- sr-users - kamailio.org

[Serusers] Contact in 200 OK response to REGISTER
by Stefan Sayer 27 Nov '06

27 Nov '06

Hi, registrar at iptel.org (0.10.99-iptelorg/tekelec-SOPv2) and ser 0.9.6 as well do not return a contact header in 200 response to REGISTER (see trace below), but in rfc (10.3) I read it's a MUST 8. The registrar returns a 200 (OK) response. The response MUST contain Contact header field values enumerating all current bindings. Each Contact value MUST feature an "expires" parameter indicating its expiration interval chosen by the registrar. The response SHOULD include a Date header field. Am I missing something (apart from the contact header) or is my register message not correct? Stefan U 192.168.5.101:5070 -> 213.192.59.75:5060 REGISTER sip:iptel.org SIP/2.0. Via: SIP/2.0/UDP 192.168.5.101:5070;branch=z9hG4bK15f6.4d8faeb.0. To: Stefan <sip:sayer@iptel.org>. From: Stefan <sip:sayer@iptel.org>;tag=0E520893-456AF6AD0009055B-B7963B90. CSeq: 10 REGISTER. Call-ID: 0D1543AB-456AF6AD00090738-B7C66B90(a)192.168.5.101. Content-Length: 0. User-Agent: Sip EXpress router(0.9.6-sems (i386/linux)). Contact: <sip:sayer@192.168.5.101:5070>. Expires: 0. . # U 213.192.59.75:5060 -> 192.168.5.101:5070 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 192.168.10.144:5070;rport=5070;received=91.64.139.240;branch=z9hG4bK15f6.4d8faeb.0. To: Stefan <sip:sayer@iptel.org>;tag=17da4c1f77cf571798ae2c71e6d4b5c0.4003. From: Stefan <sip:sayer@iptel.org>;tag=0E520893-456AF6AD0009055B-B7963B90. CSeq: 10 REGISTER. Call-ID: 0D1543AB-456AF6AD00090738-B7C66B90(a)192.168.5.101. P-Nat: Yes. WWW-Authenticate: Digest realm="iptel.org", nonce="456af887c3eac5c30c87bab4268389ee5e796372". Server: Sip EXpress router (0.10.99-iptelorg/tekelec-SOPv2 (i386/linux)). Content-Length: 0. Warning: 392 213.192.59.77:5070 "Noisy feedback tells: pid=4141 req_src_ip=213.192.59.75 req_src_port=5060 in_uri=sip:iptel.org out_uri=sip:iptel.org via_cnt==2". . # U 192.168.5.101:5070 -> 213.192.59.75:5060 REGISTER sip:iptel.org SIP/2.0. Via: SIP/2.0/UDP 192.168.5.101:5070;branch=z9hG4bK25f6.dae56b14.0. To: Stefan <sip:sayer@iptel.org>. From: Stefan <sip:sayer@iptel.org>;tag=0E520893-456AF6AD0009055B-B7963B90. CSeq: 11 REGISTER. Call-ID: 0D1543AB-456AF6AD00090738-B7C66B90(a)192.168.5.101. Content-Length: 0. User-Agent: Sip EXpress router(0.9.6-sems (i386/linux)). Contact: <sip:sayer@192.168.5.101:5070>. Expires: 0. Authorization: Digest username="sayer", realm="iptel.org", nonce="456af887c3eac5c30c87bab4268389ee5e796372", uri="sip:iptel.org", response="880ff39073b5eeb3cda68352b78de368", algorithm="MD5". . # U 213.192.59.75:5060 -> 192.168.5.101:5070 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.5.101:5070;rport=5070;received=91.64.139.240;branch=z9hG4bK25f6.dae56b14.0. To: Stefan <sip:sayer@iptel.org>;tag=17da4c1f77cf571798ae2c71e6d4b5c0.4003. From: Stefan <sip:sayer@iptel.org>;tag=0E520893-456AF6AD0009055B-B7963B90. CSeq: 11 REGISTER. Call-ID: 0D1543AB-456AF6AD00090738-B7C66B90(a)192.168.5.101. P-Nat: Yes. Server: Sip EXpress router (0.10.99-iptelorg/tekelec-SOPv2 (i386/linux)). Content-Length: 0. Warning: 392 213.192.59.77:5070 "Noisy feedback tells: pid=4140 req_src_ip=213.192.59.75 req_src_port=5060 in_uri=sip:iptel.org out_uri=sip:iptel.org via_cnt==2". . -- Stefan Sayer Media Services Development iptego GmbH Am Borsigturm 40 13507 Berlin Germany stefan.sayer(a)iptego.de www.iptego.de

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[Users] avp_subst re problem
by Andreas Heise 27 Nov '06

27 Nov '06

Hello, I tried to use avp_subst to extract the part <......> of a Remote-Party-ID $avp(s:merker)= Remote-Party-ID: "test"<sip:12345@localhost>;party=calling;id-type=subscriber;screen=yes;privacy=full avp_subst("$avp(s:merker)/$avp(s:merker1)", "/(.*)>(.*)/\1>/"); the result for merker1 is Remote-Party-ID: "test"<sip:12345@localhost> what must I change in the re to get this result? <sip:12345@localhost> thanks in advance, Andreas

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[Serusers] Call Transfers in SER + Asterisk architecture
by Walter Rodrigues Filho 26 Nov '06

26 Nov '06

Ricardo - this is an Asterisk issue. Configure your SIP.conf in * creating a sip peer with insecure=very fotr the SER peer, like this: [ser-peer] type = friend ; should you want to receive and make calls to SER context=from-ser ; the context in dialplan with extensions allowed to be accessed by SER. Here you must have PSTN extensions capability. host=200.XXX.XXX.XXX ; the IP address for SER fromdomain=domain.com ;the Domain part of uri to be verified by asterisk on the INVITE received by SER. qualify= yes ; just to check the latency between SER and Asterisk (like this, if over 2000ms Ast will report as unavailable peer). disallow=all allow=alaw allow=g729 insecure= very ; this line garantees that any username part of Request URI sent by SER in INVITE to Asterisk will be accepted by Ast and routed to the dialplan. So, if SER send an INVITE to HYPERLINK "mailto:5531332818847@domain.com.br"5531332818847(a)domain.com.br , Asterisk will look for a section of type =user in SIP conf to match the user part first, it won't find, then it will look for a type=peer. It will find and try to match the IP address as in host= ...line and will accept any username part as per insecure=very line. Then, if context=from-ser in the Ast dialplan allows this dialing string (553132818847), it will proceed from there. Hope it helps. At. Walter -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.430 / Virus Database: 268.14.16/552 - Release Date: 26/11/2006 11:30

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[Users] Improper relaying of Invite requests
by Kaushik Kumar 26 Nov '06

26 Nov '06

Hi, I am working on a project which involves testing SER with some dummy clients. I have downloaded and installed OpenSER. I am using the very basic Hello.cfg configuration file available online. I find that when SER gets a INVITE request from the caller, it does not deliver it properly to the client (callee). Some debugging revealed that it relays the request to port 5060 at the client rather than the one which was registered by the callee earlier. I tried to look at the source code of SER and found that the callee's contact information never seem to be looked up at any stage. Is this due to some missing information in my configuration file? Any feedback would be appreciated. Thank you -Kaushik PS: I have attached (inline) the configuration file ser.cfg ------------------------------------ debug=9 fork=yes log_stderror=yes listen=128.42.3.3 port=5060 children=4 check_via=no # added dns=no rev_dns=no fifo="/tmp/ser_fifo" loadmodule "/scratch/ser-0.9.7-pre1/modules/sl/sl.so" # Stateless replies loadmodule "/scratch/ser-0.9.7-pre1/modules/tm/tm.so" # Transaction management loadmodule "/scratch/ser-0.9.7-pre1/modules/rr/rr.so" # Routing and record-routing loadmodule "/scratch/ser-0.9.7-pre1/modules/maxfwd/maxfwd.so" # Max fowards check loadmodule "/scratch/ser-0.9.7-pre1/modules/usrloc/usrloc.so" # User location support loadmodule "/scratch/ser-0.9.7-pre1/modules/registrar/registrar.so" # Registrar (requires userloc and sl) loadmodule "/scratch/ser-0.9.7-pre1/modules/textops/textops.so" # Message textual operations modparam("usrloc", "db_mode", 0) # 0 forces registration data to be stored in memory modparam("rr", "enable_full_lr", 1) 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!="REGISTER") { record_route(); }; # ------------------------------------------------------------------------ # Loose Route Section # ------------------------------------------------------------------------ if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; # ------------------------------------------------------------------------ # Call Type Processing Section # ------------------------------------------------------------------------ if (uri!=myself) { append_hf("P-hint: outbound\r\n"); route(1); break; }; if (method=="ACK") { route(1); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; append_hf("P-hint: userloc applied\r\n"); route(1); } route[1] { # ------------------------------------------------------------------------ # Default Message Handler # ------------------------------------------------------------------------ if (!t_relay()) { log(2, "Failed t_relay"); sl_reply_error(); }; } route[2] { # ------------------------------------------------------------------------ # REGISTER Message Handler # ------------------------------------------------------------------------ if (!save("location")) { sl_reply_error(); }; }

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[Serusers] Analyse calling ID
by inge 26 Nov '06

26 Nov '06

Hi all, What sort of if structure can I use to test the Calling ID ? (ie : my calling id are member of subscriber table). For the moment I do something like this : "if(uri=~"^sip:119@") {" for the destination number, but I want to do the same for the calling ID. Thanks for your help. Best regards

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[Serusers]How to remove sip header field?
by Ren-Huang Liou 26 Nov '06

26 Nov '06

Dear all, I already know that there is a “remove_hf_f” function in the textops module. I can use this function to remove sip header fields which are supported by SER such as “From”, “To”, and “Expires” header fields, but it can’t remove sip header fields which are not support by SER such as “SIP-If-Match” and “SIP-ETag”. When I use “sl_send_reply” function to build and send sip response, this response contains some sip headers field which I don’t like. How can I remove unnecessary header fields? Thanks a lot, Ren-Huang Liou

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[Serusers] message_store.php problem
by Meidiana Ten 26 Nov '06

26 Nov '06

Hi, I've solve the problem of DNS so now my serweb can send IM successfully through SERWeb. But when I need to see the IM from message store tab on SERWeb, it shows nothing (only blank page on the tab). Below is my message_store.php: <? /* * $Id: message_store.php,v 1.3 2003/10/13 19:56:43 kozlik Exp $ */ require "prepend.php"; require "/var/www/phplib/oohforms.inc"; put_headers(); page_open (array("sess" => "phplib_Session", "auth" => "phplib_Auth")); class Cinst_mess{ var $mid, $src_addr, $time, $body; function Cinst_mess($mid, $src_addr, $time, $body){ $this->mid=$mid; $this->src_addr=$src_addr; $this->time=$time; $this->body=$body; } } class Cvoice_mess{ var $mid, $src_addr, $time, $subject, $file; function Cvoice_mess($mid, $src_addr, $time, $subject, $file){ $this->mid=$mid; $this->src_addr=$src_addr; $this->time=$time; $this->subject=$subject; $this->file=$file; } } do{ $db = connect_to_db(); if (!$db){ $errors[]="can't connect to sql server"; break;} if (isset($dele_im)){ $q="select from ".$config->table_message_silo." where mid=$dele_im and r_uri like 'sip:".$auth->auth["uname"]."@".$config->default_domain."%'"; $res=mySQL_query($q); if (!$res) {$errors[]="error in SQL query, line: ".__LINE__;} Header("Location: ".$sess->url("message_store.php?kvrk=".uniqID(""))); page_close(); exit; } if (isset($dele_vm)){ $q="select file from ".$config->table_voice_silo." where mid=".$dele_vm." and r_uri like 'sip:".$auth->auth["uname"]."@".$config->default_domain."%'"; $res=mySQL_query($q); if (!$res) {$errors[]="error in SQL query, line: ".__LINE__; break;} if (!MySQL_num_rows($res)) {$errors[]="Message not found or you haven't access to message"; break;} $row=MySQL_Fetch_Object($res); @unlink($config->voice_silo_dir.$row->file); $q="delete from ".$config->table_voice_silo." where mid=$dele_vm"; $res=mySQL_query($q); if (!$res) {$errors[]="error in SQL query, line: ".__LINE__;} Header("Location: ".$sess->url("message_store.php?kvrk=".uniqID(""))); page_close(); exit; } }while(false); do{ // if ($db){ if ($config->show_message_silo) { $q="select mid, src_addr, inc_time, body from ". $config->table_message_silo. " where r_uri like 'sip:".$auth->auth["uname"]. "@".$config->default_domain."%'"; $im_res=mySQL_query($q); if (!$im_res) {$errors[]="error in SQL query, line: ".__LINE__; break;} while ($row=MySQL_Fetch_Object($im_res)){ $im_arr[]=new Cinst_mess($row->mid, $row->src_addr, $row->inc_time, $row->body); } if ($config->show_voice_silo) { $q="select mid, src_addr, inc_time, subject, file from ". $config->table_voice_silo." where r_uri like 'sip:". $auth->auth["uname"]."@".$config->default_domain."%'"; $vm_res=mySQL_query($q); if (!$vm_res) { $errors[]="error in SQL query, line: ".__LINE__; break; } while ($row=MySQL_Fetch_Object($vm_res)){ $vm_arr[]=new Cvoice_mess($row->mid, $row->src_addr, $row->inc_time, $row->subject, $row->file); } } set_timezone($errors); } }while (false); ?> <!doctype html public "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> <title><?echo $config->title;?></title> <?print_html_head();?> <style type="text/css">  </style> </head> <? print_html_body_begin(7, true, true, get_user_name($errors)); echo "<br>"; print_errors($errors); // Display error print_message($message); ?> <table border="0" cellspacing="0" cellpadding="0" align="center"> <tr><td class="title" width="502">Instant messages store:</td></tr> </table><br> <?if (isset($im_arr)){?> <table border="0" cellspacing="0" cellpadding="0" align="center"> <?foreach($im_arr as $row){ if (date('Y-m-d',$row->time)==date('Y-m-d')) $time="today ".date('H:i',$row->time); else $time=date('Y-m-d H:i',$row->time) ?> <tr><td> <table border="0" cellspacing="0" cellpadding="2" bgcolor="#C1D773"> <tr><td> <table border="0" cellpadding="0" cellspacing="0" bgcolor="#FFFFFF"> <tr bgcolor="#B1C9DC" valign="top"> <td width="245" class="msgtitle"><?echo $row->src_addr;?></td> <td align="right" width="135" class="msgtitle"><?echo $time;?></td> <td align="center" width="60" class="msgtitle"><a href="<?$sess->purl("send_im.php?kvrk=".uniqid("")."&sip_addr=".rawURLEncode($row->src_addr));?>" class="msgtitle">reply</a></td> <td align="center" width="60" class="msgtitle"><a href="<?$sess->purl("message_store.php?kvrk=".uniqid("")."&dele_im=".rawURLEncode($row->mid));?>" class="msgtitle">delete</a></td> </tr> <tr> <td colspan="4" width="500"><?echo $row->body;?></td> </tr> </table> </td></tr> </table> </td></tr> <tr><td> </td></tr> <?}?> </table> <?}else{?> <div align="center">No stored instant messages</div> <br> <?}?> <?if ($default->show_voice_silo) {?> <table border="0" cellspacing="0" cellpadding="0" align="center"> <tr><td class="title" width="502">Voicemail messages store:</td></tr> </table><br> <?if (isset($vm_arr)){?> <table border="0" cellpadding="2" cellspacing="0" bgcolor="#C1D773" align="center"> <tr><td> <table border="0" cellpadding="0" cellspacing="0" bgcolor="#FFFFFF" align="center"> <tr> <td class="titleT" width="139">calling subscriber</td> <td width="2" bgcolor="#C1D773"><img src="<?echo $config->img_src_path;?>title/green_pixel.gif" width="2" height="2"></td> <td class="titleT" width="160">subject</td> <td width="2" bgcolor="#C1D773"><img src="<?echo $config->img_src_path;?>title/green_pixel.gif" width="2" height="2"></td> <td class="titleT" width="135">time</td> <td width="2" bgcolor="#C1D773"><img src="<?echo $config->img_src_path;?>title/green_pixel.gif" width="2" height="2"></td> <td class="titleT" width="60"> </td> </tr> <tr><td colspan="7" height="2" bgcolor="#C1D773"><img src="<?echo $config->img_src_path;?>title/green_pixel.gif" width="2" height="2"></td></tr> <?foreach($vm_arr as $row){ if (date('Y-m-d',$row->time)==date('Y-m-d')) $time="today ".date('H:i',$row->time); else $time=date('Y-m-d H:i',$row->time) ?> <tr valign="top"> <td align="left" class="f12" width="139"><a href="<?$sess->purl("send_im.php?kvrk=".uniqid("")."&sip_addr=".rawURLEncode($row->src_addr));?>"><?echo htmlspecialchars($row->src_addr);?></a></td> <td width="2" bgcolor="#C1D773"><img src="<?echo $config->img_src_path;?>title/green_pixel.gif" width="2" height="2"></td> <td align="left" class="f12" width="160"><a href="<?$sess->purl("ms_get_v_msg.php?kvrk=".uniqid("")."&mid=".rawURLEncode($row->mid));?>"><?echo $row->subject;?></a></td> <td width="2" bgcolor="#C1D773"><img src="<?echo $config->img_src_path;?>title/green_pixel.gif" width="2" height="2"></td> <td align="center" class="f12" width="135"><?echo $time;?></td> <td width="2" bgcolor="#C1D773"><img src="<?echo $config->img_src_path;?>title/green_pixel.gif" width="2" height="2"></td> <td align="center" class="f12" width="60"><a href="<?$sess->purl("message_store.php?kvrk=".uniqid("")."&dele_vm=".rawURLEncode($row->mid));?>" class="msgtitle">delete</a></td> </tr> <?}?> </table> </td></tr> </table> <?}else{?> <div align="center">No stored voicemail messages</div> <?}?> <?}?> <br> <?print_html_body_end();?> </html> <?page_close();?> Please tell me what's the problem. Thanx Regards, Meidiana --------------------------------- Everyone is raving about the all-new Yahoo! Mail beta.

