Re: [OpenSIPS-Users] async radius problem
Actually this is a characteristic of the async engine, not only the
AAA_Radius module. I check to see where is the best place to specify
this info.
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 01/19/2017 10:40 AM, Dragomir Haralambiev wrote:
Hi,
Thanks for your quick replay.
Please add this information in AAA_Radius module.
Best regards,
Dragomir
2017-01-19 10:25 GMT+02:00 Răzvan Crainea <raz...@opensips.org
<mailto:raz...@opensips.org>>:
Hi, Dragomir!
Currently async operations are only available for requests, not
for replies. In the onreply_route you can only use synchronous
operations for now.
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com <http://www.opensips-solutions.com>
On 01/18/2017 03:45 PM, Dragomir Haralambiev wrote:
Hello,
I have problem with async radius implementation.
OS - CEntOs 7
Opensips 2.2.2 - git hub from 15.01.2017
Here pasrt of srcipt:
onreply_route[outgoing] {
.
if (t_check_status("200")) {
async( radius_send_auth("prepayout","prepayin"), return_prepay );
}
}
route[return_prepay] {
xlog("L_ERR", "Radius return $rc");
} # end route return_prepay
Here is part of Opensips log level 6:
/usr/sbin/opensips[21026]: DBG:tm:update_totag_set: new totag
/usr/sbin/opensips[21026]: DBG:tm:insert_timer_unsafe: [2]:
0x7fee75f19bd0 (36)
/usr/sbin/opensips[21026]: DBG:tm:run_trans_callbacks:
trans=0x7fee75f19b50, callback type 64, id 0 entered
/usr/sbin/opensips[21005]: DBG:core:handle_sigs: status = 11
/usr/sbin/opensips[21005]: INFO:core:handle_sigs: child process
21026 exited by a signal 11
/usr/sbin/opensips[21005]: INFO:core:handle_sigs: core was not
generated
/usr/sbin/opensips[21005]: INFO:core:handle_sigs: terminating
due to SIGCHLD
/usr/sbin/opensips[21024]: INFO:core:sig_usr: signal 15 received
/usr/sbin/opensips[21011]: INFO:core:sig_usr: signal 15 received
/usr/sbin/opensips[21018]: INFO:core:sig_usr: signal 15 received
Where is problem?
Best regards,
Dragomir
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Re: [OpenSIPS-Users] OpenSIPs 2.2.2 warnings during start service
When starting opensips, is there any opensips process that is using more than 80% of a core? If so, can you pinpoint the PID in the opensipsctl ps command? Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 01/18/2017 11:55 PM, Ramachandran, Agalya (Contractor) wrote: Same with my case too. Regards, Agalya *From:*Users [mailto:users-boun...@lists.opensips.org] *On Behalf Of *Ahmed Munir *Sent:* Wednesday, January 18, 2017 1:31 PM *To:* OpenSIPs Users <users@lists.opensips.org> *Subject:* [OpenSIPS-Users] OpenSIPs 2.2.2 warnings during start service Hi, I'm currently seeing the warnings when I start opensips service; Jan 18 13:04:35 qorblpsisprxyd1 /usr/sbin/opensips[3089]: WARNING:core:handle_timer_job: timer job has a 150 us delay in execution Jan 18 13:04:35 qorblpsisprxyd1 /usr/sbin/opensips[3089]: WARNING:core:handle_timer_job: timer job has a 150 us delay in execution Jan 18 13:04:35 qorblpsisprxyd1 /usr/sbin/opensips[3089]: WARNING:core:handle_timer_job: timer job has a 150 us delay in execution Jan 18 13:04:35 qorblpsisprxyd1 /usr/sbin/opensips[3089]: WARNING:core:handle_timer_job: utimer job has a 229 us delay in execution Jan 18 13:04:35 qorblpsisprxyd1 /usr/sbin/opensips[3089]: INFO:core:do_action: max while loops are encountered Jan 18 13:04:35 qorblpsisprxyd1 /usr/sbin/opensips[3088]: WARNING:core:utimer_ticker: utimer task already scheduled for 190 ms (now 2470 ms), it may over lap.. I've tried to update the source code for timer.c (line#: 190) ref: https://github.com/OpenSIPS/opensips/commit/fd8f6ec442b4365da9d274af6939954246ece865?diff=split, but didn't work at all. Currently running 8 child processors, see below; [root@qorblpsisprxyd1 ]# opensips -V version: opensips 2.2.2 (x86_64/linux) flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. main.c compiled on 12:39:45 Jan 18 2017 with gcc 4.4.7 [root@qorblpsisprxyd1 ]# opensipsctl fifo ps Process:: ID=0 PID=3083 Type=attendant Process:: ID=1 PID=3085 Type=MI FIFO Process:: ID=2 PID=3086 Type=time_keeper Process:: ID=3 PID=3088 Type=timer Process:: ID=4 PID=3089 Type=SIP receiver udp:10.3.120.94:5060 <http://10.3.120.94:5060> Process:: ID=5 PID=3091 Type=SIP receiver udp:10.3.120.94:5060 <http://10.3.120.94:5060> Process:: ID=6 PID=3092 Type=SIP receiver udp:10.3.120.94:5060 <http://10.3.120.94:5060> Process:: ID=7 PID=3094 Type=SIP receiver udp:10.3.120.94:5060 <http://10.3.120.94:5060> Process:: ID=8 PID=3096 Type=SIP receiver udp:10.3.120.94:5060 <http://10.3.120.94:5060> Process:: ID=9 PID=3098 Type=SIP receiver udp:10.3.120.94:5060 <http://10.3.120.94:5060> Process:: ID=10 PID=3100 Type=SIP receiver udp:10.3.120.94:5060 <http://10.3.120.94:5060> Process:: ID=11 PID=3102 Type=SIP receiver udp:10.3.120.94:5060 <http://10.3.120.94:5060> Process:: ID=12 PID=3104 Type=Timer handler I would like to know what changes required to fix this change? Please advise. -- Regards, Ahmed Munir Chohan ___ 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] async- issues- 2.2.2
You have my answer inline.
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 01/18/2017 08:17 PM, Ramachandran, Agalya (Contractor) wrote:
Hi Razvan,
Got your point.I tried to increase the MAX_CONTENT_TYPE_LEN and
tested async call and it is working fine without crash.
But one more question.
The same piece of code am using for *sync REST_API* query too in
rest_put() method. There also print_buff is only being used.
It is working absolutely fine in the case of sync call even if the
MAX_CONTENT_TYPE_LEN – is 64.
Wondering what would be the reason in sync call, it is working and in
async it is not?
Pure luck. You are doing a buffer overflow, overwriting the data
section. Probably when doing sync calls, nobody is using the data you
are overwriting. But if it doesn't crash it doesn't mean it is ok :).
Regards,
Agalya
*From:*Users [mailto:users-boun...@lists.opensips.org] *On Behalf Of
*Razvan Crainea
*Sent:* Wednesday, January 18, 2017 4:29 AM
*To:* users@lists.opensips.org
*Subject:* Re: [OpenSIPS-Users] async- issues- 2.2.2
Hi, Ramachandran!
The print_buff buffer is declared with length MAX_CONTENT_TYPE_LEN ->
64. Writing more than 64 bytes will lead to a buffer overflow,
probably followed by a crash.
If you want to suppor longer CallIDs, just increase the size of the
buffer, or allocate the buffer with a size large enought to fit your
callid length.
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com <http://www.opensips-solutions.com>
On 01/17/2017 11:13 PM, Ramachandran, Agalya (Contractor) wrote:
Hi Liviu,
Found the exact issue what causes the crash.
In start_async_http_req, under case PUT/POST, we have the
following code blue color by default.
But for my project scenario, I need to add call-id header too
here, including that code in red color for your reference.
if (req_ctype) {
sprintf(print_buff, "Content-Type: %s", req_ctype);
header_list = curl_slist_append(header_list, print_buff);
sprintf(print_buff, "Call-Id: %s", instanceId);
header_list = curl_slist_append(header_list, print_buff);
w_curl_easy_setopt(handle, CURLOPT_HTTPHEADER, header_list);
}
If I try by removing the Call-Id header in the curl-slist, then it
works perfectly fine in case of POST as well as PUT.
How can I overcome this situation? Your guidance would help me a lot.
Regards,
Agalya
*From:* Users [mailto:users-boun...@lists.opensips.org] *On Behalf
Of *Ramachandran, Agalya (Contractor)
*Sent:* Tuesday, January 17, 2017 3:58 PM
*To:* OpenSIPS users mailling list <users@lists.opensips.org>
<mailto:users@lists.opensips.org>; Liviu Chircu
<li...@opensips.org> <mailto:li...@opensips.org>
*Subject:* Re: [OpenSIPS-Users] async- issues- 2.2.2
Hi,
Another information may help you to find this issue.
Am changing the “req_body” in the rest_methods.c, as per the REST
API server is expecting the payload value.
If I change this “req_body” in *REST_POST* as well, and if the
*Call-Id length is > 56*, it crashes in *the POST call too*.
Regards,
Agalya
*From:* Users [mailto:users-boun...@lists.opensips.org] *On Behalf
Of *Ramachandran, Agalya (Contractor)
*Sent:* Tuesday, January 17, 2017 3:03 PM
*To:* OpenSIPS users mailling list <users@lists.opensips.org
<mailto:users@lists.opensips.org>>; Liviu Chircu
<li...@opensips.org <mailto:li...@opensips.org>>
*Subject:* [OpenSIPS-Users] async- issues- 2.2.2
Hi Liviu/team,
When I try to do an asynchronous REST_PUT call in OpenSIPS 2.2.2,
I see the below observance and issues.
All the calls are made from sipp client.
*No*
*Test Scenario *
*Result/observation*
1.
Call-Id length < 50
Works perfectly fine
2
Call-Id length >54 and < 56
No crash observed. But error in curl_multiperform
3.
Call-Id length > 56
Opensips crashes at liburl.
Please find the details for test 2 and test 3.
I have tested with REST_POST with case 3, I didn’t observe any
crashes.
Please let me know what could cause this issue and how can I fix this?
*Test2:*
**
Jan 17 18:23:13 /usr/local/sbin/opensips[18554]:
ERROR:rest_client:start_async_http_req: curl_multi_perform:
Invalid multi handle
Jan 17 18:23:13 /usr/local/sbin/opensips[18554]:
ERROR:rest_client:start_async_http_req: curl_multi_remove_handle:
Invalid multi handle
*Test 3:*
(gdb) bt
#0 0x7f370bccb9bb in curl_multi_add_handle () from
/lib64/libcurl.so.4
#1 0x7f370bf05521 in start_async_http_req
(msg=msg@entry=0x7f374db14270, method=method@entry=REST_CLIENT_PUT,
url=0x7f374dad8
Re: [OpenSIPS-Users] async radius problem
Hi, Dragomir!
Currently async operations are only available for requests, not for
replies. In the onreply_route you can only use synchronous operations
for now.
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 01/18/2017 03:45 PM, Dragomir Haralambiev wrote:
Hello,
I have problem with async radius implementation.
OS - CEntOs 7
Opensips 2.2.2 - git hub from 15.01.2017
Here pasrt of srcipt:
onreply_route[outgoing] {
.
if (t_check_status("200")) {
async( radius_send_auth("prepayout","prepayin"), return_prepay );
}
}
route[return_prepay] {
xlog("L_ERR", "Radius return $rc");
} # end route return_prepay
Here is part of Opensips log level 6:
/usr/sbin/opensips[21026]: DBG:tm:update_totag_set: new totag
/usr/sbin/opensips[21026]: DBG:tm:insert_timer_unsafe: [2]:
0x7fee75f19bd0 (36)
/usr/sbin/opensips[21026]: DBG:tm:run_trans_callbacks:
trans=0x7fee75f19b50, callback type 64, id 0 entered
/usr/sbin/opensips[21005]: DBG:core:handle_sigs: status = 11
/usr/sbin/opensips[21005]: INFO:core:handle_sigs: child process 21026
exited by a signal 11
/usr/sbin/opensips[21005]: INFO:core:handle_sigs: core was not generated
/usr/sbin/opensips[21005]: INFO:core:handle_sigs: terminating due to
SIGCHLD
/usr/sbin/opensips[21024]: INFO:core:sig_usr: signal 15 received
/usr/sbin/opensips[21011]: INFO:core:sig_usr: signal 15 received
/usr/sbin/opensips[21018]: INFO:core:sig_usr: signal 15 received
Where is problem?
Best regards,
Dragomir
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Re: [OpenSIPS-Users] does OpenSIPS Control Panel work?
Hi, Robert! Yes, OpenSIPS Control Panel should work without problems. I double checked the line you get that error[1], but it doesn't seem anything wrong over there. Can you post your file on pastebin so I can check your version? [1] https://github.com/OpenSIPS/opensips-cp/blob/6.2/web/tools/admin/list_admins/template/list_admins.main.php#L152 Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 01/17/2017 09:15 PM, robert mundkowsky via Users wrote: I installed https://github.com/OpenSIPS/opensips-cp/archive/6.2.zip And I am getting a parse errors for list_admins.main.php which I assume means there is an error in the php code. Below is the error log, but note the line number might be a little different from source code because I tried adding error log messages, but forgot these will nto work, because the code will not even parse. PHP Parse error: syntax error, unexpected '}' in /var/www/opensips-cp/web/tools/admin/list_admins/template/list_admins.main.php on line 152 ___ 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] Registrar with IP authentication - selecting variables from DB
You are probably using an old version of OpenSIPS, that's why you are
not seeing negative return values.
Checking if the IP might still be ok, if your database is consistent and
has an IP for every user. So for now I would go with this solution.
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 01/17/2017 11:03 AM, maatohewetbi wrote:
I've just did it like this:
xlog("$rc");
and on Friday I got
18446744073709551615
so You were right that it was unsigned int. But now if I want to read
xlog("$rc") it has 1 value. And my table is empty.
Now I've changed script and it looks:
if ($avp(s:ip) == null ) {
xlog("no results found in DB");
xlog("$rc");
and it is ok, and works like it should, because $avp(s:ip) is NULL when
there's no records found. But I'm afraid it's not the best solution. What do
You think?
--
View this message in context:
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Re: [OpenSIPS-Users] Opensips 1.6 as a Proxy and Presence Server
I checked your script and you are not handling publish/subscribe for sequential requests. You should try to move the route(handle_presence); call for sequential requests too. Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 01/17/2017 10:07 AM, maatohewetbi wrote: Hello, can anybody help? I need blf to work. -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Opensips-1-6-as-a-Proxy-and-Presence-Server-tp7605411p7605593.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Registrar with IP authentication - selecting variables from DB
That's weird. How did you check the value and you got 1842312...? I am asking because that does look like a -2, only converted to an unsigned representation, (unsigned int)-2 = 18446744073709551614). Or was it -1? Can you control the data in the database and make sure you don't have that username when doing the query? Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 01/17/2017 10:06 AM, maatohewetbi wrote: Razvan, I've found that this conditional doesn't work: if ($rc == -2) It turned out that $rc variable is never -2, although select query(select ip from address where context_info='$fU'", "$avp(ip)"), doesn't contain any values. When I checked $rc variable its value was 1, and once it was something like 1842312...so very long digit, but it was never -2. It it possible that this value changes? -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Registrar-with-IP-authentication-selecting-variables-from-DB-tp7605514p7605592.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] opensips $Revision: 4448 $
Yes, please try that. To be honest, I've never done it using the new
accounting module, but this might do the trick.
Also, did you take a look at the multi-legging accounting?
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 01/16/2017 01:18 PM, Khalil Khamlichi wrote:
Oki, I will modify my old failure_route :
failure_route[GW_FAILOVER]{
if (t_was_cancelled()) {
exit;
}
# detect failure and redirect to next available GW
if (t_check_status("(408)|488|([56][0-9][0-9])")) {
#xlog("Failed GW $rd detected \n");
if ( use_next_gw() ) {
t_on_failure("GW_FAILOVER");
t_relay();
exit;
}
send_reply("500","All GW are down");
}
}
to :
failure_route[GW_FAILOVER]{
if (t_was_cancelled()) {
exit;
}
# detect failure and redirect to next available GW
if (t_check_status("(408)|488|([56][0-9][0-9])")) {
xlog("Failed GW $rd detected \n");
if ( use_next_gw() ) {
t_on_failure("GW_FAILOVER");
*do_accounting("db|log","failed|missed",);*
t_relay();
exit;
}
send_reply("500","All GW are down");
}
}
right ?
On Mon, Jan 16, 2017 at 11:07 AM, Răzvan Crainea <raz...@opensips.org
<mailto:raz...@opensips.org>> wrote:
I was asking you to call do_accounting() in failure route, for
each leg.
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com <http://www.opensips-solutions.com>
On 01/16/2017 12:02 PM, Khalil Khamlichi wrote:
sorry mistype, I am calling do_accounting() twice.
On Mon, Jan 16, 2017 at 9:47 AM, Khalil Khamlichi
<khamlichi.kha...@gmail.com <mailto:khamlichi.kha...@gmail.com>>
wrote:
thanks for your much appreciated help,
I am calling do_routing twice.
Here is my actual opensips.cfg :
route {
...
...
...
...
if (has_totag()) {
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route()) {
# validate the sequential request against dialog
if ( $DLG_status!=NULL &&
!validate_dialog() ) {
xlog("In-Dialog $rm from $si (callid=$ci) is not valid
according to dialog\n");
}
if (is_method("BYE")) {
# do accounting even if the transaction fails
do_accounting("db|log","failed|missed",);
} else if (is_method("INVITE")) {
# even if in most of the cases is useless, do RR for
# re-INVITEs alos, as some buggy clients do change route set
# during the dialog.
record_route();
}
...
...
...
...
$acc_extra(gwid)=$avp(gw_id);
t_on_failure("GW_FAILOVER");
do_accounting("db|log","cdr|missed",);
#NAT
if (isbflagset(NAT)) setflag(NAT);
#NAT
route(RELAY);
}
# END OF MAIN ROUTE
On Mon, Jan 16, 2017 at 8:51 AM, Răzvan Crainea
<raz...@opensips.org <mailto:raz...@opensips.org>> wrote:
Hi, Khalil!
Did you try to call the do_accounting() function for each
leg going to the next gateway?
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
<http://www.opensips-solutions.com>
On 01/15/2017 12:24 AM, Khalil Khamlichi wrote:
Hi,
I am testing opensips 2.2, 2.3 I have tried to configure
acc module to save to db failed calls on drouting
configuration, I have found that it does save only the
first failed call (that is the first gateway) it does
not save to database any other failures on second and
third gateways that are tried for the call.
is this the expected behavioure ?
Thanks for your help.
regards,
kh
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Re: [OpenSIPS-Users] Problem with switch .. case
The $rc in the switch is the return value of xlog(), which probably is
success.
Just remove the xlog after the radius_send_auth(), or store the $rc in a
variable and test that variable.
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 01/16/2017 12:24 PM, Dragomir Haralambiev wrote:
Hello,
I see problem when using switch .. case.
Opensips version 2.2.2.
Here is part of my script:
radius_send_auth("out","in");
xlog("L_ERR","Radius return $rc");
switch ($rc) {
case -1:
xlog("L_ERR", "ERROR during authentication");
sl_reply_error();
exit;
case -2:
xlog("L_ERR", "Authentication denied");
sl_reply_error();
exit;
} # end switch
xlog("L_ERR","Continue ");
This is results:
Jan 16 10:16:55 sbc-01 /usr/sbin/opensips[7820]: Radius return -1
Jan 16 10:16:55 sbc-01 /usr/sbin/opensips[7820]: Continue
Best regards,
Dragomir
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Re: [OpenSIPS-Users] opensips $Revision: 4448 $
I was asking you to call do_accounting() in failure route, for each leg.
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 01/16/2017 12:02 PM, Khalil Khamlichi wrote:
sorry mistype, I am calling do_accounting() twice.
On Mon, Jan 16, 2017 at 9:47 AM, Khalil Khamlichi
<khamlichi.kha...@gmail.com <mailto:khamlichi.kha...@gmail.com>> wrote:
thanks for your much appreciated help,
I am calling do_routing twice.
Here is my actual opensips.cfg :
route {
...
...
...
...
if (has_totag()) {
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route()) {
# validate the sequential request against dialog
if ( $DLG_status!=NULL &&
!validate_dialog() ) {
xlog("In-Dialog $rm from $si (callid=$ci) is not valid according
to dialog\n");
}
if (is_method("BYE")) {
# do accounting even if the transaction fails
do_accounting("db|log","failed|missed",);
} else if (is_method("INVITE")) {
# even if in most of the cases is useless, do RR for
# re-INVITEs alos, as some buggy clients do change route set
# during the dialog.
record_route();
}
...
...
...
...
$acc_extra(gwid)=$avp(gw_id);
t_on_failure("GW_FAILOVER");
do_accounting("db|log","cdr|missed",);
#NAT
if (isbflagset(NAT)) setflag(NAT);
#NAT
route(RELAY);
}
# END OF MAIN ROUTE
On Mon, Jan 16, 2017 at 8:51 AM, Răzvan Crainea
<raz...@opensips.org <mailto:raz...@opensips.org>> wrote:
Hi, Khalil!
Did you try to call the do_accounting() function for each leg
going to the next gateway?
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com <http://www.opensips-solutions.com>
On 01/15/2017 12:24 AM, Khalil Khamlichi wrote:
Hi,
I am testing opensips 2.2, 2.3 I have tried to configure acc
module to save to db failed calls on drouting configuration,
I have found that it does save only the first failed call
(that is the first gateway) it does not save to database any
other failures on second and third gateways that are tried
for the call.
is this the expected behavioure ?
Thanks for your help.
regards,
kh
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Re: [OpenSIPS-Users] mi_json Module
Hi, Dragomir! Parameters are sent in the params parameter. Check the example in the documentation: http://www.opensips.org/html/docs/modules/2.2.x/mi_json#id249590 This works for me: # curl -s 'http://127.0.0.1:8080/json/get_statistics?params=shmem:' Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 01/16/2017 11:07 AM, Dragomir Haralambiev wrote: Hello, I try to use mi_json modile in Opensips 2.2.2. All is OK when try to read log level: opensips_ip:Opensips_http_port/json/log_level How to use "get_statistics hmem:" ? Follow command not working: opensips_ip:Opensips_http_port/json/get_statistics%20shmem: Regards, Dragomir ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] opensips $Revision: 4448 $
Hi, Khalil! Did you try to call the do_accounting() function for each leg going to the next gateway? Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 01/15/2017 12:24 AM, Khalil Khamlichi wrote: Hi, I am testing opensips 2.2, 2.3 I have tried to configure acc module to save to db failed calls on drouting configuration, I have found that it does save only the first failed call (that is the first gateway) it does not save to database any other failures on second and third gateways that are tried for the call. is this the expected behavioure ? Thanks for your help. regards, kh ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OpenSIPs crashed
Hi, Ahmed!
Can you tell us exactly what revision of OpenSIPS you are using? Please
provide the output of the following commands:
opensips -V
opensipsctl ps
Also, during startup, is there a process who's "eating" a lot of CPU? If
so, can you pinpoint the PID to see what type of process is that?
Regarding the avp_db_query() issue, did you define a db_url parameter
for it? Also I am not sure you can do something like $var(res) =
avp_db_query(...). But anyways, this is something completely different,
so please open a different topic for it.
