan module
 
 Hello.
 
 I'm using a head version of opensips. In case when in the column
 subst_expr there is nothing, dp_translate() does not place to the output
 the repl_exp column value.
 But it should accordingly the dialplan module documentation.
 
 
 
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Re: [OpenSIPS-Users] ACC_RADIUS makes opensips crash

[Denis Putyato]

Hello, Irina

Now it works, thank you. But, if you don't mind,  one more question about 
RADIUS.
I want to use Cisco dictionary for auth. subscribers. 
Dictionary is a main file used for radius in my opensips. I insert into this 
file such string  $INCLUDE /etc/radiusclient-ng/dictionary.cisco  where 
dictionary.cisco is a cisco specific dictionary.  

a little part of opensips.cfg
{...
...
modparam(aaa_radius, sets, set1 = (User-Name = $avp(i:20), 
User-Password=$avp(i:50)))
modparam(aaa_radius, sets, set2 = (Cisco-AVPair = $var(duration)))
...
...
radius_send_auth(set1,set2);
...
...
}

After made a call I can see such string in syslog
 /usr/local/opensips/sbin/opensips[7435]: ERROR:aaa_radius:send_auth_func: 
attribute was not found in received radius message 

# cat /etc/radiusclient-ng/dictionary.cisco | grep Cisco
...
VENDOR  Cisco   9
BEGIN-VENDORCisco
ATTRIBUTE   Cisco-AVPair1   string  
vendor=Cisco 
...

-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Irina Stanescu
Sent: Thursday, July 15, 2010 6:11 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] ACC_RADIUS makes opensips crash

Hello Denis,

I think the problem in you case is that the radius dictionary does not contain 
an entry for SIP-AVP, and that is why attr is null, causing the crash.

I added a fix on the trunk and i also attached the patch to this email in case 
you don't use the trunk version. Please let me know if there are any other 
issues.

Regards,
Irina Stanescu

On Thu, Jul 15, 2010 at 3:29 PM, Denis Putyato denis7...@mail.ru wrote:
 Bogdan, i made another call which makes opensips crash