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[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

1 0

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

1 0

[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

1 0

[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

2 2

[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.

2 1

[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

2 1

[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.-

2 1

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 >

1 0

[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

2 1

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

24 Nov '06

aaa

1 0

[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

3 3

[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

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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

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[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/

3 2

[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.

6 6

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

2 1

[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 ?

2 4

[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 >

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[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

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[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

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[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.

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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.

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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>

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[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

2 1

[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/

2 1

[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

2 1

[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…

3 4

[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.

2 1

[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

2 1

[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

1 0

[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.

2 2

[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/

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[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

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[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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[Serusers] Any resolution to the presence core dump?
by SER LIST 13 Nov '06

13 Nov '06

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…

3 2

[Serusers] XCAP server
by Jayantheesh Srimushnam - TLS, Chennai 13 Nov '06

13 Nov '06

Hi All, I have downloaded the SER and trying to run XCAP server. I have gone through the README file for the XCAP server and understood that it is server simulation. Still I didn't get how to install and run it. Could anyone please through some light on this? Thanks, Jayantheesh DISCLAIMER The contents of this e-mail and any attachment(s) are confidential and intended for the named recipient(s) only. It shall not attach any liability on the originator or HCL or its affiliates. Any views or opinions presented in this email are solely those of the author and may not necessarily reflect the opinions of HCL or its affiliates. Any form of reproduction, dissemination, copying, disclosure, modification, distribution and / or publication of this message without the prior written consent of the author of this e-mail is strictly prohibited. If you have received this email in error please delete it and notify the sender immediately. Before opening any mail and attachments please check them for viruses and defect.

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[Serusers] Registering SIP UA from the internet
by "tzieleniewski" 13 Nov '06

13 Nov '06

Hi All!! According to the SER -Getting Started Manual SER is located in the local network and has an local IP adres. Is it some how possible to register the SIP UA from the outside of the local net?? I mean can I register them without giving SER an public IP adress?? If yes then what do I have to specify (what adress) in the softphones configuration in the SIP Proxy section?? Thanks 4 any help! Bests

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[Serusers] Change SDP
by Benito Caracuel 13 Nov '06

13 Nov '06

Hello All, Basic Concept: I have to change the Path MSRP into SDP field in an INVITE and relay the new INVITE. SDP EXAMPLE: v=0 o=686873200 3372497915 3372497916 IN IP4 80.27.211.205 s=SipSession c=IN IP4 80.27.211.205 t=0 0 m=message 49152 TCP/MSRP * a=direction:active a=accept-types:text/plain a=path:msrp://80.27.211.205:49152/63331682315468750;tcp When I receive the SDP, I have to change "msrp://80.27.211.205:49152", so I execute an script and I obtain the new SDP into file.txt. Now, I need to replace the old msrp path by the new sdp path, which is into file.txt In my ser.cfg: if (method=="INVITE") { exec_msg("printenv SRCIP > /tmp/invite.txt; cat >> /tmp/invite.txt; /bin/bash /usr/local/ser_snapshot/sbin/controlador"); #I have the new SDP in /tmp/test.txt exec_msg("set `cat /tmp/test.txt`; %var = $1"); #I have the new SDP in $var replace_all("msrp:\/\/[0-9\.:]{2,20}","%$var"); # I don't know why $var = null!!!! I need that $var = cat /tmp/test.txt but i don't know how to do this!!!!! t_relay(); break; }; Please help me, Thank you very much. Ben. _________________________________________________________________ Busca desde cualquier página Web con una protección excepcional. Consigue la Barra de herramientas de Windows Live hoy mismo y GRATUITAMENTE. http://www.toolbar.live.com

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[Serusers] Presence - PA AND RLS
by SER LIST 13 Nov '06

13 Nov '06

Hi, I am not sure whether PA module access the XCAP server or not? I have users presence rules configured on XCAP server, but all my clients are getting a NOTIFY w/ pending status. What function call in PA accesses the XCAP server? It seems to me that the RLS accesses the XCAP. Thanks in advance _________________________________________________________________ Buy, Load, Play. The new Sympatico / MSN Music Store works seamlessly with Windows Media Player. Just Click PLAY. http://musicstore.sympatico.msn.ca/content/viewer.aspx?cid=SMS_Sept192006

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[Serusers] Presence / Eyebeam Question
by SER LIST 13 Nov '06

13 Nov '06

I have the following issue: I have two clients (eyebeam softphones). If I modparam(pa, auth, none) then I can see the status of each other. But when modparam(pa, auth, xcap) I cannot see each other's status. I do get a POPUP window on each softphone asking me whether I want to allow the other party to see my status. Even after I say "Allow", I still can't see the status. I am able to write the presence-rules to XCAP server from the eyebeam client. Eyebeam writes the file presrules.xml and rls looks for presence-rules.xml. So I have a softlink between the two files. Do I need to do something to make this work? Thanks _________________________________________________________________ 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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[Users] exec module : why not in onreply_route ?
by Jean-François SMIGIELSKI 13 Nov '06

13 Nov '06

Hello, On 11/13/06 Daniel-Constantin Mierla wrote: [...] > basically should work in onreply route as well, I haven't tried it. You > will not have access to transaction's avps, otherwise you can play and > see if you encounter any problem. Just edit the modules/exec/exec_mod.c, > add ONREPLY_ROUTE to exported functions' flags, recompile and reinstall. > If everything is ok, then we can include it in the trunk. [...] That is what I did last week and till now, I did not *yet* meet any problem with that. The exec module is a really convenient way to test various functionnalities without having to write a custom module. Very useful even in a reply block. Regards, JF Smigielski. ________________________________________________________________________ iBELGIQUE, exprimez-vous ! http://web.ibelgique.com/

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[Users] update documentation
by Klaus Darilion 13 Nov '06

13 Nov '06

Hi Daniel,Bogdan! Please update the 1.2.x documentation (http://openser.org/docs/modules/1.2.x/): - add domainpolicy module - add new ENUM module features thanks Klaus -- Klaus Darilion nic.at

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[Users] Errors whie doing openser integration with radius server
by Sanjeev Manoli 13 Nov '06

13 Nov '06

Hi, I am getting following errors as seen in /var/log/message while running openser with radius integration, ***************************************************************************************** Nov 6 20:33:45 lx-dev monit[13565]: 'openser' start: /etc/init.d/openser Nov 6 20:33:45 lx-dev monit[13565]: 'openser' failed to start Nov 6 20:33:45 lx-dev openser: init_tcp: using epoll_lt as the io watch method (auto detected) Nov 6 20:33:45 lx-dev openser: INFO: statistics manager successfully initialized Nov 6 20:33:45 lx-dev openser: StateLess module - initializing Nov 6 20:33:45 lx-dev openser: TM - initializing... Nov 6 20:33:46 lx-dev openser: Maxfwd module- initializing Nov 6 20:33:46 lx-dev openser: AVPops - initializing Nov 6 20:33:46 lx-dev openser: TextOPS - initializing Nov 6 20:33:46 lx-dev openser: ACC - initializing Nov 6 20:33:46 lx-dev openser: AUTH module - initializing Nov 6 20:33:46 lx-dev openser: xl_parse_item: error - bad parameters Nov 6 20:33:46 lx-dev openser: ERROR:avpops:fixup_check_avp: unable to get pseudo-variable in P1 Nov 6 20:33:46 lx-dev openser: ERROR: fix_actions: fixing failed (code=-2) at cfg line 146 Nov 6 20:33:46 lx-dev openser: ERROR: fix_expr : fix_actions error ***************************************************************************************** I am using openser (Version: openser-1.1.0-tls) and radius server ( freeradius-1.1.3) along with radiusclient-ng (radiusclient-ng-0.5.2). I exactly followed the following radius integration documentation from openser web site, http://openser.org/docs/openser-radius-1.0.x.html For your reference I am also attaching the openser.cfg file that I am using. If I remove the radius integration related part from openser.cfg then my openser server starts fine, I have tested it with kphone SIP UA and it works fine. One other question is I get parse error for following 2 statements in openser.cfg (I commented them to make forward progress). Please advice the right syntax to use following modparam statements. modparam("acc", "radius_extra", "Sip-Src-IP=$si;Sip-Src-Port=$sp") modparam("avpops", "avp_aliases", "day=i:101;time=i:102") This is kind of urgent for me and I am clueless at this point so really appreciate all your help. Thanks, - San

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[Users] which is better: avp_db_load with scheme or avp_db_query?
by Alan Crosswell 13 Nov '06

13 Nov '06

Hi, I want to pull the First+Last names of the subscriber for the $ru so I can stick an RPID in a 180 or 183 response (Polycom phones display the called party name when you do this). Is it better practice to define two schemes and use avp_db_load to get the first and last name columns out of subscriber table or to use avp_db_query and do it in one shot? Thanks. /a

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[Serusers] Asterisk + SIP integration for SIPLE. Newb needs advice.
by Ryan Churches 13 Nov '06

13 Nov '06

Ive administrate several Asterisk servers, and sometimes one company will have multiple servers with their own respective users. Unfortunately users of one server are unable to see the presence state of a user on another server. I understand that SER can (a) be my presence agent or (b) can pass SUBSCRIBErs to a respective asterisk box, but doesnt that result in (a) clients subscribing to SER cannot see the presence states of local channels on asterisk (i.e. a parking spot), or (b) clients not being able to see the presence of clients SUBSCRIBing to a different asterisk server (back to square 1)? Can someone help me wrap my mind around how this could/should work. Perhaps it can't be made to work at all. All i'm looking for is a general direction to research in. The question as best i can phrase it susinctly is "How can I use SER to allow clients of disparate asterisk servers on disparate lans to have a centralized presence roster?"

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[Users] CANCEL/ACK processing
by Mark Kent 13 Nov '06

13 Nov '06

Observant readers will recognize from the following that this question is related to earlier posts from me... that is: there is something fundamental that I just don't get. If someone is eager to answer all my questions, then please search ahead for the question marks (?). Also, what is the bigger picture that I am missing? Please use as many words as necessary to explain. Thanks, -mark The players: CUSTOMER-IP: the UserAgent OPENSER-IP: my openser box OPENSER.FQDN: the fully qualified domain name of my openser box PSTNGW-IP: the PSTN gateway My openser box gets this CANCEL: U 2006/11/06 18:40:10.689785 CUSTOMER-IP:5060 -> OPENSER-IP:5060 CANCEL sip:011445551212@OPENSER.FQDN:5060 SIP/2.0 From: <sip:CUST-PHONE-NUMBER@CUSTOMER-IP:5060;transport=UDP>;tag=TAG To: <sip:011445551212@OPENSER.FQDN:5060;transport=UDP> Call-ID: CALLID@CUSTOMER-IP CSeq: 2 CANCEL Via: SIP/2.0/UDP CUSTOMER-IP:5060;branch=BRANCH Max-Forwards: 70 Supported: 100rel,replaces Content-Length: 0 It ends up in my standard stateful relay handler where I t_relay() it (should I?) and this gets generated: # U 2006/11/06 18:40:10.691301 OPENSER-IP:5060 -> PSTNGW-IP:5060 CANCEL sip:445551212@PSTNGW-IP;user=phone SIP/2.0 Record-Route: <sip:011445551212@OPENSER-IP;lr=on;ftag=TAG> From: <sip:CUST-PHONE-NUMBER@CUSTOMER-IP:5060;transport=UDP>;tag=TAG To: <sip:011445551212@OPENSER.FQDN:5060;transport=UDP> Call-ID: CALLID@CUSTOMER-IP CSeq: 2 CANCEL Via: SIP/2.0/UDP OPENSER-IP;branch=BRANCH2.0 Via: SIP/2.0/UDP CUSTOMER-IP:5060;branch=BRANCH Max-Forwards: 69 Supported: 100rel,replaces Content-Length: 0 and this also happens: # U 2006/11/06 18:40:10.691342 OPENSER-IP:5060 -> CUSTOMER-IP:5060 SIP/2.0 200 canceling From: <sip:CUST-PHONE-NUMBER@CUSTOMER-IP:5060;transport=UDP>;tag=TAG To: <sip:011445551212@OPENSER.FQDN:5060;transport=UDP>;tag=TAG2 Call-ID: CALLID@CUSTOMER-IP CSeq: 2 CANCEL Via: SIP/2.0/UDP CUSTOMER-IP:5060;branch=BRANCH Content-Length: 0 How was that last packet generated, where was it done? It's not in my openser.cfg, should I have sent back just such a message? If so, would this automagically sent message still have been sent? Next I get these two packets: # U 2006/11/06 18:40:10.761582 PSTNGW-IP:5060 -> OPENSER-IP:5060 SIP/2.0 200 OK Call-ID: CALLID@CUSTOMER-IP From: <sip:CUST-PHONE-NUMBER@CUSTOMER-IP:5060;transport=UDP>;tag=TAG To: <sip:011445551212@OPENSER.FQDN:5060;transport=UDP> Content-Length: 0 CSeq: 2 CANCEL Via: SIP/2.0/UDP OPENSER-IP;branch=BRANCH2.0,SIP/2.0/UDP CUSTOMER-IP:5060;branch=BRANCH Contact: sip:445551212@PSTNGW-IP:5060;user=phone Record-Route: <sip:011445551212@OPENSER-IP;lr=on;ftag=TAG> # U 2006/11/06 18:40:10.761616 PSTNGW-IP:5060 -> OPENSER-IP:5060 SIP/2.0 487 Request Terminated Call-ID: CALLID@CUSTOMER-IP From: <sip:CUST-PHONE-NUMBER@CUSTOMER-IP:5060;transport=UDP>;tag=TAG To: <sip:011445551212@OPENSER.FQDN:5060;transport=UDP>;tag=20505 Content-Length: 0 CSeq: 2 INVITE Via: SIP/2.0/UDP OPENSER-IP;branch=BRANCH2.0,SIP/2.0/UDP CUSTOMER-IP:5060;branch=BRANCH and my openser box replies with : # U 2006/11/06 18:40:10.762238 OPENSER-IP:5060 -> PSTNGW-IP:5060 ACK sip:445551212@PSTNGW-IP;user=phone SIP/2.0 Via: SIP/2.0/UDP OPENSER-IP;branch=BRANCH2.0 From: <sip:CUST-PHONE-NUMBER@CUSTOMER-IP:5060;transport=UDP>;tag=TAG Call-ID: CALLID@CUSTOMER-IP To: <sip:011445551212@OPENSER.FQDN:5060;transport=UDP>;tag=20505 CSeq: 2 ACK Content-Length: 0 and sends on this to the UAC: # U 2006/11/06 18:40:10.762377 OPENSER-IP:5060 -> CUSTOMER-IP:5060 SIP/2.0 487 Request Terminated Call-ID: CALLID@CUSTOMER-IP From: <sip:CUST-PHONE-NUMBER@CUSTOMER-IP:5060;transport=UDP>;tag=TAG To: <sip:011445551212@OPENSER.FQDN:5060;transport=UDP>;tag=20505 Content-Length: 0 CSeq: 2 INVITE Via: SIP/2.0/UDP CUSTOMER-IP:5060;branch=BRANCH Now, how was that generated? Is this part of the "tm" module? And then when I get this ACK: # U 2006/11/06 18:40:10.772875 CUSTOMER-IP:5060 -> OPENSER-IP:5060 ACK sip:011445551212@OPENSER.FQDN:5060 SIP/2.0 From: <sip:CUST-PHONE-NUMBER@CUSTOMER-IP:5060;transport=UDP>;tag=TAG To: <sip:011445551212@OPENSER.FQDN:5060;transport=UDP>;tag=20505 Call-ID: CALLID@CUSTOMER-IP CSeq: 2 ACK Via: SIP/2.0/UDP CUSTOMER-IP:5060;branch=BRANCH Max-Forwards: 70 Contact: <sip:CUST-PHONE-NUMBER@CUSTOMER-IP:5060;transport=UDP> Content-Length: 0 What should I do with it? I do nothing... if I should do nothing is there any special way I should do it? Or is this just handled by the tm module? What I don't get is the CANCEL is directed AT my openser box: CANCEL sip:011445551212@OPENSER.FQDN:5060 SIP/2.0 To: <sip:011445551212@OPENSER.FQDN:5060;transport=UDP> just like the ACK: ACK sip:011445551212@OPENSER.FQDN:5060 SIP/2.0 To: <sip:011445551212@OPENSER.FQDN:5060;transport=UDP>;tag=20505 and I t_relay() the CANCEL but sit on the ACK...

