Re: [OpenSIPS-Users] uac_registrant preferences
Hi, Ovidiu.
Yes, I use FQDN for registrar and somebody already said me here, that is the
possible problem.
I see next errors in the log:
Mar 16 03:48:36 node1 opensips[25954]: CRITICAL:core:timer_ticker: timer
handler lasted (218 us) for more than timer tick (100
us) -> potential timer shifting
Mar 16 03:48:56 node1 opensips[25954]: CRITICAL:core:timer_ticker: timer
handler lasted (274 us) for more than timer tick (100
us) -> potential timer shifting
Mar 16 03:49:36 node1 opensips[25954]: CRITICAL:core:timer_ticker: timer
handler lasted (267 us) for more than timer tick (100
us) -> potential timer shifting
Do you need gdb of any process of opensips?
Best regards,
Alexander Mustafin
mustafin.aleksa...@gmail.com
> 15 марта 2016 г., в 21:47, Ovidiu Sas написал(а):
>
> On Tue, Mar 15, 2016 at 8:09 AM, Alexander Mustafin
> mailto:mustafin.aleksa...@gmail.com>> wrote:
>> Hi there!
>>
>> Perhaps I don’t understand logic of this module, but I have strange problem:
>>
>> version: opensips 1.11.6-tls (x86_64/linux)
>>
>> I have 223 entries in my db. Previously, preferences of module was like
>> these:
>> modparam("uac_registrant", "hash_size", 2)
>> modparam("uac_registrant", "db_url", "SQL_URL")
>> modparam("uac_registrant", "table_name", "REGISTRANT_TABLE")
>> modparam("uac_registrant", "timer_interval", 100)
>>
>> All works fine, but I had awful CPU load, almost 100%.
>
> In this setup, assuming that you had an even distribution, you would
> have around 223/4=55 records per hash slot and each slot should be
> checked every 100/4=25s.
> Can you attach with gdb to the process to see what is happening there?
> Do you see any errors in the logs?
>
>> Then, I changed it to
>>
>> modparam("uac_registrant", "hash_size", 5)
>> modparam("uac_registrant", "db_url", "SQL_URL")
>> modparam("uac_registrant", "table_name", "REGISTRANT_TABLE")
>> modparam("uac_registrant", "timer_interval", 640)
>>
>> Processor feels good now, but I experience problem with registrations:
>> opensipsctl fifo reg_list |grep AOR|wc -l
>> 166
>>
>> Opensips shows only 166 entries in memory, so, I have no idea where is my
>> other registrations.
>>
>> How to tie hash_size, timer_interval and value in expiry column?
> See the Overview in the documentation.
> The expiry column is stores the binding's expiration time.
>
>> And why I
>> see this information for some of my numbers, where registration_t_out less
>> than last_register_sent:
>>
>> state:: REGISTERED_STATE
>> last_register_sent:: Tue Mar 15 06:25:38 2016
>> registration_t_out:: Tue Mar 15 06:18:57 2016
>> registrar:: sip:some.registrar
>> binding:: sip:777@89.88.87.866:5070
>>
>
> Do you see any errors in the logs?
> Are you using FQDNs for registrar?
>
>>
>> Thanks in advance!
>>
>> Best regards,
>> Alexander Mustafin
>> mustafin.aleksa...@gmail.com <mailto:mustafin.aleksa...@gmail.com>
>>
>>
>>
>>
>>
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>> <http://lists.opensips.org/cgi-bin/mailman/listinfo/users>
>>
>
>
>
> --
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> http://www.voipembedded.com <http://www.voipembedded.com/>
>
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[OpenSIPS-Users] uac_registrant preferences
Hi there!
Perhaps I don’t understand logic of this module, but I have strange problem:
version: opensips 1.11.6-tls (x86_64/linux)
I have 223 entries in my db. Previously, preferences of module was like these:
modparam("uac_registrant", "hash_size", 2)
modparam("uac_registrant", "db_url", "SQL_URL")
modparam("uac_registrant", "table_name", "REGISTRANT_TABLE")
modparam("uac_registrant", "timer_interval", 100)
All works fine, but I had awful CPU load, almost 100%. Then, I changed it to
modparam("uac_registrant", "hash_size", 5)
modparam("uac_registrant", "db_url", "SQL_URL")
modparam("uac_registrant", "table_name", "REGISTRANT_TABLE")
modparam("uac_registrant", "timer_interval", 640)
Processor feels good now, but I experience problem with registrations:
opensipsctl fifo reg_list |grep AOR|wc -l
166
Opensips shows only 166 entries in memory, so, I have no idea where is my other
registrations.
How to tie hash_size, timer_interval and value in expiry column? And why I see
this information for some of my numbers, where registration_t_out less than
last_register_sent:
state:: REGISTERED_STATE
last_register_sent:: Tue Mar 15 06:25:38 2016
registration_t_out:: Tue Mar 15 06:18:57 2016
registrar:: sip:s ome.registrar
binding:: sip:777@89.88.87.866:5070
Thanks in advance!
Best regards,
Alexander Mustafin
mustafin.aleksa...@gmail.com
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[OpenSIPS-Users] Hash in the uac_registrant module
Hello. Is it possible to use hash of passwords in uac_registrant table? It isn’t very secure to keep plain passwords. Best regards, Alexander Mustafin mustafin.aleksa...@gmail.com signature.asc Description: Message signed with OpenPGP using GPGMail ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Reload group cache
Hello! I used groups based on regular expressions and documentation says: "Due performance reasons (regular expression evaluation), DB cache support is available: the table content is loaded into memory at startup and all regular expressions are compiled. « But no functions for reloading cache, as I see - and I need to restart open sips for renew cache. Does reload implemented in module? Or maybe it possible to disable cache support if it not implemented? Best regards, Alexander Mustafin mustafin.aleksa...@gmail.com ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Get variables in failure route
Hi, Liviu.
Thank you very much for the advice. Now, all works fine!
Best regards,
Alexander Mustafin
mustafin.aleksa...@gmail.com
22 окт. 2014 г., в 17:20, Liviu Chircu написал(а):
> Hello Alexander,
>
> You need to specify the context of a pseudo-var. [1]
>
> For your script, you should use $(fU) and $(fd).
>
> [1]: http://www.opensips.org/Documentation/Script-CoreVar-2-1
>
> Best regards,
> Liviu Chircu
> OpenSIPS Developer
> http://www.opensips-solutions.com
> On 10/22/2014 01:13 PM, Alexander Mustafin wrote:
>> Hello.
>>
>> I need to receive variables from 407 message in failure route, but when I
>> tried it - I received a value from initial message.
>>
>> INVITE sip:05534...@sbc.sbc.sbc SIP/2.0.
>> Via: SIP/2.0/UDP 54.55.56.57:12000;rport;branch=z9hG4bK37Ny7QvNF7Hrc.
>> Max-Forwards: 70.
>> From: «User" ;tag=ct7BK7ccyaK7H.
>>
>> SIP/2.0 407 Proxy Authentication Required.
>> From: ;tag=ct7BK7ccyaK7H.
>> To: ;tag=9560995.
>>
>> failure_route[EXTERNAL_FAULT] {
>> if (t_was_cancelled()) {
>> xlog("L_INFO", "$ci|log|transaction was cancelled");
>>
>> exit;
>> }
>>
>> $var(auth_user) = $fU + "@" + $fd;
>> }
>>
>> In variable $var(auth_user) I’ve seen ...@office.sbc.sbc.
>>
>> Can I receive values from 407 message?
>>
>> Best regards,
>> Alexander Mustafin
>> mustafin.aleksa...@gmail.com
>>
>>
>>
>>
>>
>>
>> ___
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>
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[OpenSIPS-Users] Get variables in failure route
Hello.
I need to receive variables from 407 message in failure route, but when I tried
it - I received a value from initial message.
INVITE sip:05534...@sbc.sbc.sbc SIP/2.0.
Via: SIP/2.0/UDP 54.55.56.57:12000;rport;branch=z9hG4bK37Ny7QvNF7Hrc.
Max-Forwards: 70.
From: «User" ;tag=ct7BK7ccyaK7H.
SIP/2.0 407 Proxy Authentication Required.
From: ;tag=ct7BK7ccyaK7H.
To: ;tag=9560995.
failure_route[EXTERNAL_FAULT] {
if (t_was_cancelled()) {
xlog("L_INFO", "$ci|log|transaction was cancelled");
exit;
}
$var(auth_user) = $fU + "@" + $fd;
}
In variable $var(auth_user) I’ve seen ...@office.sbc.sbc.
Can I receive values from 407 message?
Best regards,
Alexander Mustafin
mustafin.aleksa...@gmail.com
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Re: [OpenSIPS-Users] Missed INVITE record in acc table
Thanks, Bogdan. I have got the next parameters in my script.
Route CREATE_DIALOG calls after some additional routes, where db_extra fields
are filling, drouting etc - but on the INVITE stage.
I’m just check missed_tables - and it not filled too, but missed and failures
calls are present.
modparam("acc", "db_table_acc", "acc")
modparam("acc", "db_table_missed_calls", "missed_calls")
modparam("acc", "early_media", 0)
modparam("acc", "report_cancels", 1)
modparam("acc", "detect_direction", 0)
modparam("acc", "failed_transaction_flag", "ACC_FAILED")
modparam("acc", "db_flag", "ACC_DO")
modparam("acc", "db_missed_flag", "ACC_MISSED")
modparam("acc", "cdr_flag", "CDR_FLAG»)
modparam("acc", "db_extra", "incoming_number=$avp(incoming_number);………)
route[CREATE_DIALOG] {
xlog("L_INFO", "$ci|log|Now we set flags for ACC module");
setflag(CDR_FLAG);
setflag(ACC_DO);
setflag(ACC_FAILED);
setflag(ACC_MISSED);
setflag(QOS_FLAG);
create_dialog("B");
if ( is_gflag("2") ) {
xlog("L_INFO", "$ci|log|Flag 2 is set. We will use topology hiding for
our request");
topology_hiding();
}
……..
}
Best regards,
Alexander Mustafin
mustafin.aleksa...@gmail.com
03 сент. 2014 г., в 13:59, Bogdan-Andrei Iancu написал(а):
> Hi Alexander,
>
> At INVITE time be sure both flags (ACC_DO - accounting is required - and
> CDR_FLAG - do it as CDR) are set before you do create_dialog().
>
> Regards,
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
> On 03.09.2014 10:52, Alexander Mustafin wrote:
>> Hi, Bogdan.
>>
>> Sure, I’m use dialog module and create dialog. In modules section I’m load
>> acc module before dialog(because, I’ve read about problem with it).