 ...
 Core was generated by `/usr/local/opensips/sbin/opensips -P 
 /var/run/opensips.pid'.
 Program terminated with signal 11, Segmentation fault.
 [New process 8833]
 #0  0xb7a0adf1 in send_auth_func (msg=0x81c113c, s1=0x81bcd34, 
 s2=0x81bcd48) at aaa_radius.c:369
 369 for(; (vp = rc_avpair_get(vp, attr-value, 0)); vp = 
 vp-next)
 (gdb) bt
 #0  0xb7a0adf1 in send_auth_func (msg=0x81c113c, s1=0x81bcd34, 
 s2=0x81bcd48) at aaa_radius.c:369
 #1  0x08056111 in do_action (a=0x81b7404, msg=0x81c113c) at 
 action.c:967
 #2  0x08054f9e in run_action_list (a=0x81b5e14, msg=0x81c113c) at 
 action.c:139
 #3  0x08057f97 in do_action (a=0x81b7680, msg=0x81c113c) at 
 action.c:706
 #4  0x08054f9e in run_action_list (a=0x81b5938, msg=0x81c113c) at 
 action.c:139
 #5  0x08057946 in do_action (a=0x81aa660, msg=0x81c113c) at 
 action.c:119
 #6  0x08054f9e in run_action_list (a=0x81aa660, msg=0x81c113c) at 
 action.c:139
 #7  0x08057f97 in do_action (a=0x81aa758, msg=0x81c113c) at 
 action.c:706
 #8  0x08054f9e in run_action_list (a=0x81aa758, msg=0x81c113c) at 
 action.c:139
 #9  0x08057f97 in do_action (a=0x81aaff4, msg=0x81c113c) at 
 action.c:706 #10 0x08054f9e in run_action_list (a=0x81aaff4, 
 msg=0x81c113c) at action.c:139
 #11 0x08057f97 in do_action (a=0x81ac2d8, msg=0x81c113c) at 
 action.c:706
 #12 0x08054f9e in run_action_list (a=0x81a9f24, msg=0x81c113c) at 
 action.c:139
 #13 0x08057946 in do_action (a=0x81b428c, msg=0x81c113c) at 
 action.c:119
 #14 0x08054f9e in run_action_list (a=0x81b1ab0, msg=0x81c113c) at 
 action.c:139
 #15 0x08057946 in do_action (a=0x81b1308, msg=0x81c113c) at 
 action.c:119
 #16 0x08054f9e in run_action_list (a=0x81b1308, msg=0x81c113c) at 
 action.c:139
 #17 0x08057f97 in do_action (a=0x81b14cc, msg=0x81c113c) at 
 action.c:706
 #18 0x08054f9e in run_action_list (a=0x81b0a70, msg=0x81c113c) at 
 action.c:139
 #19 0x08057946 in do_action (a=0x81b078c, msg=0x81c113c) at 
 action.c:119 #20 0x08054f9e in run_action_list (a=0x81ad9e0, 
 msg=0x81c113c) at action.c:139
 #21 0x08057946 in do_action (a=0x81ad50c, msg=0x81c113c) at 
 action.c:119
 #22 0x08054f9e in run_action_list (a=0x81aca9c, msg=0x81c113c) at 
 action.c:139
 #23 0x08057946 in do_action (a=0x81a9cb4, msg=0x81c113c) at 
 action.c:119
 #24 0x08054f9e in run_action_list (a=0x81a6c50, msg=0x81c113c) at 
 action.c:139
 #25 0x08057f97 in do_action (a=0x81a9d8c, msg=0x81c113c) at 
 action.c:706
 #26 0x08054f9e in run_action_list (a=0x81a47e0, msg=0x81c113c) at 
 action.c:139
 #27 0x080590bf in run_top_route (a=0x81a47e0, msg=0x81c113c) at 
 action.c:119
 #28 0x08098b9c in receive_msg (
buf=0x8178200 INVITE sip:4483...@213.170.75.90:5060 
 SIP/2.0\r\nVia: SIP/2.0/UDP 
 213.170.75.90:5050;branch=z9hG4bK6912c413;rport\r\nMax-Forwards: 
 70\r\nFrom: \3364079\ 
 sip:3364...@213.170.75.90:5050;tag=as477593c3\r\nTo: ..., len=826,
rcv_info=0xbfeaed48) at receive.c:162
 #29 0x080da834 in udp_rcv_loop () at udp_server.c:492 #30 0x0806ee80 
 in main (argc=3, argv=0xbfeaeee4) at main.c:818
 (gdb) print vp
 $1 = (VALUE_PAIR *) 0x854c8f8
 (gdb) print attr
 $2 = (DICT_ATTR *) 0x0
 (gdb)

 -Original Message-
 From: users-boun...@lists.opensips.org 
 [mailto:users-boun...@lists.opensips.org

Re: [OpenSIPS-Users] ACC_RADIUS makes opensips crash

[Denis Putyato]

Yes, Cisco dictionary included in the dictionary of the
RADIUS server and server really send attr. that i want.

I attach a pcap file (radius3.log) of session between opensips and radius 
server.

-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Irina Stanescu
Sent: Friday, July 16, 2010 5:13 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] ACC_RADIUS makes opensips crash

Hello Denis,

Is the cisco specific dictionary included in the dictionary of the
RADIUS server too?

Also, to be able to extract a certain AVP from a RADIUS reply, you
need to make sure you have configured the server to return the
attributes you want. You can find a brief tutorial on this here [1].