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[Users] Some problem in Hung up ...... help me.........
by raviprakash sunkara 13 Nov '06

13 Nov '06

Hello Users, When I keep my Sip Server into firewall_router is not hung up, But with Direct Modem to My Sip Server then it hung ups finely, Sip server listen on 192.168.2.90 with public ip xx.xx.xx.xx.xx. See the Loation tables +----------+--------+----------------------------+------------------------+------+---------------------+-------+-----------------------------------------------+-------+---------------------+-------+----------------------+-----------------------+---------+ | username | domain | contact | received | path | expires | q | callid | cseq | last_modified | flags | user_agent | socket | methods | +----------+--------+----------------------------+------------------------+------+---------------------+-------+-----------------------------------------------+-------+---------------------+-------+----------------------+-----------------------+---------+ | 102 | | sip:102@59.144.72.234:5060 | sip:59.144.72.234:5060 | NULL | 2006-11-07 03:17:42 | -1.00 | 414D9B3694024D7D94F21AF17E63460E(a)xx.xx.xx.xx | 61160 | 2006-11-07 02:47:42 | 1 | X-Lite release 1103m | udp:192.168.2.90:5060 | NULL | | 101 | | sip:101@61.17.248.68:5060 | sip:61.17.248.68:5060 | NULL | 2006-11-07 03:17:04 | -1.00 | 532BD0D57F89495CB42AFA7B7D760BE2(a)xx.xx.xx.xx | 9830 | 2006-11-07 02:47:04 | 1 | X-Lite release 1103m | udp:192.168.2.90:5060 | NULL | +----------+--------+----------------------------+------------------------+------+---------------------+-------+-----------------------------------------------+-------+---------------------+-------+----------------------+-----------------------+---------+ my openser nat configure is like route[2] { if(nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_resgiter(); force_rport(); setflag(6); } else { if (method == "INVITE") { fix_nated_sdp("2") }; fix_nated_contact() force_rport() setflag(6); }; } In X-lite log..... After the INVITE Method got response from server ( xx.xx.xx.xx.xx) .... ACK is sending like That ..... and also Bye method After the INvite is posseed Ack Send >> 192.168.2.90:5060 ... But CANCEL method send properly............ Help me -- 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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[Users] exec module : why not in onreply_route ?
by Jean-François SMIGIELSKI 13 Nov '06

13 Nov '06

Hello, all. I would know why we should avoid using the exec module exported functions outside out the request_route and on_failure_route blocks? (and so ... why it is forbidden). Answers carry useful informations (e.g. SDP) that we could want to parse with an external command. Maybe is there a more appropriate way to achieve this, and I mail you for this reason. Thanks once again, JF Smigielski ________________________________________________________________________ iBELGIQUE, exprimez-vous ! http://web.ibelgique.com/

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[Serusers] off-topic: Sofphones with Redirect Suport
by Felipe Louback 13 Nov '06

13 Nov '06

Hi all, I need a softphone that supports redirect. Let me explain: In all free softphones I download, I have an "outbound proxy" field. So, if my server returns and 302 Moved Temporarily, the softphone will get the Contact header field and try to call the user again using the outbound proxy, and the outbound will reply with an 302 and I will get an loop. I need a softphone who will use a location server to query the location of a user and after that the softphone will make the call directly, without using a proxy. Could anyone point me a softphone that will do this? Best Regards, Felipe -- Master Student - Electrical Engineering Department Computer Engineering and Telecommunications Research Group Universidade Federal de Minas Gerais - Brazil "Come to me, all you who are weary and burdened, and I will give you rest." Matthews 11:28

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[Serusers] SER pre-release ser_ctl
by SER LIST 13 Nov '06

13 Nov '06

Hello, When I try to run the ser_ctl command to do something like ser_ctl version or any other command, I get the error ser_ctl: (111, 'Connection refused'): error SER is running while I type the command. Any hints will be appreciated. _________________________________________________________________ Buy, Load, Play. The new Sympatico / MSN Music Store works seamlessly with Windows Media Player. Just Click PLAY. http://musicstore.sympatico.msn.ca/content/viewer.aspx?cid=SMS_Sept192006

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Re: [Serusers] SER pre-release - PRESENCE ISSUES
by SER LIST 12 Nov '06

12 Nov '06

Thank you. settig the xcap_root for the xcap module fixed the two errors relating to xcap_root. BUT I am still getting the two errors relating to the lookup_domain command, namely: ------------------------------------- ser: parse error (192,24-25): unknown command, missing loadmodule? ser: parse error (261,26-27): unknown command, missing loadmodule? ser: ERROR: bad config file (2 errors) ser startup failed ---------------------------------- I am using the presence example file. The excerpt from my ser.cfg file is (I have marked the LINES where the error occurs): ---- start --- loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/avp.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/dialog.so" loadmodule "/usr/local/lib/ser/modules/rls.so" loadmodule "/usr/local/lib/ser/modules/pa.so" loadmodule "/usr/local/lib/ser/modules/xcap.so" loadmodule "/usr/local/lib/ser/modules/presence_b2b.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/fifo.so" loadmodule "/usr/local/lib/ser/modules/xmlrpc.so" loadmodule "/usr/local/lib/ser/modules/xlog.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/msilo.so" { LINES TRUNCATED} if (uri==myself) { if (!lookup_domain("To")) { <-------------------------- LINE 192 xlog("L_ERR", "Unknown domain to: %tu from: %fu\n"); route(1); break; } if (method=="SUBSCRIBE") { if (!t_newtran()) { sl_reply_error(); break; }; (lines truncated) if (lookup_domain("From")) { <---------------------- LINE 261 if (lookup_user("From")) { if (is_simple_rls_target("$uid-list")) { # log(1, "it is simple subscription!\n"); # takes From UID and makes XCAP query for user's # list named "default" if (!query_resource_list("default")) { t_reply("404", "No such user list"); break; } } } } ----------- END --------------------- Any suggestions?? THanks in advance >From: samuel <samu60(a)gmail.com> >To: "SER LIST" <sergrp(a)hotmail.com> >CC: serusers(a)iptel.org, serdev(a)iptel.org >Subject: Re: [Serusers] SER pre-release - PRESENCE ISSUES >Date: Fri, 10 Nov 2006 10:14:17 +0100 > >XCAP parameters where removed from rls and pa and a new module, called >xcap contains the xcap root so you have to delete pa and rls xcap >parameter, load the new module xcap.so and set the xcap_root parameter >of xcap module to the directory having the doc structure. > >About the lookup_domain, there can be a typo. Can you post the config >file syntax?? > >Hope it helps! > >Samuel > >2006/11/10, SER LIST <sergrp(a)hotmail.com>: >>I installed the SER pre-release along w/ all modules. The problem is that >>presence now does not run (it was working w/ the presence-release-10.99). >>The errors I get are: >>---------------------------------------------------------------------------------------------------------- >>set_mod_param_regex: parameter <xcap_root> not found in module <rls> >>ser: parse error (82,20-21): Can't set module parameter >>set_mod_param_regex: parameter <auth_xcap_root> not found in module <pa> >>parse error (97,20-21): Can't set module parameter >>ser: parse error (186,24-25): unknown command, missing loadmodule? >>parse error (255,26-27): unknown command, missing loadmodule? >>ERROR: bad config file (4 errors) >>------------------------------------------------------------------------------------------------------------- >>I checked the pa module code and also the rls code and the parameters >>auth_xcap_root and xcap_root are NOT defined !!!! >> >>The error on line 185 and 255 refer to the "Lookup_domain" function. I >>verified that the domain module has been compiled and included in the >>ser.cfrg file _________________________________________________________________ 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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[Users] Forwarding requests
by Harry Harcrow 12 Nov '06

12 Nov '06

I have my SER server running on a Linux machine and I am attempting to forward some requests to a Cisco SIP server. I have been running X-lite clients on windows and the same Linux box that the SER server is running on. I have no problem completing calls between the X-lite clients OR with either X-lite client calling the Cisco SIP server directly. However, when I try to make a call from either X-lite client through the SER server I eventually get timeout methods. I am trapping those calls and wrote the following route block: route [5] { record_route(); forward(uri:host, uri:port); } I have tried replacing the forward with send() and t_relay() method calls, but nothing seems to work. The header from the X-lite client directly to the Cisco SIP server, which works is: INVITE sip:2158200@65.221.7.xxx SIP/2.0 Via: SIP/2.0/UDP 166.34.149.xxx:5061;rport;branch=z9hG4bK5963AD09C90F6F3A16F55A271C98F573 From: Harry Harcrow <sip:1000@166.34.149.xxx:5061>;tag=1279481120 To: <sip:2158200@65.221.7.xxx> Contact: <sip:1000@166.34.149.xxx:5061> Call-ID: 28E5B723-9B02-BF38-1A81-1FECC0A85D5E(a)166.34.149.xxx CSeq: 6254 INVITE Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1105d Content-Length: 313 The header from SER to the Cisco server, which does not work is: INVITE sip:2158200@65.221.7.xxx SIP/2.0 Record-Route: <sip:166.34.149.xxx;ftag=2061592395;lr=on> Via: SIP/2.0/UDP 166.34.149.xxx;branch=0 Via: SIP/2.0/UDP 166.34.149.xxx:5061;rport=5061;branch=z9hG4bK4EA9A90B8CE7CE52BFB69DA431560E5 7 From: Harry Harcrow <sip:1000@166.34.149.xxx:5061>;tag=2061592395 To: <sip:2158200@65.221.7.xxx> Contact: <sip:1000@166.34.149.xxx:5061> Call-ID: 38CA166F-59A6-E44B-F34E-9E1D6307698B(a)166.34.149.xxx CSeq: 3009 INVITE Max-Forwards: 16 Content-Type: application/sdp User-Agent: X-Lite release 1105d Content-Length: 313 The only thing that I can see different is the record_route record and an additional Via record. Does anyone have any idea what I am doing wrong? Thanks Harry Harcrow

2 1

[Serusers] Suggestion
by Eddie Johnson Jr 12 Nov '06

12 Nov '06

I am new to setting up SER and this is the test setup. 1. DHCP running from router 2. SJ Phone on laptop 192.168.X.X 3. Grandstream Budgetone 101 192.168.X.X 4. SIP Express Router 192.168.X.X 5. Asterisk Server 192.168.X.X attached to tdm400p card is an analog phone Will this work internally? There is no connection to a telco. The SJ phone is showing the following: nat/firewall: Port Restricted Cone NAT SIP calls use outbound proxy: 192.168.X.X (SER) Will I have to register the SJ, Grandstream and Asterisk PBX? Ed

1 0

[Serusers] SER not sending 486 Busy Here
by Ravi ***** 11 Nov '06

11 Nov '06

Hi Users , I am Using SER-0.96 , When I make a call from SER to Pstn it works fine I mean is all "INV"BYE"CAN". But when I recieved a call from Pstn I cannot cancel the call because X-lite is sending to pstn gateway 480 temp. service not available instead of 486 Busy Here. I am using default config. but I dont understand where it went wrong . Any Idea or help greatly appreciated Thank You. Regards, Ravi.

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Re: [Serusers] Re: How to change Caller ID in FROM field ?
by lawrence k.y. lin 10 Nov '06

10 Nov '06

Hi Kristian. Thanks for your email. I am using ser 0.9.6 and did not find uac_replace_from function. Then I did a search and found that it is in openser. I did install openser and it worked. Do you or anyone know if this function will be added into ser release ? Please advise, thanks, Larry >------------------------------ >Message: 20 >Date: Wed, 25 Oct 2006 13:11:06 +0200 >From: Kristian Larsson <kristian(a)netatonce.se> >Subject: Re: [Serusers] Re: How to change Caller ID in FROM field ? >To: "lawrence k.y. lin" <lawlin888(a)hotmail.com> >Cc: serusers(a)lists.iptel.org >Message-ID: <20061025111106.GJ30882(a)juniks.net> >Content-Type: text/plain; charset=iso-8859-15 > >On Tue, Oct 24, 2006 at 03:45:01PM -0700, lawrence k.y. lin wrote: >>Hi all, >> >> Appreciate if someone can give me a suggestion or even an answer. >> Since the call from SIP device to PSTN phone did not have Caller ID >>and our Telco carriers request for the Caller ID, we have to set a number >>before the calls are routed to PSTN gateway. >> I did it in Asterisk before with the function calls SetCallerID(CLID) >>or Set(CALLERID(number)=CLID) (depends on the Asterisk version) and these >>function calls worked. But because of the scalabliity issue, I changed to >>SER and would like to do the similar thing as in Asterisk. >> I read the SER document and awared SIP_HF_FROM has the data for the >>entired FROM field. Is there any other variables for Caller ID only ? >> I searched the mailing list for SER and did not find any related topic >>(only found one with title: caller-id with raius using sip-rpid, but it is >>irrelevant). >There is a function called uac_replace_from which >can replace values in the From header, however I >would need some help with it myself. >For example, towards one of my SIP "peers", I need >to prefix a "0" on all From headers which is no >problem, just do a uac_replace_from("0$fU","sip:0$fU@$fd"); >$fU is the from user part and $fd is the from >domain... >But how do I do if I first want to strip a few >characters of from $fU and then add a few others? >Regards, > Kristian. >-- >Kristian Larsson _________________________________________________________________ Find a local pizza place, music store, museum and morethen map the best route! http://local.live.com?FORM=MGA001

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[Serusers] fr_inv_timer + avp not working under high load
by Marcus Hunger 10 Nov '06