>> I’ve tried to set CDR_FLAG before create_dialog and after - with the same
>> result. Also, I’ve disabled CDR_FLAG and used ACC_DO only, but INVITE record
>> still absent).
>>
>> I thought the problem with MySQL, but I’ve sniffed traffic to MySQL and
>> OpenSIPS not send anything for INVITE - only for dialog and BYE.
>>
>> How can I detect what’s happened when flags is sets?
>>
>> Best regards,
>> Alexander Mustafin
>> mustafin.aleksa...@gmail.com
>>
>>
>>
>>
>> 03 сент. 2014 г., в 13:42, Bogdan-Andrei Iancu
>> написал(а):
>>
>>> Hi Alexander,
>>>
>>> If you use the CDR_FLAG, you really must use dialog module and create the
>>> dialogs during the initial INVITE - do you do that ?
>>>
>>> Regards,
>>> Bogdan-Andrei Iancu
>>> OpenSIPS Founder and Developer
>>> http://www.opensips-solutions.com
>>> On 03.09.2014 10:21, Alexander Mustafin wrote:
>>>> Hello!
>>>>
>>>> I’ve the problem with acc - now, only BYE records presents in acc table.
>>>> I’ve set the flags ACC_DO and others for initial INVITE, tried to use
>>>> CDR_FALG, but without success.
>>>>
>>>> If I use CDR_FLAG with ACC_DO for INVITE and disable it for BYE - all
>>>> records for call absent.
>>>> Maybe, other parts of my script does broke ACC behavior, but I can’t see
>>>> an error in debug log.
>>>>
>>>> version: opensips 1.11.2-tls (x86_64/linux)
>>>> flags: STATS: On, USE_IPV6, USE_TCP, USE_TLS, DISABLE_NAGLE, USE_MCAST,
>>>> SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
>>>> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
>>>> MAX_URI_SIZE 1024, BUF_SIZE 65535
>>>> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
>>>> main.c compiled on 00:09:03 Jul 3 2014 with gcc 4.4.7
>>>>
>>>>
>>>> Best regards,
>>>> Alexander Mustafin
>>>> mustafin.aleksa...@gmail.com
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
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>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>
>
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Re: [OpenSIPS-Users] Missed INVITE record in acc table
Hi, Bogdan. Sure, I’m use dialog module and create dialog. In modules section I’m load acc module before dialog(because, I’ve read about problem with it). I’ve tried to set CDR_FLAG before create_dialog and after - with the same result. Also, I’ve disabled CDR_FLAG and used ACC_DO only, but INVITE record still absent). I thought the problem with MySQL, but I’ve sniffed traffic to MySQL and OpenSIPS not send anything for INVITE - only for dialog and BYE. How can I detect what’s happened when flags is sets? Best regards, Alexander Mustafin mustafin.aleksa...@gmail.com 03 сент. 2014 г., в 13:42, Bogdan-Andrei Iancu написал(а): > Hi Alexander, > > If you use the CDR_FLAG, you really must use dialog module and create the > dialogs during the initial INVITE - do you do that ? > > Regards, > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > On 03.09.2014 10:21, Alexander Mustafin wrote: >> Hello! >> >> I’ve the problem with acc - now, only BYE records presents in acc table. >> I’ve set the flags ACC_DO and others for initial INVITE, tried to use >> CDR_FALG, but without success. >> >> If I use CDR_FLAG with ACC_DO for INVITE and disable it for BYE - all >> records for call absent. >> Maybe, other parts of my script does broke ACC behavior, but I can’t see an >> error in debug log. >> >> version: opensips 1.11.2-tls (x86_64/linux) >> flags: STATS: On, USE_IPV6, USE_TCP, USE_TLS, DISABLE_NAGLE, USE_MCAST, >> SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT >> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, >> MAX_URI_SIZE 1024, BUF_SIZE 65535 >> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. >> main.c compiled on 00:09:03 Jul 3 2014 with gcc 4.4.7 >> >> >> Best regards, >> Alexander Mustafin >> mustafin.aleksa...@gmail.com >> >> >> >> >> >> >> ___ >> Users mailing list >> Users@lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > signature.asc Description: Message signed with OpenPGP using GPGMail ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Missed INVITE record in acc table
Hello! I’ve the problem with acc - now, only BYE records presents in acc table. I’ve set the flags ACC_DO and others for initial INVITE, tried to use CDR_FALG, but without success. If I use CDR_FLAG with ACC_DO for INVITE and disable it for BYE - all records for call absent. Maybe, other parts of my script does broke ACC behavior, but I can’t see an error in debug log. version: opensips 1.11.2-tls (x86_64/linux) flags: STATS: On, USE_IPV6, USE_TCP, USE_TLS, DISABLE_NAGLE, USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. main.c compiled on 00:09:03 Jul 3 2014 with gcc 4.4.7 Best regards, Alexander Mustafin mustafin.aleksa...@gmail.com signature.asc Description: Message signed with OpenPGP using GPGMail ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Amazon EC2 and topology_hiding(U) looping ACK
Hello! I’ve a problem with looping ACK, when using topology_hiding(«U») on Amazon EC2 server with Elastic IP. All works fine without topology_hiding. I’ve set advertised_address=«54.54.54.54» (it;s example of my Elastic IP address) option in config and have no other special functions like «rewrite host()» and others in config. And now, when I receive ACK - it’s looping (some headers are cuted): U 2014/08/21 10:18:29.040856 62.62.62.62:5060 -> 10.0.0.196:5070 ACK sip:777@54.54.54.54:5070;did=d99.910e2214 SIP/2.0. From: ;tag=-45026-48be65b-339c17e3-48be65b. To: «user user";tag=UjBF6gy1Q0Z7H. Call-ID: ead1318f74d2960620e85e487372d63a55e4a2@62.62.62.62. CSeq: 1 ACK. Via: SIP/2.0/UDP 62.62.62.62:5060;branch=z9hG4bK-70ee58-b9230b46-5c11ab8. U 2014/08/21 10:18:29.041446 10.0.0.196:5070 -> 54.54.54.54:5070 ACK sip:777@54.54.54.54:5070;did=d99.910e2214 SIP/2.0. From: ;tag=-45026-48be65b-339c17e3-48be65b. To: «user user";tag=UjBF6gy1Q0Z7H. Call-ID: ead1318f74d2960620e85e487372d63a55e4a2@62.62.62.62. CSeq: 1 ACK. Via: SIP/2.0/UDP 54.54.54.54:5070;branch=z9hG4bK4d9f.070d4732.2. Via: SIP/2.0/UDP 62.62.62.62:5060;branch=z9hG4bK-70ee58-b9230b46-5c11ab8. U 2014/08/21 10:18:29.043433 54.54.54.54:5070 -> 10.0.0.196:5070 ACK sip:777@54.54.54.54:5070;did=d99.910e2214 SIP/2.0. From: ;tag=-45026-48be65b-339c17e3-48be65b. To: "ration1 ration1";tag=UjBF6gy1Q0Z7H. Call-ID: ead1318f74d2960620e85e487372d63a55e4a2@62.62.62.62. CSeq: 1 ACK. Via: SIP/2.0/UDP 54.54.54.54:5070;branch=z9hG4bK4d9f.070d4732.2. Via: SIP/2.0/UDP 62.62.62.62:5060;branch=z9hG4bK-70ee58-b9230b46-5c11ab8. What I should to do to solve this problem? Best regards, Alexander Mustafin mustafin.aleksa...@gmail.com signature.asc Description: Message signed with OpenPGP using GPGMail ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Registration stale nonce
Hello!
I’ve many problems with registered user. Their registrations periodically
failed with error: «Stale nonce».
Why this happens? Buggy clients (I use Linksys and Cisco adapters)?
What I need to do? If I increase this time, or disable check_nonce - is this
very dangerous?
Now, I’m use default value: modparam("auth", "nonce_expire", 30).
Best regards,
Alexander Mustafin
mustafin.aleksa...@gmail.com
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[OpenSIPS-Users] Negative replies retransmission
Good day! I’ve a problem with negative replies retransmission. Dialog is destroyed at this moment by upstream negative reply. How to prevent this behaviour? FS OpenSIPS AST1 AST2 0.00 |->|| |INVITE sip:1234567...@sbc.ops.lan | +SDP [0s] 0.351449 |<-|| |100 Giving a try [0.35s] 0.351919 | |-->| |INVITE sip:1234567890@192.168.0.155 | +SDP [0.35s] 0.354767 | |<--| |100 Trying [0.35s] 1.309592 | |<--| |603 [1.31s] 1.31 | |-->| |ACK sip:1234567890@192.168.0.155 [1.31s] 1.886184 | |-->| INVITE sip:1234567890@192.168.0.199 | +SDP [1.89s] 1.951046 | |<--| 100 Trying [1.95s] 6.725141 | |<--| 183 Progress | +SDP [6.73s] 6.726316 |<-| | 183 Progress | +SDP [6.73s] 6.799651 | |<--| 180 Ringing [6.8s] 6.800444 |<-| | 180 Ringing [6.8s] 13.996213 |->| | CANCEL sip:1234567...@sbc.ops.lan [14s] 14.259989 | |--->| CANCEL sip:1234567890@192.168.0.199 [14.26s] 14.321767 | |<---| 200 OK [14.32s] 14.322271 | |<---| 487 Request Terminated [14.32s] 14.322748 | |--->| ACK sip:1234567890@192.168.0.199 [14.32s] 14.323271 |<-| | 487 Request Terminated [14.32s] 14.323859 |->| | ACK sip:1234567...@sbc.ops.lan [14.32s] 14.825038 |<-| | 487 Request Terminated [14.83s] 14.825669 |->| | ACK sip:1234567...@sbc.ops.lan [14.83s] 15.826212 |<-| | 487 Request Terminated [15.83s] 15.826901 |->| | ACK sip:1234567...@sbc.ops.lan [15.83s] 17.821087 |<-| | 487 Request Terminated [17.82s] 17.821674 |->| | ACK sip:1234567...@sbc.ops.lan [17.82s] 21.827427 |<-| | 487 Request Terminated [21.83s] 21.828124 |->| | ACK sip:1234567...@sbc.ops.lan [21.83s] Best regards, Alexander Mustafin mustafin.aleksa...@gmail.com signature.asc Description: Message signed with OpenPGP using GPGMail ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Opensips 1.11 drouting route_to_gw error
Hello.
I’m testing OpenSIPS 1.11. Now my version installed from daily repo:
opensips-1.11.0.20140420.d52cb7e-1.el6.x86_64
I’ve got error in drouting module with route_to_gw function. I would export
gateway attributes after calling this function, but variable contains digits -
not my parameters.
