- sr-users - kamailio.org

[Serusers] Credentials
by Daniel Mirrha Santos 25 Nov '06

25 Nov '06

Does anybody have an example about how to use credentials to permit some kinds of calls to some users on SER? I tried this but, it stops work with no cause: if (uri=~"^sip:0800.*@.*") { if(!is_user_in("credentials", "0800")) { sl_send_reply("403", "No 0800"); log(1,"No 0800"); break; }; consume_credentials(); prefix ("5511"); route(8); break; }; Thanks, Daniel

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[Serusers] Call Transfers in SER + Asterisk architecture
by Ricardo Carvalho 25 Nov '06

25 Nov '06

Hi, I'm deploying a SER + Asterisk architecture, where SER is used as SIP registrar, and Asterisk is used for voicemail and PSTN gateway. This system is already able to make Call Transfers (Blind and Attended) internally between phones registered in SER, although, I can't make Call Transfers in some scenarios involving PSTN numbers (which need to pass through Asterisk). The problem is that when the REFER message (that carries the Refer-To number to whom the call should be transferred) gets to Asterisk, it replies with a 404 Not Found message, and the Call Transfer isn't established! Any ideas on how can I solve this problem? Thanks in advance, Ricardo.

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[Serusers] OptiPoint420 behind a NAT device
by hiSIPatH 25 Nov '06

25 Nov '06

Hi all, i have a ser setup (CentOS 4.4/ser 0.9.6) with several ip phones (SNOM320 and Xlite30) and every thing seems to work fine. The ser server has a public ip address 89.xxx.xxx.xxx and the ua are behind a NAT device (corporate FW). I want now to add some OptiPoint420 SIP but was unable to get them working. That means that the registration is ok and calls to the Optipoints from other ua (xlite or Snom) work but i was unable to place a call from Optipoint to other ua. In the trace you can notice that the INVITE is sent to ser with Src Port 38625 and Dst Port 5060. But the response from ser is sent to Src Port: 5060 (5060), Dst Port: 5060 (5060). The result is of couse ICMP Destination unreachable (Port unreachable). I have rtp proxy and ser running on the same server and my ser.cfg looks like the one from the SER GettingStarted http://siprouter.onsip.org/doc/gettingstarted/ch08s02.html). I read the doc about "handling of NAT using RTP Proxy" but was unable to change the config to get this scenario with OptiPoint working. Has anyone managed to get OptiPoint to work with ser? thx in advance? No. Time Source Destination Protocol Info 14 8.907171 89.xxx.xxx.xxx 89.xxx.xxx.xxx SIP/SDP Request: INVITE sip:8001@registrar.mydomaine.com;transport=udp, with session description Frame 14 (972 bytes on wire, 972 bytes captured) Ethernet II, Src: AminoCom_02:02:02 (00:02:02:02:02:02), Dst: AcerTech_9c:00:8d (00:00:e2:9c:00:8d) Internet Protocol, Src: 89.xxx.xxx.xxx (89.xxx.xxx.xxx), Dst: 89.xxx.xxx.xxx (89.xxx.xxx.xxx) User Datagram Protocol, Src Port: 38625 (38625), Dst Port: 5060 (5060) Session Initiation Protocol No. Time Source Destination Protocol Info 15 8.907394 89.xxx.xxx.xxx 89.xxx.xxx.xxx SIP Status: 407 Proxy Authentication Required Frame 15 (772 bytes on wire, 772 bytes captured) Ethernet II, Src: AcerTech_9c:00:8d (00:00:e2:9c:00:8d), Dst: AminoCom_02:02:02 (00:02:02:02:02:02) Internet Protocol, Src: 89.xxx.xxx.xxx (89.xxx.xxx.xxx), Dst: 89.xxx.xxx.xxx (89.xxx.xxx.xxx) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol

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[Serusers] How to enable the verbose log?
by Bin Chen 25 Nov '06

25 Nov '06

Hi, I am using ser-0.9.6, my client has some problems with it so I need to look some log of ser to find out why my request is rejected. I looked the source of ser and seems it use syslog, but I don’t know how to enable it, through config file or specify at compile time? Thanks a lot. Binary Chen

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[Users] VOIP Consultants wanted to build a Scalable ITSP Architecture Using OpenSource Softwares
by Shanti kishore Balusu 25 Nov '06

25 Nov '06

Hello Guys We are looking for VOIP Cosultants who can successfully build A Scalable ITSP Architecture Using OpenSource Softwares something like http://www.skyyconsulting.com/itsp_voip_asterisk.php. we are looking for some body who can design & build a scallable highly redundant sollution with billing which can handle 2000-5000 calls per second. we use ILBC for SIP-SIP & GSM for PSTN. Most of our PSTN hardware is Quintums & Cisco Hardware. Servers in use Dell 1850,1950,2950. we need u to give us the complete source code of what u r deploying on our servers(Very Important). UA can be a Softphone or a IP Phone or a PDA with VOIP(wifi). If u think u can a provide a good sollution for our requirement at a low cost please contact me offlist with ur sollution & pricing.. Thanx & Regards Kishore Chowdary

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[Serusers] VOIP Consultants wanted to build a Scalable ITSP Architecture Using OpenSource Softwares
by Shanti kishore Balusu 25 Nov '06

25 Nov '06

Hello Guys We are looking for VOIP Cosultants who can successfully build A Scalable ITSP Architecture Using OpenSource Softwares something like http://www.skyyconsulting.com/itsp_voip_asterisk.php. we are looking for some body who can design & build a scallable highly redundant sollution with billing which can handle 2000-5000 calls per second. we use ILBC for SIP-SIP & GSM for PSTN. Most of our PSTN hardware is Quintums & Cisco Hardware. Servers in use Dell 1850,1950,2950. we need u to give us the complete source code of what u r deploying on our servers(Very Important). UA can be a Softphone or a IP Phone or a PDA with VOIP(wifi). If u think u can a provide a good sollution for our requirement at a low cost please contact me offlist with ur sollution & pricing.. Thanx & Regards Kishore Chowdary

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[Serusers] Call Transfers in SER + Asterisk
by Arun Kumar 25 Nov '06

25 Nov '06

HI, I want to transfer calls from SER to Asterisk but I'm not able to do it. So, please guide me. Here is my ser.cfg and other config files please guide me. my trusted table contiains a record of my asterisk ip : asterisk ip any * ser.cfg ------------------ debug=5 fork=no log_stderror=yes listen=2xx.xxx.xxx.xxx # INSERT YOUR IP ADDRESS HERE port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:mypass@localhost/ser" loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" loadmodule "/usr/lib/ser/modules/uri.so" loadmodule "/usr/lib/ser/modules/uri_db.so" loadmodule "/usr/lib/ser/modules/domain.so" loadmodule "/usr/lib/ser/modules/mediaproxy.so" loadmodule "/usr/lib/ser/modules/nathelper.so" loadmodule "/usr/lib/ser/modules/textops.so" loadmodule "/usr/lib/ser/modules/avpops.so" loadmodule "/usr/lib/ser/modules/permissions.so" modparam("auth_db|permissions|uri_db|usrloc|domain", "db_url", " mysql://ser:mypass@localhost/ser") modparam("auth_db", "calculate_ha1", 1) 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","sip_asymmetrics","/etc/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/etc/ser/rtp-clients") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) 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") # ------------------------- request routing logic ------------------- # main routing logic route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { 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=="INVITE" && client_nat_test("3")) { # INSERT PROXY IP ADDRESS HERE record_route_preset(" 2xx.xxx.xxx.xxx:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if ((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403", "Use From=ID"); break; }; if (method=="INVITE") { if (!allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "user From=ID"); break; }; consume_credentials(); }; if (client_nat_test("3") || search("^Route:.*;nat=yes")){ setflag(6); use_media_proxy(); }; }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (!is_uri_host_local()) { if (is_from_local() || allow_trusted()) { route(4); route(1); } else { sl_send_reply("403", "Forbidden-two"); }; break; }; if (method=="ACK") { route(1); break; } if (method=="CANCEL") { route(1); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (!is_uri_host_local()) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); 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 (!is_uri_host_local()) { route(4); route(1); break; }; if (uri=~"^sip:011[0-9]*@") { route(4); route(5); break; }; if (!lookup("location")) { if (uri=~"^sip:[0-9]{10}@") { route(4); route(5); break; }; 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(); }; }; } route[5]{ rewritehost("Asterisk Server IP"); avp_write("i:45", "inv_timeout"); route(1); } 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(); }; } ---------------------------------- sip.conf ---------------- [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) context = default maxexpirey=180 defaultexpirey=160 autocreatepeer=yes register => 1000:test@2xx.xxx.xxx.xxx/5678 [ser] type=friend username=1000 secret=test fromdomain=2xx.xxx.xxx.xxx host=2xx.xxx.xxx.xxx [5678] type=friend username=5678 host=dynamic canreinvite=no --------------------------- exte.conf exten => 5678,1,Dial(SIP/5678,20) exten => 5678,2,Voicemail2(u5678) exten => 5678,3,MusicOnHold() exten => 5678,102,Voicemail2(b5678) exten => 5678,103,Hangup Nothing is happeing in my case that is I dont see any packet coming from ser to asterisk for asterisk to ser. please guide me. thanks for your reply. arun

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Re: [Serusers] Problem with MTU size in SER with MIPv6.
by Andrei Pelinescu-Onciul 24 Nov '06

24 Nov '06

On Nov 24, 2006 at 11:45, Jose Angel Calvo <jacc15dkz(a)yahoo.es> wrote: > Thanks for your fast answer, Andrei. I answer to your questions: > > > Could you give more details? > Of course :-) > > - what version of ser are you running (ser -V)? > The exit in my screen are: > version: ser 0.9.6 (i386/linux) > flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT > ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 > @(#) $Id: main.c,v 1.197.2.1 2005/07/25 16:56:24 andrei Exp $ > main.c compiled on 11:38:44 Feb 1 2006 with gcc 3.3.6 > > > - is it vanilla ser or a modified version? > It's a vanilla version. > > - the packet is sent over TCP? UDP? > Over UDP. > > - a coredump backtrace? (for this you'll have to start ser with > > -w directory_where_core_can_be_written, e.g. ser -f ser.cfg -w /tmp/) > > After the crash, run gdb path_to_ser/ser /tmp/core, type bt > > and send me the output. Note: the core file might have another name > > (e.g. core.3456). Just look for core.* . > I do what you say me, but thre is not a coredump backtrace. Next, i show you the syslog: Could you try again after runing: ulimit -c unlimited # you need bash for this ser -f ser.cfg -w /tmp/ (or if unlimited doesn't work, some large value, e.g. ulimit -c 100000) ? If it still doesn't work, could you send me a network dump with the INVITE causing the crash and your config, so that I can try to reproduce it? Thanks, Andrei

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[Serusers] Sip-Conf-Id Label
by Gustavo Passos Tourinho 24 Nov '06

24 Nov '06

Hey There! I was wondering how ser calculates Sip-Conf-Id that apears in Radius Details file? Thanks in advanced, Tourinho

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[Users] How can I obtain the first two digits of the header From?
by Jose Gil Navarrete 24 Nov '06

24 Nov '06

Hi. I need to obtain the first two o three digits of the header From and add them to uri? It's, because the telephones make the register with ten digits (area-code + telephone number), then in the call, the telephone sends the invites with 7 or 8 digits, it's because they don't use area code. thanks in advance. --------------------------------- Do You Yahoo!? La mejor conexión a Internet y 2GB extra a tu correo por $100 al mes. http://net.yahoo.com.mx

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[Serusers] Variabilized a data into ser.cfg
by inge 24 Nov '06

24 Nov '06

Hi all, I want to change the host of my PSTN gateway according to certain conditions. But in the aim of not rewrite all routes wich don't change (just PSNT host sometimes) I want to variabilize this data. Is it possible to do something like : if {} => gw=w.x.y.z else => gw=z.y.x.w Next in the routing plan : ... rewritehost(gw); ... ? Thanks for your help.

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[Users] Billing system with good integration with OpenSER / Documentations
by Nick De Cristofaro 24 Nov '06

24 Nov '06

Hey everyone, Good sessions yesterday at the OpenSER summit. I found the presentations very insightful and I learned a lot of new things. I am hoping to get some feedback as a good solution for a billing system that we will need to implement over here in our company that can be easily interacted through OpenSER. As I am pretty new to this I am wondering what component pieces should be setup to do this right. OpenSER as our main SIP router with database access and creating CDRs or do we need a B2BUA with this? Can anyone recommend some good starting point to look into or some publishings that would be a good read to get more insight on building this right. Some of the presentations had some good insights on scaling, redundencies and so forth and we would like to do this right from the beginning and look at our cost platforms. Anyone ever used the Hiper biling system? I saw their exhibit yesterday and it was pretty nice. Thanks Nick

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[Users] Disconnect SIP Call.
by Ricardo Martinez 24 Nov '06

24 Nov '06

Hello List. Is possible, using OPENSER, disconnect a call after a timeout is complete. For example i want to limit the call duration to one hour. Is possible to disconnect the call after this period using some module from OpenSER ?.. maybe the dialog module? Thanks! Regards, Ricardo Martinez.-

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Re: [Users] Loop?
by Daniel-Constantin Mierla 24 Nov '06

24 Nov '06

Hello, On 11/22/06 17:57, Kelvin Williams wrote: > > Yesterday, something happened—the exact I do not know just yet. > However, when looking at the logging that I have my OpenSER performing > I seen a ton of INVITES, CANCELs, OpenSER became unusable for a time > until I was able to block all SIP traffic at the firewall and let it > calm down enough to restart it. > > During this process calls could not be made (in or out). Is there a > way to prevent this type of behavior? It looked as though there were > only a handful of calls that caused a continuous loop. > that means that something is wrong with the configuration file or the sip request. The message was not properly handled and looped. If there is no need for loops in your config, you can have a check at the beginning of the script denying requests from local IP. If you need loops, then try to lower maxfwd value so it will loop less. However, the best is to detect which was the problem and fix it, and have the above fixes to prevent similar cases in future. If you faced a flood attack, the best is to look at pike module. Cheers, Daniel > What logic is recommended to test for a call that has been handled too > many times by the proxy, and to immediately drop it? > > Thanks in advance. > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users >

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[Users] help in Call parking......
by raviprakash sunkara 24 Nov '06

24 Nov '06

Hello Users I'm Doing working on Both OpenSER and Asterisk ....... 9001 and 9003 are registered in OpenSER in extension.conf [from-sip] exten=>115,1,Park() exten =>115,2.Hungup() in Feature.conf ( default park no 701) in sip.conf [9001] ... .. [9002] [9003] When 9003 dial the 115 ( Parking itself) , Asterisk Server says " U parked on 701 extension " After When 9001 dial 701 ..... it Say " 483 too many parameters ... in X-lite " , Actual it has to ring 9003, -- Executing Park("SIP/9003-085d9e10", "") in new stack == Parked SIP/9003-085d9e10 on 701. Will timeout back to extension [from-sip] s, 1 in 45 seconds -- Added extension '701' priority 1 to parkedcalls -- Playing 'digits/7' (language 'en') -- Playing 'digits/0' (language 'en') -- Playing 'digits/1' (language 'en') -- Started music on hold, class 'default', on channel 'SIP/9003-085d9e10' == Spawn extension (from-sip, s, 1) exited KEEPALIVE on 'SIP/9003-085d9e10' Nov 22 18:18:45 NOTICE[3289]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 192.168.2.5 -- Stopped music on hold on SIP/9003-085d9e10 -- Registered extension context 'park-dial' -- Added extension 'SIP/9003' priority 1 to park-dial == Timeout for SIP/9003-085d9e10 parked on 701. Returning to park-dial,SIP/9003,1 -- Executing Dial("SIP/9003-085d9e10", "SIP/9003||t") in new stack -- Called 9003 -- Got SIP response 482 "Loop Detected" back from 192.168.2.76 -- Now forwarding SIP/9003-085d9e10 to 'Local/9003@from-sip' (thanks to SIP/9003-085df878) Nov 22 18:19:20 NOTICE[3949]: chan_local.c:498 local_alloc: No such extension/context 9003@from-sip creating local channel Nov 22 18:19:20 NOTICE[3949]: app_dial.c:474 wait_for_answer: Unable to create local channel for call forward to 'Local/9003@from-sip' (cause = 0) == Everyone is busy/congested at this time (1:0/0/1) Nov 22 18:19:30 WARNING[3949]: pbx.c:2415 __ast_pbx_run: Timeout, but no rule 't' in context 'park-dial' -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 ravi.sunkara(a)hyperion-tech.com www.hyperion-tech.com

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[Serusers] aaa
by Phan Van Duc 24 Nov '06

24 Nov '06

aaa

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[Users] resent packets in siptrace
by Mark Price 23 Nov '06

