Re: [OpenSIPS-Users] External transfer fails (from Asterisk)
Bogdan-Andrei Iancu wrote: Peter den Hartog wrote: Peter den Hartog wrote: Bogdan-Andrei Iancu wrote: Hi Peter, Peter den Hartog wrote: Hello, I don't know if i'm on the right mailing list for this issue but maby i'm not the only one that had it :-). if it is opensips related, you are on the right list :) I implemented opensips and it works good, the normal calls are going great, outside/inside it all works. inside transfer (exten to exten) works to. But when an outside caller calls the office, it goes to the asterisk, and asterisk forwards it to an opensips extension. exten = x,Dial,1,(SIP/2...@opensips.org) That works great, the caller gets the right person, but when the one being called, transfer that call it gone. This is the first scenario where * is fronting OpenSIPS ...typically is the other way around :D I think it's because asterisk is trying to transfer this caller, but the extension is not there (it's in opensips ofcourse, but not in *) Normally, the call transfer (from the phone) is done via a REFER request (inside the ongoing dialog) - What I suspect is that , as * is in the path of all calls with external users, * will intercept the REFER and try to handle it locally. Try to get a trace and see if this is what happens = REFER being consumed by *, instead of passing it to the external party. Regards, Bogdan I can connect the asterisk users to the opensips users by connecting the database, but is this really needed? or is there another issue here? Do i miss something? ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users Hello Bogdan, That is correct, in Asterisk i see nothing of a new call, or a transfer.. but the phone is creating a new call on line 2, in opensips i just see a new ongoing call. (the line 2 call) and on the outside phone i hear the asterisk wait/hold music. Is there any smart solution for this? can i just forward the complete call to opensips and let asterisk only forward it, and not create the call? (it now just does a dial to the sip member in opensips) Oke a little update, i can now do blind (cold) transfers from asterisk to opensips (outside lines) but not hot transfers, then the call gets disconnected. Do you see some NOTIFY requests going around? they are used during attended transfer to inform on the new call state. Regards, Bogdan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users Nope, no NOTIFY requests. Well wat is ment was making asterisk dumb, and just let if forward a complete call.. so instead of doing a dial to an opensips extention, just make a full transfer of the call to the opensips server, and then to the extention. I'm trying it the other way arround now, as you said earlier that the opensips recieves all the calls (so is directly connected to the sip trunk) but i have some strange issue's with that 2, i can't call outside and when i call inside, the phone rings (i just made a alias) and then i can't pick it up or anything, the phone doesn't respond! Any ideas ? -- View this message in context: http://n2.nabble.com/External-transfer-fails-from-Asterisk-tp3790325p3814573.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
Re: [OpenSIPS-Users] OpenSIPS - OpenXCAP integration
Hi, With XCAP auth scheme set to basic I do observe the same issue as mentioned in the ticket. I tried changing the XCAP authentication scheme from basic to digest. After this change, xdm client posted a second PUT with Auth details, but this time the server returned 500 Internal server error. Internally, I can see an OpenXCAP failure in base64.decodestring(input) Error base64.py, line 321 ... return bin2ascii.a2b_base64(s) incorrect padding. Not sure how to resolve this. Regards, Sanjeev -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Inaki Baz Castillo Sent: Tuesday, October 13, 2009 4:39 PM To: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] OpenSIPS - OpenXCAP integration El Martes, 13 de Octubre de 2009, Sanjeev BA escribió: I use Pytho-twisted 8.2.0-1ubuntu1on Ubuntu 8.04.3 Have you checked if your issue is the same as the report?: http://openxcap.org/ticket/121 Thanks. -- Iñaki Baz Castillo i...@aliax.net ___ 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] Monit tool not working in opensips-cp
HI Bodgan,
I want to implement the dialplan module. But i was confused little bit in
the configuration part. my database table of dialplan module is
++--++--+---+---+---+--+---+
| id | dpid | pr | match_op | match_exp | match_len | subst_exp |
repl_exp | attrs |
++--++--+---+---+---+--+---+
| 1 |1 | 1 |1 | (^67.+) | 0 | (^67.+) |
1...@192.168.3.36 | |
++--++--+---+---+---+--+---+
and my opensips.cfg configuration is:
modparam(dialplan, db_url, mysql://opensips:opensip...@localhost
/opensips)
route[14]
{
$var(x) = sip:678;
dp_translate(1, $var(x)/$var(tmp));
xlog(-$var(tmp)\n);
}.
My actual intention is when the client dials the number 678... a prefix 1
should add to that numbers. Is the above configuration correct for my
architecture?.
Thanks,
Nehru.
On Mon, Oct 12, 2009 at 10:37 AM, indiver nehru nehru.i...@gmail.comwrote:
Hi Bodgan,
Thanks a lot!. I solved my problem by keeping ssl
option to 0. Monit tool working!.
Thanks,
Nehru.
On Mon, Oct 12, 2009 at 10:23 AM, Bogdan-Andrei Iancu (via Nabble)
ml-user+121376-854570...@n2.nabble.comml-user%2b121376-854570...@n2.nabble.com
wrote:
Hi Nehru,
Check the SSL option for monit, in boxes.global.inc.php - maybe CP
tried to uses https instead of http.
Regards,
Bogdan
Indiver wrote:
Hi Everyone,
I was unable to connect to monit tool of opensips control panel. I
started
monit server and loaded mi_xmlrpc module in opensips correctly. i had
given
ip address, usename, password correctly in boxes.global.inc.php file.
Still
When i tried to connect monit tool it displays i can't connect error
message. can any one suggest the right configuration of monit tool
Thanks,
Nehru.
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[OpenSIPS-Users] Dialplan module not working
HI Everyone,
I want to implement the dialplan module. But i was confused little bit in
the configuration part. my database table of dialplan module is
++--++--+---+---+---+--+---+
| id | dpid | pr | match_op | match_exp | match_len | subst_exp | repl_exp
| attrs |
++--++--+---+---+---+--+---+
| 1 |1 | 1 |1 | (^67.+) | 0 | (^67.+) | [hidden
email] | |
++--++--+---+---+---+--+---+
and my opensips.cfg configuration is:
modparam(dialplan, db_url,
mysql://opensips:opensip...@localhost/opensips)
route[14]
{
$var(x) = sip:678;
dp_translate(1, $var(x)/$var(tmp));
xlog(-$var(tmp)\n);
}.
My actual intention is when the client dials the number 678... a prefix 1
should add to that numbers. Is the above configuration correct for my
architecture?.
Thanks,
Nehru.
--
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[OpenSIPS-Users] only 4x ring try.
Hello, A quik question, but i can't find it myself, my phone's are only rinning 4 times.. i would like it to keep rinning or as long as possible. Is this an opensips setting? We have a lot of diffrent sip phones and they all do the same. Best Regards! -- View this message in context: http://n2.nabble.com/only-4x-ring-try-tp3815208p3815208.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
Re: [OpenSIPS-Users] only 4x ring try.
El Martes, 13 de Octubre de 2009, Peter den Hartog escribió: Hello, A quik question, but i can't find it myself, my phone's are only rinning 4 times.. i would like it to keep rinning or as long as possible. Is this an opensips setting? Not at all. OpenSIPS is not a PBX. -- Iñaki Baz Castillo i...@aliax.net ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] only 4x ring try.
El Martes, 13 de Octubre de 2009, Jeff Pyle escribió: It could be the TM module modparam fr_inv_timer, or perhaps the fr_inv_timer_avp doing the same thing dynamcally. If you're seeing Opensips send a CANCEL towards the UAS and a 408 towards the UAC this seems the logical cause. Yes, however those parameters require custom configuration as with the default values they allow more than 4 rings on a phone :) So IMHO what's happening is that the caller is cancelling the call after those 4 rings. For that, the mail sender could inspect the SIP flow (with ngrep or ethereal) in order to know if OpenSIPS is receiving a CANCEL from the caller. Regards. -- Iñaki Baz Castillo i...@aliax.net ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] only 4x ring try.
