- sr-users - kamailio.org

[OpenSER-Users] In 1.3 how to use set_rtp_proxy_set
by Robert Dyck 23 Nov '07

23 Nov '07

If I have only one rtp proxy should I use set_rtp_proxy_set("0")? I think openser should be smart enough to figure it out if there is only one rtp proxy. It looks like there is a cut and paste error in the 1.3 nathelper wiki. 1.6.3. set_rtp_proxy_set() Sets the Id of the rtpproxy set to be used for the next [un]force_rtp_proxy() command. This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE, BRANCH_ROUTE. Example 1-17. fix_nated_contact usage ... set_rtp_proxy_set("2"); force_rtp_proxy(); Rob ...

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Re: [Serusers] [Serdev] [Tracker] Created: (SER-332) tcp branching error
by Tomasz Zieleniewski 23 Nov '07

23 Nov '07

once again because to big attachments:) On Nov 23, 2007 11:48 AM, Tomasz Zieleniewski <tzieleniewski(a)gmail.com> wrote: > sorry I forgot to place the attachments > > > On Nov 23, 2007 11:47 AM, Tomasz Zieleniewski <tzieleniewski(a)gmail.com> wrote: > > Hi, > > > > Some more debug information: > > > > User contacts: > > ser@sen:~/ser/ser-2.0/ssp$ ./serctl/ser_ctl usrloc show touk(a)voip.rd.touk.pl > > sip:touk@192.168.0.164:1338;rinstance=191d670c84591ea3;transport=TCP 3330 -1.0 > > sip:touk@192.168.0.164:1348;rinstance=e51fd20db73fc732;transport=TCP 3405 -1.0 > > > > Contents of the location table: > > mysql> select * from location where uid like 'touk@%'; > > +----------------------+--------------------------+----------------------------------------------------------------------+----------+---------------------+----+----------------------------------------------+------+-------+----------------------------------+----------+ > > | uid | aor | contact > > | received | expires > > | q | callid | cseq > > | flags | user_agent | instance | > > +----------------------+--------------------------+----------------------------------------------------------------------+----------+---------------------+----+----------------------------------------------+------+-------+----------------------------------+----------+ > > | touk(a)voip.rd.touk.pl | sip:touk@voip.rd.touk.pl | > > sip:touk@192.168.0.164:1338;rinstance=191d670c84591ea3;transport=TCP | > > NULL | 2007-11-23 11:17:03 | -1 | > > MTQ1ZWZmZWE2MWIxZjViYmVkNzZlNmU0YTEwNjRmMTA. | 4 | 0 | X-Lite > > release 1011s stamp 41150 | NULL | > > | touk(a)voip.rd.touk.pl | sip:touk@voip.rd.touk.pl | > > sip:touk@192.168.0.164:1348;rinstance=e51fd20db73fc732;transport=TCP | > > NULL | 2007-11-23 11:18:18 | -1 | > > NDk0ZDA3ZmU5NmFlZmYyMjA5Mjc2ZTZiYTFmMTMxODY. | 4 | 0 | X-Lite > > release 1011s stamp 41150 | NULL | > > +----------------------+--------------------------+----------------------------------------------------------------------+----------+---------------------+----+----------------------------------------------+------+-------+----------------------------------+----------+ > > 2 rows in set (0.00 sec) > > > > > > Here is what does the internal proxy (192.168.0.165:7160) which is > > registrar and perform location service, > > the destination set contains both contacts and the first contact used > > is the one which is unavailable. > > this proxy sends first invite with this contact to boundary sip proxy > > Nov 23 11:20:27 sen ser[28434]: route[FORWARD]: xlset_attr(bsp_route, > > <$global_addr_bsp;lr=on>) > > Nov 23 11:20:27 sen ser[28434]: route[FORWARD]: attr_destination($bsp_route) > > Nov 23 11:20:27 sen ser[28434]: route[FORWARD]: ($fr_inv_timer) > > Nov 23 11:20:27 sen ser[28434]: route[FORWARD]: t_on_reply(ONREPLY_ROUTE) > > Nov 23 11:20:27 sen ser[28434]: route[FORWARD]: t_on_failure(FAILURE_ROUTE) > > Nov 23 11:20:27 sen ser[28434]: route[FORWARD]: t_on_branch(BRANCH_ROUTE) > > Nov 23 11:20:27 sen ser[28434]: route[FORWARD]: destination set: > > Contact: sip:touk@192.168.0.164:1338;rinstance=191d670c84591ea3;transport=TCP, > > sip:touk@192.168.0.164:1348;rinstance=e51fd20db73fc732;transpo > > Nov 23 11:20:27 sen ser[28434]: route[FORWARD]: next route: <null> > > Nov 23 11:20:27 sen ser[28434]: route[FORWARD]: message's next hop: > > sip:voip.rd.touk.pl:7060;lr=on > > Nov 23 11:20:27 sen ser[28434]: route[FORWARD]: (!t_relay()) > > Nov 23 11:20:27 sen ser[28434]: branch_route[BRANCH_ROUTE]: <INVITE> > > ruri:<sip:touk@192.168.0.164:1338;rinstance=191d670c84591ea3;transport=TCP> > > t:<sip:touk@voip.rd.touk.pl> f:<sip:hellboy@voip.rd.touk.pl> c > > Nov 23 11:20:27 sen ser[28434]: branch_route[BRANCH_ROUTE]: > > append_hf(P-Warn: forked branch) > > Nov 23 11:20:27 sen ser[28434]: branch_route[BRANCH_ROUTE]: <INVITE> > > ruri:<sip:touk@192.168.0.164:1348;rinstance=e51fd20db73fc732;transport=TCP> > > t:<sip:touk@voip.rd.touk.pl> f:<sip:hellboy@voip.rd.touk.pl> c > > Nov 23 11:20:27 sen ser[28434]: branch_route[BRANCH_ROUTE]: > > append_hf(P-Warn: forked branch) > > Nov 23 11:20:27 sen ser[28434]: ERROR: tcp_blocking_connect: poll > > error: flags 18 > > Nov 23 11:20:27 sen ser[28434]: ERROR: tcp_blocking_connect: SO_ERROR > > (111) Connection refused > > Nov 23 11:20:27 sen ser[28434]: ERROR: tcpconn_connect: > > tcp_blocking_connect failed > > Nov 23 11:20:27 sen ser[28434]: ERROR: tcp_send: connect failed > > Nov 23 11:20:27 sen ser[28434]: msg_send: ERROR: tcp_send failed > > Nov 23 11:20:27 sen ser[28434]: ERROR: t_send_branch: sending request > > on branch 1 failed > > Nov 23 11:20:27 sen ser[28434]: route[FORWARD]: drop > > > > here is what happens in boundary sip proxy: > > is receives the invite tries to do the t_rqlay but there is the tcp error. > > t_relay() fails and it sends the sl_reply_error to internal sip proxy: > > Nov 23 11:31:38 sen ser[31861]: route[FORWARD]: destination set: <null> > > Nov 23 11:31:38 sen ser[31861]: route[FORWARD]: next route: <null> > > Nov 23 11:31:38 sen ser[31861]: route[FORWARD]: message's next hop: > > sip:touk@192.168.0.164:1338;rinstance=191d670c84591ea3;transport=TCP > > Nov 23 11:31:38 sen ser[31861]: route[FORWARD]: (!t_relay()) > > Nov 23 11:31:38 sen ser[31861]: ERROR: tcp_blocking_connect: poll > > error: flags 18 > > Nov 23 11:31:38 sen ser[31861]: ERROR: tcp_blocking_connect: SO_ERROR > > (111) Connection refused > > Nov 23 11:31:38 sen ser[31861]: ERROR: tcpconn_connect: > > tcp_blocking_connect failed > > Nov 23 11:31:38 sen ser[31861]: ERROR: tcp_send: connect failed > > Nov 23 11:31:38 sen ser[31861]: msg_send: ERROR: tcp_send failed > > Nov 23 11:31:38 sen ser[31861]: ERROR: t_send_branch: sending request > > on branch 0 failed > > Nov 23 11:31:38 sen ser[31861]: route[FORWARD]: (!t_relay()) -> TRUE > > Nov 23 11:31:38 sen ser[31861]: route[FORWARD]: sl_reply_error() > > Nov 23 11:31:38 sen ser[31861]: ERROR: sl_reply_error used: I'm > > terribly sorry, server error occurred (1/SL) > > > > in the attachment there are ngrep and wireshark log from the same situation > > > > if you need more info please let me know > > waiting for your feedback > > > > Cheers > > tomasz > > > > > > > > On Nov 22, 2007 8:40 PM, Andrei Pelinescu-Onciul <andrei(a)iptel.org> wrote: > > > On Nov 16, 2007 at 14:43, Tomasz Zieleniewski (JIRA) <tracker(a)iptel.org> wrote: > > > > tcp branching error > > > > ------------------- > > > > > > > > Key: SER-332 > > > > URL: http://tracker.iptel.org/browse/SER-332 > > > > Project: SER > > > > Issue Type: Bug > > > > Components: core > > > > Affects Versions: 2.0 > > > > Environment: Linux sen 2.6.18-5-amd64 #1 SMP Tue Oct 2 20:37:02 UTC 2007 x86_64 GNU/Linux > > > > Reporter: Tomasz Zieleniewski > > > > Priority: Critical > > > > > > > > > > > > Hi, > > > > > > > > I have the following situation. > > > > I have two ser rel_2_0_0 instances one acting as a boundary proxy the > > > > other on as internal proxy (registrar, location). > > > > > > > > I have the following situation I have a user which has registered two > > > > contacs one woth tcp transport the other one with udp. > > > > > > > > All messages coming to and outgoing from my network travels through > > > > boundary sip proxy. > > > > > > > > When internal ser perform lookup_location it get to contacts and the > > > > destination set contains two records: > > > > Nov 15 11:03:13 sen ser[3673]: route[FORWARD]: destination set: > > > > Contact: sip:xlite@192.168.0.164:1221;rinstance=d35f9cda6bd79b0a;transport=TCP, > > > > sip:xlite@192.168.0.164:7060 > > > > > > > > Then internal ser forward message to boundary SIP proxy by changing > > > > the destination of the message and with the usage of t_relay(). > > > > I see from log that there are two branches involved: > > > > Nov 15 11:03:13 sen ser[3673]: branch_route[BRANCH_ROUTE]: <INVITE> > > > > ruri:<sip:xlite@192.168.0.164:1221;rinstance=d35f9cda6bd79b0a;transport=TCP> > > > > t:<sip:00@voip.rd.touk.pl> f:<sip:lkl@voip.rd.touk.pl> > > > > Nov 15 11:03:13 sen ser[3673]: branch_route[BRANCH_ROUTE]: > > > > append_hf(P-Warn: forked branch) > > > > Nov 15 11:03:13 sen ser[3673]: branch_route[BRANCH_ROUTE]: <INVITE> > > > > ruri:<sip:xlite@192.168.0.164:7060;rinstance=cfca8523e3747cb0> > > > > t:<sip:00@voip.rd.touk.pl> f:<sip:lkl@voip.rd.touk.pl> > > > > Nov 15 11:03:13 sen ser[3673]: branch_route[BRANCH_ROUTE]: > > > > append_hf(P-Warn: forked branch) > > > > > > > > both INVITES reaches boudary SIP proxy. > > > > > > > > Because first contact was unreachable there was an tcp error: > > > > ser[3516]: ERROR: tcp_blocking_connect: poll error: flags 18 > > > > Nov 15 11:03:13 sen ser[3516]: ERROR: tcp_blocking_connect: SO_ERROR > > > > (111) Connection refused > > > > Nov 15 11:03:13 sen ser[3516]: ERROR: tcpconn_connect: > > > > tcp_blocking_connect failed > > > > Nov 15 11:03:13 sen ser[3516]: ERROR: tcp_send: connect failed > > > > Nov 15 11:03:13 sen ser[3516]: msg_send: ERROR: tcp_send failed > > > > Nov 15 11:03:13 sen ser[3516]: ERROR: t_send_branch: sending request > > > > on branch 0 failed > > > > > > > > after which ser invokes sl_reply_error() which results in 477 next hop > > > > error reponse message sent to internal ser instance. > > > > The problem arises when boudary sip proxy processes second INVITE > > > > which correspond to second destination. > > > > when boundary SIP proxy receives it doesn't send it to user agent but > > > > instead reponses to internal ser with > > > > 500 I'm terribly sorry, server error occurred (1/SL) > > > > the only thing I see in the boundary SIP proxy log is: > > > > ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) > > > > > > > > Did I miss something here or this is an error? > > > > Waiting for Your feedback > > > > > > > > > > Could you make some traffic dump between the 2 proxies and send it to me > > > (andrei(a)iptel.org)? > > > It's sounds quite strange (it should work without any problems) and a > > > look at the packets might help. > > > > > > > > > Thanks, > > > Andrei > > > _______________________________________________ > > > Serdev mailing list > > > Serdev(a)lists.iptel.org > > > http://lists.iptel.org/mailman/listinfo/serdev > > > > > >

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Re: [Serusers] [Serdev] [Tracker] Created: (SER-332) tcp branching error
by Tomasz Zieleniewski 23 Nov '07

23 Nov '07

Hi, Some more debug information: User contacts: ser@sen:~/ser/ser-2.0/ssp$ ./serctl/ser_ctl usrloc show touk(a)voip.rd.touk.pl sip:touk@192.168.0.164:1338;rinstance=191d670c84591ea3;transport=TCP 3330 -1.0 sip:touk@192.168.0.164:1348;rinstance=e51fd20db73fc732;transport=TCP 3405 -1.0 Contents of the location table: mysql> select * from location where uid like 'touk@%'; +----------------------+--------------------------+----------------------------------------------------------------------+----------+---------------------+----+----------------------------------------------+------+-------+----------------------------------+----------+ | uid | aor | contact | received | expires | q | callid | cseq | flags | user_agent | instance | +----------------------+--------------------------+----------------------------------------------------------------------+----------+---------------------+----+----------------------------------------------+------+-------+----------------------------------+----------+ | touk(a)voip.rd.touk.pl | sip:touk@voip.rd.touk.pl | sip:touk@192.168.0.164:1338;rinstance=191d670c84591ea3;transport=TCP | NULL | 2007-11-23 11:17:03 | -1 | MTQ1ZWZmZWE2MWIxZjViYmVkNzZlNmU0YTEwNjRmMTA. | 4 | 0 | X-Lite release 1011s stamp 41150 | NULL | | touk(a)voip.rd.touk.pl | sip:touk@voip.rd.touk.pl | sip:touk@192.168.0.164:1348;rinstance=e51fd20db73fc732;transport=TCP | NULL | 2007-11-23 11:18:18 | -1 | NDk0ZDA3ZmU5NmFlZmYyMjA5Mjc2ZTZiYTFmMTMxODY. | 4 | 0 | X-Lite release 1011s stamp 41150 | NULL | +----------------------+--------------------------+----------------------------------------------------------------------+----------+---------------------+----+----------------------------------------------+------+-------+----------------------------------+----------+ 2 rows in set (0.00 sec) Here is what does the internal proxy (192.168.0.165:7160) which is registrar and perform location service, the destination set contains both contacts and the first contact used is the one which is unavailable. this proxy sends first invite with this contact to boundary sip proxy Nov 23 11:20:27 sen ser[28434]: route[FORWARD]: xlset_attr(bsp_route, <$global_addr_bsp;lr=on>) Nov 23 11:20:27 sen ser[28434]: route[FORWARD]: attr_destination($bsp_route) Nov 23 11:20:27 sen ser[28434]: route[FORWARD]: ($fr_inv_timer) Nov 23 11:20:27 sen ser[28434]: route[FORWARD]: t_on_reply(ONREPLY_ROUTE) Nov 23 11:20:27 sen ser[28434]: route[FORWARD]: t_on_failure(FAILURE_ROUTE) Nov 23 11:20:27 sen ser[28434]: route[FORWARD]: t_on_branch(BRANCH_ROUTE) Nov 23 11:20:27 sen ser[28434]: route[FORWARD]: destination set: Contact: sip:touk@192.168.0.164:1338;rinstance=191d670c84591ea3;transport=TCP, sip:touk@192.168.0.164:1348;rinstance=e51fd20db73fc732;transpo Nov 23 11:20:27 sen ser[28434]: route[FORWARD]: next route: <null> Nov 23 11:20:27 sen ser[28434]: route[FORWARD]: message's next hop: sip:voip.rd.touk.pl:7060;lr=on Nov 23 11:20:27 sen ser[28434]: route[FORWARD]: (!t_relay()) Nov 23 11:20:27 sen ser[28434]: branch_route[BRANCH_ROUTE]: <INVITE> ruri:<sip:touk@192.168.0.164:1338;rinstance=191d670c84591ea3;transport=TCP> t:<sip:touk@voip.rd.touk.pl> f:<sip:hellboy@voip.rd.touk.pl> c Nov 23 11:20:27 sen ser[28434]: branch_route[BRANCH_ROUTE]: append_hf(P-Warn: forked branch) Nov 23 11:20:27 sen ser[28434]: branch_route[BRANCH_ROUTE]: <INVITE> ruri:<sip:touk@192.168.0.164:1348;rinstance=e51fd20db73fc732;transport=TCP> t:<sip:touk@voip.rd.touk.pl> f:<sip:hellboy@voip.rd.touk.pl> c Nov 23 11:20:27 sen ser[28434]: branch_route[BRANCH_ROUTE]: append_hf(P-Warn: forked branch) Nov 23 11:20:27 sen ser[28434]: ERROR: tcp_blocking_connect: poll error: flags 18 Nov 23 11:20:27 sen ser[28434]: ERROR: tcp_blocking_connect: SO_ERROR (111) Connection refused Nov 23 11:20:27 sen ser[28434]: ERROR: tcpconn_connect: tcp_blocking_connect failed Nov 23 11:20:27 sen ser[28434]: ERROR: tcp_send: connect failed Nov 23 11:20:27 sen ser[28434]: msg_send: ERROR: tcp_send failed Nov 23 11:20:27 sen ser[28434]: ERROR: t_send_branch: sending request on branch 1 failed Nov 23 11:20:27 sen ser[28434]: route[FORWARD]: drop here is what happens in boundary sip proxy: is receives the invite tries to do the t_rqlay but there is the tcp error. t_relay() fails and it sends the sl_reply_error to internal sip proxy: Nov 23 11:31:38 sen ser[31861]: route[FORWARD]: destination set: <null> Nov 23 11:31:38 sen ser[31861]: route[FORWARD]: next route: <null> Nov 23 11:31:38 sen ser[31861]: route[FORWARD]: message's next hop: sip:touk@192.168.0.164:1338;rinstance=191d670c84591ea3;transport=TCP Nov 23 11:31:38 sen ser[31861]: route[FORWARD]: (!t_relay()) Nov 23 11:31:38 sen ser[31861]: ERROR: tcp_blocking_connect: poll error: flags 18 Nov 23 11:31:38 sen ser[31861]: ERROR: tcp_blocking_connect: SO_ERROR (111) Connection refused Nov 23 11:31:38 sen ser[31861]: ERROR: tcpconn_connect: tcp_blocking_connect failed Nov 23 11:31:38 sen ser[31861]: ERROR: tcp_send: connect failed Nov 23 11:31:38 sen ser[31861]: msg_send: ERROR: tcp_send failed Nov 23 11:31:38 sen ser[31861]: ERROR: t_send_branch: sending request on branch 0 failed Nov 23 11:31:38 sen ser[31861]: route[FORWARD]: (!t_relay()) -> TRUE Nov 23 11:31:38 sen ser[31861]: route[FORWARD]: sl_reply_error() Nov 23 11:31:38 sen ser[31861]: ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) in the attachment there are ngrep and wireshark log from the same situation if you need more info please let me know waiting for your feedback Cheers tomasz On Nov 22, 2007 8:40 PM, Andrei Pelinescu-Onciul <andrei(a)iptel.org> wrote: > On Nov 16, 2007 at 14:43, Tomasz Zieleniewski (JIRA) <tracker(a)iptel.org> wrote: > > tcp branching error > > ------------------- > > > > Key: SER-332 > > URL: http://tracker.iptel.org/browse/SER-332 > > Project: SER > > Issue Type: Bug > > Components: core > > Affects Versions: 2.0 > > Environment: Linux sen 2.6.18-5-amd64 #1 SMP Tue Oct 2 20:37:02 UTC 2007 x86_64 GNU/Linux > > Reporter: Tomasz Zieleniewski > > Priority: Critical > > > > > > Hi, > > > > I have the following situation. > > I have two ser rel_2_0_0 instances one acting as a boundary proxy the > > other on as internal proxy (registrar, location). > > > > I have the following situation I have a user which has registered two > > contacs one woth tcp transport the other one with udp. > > > > All messages coming to and outgoing from my network travels through > > boundary sip proxy. > > > > When internal ser perform lookup_location it get to contacts and the > > destination set contains two records: > > Nov 15 11:03:13 sen ser[3673]: route[FORWARD]: destination set: > > Contact: sip:xlite@192.168.0.164:1221;rinstance=d35f9cda6bd79b0a;transport=TCP, > > sip:xlite@192.168.0.164:7060 > > > > Then internal ser forward message to boundary SIP proxy by changing > > the destination of the message and with the usage of t_relay(). > > I see from log that there are two branches involved: > > Nov 15 11:03:13 sen ser[3673]: branch_route[BRANCH_ROUTE]: <INVITE> > > ruri:<sip:xlite@192.168.0.164:1221;rinstance=d35f9cda6bd79b0a;transport=TCP> > > t:<sip:00@voip.rd.touk.pl> f:<sip:lkl@voip.rd.touk.pl> > > Nov 15 11:03:13 sen ser[3673]: branch_route[BRANCH_ROUTE]: > > append_hf(P-Warn: forked branch) > > Nov 15 11:03:13 sen ser[3673]: branch_route[BRANCH_ROUTE]: <INVITE> > > ruri:<sip:xlite@192.168.0.164:7060;rinstance=cfca8523e3747cb0> > > t:<sip:00@voip.rd.touk.pl> f:<sip:lkl@voip.rd.touk.pl> > > Nov 15 11:03:13 sen ser[3673]: branch_route[BRANCH_ROUTE]: > > append_hf(P-Warn: forked branch) > > > > both INVITES reaches boudary SIP proxy. > > > > Because first contact was unreachable there was an tcp error: > > ser[3516]: ERROR: tcp_blocking_connect: poll error: flags 18 > > Nov 15 11:03:13 sen ser[3516]: ERROR: tcp_blocking_connect: SO_ERROR > > (111) Connection refused > > Nov 15 11:03:13 sen ser[3516]: ERROR: tcpconn_connect: > > tcp_blocking_connect failed > > Nov 15 11:03:13 sen ser[3516]: ERROR: tcp_send: connect failed > > Nov 15 11:03:13 sen ser[3516]: msg_send: ERROR: tcp_send failed > > Nov 15 11:03:13 sen ser[3516]: ERROR: t_send_branch: sending request > > on branch 0 failed > > > > after which ser invokes sl_reply_error() which results in 477 next hop > > error reponse message sent to internal ser instance. > > The problem arises when boudary sip proxy processes second INVITE > > which correspond to second destination. > > when boundary SIP proxy receives it doesn't send it to user agent but > > instead reponses to internal ser with > > 500 I'm terribly sorry, server error occurred (1/SL) > > the only thing I see in the boundary SIP proxy log is: > > ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) > > > > Did I miss something here or this is an error? > > Waiting for Your feedback > > > > Could you make some traffic dump between the 2 proxies and send it to me > (andrei(a)iptel.org)? > It's sounds quite strange (it should work without any problems) and a > look at the packets might help. > > > Thanks, > Andrei > _______________________________________________ > Serdev mailing list > Serdev(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serdev >

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Re: [Serusers] Can SER support 3-way Conference
by roayu 23 Nov '07

23 Nov '07

Hi Ram! I've installed SER (sip stack) and SEMS. I managed to start both SER but I failed to start SEMS. I got the following error message: "/usr/local/sbin/sems : Permission denied" Then I found that the permission for /usr/local/sbin/sems is not 755 and the file size of it is 0 (zero). Why am I having this error message? Can anyone there tell me why? Thanks. Roa Yu :) -----Original Message----- From: ram [mailto:talk2ram@gmail.com] Sent: Wednesday, November 21, 2007 9:06 PM To: roayu Subject: Re: [Serusers] Can SER support 3-way Conference On Nov 20, 2007 8:39 AM, roayu <roayu(a)ctisys.net> wrote: > > > > Hi there, > > > > Can SER alone support 3-way Conference w/o integrate with other > applications? > > > Hi SER is proxy not a Media gateway Look at SEM or ASTERISK for that ram

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[OpenSER-Users] CPL serial forking behaviour
by Daniel 23 Nov '07

23 Nov '07

Hi all, I'd like to understand CPL the behaviour or serial forking made by CPL. This is my CPL script (for USER A): <incoming> <lookup source="registration"> <success> <proxy ordering="sequential" /> </success> </lookup> </incoming> </cpl> The first question is : In this case, if I have 2 contacts for USER A, what is its behaviour ? I have made a couple of tests about it. USER A has 2 contacts : USERA@IP_1 and USERA@IP_2 and the CPL script as above. USER B has 1 contact : USERB@IP_3 The location table is: ID username contact q 318 USERB USERB@IP_3 1 319 USERA USERA@IP_1 1 320 USERA USERA@IP_2 1 In this case the CPL script calls both of Contacts of USER A, at the same time. Then, I've changed the two contact's q parameters: ID username contact q 318 USERB USERB@IP_3 1 319 USERA USERA@IP_1 1 320 USERA USERA@IP_2 0.1 In this case CPL script calls first USERA@IP_2 (low q paramenter) and then USERA@IP_1 (high q-value) ID username contact q 318 USERB USERB@IP_3 1 319 USERA USERA@IP_1 0.1 320 USERA USERA@IP_2 1 In this case CPL script calls first USERA@IP_1 (low q paramenter) and then USERA@IP_2 (high q-value) ID username contact q 318 USERB USERB@IP_3 1 319 USERA USERA@IP_1 0.1 320 USERA USERA@IP_2 0.5 In this case CPL script calls first USERA@IP_1 (low q paramenter) and then USERA@IP_2 (high q-value) ID username contact q 318 USERB USERB@IP_3 1 319 USERA USERA@IP_1 0.1 320 USERA USERA@IP_2 0.1 In this case the CPL script calls both of Contacts of USER A, at the same time. It seem work in the reverse mode: it call first the lowest q value and then the highest. So, does PROXY element (with ordering="sequential") care about q-parameter ? Can somebody help me? Regards, Daniel -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

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[OpenSER-Users] Help with unixODBC + sqlite driver
by kionez 22 Nov '07

22 Nov '07

Hi, i found a message on openser-user mailing list: http://openser.org/pipermail/users/2007-April/010628.html I'm trying to get working my openser with unixodbc+sqlite3, but i don't know why it fails with same errors.. i think is unixodbc openser's module, because with "isql -v mysqlitedb" i can access to my sqlite db... someone can help me? :) thanx Fulvio

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[OpenSER-Users] RINSTANCE
by Thiago Maluf 22 Nov '07

22 Nov '07

Somebody know what is RINSTANCE in SIP Contact? ---------------------------------------------------------------- THIAGO MALUF RESENDE e-mail: malufrj(a)gmail.com

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[OpenSER-Users] Edit SIP Header, Contact field
by Thiago Maluf 22 Nov '07

22 Nov '07

I Want edit the Contact field. How I do it? Thanks in advanced, Thiago -- ---------------------------------------------------------------- THIAGO MALUF RESENDE Consultor Voip e Programador WEB (Voip Developer and Web Developer) Tel: +55 21 86042100 e-mail: malufrj(a)gmail.com

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[OpenSER-Users] [CPL] ERROR:core:set_dst_uri: not enough pkg memory
by Iñaki Baz Castillo 22 Nov '07

22 Nov '07

I have a simple CPL for a user "200" that just should lookup for the contact in the "location" table: <cpl> <incoming> <lookup source="registration"> <success> <proxy /> </success> </lookup> </incoming> </cpl> In the script I do: cpl_run_script("incoming","FORCE_STATEFUL"); and I get an error: ERROR:core:set_dst_uri: not enough pkg memory ERROR:cpl_c:cpl_proxy_to_loc_set: do_action failed ERROR:cpl_c:cpl_run_script: runtime error and the script continues but with the URI changed. It's strange: it's obvious that CPL does a DB query to "location" table asking for 200@domain (I've monitorized it in mysql logs) but after that it shows the errors above and the OpenSer script continues (with the URI set to the contact URI). Why can it occur? maybe it's a bug? Thanks. -- Iñaki Baz Castillo

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Re: [OpenSER-Users] openser SBC and upper registration.
by Ovidiu Sas 21 Nov '07

21 Nov '07

This issue was already discussed. Check the following link: http://lists.openser.org/pipermail/users/2007-May/010857.html Regards, Ovidiu Sas On Nov 16, 2007 6:53 PM, rohail khan <rohail_12(a)yahoo.com> wrote: > Well save() does not work on reply route, its for request route , what I did > is from reply route i used t_check_status("200") and route it to another > route block let say route[6] and I used save_noreply() to save the contact > and it saving now but the problem is that like SBC I was modifying > registration request from UA and changing the CONTACT before relaying to my > other server but when I am saving the contact info its saving with openser > contact info no information for UA so call can still can not route to UA. > any idea > > Thanks > > > Ovidiu Sas <osas(a)voipembedded.com> wrote: > You can call save() in the reply_route. You can check there is the > response is a 200ok or not. > In this particular case you will lose some informations that are used > to handle NATed UA. > Test and see if it will fit your needs. > > > Regards, > Ovidiu sas > > On Nov 15, 2007 5:53 AM, rohail khan wrote: > > Hi > > I am trying to configure openser for uppser registration, so far I am > > relaying the registration after modifying contact in the registration of > > openser IP address but I want to save the contact only if the registration > > is sucessfull from upper registration server. If I use "save_noreply" its > > save the location no matter if the UA is registered with upper > registration > > server > > > > any idea > > thanks in advance > > > > > > > > ________________________________ > > Never miss a thing. Make Yahoo your homepage. > > _______________________________________________ > > Users mailing list > > Users(a)lists.openser.org > > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > > > > > > > ________________________________ > Be a better sports nut! Let your teams follow you with Yahoo Mobile. Try it > now.

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Re: [Serusers] Implicit Registration Set
by Jiri Kuthan 21 Nov '07

21 Nov '07

p.s. which may be good enough. we know from interop events that having multiple contacts with only one actual registration has actually caused some interop issues. anyhow -- it depends on what your use case is, some of these things -- should they pay off -- are scriptable. At 23:01 21/11/2007, Jiri Kuthan wrote: >It will return one contact. -jiri > >At 07:44 21/11/2007, Magnus Sörman wrote: >>OK, but using alias, will that give me back both contacts in the 200 OK for a Register (unfortunately I don't have the possibility to try this right now)? >> >> /Magnus >> >>-----Original Message----- >>From: Jiri Kuthan [mailto:jiri@iptel.org] >>Sent: den 15 november 2007 15:00 >>To: Magnus Sörman; serusers(a)lists.iptel.org >>Subject: Re: [Serusers] Implicit Registration Set >> >>well it depends -- doing what you are suggesting literally is quite tricky and I'm not sure this IMS feature is worth the effort. IF you would like to register and be then available under all of your aliases (both "that address" >>and "his tel-uri"), you can easily use aliases. >> >>-jiri >> >>At 14:52 15/11/2007, Magnus Sörman wrote: >>>Content-class: urn:content-classes:message >>>Content-Type: multipart/alternative; >>> boundary="----_=_NextPart_001_01C8278E.D8F42EB4" >>> >>>Hi, >>> >>>How do you/could you configure SER to use implicit registration set? >>>If e.g. a user with a sip URI registers, then both that address and his tel-uri should be registered. >>> >>>Thanks, >>> /Magnus Sorman >>>_______________________________________________ >>>Serusers mailing list >>>Serusers(a)lists.iptel.org >>>http://lists.iptel.org/mailman/listinfo/serusers >> >> >> >>-- >>Jiri Kuthan http://iptel.org/~jiri/ > > > >-- >Jiri Kuthan http://iptel.org/~jiri/ -- Jiri Kuthan http://iptel.org/~jiri/

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Re: [Serusers] Implicit Registration Set
by Jiri Kuthan 21 Nov '07

21 Nov '07

It will return one contact. -jiri At 07:44 21/11/2007, Magnus Sörman wrote: >OK, but using alias, will that give me back both contacts in the 200 OK for a Register (unfortunately I don't have the possibility to try this right now)? > > /Magnus > >-----Original Message----- >From: Jiri Kuthan [mailto:jiri@iptel.org] >Sent: den 15 november 2007 15:00 >To: Magnus Sörman; serusers(a)lists.iptel.org >Subject: Re: [Serusers] Implicit Registration Set > >well it depends -- doing what you are suggesting literally is quite tricky and I'm not sure this IMS feature is worth the effort. IF you would like to register and be then available under all of your aliases (both "that address" >and "his tel-uri"), you can easily use aliases. > >-jiri > >At 14:52 15/11/2007, Magnus Sörman wrote: >>Content-class: urn:content-classes:message >>Content-Type: multipart/alternative; >> boundary="----_=_NextPart_001_01C8278E.D8F42EB4" >> >>Hi, >> >>How do you/could you configure SER to use implicit registration set? >>If e.g. a user with a sip URI registers, then both that address and his tel-uri should be registered. >> >>Thanks, >> /Magnus Sorman >>_______________________________________________ >>Serusers mailing list >>Serusers(a)lists.iptel.org >>http://lists.iptel.org/mailman/listinfo/serusers > > > >-- >Jiri Kuthan http://iptel.org/~jiri/ -- Jiri Kuthan http://iptel.org/~jiri/

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Re: [Serusers] Linux version -information request-
by Jiri Kuthan 21 Nov '07

21 Nov '07

Almost any unless you have a real historical one -- debian, RH, gentoo, etc, ideally with 2.6 kernel. -jiri At 21:04 21/11/2007, Mary Allen (mrallen) wrote: >Content-class: urn:content-classes:message >Content-Type: multipart/alternative; > boundary="----_=_NextPart_001_01C82C79.91616C64" > >Hi, > >I need to install the SERi proxy. Would you please let me know what version of Linux should I put in my system before the installation of SERi? > >Thanks, >Mary >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

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[Serusers] Linux version -information request-
by Mary Allen (mrallen) 21 Nov '07

21 Nov '07

Hi, I need to install the SERi proxy. Would you please let me know what version of Linux should I put in my system before the installation of SERi? Thanks, Mary

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[OpenSER-Users] [OT] Looking for a good SIP PBX
by Iñaki Baz Castillo 21 Nov '07

21 Nov '07

Hi, first of all I'm sorry for this OffTopic. I'm looking for a good SIP PBX since Asterisk SIP implementation doesn't let me using it in a multidomain SIP enviroment properly with OpenSer ("From" domain not respected when generates a SIP channel, no good behaviour in parallel forking, etc). I can use it just as media server for now. So I'm looking for other SIP PBX server. One I've in mind is, of course, WeSIP, but I'd like to know if any of you has good experiences integrating other PBX's as FreeSwitch or SipX/SipFoundry. If possible, it would be really great a SIP PBX with virtual domains support, so the PBX logic for independent SIP domains would exist in the same server/PBX (less and easy maintenance). The aim of this PBX is to allow SIP incoming calls and during the execution of the dialplan maybe call to that domain users (registered in a multidomain main OpenSer). Please, any opinion or experience about it with some of those PBX's? I'm going now to install and try of all them, but some help would be really appreciated. Thanks a lot. Regards. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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Re: [Serusers] Implicit Registration Set
by Jiri Kuthan 21 Nov '07

21 Nov '07

well it depends -- doing what you are suggesting literally is quite tricky and I'm not sure this IMS feature is worth the effort. IF you would like to register and be then available under all of your aliases (both "that address" and "his tel-uri"), you can easily use aliases. -jiri At 14:52 15/11/2007, Magnus Sörman wrote: >Content-class: urn:content-classes:message >Content-Type: multipart/alternative; > boundary="----_=_NextPart_001_01C8278E.D8F42EB4" > >Hi, > >How do you/could you configure SER to use implicit registration set? >If e.g. a user with a sip URI registers, then both that address and his tel-uri should be registered. > >Thanks, > /Magnus Sorman >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

2 1

[Serusers] November 79% OFF
by Canadian Doctor Damon Herrera 21 Nov '07

21 Nov '07

Spam detection software, running on the system "mail.iptel.org", has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: Best Price for VIAGRA & CIALIS ONLINE, All Credit Cards Accepted, FREE DELIVERY http://www.foodenter.com [...] Content analysis details: (18.1 points, 5.0 required) pts rule name description ---- ---------------------- -------------------------------------------------- 0.0 UNPARSEABLE_RELAY Informational: message has unparseable relay lines 2.7 DRUG_ED_ONLINE BODY: Fast Viagra Delivery 0.5 DRUG_ED_CAPS BODY: Mentions an E.D. drug 3.5 BAYES_99 BODY: Bayesian spam probability is 99 to 100% [score: 0.9999] 1.6 URIBL_SBL Contains an URL listed in the SBL blocklist [URIs: foodenter.com] 4.1 URIBL_JP_SURBL Contains an URL listed in the JP SURBL blocklist [URIs: foodenter.com] 2.1 URIBL_WS_SURBL Contains an URL listed in the WS SURBL blocklist [URIs: foodenter.com] 3.0 URIBL_OB_SURBL Contains an URL listed in the OB SURBL blocklist [URIs: foodenter.com] 0.5 DRUGS_ERECTILE Refers to an erectile drug

1 0

[Serusers] Another alternative for handling RTP packets
by Carlos Loarca 20 Nov '07

20 Nov '07

Hi everyone: I have been working with NATHELPER or MEDIAPROXY to handle audio and video calls. However I have got into the point that there should be another way to work Firewalls and NAT transversal rather than mediaproxies and rtp-proxies. I have heard about STUN servers with ICE support but never seen it working and I wonder if this works. And I dont know any server like this, and I know that the softphones or IPphones need to support it. If I dont want to force the audio or video to come to my server, what kind of solution is the most recommendable? Best regards, /Carlos ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs

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Re: [OpenSER-Users] CPL serial call attempt
by Iñaki Baz Castillo 20 Nov '07

20 Nov '07

El Martes, 20 de Noviembre de 2007, Daniel Grotti escribió: > Hi Inaki, > Thank you for your help. > I've a question, does Serial Forking (with LCR) work well with NAT ? Sure, there is not difference. Just force RtpProxy (or MediaProxy) in each branch (when neccesary). -- Iñaki Baz Castillo

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[OpenSER-Users] Openser 1.3 trunk: unix_sock
by Helmut Kuper 20 Nov '07

20 Nov '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, after upgrading from openser 1.2 to 1.3 trunk the core commands unix_sock, sock_mode, sock_user, sock_group are missing. Does anybody know what happend to this commands ? regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHQuo/4tZeNddg3dwRAkx9AKCPRZKGfyOxbox73Asf+aaIgAWLnACfc/Uu IepPGsTGfNwJt2p1frTRSwM= =VFPA -----END PGP SIGNATURE-----

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[OpenSER-Users] Natping & register
by Stanisław Pitucha 20 Nov '07

20 Nov '07

Hey Is it possible to do with openser/nathelper what some other sip/nat gateways do? - for initial register, keep location, but pass registration packets upstream -> save("location", 0x02) - when there's a nat flag set, rewrite response to client with very low "Expire:" header - accept frequent register packets from client, but register on real server, when original expire time is reached It's basically natping done the other way - should be more stable, because when one natping doesn't reach the client, it's lost till next register / invite from that phone. With registers, phone tries to sent packets itself, so when one is lost, next one will reactivate nat anyway. Is there any module that does it now? Or does it need some avp & database hacking?

