- sr-users - kamailio.org

[Serusers] INFO: Overview page of nominations for technical board
by Greger V. Teigre 25 May '07

25 May '07

As requested, here is a page for tracking nominations and status for the technical board: http://www.iptel.org/nominations_and_voting_technical_board_2007

1 1

[Serusers] How set up proxy to process messages
by Nguyen Tuan 25 May '07

25 May '07

Hi everyone, I have problem with set up proxy to process message INVITE. I have 2 SIP users of different domain. How can I set up proxy for each domain to process message INVITE from one domain to another? Now when user 1 of domain A send INVITE message to user2 of domain B then message go to direct to server of domain B, do main route{} block of B and then send to user2. I need the INVITE message of user 1 the first go to proxy of domain A, process that message (do main route{} block) and then send that message to proxy of domain B, process it (do main route{})and then send to user 2. Thanks, Tuan.

2 1

[Serusers] ANNOUNCEMENT: New serdoc mailing list for documentation work
by Greger V. Teigre 25 May '07

25 May '07

Dear all, We have just created a new mailing list called serdoc(a)iptel.org. The list will be used for coordination, questions, doc patches, and discussions/input on any iptel.org documentation. The main ongoing efforts right now are: - Migration guide to 2.0 - Reference guide - SER - Getting Started and the configuration files - Improve module documentation Other documentation work also for other projects like sems, serweb, rtpproxy etc will be covered on this list. Developer or user, please consider subscribing to the list and contribute to an improved iptel.org documentation! http://lists.iptel.org/mailman/listinfo/serdoc Thanks, g-)

1 0

Re: [Serusers] behind 2 routers
by karl lu 24 May '07

24 May '07

Hi Ram, Thank you for your reply. Do you mean read the "NAT Traversal with Media proxy" section in get started document or search for "NAT Traversal with Media proxy" in our emial archive? Thanks, Karl ram <talk2ram(a)gmail.com> wrote: Hi look NAT Traversal with Media proxy ram On 5/22/07, root <taishan02(a)yahoo.com> wrote: Hi, I have a DSL router, which has only one port. To connect multiple computers I have another 4 ports router behind this DSL router. So SER is behind two routers. How can I configure these two routers so that I can use SER in this situation? Thanks, _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

Re: [Serusers] Frustrating error - please help
by Mahatma Sarasvati 24 May '07

24 May '07

Samuel, That was very helpful. The last line of the output made me wonder what was happening with the "ESTABLISHED" link. It was CounterPath Eyebeam UA that needed to be restarted. Somehow it was hung and was the source of the problem, not SER. Thanks for the help, M [root@vx1 ser]# lsof -i:5060 COMMAND PID USER FD TYPE DEVICE SIZE NODE NAME . . ser 4739 root 10u IPv4 35606 TCP localhost.localdomain:sip (LISTEN) ser 4739 root 11u IPv4 35607 TCP vx1.v11.com:sip (LISTEN) . . ser 4741 root 10u IPv4 35606 TCP localhost.localdomain:sip (LISTEN) ser 4741 root 11u IPv4 35607 TCP vx1.v11.com:sip (LISTEN) ser 4741 root 45u IPv4 35925 TCP vx1.v11.com:sip->10.10.30.121:ispipes (ESTABLISHED) [root@vx1 ser]# ----- Original Message ---- From: samuel <samu60(a)gmail.com> To: Mahatma Sarasvati <gnoahhb(a)yahoo.com> Cc: Andrey Kuprianov <andrey.kouprianov(a)gmail.com>; "serusers(a)iptel.org" <serusers(a)iptel.org> Sent: Thursday, May 24, 2007 9:27:16 AM Subject: Re: [Serusers] Frustrating error - please help Maybe you've got another software bind to the address SER is trying to bind to(softphone,asterisk,...). Can you check whether the port 5060 is in use? what's the output of the next comand? #lsof -i:5060 samuel. 2007/5/24, Mahatma Sarasvati <gnoahhb(a)yahoo.com>: Also a good idea. Still no effect. This installation was working great for a while. Is it possible that some other software I've installed is causing the problem? This is 2.0.0 rc2 from CVS. M ----- Original Message ---- From: Andrey Kuprianov <andrey.kouprianov(a)gmail.com> To: "serusers(a)iptel.org" < serusers(a)iptel.org> Sent: Thursday, May 24, 2007 8:23:14 AM Subject: Re: [Serusers] Frustrating error - please help Maybe just try setting fork=yes ?... On 5/24/07, Mahatma Sarasvati < gnoahhb(a)yahoo.com> wrote: > > A good hint, but no. I'm using the standard version 1.4 ser.cfg. I've only > added my 10.10.10.41 ip address. The "fork=no" command is commented out. > > Is there something else that would cause ser to start in non-forking mode? > > Thanks, > M > > > ----- Original Message ---- > From: samuel < samu60(a)gmail.com> > To: Mahatma Sarasvati <gnoahhb(a)yahoo.com> > Cc: serusers(a)iptel.org > Sent: Thursday, May 24, 2007 12:40:41 AM > Subject: Re: [Serusers] Frustrating error - please help > > Are you starting SER in non-forking mode? It looks like SER is not binding > to the 10.10.10.41 IP, only to the local IP ( 127.0.0.1) and that's why you > see in the logs that it's not able to send to a "standard" IP address: > WARNING: xl_mod_init: more IP 10.10.10.41 not used > CRITICAL: invalid sendtoparameters one possible reason is the server is > bound to localhost and attempts to send to the net > > Case you need to use 2 interfaces, I would recommend using mhomed=yes (don't > recall the exact syntax right now..). > > Hope it helps, > Samuel. > > 2007/5/24, Mahatma Sarasvati <gnoahhb(a)yahoo.com>: > > Hi All, > > > > This ser installation was working fine and I can't figure out why its not > working now. > > I'm getting an error (see last log line) every time I attempt a call to > local users. Calling out to PSTN works great. ngrep reveals that SER is not > sending any messages to the callee location. > > > > There are 2 registered local users, one registered twice. > > > > I have no idea what is going on. I'm using an unmodified version 1.4 > ser.cfg file on CentOS 5. > > > > I suspect some server configuration error? ? ? ? > > > > > > Thanks, > > M > > > > > > > > Here are the logs. > > > > > > May 23 16:03:38 vx1 ser: init_tcp: using epoll_lt as the io watch method > (auto detected) > > May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: Maxfwd module- initializing > > May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: WARNING: xl_mod_init: more > IP 10.10.10.41 not used > > May 23 16:03:38 vx1 last message repeated 8 times > > May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: AVPops - initializing > > May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: DEB: IP address of RTPPROXY > is /var/run/rtpproxy.sock > > May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: INFO: xlog null is "<null>" > > May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: INFO: udp_init: SO_RCVBUF > is initially 109568 > > May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: INFO: udp_init: SO_RCVBUF > is finally 262142 > > May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: INFO: udp_init: SO_RCVBUF > is initially 109568 > > May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: INFO: udp_init: SO_RCVBUF > is finally 262142 > > . > > . > > . > > . > > May 23 16:09:15 vx1 /usr/local/sbin/ser[1781]: rtpp_test: RTP proxy > <unix:/var/run/rtpproxy.sock> found, support for it re-enabled > > May 23 16:09:15 vx1 /usr/local/sbin/ser[1781]: ERROR: udp_send: > sendto(sock,0xb62048f4,1773,0, 10.10.100.103:5060,16): Invalid argument(22) > > > > May 23 16:09:15 vx1 /usr/local/sbin/ser[1781]: CRITICAL: invalid > sendtoparameters one possible reason is the server is bound to localhost > > and attempts to send to the net > > May 23 16:09:15 vx1 /usr/local/sbin/ser[1781]: msg_send: ERROR: udp_send > failed > > May 23 16:09:15 vx1 /usr/local/sbin/ser[1781]: ERROR: t_send_branch: > sending request on branch 0 failed > > May 23 16:09:15 vx1 /usr/local/sbin/ser[1781]: ERROR: sl_reply_error used: > Unfortunately error on sending to next hop occurred (477/SL) > > > > > > > > When I start SER I get this response: > > > > [root@vx1 ~]# service ser restart > > Stopping ser: > [ OK ] > > Starting ser: Listening on > > udp: 127.0.0.1 [127.0.0.1]:5060 > > udp: 10.10.10.41 [ 10.10.10.41]:5060 > > tcp: 127.0.0.1 [127.0.0.1]:5060 > > tcp: 10.10.10.41 [ 10.10.10.41]:5060 > > Aliases: > > tcp: vx1:5060 > > tcp: vx1.velocity11.com:5060 > > tcp: localhost:5060 > > tcp: localhost.localdomain:5060 > > udp: vx1:5060 > > udp: vx1.velocity11.com:5060 > > udp: localhost:5060 > > udp: localhost.localdomain:5060 > > > > > [ OK ] > > [root@vx1 ~]# > > > > > > > > > > > > > > > > > > > ____________________________________________________________________________________Ready > for the edge of your seat? > > Check out tonight's top picks on Yahoo! TV. > > http://tv.yahoo.com/ > > _______________________________________________ > > Serusers mailing list > > Serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > ________________________________ > Sucker-punch spam with award-winning protection. > Try the free Yahoo! Mail Beta. > _______________________________________________ > 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 ____________________________________________________________________________________Get the free Yahoo! toolbar and rest assured with the added security of spyware protection. http://new.toolbar.yahoo.com/toolbar/features/norton/index.php _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers ____________________________________________________________________________________Sick sense of humor? Visit Yahoo! TV's Comedy with an Edge to see what's on, when. http://tv.yahoo.com/collections/222

1 0

[Users] question about the openserctl ping command
by Papadopoulos Georgios 24 May '07

24 May '07

Hello all, I am using the openserct ping command to periodically check whether openser is working properly (If there is a better way to do this, please let me know). My script looks like this: SIP_URI="sip:213.5.xxx.xxx:5060" OPENSER_DIR=/root/openser TMP_FILE=/tmp/openser_ping.result $OPENSER_DIR/bin/openserctl ping $SIP_URI > $TMP_FILE 2>&1 Sometimes in TMP_FILE I see the following type of message: mkfifo: cannot create fifo `/tmp/openser_receiver_26907': File exists anaconda-ks.cfg install.log install.log.syslog mbox openser ERROR: error opening read fifo /tmp/openser_receiver_26907 At the same time openser seems to be working fine, however because the script gets this error, it restarts the server (the idea is that if ping fails then restart). Any idea why this is happening? What is the best way to make sure openser is working properly? I am assuming that a succesful ping only proves that one child is working properly. Thank you for any help George Disclaimer The information in this e-mail and any attachments is confidential. It is intended solely for the attention and use of the named addressee(s). If you are not the intended recipient, or person responsible for delivering this information to the intended recipient, please notify the sender immediately. Unless you are the intended recipient or his/her representative you are not authorized to, and must not, read, copy, distribute, use or retain this message or any part of it. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.

3 2

[Users] One way communication on calls from PSTN, two way on outbound
by Aaron Seelye 24 May '07

24 May '07

Hello all, I've based my config on the one found at http://siprouter.onsip.org/doc/gettingstarted/ch10.html, coming up with the one found here http://209.213.8.4/openser.cfg (all the xlog's are so I can follow what's going on when I run openser in a non-forking mode). I'm trying to get a simple system working where clients behind nat'ing firewalls can both send and receive calls. Currently outbound calls work fine, with two way communication, but inbound calls only have one way communication, from the nat'd client to the pstn. If anyone could give me some pointers as to what's going wrong, I'd appreciate it. TIA, Aaron Seelye

1 0

Re: [Serusers] Frustrating error - please help
by Mahatma Sarasvati 24 May '07

24 May '07

Also a good idea. Still no effect. This installation was working great for a while. Is it possible that some other software I've installed is causing the problem? This is 2.0.0 rc2 from CVS. M ----- Original Message ---- From: Andrey Kuprianov <andrey.kouprianov(a)gmail.com> To: "serusers(a)iptel.org" <serusers(a)iptel.org> Sent: Thursday, May 24, 2007 8:23:14 AM Subject: Re: [Serusers] Frustrating error - please help Maybe just try setting fork=yes ?... On 5/24/07, Mahatma Sarasvati <gnoahhb(a)yahoo.com> wrote: > > A good hint, but no. I'm using the standard version 1.4 ser.cfg. I've only > added my 10.10.10.41 ip address. The "fork=no" command is commented out. > > Is there something else that would cause ser to start in non-forking mode? > > Thanks, > M > > > ----- Original Message ---- > From: samuel <samu60(a)gmail.com> > To: Mahatma Sarasvati <gnoahhb(a)yahoo.com> > Cc: serusers(a)iptel.org > Sent: Thursday, May 24, 2007 12:40:41 AM > Subject: Re: [Serusers] Frustrating error - please help > > Are you starting SER in non-forking mode? It looks like SER is not binding > to the 10.10.10.41 IP, only to the local IP (127.0.0.1) and that's why you > see in the logs that it's not able to send to a "standard" IP address: > WARNING: xl_mod_init: more IP 10.10.10.41 not used > CRITICAL: invalid sendtoparameters one possible reason is the server is > bound to localhost and attempts to send to the net > > Case you need to use 2 interfaces, I would recommend using mhomed=yes (don't > recall the exact syntax right now..). > > Hope it helps, > Samuel. > > 2007/5/24, Mahatma Sarasvati <gnoahhb(a)yahoo.com>: > > Hi All, > > > > This ser installation was working fine and I can't figure out why its not > working now. > > I'm getting an error (see last log line) every time I attempt a call to > local users. Calling out to PSTN works great. ngrep reveals that SER is not > sending any messages to the callee location. > > > > There are 2 registered local users, one registered twice. > > > > I have no idea what is going on. I'm using an unmodified version 1.4 > ser.cfg file on CentOS 5. > > > > I suspect some server configuration error? ? ? ? > > > > > > Thanks, > > M > > > > > > > > Here are the logs. > > > > > > May 23 16:03:38 vx1 ser: init_tcp: using epoll_lt as the io watch method > (auto detected) > > May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: Maxfwd module- initializing > > May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: WARNING: xl_mod_init: more > IP 10.10.10.41 not used > > May 23 16:03:38 vx1 last message repeated 8 times > > May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: AVPops - initializing > > May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: DEB: IP address of RTPPROXY > is /var/run/rtpproxy.sock > > May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: INFO: xlog null is "<null>" > > May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: INFO: udp_init: SO_RCVBUF > is initially 109568 > > May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: INFO: udp_init: SO_RCVBUF > is finally 262142 > > May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: INFO: udp_init: SO_RCVBUF > is initially 109568 > > May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: INFO: udp_init: SO_RCVBUF > is finally 262142 > > . > > . > > . > > . > > May 23 16:09:15 vx1 /usr/local/sbin/ser[1781]: rtpp_test: RTP proxy > <unix:/var/run/rtpproxy.sock> found, support for it re-enabled > > May 23 16:09:15 vx1 /usr/local/sbin/ser[1781]: ERROR: udp_send: > sendto(sock,0xb62048f4,1773,0, 10.10.100.103:5060,16): Invalid argument(22) > > > > May 23 16:09:15 vx1 /usr/local/sbin/ser[1781]: CRITICAL: invalid > sendtoparameters one possible reason is the server is bound to localhost > > and attempts to send to the net > > May 23 16:09:15 vx1 /usr/local/sbin/ser[1781]: msg_send: ERROR: udp_send > failed > > May 23 16:09:15 vx1 /usr/local/sbin/ser[1781]: ERROR: t_send_branch: > sending request on branch 0 failed > > May 23 16:09:15 vx1 /usr/local/sbin/ser[1781]: ERROR: sl_reply_error used: > Unfortunately error on sending to next hop occurred (477/SL) > > > > > > > > When I start SER I get this response: > > > > [root@vx1 ~]# service ser restart > > Stopping ser: > [ OK ] > > Starting ser: Listening on > > udp: 127.0.0.1 [127.0.0.1]:5060 > > udp: 10.10.10.41 [10.10.10.41]:5060 > > tcp: 127.0.0.1 [127.0.0.1]:5060 > > tcp: 10.10.10.41 [10.10.10.41]:5060 > > Aliases: > > tcp: vx1:5060 > > tcp: vx1.velocity11.com:5060 > > tcp: localhost:5060 > > tcp: localhost.localdomain:5060 > > udp: vx1:5060 > > udp: vx1.velocity11.com:5060 > > udp: localhost:5060 > > udp: localhost.localdomain:5060 > > > > > [ OK ] > > [root@vx1 ~]# > > > > > > > > > > > > > > > > > > > ____________________________________________________________________________________Ready > for the edge of your seat? > > Check out tonight's top picks on Yahoo! TV. > > http://tv.yahoo.com/ > > _______________________________________________ > > Serusers mailing list > > Serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > ________________________________ > Sucker-punch spam with award-winning protection. > Try the free Yahoo! Mail Beta. > _______________________________________________ > 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 ____________________________________________________________________________________Get the free Yahoo! toolbar and rest assured with the added security of spyware protection. http://new.toolbar.yahoo.com/toolbar/features/norton/index.php

2 1

Re: [Serusers] Frustrating error - please help
by Mahatma Sarasvati 24 May '07

24 May '07

A good hint, but no. I'm using the standard version 1.4 ser.cfg. I've only added my 10.10.10.41 ip address. The "fork=no" command is commented out. Is there something else that would cause ser to start in non-forking mode? Thanks, M ----- Original Message ---- From: samuel <samu60(a)gmail.com> To: Mahatma Sarasvati <gnoahhb(a)yahoo.com> Cc: serusers(a)iptel.org Sent: Thursday, May 24, 2007 12:40:41 AM Subject: Re: [Serusers] Frustrating error - please help Are you starting SER in non-forking mode? It looks like SER is not binding to the 10.10.10.41 IP, only to the local IP (127.0.0.1) and that's why you see in the logs that it's not able to send to a "standard" IP address: WARNING: xl_mod_init: more IP 10.10.10.41 not used CRITICAL: invalid sendtoparameters one possible reason is the server is bound to localhost and attempts to send to the net Case you need to use 2 interfaces, I would recommend using mhomed=yes (don't recall the exact syntax right now..). Hope it helps, Samuel. 2007/5/24, Mahatma Sarasvati <gnoahhb(a)yahoo.com>: Hi All, This ser installation was working fine and I can't figure out why its not working now. I'm getting an error (see last log line) every time I attempt a call to local users. Calling out to PSTN works great. ngrep reveals that SER is not sending any messages to the callee location. There are 2 registered local users, one registered twice. I have no idea what is going on. I'm using an unmodified version 1.4 ser.cfg file on CentOS 5. I suspect some server configuration error? ? ? ? Thanks, M Here are the logs. May 23 16:03:38 vx1 ser: init_tcp: using epoll_lt as the io watch method (auto detected) May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: Maxfwd module- initializing May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: WARNING: xl_mod_init: more IP 10.10.10.41 not used May 23 16:03:38 vx1 last message repeated 8 times May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: AVPops - initializing May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: DEB: IP address of RTPPROXY is /var/run/rtpproxy.sock May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: INFO: xlog null is "<null>" May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: INFO: udp_init: SO_RCVBUF is initially 109568 May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: INFO: udp_init: SO_RCVBUF is finally 262142 May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: INFO: udp_init: SO_RCVBUF is initially 109568 May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: INFO: udp_init: SO_RCVBUF is finally 262142 . . . . May 23 16:09:15 vx1 /usr/local/sbin/ser[1781]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it re-enabled May 23 16:09:15 vx1 /usr/local/sbin/ser[1781]: ERROR: udp_send: sendto(sock,0xb62048f4,1773,0, 10.10.100.103:5060,16): Invalid argument(22) May 23 16:09:15 vx1 /usr/local/sbin/ser[1781]: CRITICAL: invalid sendtoparameters one possible reason is the server is bound to localhost and attempts to send to the net May 23 16:09:15 vx1 /usr/local/sbin/ser[1781]: msg_send: ERROR: udp_send failed May 23 16:09:15 vx1 /usr/local/sbin/ser[1781]: ERROR: t_send_branch: sending request on branch 0 failed May 23 16:09:15 vx1 /usr/local/sbin/ser[1781]: ERROR: sl_reply_error used: Unfortunately error on sending to next hop occurred (477/SL) When I start SER I get this response: [root@vx1 ~]# service ser restart Stopping ser: [ OK ] Starting ser: Listening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 10.10.10.41 [10.10.10.41]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 10.10.10.41 [10.10.10.41]:5060 Aliases: tcp: vx1:5060 tcp: vx1.velocity11.com:5060 tcp: localhost:5060 tcp: localhost.localdomain:5060 udp: vx1:5060 udp: vx1.velocity11.com:5060 udp: localhost:5060 udp: localhost.localdomain:5060 [ OK ] [root@vx1 ~]# ____________________________________________________________________________________Ready for the edge of your seat? Check out tonight's top picks on Yahoo! TV. http://tv.yahoo.com/ _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers ____________________________________________________________________________________ Need Mail bonding? Go to the Yahoo! Mail Q&A for great tips from Yahoo! Answers users. http://answers.yahoo.com/dir/?link=list&sid=396546091

2 1

[Users] Sometime t_really() func seem to be endless
by Bodin Bruno 24 May '07

24 May '07

Hi, I don't now why, but when call is cancel or unavaible t_relay is endless and never return anything. anyone know why ? Thank

4 15

[Serusers] SER User Group Meeting In Oslo
by Martin Hoffmann 24 May '07

24 May '07

Heisann, we are currently in the planning stage of the Second SER User Group Meeting. We agreed back in Prague to try to have it in Oslo in summer (no point in coming here in the dark season). The preliminary date for the meeting is Friday, July 13, 2007. However, since this is in the middle of the holiday seasons, I'd rather ask if this suits you all or whether we should move it to September. A more formal invitation will follow later. Best regards, Martin

6 7

[Users] failure_route problem
by Yago Bargalló 24 May '07

24 May '07

Hello, First of all thaks to Kraus for his quick reply on my last question. However I can't do openser work fine. I'm trying to use failure_route as follows, can anybody tell me which could be the problem?? (...) modparam("tm", "fr_inv_timer" , 10) # timer for response to a request modparam("tm","fr_timer", 4) # ------------------------- request routing logic ------------------- # main routing logic route{ (...) if (uri==myself) { (...) # Redirecciono cap a l'asterisk si marquen una extensió que comença per 3 if(uri=~"sip:2[0-9]+@" && method=="INVITE") { t_on_failure("1"); } (...) 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; } failure_route[1]{ rewritehostport("172.16.51.160"); #asterisk rewriteuser("2002"); append_branch(); t_relay(); exit; } With this openser.cfg the call is cancelled after fr_inv_timer and failure_route[1] doesn't work but, however, if I do this: route{ (...) if (uri==myself) { (...) # Redirecciono cap a l'asterisk si marquen una extensió que comença per 3 if(uri=~"sip:2[0-9]+@" && method=="INVITE") { à rewritehostport("172.16.51.160"); #asterisk à rewriteuser("2002"); t_relay(); exit; } (...) route(1); } If I use this the call is redirected to my asterisk but using the failure_route as my first openser.cfg it does't work. Which is my problem?? Thanks....

3 2

[Serusers] SIP Gateway
by David Borgesen 24 May '07

24 May '07

Hello, We are trying to find a SIP to SIP gateway and we fell across this project. Can someone tell me if we can accomplish the following with this software. 1) Our DID/SIP telephone provider does not support the Refer method, and as a result our SipX IP PBX can not transfer calls from extension to extension. 2) As a work around we are thinking that we could install a gateway between the DID provider and SipX where the gateway would support the Refer method and handle the transfers. This method is not ideal because all phone traffic would need to go through the gateway as opposed to a direct connection between the DID/SIP provider and the end user extension, but we can't see any other solution. So, can Ser act as a gateway as needed, or is there another solution that anyone of you can think of. Thanks DB

1 0

[Serusers] t_relay() TCP errors
by tzieleniewski 24 May '07

24 May '07

Hi, When I try to t_relay() a message through SER i get the following errors: May 22 10:31:21 rd ser[22137]: ERROR: tcp_blocking_connect: timeout 10 s elapsed from 10 s May 22 10:31:21 rd ser[22137]: ERROR: tcpconn_connect: tcp_blocking_connect failed May 22 10:31:21 rd ser[22137]: ERROR: tcp_send: connect failed May 22 10:31:21 rd ser[22137]: msg_send: ERROR: tcp_send failed Please point me what could be the reason for this? Maybe such situation is possible in case of non tcp server socket listening?? Cheers Tomasz

2 1

[Serusers] mysql dies -- accounting stopped -- call continues
by Nhadie 24 May '07

24 May '07

Hi All How can i make sure that when mysql dies, ser will not allow calls anymore? Because i had a problem, when mysql died, ser still continued to allow calls and i was not able to account it. How can i resolve this. Thank You in Advance. Regards Ronald

5 11

[Users] Loop construct in openser config file
by Steve Blair 24 May '07

24 May '07

Hello: Is it possible to implement a loop in the openser config file? Something simple that can iterate through a list of R-URI of AVP values. Thanks,Steve

3 2

[Users] (no subject)
by Yago Bargalló 24 May '07

24 May '07

Hello, First of all thaks to Kraus for his quick reply on my last question. However I can't do openser work fine. I'm trying to use failure_route as follows, can anybody tell me which could be the problem?? (...) modparam("tm", "fr_inv_timer" , 10) # timer for response to a request modparam("tm","fr_timer", 4) # ------------------------- request routing logic ------------------- # main routing logic route{ (...) if (uri==myself) { (...) # Redirecciono cap a l'asterisk si marquen una extensió que comença per 3 if(uri=~"sip:2[0-9]+@" && method=="INVITE") { t_on_failure("1"); } (...) 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; } failure_route[1]{ rewritehostport("172.16.51.160"); #asterisk rewriteuser("2002"); append_branch(); t_relay(); exit; } With this openser.cfg the call is cancelled after fr_inv_timer and failure_route[1] doesn't work but, however, if I do this: route{ (...) if (uri==myself) { (...) # Redirecciono cap a l'asterisk si marquen una extensió que comença per 3 if(uri=~"sip:2[0-9]+@" && method=="INVITE") { à rewritehostport("172.16.51.160"); #asterisk à rewriteuser("2002"); t_relay(); exit; } (...) route(1); } If I use this the call is redirected to my asterisk but using the failure_route as my first openser.cfg it does't work. Which is my problem?? Thanks....

1 0

Re: [Users] Errors with openserctl
by Henning Westerholt 24 May '07

24 May '07

On Donnerstag, 24. Mai 2007, you wrote: > Hi Henning, > > Thanks for youe response, but can you please tell me how to load the mysql > database engine ? Hi, add ' loadmodule "mysql.so" ' to your config file. Then you must also add a db_url to the module that should use the database, look for example to the usrloc module documentation. A sample setup is described at in the installation notes at: http://www.openser.org/mos/view/-OpenSER-Installation-Notes/ Henning

1 0

Re: [Users] active_watchers table
by Jeremy George 24 May '07

24 May '07

Hi Henning, Perfect. Thanks again. - Jeremy >-----Original Message----- >From: Henning Westerholt [mailto:henning.westerholt@1und1.de] >Sent: Thursday, May 24, 2007 04:39 AM >To: users(a)openser.org, 'Jeremy George' >Subject: Re: [Users] active_watchers table > >On Mittwoch, 23. Mai 2007, Jeremy George wrote: >> Hello All, >> >> I'm getting this error message: >> >> 0(0) PRESENCE:mod_init: Wrong version v1 for table <active_watchers>, need >> v4 >> 0(0) init_mod(): Error while initializing module presence >> ERROR: error while initializing modules >> >> I downloaded and compiled the current svn head (2276), and created the >> mysql active_watchers table defined in the README (see below). >> [..] >> Can anyone tell me where to find the right table? Thanks much. > >Hello Jeremy, > >the table that is defined by the README is to old. You can extract at the >moment all tables from the mysqldb.sh script. > >In the next weeks the db stuff will be updated, then all db schemas would be >up to date in a central place. > >I've attached the current table too. > >################# >INSERT INTO version (table_name, table_version) values >('active_watchers','4'); >CREATE TABLE active_watchers ( > id INT(10) UNSIGNED AUTO_INCREMENT NOT NULL, > pres_user VARCHAR(64) NOT NULL, > pres_domain VARCHAR(128) NOT NULL, > to_user VARCHAR(64) NOT NULL, > to_domain VARCHAR(128) NOT NULL, > from_user VARCHAR(64) NOT NULL, > from_domain VARCHAR(128) NOT NULL, > event VARCHAR(64) NOT NULL DEFAULT 'presence', > event_id VARCHAR(64), > to_tag VARCHAR(128) NOT NULL, > from_tag VARCHAR(128) NOT NULL, > callid VARCHAR(128) NOT NULL, > local_cseq INT(11) NOT NULL, > remove_cseq INT(11) NOT NULL, > contact VARCHAR(128) NOT NULL, > record_route TEXT, > expires INT(11) NOT NULL, > status VARCHAR(32) NOT NULL DEFAULT 'pending', > version INT(11) NOT NULL DEFAULT '0', > socket_info VARCHAR(128) NOT NULL, > local_contact VARCHAR(255) NOT NULL, > UNIQUE KEY tt_watchers (to_tag), > KEY due_activewatchers (to_domain, to_user, event), > PRIMARY KEY (id) >) Type=MyISAM; >######## > > >Best regards, > >Henning >

1 0

[Users] General for openser-2 things
by Yago Bargalló 24 May '07

24 May '07

Hello, I'm trying to redirect a call from openser to an Asterisk extension when a call is no answer. I think that this wouldn't be so difficult but I can't do it. I know that I have to use the failure_route(); with a fr_inv_timer parameter and then do a rewritehostport and a rewriteuser. My problem is that when the fr_inv_timer pass the call is cancelled and nothing else happens. In addition, I don't know where I can find the logs that I put in my openser.cfg, I have read the xlog module but I can't understand how to do it. On other hand, I have seen that there are a lot of functions that are often used but I don't know what they do internally, for example t_relay(), t_onreply(), revert_uri(),t_was_cancelled() etc etc etc and I don't know how to use it. ¿Is there any document that explain all it WELL?? Also, I've checked that if I do a call and I put the lines t_relay() and exit that call works fine but if I erase one of the two lines it don't work. Why??? Please help, I want to do a lot of things with openser but there is nothing that explain the basic functionality to do interessant things... Thanks in advance, "It will arrive the day that we will control telephony"

2 1

Re: [Serusers] Ser + Asterisk on the same box??
by John Breen 24 May '07

24 May '07

SIP wrote: > Your biggest issue, bar none, is using the loopback address for > anything externally-facing. 127.0.0.1 is called a loopback address for > a reason -- it never leaves the host itself. Just how do you expect > the SIP phones to be able to contact a 127.0.0.1 address when that > address is, in essence, themselves? > > There's something inherently flawed in your design. > > Let's back up and explain, in detail, what it is exactly you're trying > to do, and why you feel the 127.0.0.1 address should be part of this > equation? Perhaps we can suggest something if we know what you're > trying to do. > There's a very simple reason. Asterisk and SER are on the same machine. So the traffic from SER to asterisk doesn't need to leave the box. This is why I'm using the loopback interface - there's no need to use the real world one. -- Regards, John Breen HGA IT&T Solutions Tel: 08 63113000 Fax: 08 63113144 http://www.hgait.net.au Level 9, 256 Adelaide Terrace, Perth, WA 6000 Postal PO Box 3133 Perth, Adelaide Tce WA 6832 The information in this E-mail is confidential. If you are not the named & intended recipient, you must not read, use, copy, disclose or disseminate this E-mail, any attachments or the information contained in this E-mail and must immediately notify the sender & delete this E-mail immediately. Although HGA IT&T Solutions believes that this E-mail and any attachments are free of any virus or other defect which may affect a computer, it is the responsibility of the recipient to ensure that it is virus free. HGA IT&T Solutions does not accept any responsibility for any loss or damage arising in any way from the use of this E-mail or any attachments. IMPORTANT INFORMATION This is not a SPAM. The SPAM Act 2003 commenced on the 10th April 2004 and requires consent to the sending of most commercial electronic messages. If you wish to continue receiving electronic communications from HGA IT&T Solutions, you do not need to take any action. If you DO NOT wish to continue receiving electronic communications from HGA IT&T Solutions please advise us at admin(a)hgait.net.au and we will update our records accordingly.

4 3

[Users] Errors with openserctl
by kiran.bkumar＠wipro.com 24 May '07

24 May '07

Hi all, I am new user to openser, and I am trying to install openser to work as a presence server with the openims. Firstly is it possible to use the openser as a presence server. Secondly, I am getting the below error when I try to run the openserctl. [root@mkd149633 scripts]# openserctl moni ERROR: Error opening OpenSER's FIFO /tmp/openser_fifo ERROR: Make sure you have line 'fifo=/tmp/openser_fifo' in your config Also I am not able to add users , [root@mkd149633 scripts]# openserctl add wipro1 wipro1 wipro1(a)presence.mkd149633.wipro.com This command requires a database engine - none was loaded [root@mkd149633 scripts]# Looking forward to your support. Thanks and Regards, 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

2 1

[Users] active_watchers table
by Jeremy George 24 May '07

24 May '07

Hello All, I'm getting this error message: 0(0) PRESENCE:mod_init: Wrong version v1 for table <active_watchers>, need v4 0(0) init_mod(): Error while initializing module presence ERROR: error while initializing modules I downloaded and compiled the current svn head (2276), and created the mysql active_watchers table defined in the README (see below). I'm attempting to run it against the sample bla config found at: http://www.openser.org/dokuwiki/doku.php/presence:pua_bla (with some changes for appropriate environment). Can anyone tell me where to find the right table? Thanks much. - Jeremy CREATE TABLE `active_watchers` ( `id` int(10) NOT NULL auto_increment, `pres_user` varchar(64) NOT NULL, `pres_domain` varchar(128) NOT NULL, `to_user` varchar(64) NOT NULL, `to_domain` varchar(128) NOT NULL, `from_user` varchar(64) NOT NULL, `from_domain` varchar(128) NOT NULL, `event` varchar(64) NOT NULL default 'presence', `event_id` varchar(64), `to_tag` varchar(128) NOT NULL, `from_tag` varchar(128) NOT NULL, `callid` varchar(128) NOT NULL, `local_cseq` int(11) NOT NULL, `remote_cseq` int(11) NOT NULL, `contact` varchar(128) NOT NULL, `record_route` text, `expires` int(11) NOT NULL, `status` varchar(32) NOT NULL default 'pending', `version` int(11) default '0', `socket_info` varchar(128) NOT NULL, `local_contact` varchar(255) NOT NULL, PRIMARY KEY (`id`), UNIQUE KEY `tt_watchers` (`to_tag`), KEY `due_activewatchers` (`to_domain`,`to_user`,`event`) ) ENGINE=MyISAM DEFAULT CHARSET=latin1 AUTO_INCREMENT=1 ;

2 1

[Serusers] Frustrating error - please help
by Mahatma Sarasvati 24 May '07

24 May '07

Hi All, This ser installation was working fine and I can't figure out why its not working now. I'm getting an error (see last log line) every time I attempt a call to local users. Calling out to PSTN works great. ngrep reveals that SER is not sending any messages to the callee location. There are 2 registered local users, one registered twice. I have no idea what is going on. I'm using an unmodified version 1.4 ser.cfg file on CentOS 5. I suspect some server configuration error? ? ? ? Thanks, M Here are the logs. May 23 16:03:38 vx1 ser: init_tcp: using epoll_lt as the io watch method (auto detected) May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: Maxfwd module- initializing May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: WARNING: xl_mod_init: more IP 10.10.10.41 not used May 23 16:03:38 vx1 last message repeated 8 times May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: AVPops - initializing May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: DEB: IP address of RTPPROXY is /var/run/rtpproxy.sock May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: INFO: xlog null is "<null>" May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: INFO: udp_init: SO_RCVBUF is initially 109568 May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: INFO: udp_init: SO_RCVBUF is finally 262142 May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: INFO: udp_init: SO_RCVBUF is initially 109568 May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: INFO: udp_init: SO_RCVBUF is finally 262142 . . . . May 23 16:09:15 vx1 /usr/local/sbin/ser[1781]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it re-enabled May 23 16:09:15 vx1 /usr/local/sbin/ser[1781]: ERROR: udp_send: sendto(sock,0xb62048f4,1773,0,10.10.100.103:5060,16): Invalid argument(22) May 23 16:09:15 vx1 /usr/local/sbin/ser[1781]: CRITICAL: invalid sendtoparameters one possible reason is the server is bound to localhost and attempts to send to the net May 23 16:09:15 vx1 /usr/local/sbin/ser[1781]: msg_send: ERROR: udp_send failed May 23 16:09:15 vx1 /usr/local/sbin/ser[1781]: ERROR: t_send_branch: sending request on branch 0 failed May 23 16:09:15 vx1 /usr/local/sbin/ser[1781]: ERROR: sl_reply_error used: Unfortunately error on sending to next hop occurred (477/SL) When I start SER I get this response: [root@vx1 ~]# service ser restart Stopping ser: [ OK ] Starting ser: Listening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 10.10.10.41 [10.10.10.41]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 10.10.10.41 [10.10.10.41]:5060 Aliases: tcp: vx1:5060 tcp: vx1.velocity11.com:5060 tcp: localhost:5060 tcp: localhost.localdomain:5060 udp: vx1:5060 udp: vx1.velocity11.com:5060 udp: localhost:5060 udp: localhost.localdomain:5060 [ OK ] [root@vx1 ~]# ____________________________________________________________________________________Ready for the edge of your seat? Check out tonight's top picks on Yahoo! TV. http://tv.yahoo.com/

2 1

[Serusers] Ser + Asterisk on the same box??
by John Breen 24 May '07

24 May '07

Ok, let me see if I've got this right. I'm running SER and Asterisk on the same box. SER listens on the external IP of the box on port 5060, Asterisk on localhost (127.0.0.1) on port 5061. My starting point to get calls from SER able to use the real world is the SER PSTN gateway in the Getting Started document, with route[5] changed to: route[5] { rewritehostport("127.0.0.1:5061"); avp_write("i:45", "inv_timeout"); route(1); } I start SER and add accounts 306, 307 and 499. So far so good, and when I get it all up and running, SIP phones can call other SIP phones via SER. Now, if I read right, I then need to tell SER that 127.0.0.1 is a trusted account by doing INSERT INTO trusted VALUES('127.0.0.1', 'any', '^sip:.*$'); in mysql, and restarting SER. I can still dial between internal phones and that is all OK, but I cannot dial via asterisk. I don't even see calls getting there. If I then add into the trusted table the IP addresses of SIP clients, I can dial from them via asterisk. But I have to do this for every client. This is fine if there are only a limited number of IP addresses that can use the asterisk gateway, but I have users with laptops in the real world whose IP addresses change so it's obviously not an ideal solution. What am I missing? Is the Call Forwarding setup the answer? I need to use SER as a front end to asterisk because of NAT issues, but I don't want to have to add EVERY user to the trusted table individually. Regards John Breen -- Regards, John Breen HGA IT&T Solutions Tel: 08 63113000 Fax: 08 63113144 http://www.hgait.net.au Level 9, 256 Adelaide Terrace, Perth, WA 6000 Postal PO Box 3133 Perth, Adelaide Tce WA 6832 The information in this E-mail is confidential. If you are not the named & intended recipient, you must not read, use, copy, disclose or disseminate this E-mail, any attachments or the information contained in this E-mail and must immediately notify the sender & delete this E-mail immediately. Although HGA IT&T Solutions believes that this E-mail and any attachments are free of any virus or other defect which may affect a computer, it is the responsibility of the recipient to ensure that it is virus free. HGA IT&T Solutions does not accept any responsibility for any loss or damage arising in any way from the use of this E-mail or any attachments. IMPORTANT INFORMATION This is not a SPAM. The SPAM Act 2003 commenced on the 10th April 2004 and requires consent to the sending of most commercial electronic messages. If you wish to continue receiving electronic communications from HGA IT&T Solutions, you do not need to take any action. If you DO NOT wish to continue receiving electronic communications from HGA IT&T Solutions please advise us at admin(a)hgait.net.au and we will update our records accordingly.

2 1

[Users] CPU load, ERROR: append_branch: max nr of branches exceeded, etc..
by Dan Mechanic 24 May '07

24 May '07

Forgive me if this question is a little open-ended... I am new to openser and sip, and trying to learn all I can... I am looking for information regarding: ERROR: append_branch: max nr of branches exceeded I understand that this error can be avoided by increasing the value of MAX_BRANCHES and recompiling openser, but, I am curious as to what happens when this error has been reached. Here's our situation... We use openser, and have a separate openser presence server. We had a situation recently where it seemed our presence server was bombarding our proxy to the point where the load on the proxy jumped and phones became unusable. Taking down the presence server seemed to alleviate the troubles. Also, we have an intermittent issue with phone reboots and lost calls, particularly on extensions that are on more than 12 phones (MAX_BRANCHES is currently set to 12). So, my question is, is this failure (max nr of branches exceeded) an innocuous one? or could it lead to futher failures (and maybe sip message recents... thus the bombardment)... Are there known side affects of this failure? Any info would be appreciated. Thanks Much! Best, Dan Mechanic

2 1

[Users] Require Core Dump Analysis
by Martin Burns 23 May '07

23 May '07

Hi, After many many months of faultless operation, today we experienced a SEG FAULT on one of our proxies running OpenSER 1.1.0-notls: Below is a back trace (with some values masked with xxx's), however our initial analysis can not make any sense of what the problem is. Any ideas would be greatly appreciated. res.c:194 is =====> r = (db_res_t*)pkg_malloc(sizeof(db_res_t)); Here is the back trace: (gdb) bt #0 0x00000000 in ?? () #1 0x4016ad2c in new_result () at res.c:194 #2 0x4016a495 in store_result (_h=0x81301b8, _r=0xbffff198) at dbase.c:322 #3 0x404bfc0a in db_load_avp (uuid=0x81301b8, username=0x81301b8, domain=0x81301b8, attr=0x81301b8 "@X\022\bø\034\023\b", table=0x0, scheme=0x404ce9e0) at avpops_db.c:270 #4 0x404c05d6 in ops_dbload_avps (msg=0x8134cd8, sp=0x8128210, dbp=0x8128b18, use_domain=0) at avpops_impl.c:393 #5 0x404bef47 in w_dbload_avps (msg=0x81301b8, source=0x81301b8 "@X\022\bø\034\023\b", param=0x81301b8 "@X\022\bø\034\023\b") at avpops.c:1183 #6 0x0805071b in do_action (a=0x8128de8, msg=0x8134cd8) at action.c:701 #7 0x08050631 in do_action (a=0x8129478, msg=0x8134cd8) at action.c:89 #8 0x08050631 in do_action (a=0x81294a8, msg=0x8134cd8) at action.c:89 #9 0x080522c2 in run_actions (a=0x8126548, msg=0x8134cd8) at action.c:89 #10 0x08051f2e in run_top_route (a=0x81301b8, msg=0x81301b8) at action.c:151 #11 0x08071ec1 in receive_msg ( buf=0x80ea340 "REGISTER sip:xx.xxxxxxx.xxx SIP/2.0\r\nVia: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK-f1cef63e\r\nFrom: xxxxx < sip:xxxxxx@xx.xxxxxxx.xxx>;tag=fe10f354c23eb963o0\r\nTo: xxxxx < sip:xxxxxx@xx.xxxxxxx.xxx>\r\nC"..., len=475, rcv_info=0xbffff9a0) at receive.c:155 #12 0x08089654 in udp_rcv_loop () at udp_server.c:465 #13 0x08063b64 in main_loop () at main.c:925 #14 0x08064fae in main (argc=9, argv=0xbffffb24) at main.c:1477 (gdb) quit Thanks in advance Martin

1 0

[Users] OpenSER v1.2.1 released
by Bogdan-Andrei Iancu 23 May '07

23 May '07

Hello, Version 1.2.1 of OpenSER has been released. It is an update of version 1.2.0 and includes the fixes for the issues discovered in version 1.2.0, therefore all people using 1.2.0 should upgrade to this new version. The compatibility of the configuration file and database has been preserved (you do not need to do database reinstallation/migration). Detailed changelog: http://www.openser.org/pub/openser/1.2.1/ChangeLog For people using SVN, a new TAG was created ( 1.2.1 ). The development of the 1.2 branch will continue on the SVN 1.2 branch. So, if you want to synchronize with the latest changes/release in 1.2, keep using the 1.2 branch. If you want to take the 1.2.1 snapshot, use the 1.2.1 SVN tag. Source tarball can be downloaded from: http://www.openser.org/pub/openser/latest/src/ Or SourceForge: http://sourceforge.net/project/showfiles.php?group_id=139143&package_id=224… If someone wants to contribute with packages for different distros, please send the link from where can be downloaded to team(a)openser.org. Regards, Bogdan

1 0

[Serusers] acl
by flavio 23 May '07

23 May '07

---------- Forwarded message ---------- From: flavio <flavio.patria(a)gmail.com> Date: 23-mag-2007 17.35 Subject: acl To: serusers(a)list.iptel.org Cc: serweb-users(a)lists.iptel.org Hi to all, I have successfully installed serweb and I can manage acl users grants as admin. Should I configure my ser.cfg anyway?how to? Any references will be well accepted. Thanks, Flavio -- ******************************** * (o< ing. Patria Flavio * //\ phone 0823451358 * V_/_ mobile 3407873357 * ********************************

1 0

[Users] Content-Lenght problem with NOTIFYs
by Tobias Lindgren 23 May '07

23 May '07

Hi all, a customer of ours are having some trouble with the Content-Lenght header when a NOTIFY-message goes through our OpenSER 1.2.0. Here is what seem to happend. SIP NOTIFY from our PSTN-gateway to our OpenSER: --- CUT --- Content-Length: 21. . SIP/2.0 100 Trying. . --- CUT --- SIP NOTIFY from our OpenSER to Customer: --- CUT --- Content-Length: 21. . SIP/2.0 100 Trying. --- CUT --- Now, the customer has to manually do this in his OpenSER to make the UA on his side happy: if (method=="NOTIFY") { if (search_body("SIP/2\.0 100 Trying")) { xlog("L_WARN", "R: Setting Content-Length to 20\n"); remove_hf("Content-Length"); append_hf("Content-Length: 20\r\n"); } } Please advice on what we are doing wrong here? Br, /Tobias

5 10

[Users] How to use pdt module correctly?
by Charles Wang 23 May '07

23 May '07

Hi all, I try to rewrite 444111(a)aaa.com to 111(a)bbb.com with pdt module. ############# pdt table in DB ################## mysql> select * from pdt; +----+----------------------+----------+--------------+ | id | sdomain | prefix | domain | +----+----------------------+----------+--------------+ | 1 | bbb.com | 444 | bbb.com | +----+----------------------+----------+--------------+ ############## parts of my openser.cfg ##################################3 loadmodule "pdt.so" modparam("pdt", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("pdt", "db_table", "pdt") modparam("pdt", "domain_column", "sdomain") modparam("pdt", "prefix_column", "prefix") modparam("pdt", "prefix", "") modparam("pdt", "hsize_2pow", 4) modparam("pdt", "sync_time", 300) modparam("pdt", "clean_time", 600) route[0] { ........................(skip) if (is_method("INVITE")) { xlog("L_INFO", "Prefix To Domain I - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); prefix2domain("0","0"); xlog("L_INFO", "Prefix To Domain II - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); }; ........................(skip) } #################### /var/log/message ####################################### May 23 15:55:12 localhost openser[22172]: New request - M=INVITE RURI= sip:444111@aaa.com F=sip:222@aaa.com T=sip:444111@aaa.com IP=60.244.xxx.xxx ID=88BEB1DB-B447-4DAF-A3D6-212DBA3EC628(a)10.6.1.185 May 23 15:55:12 localhost openser[22172]: Prefix To Domain I - M=INVITE RURI=sip:444111@aaa.com F=sip:222@aaa.com T=sip:444111@aaa.com IP= 60.244.xxx.xxx ID=88BEB1DB-B447-4DAF-A3D6-212DBA3EC628(a)10.6.1.185 May 23 15:55:12 localhost openser[22172]: PDT:prefix2domain: no prefix found in [444111] May 23 15:55:12 localhost openser[22172]: Prefix To Domain II - M=INVITE RURI=sip:444111@aaa.com F=sip:222@aaa.com T=sip:444111@aaa.com IP= 60.244.xxx.xxx ID=88BEB1DB-B447-4DAF-A3D6-212DBA3EC628(a)10.6.1.185 -- Best Regards Charles

2 4

[Users] OpenSER Acc, Direction Detecting?
by Brandon Armstead 23 May '07

23 May '07

Hello everyone, I'm currently running into the following problem, I currently run openser SVN 1.3. With the following modparam("acc", "detect_direction", 1) ON INCOMING CALL FROM PSTN AND HANGUP ON UAC from/to username is flipped. (ERROR) ON INCOMING CALL FROM PSTN AND HANGUP ON PSTN from/to username remains 'detected' ON OUTGOING CALL FROM UAC AND HANGUP ON PSTN. from/to username is flipped. (ERROR) ON OUTGOING CALL FROM UAC AND HANGUP ON UAC from/tousername remains 'detected' Has anyone ran into this issue, or know what I can do to remain direction state? Thanks, all help is appreciated. -- Brandon Armstead

2 3

[Serusers] test
by Jan Janak 23 May '07

23 May '07

please ignore

1 0

[Users] Need starter information (newbie)
by Rizwan Hisham 23 May '07

23 May '07

Hi all, this is my first email to this list. hoping to mingle with you all from now on. I have been working on asterisk for about a year. Now my company is planning to use openser with asterisk having asterisk at backend for features and voicemail. So is there any book for starters for openser as there is for asterisk (Asterisk the future of telephony). i need a book or any helpfull link which should reveil the basic concepts of openser, how to install it, how to program it, and how to manage it. Thanx in advance -- Rizwan Hisham Software Engineer AXVOICE Inc.