It looks like:
if ( route_to_gw("$avp(gw_id)", "$avp(gw_attrs)") ) {
xlog("L_INFO", "[DROUTING] Gateway [GW=$avp(gw_id)] with attrs
$avp(gw_attrs)»);
Apr 22 15:41:01 test aaa-radius[28744]: DBG:drouting:push_gw_for_usage: setting
GW id [GW_TEST] as avp
Apr 22 15:41:01 test aaa-radius[28744]: DBG:drouting:push_gw_for_usage: setting
GW attr [A=10;B=1;C=30,G=1;I=5] as avp
Apr 22 15:41:01 test aaa-radius[28744]: [DROUTING] Gateway [GW=GW_TEST] with
attrs 286816192
Apr 22 15:41:01 test aaa-radius[28744]: DBG:core:parse_params: Parsing params
for:[286816192]
Best regards,
Alexander Mustafin
mustafin.aleksa...@gmail.com
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[OpenSIPS-Users] OpenSIPS 1.11 error in tm
Hello! I’ve just upgrade 1.10 server to 1.11 and got next error: After receive "100 Trying" server generates «Cancel". Apr 22 11:02:05 reg1.sip3.net aaa-radius[1783]: DBG:core:parse_headers: flags= Apr 22 11:02:05 reg1.sip3.net aaa-radius[1783]: DBG:sipmsgops:has_body_f: content length is zero Apr 22 11:02:05 reg1.sip3.net aaa-radius[1783]: DBG:tm:t_should_relay_response: T_code=100, new_code=100 Apr 22 11:02:05 reg1.sip3.net aaa-radius[1783]: DBG:tm:relay_reply: branch=0, save=0, relay=-1 Apr 22 11:02:05 reg1.sip3.net aaa-radius[1783]: DBG:tm:reply_received: FR_INV_TIMER = 0 Apr 22 11:02:05 reg1.sip3.net aaa-radius[1783]: DBG:tm:set_timer: relative timeout is 0 Apr 22 11:02:05 reg1.sip3.net aaa-radius[1783]: DBG:tm:insert_timer_unsafe: [1]: 0x7fbed3aa4f58 (109) Apr 22 11:02:05 reg1.sip3.net aaa-radius[1783]: DBG:tm:t_unref: UNREF_UNSAFE: [0x7fbed3aa4d08] after is 0 Apr 22 11:02:05 reg1.sip3.net aaa-radius[1783]: DBG:core:destroy_avp_list: destroying list (nil) Apr 22 11:02:05 reg1.sip3.net aaa-radius[1783]: DBG:core:receive_msg: cleaning up Apr 22 11:02:06 reg1.sip3.net aaa-radius[1786]: DBG:tm:utimer_routine: timer routine:4,tl=0x7fbed3aa4f28 next=(nil), timeout=11000 Apr 22 11:02:06 reg1.sip3.net aaa-radius[1786]: DBG:tm:timer_routine: timer routine:1,tl=0x7fbed3aa4f58 next=(nil), timeout=109 Apr 22 11:02:06 reg1.sip3.net aaa-radius[1786]: DBG:tm:build_local: using FROM=;tag=da316d62#015#012>, TO=#015#012>, CSEQ_N= Apr 22 11:02:06 reg1.sip3.net aaa-radius[1786]: DBG:tm:cancel_branch: sending cancel... Apr 22 11:02:06 reg1.sip3.net aaa-radius[1786]: DBG:tm:set_timer: relative timeout is 50 Apr 22 11:02:06 reg1.sip3.net aaa-radius[1786]: DBG:tm:insert_timer_unsafe: [4]: 0x7fbed3aa4ff0 (11100) Apr 22 11:02:06 reg1.sip3.net aaa-radius[1786]: DBG:tm:insert_timer_unsafe: [0]: 0x7fbed3aa5020 (130) Apr 22 11:02:06 reg1.sip3.net aaa-radius[1786]: DBG:tm:final_response_handler: Cancel sent out, sending 408 (0x7fbed3aa4d08) Apr 22 11:02:06 reg1.sip3.net aaa-radius[1786]: DBG:tm:t_should_relay_response: T_code=100, new_code=408 Apr 22 11:02:06 reg1.sip3.net aaa-radius[1786]: DBG:tm:t_pick_branch: picked branch 0, code 408 (prio=800) Apr 22 11:02:06 reg1.sip3.net aaa-radius[1786]: DBG:tm:is_3263_failure: dns-failover test: branch=0, last_recv=408, flags=2 Apr 22 11:02:06 reg1.sip3.net aaa-radius[1786]: DBG:tm:run_trans_callbacks: trans=0x7fbed3aa4d08, callback type 32, id 1 entered Apr 22 11:02:06 reg1.sip3.net aaa-radius[1786]: DBG:db_mysql:has_stmt_ctx: ctx found for missed_calls Maybe, some parameters of configuration are wrong, but they are works successfully in 1.10. Best regards, Alexander Mustafin mustafin.aleksa...@gmail.com signature.asc Description: Message signed with OpenPGP using GPGMail ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] uac_auth() segfault
Hello!
I can’t find commit in git for this problem.
Maybe, my problem related:
opensips[4065]: segfault at 0 ip (null) sp 7fff64ade248 error 14 in
opensips[40+14c000]
Best regards,
Alexander Mustafin
mustafin.aleksa...@gmail.com
27 марта 2014 г., в 2:08, Justin Zondagh написал(а):
> Thanks Bogdan, glad we could fix it.
>
> Regards,
> Justin
>
>
> Justin Zondagh
> zond...@gmail.com
>
> Cape Town | South Africa
> skype: jrzondagh
> m: +27 72 598 4887 | f: +27 86 546 1405
> uk: +44 20 328 99610
>
>
> On Wed, Mar 26, 2014 at 7:57 PM, Bogdan-Andrei Iancu
> wrote:
> Hello Justin,
>
> Thanks for all your help - I managed to find and fix the bug - please update
> from GIT and try again.
>
> Best regards,
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
> On 25.03.2014 21:31, Justin Zondagh wrote:
>> Hi Bogdan,
>>
>> I placed the LM_DBG as follows
>>
>> if ( is_script_func_used("uac_auth", -1) ) {
>> /* load the UAC_AUTH API as uac_auth() is invoked from
>> script */
>> if(load_uac_auth_api(&uac_auth_api)<0){
>> LM_ERR("can't load UAC_AUTH API, needed for
>> uac_auth()\n");
>> goto error;
>> }
>>
>> LM_DBG("Loaded uac_auth api as found in script");
>> }
>>
>> But nothing appears in the log on init.
>>
>>
>>
>> I'm using the uac_auth() in a failure route
>>
>>
>>
>> failure_route[ip_auth_fail]
>> {
>>
>> if (t_check_status("401")) {
>> xlog("L_INFO","[$ci] Got 401 from Proxy\n");
>>
>> avp_db_query("select password from registrant where username
>> = '$(avp(authuser){s.escape.common})'","$avp(authpass)");
>>
>> $avp(authrealm) = "";
>>
>> $avp(www) = $(hdr(WWW-Authenticate));
>> avp_subst("$avp(www)", "/Digest\s//");
>>
>> #Get the realm from the www-auth header
>> $var(numkvp) = $(avp(www){csv.count});
>> $var(i) = 0;
>> while($var(i) < $var(numkvp)) {
>>
>> $var(temp) = $(avp(www){s.select,$var(i),,});
>>
>> if ($var(temp) =~ "realm.*") {
>> $avp(authrealm) = $var(temp);
>>
>> avp_subst("$avp(authrealm)","/(realm=\")(.*)(\")/\2/");
>> #$avp(authrealm) := "asterisk";
>> }
>>
>> $var(i) = $var(i) + 1;
>> }
>>
>> xlog("L_INFO","[$ci] authrealm is [$avp(authrealm)],
>> authuser is [$avp(authuser)], authpass is [$avp(authpass)]\n");
>>
>> #No need for loop prevention as Proxy sends 183 early media
>> recording saying password is wrong, then sends 603
>>
>> if (uac_auth()) {
>> xlog("L_INFO","[$ci] Built auth, sending back to
>> Proxy\n");
>>
>> t_on_failure("ip_auth_fail");
>>
>> # the $du should really be uri as specified in
>> Record-Route in 401, but using reply's source IP for now
>> $du = "sip:" + $(si) + ":5060";
>>
>> xlog("L_INFO", "[$ci] Sending request with Auth
>> header to [$du]\n");
>> t_relay();
>>
>> }
>> }
>>
>> if (t_was_cancelled()) {
>> exit;
>> }
>>
>> }
>>
>>
>>
>>
>> Justin Zondagh
>> zond...@gmail.com
>>
>> Cape Town | South Africa
>> skype: jrzondagh
>> m: +27 72 598 4887 | f: +27 86 546 1405
>> uk: +44 20 328 99610
>>
>>
>> On Tue, Mar 25, 2014 at 7:28 PM, Bogdan-Andrei Iancu
>> wrote:
>> Hello Justin,
>>
>> In the 1.10 code, in the UAC module, in the mod_init function, the binding
>> to UAC AUTH module is to be done. See line 171 in modules/uac/uac.c
>>
>>
>> if
[OpenSIPS-Users] Problem with {s.int] transformation
Hello.
I’ve got the header in message: P-Asserted-Identity: «77"
And I need to parse «name» part for future used, but without quotes.
I’m trying:
$avp(name) = $(hdr(P-Asserted-Identity){nameaddr.name}{s.int});
but $avp(name) is null after this transformation.
How to delete quotes from this field?
Best regards,
Alexander Mustafin
mustafin.aleksa...@gmail.com
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[OpenSIPS-Users] Generate Billing-Correlation-ID
Hello. I would to use P-DCS-Billing-Info header with parameters (RFC 5503) for billing purposes. Especially, I need to generate Billing-Correlation-ID. How to implement this in script? Or, I should do this via exec or perl, or other external script? Best regards, Alexander Mustafin mustafin.aleksa...@gmail.com signature.asc Description: Message signed with OpenPGP using GPGMail ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Prevent re-INVITE to T.38
Hi, Jeff. Maybe stream_exists(regexp) in sipmsgops module will be useful for you. Best regards, Alexander Mustafin mustafin.aleksa...@gmail.com 11 марта 2014 г., в 20:07, Jeff Pyle написал(а): > Hello, > > Is there anything I can do at the proxy level to prevent a dialog from > reinviting to to T.38? I think I could detect the T.38 attributes easily > enough and respond with a 488, although I'm concerned the CSeq values would > be out of sequence for the next transaction that did make it through the > proxy to the far end. That could cause a problem, no? > > Is this something that requires a B2BUA? Is it possible from within the > OpenSIPS B2B modules to do SDP inspection of any sort? > > > - Jeff > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users signature.asc Description: Message signed with OpenPGP using GPGMail ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] radius_send_acct and dialog
Hello!
