[OpenSIPS-Users] Dialog Module and Bogus Event 8 in state 2
Hi to all, i have a lot of Dialog Module and Bogus Event 8 in state 2 in opensips logs. I have opensips 1.6.4 and i see from the sip traces that the request causing the problem is an UPDATE made from the client (Cisco) during the early state of the call (it is still ringing). Can I ignore them or what? Thank you Bye Marcello ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Radius Accounting - Acct-Statys-Type 0
Ciao Davide, l'Acct-Statys-Type è definito in più dictionary del Radius ma non può avere il valore zero. Prova a vedere un grep Acct-Statys-Type nella directory dei dictionary radius e vedi i valori che può avere. Bisogna capire da dove arriva il valore zero. Non so aiutarti di più. Ciao, Marcello On Dec 3, 2012, at 10:45 AM, Davide Dal Frà wrote: Hi , I'have a working Opensips (1.8.0) with accounting directly on Mysql database using acc module. Now i'm migrating accounting to Radius , using always acc module configured for radius. I use flag to do accounting, and my configuration is the follow: loadmodule acc.so modparam(acc , detect_direction, 0) modparam(acc , failed_transaction_flag, 3) modparam(acc , report_cancels , 1) modparam(acc , early_media, 0) modparam(acc , log_level , 1) modparam(acc , log_flag , 1) modparam(acc , log_missed_flag , 2) modparam(acc , aaa_url , radius:/etc/opensips/radius/client.conf ) modparam(acc , aaa_flag , 1) modparam(acc , aaa_missed_flag, 2) ### radius aaa module loadmodule aaa_radius.so modparam(aaa_radius , radius_config , /etc/opensips/radius/client.conf) i've removed the aaa_extra for debugging. FreeRadius is listening on another machine, but when i make a call no radius packet are sent and i can see this error in logs: Dec 3 10:08:51 sip-ngn /sbin/opensips[5498]: appended nat flag value yes Dec 3 10:08:51 sip-ngn /sbin/opensips[5476]: rc_avpair_new: unknown attribute 0 Dec 3 10:08:51 sip-ngn /sbin/opensips[5476]: ERROR:aaa_radius:rad_avp_add: failure Dec 3 10:08:51 sip-ngn /sbin/opensips[5476]: ERROR:acc:acc_aaa_request: failed to add Acct-Status-Type, 0 Acc-Status-Type is defined in dictionary, but not with value 0. Any kind of help is appreciated Thanks in advance Davide ___ 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] Radius Accounting - Acct-Statys-Type 0
The Acct-Statys-Type is defined in various Radius Dictionary but cannot be zero. Try grep Acct-Statys-Type in the directory of radius dictionary to see the values ti can have. Have to understand why it is at zero value. Sorry to not be able to help more. Bye Marcello On Dec 3, 2012, at 10:45 AM, Davide Dal Frà wrote: Hi , I'have a working Opensips (1.8.0) with accounting directly on Mysql database using acc module. Now i'm migrating accounting to Radius , using always acc module configured for radius. I use flag to do accounting, and my configuration is the follow: loadmodule acc.so modparam(acc , detect_direction, 0) modparam(acc , failed_transaction_flag, 3) modparam(acc , report_cancels , 1) modparam(acc , early_media, 0) modparam(acc , log_level , 1) modparam(acc , log_flag , 1) modparam(acc , log_missed_flag , 2) modparam(acc , aaa_url , radius:/etc/opensips/radius/client.conf ) modparam(acc , aaa_flag , 1) modparam(acc , aaa_missed_flag, 2) ### radius aaa module loadmodule aaa_radius.so modparam(aaa_radius , radius_config , /etc/opensips/radius/client.conf) i've removed the aaa_extra for debugging. FreeRadius is listening on another machine, but when i make a call no radius packet are sent and i can see this error in logs: Dec 3 10:08:51 sip-ngn /sbin/opensips[5498]: appended nat flag value yes Dec 3 10:08:51 sip-ngn /sbin/opensips[5476]: rc_avpair_new: unknown attribute 0 Dec 3 10:08:51 sip-ngn /sbin/opensips[5476]: ERROR:aaa_radius:rad_avp_add: failure Dec 3 10:08:51 sip-ngn /sbin/opensips[5476]: ERROR:acc:acc_aaa_request: failed to add Acct-Status-Type, 0 Acc-Status-Type is defined in dictionary, but not with value 0. Any kind of help is appreciated Thanks in advance Davide ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] access avp and script flags with new naming format
Hi to all, just like to know how to access an AVP with script flags set in the new ( 1.7) avp naming format. Usually in 1.6 load from DB values as: avp_db_load($fu/username,a1); avp_db_load($tu/username,a2); and after i access them with: $avp(i1:11) $avp(i2:11) Now that is not needed anymore to specify the i or s for the avp type how i can access avp 11 with script flag 1 or 2 set? Thank you Bye, Marcello ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Dialog and avp_timeout
Hi,
effectively i was using it after the loose_route().