10 Nov '06

Hi list, I am trying to realise dynamic invite-timeout-timers loading timer-settings per call from my database using avp. It seems to work perfectly on my development-system but fails nearly completly in my high loaded life-environment. According to my logs, timer values are set correctly (eg.: reply_received: FR_INV_TIMER = 15, see modules/tm/t_reply.c:1371), but most of the times the timers have no effect. My setup is similar to http://siprouter.onsip.org/doc/gettingstarted/ch09.html, but avps are not written by hand, but acquired with: modparam("tm", "fr_inv_timer_avp", "mytimer"); avp_db_load("$ruri/username","s:mytimer"); 'mytimer' is an attribute in an usr_preferences-table. Ser's version is 0.9.6. I already searched the list-archives without finding anything regarding this issue. I Also failed accessing the bug-tracker (http://bugs.sip-router.org/), it gave me an internal server error. Has anybody experienced this effect? Is this a bug? Is there a workaround? Best regards, Marcus Hunger

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[Users] Problem with TCP INVITE being bridged by OpenSER to UDP
by Russ Daigle 10 Nov '06

10 Nov '06

I have a scenario where user Alice registered with OpenSER with UDP port 5070. Then, I have user Bob sending an invite to Alice via the OpenSER using TCP. Hence, OpenSER is bridging my TCP SIP requests to UDP (and vice versa for the responses). Here is the sequence: Alice ----- TCP Invite -----> OpenSER ---------- UDP Invite ------ > Alice Alice responds back with 180 ringing, then 200 Ok. Upon getting the 200 OK, Bob sends an ACK (to complete the INVITE), followed by an Immediate BYE. THE TROUBLE IS...... the ACK and the BYE that Bob sends are dropped by OpenSER, and never forwarded onto Alice!!!! I see nothing wrong with the messages. WHat is happening? sip:~ russdaigle$ sudo tcpdump -i en0 -A -q -s 2000 port 5060 Password: tcpdump: verbose output suppressed, use -v or -vv for full protocol decode listening on en0, link-type EN10MB (Ethernet), capture size 2000 bytes The following shows an sip-invite-bye from mutator using TCP to mjsip (via OPENSER). Everything after the ACK is dropped and not sent on. Then, is a similar but UDP exchange. In that case, we get a 404 not found in response to the BYE!!! 01:17:59.553909 IP 10.10.2.6.50135 > 10.10.1.234.sip: tcp 588 INVITE sip:alice@siptest2.com:5060 SIP/2.0 Via: SIP/2.0/TCP 10.10.2.6:50135;branch=z9hG4bK85478043 From: Bob <sip:bob@siptest2.com>;tag=fromhackblah To: Alice <sip:alice@siptest2.com:5060> CSeq: 1 INVITE Call-ID: SYRGNUXWKNZNXXPXUAMUTJXXPPMXWKVJFAXUNUUTGG(a)musecurity.com Contact: <sip:bob@10.10.2.6>;transport=TCP Max-Forwards: 70 Content-Type: application/sdp Content-Length: 195 v=0 o=- 817933771 817933775 IN IP4 10.10.2.6 s=darkness c=IN IP4 10.10.2.6 t=0 0 m=audio 5000 RTP/AVP 0 4 97 a=rtpmap:0 PCMU/8000 a=rtpmap:4 G723/8000/1 a=rtpmap:97 telephone-event/8000 01:17:59.554929 IP 10.10.1.234.sip > 10.10.2.6.50135: tcp 536 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/TCP 10.10.2.6:50135;branch=z9hG4bK85478043 From: Bob <sip:bob@siptest2.com>;tag=fromhackblah To: Alice <sip:alice@siptest2.com:5060> CSeq: 1 INVITE Call-ID: SYRGNUXWKNZNXXPXUAMUTJXXPPMXWKVJFAXUNUUTGG(a)musecurity.com Server: OpenSer (1.1.0-notls (i386/darwin)) Content-Length: 0 Warning: 392 10.10.1.234:5060 "Noisy feedback tells: pid=11962 req_src_ip=10.10.2.6 req_src_port=50135 in_uri=sip:alice@siptest2.com: 5060 out_uri=sip:alice@10.10.2.6:5070 via_cnt==1" 01:17:59.555035 IP 10.10.1.234.sip > 10.10.2.6.5070: UDP, length 814 INVITE sip:alice@10.10.2.6:5070 SIP/2.0 Record-Route: <sip:10.10.1.234;r2=on;lr=on;ftag=fromhackblah> Record-Route: <sip: 10.10.1.234;transport=tcp;r2=on;lr=on;ftag=fromhackblah> Via: SIP/2.0/UDP 10.10.1.234;branch=z9hG4bK9316.e5ab8ce2.0;i=b1 Via: SIP/2.0/TCP 10.10.2.6:50135;branch=z9hG4bK85478043 From: Bob <sip:bob@siptest2.com>;tag=fromhackblah To: Alice <sip:alice@siptest2.com:5060> CSeq: 1 INVITE Call-ID: SYRGNUXWKNZNXXPXUAMUTJXXPPMXWKVJFAXUNUUTGG(a)musecurity.com Contact: <sip:bob@10.10.2.6>;transport=TCP Max-Forwards: 69 Content-Type: application/sdp Content-Length: 195 P-hint: usrloc applied v=0 o=- 817933771 817933775 IN IP4 10.10.2.6 s=darkness c=IN IP4 10.10.2.6 t=0 0 m=audio 5000 RTP/AVP 0 4 97 a=rtpmap:0 PCMU/8000 a=rtpmap:4 G723/8000/1 a=rtpmap:97 telephone-event/8000 01:17:59.667962 IP 10.10.2.6.5070 > 10.10.1.234.sip: UDP, length 364 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.1.234;branch=z9hG4bK9316.e5ab8ce2.0;i=b1 Via: SIP/2.0/TCP 10.10.2.6:50135;branch=z9hG4bK85478043 To: Alice <sip:alice@siptest2.com:5060> From: Bob <sip:bob@siptest2.com>;tag=fromhackblah Call-ID: SYRGNUXWKNZNXXPXUAMUTJXXPPMXWKVJFAXUNUUTGG(a)musecurity.com CSeq: 1 INVITE Server: mjsip stack 1.6 Content-Length: 0 01:17:59.699680 IP 10.10.2.6.5070 > 10.10.1.234.sip: UDP, length 505 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.10.1.234;branch=z9hG4bK9316.e5ab8ce2.0;i=b1 Via: SIP/2.0/TCP 10.10.2.6:50135;branch=z9hG4bK85478043 Record-Route: <sip:10.10.1.234;r2=on;lr=on;ftag=fromhackblah> Record-Route: <sip: 10.10.1.234;transport=tcp;r2=on;lr=on;ftag=fromhackblah> To: Alice <sip:alice@siptest2.com:5060> From: Bob <sip:bob@siptest2.com>;tag=fromhackblah Call-ID: SYRGNUXWKNZNXXPXUAMUTJXXPPMXWKVJFAXUNUUTGG(a)musecurity.com CSeq: 1 INVITE Server: mjsip stack 1.6 Content-Length: 0 01:17:59.699683 IP 10.10.2.6.5070 > 10.10.1.234.sip: UDP, length 728 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.1.234;branch=z9hG4bK9316.e5ab8ce2.0;i=b1 Via: SIP/2.0/TCP 10.10.2.6:50135;branch=z9hG4bK85478043 Record-Route: <sip:10.10.1.234;r2=on;lr=on;ftag=fromhackblah> Record-Route: <sip: 10.10.1.234;transport=tcp;r2=on;lr=on;ftag=fromhackblah> To: Alice <sip:alice@siptest2.com:5060>;tag=7c452a3254b2e183 From: Bob <sip:bob@siptest2.com>;tag=fromhackblah Call-ID: SYRGNUXWKNZNXXPXUAMUTJXXPPMXWKVJFAXUNUUTGG(a)musecurity.com CSeq: 1 INVITE Contact: <sip:alice@192.168.0.3:5070> Server: mjsip stack 1.6 Content-Length: 135 Content-Type: application/sdp v=0 o=- 817933771 817933775 IN IP4 10.10.2.6 s=darkness c=IN IP4 192.168.0.3 t=0 0 m=audio 21000 RTP/AVP 0 a=rtpmap:0 PCMU/8000 01:17:59.700273 IP 10.10.1.234.sip > 10.10.2.6.50135: tcp 440 SIP/2.0 180 Ringing Via: SIP/2.0/TCP 10.10.2.6:50135;branch=z9hG4bK85478043 Record-Route: <sip:10.10.1.234;r2=on;lr=on;ftag=fromhackblah> Record-Route: <sip: 10.10.1.234;transport=tcp;r2=on;lr=on;ftag=fromhackblah> To: Alice <sip:alice@siptest2.com:5060> From: Bob <sip:bob@siptest2.com>;tag=fromhackblah Call-ID: SYRGNUXWKNZNXXPXUAMUTJXXPPMXWKVJFAXUNUUTGG(a)musecurity.com CSeq: 1 INVITE Server: mjsip stack 1.6 Content-Length: 0 01:17:59.700299 IP 10.10.1.234.sip > 10.10.2.6.50135: tcp 663 SIP/2.0 200 OK Via: SIP/2.0/TCP 10.10.2.6:50135;branch=z9hG4bK85478043 Record-Route: <sip:10.10.1.234;r2=on;lr=on;ftag=fromhackblah> Record-Route: <sip: 10.10.1.234;transport=tcp;r2=on;lr=on;ftag=fromhackblah> To: Alice <sip:alice@siptest2.com:5060>;tag=7c452a3254b2e183 From: Bob <sip:bob@siptest2.com>;tag=fromhackblah Call-ID: SYRGNUXWKNZNXXPXUAMUTJXXPPMXWKVJFAXUNUUTGG(a)musecurity.com CSeq: 1 INVITE Contact: <sip:alice@192.168.0.3:5070> Server: mjsip stack 1.6 Content-Length: 135 Content-Type: application/sdp v=0 o=- 817933771 817933775 IN IP4 10.10.2.6 s=darkness c=IN IP4 192.168.0.3 t=0 0 m=audio 21000 RTP/AVP 0 a=rtpmap:0 PCMU/8000 01:17:59.804566 IP 10.10.2.6.50135 > 10.10.1.234.sip: tcp 435 ACK sip:alice@siptest2.com:5060 SIP/2.0 Route: <sip:10.10.1.234;r2=on;lr=on;ftag=fromhackblah> Route: <sip:10.10.1.234;transport=tcp;r2=on;lr=on;ftag=fromhackblah> Via: SIP/2.0/TCP 10.10.2.6:50135;branch=z9hG4bK59972 From: Bob <sip:bob@siptest2.com>;tag=fromhackblah To: Alice <sip:alice@siptest2.com:5060>;tag=7c452a3254b2e183 CSeq: 1 ACK Call-ID: SYRGNUXWKNZNXXPXUAMUTJXXPPMXWKVJFAXUNUUTGG(a)musecurity.com Max-Forwards: 70 01:17:59.839493 IP 10.10.2.6.50135 > 10.10.1.234.sip: tcp 435 BYE sip:alice@siptest2.com:5060 SIP/2.0 Route: <sip:10.10.1.234;r2=on;lr=on;ftag=fromhackblah> Route: <sip:10.10.1.234;transport=tcp;r2=on;lr=on;ftag=fromhackblah> Via: SIP/2.0/TCP 10.10.2.6:50135;branch=z9hG4bK68444 From: Bob <sip:bob@siptest2.com>;tag=fromhackblah To: Alice <sip:alice@siptest2.com:5060>;tag=7c452a3254b2e183 CSeq: 2 BYE Call-ID: SYRGNUXWKNZNXXPXUAMUTJXXPPMXWKVJFAXUNUUTGG(a)musecurity.com Max-Forwards: 70 01:18:00.199306 IP 10.10.2.6.5070 > 10.10.1.234.sip: UDP, length 728 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.1.234;branch=z9hG4bK9316.e5ab8ce2.0;i=b1 Via: SIP/2.0/TCP 10.10.2.6:50135;branch=z9hG4bK85478043 Record-Route: <sip:10.10.1.234;r2=on;lr=on;ftag=fromhackblah> Record-Route: <sip: 10.10.1.234;transport=tcp;r2=on;lr=on;ftag=fromhackblah> To: Alice <sip:alice@siptest2.com:5060>;tag=7c452a3254b2e183 From: Bob <sip:bob@siptest2.com>;tag=fromhackblah Call-ID: SYRGNUXWKNZNXXPXUAMUTJXXPPMXWKVJFAXUNUUTGG(a)musecurity.com CSeq: 1 INVITE Contact: <sip:alice@192.168.0.3:5070> Server: mjsip stack 1.6 Content-Length: 135 Content-Type: application/sdp v=0 o=- 817933771 817933775 IN IP4 10.10.2.6 s=darkness c=IN IP4 192.168.0.3 t=0 0 m=audio 21000 RTP/AVP 0 a=rtpmap:0 PCMU/8000

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[Serusers] IM and silo table in MySQL
by davor jovanovic 10 Nov '06

10 Nov '06

Hi all, I have a problem with silo table in MySQL. messages are stored in the table when user is offline, and when comes back online messages are delivered to the client. Problem is that the messages aren't erased from table after delivery . Am I missing something in my ser.cfg? I'm using ser-0.9.6 and XLite 3.0. I will appreciate any help. thanks /davor loadmodule "/usr/lib/ser/modules/msilo.so" modparam("auth_db|uri_db|usrloc|permissions|group|acc_db|msilo", "db_url", "mysql://ser:heslo@localhost/ser") modparam("msilo", "db_table", "silo") modparam("msilo", "check_time", 10) modparam("msilo", "clean_period", 3) modparam("msilo", "use_contact", 1) 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; }; ... } else if (method=="MESSAGE") { route(8); break; }; ... route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- if (!www_authorize("","subscriber")) { www_challenge("","0"); break; ... m_dump(); } route[8] { ########################## # Message Handler # ########################### if (!lookup("location")) { if (method == "MESSAGE") { if (!t_newtran()) { sl_reply_error(); break; }; if (m_store("1")) { t_reply("202", "Accepted for Later Delivery"); break; }; t_reply("503", "Service Unavailable"); break; }; }; route(1); t_on_failure("1"); } ......... failure_route[1] { if (m_store("1")) { log("MSILO: offline message stored\n"); t_reply("202", "Accepted"); }else{ log("MSILO: offline message NOT stored\n"); t_reply("503", "Service Unavailable"); };

