[OpenSIPS-Users] prepaid with call plan

2011-01-06 Thread Dani Popa
Hi, I wonder, how can be implemented with opensips prepaid system with service plans with included minutes. Can someone to give me some hints ? Thanks, Dani Popa ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman

Re: [OpenSIPS-Users] prepaid with call plan

2011-01-06 Thread Dani Popa
, DB, RADIUS, etc) and allows the billing to trigger the call > termination from outside (like billing is keep computing costs and > when there is no more credit, it notifies opensips to terminate the > call) - again, you can use here the dialog module with the dlg_end_dlg > command via

Re: [OpenSIPS-Users] prepaid with call plan

2011-01-06 Thread Dani Popa
opensips.org/html/docs/modules/1.6.x/aaa_radius.html#id249105 > > Regards, > bogdan > > Dani Popa wrote: >> Hi Bogdan, >> >> I know that opensips care just about SIP part. But my question is if >> somebody tried to make this setup with call_control and opensips.

Re: [OpenSIPS-Users] prepaid with call plan

2011-01-06 Thread Dani Popa
ol application for this. See > http://callcontrol.ag-projects.com. > > Best Regards, > > Tijmen de Mes > AG Projects > > Op 1/6/11 12:21 PM, Dani Popa schreef: >> Hi, >> >> I wonder, how can be implemented with opensips prepaid system with >> service

Re: [OpenSIPS-Users] rtp and fake hangup

2011-01-06 Thread Dani Popa
hi, you have to use mediaproxy or rtpproxy for media timeout. Dani mancyb...@gmail.com wrote: > Hi All, > > in a scenario where opensips routes calls from sip user agents to voip > carriers: > can you please confirm that the only way to be sure to prevent fraud false > hangups > is to force th

Re: [OpenSIPS-Users] prepaid with call plan

2011-01-06 Thread Dani Popa
The auth is done at > INVITE time, before sending the call to termination, while the ACC start is > done at 200 OK INVITE, when the call is established. > > BTW, the acc module in opensips can automatically do RADIUS accounting (you > do not need to do it manually). > > >

Re: [OpenSIPS-Users] prepaid with call plan

2011-01-10 Thread Dani Popa
rticular requirements . > > Adrian > > On Jan 6, 2011, at 5:40 PM, Dani Popa wrote: > > >> Hi, >> >> I'm afraid i can not use call_control, because after DebitBalance or >> MaxSessiontime, I can not categorize the call as belonging to a call

Re: [OpenSIPS-Users] About domain name and IP

2011-01-10 Thread Dani Popa
Hi, I belive you want to use opensips without domain name. Check use_domain value . Dani Chris Liu wrote: Hi All, I am new to opensips. I'd like to let users be able to login the system by domain name and ip address. For eg. our opensips server's ip is 123.123.123.123 , the d

[OpenSIPS-Users] ERROR:aaa_radius:send_auth_func: radius authentication message failed with ERROR

2011-01-13 Thread Dani Popa
Hi, I get next error when i try to use aaa_radius Jan 13 13:01:44 [10886] ERROR:aaa_radius:send_auth_func: radius authentication message failed with ERROR The opensips start normally with no error. In script config i have: modparam("aaa_radius", "radius_config", "/etc/opensips_aaa_radius/radi

Re: [OpenSIPS-Users] Billing from OpenSIPS

2011-01-14 Thread Dani Popa
Hi, For particular billing system you can use aaa_radius great opensips module to integrate your billing system with wanted variables without querying databases direct from opensips . Dani Andrew Philp wrote: Hi All,   I am looking for a way to integrate calls going throug

Re: [OpenSIPS-Users] DND and presence

2011-01-14 Thread Dani Popa
Hi, I think you should make distinction between "call DND" and "presence DND". DND for not receiving calls you can do from ACL or user credentials. For presence, DND is indicated by UACs(also you cand use fifo to change presence status for subscribers). Of course you can check presence messages f

Re: [OpenSIPS-Users] Do I have a realm configuration problem

2011-01-14 Thread Dani Popa
Hi, If you use domain for auth , check if subscriber 1000 have domain vmopensips1.skycomuk.com. Dani Gareth Blades wrote: I have just setup opensips and have a phone with a couple of lines which are registered with opensips. I have a problem when trying to place a call between the two

Re: [OpenSIPS-Users] ERROR:aaa_radius:send_auth_func: radius authentication message failed with ERROR

2011-01-18 Thread Dani Popa
do you see in the system log any errors from the radius library ? > > Regards, > Bogdan > > Dani Popa wrote: >> Hi, >> I get next error when i try to use aaa_radius >> >> >> Jan 13 13:01:44 [10886] ERROR:aaa_radius:send_auth_func: radius >> authe

[OpenSIPS-Users] acc-status-type with aaa_radius

2011-01-19 Thread Dani Popa
Hi, I'm trying to use aaa_radius for accounting, but aaa_radius modules don't advertise Acct-Session-Type to radius. I miss something or aaa_radius dont use acc engine from opensips ? Thanks, Dani ___ Users mailing list Users@lists.opensips.org http:/