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 01/14/2017 12:24 AM, Ahmed Munir wrote:
Hi,
I've just installed new version of opensips 2.2.2 on the test box and
updated by routing script, the issue currently I'm seeing alot warning
messages while starting opensips service below;
/usr/sbin/opensips[6902]: WARNING:core:handle_timer_job: utimer job
has a 283 us delay in execution
Number of children running on that server is 8 as it is 8 core processor.
I would like to know what steps do I need to take to fix this issue.
Btw, warnings only occurred during the time of starting opensips
service but not during calls.
Further added, a issue I face using avp_db_query () function i.e. when
using it as
$var(res) = avp_db_query("SELECT Outpulse_number,setid FROM
Prefix_data where Program_prefix = $var(pg_prefix)", "$avp(outpluse),
$avp(trunkid)");
failed to start opensips service due to errors below;
ERROR:avpops:__fixup_db_query_avp: no db url defined to be used by
this function
ERROR:core:fix_actions: fixing failed (code=-6) at
//etc/opensips/opensips.cfg:207
CRITICAL:core:fix_expr: fix_actions error
ERROR:core:main: failed to fix configuration with err code -6
If I add this line: avp_db_query("SELECT 1"); above to my $var(res) db
query, opensips service starts successfully.
Please advise the steps do I need to take to fix above issues.
From: Răzvan Crainea <raz...@opensips.org
<mailto:raz...@opensips.org>>
To: users@lists.opensips.org <mailto:users@lists.opensips.org>
Subject: Re: [OpenSIPS-Users] OpenSIPs crashed
Message-ID: <40f6dada-e121-a2da-b283-69dff891c...@opensips.org
<mailto:40f6dada-e121-a2da-b283-69dff891c...@opensips.org>>
Content-Type: text/plain; charset="utf-8"; Format="flowed"
Hi, Ahmed!
OpenSIPS 1.6.3 is no longer supported (since 2013), so there's not
much
we can do right now. Try upgrading your install to the latest 1.6.4
version and see if your problem is solved. Otherwise, upgrade to a
newer, supported version, preferably the latest stable release, 2.2.2.
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com <http://www.opensips-solutions.com>
On 01/12/2017 11:55 PM, Ahmed Munir wrote:
> Found coredump on one of the server, see some partial message below
> while taking the back trace;
>
>
>
> Core was generated by `/usr/sbin/opensips -P
/var/run/opensips.pid -m
> 64 -u opensips -g opensips'.
>
> Program terminated with signal 11, Segmentation fault.
>
> #0 0x7f650687a069 in sip_msg_cloner () from
> /usr/lib64/opensips/modules/tm.so
>
> Missing separate debuginfos, use: debuginfo-install
> opensips-1.6.3-notls.x86_64
>
>
>
> Please advise what might be the reason causing opensips to crash.
>
> --
> Regards,
>
> Ahmed Munir Chohan
>
>
>
>
> --
> Regards,
>
> Ahmed Munir Chohan
>
--
Regards,
Ahmed Munir Chohan
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Re: [OpenSIPS-Users] Registrar with IP authentication - selecting variables from DB
You can do something like this:
avp_db_query("select ip from address where context_info='$fU'", "$avp(ip)");
if ($rc == -2) {
# not found in db
} else if ($avp(ip) != $si) {
# reject the call
}
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 01/13/2017 01:51 PM, maatohewetbi wrote:
Still I have to check login whether it exist in table. Then I have to compare
it to IP address.
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Re: [OpenSIPS-Users] change T_fr_in_timeout
Hello!
The problem is that you are doing your logic in the onreply_route, which
is not transaction aware. You should move the logic in a name onreply
route that you engage in your main route. Something like:
route {
...
if (is_method("INVITE"))
t_on_reply(handle_timer);
...
}
onreply_route[handle_timer] {
if (t_check_status("(180)|(183)"))
$T_fr_inv_timeout = 60;
}
Hope this is a right pointer to solve your problem.
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 01/12/2017 04:40 PM, bluerain via Users wrote:
Ok, so hopefully I get some help in this forum, I'm trying to get a faster
PDD by going down the LCR list faster. I'm trying to use the fr_inv_timeout
to skip the carrier that has slow PDD. So I did the code below so that if I
don't get a 180/183 after invite within 5 second, I will terminate the call
and go to the next carrier. But if they do give 180/183 after invite within
5 second, I want to change T_fr_inv to 60 seconds so have enough time for
far end user to pick up the call. But I ran into 2 problem:
1. 180/183 seems is NOT picking up inthe onreply_route. I did a log and it
seems on 180/183 it never went into the code where I wanted to, so is this
is WRONG place to detect 180/183?
2. I also try to display the value of $T_fr_inv_timeout in onreply_route
section of the code, but it always show "ZERO"?
Anyway, any pointer will be greatly appreciated. Thank you!
loadmodule "tm.so"
modparam("tm", "fr_timeout", 5)
modparam("tm", "fr_inv_timeout", 5)
modparam("tm", "restart_fr_on_each_reply", 0)
modparam("tm", "onreply_avp_mode", 1)
onreply_route {
if (t_check_status("(180)|(183)")) {
$T_fr_inv_timeout = 60;
}
}
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Re: [OpenSIPS-Users] Registrar with IP authentication - selecting variables from DB
Yes, it is. Provision the address table in the database and use the check_source_address() functionin the script. [1] http://www.opensips.org/html/docs/modules/2.2.x/permissions#id295007 Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 01/13/2017 11:03 AM, maatohewetbi wrote: Yes, but every IP and login should be in table. How can I read variables from DB? Is it possible to do it? -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Registrar-with-IP-authentication-selecting-variables-from-DB-tp7605514p7605547.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Registrar with IP authentication - selecting variables from DB
Then simply reverse the IP check logic and do it after the user is authenticated. Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 01/11/2017 01:15 PM, maatohewetbi wrote: Yes, but I want to check sip login first, not an IP. Here is ny plan, what I want to do: - store IP, login in one table (a new on or existing one) - there will be IP and SIP logins. When a client make a registration, my script should check if this login is in table, if yes - then check IP, if it matches - allow a registration, if not - send 403 and exit. There will be another case, when a SIP login is not in this table - just allow registration without checking an IP. -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Registrar-with-IP-authentication-selecting-variables-from-DB-tp7605514p7605516.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] $rl_count in 2.2.2
Hi, Pat!
The way you are using $rl_count is wrong, because the $rl_count pseudo
variable only accepts strings or other pvars, not formatted strings
(such as "cps_$avp(trunk_group)").
To achieve what you are trying to do is to assign the name to a pvar and
feed it in the $rl_count's name:
$var(rl_name) = "cps_" + $avp(trunk_group);
$json(call_details/tg_cps) = $rl_count($var(rl_name));
Let me know how this goes.
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 01/13/2017 12:48 AM, Pat Burke wrote:
Hello,
I am trying to get the current rate limit value using $rl_count, but
it is giving the following error.
ERROR:ratelimit:pv_get_rl_count: invalid name
WARNING:core:do_assign: no value in right expression at
/etc/opensips/opensips_proxy.cfg:598
4024438929 SCRIPT:CCLIMIT:INFO: rl_limit for cps_90761 = 0
Here is the code
$json(call_details/tg_cps) = 0; # Default to 0
...
if ($avp(maxcps) != NULL && $avp(maxcps) > 0) {
if (!rl_check("cps_$avp(trunk_group)", "$avp(maxcps)")) {
xlog("L_NOTICE", "$rU SCRIPT:CPSLIMIT:DBG: Max
$avp(maxcps) cps reached for trunk group $avp(trunk_group) \n");
$avp(error_reason) = $avp(error_reason) +
$avp(trunk_group) + " CPS limit reached.";
send_reply("403", "Max CPS limit reached");
exit;
}
}
$json(call_details/tg_cps) = $rl_count("cps_$avp(trunk_group)");
xlog("L_INFO", "$rU SCRIPT:CCLIMIT:INFO: rl_limit for
cps_$avp(trunk_group) = $json(call_details/tg_cps) \n");
From the command opensipsctl fifo rl_list | grep cps_90761, I get
PIPE:: id=cps_90761 algorithm=TAILDROP limit=2 counter=0
What am I missing?
Regards,
*Pat Burke*
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Re: [OpenSIPS-Users] Registrar with IP authentication - selecting variables from DB
Yes, you can use the check_source() address function[1] just before the auth block. [1] http://www.opensips.org/html/docs/modules/2.2.x/permissions#id295007 Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 01/11/2017 12:44 PM, maatohewetbi wrote: Is there any way to make an IP authorization with registrar module? First I want to authenticate peer with IP, and then allow him to register with correct login/pass. Or is there any way to select any variable from DB? -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Registrar-with-IP-authentication-selecting-variables-from-DB-tp7605514.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] How to store variables into file
Hello,
PS: Because you are not subscribed to the list, your messages are
moderated - therefore they might not reach the list, or might reach the
list later. Please subscribe to the users list[1], otherwise the
interaction between us and you will be harder.
[1] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 01/09/2017 05:43 PM, maatohewetbi wrote:
How can I save any variable into log? I use somethin like this:
log("$Ts");
But in log file I always get:
Jan 9 16:39:14 OpenSips/sbin/opensips[23670]: $Ts
How I can get this variable in log?
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Re: [OpenSIPS-Users] 2.2.2 packaging
Actually as Nick pointed out, the rpm specs used are here: https://github.com/OpenSIPS/opensips/tree/2.2/packaging/redhat_fedora Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 01/09/2017 10:36 AM, Răzvan Crainea wrote: Hello, Nathan! You have my answer inline. Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 01/07/2017 11:10 AM, Nathan Ward wrote: Hi, I am running OpenSIPS 2.2.2 from RPMs from the opensips.org yum server. I am trying to find out where the SPEC files to generate the CentOS (/el7) RPMs are. I don’t seem to be able to find these in GitHub. Can someone help with this? Nick (CC'ed) is maintaining the SPECS for the CentOS RPMs, but I am not sure all changes are integrated in the RPM files. But AFAIK, they are based on the public specs here: https://github.com/OpenSIPS/opensips/blob/2.2/packaging/rpm/opensips.spec.CentOS I am hoping to submit a PR to move the ‘opensips’ binary to /usr/libexec, rather than a common $PATH location. We have had a problem where this binary was mistakenly run, rather than opensipsctl. Typically, daemon binaries that do not generally get run by users should be in /usr/libexec so that they are only very intentionally executed. This should probably go in to at the very least a minor version change as it may break some people’s custom init scripts etc. I am not sure I fully agree with this. /usr/libexec contain binaries that should never be run as stand-alone. But that is not OpenSIPS' case, where you can easily start it in debug mode or smth like that. For example, a similar daemon is sshd, which is located in /usr/sbin/sshd. However, internally triggered binaries, such as sftp-server are located in /usr/libexec/openssh/sftp-server. This exec should never be executed outside sshd. On my search, I note change 9e406b2b3acfd61b39ba9679f0a599b95f56f5c2 under the 2.2.2 tag, which appears to be done with something like: sed ’s/2.2.1/2.2.2/‘ Note the matches where . is ‘any’ not a literal period, so there are a lot of dates that get messed up: -* Mon Oct 12.2.19 Bogdan-Andrei Iancu <bog...@opensips.org> +* Mon Oct 12.2.29 Bogdan-Andrei Iancu <bog...@opensips.org> You are right, this is a bug in my changelog update script. I will update it now. Best regards, Răzvan ___ 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] 2.2.2 packaging
Hello, Nathan! You have my answer inline. Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 01/07/2017 11:10 AM, Nathan Ward wrote: Hi, I am running OpenSIPS 2.2.2 from RPMs from the opensips.org yum server. I am trying to find out where the SPEC files to generate the CentOS (/el7) RPMs are. I don’t seem to be able to find these in GitHub. Can someone help with this? Nick (CC'ed) is maintaining the SPECS for the CentOS RPMs, but I am not sure all changes are integrated in the RPM files. But AFAIK, they are based on the public specs here: https://github.com/OpenSIPS/opensips/blob/2.2/packaging/rpm/opensips.spec.CentOS I am hoping to submit a PR to move the ‘opensips’ binary to /usr/libexec, rather than a common $PATH location. We have had a problem where this binary was mistakenly run, rather than opensipsctl. Typically, daemon binaries that do not generally get run by users should be in /usr/libexec so that they are only very intentionally executed. This should probably go in to at the very least a minor version change as it may break some people’s custom init scripts etc. I am not sure I fully agree with this. /usr/libexec contain binaries that should never be run as stand-alone. But that is not OpenSIPS' case, where you can easily start it in debug mode or smth like that. For example, a similar daemon is sshd, which is located in /usr/sbin/sshd. However, internally triggered binaries, such as sftp-server are located in /usr/libexec/openssh/sftp-server. This exec should never be executed outside sshd. On my search, I note change 9e406b2b3acfd61b39ba9679f0a599b95f56f5c2 under the 2.2.2 tag, which appears to be done with something like: sed ’s/2.2.1/2.2.2/‘ Note the matches where . is ‘any’ not a literal period, so there are a lot of dates that get messed up: -* Mon Oct 12.2.19 Bogdan-Andrei Iancu <bog...@opensips.org> +* Mon Oct 12.2.29 Bogdan-Andrei Iancu <bog...@opensips.org> You are right, this is a bug in my changelog update script. I will update it now. Best regards, Răzvan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Install opensips to systemd
No, unfortunately there is no way to install the systemd files automatically, you have to do it manually. Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 01/08/2017 10:49 PM, Robert Dyck wrote: I had a working opensips 1.11 installed by Fedora package manager. I have been trying opensips 2.2.2 and decided to make the jump. Since 2.2.2 is not available as a package from Fedora I cloned the source. I have tried installing using menuconfig and also make but neither installs the necessary files for systemd. I see that the files exist in the source directory "packaging". Is there a way to trigger the installation of the necessary files for systemd or does it need to be done manually. I have thought of re-installing 1.11 from the package manager and then overwriting the opensips binary. The problem I see is that a package upgrade would overwrite 2.2.2 because the package manager sees opensips as an installed package. ___ 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] do_routing function first argument
Hello, Khalil!
If you check the documentation of the do_routing function[1], you will
see that the group is not mandatory. If it exists, then it will use the
route set within that group. If it does not exist, the group is
determined from the dr_group table in the database - this is just a
simple mapping between user and group. If you would like to have a more
flexible approach, where you determine the route set using a more
complex logic than a simple matching, then you will have to implement
that logic in the script to find the group, and provide it to the
do_routing() function.
I hope I answered you question now.
[1] http://www.opensips.org/html/docs/modules/2.2.x/drouting.html#id295067
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 01/08/2017 08:23 PM, Khalil Khamlichi wrote:
Well you get to my point, shouldn't it be in the database then ? whats
the point in having it hard coded inside the config file ?
On Sun, Jan 8, 2017 at 1:24 PM, Johan De Clercq <jo...@democon.be
<mailto:jo...@democon.be>> wrote:
I think that's route set id.
On 08 Jan 2017 12:12 PM, "Khalil Khamlichi"
<khamlichi.kha...@gmail.com <mailto:khamlichi.kha...@gmail.com>>
wrote:
Hi eveyone,
I have been trying to understand the first argument to the
do_routing("1","",,"$var(rule_attrs)")
Can anyone please give some explanation of the first argument
to this function ? what does the "1" stands for and where does
it fit in the logic back in the database setup (in my scenario
it's mostly drouting tables)
Thanks in advance.
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Re: [OpenSIPS-Users] Topology hiding and Route header
Hi, Adrien!
If you are talking about a pre-loaded Route header (received from the
client) for an initial INVITE, then the topology_hiding() module does
not remove it and you have to remove it yourself.
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 01/06/2017 02:44 PM, Adrien Martin wrote:
Hello,
I'm using the module topology_hiding with Opensips 2.2 but I'm not sure to
understand properly the documentation.
Is the header Route supposed to be removed ? (No problem removing and restoring
Record-Route)
In my tests Route header was not removed unless I added remove_hf("Route")
after topology_hiding.
Regards,
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Re: [OpenSIPS-Users] ERROR:drouting:populate_dr_bls: Something went wrong in add_rule_to_list
Hi, Flavio? Are you seeing the errors in the logs? Can you send them to us to check? Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 01/05/2017 02:12 PM, Flavio Goncalves wrote: Hi, I'm getting some errors on drouting and I got some routes sent to wrong gateways (some gateways being skipped). It seems a fail to insert in the blacklist, but never saw this message before, opensips 1.11. I don't know if the problems are correlated. Att, Flavio E. Goncalves V.Office Redes e Telecomunicações Ltda. Fone 48 33328590 Skype:flaviogoncalves1 Linkedin: www.linkedin.com/in/flavioegoncalves <http://www.linkedin.com/in/flavioegoncalves> ___ 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] CRITICAL:core:timer_ticker: timer handler
Hi, Flavio! Is it the latest OpenSIPS version? How many children are you using? Can you send over the output of "opensipsctl ps"? Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 01/05/2017 01:59 PM, Flavio Goncalves wrote: Hi, I'm getting some errors like below (2 simultaneous calls). No stress test. pkmem is ok, shmem is ok. CRITICAL:core:timer_ticker: timer handler lasted (503 us) for more than timer tick (100 us) -> potential timer shifting. Anyone with the same problem? Flavio E. Goncalves ___ 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] Codecs Order
Hi, Alain!
You could use the SDP transformations combine with regular expresions. I
am thinking at something like:
if ($(rb{sdp.line,m}) =~ "AVP\s*8")
xlog("PCMA is first codec!\n");
Hope this is useful :)
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 01/05/2017 12:42 PM, Alain Bieuzent wrote:
Ok, but how can i extract the value of first codec,
Actually i’m using codec_exist function, but this one only retrun TRUE
or FALSE, but in case of true, i want know if it’s the first codec or no.
Regards
*De : *Users <users-boun...@lists.opensips.org> au nom de Aqs Younas
<aqsyou...@gmail.com>
*Répondre à : *OpenSIPS users mailling list <users@lists.opensips.org>
*Date : *jeudi 5 janvier 2017 à 11:35
*À : *OpenSIPS users mailling list <users@lists.opensips.org>
*Objet : *Re: [OpenSIPS-Users] Codecs Order
media line(m) shows codec preference in order.
m=audio 13406 RTP/AVP 8 0 101
8 for PCMA
0 for PCMU
On 5 January 2017 at 15:19, Alain Bieuzent <alain.bieuz...@free.fr
<mailto:alain.bieuz...@free.fr>> wrote:
Hi all,
Is there a way to know the order of the codecs?
For exemple in this SDP :
m=audio 13406 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
How i know that the prefered codec is PCMA ?
Regards,
Alain
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Re: [OpenSIPS-Users] rtpproxy and record calls
Hi, Denis!
Regarding 1, did you try to run rtpproxy_start_recording() for both
INVITE and 200 OK?
Regarding 2, '/' is a valid character in Call-id. Therefore the problem
is at the RTPProxy side - before writing the CDR they should escape (or
transform somehow) the '/' character in something else.
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 01/04/2017 05:24 PM, Денис Путято via Users wrote:
Hello!
Is there any information about the problem?
Thank you.
--
С уважением,
Путято Денис
Best regards, Denis
14:47, 27 декабря 2016 г., Denis via Users <users@lists.opensips.org>:
Hello!
I try to use rtpproxy for call recording and have two problems
1) rtpproxy records only one way of the call (from callee to
caller). For starting rtp proxy i use
rtpproxy_engage("conrf",,"1",) function.
2) i am using top_hiding with "C" flags (which should change
callid) and i noticed, that sometime the function generates callid
with such form (for example)
DLGCH_fRUAW2B/M2NiSV1WYWRhYX9ASFowK2J+f0JXXmt5MWspSVEM.
I want to pay attention to "/" character. As i understand, because
of this character i got such error from rtpproxy
ERR:DLGCH_fRUAW2B/M2NiSV1WYWRhYX9ASFowK2J+f0JXXmt5MWspSVEM:rtpp_record_open:
can't open file
/mnt/callrecords//DLGCH_fRUAW2B/M2NiSV1WYWRhYX9ASFowK2J+f0JXXmt5MWspSVEM=cU7ZHtge63paN.a.rtp
for writing: No such file or directory (2)
rtpproxy: 2.2.alpha.20160822
Opensips: Server:: OpenSIPS (2.2.2 (x86_64/linux))
Thank you for any help.
--
С уважением, Денис.
Best regards, Denis
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Re: [OpenSIPS-Users] opensips for windows
Hi, Mohsin! OpenSIPS was designed to work on Unix systems, I never tried to compile it on Windows. Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 12/31/2016 12:32 PM, Mohsin Barbhaiwala wrote: can this opensips be compiled & run on windows. if so does anybody know ? ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips 1.6 as a Proxy and Presence Server
Hello!
Basically merging the two separate configs together should do the trick.
If you follow this[1] example, change your Proxy config as it follows:
When an initial INVITE comes in, create the dialog and publish the
information:
if (is_method("INVITE")) {
create_dialog();
# publish for both legs
dialoginfo_set("AB");
}
When a publish or subscribe methods come in handle them accordingly:
... if (is_method("PUBLISH|SUBSCRIBE")) {
route(handle_presence);
exit;
}
... route[handle_presence]
{
if(!t_newtran()){
sl_reply_error();
exit;
}
if (is_method("PUBLISH")) {
handle_publish();
} else
if (is_method("SUBSCRIBE")) {
handle_subscribe();
}
exit;
}
[1]
http://www.opensips.org/Documentation/Tutorials-Presence-PuaDialoinfoConfig
PS: Please subscribe on the mailing list, otherwise you will loose
further replies
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 01/03/2017 10:13 AM, maatohewetbi wrote:
I use Opensips 1.6.2 on my server. I want to add BLF function to it but I
don't know how. I have config where I route calls to asterisk. There are
tutorials where I can set Opensip Presence Server as standalone, with
OpenXcap as well, but there's nowhere config how to set Opensips to work on
one machine as Presence Server, and also as a Proxy at same Opensips
instance - in one config. How can I make it? Ok, I can set one Opensips as a
proxy, and other Opensips (on another machine) as a Presence Server. But how
can I connect them so that they work together?
