[OpenSIPS-Users] openSips - Asterisk and Session Timers: ACK is sent to 192.168.1.10
Hi, My setup: - 11.22.33.44 : openSIPS 1.6.3 - 11.22.33.45 : one of the Asterisk 1.6.2.13 servers - 88.77.66.55 : my public ip-address - 192.168.1.10 : my local ip-address (NAT) All is working well except Session Timers where the Re-Invite originates from Asterisk. I have a SIP trace ( http://pastebin.com/raw.php?i=NRDdaktn ) of a call initiated by a softphone on my pc (192.168.1.10). When Asterisk sends the Re-Invite (line 290) my softphone receives this Re-Invite correctly. The 100 Trying and 200 OK are also handled as it should. But on line 455 you see openSIPS forwarding the ACK to 192.168.1.10 instead of 88.77.66.55. Does anyone know why this isn't working? Thanks in advance! ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] load_balance debugging
Hi!
I have a strange problem with trying to use avps in load_balance function.
I'm trying to do balancing like this:
avp_db_query(select phone, resource from phone_resource where
phone like '%$fU%', $avp(i:111);$avp(i:112));
avp_print();
xlog(L_INFO,$fu = $avp(i:111));
if ($fu=~$avp(i:111)) {
xlog(L_INFO, AAA detected!\n);
load_balance(1, $avp(i:112));
Here is what I get in logs:
Oct 22 16:35:44 kzo2 /usr/local/sbin/opensips[8214]:
INFO:avpops:ops_print_avp: #011#011#011id=112
Oct 22 16:35:44 kzo2 /usr/local/sbin/opensips[8214]:
INFO:avpops:ops_print_avp: #011#011#011val_str=aaa / 3
Oct 22 16:35:44 kzo2 /usr/local/sbin/opensips[8214]:
INFO:avpops:ops_print_avp: p=0x7f73f6b251e8, flags=0x0002
Oct 22 16:35:44 kzo2 /usr/local/sbin/opensips[8214]:
INFO:avpops:ops_print_avp: #011#011#011id=111
Oct 22 16:35:44 kzo2 /usr/local/sbin/opensips[8214]:
INFO:avpops:ops_print_avp: #011#011#011val_str=1234565677 / 10
Oct 22 16:35:44 kzo2 /usr/local/sbin/opensips[8214]:
sip:1234565...@xx.xxx.xxx.xxx;user=phone = 1234565677
Oct 22 16:35:44 kzo2 /usr/local/sbin/opensips[8214]: AAA detected!
Oct 22 16:35:44 kzo2 /usr/local/sbin/opensips[8214]:
ERROR:load_balancer:do_load_balance: unknown resource in input string
Oct 22 16:35:44 kzo2 /usr/local/sbin/opensips[8214]:
DBG:core:comp_scriptvar: int 26 : -1 / 0
If I write the resource directly, like
load_balance(1, aaa);
everything works fine.
In the above log you can see that $avp(i:112) contains exactly the same
resource string.
What is the correct way to do this, or is there a way to debug how load
balancer searches for resources defined in database?
Thanks in advance,
Alexandr A. Alexandrov
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[OpenSIPS-Users] Problem creating Registrar database for OpenSIPS with PostgreSQL
Hello, I have successfully compiled and installed OpenSIPS 1.6.3 (no tls) with the PostgreSQL module for an OpenSIPS Registrar testing installation. I have edited the *opensipsctlrc* file to specify the connection details for a remote PostgreSQL database server I want to use as the persistence store for an OpenSIPS Registrar installation. I have set the hostname, database name, usernames and passwords. Nevertheless, when executing the *opensipsdbctl* command, I get the following error: -e \E[37;31mERROR: ~./pgpass does not exist, please create this file and support proper credentials for user postgres. -e \E[37;31mERROR: Note: you need at least postgresql= 7.3 Any ideas? Thx ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] opensips tm timer core dump
Hi Kennard,
I suppose the bt is the same ? do you still have the core file ?
Regards,
Bogdan
kennard_wh...@logitech.com wrote:
Hi Bodgen,
I replicated the error. Unfortunately the entire insert_timer_unsafe
and been in-lined and little is available:
Program terminated with signal 11, Segmentation fault.
#0 0x7f8b8356c2c2 in insert_timer_unsafe (new_tl=0x7f8b7a54e310,
list_id=WT_TIMER_LIST, ext_timeout=value optimized out) at timer.c:731
731 timer.c: No such file or directory.
in timer.c
(gdb) print tl
$1 = value optimized out
(gdb) print *tl
Cannot access memory at address 0x0
(gdb) print ptr
$2 = value optimized out
(gdb) print *ptr
Cannot access memory at address 0x0
(gdb) print *new_tl
No symbol new_tl in current context.
(gdb) up
#1 set_1timer (new_tl=0x7f8b7a54e310, list_id=WT_TIMER_LIST,
ext_timeout=value optimized out) at timer.c:904
904 in timer.c
(gdb) print *new_tl
$3 = {next_tl = 0x0, prev_tl = 0x0, ld_tl = 0x0, time_out = 0,
timer_list = 0x0, deleted = 0}
(gdb) print list
$4 = value optimized out
(gdb) print timeout
$5 = 32
(gdb) print new_tl
$6 = (struct timer_link *) 0x7f8b7a54e310
I'll keep the core for a while -- please let me know if there is
anything else I can try.
Thanks,
Kennard
Inactive hide details for Bogdan-Andrei Iancu ---10/08/2010 04:40:47
AM---Hi Kennard, Ok, keep the core next time :)Bogdan-Andrei Iancu
---10/08/2010 04:40:47 AM---Hi Kennard, Ok, keep the core next time :)
From: Bogdan-Andrei Iancu bog...@voice-system.ro
To: OpenSIPS users mailling list users@lists.opensips.org
Date: 10/08/2010 04:40 AM
Subject: Re: [OpenSIPS-Users] opensips tm timer core dump
Sent by: users-boun...@lists.opensips.org
Hi Kennard,
Ok, keep the core next time :)
Regards,
Bogdan
kennard_wh...@logitech.com wrote:
Hi Bogden,
Thanks for explaining the child processes involved -- I misunderstood
what was happening.
Unfortunately, I don't have the core anymore. My recollection is that
I couldn't print anything useful due to compiler optimization. That
said, this should re-create pretty easily, and I'll get more dumps
next time it happens.
Regards,
Kennard
Inactive hide details for Bogdan-Andrei Iancu ---10/05/2010 01:41:38
AM---Hi Kennard, The core was generated by process 22255:Bogdan-Andrei
Iancu ---10/05/2010 01:41:38 AM---Hi Kennard, The core was generated
by process 22255:
From: Bogdan-Andrei Iancu bog...@voice-system.ro
To: OpenSIPS users mailling list users@lists.opensips.org
Date: 10/05/2010 01:41 AM
Subject: Re: [OpenSIPS-Users] opensips tm timer core dump
Sent by: users-boun...@lists.opensips.org
Hi Kennard,
The core was generated by process 22255:
[22238]: INFO:core:handle_sigs: child process 22255 exited by a
signal 11
and this process also reported mem problems:
[22255]: ERROR:tm:new_t: out of mem
Can you print the tl or ptr variables in frame 0?
Regards,
Bogdan
kennard_wh...@logitech.com wrote:
Running against opensips HEAD, I got a segfault in the tm timer code.
I believe this is triggered by running out of shared memory.
The stack trace:
(gdb) where
#0 0x7fe8f8d96212 in insert_timer_unsafe (new_tl=0x7fe8f66337b0,
list_id=WT_TIMER_LIST, ext_timeout=value optimized out) at
timer.c:731
#1 set_1timer (new_tl=0x7fe8f66337b0, list_id=WT_TIMER_LIST,
ext_timeout=value optimized out) at timer.c:904
#2 0x7fe8f8d78ac8 in t_release_transaction (trans=0x7fe8f6633730)
at t_funcs.c:122
#3 0x7fe8f8d808e5 in t_unref (p_msg=value optimized out)
at t_lookup.c:1152
#4 0x00483ae5 in exec_post_req_cb ()
#5 0x0046c1e4 in receive_msg ()
#6 0x004bc77c in udp_rcv_loop ()
#7 0x0042de9c in main ()
The offending code (I believe):
if (tl-time_out==ptr-time_out) {
tl-ld_tl = ptr-ld_tl
ptr-ld_tl = 0;
tl-ld_tl-ld_tl = tl; -- SEG FAULT HERE (according to trace)
} else {
tl-ld_tl = tl;
}
Unfortunately, due to optimization I cannot dump anything useful, and
I'm not convinced the actual fault is on the line indicated. Note that
the core dump is not one of the processes that reported out of memory.
Maybe one of the other processes left the timer list in a corrupt
state?
The log file:
Sep 29 11:43:36 org-sip01 /var/run/openser/opensips-pres[22255]:
ERROR:tm:sip_msg_cloner: no more share memory
Sep 29 11:43:36 org-sip01 /var/run/openser/opensips-pres[22255]:
ERROR:tm:new_t: out of mem
Sep 29 11:43:36 org-sip01 /var/run/openser/opensips-pres[22255]:
ERROR:tm:t_newtran: new_t failed
Sep 29 11:43:36 org-sip01 /var/run/openser/opensips-pres[22254]:
WARNING:core:fm_malloc: Not enough free memory, will atempt
defragmenation
Sep 29 11:43:36
Re: [OpenSIPS-Users] {param.value, field} problem with quoted values
Hi Jeff,
I have just tried like this with 1.6 and it works:
$var(text)=sip:2165551...@192.168.32.2;rn=+1216621;ocn=9321;carrier=\ATT\;cat=\RBOC\;lata=320;
$avp(s:call_carname) = $(var(text){param.value,carrier});
xlog(carrier = $avp(s:call_carname)\n);
= carrier = ATT
But if you take it from a header, you don't have the quotes escaped. Do
you see any error in the log?