23 Nov '06

Hi, I am logging text/html messages with siptrace, and seeing that certain packets are being retransmitted several times, but am left with no way to detect that they are retransmissions and not legitimate messages. I.e. the contents of the msg field are the same, and the other fields in the table seem to be of no help. Is there a way to detect within openser that a packet is either a retransmission or being sent to multiple destinations, so that only the first one is recorded? The larger problem I am trying to solve is the above one. However, the specific one is interesting, too. The specific circumstances under which this is happening is that I'm trying to communicate between gaim 2.0 beta5 and a counterpath xten phone. 1. the gaim client will only login using tcp (it's udp transport seems to be broken.. it won't send the Authorization header). 2. messages sent from gaim arrive at the destination, but gaim does not see incoming messages... openser complains about tcp errors. 7(32169) ERROR: tcp_blocking_connect: timeout 10 s elapsed from 10 s 7(32169) ERROR: tcpconn_connect: tcp_blocking_connect failed 7(32169) ERROR: tcp_send: connect failed 7(32169) msg_send: ERROR: tcp_send failed 7(32169) ERROR:tm:t_forward_nonack: sending request failed In this context, the MESSAGE packets get sent several times, and get logged several times. Thanks, Mark Price

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[Serusers] Problem with MTU size in SER with MIPv6.
by Jose Angel Calvo 23 Nov '06

23 Nov '06

Thank you for your help and sorry for my english. I have a problem with mipv6 and a server SIP, SER (SIP Express Router). The mipv6 path is "mipv6-2.0.2" in kernel 2.6.16.24-Ubuntu. The problem occurs when a packet is too big (it must be fragmented), and go to the node what makes function of P-CSCF (Proxy in SIP) and Correspondent Node. If the P-CSCF is not a CN, all is OK. But if the node is CN, then i have the problem. The SER logs are: ................ Nov 20 18:28:25 pcscfD /usr/sbin/ser[8051]: SER: new INVITE Nov 20 18:28:25 pcscfD /usr/sbin/ser[8051]: parse_headers: flags=-1 Nov 20 18:28:25 pcscfD /usr/sbin/ser[8051]: check_via_address(2001:720:1500:1B:FCFD:FF:FE0C:103, [2001:720:1500:1B:FCFD:FF:FE0C:103], 3) Nov 20 18:28:25 pcscfD /usr/sbin/ser[8051]: ERROR: warning_builder: buffer size exceeded Nov 20 18:28:25 pcscfD /usr/sbin/ser[8051]: WARNING: warning skipped -- too big Nov 20 18:28:25 pcscfD /usr/sbin/ser[8051]: WARNING:vqm_resize: resize(0) called Nov 20 18:28:25 pcscfD /usr/sbin/ser[8051]: DEBUG: reply sent out. buf=0x8106270: SIP/2.0 1..., shmem=0xb5e8b330: SIP/2.0 1 Nov 20 18:28:25 pcscfD /usr/sbin/ser[8051]: DEBUG: _reply_light: finished Nov 20 18:28:25 pcscfD /usr/sbin/ser[8051]: DEBUG: mk_proxy: doing DNS lookup... Nov 20 18:28:25 pcscfD /usr/sbin/ser[8051]: check_via_address(2001:720:1500:1B:FCFD:FF:FE0C:103, [2001:720:1500:1B:FCFD:FF:FE0C:103], 3) Nov 20 18:28:25 pcscfD /usr/sbin/ser[8068]: DBG: tcp_main_loop: dead child 2 (shutting down?) Nov 20 18:28:25 pcscfD /usr/sbin/ser[8049]: child process 8051 exited by a signal 11 Nov 20 18:28:25 pcscfD kernel: [4296472.851000] <0>skb_under_panic: text:c02d2aad len:321 put:14 head:d62cfc00 data:d62cfbea tail:d62cfd2b end:d62cfd80 dev:eth0 Nov 20 18:28:25 pcscfD /usr/sbin/ser[8049]: core was not generated Nov 20 18:28:25 pcscfD kernel: [4297162.639000] ------------[ cut here ]------------ Nov 20 18:28:25 pcscfD /usr/sbin/ser[8049]: INFO: terminating due to SIGCHLD Nov 20 18:28:25 pcscfD kernel: [4297162.639000] kernel BUG at net/core/skbuff.c:112! Nov 20 18:28:25 pcscfD /usr/sbin/ser[8059]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8052]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8053]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8054]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8055]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8056]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8057]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8058]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8060]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8061]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8062]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8063]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8064]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8065]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8066]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8067]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8068]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8050]: INFO: signal 15 received Nov 20 18:28:25 pcscfD kernel: [4297162.639000] invalid opcode: 0000 [#3] Nov 20 18:28:25 pcscfD /usr/sbin/ser[8059]: Memory status (pkg): Nov 20 18:28:25 pcscfD /usr/sbin/ser[8052]: Memory status (pkg): Nov 20 18:28:25 pcscfD /usr/sbin/ser[8053]: Memory status (pkg): Nov 20 18:28:25 pcscfD /usr/sbin/ser[8054]: Memory status (pkg): Nov 20 18:28:25 pcscfD /usr/sbin/ser[8055]: Memory status (pkg): Nov 20 18:28:25 pcscfD /usr/sbin/ser[8056]: Memory status (pkg): ....................... Nov 20 18:28:25 pcscfD /usr/sbin/ser[8049]: hash = 2055 fragments no.: 1, unused: 0 ^I^I bucket size: 1048576 - 2097152 (first 1572864) Nov 20 18:28:25 pcscfD /usr/sbin/ser[8049]: hash = 2059 fragments no.: 1, unused: 0 ^I^I bucket size: 16777216 - 33554432 (first 31934160) Nov 20 18:28:25 pcscfD /usr/sbin/ser[8049]: TOTAL: 26 free fragments = 33537656 free bytes Nov 20 18:28:25 pcscfD /usr/sbin/ser[8049]: ----------------------------- Nov 20 18:28:25 pcscfD /usr/sbin/ser[8049]: shm_mem_destroy Nov 20 18:28:25 pcscfD /usr/sbin/ser[8049]: destroying the shared memory lock Nov 20 18:28:25 pcscfD /usr/sbin/ser[8049]: terminating due to SIGCHLD ############################################################## The problem is in the MTU size, because if the packet is not fragmented, there is not a problem. But the INVITE packet needs to fragment, what it means, i always have the same problem. My questions are: - Have anybody the same o similar problem? - The MTU size is a problem in SER? - Have anybody sent fragmented packets with SER? Thanks for your help. ______________________________________________ LLama Gratis a cualquier PC del Mundo. Llamadas a fijos y móviles desde 1 céntimo por minuto. http://es.voice.yahoo.com

2 1

Re: [Users] $ct and $hdr(Content-Type) failing
by Daniel-Constantin Mierla 23 Nov '06

23 Nov '06

Hello, On 11/22/06 18:33, Mark Price wrote: > Daniel, > > I'm using openser 1.1.0. > Using $cT works, as well as $hdr(Content-Type). > I must have had a typo in my $hdr() command yesterday. > Thanks for the help. to make it more interactive and allow users to contribute/enhance to these documents, I moved to dokuwiki latest version so it can be edited by anybody. From time to time we may generate a static version to have it as reference in case someone tries to "poison" the content. http://www.openser.org/dokuwiki/doku.php/pseudovariables:1.1.x http://www.openser.org/dokuwiki/doku.php/pseudovariables:devel Cheers, Daniel > > Mark Price > > > On 11/22/06, *Daniel-Constantin Mierla* <daniel(a)voice-system.ro > <mailto:daniel@voice-system.ro>> wrote: > > Hello, > > I tried and seems to work. There was a typo in pseudo-variables > documentation (fixed now), the short name is $cT for Content-Type > -- $ct > stands for Contact header body. > > xlog("L_ALERT","**** $hdr(Content-Type) | $cT | $rb\n"); > > **** text/plain;charset=UTF-8 | text/plain;charset=UTF-8 | test > > I used devel version, what version are you using ? > > Cheers, > Daniel > > On 11/22/06 01:36, Mark Price wrote: > > Hi, > > > > I am trying to log MESSAGE packets differentiated the > content-type of > > the packet. > > I can see in the packet dump that they have types such as > "text/html" > > and so on. > > I'm using the following statement in the code: > > if(method=="MESSAGE") { > > xlog("L_ALERT","$ct $rb\n"); > > } > > The message body specified by $rb is printing out fine, but $ct is > > printing out "<null>". > > I get the exact same result if I say: > > xlog("L_ALERT","$hdr(Content-Type) $rb\n"); > > > > Is this a bug in openser? > > > > Thanks, > > Mark Price > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org <mailto:Users@openser.org> > > http://openser.org/cgi-bin/mailman/listinfo/users > > > >

1 0

[Serusers] Problem with MTU size in SER with MIPv6.
by jose angel calvo 23 Nov '06

23 Nov '06

1 0

[Serusers] NAT and ACK
by Benito Caracuel 23 Nov '06

23 Nov '06

Hello all , This is my scenario: I have SER with Private address behind a Cisco router which Has NAT implemented. I have two UAs with public address each one. UA1 ----------------------> Router Cisco with NAT (public IP) ------------------------> SER with private IP UA2 <----------------------Router Cisco with NAT (public IP) <------------------------ SER with private IP The problem is that when UA1 sends an INVITE with SDP, UA2 reply with 200 ok with SDP but after that, SER doesn't see the ACK. I know that UA1 sends the ACK but this ACK doesn't reach SER. I think that maybe the problem is NAT. Is possible to run SER with private address? How can i solve this? Help me please. Thank you very much. Ben. _________________________________________________________________ Consigue el nuevo Windows Live Messenger http://get.live.com/messenger/overview

2 1

Re: [Serusers] Re: where is the vm.so for voice mail feature ?:
by lawrence k.y. lin 23 Nov '06

23 Nov '06

Hello list, I searched for the voice mail feature and found an example "Example 5-3. Example ser Config File" at the Admin's Guide in the following url: http://old.iptel.org/ser/doc/seruser/seruser.html#AEN1364 Inside that example, there is a statement to load vm.so as follows: loadmodule "modules/vm/vm.so" However, I cannot find vm.so under the ser-0.9.6. Please advise where I can find vm.so for voice mail feature. Thanks and Regards, Larry _________________________________________________________________ View Athletes Collections with Live Search http://sportmaps.live.com/index.html?source=hmemailtaglinenov06&FORM=MGAC01

3 5

[Users] polycom phone proxy failover question
by Zahid Mehmood 23 Nov '06

23 Nov '06

Hi, I am having difficulty configuring polycom phone to failover when one proxy server is not available. I've configured the phone to use sip naptr records that provides two proxy server names. I've configured the "retry timeout" and "retry max count" however, when proxy1 is not available, polycom phone does not send requests to proxy2. Is there any other option I need to configure? I've tried this on polycom 430 and 601 models. Thanks is advance for your help. -- Zahid Mehmood CUIT Network Systems

2 1

[Users] draft-anil-sipping-bla-03.txt (bridged line appearance) support
by Alan Crosswell 22 Nov '06

22 Nov '06

Has anybody implemented BLA support using OpenSER and/or other code? It appears a State Agent is needed to SUBSCRIBE to the UAs and do all kinds of fun stuff. We use Polycom phones which support all this stuff and are migrating cranky customers from a highly functional legacy PBX that heavily use all these "business" features. /a

1 0

[Users] incorrect documentation
by Mark Price 22 Nov '06

22 Nov '06

Hi, I see that the siptrace documentation for openser 1.1.0 says that the function siptrace() will log a packet. However, the correct function name is sip_trace(). Is there a way to go about submitting improvements/fixes to the documentation? Mark Price

2 1

[Users] Loop?
by Kelvin Williams 22 Nov '06

22 Nov '06

Yesterday, something happened-the exact I do not know just yet. However, when looking at the logging that I have my OpenSER performing I seen a ton of INVITES, CANCELs, OpenSER became unusable for a time until I was able to block all SIP traffic at the firewall and let it calm down enough to restart it. During this process calls could not be made (in or out). Is there a way to prevent this type of behavior? It looked as though there were only a handful of calls that caused a continuous loop. What logic is recommended to test for a call that has been handled too many times by the proxy, and to immediately drop it? Thanks in advance.

1 0

[Serusers] Open IMS Core released last Thursday
by Dragos Vingarzan 22 Nov '06

22 Nov '06

Hi *, as you did not had enough of me last week, here I go again: FOKUS recently launched the Open IMS Core. Keeping it short: - project at http://www.openimscore.org (SVN, mail-lists, etc on berlios) - general information at http://www.fokus.fraunhofer.de/ims/OSIMS.php?lang=en or mailto:info@open-ims.org - you get P/I/S-CSCF, HSS, SIP2IMS Gateway, CDiameterPeer (as module for SER, also standalone usable), JavaDiameterPeer (100% Java and decent performance). All in development, so if you look for stability, security or compatibility, you won't find it there yet. -Dragos -- ----------------------------------------- Dipl. Eng. Dragos Vingarzan FOKUS/NGNI Kaiserin-Augusta-Allee 31 10589 Berlin,Germany Phone +49 (0)30 - 3463 - 7385 Mobile +49 (0)163 - 159 - 5221 eMail vingarzan(a)fokus.fraunhofer.de Web www.fokus.fraunhofer.de We could change the world if God would give us the source code... -----------------------------------------------------------------

3 3

[Users] merging string and pseudo-variable in avp_write
by Benko 22 Nov '06

22 Nov '06

Hello! I'd like to implement an avpops serial forking solution as described in http://www.voice-sistem.ro/docs/avpops/ar01s08.html#ex_serial_forking However, i would like to define the adresses as "$rU(a)mydomain.org". What is the correct syntax to merge the string and the variable in avp_write? I would like to do something like this: avp_write("sip:$rU@mydomain1.org", "$serial_fork"); avp_write("sip:$rU@mydomain2.org", "$serial_fork"); where $rU should be parsed. $serial_fork would have a value of e.g. "12632421(a)mydomain1.org" then. Or does this not work at all? How else could i define it? Can someone point me to a fundamental doc where the openser.cfg-syntax is described? My well-tried trial-and-error method doesn't seem to work here... thx Christian

2 3

[Fwd: Re: [Users] Why is fix_nated_sdp appending 127.0.0.1 at the end of my c= lines?]
by nick 22 Nov '06

22 Nov '06

-------- Original Message -------- Subject: Re: [Users] Why is fix_nated_sdp appending 127.0.0.1 at the end of my c= lines? Date: Tue, 21 Nov 2006 16:46:42 +0100 From: nick <nick(a)mobilia.it> To: Ovidiu Sas <sip.nslu(a)gmail.com> References: <4562E2D7.7010403(a)mobilia.it> <4562E82E.4070503(a)pernau.at> <45631216.3000800(a)mobilia.it> <6f497e130611210650i409e6cfblfb39b0d8a0b9f2e2(a)mail.gmail.com> <45631873.8010400(a)mobilia.it> <6f497e130611210724p4b3ead9bha95cb848c29d276c(a)mail.gmail.com> Ovidiu Sas wrote: > It seems that you are trying to do a double processing of the SDP > (that's why you have 2 IPs inside SDP). Check you script if you use > force_rtp_proxy and fix_nated_sdp at the same time. > > > Regards, > Ovidiu Sas > > On 11/21/06, nick <nick(a)mobilia.it> wrote: >> Ovidiu Sas wrote: >> > check the '-d' argument. If you don't specify '-d', ngrep will only >> > listen to one interface. '-d any' will force ngrep to listen to all >> > interfaces, including the loopback. >> > >> > >> > Hope this helps, >> > Ovidiu Sas >> >> BTW, I changed my fix_nated_sdp() command to fix_nated_sdp("10") only, >> which is adding the origin IP instead of the mediaproxy one.. I don't >> mind this, but it still seems to be adding the 127.0.0.1 >> >> > Actually, I'm using mediaproxy and fix_nated_sdp at the same time, I am using fix_nated_sdp because for some reason, mediaproxy isn't changing the SDP information in the SIP requests (or at least, it wasn't beforehand). I'll have a look at where it might be overlapping. whoops forgot to cc

1 1

[Users] $ct and $hdr(Content-Type) failing
by Mark Price 22 Nov '06

22 Nov '06

Hi, I am trying to log MESSAGE packets differentiated the content-type of the packet. I can see in the packet dump that they have types such as "text/html" and so on. I'm using the following statement in the code: if(method=="MESSAGE") { xlog("L_ALERT","$ct $rb\n"); } The message body specified by $rb is printing out fine, but $ct is printing out "<null>". I get the exact same result if I say: xlog("L_ALERT","$hdr(Content-Type) $rb\n"); Is this a bug in openser? Thanks, Mark Price

2 1

[Users] new poll: location for next OpenSER event
by Daniel-Constantin Mierla 22 Nov '06

22 Nov '06

*Hello Everybody, due to very good feedback and results we got for the first OpenSER summit we want to collect your opinion about the location of the next one. It will help to make the decision since will reflect community distribution. The options are basically the continents, if you have other suggestions please use team(a)openser.org. The poll is at bottom side of right column at http://www.openser.org Cheers, Daniel *

1 0

[Serusers] re: question - digest authentication and firewall
by Yair Hakak 21 Nov '06

21 Nov '06

hi all- i have the following setup: 1. ser running on 5060 2. nathelper/rtp 3. mysql database w/digest authentication now, i have a client on a private network (NAT) behind a firewall. port 5060 tcp is on the firewall is open as well as 35000 to 37000 UDP (rtpproxy is set to run between these ports). when i plug the client into a regular, non-firewalled home router, it registers properly (register->401->register again->OK) when i put the same client behind the firewall, it 401's twice. i am 100% sure the password is correct. is there something else that needs to be open? I dont really understand the authentication mechanism - does the password come via a different port or something like that?