--
View this message in context:
http://opensips-open-sip-server.1449251.n2.nabble.com/Opensips-1-6-as-a-Proxy-and-Presence-Server-tp7605411.html
Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
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Re: [OpenSIPS-Users] opensips-cp missing
Hi, Jaap! How did you initially install opensips CP? Because most likely it is was a manual install, and you will need to restore the apache config file. You can use this example[1]. PS: please subscribe to the mailing list, otherwise you won't get further replies. [1] https://github.com/OpenSIPS/opensips-cp/blob/abafe7503dd8ad089b834cfcb7058deeda30375e/INSTALL#L30 Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 01/02/2017 10:28 PM, Jaap van der Starre wrote: After upgrade from wheezy to jessie opensips-cp is missing while my grandstream HT503 is still registered to opensips, I cannot use the control panel anymore obviously. Not Found The requested URL /opensips-cp was not found on this server. Apache/2.4.10 (Debian) Server at 192.168.2.12 Port 80 My opensips installation is 1.10. Because it is not supported anymore I should at least migrate to 1.11 now. This will only change the database and Script migration. Would it bring back opensips-cp in Apache? jaap ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] opensips performance
Hi, Agalya! I've just written an article about dimensioning OpenSIPS memory on the OpenSIPS blog[1]. Perhaps that can help you. To do further debugging, you should add some thresholds warnings to determine what function spends most of the time processing. You can do this by using msg thresholds[2]. Its also a good idea to check how DNS behaves on your system, using the dns threshold[3]. Also, monitoring statistics, like memory and load might be useful for your debug. Also, make sure bandwidth is not a problem either! PS: for the test in the article I used my personal laptop (2 cores, 8GB RAM) and reached 500CPS with 20 children. [1] https://blog.opensips.org/2016/12/29/understanding-and-dimensioning-memory-in-opensips/ [2] http://www.opensips.org/Documentation/Script-CoreParameters-2-2#toc60 [3] http://www.opensips.org/Documentation/Script-CoreParameters-2-2#toc59 Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com On 12/23/2016 08:39 PM, Ramachandran, Agalya (Contractor) wrote: Hi Bogdan/team, Am using opensips 2.2.2 version and using opensips as only proxy. In the default script, in route[relay], I have called setdsturi(); No other changes with the default script. Earlier, I have used only the default value of S_MEMORY and P_MEMORY in the /etc/default/opensips. With this default values, for the end to end established call , I could reach only 150cps. · When I tried to increase *S_MEMORY and P_MEMORY to 1024 and 64*respectively, and having opensips *children =24*, I could achieve 1000 cps. · I have also re-tuned OpenSIPS log file, by adding “-/var/log/opensips.log”. · I have set the log_level =1 When I see opensips performance with default script I see you are achieving 9000cps. Am running on VM, with 8 core CPU, 16GB RAM and processor is Intel Xeon E312xx(Sandy Bridge). *What are the places I need to take a look to tune up still, to get more performance. Your expertise would help me a lot. * * * In this particular test, I am not performing any DB operations /REST operations. This is pure proxy, using default script. Regards, Agalya ___ 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] [BLOG] Understanding and dimensioning memory in OpenSIPS
I've just posted a new article[1] on the OpenSIPS blog[2] that talks about OpenSIPS memory usage: understanding memory internals and explaining how you can dimension your OpenSIPS installation to support your customers and the services you offer. I hope this will be useful for some of you. Happy reading :)! [1] https://blog.opensips.org/2016/12/29/understanding-and-dimensioning-memory-in-opensips/ [2] https://blog.opensips.org Best wishes, -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] b2b server_address parameter
Hi, Ziv!
Are you using any advertise line in your script? If you do, you should
add it for the listener the message was received on, i.e.
listener=udp:PRIVATE_IP:5060 as PUBLIC_IP
Note that this will change the IP for all the headers, including Via,
Record-Route, etc.
If you don't want to do that, another idea (but less efficient) is to
catch the INVITE in local_route and "manually" change the Contact
header, using remove_hf() and append_hf().
Let us know how that works.
Best regards,
Răzvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 12/26/2016 09:03 AM, Ziv Gabel wrote:
I’m using to top hiding script and I’ve set the following
modparam("b2b_logic", "server_address",
"sip:sa@:5063;transport=tls")
But in the contact of the INVITE I still see the internal IP of the server.
How can I change the contact in such a scenario ?
*Ziv Gabel *l* *Professional services* *l* *CommuniTake Technologies Ltd.
* *
*M*: +972535265553 l* **Skype*: ziv_gabel l *E*: z...@communitake.com
<mailto:z...@communitake.com>
*T*: +972.4.696.8908 l *F*: +972.4.959.1654 l www.communitake.com
<http://www.communitake.com/>
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Re: [OpenSIPS-Users] custom presence bodies
Hi, Tito! Yes, the Dialog id is mandatory. You can find more info in Section 4.1.1 of RFC4235[1]. Now I am not sure what you are trying to do, but if you use the pua_dialoginfo module[2], OpenSIPS will be able to generate the Publish for each call. [1] https://tools.ietf.org/html/rfc4235#section-4.1.1 [2] http://www.opensips.org/html/docs/modules/2.2.x/pua_dialoginfo.html Best regards. Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com On 12/23/2016 11:36 PM, Tito Cumpen wrote: The closest thing that resembles the sort of body I need is using dialog_info but even that is lacking because it supposed to be used for the user you are subscribed to. Can you tell me if the dialog id is necessary during publish? I tried reading https://tools.ietf.org/html/rfc4235 <https://tools.ietf.org/html/rfc4235> but cannot find information of where it is sourced from. On Tue, Dec 20, 2016 at 7:25 AM, Tito Cumpen <t...@xsvoce.com <mailto:t...@xsvoce.com>> wrote: Thanks for your reply Razvan, I tried using the conference event with application/xml+conf but opensips is replying with these errors: : ERROR:presence:handle_subscribe: unrecognized value [conference] in Event header : INFO:presence:handle_subscribe: Missing or unsupported event header field value ERROR:presence:handle_publish: unrecognized value [conference] in Event header ERROR:presence:handle_publish: Missing or unsupported event header field value ERROR:presence:handle_publish: #011event=[conference] When I used the presence events with application/pidf+xml notifies worked but they only carried limited information \nmailto:sip%3asjoi...@domain.org>\"/>\n reduced from : . . . connected. . . . which means that the body was dismissed and the entity was taken from the ruri? any idea how to relay this body? or modify it to carry my users tag with state and endpoint enitity? Thanks, Tito On Tue, Dec 20, 2016 at 2:49 AM, Răzvan Crainea <raz...@opensips.org <mailto:raz...@opensips.org>> wrote: Hi, Tito! The Content-Type indicates how to read and parse the Message body. This is a mandatory header, and without it the client can't know how to interpret the body. The Event header is used by opensips (and clients) to figure out the event that happened. The idea is that the clients will subscribe for a particular event (presence, or conference in your case) and when that event happens, OpenSIPS will send notifies only to those clients registered for that event. Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com <http://www.opensips-solutions.com> On 12/20/2016 11:37 AM, Tito Cumpen wrote: Razvan, Thanks for pointing that out. Will my subscribers get this xml via notify regardless of the content type? say : application/xml+conf" or application/pidf+xml does the application header or Event:presence header mean anything to opensips? On Tue, Dec 20, 2016 at 12:04 AM, Răzvan Crainea <raz...@opensips.org <mailto:raz...@opensips.org>> wrote: Hi, Tito! I ran your example in an XML validator, and I noticed that your XML has an "error on line 2 at column 86: attributes construct error"[1]. If you add a space before the "state" attribute, the XML gets parsed properly[2]. [1] http://www.utilities-online.info/xmltojson/?save=5a6721b1-55f9-46c2-8bda-cf21e15e38b3-xmltojson <http://www.utilities-online.info/xmltojson/?save=5a6721b1-55f9-46c2-8bda-cf21e15e38b3-xmltojson> [2] http://www.utilities-online.info/xmltojson/?save=320f586b-c2eb-4e73-b564-3f73f0120662-xmltojson <http://www.utilities-online.info/xmltojson/?save=320f586b-c2eb-4e73-b564-3f73f0120662-xmltojson> Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com <http://www.opensips-solutions.com> On 12/20/2016 01:38 AM, Tito Cumpen wrote: I want to implement something similar to https://tools.ietf.org/html/rfc4579 <https://tools.ietf.org/html/rfc4579> On Mon, Dec 19, 2016 at 1:58 PM, Tito Cumpen <t...@xsvoce.com <mailto:t...@xsvoce.com>> wrote: Group or Devs, Is there any way to allow custom bodies during a presence publish? I am trying to implement presence notifies for a conferencing scenario to allow participants of a
Re: [OpenSIPS-Users] uac_replace_from and top_hiding
Hi, Denis! No, topology_hiding is not meant to change the From and To domains. If you want to change them, you have to do it explicitely in the script using the uac_replace_from/to() functions. There are no incompatibilities between topology_hiding and these functions, they should work just fine. Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 12/23/2016 02:49 PM, Denis via Users wrote: Hello! I wander is there any reason to use uac_replace_from function with top_hiding? Or this will not work? Because top_hiding doesn`t change "From domain" field. -- С уважением, Денис. Best regards, Denis ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips crash
Hi, Denis! It's not OpenSIPS making the core, it's the Operating System. And the answer is "yes", if OpenSIPS crashes again, the "old core" will be replaced by a new one. One idea is to configure the Operating System to write the corefile using the pid of the process in the pattern. To do that, run: # echo core.%p > /proc/sys/kernel/core_pattern Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 12/23/2016 11:35 AM, Denis wrote: One question. If in the destination directory of the core file will be located another "core" file, what will be? Would "old core" file be replaced by a new one, or Opensips makes another core file with a fresh data? Thank you. -- С уважением, Денис. Best regards, Denis 23.12.2016, 12:23, "Răzvan Crainea" <raz...@opensips.org>: Please update to the latest 2.2.2. If you still have problems, try to make sure opensips can generate a corefile[1]. [1] http://www.opensips.org/Documentation/TroubleShooting-Crash Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com <http://www.opensips-solutions.com/> On 12/23/2016 11:16 AM, Denis wrote: Hello! Server:: OpenSIPS (2.2.1 (x86_64/linux)) Today i had a crash of Opensips. Everything that i could collect is here https://yadi.sk/i/dyNnXpBr34YJQ3 Unfortunately, i could not find any fresh core file, despite of the fact that Opensips starts with -w /opensipscore option. In opensipscore i found only core file at 29 Nov. Thank you for any help. -- С уважением, Денис. Best regards, Denis ___ Users mailing list Users@lists.opensips.org <mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users , ___ Users mailing list Users@lists.opensips.org <mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips crash
Please update to the latest 2.2.2. If you still have problems, try to make sure opensips can generate a corefile[1]. [1] http://www.opensips.org/Documentation/TroubleShooting-Crash Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 12/23/2016 11:16 AM, Denis wrote: Hello! Server:: OpenSIPS (2.2.1 (x86_64/linux)) Today i had a crash of Opensips. Everything that i could collect is here https://yadi.sk/i/dyNnXpBr34YJQ3 Unfortunately, i could not find any fresh core file, despite of the fact that Opensips starts with -w /opensipscore option. In opensipscore i found only core file at 29 Nov. Thank you for any help. -- С уважением, Денис. Best regards, Denis ___ 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] Using Redis with OpenSIPS 2.1.5 error
Hi, Sami!
Did you compile OpenSIPS redis with a different compiler, or with gcc 5.4?
Can you apply this patch[1] and test again?
[1]
https://github.com/OpenSIPS/opensips/commit/25502536bb2cd03088b83c4997062841a5a238b1.patch
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 12/22/2016 07:52 PM, Sami Montour wrote:
Hello Everyone,
I am trying to use Redis with OpenSIPS 2.1.5 but I am getting an
error (see below) indicating that OpenSIPS is not able to load the
module (cachedb_redis.so). As you can see from my printout below, the
module does exist in the module library directory. Below is my
configuration about Redis, OpenSIPS, and the error.
Any help is very much appreciated. Thanks.
# opensips -c
Dec 21 11:39:04 [15262] ERROR:core:sr_load_module: could not open module
:
/usr/local/opensips/lib64/opensips/modules/cachedb_redis.so: undefined
symbol: get_redis_connection
Dec 21 11:39:04 [15262] ERROR:core:load_module: failed to load module
Dec 21 11:39:04 [15262] CRITICAL:core:yyerror: parse error in config
file /usr/local/opensips//etc/opensips/opensips.cfg, line 158, column
13-14: failed to load module cachedb_redis.so
Dec 21 11:39:04 [15262] ERROR:core:set_mod_param_regex: no module
matching cachedb_redis found Dec 21 11:39:04 [15262]
CRITICAL:core:yyerror: parse error in config file
/usr/local/opensips//etc/opensips/opensips.cfg, line 159, column
18-19: Parameter not found in module -
can't set Dec 21 11:39:04 [15262] ERROR:core:main: bad config file (2
errors)
# ll /usr/local/opensips/lib64/opensips/modules/cachedb_redis.so
-rwxr-xr-x 1 root root 209608 Dec 21 11:16
/usr/local/opensips/lib64/opensips/modules/cachedb_redis.so
opensips.cfg
CacheDB Redis module
loadmodule "cachedb_redis.so"
modparam("cachedb_redis","cachedb_url","redis:group1://localhost:6379/")
# dpkg -l | grep redis
ii libhiredis-dev:amd64 0.13.3-2
amd64
ii libhiredis0.13:amd64 0.13.3-2
amd64
ii redis-server 2:3.0.6-1 amd64
ii redis-tools 2:3.0.6-1 amd64
# opensips -V
version: opensips 2.1.5 (x86_64/linux)
flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC,
F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024,
MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024,
BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et,
sigio_rt, select.
git revision: e381dd5
main.c compiled on 11:14:40 Dec 21 2016 with gcc 5.4.0
# uname -a
Linux rainier 4.4.0-53-generic #74-Ubuntu SMP Fri Dec 2 15:59:10 UTC
2016 x86_64 x86_64 x86_64 GNU/Linux
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Re: [OpenSIPS-Users] WARNING in 2.2
Hi, Agalya! Can you check you have this commit: https://github.com/OpenSIPS/opensips/commit/fd8f6ec442b4365da9d274af6939954246ece865 Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 12/22/2016 04:43 PM, Ramachandran, Agalya (Contractor) wrote: Hi Razvan, Am using latest 2.2 version only. Regards, Agalya *From:*Users [mailto:users-boun...@lists.opensips.org] *On Behalf Of *Razvan Crainea *Sent:* Thursday, December 22, 2016 3:39 AM *To:* users@lists.opensips.org *Subject:* Re: [OpenSIPS-Users] WARNING in 2.2 Hi, Agalya! Are you using the latest OpenSIPS 2.2? If not, upgrade to the latest version and the Warning should no longer appear. Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com <http://www.opensips-solutions.com> On 12/21/2016 06:58 PM, Ramachandran, Agalya (Contractor) wrote: Hi Bogdan, I have installed OpenSIPS 2.2 version. I have no traffic running in the OpenSIPS server. But I see the below warning logs. How to avoid it? I remember you have provided some fix for this WARNING message previously in month of August I guess. I also see this message when I ran traffic, just as a proxy server at 1500 cps. Dec 21 14:32:23 /usr/local/sbin/opensips[28543]: WARNING:core:handle_timer_job: utimer job has a 2 us delay in execution Dec 21 15:39:56 /usr/local/sbin/opensips[28546]: WARNING:core:handle_timer_job: utimer job has a 2 us delay in execution Dec 21 15:52:11 /usr/local/sbin/opensips[28546]: WARNING:core:handle_timer_job: utimer job has a 8 us delay in execution Dec 21 15:53:04 /usr/local/sbin/opensips[28544]: WARNING:core:handle_timer_job: utimer job has a 2 us delay in execution Dec 21 15:56:12 /usr/local/sbin/opensips[28546]: WARNING:core:handle_timer_job: utimer job has a 2 us delay in execution Dec 21 15:56:49 /usr/local/sbin/opensips[28546]: WARNING:core:handle_timer_job: utimer job has a 2 us delay in execution Dec 21 15:57:41 /usr/local/sbin/opensips[28546]: WARNING:core:handle_timer_job: utimer job has a 2 us delay in execution Dec 21 15:58:29 /usr/local/sbin/opensips[28543]: WARNING:core:handle_timer_job: utimer job has a 2 us delay in execution Dec 21 15:59:56 /usr/local/sbin/opensips[28545]: WARNING:core:handle_timer_job: utimer job has a 9 us delay in execution Dec 21 15:59:56 /usr/local/sbin/opensips[28521]: WARNING:core:utimer_ticker: utimer task already scheduled for 87129980 ms (now 87130070 ms), it may overlap.. Regards, Agalya ___ Users mailing list Users@lists.opensips.org <mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] How to store some Sip-header value in variable ?
Hi, Kirill!
You can use the attr_avp [1] in the registrar module to attach the value
to the contact. You could do something like:
modparam("registrar", "attr_avp", "$avp(attr)")
...
if (is_method("REGISTER")) {
$avp(attr) = $hdr(CUSTOM_HEADER);
save("location");
exit;
}
After that, you will be able to see the attribute in the MI commands output.
[1] http://www.opensips.org/html/docs/modules/2.2.x/registrar.html#id293909
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 12/22/2016 09:43 AM, Denis wrote:
Hello
http://www.opensips.org/Documentation/Script-CoreVar-2-2
3.91
--
С уважением, Денис.
Best regards, Denis
22.12.2016, 10:28, "Kirill Galinurov" <k.galinu...@gmail.com>:
Hi All. I need to store some additional info about user from custom
sip header in Register request. How i can do it? Can i get this info
later from ul_dump or ul_show_contact command ?
,
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Re: [OpenSIPS-Users] WARNING in 2.2
Hi, Agalya! Are you using the latest OpenSIPS 2.2? If not, upgrade to the latest version and the Warning should no longer appear. Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 12/21/2016 06:58 PM, Ramachandran, Agalya (Contractor) wrote: Hi Bogdan, I have installed OpenSIPS 2.2 version. I have no traffic running in the OpenSIPS server. But I see the below warning logs. How to avoid it? I remember you have provided some fix for this WARNING message previously in month of August I guess. I also see this message when I ran traffic, just as a proxy server at 1500 cps. Dec 21 14:32:23 /usr/local/sbin/opensips[28543]: WARNING:core:handle_timer_job: utimer job has a 2 us delay in execution Dec 21 15:39:56 /usr/local/sbin/opensips[28546]: WARNING:core:handle_timer_job: utimer job has a 2 us delay in execution Dec 21 15:52:11 /usr/local/sbin/opensips[28546]: WARNING:core:handle_timer_job: utimer job has a 8 us delay in execution Dec 21 15:53:04 /usr/local/sbin/opensips[28544]: WARNING:core:handle_timer_job: utimer job has a 2 us delay in execution Dec 21 15:56:12 /usr/local/sbin/opensips[28546]: WARNING:core:handle_timer_job: utimer job has a 2 us delay in execution Dec 21 15:56:49 /usr/local/sbin/opensips[28546]: WARNING:core:handle_timer_job: utimer job has a 2 us delay in execution Dec 21 15:57:41 /usr/local/sbin/opensips[28546]: WARNING:core:handle_timer_job: utimer job has a 2 us delay in execution Dec 21 15:58:29 /usr/local/sbin/opensips[28543]: WARNING:core:handle_timer_job: utimer job has a 2 us delay in execution Dec 21 15:59:56 /usr/local/sbin/opensips[28545]: WARNING:core:handle_timer_job: utimer job has a 9 us delay in execution Dec 21 15:59:56 /usr/local/sbin/opensips[28521]: WARNING:core:utimer_ticker: utimer task already scheduled for 87129980 ms (now 87130070 ms), it may overlap.. Regards, Agalya ___ 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] rtpproxy and timeout socket
Yes, It seems OpenSIPS is not compatible with rtpproxy 2 version for timeout notifications. That's because OpenSIPS always strips the tcp: from the beginning of the socket, while RTPProxy 2 always waits for this prefix. Please open a bug/feature request on github for us to track this issue: https://github.com/OpenSIPS/opensips/issues Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 12/21/2016 10:59 AM, Denis wrote: Hello, Razvan! 2.2.alpha.20160822 -- С уважением, Денис. Best regards, Denis 21.12.2016, 11:58, "Răzvan Crainea" <raz...@opensips.org>: Hi, Denis! What version of rtpproxy are you using? There might be an incompatibility issue here. Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com <http://www.opensips-solutions.com/> On 12/21/2016 07:37 AM, Denis wrote: Hello! I am using rtpproxy in my VoIP network and now i want to locate it on another server rather then Opensips instance. Rtpproxy is started with such parameters: /usr/local/rtpproxy2/bin/rtpproxy -u -p -l -s udp:: -i -T 10 -n tcp::2229 -m 35000 -M 45000 -r -R -P -d INFO LOG_LOCAL5 The Proxy of rtp packet works fine but when i emulated stop transfer of rtp i see in log such string: ERR::rtpp_command_ul_handle: invalid socket name :2229 What the problem is? Thank you for any help. -- С уважением, Денис. Best regards, Denis ___ Users mailing list Users@lists.opensips.org <mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users , ___ Users mailing list Users@lists.opensips.org <mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Mi_datagram perfomance.