Regards,
Irina Stanescu


[1] http://www.opensips.org/Resources/DocsTutRadius#toc4


On Fri, Jul 16, 2010 at 12:20 PM, Denis Putyato denis7...@mail.ru wrote:
 Hello, Irina

 Now it works, thank you. But, if you don't mind,  one more question about 
 RADIUS.
 I want to use Cisco dictionary for auth. subscribers.
 Dictionary is a main file used for radius in my opensips. I insert into this 
 file such string  $INCLUDE /etc/radiusclient-ng/dictionary.cisco  where 
 dictionary.cisco is a cisco specific dictionary.

 a little part of opensips.cfg
 {...
 ...
 modparam(aaa_radius, sets, set1 = (User-Name = $avp(i:20), 
 User-Password=$avp(i:50)))
 modparam(aaa_radius, sets, set2 = (Cisco-AVPair = $var(duration)))
 ...
 ...
 radius_send_auth(set1,set2);
 ...
 ...
 }

 After made a call I can see such string in syslog
  /usr/local/opensips/sbin/opensips[7435]: ERROR:aaa_radius:send_auth_func: 
 attribute was not found in received radius message 

 # cat /etc/radiusclient-ng/dictionary.cisco | grep Cisco
 ...
 VENDOR  Cisco   9
 BEGIN-VENDORCisco
 ATTRIBUTE   Cisco-AVPair1   string  
 vendor=Cisco
 ...

 -Original Message-
 From: users-boun...@lists.opensips.org 
 [mailto:users-boun...@lists.opensips.org] On Behalf Of Irina Stanescu
 Sent: Thursday, July 15, 2010 6:11 PM
 To: OpenSIPS users mailling list
 Subject: Re: [OpenSIPS-Users] ACC_RADIUS makes opensips crash

 Hello Denis,

 I think the problem in you case is that the radius dictionary does not 
 contain an entry for SIP-AVP, and that is why attr is null, causing the 
 crash.

 I added a fix on the trunk and i also attached the patch to this email in 
 case you don't use the trunk version. Please let me know if there are any 
 other issues.

 Regards,
 Irina Stanescu

 On Thu, Jul 15, 2010 at 3:29 PM, Denis Putyato denis7...@mail.ru wrote:
 Bogdan, i made another call which makes opensips crash

 ...
 Core was generated by `/usr/local/opensips/sbin/opensips -P 
 /var/run/opensips.pid'.
 Program terminated with signal 11, Segmentation fault.
 [New process 8833]
 #0  0xb7a0adf1 in send_auth_func (msg=0x81c113c, s1=0x81bcd34,
 s2=0x81bcd48) at aaa_radius.c:369
 369 for(; (vp = rc_avpair_get(vp, attr-value, 0)); vp =
 vp-next)
 (gdb) bt
 #0  0xb7a0adf1 in send_auth_func (msg=0x81c113c, s1=0x81bcd34,
 s2=0x81bcd48) at aaa_radius.c:369
 #1  0x08056111 in do_action (a=0x81b7404, msg=0x81c113c) at
 action.c:967
 #2  0x08054f9e in run_action_list (a=0x81b5e14, msg=0x81c113c) at
 action.c:139
 #3  0x08057f97 in do_action (a=0x81b7680, msg=0x81c113c) at
 action.c:706
 #4  0x08054f9e in run_action_list (a=0x81b5938, msg=0x81c113c) at
 action.c:139
 #5  0x08057946 in do_action (a=0x81aa660, msg=0x81c113c) at
 action.c:119
 #6  0x08054f9e in run_action_list (a=0x81aa660, msg=0x81c113c) at
 action.c:139
 #7  0x08057f97 in do_action (a=0x81aa758, msg=0x81c113c) at
 action.c:706
 #8  0x08054f9e in run_action_list (a=0x81aa758, msg=0x81c113c) at
 action.c:139
 #9  0x08057f97 in do_action (a=0x81aaff4, msg=0x81c113c) at
 action.c:706 #10 0x08054f9e in run_action_list (a=0x81aaff4,
 msg=0x81c113c) at action.c:139
 #11 0x08057f97 in do_action (a=0x81ac2d8, msg=0x81c113c) at
 action.c:706
 #12 0x08054f9e in run_action_list (a=0x81a9f24, msg=0x81c113c) at
 action.c:139
 #13 0x08057946 in do_action (a=0x81b428c, msg=0x81c113c) at
 action.c:119
 #14 0x08054f9e in run_action_list (a=0x81b1ab0, msg=0x81c113c) at
 action.c:139
 #15 0x08057946 in do_action (a=0x81b1308, msg=0x81c113c) at
 action.c:119
 #16 0x08054f9e in run_action_list (a=0x81b1308, msg=0x81c113c) at
 action.c:139
 #17 0x08057f97 in do_action (a=0x81b14cc, msg=0x81c113c) at
 action.c:706
 #18 0x08054f9e in run_action_list (a=0x81b0a70, msg=0x81c113c) at
 action.c:139
 #19 0x08057946 in do_action (a=0x81b078c, msg=0x81c113c) at
 action.c:119 #20 0x08054f9e in run_action_list (a=0x81ad9e0,
 msg=0x81c113c) at action.c:139
 #21 0x08057946 in do_action (a=0x81ad50c, msg=0x81c113c) at
 action.c:119
 #22 0x08054f9e in run_action_list (a=0x81aca9c, msg=0x81c113c) at
 action.c:139
 #23 0x08057946 in do_action (a=0x81a9cb4, msg=0x81c113c) at
 action.c:119
 #24 0x08054f9e