2 2

[Users] Why is fix_nated_sdp appending 127.0.0.1 at the end of my c= lines?
by nick 21 Nov '06

21 Nov '06

Everything seems fine, the SIP messages go tot he right place, the NAT tests are finding everything as they should.. and in the o= row, they're being replaced (more as a test, I don't think it's completely neccessary) without an issue. However, in the c= line, it adds the IP I've specified (for some reason, if I leave it blank it only puts 127.0.0.1; or a mashup of the public address and localhost (as mediaproxy runs on the same machine for the moment). here are the two sections of code I've added.. these are the only places I call fix_nated_sdp() if (method=="INVITE") { if (nat_uac_test("8")) { fix_nated_sdp("10","89.xxx.xxx.16"); }; t_on_failure("1"); } else if (method == "BYE" || method == "CANCEL") { end_media_session(); }; onreply_route[1] { if (status=~"(183)|(2[0-9][0-9])") { if (client_nat_test("1")) { fix_contact(); }; if (nat_uac_test("8")) { fix_nated_sdp("10","89.xxx.xxx.16"); }; use_media_proxy(); }; } I'm using openser 1.1.0 notls, on a dual opteron server running CentOS 4.3 here is a trace of the code in action.. (names changed.. because I enjoy search and replace) U 89.xxx.xxx.8:30626 -> 89.xxx.xxx.16:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 89.xxx.xxx.16;branch=z9hG4bK674e.0d2a4f64.0. Via: SIP/2.0/UDP 192.168.20.16:38000;received=87.xxx.xxx.99;branch=z9hG4bK-d87543-b3783334b231975b-1--d87543-;rport=38000. Record-Route: <sip:89.xxx.xxx.16;lr;ftag=5c63711f>. Contact: <sip:nick@89.xxx.xxx.8:30626;rinstance=f0677dc7f7d4d2f0>. To: "Joe Blow"<sip:joe@blow.com>;tag=ee11f527. From: "Tom"<sip:tom@blow.com>;tag=5c63711f. Call-ID: NTFjYmM5Y2JhNWE5MzU2NzQ1NTMxYTg2NGQ3ZDYzNmM.. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. User-Agent: X-Lite release 1006e stamp 34025. Content-Length: 237. . v=0. o=- 1 2 IN IP4 192.168.1.67. s=CounterPath X-Lite 3.0. c=IN IP4 192.168.1.67. t=0 0. m=audio 31556 RTP/AVP 0 8 3 101. a=alt:1 1 : 0W8e8Rrk y9fAinpH 192.168.1.67 31556. a=fmtp:101 0-15. a=rtpmap:101 telephone-event/8000. a=sendrecv. # U 89.xxx.xxx.16:5060 -> 87.xxx.xxx.99:38000 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.20.16:38000;received=87.xxx.xxx.99;branch=z9hG4bK-d87543-b3783334b231975b-1--d87543-;rport=38000. Record-Route: <sip:89.xxx.xxx.16;lr;ftag=5c63711f>. Contact: <sip:nick@89.xxx.xxx.8:30626;rinstance=f0677dc7f7d4d2f0>. To: "Joe Blow"<sip:joe@blow.com>;tag=ee11f527. From: "Tom"<sip:tom@blow.com>;tag=5c63711f. Call-ID: NTFjYmM5Y2JhNWE5MzU2NzQ1NTMxYTg2NGQ3ZDYzNmM.. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. User-Agent: X-Lite release 1006e stamp 34025. Content-Length: 302. . v=0. o=- 1 2 IN IP4 89.xxx.xxx.16. s=CounterPath X-Lite 3.0. c=IN IP4 89.xxx.xxx.16127.0.0.1. t=0 0. m=audio 60070 RTP/AVP 0 8 3 101. a=alt:1 1 : 0W8e8Rrk y9fAinpH 192.168.1.67 31556. a=fmtp:101 0-15. a=rtpmap:101 telephone-event/8000. a=sendrecv. a=oldmediaip:192.168.1.67. a=oldmediaip:192.168.1.67. # U 87.xxx.xxx.99:38000 -> 89.xxx.xxx.16:5060 ACK sip:nick@89.xxx.xxx.8:30626;rinstance=f0677dc7f7d4d2f0 SIP/2.0. Via: SIP/2.0/UDP 192.168.20.16:38000;branch=z9hG4bK-d87543-7e77316e3b41bd7d-1--d87543-;rport. Max-Forwards: 70. Route: <sip:89.xxx.xxx.16;lr;ftag=5c63711f>. Contact: <sip:thomas@87.xxx.xxx.99:38000>. To: "Joe Blow"<sip:joe@blow.com>;tag=ee11f527. From: "Tom"<sip:tom@blow.com>;tag=5c63711f. Call-ID: NTFjYmM5Y2JhNWE5MzU2NzQ1NTMxYTg2NGQ3ZDYzNmM.. CSeq: 1 ACK. User-Agent: X-Lite release 1006e stamp 34025. Content-Length: 0. . # U 89.xxx.xxx.16:5060 -> 89.xxx.xxx.8:30626 ACK sip:nick@89.xxx.xxx.8:30626;rinstance=f0677dc7f7d4d2f0 SIP/2.0. Record-Route: <sip:89.xxx.xxx.16;lr=on;ftag=5c63711f>. Via: SIP/2.0/UDP 89.xxx.xxx.16;branch=z9hG4bK674e.0d2a4f64.2. Via: SIP/2.0/UDP 192.168.20.16:38000;received=87.xxx.xxx.99;branch=z9hG4bK-d87543-7e77316e3b41bd7d-1--d87543-;rport=38000. Max-Forwards: 69. Contact: <sip:thomas@87.xxx.xxx.99:38000>. To: "Joe Blow"<sip:joe@blow.com>;tag=ee11f527. From: "Tom"<sip:tom@blow.com>;tag=5c63711f. Call-ID: NTFjYmM5Y2JhNWE5MzU2NzQ1NTMxYTg2NGQ3ZDYzNmM.. CSeq: 1 ACK. User-Agent: X-Lite release 1006e stamp 34025. Content-Length: 0. P-hint: rr-enforced. . As you can see, the code is appending 127.0.0.1 for no good reason (at least that I can tell) Any ideas (bug fixes, or tricks...)? Nick

3 6

[Users] Using LCR
by Bruno Machado 21 Nov '06

21 Nov '06

Hi all I'm working with LCR and I'm trying the follow situation: I have 5 devices that are using 3 carries. Look the LCR table below: prefix | from_uri | grp_id | priority 5511 % 1 0 5512 % 2 0 5521 % 3 0 55 % 1 0 It works fine. But I want my voip uses only the route 2 wherever I call. I tried to use 'sip:1234%' in from_uri column but didnt work how I want. If I insert a new line: 55 sip:1234% 2 0 and call to '5511XXXXXX', the select route will be '1' and not '2'. It will work if I change to: 5511 sip:1234% 2 0 This way works fine. But I dont want to duplicate all the lines in the table when I want to priority a route to a specific user. The 'preferred' route is that have a longest prefix in the table. The module LCR tells: "Matching gateways are then ordered for forwarding purpose (1) according to longest user part match, (2) according to gateway's priority (3) randomly" There are a way to force Openser to select the line that has the longest 'from_uri'? Thanks a lot. Bruno Machado _______________________________________________________ Novidade no Yahoo! Mail: receba alertas de novas mensagens no seu celular. Registre seu aparelho agora! http://br.mobile.yahoo.com/mailalertas/

1 0

[Users] Using ssldump to decrypt SIP TLS messages
by Christophe Irles 21 Nov '06

21 Nov '06

Hi, I try to use ssldump in order to decrypt SIP TLS messages but without any success for the moment . TLS configuration in my "openser.cfg" is as follow: disable_tls = 0 tls_method = TLSv1 tls_certificate = "/etc/openser/user-cert.pem" tls_private_key = "/etc/openser/user-privkey.pem" I'm using ssldump with these options: ssldump -Adx -k /etc/openser/user-privkey.pem -i eth0 port 5061 But application data (cf. attached file) are always encrypted . What i'm doing wrong ? I tryed to use the certificate used by the client device too with the same result ... Thanks, Christophe

2 1

RE: [Users] innodb tables
by Watkins, Bradley 21 Nov '06

21 Nov '06

> -----Original Message----- > From: users-bounces(a)openser.org > [mailto:users-bounces@openser.org] On Behalf Of Brandon Price > Sent: Monday, November 20, 2006 8:58 AM > To: daniel(a)voice-system.ro > Cc: openser > Subject: Re: [Users] innodb tables > > What about ndbcluster? I would like to use ndbcluster for all > tables except acc which would of coarse be too big too > quickly. I hope I am not the first to try this. > I am using OpenSER successfully (not in production yet, but will be in a month or so) using ndbcluster. I am, however, using MySQL 5.1.11 (it's beta) and disk-based tables. I don't think that will make much of a difference so long as you have enough memory for your tables if you are using 5.0. Regards, - Brad The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it.

3 2

[Serusers] SER ipv6
by ali daher 21 Nov '06

21 Nov '06

Hi does anyone knows a sip client that runs on IPV6 and register using IPV6? and does it work that such a client can register on SER using IPV6? and is it a problem when the client is on windows xp and SER on linux if i want to work with IPV6? thanks Ali _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

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[Users] WINDOWS OpenSER??
by Jordi Ferre 21 Nov '06

21 Nov '06

Hi ! I'm new on this, however i'm trying to make a study about SER/OpenSER... My question is: Is There any possibility to find one OpenSER program running under Windows platform? or it's only for GNU... If yes, please tell me where... Thanks, Jordi _________________________________________________________________ Grandes éxitos, superhéroes, imitaciones, cine y TV... http://es.msn.kiwee.com/ Lo mejor para tu móvil.

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Re: [Users] insertintg 0-uri header in acc of openser 1.2
by Jayesh Nambiar 20 Nov '06

20 Nov '06

Hi Daniel, Thanks for your response. Actually I wanted the outbound_uri and inbound_uri to be accounted as it used to be in the old structure of acc table(o_uri and i_uri) The pseudo variable "$ou" logs the original uri which is same as the to_uri most of the times. I wanted to log the uri format when it goes from my openser to the provider. Is this possible. Thanks in advance, w/regards, Jayesh ----- Original Message ---- From: Daniel-Constantin Mierla <daniel(a)voice-system.ro> To: Jayesh Nambiar <voip_freak(a)yahoo.co.in> Cc: openser <users(a)openser.org> Sent: Monday, 20 November, 2006 3:37:09 PM Subject: Re: [Users] insertintg 0-uri header in acc of openser 1.2 Hello, the original uri is $ou: http://www.openser.org/docs/pseudo-variables-1.1.x.html#pv_ou Cheers, Daniel On 11/20/06 09:48, Jayesh Nambiar wrote: > Hi all, > I was just wondering what pseudo variable should be used to insert the > o_uri information in acc table of openser 1.2, since there is no o_uri > logged in the new acc structure and I need that for my accounting purpose. > Please help, > > w/regards, > Jayesh > > ------------------------------------------------------------------------ > Find out what India is talking about on - Yahoo! Answers India > <http://us.rd.yahoo.com/mail/in/yanswers/*http://in.answers.yahoo.com/> > Send FREE SMS to your friend's mobile from Yahoo! Messenger Version 8. > Get it NOW > <http://us.rd.yahoo.com/mail/in/messengertagline/*http://in.messenger.yahoo.…> > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > __________________________________________________________ Yahoo! India Answers: Share what you know. Learn something new http://in.answers.yahoo.com/

2 1

[Users] innodb tables
by Jayesh Nambiar 20 Nov '06

20 Nov '06

Hi, Can i use the default table type as INNODB for all my mysql tables in openser database. Can someone please tell me the fundamental drawback of using innodb type instead of myisam. My only concern is the limitaion of myisam to 4GB which requires continous backup of acc table data as it gets to its limit very soon. I am asking this since openser also specifies the default table type as myisam, so before changing the table type I just needed some expert advice on any performance related issues. thanks in advance, w/regards, Jayesh __________________________________________________________ Yahoo! India Answers: Share what you know. Learn something new http://in.answers.yahoo.com/

3 2

[Users] xlog performance
by Christian Schlatter 20 Nov '06

20 Nov '06

I recently did some performance testing with openser 1.1 and stumbled over a rather strange thing. The xlog command from the xlog module seems to cause a huge performance penalty. I have a rather complex config with many routing blocks and mysql queries with which I can achieve up to 100 calls per second (authenticated INVITEs) per openser instance. As soon as I add *one* xlog statement to the config the cps rate goes down to 25, and the average SIP message response time goes up from <10ms to >200ms. Has anyone else observed this behavior or could this issue be caused by a misconfiguration? thanks, Christian

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[Users] insertintg 0-uri header in acc of openser 1.2
by Jayesh Nambiar 20 Nov '06

20 Nov '06

Hi all, I was just wondering what pseudo variable should be used to insert the o_uri information in acc table of openser 1.2, since there is no o_uri logged in the new acc structure and I need that for my accounting purpose. Please help, w/regards, Jayesh __________________________________________________________ Yahoo! India Answers: Share what you know. Learn something new http://in.answers.yahoo.com/

2 1

[Serusers] How to add a file to XCAP server
by VinayG 20 Nov '06

20 Nov '06

hai, We are trying to setup xcap server (http://download.dns-hosting.info/XCAP/) with no presence functionality. we would like to use this server to store, retrieve, modify the contact/group list documents at the server. We have followed the instructions given for installation in the site( http://download.dns-hosting.info/XCAP/). Installation instructions Setup php (Skipped this part as we are not using any authentication and also commented authentication check code in index.php(given below) file of xcap-server) Setup Apache web server(no ssl support) Create a virtual host for Apache server as follows: <VirtualHost 192.168.3.209:80> ServerName 192.168.3.209 DocumentRoot /var/www/xcap-root RewriteEngine On RewriteRule /xcap-root/(.*) /xcap-root/index.php ProxyVia On # SSLEngine On # SSLCertificateFile /path/to/server.crt # SSLCertificateKeyFile /path/to//server.key # SetEnvIf User-Agent ".*MSIE.*" nokeepalive ssl-unclean-shutdown </VirtualHost> Enable php as scripting language and add index.php to the list of index files. LoadModule php4_module /usr/lib/apache/1.3/libphp4.so <IfModule mod_dir.c> DirectoryIndex index.php </IfModule> Make sure the user under which the Apache server runs has the rights to write into the root directory of the virtual host: chmod -R g+w /var/www/xcap-root chown -R apache /var/www/xcap-root Setup SIP User Agent (Skipped this part, as we are facing problems in setup of ser as presence agent) Setup XCAP server Create a file index.php under the xcap-root directory and change the database connection details to the place where SER subscriber table is located and add the IP addresses of the SER PA server in the trusted list: <? include("xcap-lib.php"); $XCAPServer = new XCAPServer(); $XCAPServer->AuthOptions = array( 'dsn' => "mysql://ser:passwd@db/ser", 'table' => "subscriber", 'usernamecol' => "username", 'passwordcol' => "password", 'cryptType' => "none", ); $XCAPServer->trustedPeers = array('21.23.228.49','21.23.228.46'); #if ($XCAPServer->authenticate() && $XCAPServer->authorize()) { if ($_SERVER['REQUEST_METHOD'] == "GET") { $XCAPServer->getFile(); } else if ($_SERVER['REQUEST_METHOD'] == "PUT") { $XCAPServer->putFile(); } else if ($_SERVER['REQUEST_METHOD'] == "DELETE") { $XCAPServer->deleteFile(); } #} ?> In effect, we are configuring apache server to work as xcap server. How to ADD/RETRIEVE a contact list file to/from the XCAP server? Any help is greatly appreciated. Thanks Vinay G

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[Users] Problems with one sided audio..
by nick 20 Nov '06