Hi, Kirill! If you want to get info from OpenSIPS remotely, you have two choices: 1. mi_xmlrpc - this goes over TCP and delivers information in XML format. There are a couple of libraries that implement this protocol, so it should be pretty easy to develop a connector. 2. mi_datagram - this goes over UDP and it is more lightweight. The payload is a bit smaller (because there is no xml tags overhead), but it is a bit harder to parse. In terms of performance, the impact on OpenSIPS is not that huge (unless you are doing >1000 queries per second). Hope this helps, Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 12/21/2016 10:36 AM, Kirill Galinurov wrote: Hello all. we want to get some statistic from our opensips. We want to use http://www.opensips.org/html/docs/modules/2.2.x/mi_datagram.html module to send some fifo commands like ul_show, reg_list, dlg_lists and others. What is the right way to get statistics from opensips in realtime. Are there any performance issues in mi_datagram module? I see that mi_datagram module used for visualizing OpenSIPS performance in https://www.youtube.com/watch?v=kfa6NuW7vgk presentation on last opensips summit. ___ 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] rtpproxy and timeout socket
Hi, Denis! What version of rtpproxy are you using? There might be an incompatibility issue here. Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 12/21/2016 07:37 AM, Denis wrote: Hello! I am using rtpproxy in my VoIP network and now i want to locate it on another server rather then Opensips instance. Rtpproxy is started with such parameters: /usr/local/rtpproxy2/bin/rtpproxy -u -p -l -s udp:: -i -T 10 -n tcp::2229 -m 35000 -M 45000 -r -R -P -d INFO LOG_LOCAL5 The Proxy of rtp packet works fine but when i emulated stop transfer of rtp i see in log such string: ERR::rtpp_command_ul_handle: invalid socket name :2229 What the problem is? Thank you for any help. -- С уважением, Денис. Best regards, Denis ___ 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] custom presence bodies
Hi, Tito! The Content-Type indicates how to read and parse the Message body. This is a mandatory header, and without it the client can't know how to interpret the body. The Event header is used by opensips (and clients) to figure out the event that happened. The idea is that the clients will subscribe for a particular event (presence, or conference in your case) and when that event happens, OpenSIPS will send notifies only to those clients registered for that event. Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 12/20/2016 11:37 AM, Tito Cumpen wrote: Razvan, Thanks for pointing that out. Will my subscribers get this xml via notify regardless of the content type? say : application/xml+conf" or application/pidf+xml does the application header or Event:presence header mean anything to opensips? On Tue, Dec 20, 2016 at 12:04 AM, Răzvan Crainea <raz...@opensips.org <mailto:raz...@opensips.org>> wrote: Hi, Tito! I ran your example in an XML validator, and I noticed that your XML has an "error on line 2 at column 86: attributes construct error"[1]. If you add a space before the "state" attribute, the XML gets parsed properly[2]. [1] http://www.utilities-online.info/xmltojson/?save=5a6721b1-55f9-46c2-8bda-cf21e15e38b3-xmltojson <http://www.utilities-online.info/xmltojson/?save=5a6721b1-55f9-46c2-8bda-cf21e15e38b3-xmltojson> [2] http://www.utilities-online.info/xmltojson/?save=320f586b-c2eb-4e73-b564-3f73f0120662-xmltojson <http://www.utilities-online.info/xmltojson/?save=320f586b-c2eb-4e73-b564-3f73f0120662-xmltojson> Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com <http://www.opensips-solutions.com> On 12/20/2016 01:38 AM, Tito Cumpen wrote: I want to implement something similar to https://tools.ietf.org/html/rfc4579 <https://tools.ietf.org/html/rfc4579> On Mon, Dec 19, 2016 at 1:58 PM, Tito Cumpen <t...@xsvoce.com <mailto:t...@xsvoce.com>> wrote: Group or Devs, Is there any way to allow custom bodies during a presence publish? I am trying to implement presence notifies for a conferencing scenario to allow participants of a conference to subscribe to the conference events. The events will send information about who connected to and disconnected with the intention of syncing the UI. I am trying to publish the following: PUBLISH sip:ap5b...@blah.org <mailto:sip%3aap5b...@blah.org> SIP/2.0. Call-ID: 861f3c86868d8a4b00276064d1205e0a@x.x.x.x.x.x <mailto:0a@x.x.x.x.x.x>. CSeq: 2 PUBLISH. From: "Dave Drummond" <sip:ddrummondah...@blah.org <mailto:sip%3addrummondah...@blah.org>>;tag=62944280_a75dd11f_3a41c2f1_a8e175ca. To: <sip:ap5b...@nurseliveconnect.org <mailto:sip%3aap5b...@nurseliveconnect.org>>. Max-Forwards: 70. User-Agent: Mobicents Sip Servlets 3.0.0-SNAPSHOT. Event: presence. Via: SIP/2.0/TCP 192.237.207.220:5080;branch=z9hG4bKa8e175ca_3a41c2f1_7d6e0f5b-7595-4432-849e-500f6d4fe940. Content-Type: application/pidf+xml. Proxy-Authorization: Digest username="ddrummond.b...@blah.org <mailto:ddrummond.b...@blah.org>",realm="blah.org <http://blah.org>",nonce="58585739212cd09ec487455386e318963e82d972f333",uri="sip:ap5b...@blah.org <mailto:sip%3aap5b...@blah.org>",response="670c0336b89491900c284014c12b62ac". Content-Length: 263. . . . . connected. . . . but I get a 415 unsupported media type Dec 19 21:53:15 cloud-server-06 /sbin/opensips[31135]: ERROR:presence_xml:xml_publ_handl: bad body format Dec 19 21:53:15 cloud-server-06 /sbin/opensips[31135]: ERROR:presence:handle_publish: in event specific publish handling Can someone tell me how I can modify the body to make this work? Thanks, Tito ___ Users mailing list Users@lists.opensips.org <mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users <http://lists.opensips.org/cgi-bin/mailman/listinfo/users> ___ Users mailing list Users@lists.opensips.org <mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users <http://lists.opensips.org/cgi-bin/mailman/listinfo/users> ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] rtpproxy compile
Hi, Denis! Can you run 'ldconfig -p | grep sndfile'? Do you see the .so libraries in the output? If not, perhaps you should run 'ldconfig'. If this still does not work, you should manually add the library's directory in the library path[1]. If you do get sndfile in the output, but it still does not work, perhaps the first configure commmand did not get the proper library, so you should re-run './configure'. PS: I also forwarded this question on the RTPProxy mailing list. If this does not work, perhaps somebody out there can help you out. [1] https://codeyarns.com/2014/01/14/how-to-add-library-directory-to-ldconfig-cache/ Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 12/20/2016 08:46 AM, Denis wrote: Hello! I want to use extractaudio utility to extract audio and write it to some .wav file. But, during compiling of this utility i get such error " extractaudio.c:325: undefined reference to `sf_open' extractaudio.c:389: undefined reference to `sf_write_short' extractaudio.c:400: undefined reference to `sf_close' extractaudio.c:389: undefined reference to `sf_write_short' collect2: error: ld returned 1 exit status Makefile:427: recipe for target 'extractaudio' failed make: *** [extractaudio] Error 1 " libsndfile1 package (and dev) had been installed early. May be, somebody, has dealt with the problem? Thank you for any help. -- С уважением, Денис. Best regards, Denis ___ 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] custom presence bodies
Hi, Tito! I ran your example in an XML validator, and I noticed that your XML has an "error on line 2 at column 86: attributes construct error"[1]. If you add a space before the "state" attribute, the XML gets parsed properly[2]. [1] http://www.utilities-online.info/xmltojson/?save=5a6721b1-55f9-46c2-8bda-cf21e15e38b3-xmltojson [2] http://www.utilities-online.info/xmltojson/?save=320f586b-c2eb-4e73-b564-3f73f0120662-xmltojson Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 12/20/2016 01:38 AM, Tito Cumpen wrote: I want to implement something similar to https://tools.ietf.org/html/rfc4579 On Mon, Dec 19, 2016 at 1:58 PM, Tito Cumpen <t...@xsvoce.com <mailto:t...@xsvoce.com>> wrote: Group or Devs, Is there any way to allow custom bodies during a presence publish? I am trying to implement presence notifies for a conferencing scenario to allow participants of a conference to subscribe to the conference events. The events will send information about who connected to and disconnected with the intention of syncing the UI. I am trying to publish the following: PUBLISH sip:ap5b...@blah.org <mailto:sip%3aap5b...@blah.org> SIP/2.0. Call-ID: 861f3c86868d8a4b00276064d1205e0a@x.x.x.x.x.x. CSeq: 2 PUBLISH. From: "Dave Drummond" <sip:ddrummondah...@blah.org <mailto:sip%3addrummondah...@blah.org>>;tag=62944280_a75dd11f_3a41c2f1_a8e175ca. To: <sip:ap5b...@nurseliveconnect.org <mailto:sip%3aap5b...@nurseliveconnect.org>>. Max-Forwards: 70. User-Agent: Mobicents Sip Servlets 3.0.0-SNAPSHOT. Event: presence. Via: SIP/2.0/TCP 192.237.207.220:5080;branch=z9hG4bKa8e175ca_3a41c2f1_7d6e0f5b-7595-4432-849e-500f6d4fe940. Content-Type: application/pidf+xml. Proxy-Authorization: Digest username="ddrummond.b...@blah.org <mailto:ddrummond.b...@blah.org>",realm="blah.org <http://blah.org>",nonce="58585739212cd09ec487455386e318963e82d972f333",uri="sip:ap5b...@blah.org <mailto:sip%3aap5b...@blah.org>",response="670c0336b89491900c284014c12b62ac". Content-Length: 263. . . . . connected. . . . but I get a 415 unsupported media type Dec 19 21:53:15 cloud-server-06 /sbin/opensips[31135]: ERROR:presence_xml:xml_publ_handl: bad body format Dec 19 21:53:15 cloud-server-06 /sbin/opensips[31135]: ERROR:presence:handle_publish: in event specific publish handling Can someone tell me how I can modify the body to make this work? Thanks, Tito ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips 2.2 and CANCEL transaction
Hi, Denis! According to the SIP RFC, if the Proxy doesn't receive any reply, it will not generate a CANCEL. Are you sending at least 100 Trying between your servers? Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 12/19/2016 12:42 PM, Denis wrote: Hello! Server:: OpenSIPS (2.2.2 (x86_64/linux)) I am going to use Opensips as top_hiding, load balancer and proxy instance. My scheme looks as follows. The first instance: top_hiding and load balancer. It listens 5060 and 5068 ports. The second instance: proxy. It listens 5065 port. The scheme of the call: Caller SIP UA -> first instance (makes top_hiding and load balancer) -> second instance (makes routing and setup destination URI) -> first instance (makes top_hiding) -> Callee SIP UA. This scheme of the call seems to be working correctly (from perspective of SIP signalling) for: - successful call (i.e. call with answer) - Cancel transaction during ringing of th callee But when i emulated situation, when callee becomes unreachable (internet problem or other) the PROXY doesn`t send CANCEL to the first instance on the third leg. tcpdump capture of the problem call you can find here. https://yadi.sk/d/ase8fSjw342eBJ Where Caller - 192.168.18.150 the first instance and the second are located on the same server (IP address - 172.31.0.10) Thank you for any help. -- Best regards, Denis -- С уважением, Путято Денис. ___ 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] How to Keep OpenSIPS in reverse path
So as far as I understand, you receive a request and you want Freeswitch
to route it back to OpenSIPS. There is no SIP mechanism that can achieve
this - you need to configure Freeswitch so that when it gets a message
from OpenSIPS, determine the gateway, and send it back. Note that you
need to set opensips's IP in the destination URI, not the request URI,
otherwise OpenSIPS will not know the destination GW.
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 12/15/2016 01:33 PM, M. Salman wrote:
Hi,
Generally there are three ways of doing this:
1) Use SIP sever as an Edge Proxy (check rfc5626, should be a quick read)
2) Create spiral, route SIP call from proxy to your media-server and
then back to proxy from media server.
3) Use SIP sever as a registrar and then forward registrations to
media sever and maintaining it. (cheap way though)
Regards,
Salman
On Thu, Dec 15, 2016 at 4:12 PM, Muhammad Naseer Bhatti
<nbha...@gmail.com <mailto:nbha...@gmail.com>> wrote:
So the call hits the proxy and dispatched to the media server.
Media server makes the gateway selection and now need to send the
call to the provider. Instead of the signaling to be sent directly
from the media server and the provider I and trying to pass the
signaling back through the proxy so the outgoing connections to
the provider will be seen by the proxy IP address not the media
server. Both the proxy and media servers are not behind nat.
--
Sent with Airmail
From: Răzvan Crainea <raz...@opensips.org>
<mailto:raz...@opensips.org>
Reply: OpenSIPS users mailling list <users@lists.opensips.org>
<mailto:users@lists.opensips.org>
Date: December 15, 2016 at 12:55:18 PM
To: users@lists.opensips.org <mailto:users@lists.opensips.org>
<users@lists.opensips.org> <mailto:users@lists.opensips.org>
Subject: Re: [OpenSIPS-Users] How to Keep OpenSIPS in reverse path
I am sorry, but I don't understand your call flow. Please present
here the call flow you have now, and the expected one.
PS: not sure why you are looking at the Via header, that's only
used for replies, not for requests.
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com <http://www.opensips-solutions.com>
On 12/14/2016 07:53 PM, Muhammad Naseer Bhatti wrote:
Hi Razvan,
I am not using REGISTER, but I guess add_path() wont’ work for
me, I am using record_route() for the INVITE though.
…
if (is_method("INVITE")) {
record_route();
}
…
On the media server I see the Via header,
INVITE sip:6054775550@64.58.228.102:
<http://sip:6054775550@64.58.228.102:> SIP/2.0
Record-Route:
<sip:64.58.228.102:;lr=on;ftag=CT60RBHd.BRAp6IAQSKwtAN3Mjj2bPKL>
Via: SIP/2.0/UDP
64.58.228.102:;branch=z9hG4bK31d2.b8fe57fe9c7545e354a93f391a1d0704.0
Via: SIP/2.0/UDP
172.16.0.101:52207;received=172.16.0.101;rport=52207;branch=z9hG4bKPjyGufepGSH2zMGfp6J.CIKvEgl87YIPFN
but when the media server sends
recv 1432 bytes from udp/[64.58.228.89]: at 12:44:12.709234:
INVITE sip:6054775550@23.29.112.144:15080
<http://sip:6054775550@23.29.112.144:15080> SIP/2.0
Via: SIP/2.0/UDP
64.58.228.89:;rport;branch=z9hG4bKF8jFHt4maUBeQ
Max-Forwards: 68
From: "Naseer" <sip:1234@64.58.228.89
<mailto:sip%3A1234@64.58.228.89>>;tag=45yZ9S13S47HF
Not sure what I am doing wrong. Here is my script,
http://pastebin.com/Cmnxnf4c <http://pastebin.com/Cmnxnf4c>
--
Sent with Airmail
From: Răzvan Crainea <raz...@opensips.org>
<mailto:raz...@opensips.org>
Reply: OpenSIPS users mailling list <users@lists.opensips.org>
<mailto:users@lists.opensips.org>
Date: December 14, 2016 at 8:34:22 PM
To: users@lists.opensips.org <mailto:users@lists.opensips.org>
<users@lists.opensips.org> <mailto:users@lists.opensips.org>
Subject: Re: [OpenSIPS-Users] How to Keep OpenSIPS in reverse path
Hi, Muhammad!
The add_path() function should only be called on REGISTER
messages, and it adds a Path header (not a Via). Do you see
this header in your REGISTER message?
For sequential requests, you should use the record_route()
mechanism.
Best regards,
Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com
<http://www.opensips-solutions.com>
On 12/14/2016 06:39 PM, Muhammad Naseer Bhatti wrote:
I am using dispatcher to distribute calls to multiple media
servers, but also want the reply to go through OpenSIPS. So
far I have tried add_path() function which add the Via header
but
Re: [OpenSIPS-Users] How to Keep OpenSIPS in reverse path
I am sorry, but I don't understand your call flow. Please present here
the call flow you have now, and the expected one.
PS: not sure why you are looking at the Via header, that's only used for
replies, not for requests.
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 12/14/2016 07:53 PM, Muhammad Naseer Bhatti wrote:
Hi Razvan,
I am not using REGISTER, but I guess add_path() wont’ work for me, I
am using record_route() for the INVITE though.
…
if (is_method("INVITE")) {
record_route();
}
…
On the media server I see the Via header,
INVITE sip:6054775550@64.58.228.102:
<http://sip:6054775550@64.58.228.102:> SIP/2.0
Record-Route:
<sip:64.58.228.102:;lr=on;ftag=CT60RBHd.BRAp6IAQSKwtAN3Mjj2bPKL>
Via: SIP/2.0/UDP
64.58.228.102:;branch=z9hG4bK31d2.b8fe57fe9c7545e354a93f391a1d0704.0
Via: SIP/2.0/UDP
172.16.0.101:52207;received=172.16.0.101;rport=52207;branch=z9hG4bKPjyGufepGSH2zMGfp6J.CIKvEgl87YIPFN
but when the media server sends
recv 1432 bytes from udp/[64.58.228.89]: at 12:44:12.709234:
INVITE sip:6054775550@23.29.112.144:15080
<http://sip:6054775550@23.29.112.144:15080> SIP/2.0
Via: SIP/2.0/UDP 64.58.228.89:;rport;branch=z9hG4bKF8jFHt4maUBeQ
Max-Forwards: 68
From: "Naseer" <sip:1234@64.58.228.89
<mailto:sip%3A1234@64.58.228.89>>;tag=45yZ9S13S47HF
Not sure what I am doing wrong. Here is my script,
http://pastebin.com/Cmnxnf4c
--
Sent with Airmail
From: Răzvan Crainea <raz...@opensips.org> <mailto:raz...@opensips.org>
Reply: OpenSIPS users mailling list <users@lists.opensips.org>
<mailto:users@lists.opensips.org>
Date: December 14, 2016 at 8:34:22 PM
To: users@lists.opensips.org <mailto:users@lists.opensips.org>
<users@lists.opensips.org> <mailto:users@lists.opensips.org>
Subject: Re: [OpenSIPS-Users] How to Keep OpenSIPS in reverse path
Hi, Muhammad!
The add_path() function should only be called on REGISTER messages,
and it adds a Path header (not a Via). Do you see this header in your
REGISTER message?
For sequential requests, you should use the record_route() mechanism.
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 12/14/2016 06:39 PM, Muhammad Naseer Bhatti wrote:
I am using dispatcher to distribute calls to multiple media servers,
but also want the reply to go through OpenSIPS. So far I have
tried add_path() function which add the Via header but FreeSWITCH
sends the call directly to the gateway not sending the call back
through OpenSIPS. I am not sure if I am doing something wrong, how
can I achieve that?
--
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Re: [OpenSIPS-Users] How to Keep OpenSIPS in reverse path
Hi, Muhammad! The add_path() function should only be called on REGISTER messages, and it adds a Path header (not a Via). Do you see this header in your REGISTER message? For sequential requests, you should use the record_route() mechanism. Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 12/14/2016 06:39 PM, Muhammad Naseer Bhatti wrote: I am using dispatcher to distribute calls to multiple media servers, but also want the reply to go through OpenSIPS. So far I have tried add_path() function which add the Via header but FreeSWITCH sends the call directly to the gateway not sending the call back through OpenSIPS. I am not sure if I am doing something wrong, how can I achieve that? -- Sent with Airmail ___ 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] [IMPORTANT] MAINTENANCE: opensips.org email and mailing lists servers migration
Hello, everyone! Both email and mailing lists servers have been successfully moved to a new platform. You are now free to use them at your convenience. Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 12/12/2016 06:38 PM, Răzvan Crainea wrote: Hello everybody! Wednesday, 14th of December 2016, 10:00 EET[1], we are planning to migrate some opensips.org servers to a new platform. Therefore the email server and mailing lists servers will be down for about 2 hours. During this timeframe[1] we recommend you to avoid sending any emails to either an @opensips.org or @lists.opensips.org email, because they might not be received and they will get lost. As soon as the migration is completed, we will get back with a notification. [1] https://www.timeanddate.com/worldclock/fixedtime.html?msg=MAINTENANCE%3A+opensips.org+email+and+mailing+lists+server+migration=20161214T10=49=2 Best regards, ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] [IMPORTANT] MAINTENANCE: opensips.org email and mailing lists servers migration
Hello everybody! Wednesday, 14th of December 2016, 10:00 EET[1], we are planning to migrate some opensips.org servers to a new platform. Therefore the email server and mailing lists servers will be down for about 2 hours. During this timeframe[1] we recommend you to avoid sending any emails to either an @opensips.org or @lists.opensips.org email, because they might not be received and they will get lost. As soon as the migration is completed, we will get back with a notification. [1] https://www.timeanddate.com/worldclock/fixedtime.html?msg=MAINTENANCE%3A+opensips.org+email+and+mailing+lists+server+migration=20161214T10=49=2 Best regards, -- Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] 回复： what is the problem with
Hi, James! That error indicates that the resources list is not valid. All reasorces should be alphanumeric and the quantity should be numeric. Are you using a resource name that has a dash or something in it (like calls_available=100)? Can you paste the resources list you are provisioning? Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 12/09/2016 03:35 AM, james wrote: Hello： I am trying with Lab-9 and add load balance from CP, Group ID, Destination URL(sip:54.160.131.18:5600), Resource, Probe Mode and description. the format of URL is wrong, but I do not know why so. -- 原始邮件 -- *发件人:* "Răzvan Crainea";<raz...@opensips.org>; *发送时间:* 2016年12月8日(星期四) 晚上8:01 *收件人:* "OpenSIPS users mailling list"<users@lists.opensips.org>; *主题:* Re: [OpenSIPS-Users] what is the problem with Hi, James! Are you specifying any resources for your destination? Resources should be something like: pstn=100;conf=10 Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 12/08/2016 01:32 PM, james wrote: Hello: If I enter this: sip:54.160.131.18:5600 from load balance on CP, after click "save", the system shows errors: !!! Data format is incorrect!. Should be name1=value1;name2=value2? what is the problem? ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] what is the problem with
Hi, James! Are you specifying any resources for your destination? Resources should be something like: pstn=100;conf=10 Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 12/08/2016 01:32 PM, james wrote: Hello: If I enter this: sip:54.160.131.18:5600 from load balance on CP, after click "save", the system shows errors: !!! Data format is incorrect!. Should be name1=value1;name2=value2? what is the problem? ___ 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] is_ip_registered in invite
Hi, Schneur! The second parameter of the is_ip_registered() function[1] should be the AOR of the caller, in the sip:SIP_USER@SIP_DOMAIN format. The source IP is only checked against the contacts of that specific subscriber. However, if I understand correctly, your problem is determining what is the correct AOR to use, because the From username and domain might be different between REGISTER and INVITE, right? If that's the case, you don't have that many choices: either you search through all registered IPs (but there is no OpenSIPS function to do that, so you'll need someting external as you've already done), or you create some sort of mappings between the REGISTER and INVITE users/formats. Or you impose your customers to comply with a specific format, that can help you figure out the mapping. [1] http://www.opensips.org/html/docs/modules/2.2.x/registrar.html#id294953 Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 12/04/2016 01:20 PM, Schneur Rosenberg wrote: Hi, I would like to check during the initial invite if the request comes from a IP that is registered (I don't care about the user credentials at this time), I use it to know if the invite is from a registered user or if it is from a unauthenticated source (DID's or hacking attempt) I can't use is_contact_registered() because not all clients send the user name in the initial invite, and they only send it in the authentication username which is absent in the initial invite, therefore I want to use is_ip_registered() but I'm having issues and I don't understand exactly what the second parameter is for, I want to check for the ip in the $si variable if it is registered (either in the contact field or in the received field). When leaving blank the AOR field, some devices work well but some don't. Due to NAT some devices register the IP in the contact field, and some in the received field, I want to try to match to either one, and it should parse the contact field that it should ignore the username from the contact field. I was doing a avp_db_query() until now, but it had 2 major issues. 1) It runs a MYSQL query on each REQUEST which reduces performance, I couldn't use memcache because IP's are dynamic in nature. 2) I use db_mode 2 on usrloc and it takes about a minute for the registration to appear in the DB and the user can't call out during that minute, and even worse if he tries multiple times and it gets rejected my iptables will block his IP. thank you S. Rosenberg ___ 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:dialog:dlg_onreq_out: No outgoing contact in the initial INVITE
Hi, Adrian!
That message is generated because the CANCEL message does not have a
Contact header.
The error message is indeed misleading. I will turn this into a debug
message and adjust the message.
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 12/03/2016 10:56 PM, Adrian Fretwell wrote:
Help please I have been stuck on this for days.
Opensips v2.2.1 and using topology hiding. Everything works just fine
but if a call gets cancelled I see the following message in the log:
Dec 3 18:41:00 sip-01
/usr/local/opensips-pxy-2.2.1/sbin/opensips[10407]:
ERROR:dialog:dlg_onreq_out: No outgoing contact in the initial INVITE
This error is generated as soon as the cancel comes in, I don't think
it is related to any other messages. Following the error I see the 487
coming in on the on_reply route.
In the script I have the usual cancel handling:
if (is_method("CANCEL")) {
if (t_check_trans()) {
if (!t_relay()) {
sl_reply_error();
}
}
exit;
}
The packet trace shows the cancel working perfectly, I just can't see
the problem. The Cancel request is shown below:
CANCEL sip:2...@vox1.lantecsip.co.uk:5060 SIP/2.0
Via: SIP/2.0/UDP 92.24.10.16:40518;branch=z9hG4bK3446811728
From: "0xx60" <sip:x...@vox1.xx.xx.uk>;tag=1329526351
To: <sip:2...@vox1.xxx.xx.uk>
Call-ID: 2345909838@192.168.6.42
CSeq: 2 CANCEL
Max-Forwards: 70
User-Agent: Yealink SIP-W52P 25.73.179.4
Content-Length: 0
Kind regards,
Adrian Fretwell
The Old School house
Top Green
Sibthorpe
Nottinghamshire
NG23 5PN.