I will try to do it after the loose_route().
Why it have to be done in this way? Just for information.
Thank you for the answer.
Regards
Marcello
On Apr 5, 2012, at 9:36 AM, Razvan Crainea wrote:
Hi, Marcello!
The block used to handle the ACK timeout is executed before loose_route or
after? It should be before.
Regards,
--
Răzvan Crainea
OpenSIPS Developer
http://www.opensips-solutions.com
On 04/04/2012 11:59 PM, Marcello Lupo wrote:
Hi,
I'm using opensips 1.6.4 with dialog support.
I use dialog default timeout to close automatically calls after 3 hours and
it works great.
Sometimes happen that some dialog remain in state 3 (200 OK received but ACK
not received) till the default_timeout is reached.
I was trying to set default_timeout to 120 seconds and change the
avp_timeout on the ACK to a greater value so the calls in state 3 will be
automatically closed form the system after 120 sec.
I read around the docs that the timeout can be changed everywhere in the
script after the dialog has been created but it is not working for me.
Every time the system give me:
DBG:dialog:get_dlg_timeout: invalid AVP value, use default timeout
and never update the timeout_avp.
I have in the config:
modparam(dialog, default_timeout, 120)
modparam(dialog, timeout_avp, $avp(i:104))
modparam(dialog, bye_on_timeout_flag, 21)
In routing block when dialog start:
create_dialog();
setflag(21);
In routing block to check ACK:
if(method==ACK $DLG_status!=NULL) {
$avp(i:104)=10800;
# $avp(i:104)=10800;
setflag(21);
}
I tried to put the avp_timeout value as INT or as STRING but no difference.
Looking in the source code seems that default_timeout is INT but timeout_avp
expect string value.
Someone can help?
Thank you
Bye
Marcello
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Re: [OpenSIPS-Users] Dialog and avp_timeout
Hi,
thank you.
In this way i cannot check the $DLG_status variable correct? I read in the docs
that this variable is available only after the loose_route.
Regards
Marcello
On Apr 5, 2012, at 10:10 AM, Razvan Crainea wrote:
Hi, Marcello!
The dialog is matched by the loose_route function. And this is when all the
dialog structures are updated. If you are changing anything after the
loose_route, the changes won't be visible in the dialog.
Regards,
--
Răzvan Crainea
OpenSIPS Developer
http://www.opensips-solutions.com
On 04/05/2012 11:06 AM, Marcello Lupo wrote:
Hi,
effectively i was using it after the loose_route().
I will try to do it after the loose_route().
Why it have to be done in this way? Just for information.
Thank you for the answer.
Regards
Marcello
On Apr 5, 2012, at 9:36 AM, Razvan Crainea wrote:
Hi, Marcello!
The block used to handle the ACK timeout is executed before loose_route or
after? It should be before.
Regards,
--
Răzvan Crainea
OpenSIPS Developer
http://www.opensips-solutions.com
On 04/04/2012 11:59 PM, Marcello Lupo wrote:
Hi,
I'm using opensips 1.6.4 with dialog support.
I use dialog default timeout to close automatically calls after 3 hours
and it works great.
Sometimes happen that some dialog remain in state 3 (200 OK received but
ACK not received) till the default_timeout is reached.
I was trying to set default_timeout to 120 seconds and change the
avp_timeout on the ACK to a greater value so the calls in state 3 will be
automatically closed form the system after 120 sec.