1 0

[OpenSER-Users] cpl-c module problems
by Aki Rehn 20 Nov '07

20 Nov '07

Hi, Im having some cpl-c module problems.. Both are related to time-switch. What it seems to me like is that bymonth parameter is parsed somehow wrong which then causes overflow. I added bit more debug and attr_name and attr_len seem to be completely off-the-track. (The actual line number of the overflow in cpl_switches.h is 763 because of the one line added.) Input: <time dtstart="20071101T090000" dtend="20171101T090000" byhour="7,8,9,10,11,12,13,14,15,16,17,18,19,20"> Output: openser[5160]: DEBUG:cpl_run_script: processing CPL node openser[5160]: DEBUG:cpl_run_script: processing time-switch node openser[5160]: DEBUG:cpl-c:run_time_switch: checking recv. time stamp <1195111240> openser[5160]: DEBUG:cpl-c:set_TZ: switching TZ as "TZ=Europe/Helsinki" openser[5160]: ---> fill = Thu Nov 15 09:20:40 2007 openser[5160]: DEBUG:cpl_c:testing, attr_name: [0], attr_len: [16] openser[5160]: DEBUG:cpl_c:run_time_node: attribute [0] found :[20071101T090000] openser[5160]: DEBUG:cpl_c:testing, attr_name: [1], attr_len: [16] openser[5160]: DEBUG:cpl_c:run_time_node: attribute [1] found :[20171101T090000] openser[5160]: DEBUG:cpl_c:testing, attr_name: [19], attr_len: [12592] openser[5160]: ERROR:cpl_c: overflow detected ip=0xb5a3c8ec ptr=0xb5a3fa6e in func. cpl_switches.h, line 764 openser[5160]: DEBUG:cpl-c:set_TZ: switching TZ as "TZ=" openser[5160]: ERROR:cpl_c:cpl_run_script: script error Second problem is the byhour attribute, which doesnt seem to be supported at all. Input: <time dtstart="20071101T090000" dtend="20171101T090000" byhour="7,8,9,10,11,12,13,14,15,16,17,18,19,20"> Output: openser[5162]: DEBUG:cpl_run_script: processing CPL node openser[5162]: DEBUG:cpl_run_script: processing time-switch node openser[5162]: DEBUG:cpl-c:run_time_switch: checking recv. time stamp <1195111623> openser[5162]: DEBUG:cpl-c:set_TZ: switching TZ as "TZ=Europe/Helsinki" openser[5162]: ---> fill = Thu Nov 15 09:27:03 2007 openser[5162]: DEBUG:cpl_c:testing, attr_name: [0], attr_len: [16] openser[5162]: DEBUG:cpl_c:run_time_node: attribute [0] found :[20071101T090000] openser[5162]: DEBUG:cpl_c:testing, attr_name: [1], attr_len: [16] openser[5162]: DEBUG:cpl_c:run_time_node: attribute [1] found :[20171101T090000] openser[5162]: DEBUG:cpl_c:testing, attr_name: [9], attr_len: [39] openser[5162]: DEBUG:cpl_c:run_time_node: attribute [9] found :[7,8,9,10,11,12,13,14,15,16,17,18,19,20] openser[5162]: ERROR:cpl_c:run_time_switch: unsupported attribute [9] found in TIME node openser[5162]: DEBUG:cpl-c:set_TZ: switching TZ as "TZ=" openser[5162]: ERROR:cpl_c:cpl_run_script: script error Br, Aki Rehn

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[OpenSER-Users] Error using openser
by Marcio Ribeiro 20 Nov '07

20 Nov '07

Hi, I'm trying to set up openser with TLS support with mysql database, on Debian. In fact, i could not even start openser with succes. When using "openserctl start": Starting OpenSER: ERROR: PID file /var/run/openser.pid does not exist -- OpenSER start failed when using "openser": Listening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 164.41.14.213 [164.41.14.213]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 164.41.14.213 [164.41.14.213]:5060 Aliases: tcp: unbvoip3.cic.unb.br:5060 tcp: localhost:5060 tcp: localhost.localdomain:5060 udp: unbvoip3.cic.unb.br:5060 udp: localhost:5060 udp: localhost.localdomain:5060 *: voip.unb.br:* but it's not running, since my "ps aux | grep openser" cannot find it. Sugestions??? Thanks, Marcio

4 6

[Serusers] Can SER support 3-way Conference
by roayu 20 Nov '07

20 Nov '07

Hi there, Can SER alone support 3-way Conference w/o integrate with other applications? Thanks. Roa Yu :-)

1 0

Re: [Serusers] Voicemail setup on SER v0.9.6
by roayu 20 Nov '07

20 Nov '07

Thanks Stefan! I will try on it. Roa Yu =) -----Original Message----- From: Stefan Sayer [mailto:stefan.sayer@iptego.de] Sent: Thursday, November 15, 2007 7:41 PM To: roayu Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] Voicemail setup on SER v0.9.6 roayu wrote: > Thanks Stefan! > > I need to confirm that after the installation, I will be having 2 SERs + 1 > SEMS, rite? ( 1 SER for normal sip, 1 SER as SIP stack, 1 SEMS) yes. this is the easiest way to do it. Regards Stefan > > Thanks. > > Roa Yu > -- Stefan Sayer VoIP Services stefan.sayer(a)iptego.de www.iptego.de iptego GmbH Am Borsigturm 40 13507 Berlin Germany Amtsgericht Charlottenburg, HRB 101010 Geschaeftsfuehrer: Alexander Hoffmann

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[Serusers] from uri???
by Roderick 19 Nov '07

19 Nov '07

Hi list, I'm looking how to forward a call to a PBX if it comes from a certain number. Just to illustrate, this would call forward to a pbx if you're dialing 123: if (uri=~"^sip:123@") { #Forward to PBX if calling 123 } What I want is something like if (from=~"^sip:123@") { #Forward to PBX if calling FROM 123 } Is there something like this??? Thanks for your help, Roderick

1 0

[OpenSER-Users] check_from() and calls from a PBX
by Iñaki Baz Castillo 19 Nov '07

19 Nov '07

Hi, I've a PBX (Asterisk) registered in OpenSer as user "pbx@domain". When Asterisk receives a call from PSTN it calls to OpenSer users with From: From: <sip:PSTN_number@domain> But of course, this can't validate "check_from()" since From username (PSTN_number) and auth username (pbx) are differents. As I solution I don't do "check_from()" if From username is "pbx" but of course, it can create security problems. Other solution is to force "From" username in PBX outgoing calls, but with this the callerid number will be "PBX" (that is not very cool for end users xD). Another solution is using "trusted" ips, but this is not valid for me since Asterisk is behind a NAT and dynamic IP, and because there can be multiples PBX for each domain of OpenSer and I don't want they to have privileges. Is there any other approach for this? Thanks for any suggestion. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

3 2

[OpenSER-Users] lookup("location") and CPL locations set
by Daniel 19 Nov '07

19 Nov '07

Hi all, I have a question: in this scenario route(1) { ..... ...... lookup ("location"); cpl_run_script("incoming", "force_stateful"); .... ... } is the location (R-URI address), re-writed by the "lookup" function, part of the CPL locations set ? Or CPL doesn't care about this and it see the locations set empty? Regards daniel -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

2 1

[OpenSER-Users] Openser performance with rtpproxy
by Stanisław Pitucha 19 Nov '07

19 Nov '07

Hey I've got a simple openser configuration working with rtpproxy. It's mostly copied from http://www.voip-info.org/wiki/view/OpenSER+And+RTPProxy but I don't handle registration and do only nat helping (for current testing I don't even save location - Contact / URI headers are enough). I've tried setting up sipp (no authorization, just invite, send 7 secs rtp, bye) and standard UAS scenario on the other end. I start to get retransmissions / timeouts when I use 2 hosts and ~70 simultaneous connections (total). Typical delay is ~250ms (not network delay - it's on gigabit LAN). I've used the same sipp scenarios for asterisk before and got up to 700 sessions running correctly, so I believe the problem is with openser config here. How can I check what causes the problems exactly? Is there any typical performance problem? I've tried googling already - couldn't find anything obvious. I'm running svn version. Thanks for ideas.

1 0

[OpenSER-Users] [CPL] ERROR:cpl-c:run_priority_switch: error parsing attr [0][9]
by Iñaki Baz Castillo 19 Nov '07

19 Nov '07

Hi, I'm just starting with CPL and I don't know if this mail should go to devel list or users list. I load a very simple CPL script sucessfully: $ openserctl fifo LOAD_CPL sip:ibc@domain /etc/openser/cpl/ibc(a)domain.xml 200 OK The XML is this one: <?xml version="1.0" encoding="UTF-8"?> <cpl> <incoming> <time-switch> <time dtstart="9" dtend="5" wkst="MO|TU|WE|TH|FR"> <lookup source="registration"> <success> <proxy /> </success> </lookup> </time> <otherwise> <location url="sip:angel@domain"> <proxy /> </location> </otherwise> </time-switch> </incoming> </cpl> But when OpenSer runs: cpl_run_script("incoming","force_stateful"); I get in the logs: ERROR:cpl-c:run_priority_switch: error parsing attr [0][9] ERROR:cpl_c:cpl_run_script: script error I can't understand. I suppose that that XML check is done when loading the file with FIFO command. Why then I get a "error parsing" during OpenSer script? is it normal? Thanks for any explanation. Best regards. -- Iñaki Baz Castillo

3 2

[OpenSER-Users] CPL serial call attempt
by Daniel 19 Nov '07

19 Nov '07

Hi all, I have this problem. I have 2 contacts for one user subscriber, so in LOCATION tables, after REGISTER session I have: ID user contact q 12 A A@IP_X 1 13 A A@IP_Y 1 14 B B@IP_Z 1 so, B calls A and A has an incoming CPL script. In this script I'd like to call the 2 A's contacts in sequential order: The script is after looup("location"), so the R-URI is just A@IP_X (because it has lowest ID) SRIPT user A: <cpl> <incoming> <location url="sip:A@IP_Y" priority="0.3"> <proxy timeout="10" ordering="sequential" /> </location> </incoming> </cpl> Whit this script I hoped to add A@IP_Y to location set (that already included A@IP_X) with priority 0.3, and then call A@IP_X as first (it has priority=q=1) and then, if I no aswer after 10sec, call A@IP_Y (that it has lowest priority). CPL don't add A@IP_Y to the location set, but it rewrites the location (it removes the old R-URI, A@IP_X) with A@IP_Y and it call only this contact. Is there a sense in what I've write? if yes, where is the problem? regards, daniel -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

4 4

[OpenSER-Users] record_route() for responses
by Klaus Darilion 19 Nov '07

19 Nov '07

Hi! I just was thinking about the follwing scenario_ client a--->proxy--->client b the proxy does record routing a calls b b send back 200 OK, but does NOT mirror the Record-route headers. in-dialog requests sent from a to b will bypass the proxy, thus bypass proxy security features (e.g. removing the credentials). Thus IMO we should have some settings which allow the proxy to add RR header also for relayed response if during request processing record-route was called and the proxy misses itself as the top RR header. what do you think? regards klaus

7 15

[OpenSER-Users] Record-Route: "did" - parameter
by Franz Edler 19 Nov '07

19 Nov '07

Hi, can anyone explain what the "did" parameter in Record-Route header does? Record-Route: <sip:192.168.0.5;lr=on;ftag=e45c4b13;did=c92.09e2afc1> I suppose it is inserted in case of a "routing spiral", because it is inserted when the INVITE request is routed through the proxy a second time. Cheers Franz

2 1

[OpenSER-Users] OpenSER + Remote Database scalability issue
by John Barry 19 Nov '07

19 Nov '07

Hello, I have notice that when I configure http-digest authentication for INVITE messages in OpenSER 1.2.2, there is a big difference in the call throughput when you use a database in a remote machine. For example, when using a Mysql database in a remote machine with a RTT= 0.187 ms, the total call throughput decreases by almost 60% in comparison when using the database in the same machine where OpenSER is installed. Why is this happening? How can improve this behavior? I would like to have the database in a separate machine. Thanks, JB74 _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

2 1

[OpenSER-Users] openxcap - GET a document
by Reinhold Buchinger 18 Nov '07

18 Nov '07

Hi, I'm trying to fetch a document from the openxcap server with a normal browser. My openxcap configuration uses the database backend and I created a db table equal to the xcap table used in OpenSER. For testing I inserted some data but I'm not sure what the fields "source" and "doc-uri" should contain. Because whenever I try to fetch a document with my webbrowser I receive a 404 Not Found. OpenXCAP doesn't show any logs... Thanks a lot for your help! Best regards, Reinhold

2 4

[OpenSER-Users] openser SBC and upper registration.
by rohail khan 16 Nov '07

16 Nov '07

Hi I am trying to configure openser for uppser registration, so far I am relaying the registration after modifying contact in the registration of openser IP address but I want to save the contact only if the registration is sucessfull from upper registration server. If I use "save_noreply" its save the location no matter if the UA is registered with upper registration server any idea thanks in advance --------------------------------- Never miss a thing. Make Yahoo your homepage.

2 1

[OpenSER-Users] save("location") not saving in openser 1.3
by Dave Singer 16 Nov '07

16 Nov '07

Any one interested in helping with OpenSER 1.3? Compiled openser 1.3 from svn, installed new test db from 1.3 svn db scripts (tables should have right structure). when a phone tries to register it hits this script section. if (!www_authorize("", "subscriber")) { xlog("L_INFO", "user didn't send credentials: f-uri <$fu> r-uri <$ru>\n"); www_challenge("", "0"); exit; }; xlog("L_INFO", "user registered: f-uri <$fu>\n"); save("location"); exit; nothing is actually saving in location table. I sometimes get extra messages in the log file: core:db_print_set: Error in snprintf usrloc:db_update_ucontact: updating database failed usrloc:wb_timer: updating contact in db failed I am able to use openserctl, from the openser server with all the same db setting as the openser.cfg file has, to make changes so It "can't" be permissions. It is able to verify with the www_authorize that we should save it. just not able to do the actual save. I'm using mysql 5.x (which is also serving up the production openser 1.2-svnserver with a diff db) and openser -V renders: version: openser 1.3.0-pre1-notls (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP, 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_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. svnrevision: unknown @(#) $Id: main.c 3015 2007-11-05 13:08:50Z henningw $ main.c compiled on 14:57:42 Nov 5 2007 with gcc 4.1.1 Thanks Dave

2 2

[OpenSER-Users] cancel going to 127.0.0.1
by Richard Bennett 16 Nov '07

16 Nov '07

Hi, I seem to remember reading about this problem before, but couldn't find any pointers by searching... On the following setup with Openser 1.1: UA -> Openser/Mediaproxy -> PSTN-Gateway most calls pass just fine and cancel fine, but some don't stop properly. If the UA cancels the call the cancel goes: gateway ip= 1.2.3.204 proxy ip= 1.2.3.201 ua ip = 81.1.2.223 U 81.1.2.223:5071 -> 1.2.3.201:5060 CANCEL sip:0475711111@sipc.example.com SIP/2.0. Via: SIP/2.0/UDP 192.168.1.11:5071;branch=z9hG4bK-2f6ae216. From: 002 <sip:101211111@sipc.example.com>;tag=a27b4fe18ca60f4ao0. To: <sip:0475711111@sipc.example.com>. U 1.2.3.201:5060 -> 127.0.0.1:5060 CANCEL sip:0475711111@sipc.example.com SIP/2.0. Record-Route: <sip:1.2.3.201;lr=on;ftag=a27b4fe18ca60f4ao0>. Via: SIP/2.0/UDP 1.2.3.201;branch=z9hG4bK7f9f.b68d8185.0. Via: SIP/2.0/UDP 192.168.1.11:5071;received=81.1.2.223;branch=z9hG4bK-2f6ae216. From: 002 <sip:101211111@sipc.example.com>;tag=a27b4fe18ca60f4ao0. To: <sip:0475711111@sipc.example.com>. I 127.0.0.1 -> 1.2.3.201 3:3 ....E..o..@.@...Y............[h.CANCEL sip:0475711111@sipc.example.com SIP/2.0. Record-Route: <sip:1.2.3.201;lr=on;ftag=a27b4fe18ca60f4ao0>. Via: SIP/2.0/UDP 1.2.3.201;branch=z9hG4bK7f9f.b68d8185.0. Via: SIP/2.0/UDP 192.168.1.11:5071;received=81.1.2.223;branch=z9hG4bK-2f6ae216. From: 002 <sip:101211111@sipc.example.com>;tag=a27b4fe18ca60f4ao0. To: <sip:0475711111@sipc.example.com>. And then it keeps looping between 127.0.0.1 and 1.2.3.201. For some reason openser sends the CANCEL to the loopback interface instead of on to the gateway. Is this a recognizable problem or should I post a full trace and config file? Thanks for looking, Richard.

2 3

[OpenSER-Users] Newbie issue: Cannot exec HelloWorldServlet on WeSIP
by Alvaro Mari Diego 16 Nov '07

16 Nov '07

Hi everyone, I'm new to WeSIP, so I've got stuck pretty fast, as expected :) I got WeSIP running on OpenSER 1.2.0. I'm trying to run a HelloWorld sip servlet to test WeSIP, which responses a 200 OK and a MESSAGE request saying "Hello, World", to any incoming request to a sip:greetings@amd-openims.test URI. I mounted my sar file, and installed it in wesip webapps with the web manager application, and got WeSIP properly connected to OpenSER' seas. Openser is configured to route any MESSAGE request to WeSIP AS. But when I try to send a message to sip:greetings@amd-openims.test from a client logged as alice(a)amd-openims.test, WeSIP console displays this: ******************************************************************** SeasMessageEvent [SeasRouter] - New message received, from 138.4.7.163:6060 to 138.4.7.163:5080 over UDP SeasTransactionEvent [SeasRouter] - hash_index = 54974,label=620758046 flags=0 ExpressMessageChannel [ExpressMChannel[0]] - peerAddress = 138.4.7.163/6060 ExpressMessageChannel [ExpressMChannel[0]] - About to process MESSAGE sip:greetings@amd-openims.test SIP/2.0 SipConnector [ExpressMChannel[0]] - createProcessor: Reusing existing processor SipProcessor [ExpressMChannel[0]] - An incoming message is being assigned SipProcessor [SipProcessor[4]] - <<<<<<<<< Request Received <<<<<<<<< MESSAGE sip:greetings@amd-openims.test SIP/2.0 Route: <sip:138.4.7.163:5080;lr=on>,<sip:iscmark@scscf.amd-openims.test:6060;lr=on;s=1;h=0;d=0> Via: SIP/2.0/UDP 138.4.7.163:6060;branch=z9hG4bKeb6d.ff7360e1.0,SIP/2.0/UDP 138.4.7.163:4060;branch=z9hG4bKeb6d.000ef08.0,SIP/2.0/UDP 138.4.7.149:63926;branch=z9hG4bK-d87543-b370c6252d6ad226-1--d87543-;rport=63926 Max-Forwards: 14 To: "Servicio HelloWorld" <sip:greetings@amd-openims.test> From: "alice" <sip:alice@amd-openims.test>;tag=737cbe08 Call-ID: Y2FiYmY0MmM2NjFhYzU5NWUzNTE2MmM5MWY2ZjdhMGI. CSeq: 2 MESSAGE Allow: INVITE,CANCEL,ACK,BYE,INFO,OPTIONS,MESSAGE,SUBSCRIBE,NOTIFY,REFER Content-Type: application/unknown? User-Agent: X-Lite release 1011s stamp 41150 P-Charging-Vector: icid-value="P-CSCFabcd4739818c00000007";icid-generated-at=138.4.7.163;orig-ioi="amd-openims.test" Content-Length: 262 SeasHashTable [SipProcessor[4]] - Inserting hashIdx:54974 label:620758046 into hashTable SipProcessor [SipProcessor[4]] - Transaction [z9hG4bKeb6d.ff7360e1.0] is new Transaction?? --> true EngineSipMapper [SipProcessor[4]] - Request from 138.4.7.163 mapped to: ConvergedEngine[WeSIP_engine].StandardHttpHost[localhost] ConvergedEngineValve [SipProcessor[4]] - Invoking Host:localhost ConvergedHostValve [SipProcessor[4]] - SipSession not found for requestMESSAGE, trying match against Rules ConvergedHostValve [SipProcessor[4]] - No servlet-mapping matched this Request. SipResponse [SipProcessor[4]] - >>>>>>>>> Sending Response >>>>>>>>> SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP 138.4.7.163:6060;branch=z9hG4bKeb6d.ff7360e1.0,SIP/2.0/UDP 138.4.7.163:4060;branch=z9hG4bKeb6d.000ef08.0,SIP/2.0/UDP 138.4.7.149:63926;branch=z9hG4bK-d87543-b370c6252d6ad226-1--d87543-;rport=63926 Max-Forwards: 14 To: "Servicio HelloWorld" <sip:greetings@amd-openims.test> From: "alice" <sip:alice@amd-openims.test>;tag=737cbe08 Call-ID: Y2FiYmY0MmM2NjFhYzU5NWUzNTE2MmM5MWY2ZjdhMGI. CSeq: 2 MESSAGE Content-Length: 0 LingerList [SipProcessor[4]] - scheduled transaction z9hG4bKeb6d.ff7360e1.0 for deletion ExpressMessageChannel [SipProcessor[4]] - sending with spiral=false SipConnector [SipProcessor[4]] - recycle: Recycling processor SipProcessor[4] ActionRouter$ActionSender [ActionSender] - Writing 509 bytes into Action Socket SeasHashTable [ExpressMChannel[0]] - Removing hashIdx:54974 label:620758046 from hashTable SeasHashTable [ExpressMChannel[0]] - Transaction with id:z9hG4bKeb6d.ff7360e1.0, hash_idx:54974, label:620758046 removed from the Seas Hash Table ExpressMessageChannel [ExpressMChannel[0]] - Done processing MESSAGE sip:greetings@amd-openims.test SIP/2.0 ******************************************************************** And, as I feared, got a 481 response in my client. I don't know what I'm doing wrong. I wonder it could be a misconfiguration in the WeSIP server.xml, or maybe something wrong in my servlet. I attach you in a zip my HelloWorld Servlet, openser.cfg file, and wesip server.xml file. Thanks in advance, Cheers, -- Alvaro

2 1

[OpenSER-Users] ERROR: init_io_wait: could not alloc epoll array
by Ariadne Ramos 16 Nov '07

16 Nov '07

I have Openser 1.1.1 installed and working fine for long time. But now when I restart the service openser with no changes made in configuration this doesn't start. I have to restart many times to get the service up again. The error that appear in logs is the following: Nov 15 11:32:06 prxc /usr/local/sbin/openser[22658]: ERROR: init_io_wait: could not alloc epoll array Nov 15 11:32:06 prxc /usr/local/sbin/openser[22658]: ERROR: tcp_receive_loop: exiting... Do you know why this is happening? I appreciate your help Thanks in advanced Ariadne

2 1

[OpenSER-Users] q-value
by Daniel 16 Nov '07

16 Nov '07

Hi, Is it possible to change q parameter from Client side? I'm using xlite and Sjphone, but I don't be able to change this value so, Proxy, by default, set as 1 every q-value. any clue? Regards, daniel -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

3 2

[OpenSER-Users] [OT] Do ITSP SIP_PSTN gateways allow REFER?
by Iñaki Baz Castillo 16 Nov '07

16 Nov '07

Hi, I will connect my OpenSer with a Nortel (SIP to PSTN gateway) for incoming/outgoings call to/from PSTN. I have a doubt: if an OpenSer user (200(a)mydomain.org) receives an incoming call from gateway, could he REFER the call to 201(a)mydomain.org? This is: do ITSP gateways allow REFER to SIP users subcribers to my OpenSer? I understand they could not accept REFER's to PSTN number because accounting issues (gateway UAC receving a REFER and calling to PSTN?). Thanks for any explanation. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

5 10

[Serusers] rtpproxy and rtcp
by Miles Scruggs 15 Nov '07

15 Nov '07

Just wondering if there is any way to collect rtcp information from rtpproxy? Thanks Miles

1 0

[OpenSER-Users] attention!!! -- mailing list domains
by Daniel-Constantin Mierla 15 Nov '07

15 Nov '07

Hello, as I see that some send emails to xyz(a)lists.openser.org and cc to xyz(a)openser.org. I want to make sure that everybody understands that only the new domain "lists.openser.org" MUST be used now. First is confusing and, also, we do not need duplicates. If you get problems with lists.openser.org, please report. The old domain "openser.org" for mailing lists will stay in place about one more month from now on, to be sure that most of discussion threads started before migration get closed. The reason at migration time was not to create chaos by not being able to reply, or not having enough time to learn about the new domain. The old mailing lists domains are just aliased to the new ones, the messages end in the same archive, same server. Thanks, Daniel

1 0

[OpenSER-Users] Permissions module addition
by Daryl G. Jurbala 15 Nov '07

15 Nov '07

I'm looking for an addition to the permissions module. I'm using the address table to match valid originators. Each originator needs to be placed into a specific route tree (in carrierroute). Right now, I can do something like: $var(orig) = "unauth"; if (allow_source_address("<fist route tree>")) { $var(orig) = "<first route tree>" } if (allow_source_address("<second route tree>")) { $var(orig) = "<second route tree>" } <etc., etc. for each route tree I have> if $var(orig)=="unauth" { sl_send_reply("403", "Unauthorized IP"); exit; } else { ....process the call..... What I'd like to see is a permissions function that would allow me to do something like: $var(orig) = allow_source_address_group() if $var(orig)=="-1" { sl_send_reply("403", "Unauthorized IP"); exit; } else { ....process the call..... Where the proposed allow_source_address_group() function would search all IPs in the address table and return either a -1 if no matching address is found, or return the value of the group column if one is found. Anyone out there willing to make these changes? If so, please contact me. I'm willing to put a bounty on this. Thanks, Daryl G. Jurbala Director of Network Operations Global Convergence Solutions Tel: +1 732 853 0513

1 0

[Serusers] ACK Problem : Not routed to NAT'd port
by Ricardo Martinez 15 Nov '07

15 Nov '07

Hello list. I'm facing a problem with my SER configuration and a call to a NAT'd endpoint. When I call a NAT'd endpoint registered in my SER proxy, the call is established, but the last ACK message is not properly routed by SER. You can see this in the next messsage flow. 100.100.100.50 : Callee 100.100.100.99 : SER 100.100.100.231 : Called U 100.100.100.50:5060 -> 100.100.100.99:5060 INVITE sip:008991000404@sip.mydomain.com SIP/2.0. Via: SIP/2.0/UDP 100.100.100.50;rport;branch=z9hG4bK404c94810000005a43b2b7ee000053dc0000010f. Contact: <sip:551100000121@100.100.100.50;user=phone>. U 100.100.100.99:5060 -> 100.100.100.50:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 100.100.100.50;rport=5060;branch=z9hG4bK404c94810000005a43b2b7ee000053dc0000 010f. U 100.100.100.99:5060 -> 100.100.100.231:62447 INVITE sip:551100000684@100.100.100.231;user=phone SIP/2.0. Record-Route: <sip:100.100.100.99;ftag=979715625143;lr=on>. Via: SIP/2.0/UDP 100.100.100.99;branch=z9hG4bKe4c7.150a3f81.0. Via: SIP/2.0/UDP 100.100.100.50;rport=5060;branch=z9hG4bK404c94810000005a43b2b7ee000053dc0000 010f. Contact: <sip:551100000121@100.100.100.50;user=phone>. U 100.100.100.231:62447 -> 100.100.100.99:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 100.100.100.99;branch=z9hG4bKe4c7.150a3f81.0,SIP/2.0/UDP 100.100.100.50;rport=5060;branch=z9hG4bK404c94810000005a43b2b7ee000053dc0000 010f. (The call is answered) U 100.100.100.231:62447 -> 100.100.100.99:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 100.100.100.99;branch=z9hG4bKe4c7.150a3f81.0,SIP/2.0/UDP 100.100.100.50;rport=5060;branch=z9hG4bK404c94810000005a43b2b7ee000053dc0000 010f. Contact: <sip:551100000684@100.100.100.231;user=phone>. U 100.100.100.99:5060 -> 100.100.100.50:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 100.100.100.50;rport=5060;branch=z9hG4bK404c94810000005a43b2b7ee000053dc0000 010f. Contact: <sip:551100000684@100.100.100.231;user=phone>. U 100.100.100.50:5060 -> 100.100.100.99:5060 ACK sip:551100000684@100.100.100.231;user=phone SIP/2.0. Via: SIP/2.0/UDP 100.100.100.50;rport;branch=z9hG4bK404c94810000005a43b2b7f00000182a00000112. Route: <sip:100.100.100.99;ftag=979715625143;lr=on>. U 100.100.100.99:5060 -> 100.100.100.231:5060 ACK sip:551100000684@100.100.100.231;user=phone SIP/2.0. Record-Route: <sip:100.100.100.99;ftag=979715625143;lr=on>. Via: SIP/2.0/UDP 100.100.100.99;branch=0. Via: SIP/2.0/UDP 100.100.100.50;rport=5060;branch=z9hG4bK404c94810000005a43b2b7f00000182a0000 0112. U 100.100.100.231:62447 -> 100.100.100.99:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 100.100.100.99;branch=z9hG4bKe4c7.150a3f81.0,SIP/2.0/UDP 100.100.100.50;rport=5060;branch=z9hG4bK404c94810000005a43b2b7ee000053dc0000 010f. Contact: <sip:551100000684@100.100.100.231;user=phone>. U 100.100.100.99:5060 -> 100.100.100.50:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 100.100.100.50;rport=5060;branch=z9hG4bK404c94810000005a43b2b7ee000053dc0000 010f. Contact: <sip:551100000684@100.100.100.231;user=phone>. U 100.100.100.50:5060 -> 100.100.100.99:5060 ACK sip:551100000684@100.100.100.231;user=phone SIP/2.0. Via: SIP/2.0/UDP 100.100.100.50;rport;branch=z9hG4bK404c94810000005a43b2b7f000001d6e00000114. Route: <sip:100.100.100.99;ftag=979715625143;lr=on>. . U 100.100.100.99:5060 -> 100.100.100.231:5060 ACK sip:551100000684@100.100.100.231;user=phone SIP/2.0. Record-Route: <sip:100.100.100.99;ftag=979715625143;lr=on>. Via: SIP/2.0/UDP 100.100.100.99;branch=0. Via: SIP/2.0/UDP 100.100.100.50;rport=5060;branch=z9hG4bK404c94810000005a43b2b7f000001d6e0000 0114. (.........) (Timeout and the call is released) U 100.100.100.231:62447 -> 100.100.100.99:5060 BYE sip:551100000121@100.100.100.50;user=phone SIP/2.0. Via: SIP/2.0/UDP 192.168.1.20;rport;branch=z9hG4bKc0a801140000002643b2a9be000036e10000001f. Route: <sip:100.100.100.99;ftag=979715625143;lr=on>. . U 100.100.100.99:5060 -> 100.100.100.50:5060 BYE sip:551100000121@100.100.100.50;user=phone SIP/2.0. Record-Route: <sip:100.100.100.99;ftag=93794163011;lr=on>. Via: SIP/2.0/UDP 100.100.100.99;branch=z9hG4bK15c7.924c3535.0. Via: SIP/2.0/UDP 192.168.1.20;received=100.100.100.231;rport=62447;branch=z9hG4bKc0a801140000 002643b2a9be000036e10000001f. U 100.100.100.50:5060 -> 100.100.100.99:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 100.100.100.99;branch=z9hG4bK15c7.924c3535.0,SIP/2.0/UDP 192.168.1.20;rport=62447;received=100.100.100.231;branch=z9hG4bKc0a801140000 002643b2a9be000036e10000001f. U 100.100.100.99:5060 -> 100.100.100.231:62447 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.1.20;rport=62447;received=100.100.100.231;branch=z9hG4bKc0a801140000 002643b2a9be000036e10000001f. I don't understand why the the ACK message is routed to port 5060 and not to the port 62447 like the INVITE or the final OK. Is this maybe a configuration problem?. My ser.cfg is based on the onsip.org configuration documents plus some modifications by me. Can someone help me here Thanks in advance Ricardo Martinez.-

3 5

[OpenSER-Users] Illegal characteres in SIP URI?
by Iñaki Baz Castillo 15 Nov '07

15 Nov '07

Hi, could somebody tell me which characteres are allowd in a SIP URI? which RFC defines it? I ask it because I do a DB query with $ru so a SQL injection it's possible if RURI contains single or double ' So at the begining of the script I'd like to reject a message if the RURI contains illegal symbols. Could be useful a core function for this? Regards. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

2 3

Re: [OpenSER-Users] Aliases table and db aliases
by Iñaki Baz Castillo 15 Nov '07

15 Nov '07

El Thursday 15 November 2007 16:08:27 escribió: > *Hi,* Better replies to the mailist ;) > Iñaki Baz Castillo wrote: > > * alias with DB_ALIASES module: > > > > table dbaliases: > > alias_username: 1234 > > alias_domain: IP_PROXY > > username: bob > > domain: IP_PROXY > > > > db_aliases_lookup() will rewrite URI when it's 1234@IP_PROXY and it'll > > become bob@IP_PROXY > > > > Later lookup("location") will look for the contact of that AOR. > > *Ok. > db_aliases talble is filled in by myself through "openserctl db_alias > add" command.* > > > * alias with ALIASES table: > > > > table aliases: > > username: 1234 > > domain: IP_PROXY > > contact: sip:bob@IP_PROXY > > > > lookup("aliases") will rewrite URI when it's 1234@IP_PROXY and it'll > > become bob@IP_PROXY > > > > Later lookup("location") will look for the contact of that AOR. > > *so, is ALIASES table also created by myself through "openserctl" > command*s ? No idea. In fact I fill these tables via SQL. > > ok now? > > *ok, thank you very much. > daniel > * Regards. -- ilimit... *Iñaki Baz Castillo* ibc(a)in.ilimit.es ÀREA SISTEMES 0034 937 333 375 VOLTA 1, PIS 5 08224 TERRASSA.BCN Aquest enviament és confidencial i està destinat únicament a la persona a qui s'ha enviat. Pot contenir informació privada sotmesa al secret professional, la distribució de la qual està prohibida per la legislació vigent.

1 0

Re: [OpenSER-Users] Aliases table and db aliases
by Daniel 15 Nov '07

15 Nov '07

Thanks. But I don't understand the use of aliases table. I have defined some subscriber users, with username (like BOB) and password (BOBPWD). So, if I wanted to access the system, I would send a REGISTER message with correct username and password (authorization required). Then my proxy address will be "sip:BOB@IP_PROXY" and my Contact will be "sip:BOB@MY _IP_ADRRESS". These information are saved into USER LOCATION table (save("location") function). Now, through DB_aliases module and dbaliases table, I can define some aliases for BOB (like "sip:1234@IP_PROXY", etc...), so if an INVITE for bob has "sip:1234@IP_PROXY" as R-URI, with db_aliases_lookup() function I can rewrite R-URI with Bob's URI ("bob@IP_PROXY") and then call the lookup("location") function. But I don't understand how I can use the ALIASES table. Are there any example explaining its use ? Sorry. Regards, daniel Iñaki Baz Castillo wrote: El Thursday 15 November 2007 14:31:52 Daniel escribi�: Hi all, what is the different between Aliases table and DB aliases table? What is their use ? "aliases" table is same as "location" table but it should be used to look for alias with the function lookup("aliases"). Of course, aliases are simpler than the complex location or aliases tables that manage more info about users, so in order to implement alias system in a simpler way the module ALIAS_DB exists and its table dbaliases. Note that dbaliases table is much simpler (it's just valid for aliases), Look at the module doc: http://www.openser.org/docs/modules/1.3.x/alias_db.html -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

2 1

[Serusers] Implicit Registration Set
by Magnus Sörman 15 Nov '07

15 Nov '07

Hi, How do you/could you configure SER to use implicit registration set? If e.g. a user with a sip URI registers, then both that address and his tel-uri should be registered. Thanks, /Magnus Sorman

1 0

[OpenSER-Users] Aliases table and db aliases
by Daniel 15 Nov '07

15 Nov '07

Hi all, what is the different between Aliases table and DB aliases table? What is their use ? regards, daniel -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

2 1

[OpenSER-Users] Presence module help
by Stanisław Pitucha 15 Nov '07

15 Nov '07

Hello Can someone tell me couple of details about presence code? I.e.: 1. There's PUBL_TYPE and WINFO_TYPE define. They are used as flags, but then they are just marking winfo / other packages. What's the effect of PUBL_TYPE on event package exactly? 2. What is the meaning of 'get_rules_doc', 'apply_auth_nbody', 'get_auth_status' funcs? Unfortunately I wasn't able figure it out from source completely. Thanks Stanisław Pitucha

2 2

[OpenSER-Users] OpenSER with multidomain configuration
by satish patel 15 Nov '07

15 Nov '07

Dear all I am going to install openser with multidomain support do anyone have sample .cfg file or configuration guide i have find many document but it is very hard to implement so just suggest me how to test and configure it ----PGP Signature-- Satish Patel mobile:- +91-9818875535 http://www.linuxbug.org --------------------------------- Never miss a thing. Make Yahoo your homepage.