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Re: [OpenSIPS-Users] Ipv6 on non-default port not working
Hi, Robert! Are you using only a single listening interface, on port 5062? Or you're doing more complex scenarios? You're saying that you are not seeing any debugging logs when a message is sent to the 5062 port? Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 12/03/2016 01:52 AM, Robert Dyck wrote: I have been doing some testing with opensips 2.2.2 using ipv6. I found that the server will only respond to a request over IPV6 if it is configured to listen on the default port. Wireshark sees a request addressed to the server but there is no reply. Running opensips in the foreground show no activity associated with a request. Netstat however shows the listening port. ( UDP 5062 ) Tried with global address and ULA. IPv4 works with non-default port. ___ 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] Conflicting information from commands 'opensipsctl ul show' and ' opensipsctl fifo list_tcp_conns '.
Hi, Rodrigo! Before removing A from the user location, did you do an opensips ul show to see what registrations OpenSIPS knows? Are there multiple registrations? Are you deleting all of them? Opening a TCP connection to opensips doesn't necessarily mean that the client also sent a REGISTER message, and therefore the client is not yet registered from SIP perspective. What you might see there (with port 48695) might be an old (bogus) registration. After a while, when the client registers, you see the correct info. Best regards, Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com On 11/29/2016 10:33 PM, Rodrigo Pimenta Carvalho wrote: Hi. I have peer A and peer B online on my OpenSIPS. After removing peer A from table location and reseting peer A, I have: Connection:: ID=29 Type=tcp State=0 Source=127.0.0.1:*52887* Destination=127.0.0.1:5060 Lifetime=1970-01-01 08:44:41 It means that peer A is online on OpenSIPS via TCP socket with port 52887. This is the result of command '/opensipsctl fifo list_tcp_conns/'. However, the commad '/opensipsctl ul show/' gives me: AOR:: intercomA_5dtUWgwgqzR6 Contact:: sip:intercomA_5dtUWgwgqzR6@127.0.0.1:*48694*;transport=TCP;ob Q= Expires:: 479 Callid:: 5694778c-6178-4e80-bf2e-7a4dc0deb5d1 Cseq:: 37504 User-agent:: n/a State:: CS_SYNC Flags:: 0 Cflags:: Socket:: tcp:127.0.0.1:5060 Methods:: 8063 Attr:: in_same_network SIP_instance:: It means that peer A is online on OpenSIPS via TCP socket with port 48694. So, I have a kind of conflict here. How can it be possible? So, if peer A calls peer B, when B answers I can see the following log: Jan 1 08:36:14 [435] ERROR:core:tcpconn_async_connect: failed to retrieve SO_ERROR [server=127.0.0.1:*48694*] (111) Connection refused Why such behavior does exist in OpenSIPS? How to avoid it? And after a while a new TCP connection appered in port 52887. Like this: Contact:: sip:intercomA_5dtUWgwgqzR6@127.0.0.1:52887;transport=TCP;ob Q= Expires:: 236 Callid:: 96672dc5-a98c-468e-a07a-aca27748791a Cseq:: 25094 User-agent:: n/a State:: CS_SYNC Flags:: 0 Cflags:: Socket:: tcp:127.0.0.1:5060 Methods:: 8063 Attr:: in_same_network SIP_instance:: Could it be a problem in OpenSIPS? Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ___ 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] B2BUA, authenticated INVITEs with "ACK for a negative reply"
On 11/29/2016 10:36 AM, Nathan Ward wrote: On 29/11/2016, at 9:26 PM, Răzvan Crainea <raz...@opensips.org> wrote: On 11/29/2016 04:09 AM, Nathan Ward wrote: On 29/11/2016, at 5:25 AM, Răzvan Crainea <raz...@opensips.org> wrote: Hi, Nathan! Have you tried calling b2b_init_request() with the "a" flag [1]? [1] http://www.opensips.org/html/docs/modules/2.2.x/b2b_logic.html#id294010 Hi, Yes I have. This passes through the authentication challenge headers in the 401/407, and then any subsequent response headers in the new INVITE. b2b_logic/logic.c:1239 calls b2b_mark_todel, after forwarding the message - because it is marked to_del, the ACK that the originator of the INVITE sends in response to the 401/407 deletes the session. I don’t understand how this flag is intended to be used, as there doesn’t seem to be anything in the code to avoid setting to_del if the response is a 401/407 (or anything >=300, actually) with auth challenge headers. All it does is pass through the headers, but as it deletes the session, a new Call-ID is issued by B2BUA when the authenticated invite is generated. Hi, Nathan! Yes, you are right, the flag simply passes the auth headers between the two legs. So you were saying that you were only using b2b for topology hiding? If so, why not using directly the topology_hiding module[1]? Sure that’s an option, however I would like to understand the B2BUA module better. What is the use case for passing authentication headers if the B2BUA instance is shut down when a challenge (401/407) passes through? Hi, Nathan! From SIP perspective, the authentication mechanism is completely independent from the message/call. That's why you can even use the same credentials for different calls, as long as the nonce does not change. So the auth server, does not have to map the credentials with a call-id. Some servers might enforce this requirement - however, unfortunately those will not work with opensips B2B. From OpenSIPS perspective, when it receives the 401/407, the transaction will be terminated, and the B2B will destroy the associated entities. When the new INVITE comes in, it will create a completly new entity, that will contain a different Callid (and will be seen as a new leg). These two entities are not corelated at all in the current code. That's why for now the current B2B implementation does not support your scenario. Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] handle_timer_job delay in execution
Hi, Adrian! What version of OpenSIPS are you using? Can you update to the latest version? Can you also run an "opensipsctl ps" and send the output back? Also, can you send the list of the modules you are using? Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 11/28/2016 04:42 PM, Adrian Fretwell wrote: Hi Razvan, No other users of the mysql, everything looks idle. The machine is quite powerful a Dell PowerEdge Server with Dual Intel Xeon CPUs (12 cores/24 threads) and 32Gb Memory and a three disk RAID 5 array. Here is the parameters from config: children=4 dns=no rev_dns=no # for customers listen=udp:XX.XX.XX.X1:5060 # for gateways listen=udp:XX.XX.XX.X2:5060 # for internal media etc. listen=udp:XX.XX.XX.X1:8060 From /etc/defaults # Amount of shared memory to allocate for the running OpenSIPS server (in Mb) S_MEMORY=512 # Amount of pkg memory to allocate for the running OpenSIPS server (in Mb) P_MEMORY=16 Kind regards, Adrian. On 28/11/16 13:42, Răzvan Crainea wrote: Are you doing any DB queries that might take a lot of time? Also, how many children are you using? Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 11/28/2016 03:33 PM, Adrian Fretwell wrote: Hi Razvan, I don't see the error regularly, I think that's what is making it hard to track down, they just seem to randomly occur. Anyway here is the output of the fifo command, I will check this as soon as I get the next error. tm:received_replies:: 177 tm:relayed_replies:: 111 tm:local_replies:: 6 tm:UAS_transactions:: 59 tm:UAC_transactions:: 3 tm:2xx_transactions:: 56 tm:3xx_transactions:: 0 tm:4xx_transactions:: 6 tm:5xx_transactions:: 0 tm:6xx_transactions:: 0 tm:inuse_transactions:: 0 Kind regards, Adrian Fretwell The Old School house Top Green Sibthorpe Nottinghamshire NG23 5PN. On 28/11/16 13:08, Răzvan Crainea wrote: Hi, Adrian! Are you still seeing those errors, even if there is no traffic? Can you check if you have any hung transactions? Just run: scripts/opensipsctl fifo get_statistics tm: Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 11/28/2016 12:57 PM, Adrian Fretwell wrote: Hello, help please. Can anyone give me some direction on how to diagnose what is causing timer job delays in execution? This SIP proxy (2.2.1) has virtually no load, you can see from the timestamps in the log extract and yet there is still a delay. I don't know where to start looking. Nov 28 08:50:37 sip-01 /usr/local/opensips-pxy-2.2.1/sbin/opensips[22176]: main: 185.40.4.47 Relay denied. Nov 28 08:51:14 sip-01 /usr/local/opensips-pxy-2.2.1/sbin/opensips[22193]: WARNING:core:handle_timer_job: timer job has a 7 us delay in execution Nov 28 08:54:22 sip-01 /usr/local/opensips-pxy-2.2.1/sbin/opensips[22178]: main: 185.40.4.198 Relay denied. Kind regards, Adrian Fretwell The Old School house Top Green Sibthorpe Nottinghamshire NG23 5PN. T: 01636 525360 M: 07850 756603 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] B2BUA, authenticated INVITEs with "ACK for a negative reply"
On 11/29/2016 04:09 AM, Nathan Ward wrote: On 29/11/2016, at 5:25 AM, Răzvan Crainea <raz...@opensips.org> wrote: Hi, Nathan! Have you tried calling b2b_init_request() with the "a" flag [1]? [1] http://www.opensips.org/html/docs/modules/2.2.x/b2b_logic.html#id294010 Hi, Yes I have. This passes through the authentication challenge headers in the 401/407, and then any subsequent response headers in the new INVITE. b2b_logic/logic.c:1239 calls b2b_mark_todel, after forwarding the message - because it is marked to_del, the ACK that the originator of the INVITE sends in response to the 401/407 deletes the session. I don’t understand how this flag is intended to be used, as there doesn’t seem to be anything in the code to avoid setting to_del if the response is a 401/407 (or anything >=300, actually) with auth challenge headers. All it does is pass through the headers, but as it deletes the session, a new Call-ID is issued by B2BUA when the authenticated invite is generated. Hi, Nathan! Yes, you are right, the flag simply passes the auth headers between the two legs. So you were saying that you were only using b2b for topology hiding? If so, why not using directly the topology_hiding module[1]? [1] http://www.opensips.org/html/docs/modules/2.2.x/topology_hiding.html Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] B2BUA, authenticated INVITEs with "ACK for a negative reply"
Hi, Nathan! Have you tried calling b2b_init_request() with the "a" flag [1]? [1] http://www.opensips.org/html/docs/modules/2.2.x/b2b_logic.html#id294010 Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 11/26/2016 06:04 AM, Nathan Ward wrote: Hi all, I am configuring an SBC with 3 legs - one to the core (i.e. to a registrar and routing server), one towards end users, and one towards a provider. My intent is to make he SBC fairly “dumb”, and not keep state of registrations etc. The provider requires registration and authentication for calls. I generate registrations from our core towards our provider for each line, through the SBC (forwarding these rather than B2BUA-ing). I also have users registering towards our core. Works great. When we forward an INVITE from our core, the B2BUA creates a new session and forwards it to the provider. The provider challenges (401), which is forwarded back towards the core. The core ACKs this challenge, and forwards the ACK to the provider. Then, the B2BUA deletes the dialog after saying "ACK for a negative reply”. This means that the subsequent authenticated INVITE generates a new instance on the B2BUA, and a new Call-ID - which causes the provider to reject this as the Call-ID is expected to be consistent across auth/unauth INVITEs. Worth noting that before we call b2b_init_request, I call “force_send_socket”, as we use loopback/virtual addresses for talking with our provider. - Is this expected behaviour? - Is there a way to tweak this so that ACK for a 401/407/etc. does not immediately tear down the B2BUA entity? - Can I avoid this by writing my own B2BUA scenario? We are using the built in “top hiding” for the moment. If code is required to permit this model I’m happy to work on this, but before I get my hands dirty I thought I’d ask around :-) We have the same behaviour from User->B2BUA->Core - however for the moment our Core doesn’t care about differing Call-ID, which is obviously a problem that I’ll be looking at next..! -- Nathan Ward ___ 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] handle_timer_job delay in execution
Are you doing any DB queries that might take a lot of time? Also, how many children are you using? Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 11/28/2016 03:33 PM, Adrian Fretwell wrote: Hi Razvan, I don't see the error regularly, I think that's what is making it hard to track down, they just seem to randomly occur. Anyway here is the output of the fifo command, I will check this as soon as I get the next error. tm:received_replies:: 177 tm:relayed_replies:: 111 tm:local_replies:: 6 tm:UAS_transactions:: 59 tm:UAC_transactions:: 3 tm:2xx_transactions:: 56 tm:3xx_transactions:: 0 tm:4xx_transactions:: 6 tm:5xx_transactions:: 0 tm:6xx_transactions:: 0 tm:inuse_transactions:: 0 Kind regards, Adrian Fretwell The Old School house Top Green Sibthorpe Nottinghamshire NG23 5PN. On 28/11/16 13:08, Răzvan Crainea wrote: Hi, Adrian! Are you still seeing those errors, even if there is no traffic? Can you check if you have any hung transactions? Just run: scripts/opensipsctl fifo get_statistics tm: Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 11/28/2016 12:57 PM, Adrian Fretwell wrote: Hello, help please. Can anyone give me some direction on how to diagnose what is causing timer job delays in execution? This SIP proxy (2.2.1) has virtually no load, you can see from the timestamps in the log extract and yet there is still a delay. I don't know where to start looking. Nov 28 08:50:37 sip-01 /usr/local/opensips-pxy-2.2.1/sbin/opensips[22176]: main: 185.40.4.47 Relay denied. Nov 28 08:51:14 sip-01 /usr/local/opensips-pxy-2.2.1/sbin/opensips[22193]: WARNING:core:handle_timer_job: timer job has a 7 us delay in execution Nov 28 08:54:22 sip-01 /usr/local/opensips-pxy-2.2.1/sbin/opensips[22178]: main: 185.40.4.198 Relay denied. Kind regards, Adrian Fretwell The Old School house Top Green Sibthorpe Nottinghamshire NG23 5PN. T: 01636 525360 M: 07850 756603 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] handle_timer_job delay in execution
Hi, Adrian! Are you still seeing those errors, even if there is no traffic? Can you check if you have any hung transactions? Just run: scripts/opensipsctl fifo get_statistics tm: Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 11/28/2016 12:57 PM, Adrian Fretwell wrote: Hello, help please. Can anyone give me some direction on how to diagnose what is causing timer job delays in execution? This SIP proxy (2.2.1) has virtually no load, you can see from the timestamps in the log extract and yet there is still a delay. I don't know where to start looking. Nov 28 08:50:37 sip-01 /usr/local/opensips-pxy-2.2.1/sbin/opensips[22176]: main: 185.40.4.47 Relay denied. Nov 28 08:51:14 sip-01 /usr/local/opensips-pxy-2.2.1/sbin/opensips[22193]: WARNING:core:handle_timer_job: timer job has a 7 us delay in execution Nov 28 08:54:22 sip-01 /usr/local/opensips-pxy-2.2.1/sbin/opensips[22178]: main: 185.40.4.198 Relay denied. Kind regards, Adrian Fretwell The Old School house Top Green Sibthorpe Nottinghamshire NG23 5PN. T: 01636 525360 M: 07850 756603 ___ 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] Want to create opensips subscriber from restful api or from web application
Hi, Vishal!
You could insert directly a subscriber in the database, using something
like:
insert into subscriber (username, domain, password, ha1, ha1b) values
('100', 'opensips.org', 'password', md5(concat(username, ':', domain,
':', password)), md5(concat(username, '@', domain, ':', domain, ':',
password)));
You can find more info here:
https://www.opensips.org/Documentation/TipsFAQ#toc2
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 11/28/2016 06:50 AM, vishal dubey wrote:
Team,
Any suggestion on my earlier request "Want to create opensips
subscriber from restful api or from web application".
I want to add subscribers from other web application. Is there any
webservice or other way available to do this?
Thanks, in advance.
On Monday, 21 November 2016 6:21 PM, vishal dubey
<dubeyvis...@yahoo.com> wrote:
Hi Team,
I want to create opensips subscriber from other web application. I
think it is possible through pi_httpd or db_httpd. but i am not able
to find any example hot to do this.
Please help.
Thanks,
Vishal
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Re: [OpenSIPS-Users] Actions that apply to all branches
Hi, Adrian! I don't think you will find such a list. Usually the idea is to do the changes that affect all branches in the main route, and if you want to do per-branch changes, do them in the branch_route. Are you sure that force_send_socket() only applies to the main branch? If you call force_send_socket() in the main route, and then add a second branch, the messages leave on different interfaces? Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 11/26/2016 12:19 PM, Adrian Fretwell wrote: Hello, I understand from the documentation that after calling append_branch() any further URI manipulations only apply to the main branch, but I am trying to understand what actions will still apply over all the branches. For example I discovered that rtpproxy_offer() appears to affect all branches, but force_send_socket() only apples to the main branch. Is there a list or rule that will help me work out what I need to apply before branching and what I can leave until later, perhaps until just before t_relay() is called? I hope my question makes sense. Kind regards, Adrian Fretwell The Old School house Top Green Sibthorpe Nottinghamshire NG23 5PN. T: 01636 525360 M: 07850 756603 This electronic message contains information from A-Squared Engineering Services which may be privileged or confidential. The information is intended to be for the use of the individual(s) or entity(s) named above. If you are not the intended recipient be aware that any disclosure, copying, distribution or use of the contents of this information is prohibited. If you have received this electronic message in error, please notify us by telephone or email to adrian.fretw...@topgreen.co.uk immediately. ___ 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] $ai transformation
Hi, Ehrny!
I've just tested, and for me it works as it should - the reply goes
through the interface it came from. Is there any chance you could send
me your script (privately)? Perhaps I can spot some problemsby looking
at it.
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 11/22/2016 07:45 PM, Ehrny wrote:
Hi Răzvan,
I’ve tried both
# force_send_socket(udp:10.197.26.170:5060);
and
$fs="udp:10.197.26.170:5060";
with no luck ((
Without any of these lines it doesn’t work either ((
Ps.. in the main request routing logic I use
force_send_socket(udp:x.x.82.39:5060);
Best Regards,
Ehrny
*From:*users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] *On Behalf Of *Razvan Crainea
*Sent:* Tuesday, November 22, 2016 8:33 PM
*To:* users@lists.opensips.org
*Subject:* Re: [OpenSIPS-Users] $ai transformation
Hi, Ehrny!
If you completely remove the force_send_socket() and any $fs settings
in onreply_route, is it doing the same thing?
Normally the reply should be automatically routed through the same
interface the request came from. Not sure why your reply goes over the
other one. I will try to replicate this and let you know if it works
or not.
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com <http://www.opensips-solutions.com>
On 11/21/2016 06:40 PM, Ehrny wrote:
Hi Răzvan,
Thanks for your help.
The call needs to be done through multi homed OpenSIPs (I don’t
use mhomed flag)
Caller -> Carrier1 -> OpenSIPs(eth1)---OpenSIPs(eth0) -> pbx ->
Callee
$var(upstream1) = $(hdr(Via)[0]{via.host}); returned the IP
address I needed
OpenSIPs(eth1) has private IP , so on requests I use
force_send_socket(udp:OpenSIPs_PUB_IP:5060) to be able to send
call to further destinations. When replies are back I need to
change send _socket back to privateIP for the answers to Carrier1.
I’ve got the IP address of the Carrier1 in the onreply_route .
onreply_route[1] {
…
if ($(var(upstream1)) == "10.250.242.74") {
force_send_socket(udp:10.197.26.170:5060);
}
…
}
It doesn’t seem to change ip address for replies .
Would you please advise me how to change send_socket in
onreply_route ?
Kind regards,
Ehrny
*From:*users-boun...@lists.opensips.org
<mailto:users-boun...@lists.opensips.org>
[mailto:users-boun...@lists.opensips.org] *On Behalf Of *Razvan
Crainea
*Sent:* Monday, November 21, 2016 1:32 PM
*To:* users@lists.opensips.org <mailto:users@lists.opensips.org>
*Subject:* Re: [OpenSIPS-Users] $ai transformation
Hi, Ehrny!
You need the IP address of whom? Caller? Callee?
$rd is null because a reply does not have a R-URI. Perhaps the
reply doesn't have a received parameter in the reply either,
that's why it is empty.
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com <http://www.opensips-solutions.com>
On 11/21/2016 12:13 PM, Ehrny wrote:
Hello Răzvan,
I need to do some routing in onreply_route[] based on
destination IP.
Tried $rd with no avail , it returns null
If I get you right regarding context, the var
$var(upstream0) = $(hdr(Via)[0]{via.received}); Is empty
also.
What is the right way to get an IP address in replies and do
further routing?
Kind regards,
Ehrny
*From:*users-boun...@lists.opensips.org
<mailto:users-boun...@lists.opensips.org>
[mailto:users-boun...@lists.opensips.org] *On Behalf Of
*Razvan Crainea
*Sent:* Monday, November 21, 2016 11:50 AM
*To:* users@lists.opensips.org <mailto:users@lists.opensips.org>
*Subject:* Re: [OpenSIPS-Users] $ai transformation
Hi, Ehrny!
You don't need to use contexts in the onreply_route[], because
that route is already ran in the context of the reply message.
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com <http://www.opensips-solutions.com>
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Re: [OpenSIPS-Users] $ai transformation
Hi, Ehrny!
If you completely remove the force_send_socket() and any $fs settings in
onreply_route, is it doing the same thing?
Normally the reply should be automatically routed through the same
interface the request came from. Not sure why your reply goes over the
other one. I willtry to replicate this and let you know if it works or not.
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 11/21/2016 06:40 PM, Ehrny wrote:
Hi Răzvan,
Thanks for your help.
The call needs to be done through multi homed OpenSIPs (I don’t use
mhomed flag)
Caller -> Carrier1 -> OpenSIPs(eth1)---OpenSIPs(eth0) -> pbx -> Callee
$var(upstream1) = $(hdr(Via)[0]{via.host}); returned the IP
address I needed
OpenSIPs(eth1) has private IP , so on requests I use
force_send_socket(udp:OpenSIPs_PUB_IP:5060) to be able to send call
to further destinations. When replies are back I need to change send
_socket back to privateIP for the answers to Carrier1.
I’ve got the IP address of the Carrier1 in the onreply_route .
onreply_route[1] {
…
if ($(var(upstream1)) == "10.250.242.74") {
force_send_socket(udp:10.197.26.170:5060);
}
…
}
It doesn’t seem to change ip address for replies .
Would you please advise me how to change send_socket in onreply_route ?
Kind regards,
Ehrny
*From:*users-boun...@lists.opensips.org
<mailto:users-boun...@lists.opensips.org>
[mailto:users-boun...@lists.opensips.org] *On Behalf Of *Razvan Crainea
*Sent:* Monday, November 21, 2016 1:32 PM
*To:* users@lists.opensips.org <mailto:users@lists.opensips.org>
*Subject:* Re: [OpenSIPS-Users] $ai transformation
Hi, Ehrny!
You need the IP address of whom? Caller? Callee?
$rd is null because a reply does not have a R-URI. Perhaps the reply
doesn't have a received parameter in the reply either, that's why it
is empty.
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com <http://www.opensips-solutions.com>
On 11/21/2016 12:13 PM, Ehrny wrote:
Hello Răzvan,
I need to do some routing in onreply_route[] based on destination IP.
Tried $rd with no avail , it returns null
If I get you right regarding context, the var
$var(upstream0) = $(hdr(Via)[0]{via.received}); Is empty also.
What is the right way to get an IP address in replies and do
further routing?
Kind regards,
Ehrny
*From:*users-boun...@lists.opensips.org
<mailto:users-boun...@lists.opensips.org>
[mailto:users-boun...@lists.opensips.org] *On Behalf Of *Razvan
Crainea
*Sent:* Monday, November 21, 2016 11:50 AM
*To:* users@lists.opensips.org <mailto:users@lists.opensips.org>
*Subject:* Re: [OpenSIPS-Users] $ai transformation
Hi, Ehrny!