1 0

[Serusers] Problem with radius-sql and ser-0.9.6
by Lokesh Kumar 10 Nov '06

10 Nov '06

Hello, I am running old ser version 0.9.6, where I am authenticating on radius and keeping the users record in default sql database of radius. But it is not authenticating, the logs are mentioned below. But it worked absolutely fine with radius users files. I have the entry for the user in radcheck file but still it is saying user not found. Can anyone give any hint where I am doing wrong. Going to the next request --- Walking the entire request list --- Waking up in 1 seconds... rad_recv: Access-Request packet from host 127.0.0.1:54086, id=241, length=262 User-Name = "211069020(a)voip.nortenet.pt" Digest-Attributes = 0x0a0b323131303639303230 Digest-Attributes = 0x0112766f69702e6e6f7274656e65742e7074 Digest-Attributes = 0x022a3435353436346634343937623539656362346335633261323364656436656532393934 3565316432 Digest-Attributes = 0x04167369703a766f69702e6e6f7274656e65742e7074 Digest-Attributes = 0x030a5245474953544552 Digest-Attributes = 0x050661757468 Digest-Attributes = 0x090a3030303030303031 Digest-Attributes = 0x08103132373935383532383139343033 Digest-Response = "2ae0ba094f508b9dff7bb56d96649875" Service-Type = Sip-Session Sip-Uri-User = "211069020" NAS-Port = 5060 NAS-IP-Address = 127.0.0.1 Processing the authorize section of radiusd.conf modcall: entering group authorize for request 1 modcall[authorize]: module "preprocess" returns ok for request 1 modcall[authorize]: module "chap" returns noop for request 1 modcall[authorize]: module "mschap" returns noop for request 1 rlm_digest: Adding Auth-Type = DIGEST modcall[authorize]: module "digest" returns ok for request 1 rlm_eap: No EAP-Message, not doing EAP modcall[authorize]: module "eap" returns noop for request 1 radius_xlat: '211069020(a)voip.nortenet.pt' rlm_sql (sql): sql_set_user escaped user --> '211069020(a)voip.nortenet.pt' radius_xlat: 'SELECT id, UserName, Attribute, Value, op FROM radcheck WHERE Username = '211069020(a)voip.nortenet.pt' ORDER BY id' rlm_sql (sql): Reserving sql socket id: 13 rlm_sql (sql): User 211069020(a)voip.nortenet.pt not found in radcheck radius_xlat: 'SELECT radgroupcheck.id,radgroupcheck.GroupName,radgroupcheck.Attribute,radgroupche ck.Value,radgroupcheck.op FROM radgroupcheck,usergroup WHERE usergroup.Username = '211069020(a)voip.nortenet.pt' AND usergroup.GroupName = radgroupcheck.GroupName ORDER BY radgroupcheck.id' radius_xlat: 'SELECT radgroupreply.id,radgroupreply.GroupName,radgroupreply.Attribute,radgrouprep ly.Value,radgroupreply.op FROM radgroupreply,usergroup WHERE usergroup.Username = '211069020(a)voip.nortenet.pt' AND usergroup.GroupName = radgroupreply.GroupName ORDER BY radgroupreply.id' rlm_sql (sql): User 211069020(a)voip.nortenet.pt not found in radgroupcheck rlm_sql (sql): Released sql socket id: 13 rlm_sql (sql): User not found modcall[authorize]: module "sql" returns notfound for request 1 modcall: leaving group authorize (returns ok) for request 1 rad_check_password: Found Auth-Type DIGEST auth: type "digest" Processing the authenticate section of radiusd.conf modcall: entering group authenticate for request 1 rlm_digest: Configuration item "User-Password" or Digest-HA1 is required for authentication. modcall[authenticate]: module "digest" returns invalid for request 1 modcall: leaving group authenticate (returns invalid) for request 1 auth: Failed to validate the user. Login incorrect: [211069020(a)voip.nortenet.pt] (from client localhost port 5060) Delaying request 1 for 1 seconds Finished request 1 Going to the next request --- Walking the entire request list --- Waking up in 1 seconds... --- Walking the entire request list --- Sending Access-Reject of id 240 to 127.0.0.1 port 54085 Waking up in 1 seconds... --- Walking the entire request list --- Sending Access-Reject of id 241 to 127.0.0.1 port 54086 Waking up in 3 seconds... --- Walking the entire request list --- Cleaning up request 0 ID 240 with timestamp 455463c8 Waking up in 1 seconds... --- Walking the entire request list --- Cleaning up request 1 ID 241 with timestamp 455463c9 Nothing to do. Sleeping until we see a request. Thanks very much Lokesh -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.409 / Virus Database: 268.14.1/527 - Release Date: 11/9/2006

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[Users] Problems with TCP REGISTER, UDP REGISTER works fine
by Russ Daigle 10 Nov '06

10 Nov '06

For some reason, I am having problems with having TCP REGISTER messages being accepted by openser. The following is a REGISTER I send to Open SER. It is never responded to. Do you see anything wrong with it? It should be registering port 5060 TCP. 00:19:42.157347 IP 10.10.2.6.50083 > 10.10.1.234.sip: tcp 345 REGISTER sip:siptest2.com:5060 SIP/2.0^M Via: SIP/2.0/TCP 10.10.2.6:50083;branch=z9hG4bK45^M From: Alice <sip:alice@siptest2.com>;tag=fromhackblah^M To: Alice <sip:alice@siptest2.com>^M CSeq: 1 REGISTER^M Call-ID: DNYEXXLURKQELGQLDAJXJADPKUJVSHKUZSFPYXDGQ(a)musecurity.com^M Contact: <sip:alice@10.10.2.6>;expires=900;transport=TCP^M Max-Forwards: 70^M ^M When I send a similar UDP register, it works fine! REGISTER sip:siptest2.com:5060 SIP/2.0^M Via: SIP/2.0/UDP 10.10.2.6:5070;branch=z9hG4bK8149125^M From: Alice <sip:alice@siptest2.com>;tag=fromhackblah^M To: Alice <sip:alice@siptest2.com>^M CSeq: 1 REGISTER^M Call-ID: PFZEHCUVOJRSHOXXOQNYONBOFKDGWUKNEWRYDJFSCTBCDJQLLNQPJUN(a)musecurity.com Contact: <sip:alice@10.10.2.6:5070>;expires=900;transport=UDP^M Max-Forwards: 70^M ^M 00:19:39.364079 IP 10.10.1.234.sip > 10.10.2.6.5070: UDP, length 593 SIP/2.0 200 OK^M Via: SIP/2.0/UDP 10.10.2.6:5070;branch=z9hG4bK8149125^M From: Alice <sip:alice@siptest2.com>;tag=fromhackblah^M To: Alice <sip:alice@siptest2.com>;tag=329cfeaa6ded039da25ff8cbb8668bd2.c3bf^M CSeq: 1 REGISTER^M Call-ID: PFZEHCUVOJRSHOXXOQNYONBOFKDGWUKNEWRYDJFSCTBCDJQLLNQPJUN(a)musecurity.com Contact: <sip:alice@10.10.2.6:5070>;expires=900^M Server: OpenSer (1.1.0-notls (i386/darwin))^M Content-Length: 0^M Warning: 392 10.10.1.234:5060 "Noisy feedback tells: pid=11956 req_src_ip=10.10 .2.6 req_src_port=5070 in_uri=sip:siptest2.com:5060 out_uri=sip:siptest2.com:506 0 via_cnt==1"^M ^M

4 4

[Users] Problem with BYE method getting a 404 not found response
by Russ Daigle 10 Nov '06

10 Nov '06

I can reproduce this easily. Do you see anything wrong with the below packet traces? Here is the sequence: Bob ----- UDP Invite -----> OpenSER ---------- UDP Invite ------ > Alice Alice responds back with 180 ringing, then 200 Ok. Upon getting the 200 OK, Bob sends an ACK (to complete the INVITE), followed by an Immediate BYE. THE TROUBLE IS...... the ACK that Bob sends is dropped (never forwarded to Alice) and the BYE that Bob sends is responded by OpenSER with a 404 not found, and never forwarded onto Alice!!!! I see nothing wrong with the messages. WHat is happening? Below is the message exchanges. Followed by "openserctl ul show" to show Alice is still registered. 01:18:19.801230 IP 10.10.2.6.5080 > 10.10.1.234.sip: UDP, length 597 INVITE sip:alice@siptest2.com:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.2.6:5080;branch=z9hG4bK15 From: Bob <sip:bob@siptest2.com>;tag=fromhackblah To: Alice <sip:alice@siptest2.com:5060> CSeq: 1 INVITE Call-ID: TUNQQQNWAYJDYCXGYAIWQEMRGOLILIBMMJMLNTFVLIBIBUORMDALF(a)musecurity.com Contact: <sip:bob@10.10.2.6:5080>;transport=UDP Max-Forwards: 70 Content-Type: application/sdp Content-Length: 195 v=0 o=- 817933771 817933775 IN IP4 10.10.2.6 s=darkness c=IN IP4 10.10.2.6 t=0 0 m=audio 5000 RTP/AVP 0 4 97 a=rtpmap:0 PCMU/8000 a=rtpmap:4 G723/8000/1 a=rtpmap:97 telephone-event/8000 01:18:19.802801 IP 10.10.1.234.sip > 10.10.2.6.5080: UDP, length 539 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 10.10.2.6:5080;branch=z9hG4bK15 From: Bob <sip:bob@siptest2.com>;tag=fromhackblah To: Alice <sip:alice@siptest2.com:5060> CSeq: 1 INVITE Call-ID: TUNQQQNWAYJDYCXGYAIWQEMRGOLILIBMMJMLNTFVLIBIBUORMDALF(a)musecurity.com Server: OpenSer (1.1.0-notls (i386/darwin)) Content-Length: 0 Warning: 392 10.10.1.234:5060 "Noisy feedback tells: pid=11955 req_src_ip=10.10.2.6 req_src_port=5080 in_uri=sip:alice@siptest2.com: 5060 out_uri=sip:alice@10.10.2.6:5070 via_cnt==1" 01:18:19.802885 IP 10.10.1.234.sip > 10.10.2.6.5070: UDP, length 734 INVITE sip:alice@10.10.2.6:5070 SIP/2.0 Record-Route: <sip:10.10.1.234;lr=on;ftag=fromhackblah> Via: SIP/2.0/UDP 10.10.1.234;branch=z9hG4bKcbde.5492991.0 Via: SIP/2.0/UDP 10.10.2.6:5080;branch=z9hG4bK15 From: Bob <sip:bob@siptest2.com>;tag=fromhackblah To: Alice <sip:alice@siptest2.com:5060> CSeq: 1 INVITE Call-ID: TUNQQQNWAYJDYCXGYAIWQEMRGOLILIBMMJMLNTFVLIBIBUORMDALF(a)musecurity.com Contact: <sip:bob@10.10.2.6:5080>;transport=UDP Max-Forwards: 69 Content-Type: application/sdp Content-Length: 195 P-hint: usrloc applied v=0 o=- 817933771 817933775 IN IP4 10.10.2.6 s=darkness c=IN IP4 10.10.2.6 t=0 0 m=audio 5000 RTP/AVP 0 4 97 a=rtpmap:0 PCMU/8000 a=rtpmap:4 G723/8000/1 a=rtpmap:97 telephone-event/8000 01:18:19.891076 IP 10.10.2.6.5070 > 10.10.1.234.sip: UDP, length 362 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.1.234;branch=z9hG4bKcbde.5492991.0 Via: SIP/2.0/UDP 10.10.2.6:5080;branch=z9hG4bK15 To: Alice <sip:alice@siptest2.com:5060> From: Bob <sip:bob@siptest2.com>;tag=fromhackblah Call-ID: TUNQQQNWAYJDYCXGYAIWQEMRGOLILIBMMJMLNTFVLIBIBUORMDALF(a)musecurity.com CSeq: 1 INVITE Server: mjsip stack 1.6 Content-Length: 0 01:18:19.957284 IP 10.10.2.6.5070 > 10.10.1.234.sip: UDP, length 420 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.10.1.234;branch=z9hG4bKcbde.5492991.0 Via: SIP/2.0/UDP 10.10.2.6:5080;branch=z9hG4bK15 Record-Route: <sip:10.10.1.234;lr=on;ftag=fromhackblah> To: Alice <sip:alice@siptest2.com:5060> From: Bob <sip:bob@siptest2.com>;tag=fromhackblah Call-ID: TUNQQQNWAYJDYCXGYAIWQEMRGOLILIBMMJMLNTFVLIBIBUORMDALF(a)musecurity.com CSeq: 1 INVITE Server: mjsip stack 1.6 Content-Length: 0 01:18:19.957363 IP 10.10.2.6.5070 > 10.10.1.234.sip: UDP, length 643 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.1.234;branch=z9hG4bKcbde.5492991.0 Via: SIP/2.0/UDP 10.10.2.6:5080;branch=z9hG4bK15 Record-Route: <sip:10.10.1.234;lr=on;ftag=fromhackblah> To: Alice <sip:alice@siptest2.com:5060>;tag=d418a42d9b1c6819 From: Bob <sip:bob@siptest2.com>;tag=fromhackblah Call-ID: TUNQQQNWAYJDYCXGYAIWQEMRGOLILIBMMJMLNTFVLIBIBUORMDALF(a)musecurity.com CSeq: 1 INVITE Contact: <sip:alice@192.168.0.3:5070> Server: mjsip stack 1.6 Content-Length: 135 Content-Type: application/sdp v=0 o=- 817933771 817933775 IN IP4 10.10.2.6 s=darkness c=IN IP4 192.168.0.3 t=0 0 m=audio 21000 RTP/AVP 0 a=rtpmap:0 PCMU/8000 01:18:19.957995 IP 10.10.1.234.sip > 10.10.2.6.5080: UDP, length 584 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.2.6:5080;branch=z9hG4bK15 Record-Route: <sip:10.10.1.234;lr=on;ftag=fromhackblah> To: Alice <sip:alice@siptest2.com:5060>;tag=d418a42d9b1c6819 From: Bob <sip:bob@siptest2.com>;tag=fromhackblah Call-ID: TUNQQQNWAYJDYCXGYAIWQEMRGOLILIBMMJMLNTFVLIBIBUORMDALF(a)musecurity.com CSeq: 1 INVITE Contact: <sip:alice@192.168.0.3:5070> Server: mjsip stack 1.6 Content-Length: 135 Content-Type: application/sdp v=0 o=- 817933771 817933775 IN IP4 10.10.2.6 s=darkness c=IN IP4 192.168.0.3 t=0 0 m=audio 21000 RTP/AVP 0 a=rtpmap:0 PCMU/8000 01:18:20.047797 IP 10.10.2.6.5080 > 10.10.1.234.sip: UDP, length 365 ACK sip:alice@siptest2.com:5060 SIP/2.0 Route: <sip:10.10.1.234;lr=on;ftag=fromhackblah> Via: SIP/2.0/UDP 10.10.2.6:5080;branch=z9hG4bK0 From: Bob <sip:bob@siptest2.com>;tag=fromhackblah To: Alice <sip:alice@siptest2.com:5060>;tag=d418a42d9b1c6819 CSeq: 1 ACK Call-ID: TUNQQQNWAYJDYCXGYAIWQEMRGOLILIBMMJMLNTFVLIBIBUORMDALF(a)musecurity.com Max-Forwards: 70 01:18:20.047803 IP 10.10.2.6.5080 > 10.10.1.234.sip: UDP, length 370 BYE sip:alice@siptest2.com:5060 SIP/2.0 Route: <sip:10.10.1.234;lr=on;ftag=fromhackblah> Via: SIP/2.0/UDP 10.10.2.6:5080;branch=z9hG4bK767774 From: Bob <sip:bob@siptest2.com>;tag=fromhackblah To: Alice <sip:alice@siptest2.com:5060>;tag=d418a42d9b1c6819 CSeq: 2 BYE Call-ID: TUNQQQNWAYJDYCXGYAIWQEMRGOLILIBMMJMLNTFVLIBIBUORMDALF(a)musecurity.com Max-Forwards: 70 01:18:20.049706 IP 10.10.1.234.sip > 10.10.2.6.5080: UDP, length 561 SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.10.2.6:5080;branch=z9hG4bK767774 From: Bob <sip:bob@siptest2.com>;tag=fromhackblah To: Alice <sip:alice@siptest2.com:5060>;tag=d418a42d9b1c6819 CSeq: 2 BYE Call-ID: TUNQQQNWAYJDYCXGYAIWQEMRGOLILIBMMJMLNTFVLIBIBUORMDALF(a)musecurity.com Server: OpenSer (1.1.0-notls (i386/darwin)) Content-Length: 0 Warning: 392 10.10.1.234:5060 "Noisy feedback tells: pid=11955 req_src_ip=10.10.1.234 req_src_port=5060 in_uri=sip: 10.10.1.234;lr=on;ftag=fromhackblah out_uri=sip: 10.10.1.234;lr=on;ftag=fromhackblah via_cnt==2" 01:18:20.421033 IP 10.10.2.6.5070 > 10.10.1.234.sip: UDP, length 643 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.1.234;branch=z9hG4bKcbde.5492991.0 Via: SIP/2.0/UDP 10.10.2.6:5080;branch=z9hG4bK15 Record-Route: <sip:10.10.1.234;lr=on;ftag=fromhackblah> To: Alice <sip:alice@siptest2.com:5060>;tag=d418a42d9b1c6819 From: Bob <sip:bob@siptest2.com>;tag=fromhackblah Call-ID: TUNQQQNWAYJDYCXGYAIWQEMRGOLILIBMMJMLNTFVLIBIBUORMDALF(a)musecurity.com CSeq: 1 INVITE Contact: <sip:alice@192.168.0.3:5070> Server: mjsip stack 1.6 Content-Length: 135 Content-Type: application/sdp v=0 o=- 817933771 817933775 IN IP4 10.10.2.6 s=darkness c=IN IP4 192.168.0.3 t=0 0 m=audio 21000 RTP/AVP 0 a=rtpmap:0 PCMU/8000 'alice' sip:~ russdaigle$ sudo openserctl ul show WARINIG: no database engine found - tried '' Dumping all contacts may take long: are you sure you want to proceed? [Y|N] y ===Domain list=== ---Domain--- name : 'aliases' size : 512 table: 0x2191b90 d_ll { n : 0 first: 0x0 last : 0x0 } ---/Domain--- ---Domain--- name : 'location' size : 512 table: 0x218fa08 d_ll { n : 1 first: 0x2193c90 last : 0x2193c90 } ...Record(0x2193c90)... domain: 'location' aor : 'alice' ~~~Contact(0x2193cd0)~~~ domain : 'location' aor : 'alice' Contact : 'sip:alice@10.10.2.6:5070' Expires : 794 q : Call-ID : 'XQQZRLTJDMQZUMPKAJZBQEVGSPRAFOXTCGRQQCSIWPKMJLTGHFC(a)musecurity.com' CSeq : 1 User-Agent: 'n/a' received : '' Path : '' State : CS_NEW Flags : 0 Sock : 10.10.1.234:5060 (0xfa6e8) Methods : 4294967295 next : 0x0 prev : 0x0 ~~~/Contact~~~~ .../Record... ---/Domain--- ===/Domain list===