I’ve got opensips-1.10.0-1.el6.x86_64 and work with radius-acct.
I’ve the next problem:
If caller ends the call - all works properly, and
radius_send_acct("acct_answer»); send Radius-Acct message. But if callee ends
the call - radius_send_acct("acct_answer») does nothing.
In log:
=
Mar 7 10:57:41 sbc aaa-radius[19583]: DBG:dialog:destroy_dlg: dlg expired or
not in list - dlg 0x7f2cb74b7120 [2109:914270467] with clid
‘458d03c9440208c0120e85ef656388572e7000a@72.72.72.72' and tags
'4-45026-3afca21-9796f6d-3afca21' ‘H6Xg8B08Hr57F'
But dialog ends successfully.
Please, help me to solve this problem and explain how aaa_radius relates to
dialog.
Best regards,
Alexander Mustafin
mustafin.aleksa...@gmail.com
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Re: [OpenSIPS-Users] drouting get_group_id behavior
Hi!
Bacause drouting do not accept .* in dr_rules - you may use dialplan module for
this. Just catch domain name with regex and $avp(dest) will store name of rule
for drouting, as example.
С уважением,
Александр Мустафин
mustafin.aleksa...@gmail.com
26 февр. 2014 г., в 20:45, Maciej Bylica написал(а):
> Hello,
>
> Thanks for reply.
> Yeah i did it by asking db for..
> avp_db_query("SELECT groupid FROM dr_groups WHERE domain =
> '$fd'","$avp(i:600)");
> and then using exactly the same avp for do_routing.
> It works, but i am still wondering how to match domain different way
> (do_routing())
>
> Thanks.
>
>
> 2014-02-25 18:52 GMT+01:00 :
> Hello,
>
> > I have the same problem on 1.9 rel.
> > | id | username | domain | groupid | description |
> > | 4 | .* | 10.10.10.5 | 0 | TEST
>
> If you don't need to match on username why not pass directly the groupid
> to `do_routing` ?
>
> do_routing("0");
>
> If you need to dynamically map between a domain and a groupid, use e.g.
>
> do_routing("$avp(10)");
>
> and an AVP table which maps from domains to groupid.
> S.
>
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[OpenSIPS-Users] Problem with aaa_radius
Hello.
I’m trying to setup new server and just copy opensips.cfg from other server.
This configuration tested and works successfully.
But now, I’ve silent problem.
OpenSIPS from yum-repo:
version: opensips 1.10.0beta-tls (x86_64/linux)
flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, DISABLE_NAGLE, USE_MCAST,
SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
@(#) $Id$
main.c compiled on 03:46:24 Aug 6 2013 with gcc 4.4.7
radiusclient-ng installed and config copied too.
When I insert radius_send_auth("radius_auth","radius_auth_resp");
in config - OpenSIPS can’t start and I have no errors in syslog.
Sets are defined in module configuration.
Syslog:
===
Feb 24 02:34:50 reg opensips: DBG:core:yyparse: loading module
/usr/lib64/opensips/modules/aaa_radius.so
Feb 24 02:34:50 reg opensips: DBG:core:set_mod_param_regex: aaa_radius matches
module aaa_radius
Feb 24 02:34:50 reg opensips: DBG:core:set_mod_param_regex: found
in module aaa_radius [/usr/lib64/opensips/modules/]
Feb 24 02:34:50 reg opensips: DBG:core:set_mod_param_regex: aaa_radius matches
module aaa_radius
Feb 24 02:34:50 reg opensips: DBG:core:set_mod_param_regex: found in
module aaa_radius [/usr/lib64/opensips/modules/]
Feb 24 02:34:50 reg opensips: DBG:core:__search_avp_map: looking for [username]
avp - found -1
Feb 24 02:34:50 reg opensips: DBG:core:new_avp_alias: added alias username with
id 1
Feb 24 02:34:50 reg opensips: DBG:core:__search_avp_map: looking for [username]
avp - found 1
Feb 24 02:34:50 reg opensips: DBG:core:__search_avp_map: looking for [rpid] avp
- found -1
Feb 24 02:34:50 reg opensips: DBG:core:new_avp_alias: added alias rpid with id 2
Feb 24 02:34:50 reg opensips: DBG:core:set_mod_param_regex: aaa_radius matches
module aaa_radius
Feb 24 02:34:50 reg opensips: DBG:core:set_mod_param_regex: found in
module aaa_radius [/usr/lib64/opensips/modules/]
Feb 24 02:34:50 reg opensips: DBG:core:__search_avp_map: looking for
[credit_time] avp - found -1
Feb 24 02:34:50 reg opensips: DBG:core:new_avp_alias: added alias credit_time
with id 3
Feb 24 02:34:50 reg opensips: DBG:core:__search_avp_map: looking for
[return_code] avp - found -1
Feb 24 02:34:50 reg opensips: DBG:core:new_avp_alias: added alias return_code
with id 4
Best regards,
Alexander Mustafin
mustafin.aleksa...@gmail.com
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Re: [OpenSIPS-Users] Save info about codecs
Thanks for help! WIlmar: This method works, but cumbersome like mine. Ovidiu: Very good module. Can I store info implemented by this module to $avp’s? Best regards, Alexander Mustafin mustafin.aleksa...@gmail.com ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Save info about codecs
Hello! I need to save info about all codecs in INVITE and 200 ОК messages(presence and their position). I was tried to use some functions in sipmsgops module, like codec_exists, but manual comparison with some list is so annoying :) Can you hint me more simple solution for it? Best regards, Alexander Mustafin mustafin.aleksa...@gmail.com ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Recommended modules
Hi, David. drouting module - http://www.opensips.org/html/docs/modules/1.10.x/drouting.html Best regards, Alexander Mustafin mustafin.aleksa...@gmail.com 11 дек. 2013 г., в 3:03, Hiers, David написал(а): > Hi, > I'm planning to run openSIPS as a redirect server in the core. > > I'm looking for least cost routing, the ability to detect dead SIP servers, > as well as pretty beefy stats and troubleshooting information. > > What modules would you run on your openSIPS server in this role? > > Thanks! > > > > David > > > > This message and any attachments are intended only for the use of the > addressee and may contain information that is privileged and confidential. If > the reader of the message is not the intended recipient or an authorized > representative of the intended recipient, you are hereby notified that any > dissemination of this communication is strictly prohibited. If you have > received this communication in error, please notify us immediately by e-mail > and delete the message and any attachments from your system. > > ___ > 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] siptrace issue
Hello!
I need to save initial INVITE - but in-dialog messages only in sip_trace table.
Is it possible?
Best regards,
Alexander Mustafin
mustafin.aleksa...@gmail.com
06 дек. 2013 г., в 15:25, Bogdan-Andrei Iancu написал(а):
> Hi Mickael,
>
> I see you commented the "trace_flag" parameter in your modparm section -
> please enable it back. Even if you do not use it from script, it is
> internally used by the module.
>
> Best regards,
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
> On 06.12.2013 09:56, Mickael Hubert wrote:
>> Hi Bigdan,
>> thanks for your answer.
>>
>> I have modify my opensips code:
>>
>> my conf:
>> setdebug(6);
>> trace_dialog();
>> setdebug();
>>
>>
>> see below my opensips.log
>>
>> Dec 6 08:38:22 core-sip-2 /usr/local/sbin/opensips[2054]:
>> DBG:dialog:build_new_dlg: new dialog 0x806aee888
>> (c=1104374-3595304320-30853@asbc1b.hexavoip.local,f=sip:+33310368002@10.84.8.10,t=sip:+33XXX@10.84.8.21;user=phone,ft=3595304320-30953)
>> on hash 558
>> Dec 6 08:38:22 core-sip-2 /usr/local/sbin/opensips[2054]:
>> DBG:core:parse_headers: flags=
>> Dec 6 08:38:22 core-sip-2 /usr/local/sbin/opensips[2054]:
>> DBG:dialog:init_leg_info: route_set , contact sip:+33XX@10.84.8.10:5060,
>> cseq 1 and bind_addr udp:10.84.8.21:5060
>> Dec 6 08:38:22 core-sip-2 /usr/local/sbin/opensips[2054]:
>> DBG:dialog:dlg_add_leg_info: set leg 0 for 0x806aee888:
>> tag=<3595304320-30953> rcseq=<0>
>> Dec 6 08:38:22 core-sip-2 /usr/local/sbin/opensips[2054]:
>> DBG:dialog:link_dlg: ref dlg 0x806aee888 with 4 -> 4 in h_entry 0x806adf248
>> - 558
>> Dec 6 08:38:22 core-sip-2 /usr/local/sbin/opensips[2054]:
>> DBG:rr:add_rr_param: adding (;xyz=e22.25982815) 0x80146c5e8
>> Dec 6 08:38:22 core-sip-2 /usr/local/sbin/opensips[2054]:
>> DBG:siptrace:sip_trace: nothing to trace...
>>
>> I don't understand this result ;)
>>
>> Have you an idea ?
>>
>> 2013/12/5 Bogdan-Andrei Iancu
>> Hello Mickael,
>>
>> Try to place the trace_dialog() between a set_debug(6) and set_debug(), to
>> see what is going on inside the function. Maybe the debug logs will give you
>> an idea.
>>
>> Best regards,
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developer
>> http://www.opensips-solutions.com
>> On 05.12.2013 09:47, Mickael Hubert wrote:
>>> Hi list,
>>> I have an issue with my siptrace module.
>>> see below my extract of opensips.cfg.
>>>
>>> I want duplicate dialogs's messages to the other server (10.84.8.201), no
>>> DB.
>>> But, it doesn't work... no sip message in my wireshark trace.
>>>
>>> Have you an idea ?
>>>
>>> Thanks in advance
>>>
>>>
>>> # - module siptrace.so ---
>>> loadmodule "siptrace.so"
>>> # - siptrace params -
>>> modparam("siptrace", "db_url", "")
>>> modparam("siptrace", "duplicate_uri","sip:10.84.8.201:9060")
>>> modparam("siptrace", "duplicate_with_hep", 1)
>>> modparam("siptrace", "enable_ack_trace", 1)
>>> modparam("siptrace", "hep_version", 2)
>>> modparam("siptrace", "trace_to_database", 0)
>>> #modparam("siptrace", "trace_flag", 22)
>>> modparam("siptrace", "hep_capture_id", 2)
>>> modparam("siptrace", "trace_on", 1)
>>>
>>> .
>>>
>>> in my code:
>>>
>>> if (is_method("INVITE"))
>>> {
>>> trace_dialog();
>>>
>>> ..