2 1

[Users] dispatcher question
by Klaus Darilion 23 May '07

23 May '07

Hi! Let's assume I have a load balancer which distributes (using hash over R-URI) calls to 4 proxies. Further, ds_mark_dst() is used to marked broken proxys. Marking a destination as inactive - does this change the routing completely (as if there were only 3 proxies in the config file) or will it still hash over the 4 proxies, but choose automatically the second entry in the resulting list if the result is the inactive proxy? regards klaus

2 3

[Users] XCAP Lite 1.0.1 released
by Adrian Georgescu 23 May '07

23 May '07

Change log xcap-lite (1.0.1) unstable; urgency=low * Added support for multiple applications and detect them based on xcap-root * Added xcap-root setting in config.php * Added pidf-manipulation application (to allow persistent publication of data using xcap server) * Detect auth_domain from XCAP URI if SIP user is formatted as user@domain * Added DELETE method The software can be downloaded from: http://download.dns-hosting.info/XCAP/xcap-lite-1.0.1.tar.gz Regards, Adrian

1 0

[Users] WeSIP: RingTone
by Helmut Kuper 23 May '07

23 May '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I play around with WeSIP's ringtone app. I try to get ringtone running in this constellation: caller -> openser sip-proxy -> openser+wesip -> -> target sip-proxy -> callee openser+wesip is loaded with both c2c and ringtone app. When openser sip-proxy relay callers INVITE to wesip, wesip response with "SIP/2.0 481 Call Leg/Transaction Does Not Exist". I dunno wheter this is an answere from ringtone app or from wesip itself. I also dunno, whether I have to config ringtone with environment specific things, like sip port, ip address or something like that or whether my constallation from above is correct. Maybe someone can help me out here :) Here is WeSIP's log, after receiving caller's INVITE: Starting WESIP Application Server VozTelecom... hk@vm-en-test0:~/wesip/wesip$ HttpConnector [main] - The HttpConnector [all:8080] has been created ExpressConnector [main] - SEAS module successfully connected at 212.6.123.118:5081 ExpressConnector [main] - trying to receive 2 socket descriptions from SEAS ExpressConnector [main] - SER socket:[212.6.123.118:5080/udp] ExpressConnector [main] - SER socket:[212.6.123.118:5080/udp] ExpressConnector [main] - SER socket:[212.6.123.118:5080/tcp] ExpressConnector [main] - SER socket:[212.6.123.118:5080/tcp] SeasRouter [SeasRouter] - SeasDispatcher Thread starts running SeasRouter [SeasRouter] - SeasDispatcher Thread starts running SeasHashTable [main] - Creating Seas Hash Table, with tablePower=17 and total entries:131072 SeasHashTable [main] - Creating Seas Hash Table, with tablePower=17 and total entries:131072 ExpressMessageChannel [ExpressMessageProcessor] - setting spiral=false ExpressMessageChannel [ExpressMessageProcessor] - setting spiral=false SipConnector [main] - The SipConnector [212.6.123.118:5080] has been created SipProcessor [main] - Starting background thread ExpressMessageChannel [ExpressMessageProcessor] - setting spiral=false ExpressMessageChannel [ExpressMessageProcessor] - setting spiral=false ExpressMessageChannel [ExpressMessageProcessor] - setting spiral=false ExpressMessageChannel [ExpressMessageProcessor] - setting spiral=false ExpressMessageChannel [ExpressMessageProcessor] - setting spiral=false ExpressMessageChannel [ExpressMessageProcessor] - setting spiral=false ExpressMessageChannel [ExpressMessageProcessor] - setting spiral=false ExpressMessageChannel [ExpressMessageProcessor] - setting spiral=false SipProcessor [main] - Background thread has been started SipConnector [main] - recycle: Recycling processor SipProcessor[0] SipProcessor [main] - Starting background thread SipProcessor [main] - Background thread has been started SipConnector [main] - recycle: Recycling processor SipProcessor[1] SipProcessor [main] - Starting background thread SipProcessor [main] - Background thread has been started SipConnector [main] - recycle: Recycling processor SipProcessor[2] SipProcessor [main] - Starting background thread SipProcessor [main] - Background thread has been started SipConnector [main] - recycle: Recycling processor SipProcessor[3] SipProcessor [main] - Starting background thread SipProcessor [main] - Background thread has been started SipConnector [main] - recycle: Recycling processor SipProcessor[4] ConvergedEngine [main] - 212.6.123.118_5080 ConvergedEngine [main] - 212.6.123.118_5080 ConvergedEngine [main] - 212.6.123.118_5080 ConvergedEngine [main] - 212.6.123.118_5080 StatisticsServlet [main] - Wrapper setted in StatisticsServlet!!! ConvergedHostDeployer [main] - Installing web application at context path /click2call-08_11_06 from URL file:/home/hk/wesip/wesip/wesipapps/click2call-08_11_06 StandardSipManager [main] - Starting StandardSipManager [main] - Force random number initialization starting StandardSipManager [main] - Force random number initialization completed StandardSipManager [main] - Start: Loading persisted sessions StandardSipManager [main] - Loading persisted sessions from SIPSESSIONS.ser StandardSipManager [main] - No persisted data file found StandardAppSessionManager [main] - Starting StandardAppSessionManager [main] - Force random number initialization starting StandardAppSessionManager [main] - Force random number initialization completed AxisServletBase [main] - In AxisServletBase init AxisServletBase [main] - Enter: getEngine() ProjectResourceBundle [main] - getBundle(org.apache.axis,org.apache.axis.i18n,resource,null,...) ProjectResourceBundle$Context [main] - loadBundle: Ignoring MissingResourceException: Can't find bundle for base name org.apache.axis.resource, locale en_US ProjectResourceBundle [main] - Created org.apache.axis.i18n.resource, linked to parent null ProjectResourceBundle [main] - getBundle(org.apache.axis,org.apache.axis.utils,resource,null,...) ProjectResourceBundle$Context [main] - loadBundle: Ignoring MissingResourceException: Can't find bundle for base name org.apache.axis.utils.resource, locale en_US ProjectResourceBundle$Context [main] - loadBundle: Ignoring MissingResourceException: Can't find bundle for base name org.apache.axis.resource, locale en_US ProjectResourceBundle [main] - Root package not found, cross link to org.apache.axis.i18n.resource ProjectResourceBundle [main] - Root package not found, cross link to org.apache.axis.i18n.resource ProjectResourceBundle [main] - org.apache.axis.i18n.resource::handleGetObject(engineFactory) EngineConfigurationFactoryFinder$1 [main] - Got EngineFactory: org.apache.axis.configuration.EngineConfigurationFactoryServlet DefaultAxisServerFactory [main] - Enter: DefaultAxisServerFactory::getServer DefaultAxisServerFactory [main] - Enter: DefaultAxisServerFactory::getEngineConfiguration DefaultAxisServerFactory [main] - Exit: DefaultAxisServerFactory::getEngineConfiguration AxisEngine [main] - Enter: AxisEngine::init ProjectResourceBundle [main] - org.apache.axis.i18n.resource::handleGetObject(attachEnabled) JavaUtils [main] - Attachment support is enabled? false ProjectResourceBundle [main] - org.apache.axis.i18n.resource::handleGetObject(attachDisabled) JavaUtils [main] - Unable to find required classes (javax.activation.DataHandler and javax.mail.internet.MimeMultipart). Attachment support is disabled. ProjectResourceBundle [main] - org.apache.axis.i18n.resource::handleGetObject(oddDigits00) ProjectResourceBundle [main] - org.apache.axis.i18n.resource::handleGetObject(badChars01) AxisEngine [main] - Exit: AxisEngine::init DefaultAxisServerFactory [main] - Exit: DefaultAxisServerFactory::getServer AxisServletBase [main] - Exit: getEngine() AxisServlet [main] - In servlet init AxisServletBase [main] - Enter: getEngine() AxisServletBase [main] - Exit: getEngine() AxisServletBase [main] - Enter: getEngine() AxisServletBase [main] - Exit: getEngine() AxisServletBase [main] - Enter: getEngine() AxisServletBase [main] - Exit: getEngine() AxisServletBase [main] - Enter: getEngine() AxisServletBase [main] - Exit: getEngine() AxisServletBase [main] - Enter: getEngine() AxisServletBase [main] - Exit: getEngine() AxisServletBase [main] - Enter: getEngine() AxisServletBase [main] - Exit: getEngine() AxisServletBase [main] - Enter: getEngine() AxisServletBase [main] - Exit: getEngine() Registrar [main] - Registering org.apache.axis.management.jmx.ServiceAdministrator@10ffb38 as axis:type=server Registry [main] - Managed= axis:type=server Registry [main] - Looking for descriptor Registry [main] - Introspecting Registry [main] - load class org.apache.axis.management.jmx.ServiceAdministrator MbeansDescriptorsIntrospectionSource [main] - Introspected attribute version public java.lang.String org.apache.axis.management.jmx.ServiceAdministrator.getVersion() null MbeansDescriptorsIntrospectionSource [main] - Setting name: ServiceAdministrator BaseModelMBean [main] - default constructor Registry [main] - Creating MBeanServer9 BaseModelMBean [main] - preRegister org.apache.axis.management.jmx.ServiceAdministrator@10ffb38 axis:type=server Registrar$ModelerBinding [main] - Registered axis:type=server in ServiceAdministrator Registrar [main] - Registering org.apache.axis.management.jmx.DeploymentAdministrator@b40ec4 as axis:type=deploy Registry [main] - Managed= axis:type=deploy Registry [main] - Looking for descriptor Registry [main] - Introspecting Registry [main] - load class org.apache.axis.management.jmx.DeploymentAdministrator MbeansDescriptorsIntrospectionSource [main] - Setting name: DeploymentAdministrator BaseModelMBean [main] - default constructor BaseModelMBean [main] - preRegister org.apache.axis.management.jmx.DeploymentAdministrator@b40ec4 axis:type=deploy Registrar$ModelerBinding [main] - Registered axis:type=deploy in DeploymentAdministrator Registrar [main] - Registering org.apache.axis.management.jmx.DeploymentQuery@edf730 as axis:type=query Registry [main] - Managed= axis:type=query Registry [main] - Looking for descriptor Registry [main] - Introspecting Registry [main] - load class org.apache.axis.management.jmx.DeploymentQuery MbeansDescriptorsIntrospectionSource [main] - Setting name: DeploymentQuery BaseModelMBean [main] - default constructor BaseModelMBean [main] - preRegister org.apache.axis.management.jmx.DeploymentQuery@edf730 axis:type=query Registrar$ModelerBinding [main] - Registered axis:type=query in DeploymentQuery AxisServletBase [main] - In AxisServletBase init AxisServletBase [main] - Enter: getEngine() ProjectResourceBundle [main] - org.apache.axis.i18n.resource::handleGetObject(engineFactory) EngineConfigurationFactoryFinder$1 [main] - Got EngineFactory: org.apache.axis.configuration.EngineConfigurationFactoryServlet DefaultAxisServerFactory [main] - Enter: DefaultAxisServerFactory::getServer DefaultAxisServerFactory [main] - Enter: DefaultAxisServerFactory::getEngineConfiguration DefaultAxisServerFactory [main] - Exit: DefaultAxisServerFactory::getEngineConfiguration AxisEngine [main] - Enter: AxisEngine::init AxisEngine [main] - Exit: AxisEngine::init DefaultAxisServerFactory [main] - Exit: DefaultAxisServerFactory::getServer AxisServletBase [main] - Exit: getEngine() ConvergedContext [main] - ConvergedContext [Click2Call SipServlet] has been started ConvergedHostDeployer [main] - Installing web application at context path /ringtone-22_02_07 from URL file:/home/hk/wesip/wesip/wesipapps/ringtone-22_02_07 StandardSipManager [main] - Starting StandardSipManager [main] - Force random number initialization starting StandardSipManager [main] - Force random number initialization completed StandardSipManager [main] - Start: Loading persisted sessions StandardSipManager [main] - Loading persisted sessions from SIPSESSIONS.ser StandardSipManager [main] - No persisted data file found StandardAppSessionManager [main] - Starting StandardAppSessionManager [main] - Force random number initialization starting StandardAppSessionManager [main] - Force random number initialization completed ConvergedContext [main] - ConvergedContext [Ringtone] has been started ConvergedHostDeployer [main] - Installing web application at context path from URL file:/home/hk/wesip/wesip/wesipapps/ROOT StandardSipManager [main] - Starting StandardSipManager [main] - Force random number initialization starting StandardSipManager [main] - Force random number initialization completed StandardSipManager [main] - Start: Loading persisted sessions StandardSipManager [main] - Loading persisted sessions from SIPSESSIONS.ser StandardSipManager [main] - No persisted data file found StandardAppSessionManager [main] - Starting StandardAppSessionManager [main] - Force random number initialization starting StandardAppSessionManager [main] - Force random number initialization completed processJars: cannot find /WEB-INF/lib ConvergedContext [main] - ConvergedContext [Welcome] has been started StandardService [main] - > Default Wesip Application ==> http://localhost:8080/ > Manager Wesip Application ==> http://localhost:8080/manager/html? StandardService [main] - > In order to access the manager application use default credentials: - username: wesip - password: wesip > This configuration is defined in wesip-users.xml inside conf folder: - file: ./conf/wesip-users.xml SeasMessageEvent [SeasRouter] - New message received, from 212.6.123.118:5060 to 212.6.123.118:5080 over UDP SeasTransactionEvent [SeasRouter] - hash_index = 46338,label=965718283 flags=0 ExpressMessageChannel [ExpressMChannel[0]] - peerAddress = 212.6.123.118/5060 ExpressMessageChannel [ExpressMChannel[0]] - About to process INVITE sip:v111@212.6.123.118:5080;rinstance=fe71c43d95b8211a SIP/2.0 SipConnector [ExpressMChannel[0]] - createProcessor: Reusing existing processor SipProcessor [ExpressMChannel[0]] - An incoming message is being assigned SipProcessor [SipProcessor[4]] - <<<<<<<<< Request Received <<<<<<<<< INVITE sip:v111@212.6.123.118:5080;rinstance=fe71c43d95b8211a SIP/2.0 Record-Route: <sip:212.6.123.118;lr=on;ftag=450c2a3b> Via: SIP/2.0/UDP 212.6.123.118;branch=z9hG4bK205b.8c060ce2.0,SIP/2.0/UDP 212.6.95.198:25952;branch=z9hG4bK-d87543-ed108070b87aca0e-1--d87543-;rport=25952 Max-Forwards: 69 Contact: <sip:v101@212.6.95.198:25952> To: "Bruening" <sip:v111@sip0.en.ewetel.de> From: "v101" <sip:v101@sip0.en.ewetel.de>;tag=450c2a3b Call-ID: 7315210f6c7d7741NTA0NmE0M2ZkM2Y4MTg0YmU4NjQzNTllM2Q1YzliZGE. CSeq: 1 INVITE Allow: INVITE,CANCEL,ACK,BYE,INFO,OPTIONS,MESSAGE,SUBSCRIBE,NOTIFY,REFER Content-Type: application/sdp User-Agent: X-Lite release 1003l stamp 30942 P-Announcement-Code: 322 P-Hint: ringtone Content-Length: 300 SeasHashTable [SipProcessor[4]] - Inserting hashIdx:46338 label:965718283 into hashTable SipProcessor [SipProcessor[4]] - Transaction [z9hG4bK205b.8c060ce2.0] is new Transaction?? --> true EngineSipMapper [SipProcessor[4]] - Request from 212.6.123.118 mapped to: ConvergedEngine[Standalone].StandardHttpHost[212.6.123.118] ConvergedEngineValve [SipProcessor[4]] - Invoking Host:212.6.123.118 ConvergedHostValve [SipProcessor[4]] - SipSession not found for requestINVITE, 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 Record-Route: <sip:212.6.123.118;lr=on;ftag=450c2a3b> Via: SIP/2.0/UDP 212.6.123.118;branch=z9hG4bK205b.8c060ce2.0,SIP/2.0/UDP 212.6.95.198:25952;branch=z9hG4bK-d87543-ed108070b87aca0e-1--d87543-;rport=25952 Max-Forwards: 69 To: "Bruening" <sip:v111@sip0.en.ewetel.de> From: "v101" <sip:v101@sip0.en.ewetel.de>;tag=450c2a3b Call-ID: 7315210f6c7d7741NTA0NmE0M2ZkM2Y4MTg0YmU4NjQzNTllM2Q1YzliZGE. CSeq: 1 INVITE Content-Length: 0 LingerList [SipProcessor[4]] - scheduled transaction z9hG4bK205b.8c060ce2.0 for deletion ExpressMessageChannel [SipProcessor[4]] - sending with spiral=false SipConnector [SipProcessor[4]] - recycle: Recycling processor SipProcessor[4] ActionRouter$ActionSender [ActionSender] - Writing 503 bytes into Action Socket SeasMessageEvent [SeasRouter] - New message received, from 212.6.123.118:5060 to 212.6.123.118:5080 over UDP SeasTransactionEvent [SeasRouter] - hash_index = 46338,label=965718283 flags=0 ExpressMessageChannel [ExpressMChannel[1]] - peerAddress = 212.6.123.118/5060 ExpressMessageChannel [ExpressMChannel[1]] - About to process ACK sip:v111@212.6.123.118:5080;rinstance=fe71c43d95b8211a SIP/2.0 ExpressMessageChannel [ExpressMChannel[1]] - Done processing ACK sip:v111@212.6.123.118:5080;rinstance=fe71c43d95b8211a SIP/2.0 ExpressMessageChannel [ExpressMChannel[0]] - Done processing INVITE sip:v111@212.6.123.118:5080;rinstance=fe71c43d95b8211a SIP/2.0 regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGU/w94tZeNddg3dwRAhs+AKCpq2ZrAwCatPUY3BCUJR/ZJYwuzQCfUEOd MuNZ3C48K5Ypzun/27JG6TI= =5nOc -----END PGP SIGNATURE-----

1 0

[Users] WeSIP: RingTone
by Helmut Kuper 23 May '07

23 May '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I play around with WeSIP's ringtone app. I try to get ringtone running in this constellation: caller -> openser sip-proxy -> openser+wesip -> -> target sip-proxy -> callee openser+wesip is loaded with both c2c and ringtone app. When openser sip-proxy relay callers INVITE to wesip, wesip response with "SIP/2.0 481 Call Leg/Transaction Does Not Exist". I dunno wheter this is an answere from ringtone app or from wesip itself. I also dunno, whether I have to config ringtone with environment specific things, like sip port, ip address or something like that or whether my constallation from above is correct. Maybe someone can help me out here :) Here is WeSIP's log, after receiving caller's INVITE: Starting WESIP Application Server VozTelecom... hk@vm-en-test0:~/wesip/wesip$ HttpConnector [main] - The HttpConnector [all:8080] has been created ExpressConnector [main] - SEAS module successfully connected at 212.6.123.118:5081 ExpressConnector [main] - trying to receive 2 socket descriptions from SEAS ExpressConnector [main] - SER socket:[212.6.123.118:5080/udp] ExpressConnector [main] - SER socket:[212.6.123.118:5080/udp] ExpressConnector [main] - SER socket:[212.6.123.118:5080/tcp] ExpressConnector [main] - SER socket:[212.6.123.118:5080/tcp] SeasRouter [SeasRouter] - SeasDispatcher Thread starts running SeasRouter [SeasRouter] - SeasDispatcher Thread starts running SeasHashTable [main] - Creating Seas Hash Table, with tablePower=17 and total entries:131072 SeasHashTable [main] - Creating Seas Hash Table, with tablePower=17 and total entries:131072 ExpressMessageChannel [ExpressMessageProcessor] - setting spiral=false ExpressMessageChannel [ExpressMessageProcessor] - setting spiral=false SipConnector [main] - The SipConnector [212.6.123.118:5080] has been created SipProcessor [main] - Starting background thread ExpressMessageChannel [ExpressMessageProcessor] - setting spiral=false ExpressMessageChannel [ExpressMessageProcessor] - setting spiral=false ExpressMessageChannel [ExpressMessageProcessor] - setting spiral=false ExpressMessageChannel [ExpressMessageProcessor] - setting spiral=false ExpressMessageChannel [ExpressMessageProcessor] - setting spiral=false ExpressMessageChannel [ExpressMessageProcessor] - setting spiral=false ExpressMessageChannel [ExpressMessageProcessor] - setting spiral=false ExpressMessageChannel [ExpressMessageProcessor] - setting spiral=false SipProcessor [main] - Background thread has been started SipConnector [main] - recycle: Recycling processor SipProcessor[0] SipProcessor [main] - Starting background thread SipProcessor [main] - Background thread has been started SipConnector [main] - recycle: Recycling processor SipProcessor[1] SipProcessor [main] - Starting background thread SipProcessor [main] - Background thread has been started SipConnector [main] - recycle: Recycling processor SipProcessor[2] SipProcessor [main] - Starting background thread SipProcessor [main] - Background thread has been started SipConnector [main] - recycle: Recycling processor SipProcessor[3] SipProcessor [main] - Starting background thread SipProcessor [main] - Background thread has been started SipConnector [main] - recycle: Recycling processor SipProcessor[4] ConvergedEngine [main] - 212.6.123.118_5080 ConvergedEngine [main] - 212.6.123.118_5080 ConvergedEngine [main] - 212.6.123.118_5080 ConvergedEngine [main] - 212.6.123.118_5080 StatisticsServlet [main] - Wrapper setted in StatisticsServlet!!! ConvergedHostDeployer [main] - Installing web application at context path /click2call-08_11_06 from URL file:/home/hk/wesip/wesip/wesipapps/click2call-08_11_06 StandardSipManager [main] - Starting StandardSipManager [main] - Force random number initialization starting StandardSipManager [main] - Force random number initialization completed StandardSipManager [main] - Start: Loading persisted sessions StandardSipManager [main] - Loading persisted sessions from SIPSESSIONS.ser StandardSipManager [main] - No persisted data file found StandardAppSessionManager [main] - Starting StandardAppSessionManager [main] - Force random number initialization starting StandardAppSessionManager [main] - Force random number initialization completed AxisServletBase [main] - In AxisServletBase init AxisServletBase [main] - Enter: getEngine() ProjectResourceBundle [main] - getBundle(org.apache.axis,org.apache.axis.i18n,resource,null,...) ProjectResourceBundle$Context [main] - loadBundle: Ignoring MissingResourceException: Can't find bundle for base name org.apache.axis.resource, locale en_US ProjectResourceBundle [main] - Created org.apache.axis.i18n.resource, linked to parent null ProjectResourceBundle [main] - getBundle(org.apache.axis,org.apache.axis.utils,resource,null,...) ProjectResourceBundle$Context [main] - loadBundle: Ignoring MissingResourceException: Can't find bundle for base name org.apache.axis.utils.resource, locale en_US ProjectResourceBundle$Context [main] - loadBundle: Ignoring MissingResourceException: Can't find bundle for base name org.apache.axis.resource, locale en_US ProjectResourceBundle [main] - Root package not found, cross link to org.apache.axis.i18n.resource ProjectResourceBundle [main] - Root package not found, cross link to org.apache.axis.i18n.resource ProjectResourceBundle [main] - org.apache.axis.i18n.resource::handleGetObject(engineFactory) EngineConfigurationFactoryFinder$1 [main] - Got EngineFactory: org.apache.axis.configuration.EngineConfigurationFactoryServlet DefaultAxisServerFactory [main] - Enter: DefaultAxisServerFactory::getServer DefaultAxisServerFactory [main] - Enter: DefaultAxisServerFactory::getEngineConfiguration DefaultAxisServerFactory [main] - Exit: DefaultAxisServerFactory::getEngineConfiguration AxisEngine [main] - Enter: AxisEngine::init ProjectResourceBundle [main] - org.apache.axis.i18n.resource::handleGetObject(attachEnabled) JavaUtils [main] - Attachment support is enabled? false ProjectResourceBundle [main] - org.apache.axis.i18n.resource::handleGetObject(attachDisabled) JavaUtils [main] - Unable to find required classes (javax.activation.DataHandler and javax.mail.internet.MimeMultipart). Attachment support is disabled. ProjectResourceBundle [main] - org.apache.axis.i18n.resource::handleGetObject(oddDigits00) ProjectResourceBundle [main] - org.apache.axis.i18n.resource::handleGetObject(badChars01) AxisEngine [main] - Exit: AxisEngine::init DefaultAxisServerFactory [main] - Exit: DefaultAxisServerFactory::getServer AxisServletBase [main] - Exit: getEngine() AxisServlet [main] - In servlet init AxisServletBase [main] - Enter: getEngine() AxisServletBase [main] - Exit: getEngine() AxisServletBase [main] - Enter: getEngine() AxisServletBase [main] - Exit: getEngine() AxisServletBase [main] - Enter: getEngine() AxisServletBase [main] - Exit: getEngine() AxisServletBase [main] - Enter: getEngine() AxisServletBase [main] - Exit: getEngine() AxisServletBase [main] - Enter: getEngine() AxisServletBase [main] - Exit: getEngine() AxisServletBase [main] - Enter: getEngine() AxisServletBase [main] - Exit: getEngine() AxisServletBase [main] - Enter: getEngine() AxisServletBase [main] - Exit: getEngine() Registrar [main] - Registering org.apache.axis.management.jmx.ServiceAdministrator@10ffb38 as axis:type=server Registry [main] - Managed= axis:type=server Registry [main] - Looking for descriptor Registry [main] - Introspecting Registry [main] - load class org.apache.axis.management.jmx.ServiceAdministrator MbeansDescriptorsIntrospectionSource [main] - Introspected attribute version public java.lang.String org.apache.axis.management.jmx.ServiceAdministrator.getVersion() null MbeansDescriptorsIntrospectionSource [main] - Setting name: ServiceAdministrator BaseModelMBean [main] - default constructor Registry [main] - Creating MBeanServer9 BaseModelMBean [main] - preRegister org.apache.axis.management.jmx.ServiceAdministrator@10ffb38 axis:type=server Registrar$ModelerBinding [main] - Registered axis:type=server in ServiceAdministrator Registrar [main] - Registering org.apache.axis.management.jmx.DeploymentAdministrator@b40ec4 as axis:type=deploy Registry [main] - Managed= axis:type=deploy Registry [main] - Looking for descriptor Registry [main] - Introspecting Registry [main] - load class org.apache.axis.management.jmx.DeploymentAdministrator MbeansDescriptorsIntrospectionSource [main] - Setting name: DeploymentAdministrator BaseModelMBean [main] - default constructor BaseModelMBean [main] - preRegister org.apache.axis.management.jmx.DeploymentAdministrator@b40ec4 axis:type=deploy Registrar$ModelerBinding [main] - Registered axis:type=deploy in DeploymentAdministrator Registrar [main] - Registering org.apache.axis.management.jmx.DeploymentQuery@edf730 as axis:type=query Registry [main] - Managed= axis:type=query Registry [main] - Looking for descriptor Registry [main] - Introspecting Registry [main] - load class org.apache.axis.management.jmx.DeploymentQuery MbeansDescriptorsIntrospectionSource [main] - Setting name: DeploymentQuery BaseModelMBean [main] - default constructor BaseModelMBean [main] - preRegister org.apache.axis.management.jmx.DeploymentQuery@edf730 axis:type=query Registrar$ModelerBinding [main] - Registered axis:type=query in DeploymentQuery AxisServletBase [main] - In AxisServletBase init AxisServletBase [main] - Enter: getEngine() ProjectResourceBundle [main] - org.apache.axis.i18n.resource::handleGetObject(engineFactory) EngineConfigurationFactoryFinder$1 [main] - Got EngineFactory: org.apache.axis.configuration.EngineConfigurationFactoryServlet DefaultAxisServerFactory [main] - Enter: DefaultAxisServerFactory::getServer DefaultAxisServerFactory [main] - Enter: DefaultAxisServerFactory::getEngineConfiguration DefaultAxisServerFactory [main] - Exit: DefaultAxisServerFactory::getEngineConfiguration AxisEngine [main] - Enter: AxisEngine::init AxisEngine [main] - Exit: AxisEngine::init DefaultAxisServerFactory [main] - Exit: DefaultAxisServerFactory::getServer AxisServletBase [main] - Exit: getEngine() ConvergedContext [main] - ConvergedContext [Click2Call SipServlet] has been started ConvergedHostDeployer [main] - Installing web application at context path /ringtone-22_02_07 from URL file:/home/hk/wesip/wesip/wesipapps/ringtone-22_02_07 StandardSipManager [main] - Starting StandardSipManager [main] - Force random number initialization starting StandardSipManager [main] - Force random number initialization completed StandardSipManager [main] - Start: Loading persisted sessions StandardSipManager [main] - Loading persisted sessions from SIPSESSIONS.ser StandardSipManager [main] - No persisted data file found StandardAppSessionManager [main] - Starting StandardAppSessionManager [main] - Force random number initialization starting StandardAppSessionManager [main] - Force random number initialization completed ConvergedContext [main] - ConvergedContext [Ringtone] has been started ConvergedHostDeployer [main] - Installing web application at context path from URL file:/home/hk/wesip/wesip/wesipapps/ROOT StandardSipManager [main] - Starting StandardSipManager [main] - Force random number initialization starting StandardSipManager [main] - Force random number initialization completed StandardSipManager [main] - Start: Loading persisted sessions StandardSipManager [main] - Loading persisted sessions from SIPSESSIONS.ser StandardSipManager [main] - No persisted data file found StandardAppSessionManager [main] - Starting StandardAppSessionManager [main] - Force random number initialization starting StandardAppSessionManager [main] - Force random number initialization completed processJars: cannot find /WEB-INF/lib ConvergedContext [main] - ConvergedContext [Welcome] has been started StandardService [main] - > Default Wesip Application ==> http://localhost:8080/ > Manager Wesip Application ==> http://localhost:8080/manager/html? StandardService [main] - > In order to access the manager application use default credentials: - username: wesip - password: wesip > This configuration is defined in wesip-users.xml inside conf folder: - file: ./conf/wesip-users.xml SeasMessageEvent [SeasRouter] - New message received, from 212.6.123.118:5060 to 212.6.123.118:5080 over UDP SeasTransactionEvent [SeasRouter] - hash_index = 46338,label=965718283 flags=0 ExpressMessageChannel [ExpressMChannel[0]] - peerAddress = 212.6.123.118/5060 ExpressMessageChannel [ExpressMChannel[0]] - About to process INVITE sip:v111@212.6.123.118:5080;rinstance=fe71c43d95b8211a SIP/2.0 SipConnector [ExpressMChannel[0]] - createProcessor: Reusing existing processor SipProcessor [ExpressMChannel[0]] - An incoming message is being assigned SipProcessor [SipProcessor[4]] - <<<<<<<<< Request Received <<<<<<<<< INVITE sip:v111@212.6.123.118:5080;rinstance=fe71c43d95b8211a SIP/2.0 Record-Route: <sip:212.6.123.118;lr=on;ftag=450c2a3b> Via: SIP/2.0/UDP 212.6.123.118;branch=z9hG4bK205b.8c060ce2.0,SIP/2.0/UDP 212.6.95.198:25952;branch=z9hG4bK-d87543-ed108070b87aca0e-1--d87543-;rport=25952 Max-Forwards: 69 Contact: <sip:v101@212.6.95.198:25952> To: "Bruening" <sip:v111@sip0.en.ewetel.de> From: "v101" <sip:v101@sip0.en.ewetel.de>;tag=450c2a3b Call-ID: 7315210f6c7d7741NTA0NmE0M2ZkM2Y4MTg0YmU4NjQzNTllM2Q1YzliZGE. CSeq: 1 INVITE Allow: INVITE,CANCEL,ACK,BYE,INFO,OPTIONS,MESSAGE,SUBSCRIBE,NOTIFY,REFER Content-Type: application/sdp User-Agent: X-Lite release 1003l stamp 30942 P-Announcement-Code: 322 P-Hint: ringtone Content-Length: 300 SeasHashTable [SipProcessor[4]] - Inserting hashIdx:46338 label:965718283 into hashTable SipProcessor [SipProcessor[4]] - Transaction [z9hG4bK205b.8c060ce2.0] is new Transaction?? --> true EngineSipMapper [SipProcessor[4]] - Request from 212.6.123.118 mapped to: ConvergedEngine[Standalone].StandardHttpHost[212.6.123.118] ConvergedEngineValve [SipProcessor[4]] - Invoking Host:212.6.123.118 ConvergedHostValve [SipProcessor[4]] - SipSession not found for requestINVITE, 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 Record-Route: <sip:212.6.123.118;lr=on;ftag=450c2a3b> Via: SIP/2.0/UDP 212.6.123.118;branch=z9hG4bK205b.8c060ce2.0,SIP/2.0/UDP 212.6.95.198:25952;branch=z9hG4bK-d87543-ed108070b87aca0e-1--d87543-;rport=25952 Max-Forwards: 69 To: "Bruening" <sip:v111@sip0.en.ewetel.de> From: "v101" <sip:v101@sip0.en.ewetel.de>;tag=450c2a3b Call-ID: 7315210f6c7d7741NTA0NmE0M2ZkM2Y4MTg0YmU4NjQzNTllM2Q1YzliZGE. CSeq: 1 INVITE Content-Length: 0 LingerList [SipProcessor[4]] - scheduled transaction z9hG4bK205b.8c060ce2.0 for deletion ExpressMessageChannel [SipProcessor[4]] - sending with spiral=false SipConnector [SipProcessor[4]] - recycle: Recycling processor SipProcessor[4] ActionRouter$ActionSender [ActionSender] - Writing 503 bytes into Action Socket SeasMessageEvent [SeasRouter] - New message received, from 212.6.123.118:5060 to 212.6.123.118:5080 over UDP SeasTransactionEvent [SeasRouter] - hash_index = 46338,label=965718283 flags=0 ExpressMessageChannel [ExpressMChannel[1]] - peerAddress = 212.6.123.118/5060 ExpressMessageChannel [ExpressMChannel[1]] - About to process ACK sip:v111@212.6.123.118:5080;rinstance=fe71c43d95b8211a SIP/2.0 ExpressMessageChannel [ExpressMChannel[1]] - Done processing ACK sip:v111@212.6.123.118:5080;rinstance=fe71c43d95b8211a SIP/2.0 ExpressMessageChannel [ExpressMChannel[0]] - Done processing INVITE sip:v111@212.6.123.118:5080;rinstance=fe71c43d95b8211a SIP/2.0 regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGU/w94tZeNddg3dwRAhs+AKCpq2ZrAwCatPUY3BCUJR/ZJYwuzQCfUEOd MuNZ3C48K5Ypzun/27JG6TI= =5nOc -----END PGP SIGNATURE-----

1 0

[Users] With SER Subscription state:- pending and Publish data is not coming with Notify
by Ambed Kumar 23 May '07

23 May '07

Hi Friends, In SER I am subscribing and i am getting 202 Ok also notify but the publish method data is is not coming to me with notify although after sending publish i am getting 200 Ok. Can any one suggest me regarding this. Thanks and Regards, Ambed

2 1

Re: [Users] Newbie question about FIFO in Openser 1.2.x
by Andreas Granig 23 May '07

23 May '07

Forgot to CC to the list. > Hi, > >> I never worked with FIFO, and I need exec one command in fifo, the >> trusted_reload ! > > OSER_FIFO has to be "/tmp/openser_fifo" in your openserctlrc. Then just > try "openserctl fifo trusted_reload", and you should get a "200 OK" if > successful. > > Cheers, > Andreas >

1 0

[Users] Newbie question about FIFO in Openser 1.2.x
by Thiago Maluf 22 May '07

22 May '07

Hi list, I never worked with FIFO, and I need exec one command in fifo, the trusted_reload ! ********** openser.cfg ********************** alias="insta.voip.nce.ufrj.br" mpath="/lib/openser/modules" children=8 tcp_children=4 debug=3 fork=yes log_facility=LOG_DAEMON log_stderror=yes group="openser" user="openser" sock_group="openser" sock_user="openser" disable_tcp=no mhomed=no server_signature=yes sock_mode=0600 unix_sock="/tmp/openser.sock" unix_sock_children=1 reply_to_via=no sip_warning=no check_via=no dns=no rev_dns=no disable_core_dump=no dns_try_ipv6=yes dns_use_search_list=yes ....... loadmodule "mi_fifo.so" modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") modparam("mi_fifo", "fifo_mode", 0660) modparam("mi_fifo", "fifo_group", "openser") modparam("mi_fifo", "fifo_user", "openser") modparam("mi_fifo", "reply_dir", "/tmp/") modparam("mi_fifo", "reply_indent", "\t") ...... **************** openserctl ***************** # $Id: openserctlrc 1827 2007-03-12 15:22:53Z bogdan_iancu $ # # openser control tool resource file # # here you can set variables used in the openserctl ## your SIP domain SIP_DOMAIN=insta.voip.nce.ufrj.br ## database type: MYSQL or PGSQL, by defaulte none is loaded DBENGINE=PGSQL ## database host DBHOST=146.164.247.209 ## database name DBNAME=openser ## database read/write user DBRWUSER=openser ## database read only user # DBROUSER=openserro ## password for database read only user DBROPW=openserro ## database super user DBROOTUSER="postgres" ## type of aliases used: DB - database aliases; UL - usrloc aliases ## - default: none ALIASES_TYPE="DB" ## control engine: FIFO or UNIXSOCK ## - default FIFO CTLENGINE="UNIXSOCK" ## path to FIFO file OSER_FIFO="/tmp/" ## check ACL names; default on (1); off (0) VERIFY_ACL=1 ## ACL names - if VERIFY_ACL is set, only the ACL names from below list ## are accepted # ACL_GROUPS="local ld int voicemail free-pstn" ## presence of serweb tables - default "no" # HAS_SERWEB="yes" ## verbose - debug purposes - default '0' VERBOSE=1 ## do (1) or don't (0) store plaintext passwords ## in the subscriber table - default '1' # STORE_PLAINTEXT_PW=0 *************************** My doubt is what my conf is correct and what is the command that I use in the Konsole!? Thanks in advanced for your dedication to someone that read to here. Thiago Maluf. ---------------------------------------------------------------- THIAGO MALUF RESENDE Consultor Voip e Programador WEB (Voip Developer and Web Developer) Tel: +55 21 86042100 e-mail: malufrj(a)gmail.com

1 0

[Users] BYE relay issue
by kaiser 22 May '07

22 May '07

Dear sir, We install 3 openser 1.1.0 in public internet, and we find a BYE relay issue to caller SIP UA. UA(a)-------Proxy1 --------Proxy2 ----Proxy 3 ---------UA(b) We make call from UA(a) to UA (b), if UA(b) send Bye first, UA(a) can not get Bye. The packet is from Proxy2 to Proxy1 as following BYE sip:1.2.3.4:5060;nat=yes;ftag=ooqjvbi7ql;lr=on SIP/2.0 Record-Route: <sip:5.6.7.8;lr=on;ftag=as1b41fbeb> Record-Route: <sip:5.6.7.8;lr=on;ftag=as1b41fbeb> Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKc74c.bd89ce13.0 Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKc74c.ad89ce13.0 Via: SIP/2.0/UDP 100.200.100.200;branch=z9hG4bKc74c.321ef5e3.0 Via: SIP/2.0/UDP 100.200.100.199:5060;branch=z9hG4bK75e8d6f3;rport=5060 From: <sip:0908900000@sip.edu.tw;user=phone>;tag=as1b41fbeb To: "0909802020 kl" <sip:0909802020@sip.edu.tw>;tag=ooqjvbi7ql Call-ID: 3c275da915f9-iylvmp2wjuh0@snom360-00041323058B CSeq: 102 BYE User-Agent: Gentrice_IPPBX Max-Forwards: 67 Content-Length: 0 Then bye is looping in proxy1. BYE sip:1.2.3.4:5060;nat=yes;ftag=ooqjvbi7ql;lr=on SIP/2.0 Record-Route: <sip:5.6.7.8;lr=on;ftag=as1b41fbeb> Record-Route: <sip:5.6.7.8;lr=on;ftag=as1b41fbeb> Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bKc74c.1714a4e7.0 Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKc74c.bd89ce13.0 Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKc74c.ad89ce13.0 Via: SIP/2.0/UDP 100.200.100.200;branch=z9hG4bKc74c.321ef5e3.0 Via: SIP/2.0/UDP 100.200.100.199:5060;branch=z9hG4bK75e8d6f3;rport=5060 From: <sip:0908900000@sip.edu.tw;user=phone>;tag=as1b41fbeb To: "0909802020 kl" <sip:0909802020@sip.edu.tw>;tag=ooqjvbi7ql Call-ID: 3c275da915f9-iylvmp2wjuh0@snom360-00041323058B CSeq: 102 BYE User-Agent: Gentrice_IPPBX Max-Forwards: 66 Content-Length: 0 We used to use default script for proxy1, but the same... And we do record_route, loose_route relay as well. Anyone know what happen? best regards Thrli

3 7

[Users] WeSIP: Click2Call Linefeed
by Helmut Kuper 22 May '07

22 May '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello WeSIPer, I found a little but annoying Error in WeSIP's C2C application. I send back a 483 reply on a C2C's first caller-INVITE to WeSIP via openser's sl-send_reply(). I expected that C2C terminates the call function, cause C2C has an error handling method for 4xx replies, but nothing happens. openser as wesip's frontend received definately the reply. By chance I found, that the reason for this was my reply message. This doesn't work with WeSIP's C2C application: sl_send_reply("483", "My message\n"); But this works: sl_send_reply("483", "My message"); It seems, that C2C's or WeSIPs reply parser has some problems with linefeeds at the end of reply's message text. I use openser 1.2.0, WeSIP 0.1.4 and click2call-08_11_06 regards Helmut PS: Is there already a fix for WeSIP to reconnect to openser's seas socket, when openser restarts? -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGUV2k4tZeNddg3dwRAsPcAKCNqGE+XTK7BvW69x4YT498Bg0dkgCbB3eL 6Me/ZKYaqxfo/qDGB8M3lg4= =jTel -----END PGP SIGNATURE-----

2 1

[Users] loose_route problem with Fritz
by Edoardo Serra 22 May '07

22 May '07

Hi guys, I have OpensER 1.1.0 in fornt of 2 Asterisks servers. My OpenSER act as load balancer for outbound calls and for registrar server for inbound calls coming from Asterisks PSTN GW to Users Some users, using AVM Fritz Fon Boxes cannot register on the proxy, they get a "483 Too Many Hops". They say it happens only if they configure the Fritx using FQDN of our proxy, if they put its IP addess in it they are having no problems... (it sounds very strange to me but I captured some trafic and had confirmation of what they're saying) Here is a SIP dialog with the error message Request-Line: REGISTER sip:sip.advancemedia.it SIP/2.0 Via: SIP/2.0/UDP 192.168.1.121:5060;branch=z9hG4bKC0F2D683E6BE8AC5F1A4A260BE120 Route: <sip:sip.advancemedia.it;lr> From: <sip:cirifax@sip.advancemedia.it>;tag=1898319928 To: <sip:cirifax@sip.advancemedia.it> Call-ID: 9ADCCD22187BDEC93B5953DD93E0F(a)192.168.1.121 CSeq: 1 REGISTER Max-Forwards: 70 User-Agent: AVM FRITZ!Box Fon ata 1020 11.04.01 (Jan 25 2006) Supported: 100rel, replaces Allow-Events: telephone-event, refer Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, UPDATE, PRACK, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE Accept: application/sdp, multipart/mixed Accept-Encoding: identity Content-Length: 0 Status-Line: SIP/2.0 483 Too Many Hops Message Header Via: SIP/2.0/UDP 192.168.1.121:5060;received=80.67.115.96;branch=z9hG4bKC0F2D683E6BE8AC5F1A4A260BE120 From: <sip:cirifax@sip.advancemedia.it>;tag=1898319928 To: <sip:cirifax@sip.advancemedia.it>;tag=b36cb132c8240044969e91db40b97d00.d5d1 Call-ID: 9ADCCD22187BDEC93B5953DD93E0F(a)192.168.1.121 CSeq: 1 REGISTER Server: OpenSer (1.1.0-notls (i386/linux)) Content-Length: 0 Warning: 392 213.92.79.131:5060 "Noisy feedback tells: pid=13326 req_src_ip=213.92.79.131 req_src_port=5060 in_uri=sip:sip.advancemedia.it out_uri=sip:sip.advancemedia.it via_cnt==71" It doesn't happen if the user put the ip of the proxy server in the configuration of the Fritz I guess it's a problem with loose routing but I cannot understand if it's a buggy implementation of SIP in the fritz or my misconfiguration Here is my openser.cfg, tnx in advance for help. Edoardo # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ # ----------- global configuration parameters ------------------------ check_via=yes # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- loadmodule "/usr/lib/openser/modules/sl.so" [...] modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "mysql://user:pass@192.168.252.5/openser") modparam("usrloc", "timer_interval", 120) modparam("auth_db", "calculate_ha1", 0) modparam("auth_db", "db_url", "mysql://user:pass@192.168.252.5/db") modparam("uri_db", "db_url", "mysql://user:pass@192.168.252.5/openser") modparam("rr", "enable_full_lr", 1) modparam("registrar", "nat_flag", 6) modparam("registrar", "max_expires", 3600) modparam("registrar", "min_expires", 60) modparam("registrar", "append_branches", 0) modparam("registrar", "desc_time_order", 1) modparam("nathelper", "natping_interval", 20) # Ping interval 20 s modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT modparam("dispatcher", "force_dst", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); exit; }; if ( (method=="OPTIONS") || (method=="SUBSCRIBE") || (method=="NOTIFY") ) { sl_send_reply("405", "Method Not Allowed"); exit; } # loose-route processing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); t_relay(); exit; }; if (!method=="REGISTER") { record_route(); }; if ((src_ip==ip.of.asterisk.1) || (src_ip==ip.of.asterisk.1)) { if (!lookup("location")) { sl_send_reply("486", "Busy here"); exit; }; t_on_reply("1"); if (!t_relay()) { sl_reply_error(); }; exit; }; if (nat_uac_test("3")) { if ((method=="REGISTER") || (method=="INVITE") || (method=="OPTIONS")) { fix_nated_contact(); force_rport(); setflag(6); # Mark as NATed } } if (method=="REGISTER") { if (!proxy_authorize("exorsa", "openser_view")) { proxy_challenge("exorsa", "0"); exit; } if (!check_to()) { sl_send_reply("403", "Digest username and URI username do NOT match! Stay away!"); exit; } save("location"); exit; }; if (method=="INVITE") { if (!proxy_authorize("exorsa", "openser_view")) { proxy_challenge("exorsa", "0"); exit; } if (!check_from()) { sl_send_reply("403", "Digest username and URI username do NOT match! Stay away!"); exit; } } if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); exit; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # !! Nathelper if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && !search("^Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); exit; }; # NAT processing of replies; apply to all transactions (for example, # re-INVITEs from public to private UA are hard to identify as # NATed at the moment of request processing); look at replies t_on_reply("1"); if ((src_ip!=ip.of.asterisk.1) && (src_ip!=ip.of.asterisk.2)) { ds_select_dst("1", "0"); } if (!t_relay()) { sl_reply_error(); }; } # !! Nathelper onreply_route[1] { # NATed transaction ? if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { fix_nated_contact(); # otherwise, is it a transaction behind a NAT and we did not # know at time of request processing ? (RFC1918 contacts) } else if (nat_uac_test("1")) { fix_nated_contact(); }; }

2 2

[Users] b2bua
by Kro Kamil 22 May '07

22 May '07

Hi, I need a b2bua to use with openser. I also don't want to use radius. what options do I have? thanks in advance. _________________________________________________________________ Live.com'u deneyin: çevrimiçi dünyanýzý bir araya getirin; haberler, spor, hava durumu ve çok daha fazlasý. http://www.live.com/getstarted

4 3

[Users] Presence, Presence_xml & NOTIFY
by mika.saari＠wipsl.com 22 May '07

22 May '07

Hi, At the moment I am using SVN version of OpenSER (2:2253M) also I have been using the 1.2 release. The problem has been the NOTIFY message which has been dropping time to time, or like in current svn release, I do not get the NOTIFY message at all anymore. I think my openser.cfg is having something wrong, but haven't been able to figure out what is the reason. I do use XCAP_Lite 1.0.0 with presence & presence_xml modules. The force_active is set to "0", because I want to have reactive buddy authorization. The XCAP side works nicely, also watchers and active_watchers are populated by OpenSER correctly. But once the SUBSCRIBE is received from client, no NOTIFY is sent to any clients which are active in watchers table. After reading the log file I found this message from it: 0(4415) PRESENCE: publ_notify: Could not get subs_dialog from database Thank you for tips, -Mika Saari

2 3

[Users] Distribution Calls with the same destination number to different servers
by Andreas Brodmann 22 May '07

22 May '07

Hi all, I am working on the following scenario: Our carrier sends all calls (SIP protocol) to my openser, which forwards the calls depending on their destination number to four asterisk servers (for load sharing reason - we're speaking of a roof of 300 concurrent calls). I have one number which leads to a conferencence (one speaker, multiple listeners), whoose number of callers will exceed the number of calls a single asterisk will hold. Therefore I want the openser to distribute the calls to this number to all four asterisk servers (to share the load). My question is: how do I configure this on openser, as I have to send all requests from the same SIP session to the same asterisk? Can anyone give me a hint on this? Thanks in advance. Andreas

1 0

[Serusers] Unable to download SER RC1
by Andrey Kuprianov 22 May '07

22 May '07

Hi, Is something wrong with siprouter.onsip.org FTP site? I've tried Firefox, IE7 and even wget, but still cant download ftp://siprouter.onsip.org/pub/ser/ser-2.0.0-rc1_src.tar.gz. What's wrong? Andrey.