2 1

[Users] certificate recommendations
by Mark Price 10 Nov '06

10 Nov '06

Can I have some recommendations about what company and what package to go with for a certificate to work with openser? I have a cert from godaddy, and it seems that it won't work with openser because of the intermediate certificate that they require you to use. Godaddy issues a certificate, a private key and an intermediate certificate (the intermediate certificate So openser loads just fine if I set: tls_certifcate=cert.pem tls_preivate_key=cert.key but the phone still fails to validate the certificate, because there is no place to specify the intermediate certificate. The intermediate certificate is the one that corresponds to the apache2 ssl directive SSLCertificateChainFile. The phone says: Registration Error: 503 - Certificate Validation Failure and the openser logs say: 7(7201) tls_accept: Error in SSL: 7(7201) tls_error: error:14094418:SSL routines:SSL3_READ_BYTES:tlsv1 alert unknown ca Thanks, Mark Price

2 1

[Users] Multiple CA
by Gregoire 10 Nov '06

10 Nov '06

Hi everybody! I am using OpenSER 1.1 with TLS. I have generate the client and server certificate with the scripts gen_rootCA.sh and gen_usercert.sh. Everything works fine, but I have generate certificate for my UA with another CA and I have added this CA to the file user-cacert.pem. When I try to connect with my UA, OpenSER logs an error like: "tls_error: error:14094418:SSL routines:SSL3_READ_BYTES:tlsv1 alert unknown ca" My file user-cacert.pem looks like: -------BEGIN CERTIFICATE------ MAOIposio..... --------END CERTIFICATE-------- -------BEGIN CERTIFICATE------ MJ809il...... --------END CERTIFICATE-------- I think that OpenSER takes only the first CA certificate and not all the followings. Did someone have some experience with that case? Regards Greg

2 4

[Users] Measuring QoS with OpenSER
by Andreas Granig 10 Nov '06

10 Nov '06

Hi, I just came across http://www.golem.de/0602/43572.html (sorry, German). It states that AVM contributed GPL'ed source code to OpenSER, which allows an UAC to send QoS measurement samples to the SIP proxy after finishing a call. The goal is to have steady QoS monitoring without changing or extending the network topology. Are these extensions somewhere in CVS or isn't it intended to be released to the public? Andy

4 7

[Serusers] Change INVITE and Replace
by Benito Caracuel 10 Nov '06

10 Nov '06

Hello all, I need help about this: When an INVITE packet is received by SER, a Perl script changes the SDP of this INVITE and the result of this script (new INVITE with new SDP) is saved in a txt file. My question is: How can I do that SER sends the new INVITE, which is in a txt file. I have tried to do this with REPLACE command but I think is not possible because It doesn`t accept "regular expression" as substitute and I don´t know how save this new SDP in variables which can be process by SER. Please help me. Thank you very much. _________________________________________________________________ Consigue el nuevo Windows Live Messenger http://get.live.com/messenger/overview

4 3

[SerUsers]: Mysql DB at different machine
by Kamal.Mann＠t-systems.com 10 Nov '06

10 Nov '06

Hi All I am trying to implement ser database (mysql part) on different machine. Ser is installed on machine say 'A' and Mysql DB is on 'B'. In ser.cfg I changed url to connect to database everywhere in ser.cfg file with machine B's ip (eg. modparam("auth_db|permissions|uri_db|usrloc","db_url", "mysql://ser:heslo@10.25.117.86/ser") n in rest too). But I am getting the following error when I checked ser logs: Nov 10 14:12:40 vm01 /usr/sbin/ser[12979]: get_connection(): Connection not found in the pool Nov 10 14:12:40 vm01 /usr/sbin/ser[12979]: new_connection(): Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2) Why it tries for local Mysql sock?? Regards Kamal Mann

2 4

[Serusers] SER pre-release - PRESENCE ISSUES
by SER LIST 10 Nov '06

10 Nov '06

I installed the SER pre-release along w/ all modules. The problem is that presence now does not run (it was working w/ the presence-release-10.99). The errors I get are: ---------------------------------------------------------------------------------------------------------- set_mod_param_regex: parameter <xcap_root> not found in module <rls> ser: parse error (82,20-21): Can't set module parameter set_mod_param_regex: parameter <auth_xcap_root> not found in module <pa> parse error (97,20-21): Can't set module parameter ser: parse error (186,24-25): unknown command, missing loadmodule? parse error (255,26-27): unknown command, missing loadmodule? ERROR: bad config file (4 errors) ------------------------------------------------------------------------------------------------------------- I checked the pa module code and also the rls code and the parameters auth_xcap_root and xcap_root are NOT defined !!!! The error on line 185 and 255 refer to the "Lookup_domain" function. I verified that the domain module has been compiled and included in the ser.cfrg file >From: "Greger V. Teigre" <greger(a)teigre.com> >To: Jiri Kuthan <jiri(a)iptel.org> >CC: SER developer mailing list <serdev(a)iptel.org>, serusers(a)iptel.org >Subject: Re: [Serusers] new major SER pre-release, Ottendorf,is out for >testing >Date: Tue, 07 Nov 2006 15:37:17 +0100 > >Just a bit more info on this pre-release adding to Jiri's long post (if you >can take it ;-): > >* The mentioned http://www.iptel.org/ser/doc/010whatsnew will be updated as >we go with new documentation. Any registered user on iptel.org can add to >or modify documentation and all help is welcome. If you are uncertain about >how to proceed, send me an email > >* As Jiri mentioned, any input and help is appreciated. If you want to >contribute, but don't know what to do, have a look at the to-dos page: >http://www.iptel.org/todos > >* This pre-release is, without basic documentation, a bit for the die-hard. >This is an overview of what we are working (and need help) on to make it >easier accessible: - A tar source package ready for simple update from >CVS (similar to the ONsip.org source package) > - Better "kickstart" instructions, including more details in What's new >0.10 (above link) > - Updated SER - Getting Started configuration files and updated SER - >Getting Started document > - Database migration script (contact me if you want to try out an >existing python migration script and test/improve) > >* Also, in addition to the ser.cfg and Nils recently commited ser-basic.cfg >in CVS, take a look at an example config file: >http://www.iptel.org/basic_changes_in_configuration_file_0 > >*An optimized NAT-handled config file can be found here: >http://www.iptel.org/ser/howtos/optimizing_the_use_of_rtp_proxy > >Happy testing! >g-) > >Jiri Kuthan wrote: >>Ladies and gentlemen, girls and boys, >> >>the new testing pre-release of the best SIP proxy server ever, SER, is >>out! >> >>This major pre-release, codenamed Ottendorf, includes fundamental >>improvements to the software. The improvements allow SER users to achieve >>significantly higher managability and reliability by better data mode, >>interop features, security, etc. They recollect voluminous feedback we >>have received from iptel customers and open source users. In particular, >>the new features include but are not limited to: >>- improved data model which eliminates any possible ambiguities. >>- key under-the-hood changes with impact on performance: timer subsystem >> overhaul (now fast, with fine resolution, high accuracy and >>variability), TCP processing overhaul (higher scalability) >>- key reliability changes: DNS cache and IP blacklisting >>- key interop changes: support for STUN-bis, presence server aligned to >>latest >> IETF development, sanity module for early detection of invalid messages >>- more management: management features are now available via XML-RPC >>remotely, >> as well as via an alternative binary protocol >>- more security: ratelimit module, more check types in permissions module >>- scripting features: select operator in combination with revised AVPs, >> named flags and routes, more types of routes (onsend, timer, branch) >>- applications: presence has been greatly updated to keep pace with >>development in the IETF/SIMPLE (includes asynchronous subscription >> authorization, extended PIDF); web interface includes many new >>managibility >> features (multidomain, templates and multilanguage to name some). New >> scripting flexibility allows safely for features like individual >>ringing >> timers and direct access to SIP message elements. >>- new tools: Nils's script execution tree tool >> http://lists.iptel.org/pipermail/serdev/2006-October/008124.html >> and refurbished ser_ctl >>see the following links for some additional details: >>- http://www.iptel.org/ser/doc/010whatsnew - >>http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/NEWS?rev=HEAD >> >> >>Accompanying software, SEMS and SERWEB have been greatly overhauled as >>well. >>SEMS now features improved performance (single-threaded model now), >>service >>composition via python-IVR-scripting and B2BUA, adaptive playout buffer >>for >>better voice quality, and more. SERWEB features new multilevel >>administrative >>access system, multidomain support, per-domain customization, remote >>xml-rpc >>access, multilanguage support, and more. The new command-line utility, >>SER_CTL, provides also many great improvements over the previous tool. >> >>Interop >>------- >>Shortly, we have been keeping the bar high through our participation in >>the IETF >>and SIPITs. SER implements latest interop changes from the IETF (STUN-bis, >>presence >>+ XCAP, to name the most important ones) and both new features and legacy >>features were extensively and successfully tested middle of October in >>SIPit. >> >>Next Steps >>----------- >>Keep in mind, this is a pre-release for testing purposes. It is now call >>on >>users to test the software hard and provide feedback. In the meantime, the >>codebase remains frozen and we only commit patches fixing various bugs. >>Once >>we get good confidence that Ottendorf has stabilized and is being >>adequately >>documented, we will declare the final release. No date has been determined >>yet. >>Also, we consider changing CVS to SVN. >> >>Again, and this is very important: the final release will be as good as >>the >>feedback we obtain for this pre-release. Please do spend your time sending >>solid feedback back to [serusers]. >> >> >>Roadmap >>------- >>We keep rolling. We are still debating with our generous contributors what >>is the next possible good things to work on, the topics include but are >>not limited >>to as of now: >>- documentation. >>- databases are important. LDAP and BDB, DB-API revision. >>- interop: identity, TLS revision >>- advanced scripting (scripting engine, vars returns, AVPs) >>- uncountable improvements (futex, dev/rtc, tm/refcnts, tcp fd cache, >>.....) >> >>Trivia >>-------- >>The release name Ottendorf is coming from a place in Saxonian Switzerland, >>where many iptel fellows and their friends spent a weekend during this >>Indian summer. >>http://www.iptel.org/jan/gallery/saxonian_switzerland_2006/ >>http://www.iptel.org/~andi/2006-09-22-Ottendorf/ >>http://www.iptel.org/~vlada/saxon_switzerland/index.html >>http://www.ohlmeier.de/albums/ottendorf_2006/index.html >> >> >>Acknowledgments >>--------------- >>This grand achievement is result of extremely many folks who did lot >>of hard work. Out of folks affiliated with iptel, Andrei and Jan spent >>immense effort on moving SER forward, with great support from Hallik >>(command-line tool), Michal, Miklos, Nils, Tomas, Vaclav (presence), Vlada >>(new STUN). In this context, I would like to thank to our company for the >>great support. Recently, Bogdan Pintea greatly helped to make the upcoming >>release better and more stable under sponsorship of his company, iptego. >>transnexus provided osp support. >> >>SEMS has been greatly improved thanks to Stefan and Raphael. >>Credits for serweb go to Karel. >> >>Same importantly, SER has been immensely benefitting of contributors, >>to name in alphabetic order at least the 'top-of-the-iceberg' fellows: >>Bogdan Pintea, Cesc Santa, Greg Fausak, Greger Teigre, Hendrik Scholz, >><http://tutpro.com/>Juha Heinänen, Klaus Darilion, Maxim Sobolev, Peter >>Nixon, Uli Abend. >> >>Great support on the mailing list has been provided by Greger V. Teigre, >>Java Rockx, Klaus Darilion, Zeus Ng, and more. >> >>Simply, this list is too short and incomplete to capture all the >>generous contributions that came in -- my appologies for everyone >>else I have missed. >> >> >>Warnings and Disclaimers >>------------------------ >>This is a TESTING version. We are encouraging people to give it intense >>testing and provide us with feedback, so that we can properly assess >>what it takes to declare it final. The volume and number of changes >>is big, caution is advised. >> >>All GPL warnings and disclaimers apply. >> >>This is SER, which is not the derivate software known as openser. We have >>no ideas of in what extent and timeline derivate projects follow >>(or not) SER. >> >>------------------------------------------------------------------------------- >> >>To Give it a Try >>----------------- >>warning: this is just a 'jump-start' for black-belt users. It is not a >>replacement for documentation or a detailed step-by-step installation >>guide. It assumes knowledge of operating system, SIP, and SER. >> >> >>PREPARATION >>- make sure mysql is running. In the default configuration, SER assumes it >> to be running on localhost, if this does not hold, you need to change >> ser.cfg. >>- make sure python is installed (needed for new command-line utilities) >>SER INSTALLATION FROM CVS >>- Download from CVS/HEAD using instructions at >>http://www.iptel.org/downloads >>- Compile it: make include_modules="mysql xmlrpc" all >>- Install it the default way of your administrative privileges allow to: >> make include_modules="mysql xmlrpc" install >>- Download command-line tool (see more documentation under >> >>http://cvs.berlios.de/cgi-bin/viewcvs.cgi/*checkout*/ser/serctl/README?rev=…) >> - cvs co serctl >>- Initiate database: >> ./scripts/mysql/ser_mysql.sh create >>START >>- Start it. Call: ser >>- Begin provisioning users with the new serctl (the capital word are just >>place holders) >> # ser_ctl domain add DOMAINNAME >> # ser_ctl user add USERNAME@DOMAINNAME -p PASSWORD >> # If you want to have Asserted Identity header for your user >> # - ser_attr add uid=UID asserted_id="PID" >> # If you want to have gateway support >> # - ser_db add attr_types name=gw_ip rich_type=string raw_type=2 \ >> description="The gateway IP for the default ser.cfg" >>default_flags=33 >> # - ser_attr add global gw_ip=GATEWAY-IP >>- more things you may wish to do: >> - familiarize yourself with current ser.cfg >> - send us feedback. >> >> >> >> >> >>-- >>Jiri Kuthan http://iptel.org/~jiri/ >> >>_______________________________________________ >>Serusers mailing list >>Serusers(a)lists.iptel.org >>http://lists.iptel.org/mailman/listinfo/serusers >> >> >> >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers _________________________________________________________________ Experience Live Search from your PC or mobile device today. http://www.live.com/?mkt=en-ca

2 1

[Serusers] New developer
by Gustavo Passos Tourinho 10 Nov '06

10 Nov '06

Hello There! I'm trying to develop some new functions to be used at ser.cfg. I was looking the serdev.pdf but I didn't figure out how can I integrate my new function with ser.cfg. Is there some sort of "how to develop new modules for ser"? Thanks in advanced, Tourinho

2 1

[Serusers] Problem in using ser_ctl
by Ajay Kolambekar 10 Nov '06

10 Nov '06

Hi, I have installed SER on Redhat Linux 9 machine. I want to use ser_ctl to add/view users or domains. When I give command set_ctl user show I am getting an error like ser_ctl: /root/Python-2.5/lib/python2.5/site-packages/_mysql.so: undefined symbol: mysql_set_character_set: ImportError'> Does anybody know how to solve this problem? Thanks in advance, 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.