Thank you! that was it :-)! it was on 20, it's off now, and it keeps rining. Thank you Jeff Pyle wrote: It could be the TM module modparam fr_inv_timer, or perhaps the fr_inv_timer_avp doing the same thing dynamcally. If you're seeing Opensips send a CANCEL towards the UAS and a 408 towards the UAC this seems the logical cause. A trace would be useful here. Links: http://www.opensips.org/html/docs/modules/devel/tm.html#id228480 http://www.opensips.org/html/docs/modules/devel/tm.html#id271154 - Jeff On 10/13/09 6:54 AM, Iñaki Baz Castillo i...@aliax.net wrote: El Martes, 13 de Octubre de 2009, Peter den Hartog escribió: Hello, A quik question, but i can't find it myself, my phone's are only rinning 4 times.. i would like it to keep rinning or as long as possible. Is this an opensips setting? Not at all. OpenSIPS is not a PBX. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- View this message in context: http://n2.nabble.com/only-4x-ring-try-tp3815208p3815926.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
Re: [OpenSIPS-Users] OpenSips as SMS-GW
Hi, OpenSIPS has a builtin SMS module, but for SMS over AT-modems - see the SMS module (http://www.opensips.org/html/docs/modules/1.5.x/sms.html). The module can control an AT-modem to send and receive SMS directly from the GSM network. Another options is to simply integrate (via DB for example) opensips with kannel (SMS gateway). Regards, Bogdan Ajay Pratap Singh Pundhir wrote: Hi , Any one have idea about can OpenSips/Openser be used as SMS-GW for the SMS over IP application ( I have OpenIMSCore Network Configured) . Is there any opensourse implimentaion of SMS over IP ?? Thanks -- --With Regards-- Ajay Pratap Singh Pundhir M.Tech International Institute of Information Technology, Bangalore. ___ 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] Dialplan module not working
Your very close..
look, your regex shows ^67.+ (starts with 67)
BUT your string does NOT start with 67. it starts with sip:67 (ie: si !=
67) so it just doesn't match.
Also, I wouldn't use the $var vars here.. instead try:
dp_translate(1, $ruri.user/$ruri.user)
THEN your shown regex will work ($ruri.user is just the dialed number, does
not contain the sip:)
-Brett
On Tue, Oct 13, 2009 at 3:05 AM, Indiver nehru.i...@gmail.com wrote:
HI Everyone,
I want to implement the dialplan module. But i was confused little bit in
the configuration part. my database table of dialplan module is
++--++--+---+---+---+--+---+
| id | dpid | pr | match_op | match_exp | match_len | subst_exp | repl_exp
| attrs |
++--++--+---+---+---+--+---+
| 1 |1 | 1 |1 | (^67.+) | 0 | (^67.+) | [hidden
email] | |
++--++--+---+---+---+--+---+
and my opensips.cfg configuration is:
modparam(dialplan, db_url,
mysql://opensips:opensip...@localhost/opensips)
route[14]
{
$var(x) = sip:678;
dp_translate(1, $var(x)/$var(tmp));
xlog(-$var(tmp)\n);
}.
My actual intention is when the client dials the number 678... a prefix 1
should add to that numbers. Is the above configuration correct for my
architecture?.
Thanks,
Nehru.
--
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Re: [OpenSIPS-Users] Dialplan module not working
Hi Nehru,
the rule matching regexp (match_exp ) expects to receive a string
starting with 67, but you pass a string with sip: in front.
Regards,
Bogdan
Indiver wrote:
HI Everyone,
I want to implement the dialplan module. But i was confused little bit in
the configuration part. my database table of dialplan module is
++--++--+---+---+---+--+---+
| id | dpid | pr | match_op | match_exp | match_len | subst_exp | repl_exp
| attrs |
++--++--+---+---+---+--+---+
| 1 |1 | 1 |1 | (^67.+) | 0 | (^67.+) | [hidden
email] | |
++--++--+---+---+---+--+---+
and my opensips.cfg configuration is:
modparam(dialplan, db_url,
mysql://opensips:opensip...@localhost/opensips)
route[14]
{
$var(x) = sip:678;
dp_translate(1, $var(x)/$var(tmp));
xlog(-$var(tmp)\n);
}.
My actual intention is when the client dials the number 678... a prefix 1
should add to that numbers. Is the above configuration correct for my
architecture?.
Thanks,
Nehru.
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Re: [OpenSIPS-Users] Are user registrations stored?
Hi, Sanjeev BA wrote: Hi, I would like to add one more observation. If the restart happens before the registration timer expiry, only then opensips remembers the previous registration. If I restart after timer expiry, the previous values are not present. make sense - why to restore a registration that is already expired and useless ??? Regards, Bogdan Any help would be appreciated. Regards, Sanjeev -Original Message- From: Sanjeev BA [mailto:as290...@samsung.com] Sent: Tuesday, October 13, 2009 12:15 PM To: 'OpenSIPS users mailling list' Subject: RE: [OpenSIPS-Users] Are user registrations stored? Here is the problem I have. I typically restart opensips multiple times and after each restart, my sip client REGISTERS with a new IP. However, opensips seems to remember all the old Contacts as I can see them listed in the 200 OK. The param is set to 0. Please advice. -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu Sent: Tuesday, October 13, 2009 12:03 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Are user registrations stored? Hi Sanjeev, Sanjeev BA wrote: Hi, Between two successive runs of opensips, are user registrations stored in a temporary file (or) DB? Could you please provide me the details of the file or table name? USRLOC module does persistence across reboots via DB (table location). But if the DB is used by the module, and how it is used, depends on the db_mode param. See: http://www.opensips.org/html/docs/modules/1.5.x/usrloc.html#id227873 By default, the mode is 0 - no persistence, so you need to explicitly set it to another value to get the DB persistence. Regards, Bogdan Regards Sanjeev ___ 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] Are user registrations stored?
Sanjeev BA wrote: Here is the problem I have. I typically restart opensips multiple times and after each restart, my sip client REGISTERS with a new IP. you restart opensips and the SIP client gets a new IP ? ? ? hard to belive it as there is no connection between the 2 actions. However, opensips seems to remember all the old Contacts as I can see them listed in the 200 OK. The param is set to 0. the param must be different than 0 to get registration persistence across restarts. Regards, Bogdan Please advice. -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu Sent: Tuesday, October 13, 2009 12:03 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Are user registrations stored? Hi Sanjeev, Sanjeev BA wrote: Hi, Between two successive runs of opensips, are user registrations stored in a temporary file (or) DB? Could you please provide me the details of the file or table name? USRLOC module does persistence across reboots via DB (table location). But if the DB is used by the module, and how it is used, depends on the db_mode param. See: http://www.opensips.org/html/docs/modules/1.5.x/usrloc.html#id227873 By default, the mode is 0 - no persistence, so you need to explicitly set it to another value to get the DB persistence. Regards, Bogdan Regards Sanjeev ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] ERROR:tm:t_forward_nonack: discarding fwd for a cancelled/6xx transaction
Hi Krunal,
could you send me (off list if necessary) a trace of the call and the
output of opensips in debug mode 6 ?
Regards,
Bogdan
Krunal Patel wrote:
Hello Bogdan,
I have your suggested code block , in my cfg.
But still getting the same.
--
Krunal Patel
On Mon, Oct 12, 2009 at 9:04 PM, Bogdan-Andrei Iancu
bog...@voice-system.ro mailto:bog...@voice-system.ro wrote:
Hi Krunal,
Looks like a race between the INVITE and CANCEL.typical way to
deal
with that is by processing the CANCEL only if it matches an INVITE
transaction:
# CANCEL processing
if (is_method(CANCEL))
{
if (t_check_trans())
t_relay();
exit;
}
do you have this in your script ?
Regards,
Bogdan
Krunal Patel wrote:
Hi,
I am getting issue like :
ERROR:tm:t_forward_nonack: discarding fwd for a cancelled/6xx
transaction
Here is the SIP trace.
U 2009/10/12 11:24:19.127160 XXX.XXX.XXX.XXX:5060 -
YYY.YYY.YYY.YYY:5060
INVITE sip:[EMAIL PROTECTED] SIP/2.0.
U 2009/10/12 11:24:19.127237 XXX.XXX.XXX.XXX:5060 -
YYY.YYY.YYY.YYY:5060
CANCEL sip:[EMAIL PROTECTED] SIP/2.0.
U 2009/10/12 11:24:19.127276 YYY.YYY.YYY.YYY:5060 -
XXX.XXX.XXX.XXX:5060
SIP/2.0 100 ci Trying...
U 2009/10/12 11:24:19.130913 YYY.YYY.YYY.YYY:5060 -
XXX.XXX.XXX.XXX:5060
SIP/2.0 200 canceling.
U 2009/10/12 11:24:19.142229 YYY.YYY.YYY.YYY:5060 -
XXX.XXX.XXX.XXX:5060
SIP/2.0 500 Server error occurred (19/SL).