2 3

[Serusers] ser.cfg buildsystem
by KUMAR 22 May '07

22 May '07

Hi all, I am having some problem with the configuration file buildsystem. Whatever changes I make, it always creates only the basic hello world configuration file only. I am using the latest package. It also gives the following message **** WARNING! Only helloworld is implemented!!!! How do I include other feature packages like NAT and authentication. Here's what i've done This configure script will generate a new local configuration to ./local/default/config.m4. ====================== Press ENTER to accept default values. SER Server IP address: IP address to listen on (202.52.247.221): 192.168.2.2 Port to listen on (5060): Public IP address of server (ENTER if same as listen IP): Domain name for this SER installation (i.e. mydomain.com, def. mytestser.org) : MYDOMAIN SER installation path (/usr/local, i.e. modules in /usr/local/lib/ser/modules): Usage: dbtype://readwriteuser:password@host/databasename Database URL (mysql://ser:heslo@localhost/ser): You may choose to turn on logging as found in the SER - Getting Started guide. This logging can be turned on and off in a live environment, and we recommend everybody to include the logging except in high-load environments. (You can also later disable it from your config.m4) Turn on logging (yes): How much log possibilities to include (low/medium/high, default high): You can base your configuration file on one of the feature packages available. Currently, the supported packages are (gettingstarted sems) gettingstarted : A set of basic configurations found in most setups sems : A configuration file that enables SER as a front-end for SEMS You can also choose to include specific components within a feature package (components). And finally, you can choose to include no prepared features, just use your own code in ./local. Your choice of feature, choose only one, default is gettingstarted (none, components or gettingstarted sems) ? You have chosen the SER - Getting Started feature package. (See SER - Getting Started document) Which feature set do you want to base your configuration on: helloworld : basic configuration auth : authentication using mysql database nat : NAT traversal handling pstn : Functionality for PSTN integration callfwd : basic call features like call forwarding **** WARNING! Only helloworld is implemented!!!! Your choice (default helloworld, auth, nat, pstn, callfwd): nat You have chosen NAT traversal, do you want to use rtpproxy (iptel.org) or mediaproxy (ag-projects)? Your choice (default rtpproxy, mediaproxy): rtpproxy Copying site templates to ./local/default... The resulting ser.config doesn't include anything except the helloworld. Thank you in advance. Kumar

2 1

Re: RES: [Users] Presence Problem - Publish Offline
by Anca Vamanu 22 May '07

22 May '07

Hello, The proper way to send Publish messages is described in RFC 3903 (see section 4.1 for event refreshing event state publication). If you don't include a Sip-If-Match header it is not considered a refresh of the previous state and thus it does not rewrite the open state previously published. regards, Anca Hildebrando Trannin wrote: >Hi Anca, > >Thanks for your answer. When I say that OFFLINE doesn't works is that I send >a PUBLISH with the basic status CLOSED (after already sent others PUBLISH >with status OPENED) and OpenSER doesn't send NOTIFY message to inform >subscribers that that contact is now OFFLINE. According to your answer it >happens because I'm not sending PUBLISH's with Sip-If-Match. Is it right? If >no, how is the correct way to send PUBLISH? Just to complement, PUBLISH with >expires = 0 that do UNPUBLISH is working fine. > >Regards, > Hildebrando Trannin > >-----Mensagem original----- >De: Anca-Maria Vamanu [mailto:anca@voice-system.ro] >Enviada em: segunda-feira, 21 de maio de 2007 06:12 >Para: Hildebrando Trannin >Cc: users(a)openser.org >Assunto: Re: [Users] Presence Problem - Publish Offline > >Hello, > >What happens here is that the presence module receives a first Publish >from your aplication (that is a Publish with no Sip-If-Match header) >having the basic status closed. In this situation, it does not send >Notify to its watches to prevent descovering that the user is in fact >online by tracking packets. > >Hildebrando Trannin wrote: > > > >>Hi all, >> >>I was testing OpenSER presence functionality using eyebeam and a >>softphone developed by my team. The OFFLINE state with eyebeam worked >>perfectly, but with my softphone it didn't work. I set force_active = >>1. XML sent by PUBLISH and OpenSER log are below. Can anyone help me? >> >> >> >I don't understand what you mean with the Offline state not working. >Could you please give more details? > > > >>Thanks, Hildebrando >> >>*PIDF XML:* >> >>* * >> >><?xml xmlns="urn:ietf:params:xml:ns:pidf" >> >>entity="sip:hildebrando@voip.maritaca"> >> >><presence> >> >><tuple id=""> >> >><status> >> >><basic>closed</basic> >> >></status> >> >><note>offline</note> >> >></tuple> >> >></presence> >> >>The basic difference between my application and eyebeam is that >>eyebeam uses *RPID XML *and I use *PIDF XML*. >> >>*OPENSER LOG:* >> >>* * >> >>3(26039) PRESENCE: update_presentity: The presentity status is >>offline; do not send notify >> >>------------------------------------------------------------------------ >> >> >> >> >> >regards, > >Anca Vamanu > > > >>_______________________________________________ >>Users mailing list >>Users(a)openser.org >>http://openser.org/cgi-bin/mailman/listinfo/users >> >> >> >> > > >

1 0

Re: [Serusers] Does SER's MYSQL work in SUSE Linux?
by Cyprian Clement 22 May '07

22 May '07

Hi Ram, Thanks for the note. I thought there may be some issues with SUSE regarding mysql since all the talk is about other Linux flavours. I will give it a try and see. Thanks, Cyp. ==== ----- Original Message ---- From: ram <talk2ram(a)gmail.com> To: Cyprian Clement <cyprian_clement(a)yahoo.co.uk> Cc: "serusers(a)iptel.org" <serusers(a)iptel.org> Sent: Sunday, 20 May, 2007 5:06:55 PM Subject: Re: [Serusers] Does SER's MYSQL work in SUSE Linux? On 5/18/07, Cyprian Clement <cyprian_clement(a)yahoo.co.uk> wrote: Hello Community, Does anyone know whether it's possible to install the current recommended MYSQL on SUSE Linux. I have a Machine with a SUSE installed on it. I have successfully installed SER 0.9.6 but now I would like to have MYSQL for authentication support. Does the SER's recommended MYSQL work with SUSE?. I only see the talk about Redhat and Debian so unsure if it works with SUSE. Many thanks for in advance. Hi any linux with mysql and ser should support try using mysql authentication any problems post here ram Cyp. ---------- Yahoo! Answers - Got a question? Someone out there knows the answer. Try it now. _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers ___________________________________________________________ What kind of emailer are you? Find out today - get a free analysis of your email personality. Take the quiz at the Yahoo! Mail Championship. http://uk.rd.yahoo.com/evt=44106/*http://mail.yahoo.net/uk

1 0

Re: [Serusers] Problem running presence modules - SER won't start
by Vaclav Kubart 22 May '07

22 May '07

There was problem that sudo didn't propagate LD_LIBRARY_PATH settings or it did, but linker didn't care or something like that. Does your ser binary has setuid/setgid bit set? In this case linker may ignore the LD_LIBRARY_PATH settings too. You can try to add the path /usr/local/lib/ser to your /etc/ld.so.conf and run ldconfig. This might help... Vaclav On Mon, May 21, 2007 at 12:32:15PM -0700, Mahatma Sarasvati wrote: > No, I'm not using sudo. Does it matter? > > M > > ----- Original Message ---- > From: Vaclav Kubart <vaclav.kubart(a)iptel.org> > To: Mahatma Sarasvati <gnoahhb(a)yahoo.com> > Cc: serusers(a)iptel.org > Sent: Wednesday, May 16, 2007 9:47:40 PM > Subject: Re: [Serusers] Problem running presence modules - SER won't start > > Hello, > how did you start ser? Aren't you using sudo? > Vaclav > > On Wed, May 16, 2007 at 11:24:07AM -0700, Mahatma Sarasvati wrote: > > Hello All, > > > > Running SER 2.0.0 rc1 from CVS (updated yesterday) on CentOS 5. > > > > I'm trying to test/use the presence functionality in SER. > > I've compiled and installed with make group_include="standard standard-dep" all > > > > I've set the environment variable: export LD_LIBRARY_PATH=/usr/local/lib/ser > > [root@vx1 modules]# echo $LD_LIBRARY_PATH > > /usr/local/lib/ser > > > > My supporting modules are here: > > > > [root@vx1 modules]# ls -l /usr/local/lib/ser/ > > total 828 > > -rwxr-xr-x 1 root root 40768 May 14 15:19 libbinrpc.so > > -rwxr-xr-x 1 root root 179044 May 14 15:19 lib_ser_cds.so > > -rwxr-xr-x 1 root root 213028 May 14 15:19 lib_ser_presence.so > > -rwxr-xr-x 1 root root 371056 May 14 15:19 lib_ser_xcap.so > > drwxr-xr-x 2 root root 4096 May 14 15:19 modules > > [root@vx1 modules]# > > > > Load module commands are in ser.cfg: > > loadmodule "/usr/local/lib/ser/modules/xmlrpc.so" > > loadmodule "/usr/local/lib/ser/modules/xcap.so" > > loadmodule "/usr/local/lib/ser/modules/pa.so" > > loadmodule "/usr/local/lib/ser/modules/presence_b2b.so" > > loadmodule "/usr/local/lib/ser/modules/rls.so" > > loadmodule "/usr/local/lib/ser/modules/dialog.so" > > > > libXML2 and libXML2-devel are installed. > > > > Yet, when I try to start SER I get this: > > May 15 13:09:24 vx1 ser: ERROR: load_module: could not open module </usr/local/lib/ser/modules/pa.so>: lib_ser_xcap.so: cannot open shared object file: No such file or directory > > May 15 13:09:24 vx1 ser: parse error (81,13-14): failed to load module > > May 15 13:09:24 vx1 ser: ERROR: load_module: could not open module </usr/local/lib/ser/modules/presence_b2b.so>: lib_ser_xcap.so: cannot open shared object file: No such file or directory > > May 15 13:09:24 vx1 ser: parse error (82,13-14): failed to load module > > May 15 13:09:24 vx1 ser: ERROR: load_module: could not open module </usr/local/lib/ser/modules/rls.so>: lib_ser_xcap.so: cannot open shared object file: No such file or directory > > May 15 13:09:24 vx1 ser: parse error (83,13-14): failed to load module > > May 15 13:09:24 vx1 ser: ERROR: load_module: could not open module </usr/local/lib/ser/modules/dialog.so>: lib_ser_cds.so: cannot open shared object file: No such file or directory > > > > Can anyone help? I think I've done everything right. Why won't SER start? > > > > Thanks, > > M > > > > Here's my modules directory: > > [root@vx1 modules]# ls -l > > total 7264 > > -rwxr-xr-x 1 root root 67000 May 14 15:19 acc_db.so > > -rwxr-xr-x 1 root root 73248 May 14 15:19 acc_syslog.so > > -rwxr-xr-x 1 root root 69946 May 14 15:19 auth_db.so > > -rwxr-xr-x 1 root root 58071 May 14 15:19 auth.so > > -rwxr-xr-x 1 root root 37345 May 14 15:19 avp_db.so > > -rwxr-xr-x 1 root root 168768 May 14 15:19 avpops.so > > -rwxr-xr-x 1 root root 79110 May 14 15:19 avp.so > > -rwxr-xr-x 1 root root 279385 May 14 15:19 ctl.so > > -rwxr-xr-x 1 root root 72896 May 14 15:19 db_ops.so > > -rwxr-xr-x 1 root root 218639 May 14 15:18 dbtext.so > > -rwxr-xr-x 1 root root 97287 May 14 15:19 dialog.so > > -rwxr-xr-x 1 root root 63003 May 14 15:19 dispatcher.so > > -rwxr-xr-x 1 root root 21604 May 14 15:19 diversion.so > > -rwxr-xr-x 1 root root 114536 May 14 15:19 domain.so > > -rwxr-xr-x 1 root root 45704 May 14 15:19 enum.so > > -rwxr-xr-x 1 root root 109624 May 14 15:19 eval.so > > -rwxr-xr-x 1 root root 84357 May 14 15:19 exec.so > > -rwxr-xr-x 1 root root 81398 May 14 15:19 fifo.so > > -rwxr-xr-x 1 root root 53442 May 14 15:19 flatstore.so > > -rwxr-xr-x 1 root root 43414 May 14 15:19 gflags.so > > -rwxr-xr-x 1 root root 87574 May 14 15:19 lcr.so > > -rwxr-xr-x 1 root root 35883 May 14 15:19 maxfwd.so > > -rwxr-xr-x 1 root root 112635 May 14 15:19 mediaproxy.so > > -rwxr-xr-x 1 root root 87485 May 14 15:19 msilo.so > > -rwxr-xr-x 1 root root 106696 May 14 15:19 mysql.so > > -rwxr-xr-x 1 root root 187822 May 14 15:19 nathelper.so > > -rwxr-xr-x 1 root root 21203 May 14 15:19 options.so > > -rwxr-xr-x 1 root root 741640 May 14 15:19 pa.so > > -rwxr-xr-x 1 root root 131893 May 14 15:19 pdt.so > > -rwxr-xr-x 1 root root 272113 May 14 15:19 permissions.so > > -rwxr-xr-x 1 root root 122158 May 14 15:19 pike.so > > -rwxr-xr-x 1 root root 232133 May 14 15:19 presence_b2b.so > > -rwxr-xr-x 1 root root 26172 May 14 15:19 print.so > > -rwxr-xr-x 1 root root 47597 May 14 15:19 ratelimit.so > > -rwxr-xr-x 1 root root 152843 May 14 15:19 registrar.so > > -rwxr-xr-x 1 root root 402652 May 14 15:19 rls.so > > -rwxr-xr-x 1 root root 114896 May 14 15:19 rr.so > > -rwxr-xr-x 1 root root 63276 May 14 15:19 sanity.so > > -rwxr-xr-x 1 root root 66942 May 14 15:19 sl.so > > -rwxr-xr-x 1 root root 222033 May 14 15:18 sms.so > > -rwxr-xr-x 1 root root 34415 May 14 15:19 speeddial.so > > -rwxr-xr-x 1 root root 120807 May 14 15:19 textops.so > > -rwxr-xr-x 1 root root 43276 May 14 15:19 timer.so > > -rwxr-xr-x 1 root root 848962 May 14 15:19 tm.so > > -rwxr-xr-x 1 root root 117593 May 14 15:19 uac.so > > -rwxr-xr-x 1 root root 38793 May 14 15:19 unixsock.so > > -rwxr-xr-x 1 root root 38979 May 14 15:19 uri_db.so > > -rwxr-xr-x 1 root root 38603 May 14 15:19 uri.so > > -rwxr-xr-x 1 root root 351468 May 14 15:19 usrloc.so > > -rwxr-xr-x 1 root root 41867 May 14 15:19 xcap.so > > -rwxr-xr-x 1 root root 100354 May 14 15:19 xlog.so > > -rwxr-xr-x 1 root root 98456 May 14 15:19 xmlrpc.so > > [root@vx1 modules]# > > > > > > > > > > > > > > > > > > ____________________________________________________________________________________Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games. > > http://get.games.yahoo.com/proddesc?gamekey=monopolyherenow > > _______________________________________________ > > Serusers mailing list > > Serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > ____________________________________________________________________________________ > Don't pick lemons. > See all the new 2007 cars at Yahoo! Autos. > http://autos.yahoo.com/new_cars.html

1 0

[Users] Customize messages [max_contacts]
by jprevedello 22 May '07

22 May '07

Hi gyus, I configure my openser to restrict the number of register with the same user using the function [max_contacts], and functioned perfectly. # Allow no more than 1 contacts per AOR modparam("registrar", "max_contacts", 1) Whenever that a user tries to register using the user that already register, the system return the following message: "Registration error:503 - Service Unavailable" I need to customize the message informing the user the reason of error, something as "user already registered" (softphone). I effected some tests, but I did not have success. Something as: if( t_check_status("503")) { t_reply("503","User already registered!!!"); exit; } Any sugestion ? Regards

2 1

[Serusers] VON Europe
by Greger V. Teigre 22 May '07

22 May '07

Hi guys! Anyone going to VON Europe in Stockholm?! I will be there, so if there it's interest, we can arrange an informal users' meeting (or maybe a SER pub meeting :-) Jan has signaled his interest of participating remotely if possible. Regardless of users' meeting or not, I'm open to meeting anyone interested, so ping me :-) I'm staying at the Rica hotel next to the fairground and can thus easily meet. g-) PS! For those of you who find it strange that SER is not present as an exhibitor: That costs money and resources (in terms of volunteer presence and preparation). As iptel.org only relies on sponsors and part-time volunteers, and nobody has raised the topic on the mailing lists, we won't be exhibiting :-) I don't know if any of the sponsors of SER are present as exhibitors, they may be.

1 0

Re: [Serusers] SER/PSTN Gateway conf
by Greger Viken Teigre 22 May '07

22 May '07

Copy the list! I don't understand your problem. The BYE will be caught by the loose route section of the config and relayed. The standard getting started should work fine. g-) ------- Original message ------- From: flavio <flavio.patria(a)gmail.com> Sent: 12.5.'07, 15:09 > 2007/5/12, Greger V. Teigre <greger(a)teigre.com>: > > Flavio, > > SER doesn't reply 200 OK to BYE message, the SIP UA does that. So, you > > are fine, the BYE reaches the UA. > > g-) > > > > That's right. > > If my gateway send INVITE message to SER, it does proxy it to UAC reached! > So UAC (SIP Phone) and Gateway (UAC in SIP scenario) are involved in a > media session. > If my gateway send BYE message to SER, it does not proxy it to UAC. > So I could send reply (200 OK) to Gateway (It works fine, but is not > solution!), but I cannot proxy BYE to UAC. > > How can proxy this? > > Thanks, > > Flavio

4 7

[Serusers] Registrar / Expires & reject frequent refreshes
by Kostas Marneris 22 May '07

22 May '07

Hello all, I have an issue with some SIP clients (like Linksys SPA 1001) that have not comply with the "expires" parameter of the Contact header of a "200 OK" Registrar Response. My SER conf : modparam("registrar", "default_expires", 1800) modparam("registrar", "min_expires", 300) modparam("registrar", "max_expires", 1800) Although the (200 OK) response contains the 'expires' parameter : Contact: <sip:userb2@192.168.2.10:5061>;expires=300;received="sip:1.2.3.4:5061" the SIP client ignores it and keeps sending REGISTERs every 5 sec (if configured so). Is there a way for SER to reject these registrations with (423 Interval Too Brief), according to RFC3261 - 21.4.17 (423 Interval Too Brief) & 10.3 (Processing REGISTER Requests). The 'retry_after()' registrar modparam is not my case. thank you, Kostas

2 1

[Serusers] Call pickup
by Stefano Blasco 22 May '07

22 May '07

Dear all, i'm trying to have call pickup with SER (i use ser 0.9.6), i mean i'd like to have the capability to pick up a call that is directed to another user while the other user telephone is ringing. Is there someone that suceeded in this features? If it is not possible to get this feature by configuriation of ser (i mean using already present ser functions), is there someone that can suggest how to implemente such features? (for istance with external scripts,or with avpops, or using fifo, or writing a new ser module). Maybe the use of fifo could help, in particular i saw the ctd.sh example that is able to iniziate a call for two users using t_uac_dlg fifo function. I tryed to use t_uac_dlg to send a CANCEL to a transaction in progress (in the 180 ringing status) (taking by hand parameters like Call-ID, tag, branch etc.), but it seems that t_uac_dlg is not able to interact with the transaction in progress because it creates one different transaction. Is it real, or am i wrong? This is some log for this attempt: ################################################################ 1(17187) INFO: fifo_server: command empty 1(17187) fifo_get_method: method: 'CANCEL' 1(17187) DEBUG: fifo_get_ruri: 'sip:801@x.x.x' 1(17187) DEBUG: fifo_get_nexthop: next hop empty 1(17187) fifo_get_headers: headers: To: <sip:801@x.x.x> From: Stefano<sip:803@x.x.x>;tag=237abc61 Call-ID: 262a660822777d26 CSeq: 2 CANCEL 1(17187) parse_headers: flags=-1 1(17187) end of header reached, state=9 1(17187) DEBUG: get_hdr_field: <To> [28]; uri=[sip:801@x.x.x] 1(17187) DEBUG: to body [<sip:801@x.x.x> ] 1(17187) get_hdr_field: cseq <CSeq>: <2> <CANCEL> 1(17187) DEBUG: fifo_uac: parse_headers succeeded 1(17187) fifo_get_body: body: 1(17187) DEBUG: add_param: tag=237abc61 1(17187) end of header reached, state=29 1(17187) DEBUG: fifo_uac: EoL -- proceeding to transaction creation 1(17187) DEBUG:tm:t_uac: next_hop=<sip:801@x.x.x> 1(17187) DEBUG: mk_proxy: doing DNS lookup... 1(17187) get_record: lookup(_sip._udp.x.x.x, 33) failed 1(17187) sip_resolvehost: no SRV record found for x.x.x, trying 'normal' lookup... 1(17187) DEBUG:destroy_avp_list: destroying list (nil) 1(17187) DEBUG: dlg2hash: 55633 1(17187) DEBUG: add_to_tail_of_timer[4]: 0xb5ba0ac0 1(17187) DEBUG: add_to_tail_of_timer[0]: 0xb5ba0ad0 1(17187) **** done consume 2(17188) SIP Request: 2(17188) method: <CANCEL> 2(17188) uri: <sip:801@x.x.x> 2(17188) version: <SIP/2.0> 2(17188) parse_headers: flags=1 2(17188) Found param type 232, <branch> = <z9hG4bK159d.9169633.0>; state=16 2(17188) end of header reached, state=5 2(17188) parse_headers: Via found, flags=1 2(17188) parse_headers: this is the first via 2(17188) After parse_msg... 2(17188) preparing to run routing scripts... 2(17188) parse_headers: flags=128 2(17188) end of header reached, state=9 2(17188) DEBUG: get_hdr_field: <To> [28]; uri=[sip:801@x.x.x] 2(17188) DEBUG: to body [<sip:801@x.x.x> ] 2(17188) get_hdr_field: cseq <CSeq>: <2> <CANCEL> 2(17188) DEBUG: get_hdr_body : content_length=0 2(17188) found end of header 2(17188) DEBUG: is_maxfwd_present: max_forwards header not found! 2(17188) DEBUG: add_param: tag=237abc61 2(17188) end of header reached, state=29 2(17188) parse_headers: flags=256 2(17188) find_first_route: No Route headers found 2(17188) loose_route: There is no Route HF 2(17188) is_local(): Realm 'x.x.x' is local 2(17188) DEBUG: t_newtran: msg id=9 , global msg id=8 , T on entrance=0xffffffff 2(17188) parse_headers: flags=-1 2(17188) parse_headers: flags=60 2(17188) t_lookup_request: start searching: hash=55633, isACK=0 2(17188) DEBUG: RFC3261 transaction matching failed 2(17188) DEBUG: t_lookup_request: no transaction found 2(17188) DBG: trans=0xb5ba1488, callback type 1, id 0 entered 2(17188) parse_headers: flags=44 2(17188) DEBUG: t_lookupOriginalT: searching on hash entry 55633 2(17188) DEBUG: RFC3261 transaction matching failed 2(17188) DEBUG: t_lookupOriginalT: no CANCEL matching found! 2(17188) DEBUG: t_lookupOriginalT completed 2(17188) DEBUG: mk_proxy: doing DNS lookup... 2(17188) get_record: lookup(_sip._udp.x.x.x, 33) failed 2(17188) sip_resolvehost: no SRV record found for x.x.x, trying 'normal' lookup... 2(17188) check_via_address(x.x.x.x, x.x.x.x, 0) ################################################################ Another thing, i saw that t_uac_cancel, instead, is able to interact with an existing transaction, by sending CANCEL to called and 487 to caller (using t_lookup_callid function to get transaction from callid and cseq). Does someone think that it is possible to modify t_uac_cancel to handle basic pickup functions? (that should be CANCEL for the called side and redirect for caller towards pickup user) This is some log for this case: ################################################################## 1(16115) INFO: fifo_server: command empty 1(16115) DEBUG: fifo_uac_cancel: ############### begin ############## 1(16115) DEBUG: fifo_uac_cancel: callid="0b29c5266716aa67" 1(16115) DEBUG: fifo_uac_cancel: cseq="2" 1(16115) created comparable call_id header field: >Call-ID: 0b29c5266716aa67 < 1(16115) created comparable cseq header field: >CSeq: 2 INVITE< 1(16115) just locked hash index 37328, looking for transactions there: 1(16115) we have a match: callid=>>Call-ID: 0b29c5266716aa67 << cseq=>>CSeq: 2<< 1(16115) DEBUG: t_lookup_callid: transaction found. 1(16115) DEBUG: fifo_uac_cancel: now calling cancel_uacs 1(16115) DEBUG: cancel_branch: sending cancel... 1(16115) DEBUG: add_to_tail_of_timer[4]: 0xb5b763f0 1(16115) DEBUG: add_to_tail_of_timer[0]: 0xb5b76400 1(16115) DEBUG: fifo_uac_cancel: ################ end ############## ################################################################## Thank you very much for your help Stefano Blasco

2 1

[Serusers] behind 2 routers
by root 22 May '07

22 May '07

Hi, I have a DSL router, which has only one port. To connect multiple computers I have another 4 ports router behind this DSL router. So SER is behind two routers. How can I configure these two routers so that I can use SER in this situation? Thanks,

1 0

[Serusers] SIP Security and Monitoring tool running on iptel.org
by Tudor Golubenco 21 May '07

21 May '07

Hi SER users and devs, During the past few months we, at Iptego, have been working on a security and monitoring application for SIP sites, one of the recurring subjects of the SER user meeting in Prague ;-). To be short, VSP is a network intrusion detection system, which can sniff the VoIP traffic, parse SIP messages and create alerts when something seems wrong. As the project evolved, we have presented our work to the iptel.org staff and, together with them, decided to make the monitoring extension of VSP available to all public. So, if you're interested in real time statistics of the iptel.org SIP site, or in the repartition of User-Agents, you can create your own graphs here: http://fly.iptel.org/vsp/ We have already received great feedback from the iptel.org guys and I hope to get even more from the rest of community. Disclaimer: The project is BETA for now and the graphs haven't been verified yet. I wouldn't use them for anything important. Cheers, Tudor Golubenco -- Dipl. Ing. Tudor Golubenco Software Engineer iptego GmbH Am Borsigturm 40, 13507 Berlin, Germany http://www.iptego.de

5 8

[Users] x-lite and nat_uac_test(16)
by Alexander Bergolth 21 May '07

21 May '07

Hi! I've noticed that (at least on my boxes) x-lite uses a different source-port for the sip-connection than the one that is announced in the Via-header. (See the example below.) This behavior results in nat_uac_test(16) always reporting that the client is behind a NAT. Can anyone confirm this behavior? Is it a bug in X-Lite or is there any good reason for it? Cheers, --leo -------------------- 8< -------------------- No. Time Source Destination Protocol Info 4 0.000808 137.208.90.164 137.208.89.45 SIP Request: REGISTER sip:wu-wien.ac.at Frame 4 (640 bytes on wire, 640 bytes captured) Ethernet II, Src: DellEsgP_85:83:ea (00:0b:db:85:83:ea), Dst: Cisco_f3:2e:80 (00:0d:29:f3:2e:80) Internet Protocol, Src: 137.208.90.164 (137.208.90.164), Dst: 137.208.89.45 (137.208.89.45) Transmission Control Protocol, Src Port: 1831 (1831), Dst Port: 5060 (5060), Seq: 1, Ack: 1, Len: 586 Source port: 1831 (1831) Destination port: 5060 (5060) Sequence number: 1 (relative sequence number) [Next sequence number: 587 (relative sequence number)] Acknowledgement number: 1 (relative ack number) Header length: 20 bytes Flags: 0x18 (PSH, ACK) Window size: 1028096 (scaled) Checksum: 0x491e [correct] Session Initiation Protocol Request-Line: REGISTER sip:wu-wien.ac.at SIP/2.0 Message Header Via: SIP/2.0/TCP 137.208.90.164:1676;branch=z9hG4bK-d87543-bd24cf102007467f-1--d87543-;rport Max-Forwards: 70 Contact: <sip:30001@137.208.90.164:1676;rinstance=7ce462f932219953;transport=TCP> To: "Alexander Bergolth"<sip:30001@wu-wien.ac.at> From: "Alexander Bergolth"<sip:30001@wu-wien.ac.at>;tag=b904b134 Call-ID: N2VjNGQxYzZlOWNiYWU5OTMwNzQ4OTZkMWU3NmRkYTE. CSeq: 1 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite release 1006e stamp 34025 Content-Length: 0 -------------------- 8< -------------------- -- e-mail ::: Alexander.Bergolth (at) wu-wien.ac.at fax ::: +43-1-31336-906050 location ::: Computer Center | Vienna University of Economics | Austria

5 10

[Users] WeSIP: Click2Call Linefeed
by Helmut Kuper 21 May '07

21 May '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello WeSIPer, I found a little but annoying Error in WeSIP's C2C application. I send back a 483 reply on a C2C's first caller-INVITE to WeSIP via openser's sl-send_reply(). I expected that C2C terminates the call function, cause C2C has an error handling method for 4xx replies, but nothing happens. openser as wesip's frontend received definately the reply. By chance I found, that the reason for this was my reply message. This doesn't work with WeSIP's C2C application: sl_send_reply("483", "My message\n"); But this works: sl_send_reply("483", "My message"); It seems, that C2C's or WeSIPs reply parser has some problems with linefeeds at the end of reply's message text. I use openser 1.2.0, WeSIP 0.1.4 and click2call-08_11_06 regards Helmut PS: Is there already a fix for WeSIP to reconnect to openser's seas socket, when openser restarts? -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGUa8R4tZeNddg3dwRAscvAKDC49yJRkA4ALOCoC5naJzr9gavMwCff7vX oEc4vGa8tEP5r0IVMhvC+jA= =dMZ6 -----END PGP SIGNATURE-----

1 0

[Users] Openser and Mediaproxy behind NAT
by Maciek Kowalski 21 May '07

21 May '07

Hi All, I am new to Openser and propably answer to my question is very simple.. I,ve been trying to run Openser with Mediaroxy behind NAT (they both work on a same machine with a private IP - 192.168.1.x). I am connected to internet through ADSL router (dynamic IP). I ve tried few configurations and I cant get it to work - I have "no audio problem" with nated users who register from outside of my LAN. I ve read that mediaproxy must listen on a public IP address, but first I thought that this is just a matter of port forwarding. And here is my question: Is it possible for this to work with described configuration? (Openser with mediaproxy both behind NAT) I ve used similar solution with Asterisk and Xtunnels and it worked fine. Thanks in advance and sorry for my English:). Best regards, Maciek Kowalski

2 1

[Serusers] sip uri transform
by Ҷ�� 21 May '07

21 May '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: Hi. I have a question. I use x_lite do client,and ser server do route. if user1 is 1001 and user2 is 1002, assumes i used 1001 to dial 999991002, I thinking in the ser.conf, first to remove the prefix(99999),then route to 1002. if used asterisk the idea is ok but I don't know ser whether to achieve? [...] Content analysis details: (9.8 points, 5.0 required) pts rule name description ---- ---------------------- -------------------------------------------------- -0.0 SPF_HELO_PASS SPF: HELO matches SPF record 4.1 FROM_ILLEGAL_CHARS From: has too many raw illegal characters -0.0 SPF_PASS SPF: sender matches SPF record 0.5 HTML_40_50 BODY: Message is 40% to 50% HTML 0.0 HTML_MESSAGE BODY: HTML included in message 0.0 BAYES_50 BODY: Bayesian spam probability is 40 to 60% [score: 0.5035] 0.0 MIME_HTML_ONLY BODY: Message only has text/html MIME parts 0.2 DNS_FROM_RFC_ABUSE RBL: Envelope sender in abuse.rfc-ignorant.org 2.2 RCVD_IN_WHOIS_INVALID RBL: CompleteWhois: sender on invalid IP block [202.105.45.47 listed in combined-HIB.dnsiplists.completewhois.com] 1.7 DNS_FROM_RFC_POST RBL: Envelope sender in postmaster.rfc-ignorant.org 1.1 HTML_MIME_NO_HTML_TAG HTML-only message, but there is no HTML tag 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.

2 1

[Users] Presence Problem - Publish Offline
by Hildebrando Trannin 21 May '07

21 May '07

Hi all, I was testing OpenSER presence functionality using eyebeam and a softphone developed by my team. The OFFLINE state with eyebeam worked perfectly, but with my softphone it didn't work. I set force_active = 1. XML sent by PUBLISH and OpenSER log are below. Can anyone help me? Thanks, Hildebrando PIDF XML: <?xml xmlns="urn:ietf:params:xml:ns:pidf" entity="sip:hildebrando@voip.maritaca"> <presence> <tuple id=""> <status> <basic>closed</basic> </status> <note>offline</note> </tuple> </presence> The basic difference between my application and eyebeam is that eyebeam uses RPID XML and I use PIDF XML. OPENSER LOG: 3(26039) PRESENCE: update_presentity: The presentity status is offline; do not send notify

2 1

[Serusers] Does SER's MYSQL work in SUSE Linux?
by Cyprian Clement 20 May '07

20 May '07

Hello Community, Does anyone know whether it's possible to install the current recommended MYSQL on SUSE Linux. I have a Machine with a SUSE installed on it. I have successfully installed SER 0.9.6 but now I would like to have MYSQL for authentication support. Does the SER's recommended MYSQL work with SUSE?. I only see the talk about Redhat and Debian so unsure if it works with SUSE. Many thanks for in advance. Cyprian. ---------- ___________________________________________________________ What kind of emailer are you? Find out today - get a free analysis of your email personality. Take the quiz at the Yahoo! Mail Championship. http://uk.rd.yahoo.com/evt=44106/*http://mail.yahoo.net/uk

2 1

[Users] Dropped calls with OpenSER 1.1.x and Asterisk >= 1.2.14
by Edoardo Serra 19 May '07

19 May '07

Hi guys, I'm having a problem about asterisk dropping calls received from OpenSER The problems was spotted at Asterisk users mailing lists http://lists.digium.com/pipermail/asterisk-users/2007-April/184875.html Asterisk complains about an ACK not received and drops the calls. It expects OpenSER to send an ACK back to its '200 OK' when the call is set up Is that a problem OpenSER should solve ??? Tnx in advance Regards Edoardo Serra WeBRainstorm S.r.l.

2 3

[Users] Getting start
by Mhmd 19 May '07

19 May '07

HI , Kindly provide me with the link , where can i get the OPENSER , thank you Mohammad

2 1

[Users] openser as b2bua
by A.M. 19 May '07

19 May '07

The openser website clearly states that OpenSER is not a B2BUA. Is there a particular architectural limitation that would prevent a module from implementing a B2BUA within OpenSER? Cheers, M

3 2

[Users] openser and postgresql
by zhaojunjie01＠st.lzu.edu.cn 19 May '07

19 May '07

Hello: I use postgresql to authorize the user.But it cann't work.Who can give me some suggestion? Thank you very much! My error log and openser.cfg fileas follows: #Error log May 18 12:24:43 th016184 /usr/local/sbin/openser[4358]: INFO: statistics manager successfully initialized May 18 12:24:43 th016184 /usr/local/sbin/openser[4358]: PostgreSQL - initializing May 18 12:24:43 th016184 /usr/local/sbin/openser[4358]: StateLess module - initializing May 18 12:24:43 th016184 /usr/local/sbin/openser[4358]: TM - initializing... May 18 12:24:43 th016184 /usr/local/sbin/openser[4358]: Maxfwd module- initializing May 18 12:24:43 th016184 /usr/local/sbin/openser[4358]: INFO:ul_init_locks: locks array size 512 May 18 12:24:43 th016184 /usr/local/sbin/openser[4358]: TextOPS - initializing May 18 12:24:43 th016184 /usr/local/sbin/openser[4358]: AUTH module - initializing May 18 12:24:43 th016184 /usr/local/sbin/openser[4358]: AUTH_DB module - initializing May 18 12:24:44 th016184 /usr/local/sbin/openser[4358]: register_udomain(): Cannot open database connection file! May 18 12:24:44 th016184 /usr/local/sbin/openser[4358]: domain_fixup(): Error while registering domain May 18 12:24:44 th016184 /usr/local/sbin/openser[4358]: ERROR: fix_actions: fixing failed (code=-1) at cfg line 145 May 18 12:24:44 th016184 /usr/local/sbin/openser[4358]: INFO:mi_fifo:mi_destroy: process hasn't been created -> nothing to kill #openser.cfg loadmodule "postgres.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" loadmodule "xlog.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) modparam("usrloc", "db_url", "postgres://openser:openserrw@localhost/openser") # -- 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", "db_url", "postgres://openser:openserrw@localhost/openser") 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=~"localhost") { # mark routing logic in request # 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=~"localhost") { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("localhost", "subscriber")) { www_challenge("localhost", "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; }

1 0

[Users] pseudovariables in rewriteuser() - OpenSER 1.1.x
by Edoardo Serra 18 May '07

18 May '07

Hi all, I have the need to call rewriteuser() with a pseudovariable. My need is to rewrite the user with the content of an extra header I put in the SIP packet coming from a PSTN gateway I tried with rewriteuser("$hdr(X-DNID)"); but it doesn't replace the string with the ocntent of the pseudovariable. Any idea ? Tnx in advance Edoardo Serra WeBRainstorm S.r.l.

3 5

[Users] "Register problem"
by Danielo Garcia 18 May '07

18 May '07

Hello all. I have a problem with register I dont know why, when I try to register a user behind NAT listening in port 5060 the evalution of the sentence "if (uri==myself){" is true and when change the port for another for example 5070 the evaluation of the same sentence is false could somebody explain me? or how can I do the sentence "if (uri==myself){" using patern matching My configuration is listen = IP-MYSERVER port = 5060 Thanks..

2 1

[Users] using of myself Core Values
by Victor Gamov 18 May '07

18 May '07

hi is it possible to do following in config if ( $avp(s:var_name) != myself ) { # do something } var_name is AVP looking like "192.168.0.1" If not -- how I can check that $avp(s:var_name) is not one of my IP-address (without using DOMAIN or other module but by core only) ? Thanks! -- CU, Victor Gamov

2 2

Re: [Users] x-lite and nat_uac_test(16)
by Alexander 'Leo' Bergolth 18 May '07

18 May '07

Hi! On 05/18/2007 05:12 PM, Andreas Granig wrote: > Alexander Bergolth wrote: >> I've noticed that (at least on my boxes) x-lite uses a different >> source-port for the sip-connection than the one that is announced in the >> Via-header. (See the example below.) > > Are you sure there isn't any NAT or ALG in between? I've done the packet dump on both ends, on a box running x-lite (with the windows-firewall disabled) and on the proxy-host. Both show the same source/destination ip-adresses and ports, so I assume there cannot be any kind of ALG. However, there is some packet filter (not doing NAT) between the box and the proxy, maybe x-lite does some firewall-detection on startup and uses the wrong port-number in the via header intentionally to enforce the use of an rtp-proxy? I'll do some tests on a box that isn't protected by a packet filter on Monday. > By default, x-lite binds to local port 5060, but you've some > non-standard ports in there. So my guess is either a non-standard > port setting in x-lite and NAT, or a faulty ALG on the NAT device. I don't have any special settings in X-Lite, ICE is disabled, "Use local IP address" is enabled. I've noticed that your version of x-lite uses UDP, mine seems to use TCP by default. Cheers, --leo > Here's a trace using x-lite 2.0 r1105d (Linux): > > U 192.168.123.129:5060 -> <public IP>:5060 > REGISTER sip:<some domain> SIP/2.0. > Via: SIP/2.0/UDP 192.168.123.129:5060;rport;branch=z9hG4<snip> > > Cheers, > Andreas -- e-mail ::: Alexander.Bergolth (at) wu-wien.ac.at fax ::: +43-1-31336-906050 location ::: Computer Center | Vienna University of Economics | Austria

2 1

[Users] "Register problem"
by Danielo Garcia 18 May '07

18 May '07

Hello all. I have a problem with register I dont know why the condition if (uri==myself) { My configuration is listen = 148.226.26.245 port = 5060 Thanks

1 0

[Users] Mediaproxy running with Openser
by Lesley Liu 18 May '07

18 May '07

Hi Users, I'm trying to setup a mediaproxy 1.7.2 running with operser 1.2.0 on Fedora 6. I can see all service running up. But when I make a call, the SDP does not change the owner address in the INVITE to callee. Openser only report following error: May 18 09:25:51 localhost /sbin/openser[2808]: error: mediaproxy/sendMediaproxyCommand(): can't connect to MediaProxy Configure as attachment, can anyone help? Thanks in advance. Regards, Lesley

3 10

[Users] Question about forking
by Tim Madorma 18 May '07

18 May '07

Hi, I want to start openser with runsv which requires that the starting process run in the foreground. My problem is that I also want to listen on a couple of different ports. When I set forking = yes, it will listen on multiple ports, but runsv won't work. When I set forking=no, then openser will only listen on the first specified port. Is there any way around this? Can I have the starting process run in the foreground and fork other processes that listen to the ports in the background? Here is the error message: WARNING: no fork mode and more than one listen address found(will use only the the first one) Here are the associated configuration lines: fork=no children=32 # Local IP address/port pairs to listen to listen=65.185.233.55:5061 listen=65.185.233.55:5062 # Alias IP address/port pairs alias=65.185.233.104:5061 alias=65.185.233.104:5062 thanks, Tim

3 14

[Users] How to log the username of caller and callee to acc table?
by Charles Wang 18 May '07

18 May '07

Hi all, I ever used SER and I know it will save the From information such as 101(a)192.168.11.200 to sip_from or from_uri in its acc table. But in my openser, I also setup the acc module. Where is my caller info? I can find a lot of number such as "2806655523" in my from_tag. I wanna store the caller and callee info in my acc table when I make an internal call or an external call. Can anybody kind to help me to solve it? or give me some tips please. PS: I got this openser.cfg from http://www.sipwise.com/wizard. Best regards, Charles ######################################################################## # Parts of my openser.cfg Configuration ######################################################################## loadmodule "acc.so" modparam("acc", "early_media", 0) modparam("acc", "failed_transaction_flag", 24) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) modparam("acc", "log_flag", 0) modparam("acc", "log_missed_flag", 0) modparam("acc", "log_level", 2) modparam("acc", "db_flag", 25) modparam("acc", "db_missed_flag", 0) modparam("acc", "db_table_acc", "acc") modparam("acc", "db_table_missed_calls", "missed_calls") modparam("acc", "db_url", "mysql://openser:openserrw@localhost /openser") modparam("acc", "acc_method_column", "method") modparam("acc", "acc_callid_column", "callid") modparam("acc", "acc_time_column", "time") modparam("acc", "acc_totag_column", "to_tag") modparam("acc", "acc_from_tag_column", "from_tag") modparam("acc", "detect_direction", 1) modparam("acc", "acc_sip_code_column", "sip_code") modparam("acc", "acc_sip_reason_column", "sip_reason") modparam("acc", "multi_leg_info", "src_leg=$avp(i:901);dst_leg=$avp(i:902)") ######################################################################## # Request route 'main' ######################################################################## route[0] { xlog("L_INFO", "New request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); force_rport(); if(msg:len > max_len) { xlog("L_INFO", "Message too big - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("513", "Message Too Big"); exit; } if (!mf_process_maxfwd_header("10")) { xlog("L_INFO", "Too many hops - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("483", "Too Many Hops"); exit; } if(!is_method("REGISTER")) { if(nat_uac_test("19")) { record_route(";nat=yes"); } else { record_route(); } } if(is_method("CANCEL") || is_method("BYE")) { end_media_session(); } if(loose_route()) { if(!has_totag()) { xlog("L_INFO", "Initial loose-routing rejected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Initial Loose-Routing Rejected"); exit; } if(nat_uac_test("19") || search("^Route:.*;nat=yes")) { fix_nated_contact(); if(!search("^Content-Length:[ ]*0")) { setbflag(6); } } if(is_method("BYE")) { setflag(24); # account failed transactions setflag(25); # account successful transactions } # mark as loose-routed for acc setflag(26); route(11); } if(is_method("REGISTER")) { route(10); } setflag(24); # account failed transactions setflag(25); # account successful transactions if(is_method("INVITE")) { route(12); } if(is_method("CANCEL") || is_method("ACK")) { route(16); } route(17); } ######################################################################## # Request route 'clear-usr-preferences-caller' ######################################################################## route[1] { xlog("L_INFO", "Clear caller preferences - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); avp_delete("$avp(s:caller_cli)/g"); avp_delete("$avp(s:clir)/g"); } ######################################################################## # Request route 'clear-usr-preferences-callee' ######################################################################## route[2] { xlog("L_INFO", "Clear callee preferences - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); avp_delete("$avp(s:callee_fr_inv_timer)/g"); avp_delete("$avp(s:cfu)/g"); } ######################################################################## # Request route 'usr-preferences-caller' ############################################################ ############ route[3] { route(1); xlog("L_INFO", "Load caller preferences for uuid '$avp(s:caller_uuid)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # load caller avps avp_db_load("$avp(s:caller_uuid)", "*"); avp_copy("$avp(s:cli)", "$avp(s:caller_cli)/d"); if(is_avp_set("$avp(s:clir)/n") && avp_check("$avp(s:clir)", "eq/i:1")) { # mark for anonymization setflag(28); } } ######################################################################## # Request route 'usr-preferences-callee' ######################################################################## route[4] { xlog("L_INFO", "Load callee preferences for uuid '$avp(s:callee_uuid)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # load callee avps avp_db_load("$avp(s:callee_uuid)", "*"); if(is_avp_set("$avp(s:cfu)/s")) { xlog("L_INFO", "Call-forward-unconditional to '$avp(s:cfu)' found - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(5); route(6); avp_delete("$avp(s:caller_uuid)/g"); avp_copy("$avp(s:callee_uuid)", "$avp(s:caller_uuid)/d"); avp_pushto("$ru", "$avp(s:cfu)"); route(3); route(13); exit; } if(is_avp_set("$avp(s:ringtimeout)/n")) { xlog("L_INFO", "Setting ring timeout to $avp(s:ringtimeout) secs - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); avp_copy("$avp(s:ringtimeout)", "$avp(s:callee_fr_inv_timer)/d"); } } ######################################################################## # Request route 'acc-caller' ############################################################ ############ route[5] { xlog("L_INFO", "Setting acc source-leg for uuid '$avp(s:caller_uuid)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); avp_printf("$avp(i:901)", "$avp(s:caller_uuid)|$fU|$fd"); } ######################################################################## # Request route 'acc-callee' ######################################################################## route[6] { xlog("L_INFO", "Setting acc destination-leg for uuid '$avp(s:callee_uuid)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); avp_printf("$avp(i:902)", "$avp(s:callee_uuid)|$rU|$rd"); } ######################################################################## # Request route 'acc-failure' ######################################################################## route[7] { xlog("L_INFO", "Accounting failed request for uuid '$avp(s:caller_uuid)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(5); route(6); resetflag(24); acc_db_request("404", "acc"); } ######################################################################## # Request route 'clir' ######################################################################## route[8] { if(isflagset(28) && !isflagset(27)) { setflag(27); xlog("L_INFO", "Anonymize caller - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); uac_replace_from("Anonymous"," sip:anonymous@anonymous.invalid"); if(is_present_hf("Privacy")) { remove_hf("Privacy"); } append_hf("Privacy: id\r\n"); } } ######################################################################## # Request route 'stop-media-proxy' ######################################################################## route[9] { if(isflagset(22)) { end_media_session(); } } ######################################################################## # Request route 'base-route-register' ######################################################################## route[10] { sl_send_reply("100", "Trying"); if(!www_authorize("", "subscriber")) { xlog("L_INFO", "Register authentication failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); www_challenge("", "0"); exit; } if(!check_to()) { xlog("L_INFO", "Spoofed To-URI detected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Spoofed To-URI Detected"); exit; } consume_credentials(); if(!search("^Contact:[ ]*\*") && nat_uac_test("19")) { fix_nated_register(); setbflag(6); } if(!save("location")) { xlog("L_ERR", "Saving contact failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_reply_error(); exit; } xlog("L_INFO", "Registration successful - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); exit; } ######################################################################## # Request route 'base-outbound' ######################################################################## route[11] { if(is_present_hf("P-Asserted-Identity")) { remove_hf("P-Asserted-Identity"); } if(is_present_hf("Remote-Party-ID")) { remove_hf("Remote-Party-ID"); } if(is_avp_set("$avp(s:caller_cli)/s")) { if(!isflagset(28)) { xlog("L_INFO", "Set caller CLI '$avp(s:caller_cli)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); append_hf("P-Asserted-Identity: <$avp(s:caller_cli)>\r\n"); } } route(8); if(isbflagset(6) && !isflagset(22)) { setflag(22); fix_nated_contact(); use_media_proxy(); t_on_reply("2"); } else { t_on_reply("1"); } xlog("L_INFO", "Request leaving server - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if(!isflagset(21)) { t_on_failure("2"); if(!isflagset(26)) { route(5); route(6); } } if(!t_relay()) { sl_reply_error(); if(is_method("INVITE") && isbflagset(6)) { end_media_session(); } } exit; } ######################################################################## # Request route 'base-route-invite' ######################################################################## route[12] { sl_send_reply("100", "Trying"); if(from_gw()) { $avp(s:caller_uuid) = "0"; xlog("L_INFO", "Call from PSTN' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); setflag(23); } else { if(allow_trusted()) { if(is_avp_set("$avp(s:peer_uuid)/s")) { # use tag-column from trusted-table as uuid for this caller avp_copy("$avp(s:peer_uuid)", "$avp(s:caller_uuid)/d"); } else { # if no uuid is set, use "0" as default uuid $avp(s:caller_uuid) = "0"; } xlog("L_INFO", "Call from trusted peer with uuid '$avp(s:caller_uuid)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if(!is_domain_local("$rd")) { xlog("L_INFO", "Rejecting peering attempt with non-local request domain - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Relaying Denied"); exit; } setflag(23); } else { if(!proxy_authorize("", "subscriber")) { xlog("L_INFO", "Proxy authentication failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); proxy_challenge("", "0"); exit; } if(!check_from()) { xlog("L_INFO", "Spoofed From-URI detected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Spoofed From-URI Detected"); exit; } consume_credentials(); } } route(3); if(nat_uac_test("19")) { setbflag(6); } route(13); } ######################################################################## # Request route 'invite-find-callee' ######################################################################## route[13] { if(lookup("aliases")) { xlog("L_INFO", "Callee was aliased - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); } route(2); if(!is_domain_local("$rd")) { setflag(20); $avp(s:callee_uuid) = "0"; route(15); } avp_delete("$avp(s:callee_uuid)"); avp_db_query("select uuid from subscriber where username = '$rU'", "$avp(s:callee_uuid)"); if(is_avp_set("$avp(s:callee_uuid)/s")) { xlog("L_INFO", "Callee is local, uuid='$avp(s:callee_uuid)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(14); } else { $avp(s:callee_uuid) = "0"; xlog("L_INFO", "Callee is not local - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(15); } exit; } ######################################################################## # Request route 'invite-to-internal' ######################################################################## route[14] { route(4); if(!lookup("location")) { xlog("L_INFO", "Local user offline - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(7); sl_send_reply("404", "User Offline"); } else { xlog("L_INFO", "Local user online - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(11); } exit; } ######################################################################## # Request route 'invite-to-external' ######################################################################## route[15] { if(isflagset(20)) { xlog("L_INFO", "Call to foreign domain - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(11); exit; } if(!isflagset(23)) { # don't allow calls relaying from PSTN to PSTN, if not explicitely forwarded if(uri =~ "^sip:0[0-9]+@") { # only route numeric users to PSTN xlog("L_INFO", "Loading PSTN gateways - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if(!load_gws()) { xlog("L_ERR", "Error loading PSTN gateways - M=$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=$ci\n"); sl_send_reply("503", "PSTN Termination Currently Unavailable"); exit; } setflag(21); t_on_failure("1"); route(11); } } xlog("L_INFO", "Call to unknown user - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(7); sl_send_reply("404", "User Not Found"); exit; } ######################################################################## # Request route 'base-route-local' ######################################################################## route[16] { t_on_reply("1"); if(t_check_trans()) { xlog("L_INFO", "Request leaving server - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if(!t_relay()) { sl_reply_error(); } } else { xlog("L_INFO", "Dropping mis-routed request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); } exit; } ######################################################################## # Request route 'base-route-generic' ######################################################################## route[17] { xlog("L_INFO", "Method not supported - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("501", "Method Not Supported Here"); exit; } # Request route 'base-filter-failover' route[18] { if(!t_check_status("408|500|503")) { xlog("L_INFO", "No failover routing needed for this response code - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(9); exit; } } # Reply route 'base-standard-reply' onreply_route[1] { xlog("L_INFO", "Reply - S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); exit; } # Reply route 'base-nat-reply' onreply_route[2] { xlog("L_INFO", "NAT-Reply - S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); if(nat_uac_test("1")) { fix_nated_contact(); } if(isbflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if(!search("^Content-Length:[ ]*0")) { use_media_proxy(); } } exit; } -- Best Regards Charles

2 2

[Serusers] No Audio
by Arun Kumar 18 May '07

18 May '07

Hi I've configured my ser server with dedicated Media proxy server. When I call from IP to IP it works fine without any problem but when I try to dialout using ser there is no audio. But if I'll enable STUN in my phone I'm able to hear audio. my phones are behind Nat( Port rest con nat). Please guide me how to setup my ser or mediaproxy so I don't have to use STUN server. thanks arun

2 5

[Users] ACK loop after receiving 404 from remote GW
by Taylor Carpenter 18 May '07

18 May '07

I am getting a ACK loop after receiving a 404 from a remote GW. I am starting the call using sipp for testing. sipp -r 1 -l 1 -d 5000 -s 8989 -p 5061 -sn uac 10.14.18.22 The ngrep trace is showing Via and From set to 127.0.0.1 for the communication between OpenSER and the sipp system. The ngrep output and the routes from the openser.cfg are at http://pastebin.ca/493807 I assume the Via/From being set to 127.0.0.1 may be the issue, which I assume I could change from sipp, but I would also like OpenSER to catch the loop. Taylor

2 1

[Users] Openser 1.2 Presence Module
by Curt Moore 18 May '07

18 May '07

Hello all. I've been experimenting with the new presence module in openser-1.2 and have a few questions which hopefully someone can help answer. I'm using an SVN checkout of the 1.2 branch, r2237, and am testing with Polycom SoundPoint IP phones, firmware 2.1.1.0052. >From what I can tell, openser 1.2 does not fully support the xpidf content type, it only supoorts the winfo content type. I've looked through the code for the presence module extensively and it appears that whenever a SUBSCRIBE comes in where the Event header is simply "Event: presence" and the Accept header is "Accept: application/xpidf+xml", the message is acceped and a 202 is sent in reply but in any NOTIFY sent from openser as a result, there is no message body and the message does not contain the XML information for pidf. Just as a test, I changed the event field in active_watchers to "presence.winfo" and noticed that with this change, openser generates a NOTIFY with a message body which contains watcherinfo. This, of course, generates a 481 from the phone since it does not know about the "Event: presence.winfo" event type. Also as a test, I tried to use the pa module as I'd had success with this module in the past since it _does_ support xpidf. In 1.2 it appears that the database structure specified in the databse setup scripts does not agree with the table structure being used in the module. As a result, I could not get the pa module to work either. Am I overlooking some small detail? Please let me know if further information is needed in order to help diagnose and resolve this issue. Cheers, -Curt

2 3

[Users] Failure_route help
by Yago Bargalló 18 May '07

18 May '07

Hello, I'm new in Openser and now I'm trying to use a failure_route but it doesn't work and I don't know why. The version that I'm using is Openser 1.2. What I'm trying is to forward a call after fr_inv_timer to Asterisk but it doesn't work. This is my openser.cfg : (...) modparam("tm", "fr_inv_timer" , 8) # ------------------------- request routing logic ------------------- # main routing logic route{ (...) if (uri==myself) { if (method=="REGISTER") { if (!www_authorize("172.16.51.150", "subscriber")) { www_challenge("172.16.51.150", "0"); exit; }; save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; if(is_method("INVITE") && (uri=~"sip:2[0-9]+@")) { t_on_failure("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; } failure_route[1] { #redirect to Asterisk rewritehostport("172.16.51.160:5060"); rewriteuser("2002"); }