Regards,
--
Anca Vamanu
www.voice-system.ro
On 10/21/2010 07:15 AM, Jeff Pyle wrote:
Hello,
I'm pulling some data out of a script variable using the {param.value,field}
string manipulator. It works fine, unless the value is in quotes. For
example, $var(ct) is like this:
sip:2165551...@192.168.32.2;rn=+1216621;ocn=9321;carrier=ATT;cat=RBOC;lata=320
I can pull out the ocn, lata and rn successfully with functions like this:
if !($(var(ct){param.value,ocn}) == '') {
$avp(s:call_ocn) = $(var(lrnct){param.value,ocn});
}
But, the following 'if' statement returns false and never sets the AVP:
if !($(var(ct){param.value,carrier}) == '') {
$avp(s:call_carname) =
$(var(lrnct){param.value,carrier});
}
If I skip the 'if' and set the AVP directly, it's empty.
This is on Opensips 1.5.3. Obviously it's possible this has already been
addressed in more recent versions. This server is scheduled for an upgrade
to 1.6 in the coming weeks, but we're not there yet. I thought I'd ask
anyway.
- Jeff
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Re: [OpenSIPS-Users] $fU read-only, calling number modification problem
Hi Maciej, default value for restore type is auto, which means you do not have to do anything for proper fixing of all messages in the dialog. Some hints: 1) are you using the rr module for doing record_route and loose_route ? This is essential for auto restore to work 2) in the outbound INVITE, in the RR header, do you see the vsf param ? 3) the replies and ACK for the INVITE are properly restored? 4) for 200 OK ACK , re-INVITEs, BYE requests, do you see the vsf param in the received Route header ? Regards, Bogdan Maciej Bylica wrote: Guys one more question. I have some problems to force opensips to restore oryginal uri that was previously replaced. I do have: - no modparams in config - route[0] is responsible for basic routing - route[5] is for proper call distribution by using lookup(location) information. In the same route i have implemented calling number modification. Just before the end of route i am arming t_relay with failure route (in case of busy for instance). - failure_route[105] is to do_routing the call to VM service outside the opensips. But just before t_relaying here i need to restore the original $fU. According to http://www.opensips.org/html/docs/modules/1.6.x/uac.html#id292928 i should use uac_restore_from() command (with default restore_mode modparam). Unfortunately the calling number once replaced cannot be restored in my case. Below you may find a snippet from debug Number replacing is generating vsf param /sbin/opensips[28268]: DBG:uac:w_replace_from: dsp=0xffd3f38c (len=0) , uri=0xffd3f394 (len=29) /sbin/opensips[28268]: DBG:uac:replace_uri: removing display [unknown] /sbin/opensips[28268]: DBG:uac:replace_uri: uri to replace [sip:48222114...@11.22.33.44] /sbin/opensips[28268]: DBG:uac:replace_uri: replacement uri is [sip:222114...@11.22.33.44] /sbin/opensips[28268]: DBG:uac:replace_uri: encode is=AAIJAgUMBAEAD3AHeQUXBR8FHwUaCRoKHAsyOQ-- len=44 /sbin/opensips[28268]: DBG:rr:add_rr_param: adding (;vsf=AAIJAgUMBAEAD3AHeQUXBR8FHwUaCRoKHAsyOQ--) 0x81c2828 then opensips constucts Busy message and in the same time without any uac_restore_from() command: /sbin/opensips[28795]: DBG:uac:restore_uri_reply: removing sip:222114...@11.22.33.44 /sbin/opensips[28795]: DBG:uac:restore_uri_reply: inserting unknown sip:48222114...@11.22.33.44 then the call failes to failure_route[105] and uac_restore_from() is generating following debug /sbin/opensips[28797]: DBG:uac:restore_uri: getting 'vsf' Route param /sbin/opensips[28797]: DBG:uac:restore_uri: route param 'vsf' not found just after that the call is hitting do_routing and t_relay to VM server. Of course calling number was not restored to original one. Could You please point me where the problem is located? Just one more info - calling number modification part of config is located in separated route[10] to be used whenever i wish in my script. Thx, Maciej. -- Bogdan-Andrei Iancu OpenSIPS Bootcamp 15 - 19 November 2010, Edison, New Jersey, USA www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Multiply location
Hi Anton, No, this is not normal and the user agent that you use has a bad SIP implementation - a CANCEL cancels the request with the same Via Branch id. RFC 3261 - section 9.1 A CANCEL constructed by a client MUST have only a single Via header field value matching the top Via value in the request being canceled. Using the same values for these header fields allows the CANCEL to be matched with the request it cancels Regards, -- Anca Vamanu www.voice-system.ro On 10/21/2010 11:21 AM, Антон Загорский wrote: Hello. My UA was registered at openSIPS under same account twice, so in the location table there are two records. When openSIPS receive INVITE on that UA it forks in two INVITEs with _SAME_ Call-ID (but a different 'branch' in the 'via' header). So, after when UA replies OK openSIPS sends CANCEL on unanswered INVITE but it CANCEL cancels established session because of both sessions have same Call-ID. Is this situation normal? WBR, Anton Zagorskiy VoIP Developer, Oyster Telecom Phone.: +7 812 601-0666 Fax: +7 812 601-0593 a.zagors...@oyster-telecom.ru www.oyster-telecom.ru ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Problem creating Registrar database for OpenSIPS with PostgreSQL
Installing postgresql-client package and configuring a ./pgpass file with the following format fixed the problem: hostname:port:database:username:password Regards, David On Fri, Oct 22, 2010 at 3:26 PM, David Santiago david.santi...@almiralabs.com wrote: Hello, I have successfully compiled and installed OpenSIPS 1.6.3 (no tls) with the PostgreSQL module for an OpenSIPS Registrar testing installation. I have edited the opensipsctlrc file to specify the connection details for a remote PostgreSQL database server I want to use as the persistence store for an OpenSIPS Registrar installation. I have set the hostname, database name, usernames and passwords. Nevertheless, when executing the opensipsdbctl command, I get the following error: -e \E[37;31mERROR: ~./pgpass does not exist, please create this file and support proper credentials for user postgres. -e \E[37;31mERROR: Note: you need at least postgresql= 7.3 Any ideas? Thx ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] opensips coredump on stop/restart
Hi Dave, The core files do not help as they need to be investigated with your binaries. Please run gdb path_to_executable path_to_corefile and then run 'bt full' and send the output. Regards, -- Anca Vamanu www.voice-system.ro On 10/21/2010 10:08 PM, Dave Singer wrote: On my production servers, both 1.6.2 and 1.6.3 versions of opensips, almost every time I restart opensips it creates a core dump. Since I'm giving it 2GB shared mem, it takes a little while to write the core to disk and start running again. So a couple questions. 1. Am I giving it more memory than it needs? Judging by how much the core dump compressed probably way to much. I'm running about 30 calls per sec with up to 1000 active calls. Calls are using mediaproxy with engage and thus also dialog. I did run into memory shortage problems before I explicitly upped it to 2GB with -m 2000 on the cmd line for starting opensips. 2. Can someone look at the core dump(s) or let me know what/how to look at them? The 2GB core dumps compressed very small and are accessible via http: opensips-1.6.2-notls.core.gz 6MB http://viper.wideideas.net/opensips-1.6.2-notls.core.gz opensips-1.6.3-tls_1.core.gz 2MB http://viper.wideideas.net/opensips-1.6.3-tls_1.core.gz opensips-1.6.3-tls_2.core.gz 2MB http://viper.wideideas.net/opensips-1.6.3-tls_2.core.gz Thanks for any help or suggestions. Dave ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] load_balance debugging
Hi Alexandr,
The second parameter of load_balance() function can not be a
pseudovariable, but only string.
Regards,
--
Anca Vamanu
www.voice-system.ro
On 10/22/2010 03:53 PM, Alexandr A. Alexandrov wrote:
Hi!
I have a strange problem with trying to use avps in load_balance function.
I'm trying to do balancing like this:
avp_db_query(select phone, resource from phone_resource where
phone like '%$fU%', $avp(i:111);$avp(i:112));
avp_print();
xlog(L_INFO,$fu = $avp(i:111));
if ($fu=~$avp(i:111)) {
xlog(L_INFO, AAA detected!\n);
load_balance(1, $avp(i:112));
Here is what I get in logs:
Oct 22 16:35:44 kzo2 /usr/local/sbin/opensips[8214]:
INFO:avpops:ops_print_avp: #011#011#011id=112
Oct 22 16:35:44 kzo2 /usr/local/sbin/opensips[8214]:
INFO:avpops:ops_print_avp: #011#011#011val_str=aaa / 3
Oct 22 16:35:44 kzo2 /usr/local/sbin/opensips[8214]:
INFO:avpops:ops_print_avp: p=0x7f73f6b251e8, flags=0x0002
Oct 22 16:35:44 kzo2 /usr/local/sbin/opensips[8214]:
INFO:avpops:ops_print_avp: #011#011#011id=111
Oct 22 16:35:44 kzo2 /usr/local/sbin/opensips[8214]:
INFO:avpops:ops_print_avp: #011#011#011val_str=1234565677 / 10
Oct 22 16:35:44 kzo2 /usr/local/sbin/opensips[8214]:
sip:1234565...@xx.xxx.xxx.xxx;user=phone = 1234565677
Oct 22 16:35:44 kzo2 /usr/local/sbin/opensips[8214]: AAA detected!
Oct 22 16:35:44 kzo2 /usr/local/sbin/opensips[8214]:
ERROR:load_balancer:do_load_balance: unknown resource in input string
Oct 22 16:35:44 kzo2 /usr/local/sbin/opensips[8214]:
DBG:core:comp_scriptvar: int 26 : -1 / 0
If I write the resource directly, like
load_balance(1, aaa);
everything works fine.