Re: [OpenSIPS-Users] Call limitation

2011-01-20 Thread Dani Popa
Hi, you can use usr_preferences avp_table and avp_load to load from db incoming/outgoing barring rules. in usr_preferences table you should have something like: | 139 | | username | domain | 4122 |2 | *barring_number* | 2011-01-11 12:19:31 | define avp_aliases i:4122 named as y

Re: [OpenSIPS-Users] acc-status-type with aaa_radius

2011-01-20 Thread Dani Popa
nd_acct() ) or > indirectly via the ACC module ? if you use it directly, you need to > manually put all RADIUS AVPs in the requests, including the > Acct-Session-Type (see the radius sets - > http://www.opensips.org/html/docs/modules/1.6.x/aaa_radius.html#id249105) > > Regards, >

Re: [OpenSIPS-Users] RADIUS interim update not work

2011-01-25 Thread Dani Popa
What do you use for accountig ? radius_send_acct or acc ? Dani Roberto Santini wrote: > Hi to all, > I am new here and I have a problem with accounting on RADIUS. > OpenSIPS sends start and stop accounting messages properly, but in case > of update sends messages with Acct-Status-Type = 0, which

Re: [OpenSIPS-Users] RADIUS interim update not work

2011-01-25 Thread Dani Popa
adius_send_acct. > > Roberto. > > On Tue, 2011-01-25 at 12:45 +0200, Dani Popa wrote: > >> What do you use for accountig ? radius_send_acct or acc ? >> >> Dani >> >> Roberto Santini wrote: >> >>> Hi to all, >>> I am new here and I ha

Re: [OpenSIPS-Users] changing SDP headers without rtpproxy installed

2011-01-27 Thread Dani Popa
Hi, No, it is not necessary, but if you don't use rtpproxy, the device behind NAT in some circumstances could not work. Dani Toyima Dias wrote: > Hello, > > Is it necessary to have a rtpproxy installed into opensips server to > be able to change/modify SDP headers in messages through the proxy

Re: [OpenSIPS-Users] Reseting the phones to register

2011-02-04 Thread Dani Popa
Hi, Depending how you use db_mode for usrloc: http://www.opensips.org/html/docs/modules/1.6.x/usrloc.html#id292952 If you user db_mode 0, then you have to restart the phone every time you restart the opensips. Dani On 02/04/11 11:40, Toyima Dias wrote: A dummy question... Every time i r

[OpenSIPS-Users] radius_send_auth timeout

2011-03-23 Thread Dani Popa
Hi all, How can i change timeout for radius_send_auth ? It is possible ? Dani ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Opensips sometime resend Accouting-request to freeradius

2011-03-24 Thread Dani Popa
You have a problem with opensips script. You set acct start flag many times and radius try each time to insert in mysql the same start query, but you can not because you have key defined on your table. Dani On 03/23/11 10:06, ha do wrote: Hi list i am test opensips 1.6.4 freeradius from ag-p

Re: [OpenSIPS-Users] Cannot store Accounting Record into mysql Using Opensips 1.6.4 + CDRTool + Freeradius + mysql

2011-03-24 Thread Dani Popa
you should enable sql in freeradius or use freeradius-xs(it have by default sql enabled) provided by ag-projects Dani On 03/24/11 13:33, Tijmen de Mes wrote: Hi, You can enable sqltrace in freeradius to see what is wrong and try to execute the queries manually. Which version of freeradius ar

[OpenSIPS-Users] SIP IM with exclusive User-Agent

2011-04-04 Thread Dani Popa
Hi, How can i send MESSAGES only to users who are registered with exclusiv User-Agent. I mean, A is registerd with User-Agent Jitsi and B is with jitsi and Cisco phone. If A try to send messages to B, i need that messages to be sent only to B on Jitsi, because cisco will replay with "Not Impe

Re: [OpenSIPS-Users] SIP IM with exclusive User-Agent

2011-04-04 Thread Dani Popa
thanks, Dani On 04/04/11 18:55, Bogdan-Andrei Iancu wrote: Hi Dani, On 04/04/2011 06:46 PM, Dani Popa wrote: Hi, How can i send MESSAGES only to users who are registered with exclusiv User-Agent. I mean, A is registerd with User-Agent Jitsi and B is with jitsi and Cisco phone. If A try to

[OpenSIPS-Users] mediaproxy KBIn and KBOut

2011-04-15 Thread Dani Popa
Hi, Mediaproxy radius request does not populate Kbin and Kbout. Also i tried to see sessions on port 25061 and also there callee_bytes and caller_bytes are 0. opensips:~# telnet localhost 25061 Trying 127.0.0.1... Connected to localhost. Escape character is '^]'. sessions [] sessions [{"from