3 2

[Users] Compiling Openser 1.2.0 on Solaris
by Lesley Liu 18 May '07

18 May '07

Hi Users, I tried to compile openser-1.2.0-tls on Solaris 10. The gcc, bison, flex, make, libiconv packages are added. However, when command "make all", the message shows the following without further information at beginning. When command "make install", it only shows the first 4 lines as following. Please advise how to solve it. Thanks a lot. ; # pwd /opt/openser-1.2.0-tls # make all yacc -d -b cfg cfg.y conflicts: 1 shift/reduce yacc -d -b cfg cfg.y conflicts: 1 shift/reduce yacc -d -b cfg cfg.y conflicts: 1 shift/reduce yacc -d -b cfg cfg.y conflicts: 1 shift/reduce Compiling action.c gcc -g -O9 -funroll-loops -Wall -mcpu=ultrasparc -mtune=ultrasparc -DNAME='"openser"' -DVERSION='"1.2.0-notls"' -DARCH='"sparc64"' -DOS='"solaris"' -DCOMPILER='"gcc 3.4.6"' -D__CPU_sparc64 -D__OS_solaris -D__SMP_no -DCFG_DIR='"/usr/local/etc/openser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DSTATISTICS -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -DUSE_SIGACTION -DHAVE_DEVPOLL -DHAVE_SELECT -c action.c -o action.o Compiling blacklists.c .... ; Regards, Lesley

2 1

[Users] Compiling Openser 1.2.0 on Solaris
by Lesley Liu 18 May '07

18 May '07

Hi Users, I tried to compile openser-1.2.0-tls on Solaris 10. The gcc, bison, flex, make, libiconv packages are added. However, when command "make all", the message shows the following without further information at beginning. When command "make install", it only shows the first 4 lines as following. Please advise how to solve it. Thanks a lot. ; # pwd /opt/openser-1.2.0-tls # make all yacc -d -b cfg cfg.y conflicts: 1 shift/reduce yacc -d -b cfg cfg.y conflicts: 1 shift/reduce yacc -d -b cfg cfg.y conflicts: 1 shift/reduce yacc -d -b cfg cfg.y conflicts: 1 shift/reduce Compiling action.c gcc -g -O9 -funroll-loops -Wall -mcpu=ultrasparc -mtune=ultrasparc -DNAME='"openser"' -DVERSION='"1.2.0-notls"' -DARCH='"sparc64"' -DOS='"solaris"' -DCOMPILER='"gcc 3.4.6"' -D__CPU_sparc64 -D__OS_solaris -D__SMP_no -DCFG_DIR='"/usr/local/etc/openser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DSTATISTICS -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -DUSE_SIGACTION -DHAVE_DEVPOLL -DHAVE_SELECT -c action.c -o action.o Compiling blacklists.c .... ; Regards, Lesley

1 0

[Users] Call forward when no answer
by Howard Tang 18 May '07

18 May '07

Hi All, I have followed a tutorial and set up Asterisk as a voice mail server. http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+With+Op… It works fine when the UA is offline. Now, I want a call forwarded to the Voice mail server when there is no answer from the UA after 60 seconds(UA is registered on the openser). What should I do? Below is my config (copy from the above link). # requests for Media server if(is_method("INVITE") && !has_totag() && uri=~"sip:\*9") { route(3); exit; } # mark transaction if user is in voicemail group if(is_method("INVITE") && !has_totag() && is_user_in("Request-URI","voicemail")) { xdbg("user [$ru] has voicemail redirection enabled\n"); # backup R-URI avp_write("$ruri", "i:10"); setflag(2); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { if(isflagset(2)) { # route to Asterisk Media Server prefix("1"); rewritehostport("10.10.10.11:5060"); route(1); } else { sl_send_reply("404", "Not Found"); exit; } }; # voicemail access # - *98 - listen caller's voice messages, being prompted for pin # - *981 - listen voice messages, being promted for mailbox and pin # - *98XXXX - leave voice message to XXXX # route[3] { # direct voicemail if (uri =~ "sip:\*98@" ) { rewriteuser("1"); xdbg("voicemail access\n"); } else if (uri =~ "sip:\*981@" ) { strip(4); rewriteuser("11"); } else if (uri =~ "sip:\*98.+@" ) { strip(3); prefix("1"); } else { xlog("unknown media extension $rU\n"); sl_send_reply("404", "Unknown media service"); exit; } # route to Asterisk Media Server rewritehostport("10.10.10.11:5060"); route(1); } failure_route[1] { if (t_was_cancelled()) { xdbg("transaction was cancelled by UAC\n"); return; } # restore initial uri avp_pushto("$ruri", "i:10"); prefix("1"); # route to Asterisk Media Server rewritehostport("10.10.10.11:5060"); resetflag(2); route(1); }

3 6

RE: AW: [Users] OpenSER crashed with Segmentation Fault using PresenceModule
by Schumann Sebastian 17 May '07

17 May '07

Thank you both very much for this hints, I will change the versions I currently use immediately. Regards Sebastian -----Original Message----- From: Anca-Maria Vamanu [mailto:anca@voice-system.ro] Sent: Thursday, May 17, 2007 5:55 PM To: Ovidiu Sas Cc: Schumann Sebastian; users(a)openser.org Subject: Re: AW: [Users] OpenSER crashed with Segmentation Fault using PresenceModule I was just about to tell you this myself. Please take the svn version, either 1.2 or the trunk. Ovidiu Sas wrote: > Hi Sebastian, > > You should use the 1.2 svn branch (or maybe the trunk). There are > several fixes already present in the tree. There's no point testing > against the original 1.2 tarball. > > > Regards, > Ovidiu Sas

1 0

RE: AW: [Users] OpenSER crashed with Segmentation Fault using PresenceModule
by Schumann Sebastian 17 May '07

17 May '07

Alright, I just used the Debian 4.0 OpenSER distribution files. I will compile it by myself now and send you the Logs again. Thanks so far. Sebastian -----Original Message----- From: Anca-Maria Vamanu [mailto:anca@voice-system.ro] Sent: Thursday, May 17, 2007 5:37 PM To: Schumann Sebastian Cc: users(a)openser.org Subject: Re: AW: [Users] OpenSER crashed with Segmentation Fault using PresenceModule I have not found the source either. It would really help if you would compile openser with debug simbols activated (they are activated in the default Makefile) to see exactly which line the segm fault occures. Then, please post again the backtrace. Thanks, Anca Schumann Sebastian wrote: >Hello, > >here is the stderr console output (just the last lines which should be important as I saw there is a message size limit), and the gdb backtrace. >Do you see any hint for the crash? > >Best regards >Sebastian > >________________________________ > >Von: Anca-Maria Vamanu [mailto:anca@voice-system.ro] >Gesendet: Do 5/17/2007 16:57 >An: Schumann Sebastian >Cc: users(a)openser.org >Betreff: Re: [Users] OpenSER crashed with Segmentation Fault using PresenceModule > > > >Hello, > >Please post the log and the gdb backtrace. > >regards, > >Anca Vamanu > >Schumann Sebastian wrote: > > > >>Dear all >> >>I am currently writing my final thesis and want to set up a Presence Core therefore. I am using OpenSER as Presence Agent (with Registrar function or now with separat Registrar, both configuration has the problem). It crashes after a certain time and certain amount of watcher entries. I cannot replicate this or deliminate to a certain action but it always happends, no matter if I clean up the tables, restart the machine or whatever. >> >>My configuration is one XCAP Server (XCAP Lite 1.0 from ag-projects, working properly), one Registrar (OpenSER with MySQL DB, which forwards SUBSCRIBE/PUBLISH to PA) and Presence Agent (OpenSER with Presence Module, MySQL DB, XMLRPC Module) and Eyebeam 1.5 Clients. All runs on VMware virtual machines. >> >>I have just three clients that try to subscribe each other. Without any Subscriptions everything works properly, XML data gets stored, PUBLISH is processed etc. Subscribing one user by another is possible and works properly. >> >>After allowing a watcher to subscription (using XCAP) and if the watcher logs in next time, the state in the watcher table changes from pending to active. It happens now and then, when a user goes online and one of his subscribed presentities is not online, that the OpenSER PA crashes. >> >>Analyzing DEBUG info and backtrace was not successful, because I found no information or hints about it, just the message segmentation fault appears. >> >>Does anyone have this problem as well or has anybody an idea about it? I can provide you logs, the dump or traces if you need it. >> >>Thanks for your help. >> >>Sebastian >> >> >> >> >> > > > > > >----------------------------------------------------------------------- - > > > 0(4591) SIP Request: > 0(4591) method: <SUBSCRIBE> > 0(4591) uri: <sip:bob@10.96.115.125> > 0(4591) version: <SIP/2.0> > 0(4591) parse_headers: flags=2 > 0(4591) Found param type 232, <branch> = <z9hG4bK0a73.6d721e06.0>; state=16 > 0(4591) end of header reached, state=5 > 0(4591) parse_headers: Via found, flags=2 > 0(4591) parse_headers: this is the first via > 0(4591) After parse_msg... > 0(4591) preparing to run routing scripts... > 0(4591) parse_headers: flags=100 > 0(4591) Found param type 232, <branch> = <z9hG4bK-d87543-666f5148e9613c52-1--d87543->; state=6 > 0(4591) Found param type 235, <rport> = <32800>; state=16 > 0(4591) end of header reached, state=5 > 0(4591) parse_headers: Via found, flags=100 > 0(4591) parse_headers: this is the second via > 0(4591) DEBUG:maxfwd:is_maxfwd_present: value = 69 > 0(4591) DEBUG: t_newtran: T on entrance=0xffffffff > 0(4591) parse_headers: flags=ffffffffffffffff > 0(4591) DEBUG:parse_to:end of header reached, state=10 > 0(4591) DBUG:parse_to: display={}, ruri={sip:bob@10.96.115.125} > 0(4591) DEBUG: get_hdr_field: <To> [25]; uri=[sip:bob@10.96.115.125] > 0(4591) DEBUG: to body [<sip:bob@10.96.115.125> >] > 0(4591) get_hdr_field: cseq <CSeq>: <1> <SUBSCRIBE> > 0(4591) DEBUG: get_hdr_body : content_length=0 > 0(4591) found end of header > 0(4591) parse_headers: flags=78 > 0(4591) t_lookup_request: start searching: hash=14240, isACK=0 > 0(4591) DEBUG: RFC3261 transaction matching failed > 0(4591) DEBUG: t_lookup_request: no transaction found > 0(4591) parse_headers: flags=ffffffffffffffff > 0(4591) PRESENCE: handle_subscribe:Missing or unsupported event header field value > 0(4591) parse_headers: flags=ffffffffffffffff > 0(4591) check_via_address(10.96.115.125, 10.96.115.125, 0) > 0(4591) PRESENCE:handle_subscribe: ERROR occured > 0(4591) DEBUG:tm:UNREF_UNSAFE: after is 0 > 0(4591) DEBUG:destroy_avp_list: destroying list (nil) > 0(4591) receive_msg: cleaning up > 0(4591) SIP Request: > 0(4591) method: <SUBSCRIBE> > 0(4591) uri: <sip:bob@10.96.115.125> > 0(4591) version: <SIP/2.0> > 0(4591) parse_headers: flags=2 > 0(4591) Found param type 232, <branch> = <z9hG4bKc989.6f967b16.0>; state=16 > 0(4591) end of header reached, state=5 > 0(4591) parse_headers: Via found, flags=2 > 0(4591) parse_headers: this is the first via > 0(4591) After parse_msg... > 0(4591) preparing to run routing scripts... > 0(4591) parse_headers: flags=100 > 0(4591) Found param type 232, <branch> = <z9hG4bK-d87543-4e0baf255d4f5c21-1--d87543->; state=6 > 0(4591) Found param type 235, <rport> = <32800>; state=16 > 0(4591) end of header reached, state=5 > 0(4591) parse_headers: Via found, flags=100 > 0(4591) parse_headers: this is the second via > 0(4591) DEBUG:maxfwd:is_maxfwd_present: value = 69 > 0(4591) DEBUG: t_newtran: T on entrance=0xffffffff > 0(4591) parse_headers: flags=ffffffffffffffff > 0(4591) DEBUG:parse_to:end of header reached, state=10 > 0(4591) DBUG:parse_to: display={}, ruri={sip:bob@10.96.115.125} > 0(4591) DEBUG: get_hdr_field: <To> [25]; uri=[sip:bob@10.96.115.125] > 0(4591) DEBUG: to body [<sip:bob@10.96.115.125> >] > 0(4591) get_hdr_field: cseq <CSeq>: <1> <SUBSCRIBE> > 0(4591) DEBUG: get_hdr_body : content_length=0 > 0(4591) found end of header > 0(4591) parse_headers: flags=78 > 0(4591) t_lookup_request: start searching: hash=39068, isACK=0 > 0(4591) DEBUG: RFC3261 transaction matching failed > 0(4591) DEBUG: t_lookup_request: no transaction found > 0(4591) parse_headers: flags=ffffffffffffffff > 0(4591) PRESENCE: handle_subscribe:Missing or unsupported event header field value > 0(4591) parse_headers: flags=ffffffffffffffff > 0(4591) check_via_address(10.96.115.125, 10.96.115.125, 0) > 0(4591) PRESENCE:handle_subscribe: ERROR occured > 0(4591) DEBUG:tm:UNREF_UNSAFE: after is 0 > 0(4591) DEBUG:destroy_avp_list: destroying list (nil) > 0(4591) receive_msg: cleaning up > 0(4591) SIP Request: > 0(4591) method: <SUBSCRIBE> > 0(4591) uri: <sip:sebastian@10.96.115.125> > 0(4591) version: <SIP/2.0> > 0(4591) parse_headers: flags=2 > 0(4591) Found param type 232, <branch> = <z9hG4bKe413.3434b775.0>; state=16 > 0(4591) end of header reached, state=5 > 0(4591) parse_headers: Via found, flags=2 > 0(4591) parse_headers: this is the first via > 0(4591) After parse_msg... > 0(4591) preparing to run routing scripts... > 0(4591) parse_headers: flags=100 > 0(4591) Found param type 232, <branch> = <z9hG4bK-d87543-2a56156bb30b720d-1--d87543->; state=6 > 0(4591) Found param type 235, <rport> = <32800>; state=16 > 0(4591) end of header reached, state=5 > 0(4591) parse_headers: Via found, flags=100 > 0(4591) parse_headers: this is the second via > 0(4591) DEBUG:maxfwd:is_maxfwd_present: value = 69 > 0(4591) DEBUG: t_newtran: T on entrance=0xffffffff > 0(4591) parse_headers: flags=ffffffffffffffff > 0(4591) DEBUG:parse_to:end of header reached, state=10 > 0(4591) DBUG:parse_to: display={}, ruri={sip:sebastian@10.96.115.125} > 0(4591) DEBUG: get_hdr_field: <To> [31]; uri=[sip:sebastian@10.96.115.125] > 0(4591) DEBUG: to body [<sip:sebastian@10.96.115.125> >] > 0(4591) get_hdr_field: cseq <CSeq>: <1> <SUBSCRIBE> > 0(4591) DEBUG: get_hdr_body : content_length=0 > 0(4591) found end of header > 0(4591) parse_headers: flags=78 > 0(4591) t_lookup_request: start searching: hash=12622, isACK=0 > 0(4591) DEBUG: RFC3261 transaction matching failed > 0(4591) DEBUG: t_lookup_request: no transaction found > 0(4591) parse_headers: flags=ffffffffffffffff > 0(4591) PRESENCE: handle_subscribe: 'expires' found > 0(4591) PRESENCE: handle_subscribe: lexpire= 3600 > 0(4591) PRESENCE: handle_subscribe: 'To' header ALREADY PARSED: <sip:sebastian@10.96.115.125> > 0(4591) PRESENCE:handle_subscribe: 'From' header not parsed > 0(4591) DEBUG: add_param: tag=e035dc2e > 0(4591) DEBUG:parse_to:end of header reached, state=29 > 0(4591) DBUG:parse_to: display={"Bob"}, ruri={sip:bob@10.96.115.125} > 0(4591) PRESENCE:handle_subscribe: generating to_tag > 0(4591) PRESENCE :proces_rr: out rr [<sip:10.96.115.125;lr=on;ftag=e035dc2e>] > 0(4591) > [p_user]= sebastian [p_domain]= 10.96.115.125 > [w_user]= bob [w_domain]= 10.96.115.125 > 0(4591) [event]= presence > [staus]= active > [expires]= 3600 > 0(4591) [to_tag]= 10.4591.1179421870.7 > [from_tag]= e035dc2e > 0(4591) PRESENCE: update_subscribtion ... > 0(4591) > [p_user]= sebastian [p_domain]= 10.96.115.125 > [w_user]= bob [w_domain]= 10.96.115.125 > 0(4591) [event]= presence > [staus]= active > [expires]= 3600 > 0(4591) [to_tag]= 10.4591.1179421870.7 > [from_tag]= e035dc2e > 0(4591) expires: 3600 > 0(4591) PRESENCE:update_subscribtion:Inserting into database: >n_query_cols:14 > 0(4591) [0] = to_user sebastian > 0(4591) [1] = to_domain 10.96.115.125 > 0(4591) [2] = from_user bob > 0(4591) [3] = from_domain 10.96.115.125 > 0(4591) [4] = event presence > 0(4591) [5] = event_id > 0(4591) [6] = callid YTUwN2U3MmViZDM3NDNmZDdkMTllNTQ4MDgyN2ExMDk. > 0(4591) [7] = to_tag 10.4591.1179421870.7 > 0(4591) [8] = from_tag e035dc2e > 0(4591) [9] = contact sip:bob@10.96.100.128:32800 > 0(4591) [10] = status active > 0(4591) [11] = cseq 1 > 0(4591) parse_headers: flags=ffffffffffffffff > 0(4591) check_via_address(10.96.115.125, 10.96.115.125, 0) > 0(4591) PRESENCE:get_subs_dialog:querying database table = active_watchers > 0(4591) PRESENCE: get_subs_dialog:The query for subscribtion for [user]= sebastian,[domain]= 10.96.115.125 for [event]= presence.winfo returned no result > 0(4591) PRESENCE:query_db_notify: Could not get subs_dialog from database > 0(4591) PRESENCE:update_subscribtion:Could not send notify for presence.winfo > 0(4591) PRESENCE:notify:dialog informations: > 0(4591) > [p_user]= sebastian [p_domain]= 10.96.115.125 > [w_user]= bob [w_domain]= 10.96.115.125 > 0(4591) [event]= presence > [staus]= active > [expires]= 3600 > 0(4591) [to_tag]= 10.4591.1179421870.7 > [from_tag]= e035dc2e > 0(4591) PRESENCE:get_xcap_tree:The query in table xcap for [username]=sebastian , domain=10.96.115.125 returned result 0(4591) PRESENCE:get_xcap_tree: xcap body: ><?xml version="1.0" encoding="UTF-8"?> ><ruleset xmlns="urn:ietf:params:xml:ns:common-policy" xmlns:pr="urn:ietf:params:xml:ns:pres-rules" xmlns:cp="www.counterpath.com/privacy-lists"> ><rule id="whitelist"> ><conditions> ><identity> ><id entity="sip:bob@10.96.115.125"/> ></identity> ></conditions> ><actions> ><pr:sub-handling>allow</pr:sub-handling> ></actions> ><transformations> ><pr:provide-services> ><pr:all-services/> ></pr:provide-services> ><pr:provide-persons> ><pr:all-persons/> ></pr:provide-persons> ><pr:provide-devices> ><pr:all-devices/> ></pr:provide-devices> ><pr:provide-all-attributes/> ></transformations> ></rule> ><rule id="blacklist"> ><conditions> ></conditions> ><actions> ><pr:sub-handling>block</pr:sub-handling> ></actions> ><transformations/> ></rule> ><cp:privacy-lists> ><cp:list name="whitelist_uri_phone"> ></cp:list> ><cp:list name="blacklist_uri_phone"> ></cp:list> ><cp:list name="whitelist_domain_phone"> ></cp:list> ><cp:list name="blacklist_domain_phone"> ></cp:list> ><cp:list name="whitelist_uri_IM"> ></cp:list> ><cp:list name="blacklist_uri_IM"> ></cp:list> ><cp:list name="whitelist_domain_IM"> ></cp:list> ><cp:list name="blacklist_domain_IM"> ></cp:list> ></cp:privacy-lists> ></ruleset> 0(4591) presence:uandd_to_uri: uri=sip:bob@10.96.115.125 > 0(4591) PRESENCE:is_watcher_allowed:node1->name= rule > 0(4591) PRESENCE:is_watcher_allowed:cond_node->name= conditions > 0(4591) PRESENCE:is_watcher_allowed:actions_node->name= actions > 0(4591) PRESENCE:is_watcher_allowed:sub_handling_node->name= sub-handling > 0(4591) PRESENCE:is_watcher_allowed:sub_handling_node->content= allow > 0(4591) PRESENCE:notify: [status]=active > 0(4591) PRESENCE:get_p_notify_body: querying presentity > 0(4591) PRESENCE: get_p_notify_body: The query returned no result >There is no presence information recorded forthe user > 0(4591) PRESENCE:notify: Could not get the notify_body > 0(4591) presence:uandd_to_uri: uri=sip:sebastian@10.96.115.125 > 0(4591) PRESENCE: notify: build notify to user= bob domain= 10.96.115.125 for event= presence > 0(4591) > [p_user]= sebastian [p_domain]= 10.96.115.125 > [w_user]= bob [w_domain]= 10.96.115.125 > 0(4591) [event]= presence > [staus]= active > [expires]= 3600 > 0(4591) [to_tag]= 10.4591.1179421870.7 > [from_tag]= e035dc2e > 0(4591) PRESENCE:build_str_hdr: expires = 3600 > 0(4591) PRESENCE:build_str_hdr: subs_expires : 3600 > 0(4591) PRESENCE: build_str_hdr: headers: >Event: presence >Contact: <sip:10.96.115.128:5060> >Subscription-State: active;expires=3600 >Content-Type: application/pidf+xml > > 0(4591) PRESENCE:notify: headers:Event: presence >Contact: <sip:10.96.115.128:5060> >Subscription-State: active;expires=3600 >Content-Type: application/pidf+xml > > 0(4591) CONTACT = sip:bob@10.96.100.128:32800 > 0(4591) presence:uandd_to_uri: uri=sip:bob@10.96.115.125 > 0(4591) DEBUG:tm:t_uac: next_hop=<sip:10.96.115.125;lr=on;ftag=e035dc2e> > 0(4591) DEBUG: mk_proxy: doing DNS lookup... > 0(4591) DEBUG:destroy_avp_list: destroying list (nil) > 0(4591) DEBUG: dlg2hash: 12619 > 0(4591) print_request_uri: sip:bob@10.96.100.128:32800 > 0(4591) DEBUG:tm:set_timer: relative timeout is 500000 > 0(4591) DEBUG: add_to_tail_of_timer[4]: 0xb5aaefec (41700000) > 0(4591) DEBUG:tm:set_timer: relative timeout is 30 > 0(4591) DEBUG: add_to_tail_of_timer[0]: 0xb5aaf008 (71) > 1(4592) DEBUG: timer routine:4,tl=0xb5aa825c next=0xb5aaefec, timeout=41700000 > 1(4592) DEBUG: retransmission_handler : request resending (t=0xb5aa8110, NOTIFY si ... ) > 1(4592) DEBUG:tm:set_timer: relative timeout is 1000000 > 1(4592) DEBUG: add_to_tail_of_timer[5]: 0xb5aa825c (42700000) > 1(4592) DEBUG: retransmission_handler : done > 1(4592) DEBUG: timer routine:4,tl=0xb5aaefec next=(nil), timeout=41700000 > 1(4592) DEBUG: retransmission_handler : request resending (t=0xb5aaeea0, NOTIFY si ... ) > 1(4592) DEBUG:tm:set_timer: relative timeout is 1000000 > 1(4592) DEBUG: add_to_tail_of_timer[5]: 0xb5aaefec (42700000) > 1(4592) DEBUG: retransmission_handler : done > 1(4592) DEBUG: timer routine:5,tl=0xb5aa825c next=0xb5aaefec, timeout=42700000 > 1(4592) DEBUG: retransmission_handler : request resending (t=0xb5aa8110, NOTIFY si ... ) > 1(4592) DEBUG:tm:set_timer: relative timeout is 2000000 > 1(4592) DEBUG: add_to_tail_of_timer[6]: 0xb5aa825c (44700000) > 1(4592) DEBUG: retransmission_handler : done > 1(4592) DEBUG: timer routine:5,tl=0xb5aaefec next=(nil), timeout=42700000 > 1(4592) DEBUG: retransmission_handler : request resending (t=0xb5aaeea0, NOTIFY si ... ) > 1(4592) DEBUG:tm:set_timer: relative timeout is 2000000 > 1(4592) DEBUG: add_to_tail_of_timer[6]: 0xb5aaefec (44700000) > 1(4592) DEBUG: retransmission_handler : done > 1(4592) DEBUG: timer routine:6,tl=0xb5aa825c next=0xb5aaefec, timeout=44700000 > 1(4592) DEBUG: retransmission_handler : request resending (t=0xb5aa8110, NOTIFY si ... ) > 1(4592) DEBUG:tm:set_timer: relative timeout is 4000000 > 1(4592) DEBUG: add_to_tail_of_timer[7]: 0xb5aa825c (48700000) > 1(4592) DEBUG: retransmission_handler : done > 1(4592) DEBUG: timer routine:6,tl=0xb5aaefec next=(nil), timeout=44700000 > 1(4592) DEBUG: retransmission_handler : request resending (t=0xb5aaeea0, NOTIFY si ... ) > 1(4592) DEBUG:tm:set_timer: relative timeout is 4000000 > 1(4592) DEBUG: add_to_tail_of_timer[7]: 0xb5aaefec (48700000) > 1(4592) DEBUG: retransmission_handler : done > 1(4592) DEBUG: timer routine:2,tl=0xb5aa9bc8 next=(nil), timeout=46 > 1(4592) DEBUG: wait_handler : removing 0xb5aa9b80 from table > 1(4592) DEBUG: delete transaction 0xb5aa9b80 > 1(4592) DEBUG: wait_handler : done > 1(4592) DEBUG: timer routine:7,tl=0xb5aa825c next=0xb5aaefec, timeout=48700000 > 1(4592) DEBUG: retransmission_handler : request resending (t=0xb5aa8110, NOTIFY si ... ) > 1(4592) DEBUG:tm:set_timer: relative timeout is 4000000 > 1(4592) DEBUG: add_to_tail_of_timer[7]: 0xb5aa825c (52700000) > 1(4592) DEBUG: retransmission_handler : done > 1(4592) DEBUG: timer routine:7,tl=0xb5aaefec next=(nil), timeout=48700000 > 1(4592) DEBUG: retransmission_handler : request resending (t=0xb5aaeea0, NOTIFY si ... ) > 1(4592) DEBUG:tm:set_timer: relative timeout is 4000000 > 1(4592) DEBUG: add_to_tail_of_timer[7]: 0xb5aaefec (52700000) > 1(4592) DEBUG: retransmission_handler : done > 1(4592) DEBUG: timer routine:7,tl=0xb5aa825c next=0xb5aaefec, timeout=52700000 > 1(4592) DEBUG: retransmission_handler : request resending (t=0xb5aa8110, NOTIFY si ... ) > 1(4592) DEBUG:tm:set_timer: relative timeout is 4000000 > 1(4592) DEBUG: add_to_tail_of_timer[7]: 0xb5aa825c (56700000) > 1(4592) DEBUG: retransmission_handler : done > 1(4592) DEBUG: timer routine:7,tl=0xb5aaefec next=(nil), timeout=52700000 > 1(4592) DEBUG: retransmission_handler : request resending (t=0xb5aaeea0, NOTIFY si ... ) > 1(4592) DEBUG:tm:set_timer: relative timeout is 4000000 > 1(4592) DEBUG: add_to_tail_of_timer[7]: 0xb5aaefec (56700000) > 1(4592) DEBUG: retransmission_handler : done > 1(4592) DEBUG: timer routine:7,tl=0xb5aa825c next=0xb5aaefec, timeout=56700000 > 1(4592) DEBUG: retransmission_handler : request resending (t=0xb5aa8110, NOTIFY si ... ) > 1(4592) DEBUG:tm:set_timer: relative timeout is 4000000 > 1(4592) DEBUG: add_to_tail_of_timer[7]: 0xb5aa825c (60700000) > 1(4592) DEBUG: retransmission_handler : done > 1(4592) DEBUG: timer routine:7,tl=0xb5aaefec next=(nil), timeout=56700000 > 1(4592) DEBUG: retransmission_handler : request resending (t=0xb5aaeea0, NOTIFY si ... ) > 1(4592) DEBUG:tm:set_timer: relative timeout is 4000000 > 1(4592) DEBUG: add_to_tail_of_timer[7]: 0xb5aaefec (60700000) > 1(4592) DEBUG: retransmission_handler : done > 1(4592) DEBUG: timer routine:7,tl=0xb5aa825c next=0xb5aaefec, timeout=60700000 > 1(4592) DEBUG: retransmission_handler : request resending (t=0xb5aa8110, NOTIFY si ... ) > 1(4592) DEBUG:tm:set_timer: relative timeout is 4000000 > 1(4592) DEBUG: add_to_tail_of_timer[7]: 0xb5aa825c (64700000) > 1(4592) DEBUG: retransmission_handler : done > 1(4592) DEBUG: timer routine:7,tl=0xb5aaefec next=(nil), timeout=60700000 > 1(4592) DEBUG: retransmission_handler : request resending (t=0xb5aaeea0, NOTIFY si ... ) > 1(4592) DEBUG:tm:set_timer: relative timeout is 4000000 > 1(4592) DEBUG: add_to_tail_of_timer[7]: 0xb5aaefec (64700000) > 1(4592) DEBUG: retransmission_handler : done > 1(4592) DEBUG: timer routine:7,tl=0xb5aa825c next=0xb5aaefec, timeout=64700000 > 1(4592) DEBUG: retransmission_handler : request resending (t=0xb5aa8110, NOTIFY si ... ) > 1(4592) DEBUG:tm:set_timer: relative timeout is 4000000 > 1(4592) DEBUG: add_to_tail_of_timer[7]: 0xb5aa825c (68700000) > 1(4592) DEBUG: retransmission_handler : done > 1(4592) DEBUG: timer routine:7,tl=0xb5aaefec next=(nil), timeout=64700000 > 1(4592) DEBUG: retransmission_handler : request resending (t=0xb5aaeea0, NOTIFY si ... ) > 1(4592) DEBUG:tm:set_timer: relative timeout is 4000000 > 1(4592) DEBUG: add_to_tail_of_timer[7]: 0xb5aaefec (68700000) > 1(4592) DEBUG: retransmission_handler : done > 1(4592) DEBUG: timer routine:7,tl=0xb5aa825c next=0xb5aaefec, timeout=68700000 > 1(4592) DEBUG: retransmission_handler : request resending (t=0xb5aa8110, NOTIFY si ... ) > 1(4592) DEBUG:tm:set_timer: relative timeout is 4000000 > 1(4592) DEBUG: add_to_tail_of_timer[7]: 0xb5aa825c (72700000) > 1(4592) DEBUG: retransmission_handler : done > 1(4592) DEBUG: timer routine:7,tl=0xb5aaefec next=(nil), timeout=68700000 > 1(4592) DEBUG: retransmission_handler : request resending (t=0xb5aaeea0, NOTIFY si ... ) > 1(4592) DEBUG:tm:set_timer: relative timeout is 4000000 > 1(4592) DEBUG: add_to_tail_of_timer[7]: 0xb5aaefec (72700000) > 1(4592) DEBUG: retransmission_handler : done > 1(4592) DEBUG: timer routine:0,tl=0xb5aa8278 next=0xb5aaf008, timeout=71 > 1(4592) DEBUG: final_response_handler:stop retr. and send CANCEL (0xb5aa8110) > 1(4592) DEBUG:tm:t_should_relay_response: T_code=0, new_code=408 > 1(4592) DEBUG:tm:t_pick_branch: picked branch 0, code 408 > 1(4592) DEBUG:tm:local_reply: branch=0, save=0, winner=0 > 1(4592) DEBUG:tm:local_reply: local transaction completed > 1(4592) DBG: trans=0xb5aa8110, callback type 256, id 0 entered > 1(4592) PRESENCE:p_tm_callback: completed with status 408 [watcher_id:0xb5aaa9cc/10.4591.1179421870.4] > 1(4592) DEBUG: add_to_tail_of_timer[2]: 0xb5aa8158 (76) > 1(4592) DEBUG: final_response_handler : done > 1(4592) DEBUG: timer routine:0,tl=0xb5aaf008 next=(nil), timeout=71 > 1(4592) DEBUG: final_response_handler:stop retr. and send CANCEL (0xb5aaeea0) > 1(4592) DEBUG:tm:t_should_relay_response: T_code=0, new_code=408 > 1(4592) DEBUG:tm:t_pick_branch: picked branch 0, code 408 > 1(4592) DEBUG:tm:local_reply: branch=0, save=0, winner=0 > 1(4592) DEBUG:tm:local_reply: local transaction completed > 1(4592) DBG: trans=0xb5aaeea0, callback type 256, id 0 entered > 1(4592) PRESENCE:p_tm_callback: completed with status 408 [watcher_id:0xb5aaacf4/10.4591.1179421870.7] > 1(4592) DEBUG: add_to_tail_of_timer[2]: 0xb5aaeee8 (76) > 1(4592) DEBUG: final_response_handler : done > 1(4592) DEBUG: timer routine:7,tl=0xb5aa825c next=0xb5aaefec, timeout=72700000 > 1(4592) DEBUG: timer routine:7,tl=0xb5aaefec next=(nil), timeout=72700000 > 1(4592) DEBUG: timer routine:2,tl=0xb5aa8158 next=0xb5aaeee8, timeout=76 > 1(4592) DEBUG: wait_handler : removing 0xb5aa8110 from table > 1(4592) DEBUG: delete transaction 0xb5aa8110 > 1(4592) DEBUG: wait_handler : done > 1(4592) DEBUG: timer routine:2,tl=0xb5aaeee8 next=(nil), timeout=76 > 1(4592) DEBUG: wait_handler : removing 0xb5aaeea0 from table > 1(4592) DEBUG: delete transaction 0xb5aaeea0 > 1(4592) DEBUG: wait_handler : done > 1(4592) PRESENCE: msg_watchers_clean:cleaning pending subscriptions > 1(4592) PRESENCE: msg_active_watchers_clean:cleaning expired watcher information > 1(4592) PRESENCE:msg_presentity_clean:cleaning expired presentity information > 1(4592) PRESENCE: msg_watchers_clean:cleaning pending subscriptions > 1(4592) PRESENCE: msg_active_watchers_clean:cleaning expired watcher information > 1(4592) PRESENCE:msg_presentity_clean:cleaning expired presentity information > > >----------------------------------------------------------------------- - > > >... > >(no debugging symbols found) >Core was generated by `openser -f openser.cfg -w .'. >Program terminated with signal 11, Segmentation fault. >#0 0x0809e332 in fm_free () >(gdb) bt >#0 0x0809e332 in fm_free () >#1 0xb7b7d67a in handle_subscribe () from /usr/lib/openser/modules/presence.so >#2 0x08051224 in do_action () >#3 0x08052f8c in run_action_list () >#4 0x080521af in do_action () >#5 0x08052f8c in run_action_list () >#6 0x080525a1 in do_action () >#7 0x08052f8c in run_action_list () >#8 0x08053247 in run_top_route () >#9 0x08074e04 in receive_msg () >#10 0x0809832d in udp_rcv_loop () >#11 0x080664f4 in main_loop () >#12 0x08067f25 in main () >(gdb) quit > >

3 2

AW: [Users] OpenSER crashed with Segmentation Fault using PresenceModule
by Schumann Sebastian 17 May '07

17 May '07

Hello, here is the stderr console output (just the last lines which should be important as I saw there is a message size limit), and the gdb backtrace. Do you see any hint for the crash? Best regards Sebastian ________________________________ Von: Anca-Maria Vamanu [mailto:anca@voice-system.ro] Gesendet: Do 5/17/2007 16:57 An: Schumann Sebastian Cc: users(a)openser.org Betreff: Re: [Users] OpenSER crashed with Segmentation Fault using PresenceModule Hello, Please post the log and the gdb backtrace. regards, Anca Vamanu Schumann Sebastian wrote: >Dear all > >I am currently writing my final thesis and want to set up a Presence Core therefore. I am using OpenSER as Presence Agent (with Registrar function or now with separat Registrar, both configuration has the problem). It crashes after a certain time and certain amount of watcher entries. I cannot replicate this or deliminate to a certain action but it always happends, no matter if I clean up the tables, restart the machine or whatever. > >My configuration is one XCAP Server (XCAP Lite 1.0 from ag-projects, working properly), one Registrar (OpenSER with MySQL DB, which forwards SUBSCRIBE/PUBLISH to PA) and Presence Agent (OpenSER with Presence Module, MySQL DB, XMLRPC Module) and Eyebeam 1.5 Clients. All runs on VMware virtual machines. > >I have just three clients that try to subscribe each other. Without any Subscriptions everything works properly, XML data gets stored, PUBLISH is processed etc. Subscribing one user by another is possible and works properly. > >After allowing a watcher to subscription (using XCAP) and if the watcher logs in next time, the state in the watcher table changes from pending to active. It happens now and then, when a user goes online and one of his subscribed presentities is not online, that the OpenSER PA crashes. > >Analyzing DEBUG info and backtrace was not successful, because I found no information or hints about it, just the message segmentation fault appears. > >Does anyone have this problem as well or has anybody an idea about it? I can provide you logs, the dump or traces if you need it. > >Thanks for your help. > >Sebastian > > >

2 1

[Users] OpenSER crashed with Segmentation Fault using Presence Module
by Schumann Sebastian 17 May '07

17 May '07

Dear all I am currently writing my final thesis and want to set up a Presence Core therefore. I am using OpenSER as Presence Agent (with Registrar function or now with separat Registrar, both configuration has the problem). It crashes after a certain time and certain amount of watcher entries. I cannot replicate this or deliminate to a certain action but it always happends, no matter if I clean up the tables, restart the machine or whatever. My configuration is one XCAP Server (XCAP Lite 1.0 from ag-projects, working properly), one Registrar (OpenSER with MySQL DB, which forwards SUBSCRIBE/PUBLISH to PA) and Presence Agent (OpenSER with Presence Module, MySQL DB, XMLRPC Module) and Eyebeam 1.5 Clients. All runs on VMware virtual machines. I have just three clients that try to subscribe each other. Without any Subscriptions everything works properly, XML data gets stored, PUBLISH is processed etc. Subscribing one user by another is possible and works properly. After allowing a watcher to subscription (using XCAP) and if the watcher logs in next time, the state in the watcher table changes from pending to active. It happens now and then, when a user goes online and one of his subscribed presentities is not online, that the OpenSER PA crashes. Analyzing DEBUG info and backtrace was not successful, because I found no information or hints about it, just the message segmentation fault appears. Does anyone have this problem as well or has anybody an idea about it? I can provide you logs, the dump or traces if you need it. Thanks for your help. Sebastian -- Sebastian Schumann Diploma Student Architecture and Design Department Slovak Telekom, a. s. T-Com, Operations Unit Production and Service Division Innovations and Enterprise Solution Subunit Address: Namestie Slobody 6, 817 62 Bratislava, Slovakia Office: Room 232, Jarabinkova 1, 821 09 Bratislava, Slovakia +421 910 643010 (mobile) sebastian.schumann(a)t-com.sk http://www.t-com.sk

2 1

[Users] failure_route
by Yago Bargalló 17 May '07

17 May '07

Hello all, I don't know why my openser.cfg doesn't work correctly, I want my openser to catch the failure_route after 10 seconds to reroute a call to a Asterisk when nobody responds the phone on an openser. Does anybody know what I'm doing wrong?? Thanks in advance!!!!!!!!! Openser.cfg: modparam("tm","fr_timer",4) modparam("tm", "fr_inv_timer" ,10) # ------------------------- request routing logic ------------------- # main routing logic (...) route{ (...) # 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("172.16.51.150", "subscriber")) { www_challenge("172.16.51.150", "0"); exit; }; save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # if(uri=~"sip:2[0-9]+@") { t_on_failure("1"); # rewritehostport("172.16.51.160:5060"); # t_relay(); # rewriteuser("2002"); # exit; } # 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; } failure_route[1] { #reroute to asterisk rewritehost("172.16.51.150"); rewriteuser("2002"); t_relay(); exit; }

2 1

[Serusers] 407 Proxy Authentication Required to a subscribed user
by flavio 17 May '07

17 May '07

Hi all, I'm testing Blind Trasfer with my IP Phones, SER and GW, in this scenario: AnalogPhone (GW) is the Transferee IP Phone1, registred to SER, is the Transferor IP Phone 2, registred to SER, is the Transfer Target When the Transferor send INVITE to put on hold the Trasferee, SER reply with 407 Proxy Authentication Requrire, as follow reported: 2007/05/17 12:45:49.370808 10.28.19.124:5060 -> 10.28.19.202:5060 INVITE sip:06XXXXXX15@10.28.19.230 SIP/2.0. Via: SIP/2.0/UDP 10.28.19.124;branch=z9hG4bKc442076492fe9954. Route: <sip:10.28.19.202;ftag=1c1272914830;lr=on>. From: <sip:06XXXXXX14@10.28.19.202>;tag=9bb7a2279efa8707. To: <sip:06XXXXXX15@10.28.19.230>;tag=1c1272914830. Contact: <sip:06XXXXXX14@10.28.19.124;user=phone>. Supported: replaces. Call-ID: 127291448221200023114(a)10.28.19.230. CSeq: 21593 INVITE. User-Agent: Grandstream BT110 1.0.8.12. Max-Forwards: 70. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE. Content-Type: application/sdp. Content-Length: 206. . v=0. o=0660522014 8000 8001 IN IP4 10.28.19.124. s=SIP Call. c=IN IP4 0.0.0.0. t=0 0. m=audio 5004 RTP/AVP 8 18 0. a=sendonly. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=rtpmap:0 PCMU/8000. a=ptime:20. # U 2007/05/17 12:45:49.371204 10.28.19.202:5060 -> 10.28.19.124:5060 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 10.28.19.124;branch=z9hG4bKc442076492fe9954. From: <sip:06XXXXXX14@10.28.19.202>;tag=9bb7a2279efa8707. To: <sip:06XXXXXX15@10.28.19.230>;tag=1c1272914830. Call-ID: 127291448221200023114(a)10.28.19.230. CSeq: 21593 INVITE. Proxy-Authenticate: Digest realm="10.28.19.202", nonce="464c4fa95f251088d01666190944e53313bab22e". Server: Sip EXpress router (0.9.6 (i386/linux)). Content-Length: 0. Warning: 392 10.28.19.202:5060 "Noisy feedback tells: pid=15940 req_src_ip=10.28.19.124 req_src_port=5060 in_uri=sip:0660522015@10.28.19.230 out_uri=sip:0660522015@10.28.19.230 via_cnt==1". My question is: why SER reply with an authentication request to INVITE, if the Transferor party is a registred user? Have you any suggestion about? I verify user subscribtion in my ser.cfg file as follow: if (!allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); } Thanks for support, Flavio -- ******************************** * (o< ing. Patria Flavio * //\ phone 0823451358 * V_/_ mobile 3407873357 * ********************************

2 4

[Serusers] looped requests, syn_branch==1 => branch value is repeated for 2 Ts
by JF 17 May '07

17 May '07

Hi all, I have observed when looping a request through SER (using t_relay_to("127.0.0.1", 5060)) that sometimes the via branch parameter value generated when the request goes out is kept equal to the value in the previous via (when this request was looped), and two equal vias go out. This causes problems with on_reply routes. On_reply routes are then called only once, instead of two. Since these are 2 independent transactions, this should happen twice. Browsing through tm code, it seems that this is caused by the usage of syn_branch == 1 in main.c (the default), leading to the following function calls when an initial INVITE loops: tm/t_fwd.c: t_forward_nonack add_uac print_uac_request tm/t_msgbuilder.c: t_calc_branch msg_translator.c: branch_builder <-- here, when syn_branch==1, the transaction 'label' value is used for the branch value, but according to tm/h_table.h, this value is "sequence number within hash collision slot". Is this the reason why some times the branch value is the same in two transactions looped through SER? The label value can perfectly be the same across 2 transactions not in the same hash slot... right? Maybe this has nothing to do with the request being looped, or it just makes this situation more probable? Anyone else can replicate this behaviour? Thanks, João Filipe Plácido -- PT Inovação

1 0

[Users] No answer when my openser 1.2.0 make a call to asterisk
by Charles Wang 17 May '07

17 May '07

Hi all, I try to make a call from my Openser 1.2.0 (SIP Proxy) to the asterisk in different machine. I use my asterisk as a trunking gateway. I can make a call from my openser to some trunking gateway such as my cisco 5300 or welltech 5250. In the same method, I try to make a call to asterisk ( sip listen on udp 5060 ) I use ngrep on my asterisk machine and list as below. But I can't find any sip debug in my asterisk CLI. In the message on my openser, I can see as below: openser[16801]: route[11] Base Outbound - M=INVITE RURI= sip:02123456789@asterisk.ip.addr:5060;transport=udp F= sip:101@openser.domain.name T=sip:00862123456789@openser.domain.name IP= 61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 But the To URI is still 00862123456789(a)openser.domain.name not 02123456789(a)asterisk.ip.addr. Is it correct? Does anybody kind to help me to solve it or give me some tips please? Best regards, Charles ##################### message on my Openser ################################# openser[16802]: New request - M=INVITE RURI= sip:00862123456789@openser.domain.name F=sip:101@openser.domain.name T= sip:00862123456789@openser.domain.name IP=61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 openser[16802]: route[12] Base Route Invite - M=INVITE RURI= sip:00862123456789@openser.domain.name F=sip:101@openser.domain.name T= sip:00862123456789@openser.domain.name IP=61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 openser[16802]: Proxy authentication failed - M=INVITE RURI= sip:00862123456789@openser.domain.name F=sip:101@openser.domain.name T= sip:00862123456789@openser.domain.name IP=61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 openser[16801]: New request - M=INVITE RURI= sip:00862123456789@openser.domain.name F=sip:101@openser.domain.name T= sip:00862123456789@openser.domain.name IP=61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 openser[16801]: route[12] Base Route Invite - M=INVITE RURI= sip:00862123456789@openser.domain.name F=sip:101@openser.domain.name T= sip:00862123456789@openser.domain.name IP=61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 openser[16801]: route[3] Usr Preferences Caller - M=INVITE RURI= sip:00862123456789@openser.domain.name F=sip:101@openser.domain.name T= sip:00862123456789@openser.domain.name IP=61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 openser[16801]: route[1] Clear Usr Preferences Caller - M=INVITE RURI= sip:00862123456789@openser.domain.name F=sip:101@openser.domain.name T= sip:00862123456789@openser.domain.name IP=61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 openser[16801]: Clear caller preferences - M=INVITE RURI= sip:00862123456789@openser.domain.name F=sip:101@openser.domain.name T= sip:00862123456789@openser.domain.name IP=61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 openser[16801]: Load caller preferences for uuid '1' - M=INVITE RURI= sip:00862123456789@openser.domain.name F=sip:101@openser.domain.name T= sip:00862123456789@openser.domain.name IP=61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 openser[16801]: route[13] Invite Find Callee - M=INVITE RURI= sip:00862123456789@openser.domain.name F=sip:101@openser.domain.name T= sip:00862123456789@openser.domain.name IP=61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 openser[16801]: route[2] Clear Usr Preferences Callee - M=INVITE RURI= sip:00862123456789@openser.domain.name F=sip:101@openser.domain.name T= sip:00862123456789@openser.domain.name IP=61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 openser[16801]: Clear callee preferences - M=INVITE RURI= sip:00862123456789@openser.domain.name F=sip:101@openser.domain.name T= sip:00862123456789@openser.domain.name IP=61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 openser[16801]: Callee is not local - M=INVITE RURI= sip:00862123456789@openser.domain.name F=sip:101@openser.domain.name T= sip:00862123456789@openser.domain.name IP=61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 openser[16801]: route[15] Invite To External - M=INVITE RURI= sip:00862123456789@openser.domain.name F=sip:101@openser.domain.name T= sip:00862123456789@openser.domain.name IP=61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 openser[16801]: Loading PSTN gateways - M=INVITE RURI= sip:00862123456789@openser.domain.name F=sip:101@openser.domain.name T= sip:00862123456789@openser.domain.name IP=61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 openser[16801]: route[11] Base Outbound - M=INVITE RURI= sip:02123456789@asterisk.ip.addr:5060;transport=udp F= sip:101@openser.domain.name T=sip:00862123456789@openser.domain.name IP= 61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 openser[16801]: route[8] Clir - M=INVITE RURI= sip:02123456789@asterisk.ip.addr:5060;transport=udp F= sip:101@openser.domain.name T=sip:00862123456789@openser.domain.name IP= 61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 localhost mediaproxy[1905]: command request 25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 192.168.11.9:57538:audio 61.217.xxx.xxx openser.domain.name local asterisk.ip.addr remote X-PRO=20build=201082 info=from:101@openser.domain.name, to:00862123456789@openser.domain.name,fromtag:1111281943,totag: localhost mediaproxy[1905]: session 25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9: started. listening on 60.250.132.204:35076 localhost mediaproxy[1905]: command execution time: 3.04 ms openser[16801]: Request leaving server - M=INVITE RURI= sip:02123456789@asterisk.ip.addr:5060;transport=udp F= sip:101@openser.domain.name T=sip:00862123456789@openser.domain.name IP= 61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 ################# my asterisk CLI ######################## [root@asterisk ~]# asterisk -rvvvvvvvvvvvvvv == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found == Binding iaxusers to mysql/asterisk/iaxfriends == Binding iaxpeers to mysql/asterisk/iaxfriends == Binding queues to mysql/asterisk/queue_table == Binding queue_members to mysql/asterisk/queue_member_table Asterisk 1.2.16, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer <markster(a)digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'show license' for details. ========================================================================= Connected to Asterisk 1.2.16 currently running on asterisk (pid = 26311) Verbosity is at least 14 -- Remote UNIX connection asterisk*CLI> sip debug SIP Debugging re-enabled asterisk*CLI> ######### my command running on asterisk machine: "ngrep -t -W byline -d any port 5060" #################### interface: any filter: (ip) and ( port 5060 ) # U 2007/05/17 13:31:35.908163 my.openser.ip.addr:5060 -> my.asterisk.ip.addr :5060 INVITE sip:03939749001@my.asterisk.ip.addr:5060;transport=udp SIP/2.0. Record-Route: <sip:my.openser.ip.addr;lr;ftag=3840196923;nat=yes>. Via: SIP/2.0/UDP my.openser.ip.addr;branch=z9hG4bKab01.ae06a6e5.0. Via: SIP/2.0/UDP 192.168.11.9:57536 ;received=61.217.xxx.xxx ;rport=57536;branch=z9hG4bK834BA777F3C7439EBBC7C4DAECC52FD4. From: 101 <sip:101@my.openser.domain.name>;tag=3840196923. To: < sip:0028863939749001@my.openser.domain.name>. Contact: <sip:101@61.217.xxx.xxx:57536>. Call-ID: 18875244-8D15-416B-92B7-DD24DA4630E2(a)192.168.11.9 . CSeq: 4807 INVITE. Max-Forwards: 69. Content-Type: application/sdp. User-Agent: X-LITE build 1082. Content-Length: 321. . v=0. o=101 45727796 45727796 IN IP4 192.168.11.9 . s=X-LITE. c=IN IP4 my.openser.ip.addr. t=0 0. m=audio 35066 RTP/AVP 0 8 3 18 98 97 101. a=rtpmap:0 pcmu/8000. a=rtpmap:8 pcma/8000. a=rtpmap:3 gsm/8000. a=rtpmap:18 G729/8000. a=rtpmap:98 iLBC/8000. a=rtpmap:97 speex/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 2007/05/17 13:31:36.325713 my.openser.ip.addr:5060 -> my.asterisk.ip.addr :5060 INVITE sip:03939749001@my.asterisk.ip.addr:5060;transport=udp SIP/2.0. Record-Route: <sip:my.openser.ip.addr;lr;ftag=3840196923;nat=yes>. Via: SIP/2.0/UDP my.openser.ip.addr;branch=z9hG4bKab01.ae06a6e5.0. Via: SIP/2.0/UDP 192.168.11.9:57536;received=61.217.xxx.xxx;rport=57536;branch=z9hG4bK834BA777F3C7439EBBC7C4DAECC52FD4. From: 101 <sip:101@my.openser.domain.name>;tag=3840196923. To: <sip:0028863939749001@my.openser.domain.name >. Contact: <sip:101@61.217.xxx.xxx:57536>. Call-ID: 18875244-8D15-416B-92B7-DD24DA4630E2(a)192.168.11.9. CSeq: 4807 INVITE. Max-Forwards: 69. Content-Type: application/sdp. User-Agent: X-LITE build 1082. Content-Length: 321. . v=0. o=101 45727796 45727796 IN IP4 192.168.11.9. s=X-LITE. c=IN IP4 my.openser.ip.addr . t=0 0. m=audio 35066 RTP/AVP 0 8 3 18 98 97 101. a=rtpmap:0 pcmu/8000. a=rtpmap:8 pcma/8000. a=rtpmap:3 gsm/8000. a=rtpmap:18 G729/8000. a=rtpmap:98 iLBC/8000. a=rtpmap:97 speex/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 2007/05/17 13:31:37.325722 my.openser.ip.addr:5060 -> my.asterisk.ip.addr :5060 INVITE sip:03939749001@my.asterisk.ip.addr:5060;transport=udp SIP/2.0. Record-Route: <sip:my.openser.ip.addr;lr;ftag=3840196923;nat=yes>. Via: SIP/2.0/UDP my.openser.ip.addr;branch=z9hG4bKab01.ae06a6e5.0. Via: SIP/2.0/UDP 192.168.11.9:57536;received=61.217.xxx.xxx;rport=57536;branch=z9hG4bK834BA777F3C7439EBBC7C4DAECC52FD4. From: 101 <sip:101@my.openser.domain.name>;tag=3840196923. To: <sip:0028863939749001@my.openser.domain.name >. Contact: <sip:101@61.217.xxx.xxx:57536>. Call-ID: 18875244-8D15-416B-92B7-DD24DA4630E2(a)192.168.11.9. CSeq: 4807 INVITE. Max-Forwards: 69. Content-Type: application/sdp. User-Agent: X-LITE build 1082. Content-Length: 321. . v=0. o=101 45727796 45727796 IN IP4 192.168.11.9. s=X-LITE. c=IN IP4 my.openser.ip.addr . t=0 0. m=audio 35066 RTP/AVP 0 8 3 18 98 97 101. a=rtpmap:0 pcmu/8000. a=rtpmap:8 pcma/8000. a=rtpmap:3 gsm/8000. a=rtpmap:18 G729/8000. a=rtpmap:98 iLBC/8000. a=rtpmap:97 speex/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 2007/05/17 13:31:39.325425 my.openser.ip.addr:5060 -> my.asterisk.ip.addr :5060 INVITE sip:03939749001@my.asterisk.ip.addr:5060;transport=udp SIP/2.0. Record-Route: <sip:my.openser.ip.addr;lr;ftag=3840196923;nat=yes>. Via: SIP/2.0/UDP my.openser.ip.addr;branch=z9hG4bKab01.ae06a6e5.0. Via: SIP/2.0/UDP 192.168.11.9:57536;received=61.217.xxx.xxx;rport=57536;branch=z9hG4bK834BA777F3C7439EBBC7C4DAECC52FD4. From: 101 <sip:101@my.openser.domain.name>;tag=3840196923. To: <sip:0028863939749001@my.openser.domain.name >. Contact: <sip:101@61.217.xxx.xxx:57536>. Call-ID: 18875244-8D15-416B-92B7-DD24DA4630E2(a)192.168.11.9. CSeq: 4807 INVITE. Max-Forwards: 69. Content-Type: application/sdp. User-Agent: X-LITE build 1082. Content-Length: 321. . v=0. o=101 45727796 45727796 IN IP4 192.168.11.9. s=X-LITE. c=IN IP4 my.openser.ip.addr . t=0 0. m=audio 35066 RTP/AVP 0 8 3 18 98 97 101. a=rtpmap:0 pcmu/8000. a=rtpmap:8 pcma/8000. a=rtpmap:3 gsm/8000. a=rtpmap:18 G729/8000. a=rtpmap:98 iLBC/8000. a=rtpmap:97 speex/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. -- Best Regards Charles -- Best Regards Charles