20 Nov '06

I have a situation where I make a call to a pstn provider, everything works correctly on the SIP level, INVITEs, OKs, ACKs and BYEs are all passing correctly through my server, but for some reason, I'm unable to get audio to work correctly on both sides. At the moment I'm using Openser 1.1.0 with mediaproxy (and a slightly modified openser.cfg based on the mediaproxy one on the openser site, I have a some options for accounting and forwarding to the pstn gateway).. 89.x.x.16 is my openser server. 89.x.x.8 is my NAT firewall, which is portforwarding all UDP from 5000 to 30000 to 192.168.1.67 (my internal machine, with X-Lite). x.x.x.53 is my PSTN provider's SIP server x.x.x.3 is my PSTN provider's media server. this is the SIP dialog: U 89.x.x.8:26380 -> 89.x.x.16:5060 INVITE sip:00390721111111@pstnprovider.com SIP/2.0. Via: SIP/2.0/UDP 192.168.1.67:26380;branch=z9hG4bK-d87543-cb65401207770316-1--d87543-;rport. Max-Forwards: 70. Contact: <sip:nick@89.x.x.8:26380>. To: "mobilia"<sip:00390721111111@pstnprovider.com>. From: "Nick Warr - Mobilia"<sip:nick@logycs.it>;tag=173a892a. Call-ID: N2YyNGJmMjBhZjYwMjY3OWExMmVmYzYyNDhjMTgzNzY.. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. User-Agent: X-Lite release 1006e stamp 34025. Content-Length: 261. . v=0. o=- 1 2 IN IP4 192.168.1.67. s=CounterPath X-Lite 3.0. c=IN IP4 192.168.1.67. t=0 0. m=audio 13234 RTP/AVP 0 98 3 101. a=alt:1 1 : AtyyaMHs +WwsY5o+ 192.168.1.67 13234. a=fmtp:101 0-15. a=rtpmap:98 iLBC/8000. a=rtpmap:101 telephone-event/8000. a=sendrecv. # U 89.x.x.16:5060 -> 89.x.x.8:26380 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 192.168.1.67:26380;branch=z9hG4bK-d87543-cb65401207770316-1--d87543-;rport=26380;received=89.x.x.8. To: "mobilia"<sip:00390721111111@pstnprovider.com>. From: "Nick Warr - Mobilia"<sip:nick@logycs.it>;tag=173a892a. Call-ID: N2YyNGJmMjBhZjYwMjY3OWExMmVmYzYyNDhjMTgzNzY.. CSeq: 1 INVITE. Server: OpenSer (1.1.0-notls (x86_64/linux)). Content-Length: 0. Warning: 392 89.x.x.16:5060 "Noisy feedback tells: pid=27910 req_src_ip=89.x.x.8 req_src_port=26380 in_uri=sip:00390721111111@pstnprovider.com out_uri=sip:00390721111111@x.x.x.53:5060 via_cnt==1". . # U 89.x.x.16:5060 -> x.x.x.53:5060 INVITE sip:00390721111111@x.x.x.53:5060 SIP/2.0. Record-Route: <sip:89.x.x.16;lr=on;ftag=173a892a>. Via: SIP/2.0/UDP 89.x.x.16;branch=z9hG4bK860f.0d8bc646.0. Via: SIP/2.0/UDP 192.168.1.67:26380;received=89.x.x.8;branch=z9hG4bK-d87543-cb65401207770316-1--d87543-;rport=26380. Max-Forwards: 69. Contact: <sip:nick@89.x.x.8:26380>. To: "mobilia"<sip:00390721111111@pstnprovider.com>. From: "Nick Warr - Mobilia"<sip:nick@logycs.it>;tag=173a892a. Call-ID: N2YyNGJmMjBhZjYwMjY3OWExMmVmYzYyNDhjMTgzNzY.. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. User-Agent: X-Lite release 1006e stamp 34025. Content-Length: 261. P-hint: outbound. . v=0. o=- 1 2 IN IP4 192.168.1.67. s=CounterPath X-Lite 3.0. c=IN IP4 192.168.1.67. t=0 0. m=audio 13234 RTP/AVP 0 98 3 101. a=alt:1 1 : AtyyaMHs +WwsY5o+ 192.168.1.67 13234. a=fmtp:101 0-15. a=rtpmap:98 iLBC/8000. a=rtpmap:101 telephone-event/8000. a=sendrecv. # U x.x.x.53:5060 -> 89.x.x.16:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 89.x.x.16;branch=z9hG4bK860f.0d8bc646.0. Via: SIP/2.0/UDP 192.168.1.67:26380;received=89.x.x.8;branch=z9hG4bK-d87543-cb65401207770316-1--d87543-;rport=26380. From: "Nick Warr - Mobilia" <sip:nick@logycs.it>;tag=173a892a. To: "mobilia" <sip:00390721111111@pstnprovider.com>. Call-ID: N2YyNGJmMjBhZjYwMjY3OWExMmVmYzYyNDhjMTgzNzY.. CSeq: 1 INVITE. Content-Length: 0. . # U 89.x.x.8:26380 -> 89.x.x.16:5060 . . # U x.x.x.53:5060 -> 89.x.x.16:5060 SIP/2.0 180 Ringing. To: "mobilia" <sip:00390721111111@pstnprovider.com>;tag=3372418527-183443. From: "Nick Warr - Mobilia" <sip:nick@logycs.it>;tag=173a892a. Call-ID: N2YyNGJmMjBhZjYwMjY3OWExMmVmYzYyNDhjMTgzNzY.. CSeq: 1 INVITE. Contact: sip:00390721111111@x.x.x.53:5060. Content-Type: application/sdp. Via: SIP/2.0/UDP 89.x.x.16;branch=z9hG4bK860f.0d8bc646.0. Via: SIP/2.0/UDP 192.168.1.67:26380;received=89.x.x.8;branch=z9hG4bK-d87543-cb65401207770316-1--d87543-;rport=26380. Content-Length: 220. . v=0. o=fra006nex 1234 21616 IN IP4 x.x.x.3. s=sip call. c=IN IP4 x.x.x.3. t=0 0. m=audio 5946 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=maxptime:20. # U 89.x.x.16:5060 -> 89.x.x.8:26380 SIP/2.0 180 Ringing. To: "mobilia" <sip:00390721111111@pstnprovider.com>;tag=3372418527-183443. From: "Nick Warr - Mobilia" <sip:nick@logycs.it>;tag=173a892a. Call-ID: N2YyNGJmMjBhZjYwMjY3OWExMmVmYzYyNDhjMTgzNzY.. CSeq: 1 INVITE. Contact: sip:00390721111111@x.x.x.53:5060. Content-Type: application/sdp. Via: SIP/2.0/UDP 192.168.1.67:26380;received=89.x.x.8;branch=z9hG4bK-d87543-cb65401207770316-1--d87543-;rport=26380. Content-Length: 220. . v=0. o=fra006nex 1234 21616 IN IP4 x.x.x.3. s=sip call. c=IN IP4 x.x.x.3. t=0 0. m=audio 5946 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=maxptime:20. # U x.x.x.53:5060 -> 89.x.x.16:5060 SIP/2.0 200 OK. To: "mobilia" <sip:00390721111111@pstnprovider.com>;tag=3372418527-183443. From: "Nick Warr - Mobilia" <sip:nick@logycs.it>;tag=173a892a. Call-ID: N2YyNGJmMjBhZjYwMjY3OWExMmVmYzYyNDhjMTgzNzY.. CSeq: 1 INVITE. Contact: sip:00390721111111@x.x.x.53:5060. Content-Type: application/sdp. Via: SIP/2.0/UDP 89.x.x.16;branch=z9hG4bK860f.0d8bc646.0. Via: SIP/2.0/UDP 192.168.1.67:26380;received=89.x.x.8;branch=z9hG4bK-d87543-cb65401207770316-1--d87543-;rport=26380. Record-Route: <sip:89.x.x.16;lr=on;ftag=173a892a>. Content-Length: 220. . v=0. o=fra006nex 1234 21616 IN IP4 x.x.x.3. s=sip call. c=IN IP4 x.x.x.3. t=0 0. m=audio 5946 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=maxptime:20. # U 89.x.x.16:5060 -> 89.x.x.8:26380 SIP/2.0 200 OK. To: "mobilia" <sip:00390721111111@pstnprovider.com>;tag=3372418527-183443. From: "Nick Warr - Mobilia" <sip:nick@logycs.it>;tag=173a892a. Call-ID: N2YyNGJmMjBhZjYwMjY3OWExMmVmYzYyNDhjMTgzNzY.. CSeq: 1 INVITE. Contact: sip:00390721111111@x.x.x.53:5060. Content-Type: application/sdp. Via: SIP/2.0/UDP 192.168.1.67:26380;received=89.x.x.8;branch=z9hG4bK-d87543-cb65401207770316-1--d87543-;rport=26380. Record-Route: <sip:89.x.x.16;lr=on;ftag=173a892a>. Content-Length: 220. . v=0. o=fra006nex 1234 21616 IN IP4 x.x.x.3. s=sip call. c=IN IP4 x.x.x.3. t=0 0. m=audio 5946 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=maxptime:20. # U 89.x.x.8:26380 -> 89.x.x.16:5060 ACK sip:00390721111111@x.x.x.53:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.67:26380;branch=z9hG4bK-d87543-2f47cb445c504950-1--d87543-;rport. Max-Forwards: 70. Route: <sip:89.x.x.16;lr;ftag=173a892a>. Contact: <sip:nick@89.x.x.8:26380>. To: "mobilia"<sip:00390721111111@pstnprovider.com>;tag=3372418527-183443. From: "Nick Warr - Mobilia"<sip:nick@logycs.it>;tag=173a892a. Call-ID: N2YyNGJmMjBhZjYwMjY3OWExMmVmYzYyNDhjMTgzNzY.. CSeq: 1 ACK. User-Agent: X-Lite release 1006e stamp 34025. Content-Length: 0. . # U 89.x.x.16:5060 -> x.x.x.53:5060 ACK sip:00390721111111@x.x.x.53:5060 SIP/2.0. Record-Route: <sip:89.x.x.16;lr=on;ftag=173a892a>. Via: SIP/2.0/UDP 89.x.x.16;branch=z9hG4bK860f.0d8bc646.2. Via: SIP/2.0/UDP 192.168.1.67:26380;received=89.x.x.8;branch=z9hG4bK-d87543-2f47cb445c504950-1--d87543-;rport=26380. Max-Forwards: 69. Contact: <sip:nick@89.x.x.8:26380>. To: "mobilia"<sip:00390721111111@pstnprovider.com>;tag=3372418527-183443. From: "Nick Warr - Mobilia"<sip:nick@logycs.it>;tag=173a892a. Call-ID: N2YyNGJmMjBhZjYwMjY3OWExMmVmYzYyNDhjMTgzNzY.. CSeq: 1 ACK. User-Agent: X-Lite release 1006e stamp 34025. Content-Length: 0. P-hint: rr-enforced. .... # U 89.x.x.16:5060 -> 89.x.x.8:26380 .... # U 89.x.x.8:26380 -> 89.x.x.16:5060 BYE sip:00390721111111@x.x.x.53:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.67:26380;branch=z9hG4bK-d87543-31344e58af13955c-1--d87543-;rport. Max-Forwards: 70. Route: <sip:89.x.x.16;lr;ftag=173a892a>. Contact: <sip:nick@89.x.x.8:26380>. To: "mobilia"<sip:00390721111111@pstnprovider.com>;tag=3372418527-183443. From: "Nick Warr - Mobilia"<sip:nick@logycs.it>;tag=173a892a. Call-ID: N2YyNGJmMjBhZjYwMjY3OWExMmVmYzYyNDhjMTgzNzY.. CSeq: 2 BYE. User-Agent: X-Lite release 1006e stamp 34025. Reason: SIP;description="User Hung Up". Content-Length: 0. . # U 89.x.x.16:5060 -> x.x.x.53:5060 BYE sip:00390721111111@x.x.x.53:5060 SIP/2.0. Record-Route: <sip:89.x.x.16;lr=on;ftag=173a892a>. Via: SIP/2.0/UDP 89.x.x.16;branch=z9hG4bK560f.bbdde784.0. Via: SIP/2.0/UDP 192.168.1.67:26380;received=89.x.x.8;branch=z9hG4bK-d87543-31344e58af13955c-1--d87543-;rport=26380. Max-Forwards: 69. Contact: <sip:nick@89.x.x.8:26380>. To: "mobilia"<sip:00390721111111@pstnprovider.com>;tag=3372418527-183443. From: "Nick Warr - Mobilia"<sip:nick@logycs.it>;tag=173a892a. Call-ID: N2YyNGJmMjBhZjYwMjY3OWExMmVmYzYyNDhjMTgzNzY.. CSeq: 2 BYE. User-Agent: X-Lite release 1006e stamp 34025. Reason: SIP;description="User Hung Up". Content-Length: 0. P-hint: rr-enforced. . # U 89.x.x.8:26380 -> 89.x.x.16:5060 BYE sip:00390721111111@x.x.x.53:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.67:26380;branch=z9hG4bK-d87543-31344e58af13955c-1--d87543-;rport. Max-Forwards: 70. Route: <sip:89.x.x.16;lr;ftag=173a892a>. Contact: <sip:nick@89.x.x.8:26380>. To: "mobilia"<sip:00390721111111@pstnprovider.com>;tag=3372418527-183443. From: "Nick Warr - Mobilia"<sip:nick@logycs.it>;tag=173a892a. Call-ID: N2YyNGJmMjBhZjYwMjY3OWExMmVmYzYyNDhjMTgzNzY.. CSeq: 2 BYE. User-Agent: X-Lite release 1006e stamp 34025. Reason: SIP;description="User Hung Up". Content-Length: 0. . # U 89.x.x.16:5060 -> x.x.x.53:5060 BYE sip:00390721111111@x.x.x.53:5060 SIP/2.0. Record-Route: <sip:89.x.x.16;lr=on;ftag=173a892a>. Via: SIP/2.0/UDP 89.x.x.16;branch=z9hG4bK560f.bbdde784.0. Via: SIP/2.0/UDP 192.168.1.67:26380;received=89.x.x.8;branch=z9hG4bK-d87543-31344e58af13955c-1--d87543-;rport=26380. Max-Forwards: 69. Contact: <sip:nick@89.x.x.8:26380>. To: "mobilia"<sip:00390721111111@pstnprovider.com>;tag=3372418527-183443. From: "Nick Warr - Mobilia"<sip:nick@logycs.it>;tag=173a892a. Call-ID: N2YyNGJmMjBhZjYwMjY3OWExMmVmYzYyNDhjMTgzNzY.. CSeq: 2 BYE. User-Agent: X-Lite release 1006e stamp 34025. Reason: SIP;description="User Hung Up". Content-Length: 0. P-hint: rr-enforced. . # U x.x.x.53:5060 -> 89.x.x.16:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 89.x.x.16;branch=z9hG4bK560f.bbdde784.0. Via: SIP/2.0/UDP 192.168.1.67:26380;received=89.x.x.8;branch=z9hG4bK-d87543-31344e58af13955c-1--d87543-;rport=26380. From: "Nick Warr - Mobilia" <sip:nick@logycs.it>;tag=173a892a. To: "mobilia" <sip:00390721111111@pstnprovider.com>;tag=3372418527-183443. Call-ID: N2YyNGJmMjBhZjYwMjY3OWExMmVmYzYyNDhjMTgzNzY.. CSeq: 2 BYE. Content-Length: 0. . # U 89.x.x.16:5060 -> 89.x.x.8:26380 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.1.67:26380;received=89.x.x.8;branch=z9hG4bK-d87543-31344e58af13955c-1--d87543-;rport=26380. From: "Nick Warr - Mobilia" <sip:nick@logycs.it>;tag=173a892a. To: "mobilia" <sip:00390721111111@pstnprovider.com>;tag=3372418527-183443. Call-ID: N2YyNGJmMjBhZjYwMjY3OWExMmVmYzYyNDhjMTgzNzY.. CSeq: 2 BYE. Content-Length: 0. . # U 89.x.x.16:5060 -> 89.x.x.8:26380 .... # Can anyone see something irregular in this dialog? Or do I need to start beating up my firewall/NAT ?

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[Users] SER not to mask the FROM field
by Mark Anthony C. Delfin 18 Nov '06

18 Nov '06

Hello Guys, I would like SER not to mask the FROM field when its sending a call to another sip proxy/gateway. What Im getting on the destination gateway is the IP of the SER. I wouldnt want SER to proxy/mask the original IP address of the UAC. Is this possible. Here is a sample callflow: UAC (192.168.10.10) >> SER (192.168.11.10) >> UAS (192.168.12.10) i would like the FROM on the UAS to look like 123456(a)192.168.10.10 and not 123456(a)192.168.11.10 which is the SER's IP. Please help. Thanks in advance.