You don't need to use contexts in the onreply_route[], because
that route is already ran in the context of the reply message.
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com <http://www.opensips-solutions.com>
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Re: [OpenSIPS-Users] $ai transformation
Hi, Ehrny!
You need the IP address of whom? Caller? Callee?
$rd is null because a reply does not have a R-URI. Perhaps the reply
doesn't have a received parameter in the reply either, that's why it is
empty.
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 11/21/2016 12:13 PM, Ehrny wrote:
Hello Răzvan,
I need to do some routing in onreply_route[] based on destination IP.
Tried $rd with no avail , it returns null
If I get you right regarding context, the var$var(upstream0) =
$(hdr(Via)[0]{via.received}); Is empty also.
What is the right way to get an IP address in replies and do further
routing?
Kind regards,
Ehrny
*From:*users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] *On Behalf Of *Razvan Crainea
*Sent:* Monday, November 21, 2016 11:50 AM
*To:* users@lists.opensips.org
*Subject:* Re: [OpenSIPS-Users] $ai transformation
Hi, Ehrny!
You don't need to use contexts in the onreply_route[], because that
route is already ran in the context of the reply message.
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com <http://www.opensips-solutions.com>
On 11/19/2016 10:40 PM, Ehrny wrote:
Dear Răzvan,
… I’ve tried to add variable to onreply_route[1]
$var(upstream0) = $(hdr(Via)[0]{via.param,received});
xlog("upstream0 = $var(upstream0) \n");
and in the log I get critical alert:
CRITICAL:tm:tm_pv_context_reply: no picked branch (-1) for a final
response
*From:*users-boun...@lists.opensips.org
<mailto:users-boun...@lists.opensips.org>
[mailto:users-boun...@lists.opensips.org] *On Behalf Of *Ehrny
*Sent:* Saturday, November 19, 2016 2:06 PM
*To:* OpenSIPS users mailling list
*Subject:* Re: [OpenSIPS-Users] $ai transformation
Hi Răzvan,
I gues so.
I’ve got t_on_reply("1"); in the route
and at the end of the script there is:
onreply_route[1] {
force_send_socket(udp:10.197.26.170:5060);
}
But it doesn’t seem to change send_socket back to priv IP addr ((
Kind regards,
Ehrny
*From:*users-boun...@lists.opensips.org
<mailto:users-boun...@lists.opensips.org>
[mailto:users-boun...@lists.opensips.org] *On Behalf Of *Razvan
Crainea
*Sent:* Friday, November 18, 2016 12:22 PM
*To:* users@lists.opensips.org <mailto:users@lists.opensips.org>
*Subject:* Re: [OpenSIPS-Users] $ai transformation
Hi, Ehrny!
Did you try setting the private socket on the reply?
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com <http://www.opensips-solutions.com>
On 11/17/2016 01:00 AM, Ehrny wrote:
Dear Răzvan,
Thanks again for the prompt help. I was able to change the
headers as needed but I’m stuck with another problem(
I’ve got opensips with two Ethernet adapters, eth1 as a
private and another one eth0 as public. Opensips works fine
when the call is coming on the public eth0 and leaves opensips
through the same public adapter. (All the GWs are behind that
public eth0 instead of one ). The problem happens when the
call comes in through the private eth1, please see the drawing
in attachment.
-sip1. After I’ve got invite from provider on the private
eth1 , I send it through the public eth0.
-sip2. I use force_send_socket(udp:PUBLIC_IP:PORT) for the
call to be able to pass through the opensips and come back
from external GW (x.x.82.139). I also change SIP Request's URI
and use uac_replace_to () to change these fields as needed.
-sip4. Opensips has got 180 Ringing from external GW
(x.x.82.139)
-sip5. Opensips tries to send it back to originator
(10.250.242.74) which is behind private NIC eth0 (10.197.26.170)
the call can not be set up because I send reply from my public
eth1
2016-11-16 18:56:14 : x.x.80.43:5060 -> 10.250.242.74:5060
SIP/2.0 *180* Ringing Via: SIP/2.0/UDP
10.250.242.74:5060;branch=*z9hG4bKqci5ec *Record-Route:
<sip:x.x.80.43;r2=on;lr;ftag=*2F81324631*;did=3a2.4667b68
<sip:x.x.80.43;r2=on;lr;ftag=2F81324631353641A405EA00;did=3a2.4667b68>>
Record-Route:
<sip:10.197.26.170;r2=on;lr;ftag=*2F81324631*;did=3a2.4667b68
<sip:10.197.26.170;r2=on;lr;ftag=2F81324631353641A405EA00;did=3a2.4667b68>>
From: sip:300...@domain.com;tag=*2F81324631*
<sip:300...@domain.com;tag=2F81324631353641A405EA00> To:
sip:300...@domain.com:5060;tag=231469dIr894
<sip:300...@domain.com:5060;tag=231469dIr894p0D461D0t66>
Call-ID: *020A3EA03A8@SFESIP4-id1-ext* CSeq: 1 INVITE Contact:
<sip:54321@x.x.82.139:5060>
I’m not sure if I do it righ
Re: [OpenSIPS-Users] $ai transformation
Hi, Ehrny!
You don't need to use contexts in the onreply_route[], because that
route is already ran in the context of the reply message.
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 11/19/2016 10:40 PM, Ehrny wrote:
Dear Răzvan,
… I’ve tried to add variable to onreply_route[1]
$var(upstream0) = $(hdr(Via)[0]{via.param,received});
xlog("upstream0 = $var(upstream0) \n");
and in the log I get critical alert:
CRITICAL:tm:tm_pv_context_reply: no picked branch (-1) for a final
response
*From:*users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] *On Behalf Of *Ehrny
*Sent:* Saturday, November 19, 2016 2:06 PM
*To:* OpenSIPS users mailling list
*Subject:* Re: [OpenSIPS-Users] $ai transformation
Hi Răzvan,
I gues so.
I’ve got t_on_reply("1"); in the route
and at the end of the script there is:
onreply_route[1] {
force_send_socket(udp:10.197.26.170:5060);
}
But it doesn’t seem to change send_socket back to priv IP addr ((
Kind regards,
Ehrny
*From:*users-boun...@lists.opensips.org
<mailto:users-boun...@lists.opensips.org>
[mailto:users-boun...@lists.opensips.org] *On Behalf Of *Razvan Crainea
*Sent:* Friday, November 18, 2016 12:22 PM
*To:* users@lists.opensips.org <mailto:users@lists.opensips.org>
*Subject:* Re: [OpenSIPS-Users] $ai transformation
Hi, Ehrny!
Did you try setting the private socket on the reply?
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com <http://www.opensips-solutions.com>
On 11/17/2016 01:00 AM, Ehrny wrote:
Dear Răzvan,
Thanks again for the prompt help. I was able to change the headers
as needed but I’m stuck with another problem(
I’ve got opensips with two Ethernet adapters, eth1 as a private
and another one eth0 as public. Opensips works fine when the call
is coming on the public eth0 and leaves opensips through the same
public adapter. (All the GWs are behind that public eth0 instead
of one ). The problem happens when the call comes in through the
private eth1, please see the drawing in attachment.
-sip1. After I’ve got invite from provider on the private eth1 ,
I send it through the public eth0.
-sip2. I use force_send_socket(udp:PUBLIC_IP:PORT) for the call
to be able to pass through the opensips and come back from
external GW (x.x.82.139). I also change SIP Request's URI and use
uac_replace_to () to change these fields as needed.
-sip4. Opensips has got 180 Ringing from external GW (x.x.82.139)
-sip5. Opensips tries to send it back to originator
(10.250.242.74) which is behind private NIC eth0 (10.197.26.170)
the call can not be set up because I send reply from my public eth1
2016-11-16 18:56:14 : x.x.80.43:5060 -> 10.250.242.74:5060
SIP/2.0 *180* Ringing Via: SIP/2.0/UDP
10.250.242.74:5060;branch=*z9hG4bKqci5ec *Record-Route:
<sip:x.x.80.43;r2=on;lr;ftag=*2F81324631*;did=3a2.4667b68
<sip:x.x.80.43;r2=on;lr;ftag=2F81324631353641A405EA00;did=3a2.4667b68>>
Record-Route:
<sip:10.197.26.170;r2=on;lr;ftag=*2F81324631*;did=3a2.4667b68
<sip:10.197.26.170;r2=on;lr;ftag=2F81324631353641A405EA00;did=3a2.4667b68>>
From: sip:300...@domain.com;tag=*2F81324631*
<sip:300...@domain.com;tag=2F81324631353641A405EA00> To:
sip:300...@domain.com:5060;tag=231469dIr894
<sip:300...@domain.com:5060;tag=231469dIr894p0D461D0t66> Call-ID:
*020A3EA03A8@SFESIP4-id1-ext* CSeq: 1 INVITE Contact:
<sip:54321@x.x.82.139:5060>
I’m not sure if I do it right way because the packet (sip5) goes
to 10.250.242.74 with the source ip of public eth0 and not the one
it should pass through to be able to come back.
What is the right way in my case to get the call through?
Thank you for all of your help,
Regards,
Ehrny
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Re: [OpenSIPS-Users] $ai transformation
Hi, Ehrny! Did you try setting the private socket on the reply? Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 11/17/2016 01:00 AM, Ehrny wrote: Dear Răzvan, Thanks again for the prompt help. I was able to change the headers as needed but I’m stuck with another problem( I’ve got opensips with two Ethernet adapters, eth1 as a private and another one eth0 as public. Opensips works fine when the call is coming on the public eth0 and leaves opensips through the same public adapter. (All the GWs are behind that public eth0 instead of one ). The problem happens when the call comes in through the private eth1, please see the drawing in attachment. -sip1. After I’ve got invite from provider on the private eth1 , I send it through the public eth0. -sip2. I use force_send_socket(udp:PUBLIC_IP:PORT) for the call to be able to pass through the opensips and come back from external GW (x.x.82.139). I also change SIP Request's URI and use uac_replace_to () to change these fields as needed. -sip4. Opensips has got 180 Ringing from external GW (x.x.82.139) -sip5. Opensips tries to send it back to originator (10.250.242.74) which is behind private NIC eth0 (10.197.26.170) the call can not be set up because I send reply from my public eth1 2016-11-16 18:56:14 : x.x.80.43:5060 -> 10.250.242.74:5060 SIP/2.0 *180*Ringing Via: SIP/2.0/UDP 10.250.242.74:5060;branch=*z9hG4bKqci5ec***Record-Route: <sip:x.x.80.43;r2=on;lr;ftag=*2F81324631*;did=3a2.4667b68 <sip:x.x.80.43;r2=on;lr;ftag=2F81324631353641A405EA00;did=3a2.4667b68>> Record-Route: <sip:10.197.26.170;r2=on;lr;ftag=*2F81324631*;did=3a2.4667b68 <sip:10.197.26.170;r2=on;lr;ftag=2F81324631353641A405EA00;did=3a2.4667b68>> From: sip:300...@domain.com;tag=*2F81324631* <sip:300...@domain.com;tag=2F81324631353641A405EA00> To: sip:300...@domain.com:5060;tag=231469dIr894 <sip:300...@domain.com:5060;tag=231469dIr894p0D461D0t66> Call-ID: *020A3EA03A8@SFESIP4-id1-ext*CSeq: 1 INVITE Contact: <sip:54321@x.x.82.139:5060> I’m not sure if I do it right way because the packet (sip5) goes to 10.250.242.74 with the source ip of public eth0 and not the one it should pass through to be able to come back. What is the right way in my case to get the call through? Thank you for all of your help, Regards, Ehrny ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips 2.2.2 and top hiding
Hi, Denis!
Do you have the topology_hiding module loaded on the OpenSIPS Proxy? Can
you remove it? My assumption is that OpenSIPS thinks you are trying to
use topology hiding, and he is mangling the CallID (because he "sees"
the marker in there).
Alternatively, you could change the topo hiding marker to something else[1].
[1]
http://www.opensips.org/html/docs/modules/2.2.x/topology_hiding.html#id249610
Best regards,
Răzvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 11/16/2016 11:58 AM, Denis wrote:
And a version of Opensips proxy - Server:: OpenSIPS (2.2.1 (x86_64/linux)).
mailto:denis7...@mail.ru
Razvan, all logs here
https://cloud.mail.ru/public/MtcT/r3p7mRkhF
mailto:denis7...@mail.ru
Razvan, i found the problem (there was in reply_route, where i have
done fix_nated_contact()). This problem is closed.
But, unfortunately, i still cannot make successful call, and i want to
ask you to analyze the new problem it this branch.
The next problem appears in Opensis proxy (we have the same scheme).
In attachment you can see a call log (trace3) and a log from syslog
(trace4). All logs from Opensips proxy (3.3.3.3).
Opensips proxy is the main proxy in my SIP network, while Opensips with
tophiding (1.1.1.1) is a new instance.
Opensips proxy serves many calls and i didn`t see such problem before.
I want to notice you on callid in 183 and 200 code, received from PSTN
GW. It is changed by Opensips proxy for unknown reason.
Thank you for any help.
mailto:denis7...@mail.ru
Hi, Denis!
Could you also send the logs for INVITE? It seems like the dialog is
storing a bogus Contact header.
PS: please attach the logs on pastebin.com, not directly in the email.
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com <http://www.opensips-solutions.com>
On 11/15/2016 04:34 PM, Denis wrote:
Re: [OpenSIPS-Users] Opensips 2.2.2 and top hiding In attachment.
mailto:denis7...@mail.ru
Can you put on pastebin the debug logs for the ACK?
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com <http://www.opensips-solutions.com>
On 11/15/2016 03:44 PM, Denis wrote:
Re: [OpenSIPS-Users] Opensips 2.2.2 and top hiding Hello Ben.
I am using loadbalacer module and using only for initial INVITE.
mailto:denis7...@mail.ru
You said you are doing load balancing as well. Are you doing load
balancing on the ACK? What module are you using (dispatcher,
loadbalancer, etc.)?
Load balancing functions can change the R-URI.
Ben Newlin
*From: *<users-boun...@lists.opensips.org>
<mailto:users-boun...@lists.opensips.org>on behalf of Denis
<denis7...@mail.ru> <mailto:denis7...@mail.ru>
*Reply-To: *OpenSIPS users mailling list <users@lists.opensips.org>
<mailto:users@lists.opensips.org>
*Date: *Tuesday, November 15, 2016 at 8:19 AM
*To: *Răzvan Crainea <raz...@opensips.org> <mailto:raz...@opensips.org>,
OpenSIPS users mailling list <users@lists.opensips.org>
<mailto:users@lists.opensips.org>
*Subject: *Re: [OpenSIPS-Users] Opensips 2.2.2 and top hiding
Hello, Razvan!
No, i don`t make any modification for that variables
" if (match_dialog() || topology_hiding_match()) {
if (!$DLG_status == NULL) {
xlog("L_INFO", "Route0:$rm was received (IPS=$si, IPD=$rd, CALLID=$ci,
FROMTAG=$ft, TOTAG=$tt, AUTH=$au) and RURI = $ru/$rd");
force_rport();
route(1);
exit;
}
}"
The information from a syslog.
"Route0:ACK was received (IPS=2.2.2.2, IPD=3.3.3.3,
CALLID=82158NWE4MmU0NmJiZDU2MzA4OWM1MGFiZjU1Zjg2YTA4NWM,
FROMTAG=b83b533d, TOTAG=6A3BE0-1AA9, AUTH=) and RURI =
sip:3364021@3.3.3.3:5068/3.3.3.3 <mailto:sip:3364021@3.3.3.3:5068/3.3.3.3>"
mailto:denis7...@mail.ru
Hi, Denis!
Are you modifying the $ru/$rd variables anyhwere in your script for that
ACK? I am seeing the R-URI of the ACK going to 3.3.3.3:5068:
ACK sip:3364021@3.3.3.3:5068 <mailto:sip:3364021@3.3.3.3:5068>SIP/2.0.
However, it should be:
ACK sip:3364021@4.4.4.4:5060 <mailto:sip:3364021@4.4.4.4:5060>SIP/2.0.
Can you try printing the $ru variable just after the
topology_hiding_match() function?
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com <http://www.opensips-solutions.com>
On 11/15/2016 02:22 PM, Denis wrote:
Opensips 2.2.2 and top hiding Hello!
I try to make top hiding using topology_hiding module.
In attachment you can find a log of unsuccessful call.
Scheme of the call
SIP UA (2.2.2.2) -> Opensips with top hiding (1.1.1.1) -> another
Opensis proxy (3.3.3.3) - > PSTN GW (4.4.4.4) -> PSTN.
As i understand, the problem is that Opensips proxy cannot send ACK (on
200 OK) to PSTN GW because RURI and Route header has similar IP, namely
3.3.3.3.
I am using "topology_hiding("C");" function for top hiding.
The call log was g
Re: [OpenSIPS-Users] Topology_Hiding adding extra VIA header
Hi, Sammy!
Most likely that WIP refers to the re-invites generated for pinging
purposes. Are you using the "R/r" flags for the create_dialog() function?
Best regards,
Răzvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 11/15/2016 09:56 PM, SamyGo wrote:
Hi Again,
Is this related to the "/Work Still in progress"/ related to
Topology_hiding module as mentioned here at changelog:
http://opensips.org/pub/opensips/2.2.2/ChangeLog
2015-10-14 Vlad Paiu
* [c0f25f7] :
Added Re-INVITE in-dialog pinging support
Controlled via the new "R" and "r" flags available to create_dialog()
as well as the new reinvite_ping_interval module param
Work still in progress :
- Properly handle late negociation between endpoints
- Ensure SDP persistency ( DB and BIN replication )
- Ensure compatibility with topology hiding ( currently the
Contact header
will be bogus when doing TH )
- Whitelist or blacklist logic ( terminate call for 481 and 408
timeout, or terminate call for anything else other than 200 and 491 )
- Extensive testing needed for race conditions specified in rfc
5407
The module paramns in my opensips.cfg look like this.
loadmodule "topology_hiding.so" modparam("topology_hiding",
"force_dialog", 1) modparam("topology_hiding", "th_callid_prefix",
"myvoip_box1")
modparam("topology_hiding", "th_passed_contact_uri_params", "account_id")
modparam("topology_hiding", "th_passed_contact_params", "+mediabx1.wholevoip.se
<http://mediabx1.wholevoip.se>;device;caller")
Looking for some answers thanks,
Regards,
Sammy
On Tue, Nov 15, 2016 at 12:19 PM, SamyGo <govoi...@gmail.com
<mailto:govoi...@gmail.com>> wrote:
Hi Razvan,
I just noticed that since Topo hiding function gives error, the
calls using this do not show any changes in CallID or Contact or any
other details , seems like topohiding is not doing it's job for such
calls anymore. !
Kindly let me know of anything further required to get this resolved.
Thanks,
Sammy.
On Mon, Nov 14, 2016 at 1:30 PM, SamyGo <govoi...@gmail.com
<mailto:govoi...@gmail.com>> wrote:
Hi Razvan,
Here is the requested data.
*INITIAL INVITE:
*Via: SIP/2.0/TLS
123.123.212.123:5061;branch=z9hG4bK442.8373b213.0;i=35f5
*
*
*200 OK from the B party as received by OpenSIPS:
*
Via: SIP/2.0/TLS
118.151.101.64:5061;branch=z9hG4bK442.9a584727.0;i=11
*200 OK as sent out by OpenSIPS:
*
Via: SIP/2.0/TLS
123.123.212.123:5061;received=123.123.212.123;rport=48664;branch=z9hG4bK442.8373b213.0;i=35f5
Via: SIP/2.0/TLS
123.123.212.123:5061;received=123.123.212.123;rport=48664;branch=z9hG4bK442.8373b213.0;i=35f5
Here is the portion of debug log where the destination Answers
the call and topology Hiding restore VIA twice.
http://pastebin.com/z7pt7cwM
Thanks for your response and time looking at this for me.
Regards,
Sammy.
On Nov 14, 2016 3:49 AM, "Răzvan Crainea" <raz...@opensips.org
<mailto:raz...@opensips.org>> wrote:
Hi, Samy!
Can you post on pastebin debugging logs related to this
call? Also, can you also post the Via headers of the initial
INVITE and for the 200 OK received by OpenSIPS?
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com <http://www.opensips-solutions.com>
On 11/12/2016 12:33 AM, SamyGo wrote:
Hi,
I'm using OpenSIPS 2.2.1 version and I'm facing a weird
situation where OpenSIPS is adding a duplicated VIA header
to the 200 OK, This only happens when I've
topology_hiding() engaged into the call.
The scenario is very simple; two users making call to each
other on the same OpenSIPS but with topology_hiding(). As
a consequence of this double VIA the caller device doesn't
trigger the ACK and hence we don't get media stream
established between devices.
*WITH TOPOLOGYHIDING:*
SIP/2.0 200 OK
Via: SIP/2.0/TLS
10.1.10.51:59231;received=7X.XX.XX.X7;rport=59231;branch=z9hG4bK-607165482-63
Via: SIP/2.0/TLS
10.1.10.51:59231;received=7X.XX.XX.X7;rport=59231;branch=z9hG4bK-607165482-63
CSeq: 1 INVITE
...
*WITHOUT TOPOHIDING:
*
SIP/2.0 200 OK
Via: SIP/2.0/TLS
10.1.10.51
Re: [OpenSIPS-Users] Topology_Hiding adding extra VIA header
Hi, Sammy!
What errors is the topo hiding function logging?
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 11/15/2016 07:19 PM, SamyGo wrote:
Hi Razvan,
I just noticed that since Topo hiding function gives error, the calls
using this do not show any changes in CallID or Contact or any other
details , seems like topohiding is not doing it's job for such calls
anymore. !
Kindly let me know of anything further required to get this resolved.
Thanks,
Sammy.
On Mon, Nov 14, 2016 at 1:30 PM, SamyGo <govoi...@gmail.com
<mailto:govoi...@gmail.com>> wrote:
Hi Razvan,
Here is the requested data.
*INITIAL INVITE:
*Via: SIP/2.0/TLS
123.123.212.123:5061;branch=z9hG4bK442.8373b213.0;i=35f5
*
*
*200 OK from the B party as received by OpenSIPS:
*
Via: SIP/2.0/TLS 118.151.101.64:5061;branch=z9hG4bK442.9a584727.0;i=11
*200 OK as sent out by OpenSIPS:
*
Via: SIP/2.0/TLS
123.123.212.123:5061;received=123.123.212.123;rport=48664;branch=z9hG4bK442.8373b213.0;i=35f5
Via: SIP/2.0/TLS
123.123.212.123:5061;received=123.123.212.123;rport=48664;branch=z9hG4bK442.8373b213.0;i=35f5
Here is the portion of debug log where the destination Answers the
call and topology Hiding restore VIA twice.
http://pastebin.com/z7pt7cwM
Thanks for your response and time looking at this for me.
Regards,
Sammy.
On Nov 14, 2016 3:49 AM, "Răzvan Crainea" <raz...@opensips.org
<mailto:raz...@opensips.org>> wrote:
Hi, Samy!
Can you post on pastebin debugging logs related to this call?
Also, can you also post the Via headers of the initial INVITE
and for the 200 OK received by OpenSIPS?