2 2

[Serusers] Problem running presence modules - SER won't start
by Mahatma Sarasvati 17 May '07

17 May '07

Hello All, Running SER 2.0.0 rc1 from CVS (updated yesterday) on CentOS 5. I'm trying to test/use the presence functionality in SER. I've compiled and installed with make group_include="standard standard-dep" all I've set the environment variable: export LD_LIBRARY_PATH=/usr/local/lib/ser [root@vx1 modules]# echo $LD_LIBRARY_PATH /usr/local/lib/ser My supporting modules are here: [root@vx1 modules]# ls -l /usr/local/lib/ser/ total 828 -rwxr-xr-x 1 root root 40768 May 14 15:19 libbinrpc.so -rwxr-xr-x 1 root root 179044 May 14 15:19 lib_ser_cds.so -rwxr-xr-x 1 root root 213028 May 14 15:19 lib_ser_presence.so -rwxr-xr-x 1 root root 371056 May 14 15:19 lib_ser_xcap.so drwxr-xr-x 2 root root 4096 May 14 15:19 modules [root@vx1 modules]# Load module commands are in ser.cfg: loadmodule "/usr/local/lib/ser/modules/xmlrpc.so" loadmodule "/usr/local/lib/ser/modules/xcap.so" loadmodule "/usr/local/lib/ser/modules/pa.so" loadmodule "/usr/local/lib/ser/modules/presence_b2b.so" loadmodule "/usr/local/lib/ser/modules/rls.so" loadmodule "/usr/local/lib/ser/modules/dialog.so" libXML2 and libXML2-devel are installed. Yet, when I try to start SER I get this: May 15 13:09:24 vx1 ser: ERROR: load_module: could not open module </usr/local/lib/ser/modules/pa.so>: lib_ser_xcap.so: cannot open shared object file: No such file or directory May 15 13:09:24 vx1 ser: parse error (81,13-14): failed to load module May 15 13:09:24 vx1 ser: ERROR: load_module: could not open module </usr/local/lib/ser/modules/presence_b2b.so>: lib_ser_xcap.so: cannot open shared object file: No such file or directory May 15 13:09:24 vx1 ser: parse error (82,13-14): failed to load module May 15 13:09:24 vx1 ser: ERROR: load_module: could not open module </usr/local/lib/ser/modules/rls.so>: lib_ser_xcap.so: cannot open shared object file: No such file or directory May 15 13:09:24 vx1 ser: parse error (83,13-14): failed to load module May 15 13:09:24 vx1 ser: ERROR: load_module: could not open module </usr/local/lib/ser/modules/dialog.so>: lib_ser_cds.so: cannot open shared object file: No such file or directory Can anyone help? I think I've done everything right. Why won't SER start? Thanks, M Here's my modules directory: [root@vx1 modules]# ls -l total 7264 -rwxr-xr-x 1 root root 67000 May 14 15:19 acc_db.so -rwxr-xr-x 1 root root 73248 May 14 15:19 acc_syslog.so -rwxr-xr-x 1 root root 69946 May 14 15:19 auth_db.so -rwxr-xr-x 1 root root 58071 May 14 15:19 auth.so -rwxr-xr-x 1 root root 37345 May 14 15:19 avp_db.so -rwxr-xr-x 1 root root 168768 May 14 15:19 avpops.so -rwxr-xr-x 1 root root 79110 May 14 15:19 avp.so -rwxr-xr-x 1 root root 279385 May 14 15:19 ctl.so -rwxr-xr-x 1 root root 72896 May 14 15:19 db_ops.so -rwxr-xr-x 1 root root 218639 May 14 15:18 dbtext.so -rwxr-xr-x 1 root root 97287 May 14 15:19 dialog.so -rwxr-xr-x 1 root root 63003 May 14 15:19 dispatcher.so -rwxr-xr-x 1 root root 21604 May 14 15:19 diversion.so -rwxr-xr-x 1 root root 114536 May 14 15:19 domain.so -rwxr-xr-x 1 root root 45704 May 14 15:19 enum.so -rwxr-xr-x 1 root root 109624 May 14 15:19 eval.so -rwxr-xr-x 1 root root 84357 May 14 15:19 exec.so -rwxr-xr-x 1 root root 81398 May 14 15:19 fifo.so -rwxr-xr-x 1 root root 53442 May 14 15:19 flatstore.so -rwxr-xr-x 1 root root 43414 May 14 15:19 gflags.so -rwxr-xr-x 1 root root 87574 May 14 15:19 lcr.so -rwxr-xr-x 1 root root 35883 May 14 15:19 maxfwd.so -rwxr-xr-x 1 root root 112635 May 14 15:19 mediaproxy.so -rwxr-xr-x 1 root root 87485 May 14 15:19 msilo.so -rwxr-xr-x 1 root root 106696 May 14 15:19 mysql.so -rwxr-xr-x 1 root root 187822 May 14 15:19 nathelper.so -rwxr-xr-x 1 root root 21203 May 14 15:19 options.so -rwxr-xr-x 1 root root 741640 May 14 15:19 pa.so -rwxr-xr-x 1 root root 131893 May 14 15:19 pdt.so -rwxr-xr-x 1 root root 272113 May 14 15:19 permissions.so -rwxr-xr-x 1 root root 122158 May 14 15:19 pike.so -rwxr-xr-x 1 root root 232133 May 14 15:19 presence_b2b.so -rwxr-xr-x 1 root root 26172 May 14 15:19 print.so -rwxr-xr-x 1 root root 47597 May 14 15:19 ratelimit.so -rwxr-xr-x 1 root root 152843 May 14 15:19 registrar.so -rwxr-xr-x 1 root root 402652 May 14 15:19 rls.so -rwxr-xr-x 1 root root 114896 May 14 15:19 rr.so -rwxr-xr-x 1 root root 63276 May 14 15:19 sanity.so -rwxr-xr-x 1 root root 66942 May 14 15:19 sl.so -rwxr-xr-x 1 root root 222033 May 14 15:18 sms.so -rwxr-xr-x 1 root root 34415 May 14 15:19 speeddial.so -rwxr-xr-x 1 root root 120807 May 14 15:19 textops.so -rwxr-xr-x 1 root root 43276 May 14 15:19 timer.so -rwxr-xr-x 1 root root 848962 May 14 15:19 tm.so -rwxr-xr-x 1 root root 117593 May 14 15:19 uac.so -rwxr-xr-x 1 root root 38793 May 14 15:19 unixsock.so -rwxr-xr-x 1 root root 38979 May 14 15:19 uri_db.so -rwxr-xr-x 1 root root 38603 May 14 15:19 uri.so -rwxr-xr-x 1 root root 351468 May 14 15:19 usrloc.so -rwxr-xr-x 1 root root 41867 May 14 15:19 xcap.so -rwxr-xr-x 1 root root 100354 May 14 15:19 xlog.so -rwxr-xr-x 1 root root 98456 May 14 15:19 xmlrpc.so [root@vx1 modules]# ____________________________________________________________________________________Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games. http://get.games.yahoo.com/proddesc?gamekey=monopolyherenow

3 2

[Users] Help (handle status)
by Jeferson Prevedello 17 May '07

17 May '07

Hi guys, I able the control of simultaneous users with the function [max_contacts], but whenever the user is blocked the softphone (X-lite) show the following message: [Registration error: 503 - Service Unavailable]. I tried to adapt the message conform code below, but not resolve: if (t_check_status("503")) { sl_send_reply("503","my new reply"); exit; } Any sugestion ? Regards Jeferson

2 1

[Users] changing log level of certain messages
by Dimo 17 May '07

17 May '07

Hi guys, I was wondering if it is possible to change the log level of some messages, namely: ERROR:auth_radius:radius_authorize_sterman: rc_auth failed rc_send_server: no reply from RADIUS server xxxx.:1812 I have alot of clutter from this one and i would like to lower its severity so it does not get logged, but i still need other messages from radius to be logged. Apart from going into the code and changing the log level do you know of any other way to do it? Best. Dimo

2 1

[Users] Mediaproxy
by Mhmd 17 May '07

17 May '07

Greeting , I have downloaded Media Proxy , but unfortunately I could not find mediaproxy.iso in you package , in you readme , you have explain to how I install it , you mentioned to mediaproxy.iso file which is missing in the package , can you please urgently advise where from can I get the full mediaproxy server . waiting your reply a.s.a.p . Best Regards Mohammad Network Engineer GCS Amman, Jordan

2 1

RE: [Users] Mediaproxy
by Mhmd 17 May '07

17 May '07

hi can anyone advise . how can i configure the mediaproxy with sipxfoundry sipxces ... if possible. as long as some files needs openser or ser . thanks Mohammad ________________________________ From: Lesley Liu [mailto:lesley1025@gmail.com] Sent: Thu 5/17/2007 12:00 PM To: Mhmd; Users(a)openser.org Subject: Re: [Users] Mediaproxy Mohammad, mediaproxy.so is one module of SER or OpenSER, it can be find in /usr/local/lib/openser/modules/ or simular path. Regards, Lesley 2007/5/17, Mhmd <mohammad(a)guestware.net>: Greeting , I have downloaded Media Proxy , but unfortunately I could not find mediaproxy.iso in you package , in you readme , you have explain to how I install it , you mentioned to mediaproxy.iso file which is missing in the package , can you please urgently advise where from can I get the full mediaproxy server . waiting your reply a.s.a.p . Best Regards Mohammad Network Engineer GCS Amman, Jordan _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

1 0

[Users] Mediaproxy
by Mhmd 17 May '07

17 May '07

Greeting , I have downloaded Media Proxy , but unfortunately I could not find mediaproxy.iso in your package , in your readme , you have explain how do i install it , you mentioned to mediaproxy.iso file which is missing in the package , can you please urgently advise where from can i get the full mediaproxy server . waiting your reply a.s.a.p . Best Regards Mohammad Network Engineer GCS Amman, Jordan

1 0

[Users] Openser trusted cache table
by Thiago Maluf 16 May '07

16 May '07

I would like to work with Module Parameter DB_MODE in Permissions Module set to 1 so I can use cache to optimize the request to function allow_trusted in trusted table . But this table may change once in a while. Taking a look in the source, I didn't find how the cache is updated without my interaction. My doubt is if Cache is updated automatically or if I have to use MI_FIFO or change the source to do this. Thanks in advanced. Thiago Maluf. -- ---------------------------------------------------------------- THIAGO MALUF RESENDE Consultor Voip e Programador WEB (Voip Developer and Web Developer) Tel: +55 21 86042100 e-mail: malufrj(a)gmail.com

1 0

[Users] 1.2.0 LCR schema error or my method of load_gw?
by Charles Wang 16 May '07

16 May '07

Hi all, I got this openser.cfg from http://www.sipwise.com/wizard . I tried to make a offnet call prefix with 00. I try to load_gws and next_gw for lcr. But I can't find out such as "select xxx from lcr" or "select xxx from gw" during these steps in my mysql.log ( I opened the log file myself ). And also it will not redirect to the voice gateway. Is it the error of my schemas ? Any idea for it please. #################### /var/log/message #################### May 16 22:52:05 XXX /usr/local/sbin/openser[29112]: New request - M=INVITE RURI=sip:00862123456789@XXX.XXX.XXX F=sip:101@XXX.XXX.XXX T= sip:00862123456789@XXX.XXX.XXX IP=61.217.XXX.XXX ID=C8F50846-1452-4EDC-9726-5A3AFECDF533(a)192.168.11.9 May 16 22:52:05 XXX /usr/local/sbin/openser[29112]: Proxy authentication failed - M=INVITE RURI=sip:00862123456789@XXX.XXX.XXX F=sip:101@XXX.XXX.XXXT= sip:00862123456789@XXX.XXX.XXX IP=61.217.XXX.XXX ID=C8F50846-1452-4EDC-9726-5A3AFECDF533(a)192.168.11.9 May 16 22:52:05 XXX /usr/local/sbin/openser[29117]: New request - M=INVITE RURI=sip:00862123456789@XXX.XXX.XXX F=sip:101@XXX.XXX.XXX T= sip:00862123456789@XXX.XXX.XXX IP=61.217.XXX.XXX ID=C8F50846-1452-4EDC-9726-5A3AFECDF533(a)192.168.11.9 May 16 22:52:05 XXX /usr/local/sbin/openser[29117]: Clear caller preferences - M=INVITE RURI=sip:00862123456789@XXX.XXX.XXX F=sip:101@XXX.XXX.XXX T= sip:00862123456789@XXX.XXX.XXX IP=61.217.XXX.XXX ID=C8F50846-1452-4EDC-9726-5A3AFECDF533(a)192.168.11.9 May 16 22:52:05 XXX /usr/local/sbin/openser[29117]: Load caller preferences for uuid '1' - M=INVITE RURI=sip:00862123456789@XXX.XXX.XXX F= sip:101@XXX.XXX.XXX T=sip:00862123456789@XXX.XXX.XXX IP=61.217.XXX.XXX ID=C8F50846-1452-4EDC-9726-5A3AFECDF533(a)192.168.11.9 May 16 22:52:05 XXX /usr/local/sbin/openser[29117]: Clear callee preferences - M=INVITE RURI=sip:00862123456789@XXX.XXX.XXX F=sip:101@XXX.XXX.XXX T= sip:00862123456789@XXX.XXX.XXX IP=61.217.XXX.XXX ID=C8F50846-1452-4EDC-9726-5A3AFECDF533(a)192.168.11.9 May 16 22:52:05 XXX /usr/local/sbin/openser[29117]: Callee is not local - M=INVITE RURI=sip:00862123456789@XXX.XXX.XXX F=sip:101@XXX.XXX.XXX T= sip:00862123456789@XXX.XXX.XXX IP=61.217.XXX.XXX ID=C8F50846-1452-4EDC-9726-5A3AFECDF533(a)192.168.11.9 May 16 22:52:05 XXX /usr/local/sbin/openser[29117]: Ready to Invite To External - M=INVITE RURI=sip:00862123456789@XXX.XXX.XXX F= sip:101@XXX.XXX.XXX T=sip:00862123456789@XXX.XXX.XXX IP=61.217.XXX.XXX ID=C8F50846-1452-4EDC-9726-5A3AFECDF533(a)192.168.11.9 May 16 22:52:05 XXX /usr/local/sbin/openser[29117]: Ready to 00 - M=INVITE RURI=sip:00862123456789@XXX.XXX.XXX F=sip:101@XXX.XXX.XXX T= sip:00862123456789@XXX.XXX.XXX IP=61.217.XXX.XXX ID=C8F50846-1452-4EDC-9726-5A3AFECDF533(a)192.168.11.9 May 16 22:52:05 XXX /usr/local/sbin/openser[29117]: Ready to Load PSTN gateways - M=INVITE RURI=sip:00862123456789@XXX.XXX.XXX F= sip:101@XXX.XXX.XXX T=sip:00862123456789@XXX.XXX.XXX IP=61.217.XXX.XXX ID=C8F50846-1452-4EDC-9726-5A3AFECDF533(a)192.168.11.9 May 16 22:52:05 XXX /usr/local/sbin/openser[29117]: Ready to NextGW - M=INVITE RURI=sip:00862123456789@XXX.XXX.XXX F=sip:101@XXX.XXX.XXX T= sip:00862123456789@XXX.XXX.XXX IP=61.217.XXX.XXX ID=C8F50846-1452-4EDC-9726-5A3AFECDF533(a)192.168.11.9 May 16 22:52:05 XXX /usr/local/sbin/openser[29117]: No PSTN gateways available - M=INVITE RURI=sip:00862123456789@XXX.XXX.XXX F= sip:101@XXX.XXX.XXX T=sip:00862123456789@XXX.XXX.XXX IP=61.217.XXX.XXX ID=C8F50846-1452-4EDC-9726-5A3AFECDF533(a)192.168.11.9 #################### /tmp/mysql.log #################### 070516 22:52:05 755 Query select password,uuid from subscriber where username='101' AND domain='xxx.xxx.xxx' 755 Query select value,attribute,type from usr_preferences where uuid='1' 755 Query select uuid from subscriber where username = '00862123456789' ===> gw table | id | gw_name | grp_id | ip_addr | port | uri_scheme | transport | strip | prefix | | 1 | my-test-gw | 1 | XXX.XXX.XXX.XXX | 5060 | 1 | 1 | 0 | NULL | ===> gw schema | Field | Type | Null | Key | Default | Extra | | id | int(10) | NO | PRI | NULL | auto_increment | | gw_name | varchar(128) | NO | UNI | NULL | | | grp_id | int(10) unsigned | NO | MUL | NULL | | | ip_addr | varchar(15) | NO | | NULL | | | port | smallint(5) unsigned | YES | | NULL | | | uri_scheme | tinyint(3) unsigned | YES | | NULL | | | transport | tinyint(3) unsigned | YES | | NULL | | | strip | tinyint(3) unsigned | YES | | NULL | | <=== it should be varchar ? | prefix | varchar(16) | YES | | NULL | | <=== default value should be Empty or NULL ? ===> gw_grp table | grp_id | grp_name | | 1 | my-test-gw | ===> gw_grp schema | Field | Type | Null | Key | Default | Extra | | grp_id | int(10) unsigned | NO | PRI | NULL | auto_increment | | grp_name | varchar(64) | NO | | NULL | | ===> lcr table | id | prefix | from_uri | grp_id | priority | | 1 | 00862123456789 | % | 1 | 10 | ===> lcr schema | Field | Type | Null | Key | Default | Extra | | id | int(10) | NO | PRI | NULL | auto_increment | | prefix | varchar(16) | NO | MUL | NULL | | | from_uri | varchar(128) | YES | MUL | NULL | | | grp_id | int(10) unsigned | NO | MUL | NULL | | | priority | tinyint(3) unsigned | NO | | NULL | | ################ parts of my openser.cfg####################################### loadmodule "lcr.so" modparam("lcr", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("lcr", "gw_table", "gw") modparam("lcr", "gw_name_column", "gw_name") modparam("lcr", "ip_addr_column", "ip_addr") modparam("lcr", "port_column", "port") modparam("lcr", "uri_scheme_column", "uri_scheme") modparam("lcr", "transport_column", "transport") modparam("lcr", "grp_id_column", "grp_id") modparam("lcr", "lcr_table", "lcr") modparam("lcr", "strip_column", "strip") modparam("lcr", "prefix_column", "prefix") modparam("lcr", "from_uri_column", "from_uri") modparam("lcr", "priority_column", "priority") modparam("lcr", "gw_uri_avp", "1400") modparam("lcr", "ruri_user_avp", "1402") modparam("lcr", "contact_avp", "1401") modparam("lcr", "fr_inv_timer_avp", "s:fr_inv_timer_avp") modparam("lcr", "fr_inv_timer", 90) modparam("lcr", "fr_inv_timer_next", 30) modparam("lcr", "rpid_avp", "s:rpid") ######################################################################## # Request route 'invite-to-external' ######################################################################## route[15] { xlog("L_INFO", "Ready to Invite To External - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if(isflagset(20)) { xlog("L_INFO", "Call to foreign domain - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(11); exit; } if(!isflagset(23)) { # don't allow calls relaying from PSTN to PSTN, if not explicitely forwarded xlog("L_INFO", "Ready to 00 - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if(uri =~ "^sip:00[0-9]+@") { xlog("L_INFO", "Ready to Load PSTN gateways - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # only route numeric users to PSTN if(!load_gws()) { xlog("L_INFO", "Ready to LoadGW - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); 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_INFO", "Ready to NextGW - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); 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(11); } } xlog("L_INFO", "Call to unknown user - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(7); sl_send_reply("404", "User Not Found"); exit; } -- Best Regards Charles

3 7

Re: [Serusers] Configuration examples
by Mahatma Sarasvati 16 May '07

16 May '07

Hello, I prefer one big example configuration. I like the current default ser.cfg that has nothing but calls to other route blocks in the main route block. That way its easy to turn off route features that you don't want or need to use. I like the new build system. When its more complete it should be extremely useful. Thanks, M ----- Original Message ---- From: SIP <sip(a)arcdiv.com> To: Nils Ohlmeier <nils(a)iptel.org> Cc: serdev(a)iptel.org; serusers(a)iptel.org Sent: Monday, May 14, 2007 4:03:09 AM Subject: Re: [Serusers] Configuration examples Nils Ohlmeier wrote: > Hello everybody, > > as we want to provide more examples for the configuration the following the > question came up: > > Do you prefer > - one big example configuration > - probably harder to understand > - easier to maintain for us > - one example config file per "feature" > - probably easier to understand your desired "feature" > - you might have problems to integrate all your desired "features" into one > config without breaking anything > - harder to maintain for us, as we have to update several files :-) > > As the configuration examples are indented to be for the community, I would > like to know what you prefer or if you see other options. > > Thanks > Nils > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > Ideally, I'd like to see: -One big configuration, well documented (per line or per block at the VERY least). -Smaller configuration examples for options that might not fit into a larger overall config (block-level, which blocks/modparams/etc in the large config would be replaced with/added by the option blocks provided; detailed documentation on the blocks and modparams and what they do/mean (perhaps with links to the proper lines in the module documentation). That way you're not keeping separate configs for each option (unless it's something that completely replaces all the rest of the config), and as more tricks/suggestions come up, it's easier to add an option or two to the options sections since they're just small blocks of code. Just an idea. N. _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers ____________________________________________________________________________________ We won't tell. Get more on shows you hate to love (and love to hate): Yahoo! TV's Guilty Pleasures list. http://tv.yahoo.com/collections/265

1 0

[Serusers] Problem running SER behind firewall
by Wei Wang 16 May '07

16 May '07

I have a problem running SER behind firewall. Here is the network diagram: |UA1|--|FW1| +--|FW|--|SER| \ / +--+ / \ |UA2|--|FW2| +--|MediaProxy| Where, UA1 and UA2 are Xlite soft-phones behind their own firewalls. SER is listening on private IP address 192.168.4.217. FW has public IP address 66.134.1.34 and forwards port 5060 to SER. The ser.cfg file is pretty much copied from SER getting start guide. When UA1 calling UA2, the call established fine but UA1 will hang up by itself after ~30 seconds. The captured IP packages on SER revealed that the last ACK received from UA2 by SER was sent to FW IP address. Since port 5060 is forwarded to SER on the FW, it caused a looping situation. The ser.cfg is listed at the end. Thanks in advance. Wei Wang wwang(a)m1global.com ======== ser.cfg ============ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode debug=3 #debug=9 fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) listen=192.168.4.217 #listen=66.134.1.36 port=5060 children=4 fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:s3rv1c3@localhost/ser" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/print.so" loadmodule "/usr/local/lib/ser/modules/xlog.so" # ----------------- setting module-specific parameters --------------- # -- 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|permissions|uri_db|usrloc", "db_url", "mysql://ser:s3rv1c3@localhost/ser") modparam("auth_db", "calculate_ha1", 1) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy", "natping_interval", 30) #modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy", "mediaproxy_socket", "/var/run/proxydispatcher.sock") modparam("mediaproxy", "sip_asymmetrics", "/usr/local/etc/ser/sip-clients") modparam("mediaproxy", "rtp_asymmetrics", "/usr/local/etc/ser/rtp-clients") modparam("registrar", "nat_flag", 6) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("xlog", "buf_size", 8192) # ------------------------- request routing logic ------------------- # main routing logic route{ # xlog("L_INFO", "Main route [From]%fu,[To]%tu,[Req-Method]%rm,[Req-RURI]%ru[IP-src]%is ...\n"); if(method != "SUBSCRIBE") { xlog("L_INFO", "\r\n===========SIP MSG==================\r\n%mb\r\n_____END SIP MSG________________________\r\n"); }; # 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"); break; }; if (msg:len >= 4086 ) { sl_send_reply("513", "Message too big"); break; }; # 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 == "INVITE" && client_nat_test("3")) { # xlog("L_INFO", "method==INVITE and nated: calling record_route_preset\n"); # IP ADDRESS Here record_route_preset("66.134.1.34:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ------------------------- # Call Tear Down Section #------------------------- if(method=="BYE" || method=="CANCEL") { #xlog("L_INFO", "RECEIVED BYE or CANCEL..."); end_media_session(); }; # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { xlog("L_INFO", "DEBUG: loose_route..."); if((method=="INVITE" || method == "REFER") && !has_totag()) { sl_send_reply("403", "Forbidden"); break; }; if(method == "INVITE") { if(!allow_trusted()) { if(!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); break; } else if(!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); } if(client_nat_test("3") || search("^Route:.*;nat=yes")) { setflag(6); use_media_proxy(); }; }; # mark routing logic in request #append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { route(4); # mark routing logic in request #append_hf("P-hint: outbound\r\n"); route(1); break; }; # 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(method == "ACK") { route(1); break; } else if(method=="CANCEL") { route(1); break; } else if(method == "INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (!uri==myself) { route(4); #append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { t_on_reply("1"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { if(method=="INVITE" || method == "ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { ############################ # REGISTER Message Handler ########################### sl_send_reply("100", "Trying"); if(!search("^Contact:[ ]*\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; if(!www_authorize("", "subscriber")) { www_challenge("", "0"); break; }; if(!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if(!save("location")) { sl_reply_error(); }; } route[3] { ############################ # INVITE Message Handler ########################### if(client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; if(!allow_trusted()) { if(!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); break; } else if(!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; }; consume_credentials(); lookup("aliases"); if(uri != myself) { route(4); route(1); break; }; if(!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(4); route(1); } route[4] { #---------------------------- # NAT Traversal Section #---------------------------- if(isflagset(6) || isflagset(7)) { if(!isflagset(8)) { setflag(8); use_media_proxy(); }; }; } onreply_route[1] { if((isflagset(6) || isflagset(7)) && (status =~ "(180)|183)|2[0-9][0-9]")) { if(!search("^Content-Length:[ ]*0")) { use_media_proxy(); }; }; if(client_nat_test("1")) { fix_nated_contact(); }; } =============== END ser.cfg ================

3 6

Re: [Serusers] dns failover
by samuel 16 May '07

16 May '07

I'm sorry but I don't know enough about the internals . Maybe Andrei (DNS) and Dragos (openIMS) can help with it. Probably the most important lines of the log are: [00m 1(17870) [01;30mdns_hash_find(scscf.open-ims.test(19), 1), h=915 [00m 1(17870) [01;30mdns_a_resovle(scscf.open-ims.test, 0) returning 0 [00m 1(17870) [01;30mdns_srv_resolve_ip("_sip._udp.scscf.open-ims.test", 0, 0), ret=0, ip=127.0.0.1 [00m 1(17870) [01;30mdns_sip_resolve(scscf.open-ims.test, 0, 0), srv0, ret=0 It looks like there's only one returning record...while in the DNS there are two. Try to post last message to openims and serdev lists and maybe developers can help with it... Just for trying, can you add a second fake SRV entry differing not only in the port but also in the hostname? Luck, Samuel. 2007/5/16, Mário Ferreira <mariojvp(a)gmail.com>: > > Hi, > > I tried with SER version ser 2.1.0-dev1, with dns cache and failover > compile options, and with use_dns_cache and > use_dns_failover turned on in the config files ( of P and I-CSCF, the > previous hop of S-CSCF) , but it doesn't work. > I attached the log corresponding to the REGISTER reception by I-CSCF and > delivery to S-CSCF. > I have two S-CSCFs and when i kill one, the message never goes to the > other S-CSCF. > > Thanks, > > Mário Ferreira > > On 5/14/07, Mário Ferreira <mariojvp(a)gmail.com> wrote: > > > > Hi, > > > > The SER version was updated, but only in the trunk of openimscore, and i > > had a working copy of a branch (with the old SER version) . > > Now, i have checked out the trunk and i will test it soon. > > > > Thanks, > > Mário Ferreira > > > > On 5/14/07, samuel < samu60(a)gmail.com> wrote: > > > > > > It looks like SER is compiled with all compilation flags for DNS > > > failover... are these features set in the config file? > > > I guess you're using openIMS SER branch, isn't it? I don't know if it > > > has been sync with latest SER 2.0 and I can not assure whether ser > > > 0.10.99-dev53-tm_fixes has SRV failover support (which I think it does > > > althought with maybe some bugs). > > > > > > From the dig output I guess you're trying to load balance between the > > > two servers responding for scscf.open-ims.test, isn't it? > > > Can you please attach the log file section where SER is trying to send > > > to this destination? > > > > > > Thanks, > > > Samuel > > > > > > 2007/5/14, Mário Ferreira < mariojvp(a)gmail.com>: > > > > > > > > Hi, > > > > > > > > # ser -V > > > > version: ser 0.10.99-dev53-tm_fixes (i386/linux) > > > > flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, > > > > DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, > > > > FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_DST_BLACKLIST > > > > 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. > > > > @(#) $Id: main.c 136 2007-02-13 17:58:42Z vingarzan $ > > > > main.c compiled on 22:49:24 Mar 26 2007 with gcc 3.4.6 > > > > > > > > # dig NAPTR open-ims.test > > > > > > > > ; <<>> DiG 9.2.4 <<>> NAPTR open-ims.test > > > > ;; global options: printcmd > > > > ;; Got answer: > > > > ;; ->>HEADER<<- opcode: QUERY, status: NOERROR, id: 56440 > > > > ;; flags: qr aa rd ra; QUERY: 1, ANSWER: 2, AUTHORITY: 1, > > > > ADDITIONAL: 3 > > > > > > > > ;; QUESTION SECTION: > > > > ;open-ims.test. IN NAPTR > > > > > > > > ;; ANSWER SECTION: > > > > open-ims.test. 86400 IN NAPTR 10 50 "s" "SIP+D2U" > > > > "" _sip._udp.open- ims.test. > > > > open-ims.test. 86400 IN NAPTR 20 50 "s" "SIP+D2T" > > > > "" _sip._tcp.open-ims.test. > > > > > > > > ;; AUTHORITY SECTION: > > > > open-ims.test. 86400 IN NS > > > > 127.0.0.1.open-ims.test . > > > > > > > > ;; ADDITIONAL SECTION: > > > > icscf.open-ims.test. 86400 IN A 127.0.0.1 > > > > _sip._udp.open-ims.test . 86400 IN SRV 0 0 5060 > > > > icscf.open-ims.test. > > > > _sip._tcp.open- ims.test. 86400 IN SRV 0 0 5060 > > > > icscf.open-ims.test. > > > > > > > > ;; Query time: 0 msec > > > > ;; SERVER: 127.0.0.1#53(127.0.0.1 ) > > > > ;; WHEN: Mon May 14 18:24:18 2007 > > > > ;; MSG SIZE rcvd: 259 > > > > > > > > # dig _sip._udp.scscf.open-ims.test. srv > > > > > > > > ; <<>> DiG 9.2.4 <<>> _sip._udp.scscf.open-ims.test. srv > > > > ;; global options: printcmd > > > > ;; Got answer: > > > > ;; ->>HEADER<<- opcode: QUERY, status: NOERROR, id: 21445 > > > > ;; flags: qr aa rd ra; QUERY: 1, ANSWER: 2, AUTHORITY: 1, > > > > ADDITIONAL: 1 > > > > > > > > ;; QUESTION SECTION: > > > > ;_sip._udp.scscf.open-ims.test. IN SRV > > > > > > > > ;; ANSWER SECTION: > > > > _sip._udp.scscf.open-ims.test. 86400 IN SRV 0 1 6060 > > > > scscf.open-ims.test. > > > > _sip._udp.scscf.open-ims.test. 86400 IN SRV 0 1 6061 > > > > scscf.open-ims.test. > > > > > > > > ;; AUTHORITY SECTION: > > > > open-ims.test. 86400 IN NS > > > > 127.0.0.1.open-ims.test. > > > > > > > > ;; ADDITIONAL SECTION: > > > > scscf.open-ims.test. 86400 IN A 127.0.0.1 > > > > > > > > ;; Query time: 0 msec > > > > ;; SERVER: 127.0.0.1#53( 127.0.0.1) > > > > ;; WHEN: Mon May 14 18:26:33 2007 > > > > ;; MSG SIZE rcvd: 165 > > > > > > > > > > > > regards, > > > > > > > > Mário Ferreira > > > > > > > > On 5/13/07, samuel <samu60(a)gmail.com> wrote: > > > > > > > > > > I have tried DNS failover and can assert is working like charm for > > > > > SER version 2.0 and further. > > > > > Which SER version are you using? > > > > > > > > > > can you check wheter you have compiled SER with DNS failover? > > > > > which is the ouput of > > > > > #ser -V > > > > > what is the ouput of > > > > > #dig NAPTR yourdomain.name.com > > > > > (replace yourdomain.name.com for the appropriate valie) > > > > > > > > > > regards, > > > > > sam. > > > > > > > > > > 2007/5/10, Mário Ferreira <mariojvp(a)gmail.com >: > > > > > > > > > > > > Hi, > > > > > > > > > > > > I am interested in using SER DNS failover, but I am getting some > > > > > > problems. > > > > > > > > > > > > A failure can be a 503 error response, ICMP errors or a timeout > > > > > > due to a not received response. > > > > > > > > > > > > Does anyone know if SER is able to detect these failures and > > > > > > then select an alternative destination for the message? > > > > > > > > > > > > If the ICMP errors can't be detected, can this be solved with > > > > > > timers? > > > > > > I turned on use_dns_cache and use_dns_failover, but when the > > > > > > message fails to be delivered to the first destination, > > > > > > SER never tries the second destination, which was configured > > > > > > with DNS SRV. > > > > > > > > > > > > regards, > > > > > > > > > > > > Mário Ferreira > > > > > > > > > > > > _______________________________________________ > > > > > > Serusers mailing list > > > > > > Serusers(a)lists.iptel.org > > > > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > > > > > > > > > > > > > > > > > > > > >

1 0

[Serusers] ANNOUNCEMENT: SER 2.0 Release Candidate 1 is available
by Greger V. Teigre 16 May '07

16 May '07

If you have 30 minutes to spare, you can set up a working SER 2.0! If you haven't taken the time to test SER 2.0 yet, it has never been simpler. Find full instructions on: http://www.iptel.org/ser_2_0_0_release_candidate_1 Questions and feedback go to these lists! Have fun, Greger

1 0

[Users] Openser - LCS 2005 incoming call
by Tsiferana 16 May '07

16 May '07

Hi, I've tried to test this scenario: windows messenger>>lcs2005>>openser>>x-lite and it works great but this scenario x-lite>>openser>>lcs2005>>windows messenger the call can't be etablished., in the network traffic is see that the lcs does not relay the invite request to windows messenger. what do I need for running this scenario? Best Regards Tsiferana --------------------------------- Ne gardez plus qu'une seule adresse mail ! Copiez vos mails vers Yahoo! Mail

2 3

[Users] Max-Forward header in Notify messages
by Toni Barata 16 May '07

16 May '07

Hi Anka, In Notify messages generated by Presence module the Max-Forward Header is missing. Some sip stacks have a strange behaviour when receiving Notify messages without Max-Forward header. Best regards, Toni

1 0

[Users] purposely dropping reply message
by Tim 16 May '07

16 May '07

Hello, How/Where would be the best place to purposely drop a particular reply message ( > 400 ) I have armed both the reply route and failure route and attempted to drop the message by calling the drop() function but this doesn't seem to work. Is there a simple method for conditionally dropping a reply message? My goal is to force the UAC that created the original request to retransmit (UDP) the request in certain scenarios, such as when ACK to an initial invite has not been fully processed by the downstream UAS but a re-invite has been received by the UAS causing the UAS to return a 400 for the re-invite. I do not want the UAC to see the 400 reply, instead I simply want the UAC to resend the re-invite because no final response has been received, presumably by this time the ACK processing will be finished and the retransmitted re-invite would be handled smoothly. I realize replies are routed based on via headers. Is there some way to short circuit this routing other than drop()? Thanks Tim Tried this from failure route if(t_check_status("400")) { xlog( "L_ERR", "400 message in failure route 2 ********** \n" ); drop(); return; } Tried this from reply route if(status == "400") { xlog( "L_ERR", "400 in reply route 2 .... dropping\n" ); drop(); return; }

3 4

[Users] trying to get 2-way audio in multi-homed openser configuration
by Taylor Carpenter 16 May '07

16 May '07

I have tried all sorts of configurations with both rtpproxy and media proxy, but I continue to end up with audio only working on one end of the call. The latest I tried to turn on rtpproxy for all calls. The call goes through but audio is only heard from the callee to the caller. Any help is greatly appreciated. Taylor

2 1

[Users] XCAP Lite 1.0 released
by Adrian Georgescu 16 May '07

16 May '07

Change log xcap-lite (1.0.0) unstable; urgency=low * Added control interface for OpenSER to inform the Presence agent about changes in the presrules document. This allows the presence server to imediately send Notify to the updated buddy list after the end-user changes his presence policy document. Thanks to Anca Vamanu and Juha Heinanen for the support provided for adding this critical functionality * The configuration config.php has changed to allow for definition of the OpenSER management interface URL (refreshWatchersURL) * OpenSER must have the mi_xmlrpc module loaded and listening on the port configured in config.php * Fixed query in updateWatchers function * OpenSER trunk version 2230 or higher is required when refreshWatchersURL setting is enabled in config.php The software can be downloaded from: http://download.dns-hosting.info/XCAP/xcap-lite-1.0.0.tar.gz Regards, Adrian

1 0

[Users] openser setup as a telephony gateway (to PSTN) for NAT'd clients
by Taylor Carpenter 16 May '07

16 May '07

I may be misunderstanding things (very probably so), but all the examples for both SER and OpenSER that I have seen either do not do NAT or NAT but are not the "end point" for a UAC to the PSTN. From what I can tell they are just sending on the SIP request to the next destination as is. The setup I am trying to accomplish is * SIP clients are all on a private network with connectivity to OpenSER directly on one interface (on the same private network) * OpenSER's other interface is on the external network (internet facing) * SIP clients are only sending telephone numbers (sip:telephone_number@*... do not know about PSTN providers) * OpenSER connects to the PSTN provider and send the number to dial that came from the SIP client * OpenSER "proxies" the entire call (with the help of rtpproxy or mediaproxy for RTP of course) * No incoming calls from the internet to OpenSER (no support for that is needed) * Registration not required for sip clients (they are all on same private network and authorized) I have found several posts, example configs, documents that have pieces of what I need (from what I can tell).. and I have tried to put it together, but it does not quite work... So is there some example that fits this type of usage? If not one then possibly several pieces from a few documents? I have been thinking that OpenSER setup as a outbound proxy configured for multihome, but everything I have seen on that just routes the calls through to where ever the SIP client was requesting as a final destination and I need to send to one destination (the PSTN provider). Any help is greatly appreciated. BTW, I was thinking the NAThelper or outbound proxy example from http://www.voip-info.org/wiki-SER+tips+and+tricks Looked close to what is needed, but have a bunch of stuff (seemingly) unneeded for my scenario and have nothing about PSTN connectivity. Thanks for taking the time to read this long post (if you made it this far). Taylor

4 6

[Serusers] Race Condition on CANCEL?
by Gustavo Passos Tourinho 16 May '07

16 May '07

Hello, Im having some problems with cancelled calls. This is the scenario: U1 Proxy U2 INVITE -->>> <<--- 100 Trying INVITE -->>> <<--- 100 Trying <<--- 100 Trying CANCEL ->>> <<-- 200 Cancelling CANCEL ->> <<-- 180 Ringing <<-- 487 Cancelled <<-- 180 Ringing <<-- 200 OK (Wrong??) <<-- 200 OK My problem is that after some time waiting for "ringing", the user cancel the call. Even that proxy responses "487" it still forward the late 200 OK. Should it forward? I guess not because the transaction was destroyed, right? Can it be a configuration problem on my ser.cfg ou it can be in t_relay implementation? Thanks in advanced. Regards, Gustavo

3 11

[Serusers] ser_fifo gives timeout
by Andrey Kuprianov 16 May '07

16 May '07

Hi guys, I dont know what the problem is.. Really.. Im out of ideas with permissions and reinstallations for serweb's CTD and IM. During my 1st installation with serweb they both worked fine. Then I had to reinstall my OS (Ubuntu 7.x btw) and now they give both timeouts.. I have even copied send_im() and write2fifo() functions separately into a PHP file and ran it from command line. Same old "408 Request Timeout".. Please, suggest something. Im basically using nat-mediaproxy.cfg for config file and I have tried setting various ownerships and permissions to serweb dir and for /tmp/serweb and /tmp/ser_fifo.. Nothing helps really. All packages are installed for php4 including php4-cli, pear, pear-log, pear-DB, XML_RPC and even classkit (which is useless here anyway).

2 1

[Serusers] Configuration examples
by Nils Ohlmeier 15 May '07

15 May '07

Hello everybody, as we want to provide more examples for the configuration the following the question came up: Do you prefer - one big example configuration - probably harder to understand - easier to maintain for us - one example config file per "feature" - probably easier to understand your desired "feature" - you might have problems to integrate all your desired "features" into one config without breaking anything - harder to maintain for us, as we have to update several files :-) As the configuration examples are indented to be for the community, I would like to know what you prefer or if you see other options. Thanks Nils

9 10

[Users] SIP SIMPLE / XCAP test bed - feedback request
by Adrian Georgescu 15 May '07

15 May '07

Hello, We setup a SIP SIMPLE testbed using the latest OpenSER / XCAP-lite. To make use of this testbed you need a recent Eyebeam commercial version for Apple or Windows. Other SIP/SIMPLE/XCAP client implementations are of course more then welcome and desired. To use this test bed register a SIP account at: http://sip2sip.info Instructions about the SIP/Presence/XCAP settings are sent by email to the registrant. We appreciate any feedback regarding the software behavior so that we can fix any bugs in the used implementations and update the wiki pages with proper configuration samples. Thank you for your support, Adrian

2 2

Re: [Serusers] SER/PSTN Gateway conf
by flavio 15 May '07

15 May '07

My Record_Route and Loose section are the follow: # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (method=="INVITE") { if (!allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; }; route(1); break; }; As follow reported, SER Record Route INVITE original INVITE message, that lack of 'Route:' header, and forward it to IP Phone. Why not for BYE? =_=" Thanks for support and big patience. ^_^ U 2007/05/15 16:14:20.789400 10.28.52.105:5060 -> 10.28.19.202:5060 INVITE sip:06605XXXXX@10.28.19.202 SIP/2.0. To: <sip:06605XXXXX@10.28.19.202>. From: "06720XXXXX" <sip:06720XXXXX@10.28.52.105>;tag=DXsf22387576811Yaz079427. Via: SIP/2.0/UDP 10.28.52.105:5060;branch=z9hG4bK2387576811429488787179425. Remote-Party-ID: "06720XXXXX" <sip:06720XXXXX@10.28.52.105>;party=calling;screen=yes;privacy=off. Contact: <sip:10.28.52.105:5060>. Call-ID: 238757681179426(a)10.28.52.105. Max-Forwards: 70. User-Agent: Netsynt-Gateway/SIP_AC_SRVlkup_3.1.12_4. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, REGISTER, NOTIFY. CSeq: 23392 INVITE. Content-Length: 274. Content-Type: application/sdp. . v=0. o=NetsyntSIP-GW-UserAgent 34373 1 IN IP4 10.28.52.105. s=SIP Call. c=IN IP4 10.28.52.105. t=0 0. m=audio 16572 RTP/AVP 18 8 0. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:8 PCMA/8000. a=silenceSupp:off - - - -. a=rtpmap:0 PCMU/8000. a=silenceSupp:off - - - -. # U 2007/05/15 16:14:20.790166 10.28.19.202:5060 -> 10.28.52.105:5060 SIP/2.0 100 trying -- your call is important to us. To: <sip:06605XXXXX@10.28.19.202>. From: "06720XXXXX" <sip:06720XXXXX@10.28.52.105>;tag=DXsf22387576811Yaz079427. Via: SIP/2.0/UDP 10.28.52.105:5060;branch=z9hG4bK2387576811429488787179425. Call-ID: 238757681179426(a)10.28.52.105. CSeq: 23392 INVITE. Server: Sip EXpress router (0.9.6 (i386/linux)). Content-Length: 0. Warning: 392 10.28.19.202:5060 "Noisy feedback tells: pid=3803 req_src_ip=10.28.52.105 req_src_port=5060 in_uri=sip:0660522014@10.28.19.202 out_uri=sip:06605XXXXX@10.28.19.124;user=phone via_cnt==1". . # U 2007/05/15 16:14:20.790257 10.28.19.202:5060 -> 10.28.19.124:5060 INVITE sip:06605XXXXX@10.28.19.124;user=phone SIP/2.0. Record-Route: <sip:10.28.19.202;ftag=DXsf22387576811Yaz079427;lr=on>. To: <sip:0660522014@10.28.19.202>. From: "06720XXXXX" <sip:06720XXXXX@10.28.52.105>;tag=DXsf22387576811Yaz079427. Via: SIP/2.0/UDP 10.28.19.202;branch=z9hG4bK4639.ea946db1.0. Via: SIP/2.0/UDP 10.28.52.105:5060;branch=z9hG4bK2387576811429488787179425. Remote-Party-ID: "06720XXXXX" <sip:06720XXXXX@10.28.52.105>;party=calling;screen=yes;privacy=off. Contact: <sip:10.28.52.105:5060>. Call-ID: 238757681179426(a)10.28.52.105. Max-Forwards: 16. User-Agent: Netsynt-Gateway/SIP_AC_SRVlkup_3.1.12_4. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, REGISTER, NOTIFY. CSeq: 23392 INVITE. Content-Length: 274. Content-Type: application/sdp. . v=0. o=NetsyntSIP-GW-UserAgent 34373 1 IN IP4 10.28.52.105. s=SIP Call. c=IN IP4 10.28.52.105. t=0 0. m=audio 16572 RTP/AVP 18 8 0. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:8 PCMA/8000. a=silenceSupp:off - - - -. a=rtpmap:0 PCMU/8000. a=silenceSupp:off - - - -. 2007/5/15, Kostas Marneris <K.Marneris(a)otenet.gr>: - Nascondi testo tra virgolette - > As far as I know and understand it seems that the > R-URI of the BYE mesg (1st line) which GW sends to SER is wrong. > > I expected to see : > BYE sip:user@SIP_Phone_IP_Address SIP/2.0 > > or the problem is that the 'Route:' header is missing from the BYE mesg > so it's not loose_routed. > > Do you Record_Route the original INVITE ? > > Check also : rfc3261 / 16.12.1 > > > Kostas > >

2 1

[Users] Memory Errors
by Martin Klisch 15 May '07

15 May '07

Hi, i have some memory errors after running OpenSER one day: May 15 02:46:56 sip-b /opt/SER/SER0/bin/openser[20106]: [ID 381148 local0.error] ERROR: build_req_buf_from_sip_req: out of memory May 15 02:46:56 sip-b /opt/SER/SER0/bin/openser[20106]: [ID 933771 local0.error] ERROR:tm:print_uac_request: no pkg_mem May 15 02:46:56 sip-b /opt/SER/SER0/bin/openser[20106]: [ID 826633 local0.error] ERROR:tm:t_forward_nonack: failure to add branches May 15 02:47:00 sip-b /opt/SER/SER0/bin/openser[20106]: [ID 381148 local0.error] ERROR: build_req_buf_from_sip_req: out of memory May 15 02:47:00 sip-b /opt/SER/SER0/bin/openser[20106]: [ID 933771 local0.error] ERROR:tm:print_uac_request: no pkg_mem May 15 02:47:00 sip-b /opt/SER/SER0/bin/openser[20106]: [ID 826633 local0.error] ERROR:tm:t_forward_nonack: failure to add branches May 15 02:47:03 sip-b /opt/SER/SER0/bin/openser[20106]: [ID 381148 local0.error] ERROR: build_req_buf_from_sip_req: out of memory May 15 02:47:03 sip-b /opt/SER/SER0/bin/openser[20106]: [ID 933771 local0.error] ERROR:tm:print_uac_request: no pkg_mem Version: OpenSER 1.2.0 OS: Solaris 10 Arch: Sparc Is there a memory leak an any module? or is my memory too low?