U 2009/10/12 11:24:19.182452 XXX.XXX.XXX.XXX:5060 -
YYY.YYY.YYY.YYY:5060
ACK sip:[EMAIL PROTECTED] SIP/2.0.
This happens when OpenSIPS gets CANCEL from caller before it
sends 100
Trying to CALLER for the INVITE.
Please let me know how to resolve it.
I have put t_was_cancelled in onreply failure route both.
Thanks in advance,
--
Krunal Patel
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Re: [OpenSIPS-Users] call to the mobile/ landline when all the contacts fail
That only leaves the way you solve having different fr_inv_timer values (ring office for 15 seconds, mobile for 20, and home for 10). I found this post http://n2.nabble.com/Storing-modules-parameters-on-AVP-as-much-as-I-can-td3300421.html#a3300421 I guess there is currently no easy way of doing this. Correct? On Mon, Oct 12, 2009 at 10:08 PM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: such redirects can be easily done via failure_route() and loading new destination from DB (see avpops, the avp_db_load and avp_db_query functions). In failure_route[], when you detected the failed call, simply load and set a new RURI and to t_relay() again - serial forking. You do not need usrloc at all in this case. Regards, Bogdan osiris123d wrote: Via a web interface the user should be able to set their conditional call forwarding(Findme/followme). So they should be able to do multiple scenario's Ring Office, Mobile and Home at same time (This seems easy enough with Parrallel forking) Ring Office first for 15 seconds, then Mobile and Home at same time for 20 and finally to voicemail (I can accomplish the first 15 seconds with the fr_inv_timer, but i am not sure how to reference the second 20 seconds. I don't think I will have issues with the first serial call and then the parrallel forking) Ring Office for 15 seconds, then Mobile for 20 seconds, next Home for 13 and finally send to Voicemail (I believe I can accomplish this with the fr_inv_timer for the first 15 seconds, but I don't know how the second and third timers are referenced) The other part of this scenario would be to set up Time Periods like Only call Office number from 8 till 5 on weekdays and only call home phone after 5pm and on weekends and etc. I am currently looking at Dynamic Routing for the Time-Based features but not sure. Bogdan-Andrei Iancu wrote: Hi, what exactly is your scenario? what you want to achieve ? Regards, Bogdan osiris123d wrote: I can see how to use the serialize and append branches, but will this allow you to first call office phone, then mobile and finally home? I have read about the contacts part in the Location database table, but the thing thats confusing about that is that the contacts has an expire parameter which makes me think the contact won't be permanent. I figured you had to use an AVP for your home and mobile. Iñaki Baz Castillo wrote: El Jueves, 25 de Junio de 2009, ASHWINI NAIDU escribió: hi all, I wanted to implement this feature under serial forking. Whn all the contacts are busy/ unavailable i would like to forward the call to the person's landline or mobile number. how can this be accomplished. can any one help me out with this issue. failure_route and append_branch. Reading the doc is also useful since serial forking (what you mean in fact) is well explained. -- Iñaki Baz Castillo i...@aliax.net ___ 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 -- -- *--*--*--*--*--* Duane *--*--*--*--*--* -- ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Statefull Load balancing
Hi there, Could anyone point me on how to do load balancing with opensips with transaction sharing between 2 of these servers. I know there was few threads about this and they usually end with UCARP or heartbeat, but I would want to load balance between 2 opensips server with them sharing transactions between themselves if that is possible? Best regards, Josip ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] ERROR:tm:t_forward_nonack: discarding fwd for a cancelled/6xx transaction
Hi!
I think we experienced something similar, as far as I remember the following
fixes the issue in conjunction with the code below that Bogdan suggested:
For INVITE you do:
t_newtran();
t_reply(100,Trying);
if(!t_relay(0x05))
{
#sl_reply_error(); # do not send error
}
The idea is to create transaction as fast as possible, and do not reply if
t_relay() fails.
For CANCEL you do, like Bogdan said:
if (is_method(CANCEL))
{
if (t_check_trans())
t_relay();
exit;
}
You get an error in log, but 500 is not sent. As far as I remember, there was
no way to fix it more cleanly.
The situation can be easy reproduced if you slow down processing of INVITE
and then send CANCEL, for example with exec_dset(very_slow_script).
--
Best Regards,
Alex Massover
VoIP RD TL
Jajah Inc.
-Original Message-
From: users-boun...@lists.opensips.org [mailto:users-
boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
Sent: Tuesday, October 13, 2009 4:17 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] ERROR:tm:t_forward_nonack: discarding fwd
for a cancelled/6xx transaction
Hi Krunal,
could you send me (off list if necessary) a trace of the call and the
output of opensips in debug mode 6 ?
Regards,
Bogdan
Krunal Patel wrote:
Hello Bogdan,
I have your suggested code block , in my cfg.
But still getting the same.
--
Krunal Patel
On Mon, Oct 12, 2009 at 9:04 PM, Bogdan-Andrei Iancu
bog...@voice-system.ro mailto:bog...@voice-system.ro wrote:
Hi Krunal,
Looks like a race between the INVITE and CANCEL.typical way
to
deal
with that is by processing the CANCEL only if it matches an
INVITE
transaction:
# CANCEL processing
if (is_method(CANCEL))
{
if (t_check_trans())
t_relay();
exit;
}
do you have this in your script ?
Regards,
Bogdan
Krunal Patel wrote:
Hi,
I am getting issue like :
ERROR:tm:t_forward_nonack: discarding fwd for a cancelled/6xx
transaction
Here is the SIP trace.
U 2009/10/12 11:24:19.127160 XXX.XXX.XXX.XXX:5060 -
YYY.YYY.YYY.YYY:5060
INVITE sip:[EMAIL PROTECTED] SIP/2.0.
U 2009/10/12 11:24:19.127237 XXX.XXX.XXX.XXX:5060 -
YYY.YYY.YYY.YYY:5060
CANCEL sip:[EMAIL PROTECTED] SIP/2.0.
U 2009/10/12 11:24:19.127276 YYY.YYY.YYY.YYY:5060 -
XXX.XXX.XXX.XXX:5060
SIP/2.0 100 ci Trying...
U 2009/10/12 11:24:19.130913 YYY.YYY.YYY.YYY:5060 -
XXX.XXX.XXX.XXX:5060
SIP/2.0 200 canceling.
U 2009/10/12 11:24:19.142229 YYY.YYY.YYY.YYY:5060 -
XXX.XXX.XXX.XXX:5060
SIP/2.0 500 Server error occurred (19/SL).
U 2009/10/12 11:24:19.182452 XXX.XXX.XXX.XXX:5060 -
YYY.YYY.YYY.YYY:5060
ACK sip:[EMAIL PROTECTED] SIP/2.0.
This happens when OpenSIPS gets CANCEL from caller before it
sends 100
Trying to CALLER for the INVITE.
Please let me know how to resolve it.
I have put t_was_cancelled in onreply failure route both.