2 1

[Serusers] SER Dispatcher Module
by Arun Kumar 15 Nov '07

15 Nov '07

Hi Here is my setup Users ----> SER ----> Asterisk ----> PSTN works fine. when ever user dial able to hear balance message and calls goes fine. Just now I replaced my setup to use Dispatcher module Users -----> SER (Dispatcher module) ----------> Asterisk Boxes -----> PSTN when user makes calls not able to hear balance message all the times but able to make PSTN calls ? In my config I just changed replace IP of my asterisk box with dispatcher config: Old config: strip(1); rewritehostpost("x.x.x.x:5060"); route(4); NAT Detet. New conifg: strip(1); ds_select_dst("1", "0"); forward(uri:host, uri:port); route(4); dispatcher file: 1 sip:x.x.x.x:5060 1 sip.y.y.y.y:5060 Need help on above. Thanks, Arun

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[Serusers] DTMF Problem
by Arun Kumar 15 Nov '07

15 Nov '07

Hi Here is my setup: USER ------> PSTN -----> Asterisk A ----> IAX2 Trunk ----> Asterisk B ---------> SER --------> Asterisk C I'm not able to receive DTMF passed by USER on Asterisk C. All my asterisk boxs are configured with same DTMF type (auto) but no luck. Please help on this issue. Thanks, Arun

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[OpenSER-Users] MyISAM or InnoDB in MySQL tables?
by Iñaki Baz Castillo 15 Nov '07

15 Nov '07

Hi, MySQL tables of OpenSer created by the script are "MyISAM". Is there any reason for that instead of using InnoDB? As I know: MyISAM: - Is faster reading and writting (except writting various entries to same table since it blocks the entire table). - No transactions or rollbacks. - Small disc space requeriments. InnoDB: - Transactions, rollbacks, restrictions in foreing keys. - Faster writting various entries in same table (entry lock instead of table lock). - More disc space and RAM requeriments. Yes, it can be obvious that for OpenSer the engine MyISAM can be a good choice but I would like to know if there is more data for this choice. Regards. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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Re: [Serusers] Voicemail setup on SER v0.9.6
by roayu 15 Nov '07

15 Nov '07

Thanks Stefan! I need to confirm that after the installation, I will be having 2 SERs + 1 SEMS, rite? ( 1 SER for normal sip, 1 SER as SIP stack, 1 SEMS) Thanks. Roa Yu -----Original Message----- From: Stefan Sayer [mailto:stefan.sayer@iptego.de] Sent: Tuesday, November 13, 2007 6:27 PM To: roayu Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] Voicemail setup on SER v0.9.6 Hello, roayu wrote: > Which version of SEMS should I download? Is it included the SER as well? > I would prefer a stable version =) I would recommend the stable SEMS 0.10.0 release. Bundled with its SER its available at ftp://ftp.iptel.org/pub/sems/sems-0.10.0-bundle.tar.gz You will find all information at http://iptel.org/sems. Regards Stefan Sayer > > Thanks. > > Roa Yu > > -----Original Message----- > From: SIP [mailto:sip@arcdiv.com] > Sent: Monday, November 12, 2007 10:53 AM > To: roayu > Subject: Re: [Serusers] Voicemail setup on SER v0.9.6 > > You CAN have those on the same box. Although it's not required. You > could have one SER on one box and then the other SER/SEMS on another box > of you preferred. > > > > roayu wrote: >> Hi Eneref, >> >> Thanks for your reply. So, do u mean that I'm gonna have the follwing >> applications in the SAME box? >> - 2 SER servers ( 1 existing SER + 1 new SER for SEMS ) with different > port >> - 1 SEMS >> >> Thanks. >> >> Roa Yu >> >> >> -----Original Message----- >> From: Eneref [mailto:eneref@arcdiv.com] >> Sent: Friday, November 09, 2007 4:14 PM >> To: roayu >> Subject: Re: [Serusers] Voicemail setup on SER v0.9.6 >> >> Yes. You need either SEMS or Asterisk or some such. Meshing SEMS with >> SER, though it LOOKS complex, is really not that bad. >> >> The easiest thing to do is to run another SER server to handle voicemail >> (on a different port from the first SER if it's on the same box) and >> forward messages to it if voicemail is required (on time outs, or >> whatever conditions you set). There are some good instructions on how >> to do that in the SEMS docs. >> >> This also makes it quite easy to swap out with Asterisk or any other >> sort of voicemail-capable server should the need arise in future. >> >> N. >> >> roayu wrote: >> >>> Hi there! >>> >>> >>> >>> Could anyone let me know how to setup Voicemail on SER v.0.9.6 ? Do I >>> need to install SEMS as well? >>> >>> >>> >>> Thanks >>> >>> >>> >>> >>> >>> Regards, >>> >>> Roa Yu >>> >>> ------------------------------------------------------------------------ >>> >>> _______________________________________________ >>> Serusers mailing list >>> Serusers(a)lists.iptel.org >>> http://lists.iptel.org/mailman/listinfo/serusers >>> >>> >> _______________________________________________ >> Serusers mailing list >> Serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >> > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers -- Stefan Sayer VoIP Services stefan.sayer(a)iptego.de www.iptego.de iptego GmbH Am Borsigturm 40 13507 Berlin Germany Amtsgericht Charlottenburg, HRB 101010 Geschaeftsfuehrer: Alexander Hoffmann

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[OpenSER-Users] pua_usrloc
by Venter, Andries <andries＠sun.ac.za> 15 Nov '07

15 Nov '07

listen=udp:196.211.9.222 port=5065 loadmodule "presence.so" loadmodule "avpops.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "mi_fifo.so" loadmodule "uri.so" loadmodule "uri_db.so" loadmodule "nathelper.so" loadmodule "textops.so" loadmodule "acc.so" loadmodule "domain.so" loadmodule "pua.so" loadmodule "pua_mi.so" loadmodule "pua_usrloc.so" modparam("usrloc|auth_db|uri_db|acc", "db_url", mysql://openser:password@localhost/openser") modparam("presence|pua", "db_url", "mysql://openser:password@localhost/openser") modparam("pua_usrloc", "default_domain", "196.211.9.222") modparam("presence", "force_active", 1) modparam("presence", "max_expires", 3600) modparam("presence", "server_address", "sip:196.211.9.222:5065") if( is_method("PUBLISH|SUBSCRIBE")) { route(4); } if (method=="REGISTER") { xlog("L_INFO", "Kyk of ek moet PUBLISH........................\n"); if(!search("^User-Agent: X-Lite")) { xlog("L_INFO", "JA - ek moet PUBLISH........................\n"); pua_set_publish(); } save("location"); exit; }; # presence handling route route[4] { xlog("L_INFO","Route 4........................................\n"); sl_send_reply("100","trying"); # absorb retransmissions if (! t_newtran()) { xlog("L_INFO", "Not new transaction.................................\n"); sl_reply_error(); exit; }; # append_to_reply("Contact: <sip:196.211.9.222:5065>\r\n"); if(is_method("PUBLISH")) { xlog("L_INFO", "Handling PUBLISH.................................\n"); handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { xlog("L_INFO", "Handling SUBSCRIBE.................................\n"); handle_subscribe(); t_release(); }; exit; }

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Re: [OpenSER-Users] pua_usrloc
by Venter, Andries <andries＠sun.ac.za> 15 Nov '07

15 Nov '07

My config is as follows listen=udp:196.211.9.222 port=5065 loadmodule "presence.so" loadmodule "avpops.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "mi_fifo.so" loadmodule "uri.so" loadmodule "uri_db.so" loadmodule "nathelper.so" loadmodule "textops.so" loadmodule "acc.so" loadmodule "domain.so" loadmodule "pua.so" loadmodule "pua_mi.so" loadmodule "pua_usrloc.so" modparam("usrloc|auth_db|uri_db|acc", "db_url", mysql://openser:password@localhost/openser") modparam("presence|pua", "db_url", "mysql://openser:password@localhost/openser") modparam("pua_usrloc", "default_domain", "196.211.9.222") modparam("presence", "force_active", 1) modparam("presence", "max_expires", 3600) modparam("presence", "server_address", "sip:196.211.9.222:5065") if( is_method("PUBLISH|SUBSCRIBE")) { route(4); } if (method=="REGISTER") { xlog("L_INFO", "Kyk of ek moet PUBLISH........................\n"); if(!search("^User-Agent: X-Lite")) { xlog("L_INFO", "JA - ek moet PUBLISH........................\n"); pua_set_publish(); } save("location"); exit; }; # presence handling route route[4] { xlog("L_INFO","Route 4........................................\n"); sl_send_reply("100","trying"); # absorb retransmissions if (! t_newtran()) { xlog("L_INFO", "Not new transaction.................................\n"); sl_reply_error(); exit; }; # append_to_reply("Contact: <sip:196.211.9.222:5065>\r\n"); if(is_method("PUBLISH")) { xlog("L_INFO", "Handling PUBLISH.................................\n"); handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { xlog("L_INFO", "Handling SUBSCRIBE.................................\n"); handle_subscribe(); t_release(); }; exit; }

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[OpenSER-Users] pua_usrloc
by Venter, Andries <andries＠sun.ac.za> 15 Nov '07

15 Nov '07

I would like to get presence published for all phones. I call pua_set_publish() before doing save ("location"), but no PUBLISH messages are generated when the REGISTER request comes through. I am using OpenSER 1.2.2. Any ideas?

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[Serusers] tcp branching errors
by Tomasz Zieleniewski 15 Nov '07

15 Nov '07

Hi, I have the following situation. I have two ser rel_2_0_0 instances one acting as a boundary proxy the other on as internal proxy (registrar, location). I have the following situation I have a user which has registered two contacs one woth tcp transport the other one with udp. All messages coming to and outgoing from my network travels through boundary sip proxy. When internal ser perform lookup_location it get to contacts and the destination set contains two records: Nov 15 11:03:13 sen ser[3673]: route[FORWARD]: destination set: Contact: sip:xlite@192.168.0.164:1221;rinstance=d35f9cda6bd79b0a;transport=TCP, sip:xlite@192.168.0.164:7060 Then internal ser forward message to boundary SIP proxy by changing the destination of the message and with the usage of t_relay(). I see from log that there are two branches involved: Nov 15 11:03:13 sen ser[3673]: branch_route[BRANCH_ROUTE]: <INVITE> ruri:<sip:xlite@192.168.0.164:1221;rinstance=d35f9cda6bd79b0a;transport=TCP> t:<sip:00@voip.rd.touk.pl> f:<sip:lkl@voip.rd.touk.pl> Nov 15 11:03:13 sen ser[3673]: branch_route[BRANCH_ROUTE]: append_hf(P-Warn: forked branch) Nov 15 11:03:13 sen ser[3673]: branch_route[BRANCH_ROUTE]: <INVITE> ruri:<sip:xlite@192.168.0.164:7060;rinstance=cfca8523e3747cb0> t:<sip:00@voip.rd.touk.pl> f:<sip:lkl@voip.rd.touk.pl> Nov 15 11:03:13 sen ser[3673]: branch_route[BRANCH_ROUTE]: append_hf(P-Warn: forked branch) both INVITES reaches boudary SIP proxy. Because first contact was unreachable there was an tcp error: ser[3516]: ERROR: tcp_blocking_connect: poll error: flags 18 Nov 15 11:03:13 sen ser[3516]: ERROR: tcp_blocking_connect: SO_ERROR (111) Connection refused Nov 15 11:03:13 sen ser[3516]: ERROR: tcpconn_connect: tcp_blocking_connect failed Nov 15 11:03:13 sen ser[3516]: ERROR: tcp_send: connect failed Nov 15 11:03:13 sen ser[3516]: msg_send: ERROR: tcp_send failed Nov 15 11:03:13 sen ser[3516]: ERROR: t_send_branch: sending request on branch 0 failed after which ser invokes sl_reply_error() which results in 477 next hop error reponse message sent to internal ser instance. The problem arises when boudary sip proxy processes second INVITE which correspond to second destination. when boundary SIP proxy receives it doesn't send it to user agent but instead reponses to internal ser with 500 I'm terribly sorry, server error occurred (1/SL) the only thing I see in the boundary SIP proxy log is: ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) Did I miss something here or this is an error? Waiting for Your feedback Best regards Tomasz

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[OpenSER-Users] OpenSER CVS
by Bogdan-Andrei Iancu 15 Nov '07

15 Nov '07

Hi, If there are good arguments against, Wednesday 14th, 12:00 AM GMT, the CVS repository from SF will be disabled. Regards, Bogdan

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[OpenSER-Users] RPID or PAI just for calls to PSTN?
by Iñaki Baz Castillo 15 Nov '07

15 Nov '07

Hi, I'll add RPID or PAI header in calls to PSTN gateway, but the fact is that I add those headers after user Authentication, even if it's a local call to other domain user and so. Could be problematic in any way to insert those headers when they are not needed? Of course, if 200@domain INVITE's 201@domain and during auth OpenSer add "RPID=+0034123123123..." I want 201 sees 200 as callerid. Maybe some UAS use received RPID or PAI as callerid? maybe I should add those headers just if the call goes to PSTN gateway? Thanks. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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[OpenSER-Users] Preserving domain name on route()
by Daniel Corbe 15 Nov '07

15 Nov '07

So, externally I'm using DNS SRV records to get traffic to my proxy server. Once it hits my proxy server though, I want to preserve the domain name portion of the URI, but still route the traffic around from server to server. Currently I'm using rewritehostport to forward traffic onto another box. This of course changes the field I'm interested in preserving. Is there another way to route traffic around? The only other option I see is to use 1 set of SRV records externally and another set of SRV records internally. Thanks. -Daniel

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[OpenSER-Users] [OT] Nortel CS2000 and "RPID" / "P-Asserted-Identity"
by Iñaki Baz Castillo 15 Nov '07

15 Nov '07

Hi, I need to send SIP traffic from my OpenSer to a Nortel CS2000 (SIP_2_PSTN gateway). I have a bad new: the provider tells me that they set the callerid with the "From" I send. I've asked them to support "RPID" or "P-Asserted-Identity" headers (still waiting for a reply). In case I can only use "From" I can modify it with UAC module, but I also know this is very anti-RFC3261 behaviour. Does anyone know if Nortel CS2000 can use "RPID" or "P-Asserted-Identity" to set the PSTN calling number? What do you think about changing From header? for example: 200(a)domain.com -> 00123223223234@IP Thanks for any response. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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[OpenSER-Users] Permissions Modile - BYE handling
by Douglas Garstang 15 Nov '07

15 Nov '07

So, I've been playing around with the permissions module. It's working ok, except... I am sending calls from Asterisk to OpenSER, and then onto an ITSP. Calls only flow in this outbound direction. There are NO inbound calls. I put the IP addresses of the Asterisk servers into the address table and used allow_source_address() to confirm the calls where allowed. However, I found that I also had to put the IP's of the ITSP's in the address table or otherwise when the remote end hangs up and sends a BYE message the call to allow_source_address() returns false. This isn't that big a deal, but it seems messy. It also means that the ITSP, being in the address list, could send calls to us, and we don't want that. How do I work around this? Do I do: if ( allow_source_address("1") || method=="BYE" ) { .... } What about CANCEL? I haven't tried that. The same thing goes for putting IP's in the trusted table and using if_trusted(). Douglas. ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs

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[OpenSER-Users] Exported Statistics
by Daniel Corbe 14 Nov '07

14 Nov '07

Hello, Several modules have "exported statistics" which may be extremely useful to me in monitoring my network. The module documentation isn't clear on how to get access to these statistics. Can anyone tell me how I would probe my OpenSER box for this information? Thanks. -Daniel

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Re: [OpenSER-Users] Problem with RINSTANCE in X-LITE 3.0
by Thiago Maluf 14 Nov '07

14 Nov '07

Hi Mr. Castillo, I tried it, but don't work. My problem is: - Each Register created by X-LITE 3.0 is a new contact because each message have parameter RINSTANCE with new value. - So OpenSER will keep in your location table all the contacts received. - I want break it, so for each Register Message Openser will update this contact in your location table. My location table (Replace My Ip Number to xxxxxxxxxxxx): id | username | domain | contact | received | path | expires | q | callid | cseq | last_modified | flags | cflags | user_agent | socket | methods ----+----------+--------+-------------------------------------------------------------+----------+------+---------------------+------+----------------------------------------------+------+---------------------+-------+--------+----------------------------------+--------------------------+--------- 36 | teste1 | | sip:teste1@xxxxxxxxx:27420;rinstance=457f14a9ad568c74 | | | 2007-11-14 16:54:05 | 0.00 | YjY5YmFkZDRkZjBmOTliOWFiNmY2ODViMmRmMTg3ZjQ. | 2 | 2007-11-14 15:54:05 | 0 | 0 | X-Lite release 1011s stamp 41150 | udp:xxxxxxxxxxxx:5060 | 5951 37 | teste1 | | sip:teste1@xxxxxxxxxxxx:9150;rinstance=d491e76ae7d17c63 | | | 2007-11-14 17:01:09 | 0.00 | OTdiNDM4Y2M4NWEzMjIwMmU5MWE1Y2E5M2E3N2VkNjY. | 2 | 2007-11-14 16:01:09 | 0 | 0 | X-Lite release 1011s stamp 41150 | udp:xxxxxxxxxxxx:5060 | 5951 38 | teste1 | | sip:teste1@xxxxxxxxxxxxx:32448;rinstance=a88d552b74e3ebbc | | | 2007-11-14 17:07:13 | 0.00 | ZTE4YTE4OTVlOGJjYTExM2IxM2RiZjFlOGMyNzI3Nzk. | 2 | 2007-11-14 16:07:13 | 0 | 0 | X-Lite release 1011s stamp 41150 | udp:xxxxxxxxxxxx:5060 | 5951 Then this problem is just with X-LITE 3.0 Already talked with CounterPath' Support and this error is knowledge them and in the next version will be removed. But still in demo this new version, and will need long time to be released. Wait your help. Thiago! -------------------------------------------------------------------------------------------- El Wednesday 14 November 2007 14:55:00 Thiago Maluf escribi?: > Hi list, > The X-Lite 3.0 create one new rinstance to each REGISTER Message. > Therefore the next register will be one new contact in location table. > Because this new register have new contact. > So it's my problem. I should in soon config my OpenSER to just allow one > register by user. And with this problem, I won't config it. > Thanks in Advanced, Ok, then you can use: modparam("registrar", "method_filtering", 1) so just contacts that "Allow" the actual method (INVITE, MESSAGE or whatever) will be called. For this you should, in each REGISTER, do a DB query (or FIFO command) to test if there was a previous entry for this location, and in that case modify the "methods" field of new entry: http://openser.svn.sourceforge.net/viewvc/openser/trunk/parser/msg_parser.h… -- I?aki Baz Castillo ibc(a)in.ilimit.es -- ---------------------------------------------------------------- THIAGO MALUF RESENDE e-mail: malufrj(a)gmail.com

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[OpenSER-Users] Problem with RINSTANCE in X-LITE 3.0
by Thiago Maluf 14 Nov '07

14 Nov '07

Hi list, The X-Lite 3.0 create one new rinstance to each REGISTER Message. Therefore the next register will be one new contact in location table. Because this new register have new contact. So it's my problem. I should in soon config my OpenSER to just allow one register by user. And with this problem, I won't config it. Thanks in Advanced, Thiago ---------------------------------------------------------------- THIAGO MALUF RESENDE Voip Developer e-mail: malufrj(a)gmail.com

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[OpenSER-Users] Clients behind the same NAT
by Daniel 14 Nov '07

14 Nov '07

HI all, I've read the solutions proposed at http://www.openser.org/dokuwiki/doku.php/examples:caller-callee-behind-same… 2 client behind the same NAT. The clients may be behind the same NAT, but on different sub-net (for example, one on 192.168.10.x and another on 192.168.2.x). In this case, checking source and destination IP, to discriminate against behind or not the same NAT, is not enough - they can't reach each othere anyway. What do you think? Regards, daniel -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

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[OpenSER-Users] $ruri PV missing from pseudovariables wiki page
by Ovidiu Sas 14 Nov '07

14 Nov '07

Hi all, It seems that somehow the description for $ruri PV is missing from the pseudovariables wiki page. Regards, Ovidiu Sas

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[OpenSER-Users] presence module - wrong Subscription-State when doing un-subscribe ?
by Pascal Maugeri 14 Nov '07

14 Nov '07

Hi I'm doing stability testing against presence module and after "several" requests an un-subscribe request (SUBSCRIBE with Expires=0) triggers a NOTIFY with a payload and the following Subscription-State header value: Subscription-State: active;expires=0 In my humble opinion this is incorrect as the RFC 3265 section says in section 3.3.6: Note that the NOTIFY messages triggered by SUBSCRIBE messages with "Expires" headers of 0 will contain a "Subscription-State" value of "terminated", and a "reason" parameter of "timeout". Ref.: http://tools.ietf.org/html/rfc3265#section-3.3.6 Best regards, Pascal

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[OpenSER-Users] Using Carrierroute - Weighted fail over instead of percentage groups...
by Kyle Stone 14 Nov '07

14 Nov '07

I'm wondering if it's possible to have weighted failover groups instead of % balanced groups.. In Example: domain register { prefix 011 { max_targets = 2 target int.domain.com { hash_index = 1 status = 1 comment = "Try me first.." } target int2.domain.com { hash_index = 2 status = 1 comment = "Try me second.." } } } And if the one with the lowest hash_index responds with a 5XX/6XX/Temporary etc error, send the call to the next highest hash_index, before looking for a shorter prefix? Essobi

2 1

[OpenSER-Users] CPL + DBTEXT
by Chris Heiser 13 Nov '07

13 Nov '07

So I'm playing with CPL, and I notice that the dbtext table isn't getting updated. Anyone seen this behavior before? I'm running 1.2.2-tls --Chris

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[OpenSER-Users] Reg. conditional processing of invite
by padmaja.rv＠vodcalabs.com 13 Nov '07

13 Nov '07

HiThanks for the reply, I however, do not know how to implement what you mentioned in your earlier mail. can you detail the steps? One of the openser users has suggested the following: ******************************************* if (is_method('INVITE')) { # check if from URI domain equals 911operator.com if ($fd == '911operator.com') { # relay stateless using request URI forward(); exit; } else { # relay statefull using request URI if (!t_relay()) { sl_reply_error(); }; exit; } } However, the function forward(); does not parse and gives error bad arguments. Can you suggest how to get this code working? I need it working asap. Thanks again, Padmaja Hi Padmaja, There is no other way around but just to implement a call back service on your proxy. Save in DB the last caller hitting the 911 service and when receiving the 8911, just load it and fwd to it. regards, bogdan Padmaja wrote: > Hi all, > > I have an issue, where the sip user dials 911 number and hangs up. The 911 > operator then calls back to the user using the call back command provided > earlier say 8911 (there is a b2bua that interprets the call back number and > routes the call to the specific user). This number is to be routed via the > openser proxy to the b2bua. However, when the 911 calls back, proxy treats > this number as a username and since there are no registrations to this 8911 > number (though this entry is present at the proxy, no device has registered > with this number), it sends back "404 Not found". How can this be > circumvented? > > Thanks, > Padmaja

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[OpenSER-Users] Drop reply
by Alex Hermann 13 Nov '07

13 Nov '07

Hello all, is it possible to completely drop a reply from failure_route? I know it is possible to change the reply with t_reply, but I need the possibility to send nothing back to the UAC. -- Greetings, Alex Hermann

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[OpenSER-Users] carrierroute Error
by Mark Baker 13 Nov '07

13 Nov '07

I am attempting to use carrierroute for the first time, and the problem that I am having is that when I attempt to place a call, I see the following in stderror: /usr/local/sbin/openser[32664]: New request - M=INVITE RURI= sip:1678XXXXXXX@domain.com F=sip:1404XXXXXXX@domain.com T= sip:1678XXXXXXX@domain.com IP=204.XXX.XXX.XXX ID=133DAB62-97D8-4A2C-8D68-BAA4EC4B310C(a)192.168.5.147 /usr/local/sbin/openser[32664]: Callee is not local - M=INVITE RURI= sip:1678XXXXXXX@domain.com F=sip:1404XXXXXXX@domain.com T= sip:1678XXXXXXX@domain.com IP=204.XXX.XXX.XXX ID=133DAB62-97D8-4A2C-8D68-BAA4EC4B310C(a)192.168.5.147 /usr/local/sbin/openser[32664]: Not normalized callee '1678XXXXXXX' to E.164format - M=INVITE RURI= sip:1678XXXXXXX@domain.com F=sip:1404XXXXXXX@domain.com T= sip:1678XXXXXXX@domain.com IP=204.XXX.XXX.XXX ID=133DAB62-97D8-4A2C-8D68-BAA4EC4B310C(a)192.168.5.147 /usr/local/sbin/openser[32664]: carrierroute:carrier_rewrite_msg: Domain too big. (We only have 5 domains, you wanted 0.) /usr/local/sbin/openser[32664]: Request leaving server, D-URI='<null>' - M=INVITE RURI=sip:1678XXXXXXX@domain.com F=sip:1404XXXXXXX@domain.com T= sip:1678XXXXXXX@domain.com IP=204.XXX.XXX.XXX ID=133DAB62-97D8-4A2C-8D68-BAA4EC4B310C(a)192.168.5.147 /usr/local/sbin/openser[32667]: New request - M=INVITE RURI= sip:1678XXXXXXX@domain.com F=sip:1404XXXXXXX@domain.com T= sip:1678XXXXXXX@domain.com IP=72.XXX.XXX.XXX ID=133DAB62-97D8-4A2C-8D68-BAA4EC4B310C(a)192.168.5.147 /usr/local/sbin/openser[32660]: NAT-Reply - S=100 D=Trying F= sip:1404XXXXXXX@domain.com T=sip:1678XXXXXXX@domain.com IP=72.XXX.XXX.XXX ID=133DAB62-97D8-4A2C-8D68-BAA4EC4B310C(a)192.168.5.147 /usr/local/sbin/openser[32667]: Proxy authentication failed - M=INVITE RURI= sip:1678XXXXXXX@domain.com F=sip:1404XXXXXXX@domain.com T= sip:1678XXXXXXX@domain.com IP=72.XXX.XXX.XXX ID=133DAB62-97D8-4A2C-8D68-BAA4EC4B310C(a)192.168.5.147 /usr/local/sbin/openser[32663]: NAT-Reply - S=407 D=Proxy Authentication Required F=sip:1404XXXXXXX@domain.com T=sip:1678XXXXXXX@domain.com IP= 72.XXX.XXX.XXX ID=133DAB62-97D8-4A2C-8D68-BAA4EC4B310C(a)192.168.5.147 /usr/local/sbin/openser[32663]: Failure route for PSTN entered - M=INVITE RURI=sip:1678XXXXXXX@domain.com F=sip:1404XXXXXXX@domain.com T= sip:1678XXXXXXX@domain.com IP=204.XXX.XXX.XXX ID=133DAB62-97D8-4A2C-8D68-BAA4EC4B310C(a)192.168.5.147 Thanks for any direction!!

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Re: [Serusers] Damaged message body
by Jiri Kuthan 13 Nov '07

13 Nov '07

At 10:55 02/11/2007, Stefan.Brozinski(a)materna.de wrote: >Hello everybody > >there is a bug in SER that removes the last byte from a message body if >this happens to be a binary zero. > >This may be the case when the Content-Type is something binary (i.e. >*not* using "Content-Type: text/something"). > >The bad code is in udp_server.c. It is present in SER 0.9.x a well as in >2.0. I have reported it in the bug tracker as SER-324 and attached a >comment which points out the bad section of the code. > >How shall I proceed to fix this bug? Ideally submit a patch to JIRA...http://tracker.iptel.org/browse/SER -jiri >Best regards >Stefan >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

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Re: [Serusers] R: R: R: Loadbalancing for interco
by Jiri Kuthan 13 Nov '07

13 Nov '07

At 18:48 12/11/2007, Stefano Capitanio wrote: >Yes, it seems really strange also to me. > >I've tried with both SER-0.9.6 and OpenSER-1.2.2 on a Gentoo-linux >And in both cases there is the same problem... > >Do you think that can be a problem of installation/compiling? >Have ever heard something like that? Not really :-( Calling forward can't change contacts -- perhaps you are mistakenly starting SER with other config file than you think? -jiri >-Stefano > >> -----Messaggio originale----- >> Da: Jiri Kuthan [mailto:jiri@iptel.org] >> Inviato: lunedì 12 novembre 2007 18.28 >> A: Stefano Capitanio; serusers(a)lists.iptel.org >> Oggetto: Re: [Serusers] R: R: Loadbalancing for interco >> >> I think there is some confusion here ... SER doesn't change contacts by >> the action referred to >> bellow. >> >> -jiri >> >> At 17:54 12/11/2007, Stefano Capitanio wrote: >> >It is the simplest ser.cfg you can imagine: >> > >> > Route { >> > forward(z.z.z.z,5060); >> > } >> > >> > >> >The message enter in SER (y.y.y.y) as: >> > >> > U x.x.x.x:5060 -> y.y.y.y:5060 >> > REGISTER sip:213.203.128.126 SIP/2.0. >> > Via: SIP/2.0/UDP >> x.x.x.x:5060;rport;branch=z9hG4bKFCDF5014775145F7BBFB33B621BC8BC0. >> > From: openser <sip:123456102@y.y.y.y>;tag=1540538748. >> > To: openser <sip:123456102@y.y.y.y>. >> > Contact: "openser" <sip:123456102@x.x.x.x:5060>. >> > Call-ID: 23FDB6F533AC4BCF840FD35F34B385FD(a)y.y.y.y. >> > CSeq: 60558 REGISTER. >> > Expires: 120. >> > Max-Forwards: 70. >> > User-Agent: X-Lite release 1103m. >> > Content-Length: 0. >> > >> >And goes out as: >> > >> > U y.y.y.y:5060 -> z.z.z.z:5060 >> > REGISTER sip:y.y.y.y SIP/2.0. >> > Via: SIP/2.0/UDP y.y.y.y:5060;branch=0. >> > Via: SIP/2.0/UDP >> 195.110.129.41:5060;rport=5060;branch=z9hG4bKFCDF5014775145F7BBFB33B621BC8 >> BC0. >> > From: openser <sip:123456102@y.y.y.y>;tag=1540538748. >> > To: openser <sip:123456102@y.y.y.y>. >> > Contact: "openser" <sip:123456102@y.y.y.y:5060>. >> > Call-ID: 23FDB6F533AC4BCF840FD35F34B385FD(a)y.y.y.y. >> > CSeq: 60558 REGISTER. >> > Expires: 120. >> > Max-Forwards: 70. >> > User-Agent: X-Lite release 1103m. >> > Content-Length: 0. >> > >> > >> > >> >> -----Messaggio originale----- >> >> Da: Jiri Kuthan [mailto:jiri@iptel.org] >> >> Inviato: lunedì 12 novembre 2007 17.41 >> >> A: Stefano Capitanio; serusers(a)lists.iptel.org >> >> Oggetto: Re: [Serusers] R: Loadbalancing for interco >> >> >> >> I haven't seen your config file, but normally it does not change >> Contacts. >> >> It changes contacts if it is configured to deal with NATs. >> >> >> >> -jiri >> >> >> >> At 17:20 12/11/2007, Stefano Capitanio wrote: >> >> >Ok, thanks for your answer, I understand your position. >> >> > >> >> >Anyway I cannot understand why SER modify the Contact header without >> any >> >> instruction about that in the config file...is there any reason >> concerning >> >> RFC compliance? >> >> > >> >> >Best regards, >> >> >Stefano >> >> > >> >> >> -----Messaggio originale----- >> >> >> Da: Jiri Kuthan [mailto:jiri@iptel.org] >> >> >> Inviato: lunedì 12 novembre 2007 17.09 >> >> >> A: Stefano Capitanio; serusers(a)lists.iptel.org >> >> >> Oggetto: Re: [Serusers] Loadbalancing for interco >> >> >> >> >> >> Well, load-balancing is not easy. To deal with issues like you are >> >> >> describing, >> >> >> your best choice is a load-balancer which is capable of working in >> >> >> transparent >> >> >> mode. We have such in our intelligence, some work, some less so, let >> me >> >> >> know >> >> >> if you need some intelligence on this. >> >> >> >> >> >> -jiri >> >> >> >> >> >> At 12:23 12/11/2007, Stefano Capitanio wrote: >> >> >> >Content-class: urn:content-classes:message >> >> >> >Content-Type: multipart/alternative; >> >> >> > boundary="----_=_NextPart_001_01C8251E.664EA61A" >> >> >> > >> >> >> >Hi, >> >> >> > >> >> >> >we are trying to do the same with our servers but we have some >> problem >> >> >> with registrations: >> >> >> >it seem that when the REGISTER message pass through SER, the host- >> part >> >> of >> >> >> Contact field is modified with the local address of SER. >> >> >> >Is it a misconfiguration problem? >> >> >> >Do you have experience in balancing also the registrations? >> >> >> > >> >> >> >Thanks, >> >> >> >Stefano >> >> >> > >> >> >> > >> >> >> >On Thu, 2007-09-13 at 11:02 +0200, inge wrote: >> >> >> > >> >> >> >> Hi all, >> >> >> > >> >> >> >> >> >> >> > >> >> >> >> Is there a way to have something like a loadbalancing on SER for >> >> >> > >> >> >> >> outgoing calls ? >> >> >> > >> >> >> >> >> >> >> > >> >> >> >> I want to distribute the calls between two gateways. Ideally, >> with a >> >> >> > >> >> >> >> coefficient (ie. 60% for the first and 40% for the second). >> >> >> > >> >> >> >> >> >> >> > >> >> >> >> Thanks for your support. >> >> >> > >> >> >> >> >> >> >> > >> >> >> >> Adrien .L >> >> >> >_______________________________________________ >> >> >> >Serusers mailing list >> >> >> >Serusers(a)lists.iptel.org >> >> >> >http://lists.iptel.org/mailman/listinfo/serusers >> >> >> >> >> >> >> >> >> >> >> >> -- >> >> >> Jiri Kuthan http://iptel.org/~jiri/ >> >> >_______________________________________________ >> >> >Serusers mailing list >> >> >Serusers(a)lists.iptel.org >> >> >http://lists.iptel.org/mailman/listinfo/serusers >> >> >> >> >> >> >> >> -- >> >> Jiri Kuthan http://iptel.org/~jiri/ >> >_______________________________________________ >> >Serusers mailing list >> >Serusers(a)lists.iptel.org >> >http://lists.iptel.org/mailman/listinfo/serusers >> >> >> >> -- >> Jiri Kuthan http://iptel.org/~jiri/ >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

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[OpenSER-Users] Checking Reply to INVITE
by Douglas Garstang 13 Nov '07

13 Nov '07

I need to be able to check the SIP response code to INVITE messages only. I don't care about BYE, CANCEL or anything else. I tried putting a hint header in the INVITE when I sent it out, but the replies came back from the upstream provider with the header removed. I tried setting a flag when an INVITE was received and using that. Didn't work. How can I do this? Is there a way to see what method a response is for? If there is, I could ignore everything that wasn't a response to an INVITE. Thanks, Doug. ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs

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[OpenSER-Users] missing cflags column in User Locations (location) Table definition in wiki
by Piotr Sobolewski 13 Nov '07

13 Nov '07

HI There is one column missing in User Locations (location) Table definition in wiki link below http://www.openser.org/dokuwiki/doku.php/database:1.2.x#user_locations missing column name is cflags. -- Piotr Sobolewski sobolewski(a)gmail.com

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[OpenSER-Users] Number of sessions
by Bruno Machado 13 Nov '07

13 Nov '07

Hi I`d like to know if is possible to use 'openserctl' to find the number of sessions OpenSER is managing. I think that it isn't possible. Is there another way? Thanks. Bruno Machado Abra sua conta no Yahoo! Mail, o único sem limite de espaço para armazenamento! http://br.mail.yahoo.com/

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[OpenSER-Users] xml body parser
by Daniel 13 Nov '07

13 Nov '07

Hi all, Is openser able to parse and interpret a xml script contained, for example, in the body of INVITE method ? If it was possible, what kind of module I'd have to use ? Regards, Daniel -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

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[OpenSER-Users] [Fwd: Re: [OpenSER-Devel] pua module - unsubscribe]
by Anca Vamanu 13 Nov '07

13 Nov '07

-------- Original Message -------- Subject: Re: [OpenSER-Devel] pua module - unsubscribe Date: Tue, 13 Nov 2007 11:57:03 +0200 From: Anca Vamanu <anca(a)voice-system.ro> To: Reinhold Buchinger <reinhold.buchinger(a)gmail.com> References: <473420DF.8020809(a)gmail.com> <47342410.4080307(a)voice-system.ro> <47348B90.8070700(a)gmail.com> <47381EEA.6090702(a)voice-system.ro> <4738CC58.6040105(a)gmail.com> Hello, The succession on checks was wrong. Please test with the latest revision. Thanks and regards, Anca Reinhold Buchinger wrote: > Hi, > > Unfortunately, there is also an unwanted side-effect. If a > subscription is renewed expires is set to -1 for the call-back > parameter (infinite subscribes). In case of an error this also > triggers a new subscription with expires = 0.... > > regards, > Reinhold > > Anca Vamanu schrieb: >> Hello, >> >> It is not about retransmission there, but about the cases in which a >> request is sent inside a dialog stored by pua but which is not >> recognized by the client (the case of a client signing off and no >> Subscribe with expires= 0 sent to pua to announce it that the dialog >> has ended). The logics are like this: if someone asked me to send a >> request (for publish it's the same) and I matched it with a stored >> dialog and it failed, try again with an initial request. This is to >> ensure a a certain reliability. >> Therefore, in that case that you mentioned, when a second attempt is >> made, it is no longer about an unsubscribe. The request sent will be >> an fetching Subscribe(one with expires= 0)- I have made the changes >> to make this case possible. However,even if an unsubscribe was >> desired, the effects are the same, the dialog is already no longer >> valid, the pua record is deleted, and a Notify is sent. >> >> regards, >> Anca Vamanu >> >> Reinhold Buchinger wrote: >>> Hi, >>> >>> Thanks for your answer. But there is still a problem with >>> subs_cback_func(...) if a retransmission is triggered. This message >>> cannot be send with expires=0. >>> >>> subs.expires= (hentity->desired_expires>0)? >>> hentity->desired_expires- (int)time(NULL)+ 10:-1; >>> >>> Regards, >>> Reinhold >>> >>> >>> Anca Vamanu schrieb: >>>> Hello, >>>> >>>> The expires= 0 case is covered in subs_build_hdr(...) : >>>> >>>> if( expires<= min_expires) >>>> subs_expires= int2str(min_expires, &len); else >>>> subs_expires= int2str(expires+ 10, &len); >>>> >>>> if the module parameter min_expires maintains its default value, >>>> which is 0. >>>> I will test to see if indeed it is not possible now to unsubscribe. >>>> >>>> regards, >>>> Anca Vamanu >>>> >>>> Reinhold Buchinger wrote: >>>>> Hi, >>>>> >>>>> Using the pua module, I cannot send a subscription with expires=0. >>>>> If you look at subs_build_hdr(...) in send_subscribe.c this case >>>>> is not covered. >>>>> >>>>> Regards, >>>>> Reinhold >>>>> >>>>> _______________________________________________ >>>>> Devel mailing list >>>>> Devel(a)lists.openser.org >>>>> http://lists.openser.org/cgi-bin/mailman/listinfo/devel >>>>> >>>>> >>>> >>>> >>> >>> >> >> > >

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Re: [Serusers] Voicemail setup on SER v0.9.6
by roayu 13 Nov '07

13 Nov '07

Which version of SEMS should I download? Is it included the SER as well? I would prefer a stable version =) Thanks. Roa Yu -----Original Message----- From: SIP [mailto:sip@arcdiv.com] Sent: Monday, November 12, 2007 10:53 AM To: roayu Subject: Re: [Serusers] Voicemail setup on SER v0.9.6 You CAN have those on the same box. Although it's not required. You could have one SER on one box and then the other SER/SEMS on another box of you preferred. roayu wrote: > Hi Eneref, > > Thanks for your reply. So, do u mean that I'm gonna have the follwing > applications in the SAME box? > - 2 SER servers ( 1 existing SER + 1 new SER for SEMS ) with different port > - 1 SEMS > > Thanks. > > Roa Yu > > > -----Original Message----- > From: Eneref [mailto:eneref@arcdiv.com] > Sent: Friday, November 09, 2007 4:14 PM > To: roayu > Subject: Re: [Serusers] Voicemail setup on SER v0.9.6 > > Yes. You need either SEMS or Asterisk or some such. Meshing SEMS with > SER, though it LOOKS complex, is really not that bad. > > The easiest thing to do is to run another SER server to handle voicemail > (on a different port from the first SER if it's on the same box) and > forward messages to it if voicemail is required (on time outs, or > whatever conditions you set). There are some good instructions on how > to do that in the SEMS docs. > > This also makes it quite easy to swap out with Asterisk or any other > sort of voicemail-capable server should the need arise in future. > > N. > > roayu wrote: > >> Hi there! >> >> >> >> Could anyone let me know how to setup Voicemail on SER v.0.9.6 ? Do I >> need to install SEMS as well? >> >> >> >> Thanks >> >> >> >> >> >> Regards, >> >> Roa Yu >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Serusers mailing list >> Serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >> >> > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

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[OpenSER-Users] how to implement calling between asterisk and asterisk?
by gonghai78 13 Nov '07

13 Nov '07

i am new to openser.my english is poor.my question is: xxxx@123 ---> openser() <---- xxxx@456 | | asterisk 123<----------->asterisk 456 how to implement calling between xxxx@123 and xxxx@456 in openser.cfg? thank you for your help!