3 8

[Serusers] PA module on 0.9.6
by Andrey Kuprianov 15 May '07

15 May '07

Hi, Got couple of questions: 1. Where is the sql file, which would prepare ser-0.9.6 DB for PA module. I just could not find it. 2. Is PA module working ok on 0.9.6? Would 0.9.7-pre8 work better? Andrey. P.S. - I know I should use Ottendorf for presence, but I'm bound to use 0.9.x for now :)

2 1

[Users] openSER + radiator radius server
by raviprakash sunkara 15 May '07

15 May '07

Hi users I'm New to Radiator Radius, but Am familar in Free radius server, for avp_load_radius("callee") or caller it getting only one value . my guess there is problem in Radiator Server , hanndler in part in radiator <Handler Service-Type=SIP-Callee-AVPs> #<AuthBy Group> PreAuthHook file:"/etc/radiator/user_domain.pl" # PostAuthHook file:"/etc/radiator/user_preferences.pl" # <AuthBy Group> AuthByPolicy ContinueWhileIgnore <AuthBy SQL> DBSource dbi:mysql:openser:localhost DBUsername openser DBAuth openserrw AuthSelect select value from usr_preferences where username='%{CheckGroup-User}' AuthSelect select value as value2 from usr_preferences where username='%{CheckGroup-User}' AuthColumnDef 0, SIP-AVP, reply NoDefault </AuthBy> <AuthBy INTERNAL> DefaultResult REJECT </AuthBy> </Handler> Radius reply only one sip-avp attribute , so i need to get multiple attribute gort sip-avp did any one help me ... in this......... -- Thanks &Regards Ravi Prakash Sunkara

1 0

[Serusers] SER 0.9.6 and MySQL 5.0 issues revisited
by Kostas Marneris 15 May '07

15 May '07

Greetings to the list, In a previous thread (http://lists.iptel.org/pipermail/serusers/2006-April/027961.html) SER and mysql 5.0 issues were discussed. The main idea was that if MySQL was restarted, SER should be able to do auto-reconnect to the database server. According to the thread (Jan Janak), the latest SER version (0.9.6) should contain a fix to the problem. However, we recently upgraded MySQL from version 4.1 to 5.0 as part of an upgade of Debian Sarge to Etch in our servers. We had no problems with SER and MySQL 4.1. We recompiled SER and linked mysql module with the MySQL 5 client library (libmysqlclient.so.15). The auto-reconnect problem seems to persist. We start MySQL, then SER and after a few SIP clients get registered, we restart the MySQL server (using Debian init scripts). We then see the following messages in ser's log: ./ser[24006]: submit_query(): MySQL server has gone away ./ser[24006]: db_insert: Error while submitting query ./ser[24006]: db_insert_ucontact(): Error while inserting contact ./ser[24006]: insert_ucontact(): Error while inserting in database Our SER config contains the following lines: fifo_db_url="mysql://ser:XXXXXX@localhost/ser" # -------------------------------------------------------------------- # Modules Section # -------------------------------------------------------------------- ... loadmodule "/opt/ser-0.9.6/lib/ser/modules/mysql.so" loadmodule "/opt/ser-0.9.6/lib/ser/modules/usrloc.so" loadmodule "/opt/ser-0.9.6/lib/ser/modules/uri_db.so" ... modparam("usrloc", "db_url", "mysql://ser:XXXXXX@localhost/ser") modparam("uri_db", "db_url", "mysql://ser:XXXXXX@localhost/ser") ... Our MySQL version is 5.0.32 (standard Debian etch packages) and we have compiled ourselves the latest stable SER version (0.9.6). Any comments are highly welcome, since we are thinking of deploying a large scale SER setup. thanks, Kostas

2 3

[Serusers] Forwarding Invites to SBC
by Wroclawski, Gregory J (Greg) 15 May '07

15 May '07

I am trying to configure the ser.cfg file so I can forward invites from a user registered on the SER to a ACME session border. I HAVE HAD NO SUCCESS. I have a IP adress for the SBC. Does anyone have a sample ser.cfg file that illustrates the "if then else' statements to effect this. Thanks in advance Greg Wroclawski

2 1

[Users] a compile problem with linphone1.6 and 1.7
by Lee Jacky 15 May '07

15 May '07

Hi,everone. now I am using openser1.1 and linphone1.6 and 1.7 version, after I configure linphone: ./configure --disable-strict, it displays :Now type 'make' to compile, and then 'make install' as root to install it. but when I execute make command, I get the errors as follows: gcc -DENABLE_TRACE -D_ORTP_SOURCE -g -O2 -o .libs/linphonec linphonec.o commands.o ../coreapi/.libs/liblinphone.so -lreadline -lncurses -Wl,--rpath -Wl,/usr/local/linphone/lib /root/linphone-1.6.0/mediastreamer2/src/.libs/libmediastreamer.so.0: undefined reference to `av_crc04C11DB7' /root/linphone-1.6.0/mediastreamer2/src/.libs/libmediastreamer.so.0: undefined reference to `av_realloc' /root/linphone-1.6.0/mediastreamer2/src/.libs/libmediastreamer.so.0: undefined reference to `av_log' /root/linphone-1.6.0/mediastreamer2/src/.libs/libmediastreamer.so.0: undefined reference to `ff_sqrt_tab' /root/linphone-1.6.0/mediastreamer2/src/.libs/libmediastreamer.so.0: undefined reference to `lzo1x_decode' /root/linphone-1.6.0/mediastreamer2/src/.libs/libmediastreamer.so.0: undefined reference to `av_d2q' /root/linphone-1.6.0/mediastreamer2/src/.libs/libmediastreamer.so.0: undefined reference to `av_random_generate_untempered_numbers' /root/linphone-1.6.0/mediastreamer2/src/.libs/libmediastreamer.so.0: undefined reference to `av_init_lls' I have serached a lot of materials, but they seems to be no help to solve this problem. Now I have no idea. And another question is : How to enable IPv6 from the settings? Any advice from you will be appreciated. Jacky ___________________________________________________________ 抢注雅虎免费邮箱3.5G容量，20M附件！ http://cn.mail.yahoo.com

2 1

[Users] Is there some functions which can append parameters for specific header field?
by 杨冬 15 May '07

15 May '07

hi, all: I've got such a problem recently. For our SIP client test, I must simulate a specific 180 message. There is a user-defined parameter in the Contact header field. I want to construct this message in openser server when it forwards the message to the caller. So I wonder there is some functions which can append a string parameter to the Contact header field. If it does, I can just add a process script into the configuration file of openser.cfg. Can anyone give some suggestions? Thanks. andy

3 3

[Users] avp question
by Jeremy George 15 May '07

15 May '07

Can someone help me with translating this ser config snippet to openser? Thanks. if (avp_db_load("$ruri/username", "s:anonreject")) { if (avp_check("s:anonreject", "eq/y/i)") && search("^(f|F)rom:.*(a|A)nonymous")) { avp_delete("$avp(s:anonreject)"); - Jeremy

5 6

[Users] Permit only one register by user!!!
by Jeferson Prevedello 14 May '07

14 May '07

Hi Guys, I able the authentication in my openser based in mysql. The method is perfect, but I perceived that the same user can authenticate many time using the same user :-( Somebody knows as permit only one register by user ? Regards Jeferson

2 2

[Users] CDRTool web admin login help
by Chahn Kim 14 May '07

14 May '07

Hi, First index.phtml opens with "document.forms:0.0 is null or not an object" script error in IE 6.0. Then subsequent login attempts just repeat the same login page. Password never been changed since the installation. Can someone guide me where to start looking at? Thanks in advance. CDRTool: 4.7-7 PHP: 4.4 John C Kim

1 0

[Users] How to use script variables in switch-cases?
by Peter Lemenkov 14 May '07

14 May '07

Hello All! I want to do something like this (using svn checkout of OpenSER): switch ($src_ip) { case $var(ip1): # do smth break; case $var(ip2): # do smth break; default: # do smth } I've got an error message (syntax error). Is it possible to use variables in such manner at all? -- With best regards!

3 6

[Users] XCAP Lite 0.9.3 released
by Adrian Georgescu 14 May '07

14 May '07

Change log xcap-lite (0.9.3) unstable; urgency=low * Fixed sync of active_watchers and watchers tables xcap-lite (0.9.2) unstable; urgency=low * Fixed fetch attribute to work with MDB2 driver * Added Content-length header The software can be downloaded from: http://download.dns-hosting.info/XCAP/xcap-lite-0.9.3.tar.gz Regards, Adrian

1 0

[Serusers] dns failover
by Mário Ferreira 14 May '07

14 May '07

Hi, I am interested in using SER DNS failover, but I am getting some problems. A failure can be a 503 error response, ICMP errors or a timeout due to a not received response. Does anyone know if SER is able to detect these failures and then select an alternative destination for the message? If the ICMP errors can't be detected, can this be solved with timers? I turned on use_dns_cache and use_dns_failover, but when the message fails to be delivered to the first destination, SER never tries the second destination, which was configured with DNS SRV. regards, Mário Ferreira

2 4

[Serusers] Question about SER 2.0 config item
by SIP 14 May '07

14 May '07

While I'm attempting to redo our SER 0.9.6 config in SER 2.0, I have a question concerning migration from an OpenSER 0.9.X module that we user in our SER config, alias_db. Currently, all our aliases lookups are done via alias_db. Since we provision DIDs on the fly via the web, and all our DIDs are E164 format forwarded to our SER server, it didn't make sense to add in the security issue of direct server access from the web server to reload the SER config anytime there was a change to the aliases. Instead, we opted for having everything in the DB where it should be, and we use the alias_db module to keep track of our aliases accordingly. So the question arises... how does one migrate this to the new SER 2.0? I'm still only barely able to grasp the whole concept of Aliases in the new SER 2.0 stuff, but our current table has information like: | 18005551212 | ourproxy.com | 1234567 | ourproxy.com | This gives the alias of 18005551212(a)ourproxy.com a rewrite of 1234567(a)ourproxy.com Simple enough concept. We add or remove aliases from the DB as necessary, and never have to touch the running SER server. So... how does one do this sort of thing now? N.

2 1

[Users] SIPit 20 summary
by Klaus Darilion 14 May '07

14 May '07

http://www1.ietf.org/mail-archive/web/sip/current/msg18959.html regards klaus

2 1

Re: [Users] avp question
by Jeremy George 14 May '07

14 May '07

Aargh, I swear I checked that. Thanks, Steve. - Jeremy >-----Original Message----- >From: Steve Blair [mailto:blairs@isc.upenn.edu] >Sent: Sunday, May 13, 2007 02:21 PM >To: 'Jeremy George' >Cc: ''Bogdan-Andrei Iancu'', users(a)openser.org >Subject: Re: [Users] avp question > > >Jeremy: > > The number of parenthesis in the avp_db_load call is unbalanced. > >-steve > >Jeremy George wrote: > >>Bogdan, >> >> Thank you very much. I'm a step closer, but I've still got something >>wrong. This code: >> >>modparam("avpops", "avp_url", "mysql://openser:<XXX>@localhost/openser2") >>modparam("avpops", "avp_table", "usr_preferences") >> >>route { >> >> if (avp_db_load("$ru/username", "$avp(s:anonreject)")) { >> >> if (avp_check("$avp(s:anonreject)", "eq/y/i)") && >> search("^(f|F)rom:.*(a|A)nonymous")) { >> >> avp_delete("$avp(s:anonreject)"); >> }; >> }; >>} >> >>yields these error messages: >> >>WARNING: no fork mode >> 0(3491) init_tcp: using epoll_lt as the io watch method (auto detected) >> 0(0) INFO: statistics manager successfully initialized >> 0(0) StateLess module - initializing >> 0(0) TM - initializing... >> 0(0) ACC - initializing >> 0(0) Maxfwd module- initializing >> 0(0) INFO:ul_init_locks: locks array size 512 >> 0(0) AUTH module - initializing >> 0(0) AUTH_DB module - initializing >> 0(0) TextOPS - initializing >> 0(0) EXEC - initializing >> 0(0) AVPops - initializing >> 0(0) ERROR:avpops:parse_check_value: unknown flag <)> >> 0(0) ERROR:avpops:fixup_check_avp: failed to parse checked value >> 0(0) ERROR: fix_actions: fixing failed (code=-1) at cfg line 74 >> 0(0) ERROR: fix_expr : fix_actions error >>ERROR: error -1 while trying to fix configuration >> 0(0) INFO:mi_fifo:mi_destroy: process hasn't been created -> nothing to >>kill >> >> Thanks for the help; I appreciate it. >> >>- Jeremy >> >> >>-----Original Message----- >>From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] >>Sent: Thursday, May 10, 2007 12:05 PM >>To: Jeremy George >>Cc: users(a)openser.org >>Subject: Re: [Users] avp question >> >>Hi Jeremy, >> >>For openser 1.2 it will be: >> >> if (avp_db_load("$ru/username", "$avp(s:anonreject)")) { >> if (avp_check("$avp(s:anonreject)", "eq/y/i)") && >>search("^(f|F)rom:.*(a|A)nonymous")) { >> avp_delete("$avp(s:anonreject)"); >> >> >>regards, >>bogdan >> >>PS: see http://www.openser.org/docs/modules/1.2.x/avpops.html >> >>Jeremy George wrote: >> >> >>> Can someone help me with translating this ser config snippet to openser? >>>Thanks. >>> >>> if (avp_db_load("$ruri/username", "s:anonreject")) { >>> if (avp_check("s:anonreject", "eq/y/i)") && >>>search("^(f|F)rom:.*(a|A)nonymous")) { >>> avp_delete("$avp(s:anonreject)"); >>> >>>- Jeremy >>> >>> >>>_______________________________________________ >>>Users mailing list >>>Users(a)openser.org >>>http://openser.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >>> >> >> >> >> >>_______________________________________________ >>Users mailing list >>Users(a)openser.org >>http://openser.org/cgi-bin/mailman/listinfo/users >> >> >> >

2 1

[Serusers] nathelper support for Ottendorf
by Ladislav Andel 12 May '07

12 May '07

Hi all, I have added nathelper support to basic ser.cfg file of Ottendorf release. It is a simple setup without any complexity of using other features such as connectivity_realm etc. You can download it at http://www.iptel.org/ser/howtos/optimizing_the_use_of_rtp_proxy File is called ser_config_2_0.txt The more complex one will come with updated document soon. Lada

2 2

Re: [Serusers] SER/PSTN Gateway conf
by flavio 12 May '07

12 May '07

Hi to all, I've updated my ser.cfg file as follow. Essentially I've updated "Call Type Processing" Section, with introduction of BYE SIP message handlling, and route[2] section. So SER now reply to BYE message from PSTN Gateway (in my scenario Phone --> PSTN --> Gateway --> SER --> SIP Phone) just with 200 OK, but does not forward BYE to SIP Phone. I hope this may be helpfully! Thanks, flavio debug=3 fork=no log_stderror=yes listen=10.28.19.202 # INSERT YOUR IP ADDRESS HERE port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://root@10.28.19.202/ser" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" modparam("auth_db|permissions|uri_db|usrloc", "db_url", "mysql://root@10.28.19.202/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("usrloc", "db_mode", 2) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("permissions", "default_allow_file", "/usr/local/etc/ser/permissions.allow") modparam("permissions", "default_deny_file", "/usr/local/etc/ser/permissions.deny") route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("16")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (method=="INVITE") { if (!allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(1); break; }; if (method=="ACK") { route(3); route(1); break; } if (method=="INVITE") { route(3); break; } else if (method=="REGISTER" || method=="BYE") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- if (!t_relay()) { sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!allow_trusted()){ if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; } }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); } lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { if (uri=~"^sip:0[0-9]*@") { # Domestic PSTN #strip(0); route(4); break; }; if(uri=~"^sip:3[0-9]*@") { route(4); break; }; if (allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","1"); break; }; }; sl_send_reply("404", "User Not Found"); break; }; route(1); } route[4]{ #----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- rewritehost("10.28.52.105"); # INSERT YOUR PSTN GATEWAY IP ADDRESS avp_write("i:45", "inv_timeout"); route(1); }

2 2

[Serusers] Around the world to get to my....
by SIP 11 May '07

11 May '07

Is the developer documentation on the Iptel.org site even close to still valid? We've just created a semi-complex AGI in Asterisk to sort of act as a gateway for locked SEMS conferences. Our users can create a conference number (if it's available), assign a pin, and then everyone who calls that conference is required to enter the correct pin in order to join. We keep track of the users logged in (contact info, since it's not always local users), and the owner has the ability to limit the number of users, etc. Ideally, I'd like to task one of my C developers to create a similar sort of gateway module for SER instead of, as we currently do, shunting everyone over to the Asterisk server for processing and logic (and voice prompts) and then shunting them over to the SER/SEMS server if they're authenticated. Of course, such a module would need to be able to do things like complete take apart the SEMS pin-collect module (we're not interested in using python for anything, nor in using XMLRPC) and hack it up to collect the right DTMF signals, passing that data on to a proper SER module. I'm just wondering if the developer docs for SER are up to date enough to even point my developers in that direction for information on the SER portion of things. N.

3 2

[Users] call park/pickup using openser
by zm23 11 May '07

11 May '07

Hi All. Has anyone implemented call park / pickup feature using OpenSER and asterisk? Or maybe OpenSER and something else ? Any help / suggestion is greatly appreciated. Thanks. -- Zahid

2 2

[Serusers] Proper handling of REFER
by Steve Blair 11 May '07

11 May '07

We just started experiencing a problem with call transfers. This use to work but we cannot figure out what broke. We are investigating our gateway software but I wanted to check with the SER community too. If we receive a REFER method we simply t_relay(). Is this the correct handling ? Thanks,Steve

3 2

[Users] Differentiate INVITEs
by Kristian Kielhofner 11 May '07

11 May '07

Hello Everyone, I am trying to implement music on hold with Asterisk. I know that this has been asked before, but I have searched through the archives and I think I am doing something a bit different... Here is the situation: Components: Phone (registered to OpenSER) Cisco GW (connects to PSTN) OpenSER (in the middle of this mess) Diagram Outgoing Call: Phone -> OpenSER (Authentication, LCR, record route, NAT traversal, yada yada) -> Cisco GW Incoming: Cisco GW -> OpenSER (locations lookup, record route, NAT traversal, yada yada) -> Phone OpenSER catches any INVITE from the phone and routes it as necessary. The above config works perfectly except for hold/music on hold. As I understand it, music on hold is implemented with a re-INVITE (from the phone) with a 0.0.0.0 address in the SDP (c=IN IP4 0.0.0.0). I think that if I had a way to check for this I could route the re-INVITE for hold to an Asterisk server to run music on hold instead of routing it to the Cisco GW (which is what happens now). A few questions: 1) Is there a better way to differentiate a re-INVITE for hold/moh vs. just a regular INVITE? 2) Is there a way to do this in OpenSER (either using some kind of matching based on SDPs or some other method)? 3) Is there any reason this wouldn't work? I'm thinking that the Asterisk side would be fairly simple, with a context like this for the incoming INVITEs to play music on hold: exten => _NXX.,1,Answer ; (can't hurt) exten => _NXX.,n,MusicOnHold(default) exten => _NXX.,n,Hangup ... or something to that effect. What do you think? Any and all comments welcome. Thanks! -- Kristian Kielhofner

3 4

[Users] SIP trunking requirements
by A.M. 11 May '07

11 May '07

Hi, I'm working on a project involving a SIP trunking provider that requires a b2bua with registration. I noticed on the home page that openser is not a b2bua. Is there an architectural limitation of openser that prevents it from being a b2bua? A common setup for the b2bua will be NAT traversal. Currently, we are using an Ingate SIParator 19 appliance but we would rather use pure software. I haven't been able to find an open source b2bua that can bind to multiple addresses for NAT traversal and register with the SIP trunk provider. Asterisk is pretty heavyweight for the basic SIP mangling I want to do. Any suggestions? Thanks, M

2 1

[Serusers] Capturing all messages for testing purposes
by POON Ben 11 May '07

11 May '07

Hi, I am setting up a testing environment with about 100 users. I would like to see who is online/registered, conversation times, protocols, etc. Basically I want to be able to capture all the messages for troubleshooting purposes, durability, compatibility, reliability, etc. Will the acc module do that for me, how do I set it up to capture the data? Thank you very much, Ben

4 6

[Serusers] unregister problem
by Pezhman Lali 11 May '07

11 May '07

hi I am using ser+asterisk(with a2billing), all of users must be registerd, for accepting call, but the unregistered users can place the calls ,with a valid number as callerid. as u now it is too critical bug for my system, thanks in advance Best Mani __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 1

[Serusers] How to run SER in background?
by karl lu 11 May '07

11 May '07

Hi, The get started documnt says if I type: ser D E, it will start ser in foreground. Then if I just type in ser, it will start ser in background. Is that right? It seems like when I only type in ser, it also runs in foreground. How can I run ser in backgroung? Thanks, Karl

2 1

[Users] [openser + Asterisk] Caller don't hear callee
by Bodin Bruno 11 May '07

11 May '07

Hello with this config, a caller can call a calle , but callee don't send any RTP message to asterisk :( What 's the problem ? Thank, route{ 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; }; if (!method=="REGISTER") record_route(); if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; if(method=="CANCEL" || method=="BYE") { setflag(10); }; if ((src_ip==127.0.0.1) && (src_port==5062) && (dst_port==5060)) { if (method=="REGISTER") { save("location"); return; } if (!lookup("location")) { sl_send_reply("404", "not found"); return; }; setflag(9); route(1); return; }; if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(1); }; if (uri==myself) { if (method=="REGISTER") { save("location"); return; }; if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; }; route(1); } route[1] { if(method=="INVITE" && !isflagset(10) && !isflagset(9)) { revert_uri(); $ru = "sip:"+$rU+"@"+$rd+":5062"; }; if (!t_relay()) { sl_reply_error(); }; exit; }

2 2

[Users] How I configure kdevelop to see the debug of any module?
by Lou 10 May '07

10 May '07

I don't debugger my module.so with kdevelop I run openser with fork=no but, I can't debug a module using kdevelop. __________________________________________________ Preguntá. Respondé. Descubrí. Todo lo que querías saber, y lo que ni imaginabas, está en Yahoo! Respuestas (Beta). ¡Probalo ya! http://www.yahoo.com.ar/respuestas

2 1

[Serusers] Problem with ser-mysql pakage on Debian
by Daniel Sailer 10 May '07

10 May '07

Hi ser specialists, I have a problem with the installtion of the "ser-mysql-module_0.9.6-0.1_i386.deb" module. If I want to install the pakage with: "dpkg -i ser-mysql-module_0.9.6-0.1_i386.deb" I get the following error: ######## ser:/home/ser# dpkg -i ser-mysql-module_0.9.6-0.1_i386.deb Wähle vormals abgewähltes Paket ser-mysql-module. (Lese Datenbank ... 79463 Dateien und Verzeichnisse sind derzeit installiert.) Entpacke ser-mysql-module (aus ser-mysql-module_0.9.6-0.1_i386.deb) ... dpkg: Abhängigkeitsprobleme verhindern Konfiguration von ser-mysql-module: ser-mysql-module hängt ab von libmysqlclient15; aber: Paket libmysqlclient15 ist nicht installiert. dpkg: Fehler beim Bearbeiten von ser-mysql-module (--install): Abhängigkeitsprobleme - lasse es unkonfiguriert Fehler traten auf beim Bearbeiten von: ser-mysql-module ######## TranslationFehler traten auf beim Bearbeiten von ######## ser:/home/ser# dpkg -i ser-mysql-module_0.9.6-0.1_i386.deb Select previously not selected pakage ser-mysql-module. (Read Database ... 79463 Files and Directorys currently installed.) Unpack ser-mysql-module (from ser-mysql-module_0.9.6-0.1_i386.deb) ... dpkg: Dependency problems prevent configuration of ser-mysql-module: ser-mysql-module depends on libmysqlclient15; but: pakage libmysqlclient15 is not installed. dpkg: Error while processing of ser-mysql-module (--install): Dependency problems - leaving unconfigured An Error occured while processing of: ser-mysql-module ######## libmysqlclient15off and libmysqlclient15-dev are installed already. Kind regards Daniel

1 0

[Users] Errors during a call and bad communication
by Bodin Bruno 10 May '07

10 May '07

Hi, anyone know something about this, when a twinkle UA call a eyebeam UA (Tls or not TLS) I get theses errors : 2(9486) ERROR: tcp_blocking_connect: timeout 10 s elapsed from 10 s 2(9486) ERROR: tcpconn_connect: tcp_blocking_connect failed 2(9486) ERROR: tcp_send: connect failed 2(9486) msg_send: ERROR: tcp_send failed 2(9486) ERROR:tm:t_forward_nonack: sending request failed 9(9500) WARNING: set_timer for 3 list called on a "detached" timer -- ignoring: 0x2adbfa551ee0 2(9486) ERROR: tcp_blocking_connect: timeout 10 s elapsed from 10 s 2(9486) ERROR: tcpconn_connect: tcp_blocking_connect failed 2(9486) ERROR: tcp_send: connect failed 2(9486) msg_send: ERROR: tcp_send failed 2(9486) ERROR:tm:t_forward_nonack: sending request failed But when eyebeam call twinkle , no errors . thank for all.

2 1

[Users] call transfer to asterik.. asterisk as an end point
by zm23 10 May '07

10 May '07

> > Hello All. > I am having some trouble with call transfers when asterisk is > the 2nd party called and I hope to benefit from your experience. > > I want to use asterisk for call park/pickup and have configured > openser to relay calls made to ruri 700-720 to asterisk running on > localhost:5069 > > > Call flow: > > phone A calls phone B (both phones are polycom) > > Phone B answers > > then phone b user presses transfer and dials 700 > > asterisk plays back 701 as the parking lot location > > phone B user presses transfer again. > at this time phone b is not disconnected from asterisk system > phone A is also connected to asterisk and hears 702 as the > parking lot location (as if asterisk places the user at priority 1 > for that > context) > > > From phone C calling 702 will connect phone C to phone A. > > > This was a specific example but this transfer problem is not > limited to call park only. It happens any time asterisk is the > second party called in call transfer. > > Thanks in advance for your help. > > -- > Zahid > > > > > > On May 8, 2007, at 1:56 PM, Christian Schlatter wrote: > > > > I think I found out why this doesn't work as expected. After phone > 1 receives REFER from phone 2, it sends a new INVITE to the > asterisk server. This INVITE includes a Replaces: header that tells > the receiver (asterisk) to replace an existing SIP dialog with the > new one. > > RFC 3891 "The SIP Replaces Header", Section 3 "UAS Behavior", defines: > > "the UA attempts to accept the new INVITE, reassign the user > interface and other resources of the matched dialog to the new > INVITE, and shut down the replaced dialog." > > But your SIP trace shows that asterisk doesn't shut down the > replaced dialog (by sending a BYE), which is the reason why phone 2 > does not get disconnected after hitting "transfer" the second time. > > > Instead of creating a new call park slot (702) when phone 1 sends > the Replaces: INVITE to asterisk, asterisk should be intelligent > enough to figure out that this INVITE actually replaces the > existing SIP dialog with phone 2. And asterisk should not create a > new park slot 702 but directly put phone 1 on hold at park slot 701 > and send a BYE to phone 2. > > Although asterisk supports the Replaces: header when used e.g. as a > gateway, I have some doubts that the call park/pickup > implementation does so too. Especially since it was designed to be > used in "PBX mode" where asterisk acts as B2BUA for all involved > call legs. > > Maybe this should be opened as a new feature/bug request on the > asterisk bug tracker. Or maybe there is a asterisk setting that > controls this behavior, I'm not really an asterisk expert myself ;-) > > -- >

1 0

RE: [Serusers] global contact list
by Barta János 10 May '07

10 May '07

Hello! Thanks Vaclav, it is clear now. I modified the contacts settings in the client, but nothing changed :-(. I received the next error messages: 3(16917) ERROR: rls_handler.c:484: XCAP query problems 3(16917) XCAP query failed What should I see in the MySQL querry? I can see now (in the global contact list case): select uid,flags from uri where username='bartaj-list' AND did='10.22.1.210' , but in the answer (Response OK) the Payload field is empty. mysql> select uid,flags from uri where username='bartaj-list' AND did='10.22.1.210'; Empty set (0.00 sec) In the right case (user contact list) the answer contained the username parameter: select uid,flags from uri where username='bartaj' AND did='10.22.1.210' Answer -> bartaj mysql> select uid,flags from uri where username='bartaj' AND did='10.22.1.210'; -------- ------- | uid | flags | -------- ------- | bartaj | 61 | -------- ------- 1 row in set (0.00 sec) mysql> Maybe is this the problem? Do I have to add any other parameters into the MYSQL database (in the global lists cases)? By, Jani -----Original Message----- From: Vaclav Kubart [mailto:vaclav.kubart@iptel.org] Sent: Friday, April 20, 2007 9:56 AM To: Barta János Cc: serusers(a)iptel.org Subject: Re: [Serusers] global contact list Hi, I'm very sorry for delayed reply but... ;-) Your config seems to be OK for first (and second) look. I have used list subscripriptions with EyeBeam in this way: I added the list URI in my contact list (exactly the same way like other contacts), so I had something like: contacts: - user1(a)somewhere.com - user2(a)somewhere.com - hr(a)somewhere.com - ... - franta-list(a)somewhere.com Where userx@... are 'normal' users and hr@... is resource list as defined in global resource list - in your case it could be bartaj-list(a)test.intra or hr2(a)test.intra. The franta-list@... is uri used to subscribe to user's resource list when I'm using my user with uri franta(a)somewhere.com (i.e. the resource list on XCAP is stored in user's directory as described in presence handbook). After the subscription is created, the list in contact window was expanded and seemd something like: - user1(a)somewhere.com - user2(a)somewhere.com - hr(a)somewhere.com | - smith(a)somewhere.com - ... - ... If I tried to use "Subscribe to contact list" feature, I think I had to put franta-list(a)somewhere.com as contact-list URI (or hr(a)somewhere.com could be used too). But in this case EyeBeam was not sending the Supported:eventlist header thus it was unusable for me... Have you seen any error messages in SER's log? Vaclav On Tue, Apr 17, 2007 at 05:01:24PM 0200, Barta János wrote: > Hello! > > Thanks for your answer again, and I'm sorry for the delayed answer, but I was out of office. > I attached my config file. > > My biggest problem is, that I don't know, what I should see :-(. > I have an eyeBeam window, that contains the "Calls & Conntats" parameters. > > In this window I can see my (user) contact list, but I didn't see the global one. > For example (the PC is my user contact list): > > -PC > -105 > -12345 > -barta > > and it is working fine, but I can not see here (in this window) the global one: > > sip:~# cat /var/www/xcap-root/rls-services/global/index > <?xml version="1.0" encoding="UTF-8"?> <rls-services> > > <service uri="sip:bartaj-list@test.intra"> <list name="Human > resources"> <entry uri="sip:test1@test.intra"> > <display-name>bartaj</display-name> > </entry> > <entry uri="sip:test2@test.intra"> > <display-name>12345</display-name> > </entry> > </list> > <packages> > <package>presence</package> > </packages> > </service> > > <service uri="sip:hr2@test.intra"> > <list name="Human resources"> > <entry uri="sip:test1@test.intra"> > <display-name>bartaj</display-name> > </entry> > <entry uri="sip:test2@test.intra"> > <display-name>12345</display-name> > </entry> > </list> > <packages> > <package>presence</package> > </packages> > </service> > > </rls-services> > > > I think in that case I should see the next tree: > > -PC > -105 > -12345 > -barta > > -Human resources > -bartaj > -12345 > > > > As You asked, I made a capture between the SER and XCAP. > > It was a little difficult, because I didn't see interesting traffic between the SER and XCAP, but a few minutes later I realised, that the SER for the communication (in that case) uses the loopback (lo) interface. > > You can see here the capture, that I made on the lo interface: > > 1 0.000000 127.0.0.1 -> 127.0.0.1 TCP 37512 > www [SYN] Seq=0 Len=0 > MSS=16396 TSV=3808162073 TSER=0 WS=0 > 2 0.000056 127.0.0.1 -> 127.0.0.1 TCP www > 37512 [SYN, ACK] Seq=0 > Ack=1 Win=32767 Len=0 MSS=16396 TSV=3808162073 TSER=3808162073 WS=0 > 3 0.000065 127.0.0.1 -> 127.0.0.1 TCP 37512 > www [ACK] Seq=1 Ack=1 > Win=32767 Len=0 TSV=3808162073 TSER=3808162073 > 4 0.000072 127.0.0.1 -> 127.0.0.1 HTTP GET > /xcap/pres-rules/users/12345/presence-rules.xml HTTP/1.1 > 5 0.000079 127.0.0.1 -> 127.0.0.1 TCP www > 37512 [ACK] Seq=1 Ack=113 > Win=32767 Len=0 TSV=3808162073 TSER=3808162073 > 6 0.000545 127.0.0.1 -> 127.0.0.1 HTTP HTTP/1.1 200 OK > 7 0.000559 127.0.0.1 -> 127.0.0.1 TCP 37512 > www [ACK] Seq=113 > Ack=262 Win=32767 Len=0 TSV=3808162074 TSER=3808162074 > 8 0.000571 127.0.0.1 -> 127.0.0.1 HTTP Continuation or non-HTTP > traffic > 9 0.000578 127.0.0.1 -> 127.0.0.1 TCP 37512 > www [ACK] Seq=113 > Ack=870 Win=32767 Len=0 TSV=3808162074 TSER=3808162074 10 0.005535 > 127.0.0.1 -> 127.0.0.1 TCP 37513 > www [SYN] Seq=0 Len=0 MSS=16396 > TSV=3808162079 TSER=0 WS=0 > 11 0.005556 127.0.0.1 -> 127.0.0.1 TCP www > 37513 [SYN, ACK] Seq=0 > Ack=1 Win=32767 Len=0 MSS=16396 TSV=3808162079 TSER=3808162079 WS=0 > 12 0.005565 127.0.0.1 -> 127.0.0.1 TCP 37513 > www [ACK] Seq=1 Ack=1 > Win=32767 Len=0 TSV=3808162079 TSER=3808162079 > 13 0.005616 127.0.0.1 -> 127.0.0.1 HTTP GET > /xcap/pres-rules/users/bartaj/presence-rules.xml HTTP/1.1 > 14 0.005623 127.0.0.1 -> 127.0.0.1 TCP www > 37513 [ACK] Seq=1 Ack=114 > Win=32767 Len=0 TSV=3808162079 TSER=3808162079 > 15 0.005989 127.0.0.1 -> 127.0.0.1 HTTP HTTP/1.1 200 OK > 16 0.006000 127.0.0.1 -> 127.0.0.1 TCP 37513 > www [ACK] Seq=114 > Ack=262 Win=32767 Len=0 TSV=3808162079 TSER=3808162079 > 17 0.006011 127.0.0.1 -> 127.0.0.1 HTTP Continuation or non-HTTP > traffic > 18 0.006017 127.0.0.1 -> 127.0.0.1 TCP 37513 > www [ACK] Seq=114 > Ack=870 Win=32767 Len=0 TSV=3808162079 TSER=3808162079 > 19 2.979491 127.0.0.1 -> 127.0.0.1 HTTP GET > /xcap/rls-services/global/index HTTP/1.1 20 2.979722 127.0.0.1 -> > 127.0.0.1 HTTP HTTP/1.1 200 OK > 21 2.979734 127.0.0.1 -> 127.0.0.1 TCP 37512 > www [ACK] Seq=209 > Ack=1126 Win=32767 Len=0 TSV=3808165053 TSER=3808165053 > 22 2.979745 127.0.0.1 -> 127.0.0.1 HTTP Continuation or non-HTTP > traffic > 23 2.979751 127.0.0.1 -> 127.0.0.1 TCP 37512 > www [ACK] Seq=209 > Ack=1887 Win=32767 Len=0 TSV=3808165053 TSER=3808165053 > > > > You can see, that in the message number 19 the SER downloaded the index file. > > This file contains correct information: > > > sip:~# wget 127.0.0.1/xcap/rls-services/global/index > --17:22:43-- http://127.0.0.1/xcap/rls-services/global/index > => `index' > Connecting to 127.0.0.1:80... connected. > HTTP request sent, awaiting response... 200 OK > Length: 761 [text/plain] > > 100%[====================================>] 761 --.--K/s > > 17:22:43 (7.26 MB/s) - `index' saved [761/761] > > sip:~# > sip:~# cat index > <?xml version="1.0" encoding="UTF-8"?> <rls-services> > > <service uri="sip:bartaj-list@test.intra"> <list name="Human > resources"> <entry uri="sip:test1@test.intra"> > <display-name>bartaj</display-name> > </entry> > <entry uri="sip:test2@test.intra"> > <display-name>12345</display-name> > </entry> > </list> > <packages> > <package>presence</package> > </packages> > </service> > > <service uri="sip:hr2@test.intra"> > <list name="Human resources"> > <entry uri="sip:test1@test.intra"> > <display-name>bartaj</display-name> > </entry> > <entry uri="sip:test2@test.intra"> > <display-name>12345</display-name> > </entry> > </list> > <packages> > <package>presence</package> > </packages> > </service> > > </rls-services> > sip:~# > > > > And at last, how can I add query_rls_services and query_resource_list log informations into the config file? > > Thx, > Jani > > > > Hi, > this part of config file is not sufficient for me, but I guess that > RLS related things stayed as in the original config you mentioned... > In which way it doesn't work? (For example it returns non OK > responses?) Please could you try to capture HTTP traffic between XCAP > server and SER? (If SER really queries the right resource list > document.) > > You can add more log information around important config parts (query_rls_services and query_resource_list functions are most important here). > > BTW the second message you sent is not subscription to resource list or user, it is subscription to event package 'sip-profile' (content of Event header field) which is not supported by SER. > > It is possible to use registration status as source for presence - in this case you have to have PA module on your server handling registrations and PA module parameter 'use_callbacks' must be set to 1. > > Vaclav > > > > -----Original Message----- > From: Vaclav Kubart [mailto:vaclav.kubart@iptel.org] > Sent: Tuesday, April 03, 2007 11:18 AM > To: Barta János > Cc: serusers(a)iptel.org > Subject: Re: [Serusers] global contact list > > > ______________________________________________________________________ > _________ Állást keres? 3300 lehetőséget kínál a CV-Online Ne > keressen tovább! Portálunkon Magyarország vezető vállalatainak több mint 3300 friss állásajánlatából válogathat! > > http://ad.adverticum.net/b/cl,1,6022,139845,209300/click.prm > sip:/opt/etc/ser# cat ./ser_pa.cfg > log_stderror=yes > debug=5 # debug level (cmd line: -dddddddddd) > check_via=no # (cmd. line: -v) > dns=no # (cmd. line: -r) > rev_dns=no # (cmd. line: -R) > port=5060 > children=2 > alias="test.intra" > #alias="ipp.test.intra" > alias="10.22.1.210" > mhomed=yes # usefull for multihomed hosts, small performance penalty > #tcp_accept_aliases=yes # accepts the tcp alias via option (see NEWS) > #tcp_poll_method="sigio_rt" > > # ------------------ module loading ---------------------------------- > > loadmodule "/opt/lib/ser/modules/sl.so" > loadmodule "/opt/lib/ser/modules/avp.so" > loadmodule "/opt/lib/ser/modules/avpops.so" > loadmodule "/opt/lib/ser/modules/tm.so" > loadmodule "/opt/lib/ser/modules/rr.so" > loadmodule "/opt/lib/ser/modules/maxfwd.so" > loadmodule "/opt/lib/ser/modules/usrloc.so" > loadmodule "/opt/lib/ser/modules/registrar.so" > loadmodule "/opt/lib/ser/modules/textops.so" > loadmodule "/opt/lib/ser/modules/mysql.so" > loadmodule "/opt/lib/ser/modules/dialog.so" > loadmodule "/opt/lib/ser/modules/rls.so" > loadmodule "/opt/lib/ser/modules/pa.so" > loadmodule "/opt/lib/ser/modules/presence_b2b.so" > loadmodule "/opt/lib/ser/modules/uri.so" > loadmodule "/opt/lib/ser/modules/uri_db.so" > loadmodule "/opt/lib/ser/modules/domain.so" > loadmodule "/opt/lib/ser/modules/fifo.so" > loadmodule "/opt/lib/ser/modules/xmlrpc.so" > loadmodule "/opt/lib/ser/modules/xlog.so" > #loadmodule "/opt/lib/ser/modules/unixsock.so" > loadmodule "/opt/lib/ser/modules/msilo.so" > > # Uncomment this if you want digest authentication # mysql.so must be > loaded ! > loadmodule "/opt/lib/ser/modules/auth.so" > #loadmodule "/opt/lib/ser/modules/auth_db.so" > > loadmodule "/opt/lib/ser/modules/acc_radius.so" > loadmodule "/opt/lib/ser/modules/mediaproxy.so" > loadmodule "/opt/lib/ser/modules/auth_radius.so" > > # ----------------- setting module-specific parameters --------------- > ############################# modparam("mediaproxy", > "mediaproxy_socket", "/var/run/mediaproxy.sock") > modparam("auth_radius","radius_config","/etc/radiusclient-ng/radiuscli > ent.conf") > > ######################### > # -- acc params -- # > ######################### > modparam("acc_radius","radius_config","/etc/radiusclient-ng/radiusclie > nt.conf") > modparam("acc_radius", "log_flag", 1) > modparam("acc_radius", "log_missed_flag", 2) > > > modparam("msilo","registrar","sip:registrar@test.intra") > modparam("msilo","use_contact",0) > modparam("msilo","expire_time",7200) > > # -- usrloc params -- > > # -- 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) > > modparam("rls", "min_expiration", 200) modparam("rls", > "max_expiration", 300) modparam("rls", "default_expiration", 300) > modparam("rls", "auth", "none") modparam("rls", "xcap_root", > "http://localhost/xcap") modparam("rls", "reduce_xcap_needs", 1) > modparam("rls", "db_mode", 1) modparam("rls", "db_url", > "mysql://xxxxxxxxxxx:yyyyyyyyy@10.22.1.222/ser_pa") > > modparam("pa", "use_db", 1) > # allow storing authorization requests for offline users into database > modparam("pa", "use_offline_winfo", 1) # how often try to remove old > stored authorization requests modparam("pa", "offline_winfo_timer", > 600) # how long stored authorization requests live modparam("pa", > "offline_winfo_expiration", 600) modparam("pa", "db_url", > "mysql://xxxxxxxxxxx:yyyyyyyyy@10.22.1.222/ser_pa") > # mode of PA authorization: none, implicit or xcap modparam("pa", > "auth", "xcap") modparam("pa", "auth_xcap_root", > "http://localhost/xcap") # do not authorize watcherinfo subscriptions > modparam("pa", "winfo_auth", "none") # use only published information > if set to 0 modparam("pa", "use_callbacks", 1) # don't accept internal > subscriptions from RLS, ... > modparam("pa", "accept_internal_subscriptions", 1) # maximum value of > Expires for subscriptions modparam("pa", > "max_subscription_expiration", 600) # maximum value of Expires for > publications modparam("pa", "max_publish_expiration", 120) # how often > test if something changes and send NOTIFY modparam("pa", > "timer_interval", 10) > > # route for generated SUBSCRIBE requests for presence > modparam("presence_b2b", "presence_route", > "<sip:127.0.0.1;transport=tcp;lr>") > # waiting time from error to new attepmt about SUBSCRIBE > modparam("presence_b2b", "on_error_retry_time", 60) # how long wait > for NOTIFY with Subscription-Status=terminated after unsubscribe > modparam("presence_b2b", "wait_for_term_notify", 33) # how long before > expiration send renewal SUBSCRIBE request modparam("presence_b2b", > "resubscribe_delta", 30) # minimal time to send renewal SUBSCRIBE > request from receiving previous response modparam("presence_b2b", > "min_resubscribe_time", 60) # default expiration timeout > modparam("presence_b2b", "default_expiration", 3600) # process > internal subscriptions to presence events modparam("presence_b2b", > "handle_presence_subscriptions", 1) > > modparam("usrloc", "db_mode", 2) > modparam("domain", "db_mode", 1) > #modparam("domain|uri_db|acc|auth_db|usrloc|msilo", "db_url", > "mysql://xxxxxxxxxxx:yyyyyyyyy@10.22.1.222/ser_pa") > modparam("domain|uri_db|auth_db|usrloc|msilo", "db_url", > "mysql://xxxxxxxxxxx:yyyyyyyyy@10.22.1.222/ser_pa") > > modparam("fifo", "fifo_file", "/tmp/ser_fifo") > > # ------------------------- request routing logic ------------------- > > # main routing logic > > route{ > # XML RPC > if (method == "POST" || method == "GET") { > create_via(); > dispatch_rpc(); > break; > } > > # 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"); > break; > }; > if (msg:len >= max_len ) { > sl_send_reply("513", "Message too big"); > break; > }; > > # 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(); > > if ((method=="INVITE") or (method=="BYE") or (method=="CANCEL")) > { > setflag(1); > } > > # 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); > break; > }; > > # 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 (!lookup_domain("To")) { > xlog("L_ERR", "Unknown domain to: %tu from: %fu\n"); > route(1); > break; > } > > if (method=="SUBSCRIBE") { > if (!t_newtran()) { > sl_reply_error(); > break; > }; > > if (@to.tag=="") { > # only for new subscriptions (with > empty to tag) > > if (lookup_user("To")) { > # existing user -> it is subscription to PA > if (handle_subscription("registrar")) { > if ((@msg.event=~"presence\.winfo")) { > # new watcher info subscription > # sends one watcher info NOTIFY message with all saved authorization requests > xlog("L_ERR", "dumping stored winfo to %fu\n"); > dump_stored_winfo("registrar", "presence"); > } > else { > # new presence subscription > if ((@msg.event=~"presence") && (%subscription_status=="pending")) { > # if offline user and new pending subscription > if (!target_online("registrar")) { > #%subscription_status="waiting"; # store it as waiting subscription > xlog("L_ERR", "storing 'pending' winfo to: %tu, from: %fu\n"); > store_winfo("registrar"); > } > } > } > } > break; > } > > if ((@msg.supported=~"eventlist")) { > # such user doesn't exist and Supported header field > # -> probably RLS subscription > > if (lookup_domain("From")) { > if (lookup_user("From")) { > if (is_simple_rls_target("$uid-list")) { > # log(1, "it is simple subscription!\n"); > # takes From UID and makes XCAP query for user's > # list named "default" > if (!query_resource_list("default")) { > t_reply("404", "No such user list"); > break; > } > } > } > } > > if (!have_flat_list()) { > # query_resource_list failed or was not called > # do standard RLS query acording to To/AOR > if (!query_rls_services()) { > log(1, "XCAP query failed\n"); > t_reply("404", "No such list URI"); > break; > } > } > > > if (!radius_www_authorize("test.intra")) { > www_challenge("test.intra", "0"); > break; > }; > handle_rls_subscription("1"); > } > else { > # not resource list subscription -> invalid user > xlog("L_ERR", "subscription to invalid user %tu\n"); > t_reply("404", "User not found"); > } > > break; > } > else { > # renewal subscriptions - try to handle it as RLS and if failed, handle it as PA subscription > # FIXME: better will be test like existing_rls_subscription() > # and existing_subscription("registrar") > if (!handle_rls_subscription("0")) { > handle_subscription("registrar"); > } > break; > } > }; > > # get user (common for all other messages than SUBSCRIBE) > if (!lookup_user("To")) { > # log(1, "Unknown user - message should be forwarded?"); > # # break; > append_hf("P-hint: unknown user\r\n"); > route(1); > break; > } > > if (method=="PUBLISH") { > if (!t_newtran()) { > # log(1, "newtran error\n"); > sl_reply_error(); > break; > }; > handle_publish("registrar"); > > # deliver messages to online user > # TODO: only if user goes from offline to online? > if (target_online("registrar")) { > # log(1, "Dumping stored messages\n"); > # dump stored messages - route it through myself (otherwise routed via DNS!) > if (m_dump("sip:127.0.0.1")) { > xlog("L_ERR", "MSILO: offline messages for %fu dumped\n"); > } > } > > break; > }; > > if (method=="NOTIFY") { > if (!t_newtran()) { > log(1, "newtran error\n"); > sl_reply_error(); > break; > }; > # handle notification sent in internal subscriptions (presence_b2b) > if (!handle_notify()) { > t_reply("481", "Unable to handle notification"); > } > break; > }; > > if (method=="MESSAGE") { > > if > (authorize_message("http://localhost/xcap")) { > > # use usrloc for delivery > if (lookup("location")) { > > log(1, "Delivering MESSAGE using usrloc\n"); > t_on_failure("1"); > if (!t_relay()) { > sl_reply_error(); > } > > break; > } > else { > # store messages for offline user > xlog("L_ERR", "MSILO: storing > MESSAGE for %tu\n"); > > if (!t_newtran()) { > log(1, "newtran error\n"); > sl_reply_error(); > break; > }; > > # store only text messages NOT isComposing... ! > if (search("^(Content-Type|c):.*application/im-iscomposing\ xml.*")) { > log(1, "it is only isComposing message - ignored\n"); > t_reply("202", "Ignored"); > break; > } > > if (m_store("0", "sip:127.0.0.1")) { > # log(1, "MSILO: offline message stored\n"); > if (!t_reply("202", "Accepted")) { > sl_reply_error(); > }; > } else { > log(1, "MSILO: error storing offline message\n"); > if (!t_reply("503", "Service Unavailable")) { > sl_reply_error(); > }; > }; > break; > } > break; > } > else { > # log(1, "unauthorized message\n"); > sl_reply("403", "Forbidden"); > } > break; > } > > if (method=="REGISTER") { > > if (!radius_www_authorize("test.intra")) { > www_challenge("test.intra", "0"); > break; > }; > > save("location"); > > # dump stored messages - route it through myself (otherwise routed via DNS!) > if (m_dump("sip:127.0.0.1")) { > xlog("L_ERR", "MSILO: offline messages for %fu dumped\n"); > } > break; > }; > > # native SIP destinations are handled using our USRLOC DB > if (!lookup("location")) { > sl_send_reply("404", "Not Found"); > break; > }; > }; > # 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(); > # }; > #} > > > #failure_route[1] { > # # forwarding failed -- check if the request was a MESSAGE > # if (!method=="MESSAGE") { break; }; > # log(1, "MSILO: MESSAGE forward failed - storing it\n"); > # > # # we have changed the R-URI with the contact address, ignore it now > # if (m_store("0", "")) { > # t_reply("202", "Accepted"); > # } else { > # log(1, "MSILO: offline message NOT stored\n"); > # t_reply("503", "Service Unavailable"); > # }; > #} > > > > route[1] > { > > if (method=="INVITE") > { > t_on_reply("1"); > t_on_failure("1"); > use_media_proxy(); > } > else if (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(); > }; > } > > > > onreply_route[1] > { > if (status=~"(183)|2[0-9][0-9]") { > use_media_proxy(); > }; > if (status=~"[3-6]0[0-9]") { > end_media_session(); > break; > } > } > failure_route[1] > { > # forwarding failed -- check if the request was a MESSAGE > if (!method=="MESSAGE") { break; }; > log(1, "MSILO: MESSAGE forward failed - storing it\n"); > # we have changed the R-URI with the contact address, ignore it now > if (m_store("0", "")) { > t_reply("202", "Accepted"); > } else { > log(1, "MSILO: offline message NOT stored\n"); > t_reply("503", "Service Unavailable"); > }; > end_media_session(); > } ___________________________________________________________________________________________ Lehet, hogy ma pont az ön kedvenc kötetét kínáljuk nagy engedménnyel? Érdemes betérni hozzánk, naponta új akcióval várjuk! http://ad.adverticum.net/b/cl,1,6022,165977,228257/click.prm

2 1

Re: [Serusers] FreeRadius brokes down during "Accounting Request"
by TZieleniewski 10 May '07