I read around the docs that the timeout can be changed everywhere in the
script after the dialog has been created but it is not working for me.
Every time the system give me:
DBG:dialog:get_dlg_timeout: invalid AVP value, use default timeout
and never update the timeout_avp.
I have in the config:
modparam(dialog, default_timeout, 120)
modparam(dialog, timeout_avp, $avp(i:104))
modparam(dialog, bye_on_timeout_flag, 21)
In routing block when dialog start:
create_dialog();
setflag(21);
In routing block to check ACK:
if(method==ACK $DLG_status!=NULL) {
$avp(i:104)=10800;
# $avp(i:104)=10800;
setflag(21);
}
I tried to put the avp_timeout value as INT or as STRING but no
difference. Looking in the source code seems that default_timeout is INT
but timeout_avp expect string value.
Someone can help?
Thank you
Bye
Marcello
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Re: [OpenSIPS-Users] Dialog and avp_timeout
Hi,
in which way i can recognize the ACK to a 200 OK and check that there is a
valid dialog attached to it to don't change the timeout_avp value for an ACK to
a non 200 OK?
Thank you
Regards
Marcello
On Apr 5, 2012, at 12:22 PM, Marcello Lupo wrote:
Hi,
thank you.
In this way i cannot check the $DLG_status variable correct? I read in the
docs that this variable is available only after the loose_route.
Regards
Marcello
On Apr 5, 2012, at 10:10 AM, Razvan Crainea wrote:
Hi, Marcello!
The dialog is matched by the loose_route function. And this is when all the
dialog structures are updated. If you are changing anything after the
loose_route, the changes won't be visible in the dialog.
Regards,
--
Răzvan Crainea
OpenSIPS Developer
http://www.opensips-solutions.com
On 04/05/2012 11:06 AM, Marcello Lupo wrote:
Hi,
effectively i was using it after the loose_route().
I will try to do it after the loose_route().
Why it have to be done in this way? Just for information.
Thank you for the answer.
Regards
Marcello
On Apr 5, 2012, at 9:36 AM, Razvan Crainea wrote:
Hi, Marcello!
The block used to handle the ACK timeout is executed before loose_route or
after? It should be before.
Regards,
--
Răzvan Crainea
OpenSIPS Developer
http://www.opensips-solutions.com
On 04/04/2012 11:59 PM, Marcello Lupo wrote:
Hi,
I'm using opensips 1.6.4 with dialog support.
I use dialog default timeout to close automatically calls after 3 hours
and it works great.
Sometimes happen that some dialog remain in state 3 (200 OK received but
ACK not received) till the default_timeout is reached.
I was trying to set default_timeout to 120 seconds and change the
avp_timeout on the ACK to a greater value so the calls in state 3 will be
automatically closed form the system after 120 sec.
I read around the docs that the timeout can be changed everywhere in the
script after the dialog has been created but it is not working for me.
Every time the system give me:
DBG:dialog:get_dlg_timeout: invalid AVP value, use default timeout
and never update the timeout_avp.
I have in the config:
modparam(dialog, default_timeout, 120)
modparam(dialog, timeout_avp, $avp(i:104))
modparam(dialog, bye_on_timeout_flag, 21)
In routing block when dialog start:
create_dialog();
setflag(21);
In routing block to check ACK:
if(method==ACK $DLG_status!=NULL) {
$avp(i:104)=10800;
# $avp(i:104)=10800;
setflag(21);
}
I tried to put the avp_timeout value as INT or as STRING but no
difference. Looking in the source code seems that default_timeout is INT
but timeout_avp expect string value.
Someone can help?
Thank you
Bye
Marcello
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Re: [OpenSIPS-Users] Dialog and avp_timeout
Hi Bogdan,
ok thank you for the answer.
I will implement it this night.
Bye,
Marcello
On Apr 5, 2012, at 6:44 PM, Bogdan-Andrei Iancu wrote:
Hi Marcello,
If it is an ACK for 200 OK, it will be an end-2-end ACK driven by Route hdrs
, so it will have loose_route returning true.
if it is an ACK for a negative reply, it will be a hop-by-hop ACL with no
Route hdrs, so it script (check the default one) will go like loose_route()
false - is ACK - t_check_tran() true - t_relay().