Re: [OpenSIPS-Users] ACC_RADIUS makes opensips crash

[Denis Putyato]

Bogdan, i made another call which makes opensips crash

...
Core was generated by `/usr/local/opensips/sbin/opensips -P 
/var/run/opensips.pid'.
Program terminated with signal 11, Segmentation fault.
[New process 8833]
#0  0xb7a0adf1 in send_auth_func (msg=0x81c113c, s1=0x81bcd34, s2=0x81bcd48) at 
aaa_radius.c:369
369 for(; (vp = rc_avpair_get(vp, attr-value, 0)); vp = vp-next)
(gdb) bt
#0  0xb7a0adf1 in send_auth_func (msg=0x81c113c, s1=0x81bcd34, s2=0x81bcd48) at 
aaa_radius.c:369
#1  0x08056111 in do_action (a=0x81b7404, msg=0x81c113c) at action.c:967
#2  0x08054f9e in run_action_list (a=0x81b5e14, msg=0x81c113c) at action.c:139
#3  0x08057f97 in do_action (a=0x81b7680, msg=0x81c113c) at action.c:706
#4  0x08054f9e in run_action_list (a=0x81b5938, msg=0x81c113c) at action.c:139
#5  0x08057946 in do_action (a=0x81aa660, msg=0x81c113c) at action.c:119
#6  0x08054f9e in run_action_list (a=0x81aa660, msg=0x81c113c) at action.c:139
#7  0x08057f97 in do_action (a=0x81aa758, msg=0x81c113c) at action.c:706
#8  0x08054f9e in run_action_list (a=0x81aa758, msg=0x81c113c) at action.c:139
#9  0x08057f97 in do_action (a=0x81aaff4, msg=0x81c113c) at action.c:706
#10 0x08054f9e in run_action_list (a=0x81aaff4, msg=0x81c113c) at action.c:139
#11 0x08057f97 in do_action (a=0x81ac2d8, msg=0x81c113c) at action.c:706
#12 0x08054f9e in run_action_list (a=0x81a9f24, msg=0x81c113c) at action.c:139
#13 0x08057946 in do_action (a=0x81b428c, msg=0x81c113c) at action.c:119
#14 0x08054f9e in run_action_list (a=0x81b1ab0, msg=0x81c113c) at action.c:139
#15 0x08057946 in do_action (a=0x81b1308, msg=0x81c113c) at action.c:119
#16 0x08054f9e in run_action_list (a=0x81b1308, msg=0x81c113c) at action.c:139
#17 0x08057f97 in do_action (a=0x81b14cc, msg=0x81c113c) at action.c:706
#18 0x08054f9e in run_action_list (a=0x81b0a70, msg=0x81c113c) at action.c:139
#19 0x08057946 in do_action (a=0x81b078c, msg=0x81c113c) at action.c:119
#20 0x08054f9e in run_action_list (a=0x81ad9e0, msg=0x81c113c) at action.c:139
#21 0x08057946 in do_action (a=0x81ad50c, msg=0x81c113c) at action.c:119
#22 0x08054f9e in run_action_list (a=0x81aca9c, msg=0x81c113c) at action.c:139
#23 0x08057946 in do_action (a=0x81a9cb4, msg=0x81c113c) at action.c:119
#24 0x08054f9e in run_action_list (a=0x81a6c50, msg=0x81c113c) at action.c:139
#25 0x08057f97 in do_action (a=0x81a9d8c, msg=0x81c113c) at action.c:706
#26 0x08054f9e in run_action_list (a=0x81a47e0, msg=0x81c113c) at action.c:139
#27 0x080590bf in run_top_route (a=0x81a47e0, msg=0x81c113c) at action.c:119
#28 0x08098b9c in receive_msg (
buf=0x8178200 INVITE sip:4483...@213.170.75.90:5060 SIP/2.0\r\nVia: 
SIP/2.0/UDP 213.170.75.90:5050;branch=z9hG4bK6912c413;rport\r\nMax-Forwards: 
70\r\nFrom: \3364079\ sip:3364...@213.170.75.90:5050;tag=as477593c3\r\nTo: 
..., len=826, 
rcv_info=0xbfeaed48) at receive.c:162
#29 0x080da834 in udp_rcv_loop () at udp_server.c:492
#30 0x0806ee80 in main (argc=3, argv=0xbfeaeee4) at main.c:818
(gdb) print vp
$1 = (VALUE_PAIR *) 0x854c8f8
(gdb) print attr
$2 = (DICT_ATTR *) 0x0
(gdb)  