Thanks in advance,
--
Krunal Patel
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Re: [OpenSIPS-Users] PDD
So Brett did Bodgan's suggestion ever work? Brett Nemeroff wrote: Yeah, this is exactly what I was thinking.. Is setting the fr_timer avp really necessary there? It doesn't seem to change.. I'll give this a shot and let you know what I get.. Thanks! -Brett On Thu, Jun 11, 2009 at 10:04 AM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: ok :) so, do something like: enable restart_fr_on_each_reply and onreply_avp_mode (see http://www.opensips.org/html/docs/modules/1.5.x/tm.html#id271347) before sending out the INVITE, set $avp(fr_timer) =2 (see http://www.opensips.org/html/docs/modules/1.5.x/tm.html#id271112) , so that if no reply is coming in 2 secs, timeout will fire. in onreply_route, when receiving 100, set $avp(fr_inv_timer) =5; the timer will be reset and the new val used. in onreply_route, when receiving 18x, set $avp(fr_inv_timer) =200; the timer will be reset and the new val used. never tried this, to be honest :D... Regards, Bogdan Brett Nemeroff wrote: Ok, let me try this a different way because I think I may be spreading my confusion. :) The behavior I'm looking for is: INVITE goes to carrier. 100 Trying MUST come back in 2 seconds THEN 18X MUST come back in 5 seconds else FAIL THEN Allow ringing for 200 seconds from what you've described, it sounds like I can't have the 2 seconds and 5 seconds be different numbers (static tm module param fr_timer). But even now.. I have fr_timer set to 5 and it's perfectly happy waiting 20 seconds between 100 and 180. Why does it not time out there? Thanks! -Brett On Thu, Jun 11, 2009 at 9:44 AM, Bogdan-Andrei Iancu bog...@voice-system.ro mailto:bog...@voice-system.ro wrote: Brett, first of all you cannot reset the timers manually from script - the timers handling is automatically done by TM, totally transparent for the script. second, the restart_fr_on_each_reply controls the fr_inv_timer. Brett Nemeroff wrote: Ok, so at first I was thinking.. what I need to do is set the fr_inv_timer to something like 10 seconds. But then in the on_reply route, check for a 18X reset the fr_inv_timer to like 200 seconds to allow the call to ring. I'm pretty confused now.. I thought the fr_timer was the timer to get a provisional reply. so as soon as you get a 100 the timer isn't used anymore. This option suggests that if you set restart_fr_on_each_reply to 1, then after you get a 100 Trying, then it will allow for fr_timer seconds again before timing out. Is that right? This of course, leads me to my next question, the documentation says that by default it does this, but I'm certainly not seeing this behavior. maybe the the 100 + 180 where too fast and close to INVITE, and no other reply before 200 OK, so visibly effect of a restart let me make sure I have this clear please: fr_timer = time from trigger (request) to ANY reply. The trigger point depends on the restart_fr_on_each_reply setting. If off, it's just from the request. If on, each provisional reply will cause the timer to be reinvoked, else it would have been ignored. In other words if fr_timer = 5 seconds and I get a 100 Trying after 500ms, and then the 183 Ringing occurs 8 seconds later, the only way the timer would be tripped is if I set reset_fr_on_each_reply=1? fr_inv_timer = the max amount of time between an initial request and a positive final reply (2XX) as said , the restart_fr_on_each_reply controls the fr_inv_timer. NOTE that setting a new avp for fr_inv_timer in onreply_route (set the usage of AVPs in onreply_route) will update the value of the timer !! interesting ;). How mixed up am I? And is restart_fr_timer_on_each_reply really default to 1? yes :) and if so, why does it not work how I expect? (ie: now, as long as I get the 100 Trying in 5 secs (fr_timer) the 18X could come 20 seconds later and everything is happy (but me). not sure I get the scenario. Regards, Bogdan Thanks! -Brett On Thu, Jun 11, 2009 at 3:14 AM, Bogdan-Andrei Iancu bog...@voice-system.ro mailto:bog...@voice-system.ro mailto:bog...@voice-system.ro mailto:bog...@voice-system.ro wrote: Hi Brett, The relevant timer are: - A - timeout at transport level, if no reply comes back - B - timeout at transaction level, if the transaction did not completed (no final response received) What may help you is the fact that the B timer may be reset after each provisional reply. see: http://www.opensips.org/html/docs/modules/1.5.x/tm.html#id271074 Regards, Bogdan Brett
Re: [OpenSIPS-Users] PDD
Haven't had the opportunity to try it yet.. will soon and report.. :) On Tue, Oct 13, 2009 at 10:04 AM, osiris123d duane.lar...@gmail.com wrote: So Brett did Bodgan's suggestion ever work? Brett Nemeroff wrote: Yeah, this is exactly what I was thinking.. Is setting the fr_timer avp really necessary there? It doesn't seem to change.. I'll give this a shot and let you know what I get.. Thanks! -Brett On Thu, Jun 11, 2009 at 10:04 AM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: ok :) so, do something like: enable restart_fr_on_each_reply and onreply_avp_mode (see http://www.opensips.org/html/docs/modules/1.5.x/tm.html#id271347) before sending out the INVITE, set $avp(fr_timer) =2 (see http://www.opensips.org/html/docs/modules/1.5.x/tm.html#id271112) , so that if no reply is coming in 2 secs, timeout will fire. in onreply_route, when receiving 100, set $avp(fr_inv_timer) =5; the timer will be reset and the new val used. in onreply_route, when receiving 18x, set $avp(fr_inv_timer) =200; the timer will be reset and the new val used. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Statefull Load balancing
We do this with relative success using DNS load balancing. Our two boxes are randomly load balanced, not precisely half half. We then use a script that fires off test SIP messages at the boxes every 60 seconds, and run a second script that removes the entry from our DNS server should one of the boxes not respond (likewise, the script re-inserts them when they start responding again.) It's not totally optimal, but it works for us. Frankly the biggest problem was sharing variables across the boxes; we ended up with the need to track concurrent number of calls on various domains, and because those calls could originate/terminate via either box, we ended up using a complicated database script inside Opensips to manage cumulative variables. Cheers, Jeff From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Josip Djuricic Sent: Tuesday, October 13, 2009 8:46 AM To: users@lists.opensips.org Subject: [OpenSIPS-Users] Statefull Load balancing Importance: High Hi there, Could anyone point me on how to do load balancing with opensips with transaction sharing between 2 of these servers. I know there was few threads about this and they usually end with UCARP or heartbeat, but I would want to load balance between 2 opensips server with them sharing transactions between themselves if that is possible? Best regards, Josip ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] multiple opensips_radius instances in CDRTool
Hello, I have a working CDRTool 6.8.0 configuration. It looks at one Opensips radius data source, and one Asterisk data source. I'm attempting to add a second Opensips radius data source. I'm having trouble. If I add a second opensips_radius instance, the second one seems to overwrite the first. It works, but the first is gone. If I add it with a different name, say, opensips_radius_sun, it doesn't show up in the available Data source drop-down box. It does, however, appear as an available data source in the permissions section when editing a user. What is the proper way to do this? Thanks, Jeff ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OpenSips as SMS-GW
Thanks, is there any open source client available for SMS service in IMS. or i have to make one from scratch. On Tue, Oct 13, 2009 at 7:32 PM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Hi, OpenSIPS has a builtin SMS module, but for SMS over AT-modems - see the SMS module (http://www.opensips.org/html/docs/modules/1.5.x/sms.html). The module can control an AT-modem to send and receive SMS directly from the GSM network. Another options is to simply integrate (via DB for example) opensips with kannel (SMS gateway). Regards, Bogdan Ajay Pratap Singh Pundhir wrote: Hi , Any one have idea about can OpenSips/Openser be used as SMS-GW for the SMS over IP application ( I have OpenIMSCore Network Configured) . Is there any opensourse implimentaion of SMS over IP ?? Thanks -- --With Regards-- Ajay Pratap Singh Pundhir M.Tech International Institute of Information Technology, Bangalore. ___ 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 -- --With Regards-- Ajay Pratap Singh Pundhir M.Tech International Institute of Information Technology, Bangalore. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] An Old OpenSER Error For A New OpenSIPS User
I am certain for anyone experienced this is an easy fix, so help me out OpenSIPS community. I am using OpenSER Source 1.2.2 on Linux version 2.6.26-2-686 (Debian 2.6.26-19) (gcc version 4.1.3 20080704 (prerelease) (Debian 4.1.2-25)). I am using Flavio's book. On page 100 (Adding Authentication with MySQL) at step 4 it states: Configure two user accounts using the openserctl utility. /sbin/openserctl add 1000 password 1...@voffice.com.br (mydomain.com) /sbin/openserctl add 1001 password 1...@voffice.com.br (mydomain.com) In either instance the system returns: Database 'MYSQL' loaded Control engine 'FIFO' loaded is_user: user counter=0 check_db_alias: alias counter=0 ERROR 1045 (28000): Access denied for user 'openser'@'localhost' (using password: YES) or ERROR 1054 (42S22) at line 1: Unknown column 'phplib_id' in 'field list' ERROR: introducing the new user'1000' to the database failed I have verified that the users openser, openserro and root have the correct passwords to access the MySQL database named openser (as one forum stated it was a password issue). I have tried several changes to the file /etc/openser/openserctlrc as several online forums indicated this was the issue. I have found this listed in many online forums but I have not been able to fix this trying several of the things that are mentioned. Can anyone on this users@lists.opensips.org please advise me? I am using OpensSER 1.2.2 as that is what the book I have procured indicates to use and I want to strictly follow what Flavio's book states to use. Thanks in advance. Respectfully, Larry Kemp Network Engineer U.S. Metropolitan Telecom, LLC Bonita Springs, FL USA ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OpenSips as SMS-GW
what is a client for SMS services ??? everything is SIP here, so maybe you need a SIP client with IM support, right? Regards, Bogdan Ajay Pratap Singh Pundhir wrote: Thanks, is there any open source client available for SMS service in IMS. or i have to make one from scratch. On Tue, Oct 13, 2009 at 7:32 PM, Bogdan-Andrei Iancu bog...@voice-system.ro mailto:bog...@voice-system.ro wrote: Hi, OpenSIPS has a builtin SMS module, but for SMS over AT-modems - see the SMS module (http://www.opensips.org/html/docs/modules/1.5.x/sms.html). The module can control an AT-modem to send and receive SMS directly from the GSM network. Another options is to simply integrate (via DB for example) opensips with kannel (SMS gateway). Regards, Bogdan Ajay Pratap Singh Pundhir wrote: Hi , Any one have idea about can OpenSips/Openser be used as SMS-GW for the SMS over IP application ( I have OpenIMSCore Network Configured) . Is there any opensourse implimentaion of SMS over IP ?? Thanks -- --With Regards-- Ajay Pratap Singh Pundhir M.Tech International Institute of Information Technology, Bangalore. ___ 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 -- --With Regards-- Ajay Pratap Singh Pundhir M.Tech International Institute of Information Technology, Bangalore. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] ERROR:tm:t_forward_nonack: discarding fwd for a cancelled/6xx transaction
Hi Alex,
That is correct, but I was interested to see the logs to see why the
parallel processing of CANCEL and INVITE leads to an 500it would
rather be a local 487 cancelledBut I need the logs to see how the
processing takes place.