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[Serusers] ERROR: warning_builder: buffer size exceeded & WARNING: warning skipped -- too big
by Tomasz Zieleniewski 13 Nov '07

13 Nov '07

Hi, I am using rel_2_0_0 from cvs. I encounter following errors in my ser output log: ERROR: warning_builder: buffer size exceeded WARNING: warning skipped -- too big Please tell what do they mean. Best regerds Tomasz

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Re: [Serusers] R: R: R: Loadbalancing for interco
by Jan Andres 13 Nov '07

13 Nov '07

Hi, On Mon, Nov 12, 2007 at 06:48:20PM +0100, Stefano Capitanio wrote: > Yes, it seems really strange also to me. > > I've tried with both SER-0.9.6 and OpenSER-1.2.2 on a Gentoo-linux > And in both cases there is the same problem... > > Do you think that can be a problem of installation/compiling? > Have ever heard something like that? Just a wild guess -- is there some network component where an application layer gateway (ALG) for SIP is enabled? Maybe the SIP ALG kernel module is loaded on the server itself and is doing the rewriting? Regards, Jan -- Jan Andres <jan.andres(a)freenet.ag> VoIP Systems Engineer phone: +49 431 9020-557 freenet Cityline GmbH, Hamburger Chaussee 2-4, D-24114 Kiel, Germany Ein Unternehmen der freenet AG Amtsgericht Kiel, HRB 6202 Geschäftsführer: Eckhard Spoerr, Axel Krieger --------------------------------------------------------------------- "The question of whether a computer can think is no more interesting than the question of whether a submarine can swim." -- E. W. Dijkstra

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[OpenSER-Users] question about the INVITE method
by 程千 13 Nov '07

13 Nov '07

hi: I isntall the openser-1.2.2-notls under the centos5. I config the INVITE method like this: if (is_method("INVITE")) { # enable Record-Route record_route(); # Is this to one of our numbers if (lookup("location")) { append_hf("P-hint: userloc applied\r\n"); } else { sl_send_reply("404","Not found"); } } route(1); ... route[1] { if (isflagset(2)) { if (is_method("INVITE")) { t_on_reply("1"); t_on_failure("1"); use_media_proxy(); } else if (is_method("BYE|CANCEL")) { end_media_session(); } } # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); } exit; } But when the end ua is not online,the ser proxy seems to be in a loop lock. Thanks a lot!

2 1

[OpenSER-Users] TCP better than UDP??
by lu 13 Nov '07

13 Nov '07

Hi, I came accros one Nokia-S60 Guideline regarding SIP setting in the NAT/Firewall traversal section which say:"It is recommended to use TCP as the transport instead of UDP since even doubled battery life can be achieved with a UI always connected to a SIP service". I need OPenSER users to comment on this since what i know is that UDP is more reliable transportation. WBR lu.

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[OpenSER-Users] ERROR: receive_fd: EOF on 15
by Dave Singer 13 Nov '07

13 Nov '07

ERROR: receive_fd: EOF on 15 child process 26410 exited by a signal 11 core was generated INFO: terminating due to SIGCHLD INFO: signal 15 received .... repeated ... INFO:mi_fifo:mi_destroy: seems that fifo child is already dead! I've added xlog lines to narrow down where it is crashing. It hasn't crashed since that so not sure exactly where it is crashing in the script. I think it is crashing some where around a avp_db_load: if ( avp_db_load("$ru/username","*/usr_preferences_to") ) It doesn't always crash on setting up a call... just when I'm not watching for it. Thanks Dave PS my first time posting to a user list so not sure just how things work for it. Pointers welcome. here is some pert info about my install: version: openser 1.2.1-notls (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP, 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_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. svnrevision: unknown @(#) $Id: main.c 1827 2007-03-12 15:22:53Z bogdan_iancu $ main.c compiled on 11:12:59 Sep 6 2007 with gcc 4.1.1

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[Serusers] Some calls do not go to voicemail
by Steven C. Blair 12 Nov '07

12 Nov '07

We are experiencing a problem for which we can use some help. When a subscriber receives a call from either the PSTN or another IP phone we set the inv_timeout to a value which represents about four rings. For most subscribers an unanswered call will timeout after the specified number of rings then, via a failure route, their call will be re-route to our Asterisk server for voicemail. Some subscribers, no set pattern of users, have their calls timeout and the caller experience silence for about 45 seconds. During this period the caller usually hangs up. I cannot seem to figure out why. We set modparam("tm", "fr_inv_timer_avp", "inv_timeout") and then avp_write("20", "inv_timeout"); before relaying an incoming call. Has anyone else experienced any similar problems? Thanks,Steve Senior Network Engineer, Information Systems and Computing Networking and Telecommunications , Suite 221A /6228 University of Pennsylvania Voice:215-573-8396 FAX:215-898-9348

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[Serusers] SER 0.9.6 "desc_time_order" issue
by Mitya 12 Nov '07

12 Nov '07

Hi Everybody, I would like to set up thefollowing with ser 0.9.6 1) an UAC should be able to register more times with the same q value note: lets say, there is a power outage on the client side and it gets a new IP from the ISP, registers again, the "old" registration has not expired yed, it is still alive, because it did not unreggistered itself 2) when SER calls this UAC, only the most recently registered one should be invited 3) to limit the maximum number of contacts is not possible (I need this feature) From the documentation: ------------------------ ----- CUT ------ 1.3.5. desc_time_order (integer) [...] If set to 1 then all contacts will be ordered in descending modification time order. In this case the most recently updated/created contact will be used. [...] 1.3.3. append_branches (integer) [...] If the parameter is set to 0, only Request-URI will be overwritten with the highest-q rated contact and the rest will be left unprocessed. [...] ----- CUT ------ I also found this thread: ------------------------- http://lists.iptel.org/pipermail/serusers/2003-March/000758.html From Jan: You will have to use modparam("registrar", "append_branches", 0) in your script and after that only the most recently updated/created contact will be used (even if there are many of them). This is exactly what I need !! my relevant config is the following: modparam("usrloc", "db_mode", 1) modparam("registrar", "append_branches", 0) modparam("registrar", "desc_time_order", 1) modparam("registrar", "max_contacts", 50) Sympthom: --------- When I call the UA, always the firstly registered one gets the INVITE, and "desc_time_order" DOES NOT work :( Can sy explain me what should I do now ? Many thanks. Mitya

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[OpenSER-Users] Mediaproxy --help required
by srinivas Antarvedi 12 Nov '07

12 Nov '07

Hello all , i have doubts regarding the mediaproxy do i have to run my openser server on public ip if i need to use the mediaproxy server on the same box as that of openser? -- Srinivas Antarvedi

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Re: [Serusers] R: R: Loadbalancing for interco
by Jiri Kuthan 12 Nov '07

12 Nov '07

I think there is some confusion here ... SER doesn't change contacts by the action referred to bellow. -jiri At 17:54 12/11/2007, Stefano Capitanio wrote: >It is the simplest ser.cfg you can imagine: > > Route { > forward(z.z.z.z,5060); > } > > >The message enter in SER (y.y.y.y) as: > > U x.x.x.x:5060 -> y.y.y.y:5060 > REGISTER sip:213.203.128.126 SIP/2.0. > Via: SIP/2.0/UDP x.x.x.x:5060;rport;branch=z9hG4bKFCDF5014775145F7BBFB33B621BC8BC0. > From: openser <sip:123456102@y.y.y.y>;tag=1540538748. > To: openser <sip:123456102@y.y.y.y>. > Contact: "openser" <sip:123456102@x.x.x.x:5060>. > Call-ID: 23FDB6F533AC4BCF840FD35F34B385FD(a)y.y.y.y. > CSeq: 60558 REGISTER. > Expires: 120. > Max-Forwards: 70. > User-Agent: X-Lite release 1103m. > Content-Length: 0. > >And goes out as: > > U y.y.y.y:5060 -> z.z.z.z:5060 > REGISTER sip:y.y.y.y SIP/2.0. > Via: SIP/2.0/UDP y.y.y.y:5060;branch=0. > Via: SIP/2.0/UDP 195.110.129.41:5060;rport=5060;branch=z9hG4bKFCDF5014775145F7BBFB33B621BC8BC0. > From: openser <sip:123456102@y.y.y.y>;tag=1540538748. > To: openser <sip:123456102@y.y.y.y>. > Contact: "openser" <sip:123456102@y.y.y.y:5060>. > Call-ID: 23FDB6F533AC4BCF840FD35F34B385FD(a)y.y.y.y. > CSeq: 60558 REGISTER. > Expires: 120. > Max-Forwards: 70. > User-Agent: X-Lite release 1103m. > Content-Length: 0. > > > >> -----Messaggio originale----- >> Da: Jiri Kuthan [mailto:jiri@iptel.org] >> Inviato: lunedì 12 novembre 2007 17.41 >> A: Stefano Capitanio; serusers(a)lists.iptel.org >> Oggetto: Re: [Serusers] R: Loadbalancing for interco >> >> I haven't seen your config file, but normally it does not change Contacts. >> It changes contacts if it is configured to deal with NATs. >> >> -jiri >> >> At 17:20 12/11/2007, Stefano Capitanio wrote: >> >Ok, thanks for your answer, I understand your position. >> > >> >Anyway I cannot understand why SER modify the Contact header without any >> instruction about that in the config file...is there any reason concerning >> RFC compliance? >> > >> >Best regards, >> >Stefano >> > >> >> -----Messaggio originale----- >> >> Da: Jiri Kuthan [mailto:jiri@iptel.org] >> >> Inviato: lunedì 12 novembre 2007 17.09 >> >> A: Stefano Capitanio; serusers(a)lists.iptel.org >> >> Oggetto: Re: [Serusers] Loadbalancing for interco >> >> >> >> Well, load-balancing is not easy. To deal with issues like you are >> >> describing, >> >> your best choice is a load-balancer which is capable of working in >> >> transparent >> >> mode. We have such in our intelligence, some work, some less so, let me >> >> know >> >> if you need some intelligence on this. >> >> >> >> -jiri >> >> >> >> At 12:23 12/11/2007, Stefano Capitanio wrote: >> >> >Content-class: urn:content-classes:message >> >> >Content-Type: multipart/alternative; >> >> > boundary="----_=_NextPart_001_01C8251E.664EA61A" >> >> > >> >> >Hi, >> >> > >> >> >we are trying to do the same with our servers but we have some problem >> >> with registrations: >> >> >it seem that when the REGISTER message pass through SER, the host-part >> of >> >> Contact field is modified with the local address of SER. >> >> >Is it a misconfiguration problem? >> >> >Do you have experience in balancing also the registrations? >> >> > >> >> >Thanks, >> >> >Stefano >> >> > >> >> > >> >> >On Thu, 2007-09-13 at 11:02 +0200, inge wrote: >> >> > >> >> >> Hi all, >> >> > >> >> >> >> >> > >> >> >> Is there a way to have something like a loadbalancing on SER for >> >> > >> >> >> outgoing calls ? >> >> > >> >> >> >> >> > >> >> >> I want to distribute the calls between two gateways. Ideally, with a >> >> > >> >> >> coefficient (ie. 60% for the first and 40% for the second). >> >> > >> >> >> >> >> > >> >> >> Thanks for your support. >> >> > >> >> >> >> >> > >> >> >> Adrien .L >> >> >_______________________________________________ >> >> >Serusers mailing list >> >> >Serusers(a)lists.iptel.org >> >> >http://lists.iptel.org/mailman/listinfo/serusers >> >> >> >> >> >> >> >> -- >> >> Jiri Kuthan http://iptel.org/~jiri/ >> >_______________________________________________ >> >Serusers mailing list >> >Serusers(a)lists.iptel.org >> >http://lists.iptel.org/mailman/listinfo/serusers >> >> >> >> -- >> Jiri Kuthan http://iptel.org/~jiri/ >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

2 1

[OpenSER-Users] testing null AVPs
by Ovidiu Sas 12 Nov '07

12 Nov '07

Hi all, How should we test for non existing AVPs? method 1: is_avp_set("$avp(i:123)") method 2: $(avp(i:123)) != null Both of the above methods are working. Is there a difference between them? If no, then maybe is_avp_set should be deprecated. Regards, Ovidiu Sas

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Re: [Serusers] R: Loadbalancing for interco
by Jiri Kuthan 12 Nov '07

12 Nov '07

I haven't seen your config file, but normally it does not change Contacts. It changes contacts if it is configured to deal with NATs. -jiri At 17:20 12/11/2007, Stefano Capitanio wrote: >Ok, thanks for your answer, I understand your position. > >Anyway I cannot understand why SER modify the Contact header without any instruction about that in the config file...is there any reason concerning RFC compliance? > >Best regards, >Stefano > >> -----Messaggio originale----- >> Da: Jiri Kuthan [mailto:jiri@iptel.org] >> Inviato: lunedì 12 novembre 2007 17.09 >> A: Stefano Capitanio; serusers(a)lists.iptel.org >> Oggetto: Re: [Serusers] Loadbalancing for interco >> >> Well, load-balancing is not easy. To deal with issues like you are >> describing, >> your best choice is a load-balancer which is capable of working in >> transparent >> mode. We have such in our intelligence, some work, some less so, let me >> know >> if you need some intelligence on this. >> >> -jiri >> >> At 12:23 12/11/2007, Stefano Capitanio wrote: >> >Content-class: urn:content-classes:message >> >Content-Type: multipart/alternative; >> > boundary="----_=_NextPart_001_01C8251E.664EA61A" >> > >> >Hi, >> > >> >we are trying to do the same with our servers but we have some problem >> with registrations: >> >it seem that when the REGISTER message pass through SER, the host-part of >> Contact field is modified with the local address of SER. >> >Is it a misconfiguration problem? >> >Do you have experience in balancing also the registrations? >> > >> >Thanks, >> >Stefano >> > >> > >> >On Thu, 2007-09-13 at 11:02 +0200, inge wrote: >> > >> >> Hi all, >> > >> >> >> > >> >> Is there a way to have something like a loadbalancing on SER for >> > >> >> outgoing calls ? >> > >> >> >> > >> >> I want to distribute the calls between two gateways. Ideally, with a >> > >> >> coefficient (ie. 60% for the first and 40% for the second). >> > >> >> >> > >> >> Thanks for your support. >> > >> >> >> > >> >> Adrien .L >> >_______________________________________________ >> >Serusers mailing list >> >Serusers(a)lists.iptel.org >> >http://lists.iptel.org/mailman/listinfo/serusers >> >> >> >> -- >> Jiri Kuthan http://iptel.org/~jiri/ >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

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Re: [Serusers] Loadbalancing for interco
by Jiri Kuthan 12 Nov '07

12 Nov '07

Well, load-balancing is not easy. To deal with issues like you are describing, your best choice is a load-balancer which is capable of working in transparent mode. We have such in our intelligence, some work, some less so, let me know if you need some intelligence on this. -jiri At 12:23 12/11/2007, Stefano Capitanio wrote: >Content-class: urn:content-classes:message >Content-Type: multipart/alternative; > boundary="----_=_NextPart_001_01C8251E.664EA61A" > >Hi, > >we are trying to do the same with our servers but we have some problem with registrations: >it seem that when the REGISTER message pass through SER, the host-part of Contact field is modified with the local address of SER. >Is it a misconfiguration problem? >Do you have experience in balancing also the registrations? > >Thanks, >Stefano > > >On Thu, 2007-09-13 at 11:02 +0200, inge wrote: > >> Hi all, > >> > >> Is there a way to have something like a loadbalancing on SER for > >> outgoing calls ? > >> > >> I want to distribute the calls between two gateways. Ideally, with a > >> coefficient (ie. 60% for the first and 40% for the second). > >> > >> Thanks for your support. > >> > >> Adrien .L >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

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[OpenSER-Users] Digest Authentication and nonce count
by Klaus Darilion 12 Nov '07

12 Nov '07

Hi! the ..._challenge() functions support adding the qop=auth to the challenge. This will cause the SIP client to use the nonce count in the digest response. Is this nonce count actually used in openser? I guess this would require stateful handling of nonce and nonce-count. regards klaus

2 2

[Serusers] Loadbalancing for interco
by Stefano Capitanio 12 Nov '07

12 Nov '07

Hi, we are trying to do the same with our servers but we have some problem with registrations: it seem that when the REGISTER message pass through SER, the host-part of Contact field is modified with the local address of SER. Is it a misconfiguration problem? Do you have experience in balancing also the registrations? Thanks, Stefano On Thu, 2007-09-13 at 11:02 +0200, inge wrote: > Hi all, > > Is there a way to have something like a loadbalancing on SER for > outgoing calls ? > > I want to distribute the calls between two gateways. Ideally, with a > coefficient (ie. 60% for the first and 40% for the second). > > Thanks for your support. > > Adrien .L

1 0

[OpenSER-Users] SIP Response code in a Failure Route Block
by Douglas Garstang 12 Nov '07

12 Nov '07

Can someone tell me why the pseudo variable $rr (SIP reply reason) is not present in a failure_route? If a call fails, this pseudo var contains the SIP response code, '404' for a Not Found for example. I would think that it should DEFINITELY be set in a failure route. OTHER pseudo variables are set in a failure route. Why not the SIP response code? This is probably the most important pseudo variable to have available in a failure route! Douglas. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

4 3

[OpenSER-Users] Forking based on XML document in SDP
by Pascal Maugeri 12 Nov '07

12 Nov '07

Hi I need to fork an INVITE request to a list of SIP URIs. The list of URIs is defined in an XML document, in the payload of the incoming request. I've read in a previous email (contributed by John Peters) that one may fork the request using avpops: > modparam("avpops","avp_aliases","myGroup=i:665") > avp_write("sip:u1@host", "$myGroup"); > avp_write(" sip:u2@host", "$myGroup"); > avp_pushto("$ruri", "$myGroup/g"); For the time being the only solution I see is to process the XML document in a external application using EXEC module that will return the list of SIP URIs for forking. Before sending the forked request(s) the XML document should be removed from the payload (I have no idea yet how to do this). Do you think there is smarter way to manage it ? Maybe I should not try to do this using OpenSER but rather forward the request to an external SIP service that will do the job ? Thanks in advance for any advice, Pascal

2 1

[OpenSER-Users] Loose route problem with ;nat=yes
by Daniel 12 Nov '07

12 Nov '07

Hi all, there is a little problem in loose_route () function. Loose_route() function is not able to recognize the address of the proxy server (in the ROUTE header field) if ROUTE header field has the ";nat=yes" tag. So, addresses like " Route: 192.168.2.1;nat=yes;lr=on;ftag=123455 " are read by loose_route as foreign address. Meanwhile "Route: 192.168.2.1;lr=on;ftag=123455" are recognized correctly. where 192.168.2.1 = proxy server address Regerds, Daniel -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

2 1

[OpenSER-Users] Reg. conditional processing of invite
by Padmaja 12 Nov '07

12 Nov '07

Hi all, I have an issue, where the sip user dials 911 number and hangs up. The 911 operator then calls back to the user using the call back command provided earlier say 8911 (there is a b2bua that interprets the call back number and routes the call to the specific user). This number is to be routed via the openser proxy to the b2bua. However, when the 911 calls back, proxy treats this number as a username and since there are no registrations to this 8911 number (though this entry is present at the proxy, no device has registered with this number), it sends back "404 Not found". How can this be circumvented? Thanks, Padmaja ----- Original Message ----- From: <users-request(a)lists.openser.org> To: <users(a)lists.openser.org> Sent: Saturday, November 03, 2007 3:46 AM Subject: Users Digest, Vol 30, Issue 6 > Send Users mailing list submissions to > users(a)lists.openser.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.openser.org/cgi-bin/mailman/listinfo/users > or, via email, send a message with subject or body 'help' to > users-request(a)lists.openser.org > > You can reach the person managing the list at > users-owner(a)lists.openser.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Users digest..." > > > Today's Topics: > > 1. Core dumped when deploying presence module (Sergio Gutierrez) > 2. Re: MediaProxy 1.9.0 - Radius (Brian Heath) > 3. Re: Forking Madness (Chris Heiser) > 4. Re: MediaProxy 1.9.0 - Radius (Mike O'Connor) > 5. Re: MediaProxy 1.9.0 - Radius (Ovidiu Sas) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Fri, 2 Nov 2007 15:59:56 -0500 > From: "Sergio Gutierrez" <saguti(a)gmail.com> > Subject: [OpenSER-Users] Core dumped when deploying presence module > To: users(a)lists.openser.org, users(a)openser.org > Message-ID: > <8a4af3e20711021359p3327c719t4ae680de4ac3e155(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hello to all members. > > Currently we are trying to deploy presence module in the following > environment: > > 1. Openser 1.2.1 > 2. Solaris 10 on Sparc System > 3. libxml 2.6.30 > > Openser starts up fine, but when happens any event as a Status change of a > user agent, Openser crashes. > Below I send an extract from the logfile, and the backtrace of the core. > Thanks in advance for your attention. > > Sergio Guti?rrez > EPM Telecomunicaciones > Medellin - Colombia > > **************************************************** >>From Openser logfile. > > 0(5493) Nuevo Requerimiento - M=SUBSCRIBE > RURI=sip:5192901@200.13.225.250F= > sip:5192902@200.13.225.250 T=sip:5192901@200.13.225.250 > IP=200.116.28.88ID=YmE2MGVhMGQzMTI3ZmU0YTkzMDJjNjIxODNmMGIwOWE. > 0(5493) comp_scriptvar: str 20 : 5192901 > 0(5493) DEBUG:maxfwd:is_maxfwd_present: value = 70 > 0(5493) parse_headers: flags=ffffffffffffffff > 0(5493) get_hdr_field: cseq <CSeq>: <1> <SUBSCRIBE> > 0(5493) DEBUG: get_hdr_body : content_length=0 > 0(5493) found end of header > 0(5493) PRESENCE: handle_subscribe: 'expires' found > 0(5493) PRESENCE: handle_subscribe: lexpire= 3600 > 0(5493) PRESENCE: handle_subscribe: 'To' header ALREADY PARSED: < > sip:5192901@200.13.225.250> > 0(5493) PRESENCE:handle_subscribe: generating to_tag > 0(5493) > [p_user]= 5192901 [p_domain]= 200.13.225.250 > [w_user]= 5192902 [w_domain]= 200.13.225.250 > 0(5493) [event]= presence > [staus]= pending > [expires]= 3600 > 0(5493) [to_tag]= 10.5493.1194034490.40 > [from_tag]= 1f783845 > 0(5493) PRESENCE: update_subscription ... > 0(5493) > [p_user]= 5192901 [p_domain]= 200.13.225.250 > [w_user]= 5192902 [w_domain]= 200.13.225.250 > 0(5493) [event]= presence > [staus]= pending > [expires]= 3600 > 0(5493) [to_tag]= 10.5493.1194034490.40 > [from_tag]= 1f783845 > 0(5493) expires: 3600 > 0(5493) PRESENCE:update_subscription:Inserting into database: > n_query_cols:14 > 0(5493) [0] = to_user 5192901 > 0(5493) [1] = to_domain 200.13.225.250 > 0(5493) [2] = from_user 5192902 > 0(5493) [3] = from_domain 200.13.225.250 > 0(5493) [4] = event presence > 0(5493) [5] = event_id > 0(5493) [6] = callid YmE2MGVhMGQzMTI3ZmU0YTkzMDJjNjIxODNmMGIwOWE. > 0(5493) [7] = to_tag 10.5493.1194034490.40 > 0(5493) [8] = from_tag 1f783845 > 0(5493) [9] = contact sip:5192902@200.116.28.88:47166 > 0(5493) [10] = status pending > 0(5493) [11] = cseq 1 > 0(5493) [12] = expires 1194038090 > 0(5493) [13] = version 0 > 0(5493) parse_headers: flags=ffffffffffffffff > 0(5493) check_via_address(200.116.28.88, 192.168.0.13, 0) > 0(5493) DBG:sl:run_sl_callbacks: callback id 0 entered > 0(5493) trace_sl_onreply_out: trace off... > 0(5493) PRESENCE: get_subs_dialog:n= 1 > 0(5493) PRESENCE:notify:dialog informations: > 0(5493) > [p_user]= 5192901 [p_domain]= 200.13.225.250 > [w_user]= 5192901 [w_domain]= 200.13.225.250 > 0(5493) [event]= presence.winfo > [staus]= active > [expires]= 3518 > 0(5493) [to_tag]= 10.5493.1194034408.34 > [from_tag]= 39320458 > 0(5493) presence:uandd_to_uri: uri=sip:5192901@200.13.225.250 > 0(5493) presence:uandd_to_uri: uri=sip:5192902@200.13.225.250 > 0(5493) presence:uandd_to_uri: uri=sip:5192901@200.13.225.250 > 0(5493) PRESENCE: notify: build notify to user= 5192901 domain= > 200.13.225.250 for event= presence.winfo > 0(5493) > [p_user]= 5192901 [p_domain]= 200.13.225.250 > [w_user]= 5192901 [w_domain]= 200.13.225.250 > 0(5493) [event]= presence.winfo > [staus]= active > [expires]= 3518 > 0(5493) [to_tag]= 10.5493.1194034408.34 > [from_tag]= 39320458 > 0(5493) PRESENCE:build_str_hdr: expires = 3518 > 0(5493) PRESENCE:build_str_hdr: subs_expires : 3518 > 0(5493) PRESENCE: build_str_hdr: headers: > Event: presence.winfo > Contact: <sip:200.13.225.250:5060> > Subscription-State: active;expires=3518 > Content-Type: application/watcherinfo+xml > > 0(5493) PRESENCE:notify: headers:Event: presence.winfo > Contact: <sip:200.13.225.250:5060> > Subscription-State: active;expires=3518 > Content-Type: application/watcherinfo+xml > > 0(5493) CONTACT = sip:5192901@200.13.254.180:17116 > 0(5493) presence:uandd_to_uri: uri=sip:5192901@200.13.225.250 > 0(5493) parse_rr_body(): No body for record-route > 0(5493) PRESENCE: notify:Send notify for presence on callback 1(5494) > DEBUG: timer routine:4,tl=fc5c9a80 next=0, timeout=353600000 > 1(5494) DEBUG: retransmission_handler : request resending (t=fc5c9910, > NOTIFY si ... ) > 1(5494) DEBUG:tm:set_timer: relative timeout is 1000000 > 1(5494) DEBUG: add_to_tail_of_timer[5]: fc5c9a80 (354600000) > 1(5494) DEBUG: retransmission_handler : done > 1(5494) DEBUG: timer routine:5,tl=fc5c9a80 next=0, timeout=354600000 > 1(5494) DEBUG: retransmission_handler : request resending (t=fc5c9910, > NOTIFY si ... ) > 1(5494) DEBUG:tm:set_timer: relative timeout is 2000000 > 1(5494) DEBUG: add_to_tail_of_timer[6]: fc5c9a80 (356600000) > 1(5494) DEBUG: retransmission_handler : done > 1(5494) DEBUG: timer routine:6,tl=fc5c9a80 next=0, timeout=356600000 > 1(5494) DEBUG: retransmission_handler : request resending (t=fc5c9910, > NOTIFY si ... ) > 1(5494) DEBUG:tm:set_timer: relative timeout is 4000000 > 1(5494) DEBUG: add_to_tail_of_timer[7]: fc5c9a80 (360600000) > 1(5494) DEBUG: retransmission_handler : done > > > ******************************************************************************** > Backtrace of the core: > > #0 0xfe7f77bc in shm_dup_subs (subs=0xffbff1b0, to_tag=Cannot access > memory > at address 0x96 > ) at notify.c:1819 > 1819 cb_param->wi_subs->to_user.s = (char*)cb_param + size; > (gdb) bt > #0 0xfe7f77bc in shm_dup_subs (subs=0xffbff1b0, to_tag=Cannot access > memory > at address 0x96 > ) at notify.c:1819 > #1 0xfe7f8238 in notify (subs=0x1779c8, watcher_subs=0x15, n_body=0x0, > force_null_body=1024) at notify.c:1562 > #2 0xfe7f8f18 in query_db_notify (p_user=0x234, p_domain=0xdff08, > event=0xfe809a28 "presence.winfo", watcher_subs=0xffbff1b0, etag=0x0) > at notify.c:957 > #3 0xfe8028a8 in update_subscription (msg=0x175fd8, subs=0xffbff1b0, > rtag=0xffbff228, to_tag_gen=1) at subscribe.c:476 > #4 0xfe805080 in handle_subscribe (msg=0x175fd8, str1=0xffbff0e0 "", > str2=0x800 <Address 0x800 out of bounds>) at subscribe.c:1252 > #5 0x00018378 in do_action (a=0x145270, msg=0x175fd8) at action.c:883 > #6 0x00019bb8 in run_action_list (a=0x145270, msg=0x175fd8) at > action.c:131 > #7 0x000192bc in do_action () > #8 0x00019bb8 in run_action_list (a=0x1452c8, msg=0x175fd8) at > action.c:131 > #9 0x00019df8 in run_top_route (a=0x143e10, msg=0x175fd8) at action.c:111 > #10 0x00043b60 in receive_msg (buf=0x12dc00 "", len=1024, > rcv_info=0xe0400) > at receive.c:156 > #11 0x0006f294 in udp_rcv_loop () at udp_server.c:465 > #12 0x000338bc in main_loop () at main.c:834 > #13 0x00035c44 in main (argc=9, argv=0xe7800) at main.c:1399 > (gdb) quit >

2 3

Re: [Serusers] Voicemail setup on SER v0.9.6
by roayu 12 Nov '07

12 Nov '07

Hi Eneref, Thanks for your reply. So, do u mean that I'm gonna have the follwing applications in the SAME box? - 2 SER servers ( 1 existing SER + 1 new SER for SEMS ) with different port - 1 SEMS Thanks. Roa Yu -----Original Message----- From: Eneref [mailto:eneref@arcdiv.com] Sent: Friday, November 09, 2007 4:14 PM To: roayu Subject: Re: [Serusers] Voicemail setup on SER v0.9.6 Yes. You need either SEMS or Asterisk or some such. Meshing SEMS with SER, though it LOOKS complex, is really not that bad. The easiest thing to do is to run another SER server to handle voicemail (on a different port from the first SER if it's on the same box) and forward messages to it if voicemail is required (on time outs, or whatever conditions you set). There are some good instructions on how to do that in the SEMS docs. This also makes it quite easy to swap out with Asterisk or any other sort of voicemail-capable server should the need arise in future. N. roayu wrote: > > Hi there! > > > > Could anyone let me know how to setup Voicemail on SER v.0.9.6 ? Do I > need to install SEMS as well? > > > > Thanks > > > > > > Regards, > > Roa Yu > > ------------------------------------------------------------------------ > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

1 0

[OpenSER-Users] MediaProxy 1.9.0 - Radius
by Mike O'Connor 11 Nov '07

11 Nov '07

Hi Guys I really think there is something wrong with MediaProxy, I can not get the radius or the mysql accounting to report any details, as best as I can see there is never any attempt to even try. There is nothing in the logs about there being anything wrong at all. I had added print statement thought out the code and I can see that the code for both mysql update and the code for sending radius updates never gets call. I have not been able to find the connection between the radius or mysql thread and the rtphandler but the print statement a in the actual calls to the mysql or radius and the print messages are never seen in the logs. Other prints added by me do get logged. Has any one actual tried the current code and got do the stop updates ? Mike

4 7

[OpenSER-Users] Openser bridging
by Max Lock 11 Nov '07

11 Nov '07

Hi Folks, I'm struggling with bridging openser between a public and a private network. The private 10.10.12.0/24 network connects openser and an asterisk server that plays a file and hangs up. I've got 2 sip clients connected to the public and private sides of the openser server. When I make calls via openser to the asterisk box, I only get audio on the private networks sip client, and neither client gets hung up correctly at the end of the call. I've put the SIP debug from the external client here: http://pastebin.com/m388a707d My openser and rtpproxy configs are here: http://pastebin.com/m3ed309d3 Can anyone give me any pointers? I've been staring at this for days :( -Cheers Max

2 3

[OpenSER-Users] Regarding the LCR module and AVPs...
by Kyle Stone 09 Nov '07

09 Nov '07

I'm using 1.2.X and the LCR module... following syntax fails at start up... $avp(i:20) = 1; if(!load_gw( $avp(i:20) ) ) { with an invalid arg to load_gw... anyone have a suggestion?

3 3

Re: [OpenSER-Users] lack of notify, subscribe problem
by Anca Vamanu 09 Nov '07

09 Nov '07

Stanisław Pitucha wrote: > ----- "Anca Vamanu" <anca(a)voice-system.ro> wrote: > >> I am not sure if snom 190 supports server presence. I've searched a >> bit >> but did not find sufficient info. >> Does the phone send other Subscribe messages apart from the ones with >> >> event: dialog? Does it send any Publish message? >> > > It has "Publish Presence" option turned on. It has also line monitoring with lights. I've found this site with example notify packets: > http://www.abptech.com/support/faqs/faq_snomphone_LED_explained.html > I have both support broken registrar on and filtering off, whatever that means for snom phone. That means it supports something. > Still - it subscribes for dialog properly (as far as I can tell from rfc4235), so I don't understand the bad event response... Or maybe openser just doesn't support 'dialog' events? > Yes, as I have mentioned before, OpenSer does not support 'dialog' event. It has support only for dialog with sla parameter (Event: dialog;sla) , used for BLA implementation (draft draft-anil-sipping-bla-03.txt). Please send a ngrep capture with a Subscribe message with Event: presence , its reply and the Notify that follows. Anca

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[Serusers] proxy_authorize and radius_proxy_authorize doesn't work for ACK
by Tomasz Zieleniewski 09 Nov '07

09 Nov '07

Hi, When I invoke proxy_authorize and radius_proxy_authorize for an ACK message the is no uid avp present. All other messages which are authorized work fine. Why is that so?? Is it the right behaviour?? Bests regards Tomasz

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Re: [Serusers] proxy_authorize and radius_proxy_authorize doesn't work for ACK
by Jiri Kuthan 09 Nov '07

09 Nov '07

At 17:49 08/11/2007, Tomasz Zieleniewski wrote: >Hi Piotr:) > >RFC only implies that proxy should not challenge the ACK message and >moreover it says that client sending ACK should duplicate >Authorization and Proxy-Authorization headers. This it what my client >does. But according to the description of the proxy_authorize function >it only verifies that credentials are valid. This function doesn't >cause the challenge response to be sent to the client. This is >performed with the usage of proxy_challenge function. >"The function verifies credentials according to RFC2617. If the >credentials are verified successfully then the function will succeed >and mark the credentials as authorized (marked credentials can be >later used by some other functions). If the function was unable to >verify the credentials for some reason then it will fail and the >script should call proxy_challenge which will challenge the user >again." >In my opinion, that is why proxy_authorize function should generate >the uid avp for ACK request if verification of credential gives >positive result. Well -- it should be stored somewhere in transaction state and someway I guess you should be able to get access to it (not sure if with or without some hack) ... but do you relaly need that? ACK has a transport function and as such, I don't see the use value for doing more processing with it than abosrbing/forwarding... -jiri -- Jiri Kuthan http://iptel.org/~jiri/

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Re: [Serusers] proxy_authorize and radius_proxy_authorize doesn't work for ACK
by Jiri Kuthan 09 Nov '07

09 Nov '07

At 15:53 08/11/2007, Tomasz Zieleniewski wrote: >Hi, > >When I invoke proxy_authorize and radius_proxy_authorize for an ACK >message the is no uid avp present. >All other messages which are authorized work fine. >Why is that so?? >Is it the right behaviour?? yes. ACk has only a transport function and does not really impact SIP processing. -jiri >Bests regards >Tomasz >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

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[Serusers] textops module - are the inserted hfs visible after insertion in the same route block?
by Tomasz Zieleniewski 09 Nov '07

09 Nov '07

Hi, I have the following problem after I insert a partical hf value woth the usage of the textops module (insert_attr_hf) the is_present_hf function returns negative result. is this correct behaviour?? Regards Tomasz

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Re: [OpenSER-Users] lack of notify, subscribe problem
by Stanisław Pitucha 09 Nov '07

09 Nov '07

----- "Anca Vamanu" <anca(a)voice-system.ro> wrote: > If you do not use an xcap server, you should set the force_active > presence module parameter in 1.2.x and presence_xml module parameter > in > 1.3 to positive value. > More info here: > http://www.openser.org/dokuwiki/doku.php/presence:module-exports I'm using force-active=1 - still, no NOTIFY information is sent. Relevant parts from config: modparam("presence|pua", "db_url", "mysql://openserrw:openserrw@192.168.0.134/openser") modparam("presence", "server_address", "sip:192.168.0.134:9876") modparam("presence", "fallback2db", 1) modparam("presence_xml", "db_url", "mysql://openserrw:openserrw@192.168.0.134/openser") modparam("presence_xml", "force_active", 1) modparam("presence_xml", "xcap_table", "xcap_xml") modparam("pua_usrloc", "default_domain", "XXX") in route: handle_publish, handle_subscribe on is_method("PUBLISH") and is_method("SUBSCRIBE") (looks mostly like http://www.openxcap.org/wiki/Installation, but I'm not using openxcap) Anything else I should have added there? > > 2. Phone issue. Trying to start subscribe on snom 190 - messages > There was an wrong match with the dialog;sla event that version 1.3 > has support for. It is fixed now. Thanks. With last svn update it stops after subscribe, I don't even get 202 OK like before. Messages from Snom 190: SUBSCRIBE sip:AAA@XXX;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.112:2078;branch=z9hG4bK-cilxh214xgmc;rport From: <sip:BBB@XXX>;tag=bhscby89jz To: <sip:AAA@XXX;user=phone> Call-ID: 3c26700c5573-eye598917qwu@snom190 CSeq: 1 SUBSCRIBE Max-Forwards: 70 Contact: <sip:BBB@192.168.0.112:2078;line=g4ev9jb3> Event: dialog Accept: application/dialog-info+xml Expires: 3600 Content-Length: 0 It gets "489 bad event" response. >From openser output: ERROR:presence:handle_subscribe: Missing or unsupported event header field value ERROR:presence:handle_subscribe: event= dialog Thanks.