-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
Sent: Thursday, July 15, 2010 3:48 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] ACC_RADIUS makes opensips crash

Hi Denis,

That's perfect - thank you.

Could you print in GDB the following values : vp , attr.

Regards,
Bogdan

Denis Putyato wrote:
 Hello, Bogdan

 Is this information you asked?

 gdb /usr/local/opensips/sbin/opensips /core 
 GNU gdb 6.8-debian
 Copyright (C) 2008 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 i486-linux-gnu...

 warning: Can't read pathname for load map: Input/output error.
 ...
 ...
 Core was generated by `/usr/local/opensips/sbin/opensips -P 
 /var/run/opensips.pid'.
 Program terminated with signal 11, Segmentation fault.
 [New process 24328]
 #0  0xb79b7df1 in send_auth_func (msg=0x81c0fd4, s1=0x81bcbcc, s2=0x81bcbe0) 
 at aaa_radius.c:369
 369   for(; (vp = rc_avpair_get(vp, attr-value, 0)); vp = vp-next)
 (gdb) bt
 #0  0xb79b7df1 in send_auth_func (msg=0x81c0fd4, s1=0x81bcbcc, s2=0x81bcbe0) 
 at aaa_radius.c:369
 #1  0x08056111 in do_action (a=0x81b729c, msg=0x81c0fd4) at action.c:967
 #2  0x08054f9e in run_action_list (a=0x81b5e14, msg=0x81c0fd4) at action.c:139
 #3  0x08057f97 in do_action (a=0x81b7518, msg=0x81c0fd4) at action.c:706
 #4  0x08054f9e in run_action_list (a=0x81b5938, msg=0x81c0fd4) at action.c:139
 #5  0x08057946 in do_action (a=0x81aa660, msg=0x81c0fd4) at action.c:119
 #6  0x08054f9e in run_action_list (a=0x81aa660, msg=0x81c0fd4) at action.c:139
 #7  0x08057f97 in do_action (a=0x81aa758, msg

[OpenSIPS-Users] ACC_RADIUS makes opensips crash

[Denis Putyato]

Hello everybody!