10 May '07

1 0

[Users] Integration to Openser with Media Server
by Danilo 10 May '07

10 May '07

Hi all, I wasn't Openser is a SIP proxy. There is Media Server. It works together with OpenSER. The OpenSER is monitoring the flow of the messges. In case of this flow to be RTP, the Openser redirects for the Media Server. How does it work? Did you understand? thanks Danilo Inovax Engenharia de Sistemas Ltda. Rio de Janeiro - RJ / Brasil

1 0

[Users] sl_send_reply with avp parameter
by Sigrid Thijs 10 May '07

10 May '07

Hi, I was wondering if it's possible to create an openser script to automatically send an error response when an INVITE request for a specific SIP URI is received. For example, an INVITE to "sip:e403@domain" would return a 403 response, and requests to "sip:eXYZ@domain.com" would return a XYZ response. I've tried using avp: # get username avp_printf ( "$avp(i:2)", "$oU" ); # get error code avp_subst ("$avp(i:2)", "/e*/\1/" ); # send reply sl_send_reply ( "$avp(i:2)", "Error" ); But apparently it's not allowed to pass an avp parameter to the sl_send_reply function. Is there another way to accomplish this? kind regards, Sigrid Thijs

2 2

[Serusers] Digest Authentication in Ottendorf
by KUMAR 10 May '07

10 May '07

Thank you Michal, I have migrated to SER-2.0 and have compiled with HONOR_MADDR, but having some problem making it work. I cannot get the digest authentication working. I am using the following code for REGISTER, if (method=="REGISTER") { if(!www_authorize("","credentials")){ www_challenge("","0"); break; }; }; For some reason, www_challenge is not recognised. And gives parse error. Can anyone please tell me how can I get the digest authentication working?? kumar On 5/7/07, Michal Matyska <michal(a)iptel.org> wrote: > Hi, > > commetns inline > > On Po, 2007-05-07 at 10:04 +0545, KUMAR wrote: > > Hi Michal, > > Sorry for my late response. Here are the packets while the transport > > UDP is used. > > regards > > kumar > > > > On 5/4/07, Michal Matyska <michal(a)iptel.org> wrote: > > > Hi Kumar, > > > > > > could you please attach also the capture (at the server is enough) of > > > the UDP both way messages? There call flow is quite strange and not > > > understandable.... (well not surprised that's M$ RTC :) > > > > > > Some comments for now: > > > 1) it seems the CLIENT IS BROKEN and does not understand lr=on flag for > > > loose routing - see how is the ACK generated !!! You can try change rr > > > module parameter to put just lr into the record-route (this might > > > explain your rr and rr_preset difference) > > Taking back, the client code is not broken, it is obsoleted only ;-) > It uses strict router procedures (RFC2543), to create the Route header > and Request-URI in subsequent within-dialog requests, but SER's > loose_route function call should be able to escape form that (and it is, > the rewritten request uri shows that). If you want to check, you should > see "after_strict" message in your debug output. > > > > 2) M$ is used to misuse maddr, there is compile time flag HONOR_MADDR > > > which you should have set to route such requests taking maddr into > > > account > > This seems to be the main issue, the client does not put maddr into the > UDP requests. Check the Route header from the client in the TCP and UDP > captures to see the difference. > > Recompile SER with the option HONOR_MADDR set and it should work. If > not, please provide SER output with debug=4 statement in you ser.cfg, > while running the TCP usecase. > > Michal > > > > > On Pá, 2007-05-04 at 09:52 +0545, KUMAR wrote: > > > > Here are the packet captures. I was not quite sure whether I have to > > > > include packet captures from server or the UAs so i've included all of > > > > them. ua_A is sending INVITE to ua_B. > > > > thank you > > > > > > > > regards > > > > kumar > > > > > > > > On 5/3/07, Klaus Darilion <klaus.mailinglists(a)pernau.at> wrote: > > > > > > > > > > > > > > > Jiri Kuthan wrote: > > > > > > actually I was suggesting SIP message dump (PCAP file) as opposed to log files. -jiri > > > > > > > > > > or ngrep dump: > > > > > > > > > > ngrep -t -W byline -d any port 5060 > > > > > > > > > > regards > > > > > klaus > > > > > > > > > > > > > > > > > > > > > > At 13:10 03/05/2007, KUMAR wrote: > > > > > >> Here is the message dump that i get when using TCP as transport. > > > > > >>> > > > > > >>> On 5/3/07, Jiri Kuthan <jiri(a)iptel.org> wrote: > > > > > >>>> I think you would have to send message dumps first so that [serusers] volunteers > > > > > >>>> have material for providing an answer. > > > > > >>>> > > > > > >>>> -jiri > > > > > >>>> > > > > > >>>> At 12:07 03/05/2007, KUMAR wrote: > > > > > >>>>> - > > > > > >>>>> Hi all, > > > > > >>>>> I already posted this message yet havent got any replies. I really > > > > > >>>>> need to find this out. > > > > > >>>>> Please anyone reply to this problem. > > > > > >>>>> > > > > > >>>>> I am using SER-0.9.6 and the problem i'm having is this. When using > > > > > >>>>> the following ser.cfg, it works well when the transport UDP is used. > > > > > >>>>> But when transport TCP is used, then it only results in one way IM, > > > > > >>>>> only from the UA from which the INVITE is being sent. Moreover, when > > > > > >>>>> record_route is used instead of record_route_preset, then even that > > > > > >>>>> one way IM doesn't work. But it works well with UDP. > > > > > >>>>> > > > > > >>>>> Can anyone please point me as to where the problem might be?? > > > > > >>>>> Thank you in advance. > > > > > >>>>> > > > > > >>>>> kumar > > > > > >>>>> > > > > > >>>>> > > > > > >>>>> Content-Type: application/octet-stream; name=ser.cfg > > > > > >>>>> X-Attachment-Id: f_f18sg2bl > > > > > >>>>> Content-Disposition: attachment; filename="ser.cfg" > > > > > >>>>> > > > > > >>>>> _______________________________________________ > > > > > >>>>> Serusers mailing list > > > > > >>>>> Serusers(a)lists.iptel.org > > > > > >>>>> http://lists.iptel.org/mailman/listinfo/serusers > > > > > >>>> > > > > > >>>> > > > > > >>>> -- > > > > > >>>> Jiri Kuthan http://iptel.org/~jiri/ > > > > > >>>> > > > > > >>>> > > > > > >> > > > > > >> Content-Type: audio/x-pn-realaudio-plugin; name="msg_dump.rar" > > > > > >> Content-Disposition: attachment; filename="msg_dump.rar" > > > > > >> X-Attachment-Id: f_f1949112 > > > > > >> > > > > > >> _______________________________________________ > > > > > >> 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 > > > > > > > >

2 1

[Users] OpenSER-1.2.0 and SEMS
by raviprakash sunkara 10 May '07

10 May '07

Hello Users Todays onwards I'm Started integrating SEMS. Previously I'm using the Asterisk + openSER1.2 .For a small change , I shifted into the SEMS for voicemail By this url ( http://openser.org/pipermail/users/2005-September/000844.html) i found t_write_reg that is not in OpenSER tm modules , I found t_write_fifo/unix How I can do the voicemail or announcement with OpenSER + SEMS Any one give me ant hints Regardinng this points -- Thanks &Regards Ravi Prakash Sunkara

3 3

[Users] Integration to openser with midia server
by Danilo 09 May '07

09 May '07

Hi all, I am using OpenSER 1.2.0. I am going to install media proxy, How does it work? How the Openser knows and obtains to monitor the flow and to set in motion the midia server? Danilo Inovax Engenharia de Sistemas Ltda. Rio de Janeiro - RJ / Brazil

2 1

[Users] TLS configuration in Fedora Core 5
by Han Gu 09 May '07

09 May '07

Hi all, I am trying to configure my openSER with TLS support and get the following error message: May 7 15:44:39 tiger openser: WARNING:init_tls: disabling compression due ZLIB problems May 7 15:44:39 tiger openser: ERRRO:init_tls: null openssl compression methods Anyone could give me a hint of what is going wrong with my settings? Thank you so much for the help. Regards. Han

3 3

[Users] Sending Voicemail to Asterisk
by Bill Neely 09 May '07

09 May '07

I am having a problem forwarding calls to asterisk for voice mail handling. What I am trying to do: When a call times out for no answer, change the address of the call by prepending 777 to the extension number. Then forward the call to asterisk, where the 777 prefix tells asterisk that it just sends this call to voicemail. Here is my openser.cfg file: ... modparam("tm", "fr_inv_timer", 20) ... route[1] { # # send it out now; use stateful forwarding as it works reliably # # even for UDP2TCP t_on_failure("2"); if (!t_relay()) { sl_reply_error(); }; exit; } failure_route[2] { if ( t_check_status("408")) if the call fails for timeout, send me a message and go to route(3) { xlog("L_ERR","rrreeeeeeeeeeeeeeeeeecalling froute2 <$rm><$ru>\n"); route(3); exit; } exit; } route[3] { rewrite the uri to prepend 777 if (subst_uri('/^sip:([0-9]+)@(.*)$/sip:777\1@\2/i')) { rewritehostport("66.xxx.20.50:5060"); the asterisk box append_branch(); xlog("L_ERR","3333333333333333333333333333333333333333after substr <$rm><$ru>\n"); if (!t_relay("udp:66.xxx.20.50:5060")) { sl_send_reply("500", "Relaying failed"); } } exit; } What happens is that the call gets forwarded to Asterisk, but asterisk does not seem to recognize that the call is destined for 777xxxxxxx, but instead tries to send it to xxxxxxx, which causes the call to be transferred back to openser. This loop goes around and around until I kill it. Here is portion of log file: 1(7856) rrreeeeeeeeeeeeeeeeeecalling froute2 <INVITE><sip:1020101@67.188.231.188:42749;rinstance=1e89b57f830c50f7> 1(7856) subst_run: running. r=0 1(7856) subst_run: matched (0, 59): [sip:1020101@67.188.231.188:42749;rinstance=1e89b57f830c50f7] 1(7856) textops: subst_uri_f: match - old uri= [sip:1020101@67.188.231.188:42749;rinstance=1e89b57f830c50f7], new uri= [sip:7771020101@67.188.231.188:42749;rinstance=1e89b57f830c50f7] 1(7856) ******* setting for branch 0 flags 0 1(7856) 3333333333333333333333333333333333333333after substr <INVITE><sip:7771020101@66.224.20.50:5060;rinstance=1e89b57f830c50f7> 1(7856) check_via_address(67.188.231.188, 67.188.231.188, 0) 1(7856) DBG:check_against_rule_list: using list dns 1(7856) DEBUG:tm:set_timer: relative timeout is 500000 1(7856) DEBUG: add_to_tail_of_timer[4]: 0x801f4b4b0 (49500000) 1(7856) DEBUG:tm:set_timer: relative timeout is 30 1(7856) DEBUG: add_to_tail_of_timer[0]: 0x801f4b4e0 (79) 1(7856) DEBUG:tm:relay_reply: branch=0, save=1, relay=-1 1(7856) DEBUG:tm:build_local: using FROM=<From: "Xantek"<sip:1020101@67.102.229.3>;tag=045b4172 >, TO=<To: "1020101"<sip:1020101@67.102.229.3> >, CSEQ_N=<CSeq: 1> 1(7856) DEBUG: cancel_branch: sending cancel... 1(7856) DEBUG:tm:set_timer: relative timeout is 500000 1(7856) DEBUG: add_to_tail_of_timer[4]: 0x801f4b3c0 (49500000) 1(7856) DEBUG:tm:set_timer: relative timeout is 30 1(7856) DEBUG: add_to_tail_of_timer[0]: 0x801f4b3f0 (79) 1(7856) DEBUG: final_response_handler : done 0(7855) udp_rcv_loop: probing packet received from 24.5.58.18 53652 0(7855) SIP Reply (status): 0(7855) version: <SIP/2.0> 0(7855) status: <407> 0(7855) reason: <Proxy Authentication Required> 0(7855) parse_headers: flags=2 0(7855) Found param type 232, <branch> = <z9hG4bK5988.480944e.1>; state=6 0(7855) Found param type 234, <received> = <67.102.229.3>; state=16 0(7855) end of header reached, state=5 0(7855) parse_headers: Via found, flags=2 0(7855) parse_headers: this is the first via 0(7855) After parse_msg... 0(7855) DEBUG:forward_reply: found module tm, passing reply to it 0(7855) DEBUG: t_check: start=0xffffffffffffffff 0(7855) parse_headers: flags=22 0(7855) Found param type 232, <branch> = <z9hG4bK-d87543-e004ef19dd229e0e-1--d87543->; state=6 0(7855) Found param type 235, <rport> = <42749>; state=16 0(7855) end of header reached, state=5 0(7855) parse_headers: Via found, flags=22 0(7855) parse_headers: this is the second via 0(7855) DEBUG: add_param: tag=as499a9c72 0(7855) DEBUG:parse_to:end of header reached, state=29 0(7855) DBUG:parse_to: display={"1020101"}, ruri={sip:1020101@67.102.229.3} 0(7855) DEBUG: get_hdr_field: <To> [52]; uri=[sip:1020101@67.102.229.3] 0(7855) DEBUG: to body ["1020101"<sip:1020101@67.102.229.3>] 0(7855) get_hdr_field: cseq <CSeq>: <1> <INVITE> 0(7855) parse_headers: flags=8 0(7855) DEBUG: t_reply_matching: hash 34965 label 239374468 branch 1 0(7855) DEBUG:tm:REF_UNSAFE: after is 1 0(7855) DEBUG: t_reply_matching: reply matched (T=0x801f4b0d8)! 0(7855) DEBUG: t_check: end=0x801f4b0d8 0(7855) DEBUG:tm:reply_received: org. status uas=180, uac[1]=0 local=0 is_invite=1) 0(7855) parse_headers: flags=8 0(7855) DEBUG:tm:build_local: using FROM=<From: "Xantek"<sip:1020101@67.102.229.3>;tag=045b4172 >, TO=<To: "1020101"<sip:1020101@67.102.229.3>;tag=as499a9c72 >, CSEQ_N=<CSeq: 1> 0(7855) DEBUG:tm:t_should_relay_response: T_code=180, new_code=407 0(7855) DEBUG:tm:t_pick_branch: picked branch 1, code 407 0(7855) DEBUG:tm:relay_reply: branch=1, save=0, relay=1 0(7855) old size: 631, new size: 549 0(7855) build_res_from_sip_res: copied size: orig:125, new: 43, rest: 506 msg= SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 67.188.231.188:42749;branch=z9hG4bK-d87543-e004ef19dd229e0e-1--d87543-;rport=42749 From: "Xantek"<sip:1020101@67.102.229.3>;tag=045b4172 To: "1020101"<sip:1020101@67.102.229.3>;tag=as499a9c72 Call-ID: YjgxM2RlZmM4MjRhMDYxN2YwYjI2MDdmZWZjMGY4NTY. CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:7771020101@66.224.20.50> Proxy-Authenticate: Digest realm="sip.xantek.cc", nonce="1aebadb3" Content-Length: 0 0(7855) DEBUG:tm:set_timer: relative timeout is 500000 0(7855) DEBUG: add_to_tail_of_timer[4]: 0x801f4b220 (49500000) 0(7855) DEBUG:tm:set_timer: relative timeout is 30 0(7855) DEBUG: add_to_tail_of_timer[0]: 0x801f4b250 (79) 0(7855) DEBUG:tm:relay_reply: sent buf=0x62deb0: SIP/2.0 4..., shmem=0x801f4aea0: SIP/2.0 4 0(7855) DEBUG: cleanup_uac_timers: RETR/FR timers reset 0(7855) DEBUG:tm:UNREF_UNSAFE: after is 0 0(7855) DEBUG:destroy_avp_list: destroying list 0x0 0(7855) receive_msg: cleaning up 0(7855) SIP Reply (status): 0(7855) version: <SIP/2.0> 0(7855) status: <200> 0(7855) reason: <OK> 0(7855) parse_headers: flags=2 0(7855) Found param type 232, <branch> = <z9hG4bK5988.480944e.0>; state=16 0(7855) end of header reached, state=5 0(7855) parse_headers: Via found, flags=2 0(7855) parse_headers: this is the first via 0(7855) After parse_msg... 0(7855) DEBUG:forward_reply: found module tm, passing reply to it 0(7855) DEBUG: t_check: start=0xffffffffffffffff 0(7855) parse_headers: flags=22 0(7855) DEBUG: add_param: tag=03422a48 0(7855) DEBUG:parse_to:end of header reached, state=29 0(7855) DBUG:parse_to: display={"1020101"}, ruri={sip:1020101@67.102.229.3} 0(7855) DEBUG: get_hdr_field: <To> [50]; uri=[sip:1020101@67.102.229.3] 0(7855) DEBUG: to body ["1020101"<sip:1020101@67.102.229.3>] 0(7855) get_hdr_field: cseq <CSeq>: <1> <CANCEL> 0(7855) DEBUG: t_reply_matching: hash 34965 label 239374468 branch 0 0(7855) DEBUG:tm:REF_UNSAFE: after is 1 0(7855) DEBUG: t_reply_matching: reply matched (T=0x801f4b0d8)! 0(7855) DEBUG: t_check: end=0x801f4b0d8 0(7855) DEBUG:tm:reply_received: org. status uas=407, uac[0]=408 local=0 is_invite=1) 0(7855) DEBUG:tm:reply_received: reply to local CANCEL processed 0(7855) DEBUG:tm:UNREF_UNSAFE: after is 0 0(7855) DEBUG:destroy_avp_list: destroying list 0x0 0(7855) receive_msg: cleaning up 0(7855) SIP Reply (status): 0(7855) version: <SIP/2.0> 0(7855) status: <487> 0(7855) reason: <Request Terminated> 0(7855) parse_headers: flags=2 0(7855) Found param type 232, <branch> = <z9hG4bK5988.480944e.0>; state=16 0(7855) end of header reached, state=5 0(7855) parse_headers: Via found, flags=2 0(7855) parse_headers: this is the first via 0(7855) After parse_msg... 0(7855) DEBUG:forward_reply: found module tm, passing reply to it 0(7855) DEBUG: t_check: start=0xffffffffffffffff 0(7855) parse_headers: flags=22 0(7855) Found param type 232, <branch> = <z9hG4bK-d87543-e004ef19dd229e0e-1--d87543->; state=6 0(7855) Found param type 235, <rport> = <42749>; state=16 0(7855) end of header reached, state=5 0(7855) parse_headers: Via found, flags=22 0(7855) parse_headers: this is the second via 0(7855) DEBUG: add_param: tag=03422a48 0(7855) DEBUG:parse_to:end of header reached, state=29 0(7855) DBUG:parse_to: display={"1020101"}, ruri={sip:1020101@67.102.229.3} 0(7855) DEBUG: get_hdr_field: <To> [50]; uri=[sip:1020101@67.102.229.3] 0(7855) DEBUG: to body ["1020101"<sip:1020101@67.102.229.3>] 0(7855) get_hdr_field: cseq <CSeq>: <1> <INVITE> 0(7855) parse_headers: flags=8 0(7855) DEBUG: t_reply_matching: hash 34965 label 239374468 branch 0 0(7855) DEBUG:tm:REF_UNSAFE: after is 1 0(7855) DEBUG: t_reply_matching: reply matched (T=0x801f4b0d8)! 0(7855) DEBUG: t_check: end=0x801f4b0d8 0(7855) DEBUG:tm:reply_received: org. status uas=407, uac[0]=408 local=0 is_invite=1) 0(7855) parse_headers: flags=8 0(7855) DEBUG:tm:build_local: using FROM=<From: "Xantek"<sip:1020101@67.102.229.3>;tag=045b4172 >, TO=<To: "1020101"<sip:1020101@67.102.229.3>;tag=03422a48 >, CSEQ_N=<CSeq: 1> 0(7855) DEBUG:tm:t_should_relay_response: T_code=407, new_code=487 0(7855) DEBUG:tm:relay_reply: branch=0, save=0, relay=-1 0(7855) DEBUG:tm:UNREF_UNSAFE: after is 0 0(7855) DEBUG:destroy_avp_list: destroying list 0x0 0(7855) receive_msg: cleaning up 0(7855) SIP Request: 0(7855) method: <ACK> 0(7855) uri: <sip:1020101@67.102.229.3> 0(7855) version: <SIP/2.0> 0(7855) parse_headers: flags=2 0(7855) Found param type 232, <branch> = <z9hG4bK-d87543-e004ef19dd229e0e-1--d87543->; state=6 0(7855) Found param type 235, <rport> = <n/a>; state=17 0(7855) end of header reached, state=5 0(7855) parse_headers: Via found, flags=2 0(7855) parse_headers: this is the first via 0(7855) After parse_msg... 0(7855) preparing to run routing scripts... 0(7855) DEBUG : sl_filter_ACK: to late to be a local ACK! 0(7855) parse_headers: flags=100 0(7855) DEBUG: add_param: tag=as499a9c72 0(7855) DEBUG:parse_to:end of header reached, state=29 0(7855) DBUG:parse_to: display={"1020101"}, ruri={sip:1020101@67.102.229.3} 0(7855) DEBUG: get_hdr_field: <To> [52]; uri=[sip:1020101@67.102.229.3] 0(7855) DEBUG: to body ["1020101"<sip:1020101@67.102.229.3>] 0(7855) get_hdr_field: cseq <CSeq>: <1> <ACK> 0(7855) DEBUG: get_hdr_body : content_length=0 0(7855) found end of header 0(7855) DEBUG: is_maxfwd_present: max_forwards header not found! 0(7855) DEBUG: add_param: tag=045b4172 0(7855) DEBUG:parse_to:end of header reached, state=29 0(7855) DBUG:parse_to: display={"Xantek"}, ruri={sip:1020101@67.102.229.3} 0(7855) parse_headers: flags=200 0(7855) find_first_route: No Route headers found 0(7855) loose_route: There is no Route HF 0(7855) grep_sock_info - checking if host==us: 12==12 && [67.102.229.3] == [67.102.229.3] 0(7855) grep_sock_info - checking if port 5060 matches port 5060 0(7855) grep_sock_info - checking if host==us: 12==12 && [67.102.229.3] == [67.102.229.3] 0(7855) grep_sock_info - checking if port 5060 matches port 5060 0(7855) lookup(): '1020101' Not found in usrloc 0(7855) grep_sock_info - checking if host==us: 12==12 && [67.102.229.3] == [67.102.229.3] 0(7855) grep_sock_info - checking if port 5060 matches port 5060 0(7855) rewrite_uri: Rewriting Request-URI with 'sip:1020101@67.188.231.188:42749;rinstance=1e89b57f830c50f7' 0(7855) parse_headers: flags=ffffffffffffffff 0(7855) DEBUG: t_newtran: T on entrance=0xffffffffffffffff 0(7855) parse_headers: flags=ffffffffffffffff 0(7855) parse_headers: flags=78 0(7855) t_lookup_request: start searching: hash=34965, isACK=1 0(7855) DEBUG: RFC3261 transaction matched, tid=-d87543-e004ef19dd229e0e-1--d87543- 0(7855) DEBUG:tm:REF_UNSAFE: after is 1 0(7855) DEBUG: t_lookup_request: transaction found (T=0x801f4b0d8) 0(7855) DEBUG: cleanup_uac_timers: RETR/FR timers reset 0(7855) DEBUG: add_to_tail_of_timer[2]: 0x801f4b158 (54) 0(7855) DEBUG:tm:UNREF_UNSAFE: after is 0 0(7855) DEBUG:destroy_avp_list: destroying list 0x0 0(7855) receive_msg: cleaning up 0(7855) SIP Request: 0(7855) method: <INVITE> 0(7855) uri: <sip:1020101@67.102.229.3> 0(7855) version: <SIP/2.0> 0(7855) parse_headers: flags=2 0(7855) Found param type 232, <branch> = <z9hG4bK-d87543-c535a22e8d568d14-1--d87543->; state=6 0(7855) Found param type 235, <rport> = <n/a>; state=17 0(7855) end of header reached, state=5 0(7855) parse_headers: Via found, flags=2 0(7855) parse_headers: this is the first via 0(7855) After parse_msg... 0(7855) preparing to run routing scripts... 0(7855) parse_headers: flags=100 0(7855) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(7855) parse_headers: flags=10 0(7855) DEBUG:parse_to:end of header reached, state=10 0(7855) DBUG:parse_to: display={"1020101"}, ruri={sip:1020101@67.102.229.3} 0(7855) DEBUG: get_hdr_field: <To> [37]; uri=[sip:1020101@67.102.229.3] 0(7855) DEBUG: to body ["1020101"<sip:1020101@67.102.229.3> ] 0(7855) DEBUG: add_param: tag=045b4172 0(7855) DEBUG:parse_to:end of header reached, state=29 0(7855) DBUG:parse_to: display={"Xantek"}, ruri={sip:1020101@67.102.229.3} 0(7855) parse_headers: flags=200 0(7855) get_hdr_field: cseq <CSeq>: <2> <INVITE> 0(7855) DEBUG: get_hdr_body : content_length=476 0(7855) found end of header 0(7855) find_first_route: No Route headers found 0(7855) loose_route: There is no Route HF 0(7855) grep_sock_info - checking if host==us: 12==12 && [67.102.229.3] == [67.102.229.3] 0(7855) grep_sock_info - checking if port 5060 matches port 5060 0(7855) grep_sock_info - checking if host==us: 12==12 && [67.102.229.3] == [67.102.229.3] 0(7855) grep_sock_info - checking if port 5060 matches port 5060 0(7855) lookup(): '1020101' Not found in usrloc 0(7855) grep_sock_info - checking if host==us: 12==12 && [67.102.229.3] == [67.102.229.3] 0(7855) grep_sock_info - checking if port 5060 matches port 5060 0(7855) rewrite_uri: Rewriting Request-URI with 'sip:1020101@67.188.231.188:42749;rinstance=1e89b57f830c50f7' 0(7855) parse_headers: flags=ffffffffffffffff 0(7855) DEBUG: t_newtran: T on entrance=0xffffffffffffffff 0(7855) parse_headers: flags=ffffffffffffffff 0(7855) parse_headers: flags=78 0(7855) t_lookup_request: start searching: hash=34962, isACK=0 0(7855) DEBUG: RFC3261 transaction matching failed 0(7855) DEBUG: t_lookup_request: no transaction found 0(7855) parse_headers: flags=ffffffffffffffff 0(7855) check_via_address(67.188.231.188, 67.188.231.188, 0) 0(7855) WARNING:vqm_resize: resize(0) called 0(7855) DEBUG:tm:_reply_light: reply sent out. buf=0x62dcf8: SIP/2.0 1..., shmem=0x801f4e420: SIP/2.0 1 0(7855) DEBUG:tm:_reply_light: finished 0(7855) DEBUG: mk_proxy: doing DNS lookup... 0(7855) check_via_address(67.188.231.188, 67.188.231.188, 0) 0(7855) DBG:check_against_rule_list: using list dns 0(7855) DEBUG:tm:set_timer: relative timeout is 500000 0(7855) DEBUG: add_to_tail_of_timer[4]: 0x801f550c8 (49600000) 0(7855) DEBUG:tm:set_timer: relative timeout is 30 0(7855) DEBUG: add_to_tail_of_timer[0]: 0x801f550f8 (79) 0(7855) DEBUG:tm:t_relay_to: new transaction fwd'ed 0(7855) DEBUG:tm:UNREF_UNSAFE: after is 0 0(7855) DEBUG:destroy_avp_list: destroying list 0x0 0(7855) receive_msg: cleaning up 1(7856) DEBUG: timer routine:4,tl=0x801f4b4b0 next=0x801f4b3c0, timeout=49500000 1(7856) DEBUG: timer routine:4,tl=0x801f4b3c0 next=0x801f4b220, timeout=49500000 1(7856) DEBUG: timer routine:4,tl=0x801f4b220 next=0x0, timeout=49500000 0(7855) SIP Reply (status): 0(7855) version: <SIP/2.0> 0(7855) status: <180> 0(7855) reason: <Ringing> 0(7855) parse_headers: flags=2 0(7855) Found param type 232, <branch> = <z9hG4bK2988.3de53f62.0>; state=16 0(7855) end of header reached, state=5 0(7855) parse_headers: Via found, flags=2 0(7855) parse_headers: this is the first via 0(7855) After parse_msg... 0(7855) DEBUG:forward_reply: found module tm, passing reply to it 0(7855) DEBUG: t_check: start=0xffffffffffffffff 0(7855) parse_headers: flags=22 0(7855) Found param type 232, <branch> = <z9hG4bK-d87543-c535a22e8d568d14-1--d87543->; state=6 0(7855) Found param type 235, <rport> = <42749>; state=16 0(7855) end of header reached, state=5 0(7855) parse_headers: Via found, flags=22 0(7855) parse_headers: this is the second via 0(7855) DEBUG: add_param: tag=2a63bc63 0(7855) DEBUG:parse_to:end of header reached, state=29 0(7855) DBUG:parse_to: display={"1020101"}, ruri={sip:1020101@67.102.229.3} 0(7855) DEBUG: get_hdr_field: <To> [50]; uri=[sip:1020101@67.102.229.3] 0(7855) DEBUG: to body ["1020101"<sip:1020101@67.102.229.3>] 0(7855) get_hdr_field: cseq <CSeq>: <2> <INVITE> 0(7855) parse_headers: flags=8 0(7855) DEBUG: t_reply_matching: hash 34962 label 653483731 branch 0 0(7855) DEBUG:tm:REF_UNSAFE: after is 1 0(7855) DEBUG: t_reply_matching: reply matched (T=0x801f54ea8)! 0(7855) DEBUG: t_check: end=0x801f54ea8 0(7855) DEBUG:tm:reply_received: org. status uas=100, uac[0]=0 local=0 is_invite=1) 0(7855) DEBUG:tm:t_should_relay_response: T_code=100, new_code=180 0(7855) DEBUG:tm:relay_reply: branch=0, save=0, relay=0 0(7855) old size: 554, new size: 493 0(7855) build_res_from_sip_res: copied size: orig:82, new: 21, rest: 472 msg= SIP/2.0 180 Ringing Via: SIP/2.0/UDP 67.188.231.188:42749;branch=z9hG4bK-d87543-c535a22e8d568d14-1--d87543-;rport=42749 Record-Route: <sip:67.102.229.3;lr;ftag=045b4172> Contact: <sip:1020101@67.188.231.188:42749;rinstance=1e89b57f830c50f7> To: "1020101"<sip:1020101@67.102.229.3>;tag=2a63bc63 From: "Xantek"<sip:1020101@67.102.229.3>;tag=045b4172 Call-ID: YjgxM2RlZmM4MjRhMDYxN2YwYjI2MDdmZWZjMGY4NTY. CSeq: 2 INVITE User-Agent: eyeBeam release 1009l stamp 37966 Content-Length: 0 0(7855) DEBUG:tm:relay_reply: sent buf=0x62d678: SIP/2.0 1..., shmem=0x801f4e420: SIP/2.0 1 0(7855) DEBUG:tm:set_timer: relative timeout is 20 0(7855) DEBUG: add_to_tail_of_timer[1]: 0x801f550f8 (69) 0(7855) DEBUG:tm:UNREF_UNSAFE: after is 0 0(7855) DEBUG:destroy_avp_list: destroying list 0x0 0(7855) receive_msg: cleaning up 1(7856) DEBUG: timer routine:4,tl=0x801f550c8 next=0x0, timeout=49600000 0(7855) SIP Request: 0(7855) method: <CANCEL> 0(7855) uri: <sip:1020101@67.102.229.3> 0(7855) version: <SIP/2.0> 0(7855) parse_headers: flags=2 0(7855) Found param type 232, <branch> = <z9hG4bK-d87543-c535a22e8d568d14-1--d87543->; state=6 0(7855) Found param type 235, <rport> = <n/a>; state=17 0(7855) end of header reached, state=5 0(7855) parse_headers: Via found, flags=2 0(7855) parse_headers: this is the first via 0(7855) After parse_msg... 0(7855) preparing to run routing scripts... 0(7855) parse_headers: flags=100 0(7855) DEBUG:parse_to:end of header reached, state=10 0(7855) DBUG:parse_to: display={"1020101"}, ruri={sip:1020101@67.102.229.3} 0(7855) DEBUG: get_hdr_field: <To> [37]; uri=[sip:1020101@67.102.229.3] 0(7855) DEBUG: to body ["1020101"<sip:1020101@67.102.229.3> ] 0(7855) get_hdr_field: cseq <CSeq>: <2> <CANCEL> 0(7855) DEBUG: get_hdr_body : content_length=0 0(7855) found end of header 0(7855) DEBUG: is_maxfwd_present: max_forwards header not found! 0(7855) DEBUG: add_param: tag=045b4172 0(7855) DEBUG:parse_to:end of header reached, state=29 0(7855) DBUG:parse_to: display={"Xantek"}, ruri={sip:1020101@67.102.229.3} 0(7855) parse_headers: flags=200 0(7855) find_first_route: No Route headers found 0(7855) loose_route: There is no Route HF 0(7855) grep_sock_info - checking if host==us: 12==12 && [67.102.229.3] == [67.102.229.3] 0(7855) grep_sock_info - checking if port 5060 matches port 5060 0(7855) grep_sock_info - checking if host==us: 12==12 && [67.102.229.3] == [67.102.229.3] 0(7855) grep_sock_info - checking if port 5060 matches port 5060 0(7855) lookup(): '1020101' Not found in usrloc 0(7855) grep_sock_info - checking if host==us: 12==12 && [67.102.229.3] == [67.102.229.3] 0(7855) grep_sock_info - checking if port 5060 matches port 5060 0(7855) rewrite_uri: Rewriting Request-URI with 'sip:1020101@67.188.231.188:42749;rinstance=1e89b57f830c50f7' 0(7855) parse_headers: flags=ffffffffffffffff 0(7855) DEBUG: t_newtran: T on entrance=0xffffffffffffffff 0(7855) parse_headers: flags=ffffffffffffffff 0(7855) parse_headers: flags=78 0(7855) t_lookup_request: start searching: hash=34962, isACK=0 0(7855) DEBUG: RFC3261 transaction matching failed 0(7855) DEBUG: t_lookup_request: no transaction found 0(7855) DEBUG: t_lookupOriginalT: searching on hash entry 34962 0(7855) DEBUG: RFC3261 transaction matched, tid=-d87543-c535a22e8d568d14-1--d87543- 0(7855) DEBUG: t_lookupOriginalT: canceled transaction found (0x801f54ea8)! 0(7855) DEBUG:tm:REF_UNSAFE: after is 1 0(7855) DEBUG: t_lookupOriginalT completed 0(7855) check_via_address(67.188.231.188, 67.188.231.188, 0) 0(7855) DEBUG:tm:set_timer: relative timeout is 500000 0(7855) DEBUG: add_to_tail_of_timer[4]: 0x801f583f8 (50600000) 0(7855) DEBUG:tm:set_timer: relative timeout is 30 0(7855) DEBUG: add_to_tail_of_timer[0]: 0x801f58428 (80) 0(7855) DEBUG: e2e_cancel: e2e cancel proceeding 0(7855) parse_headers: flags=ffffffffffffffff 0(7855) check_via_address(67.188.231.188, 67.188.231.188, 0) 0(7855) WARNING:vqm_resize: resize(0) called 0(7855) DEBUG:tm:_reply_light: reply sent out. buf=0x62dcf8: SIP/2.0 2..., shmem=0x801f57da0: SIP/2.0 2 0(7855) DEBUG:tm:_reply_light: finished 0(7855) DEBUG:tm:t_relay_to: new transaction fwd'ed 0(7855) DEBUG:tm:UNREF_UNSAFE: after is 0 0(7855) DEBUG:tm:UNREF_UNSAFE: after is 0 0(7855) DEBUG:destroy_avp_list: destroying list 0x0 0(7855) receive_msg: cleaning up 0(7855) SIP Reply (status): 0(7855) version: <SIP/2.0> 0(7855) status: <200> 0(7855) reason: <OK> 0(7855) parse_headers: flags=2 0(7855) Found param type 232, <branch> = <z9hG4bK2988.3de53f62.0>; state=16 0(7855) end of header reached, state=5 0(7855) parse_headers: Via found, flags=2 0(7855) parse_headers: this is the first via 0(7855) After parse_msg... 0(7855) DEBUG:forward_reply: found module tm, passing reply to it 0(7855) DEBUG: t_check: start=0xffffffffffffffff 0(7855) parse_headers: flags=22 0(7855) Found param type 232, <branch> = <z9hG4bK-d87543-c535a22e8d568d14-1--d87543->; state=6 0(7855) Found param type 235, <rport> = <42749>; state=16 0(7855) end of header reached, state=5 0(7855) parse_headers: Via found, flags=22 0(7855) parse_headers: this is the second via 0(7855) DEBUG: add_param: tag=2a63bc63 0(7855) DEBUG:parse_to:end of header reached, state=29 0(7855) DBUG:parse_to: display={"1020101"}, ruri={sip:1020101@67.102.229.3} 0(7855) DEBUG: get_hdr_field: <To> [50]; uri=[sip:1020101@67.102.229.3] 0(7855) DEBUG: to body ["1020101"<sip:1020101@67.102.229.3>] 0(7855) get_hdr_field: cseq <CSeq>: <2> <CANCEL> 0(7855) DEBUG: t_reply_matching: hash 34962 label 653483731 branch 0 0(7855) DEBUG:tm:REF_UNSAFE: after is 1 0(7855) DEBUG: t_reply_matching: reply matched (T=0x801f581d8)! 0(7855) DEBUG: t_check: end=0x801f581d8 0(7855) DEBUG:tm:reply_received: org. status uas=200, uac[0]=0 local=0 is_invite=0) 0(7855) DEBUG:tm:t_should_relay_response: T_code=200, new_code=200 0(7855) DEBUG:tm:relay_reply: branch=0, save=0, relay=-1 0(7855) DEBUG: add_to_tail_of_timer[2]: 0x801f58258 (55) 0(7855) DEBUG: cleanup_uac_timers: RETR/FR timers reset 0(7855) DEBUG:tm:UNREF_UNSAFE: after is 0 0(7855) DEBUG:destroy_avp_list: destroying list 0x0 0(7855) receive_msg: cleaning up 0(7855) SIP Reply (status): 0(7855) version: <SIP/2.0> 0(7855) status: <487> 0(7855) reason: <Request Terminated> 0(7855) parse_headers: flags=2 0(7855) Found param type 232, <branch> = <z9hG4bK2988.3de53f62.0>; state=16 0(7855) end of header reached, state=5 0(7855) parse_headers: Via found, flags=2 0(7855) parse_headers: this is the first via 0(7855) After parse_msg... 0(7855) DEBUG:forward_reply: found module tm, passing reply to it 0(7855) DEBUG: t_check: start=0xffffffffffffffff 0(7855) parse_headers: flags=22 0(7855) Found param type 232, <branch> = <z9hG4bK-d87543-c535a22e8d568d14-1--d87543->; state=6 0(7855) Found param type 235, <rport> = <42749>; state=16 0(7855) end of header reached, state=5 0(7855) parse_headers: Via found, flags=22 0(7855) parse_headers: this is the second via 0(7855) DEBUG: add_param: tag=2a63bc63 0(7855) DEBUG:parse_to:end of header reached, state=29 0(7855) DBUG:parse_to: display={"1020101"}, ruri={sip:1020101@67.102.229.3} 0(7855) DEBUG: get_hdr_field: <To> [50]; uri=[sip:1020101@67.102.229.3] 0(7855) DEBUG: to body ["1020101"<sip:1020101@67.102.229.3>] 0(7855) get_hdr_field: cseq <CSeq>: <2> <INVITE> 0(7855) parse_headers: flags=8 0(7855) DEBUG: t_reply_matching: hash 34962 label 653483731 branch 0 0(7855) DEBUG:tm:REF_UNSAFE: after is 1 0(7855) DEBUG: t_reply_matching: reply matched (T=0x801f54ea8)! 0(7855) DEBUG: t_check: end=0x801f54ea8 0(7855) DEBUG:tm:reply_received: org. status uas=180, uac[0]=180 local=0 is_invite=1) 0(7855) parse_headers: flags=8 0(7855) DEBUG:tm:build_local: using FROM=<From: "Xantek"<sip:1020101@67.102.229.3>;tag=045b4172 >, TO=<To: "1020101"<sip:1020101@67.102.229.3>;tag=2a63bc63 >, CSEQ_N=<CSeq: 2> 0(7855) DEBUG:tm:t_should_relay_response: T_code=180, new_code=487 0(7855) DEBUG:tm:t_pick_branch: picked branch 0, code 487 0(7855) DEBUG:t_check_status: checked status is <487> 0(7855) DEBUG:tm:relay_reply: branch=0, save=0, relay=0 0(7855) old size: 442, new size: 381 0(7855) build_res_from_sip_res: copied size: orig:93, new: 32, rest: 349 msg= SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 67.188.231.188:42749;branch=z9hG4bK-d87543-c535a22e8d568d14-1--d87543-;rport=42749 To: "1020101"<sip:1020101@67.102.229.3>;tag=2a63bc63 From: "Xantek"<sip:1020101@67.102.229.3>;tag=045b4172 Call-ID: YjgxM2RlZmM4MjRhMDYxN2YwYjI2MDdmZWZjMGY4NTY. CSeq: 2 INVITE User-Agent: eyeBeam release 1009l stamp 37966 Content-Length: 0 0(7855) DEBUG:tm:set_timer: relative timeout is 500000 0(7855) DEBUG: add_to_tail_of_timer[4]: 0x801f54ff0 (50700000) 0(7855) DEBUG:tm:set_timer: relative timeout is 30 0(7855) DEBUG: add_to_tail_of_timer[0]: 0x801f55020 (80) 0(7855) DEBUG:tm:relay_reply: sent buf=0x62dcf8: SIP/2.0 4..., shmem=0x801f546f8: SIP/2.0 4 0(7855) DEBUG: cleanup_uac_timers: RETR/FR timers reset 0(7855) DEBUG:tm:UNREF_UNSAFE: after is 0 0(7855) DEBUG:destroy_avp_list: destroying list 0x0 0(7855) receive_msg: cleaning up 0(7855) SIP Request: 0(7855) method: <ACK> 0(7855) uri: <sip:1020101@67.102.229.3> 0(7855) version: <SIP/2.0> 0(7855) parse_headers: flags=2 0(7855) Found param type 232, <branch> = <z9hG4bK-d87543-c535a22e8d568d14-1--d87543->; state=6 0(7855) Found param type 235, <rport> = <n/a>; state=17 0(7855) end of header reached, state=5 0(7855) parse_headers: Via found, flags=2 0(7855) parse_headers: this is the first via 0(7855) After parse_msg... 0(7855) preparing to run routing scripts... 0(7855) DEBUG : sl_filter_ACK: to late to be a local ACK! 0(7855) parse_headers: flags=100 0(7855) DEBUG: add_param: tag=2a63bc63 0(7855) DEBUG:parse_to:end of header reached, state=29 0(7855) DBUG:parse_to: display={"1020101"}, ruri={sip:1020101@67.102.229.3} 0(7855) DEBUG: get_hdr_field: <To> [50]; uri=[sip:1020101@67.102.229.3] 0(7855) DEBUG: to body ["1020101"<sip:1020101@67.102.229.3>] 0(7855) get_hdr_field: cseq <CSeq>: <2> <ACK> 0(7855) DEBUG: get_hdr_body : content_length=0 0(7855) found end of header 0(7855) DEBUG: is_maxfwd_present: max_forwards header not found! 0(7855) DEBUG: add_param: tag=045b4172 0(7855) DEBUG:parse_to:end of header reached, state=29 0(7855) DBUG:parse_to: display={"Xantek"}, ruri={sip:1020101@67.102.229.3} 0(7855) parse_headers: flags=200 0(7855) find_first_route: No Route headers found 0(7855) loose_route: There is no Route HF 0(7855) grep_sock_info - checking if host==us: 12==12 && [67.102.229.3] == [67.102.229.3] 0(7855) grep_sock_info - checking if port 5060 matches port 5060 0(7855) grep_sock_info - checking if host==us: 12==12 && [67.102.229.3] == [67.102.229.3] 0(7855) grep_sock_info - checking if port 5060 matches port 5060 0(7855) lookup(): '1020101' Not found in usrloc 0(7855) grep_sock_info - checking if host==us: 12==12 && [67.102.229.3] == [67.102.229.3] 0(7855) grep_sock_info - checking if port 5060 matches port 5060 0(7855) rewrite_uri: Rewriting Request-URI with 'sip:1020101@67.188.231.188:42749;rinstance=1e89b57f830c50f7' 0(7855) parse_headers: flags=ffffffffffffffff 0(7855) DEBUG: t_newtran: T on entrance=0xffffffffffffffff 0(7855) parse_headers: flags=ffffffffffffffff 0(7855) parse_headers: flags=78 0(7855) t_lookup_request: start searching: hash=34962, isACK=1 0(7855) DEBUG: RFC3261 transaction matched, tid=-d87543-c535a22e8d568d14-1--d87543- 0(7855) DEBUG:tm:REF_UNSAFE: after is 1 0(7855) DEBUG: t_lookup_request: transaction found (T=0x801f54ea8) 0(7855) DEBUG: cleanup_uac_timers: RETR/FR timers reset 0(7855) DEBUG: add_to_tail_of_timer[2]: 0x801f54f28 (55) 0(7855) DEBUG:tm:UNREF_UNSAFE: after is 0 0(7855) DEBUG:destroy_avp_list: destroying list 0x0 0(7855) receive_msg: cleaning up 1(7856) DEBUG: timer routine:4,tl=0x801f583f8 next=0x0, timeout=50600000 1(7856) DEBUG: timer routine:4,tl=0x801f54ff0 next=0x0, timeout=50700000 ^C 0(7855) INT received, program terminates 0(7855) INFO: signal 2 received 0(7855) Memory status (pkg): 0(7855) fm_status (0x615be0): 0(7855) heap size= 1048576 0(7855) used= 56472, used+overhead=95544, free=953032 0(7855) max used (+overhead)= 95544 Anybody see what I an doing wrong? -- Bill Neely Xantek, Inc. 1-866-553-3833 1-702-874-3833

1 0

[Users] How to get the most stable 1.2.x release sources?
by Anatoly Pidruchny 09 May '07

09 May '07

Hi, all, I could not find a good explanation of the releasing policy for OpenSER. I mean, 1.2.0 was released on March 12, 2007. I know that the future release is 1.3.0. But is there going to be 1.2.1 ever released? There were many bugs fixed in 1.2.x since 1.2.0 was released. I think I read a developer's reply that the best and most stable OpenSER 1.2.x release sources can only be downloaded from the SVN 1.2 branch. Is this true? And what is the trunk version in SVN? Is it only for the next release (1.3.0) and changes to the trunk do not go to 1.2.x branch automatically? The download page (http://www.openser.org/mos/view/Download/) unfortunately does not tell that to get the most stable 1.2.x release one has to use SVN Download. I think most of the OpenSER users just download the released 1.2.0 sources. By the way, looks like the so called "daily" SVN snapshots have not been posted since April 23. Is this a known problem? Thanks, Anatoly.

4 5

[Serusers] ser_ctl
by Tuan Nguyen 09 May '07

09 May '07

Hi, Can someone tell me what does is mean this error: tuandat@ubuntu:~/serctl$ ./ser_ctl domain add iptel.org ser_ctl: No module named MySQLdb: ImportError I have already create mysql database ser with ser_mysql.sh script. I can show tables of ser database but I can't add domain. thanks for response Tuan.

3 2

Re: [Serusers] Which version of SER to use?
by Greger V. Teigre 09 May '07

09 May '07

:-) Spend 30 minutes on SER 2.0; it's easier than you think. etc/ser.cfg is a decent config. g-) Eneref wrote: > If I actually knew anything about how all the new stuff worked, I'd be > happy to help. ;) Sort of a chicken and egg problem. > > Unfortunately, as late, in my copious free time (*cough*), I haven't > had the time to delve into the SER 2.0 stuff and actually figure out > for myself how it all meshes together. > > N. > > > Greger V. Teigre wrote: >> heh heh, as iptel.org is an open-source project, we need a sponsor to >> pay for a technical writer if we want one. Iptego did that for SEMS >> documentation, but we don't have a documentation sponsor for SER. I'm >> afraid paid beer does not bring us far... >> >> So, we will have to cope with the volunteers we have. An advanced >> user, like you, is as good a writer as anyone, so if you want to put >> your keyboard fingers where your mouth is ;-), you are very welcome >> to sign up as a volunteer on either of our three current >> documentation efforts: >> 1. Migration guide (editor: Simon Miles) >> 2. Reference guide (editor: yours truly) >> 3. SER - Getting Started (editor: ?) >> >> I'm in the process of coordinating the table of contents with Simon, >> but here's an attempted outline: >> http://www.iptel.org/suggestion_for_outline_of_new_ser_reference_manual >> >> Feel free to sign up for any chapter or maybe editor of the Getting >> Started guide?! >> g-) >> >> SIP wrote: >>> Oh don't worry. I won't take it the wrong way. I'll be the first to >>> tell you that the documentation for SER is pretty close to >>> unusable. But unfortunately, that's the way most open source >>> projects are. You have all these coding geniuses building them, and >>> all these systems geniuses testing them... and not a single one of >>> them remembers how to convey information to the rest of us. ;) >>> >>> What the project lacks is a full-time technical writer -- someone >>> whose job it is is to corner the developers and ask them what all of >>> this stuff means (in intricate detail) and not let them go until >>> they've fully explained all of it. Then he/she can take that >>> information and translate it into something human beings can >>> understand. >>> >>> This is one reason why good technical writers are expensive, and the >>> primary reason that most companies and projects 'make do' with >>> whatever they can find. >>> >>> If you know any good technical writers who might be willing to fly >>> off to Europe and ply some of the SER team with beer enough to get >>> them talking, we could most certainly use it. I'll pay for the beer. ;) >>> >>> N. >>> >>> >>> lists(a)infoway.net wrote: >>>> Excellent news. Thanks for the feedback. Overall, and please don't >>>> take this negatively, I think documentation for SER is difficult >>>> for beginners, so if we have to go through the learning curve >>>> anyway, we might as well do it in the newer version. >>>> >>>> Thanks again >>>> >>>> On Tue, May 8, 2007 5:04 pm, SIP <sip(a)arcdiv.com> said: >>>> >>>> >>>>> For all intents and purposes, SER 2.0 (the new one) is ready for use. >>>>> The only thing it's really waiting on to be 'stable' is some better >>>>> documentation. However, it's best, if you're just getting >>>>> started, to >>>>> deploy with the new version and maybe muddle your way through a bit >>>>> until documentation is complete. >>>>> >>>>> N. >>>>> >>>>> >>>>> lists(a)infoway.net wrote: >>>>> >>>>>> We are planning on deploying a couple of SER servers and >>>>>> integrate them with >>>>>> either SEMS or Asterisk for voicemail. However, we are a bit >>>>>> confused as to which >>>>>> version of SER to go with? From what we can tell, the latest >>>>>> stable version is >>>>>> 0.9.6. However, we know of the new 0.10.X version. Reading up on >>>>>> the site, there >>>>>> is some documentation that explains a little bit of the >>>>>> differences and/or >>>>>> migration recommendations/steps. >>>>>> >>>>>> So, in an effort to keep life as simple as possible, which >>>>>> version should we work >>>>>> with? We are afraid that if we go through a learning cycle with >>>>>> 0.9.6 we're going >>>>>> to "suffer" when migrating to the new version, either because of >>>>>> the learning >>>>>> curve or because of a potential painful migration process. >>>>>> >>>>>> Any recommendations? >>>>>> >>>>>> Thanks >>>>>> >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> 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] authentication
by Stefan Ljung (TN/EAB) 09 May '07

09 May '07

Im on ser 0.9.6. Om looking for a simple way to configure my SER to check that all INVITEs coming in on a specific IP interface are coming from clients that have a valid registration. I know about the ip.src check but how to do lookups with From: URIs in the registrar is beyond me. /Stefan

2 1

[Serusers] registrar module in ser rel 2_0_0 - errors in setting parameters
by tzieleniewski 09 May '07

09 May '07

Hi, I am configuring registrar module and I get the following errors: 0(29188) set_mod_param_regex: parameter <desc_time_order> not found in module <registrar> 0(29188) parse error (88,19-20): Can't set module parameter when I try to set such module parameters: modparam("registrar", "desc_time_order", 1) modparam("registrar", "retry_after", 30) Are they still supported? Maybe under different name? Cheers Tomasz

2 1

[Serusers] tls module documentation not available
by tzieleniewski 09 May '07

09 May '07

Hi, The tls module documentation is unavailable page - http://www.iptel.org/ser/doc/modules/tls. is it temporary?? Cheers Tomasz

2 1

[Users] How to use forward command with script variables?
by Noel R. Morais 09 May '07

09 May '07

Hi All, I am trying to do something like this: route[1] { $var(forwardTo) = "192.168.192.100"; forward($var(forwardTo)); exit; } But it do not work. you know a way to do it? Thanks Noel

3 2

[Serusers] isup-oli
by Jai Rangi 09 May '07

09 May '07

Hello, Can some please tell me how can I capture isup-oli field from a incoming call. We need to capture that to recognize the incoming call from pay phone. Thank you, -Jai

3 3

[Users] XCAP-Lite problems
by Schumann Sebastian 09 May '07

09 May '07

Dear all, starting to build up OpenSER as presence server, I had no mentionable problems. Now, I tried to implement an Apache2 based XCAP server to store presence info, buddy lists etc. there. I have a problem setting up the XCAP-Lite server v0.9. Searching the web and the mailing list did not lead to a successful answer why the following happens: After setting up Apache2 with all required modules, virtual host etc. I try to use Eyebeam 1.5 to write the presrules and resource-list.xml to the database. This failes. I have successfully installed Apache2 with pear (+ extensions), php, mysql support. I created a virtual host with the following settings: NameVirtualHost * <VirtualHost *> <Location "/"> Dav On #necessary for PUT method </Location> ServerName 172.16.167.128 DocumentRoot /var/www/ #/var/www/xcap-root didn't work for GET as well, changed RewriteEngine On RewriteLog /var/log/apache2/rewrite.log RewriteLogLevel 2 RewriteRule /xcap-root/(.*) /xcap-root/index.php ProxyVia On </VirtualHost> I putted all files from XCAP-Lite archive in /var/www/xcap-root. After using PUT method through Eyebeam (change in contact list) I receive the 204 No content message. BUT: Not the database content is changed, the putted XML file overwrites index.php instead, which means, further requests fail. Does anybody know why the access to database does not work? Password etc. in config.php is ok. Thanks in advance. Best regards Sebastian PS: As I didn't find a mail address of the author of the program, I hope he is also subscribed to that list. There is probably an error in xcap-lib.php on line 766. I think there is a space between function and XCAPclient necessary.