But in your case, you shouldn't really care too much - it an ACK, set a new
timeout just before the loose_route() - if it is an 200 OK ACK - the timeout
will be set; if a negative reply ACK, no dialog, no timeout :)
Regards,
Bogdan
On 04/05/2012 07:26 PM, Marcello Lupo wrote:
Hi,
in which way i can recognize the ACK to a 200 OK and check that there is a
valid dialog attached to it to don't change the timeout_avp value for an ACK
to a non 200 OK?
Thank you
Regards
Marcello
On Apr 5, 2012, at 12:22 PM, Marcello Lupo wrote:
Hi,
thank you.
In this way i cannot check the $DLG_status variable correct? I read in the
docs that this variable is available only after the loose_route.
Regards
Marcello
On Apr 5, 2012, at 10:10 AM, Razvan Crainea wrote:
Hi, Marcello!
The dialog is matched by the loose_route function. And this is when all
the dialog structures are updated. If you are changing anything after the
loose_route, the changes won't be visible in the dialog.
Regards,
--
Răzvan Crainea
OpenSIPS Developer
http://www.opensips-solutions.com
On 04/05/2012 11:06 AM, Marcello Lupo wrote:
Hi,
effectively i was using it after the loose_route().
I will try to do it after the loose_route().
Why it have to be done in this way? Just for information.
Thank you for the answer.
Regards
Marcello
On Apr 5, 2012, at 9:36 AM, Razvan Crainea wrote:
Hi, Marcello!
The block used to handle the ACK timeout is executed before loose_route
or after? It should be before.
Regards,
--
Răzvan Crainea
OpenSIPS Developer
http://www.opensips-solutions.com
On 04/04/2012 11:59 PM, Marcello Lupo wrote:
Hi,
I'm using opensips 1.6.4 with dialog support.
I use dialog default timeout to close automatically calls after 3 hours
and it works great.
Sometimes happen that some dialog remain in state 3 (200 OK received
but ACK not received) till the default_timeout is reached.
I was trying to set default_timeout to 120 seconds and change the
avp_timeout on the ACK to a greater value so the calls in state 3 will
be automatically closed form the system after 120 sec.
I read around the docs that the timeout can be changed everywhere in
the script after the dialog has been created but it is not working for
me.
Every time the system give me:
DBG:dialog:get_dlg_timeout: invalid AVP value, use default timeout
and never update the timeout_avp.
I have in the config:
modparam(dialog, default_timeout, 120)
modparam(dialog, timeout_avp, $avp(i:104))
modparam(dialog, bye_on_timeout_flag, 21)
In routing block when dialog start:
create_dialog();
setflag(21);
In routing block to check ACK:
if(method==ACK$DLG_status!=NULL) {
$avp(i:104)=10800;
# $avp(i:104)=10800;
setflag(21);
}
I tried to put the avp_timeout value as INT or as STRING but no
difference. Looking in the source code seems that default_timeout is
INT but timeout_avp expect string value.
Someone can help?
Thank you
Bye
Marcello
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--
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
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[OpenSIPS-Users] Dialog and avp_timeout
Hi,
I'm using opensips 1.6.4 with dialog support.
I use dialog default timeout to close automatically calls after 3 hours and it
works great.
Sometimes happen that some dialog remain in state 3 (200 OK received but ACK
not received) till the default_timeout is reached.
I was trying to set default_timeout to 120 seconds and change the avp_timeout
on the ACK to a greater value so the calls in state 3 will be automatically
closed form the system after 120 sec.
I read around the docs that the timeout can be changed everywhere in the script
after the dialog has been created but it is not working for me.
Every time the system give me:
DBG:dialog:get_dlg_timeout: invalid AVP value, use default timeout
and never update the timeout_avp.
I have in the config:
modparam(dialog, default_timeout, 120)
modparam(dialog, timeout_avp, $avp(i:104))
modparam(dialog, bye_on_timeout_flag, 21)
In routing block when dialog start:
create_dialog();
setflag(21);
In routing block to check ACK:
if(method==ACK $DLG_status!=NULL) {
$avp(i:104)=10800;
# $avp(i:104)=10800;
setflag(21);
}
I tried to put the avp_timeout value as INT or as STRING but no difference.