Re: [OpenSIPS-Users] mediaproxy KBIn and KBOut

2011-04-18 Thread Dani Popa
e machine. I wondering if is a network card driver issue or kernel issue(if so, i'm dont know how to make troubleshooting, where should i see the callee_bytes and caller_bytes in kernel stats). Dani On 04/18/11 10:43, Saúl Ibarra Corretgé wrote: On 04/15/2011 02:42 PM, Dani

[OpenSIPS-Users] b2b_init_request('top hiding')

2011-04-20 Thread Dani Popa
Hi, I have a problem using b2b_init_request with "top hiding". When i receive 200 ok for invite, opensips crash with "ERROR:nat_traversal:__dialog_confirmed: FAKED reply - exit". In core dump this is where opensips crash: #0 get_source_uri (dlg=0xb2b4bc84, type=8, _params=0xb70b3c20) at na

Re: [OpenSIPS-Users] b2b_init_request('top hiding')

2011-04-20 Thread Dani Popa
, Apr 20, 2011 at 8:11 AM, Dani Popa wrote: Hi, I have a problem using b2b_init_request with "top hiding". When i receive 200 ok for invite, opensips crash with "ERROR:nat_traversal:__dialog_confirmed: FAKED reply - exit". In core dump this is where opensips crash: #0

Re: [OpenSIPS-Users] b2b_init_request('top hiding')

2011-04-21 Thread Dani Popa
at we also hit.. but still not the same. Can you please paste the output of 'bt'** <http://opensips.svn.sourceforge.net/viewvc/opensips/branches/1.6/modules/tm/uac.c?revision=7747&view=markup> in gdb? Regards, -- Anca Vamanu OpenSIPS Developer On 04/20/2011 03:11 PM, D

[OpenSIPS-Users] mediaproxy build fail: libiptc.c:63:8: error: redefinition of â

2011-04-21 Thread Dani Popa
I'm not able tu build mediaproxy on debian. Can someone give me a hint ? Thanks, Dani root@test:/home/work/mediaproxy-2.4.4# ./setup.py build running build running build_py creating build creating build/lib.linux-i686-2.6 creating build/lib.linux-i686-2.6/mediaproxy copying mediaproxy/__init__.

Re: [OpenSIPS-Users] mediaproxy build fail: libiptc.c:63:8: error: redefinition of â

2011-04-21 Thread Dani Popa
Hi, yes, i was able to install it and run it, but i have some issues. I dont have stream statistics: caller_bytes,callee_bytes,caller_packets and callee_packets. Also, if i'm not sure if media timeout is working, because i tried to simulate a hang call (in the middle of call, i restart my har

Re: [OpenSIPS-Users] mediaproxy build fail: libiptc.c:63:8: error: redefinition of â

2011-04-21 Thread Dani Popa
883c080e30a8b9c ]--- Thanks, Dani On 04/21/11 13:51, Saúl Ibarra Corretgé wrote: On 04/21/2011 12:44 PM, Dani Popa wrote: Hi, yes, i was able to install it and run it, but i have some issues. I dont have stream statistics: caller_bytes,callee_bytes,caller_packets and callee_packets. Also, if

Re: [OpenSIPS-Users] mediaproxy build fail: libiptc.c:63:8: error: redefinition of â

2011-04-21 Thread Dani Popa
On 04/21/11 14:13, Saúl Ibarra Corretgé wrote: On 04/21/2011 01:06 PM, Dani Popa wrote: sure, Apr 21 06:06:41 test media-relay[4903]: mediaproxy.mediacontrol.StreamListenerProtocol starting on 50012 Apr 21 06:06:41 test media-relay[4903]: mediaproxy.mediacontrol.StreamListenerProtocol

Re: [OpenSIPS-Users] mediaproxy build fail: libiptc.c:63:8: error: redefinition of â

2011-04-21 Thread Dani Popa
OK, Thanks, Dani On 04/21/11 15:14, Saúl Ibarra Corretgé wrote: I'm not talking abut binding ports for streams, i'm talking about stream packets and bytes info on telnet localhost 25060. I meant the statisticas that get printed in syslog after the call is closed. [{"from_tag": "4fc7812

Re: [OpenSIPS-Users] mediaproxy build fail: libiptc.c:63:8: error: redefinition of â

2011-05-02 Thread Dani Popa
Hi, Do you have any news with this issues ? Thanks, Dani On Thu, Apr 21, 2011 at 3:31 PM, Dani Popa wrote: > OK, > > Thanks, > Dani > > > On 04/21/11 15:14, Saúl Ibarra Corretgé wrote: > >> >> I'm not talking abut binding ports for streams, i

Re: [OpenSIPS-Users] mediaproxy build fail: libiptc.c:63:8: error: redefinition of â

2011-05-03 Thread Dani Popa
Ok, Thanks, Dani On Tue, May 3, 2011 at 10:00 AM, Saúl Ibarra Corretgé wrote: > On 05/02/2011 10:58 PM, Dani Popa wrote: > >> Hi, >> >> Do you have any news with this issues ? >> >> > Unfortunately not. I didn't have time to go and fix this yet,