3 5

Re: [Serusers] Which version of SER to use?
by lists＠infoway.net 09 May '07

09 May '07

Excellent news. Thanks for the feedback. Overall, and please don't take this negatively, I think documentation for SER is difficult for beginners, so if we have to go through the learning curve anyway, we might as well do it in the newer version. Thanks again On Tue, May 8, 2007 5:04 pm, SIP <sip(a)arcdiv.com> said: > For all intents and purposes, SER 2.0 (the new one) is ready for use. > The only thing it's really waiting on to be 'stable' is some better > documentation. However, it's best, if you're just getting started, to > deploy with the new version and maybe muddle your way through a bit > until documentation is complete. > > N. > > > lists(a)infoway.net wrote: >> We are planning on deploying a couple of SER servers and integrate them with >> either SEMS or Asterisk for voicemail. However, we are a bit confused as to which >> version of SER to go with? From what we can tell, the latest stable version is >> 0.9.6. However, we know of the new 0.10.X version. Reading up on the site, there >> is some documentation that explains a little bit of the differences and/or >> migration recommendations/steps. >> >> So, in an effort to keep life as simple as possible, which version should we work >> with? We are afraid that if we go through a learning cycle with 0.9.6 we're going >> to "suffer" when migrating to the new version, either because of the learning >> curve or because of a potential painful migration process. >> >> Any recommendations? >> >> Thanks >> >> >> >> _______________________________________________ >> Serusers mailing list >> Serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >> > >

3 3

Re: [Users] Proxy/Registrar and NAT
by Bernie Hoeneisen 09 May '07

09 May '07

Hi Klaus, Alex et al. I was thinking about the same problem. My setup has two redundant SIP Proxy/Registrar (OpenSER 1.1) Servers (P1 and P2), no additonal edge proxies. It uses either MySQL cluster or Master-Master Replication to exchange the state of the location table. For NAT Processing of REGISTER I use the Nathelper module, which writes the original (inside NAT) address to the 'contact' column and the information, where the REGISTER request was received from (outside NAT socket) goes to 'received' column of the location table. e.g.: contact column: sip:behoe@192.168.1.56:2051 received column: sip:130.56.88.20:5432 When an INVITE arrives at P1 (or P2), the 'contact' column will go to the R-URI and the 'received' value will be used as the next target of the INVITE request. INVITE requests that do not arrive at the proxy, where the UA has registered, have (depending on the NAT type) some likelihood to fail. Appart from that and as long as there is no path value saved in the location table, this work fine. If there is a path value, its content will go to the Route header field and the the request is sent to the topmost Route entry. Therefore it looses the information that was stored in the 'received' column....:-( The preserve this information and to make sure that LL requests are routed via the proxy, the UA has registered to, I have some idea, that uses path. (Unfortunately not all of my UAs support 'outbound'...:-( ): When the REGISTER arrives at the Proxy P1, I would insert a Path header field with the address of P1 (i.e. the IP address and port of the proxy processing the request) Furthermore I would add a new header field parameter to the contact header field (let's call it "extsock" for the time being) containing the external NAT socket (i.e. the same information that also goes to 'received' column) e.g. Path: <sip:136.59.10.85:5060;lr>, Contact: <sip:behoe@192.168.1.56:2051;extsock=sip:130.56.88.20:5432;lr> This will then be stored in the 'path' / 'contact' columns of the location table at save(). When the INVITE arrives e.g. at P2, it does the location lookup: The Route header field will be populated with the content of the 'path' column and the R-URI will be rewritten with the 'contact' column. This means the request will be normally forwarded to P1 (topmost Route). The SIP INVITE looks e.g. as follows: INVITE sip:behoe@192.168.1.56:2051;extsock=sip:130.56.88.20:5432 SIP/2.0 Route:<sip:136.59.10.85:5060;lr> [...] After the request arrives at P1, it checks, whether there is an extsock parameter in the R-URI, and if yes, it uses its values as the next target of the request i.e. sip:130.56.88.20:5432 (Maybe P1 could even remove the extsock parameter from the R-URI at this stage to make some Nokia E-Series' phones happier...) My questions: - Is this idea feasible? Does it work in any case? Any issues with forking? - Could it easily be configured to OpenSER, or is there a change in the source code necessairy? - How much does it break SIP standards? ;-) Looking forward to your feedback. Have a nice weekend! cheers, Bernie > On Wednesday 02 May 2007 08:33, Klaus Darilion wrote: >> Hi Alex! >> >> Without having done this: You can configure the SIP proxies as load >> balancers too which distribute the load over all the proxy/registrars >> (including itself). > > I thought of that, but then every REGISTER needs to be forwarded to another > proxy to get the path info right. This result in a great amount of traffic > between the proxies and extra processing power. I want the REGISTER request > to be handled on the first host it arrives on. Then only some INVITE's need > to create inter-proxy traffic.

2 3

[Serusers] Which version of SER to use?
by lists＠infoway.net 08 May '07

08 May '07

We are planning on deploying a couple of SER servers and integrate them with either SEMS or Asterisk for voicemail. However, we are a bit confused as to which version of SER to go with? From what we can tell, the latest stable version is 0.9.6. However, we know of the new 0.10.X version. Reading up on the site, there is some documentation that explains a little bit of the differences and/or migration recommendations/steps. So, in an effort to keep life as simple as possible, which version should we work with? We are afraid that if we go through a learning cycle with 0.9.6 we're going to "suffer" when migrating to the new version, either because of the learning curve or because of a potential painful migration process. Any recommendations? Thanks

2 1

[Serusers] Clueless newbie question - integrating SER and Asterisk
by John Breen 08 May '07

08 May '07

Help! I've followed the SER documentation and searched the web like a mad thing and found nothing useful. I have an asterisk box here at home and want to put SER on it so I can use SIP clients out in the real world (on the other side of my firewall) to get to it. Now, I have got SER set up and the phones register fine and everything, BUT I cannot see how to get it to pass calls through to Asterisk. I'm not using Asterisk realtime or anything like that - just plain old asterisk with plain old config files. I thought I had it right, but calls are just not getting to asterisk so far as I can see. I know it must be possible, I just can't see how to do it. Regards John Breen

3 2

[Serusers] SER/PSTN Gateway conf
by flavio 08 May '07

08 May '07

Hi to all, I'm newbie for SER and I'm testing my own configuration to send/receive calls through my pstn gateway, according to follow simple scenario: Phone --> PSTN --> Gateway --> SER --> SIP Phone I send and receive calls successfully, but when Phone over PSTN hangs up SER reply to BYE message with 404 Not Found.Here is my ser.cfg: debug=3 fork=no log_stderror=yes listen=10.28.19.202 # INSERT YOUR IP ADDRESS HERE port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://root@10.28.19.202/ser" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" modparam("auth_db|permissions|uri_db|usrloc", "db_url", "mysql://root@10.28.19.202/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("usrloc", "db_mode", 2) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("permissions", "default_allow_file", "/usr/local/etc/ser/permissions.allow") modparam("permissions", "default_deny_file", "/usr/local/etc/ser/permissions.deny") route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("16")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { # if ((method=="INVITE" || method=="REFER") && !has_totag()) { # sl_send_reply("403", "Forbidden"); # break; # }; if (method=="INVITE") { if (!allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(1); break; }; if (method=="ACK") { route(3); route(1); break; } if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- if (!t_relay()) { sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); } lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { if (uri=~"^sip:0[0-9]*@") { # calls to PSTN #strip(0); route(4); break; }; if(uri=~"^sip:3[0-9]*@") { calls to Mobile route(4); break; }; sl_send_reply("404", "User Not Found"); break; }; route(1); } route[4]{ #----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- rewritehost("10.28.52.105"); # INSERT YOUR PSTN GATEWAY IP ADDRESS avp_write("i:45", "inv_timeout"); route(1); } Any suggestions about my issue? Thanks 4 all, Flavio

4 5

[Serusers] re-invite problem
by olivier.taylor 08 May '07

08 May '07

Hello, using ser 0.9.6 The following dialog is a re-invite, the re-invite is initiated by the callee after putting the call on hold. Problem is that SER answer with a 407 on the wrong port, Invite is coming from port 62907 and SER answer to port 5060. Problem doesn't occurs for the initial invite. We have tried with other Uas without problems, just having problems with this Ua (Avaya). Any idea? Best regards, Olivier U xxx.xx.xx.xx:62907 -> yy.yyy.yyy.yy:5060 INVITE sip:990099001@192.168.162.187 SIP/2.0. Via: SIP/2.0/UDP 192.168.162.181;branch=z9hG4bK309b82ccf. Max-Forwards: 70. Content-Length: 337. To: "QE-A" <sip:990099001@yy.yyy.yyy.yy>;tag=f7983fe553e1941. From: "QE-B" <sip:990099002@yy.yyy.yyy.yy>;tag=5bf8b4fb69fca84. Call-ID: 7c933d3d19bb76c396b995202b18eb97(a)yy.yyy.yyy.yy. CSeq: 567882513 INVITE. Route: <sip:990099002@yy.yyy.yyy.yy:5060;nat=yes;ftag=f7983fe553e1941;lr=on>. P-Asserted-Identity: "QE-B" <sip:990099002@yy.yyy.yyy.yy>. Allow: INVITE. Allow: CANCEL. Allow: OPTIONS. Allow: BYE. Allow: REFER. Allow: INFO. Allow: UPDATE. Content-Type: application/sdp. Contact: "QE-B" <sip:990099002@192.168.162.181>. Supported: replaces. User-Agent: NeuralX MxSF/v3.2.6.26. . v=0. o=990099002 1136076203 1136076205 IN IP4 192.168.162.181. s=-. c=IN IP4 192.168.162.181. t=0 0. a=sendonly. m=audio 20006 RTP/AVP 0 101 8 18 . a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:101 0-15. a=fmtp:18 annexb=no. a=ptime:20. a=rtcp:20007 IN IP4 192.168.162.181. U yy.yyy.yyy.yy:5060 -> xxx.xx.xx.xx:5060 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 192.168.162.181;branch=z9hG4bK309b82ccf;received=xxx.xx.xx.xx. To: "QE-A" <sip:990099001@yy.yyy.yyy.yy>;tag=f7983fe553e1941. From: "QE-B" <sip:990099002@yy.yyy.yyy.yy>;tag=5bf8b4fb69fca84. Call-ID: 7c933d3d19bb76c396b995202b18eb97(a)yy.yyy.yyy.yy. CSeq: 567882513 INVITE. Proxy-Authenticate: Digest realm="yy.yyy.yyy.yy", nonce="463f31b81a65dee3419ce3e4bfbecc253410cbf5". Server: Sip EXpress router (0.9.6 (i386/freebsd)). Content-Length: 0. Warning: 392 yy.yyy.yyy.yy:5060 "Noisy feedback tells: pid=49990 req_src_ip=xxx.xx.xx.xx req_src_port=62907 in_uri=sip:990099001@192.168.162.187 out_uri=sip:990099001@192.168.162.187 via_cnt==1". .

2 1

RE: [Users] lcr: load_gws(): from_uri to large
by Papadopoulos Georgios 08 May '07

08 May '07

Hi Juha, Exactly this is my point, 128 should work and 129 should fail, especially since all related char buffers are allocated properly with size [MAX_FROM_URI_LEN+1] I think that the bug is here: if (from_uri.len < MAX_FROM_URI_LEN) { strncpy(from_uri_str, from_uri.s, from_uri.len); from_uri_str[from_uri.len] = '\0'; } else { LOG(L_ERR, "load_gws(): from_uri to large\n"); return -1; } It should be if (from_uri.len <= MAX_FROM_URI_LEN) George > -----Original Message----- > From: Juha Heinanen [mailto:jh@tutpro.com] > Sent: Monday, May 07, 2007 5:44 PM > To: Papadopoulos Georgios > Cc: users(a)openser.org > Subject: [Users] lcr: load_gws(): from_uri to large > > Papadopoulos Georgios writes: > > > I get the following error from lcr module > load_gws(): > from_uri to large > This happens when the rpid is exactly > 128 characters long. If I make it > 127 long it works fine. > This is on the 1.1.x branch. Has anyone else > noticed this? > > MAX_FROM_URI_LEN constant is defined as > > > #define MAX_FROM_URI_LEN 128 > > i can change that to 256. > > -- juha > Disclaimer The information in this e-mail and any attachments is confidential. It is intended solely for the attention and use of the named addressee(s). If you are not the intended recipient, or person responsible for delivering this information to the intended recipient, please notify the sender immediately. Unless you are the intended recipient or his/her representative you are not authorized to, and must not, read, copy, distribute, use or retain this message or any part of it. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.

2 1

[Users] 2 ENUM lookups
by Joao Pereira 08 May '07

08 May '07

Hello I would like to know how to lookup to 2 ENUM trees (arpa and e164.org). I have this config: loadmodule "enum.so" modparam("enum", "domain_suffix", "e164.org.") if ( method=="INVITE" && uri=~"sip:\+[1-9][0-9]*@mydomain.com") { if(!enum_query("e164.org.")) { log(1, "log: ENUM not found :( \n"); }else{ log(1, "log: ENUM found :) \n"); if (!method=="REGISTER") record_route(); t_relay(); return; } } Thanks Regards Joao Pereira -- ______________________________________________ João Gomes Pereira FCCN Av. do Brasil, nº 101 1700-066 Lisboa tel: +351 218 440 100 - fax: +351 218 472 167 email|SIP: joao.pereira(a)fccn.pt http://www.fccn.pt ______________________________________________ --- Aviso de Confidencialidade Esta mensagem é exclusivamente destinada ao seu destinatário, podendo conter informação CONFIDENCIAL, cuja divulgação está expressamente vedada nos termos da lei. Caso tenha recepcionado indevidamente esta mensagem, solicitamos-lhe que nos comunique esse mesmo facto por esta via ou para o telefone +351 218 440 100 devendo apagar o seu conteúdo de imediato. This message is intended exclusively for its addressee. It may contain CONFIDENTIAL information protected by law. If this message has been received by error, please notify us via e-mail or by telephone +351 218 440 100 and delete it immediately. ---

3 2

Re: [Serusers] Fwd: record_route..........TCP ??
by KUMAR 07 May '07

07 May '07

Here is the message dump that i get when using TCP as transport. > > > On 5/3/07, Jiri Kuthan <jiri(a)iptel.org> wrote: > > I think you would have to send message dumps first so that [serusers] volunteers > > have material for providing an answer. > > > > -jiri > > > > At 12:07 03/05/2007, KUMAR wrote: > > >- > > >Hi all, > > >I already posted this message yet havent got any replies. I really > > >need to find this out. > > >Please anyone reply to this problem. > > > > > >I am using SER-0.9.6 and the problem i'm having is this. When using > > >the following ser.cfg, it works well when the transport UDP is used. > > >But when transport TCP is used, then it only results in one way IM, > > >only from the UA from which the INVITE is being sent. Moreover, when > > >record_route is used instead of record_route_preset, then even that > > >one way IM doesn't work. But it works well with UDP. > > > > > >Can anyone please point me as to where the problem might be?? > > >Thank you in advance. > > > > > >kumar > > > > > > > > >Content-Type: application/octet-stream; name=ser.cfg > > >X-Attachment-Id: f_f18sg2bl > > >Content-Disposition: attachment; filename="ser.cfg" > > > > > >_______________________________________________ > > >Serusers mailing list > > >Serusers(a)lists.iptel.org > > >http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > -- > > Jiri Kuthan http://iptel.org/~jiri/ > > > > > >

6 12

Re: [Users] Openser generated strange SIP packets with LCR Fallback on a second gateway
by gemalto-fw＠e-qual.fr 07 May '07

07 May '07

2 1

[Users] [openser 1.2] $fn, strange shape.
by Bodin Bruno 07 May '07

07 May '07

Hello, When I do : xlog("display name = $fn"); It show : display name = "toto titi" why it add quote ? how can I disable theses quotes ? thank

4 6

[Users] 183 early media and NATed clients
by Jeff Williams 07 May '07

07 May '07

Guys, I'm wondering what solutions there are to this problem. We have some NATed clients making outgoing calls. The NAT traversal is done using mediaproxy and works fine. However, we have a problem with 183 responses and early media ringing sounds. The problem is that before the NATed client starts sending out RTP data, mediaproxy doesn't know which IP and port to send the RTP traffic to. Some of the NETed clients don't start sending out RTP traffic until the call is connected, therefore none of the early media (ringing sounds) reaches the user - making it sound like dead air. Any ideas? Jeff

4 7

[Users] Openser generated strange SIP packets with LCR Fallback on a second gateway
by mailinglist-openser-users 07 May '07

07 May '07

gallois:/var/log# sudo /usr/bin/ngrep -W byline "" "port 5060" |more filter: (ip or ip6) and ( port 5060 ) # U phone:5060 -> openser:5060 INVITE sip:xxxxxxxx@sip1.e-qual.fr SIP/2.0. Call-ID: call-717C8C1D-CEDE-2910-0D00-26@phone. Contact: <sip:phone>. Content-Length: 221. Content-Type: application/sdp. CSeq: 1 INVITE. From: <sip:sip1.e-qual.fr>;tag=c0a80843-16. Max-Forwards: 70. Session-GUID: 875967334-858992696-842282596-875782963. Supported: 100rel. To: <sip:xxxxxxxx@sip1.e-qual.fr>. User-Agent: Quintum/1.0.0. Via: SIP/2.0/UDP phone;branch=z9hG4bK-tenor-c0a8-0843-001c. Quintum: 0c01030b0237370501001301000f0b413031322d323032334332. . v=0. o=Quintum 17 17 IN IP4 phone. s=VoipCall. c=IN IP4 phone. t=0 0. m=audio 10274 RTP/AVP 4 101. c=IN IP4 phone. a=rtpmap:4 g723/8000/1. a=ptime:30. a=rtpmap:101 telephone-event/8000/1. a=sendrecv. # U openser:5060 -> phone:5060 SIP/2.0 100 Giving a try. Call-ID: call-717C8C1D-CEDE-2910-0D00-26@phone. CSeq: 1 INVITE. From: <sip:sip1.e-qual.fr>;tag=c0a80843-16. To: <sip:xxxxxxxx@sip1.e-qual.fr>. Via: SIP/2.0/UDP phone;branch=z9hG4bK-tenor-c0a8-0843-001c. Server: OpenSER (1.2.0-notls (i386/linux)). Content-Length: 0. . # U openser:5060 -> gateway1:5060 INVITE sip:xxxxxxxxx@gateway1:5060;transport=udp SIP/2.0. Record-Route: <sip:openser;lr=on;ftag=c0a80843-16>. Call-ID: call-717C8C1D-CEDE-2910-0D00-26@phone. Contact: <sip:phone>. Content-Length: 221. Content-Type: application/sdp. CSeq: 1 INVITE. From: <sip:sip1.e-qual.fr>;tag=c0a80843-16. Max-Forwards: 69. Session-GUID: 875967334-858992696-842282596-875782963. Supported: 100rel. To: <sip:xxxxxxxx@sip1.e-qual.fr>. User-Agent: Quintum/1.0.0. Via: SIP/2.0/UDP openser;branch=z9hG4bKe7a2.b7f82867.0. Via: SIP/2.0/UDP phone;branch=z9hG4bK-tenor-c0a8-0843-001c. Quintum: 0c01030b0237370501001301000f0b413031322d323032334332. P-hint: outbound PSTN. . v=0. o=Quintum 17 17 IN IP4 phone. s=VoipCall. c=IN IP4 phone. t=0 0. m=audio 10274 RTP/AVP 4 101. c=IN IP4 phone. a=rtpmap:4 g723/8000/1. a=ptime:30. a=rtpmap:101 telephone-event/8000/1. a=sendrecv. # U gateway1:5060 -> openser:5060 SIP/2.0 488 Not acceptable here. Via: SIP/2.0/UDP openser;branch=z9hG4bKe7a2.b7f82867.0;received=openser. Via: SIP/2.0/UDP phone;branch=z9hG4bK-tenor-c0a8-0843-001c. From: <sip:sip1.e-qual.fr>;tag=c0a80843-16. To: <sip:xxxxxxxx@sip1.e-qual.fr>;tag=as714f5f56. Call-ID: call-717C8C1D-CEDE-2910-0D00-26@phone. CSeq: 1 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Contact: <sip:xxxxxxxxx@gateway1>. Content-Length: 0. . # U openser:5060 -> gateway1:5060 ACK sip:xxxxxxxxx@gateway1:5060;transport=udp SIP/2.0. Via: SIP/2.0/UDP openser;branch=z9hG4bKe7a2.b7f82867.0. From: <sip:sip1.e-qual.fr>;tag=c0a80843-16. Call-ID: call-717C8C1D-CEDE-2910-0D00-26@phone. To: <sip:xxxxxxxx@sip1.e-qual.fr>;tag=as714f5f56. CSeq: 1 ACK. User-Agent: OpenSER (1.2.0-notls (i386/linux)). Content-Length: 0. . # U openser:5060 -> gateway2:5060 INVITE sip:xxxxxxxx@gateway2:5060;transport=udp..t SIP/2.0. Record-Route: <sip:openser;lr=on;ftag=c0a80843-16>. Call-ID: call-717C8C1D-CEDE-2910-0D00-26@phone. Contact: <sip:phone>. Content-Length: 221. Content-Type: application/sdp. CSeq: 1 INVITE. From: <sip:sip1.e-qual.fr>;tag=c0a80843-16. Max-Forwards: 69. Session-GUID: 875967334-858992696-842282596-875782963. Supported: 100rel. To: <sip:xxxxxxxx@sip1.e-qual.fr>. User-Agent: Quintum/1.0.0. Via: SIP/2.0/UDP openser;branch=z9hG4bKe7a2.b7f82867.1. Via: SIP/2.0/UDP phone;branch=z9hG4bK-tenor-c0a8-0843-001c. Quintum: 0c01030b0237370501001301000f0b413031322d323032334332. P-hint: outbound PSTN. . v=0. o=Quintum 17 17 IN IP4 phone. s=VoipCall. c=IN IP4 phone. t=0 0. m=audio 10274 RTP/AVP 4 101. c=IN IP4 phone. a=rtpmap:4 g723/8000/1. a=ptime:30. a=rtpmap:101 telephone-event/8000/1. a=sendrecv. # U openser:5060 -> gateway2:5060 INVITE sip:xxxxxxxx@gateway2:5060;transport=udp..t SIP/2.0. Record-Route: <sip:openser;lr=on;ftag=c0a80843-16>. Call-ID: call-717C8C1D-CEDE-2910-0D00-26@phone. Contact: <sip:phone>. Content-Length: 221. Content-Type: application/sdp. CSeq: 1 INVITE. From: <sip:sip1.e-qual.fr>;tag=c0a80843-16. Max-Forwards: 69. Session-GUID: 875967334-858992696-842282596-875782963. Supported: 100rel. To: <sip:xxxxxxxx@sip1.e-qual.fr>. User-Agent: Quintum/1.0.0. Via: SIP/2.0/UDP openser;branch=z9hG4bKe7a2.b7f82867.1. Via: SIP/2.0/UDP phone;branch=z9hG4bK-tenor-c0a8-0843-001c. Quintum: 0c01030b0237370501001301000f0b413031322d323032334332. P-hint: outbound PSTN. . v=0. o=Quintum 17 17 IN IP4 phone. s=VoipCall. c=IN IP4 phone. t=0 0. m=audio 10274 RTP/AVP 4 101. c=IN IP4 phone. a=rtpmap:4 g723/8000/1. a=ptime:30. a=rtpmap:101 telephone-event/8000/1. a=sendrecv. # U openser:5060 -> gateway2:5060 INVITE sip:xxxxxxxx@gateway2:5060;transport=udp..t SIP/2.0. Record-Route: <sip:openser;lr=on;ftag=c0a80843-16>. Call-ID: call-717C8C1D-CEDE-2910-0D00-26@phone. Contact: <sip:phone>. Content-Length: 221. Content-Type: application/sdp. CSeq: 1 INVITE. From: <sip:sip1.e-qual.fr>;tag=c0a80843-16. Max-Forwards: 69. Session-GUID: 875967334-858992696-842282596-875782963. Supported: 100rel. To: <sip:xxxxxxxx@sip1.e-qual.fr>. User-Agent: Quintum/1.0.0. Via: SIP/2.0/UDP openser;branch=z9hG4bKe7a2.b7f82867.1. Via: SIP/2.0/UDP phone;branch=z9hG4bK-tenor-c0a8-0843-001c. Quintum: 0c01030b0237370501001301000f0b413031322d323032334332. P-hint: outbound PSTN. . v=0. o=Quintum 17 17 IN IP4 phone. s=VoipCall. c=IN IP4 phone. t=0 0. m=audio 10274 RTP/AVP 4 101. c=IN IP4 phone. a=rtpmap:4 g723/8000/1. a=ptime:30. a=rtpmap:101 telephone-event/8000/1. a=sendrecv. # U openser:5060 -> gateway2:5060 INVITE sip:xxxxxxxx@gateway2:5060;transport=udp..t SIP/2.0. Record-Route: <sip:openser;lr=on;ftag=c0a80843-16>. Call-ID: call-717C8C1D-CEDE-2910-0D00-26@phone. Contact: <sip:phone>. Content-Length: 221. Content-Type: application/sdp. CSeq: 1 INVITE. From: <sip:sip1.e-qual.fr>;tag=c0a80843-16. Max-Forwards: 69. Session-GUID: 875967334-858992696-842282596-875782963. Supported: 100rel. To: <sip:xxxxxxxx@sip1.e-qual.fr>. User-Agent: Quintum/1.0.0. Via: SIP/2.0/UDP openser;branch=z9hG4bKe7a2.b7f82867.1. Via: SIP/2.0/UDP phone;branch=z9hG4bK-tenor-c0a8-0843-001c. Quintum: 0c01030b0237370501001301000f0b413031322d323032334332. P-hint: outbound PSTN. . v=0. o=Quintum 17 17 IN IP4 phone. s=VoipCall. c=IN IP4 phone. t=0 0. m=audio 10274 RTP/AVP 4 101. c=IN IP4 phone. a=rtpmap:4 g723/8000/1. a=ptime:30. a=rtpmap:101 telephone-event/8000/1. a=sendrecv.

2 1

[Users] lcr: load_gws(): from_uri to large
by Papadopoulos Georgios 07 May '07

07 May '07

Hello all, I get the following error from lcr module load_gws(): from_uri to large This happens when the rpid is exactly 128 characters long. If I make it 127 long it works fine. This is on the 1.1.x branch. Has anyone else noticed this? thank you George Disclaimer The information in this e-mail and any attachments is confidential. It is intended solely for the attention and use of the named addressee(s). If you are not the intended recipient, or person responsible for delivering this information to the intended recipient, please notify the sender immediately. Unless you are the intended recipient or his/her representative you are not authorized to, and must not, read, copy, distribute, use or retain this message or any part of it. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.

2 1

[Users] Openser used to route call between PSTN to/from remote voice gateways with routing different in each direction ( originate from PSTN or originate from remote voice gateway )
by mailinglist-openser-users 07 May '07

07 May '07

Version: Openser 1.2.0 Platform: PSTN <-> Central Gateways <-- OpenSer --> Remote Gateways <-- >PABX We have 4 central Voice Gateway connected to local PSTN and several tens of customer remote voice gateways connected to PABX. On our central voice gateways, we have several PSTN provider, and these providers have different prices for different location ( so calling a cell phone should be rather made by a voice gateway connected to a particular PSTN provider ). As well, we provide SDA numbers for incoming call from PSTN on our central gateway and that should be routed to a remote customer voice gateway. But if customerA call the SDA of customerB, customerA should go through PSTN by our central voice gateway and then our central voice gateway route the call to CustomerB. So all remote gateways should not know any of the SDA route, only our central gateway should use this routes. So basically we have 2 kinds of routing to be done by OpenSer: o For our Central Gateway - For SDA11, try gateway Customer1 - For SDA12, try gateway Customer1 - For SDA21, try gateway Customer2 - For SDA22, try gateway Customer2 - For any other unknown number, no gateway should be returned ( I don't want a wrong number routed to our gateway from PSTN to generate another call on PSTN ) o For our remote customer gateways, I should have routing like this: - for cell phone, try gateway 1, if that gateway1 fail then try randomly gateway 2,3 or 4 ( and if possible the less loaded of these 3 gateways ) - for spain try gateway2, if that gateway2 fail, then try randomly gateway 1,3 or 4 ( and if possible the less loaded of these 3 gateways ) - for all other routes, try randomly gateway 1,2,3,4 ( and if possible the less loaded of these 4 gateways ) So I have several constraint: - two kind of routing should be made either if the call is coming from the central gateway or from a remote customer gateway ( differenlty said: routing different if call originate from PSTN or originate from remote voice gateway ) - for customer remote gateways, there are some preferred central gateway for certains location (e.g international call ) but if these gateways fail, fallback should occur to others central gateway - the less is hardcoded in the configuration file, the better is is. - how to try first the currently less loaded gateway when several gateways are available ( is it possible in openser has it is only dialog statefull and not call statefull ? ) What is the best option in OpenSer to achieve this ? Is the LCR module with "from_uri" and route by "prefix" can handle all this ? Where can we find complex examples of the LCR module ? Regards, Jerome. -- Jerome

1 0

[Users] do I need openser FIFO?
by Joao Pereira 07 May '07

07 May '07

Hello I'm using SER/OpenSER for a long time and I always used the FIFO. My question is, do I really need to use the FIFO? What OpenSER functionalities rely in the FIFO? Just Serweb? Thanks regards Joao Pereira -- ______________________________________________ João Gomes Pereira FCCN Av. do Brasil, nº 101 1700-066 Lisboa tel: +351 218 440 100 - fax: +351 218 472 167 email|SIP: joao.pereira(a)fccn.pt http://www.fccn.pt ______________________________________________ --- Aviso de Confidencialidade Esta mensagem é exclusivamente destinada ao seu destinatário, podendo conter informação CONFIDENCIAL, cuja divulgação está expressamente vedada nos termos da lei. Caso tenha recepcionado indevidamente esta mensagem, solicitamos-lhe que nos comunique esse mesmo facto por esta via ou para o telefone +351 218 440 100 devendo apagar o seu conteúdo de imediato. This message is intended exclusively for its addressee. It may contain CONFIDENTIAL information protected by law. If this message has been received by error, please notify us via e-mail or by telephone +351 218 440 100 and delete it immediately. ---

2 1

[Serusers] SER/PSTN Gateway conf
by flavio 07 May '07

07 May '07

Hi to all, I'm newbie for SER and I'm testing my own configuration to send/receive calls through my pstn gateway, according to follow simple scenario: Phone --> PSTN --> Gateway --> SER --> SIP Phone I send and receive calls successfully, but when Phone over PSTN hangs up SER reply to BYE message with 404 Not Found.Here is my ser.cfg: debug=3 fork=no log_stderror=yes listen=10.28.19.202 # INSERT YOUR IP ADDRESS HERE port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://root@10.28.19.202/ser" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" modparam("auth_db|permissions|uri_db|usrloc", "db_url", "mysql://root@10.28.19.202/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("usrloc", "db_mode", 2) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("permissions", "default_allow_file", "/usr/local/etc/ser/permissions.allow") modparam("permissions", "default_deny_file", "/usr/local/etc/ser/permissions.deny") route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("16")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { # if ((method=="INVITE" || method=="REFER") && !has_totag()) { # sl_send_reply("403", "Forbidden"); # break; # }; if (method=="INVITE") { if (!allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(1); break; }; if (method=="ACK") { route(3); route(1); break; } if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- if (!t_relay()) { sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); } lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { if (uri=~"^sip:0[0-9]*@") { # calls to PSTN #strip(0); route(4); break; }; if(uri=~"^sip:3[0-9]*@") { calls to Mobile route(4); break; }; sl_send_reply("404", "User Not Found"); break; }; route(1); } route[4]{ #----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- rewritehost("10.28.52.105"); # INSERT YOUR PSTN GATEWAY IP ADDRESS avp_write("i:45", "inv_timeout"); route(1); } Any suggestions about my issue? Thanks 4 all, Flavio -- ******************************** * (o< ing. Patria Flavio * //\ phone 0823451358 * V_/_ mobile 3407873357 * ********************************

1 0

[Users] HOWTO submit a patch in a correct way?
by Peter Lemenkov 06 May '07

06 May '07

Hello All! I need a simple feature in OpenSER - parameter in Jabber-module to set up jabber's priority. So I took a look into the sources and created a simple patch (attached). Are there any formal procedures that I must follow for accepting my patch? -- With best regards!

2 1

[Users] E1 config for chile
by laurent schweizer 04 May '07

04 May '07

Hello, I try to configure a Patton smart node with a E1 for the Chile. I need to know wich parameters I must set for Chile. If someone have informations it's welcoms. Laurent

1 0

[Users] Passing predefined string-value to RADIUS accounting server
by Peter Lemenkov 04 May '07

04 May '07

Hello All! I've encountered some difficulties. I need to pass predefined string value ("REALM" actually) to accounting RADIUS server. I want to did it in way similar to: modparam("acc", "radius_extra", "User-Name=$fU; Sip-User-Realm=$ar") But I need to set realm for all radius-accounting requests to some value ("MYREALM" for example). I tried set $ar explicitly before accounting this way $ar = "MYREALM"; but failed with error "invalid left operand in assignment". Does $ar read-only variable? Another my try was to pass new realm value in module configuration section: modparam("acc", "radius_extra", "User-Name=$fU; Sip-User-Realm=MYREALM") It failed with message "parse failed in <Sip-User-Realm=MYREALM> around position ..." I can't set up custom variable before route{} sections so I can't define my $var(realm) into module configuration line. Are any ways to solve my problem exists? -- With best regards!

2 1

[Users] forcing tcp transport
by Will Quan 04 May '07

04 May '07

I am setting up the proxy so the transport protocol is forced to tcp when packet size is > 1300b. The problem is when the originator abandons the call, the CANCEL msg sent over UDP and I really want it to use the TCP transport, as the INVITE did. Is there a way to achieve this?--will

2 1

[Users] OpenSER 1.2 and Flatstore Accounting
by Brandon Armstead 04 May '07

04 May '07

Hello my fellow Flatstore Accounting users, There seems to be some issues with Flatstore and Accounting in OpenSER 1.2, it does not seem to properly account/log extras? Perhaps I am incorrectly using it, but the syntax still remains modparam("acc", "log_extra", "foo=$pvar"). Is there anything I'm missing? Thanks! Regards, Brandon.

2 1

[Users] rtpproxy CPU usage issue..
by Sajith T S 04 May '07

04 May '07

Hi, Have any of you folks seen an abnormal CPU usage issue with rtpprpxy? At times, "top" shows rtpproxy uses 100% CPU (sometimes 99%, sometimes even 101%!), and it stops responding to openser. I have to restart everything now. A small thread in Serdev list recommends upgrading rtpproxy to latest in CVS: http://lists.iptel.org/pipermail/serdev/2006-June/007459.html Mine was fairly old, so I upgraded, but without much progress. What really bugs is I don't know what triggers it. Backtraces suggest that this occurs in different places in rtpproxy, before and after upgrading. Before upgrade: (gdb) where #0 0xb7f49410 in ?? () #1 0xbffd6d88 in ?? () #2 0x00000076 in ?? () #3 0xbffd2410 in ?? () #4 0x4a8d9fd1 in sendto () from /lib/tls/libc.so.6 #5 0x0804ac0c in main (argc=7, argv=0xbffd6e14) at main.c:1467 After upgrade: (gdb) where #0 0xb7f0a410 in ?? () #1 0xbfcb0258 in ?? () #2 0xbfcadfc0 in ?? () #3 0xbfcab8e0 in ?? () #4 0x00bb8dd1 in recvfrom () from /lib/tls/libc.so.6 #5 0x0804a568 in main (argc=7, argv=0xbfcb02e4) at main.c:1364 Any clues? Is this a known problem and is there a patch or some sort ofworkaround? Is there anything I need to upgrade? (This is CentOS 4.4, FWIW) Anything that should be done in openser configuration? Thanks much in advance. - Sajith.

3 2

[Users] Abyss XMLRPC server CRLF problem
by jh＠tutpro.com 04 May '07

04 May '07

this discussion on abyss xmlrpc server reminds me about a problem on its handling of CRLF, which needs to be included in a string when applications use t_uac_dlg or send_publish mi commands to send sip requests to openser. if abyss receives CRLF in a string, it drops CR and only LF is left. if abyss server receives CRCRLF, it results in LFLF. abyss does retain CR in a string if it receives if coded as . however, none of xmlrpc clients that i have tried (python or php) allows me to send out on the wire  because they (according to xmlrpc spec) convert & to & and i get on the wire &#13;, which abyss does not convert to CR. the problem results from abyss author's own interpretation of xmlrpc specification: http://xmlrpc-c.sourceforge.net/doc/libxmlrpc.html#xmlrpc_value XML-RPC strings deserve a little more detailed explanation because of some nonsense in the definition of them in the XML-RPC specification. The spec says a string's value is arbitrary 8 bit bytes, but also says that it is part of an XML document. An XML document is composed of XML characters, and the spec doesn't mention any way of encoding those arbitrary bytes into XML characters. It appears that the author was confused about how XML works. as result, i have found no way (without hacking xmlrpc client libraries) to get CRLF via abyss server to openser. does anybody have a suggestion? -- juha

1 0

[Users] [Fwd: [Serdev] Contributing new module "prefix_route" to SER]
by Klaus Darilion 04 May '07

04 May '07

FYI: A new ser module which could probably easily ported to openser too (if someone needs this feature) regards klaus -------- Original Message -------- Subject: [Serdev] Contributing new module "prefix_route" to SER Date: Wed, 25 Apr 2007 16:52:39 +0200 From: Alfred E. Heggestad <aeh(a)db.org> To: serdev(a)iptel.org Hi Here is a contribution to the SER community; a new module called "prefix_route" that does prefix-based routing from rules stored in the database. The work was sponsored by my employer, Telio Telecom. The source code is attached, and can also be downloaded from here: http://aeh.db.org/patch/prefix_route-20070425.tar.gz The module has been tested with ser-0.9.6 and ser-0.9.7-pre8 from CVS. It could also be ported to SER 2.0 if there is any interest. The idea with this module is to move the logic of selecting PSTN gateway from ser.cfg to the database. In the database you will have a table called e.g. "prefix_route" which has one entry per rule, matching a prefix to a route number: prefix route ------ ----- 0045 -> route 3 0046 -> route 4 0047 -> route 5 for incoming requests, you can call the prefix_route() function in your ser.cfg - it will then try to match the username part of the request-uri to the "best" prefix, i.e. closest match. if there is a match, it will jump to that routing block in ser.cfg. all referenced route numbers must be declared in ser.cfg, otherwise it will not load. The prefix_route table is populated from the DB at boot or with the "prefix_reload" FIFO command, into a RAM-based tree that is used in execution time. For more documentation, please see the README file ;) I hope that this module will be useful for you, any feedback will be appreciated! /alfred _______________________________________________ Serdev mailing list Serdev(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serdev

2 1

[Users] make OpenSER 1.2.0: osp/osp.h: No such file or directory
by CSB 04 May '07

04 May '07

Attempting to make OpenSER 1.2.0 and getting the following errors: In file included from destination.c:39: usage.h:34:21: error: osp/osp.h: No such file or directory In file included from destination.c:39: usage.h:52: error: syntax error before âOSPTTRANHANDLEâ usage.h:53: error: syntax error before âOSPTTRANHANDLEâ destination.c: In function âospConvertAddressâ: destination.c:470: warning: implicit declaration of function âsnprintfâ destination.c:470: warning: incompatible implicit declaration of built-in function âsnprintfâ destination.c:472: warning: incompatible implicit declaration of built-in function âsnprintfâ make[1]: *** [destination.o] Error 1 make[1]: Leaving directory `/usr/src/openser-1.2.0-notls/modules/osp' make: *** [modules] Error 2 FC4 rpm -q libxml2-devel libxml2-devel-2.6.20-1.FC4 rpm -q curl-devel curl-devel-7.13.1-5.fc4 Any advice appreciated. Regards Cameron

2 1

RE: [Users] ACC : log_fmt equivalent in 1.2 ?
by Will Quan 04 May '07

04 May '07

thanks daniel,it looks like i can only append to the 'base' fields of the log. i was looking to reformat the whole syslog, including those those first few starting with timestamp, method, etc..thanks,-will> Date: Thu, 3 May 2007 21:04:21 +0300> From: daniel(a)voice-system.ro> To: sip(a)voiplabs.net> CC: users(a)openser.org> Subject: Re: [Users] ACC : log_fmt equivalent in 1.2 ?> > Hello,> > you can get it via the log_extra parameter:> http://www.openser.org/docs/modules/1.2.x/acc.html#AEN276> http://openser.org/dokuwiki/doku.php/install:1.1.x-to-1.2.x#acc> > Cheers,> Daniel> > On 05/03/07 18:03, Will Quan wrote:> > In version 1.1.0 it was possible to specify the log_fmt for the ACC > > records that go into the syslog.> > modparam("acc", "log_fmt", "miocfsur");> >> > It appears this has been removed in 1.2, so I was wondering if there > > is a new way to achieve the same result.> > That is, produce ACC records with the same syslog format as my 1.3 > > version.> >> > Thanks,> > Will> > ------------------------------------------------------------------------> >> > _______________________________________________> > Users mailing list> > Users(a)openser.org> > http://openser.org/cgi-bin/mailman/listinfo/users> >

2 1

[Users] uac_replace_from and avp
by CSB 03 May '07

03 May '07

In OpenSER 1.1 I use uac_replace_from to rewrite the from details. I want the rewritten from to be based on an avp value but I don't think that is possible. Am I right? If so, is it possible in 1.2? On a related point, is anyone working on fixing the CSEQ issue with the uac module or is there a plan for this? I know it's been on the todo list for a while. Regards Cameron

3 3

Res: [Users] LCR module (1.2.0)
by Bruno Machado 03 May '07

03 May '07

Hi Heinanen When I was using the version 1.1.0, I needed to set a preferred route to my voip 123. This way, it was necessary to insert this underscores after 0. They match any single character (like dot in regular expressions). I used it because the prefix needs to match the longest user part. prefix-------------------from_uri-------grp_id--------priority 0123456------------------------%------------ 1 ---------------1 0________-------------sip:123@.*----------- 2 ---------------0 If I call to number 01234560000 with voip 777, the GW 1 will be selected. Instead, when I use voip 123, the GW 2 is returned. This way, all the calls originated by voip 123 having a size of user part equal 7 or greater, will match with the second line. It does not happens with I have these lines (without the underscores): prefix-------------------from_uri-------grp_id--------priority 0123456------------------------%------------ 1 ---------------1 0-----------------------sip:123@.*----------- 2 ---------------0 In this case, GW 1 will be always selected to any voip. It works in version 1.1.0, but I need to set this behavior in 1.2.0. I know that from_uri in this version is a regular expression. So, I added a '^' in the beginning. prefix-------------------from_uri-------grp_id--------priority 0123456-------------------------------------- 1 ---------------1 0________------------^sip:123@.*--------- 2 ---------------0 The version 1.2.0 always use the LCR in memory. I looked for the query used by 1.1.0 in lcr_mod.h but I didn't find it. So, I don't know if the underscores is necessary because it is used by the MySQL. In my test, just the GW 1 was selected when I called using voip 123. Using other words: I want to route all the calls originated by voip 123 through GW 2 (when beginning with a 0) and the others to GW 1. Thanks for your help, Heinanen. Bruno ----- Mensagem original ---- De: Juha Heinanen <jh(a)tutpro.com> Para: Bruno Machado <bsmufu(a)yahoo.com.br> Cc: users(a)openser.org Enviadas: Quinta-feira, 3 de Maio de 2007 12:36:33 Assunto: [Users] LCR module (1.2.0) Bruno Machado writes: > prefix----------from_uri-------grp_id--------priority > 0123456------------------------------ 1 ---------------1 > 0________------^sip:123.*--------- 2 ---------------0 > > Ok. The LCR documentation tells that the column 'from_uri' must be a > regular expression in this version. The route selected is the first > line. I'd like to route the voip number 123 through GW 2, but it > doesn't happens. i don't understand based on the above, how your lcr table looks like and which uri you are trying to call. what are the underscores after 0? prefix is just a prefix, nothing else. -- juha __________________________________________________ Fale com seus amigos de graça com o novo Yahoo! Messenger http://br.messenger.yahoo.com/

2 1

[Users] ACC : log_fmt equivalent in 1.2 ?
by Will Quan 03 May '07

03 May '07

In version 1.1.0 it was possible to specify the log_fmt for the ACC records that go into the syslog. modparam("acc", "log_fmt", "miocfsur");It appears this has been removed in 1.2, so I was wondering if there is a new way to achieve the same result. That is, produce ACC records with the same syslog format as my 1.3 version.Thanks,Will

2 1

[Users] Rejecting SUBSCRIBE/NOTIFY messages
by Andreas Sikkema 03 May '07

03 May '07

Hi, I've been googling for a while and I haven't been able to find out how to do the following: I would like to reject all SUBSCRIBE and NOTIFY messages received by my OpenSER server. How do I do properly? if (method=="SUBSCRIBE") { }; if (method=="NOTIFY") { }; I couldn't find any examples and Google also didn't find anything that pointed me in the right direction. Thanks! -- Andreas Sikkema

4 5

Res: [Users] LCR module (1.2.0)
by Bruno Machado 03 May '07

03 May '07

Other thing: the 1.2.0 is using the priority in contrary? prefix----------from_uri-------grp_id--------priority 0123456------------------------------ 1 ---------------1 0123456----------------------------- 2 ---------------0 When I called to 01234560000, GW 1 was selected. In version 1.1.0, the GW 2 will be selected because it has the minor number. Bruno ----- Mensagem original ---- De: Juha Heinanen <jh(a)tutpro.com> Para: Bruno Machado <bsmufu(a)yahoo.com.br> Cc: users(a)openser.org Enviadas: Quinta-feira, 3 de Maio de 2007 12:36:33 Assunto: [Users] LCR module (1.2.0) Bruno Machado writes: > prefix----------from_uri-------grp_id--------priority > 0123456------------------------------ 1 ---------------1 > 0________------^sip:123.*--------- 2 ---------------0 > > Ok. The LCR documentation tells that the column 'from_uri' must be a > regular expression in this version. The route selected is the first > line. I'd like to route the voip number 123 through GW 2, but it > doesn't happens. i don't understand based on the above, how your lcr table looks like and which uri you are trying to call. what are the underscores after 0? prefix is just a prefix, nothing else. -- juha __________________________________________________ Fale com seus amigos de graça com o novo Yahoo! Messenger http://br.messenger.yahoo.com/

1 0

[Users] LCR module (1.2.0)
by Bruno Machado 03 May '07

03 May '07

Hi I'm in trouble with the new version 1.2.0. Before migration to 1.2.0, the LCR table was working without problems. The table has these lines now: prefix----------from_uri-------grp_id--------priority 0123456------------------------------ 1 ---------------1 0________------^sip:123.*--------- 2 ---------------0 Ok. The LCR documentation tells that the column 'from_uri' must be a regular expression in this version. The route selected is the first line. I'd like to route the voip number 123 through GW 2, but it doesn't happens. I already reloaded the table, restarted the proxy... all this things. Certainly, I'm doing something wrong. What I should do? Thanks in advance. Bruno Machado __________________________________________________ Fale com seus amigos de graça com o novo Yahoo! Messenger http://br.messenger.yahoo.com/

2 1

[Users] Increase or Decrease openser debug value without restarting openser
by mailinglist-openser-users 03 May '07

03 May '07

Hi, openser 1.2.0. In the "openser.cfg", there is the "debug" variable which increase or decrease the verbosity of openser and its modules as well as the xlog I would like to know if it is possible to change the debug value without having to restart openser in order to be able to troubleshoot a problem without having to restart openser. Regards, Jerome -- Jerome

2 1

[Users] mediaproxy errors
by Helge Waastad 03 May '07

03 May '07

Hi, I'm running mediaproxy 1.8.2 and have started to se: error: uncaptured python exception, closing channel <rtphandler.CommandHandler connected <ip-address>:42224 at 0x2a99dff518> (socket.error:(9, 'Bad file descriptor') [/usr/lib64/python2.3/asyncore.py|readwrite|88] [/usr/lib64/python2.3/asyncore.py|handle_write_event|397] [/etc/mediaproxy/modules/rtphandler.py|handle_write|453] [/usr/lib64/python2.3/asyncore.py|send|337]) does anyone knw what this means? br hw -- Helge Waastad Senior developer Smartnet / Datametrix Et selskap i Ignis gruppen Martin Lingesvei 25 1367 Snarøya Postboks 494 1327 Lysaker Telefon: +47 67830017 Mobil: +47 48095461 Support: +47 67830045 - support(a)smartnet.no www.smartnet.no

2 2

[Serusers] record_route..........TCP ??
by KUMAR 03 May '07

03 May '07

Hi all, I am using SER-0.9.6 and the problem i'm having is this. When using the following ser.cfg, it works well when the transport UDP is used. But when transport TCP is used, then it only results in one way IM, only from the UA from which the INVITE is being sent. Moreover, when record_route is used instead of record_route_preset, then even that one way IM doesn't work. But it works well with UDP. Can anyone please point me as to where the problem might be?? Thank you in advance. kumar

2 2

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users