Looking in the source code seems that default_timeout is INT but timeout_avp
expect string value.
Someone can help?
Thank you
Bye
Marcello
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[OpenSIPS-Users] xmlrpc fields syntax not equal in the output
Hi to all, in our system we have more or less 1900 AOR registered. We use the mi_xmlrpc module to query the proxy on the realtime AOR in the system. The issue is that for some records the mi_xmlrpc answer with AOR and Contact in the same field and for other it splits in 2 different fields. On 1900 about 1800 are in the same field and the rest in separate field. Can you tell us why it is happening? It is a problem to parse the output. Es: same field valuestringAOR:: 1234567 Contact:: sip:1234567@10.10.10.10:5060 Q=0.1 /string/value valuestringExpires:: 829 /string/value valuestringCallid:: 99f8cfddeb296001790a3a5260eae...@void.foo.org /string/value valuestringCseq:: 1085424177 /string/value valuestringUser-agent:: Patton SN4634 3BIS UI MxSF v3.2.8.45 00A0BA037DA9 R4.2 2007-09-19 H323 SIP BRI /string/value valuestringPath:: sip:10.10.10.2;lr /string/value valuestringState:: CS_SYNC /string/value valuestringFlags:: 0 /string/value valuestringCflag:: 0 /string/value valuestringSocket:: udp:10.10.10.10:5060 /string/value valuestringMethods:: 4294967295 /string/value Es: separate fields: valuestringAOR:: 09876543 /string/value valuestringContact:: sip: 09876543@10.10.10.10:5060 Q=0.1 /string/value valuestringExpires:: 3517 /string/value valuestringCallid:: 554b5a6c-965@172.16.64.245 /string/value valuestringCseq:: 10 /string/value valuestringUser-agent:: OpenSER (1.3.2-notls (arm/linux)) /string/value valuestringPath:: sip:10.10.10.2;lr /string/value valuestringState:: CS_SYNC /string/value valuestringFlags:: 0 /string/value valuestringCflag:: 0 /string/value valuestringSocket:: udp:10.10.10.10:5060 /string/value valuestringMethods:: 4294967295 /string/value Have any clues? Thank you Bye Marcello ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Dialogs remain stale on status 5
Hi to all, using opensips 1.6.4 sometimes happen to me that some dialogs remains stale in the list of dlg_list command in state 5. In my case if i try to end a dialog that is not in state 3 or 4. May be now it is a problem of cleaning the code (i'm still developing the implementation of dialogs at the moment) Can someone tell me why it happen and most of all if there is a way to delete them without restarting the Opensips? I'm working on a web interface to manage the calls of our system and don't want to have lot of stale dialogs to appear there. Thank you Regards Marcello ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Opensips geographic redundancy
Hi to all, i have a problem and hope someone can give me a clue on it. We have an Opensips infrastructure with heartbeat HA and it is working fine. I'm implementing the geographic redundancy on our VoIP infrastructure with opensips 1.6.4. There are 2 different locations. 2 server in each location (one running the Opensips and the other running the mysql DB). The 2 structures are totally independent but share the same DB data (with mysql master-slave-master replication). So each location can handle customer requests independently, the registrations are duplicated from one server to the other to maintain all the CPE's reachable from both servers. Now I'd like to use the DNS SRV records to let the CPEs to use the 2 servers in a kind of load balance. I discovered that lot of CPE are not implementing the SRV records logic in a correct way. Patton CPE with 5.7 firmware make the first REGISTER (without authentication) on one server and after the server answer back with 401 Unauthorized the CPE retry the REGISTER (with authentication) on the other server that reject it because an invalid nonce is found (it was generated from the other server obviously). It should continue the session with the same server it started with the first REGISTER because it received an answer and the server is alive. Asterisk 1.6 is making the REGISTER properly (one for each server) but place all the calls through only one server even if it is down. I'm sure will find that lot of other CPE will have trouble with SRV records correct implementation. I'd like to find a way to do it in a CPE software independent way. So i'm starting to search another solution that let me implement the geographic redundancy without the SRV records but I'm short of ideas now without inserting any Single Point of Failure in the system. I thought to a front end proxy (in HA redundancy like the one of now) to be a load balancer to the other 2 proxies but in any way i don't' have a geographic redundancy. Have you any suggestions? Thank you all. Regards, Marcello ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips geographic redundancy