3 2

Re: [Users] T.38 together with OpenSER+RTPProxy and OpenPBX not working
by Ovidiu Sas 18 Nov '06

18 Nov '06

Hi Bastien, If you see incoming and outgoing udptl traffic on both SIP-FAX devices, capture it and load the file on wireshark. Check the T.38 signaling and see where it is failing and why. As an example/reference, use a capture from a working case. Regards, Ovidiu Sas On 11/17/06, Bastian Schern <ml02(a)in-bln.de> wrote: > Hi Bogdan, > > I think it has nothing to do with a deadlock. I see the incoming and > outgoing UDPTL traffic on both SIP-FAX devices. > > Do you got a suggestion for me, what I could do to locate and resolve the > problem. > > Cheers > Bastian > > --On Donnerstag, 16. November 2006 14:12 +0200 Bogdan-Andrei Iancu > <bogdan(a)voice-system.ro> wrote: > > > Hi Bastin, > > > > probably there is kind of a RTP deadlock between the RTPproxy and > > OpenPBX - both are waiting for the other party to send the media before > > sending on its turn. > > > > try something like: > > if (nat_uac_test("8")) { > > force_rtp_proxy(); > > } else { > > force_rtp_proxy("r"); > > } > > > > refer to http://www.openser.org/docs/modules/1.2.x/nathelper.html for > > more details. > > > > regards, > > bogdan > > > > > > Bastian Schern wrote: > > > >> Hi everybody, > >> > >> I've got a question: Does anybody get T.38 working between OpenPBX and > >> OpenSER + RTPProxy? > >> > >> The following scenarios are working fine: > >> > >> 1: SIP-FAX1 (pub. IP) <-> OpenPBX <-> SIP-FAX2 (pub. IP) > >> 2: SIP-FAX1 (NAT) <-> RTPProxy+OpenSER <-> SIP-FAX2 (pub. IP) > >> 3: SIP-FAX1 (NAT) <-> RTPProxy+OpenSER <-> SIP-FAX2 (NAT) > >> 4: SIP-FAX1 (pub. IP) <-> OpenSER <-> OpenPBX <-> SIP-FAX2 (pub. IP) > >> > >> > >> But the following scenario is not working: > >> > >> SIP-FAX1 (NAT) <-> RTPProxy+OpenSER <-> OpenPBX <-> SIP-FAX2 (pub. IP) > >> > >> > >> > >> It seems to be that the RTPproxy has a problem with the OpenPBX but no > >> Problem with SIP-FAX-ATAs. > >> > >> Does anybody have an idea what's going wrong??? > >> > >> > >> Cheers > >> Bastian > >> > >> _______________________________________________ > >> 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 >

1 0

[Users] T.38 together with OpenSER+RTPProxy and OpenPBX not working
by Bastian Schern 18 Nov '06

18 Nov '06

Hi everybody, I've got a question: Does anybody get T.38 working between OpenPBX and OpenSER + RTPProxy? The following scenarios are working fine: 1: SIP-FAX1 (pub. IP) <-> OpenPBX <-> SIP-FAX2 (pub. IP) 2: SIP-FAX1 (NAT) <-> RTPProxy+OpenSER <-> SIP-FAX2 (pub. IP) 3: SIP-FAX1 (NAT) <-> RTPProxy+OpenSER <-> SIP-FAX2 (NAT) 4: SIP-FAX1 (pub. IP) <-> OpenSER <-> OpenPBX <-> SIP-FAX2 (pub. IP) But the following scenario is not working: SIP-FAX1 (NAT) <-> RTPProxy+OpenSER <-> OpenPBX <-> SIP-FAX2 (pub. IP) It seems to be that the RTPproxy has a problem with the OpenPBX but no Problem with SIP-FAX-ATAs. Does anybody have an idea what's going wrong??? Cheers Bastian

3 5

[Users] t_relay without 100 trying
by Klaus Darilion 17 Nov '06

17 Nov '06

Hi! Is it possible to deactivate the 100 Trying during t_relay? I want to use sl_send_reply("100","trying"), thus the implicit 100 trying during t_relay is not needed. regards klaus -- Klaus Darilion nic.at

5 8

[Serusers] ser ottendorf flags does not parsing strings
by samuel 17 Nov '06

17 Nov '06

I can not use "nat.uac" and "nat.uas" as registrar flags: 0(12872) str2s: ERROR: unexpected char N in NATuac 0(12872) set_mod_param_regex: 'registrar' matches module 'registrar' 0(12872) find_param_export: found <load_nat_flag> in module registrar [/usr/local/lib/ser/modules/registrar.so] 0(12872) set_mod_param_regex: found <load_nat_flag> in module registrar [/usr/local/lib/ser/modules/registrar.so] 0(12872) str2s: ERROR: unexpected char N in NATuas

4 13

[Users] OpenSer and Asterisk
by Marnus van Niekerk 17 Nov '06

17 Nov '06

3 4

RE: [Users] Radius extra: integer attribute type how to
by Pletli Antal 17 Nov '06

17 Nov '06

Bogdan, Thank you.I did it, but doesn't work. I will try to implement my own radius extra solution. Antal -----Original Message----- From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] Sent: Thursday, November 16, 2006 1:19 PM To: Pletli Antal Cc: users(a)openser.org Subject: Re: [Users] Radius extra: integer attribute type how to Hi Antal, If I'm not wrong, all extra accounting information is added as string (it's a limitation because of using several backend), so you may try configuring the attribute in the RADIUS dictionary as string and not integer. regards, bogdan Pletli Antal wrote: > Hello, > > I'm trying the radius-extra feature in openser cvs snapshot. > > In my openser.cfg I set up the following parameter: > > modparam("acc", "radius_extra", > "Acct-Session-Time=$avp(call_length)") > > modparam( "avpops", "avp_aliases", "start_timestamp=i:100") > modparam( "avpops", "avp_aliases", "end_timestamp=i:101") > modparam( "avpops", "avp_aliases", "call_length=i:102") > > (...) > > avp_op("$avp(end_timestamp)","sub/$avp(start_timestamp)"); > > avp_op("$avp(end_timestamp)/$avp(call_length)","sub/$avp(start_timesta > mp)"); > > > > I have experienced that when I sent an Accounting-Stop request to > radius server, it received the integer value as an string and it > decoded the value to the chr code of the "3" > > In the radius log: > > * Attr: 46 - Acct-Delay-Time len: 6 val: 51 > > In the openser.log: > > Nov 14 16:12:09 sip2 ./openser[727]: INFO:avpops:print_avp: > p=0x40623b88, flags=0x0000 > Nov 14 16:12:09 sip2 ./openser[727]: INFO: id=<102> > Nov 14 16:12:09 sip2 ./openser[727]: INFO: > val_int=<3> > > How can I set the corresponding integer attribute in the request? > > Thanks in advance, > > Antal > >----------------------------------------------------------------------- >- > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > >

1 0

[Users] Not Hung up Properly...................
by raviprakash sunkara 17 Nov '06

17 Nov '06

Hello Users.... For me The Hung up ( Bye Request not coming to SIP server ) not properly taking Take............... :P Below I metioned the Ngrepped messages.... Help me ........ 2006/11/14 03:01:04.129657 61.17.248.68:5831 -> 192.168.2.90:5060 INVITE sip:102@24.99.145.14 SIP/2.0..v: SIP/2.0/UDP 192.168.2.20:5060;rport;branch=z9hG4bK3F90EDDDCA6F4BD7A0B044CB585A28B9..f: siri <sip:101@24.99.145.1 4>;tag=3410236415..t: <sip:102@24.99.145.14>..m: <sip:101@192.168.2.20:5060>..i: CB3623BF-158D-4124-A1D6-877D2673BBD4@192.168.2.20..CSeq: 41383 INVITE.. Max-Forwards: 70..c: application/sdp..User-Agent: X-Lite release 1103m..l: 135....v=0..o=101 9577234 9577250 IN IP4 192.168.2.20..s=X-Lite..c=IN IP4 192 .168.2.20..t=0 0..m=audio 12222 RTP/AVP 0..a=rtpmap:0 pcmu/8000.. U 2006/11/14 03:01:04.130618 192.168.2.90:5060 -> 61.17.248.68:5831 SIP/2.0 407 Proxy Authentication Required..v: SIP/2.0/UDP 192.168.2.20:5060 ;rport=5831;branch=z9hG4bK3F90EDDDCA6F4BD7A0B044CB585A28B9;received= 61.17.248 .68..f: siri <sip:101@24.99.145.14>;tag=3410236415..t: < sip:102@24.99.145.14>;tag=952b5d87ce2effa04bc4206000ec51df.2411..i:CB3623BF-158D-4124-A1D6-877D 2673BBD4@192.168.2.20..CSeq: 41383 INVITE..Proxy-Authenticate: Digest realm="24.99.145.14", nonce="455978ece372d35484e2a5f0010e0766a823f3a6"..Server: Op enSer (1.1.0-notls (i386/linux))..Content-Length: 0..Warning: 392 192.168.2.90:5060 "Noisy feedback tells: pid=4701 req_src_ip=61.17.248.68req_src_por t=5831 in_uri=sip:102@24.99.145.14 out_uri=sip:102@24.99.145.14via_cnt==1".... U 2006/11/14 03:01:04.628174 61.17.248.68:5831 -> 192.168.2.90:5060 ACK sip:102@24.99.145.14 SIP/2.0..v: SIP/2.0/UDP 192.168.2.20:5060;rport;branch=z9hG4bK3F90EDDDCA6F4BD7A0B044CB585A28B9..f: siri <sip:101@24.99.145.14>; tag=3410236415..t: <sip:102@24.99.145.14>;tag= 952b5d87ce2effa04bc4206000ec51df.2411..m: <sip:101@192.168.2.20:5060>..i: CB3623BF-158D-4124-A1D6-877D2673 BBD4@192.168.2.20..CSeq: 41383 ACK..Max-Forwards: 70..l: 0.... U 2006/11/14 03:01:04.642913 61.17.248.68:5831 -> 192.168.2.90:5060 INVITE sip:102@24.99.145.14 SIP/2.0..v: SIP/2.0/UDP 192.168.2.20:5060;rport;branch=z9hG4bKB90C342E73C741EE87BEC21516FF0FF6..f: siri <sip:101@24.99.145.1 4>;tag=3410236415..t: <sip:102@24.99.145.14>..m: <sip:101@192.168.2.20:5060>..i: CB3623BF-158D-4124-A1D6-877D2673BBD4@192.168.2.20..CSeq: 41384 INVITE.. Proxy-Authorization: Digest username="101",realm="24.99.145.14 ",nonce="455978ece372d35484e2a5f0010e0766a823f3a6",response="3f35a6a54ac7c9695824c5443a9bf 512",uri="sip:102@24.99.145.14"..Max-Forwards: 70..c: application/sdp..User-Agent: X-Lite release 1103m..l: 135....v=0..o=101 9577234 9577250 IN IP4 192 .168.2.20..s=X-Lite..c=IN IP4 192.168.2.20..t=0 0..m=audio 12222 RTP/AVP 0..a=rtpmap:0 pcmu/8000.. U 2006/11/14 03:01:04.644799 192.168.2.90:5060 -> 61.17.248.68:5831 SIP/2.0 100 trying -- your call is important to us..v: SIP/2.0/UDP 192.168.2.20:5060 ;rport=5831;branch=z9hG4bKB90C342E73C741EE87BEC21516FF0FF6;received= 61.17.248.68..f: siri <sip:101@24.99.145.14>;tag=3410236415..t: < sip:102@24.99.145.14>..i: CB3623BF-158D-4124-A1D6-877D2673BBD4@192.168.2.20..CSeq: 4138 4 INVITE..Server: OpenSer (1.1.0-notls (i386/linux))..Content-Length: 0..Warning: 392 192.168.2.90:5060 "Noisy feedback tells: pid=4701 req_src_ip=61.1 7.248.68 req_src_port=5831 in_uri=sip:102@24.99.145.14 out_uri= sip:102@192.168.2.45:5060 via_cnt==1".... U 2006/11/14 03:01:04.645093 192.168.2.90:5060 -> 61.17.248.68:5060 INVITE sip:102@192.168.2.45:5060 SIP/2.0..Record-Route: <sip:192.168.2.90;lr=on;ftag=3410236415>..Via: SIP/2.0/UDP 192.168.2.90;branch=z9hG4bK535d.31f40 331.0..v: SIP/2.0/UDP 192.168.2.20:5060;received=61.17.248.68;rport=5831;branch=z9hG4bKB90C342E73C741EE87BEC21516FF0FF6..f: siri <sip:101@24.99.145.14>; tag=3410236415..t: <sip:102@24.99.145.14>..m: <sip:101@61.17.248.68:5831>..i: CB3623BF-158D-4124-A1D6-877D2673BBD4@192.168.2.20..CSeq: 41384 INVITE..Pro xy-Authorization: Digest username="101",realm="24.99.145.14 ",nonce="455978ece372d35484e2a5f0010e0766a823f3a6",response="3f35a6a54ac7c9695824c5443a9bf512 ",uri="sip:102@24.99.145.14"..Max-Forwards: 69..c: application/sdp..User-Agent: X-Lite release 1103m..Content-Length: 162....v=0..o=101 9577234 9577250 IN IP4 192.168.2.20..s=X-Lite..c=IN IP4 61.17.248.68..t=0 0..m=audio 12222 RTP/AVP 0..a=rtpmap:0 pcmu/8000..a=oldmediaip:192.168.2.20.. U 2006/11/14 03:01:05.223317 192.168.2.90:5060 -> 61.17.248.68:5060 INVITE sip:102@192.168.2.45:5060 SIP/2.0..Record-Route: <sip:192.168.2.90;lr=on;ftag=3410236415>..Via: SIP/2.0/UDP 192.168.2.90;branch=z9hG4bK535d.31f40 331.0..v: SIP/2.0/UDP 192.168.2.20:5060;received=61.17.248.68;rport=5831;branch=z9hG4bKB90C342E73C741EE87BEC21516FF0FF6..f: siri <sip:101@24.99.145.14>; tag=3410236415..t: <sip:102@24.99.145.14>..m: <sip:101@61.17.248.68:5831>..i: CB3623BF-158D-4124-A1D6-877D2673BBD4@192.168.2.20..CSeq: 41384 INVITE..Pro xy-Authorization: Digest username="101",realm="24.99.145.14 ",nonce="455978ece372d35484e2a5f0010e0766a823f3a6",response="3f35a6a54ac7c9695824c5443a9bf512 ",uri="sip:102@24.99.145.14"..Max-Forwards: 69..c: application/sdp..User-Agent: X-Lite release 1103m..Content-Length: 162....v=0..o=101 9577234 9577250 IN IP4 192.168.2.20..s=X-Lite..c=IN IP4 61.17.248.68..t=0 0..m=audio 12222 RTP/AVP 0..a=rtpmap:0 pcmu/8000..a=oldmediaip:192.168.2.20.. U 2006/11/14 03:01:05.674174 61.17.248.68:5060 -> 192.168.2.90:5060 SIP/2.0 100 Trying..v: SIP/2.0/UDP 192.168.2.90;branch= z9hG4bK535d.31f40331.0;received=24.99.145.14..v: SIP/2.0/UDP 192.168.2.20:5060;received=61.17.248 .68;rport=5831;branch=z9hG4bKB90C342E73C741EE87BEC21516FF0FF6..f: siri < sip:101@24.99.145.14>;tag=3410236415..t: <sip:102@24.99.145.14 >;tag=3539775680.. m: <sip:102@192.168.2.45:5060>..Record-Route: <sip:192.168.2.90;lr=on;ftag=3410236415>..i: CB3623BF-158D-4124-A1D6-877D2673BBD4@192.168.2.20..CSeq: 4138 4 INVITE..Server: X-Lite release 1103m..l: 0.... ----------------------------------------------------------- Now the starts Ring at both ends points .......... -------------------------------------------------- When the phone is pick up ............ it send the Reply only 200 Ack......... 2006/11/14 03:01:06.263035 192.168.2.90:5060 -> 61.17.248.68:5831 SIP/2.0 180 Ringing..v: SIP/2.0/UDP 192.168.2.20:5060;received= 61.17.248.68;rport=5831;branch=z9hG4bKB90C342E73C741EE87BEC21516FF0FF6..f: siri <sip:101@ 24.99.145.14>;tag=3410236415..t: <sip:102@24.99.145.14>;tag=3539775680..m: <sip:102@61.17.248.68:5060>..Record-Route: <sip:192.168.2.90 ;lr=on;ftag=34102 36415>..i: CB3623BF-158D-4124-A1D6-877D2673BBD4@192.168.2.20..CSeq: 41384 INVITE..Server: X-Lite release 1103m..l: 0.... U 2006/11/14 03:01:22.633879 61.17.248.68:5060 -> 192.168.2.90:5060 SIP/2.0 200 Ok..v: SIP/2.0/UDP 192.168.2.90;branch=z9hG4bK535d.31f40331.0 ;received=24.99.145.14..v: SIP/2.0/UDP 192.168.2.20:5060;received= 61.17.248.68; rport=5831;branch=z9hG4bKB90C342E73C741EE87BEC21516FF0FF6..f: siri < sip:101@24.99.145.14>;tag=3410236415..t: <sip:102@24.99.145.14>;tag=3539775680..m: < sip:102@192.168.2.45:5060>..Record-Route: <sip:192.168.2.90;lr=on;ftag=3410236415>..i: CB3623BF-158D-4124-A1D6-877D2673BBD4@192.168.2.20..CSeq: 41384 IN VITE..c: application/sdp..Server: X-Lite release 1103m..l: 135....v=0..o=102 6553093 6570062 IN IP4 192.168.2.45..s=X-Lite..c=IN IP4 192.168.2.45..t=0 0 ..m=audio 12222 RTP/AVP 0..a=rtpmap:0 pcmu/8000.. U 2006/11/14 03:01:22.634713 192.168.2.90:5060 -> 61.17.248.68:5831 SIP/2.0 200 Ok..v: SIP/2.0/UDP 192.168.2.20:5060;received=61.17.248.68;rport=5831;branch=z9hG4bKB90C342E73C741EE87BEC21516FF0FF6..f: siri <sip:101@24.99 .145.14>;tag=3410236415..t: <sip:102@24.99.145.14>;tag=3539775680..m: < sip:102@61.17.248.68:5060>..Record-Route: <sip:192.168.2.90 ;lr=on;ftag=3410236415 >..i: CB3623BF-158D-4124-A1D6-877D2673BBD4@192.168.2.20..CSeq: 41384 INVITE..c: application/sdp..Server: X-Lite release 1103m..l: 135....v=0..o=102 6553 093 6570062 IN IP4 192.168.2.45..s=X-Lite..c=IN IP4 192.168.2.45..t=00..m=audio 12222 RTP/AVP 0..a=rtpmap:0 pcmu/8000.. ---------------------------------- That this Time when the User agents are Hung up the Bye Request not coming to sip server at all........... Below logs are from x-lite after the Invite method processed SEND TIME: 6581140 SEND >> 24.99.145.14:5060 SIP/2.0 200 Ok v: SIP/2.0/UDP 192.168.2.90;branch=z9hG4bK535d.31f40331.0;received= 24.99.145.14 v: SIP/2.0/UDP 192.168.2.20:5060;received=61.17.248.68 ;rport=5831;branch=z9hG4bKB90C342E73C741EE87BEC21516FF0FF6 f: siri <sip:101@24.99.145.14>;tag=3410236415 t: <sip:102@24.99.145.14>;tag=3539775680 m: <sip:102@192.168.2.45:5060> Record-Route: <sip:192.168.2.90;lr=on;ftag=3410236415> i: CB3623BF-158D-4124-A1D6-877D2673BBD4(a)192.168.2.20 CSeq: 41384 INVITE c: application/sdp Server: X-Lite release 1103m l: 135 v=0 o=102 6553093 6570062 IN IP4 192.168.2.45 s=X-Lite c=IN IP4 192.168.2.45 t=0 0 m=audio 12222 RTP/AVP 0 a=rtpmap:0 pcmu/8000 SEND TIME: 6602609 SEND >> 192.168.2.90:5060 BYE sip:101@61.17.248.68:5831 SIP/2.0 v: SIP/2.0/UDP 192.168.2.45:5060 ;rport;branch=z9hG4bK827A50D7ABED4E0ABA350E5487A52AB9 f: <sip:102@24.99.145.14>;tag=3539775680 t: siri <sip:101@24.99.145.14>;tag=3410236415 m: <sip:102@192.168.2.45:5060> Route: <sip:192.168.2.90;lr=on;ftag=3410236415> i: CB3623BF-158D-4124-A1D6-877D2673BBD4(a)192.168.2.20 CSeq: 31544 BYE Max-Forwards: 70 User-Agent: X-Lite release 1103m l: 0 SEND TIME: 6604140 SEND >> 192.168.2.90:5060 BYE sip:101@61.17.248.68:5831 SIP/2.0 v: SIP/2.0/UDP 192.168.2.45:5060 ;rport;branch=z9hG4bK827A50D7ABED4E0ABA350E5487A52AB9 f: <sip:102@24.99.145.14>;tag=3539775680 t: siri <sip:101@24.99.145.14>;tag=3410236415 m: <sip:102@192.168.2.45:5060> Route: <sip:192.168.2.90;lr=on;ftag=3410236415> i: CB3623BF-158D-4124-A1D6-877D2673BBD4(a)192.168.2.20 CSeq: 31544 BYE Max-Forwards: 70 User-Agent: X-Lite release 1103m l: 0 ECEIVE TIME: 6606187 RECEIVE << 24.99.145.14:5060 SEND TIME: 6607390 SEND >> 192.168.2.90:5060 BYE sip:101@61.17.248.68:5831 SIP/2.0 v: SIP/2.0/UDP 192.168.2.45:5060 ;rport;branch=z9hG4bK827A50D7ABED4E0ABA350E5487A52AB9 f: <sip:102@24.99.145.14>;tag=3539775680 t: siri <sip:101@24.99.145.14>;tag=3410236415 m: <sip:102@192.168.2.45:5060> Route: <sip:192.168.2.90;lr=on;ftag=3410236415> i: CB3623BF-158D-4124-A1D6-877D2673BBD4(a)192.168.2.20 CSeq: 31544 BYE Max-Forwards: 70 User-Agent: X-Lite release 1103m l: 0 RECEIVE TIME: 6666312 RECEIVE << 24.99.145.14:5060 _ -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 ravi.sunkara(a)hyperion-tech.com www.hyperion-tech.com