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Re: [OpenSER-Users] Presence module - NOTIFY not sent after un-SUBSCRIBE
by Pascal Maugeri 09 Nov '07

09 Nov '07

OK I got it now working after the update you recommended it. When I un-subscribe I get now the final notify with: Subscription-State: terminated;reason=timeout Thank you very much to everybody for your help and prompt responses. Regards, Pascal On Nov 9, 2007 11:19 AM, Anca Vamanu <anca(a)voice-system.ro> wrote: > > You have to take the latest svn repository for the 1.2.x branch: > > # svn co > https://openser.svn.sourceforge.net/svnroot/openser/branches/1.2 openser > > More info: http://www.openser.org/mos/view/Download/. > > Anca > > > Pascal Maugeri wrote: > > Do you mean I have to update again with 1.2.2 or should I update to "trunk" ? > > > > Thanks > > Pascal > > > > On Nov 9, 2007 11:00 AM, Anca Vamanu <anca(a)voice-system.ro> wrote: > > > >> Hello Pascal , > >> > >> Indeed the 1.2.x version did not sent Notify when unSubscribe was > >> received. It has been fixed in trunk but a back port was forgotten. > >> It was fixed with the latest commit. > >> > >> Thanks and regards, > >> Anca Vamanu > >> > >> > >> Pascal Maugeri wrote: > >> > >>> Hi > >>> > >>> I've update to 1.2.2 > >>> (http://openser.svn.sourceforge.net/svnroot/openser/tags/1.2.2/) and I > >>> get the same problem. > >>> > >>> This is the log of openser I get when I try an unsubscribe: > >>> > >>> 6(18125) PRESENCE:update_subscription: querying database > >>> 6(18125) PRESENCE:update_subscription: expires =0 -> deleting from database > >>> 6(18125) parse_headers: flags=ffffffffffffffff > >>> 6(18125) check_via_address(10.1.1.7, 10.1.1.7, 0) > >>> 6(18125) PRESENCE: get_subs_dialog:The query for subscribtion for > >>> [user]= pascal,[domain]= mydomain.net for [event]= presence.winfo > >>> returned no result > >>> 6(18125) PRESENCE:query_db_notify: Could not get subs_dialog from database > >>> 6(18125) PRESENCE:update_subscription:Could not send notify for presence.winfo > >>> 6(18125) DEBUG: cleanup_uac_timers: RETR/FR timers reset > >>> 6(18125) DEBUG: add_to_tail_of_timer[2]: 0xb60af2b8 (414) > >>> 6(18125) DEBUG:tm:UNREF_UNSAFE: after is 0 > >>> 6(18125) DEBUG:destroy_avp_list: destroying list (nil) > >>> 6(18125) receive_msg: cleaning up > >>> 13(18132) DEBUG: timer routine:2,tl=0xb60af2b8 next=(nil), timeout=414 > >>> 13(18132) DEBUG: wait_handler : removing 0xb60af270 from table > >>> 13(18132) DEBUG: delete transaction 0xb60af270 > >>> 13(18132) DEBUG: wait_handler : done > >>> > >>> It looks there is something wrong with my configuration > >>> "query_db_notify: Could not get subs_dialog from database", doesn't it > >>> ? > >>> > >>> FYI, subscription and notifications work OK. > >>> > >>> Regards > >>> Pascal > >>> > >>> > >>> On Nov 8, 2007 9:40 AM, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> wrote: > >>> > >>> > >>>> Hi Pascal, > >>>> > >>>> be sure you are using the latest version for 1.2 - try using the SVN > >>>> snapshots for 1.2.2 > >>>> > >>>> Regards, > >>>> Bogdan > >>>> > >>>> > >>>> Pascal Maugeri wrote: > >>>> > >>>> > >>>>> Hi > >>>>> > >>>>> I'm testing the presence module (OpenSER 1.2) and noticed that there > >>>>> is no final NOTIFY request sent when I send a SUBSCRIBE to > >>>>> un-subscribe from a subscription. > >>>>> > >>>>> Is there anything to be configured ? > >>>>> > >>>>> Regards, > >>>>> Pascal > >>>>> > >>>>> > >>>>> _______________________________________________ > >>>>> Users mailing list > >>>>> Users(a)lists.openser.org > >>>>> http://lists.openser.org/cgi-bin/mailman/listinfo/users > >>>>> > >>>>> > >>>>> > >>>>> > >>> _______________________________________________ > >>> Users mailing list > >>> Users(a)lists.openser.org > >>> http://lists.openser.org/cgi-bin/mailman/listinfo/users > >>> > >>> > >>> > >> > > > > > >

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Re: [OpenSER-Users] Storing SIP Response Codes with avpops?
by Douglas Garstang 09 Nov '07

09 Nov '07

Daniel, I found that I was able to do this, and openser accepts it, and inserts a database record... avp_db_query("insert into sip_responses (response_code, response_reason) values ('$rs','abc')"); That is in a failure block. However, in the failure block, $rs is 0. I tried putting the avp_db_query into the reply block, and openser tells me that avp_db_query isn't allowed in the reply block, SO I then tried putting this: avp_write("$rs","$avp(result)"); into the reply block, and openser tells me that avp_write can't be used in a reply block. Argh! I just want to get a sip response code for the call and put it into the database. I'm using version 1.1. Douglas. ----- Original Message ---- From: Daniel-Constantin Mierla <daniel(a)voice-system.ro> To: Douglas Garstang <dougmig33(a)yahoo.com> Cc: users(a)lists.openser.org Sent: Thursday, November 8, 2007 1:30:50 PM Subject: Re: [OpenSER-Users] Storing SIP Response Codes with avpops? What version are you using? I just tested with trunk version and all ok? For the other issue, just set: modparam("avpops", "avp_table", "usr_preferences") Daniel On 11/08/07 23:20, Douglas Garstang wrote: > Thanks, but when I try that, I get the following error from openser: > > Nov 7 23:17:32 localhost openser[7035]: ERROR:xl_parse_spec: bad pvar > name "rr{s.escape.common})') " > Nov 7 23:17:32 localhost openser[7035]: > ERROR:avpops:fixup_db_query_avp: wrong format[insert into > sip_responses (response_code, response_reason) values ('$rs', > '$(rr{s.escape.common})') ] > Nov 7 23:17:32 localhost openser[7035]: ERROR: fix_actions: fixing > failed (code=-1) at cfg line 139 > > OpenSER also complains that i need to specify an avp table, but since > I am specifying the table in the query, I don't understand why that's > required. > > Douglas. > > > ----- Original Message ---- > From: Daniel-Constantin Mierla <daniel(a)voice-system.ro> > To: Douglas Garstang <dougmig33(a)yahoo.com> > Cc: users(a)lists.openser.org > Sent: Thursday, November 8, 2007 11:57:35 AM > Subject: Re: [OpenSER-Users] Storing SIP Response Codes with avpops? > > Hello, > > On 11/08/07 21:40, Douglas Garstang wrote: > > All, > > > > I'd like to store SIP response codes in mysql using avpops. Is this > > possible? > > > > I tried to do this a few years ago and couldn't find any > > documentation. Unfortunately, I still can't find any. > > > > Can someone post an example? All I can find online is reference > > documentation for avpops. > > > > I'd really like to see a full example with the modparams defining the > > table structure, and how I would use avp_write and avp_db_store to > > write the $rr response code. > the easiest way is to use avp_db_query(), you can give as parameter any > sql query. In your case, you can use in onreply_route: > > avp_db_query("insert into sip_responses (response_code, response_reason) > values ('$rs', '$(rr{s.escape.common})') "); > > You need to create the table sip_responses, with the two string columns. > > Daniel > > > > > Thanks. > > Douglas. > > > > > > __________________________________________________ > > Do You Yahoo!? > > Tired of spam? Yahoo! Mail has the best spam protection around > > http://mail.yahoo.com > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Users mailing list > > Users(a)lists.openser.org <mailto:Users@lists.openser.org> > > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > > > > __________________________________________________ > Do You Yahoo!? > Tired of spam? Yahoo! Mail has the best spam protection around > http://mail.yahoo.com __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[OpenSER-Users] [Fwd: Re: Presence module - NOTIFY not sent after un-SUBSCRIBE]
by Anca Vamanu 09 Nov '07

09 Nov '07

-------- Original Message -------- Subject: Re: [OpenSER-Users] Presence module - NOTIFY not sent after un-SUBSCRIBE Date: Fri, 09 Nov 2007 12:19:57 +0200 From: Anca Vamanu <anca(a)voice-system.ro> To: Pascal Maugeri <pascal.maugeri1(a)gmail.com> References: <990aed650711050832u17bcebeey5fea7e4b593ebf2b(a)mail.gmail.com> <990aed650711071336i16268d29v7826e6c40ec580dd(a)mail.gmail.com> <4732CB99.2020104(a)voice-system.ro> <990aed650711090100l16fc5cd2h32c94ad0504ac12a(a)mail.gmail.com> <47342FBF.7060208(a)voice-system.ro> <990aed650711090210q4112985bt16cbbd28255cdd5f(a)mail.gmail.com> You have to take the latest svn repository for the 1.2.x branch: # svn co https://openser.svn.sourceforge.net/svnroot/openser/branches/1.2 openser More info: http://www.openser.org/mos/view/Download/. Anca Pascal Maugeri wrote: > Do you mean I have to update again with 1.2.2 or should I update to "trunk" ? > > Thanks > Pascal > > On Nov 9, 2007 11:00 AM, Anca Vamanu <anca(a)voice-system.ro> wrote: > >> Hello Pascal , >> >> Indeed the 1.2.x version did not sent Notify when unSubscribe was >> received. It has been fixed in trunk but a back port was forgotten. >> It was fixed with the latest commit. >> >> Thanks and regards, >> Anca Vamanu >> >> >> Pascal Maugeri wrote: >> >>> Hi >>> >>> I've update to 1.2.2 >>> (http://openser.svn.sourceforge.net/svnroot/openser/tags/1.2.2/) and I >>> get the same problem. >>> >>> This is the log of openser I get when I try an unsubscribe: >>> >>> 6(18125) PRESENCE:update_subscription: querying database >>> 6(18125) PRESENCE:update_subscription: expires =0 -> deleting from database >>> 6(18125) parse_headers: flags=ffffffffffffffff >>> 6(18125) check_via_address(10.1.1.7, 10.1.1.7, 0) >>> 6(18125) PRESENCE: get_subs_dialog:The query for subscribtion for >>> [user]= pascal,[domain]= mydomain.net for [event]= presence.winfo >>> returned no result >>> 6(18125) PRESENCE:query_db_notify: Could not get subs_dialog from database >>> 6(18125) PRESENCE:update_subscription:Could not send notify for presence.winfo >>> 6(18125) DEBUG: cleanup_uac_timers: RETR/FR timers reset >>> 6(18125) DEBUG: add_to_tail_of_timer[2]: 0xb60af2b8 (414) >>> 6(18125) DEBUG:tm:UNREF_UNSAFE: after is 0 >>> 6(18125) DEBUG:destroy_avp_list: destroying list (nil) >>> 6(18125) receive_msg: cleaning up >>> 13(18132) DEBUG: timer routine:2,tl=0xb60af2b8 next=(nil), timeout=414 >>> 13(18132) DEBUG: wait_handler : removing 0xb60af270 from table >>> 13(18132) DEBUG: delete transaction 0xb60af270 >>> 13(18132) DEBUG: wait_handler : done >>> >>> It looks there is something wrong with my configuration >>> "query_db_notify: Could not get subs_dialog from database", doesn't it >>> ? >>> >>> FYI, subscription and notifications work OK. >>> >>> Regards >>> Pascal >>> >>> >>> On Nov 8, 2007 9:40 AM, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> wrote: >>> >>> >>>> Hi Pascal, >>>> >>>> be sure you are using the latest version for 1.2 - try using the SVN >>>> snapshots for 1.2.2 >>>> >>>> Regards, >>>> Bogdan >>>> >>>> >>>> Pascal Maugeri wrote: >>>> >>>> >>>>> Hi >>>>> >>>>> I'm testing the presence module (OpenSER 1.2) and noticed that there >>>>> is no final NOTIFY request sent when I send a SUBSCRIBE to >>>>> un-subscribe from a subscription. >>>>> >>>>> Is there anything to be configured ? >>>>> >>>>> Regards, >>>>> Pascal >>>>> >>>>> >>>>> _______________________________________________ >>>>> Users mailing list >>>>> Users(a)lists.openser.org >>>>> http://lists.openser.org/cgi-bin/mailman/listinfo/users >>>>> >>>>> >>>>> >>>>> >>> _______________________________________________ >>> Users mailing list >>> Users(a)lists.openser.org >>> http://lists.openser.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >> > >

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[OpenSER-Users] TLS Registrations start dropping?
by Chris Heiser 09 Nov '07

09 Nov '07

Information on TLS and OpenSER with this specific issues seems to be scarce. I have 2 different setups. One with about 130 registrations, one with about 250. Both with SSLv23 turned on, and TLS is enabled in the phones. What's odd is after some unknown amount of time, phones start dropping like files and fail to re-register. (Our registration timer is set at 60 seconds to aid in failover). What's interesting is it appears that the SSL handshake completely fails when this starts to happen. The requests from the phones are smaller in size (733 bytes about 970 for a good one). Suspicion would normally lie with the phones. They're obviously not doing something right! Here's the kicker. If I fail over to another OpenSER proxy (we move the IP over, use the same certs, etc...) everyone comes back up, and everything is happy for hours, maybe a day, maybe a week, and then it all starts going downhill again. I've increased the number of OpenSER children, I've played with the tcp_persistent_flag. None of these things have stopped the madness. Anyone have any ideas or thoughts? --Chris

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[OpenSER-Users] Presence module - NOTIFY not sent after un-SUBSCRIBE
by Pascal Maugeri 09 Nov '07

09 Nov '07

Hi I'm testing the presence module (OpenSER 1.2) and noticed that there is no final NOTIFY request sent when I send a SUBSCRIBE to un-subscribe from a subscription. Is there anything to be configured ? Regards, Pascal

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[OpenSER-Users] registrations not saved in the mysql database
by Ovidiu Sas 09 Nov '07

09 Nov '07

Hi all, I was trying to keep persistent registrations in the mysql db, but it seems that something is broken in the latest svn: version: openser 1.3.0-pre1-notls (i386/linux) flags: STATS: Off, EXTRA_DEBUG, USE_IPV6, USE_TCP, 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_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. svnrevision: 2:3034M @(#) $Id: main.c 3015 2007-11-05 13:08:50Z henningw $ main.c compiled on 11:41:03 Nov 6 2007 with gcc 4.1.2 The config that I used for testing is pretty simple: # ----------- global configuration parameters ------------------------ debug=9 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) children=1 disable_tcp=yes # ------------------ module loading ---------------------------------- mpath="/usr/local/lib/openser/modules/" loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "mi_fifo.so" modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") loadmodule "usrloc.so" modparam("usrloc", "db_url", "mysql://openser:openserrw@127.0.0.1/openser") modparam("usrloc", "db_mode", 1) loadmodule "registrar.so" #------------------------- request routing logic ------------------- route{ if (!method=="REGISTER") record_route(); if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(1); }; if (uri==myself) { if (method=="REGISTER") { save("location"); exit; }; if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { if (!t_relay()) { sl_reply_error(); }; exit; } Here are the relevant debug logs: Nov 6 12:19:04 localhost /usr/local/sbin/openser[8963]: core:db_double2str: Error in snprintf Nov 6 12:19:04 localhost /usr/local/sbin/openser[8963]: mysql:val2str: error while converting string to double Nov 6 12:19:04 localhost /usr/local/sbin/openser[8963]: core:db_print_values: Error while converting value to string Nov 6 12:19:04 localhost /usr/local/sbin/openser[8963]: usrloc:db_insert_ucontact: inserting contact in db failed Nov 6 12:19:04 localhost /usr/local/sbin/openser[8963]: usrloc:insert_ucontact: failed to insert in database I can see the in memory registration via `openserctl ul show' command, but the mysql database is not updated (verified via `openserctl db show location') Regards, Ovidiu Sas

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Re: [OpenSER-Users] CPL "Location" and "Proxy" behavior
by Bogdan-Andrei Iancu 09 Nov '07

09 Nov '07

Hi Daniel, Could please check the latest SVN version of 1.3 - I made a fix there. If everything ok, I will backport it to 1.2 also. Regards, Bogdan Daniel Grotti wrote: > *Hi Bogdan,* >> Hi Daniel, >> >> Hmmm....I see - you say (for outgoing scenario) that the location set >> is initialized with the A destination instead of B destination, right? > *yes, it's right.* >> If so, please open a bug report on the tracker. > *well, on the bug Repert, if I don't set any location (as in the last > script I have sended), I can see this DEBUG line: > " DEBUG:cpl_proxy_to_loc_set: rewriting Request-URI with < > A@A_ip_address>" > > where : "A@A_ip_address" is the caller IP address, not the destination > address (that is B). > * > > *regards. > daniel > * >> regards, >> Bogdan >> > > > >> Daniel Grotti wrote: >>> Hi bogdan, >>> If it's my cpl outgoing script configuration, from USER A >>> ****************************************************** >>> <cpl> >>> <outgoing> >>> <address-switch field="destination" subfield="user"> >>> <address is="B"> >>> <proxy /> >>> </address> >>> </address-switch> >>> </outgoing> >>> <cpl> >>> ******************************************************* >>> When I make a call from A to B, then the script has called, but in >>> debug I can see that CPL rewrites R-URI with source address >>> (A@A_ip_address) instead of destination address (B@B_ip_address). So >>> the message goes back to user A. >>> >>> regards, >>> Daniel >>> >>> Bogdan-Andrei Iancu wrote: >>>> Hi Daniel, >>>> >>>> there are no differences - the implementation follows the RFC. For >>>> incoming calls, the location set is empty, but it can be populated >>>> by the CPL script during execution. So, when you do "proxy", you >>>> may have multiple entries in the location set. >>>> >>>> The comment in the cpl_sig.c file refers to some implementation >>>> details - how to process the available location set, depeding if it >>>> is the first time doing proxy (for the call) or a second time. >>>> >>>> regards, >>>> bogdan >>>> >>>> daniel grotti wrote: >>>>> Hi all, >>>>> this is what CPL_SIG.c says about forwarding a given message to >>>>> the given location set: >>>>> >>>>> " forwards the msg to the given location set; if flags has set the >>>>> CPL_PROXY_DONE, all locations will be added as branches, >>>>> otherwise, the first one will set as RURI (this is ha case when >>>>> this is the first proxy of the message) >>>>> The given list of location will be freed, returning 0 instead. >>>>> Returns: 0 - OK " >>>>> >>>>> >>>>> My question is : which and what is "the first one" location ? Can >>>>> you explain it better? >>>>> Because RFC 3880 say that : "For the incoming top-level call >>>>> processing action, the location set is initialized to the empty set. >>>>> For the outgoing action, it is initialized to the destination >>>>> address of the call." >>>>> >>>>> So RFC is very clear. I think there are considerable differences >>>>> between RFC and the behavior of CPL interpreter. >>>>> >>>>> Thanks. >>>>> Daniel >>>>> >> >> >

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[OpenSER-Users] CPL processing in OpenSER
by Chris Heiser 09 Nov '07

09 Nov '07

So voip-info.org's wiki page on CPL-C notest that the XML processing is resource consuming and thus common features can be implemented in other OpenSER modules. I was under the impression that the upload of a CPL script into openser caused a binary compile of the script that gets used at runtime to eliminate the terrible overhead of libxml, etc... Or am I completely off base? --Chris

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[OpenSER-Users] lack of notify, subscribe problem
by Stanisław Pitucha 09 Nov '07

09 Nov '07

Hey I've got 2 issues with pub/sub: 1. Openser does correct subscription on grandstream 2020 phone. But then anything happens: register, invite, whatever from monitored phone, openser doesn't generate any notifications for subscribers. Any pointers on it? Is there some variable, that I should set to enable those? 2. Phone issue. Trying to start subscribe on snom 190 - messages sent are: ----- SUBSCRIBE ... ... Event: dialog Accept: application/dialog-info+xml ----- 202 OK ----- NOTIFY ... Event: dialog;sla Subscription-State: active;expires=3600\r\n ----- 489 Bad Event ----- Anything I can do about it? I've tried it with svn version with the same effect. Couldn't really find anything in docs that I could change in config. pub/sub is handled with simple handle_publish/handle_subscribe - nothing fancy in config - just location storing on register and message passing. Thanks for all info.

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[OpenSER-Users] UAC_REPLACE_FROM function
by Allan Chao ( 趙廷睿 ) 09 Nov '07

09 Nov '07

Dear All: I want to use uac_replace_from function to change from header username, it will be successed in the first time , but when i use in the second time , the from header username become <username1 username2> . but i want it become <username2> , how can i solve it , thanks. allan

2 1

[OpenSER-Users] cdrtool radus openser.cfg configuration
by ayeddjalil 09 Nov '07

09 Nov '07

Hi every one It's my first time there, I am trying to install cdrtool, I installed openser with radius and mysql accounting, I follow the install file with cdrtool, but it seam it's not very clear I added these line in my openser.cfg as they said in the install file 268 loadmodule "acc.so" 269 270 modparam("acc", "failed_transaction_flag", 1) 271 modparam("acc", "report_cancels", 0) 272 modparam("acc", "report_ack", 0) 273 modparam("acc", "early_media", 0) 274 275 modparam("acc", "log_level", 1) 276 modparam("acc", "log_flag", 1) 277 modparam("acc", "log_missed_flag", 1) 278 279 modparam("acc", "radius_config", "/etc/openser/radius/client.conf") 280 modparam("acc", "radius_flag", 1) 281 modparam("acc", "radius_missed_flag", 1) 282 modparam("acc", "radius_extra", "Sip-RPid=$avp(s:rpid); \ 283 Source-IP=$si; \ 284 Source-Port=$sp; \ 285 Canonical-URI=$avp(can_uri); \ 286 Billing-Party=$avp(billing_party); \ 287 Divert-Reason=$avp(s:divert_reason); \ 288 X-RTP-Stat=$avp(s:rtp_statistics); \ 289 From-Header=$hdr(from); \ 290 User-Agent=$hdr(user-agent); \ 291 Contact=$hdr(contact); \ 292 Event=$hdr(event); \ 293 SIP-Proxy-IP=$avp(s:sip_proxy_ip)") alone opense do not start intill I added this loadmodule "auth_radius.so" loadmodule "group_radius.so" loadmodule "avp_radius.so" modparam("acc", "service_type", 15) modparam("acc", "radius_extra", "Sip-Src-IP=$si;Sip-Src-Port=$sp") modparam("acc|auth_radius|group_radius|avp_radius", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") the next problem is when adding setflag(ACCOUNTING_FLAG); avp_write("SER_IP","$avp(s:sip-proxy)"); avp_write("$ru", "$avp(can_uri)"); errors comes, here what I get from the command /sbin/openser -c 0(22857) parse error (123,9-24): syntax error 0(22857) parse error (123,9-24): missing '(' or ')'? 0(22857) parse error (123,9-24): bad command: missing ';'? 0(22857) parse error (123,24-25): bad command!) 0(22857) parse error (124,24-25): unknown command, missing loadmodule? 0(22857) parse error (125,25-26): unknown command, missing loadmodule? ERROR: bad config file (6 errors) 0(22857) INFO:mi_fifo:mi_destroy:memory for the child's mi_fifo_pid was not allocated -> nothing to destroy is thier any one has a sample of openser.cfg file for cdrtool Please help -- View this message in context: http://www.nabble.com/cdrtool-radus-openser.cfg-configuration-tf4776366.htm… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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Re: [Serusers] Spam Re: Voicemail setup on SER v0.9.6
by roayu 09 Nov '07

09 Nov '07

Hi Mashudi, Thanks for your reply. From the documentation, it seems that I need to install both SER and SEMS under the same directory. 1) I've already installed the SER under /usr/src, can I also install SEMS under /usr/src instead of /usr/local/src ? 2) By the way, I would like to know that does this voicemail able to trigger the voicemail on the IP hard phone? For instance, if there's a new voicemail, can the IP hard phone got the signal of the new voicemail? Thanks. Regards, Roa Yu -----Original Message----- From: mashudi [mailto:mashudiflexi@telkom.co.id] Sent: Saturday, November 10, 2007 2:04 AM To: roayu Subject: Spam Re: [Serusers] Voicemail setup on SER v0.9.6 dear roayu, you can read on http://www.iptel.org/sems/sems_application_development_tutorial just follow the document, you will able run simple voice mail. roayu wrote: > ********************** > Your mail has been scanned by > Telkom Anti Virus System > *********************** > > Hi there! > > > > Could anyone let me know how to setup Voicemail on SER v.0.9.6 ? Do I > need to install SEMS as well? > > > > Thanks > > > > > > Regards, > > Roa Yu > > ------------------------------------------------------------------------ > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

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[Serusers] Voicemail setup on SER v0.9.6
by roayu 09 Nov '07

09 Nov '07

Hi there! Could anyone let me know how to setup Voicemail on SER v.0.9.6 ? Do I need to install SEMS as well? Thanks Regards, Roa Yu

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[OpenSER-Users] Forking Madness
by Chris Heiser 09 Nov '07

09 Nov '07

So I've been all over the place looking at various examples, threads, issues, and solutions to all sorts of forking scenarios. What I haven't been able to find is a solution to my odd corner case. Let me explain the forking scenario: Call comes in for user A: - Use avp_db_load to find that user A wants to parallel fork to 123 and 456 - Call to 123 fails because there's no registration - Call to 456 times out == At this point we serial fork to the next step which might hbe - Just call my cell What's unclear to me is how to actually achieve this logic. I mean, getting the initial parallel fork is quite simple using avp_pushto. What's unclear is how to setup the failure route to know that you can ignore a failure in any of the forks until the last fork as failed, and then it's time to move on. Something like, "are there other branches pending?" would be a nice check to determine if the failure route should just drop or if it should append a branch (or two) for the next serial fork. Comments? Ideas? Thoughts? Am I totally insane? Should I be using a specific module for this? --Chris Heiser

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[OpenSER-Users] openser + cdrtool and multiple cdrs
by Morten Isaksen 08 Nov '07

08 Nov '07

Hi! I few times a day I have 2 cdrs for one call but with reverse direction. I have looked at the radius server detail records and as far as I can see this happens: A Acct-Status-Type = Start with Acct-Session-Id = X1 and Sip-To-Tag = X2 and Sip-From-Tag = X3 is logged. I few seconds later Acct-Status-Type = Start with Acct-Session-Id = X1 and Sip-To-Tag = X3 and Sip-From-Tag = X2 is logged. (To and From tag is reversed) When the call is over a Update and a Stop is logged. The Acct-Status-Type is not a unique key in the raddact table and there for I get a CDR for each direction. My theory is that this happens when the server recieves a re-INVITE. Can I prevent this in openser.cfg? Is it safe to add Acct-Session-Id as a unique key to the raddact table? Or is the Session id not globaly unique? -- Morten Isaksen http://www.misak.dk/blog/

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Re: [OpenSER-Users] Storing SIP Response Codes with avpops?
by Douglas Garstang 08 Nov '07

08 Nov '07

Thanks, but when I try that, I get the following error from openser: Nov 7 23:17:32 localhost openser[7035]: ERROR:xl_parse_spec: bad pvar name "rr{s.escape.common})') " Nov 7 23:17:32 localhost openser[7035]: ERROR:avpops:fixup_db_query_avp: wrong format[insert into sip_responses (response_code, response_reason) values ('$rs', '$(rr{s.escape.common})') ] Nov 7 23:17:32 localhost openser[7035]: ERROR: fix_actions: fixing failed (code=-1) at cfg line 139 OpenSER also complains that i need to specify an avp table, but since I am specifying the table in the query, I don't understand why that's required. Douglas. ----- Original Message ---- From: Daniel-Constantin Mierla <daniel(a)voice-system.ro> To: Douglas Garstang <dougmig33(a)yahoo.com> Cc: users(a)lists.openser.org Sent: Thursday, November 8, 2007 11:57:35 AM Subject: Re: [OpenSER-Users] Storing SIP Response Codes with avpops? Hello, On 11/08/07 21:40, Douglas Garstang wrote: > All, > > I'd like to store SIP response codes in mysql using avpops. Is this > possible? > > I tried to do this a few years ago and couldn't find any > documentation. Unfortunately, I still can't find any. > > Can someone post an example? All I can find online is reference > documentation for avpops. > > I'd really like to see a full example with the modparams defining the > table structure, and how I would use avp_write and avp_db_store to > write the $rr response code. the easiest way is to use avp_db_query(), you can give as parameter any sql query. In your case, you can use in onreply_route: avp_db_query("insert into sip_responses (response_code, response_reason) values ('$rs', '$(rr{s.escape.common})') "); You need to create the table sip_responses, with the two string columns. Daniel > > Thanks. > Douglas. > > > __________________________________________________ > Do You Yahoo!? > Tired of spam? Yahoo! Mail has the best spam protection around > http://mail.yahoo.com > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org > http://lists.openser.org/cgi-bin/mailman/listinfo/users > __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[OpenSER-Users] Storing SIP Response Codes with avpops?
by Douglas Garstang 08 Nov '07

08 Nov '07

All, I'd like to store SIP response codes in mysql using avpops. Is this possible? I tried to do this a few years ago and couldn't find any documentation. Unfortunately, I still can't find any. Can someone post an example? All I can find online is reference documentation for avpops. I'd really like to see a full example with the modparams defining the table structure, and how I would use avp_write and avp_db_store to write the $rr response code. Thanks. Douglas. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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Re: [OpenSER-Users] OpenSER SIP registrar - third party registration
by Klaus Darilion 08 Nov '07

08 Nov '07

Pascal Maugeri wrote: > Thanks for your prompt answer. > > On 10/19/07, Klaus Darilion <klaus.mailinglists(a)pernau.at> wrote: >> It depends on what kind of information should be stored in the other >> server. If you "your server" needs to see the same contact information >> as openser received you could use t_replicate to replicate the REGISTER >> to "your server". In this case "your server" must not challenge the >> REGISTER. > > My server does not need to store any information but the information > that a user has been registered. > > I believe the solution I need is to use this t_replicate but I need to > replicate the register once the registration is achieved successfully. > So how can I do this replicate to my server once the complete dialog > REGISTER-401-REGISTER-OK has been achieved between the UAC and OpenSER > ? Just use t_replicate after save(), e.g: http://lists.iptel.org/pipermail/serdev/2006-June/007518.html regards klaus ps: please Cc: the mailing-list

3 2

[OpenSER-Users] [AS WeSIP & OpenIMSCore]
by Alscion MailingList 08 Nov '07

08 Nov '07

Hi, Does anybody successed to trigg the WeSIP AS with Fokus OpenIMSCore? Do you have some informations about that? Thx.

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[OpenSER-Users] X-lite (for Linux) and BYE Scenario
by Daniel Grotti 08 Nov '07

08 Nov '07

Hi All, I've noticed that using X-lite (for Linux) there are some problems with BYE method. When X-lite generates BYE, it puts in the Route Header field the address known through the record-route added by proxy. As "SER - Gettin Started" shown, an address like : IP_proxy:5060;nat=yes: " /.... .... if(method=="INVITE" && client_nat_test("7")) { record_route_preset ("proxy_ip_address :5060;nat=yes"); .... } /" 1. When we have a BYE message sended by X-lite (user A), it is like this: "/Request-Line: BYE sip: B@proxy_ip_address : 5060 SIP/2.0 Via: SIP/2.0/UDP A_address:5061; rport;branch=xyz From: <sip: A@proxy_ip_address:5061>;tag=123 To: <sip: B@proxy_ip_address:5060>;tag=456 Contact:<sip:A@A_address:5061> * Route: <sip: proxy_ip_address:5060; nat=yes; ftag=9098789; lr=on>* .... ..... .... "/ 2. Now the message BYE is parsed by the Proxy, and it uses the loose_route(): /if(loose_route())/ { forward the message }; but Proxy doesn't recognizes its address and then rewrites the R-URI with the address in Route header field and sends the BYE message.....to itself! then Proxy parses the new BYE (sended by itself) with the URI :/* <sip: proxy_ip_address:5060; nat=yes; ftag=9098789; lr=on>,*/ the the answer to user A is : 404 Destination not found! Why doesn't OpenSER, in the loose_route() function, recognize its address in Route header field? This is what should it do. And then, delete the route header field and then forward the message using R-URI. I think there is something wrong.... Regards, Daniel -- --------------------------------------------------- Daniel Grotti DEIS - University of Bologna Via Venezia, 52 - 47023 Cesena (FC) - ITALY Contacts e-mail : daniel.grotti(a)unibo.it Skype name : Daniel Grotti ---------------------------------------------------

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[Serusers] What are the possible HFNAME values in hf selects in ser rel_2_0_0?
by Tomasz Zieleniewski 08 Nov '07

08 Nov '07

Hi, What are the possible HFNAME values in hf selects in ser rel_2_0_0? When I try to add my own header and then use the a select for it the select is empty. I do it for P-Asserted-Identity header I did the following selects and no results: xlog("L_INFO", " route[AUTHENTICATION]: P-ASS: %(a)hf_value.p-asserted-identity \n"); xlog("L_INFO", " route[AUTHENTICATION]: P-ASS: %(a)hf_value.P_Asserted_Identity \n"); xlog("L_INFO", " route[AUTHENTICATION]: P-ASS: %(a)hf_value.p_asserted_identity \n"); xlog("L_INFO", " route[AUTHENTICATION]: P-ASS: %(a)hf_value2.p-asserted-identity \n"); xlog("L_INFO", " route[AUTHENTICATION]: P-ASS: %(a)hf_value2.P_Asserted_Identity \n"); xlog("L_INFO", " route[AUTHENTICATION]: P-ASS: %(a)hf_value2.p_asserted_identity \n"); Please point me what do I miss. Regards Tomasz

2 1

[Serusers] (no subject)
by Shad Mortazavi 08 Nov '07

08 Nov '07

Shad Mortazavi UK Managing Director Nexus Management (EMEA) Ltd -------------------- T: +44 (0)844 484 2574 M: +44 (0)781 721 0758 Sent from Windows Mobile.

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Re: [OpenSER-Users] CPL "Location" and "Proxy" behavior
by Bogdan-Andrei Iancu 08 Nov '07

08 Nov '07

Hi Daniel, Hmmm....I see - you say (for outgoing scenario) that the location set is initialized with the A destination instead of B destination, right? If so, please open a bug report on the tracker. regards, Bogdan Daniel Grotti wrote: > Hi bogdan, > If it's my cpl outgoing script configuration, from USER A > ****************************************************** > <cpl> > <outgoing> > <address-switch field="destination" subfield="user"> > <address is="B"> > <proxy /> > </address> > </address-switch> > </outgoing> > <cpl> > ******************************************************* > When I make a call from A to B, then the script has called, but in > debug I can see that CPL rewrites R-URI with source address > (A@A_ip_address) instead of destination address (B@B_ip_address). So > the message goes back to user A. > > regards, > Daniel > > Bogdan-Andrei Iancu wrote: >> Hi Daniel, >> >> there are no differences - the implementation follows the RFC. For >> incoming calls, the location set is empty, but it can be populated by >> the CPL script during execution. So, when you do "proxy", you may >> have multiple entries in the location set. >> >> The comment in the cpl_sig.c file refers to some implementation >> details - how to process the available location set, depeding if it >> is the first time doing proxy (for the call) or a second time. >> >> regards, >> bogdan >> >> daniel grotti wrote: >>> Hi all, >>> this is what CPL_SIG.c says about forwarding a given message to the >>> given location set: >>> >>> " forwards the msg to the given location set; if flags has set the >>> CPL_PROXY_DONE, all locations will be added as branches, >>> otherwise, the first one will set as RURI (this is ha case when >>> this is the first proxy of the message) >>> The given list of location will be freed, returning 0 instead. >>> Returns: 0 - OK " >>> >>> >>> My question is : which and what is "the first one" location ? Can >>> you explain it better? >>> Because RFC 3880 say that : "For the incoming top-level call >>> processing action, the location set is initialized to the empty set. >>> For the outgoing action, it is initialized to the destination >>> address of the call." >>> >>> So RFC is very clear. I think there are considerable differences >>> between RFC and the behavior of CPL interpreter. >>> >>> Thanks. >>> Daniel >>>

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[OpenSER-Users] CPL "Location" and "Proxy" behavior
by daniel grotti 08 Nov '07

08 Nov '07

Hi all, this is what CPL_SIG.c says about forwarding a given message to the given location set: " forwards the msg to the given location set; if flags has set the CPL_PROXY_DONE, all locations will be added as branches, otherwise, the first one will set as RURI (this is ha case when this is the first proxy of the message) The given list of location will be freed, returning 0 instead. Returns: 0 - OK " My question is : which and what is "the first one" location ? Can you explain it better? Because RFC 3880 say that : "For the incoming top-level call processing action, the location set is initialized to the empty set. For the outgoing action, it is initialized to the destination address of the call." So RFC is very clear. I think there are considerable differences between RFC and the behavior of CPL interpreter. Thanks. Daniel ----------------------------- Daniel Grotti DEIS - Universita' di Bologna ----------------------------- Via Venezia, 52 47023 Cesena (FC) - ITALY ----------------------------- email:daniel.grotti@unibo.it -----------------------------

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[Serusers] Openser bridging
by Max Lock 08 Nov '07

08 Nov '07

Hi Folks, I'm still struggling with bridging openser between a public and a private network. The private 10.10.12.0/24 network connects openser and an asterisk server that plays a file and hangs up. I've got 2 sip clients connected to the public and private sides of the openser server. When I make calls via openser to the asterisk box, I only get audio on the private networks sip client, and neither client gets hung up correctly at the end of the call. I've put the SIP debug from the external client here: http://pastebin.com/m388a707d My openser and rtpproxy configs are here: http://pastebin.com/m3ed309d3 Can anyone give me any pointers? I've been staring at this for days :( -Cheers Max

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[Serusers] Integrate LCS & SER Server
by elilkumaran s 07 Nov '07

07 Nov '07

Sir/Madam I'm a System administrator of Institute. I use MS LCS 2005 SP1 Enterprise edition & Office Communicator in my work place. I have linux SER Server & CISCO ATA 186 Telephone adapter. I have to run to Messenger client X-Lite for SER Server and Office Communicator for LCS. Because I place TP in Guard Office they can call Staff through SER server / X-Lite. But can't call Office Communicator! CISCO ATA 186 can't registering in LCS, I thing it not possible directly. ( Because LCS use Active Directory Account through Kerbas & NTLM authentication). So I want integrate LCS & SER Server to facilitate TP call to Office Communicator & Office Communicator call TP. I use a pool in LCS as normal TCP (not TLS). SER Server also use normal (with out DB -Any one can registrar with any name) Could you kindly tell the process for integrate LCS & SER Server? Thank You Your truly Elil

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[OpenSER-Users] Core dumped when deploying presence module
by Sergio Gutierrez 07 Nov '07