Hi Dave, thank you for your answer but this was a solution that i prefer to avoid if possible...We are AS and LIR on internet and we are already making BGP on 3 different links on our infrastructure so we can do it as last resort but it is a nightmare to let all that work with application level checks and routers around... Often there can be routing/connectivity problems that will cause the IP class to be routed but if the heartbeat is not aware of this there will be problems for sure. The goal is to use both the system in parallel. Thank you anyway. Regards, Marcello On Apr 29, 2011, at 7:07 PM, Dave Singer wrote: Marcello, Might check into getting an IP with BGP (Border Gateway Protocol) or some other IP routing protocol. Not all ISPs offer these services and it is usually offered on a subnet of addresses not a single address. If your providers/contacts can't do this or don't have good answers you might check with www.spectrumnet.us. Dave On Fri, Apr 29, 2011 at 9:22 AM, Marcello Lupo ml...@itspecialist.it wrote: Hi to all, i have a problem and hope someone can give me a clue on it. We have an Opensips infrastructure with heartbeat HA and it is working fine. I'm implementing the geographic redundancy on our VoIP infrastructure with opensips 1.6.4. There are 2 different locations. 2 server in each location (one running the Opensips and the other running the mysql DB). The 2 structures are totally independent but share the same DB data (with mysql master-slave-master replication). So each location can handle customer requests independently, the registrations are duplicated from one server to the other to maintain all the CPE's reachable from both servers. Now I'd like to use the DNS SRV records to let the CPEs to use the 2 servers in a kind of load balance. I discovered that lot of CPE are not implementing the SRV records logic in a correct way. Patton CPE with 5.7 firmware make the first REGISTER (without authentication) on one server and after the server answer back with 401 Unauthorized the CPE retry the REGISTER (with authentication) on the other server that reject it because an invalid nonce is found (it was generated from the other server obviously). It should continue the session with the same server it started with the first REGISTER because it received an answer and the server is alive. Asterisk 1.6 is making the REGISTER properly (one for each server) but place all the calls through only one server even if it is down. I'm sure will find that lot of other CPE will have trouble with SRV records correct implementation. I'd like to find a way to do it in a CPE software independent way. So i'm starting to search another solution that let me implement the geographic redundancy without the SRV records but I'm short of ideas now without inserting any Single Point of Failure in the system. I thought to a front end proxy (in HA redundancy like the one of now) to be a load balancer to the other 2 proxies but in any way i don't' have a geographic redundancy. Have you any suggestions? Thank you all. Regards, Marcello ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- David Singer ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Core dump on xl_get_pai in Openser 1.2.2
Hi to all, i write to this list because i hope some developer that follwed the OpenSer development can help me. Any way we are starting to move to OpenSIPS 1.6 now but it will require time because of many changes around. I implemented the P-Asserted-Identity and P-Preferred-Identity Header management on our OpenSer 1.2.2. On the development server all is working properly but when i activate the new config on our production system i start to get crashes of the openser process: openser[31483]: segfault at rip 00487853 rsp 7fff94c9f1a0 error 4 openser[6511]: segfault at 0025 rip 004214ce rsp 7fff80f05480 error 4 openser[8967] general protection rip:4214ce rsp:7fff05fbb530 error:0 openser[9872] general protection rip:4214ce rsp:7fffd2b630d0 error:0 I checked the core dump with gdb and i have this info from the core: Core was generated by `/usr/sbin/openser'. Program terminated with signal 11, Segmentation fault. #0 xl_get_pai (msg=0x2b4ad9030d00, res=0x7fffd2b63190, param=value optimized out, flags=value optimized out) at items.c:1050 1050res-rs.len = get_pai(msg)-uri.len; We are using the 1.2.2 version and i know it is quite old but i like to understand if it happen for some known bug or if i made something wrong. For the moment is not possible for us to manage the upgrade to a newer version of the proxy till we will not have it redundant on another location. Thank you for help. Bye, Marcello ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