>>>
>>> --
>>> Cordialement
>>>
>>> HUBERT Mickaël
>>> Ingénieur VOIP - Hexanet
>>>
>>>
>>>
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>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
>>
>> --
>> Cordialement
>>
>> HUBERT Mickaël
>> Ingénieur VOIP - Hexanet
>>
>>
>
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Re: [OpenSIPS-Users] ACC and failed calls
I’m puzzled :)
I know about modparam("acc", "db_table_missed_calls", "acc»), but this does not
change the behavior of the string generation.
1) I do a call to GW1 (do_routing)
2) GW1 return 503 Error
3) I catch an error and do use_next_gw() - call send to GW2
4)GW2 is accept the call
5)Call and dialog successfully ended
The ACC module generates next rows:
1) INVITE with sip_code=200 (columns are filled by avp) to GW2
2) INVITE with sip_code=200 (almost all columns are empty) to GW1
3) BYE with sip_code=200 to GW2
4) INVITE with sip_code=503 (columns are filled by avp) in missed_calls to GW1.
I’m confused only this raw, with empty columns AND it’s sip_code and think that
incorrect behavior.
Best regards,
Alexander Mustafin
mustafin.aleksa...@gmail.com
03 дек. 2013 г., в 14:24, Răzvan Crainea написал(а):
> Hi, Alexander!
>
> Indeed, the failed legs will be stored in the missed_call table. A workaround
> would be to change the table by setting the following parameter:
>
> modparam("acc", "db_table_missed_calls", "acc")
>
> Best regards,
>
> Razvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solutions.com
>
> On 12/03/2013 07:30 AM, Alexander Mustafin wrote:
>> OK, Răzvan, maybe my avp’s are null at this moment - much more
>> important, that really sip code not presented in this row. Completely
>> filled row for this leg creates in missed_call table. I.e. 2 rows
>> creates for one leg with error - one in acc table and one in missed_calls.
>>
>>
>> Best regards,
>> Alexander Mustafin
>> mustafin.aleksa...@gmail.com <mailto:mustafin.aleksa...@gmail.com>
>>
>>
>>
>>
>> 02 дек. 2013 г., в 15:07, Răzvan Crainea > <mailto:raz...@opensips.org>> написал(а):
>>
>>> Hi, Alexander!
>>>
>>> Are you sure you are populating all the multi_leg_info AVPs for each
>>> leg? In your case, you have two legs, therefore you should populate
>>> each variable two times - once for the initial leg and second for the
>>> redirect.
>>>
>>> Best regards,
>>>
>>> Razvan Crainea
>>> OpenSIPS Core Developer
>>> http://www.opensips-solutions.com <http://www.opensips-solutions.com/>
>>>
>>> On 11/30/2013 05:07 AM, Alexander Mustafin wrote:
>>>> Hi, Răzvan.
>>>>
>>>> Thank you for this hint!
>>>>
>>>> I set multi-leg acc support now, and it’s almost works )))
>>>>
>>>> 1) If first GW drop my call and return error - ACC generate row in table
>>>> for this leg, BUT sip_code is 200 (not error code)
>>>> 2) I set modparam("acc", "multi_leg_info»,…) with avp I would to store
>>>> in database, but almost all columns are empty for unsuccessfull leg -
>>>> for other legs they are present.
>>>>
>>>>
>>>> Best regards,
>>>> Alexander Mustafin
>>>> mustafin.aleksa...@gmail.com
>>>> <mailto:mustafin.aleksa...@gmail.com><mailto:mustafin.aleksa...@gmail.com>
>>>>
>>>>
>>>>
>>>>
>>>> 29 нояб. 2013 г., в 14:44, Răzvan Crainea >>> <mailto:raz...@opensips.org>
>>>> <mailto:raz...@opensips.org>> написал(а):
>>>>
>>>>> Hi, Alexander!
>>>>>
>>>>> If you want to have multiple rows for each leg, then you should use
>>>>> multi-leg acc support[1]. Note that you should not use the CDR flag,
>>>>> since you are doing old two-steps accounting.
>>>>>
>>>>> [1] http://www.opensips.org/html/docs/modules/1.8.x/acc#multi-call-legs
>>>>>
>>>>> Best regards,
>>>>>
>>>>> Razvan Crainea
>>>>> OpenSIPS Core Developer
>>>>> http://www.opensips-solutions.com
>>>>>
>>>>> On 11/28/2013 02:36 PM, Alexander Mustafin wrote:
>>>>>> If first gateway in drouting return error - I need a CDR with sip code
>>>>>> of error. Then, if next gw is setup the call - I need a CDR, after BYE
>>>>>> message.
>>>>>>
>>>>>> First attempt - generate one row INVITE in the table (with error code),
>>>>>> and second attempt generate two rows - INVITE and BYE
>>>>>>
>>>>>> I believe that it should work like that!
>>>>>>
>>>>>> Best regards,
>>>>>> Alexander Mustafin
>
Re: [OpenSIPS-Users] ACC and failed calls
OK, Răzvan, maybe my avp’s are null at this moment - much more important, that
really sip code not presented in this row. Completely filled row for this leg
creates in missed_call table. I.e. 2 rows creates for one leg with error - one
in acc table and one in missed_calls.
Best regards,
Alexander Mustafin
mustafin.aleksa...@gmail.com
02 дек. 2013 г., в 15:07, Răzvan Crainea написал(а):
> Hi, Alexander!
>
> Are you sure you are populating all the multi_leg_info AVPs for each
> leg? In your case, you have two legs, therefore you should populate each
> variable two times - once for the initial leg and second for the redirect.
>
> Best regards,
>
> Razvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solutions.com
>
> On 11/30/2013 05:07 AM, Alexander Mustafin wrote:
>> Hi, Răzvan.
>>
>> Thank you for this hint!
>>
>> I set multi-leg acc support now, and it’s almost works )))
>>
>> 1) If first GW drop my call and return error - ACC generate row in table
>> for this leg, BUT sip_code is 200 (not error code)
>> 2) I set modparam("acc", "multi_leg_info»,…) with avp I would to store
>> in database, but almost all columns are empty for unsuccessfull leg -
>> for other legs they are present.
>>
>>
>> Best regards,
>> Alexander Mustafin
>> mustafin.aleksa...@gmail.com <mailto:mustafin.aleksa...@gmail.com>
>>
>>
>>
>>
>> 29 нояб. 2013 г., в 14:44, Răzvan Crainea > <mailto:raz...@opensips.org>> написал(а):
>>
>>> Hi, Alexander!
>>>
>>> If you want to have multiple rows for each leg, then you should use
>>> multi-leg acc support[1]. Note that you should not use the CDR flag,
>>> since you are doing old two-steps accounting.
>>>
>>> [1] http://www.opensips.org/html/docs/modules/1.8.x/acc#multi-call-legs
>>>
>>> Best regards,
>>>
>>> Razvan Crainea
>>> OpenSIPS Core Developer
>>> http://www.opensips-solutions.com
>>>
>>> On 11/28/2013 02:36 PM, Alexander Mustafin wrote:
>>>> If first gateway in drouting return error - I need a CDR with sip code
>>>> of error. Then, if next gw is setup the call - I need a CDR, after BYE
>>>> message.
>>>>
>>>> First attempt - generate one row INVITE in the table (with error code),
>>>> and second attempt generate two rows - INVITE and BYE
>>>>
>>>> I believe that it should work like that!
>>>>
>>>> Best regards,
>>>> Alexander Mustafin
>>>> mustafin.aleksa...@gmail.com <mailto:mustafin.aleksa...@gmail.com>
>>>>
>>>>
>>>>
>>>>
>>>> 28 нояб. 2013 г., в 15:42, Alexander Mustafin
>>>> mailto:mustafin.aleksa...@gmail.com>>
>>>> написал(а):
>>>>
>>>>> Hi, Razvan!
>>>>>
>>>>> I’m tried set ACC_FAILED in places, where call may be failed. In
>>>>> request route too.
>>>>>
>>>>> Best regards,
>>>>> Alexander Mustafin
>>>>> mustafin.aleksa...@gmail.com <mailto:mustafin.aleksa...@gmail.com>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> 28 нояб. 2013 г., в 15:35, Răzvan Crainea >>>> <mailto:raz...@opensips.org>> написал(а):
>>>>>
>>>>>> Hi, Alexander!
>>>>>>
>>>>>> Have you tried setting the ACC_FAILED flag in the request route?
>>>>>>
>>>>>> Best regards,
>>>>>>
>>>>>> Razvan Crainea
>>>>>> OpenSIPS Core Developer
>>>>>> http://www.opensips-solutions.com <http://www.opensips-solutions.com/>
>>>>>>
>>>>>> On 11/28/2013 10:35 AM, Alexander Mustafin wrote:
>>>>>>> Hello!
>>>>>>>
>>>>>>> I need to store all CDRs for all calls, but some failed calls are
>>>>>>> not to
>>>>>>> handled by ACC.
>>>>>>>
>>>>>>> modparam("acc", "failed_transaction_flag", "ACC_FAILED»)
>>>>>>>
>>>>>>> failure_route[MISSED_CALL] {
>>>>>>> if (t_check_status("[4|5][0-9][0-9]")) {
>>>>>>> setflag(ACC_FAILED);
>>>>>>> }
>&g
Re: [OpenSIPS-Users] Error with t_check_status
Hi!
Can I use lookahead and lookbehind statements in this function (e.g. I would
cath all errors, exclude 404, 486) ?
Best regards,
Alexander Mustafin
mustafin.aleksa...@gmail.com
28 нояб. 2013 г., в 12:16, Alexander Mustafin
написал(а):
> Hello!
>
> I was write the next rule in script:
> if (t_check_status("(?!486)(?!404)[4|5][0-9][0-9]")) {
>
> And OpenSIPS wasn’t restart succesfully.
>
> opensips -C /etc/opensips/opensips.cfg check was good, and no errors in debug.
>
> With if (t_check_status("[4|5][0-9][0-9]")) works fine.
>
>
> version: opensips 1.10.0-tls (x86_64/linux) daily
>
>
> Best regards,
> Alexander Mustafin
> mustafin.aleksa...@gmail.com
>
>
>
>
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Re: [OpenSIPS-Users] ACC and failed calls
Hi, Răzvan.
Thank you for this hint!
I set multi-leg acc support now, and it’s almost works )))
1) If first GW drop my call and return error - ACC generate row in table for
this leg, BUT sip_code is 200 (not error code)
2) I set modparam("acc", "multi_leg_info»,…) with avp I would to store in
database, but almost all columns are empty for unsuccessfull leg - for other
legs they are present.
Best regards,
Alexander Mustafin
mustafin.aleksa...@gmail.com
29 нояб. 2013 г., в 14:44, Răzvan Crainea написал(а):
> Hi, Alexander!