07 Nov '07

Hello to all members. Currently we are trying to deploy presence module in the following environment: 1. Openser 1.2.1 2. Solaris 10 on Sparc System 3. libxml 2.6.30 Openser starts up fine, but when happens any event as a Status change of a user agent, Openser crashes. Below I send an extract from the logfile, and the backtrace of the core. Thanks in advance for your attention. Sergio Gutiérrez EPM Telecomunicaciones Medellin - Colombia **************************************************** >From Openser logfile. 0(5493) Nuevo Requerimiento - M=SUBSCRIBE RURI=sip:5192901@200.13.225.250F= sip:5192902@200.13.225.250 T=sip:5192901@200.13.225.250 IP=200.116.28.88ID=YmE2MGVhMGQzMTI3ZmU0YTkzMDJjNjIxODNmMGIwOWE. 0(5493) comp_scriptvar: str 20 : 5192901 0(5493) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(5493) parse_headers: flags=ffffffffffffffff 0(5493) get_hdr_field: cseq <CSeq>: <1> <SUBSCRIBE> 0(5493) DEBUG: get_hdr_body : content_length=0 0(5493) found end of header 0(5493) PRESENCE: handle_subscribe: 'expires' found 0(5493) PRESENCE: handle_subscribe: lexpire= 3600 0(5493) PRESENCE: handle_subscribe: 'To' header ALREADY PARSED: < sip:5192901@200.13.225.250> 0(5493) PRESENCE:handle_subscribe: generating to_tag 0(5493) [p_user]= 5192901 [p_domain]= 200.13.225.250 [w_user]= 5192902 [w_domain]= 200.13.225.250 0(5493) [event]= presence [staus]= pending [expires]= 3600 0(5493) [to_tag]= 10.5493.1194034490.40 [from_tag]= 1f783845 0(5493) PRESENCE: update_subscription ... 0(5493) [p_user]= 5192901 [p_domain]= 200.13.225.250 [w_user]= 5192902 [w_domain]= 200.13.225.250 0(5493) [event]= presence [staus]= pending [expires]= 3600 0(5493) [to_tag]= 10.5493.1194034490.40 [from_tag]= 1f783845 0(5493) expires: 3600 0(5493) PRESENCE:update_subscription:Inserting into database: n_query_cols:14 0(5493) [0] = to_user 5192901 0(5493) [1] = to_domain 200.13.225.250 0(5493) [2] = from_user 5192902 0(5493) [3] = from_domain 200.13.225.250 0(5493) [4] = event presence 0(5493) [5] = event_id 0(5493) [6] = callid YmE2MGVhMGQzMTI3ZmU0YTkzMDJjNjIxODNmMGIwOWE. 0(5493) [7] = to_tag 10.5493.1194034490.40 0(5493) [8] = from_tag 1f783845 0(5493) [9] = contact sip:5192902@200.116.28.88:47166 0(5493) [10] = status pending 0(5493) [11] = cseq 1 0(5493) [12] = expires 1194038090 0(5493) [13] = version 0 0(5493) parse_headers: flags=ffffffffffffffff 0(5493) check_via_address(200.116.28.88, 192.168.0.13, 0) 0(5493) DBG:sl:run_sl_callbacks: callback id 0 entered 0(5493) trace_sl_onreply_out: trace off... 0(5493) PRESENCE: get_subs_dialog:n= 1 0(5493) PRESENCE:notify:dialog informations: 0(5493) [p_user]= 5192901 [p_domain]= 200.13.225.250 [w_user]= 5192901 [w_domain]= 200.13.225.250 0(5493) [event]= presence.winfo [staus]= active [expires]= 3518 0(5493) [to_tag]= 10.5493.1194034408.34 [from_tag]= 39320458 0(5493) presence:uandd_to_uri: uri=sip:5192901@200.13.225.250 0(5493) presence:uandd_to_uri: uri=sip:5192902@200.13.225.250 0(5493) presence:uandd_to_uri: uri=sip:5192901@200.13.225.250 0(5493) PRESENCE: notify: build notify to user= 5192901 domain= 200.13.225.250 for event= presence.winfo 0(5493) [p_user]= 5192901 [p_domain]= 200.13.225.250 [w_user]= 5192901 [w_domain]= 200.13.225.250 0(5493) [event]= presence.winfo [staus]= active [expires]= 3518 0(5493) [to_tag]= 10.5493.1194034408.34 [from_tag]= 39320458 0(5493) PRESENCE:build_str_hdr: expires = 3518 0(5493) PRESENCE:build_str_hdr: subs_expires : 3518 0(5493) PRESENCE: build_str_hdr: headers: Event: presence.winfo Contact: <sip:200.13.225.250:5060> Subscription-State: active;expires=3518 Content-Type: application/watcherinfo+xml 0(5493) PRESENCE:notify: headers:Event: presence.winfo Contact: <sip:200.13.225.250:5060> Subscription-State: active;expires=3518 Content-Type: application/watcherinfo+xml 0(5493) CONTACT = sip:5192901@200.13.254.180:17116 0(5493) presence:uandd_to_uri: uri=sip:5192901@200.13.225.250 0(5493) parse_rr_body(): No body for record-route 0(5493) PRESENCE: notify:Send notify for presence on callback 1(5494) DEBUG: timer routine:4,tl=fc5c9a80 next=0, timeout=353600000 1(5494) DEBUG: retransmission_handler : request resending (t=fc5c9910, NOTIFY si ... ) 1(5494) DEBUG:tm:set_timer: relative timeout is 1000000 1(5494) DEBUG: add_to_tail_of_timer[5]: fc5c9a80 (354600000) 1(5494) DEBUG: retransmission_handler : done 1(5494) DEBUG: timer routine:5,tl=fc5c9a80 next=0, timeout=354600000 1(5494) DEBUG: retransmission_handler : request resending (t=fc5c9910, NOTIFY si ... ) 1(5494) DEBUG:tm:set_timer: relative timeout is 2000000 1(5494) DEBUG: add_to_tail_of_timer[6]: fc5c9a80 (356600000) 1(5494) DEBUG: retransmission_handler : done 1(5494) DEBUG: timer routine:6,tl=fc5c9a80 next=0, timeout=356600000 1(5494) DEBUG: retransmission_handler : request resending (t=fc5c9910, NOTIFY si ... ) 1(5494) DEBUG:tm:set_timer: relative timeout is 4000000 1(5494) DEBUG: add_to_tail_of_timer[7]: fc5c9a80 (360600000) 1(5494) DEBUG: retransmission_handler : done ******************************************************************************** Backtrace of the core: #0 0xfe7f77bc in shm_dup_subs (subs=0xffbff1b0, to_tag=Cannot access memory at address 0x96 ) at notify.c:1819 1819 cb_param->wi_subs->to_user.s = (char*)cb_param + size; (gdb) bt #0 0xfe7f77bc in shm_dup_subs (subs=0xffbff1b0, to_tag=Cannot access memory at address 0x96 ) at notify.c:1819 #1 0xfe7f8238 in notify (subs=0x1779c8, watcher_subs=0x15, n_body=0x0, force_null_body=1024) at notify.c:1562 #2 0xfe7f8f18 in query_db_notify (p_user=0x234, p_domain=0xdff08, event=0xfe809a28 "presence.winfo", watcher_subs=0xffbff1b0, etag=0x0) at notify.c:957 #3 0xfe8028a8 in update_subscription (msg=0x175fd8, subs=0xffbff1b0, rtag=0xffbff228, to_tag_gen=1) at subscribe.c:476 #4 0xfe805080 in handle_subscribe (msg=0x175fd8, str1=0xffbff0e0 "", str2=0x800 <Address 0x800 out of bounds>) at subscribe.c:1252 #5 0x00018378 in do_action (a=0x145270, msg=0x175fd8) at action.c:883 #6 0x00019bb8 in run_action_list (a=0x145270, msg=0x175fd8) at action.c:131 #7 0x000192bc in do_action () #8 0x00019bb8 in run_action_list (a=0x1452c8, msg=0x175fd8) at action.c:131 #9 0x00019df8 in run_top_route (a=0x143e10, msg=0x175fd8) at action.c:111 #10 0x00043b60 in receive_msg (buf=0x12dc00 "", len=1024, rcv_info=0xe0400) at receive.c:156 #11 0x0006f294 in udp_rcv_loop () at udp_server.c:465 #12 0x000338bc in main_loop () at main.c:834 #13 0x00035c44 in main (argc=9, argv=0xe7800) at main.c:1399 (gdb) quit

3 2

Re: [Serusers] Integrate LCS & SER Server
by Jiri Kuthan 07 Nov '07

07 Nov '07

At 18:17 05/11/2007, elilkumaran s wrote: >Sir/Madam > >I'm a System administrator of Institute. > >I use MS LCS 2005 SP1 Enterprise edition & Office Communicator in my work place. >I have linux SER Server & CISCO ATA 186 Telephone adapter. > >I have to run to Messenger client X-Lite for SER Server and Office Communicator for LCS. > >Because I place TP in Guard Office they can call Staff through SER server / X-Lite. >But can't call Office Communicator! Consider this "howto": - log in as administrator in the Windows machine on which LCS is running - start debian installation CD - when asked "replace existing partitions", answer yes - when finished, install SER debian packages, along with mysql - continue testing your SIP end-devices ;-) >CISCO ATA 186 can't registering in LCS, I thing it not possible directly. ( Because LCS use Active Directory Account through Kerbas & NTLM authentication). >So I want integrate LCS & SER Server to facilitate TP call to Office Communicator & Office Communicator call TP. >I use a pool in LCS as normal TCP (not TLS). SER Server also use normal (with out DB -Any one can registrar with any name) > >Could you kindly tell the process for integrate LCS & SER Server? Seriously -- why would you like to have two servers when you can have one, whichever you prefer better? -jiri >Thank You >Your truly >Elil > > > >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

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[OpenSER-Users] mi_fifo leaves little openser_receiver_* fifos on the floor
by Chris Heiser 07 Nov '07

07 Nov '07

I have a cronjob that periodically runs openserctl to gather stats out of openser. I've noticed that in my fifo directory, there are anywhere from 1.5 ~ 3k openser_receiver files laying around. Is there something screwy with mi_fifo in v1.2.2-tls that's preventing it from cleaning up after itself? --Chris

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Re: [OpenSER-Users] OpenSER as a conditional stateless proxy
by lu 07 Nov '07

07 Nov '07

Hi, Iam too interested in this discussion, can some one explicitly elaborate how iam gonna satisfy all the conditions in a situation whereby OpenSER is relaying to Asterisk for local PSTN calls and to another a voip provider for International PSTN calls, ofcourse this applies for any numeric uri. I have a Sipwise generated config with offnet-termination which i tried to edit route [5] 'invite-find-calee' using t_relay() and rewritehostport conditions, but my concern is the role of LCR module which is supposed to be taking care of the GWs especially in route[7] "Request route invite-to-external", will it gonna disrupt/conflict my set conditions?? ######################################################################## # Request route 'invite-find-callee' ######################################################################## route[5] { if(alias_db_lookup("dbaliases")) { xlog("L_INFO", "Callee was aliased - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); #CHECK WHERE THE REQUEST IS FROM???-edit by me if($fd=="sip.domain") { #FORWARD THEM TO ASTERISK -edit by me rewritehostport("asterisk IP"); #AND THEN TAKE THEM TO REQUEST ROUTE 'base-outbound'-edit by me route(3); } else if(uri=~"Sip:00[0-9]+@"){ #RELAY THEM TO INTERNETIONAL PSTN-edit by me t_relay("IP_of_international_PSTN_ provider"); route (3); } if(!is_domain_local("$rd")) { setflag(20); route(7); } if(does_uri_exist()) { xlog("L_INFO", "Callee is local - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(6); } else { xlog("L_INFO", "Callee is not local - M=$rm RURI=$ru F=$fu T=$tu IP= $si ID=$ci\n"); route(7); } exit; } ######################################################################## # Request route 'invite-to-external' ######################################################################## route[7] { if(isflagset(20)) { xlog("L_INFO", "Call to foreign domain - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(3); exit; } if(!isflagset(23)) { # don't allow calls relaying from PSTN to PSTN, if not explicitely forwarded if(uri =~ "^sip:[0-9]+@") { # only route numeric users to PSTN if(!load_gws()) { xlog("L_ERR", "Error loading PSTN gateways - M=$rm RURI=$ru F=$fu T= $tu IP=$si ID=$ci\n"); sl_send_reply("503", "PSTN Termination Currently Unavailable"); exit; } if(!next_gw()) { xlog("L_ERR", "No PSTN gateways available - M=$rm RURI=$ru F=$fu T= $tu IP=$si ID=$ci\n"); sl_send_reply("503", "PSTN Termination Currently Unavailable"); exit; } setflag(21); t_on_failure("1"); route(3); } } xlog("L_INFO", "Call to unknown user - M=$rm RURI=$ru F=$fu T=$tu IP= $si ID=$ci\n"); sl_send_reply("404", "User Not Found"); exit; } WBR, LU. On Tue, 2007-11-06 at 06:35 +0000, users-request(a)lists.openser.org wrote: > Send Users mailing list submissions to > users(a)lists.openser.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.openser.org/cgi-bin/mailman/listinfo/users > or, via email, send a message with subject or body 'help' to > users-request(a)lists.openser.org > > You can reach the person managing the list at > users-owner(a)lists.openser.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Users digest..." > > > Today's Topics: > > 1. Re: Openser as a conditional stateless proxy (Christian Schlatter) > 2. Presence RLS module: initialization error (Schumann Sebastian) > 3. CPL and XML schema (Daniel) > 4. Re: MediaProxy 1.9.0 - Radius (Miles Scruggs) > 5. Re: Is it possible to insert avp to reply message? (Tung Tran) > 6. Re: Is it possible to insert avp to reply message? > (I?aki Baz Castillo) > 7. Cellphones and voicemail (Chris Heiser) > 8. dispatcher with GeoIP (Victor Sergeev) > 9. Re: Cellphones and voicemail (Christian Schlatter) > 10. Re: retransmissions after final response (Vasile Zaharia) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Mon, 05 Nov 2007 10:16:18 -0500 > From: Christian Schlatter <cs(a)unc.edu> > Subject: Re: [OpenSER-Users] Openser as a conditional stateless proxy > To: Padmaja <padmaja.rv(a)vodcalabs.com> > Cc: users(a)lists.openser.org > Message-ID: <472F33C2.90900(a)unc.edu> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Padmaja wrote: > > Hi, > > > > Thanks for the reply, > > Could you provide me the example script for stateless forwarding of > > invite when a call is received from say a 911 operator to the server? > > I suppose this should be implemented by checking the From field of the > > invite and if it is from a certain 911, then it should be forwarded > > statelessly (or is this logic incorrect?). I am new to the Openser > > stuff, so I do not wish to change the script myself. > > > Your logic is correct, you can either check the From: header or the > request URI or any other part of an incoming SIP message to make the > routing decision. > > A simple example looks like: > > if (is_method('INVITE')) > { > # check if from URI domain equals 911operator.com > if ($fd == '911operator.com') > { > # relay stateless using request URI > forward(); > exit; > } > else > { > # relay statefull using request URI > if (!t_relay()) > { > sl_reply_error(); > }; > exit; > } > } > > Please always CC users(a)lists.openser.org so that others can take > advantage of the discussion. > > /Christian > > > > > Thanks again, > > Padmaja > > > > ----- Original Message ----- From: "Christian Schlatter" <cs(a)unc.edu> > > To: "Padmaja" <padmaja.rv(a)vodcalabs.com> > > Cc: <users(a)lists.openser.org> > > Sent: Monday, November 05, 2007 4:53 AM > > Subject: Re: [OpenSER-Users] Openser as a conditional stateless proxy > > > > > >> Padmaja wrote: > >>> Hi all, > >>> > >>> Can any one tell me if the same running instance of Openser can be > >>> configured as a stateful proxy for some user accounts and for some > >>> numbers like emergency services, it acts like a stateless proxy, just > >>> forwarding the request to the destination? > >> > >> Yes, it only depends on what forwarding function is used in the > >> routing script: > >> > >> - t_forward(..) --> stateless > >> > >> - t_relay(..) --> stateful > >> > >> /Christian

2 1

[OpenSER-Users] SIP Topology Question
by Brandon Armstead 07 Nov '07

07 Nov '07

Hey Guys, just looking for some input on this, thanks! Basic Topology ------------------------------- Core Route Servers - Server A, B Registration Server, Collects USRLOC which all reports to the SAME DATABASE. Server C (User A initial registration server) User A -> it will tell me location Server D User A -> it will tell me location Any complications, I should be aware of? So Call Flow will look like so: INBOUND CALL: (Server A|Server B) -> (Server C|Server D) -> (END USER) Now should I be weary of any "lockings" such as a client end such as cable/dsl modem, the ata itself, etc? Thanks. -- Brandon Armstead

1 0

[Serusers] tracker.iptel.org down
by Stefan Sayer 06 Nov '07

06 Nov '07

hi, is there a way to automatically restart it when it crashes? thanks stefan -- Stefan Sayer VoIP Services stefan.sayer(a)iptego.de www.iptego.de iptego GmbH Am Borsigturm 40 13507 Berlin Germany Amtsgericht Charlottenburg, HRB 101010 Geschaeftsfuehrer: Alexander Hoffmann

1 0

[OpenSER-Users] Redundant mysql
by Antonio Reale 06 Nov '07

06 Nov '07

Hi all, I'm trying to setup Openser with two Mysql server configured as Master-Master. Is it possible to configure Openser to read data from the first Mysql server (192.168.1.4) and if that fails to read data from the second Mysql server (192.168.1.5)? If not, which alternatives I have to instruct Openser to use the second Mysql server when the first is down? Thanks in advance. -- Antonio

4 4

[OpenSER-Users] Is it possible to insert avp to reply message?
by Tung Tran 06 Nov '07

06 Nov '07

3 4

[OpenSER-Users] Proxy Authroization Failed
by kiran.bkumar＠wipro.com 06 Nov '07

06 Nov '07

Hi All, I am trying to make a call from userA to userB, but the call is failing and I am seeing "Proxy Authrorization Fail" in the logs. Have attached the logs and configuration file. Please help me debug this problem. Thanks, Kiran. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

1 0

[OpenSER-Users] [IMC module] What are imc tables for?
by Iñaki Baz Castillo 06 Nov '07

06 Nov '07

Hi, in order to use chat rooms I'd like to manage them just from a web interface that handles imc_rooms and imc_members table, but sincerely I don't know how to use them, I don't see doc about hot to use the tables. What I'd like is to have "fixed" chat rooms without the need of creating them with commands, but via web. Could anyone explain me how to use those tables? Please, explain me with this example: - I want a chat room for domain "domain1.org" and name "chat1". - I want three members for that room: - user1(a)domain1.org - user2(a)domain1.org - other_user(a)other_domain.org Could you fill the tables please? ** imc_rooms table ** name domain flag ** imc_members table ** username domain room flag What is flag for? I suppose it's for features as private/public tables and so. Maybe it's not possible to manage those tables and the only way is using chat commands? Thanks for any help and best regards. -- Iñaki Baz Castillo

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[OpenSER-Users] [group module] Realtime in 're_grp' table?
by Iñaki Baz Castillo 06 Nov '07

06 Nov '07

Hi, I need the possibility of users in many groups. This is not allowed by "grp" table where username, domain and grp are UNIQUE key. So It could be valid for me using "re_grp" table and managing numeric groups. But "re_grp" table doesn't allow realtime (if I do a INSERT or UPDATE in the table I need to restat OpenSer). Is not possible to use "re_grp" table in realtime mode? Thanks. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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Re: [OpenSER-Users] mi_xmlrpc.so: undefined symbol: ConfReadServerFile
by Bogdan-Andrei Iancu 06 Nov '07

06 Nov '07

Hi Gregorio, Probably you need to link the module against some other lib provided by xmlrpc-c package. Try to use the latest 1.2.x (stable) or 1.3.x (testing) as they have some auto lib detection. Otherwise, you need to change in modules/mi_xmlrpc/Makefile the list of libraries to be used for linking the module. regards, bogdan Gregorio wrote: > > Hi all. > > I have a problem starting ‘mi_xmlrpc’ module. This is my configuration: > > My system is a CentOS 5. > > OpenSER revision 2327, as suggested in > http://uctimsclient.berlios.de/xcap_server_howto.html. > > ‘xmlrpc-c-0.9.10’ correctly installed. > > ‘xmlrpc-c-devel-0.9.10-1.i386.rpm’ installed but with ‘nodeps’ because > it didn´t find dependencies installed without RPMs. > > Starting openSer I get this: > > Oct 23 07:50:21 localhost openser: ERROR: load_module: could not open > module </usr/local/lib/openser/modules/mi_xmlrpc.so>: > /usr/local/lib/openser/modules/mi_xmlrpc.so: undefined symbol: > ConfReadServerFile > > I have checked ‘mi_xmlrpc.so’ and that symbol is undefined. > > Any idea of what is wrong? > > Thanks a lot. > > Note: some weeks ago I successfully installed this same version of > openSER in a similar machine but It broke L, so I know it can be done. > I must be missing something. > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users >

1 0

[OpenSER-Users] maddr in contact
by Allan Chao ( 趙廷睿 ) 06 Nov '07

06 Nov '07

Hi : I have a flow UE -> gateway (use openser and runs proxy mode) ( ip : 192.168.1.2) -> SIP Proxy ( ip: 192.168.1.3), if i have two user , UE1(192.168.1.5) and UE2(192.168.1.6) send REGISTER request to SIP proxy through gateway, but our gateway add a maddr = "192.168.1.2" string in contact header,so the contact in REGISTER becomes <username@ host ; maddr="192.168.1.2">. now , if UE1 send INVITE message to UE2, how does sip proxy to do if receive INVITE message? it will send invite message to UE2 through gateway ( maddr parameter) ? and does openser has support maddr in contact or not . thx. allan

2 1

[OpenSER-Users] Openser as a conditional stateless proxy
by Padmaja 06 Nov '07

06 Nov '07

Hi all, Can any one tell me if the same running instance of Openser can be configured as a stateful proxy for some user accounts and for some numbers like emergency services, it acts like a stateless proxy, just forwarding the request to the destination? Thanks, Padmaja ----- Original Message ----- From: <users-request(a)lists.openser.org> To: <users(a)lists.openser.org> Sent: Saturday, November 03, 2007 10:36 AM Subject: Users Digest, Vol 30, Issue 7 > Send Users mailing list submissions to > users(a)lists.openser.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.openser.org/cgi-bin/mailman/listinfo/users > or, via email, send a message with subject or body 'help' to > users-request(a)lists.openser.org > > You can reach the person managing the list at > users-owner(a)lists.openser.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Users digest..." > > > Today's Topics: > > 1. proper call tear down (Miles Scruggs) > 2. gateway aspect of LCR module (Miles Scruggs) > 3. Re: gateway aspect of LCR module (Ovidiu Sas) > 4. Re: MediaProxy 1.9.0 - Radius (Jeremy McNamara) > 5. Reg. conditional processing of invite (Padmaja) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Fri, 2 Nov 2007 14:11:56 -0700 > From: Miles Scruggs <miles.scruggs(a)wideideas.com> > Subject: [OpenSER-Users] proper call tear down > To: users(a)lists.openser.org > Message-ID: <10FE258E-087F-4736-9D18-AF367F358C8C(a)wideideas.com> > Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes > > Hi, > > This is probably more SIP related than openSER, but I'll ask it here > anyway > > When we were at VON I brought up the subject of accounting and > verifying proper call tear down, and it seems the ideal solution from > a sip perspective is RFC 4028 (SIP session timers) which tracks the > actual sip session with the UA. What I'm starting to realize is that > support for this is very limited and I haven't found much evidence if > any in the devices where network failures are most likely to occur > ATAs and end user devices. > > This seems to leave me relying on the media stream, but this also has > its issues with network issues occurring when calls are placed on > hold. Does anyone have any good suggestions for accurate accounting > when dealing with end users, or is what I described as good as I'm > going to get, until I find some solutions that support 4028? > > Cheers > > Miles Scruggs > Wide Ideas | Operations | miles.scruggs(a)wideideas.com | 509.525.6522 > ext 4880 > > > > > > > > ------------------------------ > > Message: 2 > Date: Fri, 2 Nov 2007 14:15:30 -0700 > From: Miles Scruggs <miles.scruggs(a)wideideas.com> > Subject: [OpenSER-Users] gateway aspect of LCR module > To: users(a)lists.openser.org > Message-ID: <7051067C-FB06-43AB-841F-227A7AC5C138(a)wideideas.com> > Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes > > Hi guys, > > Great to meet many of you this week at VON and looking forward to > getting together again soon. > > We have a question about the gateway aspect which is part of the LCR > module. We are needing to route to various carriers which the module > handles great, but we are trying to figure out if there is an elegant > way to handle the situation where a carrier provides us with multiple > proxies. For instance if carrier A provides proxy1 and proxy2, > carrier B etc.... When we get the AVP back which will be our stack to > do serial forking the sequence is dependent on the proxy response (or > lack of response) > > For instance if we get a 400 error back from either proxy there isn't > much use in trying the other proxy we need to more along to a > completely different carrier. For 500 errors or time outs will > continue on using the same carrier. The obvious ways to do this: to > abandon the gateway aspect all together and just build logic which we > could use to process what would appear to be a multi demential array > instead of single diminutional array. I'm just writing to see if > there is a nice out of the box way to handle this scenario, also if > not, is this something which developers would be interested in having > added to the module? > > Cheers > > Miles Scruggs > Wide Ideas | Operations | miles.scruggs(a)wideideas.com | 509.525.6522 > ext 4880 > > > > > > > > ------------------------------ > > Message: 3 > Date: Fri, 2 Nov 2007 19:33:44 -0400 > From: "Ovidiu Sas" <sip.nslu(a)gmail.com> > Subject: Re: [OpenSER-Users] gateway aspect of LCR module > To: "Miles Scruggs" <miles.scruggs(a)wideideas.com> > Cc: users(a)lists.openser.org > Message-ID: > <6f497e130711021633r5aa07f69u1d0e5410fd27abef(a)mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > Hi Miles, > > It was nice meeting you at VoN. > > With respect to your question, there are a few options. > First option would be to mess with the avp. As soon as you get a reply > back that should discard the entire gateway group, remove all the > remaining gateway (belonging to that particular group) from the avp > before calling next_gw(). > The second option would be to create a new function: next_gw_group(). > > This is just the top of my head (so there might be some details that > needs to be worked out). > > > On the other hand, there is a new module called carrierroute: > http://www.openser.org/docs/modules/1.3.x/carrierroute.html > Check this module out. It may provide some extra flexibility. > > > Regards, > Ovidiu Sas > > On 11/2/07, Miles Scruggs <miles.scruggs(a)wideideas.com> wrote: >> Hi guys, >> >> Great to meet many of you this week at VON and looking forward to >> getting together again soon. >> >> We have a question about the gateway aspect which is part of the LCR >> module. We are needing to route to various carriers which the module >> handles great, but we are trying to figure out if there is an elegant >> way to handle the situation where a carrier provides us with multiple >> proxies. For instance if carrier A provides proxy1 and proxy2, >> carrier B etc.... When we get the AVP back which will be our stack to >> do serial forking the sequence is dependent on the proxy response (or >> lack of response) >> >> For instance if we get a 400 error back from either proxy there isn't >> much use in trying the other proxy we need to more along to a >> completely different carrier. For 500 errors or time outs will >> continue on using the same carrier. The obvious ways to do this: to >> abandon the gateway aspect all together and just build logic which we >> could use to process what would appear to be a multi demential array >> instead of single diminutional array. I'm just writing to see if >> there is a nice out of the box way to handle this scenario, also if >> not, is this something which developers would be interested in having >> added to the module? >> >> Cheers >> >> Miles Scruggs >> Wide Ideas | Operations | miles.scruggs(a)wideideas.com | 509.525.6522 >> ext 4880 >> >> >> >> >> >> _______________________________________________ >> Users mailing list >> Users(a)lists.openser.org >> http://lists.openser.org/cgi-bin/mailman/listinfo/users >> > > > > ------------------------------ > > Message: 4 > Date: Fri, 02 Nov 2007 22:08:06 -0400 > From: Jeremy McNamara <jj(a)nufone.net> > Subject: Re: [OpenSER-Users] MediaProxy 1.9.0 - Radius > To: Brian Heath <brian(a)telethra.com> > Cc: users(a)lists.openser.org > Message-ID: <472BD806.80506(a)nufone.net> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Brian Heath wrote: >> If you've stopped using MediaProxy, what do you use now? I'm >> currently >> seeking an alternative to MediaProxy, and would appreciate some ideas. >> All >> I need is NAT traversal - but something does doesn't route media (voice) >> through the OpenSER server... >> > > > > As I suggested to Mike, use Asterisk. You get full transcoding > support, proper call accounting and the ability to do calling > applications (ivr, etc) , if you want. > > > > You can fix the NAT problems in the SDP from within openser.cfg. > > > > > Jeremy > > > > ------------------------------ > > Message: 5 > Date: Sat, 3 Nov 2007 10:37:52 +0530 > From: "Padmaja" <padmaja.rv(a)vodcalabs.com> > Subject: [OpenSER-Users] Reg. conditional processing of invite > To: <users(a)lists.openser.org> > Message-ID: <002201c81dd7$7e32ed20$0f32a8c0@systemtest3> > Content-Type: text/plain; format=flowed; charset="iso-8859-1"; > reply-type=original > > Hi all, > > I have an issue, where the sip user dials 911 number and hangs up. The 911 > operator then calls back to the user using the call back command provided > earlier say 8911 (there is a b2bua that interprets the call back number > and > routes the call to the specific user). This number is to be routed via the > openser proxy to the b2bua. However, when the 911 calls back, proxy treats > this number as a username and since there are no registrations to this > 8911 > number (though this entry is present at the proxy, no device has > registered > with this number), it sends back "404 Not found". How can this be > circumvented? > > Thanks, > Padmaja > > > ----- Original Message ----- > From: <users-request(a)lists.openser.org> > To: <users(a)lists.openser.org> > Sent: Saturday, November 03, 2007 3:46 AM > Subject: Users Digest, Vol 30, Issue 6 > > >> Send Users mailing list submissions to >> users(a)lists.openser.org >> >> To subscribe or unsubscribe via the World Wide Web, visit >> http://lists.openser.org/cgi-bin/mailman/listinfo/users >> or, via email, send a message with subject or body 'help' to >> users-request(a)lists.openser.org >> >> You can reach the person managing the list at >> users-owner(a)lists.openser.org >> >> When replying, please edit your Subject line so it is more specific >> than "Re: Contents of Users digest..." >> >> >> Today's Topics: >> >> 1. Core dumped when deploying presence module (Sergio Gutierrez) >> 2. Re: MediaProxy 1.9.0 - Radius (Brian Heath) >> 3. Re: Forking Madness (Chris Heiser) >> 4. Re: MediaProxy 1.9.0 - Radius (Mike O'Connor) >> 5. Re: MediaProxy 1.9.0 - Radius (Ovidiu Sas) >> >> >> ---------------------------------------------------------------------- >> >> Message: 1 >> Date: Fri, 2 Nov 2007 15:59:56 -0500 >> From: "Sergio Gutierrez" <saguti(a)gmail.com> >> Subject: [OpenSER-Users] Core dumped when deploying presence module >> To: users(a)lists.openser.org, users(a)openser.org >> Message-ID: >> <8a4af3e20711021359p3327c719t4ae680de4ac3e155(a)mail.gmail.com> >> Content-Type: text/plain; charset="iso-8859-1" >> >> Hello to all members. >> >> Currently we are trying to deploy presence module in the following >> environment: >> >> 1. Openser 1.2.1 >> 2. Solaris 10 on Sparc System >> 3. libxml 2.6.30 >> >> Openser starts up fine, but when happens any event as a Status change of >> a >> user agent, Openser crashes. >> Below I send an extract from the logfile, and the backtrace of the core. >> Thanks in advance for your attention. >> >> Sergio Guti?rrez >> EPM Telecomunicaciones >> Medellin - Colombia >> >> **************************************************** >>>From Openser logfile. >> >> 0(5493) Nuevo Requerimiento - M=SUBSCRIBE >> RURI=sip:5192901@200.13.225.250F= >> sip:5192902@200.13.225.250 T=sip:5192901@200.13.225.250 >> IP=200.116.28.88ID=YmE2MGVhMGQzMTI3ZmU0YTkzMDJjNjIxODNmMGIwOWE. >> 0(5493) comp_scriptvar: str 20 : 5192901 >> 0(5493) DEBUG:maxfwd:is_maxfwd_present: value = 70 >> 0(5493) parse_headers: flags=ffffffffffffffff >> 0(5493) get_hdr_field: cseq <CSeq>: <1> <SUBSCRIBE> >> 0(5493) DEBUG: get_hdr_body : content_length=0 >> 0(5493) found end of header >> 0(5493) PRESENCE: handle_subscribe: 'expires' found >> 0(5493) PRESENCE: handle_subscribe: lexpire= 3600 >> 0(5493) PRESENCE: handle_subscribe: 'To' header ALREADY PARSED: < >> sip:5192901@200.13.225.250> >> 0(5493) PRESENCE:handle_subscribe: generating to_tag >> 0(5493) >> [p_user]= 5192901 [p_domain]= 200.13.225.250 >> [w_user]= 5192902 [w_domain]= 200.13.225.250 >> 0(5493) [event]= presence >> [staus]= pending >> [expires]= 3600 >> 0(5493) [to_tag]= 10.5493.1194034490.40 >> [from_tag]= 1f783845 >> 0(5493) PRESENCE: update_subscription ... >> 0(5493) >> [p_user]= 5192901 [p_domain]= 200.13.225.250 >> [w_user]= 5192902 [w_domain]= 200.13.225.250 >> 0(5493) [event]= presence >> [staus]= pending >> [expires]= 3600 >> 0(5493) [to_tag]= 10.5493.1194034490.40 >> [from_tag]= 1f783845 >> 0(5493) expires: 3600 >> 0(5493) PRESENCE:update_subscription:Inserting into database: >> n_query_cols:14 >> 0(5493) [0] = to_user 5192901 >> 0(5493) [1] = to_domain 200.13.225.250 >> 0(5493) [2] = from_user 5192902 >> 0(5493) [3] = from_domain 200.13.225.250 >> 0(5493) [4] = event presence >> 0(5493) [5] = event_id >> 0(5493) [6] = callid YmE2MGVhMGQzMTI3ZmU0YTkzMDJjNjIxODNmMGIwOWE. >> 0(5493) [7] = to_tag 10.5493.1194034490.40 >> 0(5493) [8] = from_tag 1f783845 >> 0(5493) [9] = contact sip:5192902@200.116.28.88:47166 >> 0(5493) [10] = status pending >> 0(5493) [11] = cseq 1 >> 0(5493) [12] = expires 1194038090 >> 0(5493) [13] = version 0 >> 0(5493) parse_headers: flags=ffffffffffffffff >> 0(5493) check_via_address(200.116.28.88, 192.168.0.13, 0) >> 0(5493) DBG:sl:run_sl_callbacks: callback id 0 entered >> 0(5493) trace_sl_onreply_out: trace off... >> 0(5493) PRESENCE: get_subs_dialog:n= 1 >> 0(5493) PRESENCE:notify:dialog informations: >> 0(5493) >> [p_user]= 5192901 [p_domain]= 200.13.225.250 >> [w_user]= 5192901 [w_domain]= 200.13.225.250 >> 0(5493) [event]= presence.winfo >> [staus]= active >> [expires]= 3518 >> 0(5493) [to_tag]= 10.5493.1194034408.34 >> [from_tag]= 39320458 >> 0(5493) presence:uandd_to_uri: uri=sip:5192901@200.13.225.250 >> 0(5493) presence:uandd_to_uri: uri=sip:5192902@200.13.225.250 >> 0(5493) presence:uandd_to_uri: uri=sip:5192901@200.13.225.250 >> 0(5493) PRESENCE: notify: build notify to user= 5192901 domain= >> 200.13.225.250 for event= presence.winfo >> 0(5493) >> [p_user]= 5192901 [p_domain]= 200.13.225.250 >> [w_user]= 5192901 [w_domain]= 200.13.225.250 >> 0(5493) [event]= presence.winfo >> [staus]= active >> [expires]= 3518 >> 0(5493) [to_tag]= 10.5493.1194034408.34 >> [from_tag]= 39320458 >> 0(5493) PRESENCE:build_str_hdr: expires = 3518 >> 0(5493) PRESENCE:build_str_hdr: subs_expires : 3518 >> 0(5493) PRESENCE: build_str_hdr: headers: >> Event: presence.winfo >> Contact: <sip:200.13.225.250:5060> >> Subscription-State: active;expires=3518 >> Content-Type: application/watcherinfo+xml >> >> 0(5493) PRESENCE:notify: headers:Event: presence.winfo >> Contact: <sip:200.13.225.250:5060> >> Subscription-State: active;expires=3518 >> Content-Type: application/watcherinfo+xml >> >> 0(5493) CONTACT = sip:5192901@200.13.254.180:17116 >> 0(5493) presence:uandd_to_uri: uri=sip:5192901@200.13.225.250 >> 0(5493) parse_rr_body(): No body for record-route >> 0(5493) PRESENCE: notify:Send notify for presence on callback 1(5494) >> DEBUG: timer routine:4,tl=fc5c9a80 next=0, timeout=353600000 >> 1(5494) DEBUG: retransmission_handler : request resending (t=fc5c9910, >> NOTIFY si ... ) >> 1(5494) DEBUG:tm:set_timer: relative timeout is 1000000 >> 1(5494) DEBUG: add_to_tail_of_timer[5]: fc5c9a80 (354600000) >> 1(5494) DEBUG: retransmission_handler : done >> 1(5494) DEBUG: timer routine:5,tl=fc5c9a80 next=0, timeout=354600000 >> 1(5494) DEBUG: retransmission_handler : request resending (t=fc5c9910, >> NOTIFY si ... ) >> 1(5494) DEBUG:tm:set_timer: relative timeout is 2000000 >> 1(5494) DEBUG: add_to_tail_of_timer[6]: fc5c9a80 (356600000) >> 1(5494) DEBUG: retransmission_handler : done >> 1(5494) DEBUG: timer routine:6,tl=fc5c9a80 next=0, timeout=356600000 >> 1(5494) DEBUG: retransmission_handler : request resending (t=fc5c9910, >> NOTIFY si ... ) >> 1(5494) DEBUG:tm:set_timer: relative timeout is 4000000 >> 1(5494) DEBUG: add_to_tail_of_timer[7]: fc5c9a80 (360600000) >> 1(5494) DEBUG: retransmission_handler : done >> >> >> ******************************************************************************** >> Backtrace of the core: >> >> #0 0xfe7f77bc in shm_dup_subs (subs=0xffbff1b0, to_tag=Cannot access >> memory >> at address 0x96 >> ) at notify.c:1819 >> 1819 cb_param->wi_subs->to_user.s = (char*)cb_param + size; >> (gdb) bt >> #0 0xfe7f77bc in shm_dup_subs (subs=0xffbff1b0, to_tag=Cannot access >> memory >> at address 0x96 >> ) at notify.c:1819 >> #1 0xfe7f8238 in notify (subs=0x1779c8, watcher_subs=0x15, n_body=0x0, >> force_null_body=1024) at notify.c:1562 >> #2 0xfe7f8f18 in query_db_notify (p_user=0x234, p_domain=0xdff08, >> event=0xfe809a28 "presence.winfo", watcher_subs=0xffbff1b0, etag=0x0) >> at notify.c:957 >> #3 0xfe8028a8 in update_subscription (msg=0x175fd8, subs=0xffbff1b0, >> rtag=0xffbff228, to_tag_gen=1) at subscribe.c:476 >> #4 0xfe805080 in handle_subscribe (msg=0x175fd8, str1=0xffbff0e0 "", >> str2=0x800 <Address 0x800 out of bounds>) at subscribe.c:1252 >> #5 0x00018378 in do_action (a=0x145270, msg=0x175fd8) at action.c:883 >> #6 0x00019bb8 in run_action_list (a=0x145270, msg=0x175fd8) at >> action.c:131 >> #7 0x000192bc in do_action () >> #8 0x00019bb8 in run_action_list (a=0x1452c8, msg=0x175fd8) at >> action.c:131 >> #9 0x00019df8 in run_top_route (a=0x143e10, msg=0x175fd8) at >> action.c:111 >> #10 0x00043b60 in receive_msg (buf=0x12dc00 "", len=1024, >> rcv_info=0xe0400) >> at receive.c:156 >> #11 0x0006f294 in udp_rcv_loop () at udp_server.c:465 >> #12 0x000338bc in main_loop () at main.c:834 >> #13 0x00035c44 in main (argc=9, argv=0xe7800) at main.c:1399 >> (gdb) quit >>

4 3

[OpenSER-Users] dispatcher with GeoIP
by Victor Sergeev 06 Nov '07

06 Nov '07

Hello, I'm wondering if its a good idea to tie dispatcher module with GeoIP. The point is to select gateway group depending on IP from which user making a call. In such way you can select the gateway which should be closed to the user. E.g. some users will use USA gateways another EU or Asia gateways. Victor

2 1

[OpenSER-Users] CPL and XML schema
by Daniel 06 Nov '07

06 Nov '07

Hi everybody, Is it possible for OpenSER to use XML Schema for CPL instead of DTD file ? It would be more interesting use XLM schema, isn't it? Thanx daniel -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

2 1

Re: [OpenSER-Users] Presence RLS module: initialization error
by Schumann Sebastian 06 Nov '07