 

There is a problem with radius_send_auth(); function.

This function is called from request route and after opensips received 
“Access-Accept” from radius server it is crashes with such error:

 

Jul 14 14:59:02 kam /usr/local/opensips/sbin/opensips[21556]: 
DBG:aaa_radius:send_auth_func: radius authentication message sent 

Jul 14 14:59:02 kam kernel: [118250.317522] opensips[21556]: segfault at 24 ip 
b7a53df1 sp bf8f5590 error 4 in aaa_radius.so[b7a51000+7000]

Jul 14 14:59:02 kam /usr/local/opensips/sbin/opensips[21561]: 
CRITICAL:core:receive_fd: EOF on 7

 

 

Opensips.cfg:

…

…

modparam(aaa_radius, radius_config, 
/etc/radiusclient-ng/radiusclient.conf)

modparam(aaa_radius, sets, set1 = (User-Name = $avp(i:20), 
User-Password=$avp(i:50)))

modparam(aaa_radius, sets, set2 = (Session-Timeout = $var(time)))

…

…

route [7] {

…

radius_send_auth(set1,set2);

…

…

}

 

 

$avp(i:20) – tel. number

$avp(i:50) – empty value. Has been inserted because radius server didn’t accept 
request from opensips without User-Password field

Session-Timeout is ONLY one attribute that opensips receives from radius server.

#cat /etc/radiusclient-ng/dictionary | grep Session-Timeout

ATTRIBUTE Session-Timeout27   
integer

 

Thank you for any help.

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

Re: [OpenSIPS-Users] ACC_RADIUS makes opensips crash

[Denis Putyato]

Hello, Bogdan

Is this information you asked?

gdb /usr/local/opensips/sbin/opensips /core 
GNU gdb 6.8-debian
Copyright (C) 2008 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 i486-linux-gnu...