2 2

[Serusers] new major SER pre-release, Ottendorf, is out for testing
by Jiri Kuthan 10 Nov '06

10 Nov '06

Ladies and gentlemen, girls and boys, the new testing pre-release of the best SIP proxy server ever, SER, is out! This major pre-release, codenamed Ottendorf, includes fundamental improvements to the software. The improvements allow SER users to achieve significantly higher managability and reliability by better data mode, interop features, security, etc. They recollect voluminous feedback we have received from iptel customers and open source users. In particular, the new features include but are not limited to: - improved data model which eliminates any possible ambiguities. - key under-the-hood changes with impact on performance: timer subsystem overhaul (now fast, with fine resolution, high accuracy and variability), TCP processing overhaul (higher scalability) - key reliability changes: DNS cache and IP blacklisting - key interop changes: support for STUN-bis, presence server aligned to latest IETF development, sanity module for early detection of invalid messages - more management: management features are now available via XML-RPC remotely, as well as via an alternative binary protocol - more security: ratelimit module, more check types in permissions module - scripting features: select operator in combination with revised AVPs, named flags and routes, more types of routes (onsend, timer, branch) - applications: presence has been greatly updated to keep pace with development in the IETF/SIMPLE (includes asynchronous subscription authorization, extended PIDF); web interface includes many new managibility features (multidomain, templates and multilanguage to name some). New scripting flexibility allows safely for features like individual ringing timers and direct access to SIP message elements. - new tools: Nils's script execution tree tool http://lists.iptel.org/pipermail/serdev/2006-October/008124.html and refurbished ser_ctl see the following links for some additional details: - http://www.iptel.org/ser/doc/010whatsnew - http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/NEWS?rev=HEAD Accompanying software, SEMS and SERWEB have been greatly overhauled as well. SEMS now features improved performance (single-threaded model now), service composition via python-IVR-scripting and B2BUA, adaptive playout buffer for better voice quality, and more. SERWEB features new multilevel administrative access system, multidomain support, per-domain customization, remote xml-rpc access, multilanguage support, and more. The new command-line utility, SER_CTL, provides also many great improvements over the previous tool. Interop ------- Shortly, we have been keeping the bar high through our participation in the IETF and SIPITs. SER implements latest interop changes from the IETF (STUN-bis, presence + XCAP, to name the most important ones) and both new features and legacy features were extensively and successfully tested middle of October in SIPit. Next Steps ----------- Keep in mind, this is a pre-release for testing purposes. It is now call on users to test the software hard and provide feedback. In the meantime, the codebase remains frozen and we only commit patches fixing various bugs. Once we get good confidence that Ottendorf has stabilized and is being adequately documented, we will declare the final release. No date has been determined yet. Also, we consider changing CVS to SVN. Again, and this is very important: the final release will be as good as the feedback we obtain for this pre-release. Please do spend your time sending solid feedback back to [serusers]. Roadmap ------- We keep rolling. We are still debating with our generous contributors what is the next possible good things to work on, the topics include but are not limited to as of now: - documentation. - databases are important. LDAP and BDB, DB-API revision. - interop: identity, TLS revision - advanced scripting (scripting engine, vars returns, AVPs) - uncountable improvements (futex, dev/rtc, tm/refcnts, tcp fd cache, .....) Trivia -------- The release name Ottendorf is coming from a place in Saxonian Switzerland, where many iptel fellows and their friends spent a weekend during this Indian summer. http://www.iptel.org/jan/gallery/saxonian_switzerland_2006/ http://www.iptel.org/~andi/2006-09-22-Ottendorf/ http://www.iptel.org/~vlada/saxon_switzerland/index.html http://www.ohlmeier.de/albums/ottendorf_2006/index.html Acknowledgments --------------- This grand achievement is result of extremely many folks who did lot of hard work. Out of folks affiliated with iptel, Andrei and Jan spent immense effort on moving SER forward, with great support from Hallik (command-line tool), Michal, Miklos, Nils, Tomas, Vaclav (presence), Vlada (new STUN). In this context, I would like to thank to our company for the great support. Recently, Bogdan Pintea greatly helped to make the upcoming release better and more stable under sponsorship of his company, iptego. transnexus provided osp support. SEMS has been greatly improved thanks to Stefan and Raphael. Credits for serweb go to Karel. Same importantly, SER has been immensely benefitting of contributors, to name in alphabetic order at least the 'top-of-the-iceberg' fellows: Bogdan Pintea, Cesc Santa, Greg Fausak, Greger Teigre, Hendrik Scholz, <http://tutpro.com/>Juha Heinänen, Klaus Darilion, Maxim Sobolev, Peter Nixon, Uli Abend. Great support on the mailing list has been provided by Greger V. Teigre, Java Rockx, Klaus Darilion, Zeus Ng, and more. Simply, this list is too short and incomplete to capture all the generous contributions that came in -- my appologies for everyone else I have missed. Warnings and Disclaimers ------------------------ This is a TESTING version. We are encouraging people to give it intense testing and provide us with feedback, so that we can properly assess what it takes to declare it final. The volume and number of changes is big, caution is advised. All GPL warnings and disclaimers apply. This is SER, which is not the derivate software known as openser. We have no ideas of in what extent and timeline derivate projects follow (or not) SER. ------------------------------------------------------------------------------- To Give it a Try ----------------- warning: this is just a 'jump-start' for black-belt users. It is not a replacement for documentation or a detailed step-by-step installation guide. It assumes knowledge of operating system, SIP, and SER. PREPARATION - make sure mysql is running. In the default configuration, SER assumes it to be running on localhost, if this does not hold, you need to change ser.cfg. - make sure python is installed (needed for new command-line utilities) SER INSTALLATION FROM CVS - Download from CVS/HEAD using instructions at http://www.iptel.org/downloads - Compile it: make include_modules="mysql xmlrpc" all - Install it the default way of your administrative privileges allow to: make include_modules="mysql xmlrpc" install - Download command-line tool (see more documentation under http://cvs.berlios.de/cgi-bin/viewcvs.cgi/*checkout*/ser/serctl/README?rev=…) - cvs co serctl - Initiate database: ./scripts/mysql/ser_mysql.sh create START - Start it. Call: ser - Begin provisioning users with the new serctl (the capital word are just place holders) # ser_ctl domain add DOMAINNAME # ser_ctl user add USERNAME@DOMAINNAME -p PASSWORD # If you want to have Asserted Identity header for your user # - ser_attr add uid=UID asserted_id="PID" # If you want to have gateway support # - ser_db add attr_types name=gw_ip rich_type=string raw_type=2 \ description="The gateway IP for the default ser.cfg" default_flags=33 # - ser_attr add global gw_ip=GATEWAY-IP - more things you may wish to do: - familiarize yourself with current ser.cfg - send us feedback. -- Jiri Kuthan http://iptel.org/~jiri/

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[Users] auth_radius - Initializing problem
by exploiz 10 Nov '06

10 Nov '06

Hello all, I am install openSER follow (http://www.openser.org/docs/openser-radius-1.0.x.html). I have problem when run openser, below is a debug. And when remove *_radius.so the program can work. what's going wrong Thank you for all advice. BR, exploiz -------------------------------- LOG --------------------------------------------------- textops - initializing 0(0) DEBUG: init_mod: xlog 0(0) XLOG: initializing ... 0(0) DEBUG: init_mod: acc 0(0) ACC - initializing 0(0) find_export: found <load_tm> in module tm [/usr/local/lib/openser/modules/tm.so] 0(0) DEBUG: init_mod: auth 0(0) AUTH module - initializing 0(0) find_export: found <sl_send_reply> in module sl_module [/usr/local/lib/openser/modules/sl.so] 0(0) DEBUG: init_mod: auth_radius 0(0) auth_radius - Initializing 0(0) auth_radius: No `Cisco' vendor in Radius dictionary 0(0) find_export: found <bind_auth> in module auth [/usr/local/lib/openser/modules/auth.so] 0(0) ERROR: auth_radius: can't get code for the Digest-Response attribute 0(0) init_mod(): Error while initializing module auth_radius ERROR: error while initializing modules 0(0) XLOG: destroy module ... 0(0) DEBUG: tm_shutdown : start 0(0) DEBUG: unlink_timer_lists : emptying DELETE list 0(0) DEBUG: tm_shutdown : emptying hash table 0(0) DEBUG: tm_shutdown : releasing timers 0(0) DEBUG: tm_shutdown : removing semaphores 0(0) DEBUG: tm_shutdown : destroying tmcb lists 0(0) DEBUG: tm_shutdown : done 0(81840) shm_mem_destroy 0(81840) destroying the shared memory lock --------------------------------------------------------------------------- -------openser.cfg-------------------------------------------- .... loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/xlog.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/acc.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_radius.so" loadmodule "/usr/local/lib/openser/modules/group_radius.so" loadmodule "/usr/local/lib/openser/modules/avp_radius.so" #loadmodule "/usr/local/lib/openser/modules/auth_db.so" .... ---------------------------------------------------------------------------------------------- -- บุญรักษาทุกทั่น exploiz

3 3

Re[2]: [Users] pstn-to-openser, 403 Forbiden
by Ion Minzu 09 Nov '06

09 Nov '06

Hello Norman, thanks, Norman after I have closed the domain authorisation (proxy_authorize), everything works. It looks for me too sophisticated to use permission and domain module. I will try another method. now I want to make restrictions for voip users, I mean who has the right to make a call to pstn and who has not. someone has some ideeas? Wednesday, November 8, 2006, 2:35:34 PM, you wrote: > Hello Ion, > Perhaps the "pstn" device is not responding to the authorization > request. A SIP trace will show you if this is happening. > "ngrep -W byline port 5060" should be all you need to trace SIP messages > (if you're listening on eth1, then you might want to add "-d eth1" when > invoking ngrep). Older versions of ngrep did not have the "-W byline" > option. If this is your case, you probably should upgrade ngrep. > A possible solution to you problem might be to use the "permissions" > module and the "trusted" table. Then, instead of > (www_authorize/www_challenge or in addition to, depending on your > particular situation) you might want to use the "allow_trusted()" > command to authenticate incoming INVITEs. > Below is a little example > <snip> > if (!is_from_local() && !allow_trusted()) { > if (!proxy_authorize("","subscriber")) { > proxy_challenge("","0"); > exit; > } else if (!check_from()) { > sl_send_reply("403", "Use From=ID"); > exit; > }; > }; > </snip> > Regards, > Norm > Ion Minzu wrote: >> Hello , >> >> I have connected openser with pstn through cisco. when I make a >> call from voip network to pstn it's ok. >> but from pstn to voip I have a problem:openser answers 403 forbiden. >> in openser I do the authorisation on mysql, I have disabled authorisation on sip >> gateway: >> >> if (src_ip!=X.X.X.X) { >> if (!www_authorize("DOMAIN.COM","subscriber")) { >> www_challenge("DOMAIN.COM","0"); >> exit; >> } >> }; >> >> What is the problem? >> >> X.X.X.X is cisco >> >> U X.X.X.X:54177 -> 172.17.6.2:5060 >> INVITE sip:820022@172.17.6.2:5060 SIP/2.0..Via: SIP/2.0/UDP >> X.X.X.X:5060..From: >> <sip:022250699@X.X.X.X>;tag=1A0FBC30-1472..To: <sip:820022@172.1 >> 7.6.2>..Date: Wed, 08 Nov 2006 11:03:14 GMT..Call-ID: >> 906DA628-6E4F11DB-9034EA4F-E981BA1F@X.X.X.X..Supported: >> timer,100rel..Min-SE: 1800..Cisco-Guid >> : 2422905184-1850675675-2419190351-3917593119..User-Agent: >> Cisco-SIPGateway/IOS-12.x..Allow: INVITE, OPTIONS, BYE, CANCEL, >> ACK, PRACK, COMET, REFER, SUBS >> CRIBE, NOTIFY, INFO..CSeq: 101 INVITE..Max-Forwards: >> 6..Remote-Party-ID: >> <sip:022250699@X.X.X.X>;party=calling;screen=yes;privacy=off..Timestamp: >> 116 >> 2983794..Contact: <sip:022250699@X.X.X.X:5060>..Expires: >> 180..Allow-Events: telephone-event..Content-Type: >> application/sdp..Content-Length: 235....v= >> 0..o=CiscoSystemsSIP-GW-UserAgent 1226 5023 IN IP4 X.X.X.X..s=SIP >> Call..c=IN IP4 X.X.X.X..t=0 0..m=audio 16642 RTP/AVP 18 19..c=IN IP4 >> X.X.X.X..a=rtpmap:18 G729/8000..a=fmtp:18 >> annexb=no..a=rtpmap:19 CN/8000..a=ptime:20.. >> # >> U 172.17.6.2:5060 -> X.X.X.X:5060 >> SIP/2.0 403 Use From=ID..Via: SIP/2.0/UDP X.X.X.X:5060..From: >> <sip:022250699@X.X.X.X>;tag=1A0FBC30-1472..To: >> >> <sip:820022@172.17.6.2>;tag=329cfeaa6ded039da25ff8cbb8668bd2.13ec..Call-ID: >> 906DA628-6E4F11DB-9034EA4F-E981BA1F@X.X.X.X..CSeq: 101 >> INVITE..Server: OpenSer (1.1.0-tls (x86_64/linux))..C >> ontent-Length: 0..Warning: 392 172.17.6.2:5060 "Noisy >> feedback tells: pid=32240 req_src_ip=X.X.X.X req_src_port=54177 >> in_uri=sip:820022@172.17.6.2:5 >> 060 out_uri=sip:820022@172.17.6.2:5060 via_cnt==1".... >> # >> U X.X.X.X:54177 -> 172.17.6.2:5060 >> ACK sip:820022@172.17.6.2:5060 SIP/2.0..Via: SIP/2.0/UDP >> X.X.X.X:5060..From: >> <sip:022250699@X.X.X.X>;tag=1A0FBC30-1472..To: <sip:820022@172.17.6 >> .2>;tag=329cfeaa6ded039da25ff8cbb8668bd2.13ec..Date: Wed, 08 Nov >> 2006 11:03:14 GMT..Call-ID: >> 906DA628-6E4F11DB-9034EA4F-E981BA1F@X.X.X.X..Max-Forward >> s: 6..Content-Length: 0..CSeq: 101 ACK.... >> >> >> Best regards, >> Ion Minzu, >> Specialist Tehnologii Informationale, >> Administrator de sistem al Centrului de certificare, >> Administrator VoIP, >> I.S."Centrul de Telecomunicatii Speciale", >> tel:250-517 (office), 069501208 (mob), 382869185 (ICQ) >> mailto:ion.minzu@cts.md >> >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >> >> >> -- Best regards, Ion mailto:ion.minzu@cts.md

3 2

[Users] OpenSER Summit - BoF slides
by Daniel-Constantin Mierla 09 Nov '06

09 Nov '06

Hello, the slides presented Tuesday, during BoF, are now available online: http://www.openser.org/events/2006-OpenSER-Summit/slides/ Those from the Wednesday, need to be collected in the final version and will show up there as they arrive. Also, we are collecting photos/videos and any other story about the summit, please sent them to us (team(a)openser.org) to make them available. For the pictures/videos will be good if you can make them available for download somewhere, rather than sending via email. Photo albums will be generated and uploaded these days, to be easier to browse. Briefings will be posted as well on the web site. Thanks to all participants there, many of them resisted the marathon of slides, the event lasting one more hour than VoN conference, being the ones locking up the ICC at the end of the day :-) . Cheers, Daniel

1 0

[Serusers] Problem in using ser_ctl
by Ajay Kolambekar 09 Nov '06

09 Nov '06

Hi, I have installed SER on Redhat Linux 9 machine. I want to use ser_ctl to add/view users or domains. When I give command set_ctl user show I am getting an error like ser_ctl: /root/Python-2.5/lib/python2.5/site-packages/_mysql.so: undefined symbol: mysql_set_character_set: ImportError'> Does anybody know how to solve this problem? Thanks in advance, 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.