>
> If you want to have multiple rows for each leg, then you should use multi-leg
> acc support[1]. Note that you should not use the CDR flag, since you are
> doing old two-steps accounting.
>
> [1] http://www.opensips.org/html/docs/modules/1.8.x/acc#multi-call-legs
>
> Best regards,
>
> Razvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solutions.com
>
> On 11/28/2013 02:36 PM, Alexander Mustafin wrote:
>> If first gateway in drouting return error - I need a CDR with sip code
>> of error. Then, if next gw is setup the call - I need a CDR, after BYE
>> message.
>>
>> First attempt - generate one row INVITE in the table (with error code),
>> and second attempt generate two rows - INVITE and BYE
>>
>> I believe that it should work like that!
>>
>> Best regards,
>> Alexander Mustafin
>> mustafin.aleksa...@gmail.com <mailto:mustafin.aleksa...@gmail.com>
>>
>>
>>
>>
>> 28 нояб. 2013 г., в 15:42, Alexander Mustafin
>> mailto:mustafin.aleksa...@gmail.com>>
>> написал(а):
>>
>>> Hi, Razvan!
>>>
>>> I’m tried set ACC_FAILED in places, where call may be failed. In
>>> request route too.
>>>
>>> Best regards,
>>> Alexander Mustafin
>>> mustafin.aleksa...@gmail.com <mailto:mustafin.aleksa...@gmail.com>
>>>
>>>
>>>
>>>
>>> 28 нояб. 2013 г., в 15:35, Răzvan Crainea >> <mailto:raz...@opensips.org>> написал(а):
>>>
>>>> Hi, Alexander!
>>>>
>>>> Have you tried setting the ACC_FAILED flag in the request route?
>>>>
>>>> Best regards,
>>>>
>>>> Razvan Crainea
>>>> OpenSIPS Core Developer
>>>> http://www.opensips-solutions.com <http://www.opensips-solutions.com/>
>>>>
>>>> On 11/28/2013 10:35 AM, Alexander Mustafin wrote:
>>>>> Hello!
>>>>>
>>>>> I need to store all CDRs for all calls, but some failed calls are not to
>>>>> handled by ACC.
>>>>>
>>>>> modparam("acc", "failed_transaction_flag", "ACC_FAILED»)
>>>>>
>>>>> failure_route[MISSED_CALL] {
>>>>>if (t_check_status("[4|5][0-9][0-9]")) {
>>>>> setflag(ACC_FAILED);
>>>>> }
>>>>> }
>>>>>
>>>>> But call which rejected with 503 (example) error is missed in acc table.
>>>>>
>>>>> Best regards,
>>>>> Alexander Mustafin
>>>>> mustafin.aleksa...@gmail.com
>>>>> <mailto:mustafin.aleksa...@gmail.com><mailto:mustafin.aleksa...@gmail.com>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> ___
>>>>> Users mailing list
>>>>> Users@lists.opensips.org <mailto:Users@lists.opensips.org>
>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>
>>>>
>>>> ___
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>>>> Users@lists.opensips.org <mailto:Users@lists.opensips.org>
>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>
>>
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
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Re: [OpenSIPS-Users] ACC and failed calls
If first gateway in drouting return error - I need a CDR with sip code of
error. Then, if next gw is setup the call - I need a CDR, after BYE message.
First attempt - generate one row INVITE in the table (with error code), and
second attempt generate two rows - INVITE and BYE
I believe that it should work like that!
Best regards,
Alexander Mustafin
mustafin.aleksa...@gmail.com
28 нояб. 2013 г., в 15:42, Alexander Mustafin
написал(а):
> Hi, Razvan!
>
> I’m tried set ACC_FAILED in places, where call may be failed. In request
> route too.
>
> Best regards,
> Alexander Mustafin
> mustafin.aleksa...@gmail.com
>
>
>
>
> 28 нояб. 2013 г., в 15:35, Răzvan Crainea написал(а):
>
>> Hi, Alexander!
>>
>> Have you tried setting the ACC_FAILED flag in the request route?
>>
>> Best regards,
>>
>> Razvan Crainea
>> OpenSIPS Core Developer
>> http://www.opensips-solutions.com
>>
>> On 11/28/2013 10:35 AM, Alexander Mustafin wrote:
>>> Hello!
>>>
>>> I need to store all CDRs for all calls, but some failed calls are not to
>>> handled by ACC.
>>>
>>> modparam("acc", "failed_transaction_flag", "ACC_FAILED»)
>>>
>>> failure_route[MISSED_CALL] {
>>> if (t_check_status("[4|5][0-9][0-9]")) {
>>> setflag(ACC_FAILED);
>>> }
>>> }
>>>
>>> But call which rejected with 503 (example) error is missed in acc table.
>>>
>>> Best regards,
>>> Alexander Mustafin
>>> mustafin.aleksa...@gmail.com <mailto:mustafin.aleksa...@gmail.com>
>>>
>>>
>>>
>>>
>>>
>>>
>>> ___
>>> Users mailing list
>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>
>> ___
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>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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Re: [OpenSIPS-Users] ACC and failed calls
Hi, Razvan!
I’m tried set ACC_FAILED in places, where call may be failed. In request route
too.
Best regards,
Alexander Mustafin
mustafin.aleksa...@gmail.com
28 нояб. 2013 г., в 15:35, Răzvan Crainea написал(а):
> Hi, Alexander!
>
> Have you tried setting the ACC_FAILED flag in the request route?
>
> Best regards,
>
> Razvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solutions.com
>
> On 11/28/2013 10:35 AM, Alexander Mustafin wrote:
>> Hello!
>>
>> I need to store all CDRs for all calls, but some failed calls are not to
>> handled by ACC.
>>
>> modparam("acc", "failed_transaction_flag", "ACC_FAILED»)
>>
>> failure_route[MISSED_CALL] {
>> if (t_check_status("[4|5][0-9][0-9]")) {
>> setflag(ACC_FAILED);
>> }
>> }
>>
>> But call which rejected with 503 (example) error is missed in acc table.
>>
>> Best regards,
>> Alexander Mustafin
>> mustafin.aleksa...@gmail.com <mailto:mustafin.aleksa...@gmail.com>
>>
>>
>>
>>
>>
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
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[OpenSIPS-Users] ACC and failed calls
Hello!
I need to store all CDRs for all calls, but some failed calls are not to
handled by ACC.
modparam("acc", "failed_transaction_flag", "ACC_FAILED»)
failure_route[MISSED_CALL] {
if (t_check_status("[4|5][0-9][0-9]")) {
setflag(ACC_FAILED);
}
}
But call which rejected with 503 (example) error is missed in acc table.
Best regards,
Alexander Mustafin
mustafin.aleksa...@gmail.com
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[OpenSIPS-Users] Error with t_check_status
Hello!
I was write the next rule in script:
if (t_check_status("(?!486)(?!404)[4|5][0-9][0-9]")) {
And OpenSIPS wasn’t restart succesfully.
opensips -C /etc/opensips/opensips.cfg check was good, and no errors in debug.
With if (t_check_status("[4|5][0-9][0-9]")) works fine.
version: opensips 1.10.0-tls (x86_64/linux) daily
Best regards,
Alexander Mustafin
mustafin.aleksa...@gmail.com
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Re: [OpenSIPS-Users] Dialplan dp_translate error
Thanks, Liviu!
My bad - the rule is wrong. And in the debug I’ve seen error:
Nov 19 03:20:02 ops /usr/sbin/opensips[31841]: DBG:dialplan:build_rule:
Compiling ^\d{3,6}\%23\d*$ expression with flag: 0
Nov 19 03:20:02 ops /usr/sbin/opensips[31841]: DBG:dialplan:build_rule:
building subst rule
Nov 19 03:20:02 ops /usr/sbin/opensips[31841]: DBG:dialplan:build_rule:
references:0 , max:2
Nov 19 03:20:02 ops /usr/sbin/opensips[31841]: ERROR:dialplan:build_rule:
repl_exp uses a non existing subexpression
Nov 19 03:20:02 ops /usr/sbin/opensips[31841]: WARNING:dialplan:dp_load_db:
failed to build rule -> skipping
Nov 19 03:20:02 ops /usr/sbin/opensips[31841]: DBG:core:db_free_rows: freeing 3
rows
Previously, dp_reload don’t put anything in debug. And I don’t check log. :))
Best regards,
Alexander Mustafin
mustafin.aleksa...@gmail.com
18 нояб. 2013 г., в 20:47, Liviu Chircu написал(а):
> Hello again,
>
> On second thoughts, I can tell you're running a post 1.9 version of OpenSIPS.
> Now in debug mode you should see all the rules being dumped in the logs with
> every "reload" operation, either at startup or through MI.
>
> Could you check if the DPID: 3 rules actually appear listed in that very
> verbose dump?
>
> Best regards,
> Liviu Chircu
> OpenSIPS Developer
> http://www.opensips-solutions.com
> On 11/18/2013 01:54 PM, Alexander Mustafin wrote:
>> Hi.
>>
>> I’m using some translation rules for various purposes.
>> I’ve got dial plan rule with dpid=3, but when I try to do dp_translate(«3»…)
>> - I see next message in the log:
>>
>> Nov 18 11:38:28 ops /usr/sbin/opensips[31844]: DBG:dialplan:dp_get_ivalue:
>> searching 4
>> Nov 18 11:38:28 ops /usr/sbin/opensips[31844]: DBG:dialplan:dp_translate_f:
>> dpid is 3
>> Nov 18 11:38:28 ops /usr/sbin/opensips[31844]: DBG:dialplan:dp_get_svalue:
>> searching 19
>> Nov 18 11:38:28 ops /usr/sbin/opensips[31844]: DBG:dialplan:dp_translate_f:
>> input is 1001%23777
>> Nov 18 11:38:28 ops /usr/sbin/opensips[31844]: DBG:dialplan:dp_translate_f:
>> no information available for dpid 3
>>
>> Rule in database:
>> 3 3 0 1 ^\d{3,6}\%23\d*$ 0 ^\d{3,6}\%23\d*$ \2 0
>>
>> I was tried dp_reload and restart server - but unsuccessfully. If I use same
>> rule with dpid=1 - all works fine ((
>>
>> Best regards,
>> Alexander Mustafin
>> mustafin.aleksa...@gmail.com
>>
>>
>>
>>
>>
>>
>> ___
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>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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[OpenSIPS-Users] Dialplan dp_translate error
Hi.
I’m using some translation rules for various purposes.