In the above log you can see that $avp(i:112) contains exactly the same
resource string.
What is the correct way to do this, or is there a way to debug how load
balancer searches for resources defined in database?
Thanks in advance,
Alexandr A. Alexandrov
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Re: [OpenSIPS-Users] Core dump in svn head
Hi Dave,
the crash refers to an invalid dst_uri and I see in your failure route
you are not using any $du related op Also, I see you add a single
new branch (via RURI) (no parallel forking)
Also, in branch_route, you do not do any dst_uri or ruri ops - you work
only on headers ?
Could you print from gdb:
frame 1 : request-dst_uri
frame 2: *next_hop
frame 3: p_msg-dst_uri
Best regards,
Bogdan
Dave May wrote:
Yes, I do set a branch_route to strip RPID header and strip/set the
P-Asserted-Identity header.
I'm using drouting, and that portion of the failure_route is based on
the code from Flavio's book (with the addition of e164 mangling,
t_on_branch to handle the header stuff mentioned above, and t_on_failure
to handle multiple failures):
if(isflagset(10)){
if (!t_check_status(408|[56][0-9][0-9])) {
# this is not a GW failure
exit;
}
if (use_next_gw()) {
xlog($ci\t$C(rc)$si $rm$C(xx)
$C(xr)failure_route[1]$C(xx)
$C(xr)use_next_gw()=$C(rg)$ru$C(xx)\n);
### Need to duplicate e164 check from original
do_routing() call
### Want to do this in branch_route, but not
allowed to use is_uri_user_e164() there
###
#
# Does the gateway entry request e164 format?
if ($avp(s:dr_gw_attrs) != )
($(avp(s:dr_gw_attrs){param.value,format})==e164) {
xlog($ci\t$C(rc)$si $rm$C(xx)
failure_route[1] Gateway requires e164 format\n);
# Check and fix RURI
if (is_uri_user_e164($ru)) {
xlog($ci\t$C(rc)$si $rm$C(xx)
failure_route[1] $ru is in e164 format\n);
} else {
xlog($ci\t$C(rc)$si $rm$C(xx)
failure_route[1] $ru is not in e164 format\n);
dp_translate(3);
}
}
# Need to ensure that failures on the failure
are also handled
t_on_branch(1);
t_on_failure(1);
xlog($ci\t$C(rc)$si $rm$C(xx)
failure_route[1] t_relay() next rU = $rU fU=$fU
tU=$tU du=$du\n);
setdebug(4);
t_relay();
setdebug();
xlog($ci\t$C(rc)$si $rm$C(xx)
failure_route[1] t_relay() complete\n);
exit;
} else {
xlog($ci\t$C(rc)$si $rm$C(xx)
$C(xr)failure_route[1]-$C(xx) isflagset(10) $C(sr)-- use_next_gw()
FAILED!!!$C(xx)\n);
t_reply(503, Service not available, no more
gateways);
exit;
}
Dave.
-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei
Iancu
Sent: Wednesday, October 13, 2010 4:43 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Core dump in svn head
Hi Dave,
In failure route, how do you add the new destination/branch ?
Also, do you have a branch route set ?
Regards,
Bogdan
Dave May wrote:
Over the past couple weeks I have been getting occasional segfaults
just
prior to (or perhaps in the process of) a t_relay() in my
failure_route.
Still haven't gotten to the bottom of the root cause in my config, but
I
was able to find and fix the symptomatic code in the
pre_print_uac_request() function in t_fwd.c (based on Bogdan's
previous
fix for a similar problem).
Backtrace:
#0 0xb763ba55 in memcpy () from /lib/tls/i686/cmov/libc.so.6
#1 0xb72ede1d in pre_print_uac_request (t=0x972ddf78, branch=1,
request=0x972cec64) at t_fwd.c:177
#2 0xb72eeac0 in add_uac (t=0x972ddf78, request=0x7fff,
uri=0xbf8a250c, next_hop=0xbf8a2514, path=0xb7315100, proxy=0x0) at
t_fwd.c:401
#3 0xb72efbc0 in t_forward_nonack (t=0x972ddf78, p_msg=0xb7314de0,
proxy=0x0) at t_fwd.c:660
#4 0xb72fbdbf in w_t_relay (p_msg=0xb7314de0, proxy=0x0, flags=value
optimized out) at tm.c:1116
#5 0x08057810 in do_action (a=0x819f924, msg=0xb7314de0) at
action.c:1155
#6 0x080557ff in run_action_list (a=0x819e7cc, msg=0xb7314de0) at
action.c:140
#7 0x0805981e in do_action (a=0x819fed4, msg=0xb7314de0) at
action.c:821
#8 0x080557ff in run_action_list (a=0x819e584, msg=0xb7314de0) at
action.c:140
#9 0x0805981e in do_action (a=0x81a0390, msg=0xb7314de0) at
action.c:821
#10 0x080557ff in
Re: [OpenSIPS-Users] Dialplan problem ?
Hi Marcio,
first of all be sure you are using the latest SVN check out from 1.6 branch.
What is really interesting in your case I do not see any err / warning
message...and the code is generating err/warn messages before destroying
the rule.
What is happening is that during the DB load, when a new rule is
processed, it is discarded due some issue, but this issue is not logged
.
Don't you see any warning as:
failed to build rule - skipping
?
Best regards,
Bogdan
Marcio Veloso Antunes wrote:
Hi,
The dialplan still not working...
Should i change for other version than OpenSIPS 1.6.3 ?
Thanks in advance,
Marcio
Em Qua 13 Out 2010, às 18:01:48, Marcio Veloso Antunes escreveu:
Hi Bogdan,
Thanks for your fast reply...
The version is 1.6.3. The strange thing is that it was working, but after
i entered new rules it stopped, and even after emptying the table and
reinserting just that 2 routes it still not working.
I've tryed 'opensipsctl dialplan reload' but still not working.
Silly question: Could this problem be related with 'id' column? I'am
asking based on the fact that initially it was working...
Thanks again,
Marcio
Em Qua 13 Out 2010, às 17:52:26, Bogdan-Andrei Iancu escreveu:
Hi Marcio,
The answer is:
Oct 13 17:05:47 perseu /sbin/opensips[13077]: DBG:dialplan:build_rule:
references:1 , max:1
Oct 13 17:05:47 perseu /sbin/opensips[13077]: DBG:dialplan:destroy_rule:
destroying rule with priority 1
It looks like opensips rejects the rules while loading them at startup,
so basically you end up with no rule at runtime.
what opensips version are you using ?
Regards,
Bogdan
Marcio Veloso Antunes wrote:
Hi guys,
Sorry to bother you, but i can't find the problem why this is not
working:
Actual dialplan:
r...@perseu:/etc/opensips# opensipsctl dialplan show
dialplan tables
id | dpid | pr | match_op | match_exp | match_len | subst_exp |
repl_exp | attrs
+--++--+---+---+---
-- +--+---
18 | 1 | 0 | 1 | ^00[1-9][0-9]+ | 0 | ^0(0[1-9][0-9]+) | \1 |
19 | 1 | 1 | 1 | ^00[1-9][1-9][0-9]{8} | 0 | ^0(0[1-9][1-9][0-9]{8}) |
\1 |
(2 rows)
String being tested: '002185392949'
---
-
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Bogdan-Andrei Iancu
OpenSIPS Bootcamp
15 - 19 November 2010, Edison, New Jersey, USA
www.voice-system.ro
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Services management - question about proper module
Hi Maciej Maciej Bylica wrote: Hi, Have anyone tried to use usr_preferences, AVPops to determine the service to be fetched by the script? That is the the proper module for handling generic attribute. Uisng AVPops module you can load from db, for a certain user, a certain attribute (via avp_db_load ). You may use different attributes (AVPs) for different services - like one attreibute to be URI for permanent call fwd other for being URI for busy redirect. Then i am planning to use switch statement to add different prefixes before the called number and t_relay to asterisk server to do the rest. keep in mind that certain ops can be done on opensips (like call fwd), you do not need asterisk. Regards, Bogdan Is this proper point of view? Thx, Maciej. Hello. I am planning to provide opensips with a kind of mechanism to manage customer services/features like call-forward/VM/follow-me and so on. It should work in following way: If $rU is provided in subscriber table then user enabled service name is obtained from some db table. On the basis of that value opensips should do the magic :) The question is what kind of module is the best to follow. Is it AVPops or maybe there is another way to achieve my goal. What are pros and cons. Thx in advance, Maciej. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS Bootcamp 15 - 19 November 2010, Edison, New Jersey, USA www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Static Routes in OpenSIPS proxy
Hello Andrea, If you check the default opensips.cfg, you can see that there is a step where only the initial requests are getting there - starting from that point you can implement your static routing using if statements, checking the $rU (request username ) and setting new destination (writing in $rd) Regards, Bogdan andrea wrote: Hi, i'm a beginner of OpenSIPS, i've installed 1.6 version using standard configuration. I would like to know i how add static routes (or SIP trunk) in a proxy server based OpenSIPS. In particular, Can you suggest me a modification in the configuration file so that this proxy can route the calls incoming from an IP-PABX sip to phones registerd to others IP-PABX via SIP trunks? Thanks in advance. Andrea -- Bogdan-Andrei Iancu OpenSIPS Bootcamp 15 - 19 November 2010, Edison, New Jersey, USA www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] How opensis can manage different ports.
put more listening definitions: listen=udp:ip:port1 listen=udp:ip:port2 Regards, Bogdan Marcella wrote: KINGSSQUEEN ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS Bootcamp 15 - 19 November 2010, Edison, New Jersey, USA www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] [OpenSIPS LiveDVD]Register OK but call failed:prompt 420 invalid destination
bla, emailing via subject :DI wonder what the email body is good for in this case :P Check the dialplan rules - your opensips is not recognizing the local subscribers - the rules in DP must match the local subscribers Regards, Bogdan Marcella wrote: KINGSSQUEEN ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS Bootcamp 15 - 19 November 2010, Edison, New Jersey, USA www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] opensips tm timer core dump
Yes to both. Bt is below. I'll recompile with with optimization and
reproduce the problem today.