[OpenSIPS-Users] opensips 1_6_X tls crash opensips

2011-05-18 Thread Dani Popa
root@test:/opensips_1_6# opensips -V version: opensips 1.6.4-2-tls (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, DISABLE_NAGLE, USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_

Re: [OpenSIPS-Users] mediaproxy build fail: libiptc.c:63:8: error: redefinition of â

2011-05-18 Thread Dani Popa
Hi, do you have news about this mediaproxy issues ? Thanks, Dani On 05/03/11 11:52, Dani Popa wrote: Ok, Thanks, Dani On Tue, May 3, 2011 at 10:00 AM, Saúl Ibarra Corretgé mailto:s...@ag-projects.com>> wrote: On 05/02/2011 10:58 PM, Dani Popa wrote: Hi, Do yo

Re: [OpenSIPS-Users] media-dispatcher and media relay connection problem

2011-05-26 Thread Dani Popa
Hi Liviu, What kernel do you have on running media-relay machine ? Thanks, Dani On 05/26/11 11:14, Barsan Liviu wrote: Hi, With the python-gnutls update to 1.2.1 the mediaproxy works fine. A suggestion: would be welcome a minimal install guide for Ubuntu/Debian, for example I spent several d

Re: [OpenSIPS-Users] OpenXCAP - failed to create OpenXCAP 2.0.0: Document is empty

2011-06-06 Thread Dani Popa
when start openxcap, it try to take schema from www.w3.org/2001/xml.xsd and www.w3.org doesn't responde. I changed schemaLocation in "/usr/local/pymodules/python2.6/xcap/appusage/xml-schemas/xcap-directory.xsd" and pointed to local file. Dani On 06/06/11 03:01, duane.lar...@gmail.com wrote

Re: [OpenSIPS-Users] OpenXCAP - failed to create OpenXCAP 2.0.0: Document is empty

2011-06-06 Thread Dani Popa
greate thanks, Dani On 06/06/11 16:18, Saúl Ibarra Corretgé wrote: Hi Dani, On Jun 6, 2011, at 3:07 PM, Dani Popa wrote: when start openxcap, it try to take schema from www.w3.org/2001/xml.xsd and www.w3.org doesn't responde. I changed schemaLocation in "/usr/local/pymodules

[OpenSIPS-Users] moh or in dialog media insertion with opensips as sip proxy

2011-06-16 Thread Dani Popa
Hi all, I looked on the internet for MOH with opensips as sip proxy(not b2b) and other media servers (sems,asterisk,etc). The answers on internet was that is not possible because SIP implementation and because sems,asterisk are full implemented sip servers(invite from opensips to media server

Re: [OpenSIPS-Users] moh or in dialog media insertion with opensips as sip proxy

2011-06-16 Thread Dani Popa
Hi, I thought so, but I needed confirmation. Thanks Adrian, Dani On 06/16/11 15:46, Adrian Georgescu wrote: You cannot do this reliably unless you insert a B2BUA in the call flow. Adrian On Jun 16, 2011, at 2:11 PM, Dani Popa wrote: Hi all, I looked on the internet for MOH with

[OpenSIPS-Users] 30x redirect for register

2011-06-20 Thread Dani Popa
Hi all, It is viable solution to use 30(1|2|5) redirect for REGISTER sip messages ? Thanks, Dani ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Send 487 request terminbated while a cancel recieved from UAC

2012-12-08 Thread Dani Popa
As far as I know, opensips send 487, when receiving 200ok, when forking On Dec 5, 2012 8:19 AM, "M.Khaled W Chehab" wrote: > > Dears , > > > > How to send a 487 request terminated and drop the call directly if the UA send a cancel ,since now I am sending 200 canceling to UA and send a cancel

[OpenSIPS-Users] add sdp on 180 Ringing and play file as ringback tone

2013-02-12 Thread Dani Popa
Hi, I wondering if it posiible to add sdp on 180 ringing in order to play some ringing tone. The ideea si that i want to play from rtpproxy with rtpproxy_stream2uac/rtpproxy_stream2uas some music as ringback tone to calling party if it's online. -- Dani

Re: [OpenSIPS-Users] add sdp on 180 Ringing and play file as ringback tone

2013-02-12 Thread Dani Popa
I just want to play media on replay route in case of 18[013] reply, so i'm sure the user was alerted if i got one of them, i'm pretty sure is not the case from the link below and also inserted media is not a fake ringback. Thanks anyway! Dani Popa On Feb 13, 2013, at 0:56, Daniel Go

Re: [OpenSIPS-Users] add sdp on 180 Ringing and play file as ringback tone

2013-02-12 Thread Dani Popa
you want to do is a 183 with early media, > not just append an SDP to a 180. > > Good luck though:) > > > > -dg > > > On Tue, Feb 12, 2013 at 3:28 PM, Dani Popa wrote: > >> I just want to play media on replay route in case of 18[013] reply, so >> i'