1 0

[Serusers] SER database migration tool
by Jan Janak 17 Nov '06

17 Nov '06

Hello, A new SER database migration tool is available at: http://iptel.org/ser/migrate_db The tool can be used to migrate the contents of the old SER database used in releases 0.9.x into the new schema used in 0.10.x. Please report bugs to serdev(a)iptel.org Jan.

1 0

RE: [Users] Presence server with MySQL >=5.0.26
by Watkins, Bradley 16 Nov '06

16 Nov '06

> -----Original Message----- > From: users-bounces(a)openser.org > [mailto:users-bounces@openser.org] On Behalf Of Mike Williams > Sent: Thursday, November 16, 2006 12:31 PM > To: users(a)openser.org > Subject: Re: [Users] Presence server with MySQL >=5.0.26 > > All, > > I would still be in favor of moving all of the tables over to > having the primary key be an autoincremented unsigned int. > > > Mike Williams I'm certainluy not opposed to this, especially with the "oh yeah, UTF-8" clue bat Norman hit me with. ;) I was mostly pointing out that there is nothing inherent in MySQL 5.0.26 and up that precludes this table definition (unless you're using UTF-8, of course). - Brad The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it.

1 0

RE: [Users] Presence server with MySQL >=5.0.26
by Watkins, Bradley 16 Nov '06

16 Nov '06

I originally sent this to the OP instead of the list by mistake: This sounds like a bug in the package you're using or MySQL 5.0.26 itself. The storage requirements for the key should be 388 bytes, which is much less (obviously) than the maximum key length of a MyISAM table. Empirically, I tested this on MySQL 5.0.22 (Fedora RPMs) and MySQL 5.1.11 (RPMs from MySQL's website) with the original values and they both worked. The documentation for 5.0 and 5.1 on MySQL's site both state the 1000 byte limitation (that is, without code modification). As an additional data point not in the original e-mail above, using the 5.0.27 .tar.gz from MySQL also has no problems with this table definition. I've even gone so far as doubling the size of the fields used for the PK, and none of the aforementioned versions balked. With the original sizes, it should even work in older (I think pre 4.1.12?) versions of MySQL where the max key length is 500 bytes. I really don't think there's a problem here. Regards, - Brad > -----Original Message----- > From: users-bounces(a)openser.org > [mailto:users-bounces@openser.org] On Behalf Of Bogdan-Andrei Iancu > Sent: Thursday, November 16, 2006 7:06 AM > To: Nicolas Palix; Anca Vamanu > Cc: users(a)openser.org > Subject: Re: [Users] Presence server with MySQL >=5.0.26 > > Hi Nicolas, > > I think the solution will be to have a primary key a simple > unsigned int outincrement field in the table and move the > primary key as unique. > > I will ask Anca to do this change as most probably it will be > a good idea to do it for the rest of the tables also. > > regards, > Bogdan > > Nicolas Palix wrote: > > >Hi, > > > > > >The primary key of the watchers is too long with this > version of MySQL: > >"Specified key was too long; max key length is 1000 bytes" > > > >Downgrading "[pw]_domain" from 128 to 96. Works. > > > >CREATE TABLE `watchers` ( > > `p_user` varchar(64) NOT NULL, > > `p_domain` varchar(96) NOT NULL, > > `w_user` varchar(64) NOT NULL, > > `w_domain` varchar(96) NOT NULL, > > `subs_status` varchar(64) NOT NULL, > > `reason` varchar(64) NOT NULL, > > PRIMARY KEY (`p_user`,`p_domain`,`w_user`,`w_domain`) > >) ENGINE=MyISAM; > > > >Regards > > > > > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > > The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it.

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Re: [Users] Can We Use the Stun server for NAT supporting,
by Daniel-Constantin Mierla 16 Nov '06

16 Nov '06

Hello, On 11/15/06 14:08, raviprakash sunkara wrote: > Hello Daniel > Thanks for reply me... > > if U keep My Sip Server direct from Modem( ISP ) to System , Its > working fine, > But problem is when keep My sip Server Behind the Router/Firewall it > not working, UAC's behind the NAT is not Hungup and Audio is Fine at > end points... > Help me ............ check what happens on the firewall/nat, if the messages are forwarded properly between public and private networks. ngrep or ethereal should help here ... Cheers, Daniel > > > On 11/15/06, *Daniel-Constantin Mierla* <daniel(a)voice-system.ro > <mailto:daniel@voice-system.ro>> wrote: > > Hello, > > stun is recommended when the client properly implements it and > does not > sit behind a symmetric NAT (like linux box). For these cases you > need to > use server side nat traversal, look at nathelper/rtpptoxy or > mediaproxy/mediaproxy. > > Cheers, > Daniel > > > On 11/13/06 16:29, raviprakash sunkara wrote: > > Hello Users, > > > > Can I use the Stun server to Support the NAT. > > > > I tried a lot , My UAC's are Not Hungupping in Behind the > NAT/Router > > > > My SIP ip is 192.168.2.75 <http://192.168.2.75> > <http://192.168.2.75 > > > My Public IP is xx.xx.xx.xx > > > > I didn't get any Bye request to SIP server, except the Bye all the > > other Messages are Requested to MY SIP Server > > > > in openser.cfg > > i > > I'm using X-lites for testing in that > > RTP port 12222 > > NAT firewall ip is 192.168.2.1 <http://192.168.2.1> > <http://192.168.2.1> > > Sip domain is xx.xx.xx.xx > > Sip proxy xx.xx.xx.xx > > ______________________________________________ > > modparam ("nathelper",rtpproxy_sock","udp: 192.168.2.75:12222 > <http://192.168.2.75:12222> > > <http://192.168.2.75:12222>" > > > > if(nat_uac_test("19")) > > { > > if(method=="REGISTER") > > { > > fix_nated_register(); > > force_rport(); > > setflag(6); > > } else > > { > > > > if(method=="INVITE") > > { > > fix_nated_sdp("3") > > }; > > fix_nated_contact(); > > }; > > }; > > ______________________________________________ > > -- > > Thanks and Regards > > Ravi Prakash Sunkara > > ravi.sunkara(a)hyperion-tech.com > <mailto:ravi.sunkara@hyperion-tech.com> > <mailto:ravi.sunkara@hyperion-tech.com > <mailto:ravi.sunkara@hyperion-tech.com>> > > M:+91 9985077535 > > O:+91 40 23114549 > > F:+91 40 40208727 > > ravi.sunkara(a)hyperion-tech.com > <mailto:ravi.sunkara@hyperion-tech.com> <mailto: > ravi.sunkara(a)hyperion-tech.com > <mailto:ravi.sunkara@hyperion-tech.com>> > > www.hyperion-tech.com <http://www.hyperion-tech.com> > <http://www.hyperion-tech.com> > > > ------------------------------------------------------------------------ > > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org <mailto:Users@openser.org> > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > > -- > Thanks and Regards > Ravi Prakash Sunkara > ravi.sunkara(a)hyperion-tech.com <mailto:ravi.sunkara@hyperion-tech.com> > M:+91 9985077535 > O:+91 40 23114549 > F:+91 40 40208727 > ravi.sunkara(a)hyperion-tech.com <mailto:ravi.sunkara@hyperion-tech.com> > www.hyperion-tech.com <http://www.hyperion-tech.com>

1 0

[Users] Radius extra: integer attribute type how to
by Pletli Antal 16 Nov '06

16 Nov '06

Hello, I'm trying the radius-extra feature in openser cvs snapshot. In my openser.cfg I set up the following parameter: modparam("acc", "radius_extra", "Acct-Session-Time=$avp(call_length)") modparam( "avpops", "avp_aliases", "start_timestamp=i:100") modparam( "avpops", "avp_aliases", "end_timestamp=i:101") modparam( "avpops", "avp_aliases", "call_length=i:102") (...) avp_op("$avp(end_timestamp)","sub/$avp(start_timestamp)"); avp_op("$avp(end_timestamp)/$avp(call_length)","sub/$avp(start_timestamp )"); I have experienced that when I sent an Accounting-Stop request to radius server, it received the integer value as an string and it decoded the value to the chr code of the "3" In the radius log: * Attr: 46 - Acct-Delay-Time len: 6 val: 51 In the openser.log: Nov 14 16:12:09 sip2 ./openser[727]: INFO:avpops:print_avp: p=0x40623b88, flags=0x0000 Nov 14 16:12:09 sip2 ./openser[727]: INFO: id=<102> Nov 14 16:12:09 sip2 ./openser[727]: INFO: val_int=<3> How can I set the corresponding integer attribute in the request? Thanks in advance, Antal

2 1

[Users] Presence server with MySQL >=5.0.26
by Nicolas Palix 16 Nov '06

16 Nov '06

Hi, The primary key of the watchers is too long with this version of MySQL: "Specified key was too long; max key length is 1000 bytes" Downgrading "[pw]_domain" from 128 to 96. Works. CREATE TABLE `watchers` ( `p_user` varchar(64) NOT NULL, `p_domain` varchar(96) NOT NULL, `w_user` varchar(64) NOT NULL, `w_domain` varchar(96) NOT NULL, `subs_status` varchar(64) NOT NULL, `reason` varchar(64) NOT NULL, PRIMARY KEY (`p_user`,`p_domain`,`w_user`,`w_domain`) ) ENGINE=MyISAM; Regards -- Nicolas Palix http://phoenix.labri.fr/people/palix/

2 1

[Fwd: Re: [Users] Presence module and Snom phones]
by Anca Vamanu 16 Nov '06

16 Nov '06

1 0

[Users] module registrar and path_mode
by Gregoire 16 Nov '06

16 Nov '06

Hi everybody! I want to use the Path support in the module registrar. So I set, in my configuration file, the following parameters: loadmodule "/usr/lib/openser/modules/registrar.so" modparam("registrar", "use_path", 1) modparam("registrar", "path_mode", 2) I expected that OpenSER rejects all REGISTER requests that not have the Path support indicated but in fact it accepts the REGISTER and reply with a 200 OK instead of a 420 - Bad Extension. I am using OpenSER 1.1.0 with TLS support in a debian package. Does anyone have some experience with this module? Thanks Greg

2 3

[Serusers] (no subject)
by ali daher 15 Nov '06

15 Nov '06

I have configured SER on IPV6 on a linux suse 10.1, and i changed ser.cfg so that the server supports IPV6 my clients are on a windows machine and they run also on ipv6 i'm facing a problems: i did set the sip account for a client , after that it rediscovers the network but it can not register and i get this message: registration error 503 service unavaialable so what is the problem and what can i do? regards _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

2 2

[Serusers] Using SER with TLS
by Lami, Avner 15 Nov '06

15 Nov '06

Hello, We have SIP equipment that can work with UDP and TCP. We made many tests with SER and we didn't find any interoperability problems yet. We need to find solution for one of our customers and we want to check whether we can use SER. Our customer use Microsoft LCS 2005 with TLS. Because our equipment can't work with TLS we want to check if we can use SER as proxy that will connect between our equipment and the LCS and will make the translation between TCP to TLS and vice versa. Our questions are: 1) How can we use SER for that solution? 2) Did you make interoperability tests between SER and Microsoft LCS using TLS? 3) Where can we find documentation regarding such scenarios? Regards, Al

2 1

[Serusers] 603 Decline Problem
by pol 15 Nov '06

15 Nov '06

Hi, We are using SER inside a platform we are developing. We have noticed that when the called user reject the call when it hasn't established, SER sends multiple 603 Declined messages to the caller user. I think SER should send just one 603 Declined messages. Is it correct? Any idea? Thanks in advance,

2 1

[Serusers] Is it possible in Ser?!
by rjcarvalho＠reit.up.pt 15 Nov '06

15 Nov '06

Hi all, Is it possible to configure Ser to act just like a phone which when receives a ?302 Moved Temporarily? replay, generates and sends an INVITE to the ?Contact? shown in that 302 message? Is it possible? How can I generate that INVITE? Thanks in advance, Ricardo.

2 1

[Serusers] Weird Signaling
by Gustavo Passos Tourinho 15 Nov '06

15 Nov '06

Hey There. I'm having a weird signaling. When the proxy receives an ACK from caller, it puts an route at the end of packet and then forwards it. I have no idea what i could be. Does anyone have a clue? Thanks in advanced, Tourinho

3 3

[Serusers] Ser and OpenVZ
by Shidan 15 Nov '06

15 Nov '06

Hi, I'm trying to get ser to run on openvz but I get the following error: 'Too much shared memory demanded: 33554432' removing -DSHM_MMAP doesnt help, but starting ser in non forked (debugging) mode does. Anyone have any successfull experience with openvz or can give me a pointer?