I’ve got dial plan rule with dpid=3, but when I try to do dp_translate(«3»…) -
I see next message in the log:
Nov 18 11:38:28 ops /usr/sbin/opensips[31844]: DBG:dialplan:dp_get_ivalue:
searching 4
Nov 18 11:38:28 ops /usr/sbin/opensips[31844]: DBG:dialplan:dp_translate_f:
dpid is 3
Nov 18 11:38:28 ops /usr/sbin/opensips[31844]: DBG:dialplan:dp_get_svalue:
searching 19
Nov 18 11:38:28 ops /usr/sbin/opensips[31844]: DBG:dialplan:dp_translate_f:
input is 1001%23777
Nov 18 11:38:28 ops /usr/sbin/opensips[31844]: DBG:dialplan:dp_translate_f: no
information available for dpid 3
Rule in database:
3 3 0 1 ^\d{3,6}\%23\d*$0
^\d{3,6}\%23\d*$\2 0
I was tried dp_reload and restart server - but unsuccessfully. If I use same
rule with dpid=1 - all works fine ((
Best regards,
Alexander Mustafin
mustafin.aleksa...@gmail.com
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Re: [OpenSIPS-Users] Dynamic Routing for calls towards gateways
Hi, Ollie!
What are you really want to do with calls from another gateway? In practice,
these calls are dropping with error 404.
Or you may to do something in other route...
> if (is_method("INVITE")) {
>
> if(!is_from_gw()) {
>
> route(ROUTE_FOR_MISTERY_CALLS);
>
> }
}
Best regards,
Alexander Mustafin
mustafin.aleksa...@gmail.com
14 нояб. 2013 г., в 20:24, Ollie Potts написал(а):
> Hi all,
>
> I’ve noticed some very strange behaviour with our opensips config and can’t
> seem to figure out why it is happening.
>
> I use this method (brief) to determine when to use dynamic routing,
>
> if (is_method("INVITE")) {
>
> t_on_failure("missed_call");
>
> if(!is_from_gw()) {
>
> ### xlog("Call from downstream account");
> if(do_routing("0")){ }
>
> }
>
> }
>
> In this case, it uses drouting for any invites that do not originate from our
> gateways. However, if we receive a call from a gateway destined for a number
> that is not local, it then tries to send the call back to the gateway using
> dynamic routing, causing a loop. The invite messages loop 10 times and then
> the failure route is called.
>
> I would like it to call the failure route immediately if it cannot find the
> number locally, rather than send an invite back to the originating gateway.
>
> Is there any way I can do this?
>
> Thanks,
> Ollie
>
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Re: [OpenSIPS-Users] [OpenSIPS-Devel] [RELEASES] Planing OpenSIPS 1.11.0 major release
Hi, Bogdan! I want to request for $DLG_end_reason_q850 (or like this) in Dialog module, which implements Q.850 codes for dialog end. I think, this variable much more useful then $DLG_end_reason. I know - Q.850 (Q.931) codes are not 1-to-1 compatible with SIP-codes, but users need some simple codes, like 16, 17,18 etc. Best regards, Alexander Mustafin mustafin.aleksa...@gmail.com ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Store dialog value and fetch it
Hello!
I’ve tried many variants, but unsuccessfully.
After BYE message, dialog change state to [5] and fetch_dlg_val doesn’t export
any stored values.
Are the other ways to save dialog-related values after BYE message?
Best regards,
Alexander Mustafin
mustafin.aleksa...@gmail.com
09 нояб. 2013 г., в 15:33, Alexander Mustafin
написал(а):
> Hi, Răzvan!
>
> I’m using topology_hiding() function - and loose_route() will always false.
> match_dialog() in this route return TRUE and $DLG_status return 5 (dialog
> ended)
>
> route[BYE] {
> xlog("L_DEBUG", "~ Enter in [BYE] section ~");
> xlog(" DIALOG status [$DLG_status]"); #DEBUG
>
> $avp(radius_user) = $dlg_val("incoming_gate");
> $var(setup_time) = $dlg_val("start_time»);
> }
>
> Nov 9 09:23:54 ops /usr/sbin/opensips[26665]: ~ Enter in [BYE] section
> ~
> Nov 9 09:23:54 ops /usr/sbin/opensips[26665]: DIALOG status [5]
> Nov 9 09:23:54 ops /usr/sbin/opensips[26665]: DBG:dialog:fetch_dlg_value:
> looking for <"incoming_gate">
> Nov 9 09:23:54 ops /usr/sbin/opensips[26665]: DBG:dialog:fetch_dlg_value:
> var NOT found!
> Nov 9 09:23:54 ops /usr/sbin/opensips[26665]: ERROR:core:do_assign: no value
> in right expression
> Nov 9 09:23:54 ops /usr/sbin/opensips[26665]: ERROR:core:do_assign: error at
> line: 501
> Nov 9 09:23:54 ops /usr/sbin/opensips[26665]: DBG:dialog:fetch_dlg_value:
> looking for <"start_time">
> Nov 9 09:23:54 ops /usr/sbin/opensips[26665]: DBG:dialog:fetch_dlg_value:
> var NOT found!
> Nov 9 09:23:54 ops /usr/sbin/opensips[26665]: ERROR:core:do_assign: no value
> in right expression
> Nov 9 09:23:54 ops /usr/sbin/opensips[26665]: ERROR:core:do_assign: error at
> line: 502
>
>
> Best regards,
> Alexander Mustafin
> mustafin.aleksa...@gmail.com
>
>
>
>
> 08 нояб. 2013 г., в 17:03, Răzvan Crainea написал(а):
>
>> So basically you're saying that after the loose_route() call you can no
>> longer access the dialog variable? Are you sure you are calling the BYE
>> route only in this place?
>> You should check if you are really in a dialog context. Inside the BYE
>> route, just print the $DLG_status variable. If it is NULL, then
>> match_dialog() and loose_route() failed to match a dialog, therefore you
>> won't be able to retrieve the value.
>> Adding the match_dialog() call inside the BYE route only makes sense if the
>> dialog was not mached yet. So in your case, it makes sense only if you call
>> route(BYE); from a different part of script.
>>
>> You can't really extract internal dialog values, unless you explicitely save
>> them as values, or take them from the request.
>
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Re: [OpenSIPS-Users] Store dialog value and fetch it
Hi, Răzvan!
I’m using topology_hiding() function - and loose_route() will always false.
match_dialog() in this route return TRUE and $DLG_status return 5 (dialog ended)
route[BYE] {
xlog("L_DEBUG", "~ Enter in [BYE] section ~");
xlog(" DIALOG status [$DLG_status]"); #DEBUG
$avp(radius_user) = $dlg_val("incoming_gate");
$var(setup_time) = $dlg_val("start_time»);
}
Nov 9 09:23:54 ops /usr/sbin/opensips[26665]: ~ Enter in [BYE] section
~
Nov 9 09:23:54 ops /usr/sbin/opensips[26665]: DIALOG status [5]
Nov 9 09:23:54 ops /usr/sbin/opensips[26665]: DBG:dialog:fetch_dlg_value:
looking for <"incoming_gate">
Nov 9 09:23:54 ops /usr/sbin/opensips[26665]: DBG:dialog:fetch_dlg_value: var
NOT found!
Nov 9 09:23:54 ops /usr/sbin/opensips[26665]: ERROR:core:do_assign: no value
in right expression
Nov 9 09:23:54 ops /usr/sbin/opensips[26665]: ERROR:core:do_assign: error at
line: 501
Nov 9 09:23:54 ops /usr/sbin/opensips[26665]: DBG:dialog:fetch_dlg_value:
looking for <"start_time">
Nov 9 09:23:54 ops /usr/sbin/opensips[26665]: DBG:dialog:fetch_dlg_value: var
NOT found!
Nov 9 09:23:54 ops /usr/sbin/opensips[26665]: ERROR:core:do_assign: no value
in right expression
Nov 9 09:23:54 ops /usr/sbin/opensips[26665]: ERROR:core:do_assign: error at
line: 502
Best regards,
Alexander Mustafin
mustafin.aleksa...@gmail.com
08 нояб. 2013 г., в 17:03, Răzvan Crainea написал(а):
> So basically you're saying that after the loose_route() call you can no
> longer access the dialog variable? Are you sure you are calling the BYE route
> only in this place?
> You should check if you are really in a dialog context. Inside the BYE route,
> just print the $DLG_status variable. If it is NULL, then match_dialog() and
> loose_route() failed to match a dialog, therefore you won't be able to
> retrieve the value.
> Adding the match_dialog() call inside the BYE route only makes sense if the
> dialog was not mached yet. So in your case, it makes sense only if you call
> route(BYE); from a different part of script.
>
> You can't really extract internal dialog values, unless you explicitely save
> them as values, or take them from the request.
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Re: [OpenSIPS-Users] Store dialog value and fetch it
Hi, Răzvan!
Thanks for your reply.
I’m fetching values after match_dialog function call, but in the other route.
In this piece of script, values are fetching successfully, but in the
route[BYE] aren’t.
if(match_dialog()) {
xlog("Dialog matched for [$dlg_val(outgoing_gate) AND
$dlg_val(incoming_gate)] request");
loose_route();
if (is_method("BYE")) {
route(BYE);
}
Should I to call match_dialog() a second time in route[BYE] for fetching this
values?
P.S. Which a best way for extracting all values of dialog (mangled_from_uri,
mangled_to_uri, etc.)?
Best regards,
Alexander Mustafin
mustafin.aleksa...@gmail.com
08 нояб. 2013 г., в 14:58, Răzvan Crainea написал(а):
> Hi, Alexander!
>
> Are you fetching the value before the loose_route() function call? It should
> be available only after loose_route() or match_dialog() are executed on the
> BYE request.
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solutions.com
>
> On 11/08/2013 05:37 AM, Alexander Mustafin wrote:
>> Hi!
>>
>> I’m want pass to dialog additional values, such as incoming and outgoing
>> gate from drouting.
>>
>> I’m use store_dlg_value("incoming_gate»,"$avp(gw_id)") and this value
>> successfully pass to dialog (I can see it in dialog table in MySQL) and
>> in log:
>>
>> Nov 8 04:27:28 ops /usr/sbin/opensips[21074]: DBG:dialog:new_dlg_val:
>> inserting =
>>
>> But when I receive BYE - I want to extract this values from dialog and
>> use it in radius function:
>> $var(outgoing_gate) = $dlg_val("incoming_gate»);
>>
>> I see next messages in log:
>> Nov 8 04:27:31 ops /usr/sbin/opensips[21074]:
>> DBG:dialog:fetch_dlg_value: looking for <"incoming_gate">
>> Nov 8 04:27:31 ops /usr/sbin/opensips[21074]:
>> DBG:dialog:fetch_dlg_value: var NOT found!