[OpenSIPS-Users] msilo on failure_route

2013-02-15 Thread Dani Popa
log("MSILO: offline message NOT stored\n"); t_reply("503", "Service Unavailable"); }; } -- Dani Popa ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] msilo on failure_route

2013-02-15 Thread Dani Popa
Five SIP clients with the same username. Dani On Fri, Feb 15, 2013 at 12:19 PM, Dani Popa wrote: > Hi, > > Regarding msilo module and example from the documentation, one simple > question: > > if i have 5 clients already registered and non of them know IM(message sip

Re: [OpenSIPS-Users] msilo on failure_route

2013-02-15 Thread Dani Popa
d failure route is triggered when the transaction > fails). > > So the final answer -> one time. > > Regards > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > > On 02/15/2013 12:22 PM, Dani Popa wrote: > > Five SIP clients

Re: [OpenSIPS-Users] msrp relay

2013-02-18 Thread Dani Popa
104. > > > > > > > > > > From: Saúl Ibarra Corretgé > > To: OpenSIPS users mailling list > > Sent: Monday, February 18, 2013 3:10 PM > > Subject: Re: [OpenSIPS-Users] msrp relay > > > > > > On Feb 18, 2013, at 5:03 AM, nguyen khue wrote: > > > > > Hi all, > > > > > > How I can integrates msrprelay (msrprelay.org) with opensips to make > File Transfer session between SIP end-points located behind NAT?. Please > guide me. > > > I tested file transfer between two SIP end-points in LAN and it worked > successful. > > > > > > > What problems did you ran into? Did you follow the installation guide > http://msrprelay.org/projects/msrprelay/wiki/InstallationGuide ? > > > > Regards, > > > > -- > > 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 > > -- > Saúl Ibarra Corretgé > AG Projects > > > > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Dani Popa ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] msrp relay

2013-02-19 Thread Dani Popa
eb 18, 2013 at 4:50 PM, Saúl Ibarra Corretgé wrote: > > On Feb 18, 2013, at 2:26 PM, Dani Popa wrote: > > > Hi, > > > > I think it's more helpful if you can give us calltrace in case of using > msrp, sipproxy and of course 2 sip clients. Msrprelay it's a

Re: [OpenSIPS-Users] Opensips 1.9 - Radius accounting

2013-04-09 Thread Dani Popa
mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Dani Popa ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Opensips 1.9 - Radius accounting

2013-04-09 Thread Dani Popa
tp7585735.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] Too many RFCs ????

2013-04-29 Thread Dani Popa
__**_ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-**bin/mailman/listinfo/users<http://lists.opensips.org/cgi-bin/mailman/listinfo/users> > -- Dani Popa ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

[OpenSIPS-Users] rtpproxy

2013-05-24 Thread Dani Popa
ckets, LatePackets, LostPackets. I know, some of you will recomand mediaproxy and it's not good for me, because i chosed to use rtpproxy because, i can insert and record media in curent stream. So the question is: there is any way to have such information at the end of call? Thanks, --

[OpenSIPS-Users] (no subject)

2013-05-24 Thread Dani Popa
(B) (A)trying <-opensips <-trying(B) (A)ringing <-opensips <-ringing(B) (A)progress <-opensips (A)200ok <-opensips <-200OK(B) (A) ACK ->opensips ->ACK(B) .... -- Dani Popa ___

[OpenSIPS-Users] insert 183 Progress with SDP in call dialog

2013-05-24 Thread Dani Popa
(B) (A)trying <-opensips <-trying(B) (A)ringing <-opensips <-ringing(B) (A)progress <-opensips (A)200ok <-opensips <-200OK(B) (A) ACK ->opensips ->ACK(B) -- Dani Popa ___ Users mailing l

[OpenSIPS-Users] media stream extra info after hangup

2013-05-24 Thread Dani Popa
ckets, LatePackets, LostPackets. I know, some of you will recomand mediaproxy and it's not good for me, because i chosed to use rtpproxy because, i can insert and record media in curent stream. So the question is: there is any way to have such information at the end of call? Thanks, --

[OpenSIPS-Users] radius acc in local_route on dialog timeout

2013-06-18 Thread Dani Popa
cc_db_request("200 Dialog Timeout", "acc"); } } Thanks, -- Dani Popa ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] radius acc in local_route on dialog timeout

2013-06-20 Thread Dani Popa
any ideea ? On Tue, Jun 18, 2013 at 7:10 PM, Dani Popa wrote: > Hi all, > I use acc with radius and when i set accountig flag in local_route i > dont receive any accountig request on radius server. As I see local_route > was hit twice on "dialog timeout" and i dont u