(gdb) where
#0 0x7f8b8356c2c2 in insert_timer_unsafe (new_tl=0x7f8b7a54e310,
list_id=WT_TIMER_LIST, ext_timeout=value optimized out) at
timer.c:731
#1 set_1timer (new_tl=0x7f8b7a54e310, list_id=WT_TIMER_LIST,
ext_timeout=value optimized out) at timer.c:904
#2 0x7f8b8354eb18 in t_release_transaction (trans=0x7f8b7a54e290)
at t_funcs.c:122
#3 0x7f8b83556959 in t_unref (p_msg=value optimized out)
at t_lookup.c:1152
#4 0x00483b85 in exec_post_cb (msg=0x7ab158) at script_cb.c:198
#5 exec_post_req_cb (msg=0x7ab158) at script_cb.c:216
#6 0x0046c284 in receive_msg (
buf=0x775f00 PUBLISH sip:kwlt-b000...@sip.sightspeed.org SIP/2.0\r
\nRecord-Route: sip:74.217.68.83;r2=dn;lr=on\r\nRecord-Route:
sip:74.217.68.83;r2=up;lr=on\r\nVia: SIP/2.0/UDP
74.217.68.83;branch=z9hG4bK-d8754z-d29..., len=722,
rcv_info=0x7fff20f62330) at receive.c:165
#7 0x004bc81c in udp_rcv_loop () at udp_server.c:492
#8 0x0042deb5 in main_loop (argc=value optimized out,
argv=value optimized out) at main.c:823
#9 main (argc=value optimized out, argv=value optimized out)
at main.c:1388
(gdb) bt full
#0 0x7f8b8356c2c2 in insert_timer_unsafe (new_tl=0x7f8b7a54e310,
list_id=WT_TIMER_LIST, ext_timeout=value optimized out) at
timer.c:731
ptr = value optimized out
#1 set_1timer (new_tl=0x7f8b7a54e310, list_id=WT_TIMER_LIST,
ext_timeout=value optimized out) at timer.c:904
timeout = 32
__FUNCTION__ = set_1timer
#2 0x7f8b8354eb18 in t_release_transaction (trans=0x7f8b7a54e290)
at t_funcs.c:122
No locals.
#3 0x7f8b83556959 in t_unref (p_msg=value optimized out)
at t_lookup.c:1152
kr = 0
__FUNCTION__ = t_unref
#4 0x00483b85 in exec_post_cb (msg=0x7ab158) at script_cb.c:198
No locals.
#5 exec_post_req_cb (msg=0x7ab158) at script_cb.c:216
No locals.
#6 0x0046c284 in receive_msg (
buf=0x775f00 PUBLISH sip:kwlt-b000...@sip.sightspeed.org SIP/2.0\r
\nRecord-Route: sip:74.217.68.83;r2=dn;lr=on\r\nRecord-Route:
sip:74.217.68.83;r2=up;lr=on\r\nVia: SIP/2.0/UDP
74.217.68.83;branch=z9hG4bK-d8754z-d29..., len=722,
rcv_info=0x7fff20f62330) at receive.c:165
From: Bogdan-Andrei Iancu bog...@voice-system.ro
To: OpenSIPS users mailling list users@lists.opensips.org
Date: 10/22/2010 07:02 AM
Subject:Re: [OpenSIPS-Users] opensips tm timer core dump
Sent by:users-boun...@lists.opensips.org
Hi Kennard,
I suppose the bt is the same ? do you still have the core file ?
Regards,
Bogdan
kennard_wh...@logitech.com wrote:
Hi Bodgen,
I replicated the error. Unfortunately the entire insert_timer_unsafe
and been in-lined and little is available:
Program terminated with signal 11, Segmentation fault.
#0 0x7f8b8356c2c2 in insert_timer_unsafe (new_tl=0x7f8b7a54e310,
list_id=WT_TIMER_LIST, ext_timeout=value optimized out) at timer.c:731
731 timer.c: No such file or directory.
in timer.c
(gdb) print tl
$1 = value optimized out
(gdb) print *tl
Cannot access memory at address 0x0
(gdb) print ptr
$2 = value optimized out
(gdb) print *ptr
Cannot access memory at address 0x0
(gdb) print *new_tl
No symbol new_tl in current context.
(gdb) up
#1 set_1timer (new_tl=0x7f8b7a54e310, list_id=WT_TIMER_LIST,
ext_timeout=value optimized out) at timer.c:904
904 in timer.c
(gdb) print *new_tl
$3 = {next_tl = 0x0, prev_tl = 0x0, ld_tl = 0x0, time_out = 0,
timer_list = 0x0, deleted = 0}
(gdb) print list
$4 = value optimized out
(gdb) print timeout
$5 = 32
(gdb) print new_tl
$6 = (struct timer_link *) 0x7f8b7a54e310
I'll keep the core for a while -- please let me know if there is
anything else I can try.
Thanks,
Kennard
Inactive hide details for Bogdan-Andrei Iancu ---10/08/2010 04:40:47
AM---Hi Kennard, Ok, keep the core next time :)Bogdan-Andrei Iancu
---10/08/2010 04:40:47 AM---Hi Kennard, Ok, keep the core next time :)
From: Bogdan-Andrei Iancu bog...@voice-system.ro
To: OpenSIPS users mailling list users@lists.opensips.org
Date: 10/08/2010 04:40 AM
Subject: Re: [OpenSIPS-Users] opensips tm timer core dump
Sent by: users-boun...@lists.opensips.org
Hi Kennard,
Ok, keep the core next time :)
Regards,
Bogdan
kennard_wh...@logitech.com wrote:
Hi Bogden,
Thanks for explaining the child processes involved -- I misunderstood
what was happening.
Unfortunately, I don't have the core anymore. My recollection is that
I couldn't print anything useful due to compiler optimization. That
said, this should re-create pretty easily, and I'll get more dumps
next time it happens.
Regards,
Kennard
Inactive hide details for Bogdan-Andrei Iancu ---10/05/2010 01:41:38
AM---Hi Kennard, The core was generated by process 22255:Bogdan-Andrei
Iancu
Re: [OpenSIPS-Users] How to t_relay() from two send socket?
Hi CheeWii, Do you want to received the a INVITE request and to send it to two destinations , one on private interface and one on public interface? Did I get it right ? Regards, Bogdan CheeWii wrote: Hi, My OpenSIPS server has two network cards. One is public ip address such as 202.102.XX.XX,and the other is private ip address,such as 10.0.1.5. Now ,I want to forward INVITE ,BYE,CANCEL from 10.0.1.5 to 10.0.1.6,while forward MESSAGE and REGISTER from 202.102.XX.XX to 202.102.YY.YY. I have test use force_send_socket to control the send sockets to different kinds of messages,but it seems to unuseful. It just send message from one socket. How can we accomplish this demo? Thanks a lot. CheeWii ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS Bootcamp 15 - 19 November 2010, Edison, New Jersey, USA www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OpenSIPS core dumps
Any chance with the backtraces ? Regards, Bogdan Anca Vamanu wrote: Hi, You need to inspect them with gdb, run: gdb path_to_opensips_executable path_to_corefile, and then run 'bt full' and send the output. Regards, -- Anca Vamanu www.voice-system.ro On 10/14/2010 10:12 PM, thrillerbee wrote: I have this info from dmesg: [1985853.285221] opensips[30865]: segfault at 10 ip 7f43899ce21f sp 7fff8de1cf40 error 4 in db_flatstore.so[7f43899cb000+5000] [1985856.379671] opensips[30858]: segfault at 10 ip 7f43899ce21f sp 7fff8de1cf40 error 4 in db_flatstore.so[7f43899cb000+5000] [1985896.961279] opensips[30868]: segfault at 10 ip 7f43899ce21f sp 7fff8de1cf40 error 4 in db_flatstore.so[7f43899cb000+5000] [2000131.245512] opensips[17672]: segfault at 10 ip 7fd0f21fb21f sp 7fff3a3b4f00 error 4 in db_flatstore.so[7fd0f21f8000+5000] [2000161.735962] opensips[17668]: segfault at 10 ip 7fd0f21fb21f sp 7fff3a3b4e40 error 4 in db_flatstore.so[7fd0f21f8000+5000] [2000167.299402] opensips[17670]: segfault at 10 ip 7fd0f21fb21f sp 7fff3a3b4f00 error 4 in db_flatstore.so[7fd0f21f8000+5000] On Thu, Oct 14, 2010 at 1:57 PM, thrillerbee thriller...@gmail.com mailto:thriller...@gmail.com wrote: When OpenSIPS crashes, three corefiles are generated that are 2.1GB in size. How do I use these files to understand what's causing the crash? Thanks. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS Bootcamp 15 - 19 November 2010, Edison, New Jersey, USA www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] generate CDRs using sermyadmin
Hi Leon, maybe you should consider using OpenSIPS Control Panel and the CDR procedure provided by this tool... Of course, you can use only the CDR procedure (it is only at mysql level). See: http://opensips-cp.sourceforge.net/htmldoc/cdrviewer.html http://opensips-cp.svn.sourceforge.net/viewvc/opensips-cp/branches/4.0/config/tools/system/cdrviewer/ Regards, Bogdan Leon Li wrote: I found the right procedure is opensips.opensips. however the generate-cdrs.sh ran without creating any cdrs. How do I troubleshoot this? Regards, Leon *From:* users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] *On Behalf Of *Leon Li *Sent:* Friday, 15 October 2010 12:07 PM *To:* OpenSIPS users mailling list *Subject:* [OpenSIPS-Users] generate CDRs using sermyadmin Hi, I knew sermyadmin is discontinue any more sadly. But the question is how to generate CDRs using the generate-cdrs.sh in sermyadmin? I got an error. ERROR 1305 (42000) at line 1: PROCEDURE opensips.sermyadmin does not exist But the mentioned procedure is not included in sermyadmin package? Any assistance? Cheers Leon ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS Bootcamp 15 - 19 November 2010, Edison, New Jersey, USA www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Problem creating tables for OpenSIPS with PostgreSQL