Regards,
Bogdan
Alex Massover wrote:
Hi!
I think we experienced something similar, as far as I remember the following
fixes the issue in conjunction with the code below that Bogdan suggested:
For INVITE you do:
t_newtran();
t_reply(100,Trying);
if(!t_relay(0x05))
{
#sl_reply_error(); # do not send error
}
The idea is to create transaction as fast as possible, and do not reply if
t_relay() fails.
For CANCEL you do, like Bogdan said:
if (is_method(CANCEL))
{
if (t_check_trans())
t_relay();
exit;
}
You get an error in log, but 500 is not sent. As far as I remember, there was
no way to fix it more cleanly.
The situation can be easy reproduced if you slow down processing of INVITE
and then send CANCEL, for example with exec_dset(very_slow_script).
--
Best Regards,
Alex Massover
VoIP RD TL
Jajah Inc.
-Original Message-
From: users-boun...@lists.opensips.org [mailto:users-
boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
Sent: Tuesday, October 13, 2009 4:17 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] ERROR:tm:t_forward_nonack: discarding fwd
for a cancelled/6xx transaction
Hi Krunal,
could you send me (off list if necessary) a trace of the call and the
output of opensips in debug mode 6 ?
Regards,
Bogdan
Krunal Patel wrote:
Hello Bogdan,
I have your suggested code block , in my cfg.
But still getting the same.
--
Krunal Patel
On Mon, Oct 12, 2009 at 9:04 PM, Bogdan-Andrei Iancu
bog...@voice-system.ro mailto:bog...@voice-system.ro wrote:
Hi Krunal,
Looks like a race between the INVITE and CANCEL.typical way
to
deal
with that is by processing the CANCEL only if it matches an
INVITE
transaction:
# CANCEL processing
if (is_method(CANCEL))
{
if (t_check_trans())
t_relay();
exit;
}
do you have this in your script ?
Regards,
Bogdan
Krunal Patel wrote:
Hi,
I am getting issue like :
ERROR:tm:t_forward_nonack: discarding fwd for a cancelled/6xx
transaction
Here is the SIP trace.
U 2009/10/12 11:24:19.127160 XXX.XXX.XXX.XXX:5060 -
YYY.YYY.YYY.YYY:5060
INVITE sip:[EMAIL PROTECTED] SIP/2.0.
U 2009/10/12 11:24:19.127237 XXX.XXX.XXX.XXX:5060 -
YYY.YYY.YYY.YYY:5060
CANCEL sip:[EMAIL PROTECTED] SIP/2.0.
U 2009/10/12 11:24:19.127276 YYY.YYY.YYY.YYY:5060 -
XXX.XXX.XXX.XXX:5060
SIP/2.0 100 ci Trying...
U 2009/10/12 11:24:19.130913 YYY.YYY.YYY.YYY:5060 -
XXX.XXX.XXX.XXX:5060
SIP/2.0 200 canceling.
U 2009/10/12 11:24:19.142229 YYY.YYY.YYY.YYY:5060 -
XXX.XXX.XXX.XXX:5060
SIP/2.0 500 Server error occurred (19/SL).
U 2009/10/12 11:24:19.182452 XXX.XXX.XXX.XXX:5060 -
YYY.YYY.YYY.YYY:5060
ACK sip:[EMAIL PROTECTED] SIP/2.0.
This happens when OpenSIPS gets CANCEL from caller before it
sends 100
Trying to CALLER for the INVITE.
Please let me know how to resolve it.
I have put t_was_cancelled in onreply failure route both.
Thanks in advance,
--
Krunal Patel
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Re: [OpenSIPS-Users] External transfer fails (from Asterisk)
Peter den Hartog wrote: Hello Bogdan, That is correct, in Asterisk i see nothing of a new call, or a transfer.. but the phone is creating a new call on line 2, in opensips i just see a new ongoing call. (the line 2 call) and on the outside phone i hear the asterisk wait/hold music. Is there any smart solution for this? can i just forward the complete call to opensips and let asterisk only forward it, and not create the call? (it now just does a dial to the sip member in opensips) Oke a little update, i can now do blind (cold) transfers from asterisk to opensips (outside lines) but not hot transfers, then the call gets disconnected. Do you see some NOTIFY requests going around? they are used during attended transfer to inform on the new call state. Nope, no NOTIFY requests. Well wat is ment was making asterisk dumb, and just let if forward a complete call.. so instead of doing a dial to an opensips extention, just make a full transfer of the call to the opensips server, and then to the extention. I'm trying it the other way arround now, as you said earlier that the opensips recieves all the calls (so is directly connected to the sip trunk) but i have some strange issue's with that 2, i can't call outside and when i call inside, the phone rings (i just made a alias) and then i can't pick it up or anything, the phone doesn't respond! Hi Peter, have you checked the SIP trace to see why the call is not established when you pick up the ringing phone ? Regards, Bogdan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] DNS failover on 6xx reply
Hello, Help / advice on the following issue would be much appreciated. We are running a setup where a front-end opensips box is relaying calls to two or more end-client HMP machines (e.g., asterisk). We are using DNS/SRV records to failover between the end-client machines - which seems to be working fine when the actual HMP service is completely stopped. In some cases tough, the HMP service is running in a particular state where it is returning 603 replies which in turn seem to be breaking the DNS failover protocol. Note: in a separate setup we are successfully running a transaction/dispatcher based failover configuration for which we needed the following: modparam(tm, disable_6xx_block, 1) So the question is - is there any way to deal with 6xx response codes such that the DNS failover mechanism works (perhaps by using a setting similar to the one above) ? Thank you, Ion ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] CANCEL and INVITE Routing issues
I am noticing two issues that I think are due to myself using Multiple
Domains for users
1. When a user calls someone via a SIP trunk and cancels the call the SIP
proxy never sends the CANCEL message to the SIP Trunk provider. In my Main
Route I have the following code so I should be handling the CANCEL message
just fine.
#CANCEL processing
if (is_method(CANCEL)) {
if (t_check_trans()) {
end_media_session();
t_relay();
};
exit;
}
2. When a user calls someone via a SIP trunk the invite is sent to the SIP
trunk provider but also to an IP address that resolves to the Caller's
domain. I only want the SIP proxy to send the INVITE to the SIP Trunk
Provider since there is no SIP proxy at the @domain.com site.
For testing purposes the irock.com domain is used for one of the users (I
don't own irock.com I am just using that for testing). When
9x12x32...@irock.com calls 9x18x13182 the following is performed in the
config
t_relay(udp:pt1.vitelity.net:5060);
So the invite that gets sent out to my SIP Trunk provider is this
U 6x.80.xxx.14:5060 - 64.2.142.93:5060
INVITE sip:9x18x13...@irock.com SIP/2.0.
Record-Route: sip:6x.80.xxx.14;lr=on;ftag=2f27602c;nat=yes;did=535.18b17.
Via: SIP/2.0/UDP 6x.80.xxx.14;branch=z9hG4bK0de8.b68a49e7.0.
Via: SIP/2.0/UDP
192.168.100.80:22894;received=192.251.125.xx;branch=z9hG4bK-d8754z-02edb391ba5fad01-1---d8754z-;rport=10076.
Max-Forwards: 69.