2 1

[Users] Presence module and Snom phones
by Peter Gradwell 15 Nov '06

15 Nov '06

Hi We are wondering if anyone has used the new presence module with Snom phones because it looks like it just doesn't work with the subscribe types that snom use. Basically snom uses Event: dialog in the headers which the module doesn't support. Here is the subscribe after I hacked it to allow Event: dialog U 2006/11/15 14:22:04.699050 193.84.87.60:5082 -> 193.111.201.74:5060 SUBSCRIBE sip:1000305@lon-sip-1.gradwell.net;user=phone SIP/2.0. To: <sip:1000305@lon-sip-1.gradwell.net;user=phone>. From: <sip:1000305@lon-sip-1.gradwell.net>;tag=23c9f970. Via: SIP/2.0/UDP 193.84.87.60:5082;branch=z9hG4bK-d87543-65b974429da21b246a5f-1-cHAyZGM4N jUxNzk0YjIyNTJkNjdmNw..-d87543-. Call-ID: 0c296edb42d0dd85472907b1da99411d. CSeq: 452392058 SUBSCRIBE. Contact: <sip:CNOgG7HbGwkzlYrxgc7RjID5rBqkrZFREwe644KJ40.@193.84.87.60:5082;line= 27nd4hcq>. Expires: 3600. Max-Forwards: 69. Accept: application/dialog-info+xml. Event: dialog. Content-Length: 0. . U 2006/11/15 14:22:04.707409 193.111.201.74:5060 -> 193.84.87.60:5082 SIP/2.0 200 OK. To: <sip:1000305@lon-sip-1.gradwell.net;user=phone>;tag=20c56ba31bbe85b9bc9d 0e09f876aff7.24f5. From: <sip:1000305@lon-sip-1.gradwell.net>;tag=23c9f970. Via: SIP/2.0/UDP 193.84.87.60:5082;branch=z9hG4bK-d87543-65b974429da21b246a5f-1-cHAyZGM4N jUxNzk0YjIyNTJkNjdmNw..-d87543-. Call-ID: 0c296edb42d0dd85472907b1da99411d. CSeq: 452392058 SUBSCRIBE. Expires: 3600. Server: OpenSer (1.2.0-dev6-notls (i386/linux)). Content-Length: 0. Warning: 392 193.111.201.74:5060 "Noisy feedback tells: pid=14435 req_src_ip=193.84.87.60 req_src_port=5082 in_uri=sip:1000305@lon-sip-1.gradwell.net;user=phone out_uri=sip:1000305@lon-sip-1.gradwell.net via_cnt==1". . Then openser sends out this notify U 2006/11/15 14:22:04.733404 193.111.201.74:5060 -> 193.84.87.60:5082 NOTIFY sip:CNOgG7HbGwkzlYrxgc7RjID5rBqkrZFREwe644KJ40.@193.84.87.60:5082;line=2 7nd4hcq SIP/2.0. Via: SIP/2.0/UDP 193.111.201.74;branch=z9hG4bKfc5a.9a187a95.0. To: sip:1000301@lon-sip-1.gradwell.net;tag=9d6ee568. From: sip:1000301@lon-sip-1.gradwell.net;tag=10.14435.1163600524.9. CSeq: 183348005 NOTIFY. Call-ID: d250aa54d4fab3ffa3e96ac7ea73303f. User-Agent: OpenSer (1.2.0-dev6-notls (i386/linux)). Event: dialog. Contact: <sip:1000301@lon-sip-1.gradwell.net>. Subscription-State: pending;expires=3600. Content-Type: application/watcherinfo+xml. . Which is most significantly the wrong content type for snoms (as the subscribe said, they want application/dialog-info+xml) and also doesn't have a message body on the notify - so the snom tells openser where to go: U 2006/11/15 14:22:04.841722 193.84.87.60:5082 -> 193.111.201.74:5060 SIP/2.0 489 Bad Event. To: <sip:1000301@lon-sip-1.gradwell.net>;tag=9d6ee568. From: <sip:1000301@lon-sip-1.gradwell.net>;tag=10.14435.1163600524.9. Via: SIP/2.0/UDP 193.111.201.74;branch=z9hG4bKfc5a.9a187a95.0. Call-ID: d250aa54d4fab3ffa3e96ac7ea73303f. CSeq: 183348005 NOTIFY. Content-Length: 0. . Thoughts welcome! Thanks peter -- peter gradwell. gradwell dot com Ltd. http://www.gradwell.com/ -- engineering & hosting services for email, web and voip -- -- http://www.peter.me.uk/ -- http://www.voip.org.uk/ --

2 1

[Users] AVPs iteration
by tele 15 Nov '06

15 Nov '06

Hi all, In the pseudo variables docs page: $avp(id[N]) - represents the value of N-th AVP identified by 'id'. This means if i do an avp_db_query that return N avps i can access to the value of avps with $avp(id[N]). But if i don't know how many record are returned from the query how can i iterate over an array of avps ? I've something like: avp_db_query("SELECT n.number,n.type,n.translated FROM numerations AS n, users AS u, associations AS a WHERE a.numeration_id = n.id AND a.user_id = u.id AND u.lid = '$tU'","$avp(cli);$avp(type);$avp(translated)") so i can access to the record with $avp(cli[0]) $avp(cli[1]) etc.. i want to append all contact returned to the destination uri and with the record $avp(translated) "true" i want to do some processing. but if i can't iterate over array of avps it's difficult. thanks, :tele

2 2

[Users] Can We Use the Stun server for NAT supporting,
by raviprakash sunkara 15 Nov '06

15 Nov '06

Hello Users, Can I use the Stun server to Support the NAT. I tried a lot , My UAC's are Not Hungupping in Behind the NAT/Router My SIP ip is 192.168.2.75 My Public IP is xx.xx.xx.xx I didn't get any Bye request to SIP server, except the Bye all the other Messages are Requested to MY SIP Server in openser.cfg i I'm using X-lites for testing in that RTP port 12222 NAT firewall ip is 192.168.2.1 Sip domain is xx.xx.xx.xx Sip proxy xx.xx.xx.xx ______________________________________________ modparam ("nathelper",rtpproxy_sock","udp:192.168.2.75:12222" if(nat_uac_test("19")) { if(method=="REGISTER") { fix_nated_register(); force_rport(); setflag(6); } else { if(method=="INVITE") { fix_nated_sdp("3") }; fix_nated_contact(); }; }; ______________________________________________ -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 ravi.sunkara(a)hyperion-tech.com www.hyperion-tech.com

2 1

[Users] Help me for the Expiration of the Openser
by ShivaSankar Padhy 15 Nov '06

15 Nov '06

Dear All, I need to put a silly question to you, What should I change in the Openser so that it will expire after a certain period of time.?? Seems to be simple but I dont know this. Hoping for a Positive response Thanks and Regards ShivaSankar

2 1

[Users] voicemail
by Muhammad Malik 15 Nov '06

15 Nov '06

Dear All, I am trying to setup voicemail on SER using SEMS. I am new to SER. Can anyone please points to any good tutorial on installing voicemail using SER. Please help me out. Regards, -Muhammad

2 1

[Users] CDRTool with OpenSER
by Prosper Luteganya 14 Nov '06

14 Nov '06

Hi List, Is there anybody who has successfully deployed (commercially) CDRTool with OpenSER ?. regards Prosper

1 0

[Serusers] Making a presence server
by inge 14 Nov '06

14 Nov '06

Hi all, I find a way to make a presence server for my SER server. I read about it on this mailing list and also on the iptel tutorial. But, I'm lost. Can someone help me to give me the best way for make this ? My server is 0.9.4 and it is in production, so maybe can I use a standalone server ? Thanks for help. My best regards, Adrien

2 2

[Serusers] Cisco ATA186 setup question
by Steve Blair 14 Nov '06

14 Nov '06

Hello: Does anyone on this list have experience with a Cisco ATA186 running 3.0.2 software? We have a working ATA186 running 2.10 software. We upgraded to have the ability to add a display name to the from header but the new software seems to have trouble registering when the proxy name is an SRV record. I've set the ConnectMode flag to several values including "0x30060400" but this does not work. Any thoughts you might have on how to get this thing to register will be appreciated. Thanks,Steve

1 0

RE: [Users] UAC module. CANCEL request handling.
by Leonid Fainshtein 14 Nov '06

14 Nov '06

Hi Bogdan, I've added the problem to the tracker. The requestID is 1596331. Leonid -----Original Message----- From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] Sent: Tuesday, November 14, 2006 1:03 PM To: Leonid Fainshtein Cc: users(a)openser.org Subject: Re: [Users] UAC module. CANCEL request handling. Hi Leonid, no, there is no automatic way of replacing the FROM hdr in CANCEL - you need to do it from script, in a similar manner as for INVITE. For your request makes a lot of sense and I will try to see if something can be done here; but please upload this request on the tracker not to forget about it. Thanks and regards, bogdan Leonid Fainshtein wrote: >Hi, >I try to use the "UAC module" for replacing header field "From". >I defined the automatic mode for "From" restore >(modparam("uac","from_restore_mode","auto")) and change "From" only for >INVITE requests. >It looks like I also have to change manually header filed "From" also >in CANCEL requests. It is because UAC (call originator) sends CANCEL >without header "Route" and therefore OpenSer can't perform the >automatic replacement. >Do I miss anything? Is there a way to implement automatic header field >"From" handling for the CANCEL requests? > >Thank you in advance, >Leonid Fainshtein > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > > >

2 1

[Users] new presence server
by Daniel-Constantin Mierla 14 Nov '06

14 Nov '06

Hello, a new presence server which follows the standards and latest draft of SIMPLE has been committed on CVS. With this occasion I want to introduce Anca Vamanu, a new developer which will be in charge of maintaining the presence related stuff. The new module passed successfully a period of tests with common SIP UAs having SIMPLE extensions. It is still long way until we get full SIMPLE extensions, but we have the most desired functionalities: presence info, watcher info and XCAP. Some of you already tested it, and would be great if we can collect the experiences everyone encounters with presence extensions and different UA. There is a dokuwiki page where everybody can contribute: http://www.openser.org/dokuwiki/doku.php/presence:presence-module Since there are not so many mature implementation in client side, the testing proves to be a difficult process, and your help is very welcome. The dokuwiki page includes some sample config files that can be used as starting point. The news on web site and the readme for the module: http://www.openser.org/index.php?option=com_content&task=view&id=62&Itemid=9 http://www.openser.org/docs/modules/devel/presence.html Next on track is a generic module which will be able to publish presence information in behalf of dummy entities. For example will publish online/offline for hardphones registered on user location table. Looking forward to your feedback. Cheers, Daniel

7 9

[Users] presence server
by henri keski-sikkila 14 Nov '06

14 Nov '06

Hello, If I try to subsribe presence information from the user that doesn't exist I get notify-message that doesn't include body. There is Content-Type: application/pidf+xml but body is empty. If there isn't such user's presence information at all why I get notify-message without body. NOTIFY sip:+3586xxxxxxx@x.x.x.x:5060 SIP/2.0 Max-Forwards: 10 Via: SIP/2.0/UDP x.x.x.x;branch=z9hG4bK09dc.6e96f68.0 Via: SIP/2.0/UDP x.x.x.x:xxxx;branch=z9hG4bK09dc.bb749775.0 To: sip:+3586xxxxxxx@domain.fi;tag=1059233066 From: sip:+3586xxxxxxx@domain.fi;tag=10.27098.1163501233.274 CSeq: 21 NOTIFY Call-ID: xxxxxxx(a)10.0.0.5 User-Agent: OpenSer (1.2.0-dev6-notls (i386/linux)) Event: presence Contact: <sip:+3586xxxxxxx@domain.fi> Subscription-State: active;expires=600 Content-Type: application/pidf+xml Contact: <sip:+3586xxxxxxx@domain.fi> Subscription-State: active;expires=600 Content-Type: application/pidf+xml Best Regards, Henri

2 2

[Serusers] Serweb - Unable include html files, php errors
by "tzieleniewski" 14 Nov '06

14 Nov '06

Hi All!! I've installed serweb 0.9.4 acording to the INSTALL file attached with the distribution. Im running an apache2 serwer. Whenever i open the ../admin/index.php page I have the following errors: Warning: virtual() [function.virtual]: Unable to include '/usr/local/serweb-0.9.4/html/domains/_default/prolog.html' - request execution failed in /usr/local/serweb-0.9.4/html/page.php on line 79 localhost admin interface Warning: virtual() [function.virtual]: Unable to include '/usr/local/serweb-0.9.4/html/domains/_default/separator.html' - request execution failed in /usr/local/serweb-0.9.4/html/page.php on line 81 And at the same time I have the following error in the apache error log: Warning: virtual() [function.virtual]: Unable to include '/usr/local/serweb-0.9.4/html/domains/_default/prolog.html' - request execution failed in /usr/local/serweb-0.9.4/html/page.php on line 79 localhost admin interface Warning: virtual() [function.virtual]: Unable to include '/usr/local/serweb-0.9.4/html/domains/_default/separator.html' - request execution failed in /usr/local/serweb-0.9.4/html/page.php on line 81 After loging in into the users.php page when I try to go to "open proviliges" I get the following displayed error in the browser: Not Found The requested URL /usr/local/serweb-0.9.4/html/admin/list_of_admins.php was not found on this server. Apache/2.2.3 (Debian) PHP/4.4.4-6 Server at localhost Port 80 and the following one in the error log: [Tue Nov 14 12:46:23 2006] [error] [client 127.0.0.1] File does not exist: /var/www/usr, referer: http://localhost/serweb/admin/users.php?kvrk=4559ab68db15c Thanks for any help and Your time Bests Tomasz

1 0

[Users] DST and accounting
by Dimo 14 Nov '06

14 Nov '06

Hello, I have sent this a while ago but it did not come, so if you are getting the mail for the second time sorry. You may be aware of the accounting problem with Daylight Savings Time. Currently in Radius accounting we have a timestamp of the event acoording to system clock (i think). So if a call starts soon before a daylight saving change and ends soon after it, it may have too big or negative duration, depending on whethere the clock moved forward or backward. Is there a way to make accounting take notice of that (maybe accounting in gmt+offset or something?) Any ideas appreciated, as you know DST is comming soon for europe at least. Dimitar

3 4

[Users] Q:ABOUTE:SET UP MESSAGE FLOW
by Alejandro Sanchez 14 Nov '06

14 Nov '06

Hi evreybody. I need to set up the next message flow Tel.A = Phone A Tel.B = Phone B B = Balancer (openser 1.1) SS = Sip Server (openser 1.1) Tel.A B SS(get location) Tel.B INVITE ------------> INVITE -------> INVITE <------- INVITE ---------------------------> 200 OK <--------------------------- 200 OK -------> 200 OK <------- 200 OK <------------ 1.How can i send from SS to B the INVITE with the location of the Tel.B without B thiks that is a new INVITE. 2.Then route the INVITE to the Tel.B What module can help me to set up this flow? Alejandro Sanchez Thank's in Advanced. ___________________________________________________________ Do You Yahoo!? La mejor conexión a Internet y <b >2GB</b> extra a tu correo por $100 al mes. http://net.yahoo.com.mx

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[Users] Newbie questions
by Saúl Ibarra 14 Nov '06

14 Nov '06

Hi I'm new at using OpenSER. I plan to use OpenSER with a SIP proxy (rtpproxy?, mediaproxy?) in order to handle SIP client registrations and connections behind NAT, and then I want to pass the control to an Asterisk server, in order to make the call over an IAX trunk. I've been reading quite a lot documentation on both SER and OpenSER, but I can't figure out how to do it properly. Is there an example or a guide I can check? Thanks in advance. -- Saúl -- "Some people say why, other just say, why not." http://saghul.wordpress.com/

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[Users] UAC module. CANCEL request handling.
by Leonid Fainshtein 14 Nov '06

14 Nov '06

Hi, I try to use the "UAC module" for replacing header field "From". I defined the automatic mode for "From" restore (modparam("uac","from_restore_mode","auto")) and change "From" only for INVITE requests. It looks like I also have to change manually header filed "From" also in CANCEL requests. It is because UAC (call originator) sends CANCEL without header "Route" and therefore OpenSer can't perform the automatic replacement. Do I miss anything? Is there a way to implement automatic header field "From" handling for the CANCEL requests? Thank you in advance, Leonid Fainshtein

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[Users]Can OpenSER uses same TCP session from the registration in some TLS case?
by Pat wang 14 Nov '06

14 Nov '06

Hi, I am working on the TLS with OpenSER and I found an something interesting. If an INVITE received by OpenSER with a Request-URI of the client AOR then OpenSER proxy the request via the same TCP session established during registration. If the INVITE has a Request-URI of the client's contact URI then OpenSER start a new TCP session. For example: One Eyebeam registers with AOR as: user1(a)tlstest.com and has a contact URI of user1(a)192.168.100.1. When an INVITE contains a RURI as user1(a)tlstest.com then OpenSER look up the location table and replace it with the user1(a)192.168.100.1 then proxy it via the same TCP session as the one during the registration. If an INVITE already has RURI as user1(a)192.168.100.1 then OpenSER proxy this INVITE by starting a new TCP session and Eyebeam does not accept the new session. The call will be failed in this case. I can't find any solution to fix this issue. Could someone know this issue please give some advice? Thanks, Patrick _________________________________________________________________ All-in-one security and maintenance for your PC. Get a free 90-day trial! http://clk.atdmt.com/MSN/go/msnnkwlo0050000002msn/direct/01/?href=http://ww…

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[Users] OpenSER not working after running for several days
by Papadopoulos Georgios 14 Nov '06

14 Nov '06

Hi Richard, This is a very interesting discussion. However I don't think our situations are similar. Just to make it clear, in your case, did OpenSER slow down or stop working at all? Was it something that happened gradually? In my case, it runs fine for several days. There are no delays, no slowing down of any kind. Then suddenly it stops responding. There is no script change, nothing at all. Actually most of the times it happened during the night when there is nobody around "playing" with DNS, network settings etc. So what could be causing this? Can I set up some sort of logging that will help find the cause of this? I have no clue what kind of information I should be logging. Most of my troubleshooting is done with ngrep. I could redirect this to a log file, but with the volume of traffic that we have, a few hours would be enough to fill the disk. 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.