The call to opensipsdbctl create is asking me for the password on every single table created. But the real problem is that after several tables have been created the following error happens: ... ... -e Creating core table: drouting Password for user almira: NOTICE: CREATE TABLE will create implicit sequence dr_gateways_gwid_seq for serial column dr_gateways.gwid NOTICE: CREATE TABLE / PRIMARY KEY will create implicit index dr_gateways_pkey for table dr_gateways NOTICE: CREATE TABLE will create implicit sequence dr_rules_ruleid_seq for serial column dr_rules.ruleid NOTICE: CREATE TABLE / PRIMARY KEY will create implicit index dr_rules_pkey for table dr_rules NOTICE: CREATE TABLE will create implicit sequence dr_gw_lists_id_seq for serial column dr_gw_lists.id NOTICE: CREATE TABLE / PRIMARY KEY will create implicit index dr_gw_lists_pkey for table dr_gw_lists NOTICE: CREATE TABLE will create implicit sequence dr_groups_id_seq for serial column dr_groups.id NOTICE: CREATE TABLE / PRIMARY KEY will create implicit index dr_groups_pkey for table dr_groups -e Creating core table: load_balancer Password for user almira: NOTICE: CREATE TABLE will create implicit sequence load_balancer_id_seq for serial column load_balancer.id NOTICE: CREATE TABLE / PRIMARY KEY will create implicit index load_balancer_pkey for table load_balancer Password for user almira: ERROR: permission denied to create role -e \E[37;32mWARNING: Create user in database failed, perhaps they already exist? Try to continue.. Password for user almira: Password for user almira: Password for user almira: Password for user almira: [: 300: acc: unexpected operator [: 300: acc: unexpected operator Any hint about where to have a look in order to fix this? Regards, David ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] fr_timer fr_inv_timer
Hi all, I setup opensips with the tm module for call forward on timeout. It works great but I would like to perform different actions depending on if it is the fr_timer or the fr_inv_timer, and I can find no way of telling from the script which timer was hit. How could I gain access to this information? I am running 1.6.2 Thanks!!! Jesse ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Request Time out problem
Hi James, As I see, the INVITE has in URI sip:ja...@198.168.0.1 INVITE sip:ja...@198.162.0.1 SIP/2.0 without any port indication, so the default 5060 is assumed. The proxy cannot automatically discover what's the right port on the next hop if not instructed by RURI or if not discovered via DNS SRV (not your case as you use IPs). So, the dialling party is the faulty one (198.162.0.1:9013 ) as it does not provide a correct location of callee. Regards, Bogdan James Mbuthia wrote: Hi guys, Hope you can help me out. I have a situation where my Opensips box is acting as a proxy/registration and location server. I have also changed my listening port from 5060 to 5059. The registration process works fine and my UAS gets a 200 response, its IP is 198.162.0.1:1167. However am getting a Request Timeout problem because once the proxy successfuly authenticates the UAC its forwading the INVITE request to my ip:5060. Since opensips is not configured on that port it gets a 408 Request Time out. Below is a snapshot of the headers from a sip trace. udp:198.162.0.1:5059 - udp:198.162.0.1:7013 SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 198.162.0.1:1167;rport=7013;branch=z9hG4bKynybt From:sip:j...@198.162.0.1;tag=2325231 To:sip:ja...@198.162.0.1 ;tag=5594793ec2abca4d36972b7e2bfde24c.9363 Call-ID:ysu...@198.162.0.1 CSeq: 1 INVITE Proxy-Authenticate: Digest realm=198.162.0.1, nonce=4cb83402000547477b91a54ae9e5a553b9b6b633d830 Server: OpenSIPS (1.6.2-notls (i386/linux)) Content-Length: 0 udp:198.162.0.1:9013 - udp:198.162.0.1:5059 INVITE sip:ja...@198.162.0.1 SIP/2.0 Via: SIP/2.0/UDP 198.162.0.1:1167;rport=1645;branch=z9hG4bKynybt From:sip:j...@198.162.0.1;tag=2325231 To:sip:ja...@198.162.0.1 Call-ID:ysu...@198.162.0.1 CSeq: 3 INVITE Contact: sip:ja...@198.162.0.1 Proxy-Authorization: Digest username=jm21, realm=198.162.0.1, nonce=4cb83402000547477b91a54ae9e5a553b9b6b633d830, uri=sip:ja...@198.162.0.1, response=53a86a6afd0d9fc9a118cb4a304f75d6, algorithm=MD5 Content-type: application/sdp Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE udp:198.162.0.1:5059 - udp:198.162.0.1:5060 INVITE sip:ja...@198.162.0.1 SIP/2.0 Record-Route: sip:198.162.0.1:5059;lr=on Via: SIP/2.0/UDP 198.162.0.1:5059;branch=z9hG4bKa8e9.181ea9f5.0 Via: SIP/2.0/UDP 198.162.0.1:1167;received=198.162.0.1;rport=9013;branch=z9hG4bKynybt From:sip:j...@198.162.0.1;tag=2325231 To:sip:ja...@198.162.0.1 Call-ID:ysu...@198.162.0.1 CSeq: 3 INVITE Contact: sip:ja...@198.162.0.1 Content-type: application/sdp Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE udp:198.162.0.1:5059 - udp:198.162.0.1:9013 SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP 198.162.0.1:1167;rport=9013;branch=z9hG4bKynybt From:sip:j...@198.162.0.1;tag=2325231 To:sip:ja...@198.162.0.1 ;tag=a2fc028f8d02e78ceae2b814452c9bc4-9363 Call-ID:ysu...@198.162.0.1 CSeq: 3 INVITE Server: OpenSIPS (1.6.2-notls (i386/linux)) Content-Length: 0 My question is, why is the proxy routing the request to 198.162.0.1:5060 even after Opensips is not listening on that port? Is it possible to get it to route the request to the UAS i.e 198.162.0.1:1167? ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS Bootcamp 15 - 19 November 2010, Edison, New Jersey, USA www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Static Routes in OpenSIPS proxy
Couldn't the dynamic routing module also be used depending on the prefixes at each individal remote pbx system? On Oct 22, 2010 12:29 PM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Hello Andrea, If you check the default opensips.cfg, you can see that there is a step where only the initial requests are getting there - starting from that point you can implement your static routing using if statements, checking the $rU (request username ) and setting new destination (writing in $rd) Regards, Bogdan andrea wrote: Hi, i'm a beginner of OpenSIPS, i've installed 1.6 version using standard con... Bogdan-Andrei Iancu OpenSIPS Bootcamp 15 - 19 November 2010, Edison, New Jersey, USA www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http:/... ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Dispatcher failover re-enabling with probing
Hi,
what group are you using for your destination (in dispatcher) ? if 0,
use another one :D...there is an issue there...
Regards,
Bogdan
thrillerbee wrote:
I could still use some help on understanding what I'm missing that is
preventing gws from transitioning back into the 'active' state from
'probing'. Currently, I have to babysit this OpenSIPS instance.
Again, to summarize, when dispatcher detects a failure, it puts the gw
into 'probing' state begins sending OPTIONS messages to the failed
gw. Even though the gateway begins replying with 200s, dispatcher
never returns it to an 'active' state. I have to restart OpenSIPS to
return it to an 'active' state.
Here are the relevant pieces of my config script:
# - dispatcher params -
modparam(dispatcher, flags, 2)
modparam(dispatcher, dst_avp, $avp(i:271))
modparam(dispatcher, attrs_avp, $avp(i:272))
modparam(dispatcher, grp_avp, $avp(i:273))
modparam(dispatcher, cnt_avp, $avp(i:274))
modparam(dispatcher, ds_ping_interval, 1)
modparam(dispatcher, ds_probing_threshhold, 32)
modparam(dispatcher, ds_probing_mode, 0)
modparam(dispatcher, options_reply_codes, 501, 403, 200)
failure_route[1]
{
if (t_was_cancelled()) {
exit;
}
if ((t_check_status(408)) (t_local_replied(last)))
{
xlog(L_ERR,Gateway Failure! $ci\n);
ds_mark_dst(p);
t_on_failure(1);
t_relay();
}
}
Again, any assistance would be greatly appreciated.
Thanks.
On Mon, Oct 18, 2010 at 7:31 AM, thrillerbee thriller...@gmail.com
mailto:thriller...@gmail.com wrote:
Anca,
I have configured the ds_probing_threshold parameter which allows
me to adjust when I gw goes from active to probing.
However, my issue is getting the gw back to active. For some
reason, it will never transition back - even with successful 200
OK responses to the OPTIONS messages that are triggered when a gw
goes to probing. The examples below show that I can't even
force it back to active from probing to active with MI commands -
I have to restart OpenSIPS.
Thanks.
On Mon, Oct 18, 2010 at 5:50 AM, Anca Vamanu a...@opensips.org
mailto:a...@opensips.org wrote:
Hi Thrillerbee,
You can try to adjust the time when a gateway state is
changed into probing by setting the ds_probing_threshhold ||
parameter
(http://www.opensips.org/html/docs/modules/devel/dispatcher.html#id250525).