06 Nov '07

Hi Anca Thank you, it works with pua.so and db parameter for pua included. Regards Sebastian -----Original Message----- From: Anca Vamanu [mailto:anca@voice-system.ro] Sent: Tuesday, November 06, 2007 12:11 PM To: Schumann Sebastian Cc: users(a)openser.org Subject: Re: [OpenSER-Users] Presence RLS module: initialization error Hello, The bind_libxml_command is from pua modules, that rls depends on. You must load this also. There was a miss out in the documentation that is now fixed. regards, Anca Vamanu Schumann Sebastian wrote: > Dear all > > I tried to configure OpenSER as RLS. I have error message using rls > module. I did setting acc. openser.org module documentation for rls, > including library functionality of presence and so on. > > Trying to start OpenSER, I get the follwing error regarding RLS: > > Nov 5 17:29:57 [22819] DBG:core:find_cmd_export_t: <bind_libxml_api> > not found > Nov 5 17:29:57 [22819] ERROR:rls:mod_init: can't import bind_libxml_api > Nov 5 17:29:57 [22819] ERROR:core:init_mod: failed to initialize > module rls > Nov 5 17:29:57 [22819] ERROR:core:main: error while initializing modules > > Can anyone help, where to get bind_libxml_api ? > > Thank you. > Sebastian > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org > http://lists.openser.org/cgi-bin/mailman/listinfo/users >

1 0

[OpenSER-Users] Presence RLS module: initialization error
by Schumann Sebastian 06 Nov '07

06 Nov '07

Dear all I tried to configure OpenSER as RLS. I have error message using rls module. I did setting acc. openser.org module documentation for rls, including library functionality of presence and so on. Trying to start OpenSER, I get the follwing error regarding RLS: Nov 5 17:29:57 [22819] DBG:core:find_cmd_export_t: <bind_libxml_api> not found Nov 5 17:29:57 [22819] ERROR:rls:mod_init: can't import bind_libxml_api Nov 5 17:29:57 [22819] ERROR:core:init_mod: failed to initialize module rls Nov 5 17:29:57 [22819] ERROR:core:main: error while initializing modules Can anyone help, where to get bind_libxml_api ? Thank you. Sebastian

2 1

[OpenSER-Users] Announcement: devel.openser.org available
by Henning Westerholt 06 Nov '07

06 Nov '07

Hello, i'm happy to announce that the 'devel.openser.org' server, sponsored from my company 1&1 Internet AG, is now available. This machine hosts services for developers, but some of them are probably also useful for users: 1. Daily creation of debian builds (etch i386 and amd64) of the trunk for easily testing. 2. A read only svn mirror of the complete repository for better subversion performance especially outside the USA. 3. The build and testing framework "buildbot" for continous monitoring of the status from trunk and branches. 4. Doxygen code documentation with graph generation for better understanding of the code and module dependencies. You'll find at http://devel.openser.org/ the indivdual addresses of the services. This page will probably be moved to www.openser.org after the migration of these server. Best regards, Henning Westerholt -- Henning Westerholt - Development Consumer Products / DSL Core 1&1 Internet AG, Ernst-Frey-Str. 9, 76135 Karlsruhe, Deutschland Amtsgericht Montabaur HRB 6484 * Vorstand: Henning Ahlert, Ralph Dommermuth, Matthias Ehrlich, Andreas Gauger, Thomas Gottschlich, Matthias Greve, Robert Hoffmann, Norbert Lang, Achim Weiss * Aufsichtsratsvorsitzender: Michael Scheeren

1 1

Re: [OpenSER-Users] retransmissions after final response
by Vasile Zaharia 06 Nov '07

06 Nov '07

Hi Bogdan, all, I've made more digging and I need your opinion on my findings. Pre-condition: run an AS in forked mode with 2 children. 1. AS sends a request (RQ). 2. AS receives a "408 Request Timeout" reply (RPL) for RQ. 3. After some time (about 30 seconds) AS receives another "408 Request Timeout", initiated at another network node. 4. At this point, you want to send another request from AS. Result: Very often, the replies for the last request cannot be matched with its transaction in "tm" module, so request retransmissions occur. After a few days I've found that in "t_reply_matching" function from "t_lookup.c" file, just before returning (at the end of the function, not in the case when the reply gets matched) T is set to zero (set_t(0)). So, at step 3, the reply cannot be matched with an existing transaction and T is set to zero in a child process. If a reply of the request sent at step 4 gets handled by exactly the child process that last set T to zero, the transaction will not get matched (see t_check() and reply_received() functions), so the retransmission timer will hit => request retransmissions!!! I think the problem is that T is set to zero in t_reply_matching() when a reply cannot be matched. I've changed that to set_t(T_UNDEFINED) and everything seems to work just fine. What do you say Bogdan? Is this change safe? Is this the fix for my problem? Thanks, Sica. ----- Original Message ---- From: Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> To: Vasile Zaharia <sicavz(a)yahoo.com> Cc: users(a)openser.org Sent: Tuesday, October 30, 2007 9:49:38 AM Subject: Re: [OpenSER-Users] retransmissions after final response Salut Sica, So you say that openser does retransmission even after receiving the 200 OK for the request? are you sure it is about the same transaction? Could you provide the debug log showing also the retransmission events? (just send me the entire log). Salutari, Bogdan Vasile Zaharia wrote: > Salut Bogdan, > > Thank you for your response. > I think the reply is correct, because in the log file I can see: > > 2007-10-15 06:28:48.545|t_lookup.c:824|t_reply_matching|DEBUG: t_reply_matching: reply matched (T=0xb5e98bd8)!\n > 2007-10-15 06:28:48.545|t_lookup.c:924|t_check|DEBUG: t_check: end=0xb5e98bd8\n > 2007-10-15 06:28:48.546|t_hooks.c:203|run_trans_callbacks|DBG: trans=0xb5e98bd8, callback type 256, id 0 entered\n > 2007-10-15 06:28:52.754|timer.c:434|wait_handler|DEBUG: wait_handler : removing 0xb5e98bd8 from table \n > 2007-10-15 06:28:52.754|timer.c:714|remove_timer_unsafe|DEBUG: unlinking timer: tl=0xb5e98d54, timeout=364, group=0\n > 2007-10-15 06:28:52.753|timer.c:239|delete_cell|DEBUG: delete transaction 0xb5e98bd8\n > 2007-10-15 06:28:52.753|timer.c:443|wait_handler|DEBUG: wait_handler : done\n > > 2007-10-15 06:29:15.378|t_lookup.c:824|t_reply_matching|DEBUG: t_reply_matching: reply matched (T=0xb5e98bd8)!\n > 2007-10-15 06:29:15.378|t_lookup.c:924|t_check|DEBUG: t_check: end=0xb5e98bd8\n > 2007-10-15 06:29:15.378|t_hooks.c:203|run_trans_callbacks|DBG: trans=0xb5e98bd8, callback type 256, id 0 entered\n > 2007-10-15 06:29:19.871|timer.c:434|wait_handler|DEBUG: wait_handler : removing 0xb5e98bd8 from table \n > 2007-10-15 06:29:19.871|timer.c:714|remove_timer_unsafe|DEBUG: unlinking timer: tl=0xb5e98d54, timeout=391, group=0\n > 2007-10-15 06:29:19.871|timer.c:239|delete_cell|DEBUG: delete transaction 0xb5e98bd8\n > 2007-10-15 06:29:19.871|timer.c:443|wait_handler|DEBUG: wait_handler : done\n > > 2007-10-15 06:29:21.853|t_lookup.c:824|t_reply_matching|DEBUG: t_reply_matching: reply matched (T=0xb5e98bd8)!\n [22569] > 2007-10-15 06:29:21.853|t_lookup.c:924|t_check|DEBUG: t_check: end=0xb5e98bd8\n [22569] > 2007-10-15 06:29:25.894|timer.c:434|wait_handler|DEBUG: wait_handler : removing 0xb5e98bd8 from table \n > 2007-10-15 06:29:25.894|timer.c:714|remove_timer_unsafe|DEBUG: unlinking timer: tl=0xb5e98d54, timeout=397, group=0\n > 2007-10-15 06:29:25.894|timer.c:239|delete_cell|DEBUG: delete transaction 0xb5e98bd8\n > 2007-10-15 06:29:25.894|timer.c:443|wait_handler|DEBUG: wait_handler : done\n > > 2007-10-15 06:29:50.049|t_lookup.c:824|t_reply_matching|DEBUG: t_reply_matching: reply matched (T=0xb5e98bd8)!\n > 2007-10-15 06:29:50.049|t_lookup.c:924|t_check|DEBUG: t_check: end=0xb5e98bd8\n > 2007-10-15 06:29:50.051|t_hooks.c:203|run_trans_callbacks|DBG: trans=0xb5e98bd8, callback type 256, id 0 entered\n > 2007-10-15 06:29:55.014|timer.c:434|wait_handler|DEBUG: wait_handler : removing 0xb5e98bd8 from table \n > 2007-10-15 06:29:55.014|timer.c:714|remove_timer_unsafe|DEBUG: unlinking timer: tl=0xb5e98d54, timeout=425, group=0\n > 2007-10-15 06:29:55.014|timer.c:239|delete_cell|DEBUG: delete transaction 0xb5e98bd8\n > 2007-10-15 06:29:55.014|timer.c:443|wait_handler|DEBUG: wait_handler : done\n > > 2007-10-15 06:31:45.071|t_lookup.c:824|t_reply_matching|DEBUG: t_reply_matching: reply matched (T=0xb5e98bd8)!\n > 2007-10-15 06:31:45.071|t_lookup.c:924|t_check|DEBUG: t_check: end=0xb5e98bd8\n > 2007-10-15 06:31:45.071|t_hooks.c:203|run_trans_callbacks|DBG: trans=0xb5e98bd8, callback type 256, id 0 entered\n > 2007-10-15 06:31:49.834|timer.c:434|wait_handler|DEBUG: wait_handler : removing 0xb5e98bd8 from table \n > 2007-10-15 06:31:49.834|timer.c:714|remove_timer_unsafe|DEBUG: unlinking timer: tl=0xb5e98d54, timeout=540, group=0\n > 2007-10-15 06:31:49.834|timer.c:239|delete_cell|DEBUG: delete transaction 0xb5e98bd8\n > 2007-10-15 06:31:49.834|timer.c:443|wait_handler|DEBUG: wait_handler : done\n > > Maybe I don't understand it correctly? It's about T=0xb5e98bd8. It gets matched, removed, deleted. Many times. > > The situation was reproduced using 2 different clients: sipp and a proprietary client (using two opensource libraries: osip and exosip). > > What do you think? > > Thanks, > Sica. > > > ----- Original Message ---- > From: Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> > To: Vasile Zaharia <sicavz(a)yahoo.com> > Cc: users(a)openser.org > Sent: Tuesday, October 30, 2007 6:35:10 AM > Subject: Re: [OpenSER-Users] retransmissions after final response > > > Salut Sica, > > The problem may reside in two places: > 1) the reply is not correct and the client is not able to match it > to the sent request (and keeps retransmitting) > 2) the client is not able to match (due whatever other bugs) the > reply. > > Salutari, > Bogdan > > Vasile Zaharia wrote: > >> Hi all, >> >> >> I'm developing a Presence AS OpenSER module (derived from the OpenSER >> > Presence module) > >> and, from time to time, it happens the next situation: >> - AS sends a NOTIFY message >> - the client receives the NOTIFY and responds with 200 OK >> - AS receives 200 OK response, but it starts to retransmit the >> > original NOTIFY and it > >> retransmits it until it gets back a 408 Request timeout. >> >> Please note that I use "tm" module for sending SIP messages, so I >> > don't > >> handle retransmissions explicitly in my module. >> Also, note that it does not happen all the time, but from time to >> > time. > >> Does anyone else met a similar situation? >> >> >> Thanks, >> Sica. >> >> >> >> >> __________________________________________________ >> Do You Yahoo!? >> Tired of spam? Yahoo! Mail has the best spam protection around >> http://mail.yahoo.com >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >> >> >> > > > > > > __________________________________________________ > Do You Yahoo!? > Tired of spam? Yahoo! Mail has the best spam protection around > http://mail.yahoo.com > > __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

1 0

[OpenSER-Users] Cellphones and voicemail
by Chris Heiser 06 Nov '07

06 Nov '07

Trying to send a user's call to their cellphone can be problematic, especially when their phone is off, bad signal, etc... I know some carriers send q931 messages back when a call is redirected (sometimes), but not all do. Is it possible to make a determination in OpenSER (if we know we're sending to a cell phone) that the call was picked up too quickly? That's really the ugly problem. If I could say, "Oh yeah, you're cellphone picked up in less than 1 second, a real person probably didn't do that." and drop that fork, that would be awesome. --Chris

2 1

[Serusers] Anyone doing something to handle RFC3326?
by SIP 05 Nov '07

05 Nov '07

We've been looking recently at RFC 3326 (Reason header fields for SIP) primarily to help with logging in some of our UAs. While I know that SER can essentially tack on any header anywhere, I'm wondering if anyone's come up with a solution for handling something like the Call Completed Elsewhere reason tag: Reason: SIP ;cause=200 ;text="Call completed elsewhere" Essentially, this would be sent in a CANCEL message to all clients that did NOT pick up the SIP call. Since SER forks off a separate INVITE to any contact in the location table matching the intended recipient, it would be nice to fire off a reason for a cancel to some UAs which would then know not to bother logging a call as 'missed' in the case of another phone picking it up. I'm sure there's a way to hack this using the current functionality (or if not, a slight modification of current functionality), but if anyone's done it and has a code snippet, that would make life easier. N.

2 1

Re: [OpenSER-Users] MediaProxy 1.9.0 - Radius
by Brian Heath 05 Nov '07

05 Nov '07

If you've stopped using MediaProxy, what do you use now? I'm currently seeking an alternative to MediaProxy, and would appreciate some ideas. All I need is NAT traversal - but something does doesn't route media (voice) through the OpenSER server... Thanks, Brian ----- Original Message ----- From: "Jeremy McNamara" <jj(a)nufone.net> To: "Mike O'Connor" <mike(a)pineview.net> Cc: <users(a)lists.openser.org> Sent: Friday, November 02, 2007 7:30 AM Subject: Re: [OpenSER-Users] MediaProxy 1.9.0 - Radius > Mike O'Connor wrote: >> Hi Guys >> >> I really think there is something wrong with MediaProxy, I can not get >> the radius or the mysql accounting to report any details, as best as I >> can see there is never any attempt to even try. >> > > > > I totally scratched MediaProxy off of the list when the author of the > code told me to take my questions to the openser mailing list. > Good luck finding support - I even offered to pay and was completely > ignored. > > > My suggestion would be to use Asterisk as your media proxy and use one > of the various radius connectors (if you absolutely require radius) > > > > > Jeremy > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org > http://lists.openser.org/cgi-bin/mailman/listinfo/users

6 7

Re: [OpenSER-Users] Openser as a conditional stateless proxy
by Christian Schlatter 05 Nov '07

05 Nov '07

Padmaja wrote: > Hi, > > Thanks for the reply, > Could you provide me the example script for stateless forwarding of > invite when a call is received from say a 911 operator to the server? > I suppose this should be implemented by checking the From field of the > invite and if it is from a certain 911, then it should be forwarded > statelessly (or is this logic incorrect?). I am new to the Openser > stuff, so I do not wish to change the script myself. Your logic is correct, you can either check the From: header or the request URI or any other part of an incoming SIP message to make the routing decision. A simple example looks like: if (is_method('INVITE')) { # check if from URI domain equals 911operator.com if ($fd == '911operator.com') { # relay stateless using request URI forward(); exit; } else { # relay statefull using request URI if (!t_relay()) { sl_reply_error(); }; exit; } } Please always CC users(a)lists.openser.org so that others can take advantage of the discussion. /Christian > > Thanks again, > Padmaja > > ----- Original Message ----- From: "Christian Schlatter" <cs(a)unc.edu> > To: "Padmaja" <padmaja.rv(a)vodcalabs.com> > Cc: <users(a)lists.openser.org> > Sent: Monday, November 05, 2007 4:53 AM > Subject: Re: [OpenSER-Users] Openser as a conditional stateless proxy > > >> Padmaja wrote: >>> Hi all, >>> >>> Can any one tell me if the same running instance of Openser can be >>> configured as a stateful proxy for some user accounts and for some >>> numbers like emergency services, it acts like a stateless proxy, just >>> forwarding the request to the destination? >> >> Yes, it only depends on what forwarding function is used in the >> routing script: >> >> - t_forward(..) --> stateless >> >> - t_relay(..) --> stateful >> >> /Christian >> >

1 0

[OpenSER-Users] proper call tear down
by Miles Scruggs 05 Nov '07

05 Nov '07

Hi, This is probably more SIP related than openSER, but I'll ask it here anyway When we were at VON I brought up the subject of accounting and verifying proper call tear down, and it seems the ideal solution from a sip perspective is RFC 4028 (SIP session timers) which tracks the actual sip session with the UA. What I'm starting to realize is that support for this is very limited and I haven't found much evidence if any in the devices where network failures are most likely to occur ATAs and end user devices. This seems to leave me relying on the media stream, but this also has its issues with network issues occurring when calls are placed on hold. Does anyone have any good suggestions for accurate accounting when dealing with end users, or is what I described as good as I'm going to get, until I find some solutions that support 4028? Cheers Miles Scruggs Wide Ideas | Operations | miles.scruggs(a)wideideas.com | 509.525.6522 ext 4880

2 1

Re: [Serusers] [Serdev] need help - Call Forwarding
by Jiri Kuthan 05 Nov '07

05 Nov '07

This should in (poor) theory work. I would check by watching the mysql port number traffic what data gets actually fetched before making the next step. At 13:01 01/11/2007, Piyush.Bansal(a)relianceada.com wrote: >Hi All, > >We're trying to test Call Forwarding Unconditional (CFU) using SER-2.0.0-rc1. We're adding the subscriber's attribute using following command - >./ser_ctl attrs set 50001(a)192.168.112.14 fwd_always_target=50(a)192.168.112.14 > >After that we tried to call to 50001. After receiving the call, SER fails to fetch the attribute (fwd_always_target) value from the DB. How do you know SER fails to fetch it from DB? -jiri >Even, the check "if ($tu.fwd_always_target)" fails in this case.. > >The config file is also attahced with the mail. > > > >Can anybody tell me why this is happening?????? Plz reply ASAP..... > > >[] > > > >The information contained in this e-mail message is intended only for the use of the individual or entity to which it is addressed. If you are not the intended recipient, you should return it to the sender immediately. Please note that while we scan all e-mails for viruses we cannot guarantee that any e-mail is virus-free and accept no liability for any damage caused by any virus transmitted by this email. > > > >_______________________________________________ >Serdev mailing list >Serdev(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serdev -- Jiri Kuthan http://iptel.org/~jiri/

1 0

[Serusers] [need help] SER Connectvity
by Piyush.Bansal＠relianceada.com 03 Nov '07

03 Nov '07

Hi All, We are using SER-0.9.6. We want it to be configured in a way that it can have two interfaces towards two different application. One interface will be over TLS and another will be over UDP. Can we do that???? If yes, then how??? thanx in advance..... regds, --Piyush Bansal The information contained in this e-mail message is intended only for the use of the individual or entity to which it is addressed. If you are not the intended recipient, you should return it to the sender immediately. Please note that while we scan all e-mails for viruses we cannot guarantee that any e-mail is virus-free and accept no liability for any damage caused by any virus transmitted by this email.

1 0

[Serusers] serctl
by guojm 03 Nov '07

03 Nov '07

Spam detection software, running on the system "mail.iptel.org", has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: I USER SER2.0, RUN serctl ps show "500 Command 'ps' not found" why [...] Content analysis details: (7.6 points, 5.0 required) pts rule name description ---- ---------------------- -------------------------------------------------- -0.0 SPF_HELO_PASS SPF: HELO matches SPF record -0.0 SPF_PASS SPF: sender matches SPF record 0.0 BAYES_50 BODY: Bayesian spam probability is 40 to 60% [score: 0.5027] 0.2 MIME_BASE64_NO_NAME RAW: base64 attachment does not have a file name 1.9 MIME_BASE64_TEXT RAW: Message text disguised using base64 encoding 0.2 MIME_BASE64_BLANKS RAW: Extra blank lines in base64 encoding 1.9 DNS_FROM_RFC_BOGUSMX RBL: Envelope sender in bogusmx.rfc-ignorant.org 0.2 DNS_FROM_RFC_ABUSE RBL: Envelope sender in abuse.rfc-ignorant.org 1.4 DNS_FROM_RFC_WHOIS RBL: Envelope sender in whois.rfc-ignorant.org 1.7 DNS_FROM_RFC_POST RBL: Envelope sender in postmaster.rfc-ignorant.org

1 0

[OpenSER-Users] gateway aspect of LCR module
by Miles Scruggs 02 Nov '07

02 Nov '07

Hi guys, Great to meet many of you this week at VON and looking forward to getting together again soon. We have a question about the gateway aspect which is part of the LCR module. We are needing to route to various carriers which the module handles great, but we are trying to figure out if there is an elegant way to handle the situation where a carrier provides us with multiple proxies. For instance if carrier A provides proxy1 and proxy2, carrier B etc.... When we get the AVP back which will be our stack to do serial forking the sequence is dependent on the proxy response (or lack of response) For instance if we get a 400 error back from either proxy there isn't much use in trying the other proxy we need to more along to a completely different carrier. For 500 errors or time outs will continue on using the same carrier. The obvious ways to do this: to abandon the gateway aspect all together and just build logic which we could use to process what would appear to be a multi demential array instead of single diminutional array. I'm just writing to see if there is a nice out of the box way to handle this scenario, also if not, is this something which developers would be interested in having added to the module? Cheers Miles Scruggs Wide Ideas | Operations | miles.scruggs(a)wideideas.com | 509.525.6522 ext 4880

2 1

[Serusers] Damaged message body
by Stefan.Brozinski＠materna.de 02 Nov '07

02 Nov '07

Hello everybody there is a bug in SER that removes the last byte from a message body if this happens to be a binary zero. This may be the case when the Content-Type is something binary (i.e. *not* using "Content-Type: text/something"). The bad code is in udp_server.c. It is present in SER 0.9.x a well as in 2.0. I have reported it in the bug tracker as SER-324 and attached a comment which points out the bad section of the code. How shall I proceed to fix this bug? Best regards Stefan

2 1

[OpenSER-Users] Set up two SIP to PSTN calls and then connect them
by CSB 02 Nov '07

02 Nov '07

Is there any update regarding the click2dial plugin that was planned to be introduced to the trunk? Regards Cameron

4 8

[OpenSER-Users] Traffic
by Gerson A. Matiolli 02 Nov '07

02 Nov '07

Hi, all I am using Openser 1.2.2 - tls. I have 400 registered users. Everything works well as traffic is low. If traffic is high, the calls are not completed (Busy tone) Can anyone help me?

3 3

[OpenSER-Users] Reg. Subscriber table
by Padmaja 02 Nov '07

02 Nov '07

Hi all, I am using the openserctl command to add users to the Openser database and I can verify if these numbers are added to the Subscriber table in the mysql database.However, if I try to look into the Subscriber table through Webmin, I do not see all the users but only a few. What should I do to reflect all the users through webmin as well? Thank You, Padmaja ----- Original Message ----- From: <users-request(a)lists.openser.org> To: <users(a)lists.openser.org> Sent: Friday, November 02, 2007 3:27 PM Subject: Users Digest, Vol 30, Issue 2 > Send Users mailing list submissions to > users(a)lists.openser.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.openser.org/cgi-bin/mailman/listinfo/users > or, via email, send a message with subject or body 'help' to > users-request(a)lists.openser.org > > You can reach the person managing the list at > users-owner(a)lists.openser.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Users digest..." > > > Today's Topics: > > 1. Re: Set up two SIP to PSTN calls and then connect them > (Andreas Granig) > 2. Traffic (Gerson A. Matiolli) > 3. Re: How to expose the expires value in REGISTER (Robert Dyck) > 4. Beep in audio stream. (Marc Dirix) > 5. Is it possible to insert avp to reply message? (Tung Tran) > 6. Re: Set up two SIP to PSTN calls and then connect them > (Bogdan-Andrei Iancu) > 7. Re: Set up two SIP to PSTN calls and then connect them (CSB) > 8. Re: Is it possible to insert avp to reply message? > (I?aki Baz Castillo) > 9. maddr in contact (Allan Chao ( ??? )) > 10. Re: Set up two SIP to PSTN calls and then connect them > (I?aki Baz Castillo) > 11. Re: Traffic (Henning Westerholt) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Thu, 01 Nov 2007 13:07:42 +0100 > From: Andreas Granig <agranig(a)sipwise.com> > Subject: Re: [OpenSER-Users] Set up two SIP to PSTN calls and then > connect them > To: CSB <kjcsb(a)xnet.co.nz> > Cc: users(a)lists.openser.org > Message-ID: <4729C18E.7040507(a)sipwise.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > It's already included, see > http://svn.berlios.de/viewcvs/sems/trunk/apps/click2dial/ > > Andreas > > CSB wrote: >> Is there any update regarding the click2dial plugin that was planned to >> be introduced to the trunk? >> >> >> >> Regards >> >> >> >> Cameron >> > > > > ------------------------------ > > Message: 2 > Date: Thu, 01 Nov 2007 10:35:49 -0200 > From: "Gerson A. Matiolli" <gerson(a)cambridgetelecom.com.br> > Subject: [OpenSER-Users] Traffic > To: users(a)lists.openser.org > Message-ID: <1193920549.5276.9.camel@jupiter2> > Content-Type: text/plain > > Hi, all > > I am using Openser 1.2.2 - tls. > > I have 400 registered users. > > Everything works well as traffic is low. > > If traffic is high, the calls are not completed (Busy tone) > > Can anyone help me? > > > > > > > > > ------------------------------ > > Message: 3 > Date: Thu, 1 Nov 2007 09:12:20 -0700 > From: Robert Dyck <rob.dyck(a)telus.net> > Subject: Re: [OpenSER-Users] How to expose the expires value in > REGISTER > To: Christian Schlatter <cs(a)unc.edu> > Cc: users(a)lists.openser.org > Message-ID: <200711010912.20409.rob.dyck(a)telus.net> > Content-Type: text/plain; charset="iso-8859-1" > > On Wednesday 31 October 2007, Christian Schlatter wrote: >> Robert Dyck wrote: >> > I am wondering how to expose and test the value of the expires >> > parameter >> > in a REGISTER request. >> > >> > I am experimenting with openser as the basis for a home phone network. >> > I >> > use multiple devices with the same user ID. They register locally ( >> > with >> > no reply ) and with an external service provider. The contacts are >> > mangled to show the public address of openser. Multiple registrations >> > result in a single AOR at the external registrar. Incoming calls from >> > the >> > outside are forked and ring the local phones. Local phones can also >> > call >> > each other without the hairpin problem associated with STUN enabled >> > phones. >> > >> > The problem is that a softphone will deregister when it is closed or >> > its >> > profile changes. This would deregister the AOR at the external >> > registrar. >> > The remaining phones could not receive calls from the outside until >> > they >> > refreshed their registrations. >> > >> > I would like to prevent deregistration at the external registrar unless >> > the phone that was deregistering was the only remaining one. The first >> > step would be to identify REGISTER messages where the expires value is >> > equal to zero. >> >> Both 'Expires' header and 'expires' contact uri parameter have to be >> checked like e.g. >> >> if ((is_present_hf("Expires") && $(hdr(Expires){s.int}) == 0) || >> ($(ct{param.value,expires}) == '0')) >> { >> # someone tries to unregister >> } >> >> Have a look at >> http://www.openser.org/dokuwiki/doku.php/transformations:1.2.x if you're >> not familiar with the PV transformations introduced with 1.2. > > I am indeed unfamiliar with PV transformations. I will have a look it. I > was > afraid I might have to do something ugly with regular expressions. I > probably > should not put off upgrading any longer. > > Thanks, Rob > > > > > > ------------------------------ > > Message: 4 > Date: Thu, 1 Nov 2007 21:35:06 +0100 > From: Marc Dirix <marc(a)electronics-design.nl> > Subject: [OpenSER-Users] Beep in audio stream. > To: users(a)lists.openser.org > Message-ID: > <871BA93E-E51A-4297-906D-789BA5798461(a)electronics-design.nl> > Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes > > Hi, > > I'm currently setting op an openser server. > My setup at the moment is as follows: > > registrar (pstn) <=> yate (sip server) <=> openser <=> sip_phone. > > As I make a call with the sip phone to a pstn line, the rtp stream is > forwarded from > the yate server to openser, which acts as en media proxy with > rtpproxy. During > the call I get very annoying beeps every 2 or 3 seconds. > > The beeps sound a bit like cost-beeps or somethin. > > When I connect the phone directly to the yate server however, which > then starts > acting as media-proxy, I do not get any beeps. > > Furthermore, if I remove force_rtp_forward() from openser config, it > stops being proxy for the stream, but still I get these annoying > beeps. Excluding any problems with rtpproxy. > > > Clearly, the registrar sends these beeps, but he doesn't send them > when I connect with yate. > Am I missing something that can trigger this behaviour? > > Thanks, > > Marc Dirix > > > > > > ------------------------------ > > Message: 5 > Date: Fri, 2 Nov 2007 09:41:33 +0700 > From: Tung Tran <tr.tung(a)gmail.com> > Subject: [OpenSER-Users] Is it possible to insert avp to reply > message? > To: <users(a)lists.openser.org> > Message-ID: <200711294133.621270@VGN-TXN15P> > Content-Type: text/plain; charset="us-ascii" > > An HTML attachment was scrubbed... > URL: > http://lists.openser.org/pipermail/users/attachments/20071102/26276838/atta… > > ------------------------------ > > Message: 6 > Date: Fri, 02 Nov 2007 05:30:45 +0200 > From: Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> > Subject: Re: [OpenSER-Users] Set up two SIP to PSTN calls and then > connect them > To: Andreas Granig <agranig(a)sipwise.com> > Cc: users(a)lists.openser.org > Message-ID: <472A99E5.9000401(a)voice-system.ro> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Hi, > > Or you can use this script, with no external dependency: > > http://openser.svn.sourceforge.net/viewvc/openser/branches/1.2/examples/web… > > regards, > Bogdan > > Andreas Granig wrote: >> It's already included, see >> http://svn.berlios.de/viewcvs/sems/trunk/apps/click2dial/ >> >> Andreas >> >> CSB wrote: >> >>> Is there any update regarding the click2dial plugin that was planned to >>> be introduced to the trunk? >>> >>> >>> >>> Regards >>> >>> >>> >>> Cameron >>> >>> >> >> _______________________________________________ >> Users mailing list >> Users(a)lists.openser.org >> http://lists.openser.org/cgi-bin/mailman/listinfo/users >> >> > > > > > ------------------------------ > > Message: 7 > Date: Fri, 2 Nov 2007 16:32:53 +1300 > From: "CSB" <kjcsb(a)xnet.co.nz> > Subject: Re: [OpenSER-Users] Set up two SIP to PSTN calls and then > connect them > To: "'Andreas Granig'" <agranig(a)sipwise.com> > Cc: users(a)lists.openser.org > Message-ID: <003401c81d01$0d9d8920$28d89b60$(a)co.nz> > Content-Type: text/plain; charset="us-ascii" > > Thanks. > > I currently use OpenSER and Asterisk and I can get the call set up using > ctd.sh. The question I have relates to the accounting. Using the ctd.sh > script is there a way to get the CDR records written from OpenSER? If I >understand correctly, OpenSER drops out of the call signalling and will not > receive any BYEs so accounting will be impossible; am I correct? Asterisk > will record the calls but billing them appropriately using those records > would be problematic (I think). > > If using the SEMS option, am I correct in thinking that it would be > possible > to use the accounting records from OpenSER? > > Regards > > Cameron > > -----Original Message----- > From: Andreas Granig [mailto:agranig@sipwise.com] > Sent: Friday, 2 November 2007 1:08 a.m. > To: CSB > Cc: users(a)lists.openser.org > Subject: Re: Set up two SIP to PSTN calls and then connect them > > It's already included, see > http://svn.berlios.de/viewcvs/sems/trunk/apps/click2dial/ > > Andreas > > CSB wrote: >> Is there any update regarding the click2dial plugin that was planned to >> be introduced to the trunk? >> >> >> >> Regards >> >> >> >> Cameron >> > > > > > ------------------------------ > > Message: 8 > Date: Fri, 2 Nov 2007 09:39:31 +0100 > From: I?aki Baz Castillo <ibc(a)in.ilimit.es> > Subject: Re: [OpenSER-Users] Is it possible to insert avp to reply > message? > To: users(a)lists.openser.org > Message-ID: <200711020939.31335.ibc(a)in.ilimit.es> > Content-Type: text/plain; charset="ISO-8859-1" > > El Friday 02 November 2007 03:41:33 Tung Tran escribi?: >> Hi all, > > Please, when creating a **new** mail press "create new mail", but don't > press "Reply" on any other mail of any other thread. If you do so your > mail > will appear contained in a wrong thread, broking it and make it very > difficult to understand. > > Thanks. > > > -- > I?aki Baz Castillo > ibc(a)in.ilimit.es > > > > ------------------------------ > > Message: 9 > Date: Fri, 2 Nov 2007 17:15:45 +0800 > From: Allan Chao ( ??? ) <AllanChao(a)taiwanmobile.com> > Subject: [OpenSER-Users] maddr in contact > To: <users(a)lists.openser.org> > Message-ID: > <970C4ACFBCFD2349B388E8907A2A7B5E80D2B4(a)TCCEXCH12.pcdc.com.tw> > Content-Type: text/plain; charset="big5" > > Hi : > > I have a flow UE -> gateway (use openser and runs proxy mode) ( ip : > 192.168.1.2) -> SIP Proxy ( ip: 192.168.1.3), > > if i have two user , UE1(192.168.1.5) and UE2(192.168.1.6) send REGISTER > request to SIP proxy through gateway, > > but our gateway add a maddr = "192.168.1.2" string in contact header,so > the contact in REGISTER becomes <username@ host ; maddr="192.168.1.2">. > > now , if UE1 send INVITE message to UE2, how does sip proxy to do if > receive INVITE message? it will send invite message to UE2 through > gateway ( maddr parameter) ? > and does openser has support maddr in contact or not . thx. > > > allan >

1 0

[Serusers] Suresh is your Yaar! :)
by Suresh 02 Nov '07

02 Nov '07

Spam detection software, running on the system "mail.iptel.org", has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: Suresh Hyderabad Suresh Arunachalam wants you to join Yaari! Is Suresh your friend? Please respond or Suresh might think you said no :( To stop receiving emails from Yaari.com, click here. If you have any concerns regarding the content of this message, please email abuse(a)yaari.com. Yaari LLC, 358 Angier Ave, Atlanta, GA 30312 [...] Content analysis details: (6.5 points, 5.0 required) pts rule name description ---- ---------------------- -------------------------------------------------- 0.4 X_PRIORITY_HIGH Sent with 'X-Priority' set to high -0.0 SPF_HELO_PASS SPF: HELO matches SPF record -0.0 SPF_PASS SPF: sender matches SPF record 3.2 URI_UNSUBSCRIBE URI: URI contains suspicious unsubscribe link 1.2 HTML_IMAGE_ONLY_20 BODY: HTML: images with 1600-2000 bytes of words -2.6 BAYES_00 BODY: Bayesian spam probability is 0 to 1% [score: 0.0000] 0.0 HTML_MESSAGE BODY: HTML included in message 0.0 MIME_HTML_ONLY BODY: Message only has text/html MIME parts 0.2 MIME_BASE64_NO_NAME RAW: base64 attachment does not have a file name 3.0 URIBL_OB_SURBL Contains an URL listed in the OB SURBL blocklist [URIs: yaari.com] 0.9 HTML_SHORT_LINK_IMG_3 HTML is very short with a linked image 0.2 AWL AWL: From: address is in the auto white-list The original message was not completely plain text, and may be unsafe to open with some email clients; in particular, it may contain a virus, or confirm that your address can receive spam. If you wish to view it, it may be safer to save it to a file and open it with an editor.

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[OpenSER-Users] Beep in audio stream.
by Marc Dirix 02 Nov '07

02 Nov '07

Hi, I'm currently setting op an openser server. My setup at the moment is as follows: registrar (pstn) <=> yate (sip server) <=> openser <=> sip_phone. As I make a call with the sip phone to a pstn line, the rtp stream is forwarded from the yate server to openser, which acts as en media proxy with rtpproxy. During the call I get very annoying beeps every 2 or 3 seconds. The beeps sound a bit like cost-beeps or somethin. When I connect the phone directly to the yate server however, which then starts acting as media-proxy, I do not get any beeps. Furthermore, if I remove force_rtp_forward() from openser config, it stops being proxy for the stream, but still I get these annoying beeps. Excluding any problems with rtpproxy. Clearly, the registrar sends these beeps, but he doesn't send them when I connect with yate. Am I missing something that can trigger this behaviour? Thanks, Marc Dirix

3 3

[Serusers] Something I've never fully understood about AVP_Check
by SIP 01 Nov '07

01 Nov '07

Both the SER and OpenSER documentation list flags in avp_check: flags = 'g' | 'G' | 'i' | 'I' But no one seems to list exactly what those flags do. Sure, /g for global, and /i for case insensitive. But what are the capital versions? The same thing just variants thereof? Do they mean something special? N.

2 1

[OpenSER-Users] How to expose the expires value in REGISTER
by Robert Dyck 01 Nov '07

01 Nov '07

I am wondering how to expose and test the value of the expires parameter in a REGISTER request. I am experimenting with openser as the basis for a home phone network. I use multiple devices with the same user ID. They register locally ( with no reply ) and with an external service provider. The contacts are mangled to show the public address of openser. Multiple registrations result in a single AOR at the external registrar. Incoming calls from the outside are forked and ring the local phones. Local phones can also call each other without the hairpin problem associated with STUN enabled phones. The problem is that a softphone will deregister when it is closed or its profile changes. This would deregister the AOR at the external registrar. The remaining phones could not receive calls from the outside until they refreshed their registrations. I would like to prevent deregistration at the external registrar unless the phone that was deregistering was the only remaining one. The first step would be to identify REGISTER messages where the expires value is equal to zero. Any thoughts? Thanks Rob

2 3

[Serusers] need help - Call Forwarding
by Piyush.Bansal＠relianceada.com 01 Nov '07

01 Nov '07

Hi All, We're trying to test Call Forwarding Unconditional (CFU) using SER-2.0.0-rc1. We're adding the subscriber's attribute using following command - ./ser_ctl attrs set 50001(a)192.168.112.14 fwd_always_target=50(a)192.168.112.14 After that we tried to call to 50001. After receiving the call, SER fails to fetch the attribute (fwd_always_target) value from the DB. Even, the check "if ($tu.fwd_always_target)" fails in this case.. The config file is also attahced with the mail. Can anybody tell me why this is happening?????? Plz reply ASAP..... The information contained in this e-mail message is intended only for the use of the individual or entity to which it is addressed. If you are not the intended recipient, you should return it to the sender immediately. Please note that while we scan all e-mails for viruses we cannot guarantee that any e-mail is virus-free and accept no liability for any damage caused by any virus transmitted by this email.