warning: Can't read pathname for load map: Input/output error.
...
...
Core was generated by `/usr/local/opensips/sbin/opensips -P 
/var/run/opensips.pid'.
Program terminated with signal 11, Segmentation fault.
[New process 24328]
#0  0xb79b7df1 in send_auth_func (msg=0x81c0fd4, s1=0x81bcbcc, s2=0x81bcbe0) at 
aaa_radius.c:369
369 for(; (vp = rc_avpair_get(vp, attr-value, 0)); vp = vp-next)
(gdb) bt
#0  0xb79b7df1 in send_auth_func (msg=0x81c0fd4, s1=0x81bcbcc, s2=0x81bcbe0) at 
aaa_radius.c:369
#1  0x08056111 in do_action (a=0x81b729c, msg=0x81c0fd4) at action.c:967
#2  0x08054f9e in run_action_list (a=0x81b5e14, msg=0x81c0fd4) at action.c:139
#3  0x08057f97 in do_action (a=0x81b7518, msg=0x81c0fd4) at action.c:706
#4  0x08054f9e in run_action_list (a=0x81b5938, msg=0x81c0fd4) at action.c:139
#5  0x08057946 in do_action (a=0x81aa660, msg=0x81c0fd4) at action.c:119
#6  0x08054f9e in run_action_list (a=0x81aa660, msg=0x81c0fd4) at action.c:139
#7  0x08057f97 in do_action (a=0x81aa758, msg=0x81c0fd4) at action.c:706
#8  0x08054f9e in run_action_list (a=0x81aa758, msg=0x81c0fd4) at action.c:139
#9  0x08057f97 in do_action (a=0x81aaff4, msg=0x81c0fd4) at action.c:706
#10 0x08054f9e in run_action_list (a=0x81aaff4, msg=0x81c0fd4) at action.c:139
#11 0x08057f97 in do_action (a=0x81ac2d8, msg=0x81c0fd4) at action.c:706
#12 0x08054f9e in run_action_list (a=0x81a9f24, msg=0x81c0fd4) at action.c:139
#13 0x08057946 in do_action (a=0x81b428c, msg=0x81c0fd4) at action.c:119
#14 0x08054f9e in run_action_list (a=0x81b1ab0, msg=0x81c0fd4) at action.c:139
#15 0x08057946 in do_action (a=0x81b1308, msg=0x81c0fd4) at action.c:119
#16 0x08054f9e in run_action_list (a=0x81b1308, msg=0x81c0fd4) at action.c:139
#17 0x08057f97 in do_action (a=0x81b14cc, msg=0x81c0fd4) at action.c:706
#18 0x08054f9e in run_action_list (a=0x81b0a70, msg=0x81c0fd4) at action.c:139
#19 0x08057946 in do_action (a=0x81b078c, msg=0x81c0fd4) at action.c:119
#20 0x08054f9e in run_action_list (a=0x81ad9e0, msg=0x81c0fd4) at action.c:139
#21 0x08057946 in do_action (a=0x81ad50c, msg=0x81c0fd4) at action.c:119
#22 0x08054f9e in run_action_list (a=0x81aca9c, msg=0x81c0fd4) at action.c:139
#23 0x08057946 in do_action (a=0x81a9cb4, msg=0x81c0fd4) at action.c:119
#24 0x08054f9e in run_action_list (a=0x81a6c50, msg=0x81c0fd4) at action.c:139
#25 0x08057f97 in do_action (a=0x81a9d8c, msg=0x81c0fd4) at action.c:706
#26 0x08054f9e in run_action_list (a=0x81a47e0, msg=0x81c0fd4) at action.c:139
#27 0x080590bf in run_top_route (a=0x81a47e0, msg=0x81c0fd4) at action.c:119
#28 0x08098b9c in receive_msg (
buf=0x8178200 INVITE sip:3364...@1.1.1.1:5060 SIP/2.0\r\nVia: SIP/2.0/UDP 
1.1.1.1:5050;branch=z9hG4bK7d781018;rport\r\nMax-Forwards: 70\r\nFrom: 
\3364079\ sip:3364...@1.1.1.1:5050;tag=as3bb11c83\r\nTo: ..., len=826, 
rcv_info=0xbfb599f8) at receive.c:162
#29 0x080da834 in udp_rcv_loop () at udp_server.c:492
#30 0x0806ee80 in main (argc=3, argv=0xbfb59b94) at main.c:818

-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
Sent: Wednesday, July 14, 2010 7:37 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] ACC_RADIUS makes opensips crash

Hi Denis,

do you get a coredump file? if so, could you get a bracktrace from it 
and post it here?

Regards,
Bogdan

Denis Putyato wrote:

 Hello everybody!

 There is a problem with radius_send_auth(); function.

 This function is called from request route and after opensips received 
 “Access-Accept” from radius server it is crashes with such error:

 Jul 14 14:59:02 kam /usr/local/opensips/sbin/opensips[21556]: 
 DBG:aaa_radius:send_auth_func: radius authentication message sent

 Jul 14 14:59:02 kam kernel: [118250.317522] opensips[21556]: segfault 
 at 24 ip b7a53df1 sp bf8f5590 error 4 in aaa_radius.so[b7a51000+7000]

 Jul 14 14:59:02 kam /usr/local/opensips/sbin/opensips[21561]: 
 CRITICAL:core:receive_fd: EOF on 7

 Opensips.cfg:

 …

 …

 modparam(aaa_radius, radius_config, 
 /etc/radiusclient-ng/radiusclient.conf)

 modparam(aaa_radius, sets, set1 = (User-Name = $avp(i:20), 
 User-Password=$avp(i:50)))

 modparam(aaa_radius, sets, set2 = (Session-Timeout = $var(time)))

 …

 …

 route [7] {

 …

 radius_send_auth(set1,set2);

 …

 …

 }

 $avp(i:20) – tel. number

 $avp(i:50) – empty value. Has been inserted because radius server 
 didn’t accept request from opensips without User-Password field

 Session-Timeout is ONLY one attribute that opensips receives