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[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

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