Contact: sip:9x12x32...@192.251.125.xx:10076;transport=udp.
To: sip:19x18x13...@irock.com.
From: Duanesip:9x12x32...@irock.com;tag=2f27602c.
Call-ID: OTkwYTc4MjY0NTU0NjFhMGU2YjgxNDQ5N2JjYjg5YmE..
CSeq: 2 INVITE.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
INFO.
Content-Type: application/sdp.
User-Agent: Bria release 2.5.4 stamp 53956.
Content-Length: 258.
P-hint: route(3)|setflag7,forcerport,fix_contact.
P-hint: inbound-outbound .
P-hint: Route[6]: mediaproxy .
.
v=0.
o=- 9 2 IN IP4 192.168.100.80.
s=CounterPath Bria.
c=IN IP4 6x.80.xxx.13.
t=0 0.
m=audio 10854 RTP/AVP 107 0 8 18 101.
a=sendrecv.
a=rtpmap:107 BV32/16000.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=yes.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
I think the issue is the domain part in the INVITE part of that SIP message
(INVITE sip:9018313...@irock.com SIP/2.0.) because when the softphone user
9x12x32...@irock.com cancels the call I see the following SIP messages
U 192.251.125.xx:10076 - 6x.80.xxx.14:5060
CANCEL sip:19x18x13...@irock.com SIP/2.0.
Via: SIP/2.0/UDP
192.168.100.80:22894;branch=z9hG4bK-d8754z-02edb391ba5fad01-1---d8754z-;rport.
Max-Forwards: 70.
To: sip:19x18x13...@irock.com.
From: Duanesip:9x12x32...@irock.com;tag=2f27602c.
Call-ID: OTkwYTc4MjY0NTU0NjFhMGU2YjgxNDQ5N2JjYjg5YmE..
CSeq: 2 CANCEL.
Proxy-Authorization: Digest
username=9x12x32009,realm=irock.com,nonce=4acfa38a000243e104b0d51fed793230657dae910da5,uri=si
p:19x18x13...@irock.com,response=7364463734bed01be73dc4cd92742fc1,cnonce=2f2d71fe87bfdcd1325e1cb15210d1ef,nc=0002,qop=auth,
algorithm=MD5.
User-Agent: Bria release 2.5.4 stamp 53956.
Content-Length: 0.
.
U 6x.80.xxx.14:5060 - 192.251.125.xx:10076
SIP/2.0 200 canceling.
Via: SIP/2.0/UDP
192.168.100.80:22894;branch=z9hG4bK-d8754z-02edb391ba5fad01-1---d8754z-;rport=10076;received=192.251.125.xx.
To: sip:19x18x13...@irock.com;tag=d733b86164332c9143cd163ddc2dfbf1-6635.
From: Duanesip:9x12x32...@irock.com;tag=2f27602c.
Call-ID: OTkwYTc4MjY0NTU0NjFhMGU2YjgxNDQ5N2JjYjg5YmE..
CSeq: 2 CANCEL.
Server: .
Content-Length: 0.
.
U 6x.80.xxx.14:5060 - 192.251.125.xx:10076
SIP/2.0 487 Request Terminated.
Via: SIP/2.0/UDP
192.168.100.80:22894;branch=z9hG4bK-d8754z-02edb391ba5fad01-1---d8754z-;rport=10076;received=192.251.125.xx.
To: sip:19x18x13...@irock.com;tag=d733b86164332c9143cd163ddc2dfbf1-6635.
From: Duanesip:9x12x32...@irock.com;tag=2f27602c.
Call-ID: OTkwYTc4MjY0NTU0NjFhMGU2YjgxNDQ5N2JjYjg5YmE..
CSeq: 2 INVITE.
Server: .
Content-Length: 0.
.
U 192.251.125.xx:10076 - 6x.80.xxx.14:5060
ACK sip:19x18x13...@irock.com SIP/2.0.
Via: SIP/2.0/UDP
192.168.100.80:22894;branch=z9hG4bK-d8754z-02edb391ba5fad01-1---d8754z-;rport.
Max-Forwards: 70.
To: sip:19x18x13...@irock.com;tag=d733b86164332c9143cd163ddc2dfbf1-6635.
From: Duanesip:9x12x32...@irock.com;tag=2f27602c.
Call-ID: OTkwYTc4MjY0NTU0NjFhMGU2YjgxNDQ5N2JjYjg5YmE..
CSeq: 2 ACK.
Content-Length: 0.
.
U 6x.80.xxx.14:5060 - 192.251.125.xx:10076
SIP/2.0 487 Request Terminated.
Via: SIP/2.0/UDP
192.168.100.80:22894;branch=z9hG4bK-d8754z-02edb391ba5fad01-1---d8754z-;rport=10076;received=192.251.125.xx.
To: sip:19x18x13...@irock.com;tag=d733b86164332c9143cd163ddc2dfbf1-6635.
From: Duanesip:9x12x32...@irock.com;tag=2f27602c.
Call-ID: OTkwYTc4MjY0NTU0NjFhMGU2YjgxNDQ5N2JjYjg5YmE..
CSeq: 2 INVITE.
Server: .
Content-Length: 0.
So my SIP Proxy is never telling
Re: [OpenSIPS-Users] DNS failover on 6xx reply
Salut Ion, based on what RFC 3263 and RFC 3261say, the DNS failover is trigger (at signalling level) only by local 408 (local timeout) and received 503. The 603 does not trigger DNS failover (at least according to the specs). Regards, Bogdan Ion Constantinescu wrote: Hello, Help / advice on the following issue would be much appreciated. We are running a setup where a front-end opensips box is relaying calls to two or more end-client HMP machines (e.g., asterisk). We are using DNS/SRV records to failover between the end-client machines - which seems to be working fine when the actual HMP service is completely stopped. In some cases tough, the HMP service is running in a particular state where it is returning 603 replies which in turn seem to be breaking the DNS failover protocol. Note: in a separate setup we are successfully running a transaction/dispatcher based failover configuration for which we needed the following: modparam(tm, disable_6xx_block, 1) So the question is - is there any way to deal with 6xx response codes such that the DNS failover mechanism works (perhaps by using a setting similar to the one above) ? Thank you, Ion ___ 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] check_source_address() failures
Hello, A request arrives at Opensips 1.6 from address 175.88.228.19:5068 on UDP. The address table contains: ++-+--+--+--+---+--+--+ | id | grp | ip | mask | port | proto | pattern | context_info | ++-+--+--+--+---+--+--+ | 38 | 10 | 175.88.228.0 | 24 |0 | udp | ^sip:.*$ | src_test | ++-+--+--+--+---+--+--+ I would expect if(check_source_address(10)) to succeed, yet it does not. Any thoughts? I'm using check_source_address() elsewhere in my config to emulate the old allow_trusted, and that works just fine. Thanks, Jeff ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] check_source_address() failures
I know on mine I had to set port to 5060 (5068 in your case) or whatever you want, otherwise it would not work for me. Im sure you can use a * for any port number as well, but I have not tried that. Jeff Pyle wrote: Hello, A request arrives at Opensips 1.6 from address 175.88.228.19:5068 on UDP. The address table contains: ++-+--+--+--+---+--+--+ | id | grp | ip | mask | port | proto | pattern | context_info | ++-+--+--+--+---+--+--+ | 38 | 10 | 175.88.228.0 | 24 |0 | udp | ^sip:.*$ | src_test | ++-+--+--+--+---+--+--+ I would expect if(check_source_address(10)) to succeed, yet it does not. Any thoughts? I'm using check_source_address() elsewhere in my config to emulate the old allow_trusted, and that works just fine. Thanks, Jeff ___ 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] Are user registrations stored?