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com <http://www.opensips-solutions.com>
On 11/12/2016 12:33 AM, SamyGo wrote:
Hi,
I'm using OpenSIPS 2.2.1 version and I'm facing a weird
situation where OpenSIPS is adding a duplicated VIA header to
the 200 OK, This only happens when I've topology_hiding()
engaged into the call.
The scenario is very simple; two users making call to each
other on the same OpenSIPS but with topology_hiding(). As a
consequence of this double VIA the caller device doesn't
trigger the ACK and hence we don't get media stream
established between devices.
*WITH TOPOLOGYHIDING:*
SIP/2.0 200 OK
Via: SIP/2.0/TLS
10.1.10.51:59231;received=7X.XX.XX.X7;rport=59231;branch=z9hG4bK-607165482-63
Via: SIP/2.0/TLS
10.1.10.51:59231;received=7X.XX.XX.X7;rport=59231;branch=z9hG4bK-607165482-63
CSeq: 1 INVITE
...
*WITHOUT TOPOHIDING:
*
SIP/2.0 200 OK
Via: SIP/2.0/TLS
10.1.10.51:59223;received=7X.XX.XX.X7;rport=59223;branch=z9hG4bK-607166212-58
CSeq: 1 INVITE
The only difference between the two scenarios is the function
topology_hiding(); is commented out.
It seems like a bug to me, can anyone guide me here validate
this.
*
OpenSIPS Version:*
version: opensips 2.2.1 (x86_64/linux)
flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP,
PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144,
MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
git revision: 68ace2e
main.c compiled on 18:34:37 Sep 28 2016 with gcc 4.8
Thanks,
Sammy
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Re: [OpenSIPS-Users] Opensips 2.2.2 and top hiding
Hi, Denis!
Could you also send the logs for INVITE? It seems like the dialog is
storing a bogus Contact header.
PS: please attach the logs on pastebin.com, not directly in the email.
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 11/15/2016 04:34 PM, Denis wrote:
Re: [OpenSIPS-Users] Opensips 2.2.2 and top hiding In attachment.
mailto:denis7...@mail.ru
Can you put on pastebin the debug logs for the ACK?
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com <http://www.opensips-solutions.com>
On 11/15/2016 03:44 PM, Denis wrote:
Re: [OpenSIPS-Users] Opensips 2.2.2 and top hiding Hello Ben.
I am using loadbalacer module and using only for initial INVITE.
mailto:denis7...@mail.ru
You said you are doing load balancing as well. Are you doing load
balancing on the ACK? What module are you using (dispatcher,
loadbalancer, etc.)?
Load balancing functions can change the R-URI.
Ben Newlin
*From: *<users-boun...@lists.opensips.org>
<mailto:users-boun...@lists.opensips.org>on behalf of Denis
<denis7...@mail.ru> <mailto:denis7...@mail.ru>
*Reply-To: *OpenSIPS users mailling list <users@lists.opensips.org>
<mailto:users@lists.opensips.org>
*Date: *Tuesday, November 15, 2016 at 8:19 AM
*To: *Răzvan Crainea <raz...@opensips.org>
<mailto:raz...@opensips.org>, OpenSIPS users mailling list
<users@lists.opensips.org> <mailto:users@lists.opensips.org>
*Subject: *Re: [OpenSIPS-Users] Opensips 2.2.2 and top hiding
Hello, Razvan!
No, i don`t make any modification for that variables
" if (match_dialog() || topology_hiding_match()) {
if (!$DLG_status == NULL) {
xlog("L_INFO", "Route0:$rm was received (IPS=$si, IPD=$rd, CALLID=$ci,
FROMTAG=$ft, TOTAG=$tt, AUTH=$au) and RURI = $ru/$rd");
force_rport();
route(1);
exit;
}
}"
The information from a syslog.
"Route0:ACK was received (IPS=2.2.2.2, IPD=3.3.3.3,
CALLID=82158NWE4MmU0NmJiZDU2MzA4OWM1MGFiZjU1Zjg2YTA4NWM,
FROMTAG=b83b533d, TOTAG=6A3BE0-1AA9, AUTH=) and RURI =
sip:3364021@3.3.3.3:5068/3.3.3.3
<mailto:sip:3364021@3.3.3.3:5068/3.3.3.3>"
mailto:denis7...@mail.ru
Hi, Denis!
Are you modifying the $ru/$rd variables anyhwere in your script for
that ACK? I am seeing the R-URI of the ACK going to 3.3.3.3:5068:
ACK sip:3364021@3.3.3.3:5068 <mailto:sip:3364021@3.3.3.3:5068>SIP/2.0.
However, it should be:
ACK sip:3364021@4.4.4.4:5060 <mailto:sip:3364021@4.4.4.4:5060>SIP/2.0.
Can you try printing the $ru variable just after the
topology_hiding_match() function?
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com <http://www.opensips-solutions.com>
On 11/15/2016 02:22 PM, Denis wrote:
Opensips 2.2.2 and top hiding Hello!
I try to make top hiding using topology_hiding module.
In attachment you can find a log of unsuccessful call.
Scheme of the call
SIP UA (2.2.2.2) -> Opensips with top hiding (1.1.1.1) -> another
Opensis proxy (3.3.3.3) - > PSTN GW (4.4.4.4) -> PSTN.
As i understand, the problem is that Opensips proxy cannot send ACK
(on 200 OK) to PSTN GW because RURI and Route header has similar IP,
namely 3.3.3.3.
I am using "topology_hiding("C");" function for top hiding.
The call log was gathered from 1.1.1.1
Thank you for any help.
P.S. On 1.1.1.1 i also try to make load balancing.
mailto:denis7...@mail.ru
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Re: [OpenSIPS-Users] [OpenSIPS-users] FOSDEM Real Time Communications devroom seeks speakers / volunteers
Hi, Saul! Thanks for the heads-up, we'll definitely be there! See you in Brussels! Cheers, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 11/14/2016 11:39 PM, Saúl Ibarra Corretgé wrote: Hi there! It’s that time of the year again, time for FOSDEM paper submissions! Next FOSDEM we’ll have a “Real Time Communications” devroom, which is a good fit for OpenSIPS and all things VoIP / RTC. All the information regarding the process is available here: https://lists.fosdem.org/pipermail/fosdem/2016-October/002481.html I’m part of the organising team, so please do reach out if you have any questions / problems. Cheers, -- Saúl Ibarra Corretgé AG Projects ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips 2.2.2 and top hiding
Can you put on pastebin the debug logs for the ACK?
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 11/15/2016 03:44 PM, Denis wrote:
Re: [OpenSIPS-Users] Opensips 2.2.2 and top hiding Hello Ben.
I am using loadbalacer module and using only for initial INVITE.
mailto:denis7...@mail.ru
You said you are doing load balancing as well. Are you doing load
balancing on the ACK? What module are you using (dispatcher,
loadbalancer, etc.)?
Load balancing functions can change the R-URI.
Ben Newlin
*From: *<users-boun...@lists.opensips.org> on behalf of Denis
<denis7...@mail.ru>
*Reply-To: *OpenSIPS users mailling list <users@lists.opensips.org>
*Date: *Tuesday, November 15, 2016 at 8:19 AM
*To: *Răzvan Crainea <raz...@opensips.org>, OpenSIPS users mailling
list <users@lists.opensips.org>
*Subject: *Re: [OpenSIPS-Users] Opensips 2.2.2 and top hiding
Hello, Razvan!
No, i don`t make any modification for that variables
" if (match_dialog() || topology_hiding_match()) {
if (!$DLG_status == NULL) {
xlog("L_INFO", "Route0:$rm was received (IPS=$si, IPD=$rd,
CALLID=$ci, FROMTAG=$ft, TOTAG=$tt, AUTH=$au) and RURI = $ru/$rd");
force_rport();
route(1);
exit;
}
}"
The information from a syslog.
"Route0:ACK was received (IPS=2.2.2.2, IPD=3.3.3.3,
CALLID=82158NWE4MmU0NmJiZDU2MzA4OWM1MGFiZjU1Zjg2YTA4NWM,
FROMTAG=b83b533d, TOTAG=6A3BE0-1AA9, AUTH=) and RURI =
sip:3364021@3.3.3.3:5068/3.3.3.3"
mailto:denis7...@mail.ru
Hi, Denis!
Are you modifying the $ru/$rd variables anyhwere in your script for
that ACK? I am seeing the R-URI of the ACK going to 3.3.3.3:5068:
ACK sip:3364021@3.3.3.3:5068 <mailto:sip:3364021@3.3.3.3:5068>SIP/2.0.
However, it should be:
ACK sip:3364021@4.4.4.4:5060 <mailto:sip:3364021@4.4.4.4:5060>SIP/2.0.
Can you try printing the $ru variable just after the
topology_hiding_match() function?
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com <http://www.opensips-solutions.com>
On 11/15/2016 02:22 PM, Denis wrote:
Opensips 2.2.2 and top hiding Hello!
I try to make top hiding using topology_hiding module.
In attachment you can find a log of unsuccessful call.
Scheme of the call
SIP UA (2.2.2.2) -> Opensips with top hiding (1.1.1.1) -> another
Opensis proxy (3.3.3.3) - > PSTN GW (4.4.4.4) -> PSTN.
As i understand, the problem is that Opensips proxy cannot send ACK
(on 200 OK) to PSTN GW because RURI and Route header has similar IP,
namely 3.3.3.3.
I am using "topology_hiding("C");" function for top hiding.
The call log was gathered from 1.1.1.1
Thank you for any help.
P.S. On 1.1.1.1 i also try to make load balancing.
mailto:denis7...@mail.ru
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Re: [OpenSIPS-Users] Opensips 2.2.2 and top hiding
Hi, Denis!
Are you modifying the $ru/$rd variables anyhwere in your script for that
ACK? I am seeing the R-URI of the ACK going to 3.3.3.3:5068:
ACK sip:3364021@3.3.3.3:5068 SIP/2.0.
However, it should be:
ACK sip:3364021@4.4.4.4:5060 SIP/2.0.
Can you try printing the $ru variable just after the
topology_hiding_match() function?
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 11/15/2016 02:22 PM, Denis wrote:
Opensips 2.2.2 and top hiding Hello!
I try to make top hiding using topology_hiding module.
In attachment you can find a log of unsuccessful call.
Scheme of the call
SIP UA (2.2.2.2) -> Opensips with top hiding (1.1.1.1) -> another
Opensis proxy (3.3.3.3) - > PSTN GW (4.4.4.4) -> PSTN.
As i understand, the problem is that Opensips proxy cannot send ACK
(on 200 OK) to PSTN GW because RURI and Route header has similar IP,
namely 3.3.3.3.
I am using "topology_hiding("C");" function for top hiding.
The call log was gathered from 1.1.1.1
Thank you for any help.
P.S. On 1.1.1.1 i also try to make load balancing.
mailto:denis7...@mail.ru
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Re: [OpenSIPS-Users] Topology_Hiding adding extra VIA header
Hi, Samy!
Can you make sure you are not calling topology_hiding() twice on the
same request? Can you put an xlog just before each topology_hiding()
apearence in your code to make sure?
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 11/14/2016 08:30 PM, SamyGo wrote:
Hi Razvan,
Here is the requested data.
*INITIAL INVITE:
*Via: SIP/2.0/TLS 123.123.212.123:5061;branch=z9hG4bK442.8373b213.0;i=35f5
*
*
*200 OK from the B party as received by OpenSIPS:
*
Via: SIP/2.0/TLS 118.151.101.64:5061;branch=z9hG4bK442.9a584727.0;i=11
*200 OK as sent out by OpenSIPS:
*
Via: SIP/2.0/TLS
123.123.212.123:5061;received=123.123.212.123;rport=48664;branch=z9hG4bK442.8373b213.0;i=35f5
Via: SIP/2.0/TLS
123.123.212.123:5061;received=123.123.212.123;rport=48664;branch=z9hG4bK442.8373b213.0;i=35f5
Here is the portion of debug log where the destination Answers the
call and topology Hiding restore VIA twice.
http://pastebin.com/z7pt7cwM
Thanks for your response and time looking at this for me.
Regards,
Sammy.
On Nov 14, 2016 3:49 AM, "Răzvan Crainea" <raz...@opensips.org
<mailto:raz...@opensips.org>> wrote:
Hi, Samy!
Can you post on pastebin debugging logs related to this call?
Also, can you also post the Via headers of the initial INVITE and
for the 200 OK received by OpenSIPS?
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com <http://www.opensips-solutions.com>
On 11/12/2016 12:33 AM, SamyGo wrote:
Hi,
I'm using OpenSIPS 2.2.1 version and I'm facing a weird situation
where OpenSIPS is adding a duplicated VIA header to the 200 OK,
This only happens when I've topology_hiding() engaged into the call.
The scenario is very simple; two users making call to each other
on the same OpenSIPS but with topology_hiding(). As a consequence
of this double VIA the caller device doesn't trigger the ACK and
hence we don't get media stream established between devices.
*WITH TOPOLOGYHIDING:*
SIP/2.0 200 OK
Via: SIP/2.0/TLS
10.1.10.51:59231;received=7X.XX.XX.X7;rport=59231;branch=z9hG4bK-607165482-63
Via: SIP/2.0/TLS
10.1.10.51:59231;received=7X.XX.XX.X7;rport=59231;branch=z9hG4bK-607165482-63
CSeq: 1 INVITE
...
*WITHOUT TOPOHIDING:
*
SIP/2.0 200 OK
Via: SIP/2.0/TLS
10.1.10.51:59223;received=7X.XX.XX.X7;rport=59223;branch=z9hG4bK-607166212-58
CSeq: 1 INVITE
The only difference between the two scenarios is the function
topology_hiding(); is commented out.
It seems like a bug to me, can anyone guide me here validate this.
*
OpenSIPS Version:*
version: opensips 2.2.1 (x86_64/linux)
flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC,
F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN
16, MAX_URI_SIZE 1024, BUF_SIZE 65535
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
git revision: 68ace2e
main.c compiled on 18:34:37 Sep 28 2016 with gcc 4.8
Thanks,
Sammy
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Re: [OpenSIPS-Users] OpenSIPS reopen TCP connectios and sends INVITE, but not BYE. How to change it?
I got you now: so you are trying to set the tcp_no_new_conn_bflag in the
reply_route, but OpenSIPS still tries to connect to the client?
After you added the code in reply_received function, OpenSIPS still
tries to connect?
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 11/14/2016 06:06 PM, Federico Edorna wrote:
Hi Razvan, thanks for your response
I agree that it is dangerous to try to open a new tcp connection,
that's why we want to set always the flag and never try to open a new
tcp connection to the UAC.
What I'm trying to say is that setting tcp_no_new_conn_bflag doesn't
seem to work for a reply, for example what I've described in my
previous email. When opensips receives a reply from the callee (and
has to do the relay to the caller) but the caller tcp connection has
gone, opensips will try to open a new connection, even with the flag
set. It is not a common scenario, but it happens sometimes, that the
tcp connection is reseted before the call is answered.
Maybe I cannot explain the problem in my English :(, please let me know...
Best Regards
Federico
On Mon, Nov 14, 2016 at 11:24 AM, Răzvan Crainea <raz...@opensips.org
<mailto:raz...@opensips.org>> wrote:
Hi, Federico!
Not sure I understand your problem. That flag indicates OpenSIPS
to avoid opening a new connection if he doesn't have one
available. Therefore, if the connection to the caller closes
between INVITE and 200 OK, that flag prevents OpenSIPS from
opening a new one.
Why would you like to get rid of the TCP SYN message? That happens
and the TCP layer, saying that the data arrived successfully. Why
would you like to prevent that?
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com <http://www.opensips-solutions.com>
On 11/14/2016 04:05 PM, Federico Edorna wrote:
Hi Răzvan,
related to this topic, it seems that tcp_no_new_conn_bflag is not
working on "on_reply" routes
I've tried changing modules/tm/t_reply.c (opensips 2.2), using
something like this:
if (tcp_no_new_conn_bflag)
tcp_no_new_conn = 1;
in "relay_reply" function and now opensips doesn't try to open a
new tcp connection. Without this code I cannot manage to avoid
the TCP SYN from opensips to client when receiving a reply and
tcp connection is not available.
Just to clarify, the scenario is something like this:
AopensipsB
---INVITE--->
---INVITE--->
<---100 Trying---
<---100 Trying---
<---183 Session Progress---
<---183 Session Progress---
--- At this point I wait opensips to close tcp connection
(tcp_connection_lifetime=10) and then "B" answers the call
<---200 OK---
Thanks!
Federico
On Thu, Oct 27, 2016 at 4:58 AM, Răzvan Crainea
<raz...@opensips.org <mailto:raz...@opensips.org>> wrote:
Hi, Rodrigo!
Having OpenSIPS opening TCP connections towards client is a
bit dangerous, especially if the clients are behind NAT.
That's because most likely you will not be able to reach
them, and opensips will get stuck trying to connect (until it
triggers a timeout). That's why the best way to go is to try
to keep the connection (ideally opened by the client at
REGISTER) as much as possible. This is usually done by
pinging (as discussed in a previous email). So my suggestion
is to try to avoid opening new TCP connections with clients,
unless you really know they will always be reachable.
The behavior you are describing (INVITE vs BYE handling),
might be related to the fact that you are setting the
tcp_no_new_conn_bflag[1] flag for BYE messages, but not for
INVITEs. Is this correct? If not, do you see any errors in
the script?
[1]
http://www.opensips.org/Documentation/Script-CoreParameters-2-2#toc101
<http://www.opensips.org/Documentation/Script-CoreParameters-2-2#toc101>
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com <http://www.opensips-solutions.com>
On 10/26/2016 10:59 PM, Rodrigo Pimenta Carvalho wrote:
Hi.
After some log debug I have observed the following behavior
in the OpenSISP (2.2.1):
When OpenSIPS has to send a SIP INVITE to a peer through a
TCP connection that was closed before by some way, OpenSIPS
open a new one and then sends the SIP message to the peer
successfully.
However, when OpenSIPS has to send a SIP BYE to a peer
through a TCP connection that was closed before, OpenSIPS
open a new one, but doesn't send the SIP BYE. In this case
SIP BYE is discarded.
How to change the behavior of OpenSI
Re: [OpenSIPS-Users] OpenSIPS reopen TCP connectios and sends INVITE, but not BYE. How to change it?
Hi, Federico! Not sure I understand your problem. That flag indicates OpenSIPS to avoid opening a new connection if he doesn't have one available. Therefore, if the connection to the caller closes between INVITE and 200 OK, that flag prevents OpenSIPS from opening a new one. Why would you like to get rid of the TCP SYN message? That happens and the TCP layer, saying that the data arrived successfully. Why would you like to prevent that? Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 11/14/2016 04:05 PM, Federico Edorna wrote: Hi Răzvan, related to this topic, it seems that tcp_no_new_conn_bflag is not working on "on_reply" routes I've tried changing modules/tm/t_reply.c (opensips 2.2), using something like this: if (tcp_no_new_conn_bflag) tcp_no_new_conn = 1; in "relay_reply" function and now opensips doesn't try to open a new tcp connection. Without this code I cannot manage to avoid the TCP SYN from opensips to client when receiving a reply and tcp connection is not available. Just to clarify, the scenario is something like this: AopensipsB ---INVITE---> ---INVITE---> <---100 Trying--- <---100 Trying--- <---183 Session Progress--- <---183 Session Progress--- --- At this point I wait opensips to close tcp connection (tcp_connection_lifetime=10) and then "B" answers the call <---200 OK--- Thanks! Federico On Thu, Oct 27, 2016 at 4:58 AM, Răzvan Crainea <raz...@opensips.org <mailto:raz...@opensips.org>> wrote: Hi, Rodrigo! Having OpenSIPS opening TCP connections towards client is a bit dangerous, especially if the clients are behind NAT. That's because most likely you will not be able to reach them, and opensips will get stuck trying to connect (until it triggers a timeout). That's why the best way to go is to try to keep the connection (ideally opened by the client at REGISTER) as much as possible. This is usually done by pinging (as discussed in a previous email). So my suggestion is to try to avoid opening new TCP connections with clients, unless you really know they will always be reachable. The behavior you are describing (INVITE vs BYE handling), might be related to the fact that you are setting the tcp_no_new_conn_bflag[1] flag for BYE messages, but not for INVITEs. Is this correct? If not, do you see any errors in the script? [1] http://www.opensips.org/Documentation/Script-CoreParameters-2-2#toc101 <http://www.opensips.org/Documentation/Script-CoreParameters-2-2#toc101> Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com <http://www.opensips-solutions.com> On 10/26/2016 10:59 PM, Rodrigo Pimenta Carvalho wrote: Hi. After some log debug I have observed the following behavior in the OpenSISP (2.2.1): When OpenSIPS has to send a SIP INVITE to a peer through a TCP connection that was closed before by some way, OpenSIPS open a new one and then sends the SIP message to the peer successfully. However, when OpenSIPS has to send a SIP BYE to a peer through a TCP connection that was closed before, OpenSIPS open a new one, but doesn't send the SIP BYE. In this case SIP BYE is discarded. How to change the behavior of OpenSIPS to make it to send the SIP BYE is such case? I'm looking for ways of fix or workaround of a TCP tear down connection that happens during dialogs. Any hint will be very helpful! RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ___ Users mailing list Users@lists.opensips.org <mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users <http://lists.opensips.org/cgi-bin/mailman/listinfo/users> ___ Users mailing list Users@lists.opensips.org <mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users <http://lists.opensips.org/cgi-bin/mailman/listinfo/users> ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Topology_Hiding adding extra VIA header
Hi, Samy! Can you post on pastebin debugging logs related to this call? Also, can you also post the Via headers of the initial INVITE and for the 200 OK received by OpenSIPS? Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 11/12/2016 12:33 AM, SamyGo wrote: Hi, I'm using OpenSIPS 2.2.1 version and I'm facing a weird situation where OpenSIPS is adding a duplicated VIA header to the 200 OK, This only happens when I've topology_hiding() engaged into the call. The scenario is very simple; two users making call to each other on the same OpenSIPS but with topology_hiding(). As a consequence of this double VIA the caller device doesn't trigger the ACK and hence we don't get media stream established between devices. *WITH TOPOLOGYHIDING:* SIP/2.0 200 OK Via: SIP/2.0/TLS 10.1.10.51:59231;received=7X.XX.XX.X7;rport=59231;branch=z9hG4bK-607165482-63 Via: SIP/2.0/TLS 10.1.10.51:59231;received=7X.XX.XX.X7;rport=59231;branch=z9hG4bK-607165482-63 CSeq: 1 INVITE ... *WITHOUT TOPOHIDING: * SIP/2.0 200 OK Via: SIP/2.0/TLS 10.1.10.51:59223;received=7X.XX.XX.X7;rport=59223;branch=z9hG4bK-607166212-58 CSeq: 1 INVITE The only difference between the two scenarios is the function topology_hiding(); is commented out. It seems like a bug to me, can anyone guide me here validate this. * OpenSIPS Version:* version: opensips 2.2.1 (x86_64/linux) flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. git revision: 68ace2e main.c compiled on 18:34:37 Sep 28 2016 with gcc 4.8 Thanks, Sammy ___ 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] No TLS-related files in OpenSIPS 1.11.9 src.tar.gz
Hi, Chen-Che! You are right; the building system has changed a bit and I forgot to update the version in the new files! I've done a fix on all affected branches. However, they will only be visible in the next release. Until then, you can apply this patch[1] on the current archive in order to generate debs with the proper version. Thanks for reporting this! [1] https://github.com/OpenSIPS/opensips/commit/bf7f401109312dc2a914350866545bb44b4a9f7b.patch Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 11/14/2016 04:23 AM, Chen-Che Huang wrote: Hi Răzvan, I visited http://opensips.org/pub/opensips/1.11.9/ and called wget http://opensips.org/pub/opensips/1.11.9/opensips-1.11.9-tls.tar.gz to download the archive with TLS files. Then, I removed the comment of #TLS=1 in the Makefile and started to generate deb files by ``sudo TLS=1 make deb''. During the process, it seemed that some problem occurred (shown below) and the generated deb files were not with the expected version 1.11.9. For instance, the main deb is opensips_1.11.8-1_amd64.deb rather than opensips_1.11.9-1_amd64.deb. If anything is not clear, please feel free to let me know. Thanks. Best regards, Chen-Che dh_clean make[1]: Leaving directory `/home/ubuntu/opensips-1.11.9/opensips-1.11.9-tls' dpkg-source -b opensips-1.11.9-tls dpkg-source: warning: no source format specified in debian/source/format, see dpkg-source(1) dpkg-source: info: using source format `1.0' dpkg-source: warning: source directory 'opensips-1.11.9-tls' is not - 'opensips-1.11.8' dpkg-source: info: building opensips in opensips_1.11.8-1.tar.gz dpkg-source: info: building opensips in opensips_1.11.8-1.dsc debian/rules build make[1]: Entering directory `/home/ubuntu/opensips-1.11.9/opensips-1.11.9-tls' dh_testdir -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/No-TLS-related-files-in-OpenSIPS-1-11-9-src-tar-gz-tp7604883p7604980.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] usage of setdsturi
Hi, Agalya!