Re: [OpenSIPS-Users] radius acc in local_route on dialog timeout

2013-06-24 Thread Dani Popa
ry file in one or more places and opensips does not show it to you > unless you have debug on. > > Regards, > Qasim > > > On Thu, Jun 20, 2013 at 11:25 PM, Dani Popa wrote: > >> any ideea ? >> >> >> On Tue, Jun 18, 2013 at 7:10 PM, Dani Popa

Re: [OpenSIPS-Users] radius acc in local_route on dialog timeout

2013-06-24 Thread Dani Popa
). > I suggest you to use the manual accounting in this case. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > > On 06/18/2013 07:10 PM, Dani Popa wrote: > > Hi all, > I use acc with radius and when i s

Re: [OpenSIPS-Users] Opensips + asterisk 1.4

2013-07-17 Thread Dani Popa
>>> Willian Mazzardo >>> Depto TI - SYSSVOIP >>> www.syssvoip.com.br >>> 55 3537 2030 >>> >>> ___ >>> 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 > > -- Dani Popa ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Opensips + asterisk 1.4

2013-07-17 Thread Dani Popa
azzardo > Depto TI - SYSSVOIP > www.syssvoip.com.br > 55 3537 2030 > > > 2013/7/17 Dani Popa > >> set opensips peer to insecure=port,invite >> >> >> On Wed, Jul 17, 2013 at 1:12 PM, Willian Mazzardo - SYSSVOIP < >> will...@syssvoip.com.br&

Re: [OpenSIPS-Users] Opensips + asterisk 1.4

2013-07-17 Thread Dani Popa
xten => _X.,1,DeadAGI(a2billing.php) > > > Willian Mazzardo > Depto TI - SYSSVOIP > www.syssvoip.com.br > 55 3537 2030 > > > 2013/7/17 Dani Popa > >> what contex hit invite from opensips ? >> >> >> On Wed, Jul 17, 2013 at 1:24 PM, Willian Mazz

[OpenSIPS-Users] handle on reply for keepalive OPTIONS sip packet

2013-09-13 Thread Dani Popa
There is any way to handle replay for sip keepalive OPTIONS packet when using nathelper module ? Thanks, -- Dani Popa ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] handle on reply for keepalive OPTIONS sip packet

2013-09-13 Thread Dani Popa
gards, > > Aamir Chougule > Cell: 08097989101 > Skype-ID: aamir_ryu > > --- Sent from my BlackBerry --- > > -Original Message- > From: Dani Popa > Sender: users-boun...@lists.opensips.org > Date: Fri, 13 Sep 2013 13:12:51 > To: OpenSIPS users mail

[OpenSIPS-Users] Invite with Replaces header

2013-11-04 Thread Dani Popa
Hi all, There is any way to check if Opensips instance have dialog in any state defined by Replaces Header of new incoming call ? -- Dani Popa ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] MediaProxy scalability FAQ updated

2014-03-05 Thread Dani Popa
___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -- Dani Popa ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] MediaProxy scalability FAQ updated

2014-03-06 Thread Dani Popa
____ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -- Dani Popa ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] 30x redirect for register

2011-06-21 Thread Dani Popa
r REGISTER.Probably you need to explicitly test with the UACs you want use. Regards, Bogdan On 06/20/2011 07:08 PM, Dani Popa wrote: Hi all, It is viable solution to use 30(1|2|5) redirect for REGISTER sip messages ? Thanks, Dani ___ Users ma

Re: [OpenSIPS-Users] one way media stream

2011-06-29 Thread Dani Popa
Hi, As far as i know it's hard to insert media from other sources in proxy mode for situation like call hold or in call media insert. If you find a solution, please let me know. Dani On 06/29/11 10:06, Barsan Liviu wrote: Hi, Yes, exactly. And obviously for this we want just one way stream

Re: [OpenSIPS-Users] How to check active calls

2011-07-11 Thread Dani Popa
> ##} > ## > ##if (!db_check_to()) > ##{ > ##sl_send_reply("403","Forbidden auth ID"); > ##exit; > ##} > > if (!save("location")) > sl_reply_error(); > >

Re: [OpenSIPS-Users] Problem with siptrace module

2011-07-12 Thread Dani Popa
ip:proxy@192.1.8.2.154")modparam("dispatcher", "ds_ping_interval", > 30)modparam("dispatcher", "ds_probing_threshhold", 2)modparam("dispatcher", > "ds_probing_mode", 1)modparam("dispatcher", "list_file", > "/usr/local/etc/opensips/dispatcher.list")# modparam("dispatcher", > "force_dst", 1) modparam("siptrace", "db_url", > "mysql://root:Viamonte1621@localhost/opensips")modparam("siptrace", > "enable_ack_trace", 1)modparam("siptrace", "trace_on", > 1)modparam("siptrace", "table", "sip_trace")modparam("siptrace", > "trace_flag", 22) #modparam("acc", "log_level", 1)#modparam("acc", > "log_flag", 1)#modparam("acc", "db_url", > "mysql://root:Viamonte1621@localhost/opensips") route{ > setflag(22);setbflag(22);sip_trace();if ( > !mf_process_maxfwd_header("10") ){ > sl_send_reply("483","To Many Hops"); drop();}; > if (is_method("OPTIONS")) {options_reply(); > exit;}ds_select_dst("1", "0");if > ($retcodexlog("[Redmond] Service full\n"); > sl_send_reply("500","Service full");exit;} > forward();#t_relay();#sip_trace();} failure_route[1] > {if (t_check_status("(408)|(5[0-9][0-9])")) { > ds_mark_dst();if (ds_select_dst("1", "0")) > {forward();} else > { t_reply("503", "Service > Unavailable");}}} onreply_route { > setflag(22);setflag(22);sip_trace();} ThanksDiego > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -- Dani Popa ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] How to limit calls to specific number