Regards,
--
Anca Vamanu
www.voice-system.ro http://www.voice-system.ro
On 10/16/2010 08:01 AM, thrillerbee wrote:
I have been able to get the dispatcher module to detect a gw
failure and put it into a probing state route traffic
elsewhere. However, when the gw returns ( begins responding
to OPTIONS with 200s), dispatcher never puts it back in the
active state.
In fact, I cannot even manually put the server back in the
active state with mi functions. I can put it in an inactive
state, but it returns to probing if I try to set it to active:
ogw1:~# opensipsctl fifo ds_list
SET_NO:: 1
SET:: 0
URI:: sip:12.121.80.38 flag=P
URI:: sip:12.121.80.39 flag=A
URI:: sip:12.121.80.40 flag=A
ogw1:~# opensipsctl fifo ds_set_state i 0 sip:12.121.80.38
ogw1:~# opensipsctl fifo ds_list
SET_NO:: 1
SET:: 0
URI:: sip:12.121.80.38 flag=I
URI:: sip:12.121.80.39 flag=A
URI:: sip:12.121.80.40 flag=A
ogw1:~# opensipsctl fifo ds_set_state a 0 sip:12.121.80.38
ogw1:~# opensipsctl fifo ds_list
SET_NO:: 1
SET:: 0
URI:: sip:12.121.80.38 flag=P
URI:: sip:12.121.80.39 flag=A
URI:: sip:12.121.80.40 flag=A
Is there some setting that I am missing that allows gateways
to transition from probing to active?
Thanks.
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OpenSIPS Bootcamp
15 - 19 November 2010, Edison, New Jersey, USA
www.voice-system.ro
Re: [OpenSIPS-Users] Request Time out problem
Hi Bogdan, I figured that out after I went through the SIP rfc in more detail. Thanks for your help though. james On Fri, Oct 22, 2010 at 7:04 PM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Hi James, As I see, the INVITE has in URI sip:ja...@198.168.0.1sip%3aja...@198.168.0.1 INVITE sip:ja...@198.162.0.1 sip%3aja...@198.162.0.1 SIP/2.0 without any port indication, so the default 5060 is assumed. The proxy cannot automatically discover what's the right port on the next hop if not instructed by RURI or if not discovered via DNS SRV (not your case as you use IPs). So, the dialling party is the faulty one (198.162.0.1:9013 ) as it does not provide a correct location of callee. Regards, Bogdan James Mbuthia wrote: Hi guys, Hope you can help me out. I have a situation where my Opensips box is acting as a proxy/registration and location server. I have also changed my listening port from 5060 to 5059. The registration process works fine and my UAS gets a 200 response, its IP is 198.162.0.1:1167. However am getting a Request Timeout problem because once the proxy successfuly authenticates the UAC its forwading the INVITE request to my ip:5060. Since opensips is not configured on that port it gets a 408 Request Time out. Below is a snapshot of the headers from a sip trace. udp:198.162.0.1:5059 - udp:198.162.0.1:7013 SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 198.162.0.1:1167;rport=7013;branch=z9hG4bKynybt From:sip:j...@198.162.0.1 sip%3aj...@198.162.0.1;tag=2325231 To:sip:ja...@198.162.0.1 sip%3aja...@198.162.0.1 ;tag=5594793ec2abca4d36972b7e2bfde24c.9363 Call-ID:ysu...@198.162.0.1 call-id%3aysu...@198.162.0.1 CSeq: 1 INVITE Proxy-Authenticate: Digest realm=198.162.0.1, nonce=4cb83402000547477b91a54ae9e5a553b9b6b633d830 Server: OpenSIPS (1.6.2-notls (i386/linux)) Content-Length: 0 udp:198.162.0.1:9013 - udp:198.162.0.1:5059 INVITE sip:ja...@198.162.0.1 sip%3aja...@198.162.0.1 SIP/2.0 Via: SIP/2.0/UDP 198.162.0.1:1167;rport=1645;branch=z9hG4bKynybt From:sip:j...@198.162.0.1 sip%3aj...@198.162.0.1;tag=2325231 To:sip:ja...@198.162.0.1 sip%3aja...@198.162.0.1 Call-ID:ysu...@198.162.0.1 call-id%3aysu...@198.162.0.1 CSeq: 3 INVITE Contact: sip:ja...@198.162.0.1 sip%3aja...@198.162.0.1 Proxy-Authorization: Digest username=jm21, realm=198.162.0.1, nonce=4cb83402000547477b91a54ae9e5a553b9b6b633d830, uri=sip:ja...@198.162.0.1 sip%3aja...@198.162.0.1, response=53a86a6afd0d9fc9a118cb4a304f75d6, algorithm=MD5 Content-type: application/sdp Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE udp:198.162.0.1:5059 - udp:198.162.0.1:5060 INVITE sip:ja...@198.162.0.1 sip%3aja...@198.162.0.1 SIP/2.0 Record-Route: sip:198.162.0.1:5059;lr=on Via: SIP/2.0/UDP 198.162.0.1:5059;branch=z9hG4bKa8e9.181ea9f5.0 Via: SIP/2.0/UDP 198.162.0.1:1167;received=198.162.0.1;rport=9013;branch=z9hG4bKynybt From:sip:j...@198.162.0.1 sip%3aj...@198.162.0.1;tag=2325231 To:sip:ja...@198.162.0.1 sip%3aja...@198.162.0.1 Call-ID:ysu...@198.162.0.1 call-id%3aysu...@198.162.0.1 CSeq: 3 INVITE Contact: sip:ja...@198.162.0.1 sip%3aja...@198.162.0.1 Content-type: application/sdp Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE udp:198.162.0.1:5059 - udp:198.162.0.1:9013 SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP 198.162.0.1:1167;rport=9013;branch=z9hG4bKynybt From:sip:j...@198.162.0.1 sip%3aj...@198.162.0.1;tag=2325231 To:sip:ja...@198.162.0.1 sip%3aja...@198.162.0.1 ;tag=a2fc028f8d02e78ceae2b814452c9bc4-9363 Call-ID:ysu...@198.162.0.1 call-id%3aysu...@198.162.0.1 CSeq: 3 INVITE Server: OpenSIPS (1.6.2-notls (i386/linux)) Content-Length: 0 My question is, why is the proxy routing the request to 198.162.0.1:5060 even after Opensips is not listening on that port? Is it possible to get it to route the request to the UAS i.e 198.162.0.1:1167? ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS Bootcamp 15 - 19 November 2010, Edison, New Jersey, USA www.voice-system.ro ___ 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] ERROR:tm:t_forward_nonack: no branch for forwarding
Hi Najib, I guess you are using an opensips prior to 1.6 (by looking at the error messages)...As the message says , the t_relay() has no valid new destination where to send the request Check for prior error - maybe the RURI was not valid, so the destination you set was discarded - no valid destination Regards, Bogdan Najib Hara wrote: Hi everybody, I don't know if this problem is already solved but I can't find any related information. I'm trying to relay requests through OpenSIPS to another OpenSIPS server together on the same machine. And I got this each time: Oct 18 11:53:00 Opensips opensips1[17997]: ERROR:tm:t_forward_nonack: no branch for forwarding Oct 18 11:53:00 Opensips opensips1[17997]: ERROR:tm:w_t_relay: t_forward_nonack failed .. Oct 18 11:53:00 Opensips opensips1[17997]: CRITICAL:tm:t_should_relay_response: pick_branch failed (lowest==-1) for code 488 My t_relay function is used on the request route. Anyone could help me with this ? Thanks in advance ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS Bootcamp 15 - 19 November 2010, Edison, New Jersey, USA www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Static Routes in OpenSIPS proxy
Of course it did, but the subject was static routingnot dynamic :)... Regards, Bogdan Duane Larson wrote: Couldn't the dynamic routing module also be used depending on the prefixes at each individal remote pbx system? On Oct 22, 2010 12:29 PM, Bogdan-Andrei Iancu bog...@voice-system.ro mailto:bog...@voice-system.ro wrote: Hello Andrea, If you check the default opensips.cfg, you can see that there is a step where only the initial requests are getting there - starting from that point you can implement your static routing using if statements, checking the $rU (request username ) and setting new destination (writing in $rd) Regards, Bogdan andrea wrote: Hi, i'm a beginner of OpenSIPS, i've installed 1.6 version using standard con... Bogdan-Andrei Iancu OpenSIPS Bootcamp 15 - 19 November 2010, Edison, New Jersey, USA www.voice-system.ro http://www.voice-system.ro ___ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org http:/... ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS Bootcamp 15 - 19 November 2010, Edison, New Jersey, USA www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Request Time out problem