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Re: [Serusers] FW: Get RealTime Online Users status
by roayu 01 Nov '07

01 Nov '07

Hi Michal, How to check whether the SIP ALG is enabled from the trace? Thanks. Cheers, Roa Yu :) -----Original Message----- From: roayu Sent: Monday, October 22, 2007 3:13 PM To: 'Michal Matyska' Subject: RE: [Serusers] Get RealTime Online Users status Thanks Michal! I will try on it. Thanks. Cheers, Roa Yu -----Original Message----- From: Michal Matyska [mailto:michal@iptel.org] Sent: Friday, October 19, 2007 11:55 PM To: roayu Cc: serusers(a)lists.iptel.org Subject: RE: [Serusers] Get RealTime Online Users status Looks you have SIP ALG enabled on your NAT, which breaks the call transfer (no surprise, I haven't seen working SIP ALG yet). Check OPTIONS request sent by SJphone and received by SER, check NOTIFY sent by SJphone and received by SER. Disable the SIP ALG and try again. If you can't disable it, change the NAT box itself. Michal On Fri, 2007-10-19 at 14:42 +0800, roayu wrote: > Hi Michal, > > I've attached the config file as well as the network capture so that you can > help me to find out the problem. But I'm not sure whether I will gonna guide > u to the answer 42 or not. :p > > Thanks for your help! > > > Cheers, > Roa Yu > > > -----Original Message----- > From: Michal Matyska [mailto:michal@iptel.org] > Sent: Friday, October 19, 2007 1:17 PM > To: roayu > Cc: Jiri Kuthan; serusers(a)lists.iptel.org > Subject: Re: [Serusers] Get RealTime Online Users status > > Hi, > > unless you post the network capture and your config file (but make sure > you don't disclose anything you want to keep secret) there is nobody who > can help you. > > The only answer is: your script is broken, except the answer 42, but I > don't think your question was the ultimate question about life, universe > etc. ;-) > > Michal > > P.S. If you haven't read "Hitchhikers guide to the universe" don't panic > and just forget the answer 42. > > On Fri, 2007-10-19 at 11:35 +0800, roayu wrote: > > Hi Jiri! > > > > I've captured and analyzed those SIP message dumps. I found that when > > Sjphone A sent a NOTIFY signal to the SER (after Sjphone A accepted the > > REFER), SER didn't pass the NOTIFY to the SPA942 to terminate the call > > between them. > > > > Therefore, Sjphone A unable to INVITE Sjphone B to bridge up the > connection. > > May I know what causes this? > > > > > > Thanks for your precious time and reply! > > > > > > Cheers, > > Roa Yu :) > > > > > > -----Original Message----- > > From: Jiri Kuthan [mailto:jiri@iptel.org] > > Sent: Thursday, October 18, 2007 4:28 PM > > To: roayu > > Cc: serusers(a)lists.iptel.org > > Subject: Re: [Serusers] Get RealTime Online Users status > > > > At 04:59 18/10/2007, roayu wrote: > > >Hi Jiri! > > > > > >Actually, I'm using Sjphone and SPA942 to do the call transfer. I'm > calling > > >using Sjphone A to SPA942, then use the SPA942 to transfer the call to > > >Sjphone B. > > > > > > SJPhone A --> SPA942 --> SJPhone B > > > > > >After a short conversation with SJPhone B, then I press the 'xfer' button > > on > > >the 2nd time so that both SJPhone A and SJPhone B can talk to each other. > > >But the call couldn't be transferred to SJPhone B whereas when I used > > >Asterisk, it's able to do so. > > > > > >I've tried to disable the Digest-qop but it still couldn't perform the > > >transfer properly. What other settings that I need to configure on SER? > > > > I'm afraid you've gotta analyze SIP message dumps to figure out what's > gone > > wrong. -jiri > > > > > > >Thanks and really appreciate on your reply. > > > > > >Cheers, > > >Roa Yu > > > > > > > > >-----Original Message----- > > >From: Jiri Kuthan [mailto:jiri@iptel.org] > > >Sent: Thursday, October 18, 2007 9:16 AM > > >To: roayu > > >Cc: serusers(a)lists.iptel.org > > >Subject: Re: [Serusers] Get RealTime Online Users status > > > > > >At 10:37 10/10/2007, roayu wrote: > > >>Oh, I got it! Thanks Jiri! > > >> > > >>Btw, I have another questions. > > >>1) I would like to know can call transfer being done when the > > >>connection is P2P? > > > > > >you mean without a proxy? sure it can. > > > > > >> When I tried to use relay (mediaproxy) to do the call transfer, > > >it's > > >>able to transfer the call to the other party whereas when the > condition > > >is > > >>P2P, it just can't pass the correct signal. > > > > > >that's apparently unrelated to use of media proxy. you would have to > > >check the SIP dumps what's going wrong. Unfortunately I can't promise > > >I would help to review those -- my to-do-list is endless these days. > > >Perhaps someone else on the list will. > > > > > > > > >>2) Can SER support SPA942 ? Or is there some settings that I need > to > > >>configure on SER ? > > > > > >yes. > > >well for some SPAs you may need to disable digest-qop since they have a > > bug. > > >(don't ask me which though). > > > > > >-jiri > > > > > > > > > > > >>Thanks. > > >> > > >>Cheers, > > >>Roa Yu > > >> > > >>-----Original Message----- > > >>From: Jiri Kuthan [mailto:jiri@iptel.org] > > >>Sent: Wednesday, October 10, 2007 2:56 PM > > >>To: roayu; SIP > > >>Cc: serusers(a)lists.iptel.org > > >>Subject: Re: [Serusers] Get RealTime Online Users status > > >> > > >>yes, set usrloc's database mode to 1. -jiri > > >> > > >>At 03:32 10/10/2007, roayu wrote: > > >> > > >>>Thanks for your reply. I found that it's only update after around 30 > > >>>seconds. Is there anyway that I can fasten the update on the database? > > >>> > > >>>Thanks. > > >>> > > >>>Cheers, > > >>>Roa Yu > > >>> > > >>>-----Original Message----- > > >>>From: SIP [mailto:sip@arcdiv.com] > > >>>Sent: Tuesday, October 09, 2007 9:14 PM > > >>>To: roayu > > >>>Cc: serusers(a)lists.iptel.org > > >>>Subject: Re: [Serusers] Get RealTime Online Users status > > >>> > > >>>roayu wrote: > > >>>> > > >>>> Hi there! > > >>>> > > >>>> Can anyone tell me how to get the RealTime Online user status other > > >>>> than using command '*serctl ul show*'? Or how can I store the > realtime > > >>>> online user status to the MySQL db? > > >>>> > > >>>> Thanks. > > >>>> > > >>>> Cheers, > > >>>> > > >>>> Roa Yu J > > >>>> > > >>>> > > ------------------------------------------------------------------------ > > >>>> > > >>>> _______________________________________________ > > >>>> Serusers mailing list > > >>>> Serusers(a)lists.iptel.org > > >>>> http://lists.iptel.org/mailman/listinfo/serusers > > >>>> > > >>>The location table in the MySQL database stores what are, in effect, > > >>>currently online users. It is, however, only an approximation. When a > UA > > >>>registers, it sends an expiration time on its registration and that > gets > > >>>stored in the location table. If the user disconnects without > cancelling > > >>>his registration (some UAs send am expire register message to 'log out' > > > >>>and some don't), then the data may still be in the table until the > > >>>expire time occurs. Realistically, though, you can get a good idea of > > >>>currenly online users using the location table and while it may not be > > >>>100% accurate, it's close enough for government work, as it were. > > >>> > > >>>N. > > >>>_______________________________________________ > > >>>Serusers mailing list > > >>>Serusers(a)lists.iptel.org > > >>>http://lists.iptel.org/mailman/listinfo/serusers > > >> > > >> > > >> > > >>-- > > >>Jiri Kuthan http://iptel.org/~jiri/ > > >>_______________________________________________ > > >>Serusers mailing list > > >>Serusers(a)lists.iptel.org > > >>http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > > > > >-- > > >Jiri Kuthan http://iptel.org/~jiri/ > > >_______________________________________________ > > >Serusers mailing list > > >Serusers(a)lists.iptel.org > > >http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > -- > > Jiri Kuthan http://iptel.org/~jiri/ > > _______________________________________________ > > Serusers mailing list > > Serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

3 2

[Serusers] Doucumention search page is not working
by Tomasz Zieleniewski 01 Nov '07

01 Nov '07

Hi, Can some please check and set it up. Thanks Tomasz

1 0

Re: [Serusers] FW: Get RealTime Online Users status
by roayu 01 Nov '07

01 Nov '07

Hi Atle! Ya, sometimes I found that there is something unusual happened as the UA1 will send the INVITE directly + UA2 public IP to UA2 without going through SER. Is this what you are mentioning on the ALG issue? Thanks! Cheers, Roa Yu -----Original Message----- From: Atle Samuelsen [mailto:clona@cyberhouse.no] Sent: Thursday, November 01, 2007 4:04 PM To: roayu Cc: Michal Matyska; serusers(a)lists.iptel.org Subject: Re: [Serusers] FW: Get RealTime Online Users status HI Roa Yo, Look at the signalling, If you for instance see the callID with a private IP, and everything else with public IP from the UA.. Then you are fairly sure there is something happening on the way (ALG). If you know a user is behind nat, you know he should have private IP's in his signalling but still has public IP's. Also if you know your sip-device always uses port 5060, and suddenly you see traffic from one sip-device that sends from port 15321 and with a public IP.. you know something is happening on the way. Just remember,, it CAN look like a alg if ser has been in the picture since ser can do alot of funnie things with the signalling if you ask it to ;-) - Atle * roayu <roayu(a)ctisys.net> [071101 08:37]: > > Hi Michal, > > How to check whether the SIP ALG is enabled from the trace? > > > Thanks. > > Cheers, > Roa Yu :) > > > > -----Original Message----- > From: roayu > Sent: Monday, October 22, 2007 3:13 PM > To: 'Michal Matyska' > Subject: RE: [Serusers] Get RealTime Online Users status > > Thanks Michal! > > I will try on it. > > Thanks. > > Cheers, > Roa Yu > > -----Original Message----- > From: Michal Matyska [mailto:michal@iptel.org] > Sent: Friday, October 19, 2007 11:55 PM > To: roayu > Cc: serusers(a)lists.iptel.org > Subject: RE: [Serusers] Get RealTime Online Users status > > Looks you have SIP ALG enabled on your NAT, which breaks the call > transfer (no surprise, I haven't seen working SIP ALG yet). > > Check OPTIONS request sent by SJphone and received by SER, check NOTIFY > sent by SJphone and received by SER. > > Disable the SIP ALG and try again. If you can't disable it, change the > NAT box itself. > > Michal > > > On Fri, 2007-10-19 at 14:42 +0800, roayu wrote: > > Hi Michal, > > > > I've attached the config file as well as the network capture so that you > can > > help me to find out the problem. But I'm not sure whether I will gonna > guide > > u to the answer 42 or not. :p > > > > Thanks for your help! > > > > > > Cheers, > > Roa Yu > > > > > > -----Original Message----- > > From: Michal Matyska [mailto:michal@iptel.org] > > Sent: Friday, October 19, 2007 1:17 PM > > To: roayu > > Cc: Jiri Kuthan; serusers(a)lists.iptel.org > > Subject: Re: [Serusers] Get RealTime Online Users status > > > > Hi, > > > > unless you post the network capture and your config file (but make sure > > you don't disclose anything you want to keep secret) there is nobody who > > can help you. > > > > The only answer is: your script is broken, except the answer 42, but I > > don't think your question was the ultimate question about life, universe > > etc. ;-) > > > > Michal > > > > P.S. If you haven't read "Hitchhikers guide to the universe" don't panic > > and just forget the answer 42. > > > > On Fri, 2007-10-19 at 11:35 +0800, roayu wrote: > > > Hi Jiri! > > > > > > I've captured and analyzed those SIP message dumps. I found that when > > > Sjphone A sent a NOTIFY signal to the SER (after Sjphone A accepted the > > > REFER), SER didn't pass the NOTIFY to the SPA942 to terminate the call > > > between them. > > > > > > Therefore, Sjphone A unable to INVITE Sjphone B to bridge up the > > connection. > > > May I know what causes this? > > > > > > > > > Thanks for your precious time and reply! > > > > > > > > > Cheers, > > > Roa Yu :) > > > > > > > > > -----Original Message----- > > > From: Jiri Kuthan [mailto:jiri@iptel.org] > > > Sent: Thursday, October 18, 2007 4:28 PM > > > To: roayu > > > Cc: serusers(a)lists.iptel.org > > > Subject: Re: [Serusers] Get RealTime Online Users status > > > > > > At 04:59 18/10/2007, roayu wrote: > > > >Hi Jiri! > > > > > > > >Actually, I'm using Sjphone and SPA942 to do the call transfer. I'm > > calling > > > >using Sjphone A to SPA942, then use the SPA942 to transfer the call to > > > >Sjphone B. > > > > > > > > SJPhone A --> SPA942 --> SJPhone B > > > > > > > >After a short conversation with SJPhone B, then I press the 'xfer' > button > > > on > > > >the 2nd time so that both SJPhone A and SJPhone B can talk to each > other. > > > >But the call couldn't be transferred to SJPhone B whereas when I used > > > >Asterisk, it's able to do so. > > > > > > > >I've tried to disable the Digest-qop but it still couldn't perform the > > > >transfer properly. What other settings that I need to configure on SER? > > > > > > I'm afraid you've gotta analyze SIP message dumps to figure out what's > > gone > > > wrong. -jiri > > > > > > > > > >Thanks and really appreciate on your reply. > > > > > > > >Cheers, > > > >Roa Yu > > > > > > > > > > > >-----Original Message----- > > > >From: Jiri Kuthan [mailto:jiri@iptel.org] > > > >Sent: Thursday, October 18, 2007 9:16 AM > > > >To: roayu > > > >Cc: serusers(a)lists.iptel.org > > > >Subject: Re: [Serusers] Get RealTime Online Users status > > > > > > > >At 10:37 10/10/2007, roayu wrote: > > > >>Oh, I got it! Thanks Jiri! > > > >> > > > >>Btw, I have another questions. > > > >>1) I would like to know can call transfer being done when the > > > >>connection is P2P? > > > > > > > >you mean without a proxy? sure it can. > > > > > > > >> When I tried to use relay (mediaproxy) to do the call > transfer, > > > >it's > > > >>able to transfer the call to the other party whereas when the > > condition > > > >is > > > >>P2P, it just can't pass the correct signal. > > > > > > > >that's apparently unrelated to use of media proxy. you would have to > > > >check the SIP dumps what's going wrong. Unfortunately I can't promise > > > >I would help to review those -- my to-do-list is endless these days. > > > >Perhaps someone else on the list will. > > > > > > > > > > > >>2) Can SER support SPA942 ? Or is there some settings that I need > > to > > > >>configure on SER ? > > > > > > > >yes. > > > >well for some SPAs you may need to disable digest-qop since they have a > > > bug. > > > >(don't ask me which though). > > > > > > > >-jiri > > > > > > > > > > > > > > > >>Thanks. > > > >> > > > >>Cheers, > > > >>Roa Yu > > > >> > > > >>-----Original Message----- > > > >>From: Jiri Kuthan [mailto:jiri@iptel.org] > > > >>Sent: Wednesday, October 10, 2007 2:56 PM > > > >>To: roayu; SIP > > > >>Cc: serusers(a)lists.iptel.org > > > >>Subject: Re: [Serusers] Get RealTime Online Users status > > > >> > > > >>yes, set usrloc's database mode to 1. -jiri > > > >> > > > >>At 03:32 10/10/2007, roayu wrote: > > > >> > > > >>>Thanks for your reply. I found that it's only update after around 30 > > > >>>seconds. Is there anyway that I can fasten the update on the > database? > > > >>> > > > >>>Thanks. > > > >>> > > > >>>Cheers, > > > >>>Roa Yu > > > >>> > > > >>>-----Original Message----- > > > >>>From: SIP [mailto:sip@arcdiv.com] > > > >>>Sent: Tuesday, October 09, 2007 9:14 PM > > > >>>To: roayu > > > >>>Cc: serusers(a)lists.iptel.org > > > >>>Subject: Re: [Serusers] Get RealTime Online Users status > > > >>> > > > >>>roayu wrote: > > > >>>> > > > >>>> Hi there! > > > >>>> > > > >>>> Can anyone tell me how to get the RealTime Online user status other > > > > >>>> than using command '*serctl ul show*'? Or how can I store the > > realtime > > > >>>> online user status to the MySQL db? > > > >>>> > > > >>>> Thanks. > > > >>>> > > > >>>> Cheers, > > > >>>> > > > >>>> Roa Yu J > > > >>>> > > > >>>> > > > ------------------------------------------------------------------------ > > > >>>> > > > >>>> _______________________________________________ > > > >>>> Serusers mailing list > > > >>>> Serusers(a)lists.iptel.org > > > >>>> http://lists.iptel.org/mailman/listinfo/serusers > > > >>>> > > > >>>The location table in the MySQL database stores what are, in effect, > > > >>>currently online users. It is, however, only an approximation. When a > > UA > > > >>>registers, it sends an expiration time on its registration and that > > gets > > > >>>stored in the location table. If the user disconnects without > > cancelling > > > >>>his registration (some UAs send am expire register message to 'log > out' > > > > > >>>and some don't), then the data may still be in the table until the > > > >>>expire time occurs. Realistically, though, you can get a good idea of > > > > >>>currenly online users using the location table and while it may not > be > > > >>>100% accurate, it's close enough for government work, as it were. > > > >>> > > > >>>N. > > > >>>_______________________________________________ > > > >>>Serusers mailing list > > > >>>Serusers(a)lists.iptel.org > > > >>>http://lists.iptel.org/mailman/listinfo/serusers > > > >> > > > >> > > > >> > > > >>-- > > > >>Jiri Kuthan http://iptel.org/~jiri/ > > > >>_______________________________________________ > > > >>Serusers mailing list > > > >>Serusers(a)lists.iptel.org > > > >>http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > > > > > > > > >-- > > > >Jiri Kuthan http://iptel.org/~jiri/ > > > >_______________________________________________ > > > >Serusers mailing list > > > >Serusers(a)lists.iptel.org > > > >http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > > > > > -- > > > Jiri Kuthan http://iptel.org/~jiri/ > > > _______________________________________________ > > > Serusers mailing list > > > Serusers(a)lists.iptel.org > > > http://lists.iptel.org/mailman/listinfo/serusers > > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] ser 0.9.6 memory problems
by Mitya 31 Oct '07

31 Oct '07

Hi All, I have a SER 0.9.6, and sometimes I have memory allocation problem somehow :( Oct 29 19:58:38 sip ser[23161]: processing REGISTER received from x.x.x.x Oct 29 19:58:38 sip ser[23161]: DEBUG: t_newtran: msg id=23950 , global msg id=23949 , T on entrance=0xffffffff Oct 29 19:58:38 sip ser[23161]: parse_headers: flags=-1 Oct 29 19:58:38 sip ser[23161]: DEBUG: get_hdr_body : content_length=0 Oct 29 19:58:38 sip ser[23161]: found end of header Oct 29 19:58:38 sip ser[23161]: parse_headers: flags=60 Oct 29 19:58:38 sip ser[23161]: t_lookup_request: start searching: hash=31874, isACK=0 Oct 29 19:58:38 sip ser[23161]: DEBUG: RFC3261 transaction matching failed Oct 29 19:58:38 sip ser[23161]: DEBUG: t_lookup_request: no transaction found Oct 29 19:58:38 sip ser[23161]: ERROR: sip_msg_cloner: cannot allocate memory Oct 29 19:58:38 sip ser[23161]: DEBUG:destroy_avp_list: destroying list (nil) Oct 29 19:58:38 sip ser[23161]: ERROR: new_t: out of mem: Oct 29 19:58:38 sip ser[23161]: ERROR: t_newtran: new_t failed As I read in the mailing lists, it occurs, when SER has a lots of live, opened transactions. It's a test SER, I think we did not have any opened transactions (maybe the failed ones). Is there any mechanism in SER, what drops the too old opened (most likely failed transactions) transactions ? Can I get the current used shared memory somehow ? What else can sause this problem ? Thanks, Mitya

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[OpenSER-Users] von boston - openser sessions in rooms 205C and 206A
by Daniel-Constantin Mierla 31 Oct '07

31 Oct '07

Hello, because are a lot of signs and banners, some attendees may get it faster via email: the openser sessions are in 205A for openser admin training session and 206A for BoF. The are on the right side of the building, the second floor (from the registration desk, turn right and walk to the end of corridor, where you can get the elevators). Cheers, Daniel

1 0

[OpenSER-Users] port modification is sdp
by Konstantinos Koutsopoulos 31 Oct '07

31 Oct '07

Hi, I am deploying openser 1.2.2-notls on a machine with public IP. On the same machine I run a sip client (A) that registers with the above openser. On an another machine behind NAT (typical adsl router) I use a sip client (B) to register with the above openser and try to invite sip client A. I do not use any NAT-traversal techniques (the outgoing packets from B are modified by NAT and reach openser, responses are received by B through NAT due to statefull (?) routing on the NAT). The SIP messages reach their destination properly and moreover the IP addresses in the SDP (from B) are updated by openser to indicate the external interface of NAT. This is very useful in my case since the client A sends the media packets to the proper address (I also configure virtual servers on the adsl router to have these forwarded to B). However, the port numbers in the SDP are also updated (usually by an offset of +16) which hinders the media session from A to B since B is expecting the packets on other ports. Is there any way to have the ports remain unaltered during the above operation? Thanks in advance, Kostas Below is my openser.cfg # # $Id: openser.cfg 1676 2007-02-21 13:16:34Z bogdan_iancu $ # # simple quick-start config script # Please refer to the Core CookBook at http://www.openser.org/dokuwiki/doku.php # for a explanation of possible statements, functions and parameters. # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) children=4 # Uncomment these lines to enter debugging mode #fork=no #log_stderror=yes # port=5060 # uncomment the following lines for TLS support #disable_tls = 0 #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/opt/openser/etc/openser/tls/user/user-cert.pem" #tls_private_key = "/opt/openser/etc/openser/tls/user/user-privkey.pem" #tls_ca_list = "/opt/openser/etc/openser/tls/user/user-calist.pem" # ------------------ module loading ---------------------------------- #set module path mpath="/lib/openser/modules/" # Uncomment this if you want to use SQL database loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "mi_fifo.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "auth.so" loadmodule "auth_db.so" # ----------------- setting module-specific parameters --------------- # -- mi_fifo params -- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS #if(uri=~"@tls_domain1.net") { # t_relay("tls:domain1.net"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay("tls:domain2.net"); # exit; #} route(1); }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("......", "subscriber")) { www_challenge(".....", "0"); exit; }; save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; }

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[Serusers] lcr module error - no prefix column and no avp visible
by Tomasz Zieleniewski 31 Oct '07

31 Oct '07

Hi, During the lcr module set up I encountered the following problems: a) there is no prefix column in the gw table so when the modules is initialized it returns an error b) the ip_addr column in gw table is an int(1o) type how should I specify it to get the IP like 192.168.0.74? c) when I invoke the load_gw() there is the debug info that gw_uri_avp was created but when I then try to print it I get that avp not found, what is wrong? log below: Oct 31 11:38:24 sen ser[12273]: load_gws(): DEBUG: Added gw_uri_avp < sip:touk1@138.64.116.11:5070> Oct 31 11:38:24 sen ser[12273]: AVP '$gw_uri_avp' not found Waiting for Your feedback Regards Tomasz

1 0

[OpenSER-Users] Generating an INVITE with multiple Contacts
by Juan Carlos C. y Castro 31 Oct '07

31 Oct '07

(Please CC me as I'm not on the list) Hello, I'm having an OpenSER installation being submitted to a compliance test, and one of the many tests requires that OpenSER replies to a correctly authenticated REGISTER with many Contact headers with 200 OK messages for each of the contacts, in addition to storing all the contacts in the location database. Questions: 1) Do I need some special configuration for that? Shouldn't the radius_www_authorize() function (I'm authenticating against Radius) take care of it all? 2) What software could I use to generate a REGISTER request with multiple Contacts? Thanks, Juan

3 2

[Serusers] ERROR: in ser_ctl :: Release-2.0.0-rc1
by Piyush.Bansal＠relianceada.com 31 Oct '07

31 Oct '07

hi All, We're using SER Release-2.0.0.rc1. While trying to add a domain using "./ser_ctl domain add adc.com" command we're facing the following error - ser_ctl: digest_realm: Unknown attribute My doubt is, if I look at the database, there is no field named digest_realm in the domain table.... Can anybody explain, why this is happening.... thanx 'n' regds, --Piyush The information contained in this e-mail message is intended only for the use of the individual or entity to which it is addressed. If you are not the intended recipient, you should return it to the sender immediately. Please note that while we scan all e-mails for viruses we cannot guarantee that any e-mail is virus-free and accept no liability for any damage caused by any virus transmitted by this email.

1 0

[OpenSER-Users] OpenXcap exception: TypeError in twisted.web2.stream.IByteStream
by Gregorio 30 Oct '07

30 Oct '07

Hi all. I get an exception in openXcap when I send a GET to retrieve the resources file. I explain the process I followed: ----->Send GET to openXcap. <----Receive 404, because there is not file. -----> Send PUT to openXcap. The body contains a resource-list.xml document 'with no data': Body; <?xml version='1.0' encoding='UTF-8'?> <resource-lists xmlns='urn:ietf:params:xml:ns:resource-lists' xmlns:cp='urn:ietf:params:xml:ns:copycontrol'> <list></list> </resource-lists> <----Receive 200 OK. So openXcap likes my document. -----> Send GET to openXcap. <---- Receive 500 "An error occurred rendering the requested page. More information is available in the server log". So the same document that was accepted in the PUT throws an exception. The server log says: Oct 30 15:01:55 localhost openxcap[2791]: [-] Exception rendering: Oct 30 15:01:55 localhost openxcap[2791]: [-] Traceback (most recent call last): Oct 30 15:01:55 localhost openxcap[2791]: [-] File "/usr/lib/python2.4/site-packages/twisted/internet/posixbase.py", line 228, in mainLoop Oct 30 15:01:55 localhost openxcap[2791]: [-] self.runUntilCurrent() Oct 30 15:01:55 localhost openxcap[2791]: [-] File "/usr/lib/python2.4/site-packages/twisted/internet/base.py", line 533, in runUntilCurrent Oct 30 15:01:55 localhost openxcap[2791]: [-] f(*a, **kw) Oct 30 15:01:55 localhost openxcap[2791]: [-] File "/usr/lib/python2.4/site-packages/twisted/internet/defer.py", line 239, in callback Oct 30 15:01:55 localhost openxcap[2791]: [-] self._startRunCallbacks(result) Oct 30 15:01:55 localhost openxcap[2791]: [-] File "/usr/lib/python2.4/site-packages/twisted/internet/defer.py", line 304, in _startRunCallbacks Oct 30 15:01:55 localhost openxcap[2791]: [-] self._runCallbacks() Oct 30 15:01:55 localhost openxcap[2791]: [-] --- <exception caught here> --- Oct 30 15:01:55 localhost openxcap[2791]: [-] File "/usr/lib/python2.4/site-packages/twisted/internet/defer.py", line 317, in _runCallbacks Oct 30 15:01:55 localhost openxcap[2791]: [-] self.result = callback(self.result, *args, **kw) Oct 30 15:01:55 localhost openxcap[2791]: [-] File "/usr/lib/python2.4/site-packages/xcap/resource.py", line 51, in sendResponse Oct 30 15:01:55 localhost openxcap[2791]: [-] response = http.Response(response.code, stream=response.data) Oct 30 15:01:55 localhost openxcap[2791]: [-] File "/usr/lib/python2.4/site-packages/twisted/web2/http.py", line 109, in __init__ Oct 30 15:01:55 localhost openxcap[2791]: [-] self.stream = IByteStream(stream) Oct 30 15:01:55 localhost openxcap[2791]: [-] exceptions.TypeError: ('Could not adapt', array('c', "<?xml version='1.0' encoding='UTF-8'?><resource-lists xmlns='urn:ietf:params:xml:ns:resource-lists' xmlns:cp='urn:ietf:params:xml:ns:copycontrol'><list></list></resource-lists> "), <InterfaceClass twisted.web2.stream.IByteStream>) Anyone know what is wrong? Thanks and best regards. Gregorio

1 0

[OpenSER-Users] OpenXCAP server, twisted or Apache ?
by KevinKinnan 30 Oct '07

30 Oct '07

OpenXCAP has already included twisted, do I need to install Apache? I don't much about twisted. It seems I need to create a directory for xcap root, because the web browser gives me "XCAP root not found for uri: http://127.0.0.1:8000". And, how do I do that? Thank you Kevin, _________________________________________________________________ MSN 中文网，最新时尚生活资讯，白领聚集门户。 http://cn.msn.com

2 1

[Serusers] audio and auth problems?
by Max Lock 30 Oct '07

30 Oct '07

Hi Folks, I have 2 questions, I'm using openser 1.1.x and rtpproxy bridging between 2 networks, one public, one private. a pstn gateway in the private network, and clients in the public network. firstly, when I make a call check_from() fails, unless I comment it out, why is this? # if(!check_from()) # { # sl_send_reply("403", "Spoofed From-URI Detected"); # exit; # } and secondly. I have rtpproxy and openser listening on both public and private interfaces, with mhomed=1 set. whenever I make a call, everything works ok, but I have no audio? where should I look? -Cheers Max

1 0

[OpenSER-Users] How to compare $si with $dr
by Richard Bennett 30 Oct '07

30 Oct '07

Hi, I am trying to compare the values of 'IP source address' ($si) and 'Domain in SIP Request's URI' ($rd) to see if they match in Openser 1.1. Probably I have missed the relevant documentation, I checked http://www.openser.org/dokuwiki/doku.php/pseudovariables:1.1.x and http://www.openser.org/dokuwiki/doku.php/core-cookbook:1.1.x . Basically I'd like to do this: if($si == $rd){ #do something } but that gives a syntax error. I also tried this: if(src_ip==$rd){ } but it seems the pseudo variable isn't being expanded before the evaluation, as I don't get a match even though debugging shows $si and $rd to be the same. Also, is there anyway to pass a pseudo variable to rewritehost() or an equivalent function? Thanks for any pointers, Richard

2 1

Re: [OpenSER-Users] ALG error workarounds
by Phil D'Amore 30 Oct '07

30 Oct '07

Richard: Sorry if this breaks threading, I don't have the original message. In the cases where it doesn't work, can you confirm that the Contact URI in the 200 actually contains a public address as it leaves your network? What you're describing makes me think it's not, and it looks like the following to me: GW sends 200 with private address in Contact. This private address leaks out of your network. This private address happens to fall within the range of your customer's private network. The 200 hits the remote router. The ALG leaves it alone (for now). The PAP2T reads the Contact in the 200 and pulls the private address from it. It targets the ACK to this private address, and sends it. The ALG sees this, and notices the RURI contains a private address from it's own local network. It PATs this address (hence the port 2021, it just picks the next port, since 2020 was already used by your PAP2T). The ACK now contains the offending address in the RURI. When your openser instance gets it, it just relays it like it was told to do based on loose routing, and the call drops. The thing about those ALGs is that they will rewrite *anything* that matches the access list associated with the nat pool address, even if it has nothing to do with any real IP traffic flowing through the thing. Hope that gives you something useful. Phil

4 5

[OpenSER-Users] Get more data in AVPs after "lookup()"
by Iñaki Baz Castillo 30 Oct '07

30 Oct '07

Hi, I'd like to extend the "location" table with more attributes and get them as AVP's after "lookup()" function. Is it any automatic way to do that? I mean something as "auth_db" module's "load_credentials" that load AVP's with more subscriber data. I don't find it so I think I'll need to do a custom SQL query after "lookup()" (in branch_route) and get the custom fields. Anyway I ask it: is there any way I don't know to get this "easily"? Thanks. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

2 6

[OpenSER-Users] use_domain param in uri_db module
by Morten Isaksen 30 Oct '07

30 Oct '07

Hi! Runing OpenSER 1.1.1. In the uri_db module I have modparam("uri_db", "use_uri_table", 1) modparam("uri_db", "use_domain", 0) But the use_domain setting seems to be ignored. When I check the mysql.log i see select username from uri where username='XX102208' AND domain='XX.61.139.21' AND uri_user='XX102208' and check_from and check_to fails if the domain part does not match. Is this a bug or am I missing something? -- Morten Isaksen http://www.misak.dk/blog/

3 4

[OpenSER-Users] Accessing From header parameters
by Andreas Granig 30 Oct '07

30 Oct '07

Hi, I've found that Linksys SPA9000 (some sort of IP PBX) signals extensions in a "ref" parameter of From-Header, like: From: <sip:line1@matrix.sipura.com>;tag=1234;ref=105 Is there a way to easily access this parameter? Seems like... $var(ext) = $(fu{uri.param,ref}); ... doesn't catch it because it's not part of $fu. Any other ways except trying to assign $hdr(From) to an AVP and extract it using avp_subst(...)? Andreas

2 2

[Serusers] lcr module question - how to implement nice routing logic?
by Tomasz Zieleniewski 30 Oct '07

30 Oct '07

Hi, Is it possible to use lcr module so that when the routing logic checks it won't change the request URI but only change the destination of the SIP message?? I would like to utilize this module to select if the outgoing message should be forwarded to boundary SIP Proxy or to SIP/PSTN Gateway. Could You Please give me some clues or point another way to do this? Thank You for any help. Cheers TOmasz

1 0

[OpenSER-Users] Cluster Setup / openser location table entries "gone"
by Brandon Armstead 30 Oct '07

30 Oct '07

Hello Guys, I'm clustering subscribers at the database level, however it seems this one specific server I added into cluster that uses usrloc, is removing location entries, completely randomly? I've turned debug=9, but all I can see is... a bunch of: "core:db_free_rows: freeing 0 rows" over and over, this is with revision 2602 and latest trunk from openser. Which is thus on a usrloc mode of (3) causing essentially inbound calls to stop working. Any ideas anyone? Anyone with similiar issue? Every other server is running just fine, just only when this SINGLE server connects / runs to the rest of the cluster (MySQL) -- it starts dropping openser's locations. Any help appreciated, thanks! P.S. Also tried with libmysqlclient libs from 5.2.0-beta and 5.0 stable. -- Brandon Armstead

2 2

Re: [OpenSER-Users] [OpenSER-Devel] PUA/PUA_MI Error:
by Schumann Sebastian 30 Oct '07

30 Oct '07

Hello Anca With the current release 2993 this works. In general I use no etag param. I know what you mean by that it changes (for re-updates initiated by PUA, last numbers change e.g. from 1.0 to 2.1). I have now a different error though, even worse: First sending publish with infinite Expires: -1 works, also Expires: 0 afterwards deletes the entry in the presentity table (works as well). But if I wanna do another publish after this scenario, the whole OpenSER PUA gets stuck. It doesn't accept anymore MI publication. Also DEBUG log stops. The last lines I see in the debug are DBG:pua_mi:mi_pua_publish: send publish DPG: pua:send_publish: pres_uri=test(a)10.96.115.125 And then it freezes, no more debug information and MI request acceptance (also no reply in MI output FIFO) SIP messagesd for first publishs with Expires: -1 and 0 are exchanged (Publish/200 OK and Publish/200 OK) but for sending a Expires: -1 request again, no SIP message is created as well. What can be the error here? Stange it works once but afterwards not. Can it be that Expires: 0 deletes the presentity table entry but pua table entry is still there? I tried as well to delete that one manually before performing second publish with Expires: -1 but it doesn't work either. Regards Sebastian -----Original Message----- From: Anca Vamanu [mailto:anca@voice-system.ro] Sent: Monday, October 29, 2007 4:04 PM To: Schumann Sebastian Cc: users(a)openser.org; devel(a)openser.org Subject: Re: [OpenSER-Devel] PUA/PUA_MI Error: Hello, If you set the etag parameter when sending the "Expires: 0 " command, it might be that it is no longer valid. This happens if you first give a command with Expires value grater than the maximum accepted value by presence module. Then, the pua will generate itself updates such that the cumulated period of validity to equal the Expires in the first request. And for each new publish a different etag is replied.This is especially true if you have previously used an Expires: -1. On the other hand, if you do not set the etag param, there was a bug in the hash table searching function that was fixed in version 2993, that applied for the commands with no etag. Have you tested with this version? regards, Anca Vamanu Schumann Sebastian wrote: > Dear all > > After long evaluation I still have the following error and no solution > yet to solve it. > > I am using OpenSER current trunk version, but also with older versions > (e.g. 2597, 2619) I am having the problem. > > With version 2404 it seems to work, although I cannot use that version > and have to upgrade OpenSER to a newer release due to table consistencies. > > When I use PUA_MI to create a Publish, everything works fine. > "Expires: -1" works for infinite state. But when I want to expire this > state ("Expires: 0"), the PUA replies: > search_htable: record not found and later > send_publish: request for a publish with expires 0 and no record found > > There is most likely a problem in pua/hash.c or pua/send_publish.c but > I cannot find it. > > Can anyone help? > > Regards > Sebastian > ------------------------------------------------------------------------ > > _______________________________________________ > Devel mailing list > Devel(a)lists.openser.org > http://lists.openser.org/cgi-bin/mailman/listinfo/devel >

2 1

[Serusers] SERCTL Utility for SER Release 2.0.0-rc1
by Piyush.Bansal＠relianceada.com 30 Oct '07

30 Oct '07

hi All, We've downloaded the SERCTL Utility from http://www.iptel.org/ser_2_0_0_release_candidate_1. It seems that the Makefile of this utility is written for Linux. Can we have the same which runs on Unix platform (Solaris 10 OS). Can anybody tell me that whether we can use the serctl utility which is there in default installation (/PREFIX_PATH/sbin/serctl). The default script seems to be the same as that of SER-0.9.6 i.e. it expects the subscriber and uri_db table which is not the part of Release-2.0.0-rc1. plz reply ASAP.... thanx 'n' regds, --Piyush Bansal The information contained in this e-mail message is intended only for the use of the individual or entity to which it is addressed. If you are not the intended recipient, you should return it to the sender immediately. Please note that while we scan all e-mails for viruses we cannot guarantee that any e-mail is virus-free and accept no liability for any damage caused by any virus transmitted by this email.

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Re: [OpenSER-Users] Acc module question
by [ACTIVENDY] NOC 30 Oct '07

30 Oct '07

[ACTIVENDY] NOC wrote: > Hi Bodgan > > I got INVITEs in my db, but OpenSER insert the INVITE message into > my db when get a 200 code response to this INVITE. I would like to > insert this INVITE before it, when the phone is still ringing, cause i > need to know the setup time and when there is a call attempt in my > system. Is this possible? > > Thanks, > > Sergi > > Bogdan-Andrei Iancu wrote: >> Hi Sergi, >> >> The START acc event (corresponding to INVITE) has no dependency to >> the ACK. Do you get any INVITE acc in your DB? >> >> regards, >> Bogdan >> >> [ACTIVENDY] NOC wrote: >>> Hi, >>> >>> I need to know when a channel of my media gateway is busy >>> (included when the call is still ringing, etc). I have my Openser >>> working with acc module storing in a mysql database. I don't use >>> media proxy server. >>> I can see the media gateway active calls in the acc table >>> calculating the INVITEs whithout BYEs call registers. But I see >>> Openser doesn't insert the INVITE register still and ACK arrives. It >>> means I just can view in my database the number of active calls but >>> i cannot know how many call attempts there's in the media gateway. >>> How can I do to Openser insert the INVITE register to my database >>> when an INVITE messages arrives and don't wait for the ACK message? >>> >>> Thanks in advance, >>> >>> Sergi >>> >>> _______________________________________________ >>> Users mailing list >>> Users(a)openser.org >>> http://openser.org/cgi-bin/mailman/listinfo/users >>> >> >> >

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