2011-07-13 Thread Dani Popa
HI, first aaa_radius_auth and specific sql procedure in sql server. the second asterisk/freeswitch load balncing Dani On 07/12/11 17:06, duane.lar...@gmail.com wrote: For your first question would this work? http://www.ag-projects.com/projects-products-96/535-call-control For your second q

[OpenSIPS-Users] Not enough free memory, will atempt defragmenation

2011-08-04 Thread Dani Popa
Hi, How can i solve this kind of problems ? Opensips doesn't crash, but it not respond to any sip requests. Aug 3 07:36:48 test /usr/local/sbin/opensips[29094]: WARNING:core:fm_malloc: Not enough free memory, will atempt defragmenation Aug 3 07:36:48 test /usr/local/sbin/opensips[29094]:

[OpenSIPS-Users] all sip body headers regarding video removed

2011-08-04 Thread Dani Popa
Hi all, How can i remove all sip video body headers regardin video. Should i remove any line from body after "m=video", or how. Please give me a hint, if you have. Thanks, Dani ___ Users mailing list Users@lists.opensips.org http://lists.opensips.o

Re: [OpenSIPS-Users] all sip body headers regarding video removed

2011-08-04 Thread Dani Popa
wrote: Hi Dani, Why would you do that? If you don't want to allow video, you can simply replace the video port in the "m=" line with 0. Regards, Razvan Crainea OpenSIPS Developer On 04.08.2011 16:58, Dani Popa wrote: Hi all, How can i remove all sip video body headers regard

Re: [OpenSIPS-Users] Not enough free memory, will atempt defragmenation

2011-08-04 Thread Dani Popa
, Razvan Crainea wrote: Hi Dani, It seems you are out of memory. What version of OpenSIPS are you using? Regards, Razvan Crainea OpenSIPS Developer On 04.08.2011 16:07, Dani Popa wrote: Hi, How can i solve this kind of problems ? Opensips doesn't crash, but it not respond to any sip req

Re: [OpenSIPS-Users] Not enough free memory, will atempt defragmenation

2011-08-04 Thread Dani Popa
Ok, thanks for quick response. Dani On 08/04/11 18:26, Vlad Paiu wrote: Hello, Is it possible that you upgrade to 1.7 ? It is possible that this issue was fixed in the latest OpenSIPS version. If not, go to Makefile.defs, uncomment the line with -DDBG_QM_MALLOC \ and comment the line w

Re: [OpenSIPS-Users] all sip body headers regarding video removed

2011-08-04 Thread Dani Popa
0. [1] http://www.opensips.org/html/docs/modules/devel/textops.html#id293910 Regards, Razvan Crainea OpenSIPS Developer On 04.08.2011 18:03, Dani Popa wrote: Hi, In fact, i have some problems with one of my pstn gw's that send "400 Incorrect content length", i think, because

[OpenSIPS-Users] subscribe non case sensitive user from sip uri

2011-08-04 Thread Dani Popa
Hi, it is somehow that username from sip uri to be non case sensitive when we talk about presence and xcap storage? I mean, if userA add userB, in his contact list, i need userA to be able to add userB even he add him(type) as USERB. Dani ___ Us

Re: [OpenSIPS-Users] subscribe non case sensitive user from sip uri

2011-08-05 Thread Dani Popa
Hi, Ok, but also, registrar module support non "case sensitive" sip username. -- Dani Popa On 8/5/11 11:40 AM, Vlad Paiu wrote: Hello, What you're asking for is against the RFC 3261 URI comparison rules, which states that comparison of the userinfo part of the URI shoul

[OpenSIPS-Users] msilo and server_header

2011-08-16 Thread Dani Popa
Hi, When using m_store("$ru") the SIP messages sent back to sender have default server_header and not the one i rewrite it. Dani ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] subscribe non case sensitive user from sip uri

2011-08-18 Thread Dani Popa
RFC compliant. Regards, Bogdan On 08/05/2011 07:30 PM, Dani Popa wrote: Hi, Ok, but also, registrar module support non "case sensitive" sip username. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-b