OK, cool James Mbuthia wrote: Hi Bogdan, I figured that out after I went through the SIP rfc in more detail. Thanks for your help though. james On Fri, Oct 22, 2010 at 7:04 PM, Bogdan-Andrei Iancu bog...@voice-system.ro mailto:bog...@voice-system.ro wrote: Hi James, As I see, the INVITE has in URI sip:ja...@198.168.0.1 mailto:sip%3aja...@198.168.0.1 INVITE sip:ja...@198.162.0.1 mailto:sip%3aja...@198.162.0.1 SIP/2.0 without any port indication, so the default 5060 is assumed. The proxy cannot automatically discover what's the right port on the next hop if not instructed by RURI or if not discovered via DNS SRV (not your case as you use IPs). So, the dialling party is the faulty one (198.162.0.1:9013 http://198.162.0.1:9013 ) as it does not provide a correct location of callee. Regards, Bogdan James Mbuthia wrote: Hi guys, Hope you can help me out. I have a situation where my Opensips box is acting as a proxy/registration and location server. I have also changed my listening port from 5060 to 5059. The registration process works fine and my UAS gets a 200 response, its IP is 198.162.0.1:1167 http://198.162.0.1:1167. However am getting a Request Timeout problem because once the proxy successfuly authenticates the UAC its forwading the INVITE request to my ip:5060. Since opensips is not configured on that port it gets a 408 Request Time out. Below is a snapshot of the headers from a sip trace. udp:198.162.0.1:5059 http://198.162.0.1:5059 - udp:198.162.0.1:7013 http://198.162.0.1:7013 SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 198.162.0.1:1167;rport=7013;branch=z9hG4bKynybt From:sip:j...@198.162.0.1 mailto:sip%3aj...@198.162.0.1;tag=2325231 To:sip:ja...@198.162.0.1 mailto:sip%3aja...@198.162.0.1 ;tag=5594793ec2abca4d36972b7e2bfde24c.9363 Call-ID:ysu...@198.162.0.1 mailto:call-id%3aysu...@198.162.0.1 CSeq: 1 INVITE Proxy-Authenticate: Digest realm=198.162.0.1, nonce=4cb83402000547477b91a54ae9e5a553b9b6b633d830 Server: OpenSIPS (1.6.2-notls (i386/linux)) Content-Length: 0 udp:198.162.0.1:9013 http://198.162.0.1:9013 - udp:198.162.0.1:5059 http://198.162.0.1:5059 INVITE sip:ja...@198.162.0.1 mailto:sip%3aja...@198.162.0.1 SIP/2.0 Via: SIP/2.0/UDP 198.162.0.1:1167;rport=1645;branch=z9hG4bKynybt From:sip:j...@198.162.0.1 mailto:sip%3aj...@198.162.0.1;tag=2325231 To:sip:ja...@198.162.0.1 mailto:sip%3aja...@198.162.0.1 Call-ID:ysu...@198.162.0.1 mailto:call-id%3aysu...@198.162.0.1 CSeq: 3 INVITE Contact: sip:ja...@198.162.0.1 mailto:sip%3aja...@198.162.0.1 Proxy-Authorization: Digest username=jm21, realm=198.162.0.1, nonce=4cb83402000547477b91a54ae9e5a553b9b6b633d830, uri=sip:ja...@198.162.0.1 mailto:sip%3aja...@198.162.0.1, response=53a86a6afd0d9fc9a118cb4a304f75d6, algorithm=MD5 Content-type: application/sdp Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE udp:198.162.0.1:5059 http://198.162.0.1:5059 - udp:198.162.0.1:5060 http://198.162.0.1:5060 INVITE sip:ja...@198.162.0.1 mailto:sip%3aja...@198.162.0.1 SIP/2.0 Record-Route: sip:198.162.0.1:5059;lr=on Via: SIP/2.0/UDP 198.162.0.1:5059;branch=z9hG4bKa8e9.181ea9f5.0 Via: SIP/2.0/UDP 198.162.0.1:1167;received=198.162.0.1;rport=9013;branch=z9hG4bKynybt From:sip:j...@198.162.0.1 mailto:sip%3aj...@198.162.0.1;tag=2325231 To:sip:ja...@198.162.0.1 mailto:sip%3aja...@198.162.0.1 Call-ID:ysu...@198.162.0.1 mailto:call-id%3aysu...@198.162.0.1 CSeq: 3 INVITE Contact: sip:ja...@198.162.0.1 mailto:sip%3aja...@198.162.0.1 Content-type: application/sdp Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE udp:198.162.0.1:5059 http://198.162.0.1:5059 - udp:198.162.0.1:9013 http://198.162.0.1:9013 SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP 198.162.0.1:1167;rport=9013;branch=z9hG4bKynybt From:sip:j...@198.162.0.1 mailto:sip%3aj...@198.162.0.1;tag=2325231 To:sip:ja...@198.162.0.1 mailto:sip%3aja...@198.162.0.1 ;tag=a2fc028f8d02e78ceae2b814452c9bc4-9363 Call-ID:ysu...@198.162.0.1 mailto:call-id%3aysu...@198.162.0.1 CSeq: 3 INVITE Server: OpenSIPS (1.6.2-notls (i386/linux)) Content-Length: 0 My question is, why is the proxy routing the request to 198.162.0.1:5060 http://198.162.0.1:5060 even after Opensips is not listening on that port? Is it possible to get it to route the request to the UAS i.e 198.162.0.1:1167 http://198.162.0.1:1167?
Re: [OpenSIPS-Users] Static Routes in OpenSIPS proxy
Very true. You got me there ;) On Oct 22, 2010 1:12 PM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Of course it did, but the subject was static routingnot dynamic :)... Regards, Bogdan Duane Larson wrote: Couldn't the dynamic routing module also be used depending on the prefix... bog...@voice-system.ro mailto:bog...@voice-system.ro wrote: Hello Andrea, If you ... www.voice-system.ro http://www.voice-system.ro ___ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org http:/... ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS Bootcamp 15 - 19 November 2010, Edison, New Jersey, USA www.voice-syste... 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] Dispatcher failover re-enabling with probing
Thank you!
I'll give that a shot report back.
Ryan
On Fri, Oct 22, 2010 at 12:07 PM, Bogdan-Andrei Iancu
bog...@voice-system.ro wrote:
Hi,
what group are you using for your destination (in dispatcher) ? if 0,
use another one :D...there is an issue there...
Regards,
Bogdan
thrillerbee wrote:
I could still use some help on understanding what I'm missing that is
preventing gws from transitioning back into the 'active' state from
'probing'. Currently, I have to babysit this OpenSIPS instance.
Again, to summarize, when dispatcher detects a failure, it puts the gw
into 'probing' state begins sending OPTIONS messages to the failed
gw. Even though the gateway begins replying with 200s, dispatcher
never returns it to an 'active' state. I have to restart OpenSIPS to
return it to an 'active' state.
Here are the relevant pieces of my config script:
# - dispatcher params -
modparam(dispatcher, flags, 2)
modparam(dispatcher, dst_avp, $avp(i:271))
modparam(dispatcher, attrs_avp, $avp(i:272))
modparam(dispatcher, grp_avp, $avp(i:273))
modparam(dispatcher, cnt_avp, $avp(i:274))
modparam(dispatcher, ds_ping_interval, 1)
modparam(dispatcher, ds_probing_threshhold, 32)
modparam(dispatcher, ds_probing_mode, 0)
modparam(dispatcher, options_reply_codes, 501, 403, 200)
failure_route[1]
{
if (t_was_cancelled()) {
exit;
}
if ((t_check_status(408)) (t_local_replied(last)))
{
xlog(L_ERR,Gateway Failure! $ci\n);
ds_mark_dst(p);
t_on_failure(1);
t_relay();
}
}
Again, any assistance would be greatly appreciated.
Thanks.
On Mon, Oct 18, 2010 at 7:31 AM, thrillerbee thriller...@gmail.com
mailto:thriller...@gmail.com wrote:
Anca,
I have configured the ds_probing_threshold parameter which allows
me to adjust when I gw goes from active to probing.
However, my issue is getting the gw back to active. For some
reason, it will never transition back - even with successful 200
OK responses to the OPTIONS messages that are triggered when a gw
goes to probing. The examples below show that I can't even
force it back to active from probing to active with MI commands -
I have to restart OpenSIPS.
Thanks.
On Mon, Oct 18, 2010 at 5:50 AM, Anca Vamanu a...@opensips.org
mailto:a...@opensips.org wrote:
Hi Thrillerbee,
You can try to adjust the time when a gateway state is
changed into probing by setting the ds_probing_threshhold ||
parameter
(
http://www.opensips.org/html/docs/modules/devel/dispatcher.html#id250525).
Regards,
--
Anca Vamanu
www.voice-system.ro http://www.voice-system.ro
On 10/16/2010 08:01 AM, thrillerbee wrote:
I have been able to get the dispatcher module to detect a gw
failure and put it into a probing state route traffic
elsewhere. However, when the gw returns ( begins responding
to OPTIONS with 200s), dispatcher never puts it back in the
active state.
In fact, I cannot even manually put the server back in the
active state with mi functions. I can put it in an inactive
state, but it returns to probing if I try to set it to active:
ogw1:~# opensipsctl fifo ds_list
SET_NO:: 1
SET:: 0
URI:: sip:12.121.80.38 flag=P
URI:: sip:12.121.80.39 flag=A
URI:: sip:12.121.80.40 flag=A
ogw1:~# opensipsctl fifo ds_set_state i 0 sip:12.121.80.38
ogw1:~# opensipsctl fifo ds_list
SET_NO:: 1
SET:: 0
URI:: sip:12.121.80.38 flag=I
URI:: sip:12.121.80.39 flag=A
URI:: sip:12.121.80.40 flag=A
ogw1:~# opensipsctl fifo ds_set_state a 0 sip:12.121.80.38
ogw1:~# opensipsctl fifo ds_list
SET_NO:: 1
SET:: 0
URI:: sip:12.121.80.38 flag=P
URI:: sip:12.121.80.39 flag=A
URI:: sip:12.121.80.40 flag=A
Is there some setting that I am missing that allows gateways
to transition from probing to active?
Thanks.
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Re: [OpenSIPS-Users] opensips tm timer core dump
Hi Bogdan,
I believe I found the problem. When sip_msg_cloner() within build_cell()
fails due to out-of-mem, and dangling pointer to the cell is left in the
global transaction pointer. Later on the post_cb() code attempts to clean
this up, and resurrects the now-free memory, and in particular puts it on
a wait timer. My guess is that later on this memory is allocated into a new
transaction which eventually uses the same wait timer link, and the two
threads fight it out.
Adding a set_t(0) fixes the problem. While looking, I believe I found a
memory leak in uac creation code that is also triggered by out of memory
conditions. Please see attached patch.
The problem also manifest itself as a core dump with insert_timer_unsafe.
Once I disabled optimization, I saw multiple variations within this
function. The most common was that ptr==tl after the search. The sequence
of events for this is: new cell created, stored into global T, free'd,
added to wait-list by cleanup code from global T, memory re-allocated into
new cell, timer link zero'd, and then added again to wait list. At least
that is my best guess.