>>
>> I thought that dialog is died before I tried to fetch value, but it
>> still alive at this moment:
>> Nov 8 04:27:31 ops /usr/sbin/opensips[21074]: DBG:dialog:unref_dlg:
>> unref dlg 0x7ffa5a66f2d8 with 2 -> 3 in entry 0x7ffa5a63c318
>>
>> Need help! ((
>>
>> Best regards,
>> Alexander Mustafin
>> mustafin.aleksa...@gmail.com <mailto:mustafin.aleksa...@gmail.com>
>>
>>
>>
>>
>>
>>
>> ___
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>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
> ___
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[OpenSIPS-Users] Store dialog value and fetch it
Hi!
I’m want pass to dialog additional values, such as incoming and outgoing gate
from drouting.
I’m use store_dlg_value("incoming_gate»,"$avp(gw_id)") and this value
successfully pass to dialog (I can see it in dialog table in MySQL) and in log:
Nov 8 04:27:28 ops /usr/sbin/opensips[21074]: DBG:dialog:new_dlg_val:
inserting =
But when I receive BYE - I want to extract this values from dialog and use it
in radius function:
$var(outgoing_gate) = $dlg_val("incoming_gate»);
I see next messages in log:
Nov 8 04:27:31 ops /usr/sbin/opensips[21074]: DBG:dialog:fetch_dlg_value:
looking for <"incoming_gate">
Nov 8 04:27:31 ops /usr/sbin/opensips[21074]: DBG:dialog:fetch_dlg_value: var
NOT found!
I thought that dialog is died before I tried to fetch value, but it still alive
at this moment:
Nov 8 04:27:31 ops /usr/sbin/opensips[21074]: DBG:dialog:unref_dlg: unref dlg
0x7ffa5a66f2d8 with 2 -> 3 in entry 0x7ffa5a63c318
Need help! ((
Best regards,
Alexander Mustafin
mustafin.aleksa...@gmail.com
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Re: [OpenSIPS-Users] {s.select} and hex separators
Thanks, Bogdan.
I was tried it, but I made a mistake in whole transformation. Now it works
perfectly!
Best regards,
Alexander Mustafin
mustafin.aleksa...@gmail.com
30.10.2013, в 23:41, Bogdan-Andrei Iancu написал(а):
> $(fU{s.unescape.user}{s.select,1,#})
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[OpenSIPS-Users] {s.select} and hex separators
Hello!
I've got calls with FROM field looks like From: "Username"
;tag=5B353042
where %23 - is separator #. I need to save value before separator and trying to
use {s.select}, but this transformation not works.
I need function with separator, because VALUE may be random, but separator is
always #.
I was tried to use {s.escape.user} transformation, but unsuccessfully.
Any solutions or other ways?
Best regards,
Alexander Mustafin
mustafin.aleksa...@gmail.com
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Re: [OpenSIPS-Users] Dialog topology_hiding cut off all via
It's very good news. Is this applied for any versions? Best regards, Alexander Mustafin mustafin.aleksa...@gmail.com 21.10.2013, в 17:55, Nick Altmann написал(а): > We have daily builds in yum repository. It builds every night after new git > commit. It should work. > > -- > Nick > > > 2013/10/21 Bogdan-Andrei Iancu > Hello all, > > 1.10 had a problem with dialog based TH - this was fixed (thanks to Nick > Altman) on 18th of October. > > So please pull sources form GIT repo, not sure if the change did already > propagated to the YUM/DEB repos yet. > > Regards, > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > On 10/16/2013 07:33 PM, Alexander Mustafin wrote: >> >> Hi, Brett. >> >> OpenSIPS 1.9.1-tls from official yum repo works fine with the same config. >> >> I'm installed 1.10.0 from official repo too. >> >> Best regards, >> Alexander Mustafin >> mustafin.aleksa...@gmail.com >> >> >> >> >> 16.10.2013, в 18:55, Brett Nemeroff написал(а): >> >>> I did notice some weird behavior in 1.10 with topology hiding. >>> >>> Specifically I've seen route headers not get added back on when they >>> should; even tho the logs clearly show they are, they never make it on the >>> wire. >>> >>> This problem doesn't exist in 1.9. Not sure if it's the same issue you are >>> experiencing. Can you change to 1.9 and see if the problem goes away? >>> >>> >>> -Brett >>> >>> -- >>> Brett Nemeroff >>> Sent with Airmail >>> >>> On October 16, 2013 at 7:17:31 AM, Alexander Mustafin >>> (mustafin.aleksa...@gmail.com) wrote: >>> >>>> Hello! >>>> >>>> Any ideas? I'm still stuck on this problem (( >>>> >>>> All VIA's are deleting and no inserting by topology_hiding() >>>> >>>> Best regards, >>>> Alexander Mustafin >>>> mustafin.aleksa...@gmail.com >>>> >>>> >>>> >>>> >>>> 10.10.2013, в 10:27, Alexander Mustafin >>>> написал(а): >>>> >>>>> Hello! >>>>> >>>>> I'm testing OpenSIPS 1.10.0 version now and I've strange behavior of >>>>> topology_hiding() in dialog module. >>>>> >>>>> topology_hiding() function cut off all VIA headers and FreeSWITCH drop >>>>> this messages when receive it. >>>>> >>>>> >>>>> U 2013/10/09 20:58:07.836413 192.168.56.101:5060 -> 192.168.56.102:5080 >>>>> INVITE sip:888@192.168.56.102:5080 SIP/2.0. >>>>> Max-Forwards: 19. >>>>> From: ;tag=1899889404. >>>>> To: . >>>>> Call-ID: 799572721@192.168.56.1. >>>>> CSeq: 68 INVITE. >>>>> User-Agent: YATE/4.3.0. >>>>> Contact: . >>>>> Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO. >>>>> Content-Type: application/sdp. >>>>> Content-Length: 481. >>>>> >>>>> I found the next message in log: >>>>> DBG:dialog:dlg_del_vias: Delete via [Via: SIP/2.0/UDP >>>>> 192.168.56.1:5060;rport;branch=z9hG4bK1863223967#015#012] >>>>> >>>>> Why this happens? >>>>> >>>>> Best regards, >>>>> Alexander Mustafin >>>>> mustafin.aleksa...@gmail.com >>>>> >>>>> >>>>> >>>>> >>>>> >>>> >>>> ___ >>>> 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Dialog topology_hiding cut off all via
Hi, Brett. OpenSIPS 1.9.1-tls from official yum repo works fine with the same config. I'm installed 1.10.0 from official repo too. Best regards, Alexander Mustafin mustafin.aleksa...@gmail.com 16.10.2013, в 18:55, Brett Nemeroff написал(а): > I did notice some weird behavior in 1.10 with topology hiding. > > Specifically I've seen route headers not get added back on when they should; > even tho the logs clearly show they are, they never make it on the wire. > > This problem doesn't exist in 1.9. Not sure if it's the same issue you are > experiencing. Can you change to 1.9 and see if the problem goes away? > > > -Brett > > -- > Brett Nemeroff > Sent with Airmail > > On October 16, 2013 at 7:17:31 AM, Alexander Mustafin > (mustafin.aleksa...@gmail.com) wrote: > >> Hello! >> >> Any ideas? I'm still stuck on this problem (( >> >> All VIA's are deleting and no inserting by topology_hiding() >> >> Best regards, >> Alexander Mustafin >> mustafin.aleksa...@gmail.com >> >> >> >> >> 10.10.2013, в 10:27, Alexander Mustafin >> написал(а): >> >>> Hello! >>> >>> I'm testing OpenSIPS 1.10.0 version now and I've strange behavior of >>> topology_hiding() in dialog module. >>> >>> topology_hiding() function cut off all VIA headers and FreeSWITCH drop >>> this messages when receive it. >>> >>> >>> U 2013/10/09 20:58:07.836413 192.168.56.101:5060 -> 192.168.56.102:5080 >>> INVITE sip:888@192.168.56.102:5080 SIP/2.0. >>> Max-Forwards: 19. >>> From: ;tag=1899889404. >>> To: . >>> Call-ID: 799572721@192.168.56.1. >>> CSeq: 68 INVITE. >>> User-Agent: YATE/4.3.0. >>> Contact: . >>> Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO. >>> Content-Type: application/sdp. >>> Content-Length: 481. >>> >>> I found the next message in log: >>> DBG:dialog:dlg_del_vias: Delete via [Via: SIP/2.0/UDP >>> 192.168.56.1:5060;rport;branch=z9hG4bK1863223967#015#012] >>> >>> Why this happens? >>> >>> Best regards, >>> Alexander Mustafin >>> mustafin.aleksa...@gmail.com >>> >>> >>> >>> >>> >> >> ___ >> 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 topology_hiding cut off all via
Hello! Any ideas? I'm still stuck on this problem (( All VIA's are deleting and no inserting by topology_hiding() Best regards, Alexander Mustafin mustafin.aleksa...@gmail.com 10.10.2013, в 10:27, Alexander Mustafin написал(а): > Hello! > > I'm testing OpenSIPS 1.10.0 version now and I've strange behavior of > topology_hiding() in dialog module. > > topology_hiding() function cut off all VIA headers and FreeSWITCH drop this > messages when receive it. > > > U 2013/10/09 20:58:07.836413 192.168.56.101:5060 -> 192.168.56.102:5080 > INVITE sip:888@192.168.56.102:5080 SIP/2.0. > Max-Forwards: 19. > From: ;tag=1899889404. > To: . > Call-ID: 799572721@192.168.56.1. > CSeq: 68 INVITE. > User-Agent: YATE/4.3.0. > Contact: . > Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO. > Content-Type: application/sdp. > Content-Length: 481. > > I found the next message in log: > DBG:dialog:dlg_del_vias: Delete via [Via: SIP/2.0/UDP > 192.168.56.1:5060;rport;branch=z9hG4bK1863223967#015#012] > > Why this happens? > > Best regards, > Alexander Mustafin > mustafin.aleksa...@gmail.com > > > > > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Dialog topology_hiding cut off all via
Hello! I'm testing OpenSIPS 1.10.0 version now and I've strange behavior of topology_hiding() in dialog module. topology_hiding() function cut off all VIA headers and FreeSWITCH drop this messages when receive it. U 2013/10/09 20:58:07.836413 192.168.56.101:5060 -> 192.168.56.102:5080 INVITE sip:888@192.168.56.102:5080 SIP/2.0. Max-Forwards: 19. From: ;tag=1899889404. To: . Call-ID: 799572721@192.168.56.1. CSeq: 68 INVITE. User-Agent: YATE/4.3.0. Contact: . Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO. Content-Type: application/sdp. Content-Length: 481. I found the next message in log: DBG:dialog:dlg_del_vias: Delete via [Via: SIP/2.0/UDP 192.168.56.1:5060;rport;branch=z9hG4bK1863223967#015#012] Why this happens? Best regards, Alexander Mustafin mustafin.aleksa...@gmail.com ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