Re: [OpenSIPS-Users] Not enough free memory, will atempt defragmenation

2011-08-19 Thread Dani Popa
Hi, Where should i find memory dump ? I have something in logs about memory. I'll attach an file. Please let me know if this is what you need. I also increased PKG_MEM_POOL_SIZE = 8 *1024 * 1024, and shared mem to 256, and also updated opensips 1.6.4 to latest svn revision, i think. root@

Re: [OpenSIPS-Users] Not enough free memory, will atempt defragmenation

2011-08-19 Thread Dani Popa
Hi again, i also saw that i compiled opensips with libxmlrpc-c3-dev and libxmlrpc-c3 and i was warned somewhere that i'll compile it on my own risk. Now i removed libxmlrpc-c3-dev and libxmlrpc-c3 and i compiled with libxmlrpc-c++4-dev without warnings. Let's see what we will get! Thanks,

Re: [OpenSIPS-Users] Not enough free memory, will atempt defragmenation

2011-08-19 Thread Dani Popa
Hi, True, i changed wrong the Makefiles.defs. I dont know if you need this: if i change Makefile.defs as: DEFS+= $(extra_defs) \ . . . . -DCHANGEABLE_DEBUG_LEVEL \ #-DF_MALLOC \ -DDBG_QM_MALLOC \ #-DDBG_F_MALLOC \ opensips will not be compiled with -DDBG_QM

Re: [OpenSIPS-Users] Not enough free memory, will atempt defragmenation

2011-08-19 Thread Dani Popa
get only the memory dump without all runtime logs from mem debugger. Regards, Bogdan On 08/19/2011 01:13 PM, Dani Popa wrote: Hi, True, i changed wrong the Makefiles.defs. I dont know if you need this: if i change Makefile.defs as: DEFS+= $(extra_defs

Re: [OpenSIPS-Users] Not enough free memory, will atempt defragmenation

2011-08-19 Thread Dani Popa
to get only the memory dump without all runtime logs from mem debugger. Regards, Bogdan On 08/19/2011 01:13 PM, Dani Popa wrote: Hi, True, i changed wrong the Makefiles.defs. I dont know if you need this: if i change Makefile.defs as: DEFS+= $(extra_defs

Re: [OpenSIPS-Users] Not enough free memory, will atempt defragmenation

2011-08-19 Thread Dani Popa
the memory dump without all runtime logs from mem debugger. Regards, Bogdan On 08/19/2011 01:13 PM, Dani Popa wrote: Hi, True, i changed wrong the Makefiles.defs. I dont know if you need this: if i change Makefile.defs as: DEFS+= $(extra_defs) \ . . . . -DCHANGEABLE_DEBUG_LEVEL

Re: [OpenSIPS-Users] Not enough free memory, will atempt defragmenation

2011-08-19 Thread Dani Popa
in the DBG support, set: memlog=6 memdump=1 in order to get only the memory dump without all runtime logs from mem debugger. Regards, Bogdan On 08/19/2011 01:13 PM, Dani Popa wrote: Hi, True, i changed wrong the Makefiles.defs. I dont know if you need this: if i change Makefile.defs

Re: [OpenSIPS-Users] Ability to tell active calls per customer

2011-08-22 Thread Dani Popa
Hi, I think you could use dialog profile, but not sure. Dani On 08/19/11 23:17, Robert Thomas wrote: Hi, I have a load balancer module to distribute calls among my Gateways. I can use the lb_list command to see the active calls per gw, but I would like something similar to graph my customer

[OpenSIPS-Users] 412 Conditional Request Failed

2011-08-23 Thread Dani Popa
Hi, should "412 Conditional Request Failed" for PUBLISH (if SIP-If-Match doesn't match) work with opensips 1.6.4 ? Thanks, Dani ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Not enough free memory, will atempt defragmenation

2011-08-23 Thread Dani Popa
compiled on 05:48:37 Aug 19 2011 with gcc 4.5.2 Thanks, Dani Popa On 08/19/11 17:31, Bogdan-Andrei Iancu wrote: Hi Dani, You can not have comments in multi-line assignments So, instead of DEFS+= $(extra_defs) \ . -DCHANGEABLE_DEBUG_LEVEL \ #-DF_MALLOC

Re: [OpenSIPS-Users] msilo and server_header

2011-08-23 Thread Dani Popa
Thanks, Dani On 08/19/11 18:01, Bogdan-Andrei Iancu wrote: Hi Dani, In your case opensips will act as UAC (not server), so you need to define your custom user_agent_header: http://www.opensips.org/Resources/DocsCoreFcn17#toc96 Regards, Bogdan On 08/16/2011 03:12 PM, Dani Popa wrote

Re: [OpenSIPS-Users] Not enough free memory, will atempt defragmenation

2011-08-24 Thread Dani Popa
, set: memlog=6 memdump=1 in order to get only the memory dump without all runtime logs from mem debugger. Regards, Bogdan On 08/19/2011 01:13 PM, Dani Popa wrote: Hi, True, i changed wrong the Makefiles.defs. I dont know if you need this: if i change Makefile.defs as: DEFS

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