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[Users] TCP REGISTER
by Russ Daigle 14 Nov '06

14 Nov '06

When I REGISTER for UDP, I receive subsequent requests for that user. However, when I perform a TCP REGISTER, even though the register returns 200 OK, no requests make it to the user. Is this supported? It should be. However, I noticed when you do a "openserctl ul show" command, it shows users, but it doens't indicate whether they are UDP or TCP registered. -Russ

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[Users] UAC's aren't hung up, Audio is fine in 2 way
by raviprakash sunkara 14 Nov '06

14 Nov '06

Hello Users, tried a lot , My UAC's are Not Hungupping in Behind the NAT/Router And also My sip proxy is also under the Behind NAT My SIP ip is 192.168.2.75 My Public IP is xx.xx.xx.xx I didn't get any Bye request to SIP server, except the Bye all the other Messages are Requested to MY SIP Server in openser.cfg i I'm using X-lites for testing in that RTP port 12222 NAT firewall ip is 192.168.2.1 Sip domain is xx.xx.xx.xx Sip proxy xx.xx.xx.xx ______________________________________________ modparam ("nathelper",rtpproxy_sock","udp: 192.168.2.75:12222" if(nat_uac_test("19")) { if(method=="REGISTER") { fix_nated_register(); force_rport(); setflag(6); } else { if(method=="INVITE") { fix_nated_sdp("3") }; fix_nated_contact(); }; }; -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 ravi.sunkara(a)hyperion-tech.com www.hyperion-tech.com

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[Serusers] retr_timer1p1
by Andrey Kuprianov 14 Nov '06

14 Nov '06

Hi Guys, Is retr_timer1p1 still valid for SER 0.9.6? Andrey.

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[Serusers] SER pre-release - Presence and NOTIFY
by SER LIST 14 Nov '06

14 Nov '06

Hi, I have SER Proxy and PRESENCE running on different machines. REGISTERS are being forwarded by SER Proxy to PRESENCE and PRESENCE is storing the location information in "location" db. What I am seeing is that when the PRESENCE Server sends NOTIFYs to watchers, it sends them to the SER PROXY (destination IP Is that of SER PROXY) but the URI in NOTIFY is that of the watchers LOCATION INFO. Is this correct? I would like the NOTIFY URI to contain the SIP DOMAIN and not the contact info. e.g. user 444 registers with SER PROXY (sip.xx.com). It is registering from 10.1.1.1:12321 (location info). SER PROXY sends the REGISTER TO Presence server which stores this info in location database (location is 444@10.1.1.1:12321). Now when Presence Server sends NOTIFY, it sends it to the SER PROXY IP BUT the NOTIFY URI IS: NOTIFY sip:444@10.1.1.1:12321 SIP/2.0 How can I have the NOTIFY URI to be : NOTIFY sip:444@sip.xx.com SIP/2.0 Any suggestions would be higlhy appreciated. _________________________________________________________________ Ready for the world's first international mobile film festival celebrating the creative potential of today's youth? Check out Mobile Jam Fest for your a chance to WIN $10,000! www.mobilejamfest.com

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[Serusers] Presence Client
by Ajay Kolambekar 14 Nov '06

14 Nov '06

Hi, To test the presence capability of SER which presence client application I can use? Regards, Ajay DISCLAIMER ========== This e-mail may contain privileged and confidential information which is the property of Persistent Systems Pvt. Ltd. It is intended only for the use of the individual or entity to which it is addressed. If you are not the intended recipient, you are not authorized to read, retain, copy, print, distribute or use this message. If you have received this communication in error, please notify the sender and delete all copies of this message. Persistent Systems Pvt. Ltd. does not accept any liability for virus infected mails.

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Re: [Serusers] Any resolution to the presence core dump?
by SER LIST 14 Nov '06

14 Nov '06

Yes. It seems to be fixed in the CVS version. I am running the CVS version since four days and I have not had a core dump. I am testing further and will post any issues I find. Thanks >From: Vaclav Kubart <vaclav.kubart(a)iptel.org> >To: SER LIST <sergrp(a)hotmail.com> >CC: serusers(a)iptel.org >Subject: Re: [Serusers] Any resolution to the presence core dump? >Date: Mon, 13 Nov 2006 14:25:16 +0100 > >It could be corrected in CVS version, but it is not sure. (There were >few bugs in presence snapshot 4.1 corrected in CVS.) > Vaclav > >On Sun, Nov 05, 2006 at 10:32:34PM -0500, SER LIST wrote: > > I was wondering whether a patch or a new release is available to the >issue > > of presence core dumping on a regular basis that was reported short >while > > back? Thanks in advance. > > > > >The version 0.10.99 of presence snapshot core dumps on a regular > > basis after running for a certain time. Doing a gdb shows that the core > > dump occurs at msg_queue_destroy: > > >#0 0x001b87f2 in msg_queue_destroy (q=0xb6293e98) > > > at /usr/local/src/ser-0.10.99-dev35-pa-4.1/lib/../fastlock.h:147 > > >147 asm volatile( > > > > > > > _________________________________________________________________ > > Not only does Windows Live? OneCare? provide all-in-one PC care to keep > > your computer protected and well-maintained, but it also makes creating > > backup files a breeze. Try it today! > > >http://ideas.live.com/programpage.aspx?versionid=b2456790-90e6-4d28-9219-5d… > > > > _______________________________________________ > > Serusers mailing list > > Serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers _________________________________________________________________ Not only does Windows Live OneCare provide all-in-one PC care to keep your computer protected and well-maintained, but it also makes creating backup files a breeze. Try it today! http://ideas.live.com/programpage.aspx?versionid=b2456790-90e6-4d28-9219-5d…

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[Serusers] Problem with SER modules, mediaproxy.so and domain.so
by jk 14 Nov '06

14 Nov '06

Hello, I am totally confused, Installed ser rpm on FC5, with all the modules, If I start ser with out mediaproxy.so and domain.so modules every thing works great. But When I add these two mosules I start getting in problem and it does not start. Here is the error, 1(0) DEBUG: init_mod_child (-1): tm 1(0) DEBUG: callid: '74fe3fe7-0(a)xxx.xxx.xxx.xxx' 1(0) DEBUG: init_mod_child (-1): usrloc 1(0) get_connection(): Connection not found in the pool 2(0) INFO: fifo process starting: 11768 2(0) DEBUG: init_mod_child (-2): tm 2(0) DEBUG: callid: '74fe3fe7-0(a)xxx.xxx.xxx.xxx' 2(0) DEBUG: init_mod_child (-2): usrloc 2(0) get_connection(): Connection not found in the pool 0(11766) DEBUG: init_mod_child (1): tm 0(11766) DEBUG: callid: '74fe3fe7-11766(a)xxx.xxx.xxx.xxx' 0(11766) DEBUG: init_mod_child (1): usrloc 0(11766) get_connection(): Connection not found in the pool 1(11767) DEBUG: init_mod_child (-1): auth_db 0(11766) DEBUG: init_mod_child (1): auth_db 1(11767) get_connection(): Connection found in the pool 0(11766) get_connection(): Connection found in the pool 1(11767) DEBUG: init_mod_child (-1): uri_db 0(11766) DEBUG: init_mod_child (1): uri_db 1(11767) get_connection(): Connection found in the pool 0(11766) get_connection(): Connection found in the pool 1(11767) DEBUG: init_mod_child (-1): domain 0(11766) 1(11767) DEBUG: init_mod_child (1): domain DEBUG: init_mod_child (-1): nathelper 0(11766) get_connection(): Connection not found in the pool 1(11767) DEBUG: init_mod_child (-1): avpops 1(11767) get_connection(): Connection found in the pool 1(11767) DEBUG: init_mod_child (-1): permissions 2(11768) DEBUG: init_mod_child (-2): auth_db 2(11768) get_connection(): Connection found in the pool 2(11768) DEBUG: init_mod_child (-2): uri_db 2(11768) get_connection(): Connection found in the pool 2(11768) DEBUG: init_mod_child (-2): domain 2(11768) DEBUG: init_mod_child (-2): nathelper 2(11768) DEBUG: init_mod_child (-2): avpops 2(11768) get_connection(): Connection found in the pool 2(11768) DEBUG: init_mod_child (-2): permissions 2(11768) get_connection(): Connection found in the pool * 0(11766) new_connection(): Access denied for user 'serro'@'localhost' (using password: YES) I have not idea why it is using serror@localhost to connect to db. I will post my config files (ser.cfg and serctl) below. * 0(11766) db_init(): Could not create a connection * 0(11766) ERROR:domain_db_init: cannot initialize database connection 0(11766) ERROR: domain:child_init(): Unable to connect to the database 0(11766) init_mod_child(): Error while initializing module domain 0(11766) main_dontfork: init_child failed 2(11768) INFO: signal 15 received 2(11768) Memory status (pkg):* 2(11768) 1(11767) INFO: signal 15 received 1(11767) Memory status (pkg): 1(11767) fm_status (0x811c800): 1(11767) heap size= 1048576 Top part of ser.cfg file [root@ ser]# more ser.cfg # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=5 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ #listen=192.168.2.201 listen=209.XXX.XXX.XXX check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 #children=4 fifo="/tmp/ser_fifo" a fifo_db_url="mysql://username:password@host/ser_db" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" loadmodule "/usr/lib/ser/modules/uri_db.so" loadmodule "/usr/lib/ser/modules/uri.so" loadmodule "/usr/lib/ser/modules/domain.so" loadmodule "/usr/lib/ser/modules/mediaproxy.so" loadmodule "/usr/lib/ser/modules/nathelper.so" loadmodule "/usr/lib/ser/modules/textops.so" loadmodule "/usr/lib/ser/modules/avpops.so" loadmodule "/usr/lib/ser/modules/permissions.so" #loadmodule "/usr/lib/ser/modules/domain.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/lib/ser/modules/auth.so" #loadmodule "/usr/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- modparam("auth_db|permissions|uri_db|usrloc", "db_url", "mysql://username:password@host/ser_db") modparam("auth_db", "calculate_ha1", 0) 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","sip_asymmetrics","/etc/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/etc/ser/rtp-clients") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) 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("avpops", "avp_url", "mysql://username:password@host/ser_db") modparam("avpops", "avp_table", "usr_preferences") # main routing logic route { SERCTL file # configuration for starting/stopping ser PID_FILE=/var/run/ser.pid SYSLOG=1 # 0=output to console, 1=output to syslog STARTOPTIONS= # for example -dddd DIR=`dirname $0` SERBIN=$DIR/ser AWK=awk # ser's FIFO server if [ -z "$SER_FIFO" ]; then SER_FIFO=/tmp/ser_fifo fi # period in which stats are reprinted if [ -z "$WATCH_PERIOD" ] ; then WATCH_PERIOD=2 fi # SQL config if [ -z "$SQL_DB" ] ; then SQL_DB=ser_DB fi if [ -z "$SQL_HOST" ] ; then SQL_HOST=SER_HOST //not localhost fi if [ -z "$SQL_USER" ] ; then SQL_USER=username fi # the read-only user for whom password may be stored here if [ -z "$RO_USER" ] ; then RO_USER=usename fi if [ -z "$RO_PW" ] ; then RO_PW=password fi # binaries GENHA1='gen_ha1' MYSQL='mysql' SER='sr' LAST_LINE='tail -1' # ACL name verification VERIFY_ACL=1 ACL_GROUPS="local ld int voicemail free-pstn prepaid" VERSION='$Revision: 1.83.2.2 $' NOTE: If I dont add mediaproxy.so and domain.so then everything works lgreate Ser verision [root@ ser]# rpm -qa | grep ser- ser-serweb-0.9.6-6.fc5 ///This also did not work, problem with classkit and runkit extension. If any one could make it work on FC5 with PHP 5.1, Please share. ser-mysql-0.9.6-6.fc5 ser-0.9.6-6.fc5 I am tired like dead, Any idea will be help. Thank you, -JK

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[Serusers] Please help with radius-ser mysql setup
by Lokesh Kumar 13 Nov '06

13 Nov '06

Hello List, I had setup ser-0.9.6 and freeradius-1.1.3. I am able to authenticate sip users via radius using entries in mysql database, but unable to do group membership checking with mysql database, but if I use users text file for group checking it works fine. The mentioned below link have the logs and entries in mysql database. HYPERLINK "http://pastebin.ca/247613"http://pastebin.ca/247613 As Mr. Greger suggested in previous post not to challenge options message (second message) , for that I included one line in users file like @mydomain.pt NAS-Port == 0, NAS-IP-Address == "127.0.0.1", Auth-Type := Accept And authentication works fine, is it right way to avoid challenging 2 options, or is it creating problem in group membership checking. My radiusd.conf file has sql and files entries under authorize section in the order of sql files Please suggest something. Thanks you very much Lokesh -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.409 / Virus Database: 268.14.3/531 - Release Date: 11/12/2006

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[Serusers] Help needed with radius-mysql and group authorization checking
by Lokesh Kumar 13 Nov '06

13 Nov '06

Hello List, I have problem with the group membership checking of ser calls with radius using mysql module. Well group checking is working fine with the users file but when I try to do that checking with mysql entries it always getting failed. Below mentioned is section of my users file 211069020(a)mydomain.pt Sip-Group == "int", Auth-Type := Accept Reply-Message = "Authorized" What should be the entries corresponding to the above mentioned in mysql tables of radgroupcheck and radgroupreply. I am using freeradius-1.1.3 with default values in sql.conf file. Here are the table entries of mysql. mysql> select * from radcheck; +----+----------------------------+-----------+----+-----------+ | id | UserName | Attribute | op | Value | +----+----------------------------+-----------+----+-----------+ | 2 | 211069020(a)mydomain.pt | Password | == | 211069020 | | 33 | 211069020(a)mydomain.pt | Auth-Type | := | Digest | +----+----------------------------+-----------+----+-----------+ 2 rows in set (0.00 sec) My sip clients can authenticate with mysql entries but can not do group authorization. Please suggest something Thanks very much Lokesh -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.409 / Virus Database: 268.14.3/531 - Release Date: 11/12/2006

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[Users] The Summit picx are So Good,
by raviprakash sunkara 13 Nov '06

13 Nov '06

Hello Users , Thanks to *Bogdan-Andrei Iancu* *Daniel-Constantin Mierla* *Dmitry Isakbayev* *Elena-Ramona Modroiu* *Glenn Dalgliesh* *Klaus Darilion* *Mike Williams* *Norman Brandinger I'M very glad to see in U in Summit pics on Nov 7-9 , * -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 ravi.sunkara(a)hyperion-tech.com www.hyperion-tech.com

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[Serusers] SER IPV6
by ali daher 13 Nov '06

13 Nov '06

Hi all I am new with Ser i installed Ser 0.9.3 on a linux suse 10.1 machine i have two clients running supporting ipv6 . the 2 clients and the linux are connected through a switch. i want to send SIP messages between clients over IPV6. i can not register clients on SER using the IPV6. i was told to change the ser.cfg file daoes anyone have idea how to make ser work on ipv6, and support SIP messages on IPV6? thanks in advance for your help regards _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

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[Users] OpenSER Summit: photos and slides
by Daniel-Constantin Mierla 13 Nov '06

13 Nov '06

Hello, first photo album from the OpenSER summit days is available at: http://www.openser.org/events/2006-OpenSER-Summit/photos/Nov.7-8/ Also, the slides are collected in their last version and uploaded to: http://www.openser.org/events/2006-OpenSER-Summit/slides/ We like to thank to all people attending the event, the number of participants was a pleasant surprise even for VoN organizers. During Wednesday we were able to meet lot of community members and listen to very interesting speeches. Since the slides do not reflect entirely the presentation, in the near future we will post comments about each one showed there. We are still collecting photos, so expect more in the next days, maybe some videos as well. The event was more focused on OpenSER capabilities and business cases, rather than technical discussions. This leaded to a frequent question: when a technical meeting will be hold? We started investigation for time and place, someone already committed to host the event, proposals are very welcome. On the other hand, we got lot of emails asking about similar events in other places of the world, traveling between continents not being very convenient -- we look to do such meetings as well, further details will follow on the mailing lists. Cheers, Daniel

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[Serusers] Pseudo-variables
by Benito Caracuel 13 Nov '06

13 Nov '06

Hello All, I would like to do: exec_msg("cat /tmp/test.txt > %ci"); But %ci has always the same value!! How Can I change the value of a pseudo-variable? Thank you. Ben, _________________________________________________________________ Prueba algunos de los nuevos servicios en línea que te ofrece Windows Live Ideas: tan nuevos que ni siquiera se han publicado oficialmente todavía. http://ideas.live.com

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[Serusers] (no subject)
by ali daher 13 Nov '06

13 Nov '06

Hi I want to post to the list my email is "daher_ali(a)hotmail.com" regards _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

1 0

[Serusers] added sipgrep utility
by Vaclav Kubart 13 Nov '06

13 Nov '06

There was added sipgrep utility written by Alexandr Dubovikov <shurik(a)start4.info> into SER's source tree (utils/sipgrep). It is a wrapper on ngrep which * filters SIP messages according number in To or From header fields * displays From tag, To tag, Call-ID and branch in different colours, thus it is possible to trace dialogs or transactions by "one look into message" ... this is really great! ;-) * It can store received messages into a file and show them (together) For a bit more info see README. For other info contact author. Vaclav

1 0

[Serusers] Answering with music?
by Evert 13 Nov '06

13 Nov '06

Hi all! For testing purposes I want certain extensions being answered with streamed music. I guess this requires certain routing in SER (and perhaps a SIP client?) Has anyone done something similar before? Where/how should I start? Regards, Evert

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