The setdsturi() function only accepts strings as parameters, not
pseudo-variables[1]. As Ben suggested, the $du pseudo-variable is more
flexible and recommended.
[1] http://www.opensips.org/Documentation/Script-CoreFunctions-2-2#toc49
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 11/10/2016 11:58 PM, Newlin, Ben wrote:
I would recommend just using $du. [1]
$du = “sip:” + $var(Fqdn) + “:5060”;
[1] http://www.opensips.org/Documentation/Script-CoreVar-2-2#toc35
Ben Newlin
*From: *<users-boun...@lists.opensips.org> on behalf of "Ramachandran,
Agalya (Contractor)" <agalya_ramachand...@comcast.com>
*Reply-To: *OpenSIPS users mailling list <users@lists.opensips.org>
*Date: *Thursday, November 10, 2016 at 4:35 PM
*To: *OpenSIPS users mailling list <users@lists.opensips.org>
*Subject: *[OpenSIPS-Users] usage of setdsturi
Hi team,
I have a question in usage of setdsturi().
When I hardcode the uri in the function, such as
setdsturi(“sip:t...@test.com:5060”) – this works.
But why I try to use script variable, it complains as bad_uri.
$var(test) = "sip:"+$var(Fqdn)+ ":5060";
setdsturi("$var(test)");
How do I setdsturi() dynamically, with the value in script variable
and not by hardcoding?
Regards,
Agalya
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Re: [OpenSIPS-Users] How to make OpenSIPS send SIP BYE, by configuration, , before dialog timout?
Hi, Rodrigo! Sorry, I've just seen the message, I've missed it earlier. As far as I understand, OpenSIPS is listening on two interfaces: 127.0.01:5060 and 192.168.0.101:5060. Is the UPDATE coming on the same TCP connection as the initial one? Or the client opens a new connection for it, over the PUBLIC interface? Could you send over (privately) a PCAP trace? Also, you'd probably need to make sure that you call fix_nated_contact() and force_rport() on the UPDATE request. Also, are you setting the tcp_accept_aliases[1] or force_tcp_alias()[2] in your script? [1] http://www.opensips.org/Documentation/Script-CoreParameters-2-2#toc95 [2] http://www.opensips.org/Documentation/Script-CoreFunctions-2-2#force_tcp_alias Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 11/10/2016 09:59 PM, Rodrigo Pimenta Carvalho wrote: Hi Razvan. I answered your questions yesterday. I'm not sure if you saw my message. Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 *De:* users-boun...@lists.opensips.org <users-boun...@lists.opensips.org> em nome de Răzvan Crainea <raz...@opensips.org> *Enviado:* quarta-feira, 9 de novembro de 2016 08:29 *Para:* users@lists.opensips.org *Assunto:* Re: [OpenSIPS-Users] How to make OpenSIPS send SIP BYE, by configuration, , before dialog timout? Hi, Rodrigo! The only HACK that I can think of is when you get the BYE message, set the dialog timeout to 0, match it against the dialog, and then drop the message. OpenSIPS will behave as if the dialog expired in that moment. However, you seem to have a flow logic - most likely the Contact header in the BYE is not correct. Could you send us a trace to help you figure out what the problem is? Also, did you try to validate the message against the dialog[1] and fix it accordingly[2]? [1] http://www.opensips.org/html/docs/modules/2.2.x/dialog#id295894 [2] http://www.opensips.org/html/docs/modules/2.2.x/dialog#id295982 Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com Home — OpenSIPS Solutions <http://www.opensips-solutions.com/> www.opensips-solutions.com OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities. On 11/08/2016 10:07 PM, Rodrigo Pimenta Carvalho wrote: Hi. Dialogs in my OpenSIPS is programmed to finish after 60 seconds. (timeout = 1 minute). So, when 2 peers are in a dialog, OpenSIPS sends SIP BYE to both peers, automatically after 60 seconds. Is it possible to make OpenSIPS send this exact kind of SIP BYE to both peers, before the dialog timeout? I mean, in a configured way (opensips.cfg)? When OpenSISP sends SIP BYE automatically, both peers receive the SIP BYE correctly. However, when a peer sends SIP BYE, it reaches the OpenSIPS, but OpenSIPS is unable to forward this SIP BYE. Due to some unknown reason, in this moment there is no open socket to communicate with such peer. That is why I would like to make OpenSIPS send 'its own' SIP BYE, and see if such idea will simulate a normal situation, until I discover why there is a socket problem. Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Common points in B2B top hiding
Yes, it will, if you are using OpenSIPS 2.1 or 2.2 and pass the "C" flag to topology_hiding() function[1]. [1] http://www.opensips.org/html/docs/modules/2.2.x/topology_hiding.html#id293540 Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 11/10/2016 11:14 AM, Денис Путято wrote: Re: [OpenSIPS-Users] Common points in B2B top hiding Hello, Razvan! Does "topology hiding" make full top hiding? For example, if callid will include SIP UA IP address, will "topology hiding" hide it? Thank you. mailto:denis7...@mail.ru Hi, Denis! Take a look at the "a" flag used by the b2b_init_request()[1] function. This might help you achieve what you want. May I ask you why you are using the b2b topology hiding, and not the dedicated module for topology hiding[2]? It would be far more easier to use the latter one in most of the cases. [1] http://www.opensips.org/html/docs/modules/2.2.x/b2b_logic.html#id294010 [2] http://www.opensips.org/html/docs/modules/2.2.x/topology_hiding.html Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com <http://www.opensips-solutions.com> On 11/08/2016 04:19 PM, Denis wrote: Re: [OpenSIPS-Users] Common points in B2B top hiding Hello, Razvan! No, it is not quite what i am looking for. For example, before top hiding i make authentication procedure during which i got login. For some call processing logging i need to translate this login to some proxy instance. How can i do that? mailto:denis7...@mail.ru Hi, Denis! You can copy headers from the initial request to the B2B one by using the custom headers[1]. [1] http://www.opensips.org/html/docs/modules/2.2.x/b2b_logic.html#id293556 Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com <http://www.opensips-solutions.com> On 11/08/2016 03:41 PM, Denis wrote: Common points in B2B top hiding Hello! Sorry that it was, but are there any common points between received INVITE and new generated INVITE while using B2B top hiding? I need some points, gotten in local_route, which will uniquely identified the call and which i can use, for example, to add some headers (based on that points). Thank you for any help. mailto:denis7...@mail.ru ___ Users mailing list Users@lists.opensips.org <mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org <mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Rtpproxy and IPV4 IPV6 interworking
Hi, Robert! Yes, in cases where you don't need IPv6, use II for those requests. Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 11/09/2016 07:12 PM, Robert Dyck wrote: I should have described the scenario in more detail. The rtproxy is in bridge mode because two addresses were specified. This was to accommodate IPV4 - IPV6 interworking. However the rtpproxy is also to be used for NAT traversal. This is not a bridge in the physical sense because there only one interface. For NAT traversal the IPV6 address should be ignored. Am I correct in thinking that one should use either II flags or EE flags depending on the order of the addresses given to rtpproxy? Thank you for taking the time for this. On November 9, 2016 12:23:09 PM you wrote: Hi, Robert! Yes, the I and E parameters are mandatory, and they should describe how the RTP will flow. For example if the flow is from IPv4 to IPv6, you should use EI; if the flow is from IPv4 to IPv6, then you should use IE. And so on, depending on the call flow. Regarding the address parameter, that is used when you want to overwrite the address indicated by RTPProxy. This is used mainly for setups where RTPProxy is behind NAT and the address inidcated is the private one. You should swap this IP with the public advertised one. Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 11/08/2016 09:51 PM, Robert Dyck wrote: Thank you Assuming rtpproxy was started with IPV4 as the first address and IPV6 as the second, then in the NAT scenario, are the II flags mandatory in offer/answer? Slightly off topic, what sort of scenario would require the address parameter for offer/answer? On November 8, 2016 09:57:30 AM Răzvan Crainea wrote: Hi, Robert! See my answers inline. Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 11/08/2016 02:15 AM, Robert Dyck wrote: I have some question regarding rtpproxy capabilities in relation to IPV4-IPV6 interworking. The articles I have read say that you need to assign an address from each address family to rtpproxy. They go on to say that rtpproxy will then be in bridged mode. Others define bridge mode as assigning two interfaces to rtpproxy. As long as you have RTPProxy listening on two IPs, you have it set in bridge mode. It doesn't matther whether one of them is IPv6, or both are. If the IPV4 and IPV6 addresses are on the same interface, is the rtpproxy indeed in bridged mode? Should one avoid the use of engage_rtpproxy? Yes, as stated above, RTPProxy is in bridged mode and you should avoid using engage_rtpproxy(). That's because the function can't know/decide which interface is which and cannot map with the RTPProxy's one. Assuming that IPV4- IPV6 interworking is actually possible using opensips and rtpproxy, does that mean that an instance of rtpproxy is not available to enable NAT traversal - would NAT traversal require using another instance of rtpproxy using a single IPV4 address? No, you don't need an extra instance - a single instance will do both bridging and nat traversal. Furthermore is the multihome parameter relevant to IPV4-IPV6 interworking if opensips only listens on one interface? The multihome parameter is only relevant for OpenSIPS, it doesn't influence RTPProxy's behavior at all. ___ 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] Migrating form 1.11.5 to 1.11.9
Hi, Alain!
Somewhere between 1.11.5 and 1.11.9 we figured out a problem related to
mysql queries: if your database entry is NULL, we can't just simply set
a NULL value to an AVP, because that just deletes the previous value. So
you end up in a very inconsistent state. That's why we decided to set
the value "" to all the fields that are NULL in the database (this
token is used in serveral other scenarios, that's why we picked this name).
Therefore, in order to properly test if a column is provisioned NULL in
the database is:
if ($avp(redirect) != "") {
}
Let us know how this goes.
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 11/09/2016 05:06 PM, Alain Bieuzent wrote:
Hi All,
I’m trying to migrate form 1.11.5 to 1.11.9 and i have a problem to
test the value of a mysql query when the reseult is null.
In 1.11.5 this test « if (!$avp(redirect) == NULL) » works, but
doesn’t work in 1.11.9.
Is there a change in code ?
What is the corect way to test a null value from avp ?
Regards
Alain.
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Re: [OpenSIPS-Users] How to make OpenSIPS send SIP BYE, by configuration, , before dialog timout?
Hi, Rodrigo! The only HACK that I can think of is when you get the BYE message, set the dialog timeout to 0, match it against the dialog, and then drop the message. OpenSIPS will behave as if the dialog expired in that moment. However, you seem to have a flow logic - most likely the Contact header in the BYE is not correct. Could you send us a trace to help you figure out what the problem is? Also, did you try to validate the message against the dialog[1] and fix it accordingly[2]? [1] http://www.opensips.org/html/docs/modules/2.2.x/dialog#id295894 [2] http://www.opensips.org/html/docs/modules/2.2.x/dialog#id295982 Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 11/08/2016 10:07 PM, Rodrigo Pimenta Carvalho wrote: Hi. Dialogs in my OpenSIPS is programmed to finish after 60 seconds. (timeout = 1 minute). So, when 2 peers are in a dialog, OpenSIPS sends SIP BYE to both peers, automatically after 60 seconds. Is it possible to make OpenSIPS send this exact kind of SIP BYE to both peers, before the dialog timeout? I mean, in a configured way (opensips.cfg)? When OpenSISP sends SIP BYE automatically, both peers receive the SIP BYE correctly. However, when a peer sends SIP BYE, it reaches the OpenSIPS, but OpenSIPS is unable to forward this SIP BYE. Due to some unknown reason, in this moment there is no open socket to communicate with such peer. That is why I would like to make OpenSIPS send 'its own' SIP BYE, and see if such idea will simulate a normal situation, until I discover why there is a socket problem. Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ___ 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] Rtpproxy and IPV4 IPV6 interworking
Hi, Robert! Yes, the I and E parameters are mandatory, and they should describe how the RTP will flow. For example if the flow is from IPv4 to IPv6, you should use EI; if the flow is from IPv4 to IPv6, then you should use IE. And so on, depending on the call flow. Regarding the address parameter, that is used when you want to overwrite the address indicated by RTPProxy. This is used mainly for setups where RTPProxy is behind NAT and the address inidcated is the private one. You should swap this IP with the public advertised one. Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 11/08/2016 09:51 PM, Robert Dyck wrote: Thank you Assuming rtpproxy was started with IPV4 as the first address and IPV6 as the second, then in the NAT scenario, are the II flags mandatory in offer/answer? Slightly off topic, what sort of scenario would require the address parameter for offer/answer? On November 8, 2016 09:57:30 AM Răzvan Crainea wrote: Hi, Robert! See my answers inline. Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 11/08/2016 02:15 AM, Robert Dyck wrote: I have some question regarding rtpproxy capabilities in relation to IPV4-IPV6 interworking. The articles I have read say that you need to assign an address from each address family to rtpproxy. They go on to say that rtpproxy will then be in bridged mode. Others define bridge mode as assigning two interfaces to rtpproxy. As long as you have RTPProxy listening on two IPs, you have it set in bridge mode. It doesn't matther whether one of them is IPv6, or both are. If the IPV4 and IPV6 addresses are on the same interface, is the rtpproxy indeed in bridged mode? Should one avoid the use of engage_rtpproxy? Yes, as stated above, RTPProxy is in bridged mode and you should avoid using engage_rtpproxy(). That's because the function can't know/decide which interface is which and cannot map with the RTPProxy's one. Assuming that IPV4- IPV6 interworking is actually possible using opensips and rtpproxy, does that mean that an instance of rtpproxy is not available to enable NAT traversal - would NAT traversal require using another instance of rtpproxy using a single IPV4 address? No, you don't need an extra instance - a single instance will do both bridging and nat traversal. Furthermore is the multihome parameter relevant to IPV4-IPV6 interworking if opensips only listens on one interface? The multihome parameter is only relevant for OpenSIPS, it doesn't influence RTPProxy's behavior at all. ___ 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] How to find out which TCP sockes OpenSIPS is listening on?
Hi, Rodrigo! So are you interested in finding the opened TCP connections? If so, you should try the list_tcp_conns command[1]. [1] http://www.opensips.org/Documentation/Interface-CoreMI-2-2#toc5 Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 11/08/2016 06:10 PM, Rodrigo Pimenta Carvalho wrote: Hi. Is it possible to find which is every socket that is currently opened to opensips listens on SIP messages from peers, while using TCP? I have examined opensipsctl command, but it doesn't show the sockets. I need see if a new socket is being created and opened when a peer sends a SIP UPDATE. Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ___ 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] Common points in B2B top hiding
Hi, Denis! Take a look at the "a" flag used by the b2b_init_request()[1] function. This might help you achieve what you want. May I ask you why you are using the b2b topology hiding, and not the dedicated module for topology hiding[2]? It would be far more easier to use the latter one in most of the cases. [1] http://www.opensips.org/html/docs/modules/2.2.x/b2b_logic.html#id294010 [2] http://www.opensips.org/html/docs/modules/2.2.x/topology_hiding.html Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 11/08/2016 04:19 PM, Denis wrote: Re: [OpenSIPS-Users] Common points in B2B top hiding Hello, Razvan! No, it is not quite what i am looking for. For example, before top hiding i make authentication procedure during which i got login. For some call processing logging i need to translate this login to some proxy instance. How can i do that? mailto:denis7...@mail.ru Hi, Denis! You can copy headers from the initial request to the B2B one by using the custom headers[1]. [1] http://www.opensips.org/html/docs/modules/2.2.x/b2b_logic.html#id293556 Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com <http://www.opensips-solutions.com> On 11/08/2016 03:41 PM, Denis wrote: Common points in B2B top hiding Hello! Sorry that it was, but are there any common points between received INVITE and new generated INVITE while using B2B top hiding? I need some points, gotten in local_route, which will uniquely identified the call and which i can use, for example, to add some headers (based on that points). Thank you for any help. mailto:denis7...@mail.ru ___ Users mailing list Users@lists.opensips.org <mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Common points in B2B top hiding
Hi, Denis! You can copy headers from the initial request to the B2B one by using the custom headers[1]. [1] http://www.opensips.org/html/docs/modules/2.2.x/b2b_logic.html#id293556 Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 11/08/2016 03:41 PM, Denis wrote: Common points in B2B top hiding Hello! Sorry that it was, but are there any common points between received INVITE and new generated INVITE while using B2B top hiding? I need some points, gotten in local_route, which will uniquely identified the call and which i can use, for example, to add some headers (based on that points). Thank you for any help. mailto:denis7...@mail.ru ___ 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] Regarding Option request .
Hello!
You chould use the is_from_gw()[1] function to detect if a message is
coming from a gateway. So basically your code should look like this:
if (method == "OPTIONS" && is_from_gw()) {
sl_send_reply("200", "OK");
exit;
}
[1] http://www.opensips.org/html/docs/modules/1.11.x/drouting.html#id295343
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 11/08/2016 10:05 AM, Sasmita Panda wrote:
Hi All ,
I am using opensips-1.11 . I have a requirement that when an
"Option" request will come to my proxy from the gateways I have added
in my dr_gateways table then only I will process the request and will
send 200 OK .
Previously , I was sending 200 Ok to all the Options requests ,
So I have written the config file like bellow .
route{
if (method=="OPTIONS") {
sl_send_reply("200", "OK");
exit;
}
...
...
route(1);
}
I have tried many possible ways but its not happening . What should I
do to achieve my goal ?
*/Thanks & Regards/*
/Sasmita Panda/
/Network Testing and Software Engineer/
/3CLogic , ph:07827611765/
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Re: [OpenSIPS-Users] Rtpproxy and IPV4 IPV6 interworking
Hi, Robert! See my answers inline. Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 11/08/2016 02:15 AM, Robert Dyck wrote: I have some question regarding rtpproxy capabilities in relation to IPV4-IPV6 interworking. The articles I have read say that you need to assign an address from each address family to rtpproxy. They go on to say that rtpproxy will then be in bridged mode. Others define bridge mode as assigning two interfaces to rtpproxy. As long as you have RTPProxy listening on two IPs, you have it set in bridge mode. It doesn't matther whether one of them is IPv6, or both are. If the IPV4 and IPV6 addresses are on the same interface, is the rtpproxy indeed in bridged mode? Should one avoid the use of engage_rtpproxy? Yes, as stated above, RTPProxy is in bridged mode and you should avoid using engage_rtpproxy(). That's because the function can't know/decide which interface is which and cannot map with the RTPProxy's one. Assuming that IPV4- IPV6 interworking is actually possible using opensips and rtpproxy, does that mean that an instance of rtpproxy is not available to enable NAT traversal - would NAT traversal require using another instance of rtpproxy using a single IPV4 address? No, you don't need an extra instance - a single instance will do both bridging and nat traversal. Furthermore is the multihome parameter relevant to IPV4-IPV6 interworking if opensips only listens on one interface? The multihome parameter is only relevant for OpenSIPS, it doesn't influence RTPProxy's behavior at all. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] B2B top hiding
Hi, Denis!
Are you seeing any other ERROR messages in the logs? When you see those
errors in the log, does OpenSIPS send any message out (even to himself)?
Can you provide a full trace?
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 11/07/2016 03:20 PM, Denis wrote:
Re: [OpenSIPS-Users] B2B top hiding Hello, Razvan!
I want to make some input point to my SIP network which will make
authentication and topology hiding.
Authentication works (i checked it by inserting in client SIP UA wrong
password), but b2b doesn`t.
When i try to setting port before b2b_init_request i see in log "
ERROR:b2b_logic:create_top_hiding_entities: failed to create new b2b
server instance"
mailto:denis7...@mail.ru <mailto:d.puty...@ptl.ru>
Hi, Denis!
Can you please detail a bit what you are trying to achieve? In the
trace attached, all I can see is that OpenSIPS doesn't authenticate
the UA - I don't see why it should even try to send the INVITE anywhere.
However, if you are saying that you see both lines in the log, it
means at a certain point the INVITE was authenticated and the
topo-hiding was engaged. Did you try setting the port before calling
b2b_init_request(), in route(1)?
BR
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com <http://www.opensips-solutions.com>
On 11/07/2016 02:20 PM, Denis wrote:
B2B top hiding Hello!
Opensips 2.2.2.
I want to make an installation of Opensips which will include auth
and top hiding functionality.
First of all, in the main route, i catch INVITE
"if (is_method("INVITE") && !has_totag()) {
route(1);
exit;
}"
then, in route [1], i make some auth procedure using
"pv_proxy_authorize" and "proxy_authorize" functions
if auth procedure is successful i make
"xlog("L_INFO", "Route1:$rm was received with auth (IP=$si, IPD=$rd,
CALLID=$ci, FROMTAG=$ft, TOTAG=$tt, AUTH=$au) AND B2B prepared");
b2b_init_request("top hiding");
exit;"
In the local route i try to edit port of the destination, using
"xlog("L_INFO", "Local:$rm was received with auth (IP=$si, IPD=$rd,
CALLID=$ci, FROMTAG=$ft, TOTAG=$tt, AUTH=$au)");
rewritehostport("1.1.1.1:5065");"
where 1.1.1.1 - ip address of Opensips.
In attachment you can see debug of unsuccessful call, where
1.1.1.1 - ip address of Opensips.
2.2.2.2 - caller SIP UA
I can see in syslog both messages, in local route and in route [1].
As you can see from the debug Opensips doesn`t try to send INVITE to
1.1.1.1:5065.
Thank you for any help.
mailto:denis7...@mail.ru
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