Hi, I think I did not state my problem clearly. I restart both opensips and my sip client. After restart of both opensips and sip client, the sip client registers with a new Contact. Since I DO NOT want registration persistence across restarts, I have set the db_mode param to 0. Inspite of this, opensips remembers the previous Contacts of my sip client. (I can see the older contacts in 200 OK of REGISTER) This is my issue. Thanks and Regards, Sanjeev -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu Sent: Tuesday, October 13, 2009 11:15 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Are user registrations stored? Sanjeev BA wrote: Here is the problem I have. I typically restart opensips multiple times and after each restart, my sip client REGISTERS with a new IP. you restart opensips and the SIP client gets a new IP ? ? ? hard to belive it as there is no connection between the 2 actions. However, opensips seems to remember all the old Contacts as I can see them listed in the 200 OK. The param is set to 0. the param must be different than 0 to get registration persistence across restarts. Regards, Bogdan Please advice. -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu Sent: Tuesday, October 13, 2009 12:03 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Are user registrations stored? Hi Sanjeev, Sanjeev BA wrote: Hi, Between two successive runs of opensips, are user registrations stored in a temporary file (or) DB? Could you please provide me the details of the file or table name? USRLOC module does persistence across reboots via DB (table location). But if the DB is used by the module, and how it is used, depends on the db_mode param. See: http://www.opensips.org/html/docs/modules/1.5.x/usrloc.html#id227873 By default, the mode is 0 - no persistence, so you need to explicitly set it to another value to get the DB persistence. Regards, Bogdan Regards Sanjeev ___ 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 as SMS-GW
yes i am talking about SIP Client which support SMS. 3GPP has recently released standard for SMS over IP, so i am talking about the Softphone which support SMS service, i.e. which can register as a SMS capable device, and can Send/Recieve SMS in the format described in Stsndards. One idea which came to my mind is to add one more tab in UCTIMSClient for SMS. On Wed, Oct 14, 2009 at 2:37 AM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: what is a client for SMS services ??? everything is SIP here, so maybe you need a SIP client with IM support, right? Regards, Bogdan Ajay Pratap Singh Pundhir wrote: Thanks, is there any open source client available for SMS service in IMS. or i have to make one from scratch. On Tue, Oct 13, 2009 at 7:32 PM, Bogdan-Andrei Iancu bog...@voice-system.ro mailto:bog...@voice-system.ro wrote: Hi, OpenSIPS has a builtin SMS module, but for SMS over AT-modems - see the SMS module (http://www.opensips.org/html/docs/modules/1.5.x/sms.html ). The module can control an AT-modem to send and receive SMS directly from the GSM network. Another options is to simply integrate (via DB for example) opensips with kannel (SMS gateway). Regards, Bogdan Ajay Pratap Singh Pundhir wrote: Hi , Any one have idea about can OpenSips/Openser be used as SMS-GW for the SMS over IP application ( I have OpenIMSCore Network Configured) . Is there any opensourse implimentaion of SMS over IP ?? Thanks -- --With Regards-- Ajay Pratap Singh Pundhir M.Tech International Institute of Information Technology, Bangalore. ___ 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 -- --With Regards-- Ajay Pratap Singh Pundhir M.Tech International Institute of Information Technology, Bangalore. ___ 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 -- --With Regards-- Ajay Pratap Singh Pundhir M.Tech International Institute of Information Technology, Bangalore. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] OpenSIPS returns 400 Bad Request for PUBLISH
Hi, I am observing an error response from OpenSIPS. It returns 400 Bad Request for the following PUBLISH request. PUBLISH sip:tes...@imsdemo.com SIP/2.0 Expires: 3600 Event: presence Privacy: None Accept-Contact: *;+g.3gpp.Pua;explicit Contact: sip:tes...@10.10.10.20 Call-ID: 2189952...@imsdemo.com CSeq: 1 PUBLISH Max-Forwards: 70 From: sip:tes...@imsdemo.com;tag=1842963402 To: sip:tes...@imsdemo.com Via:SIP/2.0/UDP 10.254.140.195:5060;branch=z9hG4bK132770389smg;transport=UDP Content-Length: 0 Regards, Sanjeev ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OpenSIPS returns 400 Bad Request for PUBLISH
Error logs from OpenSIPS console.
Oct 14 14:01:46 [4880] DBG:core:parse_msg: method: PUBLISH
Oct 14 14:01:46 [4880] DBG:core:parse_msg: uri:
sip:tes...@imsdemo.com
Oct 14 14:01:46 [4880] DBG:core:parse_msg: version: SIP/2.0
Oct 14 14:01:46 [4880] DBG:core:parse_headers: flags=2
Oct 14 14:01:46 [4880] DBG:core:get_hdr_field: cseq CSeq: 1 PUBLISH
Oct 14 14:01:46 [4880] DBG:core:parse_to: end of header reached, state=10
Oct 14 14:01:46 [4880] DBG:core:parse_to: display={},
ruri={sip:tes...@imsdemo.com}
Oct 14 14:01:46 [4880] DBG:core:get_hdr_field: To [27];
uri=[sip:tes...@imsdemo.com]
Oct 14 14:01:46 [4880] DBG:core:get_hdr_field: to body
[sip:tes...@imsdemo.com
]
Oct 14 14:01:46 [4880] DBG:core:parse_via_param: found param type 232,
branch = z9hG4bK2559870494smg; state=6
Oct 14 14:01:46 [4880] DBG:core:parse_via_param: found param type 238,
transport = UDP; state=16
Oct 14 14:01:46 [4880] DBG:core:parse_via: end of header reached, state=5
Oct 14 14:01:46 [4880] DBG:core:parse_headers: via found, flags=2
Oct 14 14:01:46 [4880] DBG:core:parse_headers: this is the first via
Oct 14 14:01:46 [4880] DBG:core:receive_msg: After parse_msg...
Oct 14 14:01:46 [4880] DBG:core:receive_msg: preparing to run routing
scripts...
Oct 14 14:01:46 [4880] DBG:maxfwd:is_maxfwd_present: value = 70
Oct 14 14:01:46 [4880] DBG:core:parse_headers: flags=80
Oct 14 14:01:46 [4880] DBG:uri:has_totag: no totag
Oct 14 14:01:46 [4880] DBG:core:parse_headers: flags=78
Oct 14 14:01:46 [4880] DBG:tm:t_lookup_request: start searching: hash=30511,
isACK=0
Oct 14 14:01:46 [4880] DBG:tm:matching_3261: RFC3261 transaction matching
failed
Oct 14 14:01:46 [4880] DBG:tm:t_lookup_request: no transaction found
Oct 14 14:01:46 [4880] DBG:core:parse_headers: flags=200
Oct 14 14:01:46 [4880] DBG:core:get_hdr_field: content_length=0
Oct 14 14:01:46 [4880] DBG:core:get_hdr_field: found end of header
Oct 14 14:01:46 [4880] DBG:rr:find_first_route: No Route headers found
Oct 14 14:01:46 [4880] DBG:rr:loose_route: There is no Route HF
Oct 14 14:01:46 [4880] ERROR:rr:w_record_route: Double attempt to
record-route
Oct 14 14:01:46 [4880] DBG:core:grep_sock_info: checking if host==us: 11==12
[imsdemo.com] == [10.89.10.235]
Oct 14 14:01:46 [4880] DBG:core:grep_sock_info: checking if port 5060
matches port 5060
Oct 14 14:01:46 [4880] DBG:core:parse_headers: flags=
Oct 14 14:01:46 [4880] DBG:presence:search_event: start event= [presence]
Oct 14 14:01:46 [4880] DBG:presence:handle_publish: SIP-If-Match header not
found
Oct 14 14:01:46 [4880] DBG:presence:generate_ETag: etag=
a.1255496470.4880.1.0 / 21
Oct 14 14:01:46 [4880] DBG:presence:handle_publish: new etag =
a.1255496470.4880.1.0
Oct 14 14:01:46 [4880] DBG:presence:handle_publish: Expires header found,
value= 3600
Oct 14 14:01:46 [4880] ERROR:presence:handle_publish: No E-Tag and no body
found
Oct 14 14:01:46 [4880] DBG:core:parse_headers: flags=
Oct 14 14:01:46 [4880] DBG:core:check_via_address: params 10.254.140.195,
10.254.140.195, 0
Oct 14 14:01:46 [4880] DBG:sl:run_sl_callbacks: callback id 0 entered
Oct 14 14:01:46 [4880] DBG:tm:t_newtran: transaction on entrance=(nil)
Oct 14 14:01:46 [4880] DBG:core:parse_headers: flags=
Oct 14 14:01:46 [4880] DBG:core:parse_headers: flags=78
Oct 14 14:01:46 [4880] DBG:tm:t_lookup_request: start searching: hash=30511,
isACK=0
Oct 14 14:01:46 [4880] DBG:tm:matching_3261: RFC3261 transaction matching
failed
Oct 14 14:01:46 [4880] DBG:tm:t_lookup_request: no transaction found
Oct 14 14:01:46 [4880] DBG:tm:run_reqin_callbacks: trans=0xb3a09a30,
callback type 1, id 0 entered
Oct 14 14:01:46 [4880] DBG:tm:cleanup_uac_timers: RETR/FR timers reset
Oct 14 14:01:46 [4880] DBG:tm:insert_timer_unsafe: [2]: 0xb3a09a78 (41)
Oct 14 14:01:46 [4880] DBG:tm:t_unref: UNREF_UNSAFE: after is 0
Oct 14 14:01:46 [4880] DBG:core:destroy_avp_list: destroying list (nil)
Oct 14 14:01:46 [4880] DBG:core:receive_msg: cleaning up
Oct 14 14:01:51 [4885] DBG:tm:timer_routine: timer routine:2,tl=0xb3a09a78
next=(nil), timeout=41
Oct 14 14:01:51 [4885] DBG:tm:wait_handler: removing 0xb3a09a30 from table
Oct 14 14:01:51 [4885] DBG:tm:delete_cell: delete transaction 0xb3a09a30
Oct 14 14:01:51 [4885] DBG:tm:wait_handler: done
Oct 14 14:02:50 [4885] DBG:presence:update_db_subs: delete expired
Oct 14 14:02:50 [4885] DBG:db_mysql:db_mysql_do_prepared_query:
conn=0x8167ee8 (tail=135699504) MC=0x816ba48
Oct 14 14:02:50 [4885] DBG:db_mysql:db_mysql_do_prepared_query: new
query=|delete from active_watchers where expires?|
Oct 14 14:02:50 [4885] DBG:db_mysql:db_mysql_do_prepared_query: new
statement(0x8169f70) on connection: (0x8167ee8) 0x8169c30
Oct 14 14:02:50 [4885] DBG:db_mysql:db_mysql_do_prepared_query: set values
for the statement run
Oct 14 14:02:50 [4885] DBG:db_mysql:db_mysql_val2bind: added val (0): len=4;
type=3; is_null=0
Oct 14 14:02:50 [4885]
Re: [OpenSIPS-Users] OpenSIPS returns 400 Bad Request for PUBLISH
Error messages are quite descriptive ;) You've tried to record_route
that PUBLISH request twice... and also the content length it's zero so
what does that PUBLISH actulally 'mean'?