Regards,
Kennard
(See attached file: opensips-tm-cell.patch)
From: Bogdan-Andrei Iancu bog...@voice-system.ro
To: OpenSIPS users mailling list users@lists.opensips.org
Date: 10/22/2010 07:02 AM
Subject:Re: [OpenSIPS-Users] opensips tm timer core dump
Sent by:users-boun...@lists.opensips.org
Hi Kennard,
I suppose the bt is the same ? do you still have the core file ?
Regards,
Bogdan
kennard_wh...@logitech.com wrote:
Hi Bodgen,
I replicated the error. Unfortunately the entire insert_timer_unsafe
and been in-lined and little is available:
Program terminated with signal 11, Segmentation fault.
#0 0x7f8b8356c2c2 in insert_timer_unsafe (new_tl=0x7f8b7a54e310,
list_id=WT_TIMER_LIST, ext_timeout=value optimized out) at timer.c:731
731 timer.c: No such file or directory.
in timer.c
(gdb) print tl
$1 = value optimized out
(gdb) print *tl
Cannot access memory at address 0x0
(gdb) print ptr
$2 = value optimized out
(gdb) print *ptr
Cannot access memory at address 0x0
(gdb) print *new_tl
No symbol new_tl in current context.
(gdb) up
#1 set_1timer (new_tl=0x7f8b7a54e310, list_id=WT_TIMER_LIST,
ext_timeout=value optimized out) at timer.c:904
904 in timer.c
(gdb) print *new_tl
$3 = {next_tl = 0x0, prev_tl = 0x0, ld_tl = 0x0, time_out = 0,
timer_list = 0x0, deleted = 0}
(gdb) print list
$4 = value optimized out
(gdb) print timeout
$5 = 32
(gdb) print new_tl
$6 = (struct timer_link *) 0x7f8b7a54e310
I'll keep the core for a while -- please let me know if there is
anything else I can try.
Thanks,
Kennard
Inactive hide details for Bogdan-Andrei Iancu ---10/08/2010 04:40:47
AM---Hi Kennard, Ok, keep the core next time :)Bogdan-Andrei Iancu
---10/08/2010 04:40:47 AM---Hi Kennard, Ok, keep the core next time :)
From: Bogdan-Andrei Iancu bog...@voice-system.ro
To: OpenSIPS users mailling list users@lists.opensips.org
Date: 10/08/2010 04:40 AM
Subject: Re: [OpenSIPS-Users] opensips tm timer core dump
Sent by: users-boun...@lists.opensips.org
Hi Kennard,
Ok, keep the core next time :)
Regards,
Bogdan
kennard_wh...@logitech.com wrote:
Hi Bogden,
Thanks for explaining the child processes involved -- I misunderstood
what was happening.
Unfortunately, I don't have the core anymore. My recollection is that
I couldn't print anything useful due to compiler optimization. That
said, this should re-create pretty easily, and I'll get more dumps
next time it happens.
Regards,
Kennard
Inactive hide details for Bogdan-Andrei Iancu ---10/05/2010 01:41:38
AM---Hi Kennard, The core was generated by process 22255:Bogdan-Andrei
Iancu ---10/05/2010 01:41:38 AM---Hi Kennard, The core was generated
by process 22255:
From: Bogdan-Andrei Iancu bog...@voice-system.ro
To: OpenSIPS users mailling list users@lists.opensips.org
Date: 10/05/2010 01:41 AM
Subject: Re: [OpenSIPS-Users] opensips tm timer core dump
Sent by: users-boun...@lists.opensips.org
Hi Kennard,
The core was generated by process 22255:
[22238]: INFO:core:handle_sigs: child process 22255 exited by a
signal 11
and this process also reported mem problems:
[22255]: ERROR:tm:new_t: out of mem
Can you print the tl or ptr variables in frame 0?
Regards,
Bogdan
kennard_wh...@logitech.com wrote:
Running against opensips HEAD, I got a segfault in the tm timer code.
I believe this is triggered by running out of shared memory.
The stack trace:
(gdb) where
#0 0x7fe8f8d96212 in insert_timer_unsafe (new_tl=0x7fe8f66337b0,
list_id=WT_TIMER_LIST, ext_timeout=value optimized out) at
timer.c:731
#1 set_1timer (new_tl=0x7fe8f66337b0,
Re: [OpenSIPS-Users] Dispatcher failover re-enabling with probing
Bogan,
That resolved it. Thanks for the advice.
Ryan
On Fri, Oct 22, 2010 at 12:32 PM, thrillerbee thriller...@gmail.com wrote:
Thank you!
I'll give that a shot report back.
Ryan
On Fri, Oct 22, 2010 at 12:07 PM, Bogdan-Andrei Iancu
bog...@voice-system.ro wrote:
Hi,
what group are you using for your destination (in dispatcher) ? if 0,
use another one :D...there is an issue there...
Regards,
Bogdan
thrillerbee wrote:
I could still use some help on understanding what I'm missing that is
preventing gws from transitioning back into the 'active' state from
'probing'. Currently, I have to babysit this OpenSIPS instance.
Again, to summarize, when dispatcher detects a failure, it puts the gw
into 'probing' state begins sending OPTIONS messages to the failed
gw. Even though the gateway begins replying with 200s, dispatcher
never returns it to an 'active' state. I have to restart OpenSIPS to
return it to an 'active' state.
Here are the relevant pieces of my config script:
# - dispatcher params -
modparam(dispatcher, flags, 2)
modparam(dispatcher, dst_avp, $avp(i:271))
modparam(dispatcher, attrs_avp, $avp(i:272))
modparam(dispatcher, grp_avp, $avp(i:273))
modparam(dispatcher, cnt_avp, $avp(i:274))
modparam(dispatcher, ds_ping_interval, 1)
modparam(dispatcher, ds_probing_threshhold, 32)
modparam(dispatcher, ds_probing_mode, 0)
modparam(dispatcher, options_reply_codes, 501, 403, 200)
failure_route[1]
{
if (t_was_cancelled()) {
exit;
}
if ((t_check_status(408)) (t_local_replied(last)))
{
xlog(L_ERR,Gateway Failure! $ci\n);
ds_mark_dst(p);
t_on_failure(1);
t_relay();
}
}
Again, any assistance would be greatly appreciated.
Thanks.
On Mon, Oct 18, 2010 at 7:31 AM, thrillerbee thriller...@gmail.com
mailto:thriller...@gmail.com wrote:
Anca,
I have configured the ds_probing_threshold parameter which allows
me to adjust when I gw goes from active to probing.
However, my issue is getting the gw back to active. For some
reason, it will never transition back - even with successful 200
OK responses to the OPTIONS messages that are triggered when a gw
goes to probing. The examples below show that I can't even
force it back to active from probing to active with MI commands -
I have to restart OpenSIPS.
Thanks.
On Mon, Oct 18, 2010 at 5:50 AM, Anca Vamanu a...@opensips.org
mailto:a...@opensips.org wrote:
Hi Thrillerbee,
You can try to adjust the time when a gateway state is
changed into probing by setting the ds_probing_threshhold ||
parameter
(
http://www.opensips.org/html/docs/modules/devel/dispatcher.html#id250525
).
Regards,
--
Anca Vamanu
www.voice-system.ro http://www.voice-system.ro
On 10/16/2010 08:01 AM, thrillerbee wrote:
I have been able to get the dispatcher module to detect a gw
failure and put it into a probing state route traffic
elsewhere. However, when the gw returns ( begins responding
to OPTIONS with 200s), dispatcher never puts it back in the
active state.
In fact, I cannot even manually put the server back in the
active state with mi functions. I can put it in an inactive
state, but it returns to probing if I try to set it to active:
ogw1:~# opensipsctl fifo ds_list
SET_NO:: 1
SET:: 0
URI:: sip:12.121.80.38 flag=P
URI:: sip:12.121.80.39 flag=A
URI:: sip:12.121.80.40 flag=A
ogw1:~# opensipsctl fifo ds_set_state i 0 sip:12.121.80.38
ogw1:~# opensipsctl fifo ds_list
SET_NO:: 1
SET:: 0
URI:: sip:12.121.80.38 flag=I
URI:: sip:12.121.80.39 flag=A
URI:: sip:12.121.80.40 flag=A
ogw1:~# opensipsctl fifo ds_set_state a 0 sip:12.121.80.38
ogw1:~# opensipsctl fifo ds_list
SET_NO:: 1
SET:: 0
URI:: sip:12.121.80.38 flag=P
URI:: sip:12.121.80.39 flag=A
URI:: sip:12.121.80.40 flag=A
Is there some setting that I am missing that allows gateways
to transition from probing to active?
Thanks.
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Re: [OpenSIPS-Users] How to t_relay() from two send socket?
Hi,Bogdan No,I want to received a INVITE request and a MESSAGE request, then I send MESSAGE on public interface and send INVITE on private interface.How can I accomplish this? Regards, CheeWii 2010/10/23 Bogdan-Andrei Iancu bog...@voice-system.ro Hi CheeWii, Do you want to received the a INVITE request and to send it to two destinations , one on private interface and one on public interface? Did I get it right ? Regards, Bogdan CheeWii wrote: Hi, My OpenSIPS server has two network cards. One is public ip address such as 202.102.XX.XX,and the other is private ip address,such as 10.0.1.5. Now ,I want to forward INVITE ,BYE,CANCEL from 10.0.1.5 to 10.0.1.6,while forward MESSAGE and REGISTER from 202.102.XX.XX to 202.102.YY.YY. I have test use force_send_socket to control the send sockets to different kinds of messages,but it seems to unuseful. It just send message from one socket. How can we accomplish this demo? Thanks a lot. CheeWii ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS Bootcamp 15 - 19 November 2010, Edison, New Jersey, USA www.voice-system.ro ___ 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