Maybe if you show us some little piece of your script (the one andling
the publish stuff) we could help you on the first error.
On Wed, Oct 14, 2009 at 7:19 AM, Sanjeev BA as290...@samsung.com wrote:
Error logs from OpenSIPS console.
Oct 14 14:01:46 [4880] DBG:core:parse_msg: method: PUBLISH
Oct 14 14:01:46 [4880] DBG:core:parse_msg: uri:
sip:tes...@imsdemo.com
Oct 14 14:01:46 [4880] DBG:core:parse_msg: version: SIP/2.0
Oct 14 14:01:46 [4880] DBG:core:parse_headers: flags=2
Oct 14 14:01:46 [4880] DBG:core:get_hdr_field: cseq CSeq: 1 PUBLISH
Oct 14 14:01:46 [4880] DBG:core:parse_to: end of header reached, state=10
Oct 14 14:01:46 [4880] DBG:core:parse_to: display={},
ruri={sip:tes...@imsdemo.com}
Oct 14 14:01:46 [4880] DBG:core:get_hdr_field: To [27];
uri=[sip:tes...@imsdemo.com]
Oct 14 14:01:46 [4880] DBG:core:get_hdr_field: to body
[sip:tes...@imsdemo.com
]
Oct 14 14:01:46 [4880] DBG:core:parse_via_param: found param type 232,
branch = z9hG4bK2559870494smg; state=6
Oct 14 14:01:46 [4880] DBG:core:parse_via_param: found param type 238,
transport = UDP; state=16
Oct 14 14:01:46 [4880] DBG:core:parse_via: end of header reached, state=5
Oct 14 14:01:46 [4880] DBG:core:parse_headers: via found, flags=2
Oct 14 14:01:46 [4880] DBG:core:parse_headers: this is the first via
Oct 14 14:01:46 [4880] DBG:core:receive_msg: After parse_msg...
Oct 14 14:01:46 [4880] DBG:core:receive_msg: preparing to run routing
scripts...
Oct 14 14:01:46 [4880] DBG:maxfwd:is_maxfwd_present: value = 70
Oct 14 14:01:46 [4880] DBG:core:parse_headers: flags=80
Oct 14 14:01:46 [4880] DBG:uri:has_totag: no totag
Oct 14 14:01:46 [4880] DBG:core:parse_headers: flags=78
Oct 14 14:01:46 [4880] DBG:tm:t_lookup_request: start searching: hash=30511,
isACK=0
Oct 14 14:01:46 [4880] DBG:tm:matching_3261: RFC3261 transaction matching
failed
Oct 14 14:01:46 [4880] DBG:tm:t_lookup_request: no transaction found
Oct 14 14:01:46 [4880] DBG:core:parse_headers: flags=200
Oct 14 14:01:46 [4880] DBG:core:get_hdr_field: content_length=0
Oct 14 14:01:46 [4880] DBG:core:get_hdr_field: found end of header
Oct 14 14:01:46 [4880] DBG:rr:find_first_route: No Route headers found
Oct 14 14:01:46 [4880] DBG:rr:loose_route: There is no Route HF
Oct 14 14:01:46 [4880] ERROR:rr:w_record_route: Double attempt to
record-route
Oct 14 14:01:46 [4880] DBG:core:grep_sock_info: checking if host==us: 11==12
[imsdemo.com] == [10.89.10.235]
Oct 14 14:01:46 [4880] DBG:core:grep_sock_info: checking if port 5060
matches port 5060
Oct 14 14:01:46 [4880] DBG:core:parse_headers: flags=
Oct 14 14:01:46 [4880] DBG:presence:search_event: start event= [presence]
Oct 14 14:01:46 [4880] DBG:presence:handle_publish: SIP-If-Match header not
found
Oct 14 14:01:46 [4880] DBG:presence:generate_ETag: etag=
a.1255496470.4880.1.0 / 21
Oct 14 14:01:46 [4880] DBG:presence:handle_publish: new etag =
a.1255496470.4880.1.0
Oct 14 14:01:46 [4880] DBG:presence:handle_publish: Expires header found,
value= 3600
Oct 14 14:01:46 [4880] ERROR:presence:handle_publish: No E-Tag and no body
found
Oct 14 14:01:46 [4880] DBG:core:parse_headers: flags=
Oct 14 14:01:46 [4880] DBG:core:check_via_address: params 10.254.140.195,
10.254.140.195, 0
Oct 14 14:01:46 [4880] DBG:sl:run_sl_callbacks: callback id 0 entered
Oct 14 14:01:46 [4880] DBG:tm:t_newtran: transaction on entrance=(nil)
Oct 14 14:01:46 [4880] DBG:core:parse_headers: flags=
Oct 14 14:01:46 [4880] DBG:core:parse_headers: flags=78
Oct 14 14:01:46 [4880] DBG:tm:t_lookup_request: start searching: hash=30511,
isACK=0
Oct 14 14:01:46 [4880] DBG:tm:matching_3261: RFC3261 transaction matching
failed
Oct 14 14:01:46 [4880] DBG:tm:t_lookup_request: no transaction found
Oct 14 14:01:46 [4880] DBG:tm:run_reqin_callbacks: trans=0xb3a09a30,
callback type 1, id 0 entered
Oct 14 14:01:46 [4880] DBG:tm:cleanup_uac_timers: RETR/FR timers reset
Oct 14 14:01:46 [4880] DBG:tm:insert_timer_unsafe: [2]: 0xb3a09a78 (41)
Oct 14 14:01:46 [4880] DBG:tm:t_unref: UNREF_UNSAFE: after is 0
Oct 14 14:01:46 [4880] DBG:core:destroy_avp_list: destroying list (nil)
Oct 14 14:01:46 [4880] DBG:core:receive_msg: cleaning up
Oct 14 14:01:51 [4885] DBG:tm:timer_routine: timer routine:2,tl=0xb3a09a78
next=(nil), timeout=41
Oct 14 14:01:51 [4885] DBG:tm:wait_handler: removing 0xb3a09a30 from table
Oct 14 14:01:51 [4885] DBG:tm:delete_cell: delete transaction 0xb3a09a30
Oct 14 14:01:51 [4885] DBG:tm:wait_handler: done
Oct 14 14:02:50 [4885] DBG:presence:update_db_subs: delete expired
Oct 14 14:02:50 [4885] DBG:db_mysql:db_mysql_do_prepared_query:
conn=0x8167ee8 (tail=135699504) MC=0x816ba48
Oct 14 14:02:50 [4885]