1 0

[Users] openser + presence module (presence.so) is not compiling
by prateek kapoor 09 Nov '06

09 Nov '06

Hi All, I have installed openser version from 1.1.0 debian package installer and it is working fine. But when I am trying to include the presence module (presence.so), it is not there. When I am trying to recompile it from source tarball I am getting a lot of dependency errors. So can anyone tell me from where I can get the 'presence.so' ? I can see a 'pa' (presence agent) module. I would also like to know what is the difference between 'pa' and 'presence' modules ? Any help will be really appreciated. Regards, Prateek --------------------------------- Everyone is raving about the all-new Yahoo! Mail beta.

2 1

[serusers]: Presence Module
by Kamal.Mann＠t-systems.com 09 Nov '06

09 Nov '06

Hi All I need to use SER as a stand alone presence server. How to install/configure ser as presence server? Any kind of related links, documents etc. Do I need some external libraries too for compiling SER code to run it as presence server or any binary package for all? Is their any proper document is already present? Thanks in anticipation Kamal Mann

2 2

[Users] usrloc-cl
by Mark Price 09 Nov '06

09 Nov '06

Hi, It seems that it has been about a year since usrloc-cl was mentioned on this list: http://www.openser.org/pipermail/users/2005-October/001424.html What is the status of usrloc-cl on openser? Has it been integrated into openser's usrloc as the referenced email suggests is a good idea? Is the usrloc-cl functionality available in the openser codebase? Thanks, Mark Price

2 1

[Users] openser summit
by Klaus Darilion 08 Nov '06

08 Nov '06

Hi guys! It was a pleasure for me to attend the openser summit and meat you all - it's funny to map faces to names only known from the mailing list. Although my presentation exceeded the suggested timeframe I could not explain all things in detail. Thus, If there are questions left, please feel free to ask me - the mailinglist is preferred. regards klaus PS: The VON t-mobile login works also at airport Tegel :-)

1 0

[Serusers] Problems with CANCEL and INVITE
by Benito Caracuel 08 Nov '06

08 Nov '06

Hello all, I have a problem when I try to cancel an invite, because I have modified the invite method. In my ser.cfg: if ((method=="INVITE")&&(!lookup("location"))) { exec_msg("printenv SRCIP; /bin/bash /usr/local/ser_snapshot/sbin/sms"); sl_send_reply("100", "Trying"); exec_msg("sleep 10"); if (!lookup("location")){ sl_send_reply("100", "Trying"); exec_msg("sleep 10"); } else { t_relay(); break; } if (!lookup("location")){ sl_send_reply("100", "Trying"); exec_msg("sleep 10"); } else { t_relay(); break; } if (!lookup("location")){ sl_send_reply("404", "Not on-line"); break; } else { t_relay(); break; } }; If I send an INVITE to user2, who is off-line, SER sends a SMS to this user2 and meanwhile , sends TRYINGS to me, until user2 sends a REGISTER. I have a problem, because If I send a CANCEL, SER still sends TRYINGS to me and SER doesn`t send "487 Request Terminated" and SER doesn´t associate CANCEL and INVITE. How can I solve this? any idea? I think that maybe, If I use a mysql table to save CSEQ of CANCEL, and before SER sends TRYINGS, SER checks if CSEQ is into table but I don´t know if SER can save CANCEL information in a mysql table. Could someone help me, please? Thank you very much. _________________________________________________________________ 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

1 0

[Serusers] Accounting mismatch
by inge 08 Nov '06

08 Nov '06

Hi all, When my users make a call, the domain field in "acc" table from SER database have a constant FQDN something like aa.mydomain.ext. But there is some case where the domain field contain IP of SER, not FQDN. The problem is that when I consult accounting page I only see the call where the domain field contains aa.mydomain.ext. Is it possible to see both call : FQDN/IP ? Thank for your help. Best regards.

1 0

[Serusers] Cancelling an INVITE
by Benito Caracuel 08 Nov '06

08 Nov '06

Hello all, I have a problem when I try to cancel an invite, because I have modified the invite method. I'm going to explain the idea behind the problem, which is related to what we call "OffLine Invitation". An "Offline Invitation" consists of the SER set up as to DON'T send a "404 Not Found" when the Invitated party isn't registered. What is done is to keep the SER sending "100 Trying" for a while, and to send meanwhile a SMS to the Invitated -but not registered- party. That SMS is understood by the Application that's running at the Invited party as a "Register Request", so the Invitated party goes for a Register. That works OK, but the problem comes up when you try to CANCEL the Invitation request. In that case, as the Invitated party isn't still registered, it cannot take care of the CANCEL's lifecicle, so SER has to deal with it, "emulating" that it is the Invitated party. As to do this, we assume that no race condition is produced, and neither a Final response has been sent by the Invitated party. So what is needed is, according to SIP protocol, to answer to the CANCEL with "200 OK" and to the original INVITE with a "487 Request Terminated". The problem is obviously doing the INVITE - CANCEL match at the time the CANCEL gets to SER. We know there are scripts/procedures and tables in the SER architecture that can be used to do other tasks (as it is the "saving" of REGISTER packet onto a DB table), but, to be honest, we are a bit confused about if it can be done, or if it should need SER ad-hoc programming, or if there is a way to use an specific SER DB Table to do it... I post along my SER.CFG file, and an idea about how we were thinking that it could be solved (that is, a big picture; what we need indeed is to know if it is a good idea (having in mind SER architecture and scripting capabilities) or it cannot be done. Please, if it can be done, we'd like to hear a little more about what tables can be used, or if it's needed to create another new one, or at least, a clue to solve it. In my ser.cfg: if ((method=="INVITE")&&(!lookup("location"))) { exec_msg("printenv SRCIP; /bin/bash /usr/local/ser_snapshot/sbin/sms"); sl_send_reply("100", "Trying"); exec_msg("sleep 10"); if (!lookup("location")){ sl_send_reply("100", "Trying"); exec_msg("sleep 10"); } else { t_relay(); break; } if (!lookup("location")){ sl_send_reply("100", "Trying"); exec_msg("sleep 10"); } else { t_relay(); break; } if (!lookup("location")){ sl_send_reply("404", "Not on-line"); break; } else { t_relay(); break; } }; If I send an INVITE to user2, who is off-line (not registered), SER sends a SMS to this user2 and meanwhile, sends TRYINGS to me, until user2 becomes REGISTERED (the sent SMS makes the Invited party Register, so it can receive the Invite, as It was told before). I have a problem, because If I send a CANCEL, SER still sends TRYINGS to me and SER doesn`t send "487 Request Terminated" and SER doesn´t associate CANCEL and INVITE. How can I solve this? any idea? I think that maybe, If I use a mysql table to save CSEQ of CANCEL, and before SER sends TRYINGS (I mean the next TRYINGs following the first), SER checks if CSEQ is into table, but I don´t know if SER can save CANCEL information in a mysql table. I know that there is a command "save" to save REGISTER packet into location table. Is there any similar command to save CSEQ of CANCEL (and/or other, as Call-Id, Branch, Request-Uri and To) in one table? Could someone help me, please? We are in a hurry with our customer, so fast key to a solution will be really appreciated. Thank you very much. _________________________________________________________________ Llama a tus amigos de PC a PC: ¡Es GRATIS! http://get.live.com/messenger/overview

1 0

[Users] Not Hung Up when i Keep the Server in Router_firewall( NETGEAR)
by raviprakash sunkara 08 Nov '06

08 Nov '06

Hello User, A small Question , When I My Sip Server without any firewall_ Router .. Hung up is working Fine,,,,,, But I Keep Sip Server with in a Firewall_router, it not Hung upping , When i saw logs from X-lite in Bye method it Changing the uri into Private IP, But should it public IP( THis is not happening ) Please help me............. -- 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

[Serusers] mediaproxy and ser ottendorf incompatible right now
by samuel 08 Nov '06

08 Nov '06

I was trying to use mediaproxy module with SER ottendorf and got next error: error: mediaproxy/mod_init(): can't find is_from_local and/or is_uri_host_local functions. Check if domain.so is loaded domain.so is effectively loaded BUT these functions have been replaced by is_local,lookup_domain, and get_did in new SER version Is AG Projects taking care of this issue?!? Can someone shed dome light on this issue!??!?!? Thanks a lot, Samuel.

1 0

Re: [Users] TM : retransmission timers
by Rao Ramaratnamma 08 Nov '06

08 Nov '06

Greg, The email from you and others is giving me the type of most direct fair insights I am looking for . You know, just some practical "what is all this about" type of things for my selforientation. Just that openser link you are attaching unfortunately looks too much like openser marketing material. It is saying that ser was discontinued but actually the information I'm getting from others telling me that it has some features which are very important and openser do not have. With all respect, it is understandable that openser people write nicely about themselves and less nicely about others but it remains so confusing that I do not have confidence to move with this particular material. The openser info just appears too little credible for me. I think I am beginning to get the bottom line and I hope I will create my mind soon and if people have other information for me pls send it. Especially, objective technical information including this timer discussion is helping me. You know, better understanding what is the difference in "what they think is important". That would surely help me beyound the filosophical "cats or dogs, who is a better friend of man" information aqusition phase people surely like to be engaged in :-) Thank you all guys for keeping helping me a lot, rr ----- Original Message ---- From: Greg Fausak <lgfausak(a)gmail.com> To: Rao Ramaratnamma <raramarat(a)yahoo.com> Cc: Christian Schlatter <cs(a)unc.edu>; users(a)openser.org Sent: Wednesday, November 8, 2006 5:21:41 AM Subject: Re: [Users] TM : retransmission timers You can go to openser.org's website and read about howhow openser and ser are related. http://www.openser.org/index.php?option=com_content&task=view&id=40&Itemid=… They are now two separate bodies of code, maintained by two different groups. Each one has gone off and implemented and support what they think is important. Both projects are open source projects, anyone is certainly welcome to contribute to either. -g On Nov 7, 2006, at 4:15 PM, Rao Ramaratnamma wrote: the ser ottendorf announcement does mention improved timers. Cannot openser include this feature too and cannot I merge ser with openser for good timers? I am still trying to understand the difference between ser and openser but standart compliance seems to be very important matter! Cannot people provide me with some hints? I am sure that I am not the only who is asking the difference between ser and openser. ser documentation does not appear uptodate, but the software as sannounced appears impressive. I have already asked this question but did not receive any answer. thank you in advance! rr ----- Original Message ---- From: Christian Schlatter <cs(a)unc.edu> To: users(a)openser.org Sent: Tuesday, November 7, 2006 10:52:56 PM Subject: Re: [Users] TM : retransmission timers Greg Fausak wrote: > Hello, > > I believe this is a well known bug. > Granularity of timers is 1 second. So, if you sign up for a timer to > be fired in 1 second it will happen anywhere between 0 seconds and 1 > second. > 2 seconds will happen between 1 and 2 seconds. I usually set up my > timers to be 2, 2, 4, 8. There are VOIP providers that are pretty > sticky about > the first 500ms. If you are using one of them you're out of luck. Yes, there is a timer process that wakes up every second to perform retransmissions. I was actually quite surprised that OpenSER, which is known to be very standards compliant, does not follow the RFC 3261 retransmission timeouts. On the other hand, the RFC 3261 timeout values are just suggestions and standards compliant SIP UA must accept shorter timeouts. Still it would be nice if OpenSER would support sub second timers, this would allow for shorter fail-over times. Christian > > I believe SER has made timer changes to support more exact timer > intervals. They are a completely different camp, with a different feature > set (although they share the same roots). > > -g > > > On 11/7/06, Jean-François SMIGIELSKI <jf-smig(a)ibelgique.com> wrote: >> Hello, >> >> I made strange observations about the intervals between >> retransmissions with the TM module. >> In my experiments, I used the default parameters for the TM module >> timers, and I sent an INVITE that cannot receive answers (it has a >> well known R-URI pattern that is forwarded to a place and port that >> nobody listen). >> >> When reading RFC3261, I expected to see intervals between >> retransmissions of |500ms|1s|2s|4s|8s|16s|. 7 transmissions, during 32s. >> >> But with OpenSER, (I have tested with the debian package 1.1.0-5 on a >> debian etch, and the cvs sources for 1.1.0 or 1.0.1compiled by >> myself), I can see intervals like <500ms, 2s, 4s, 4s,4s, ... until 26s >> are spent (9 sendings). The first interval is sometomes very short >> (40ms). >> >> Altough I like the sequence of 4s separated transmissions, I do not >> know why the first interval is so short, and why there is no sending >> after 1s. >> >> Did anybody observed such behaviours? Are they normal? >> >> Thanks in advance! >> >> JF Smigielski. >> >> >> ________________________________________________________________________ >> iBELGIQUE, exprimez-vous ! >> http://web.ibelgique.com/ >> >> _______________________________________________ >> 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 _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

1 0

[Users] TM : retransmission timers
by Jean-François SMIGIELSKI 08 Nov '06

08 Nov '06

Hello, I made strange observations about the intervals between retransmissions with the TM module. In my experiments, I used the default parameters for the TM module timers, and I sent an INVITE that cannot receive answers (it has a well known R-URI pattern that is forwarded to a place and port that nobody listen). When reading RFC3261, I expected to see intervals between retransmissions of |500ms|1s|2s|4s|8s|16s|. 7 transmissions, during 32s. But with OpenSER, (I have tested with the debian package 1.1.0-5 on a debian etch, and the cvs sources for 1.1.0 or 1.0.1compiled by myself), I can see intervals like <500ms, 2s, 4s, 4s,4s, ... until 26s are spent (9 sendings). The first interval is sometomes very short (40ms). Altough I like the sequence of 4s separated transmissions, I do not know why the first interval is so short, and why there is no sending after 1s. Did anybody observed such behaviours? Are they normal? Thanks in advance! JF Smigielski. ________________________________________________________________________ iBELGIQUE, exprimez-vous ! http://web.ibelgique.com/

5 6

[Serusers] grandstream configuration -- help
by Ravi ***** 08 Nov '06

08 Nov '06

Hi users , I am using ser-0.96 , and every thing seems to work fine . but when I make calls from grandstream ATA .. its is working properly but the problem is when the other party hung the phone SER cannot send BYE packets to grandstream NAT ip . It is sending to grandstream private-ip which inturns delayed for acct.stop packet unless grandstream hung the phone x-lite, is O.K and its sending to SER as tel.no@Nat-ip-address but grandstream is sending to SER as tel.no@private-ip-address so in record-route SER is strictly following this and making some small mistake,, like sending Bye request to private-ip How to solve this : -) Thank You. Regards, Ravi.

2 1

[Serusers] Problem with CANCEL messages and Cisco 7905
by Kristian Larsson 08 Nov '06

08 Nov '06

Hey! This is my first time posting to the list, although I'm a long time reader :) I have searched the archives as well as the internet and found no solution or answer to this so far. Perhaps my Google Kung Fu isn't what it should be. Anyhow, I have a cisco 7905 phone working together with a SER machine. When dialing to the 7905 phone from another phone and hanging up on the caller phone before answering the 7905, ie CANCEL before the call has actually been setup, I get a "SIP/2.0 481 Call Leg/Transaction Does Not Exist." back from the Cisco phone. It's a pretty basic configuration (basically the one used in the examples for call forwarding). What I want to know is how the phone identifies a Call Leg/Transaction. I have used ngrep quite extensively and found nothing. Most values in both the INVITE and CANCEL message are equal. I've also got a Thomson ATA box which does not show the same behaviour despite being connected to the same SER. What values are used to identify a transaction, how can I see if two messages are of the same transaction? Perhaps this is actually a user agent problem, but I don't really know where to turn for help with this 7905 (Cisco only provides support if you use their Call Manager solution). regards, Kristian. -- Kristian Larsson KLL-RIPE Network Engineer Net At Once [AS35706] email: kristian(a)netatonce.se irc: kll@ircnet phone: +46 470 592717 cell: +46 704 910401

2 2

[Users] pstn-to-openser, 403 Forbiden
by Ion Minzu 08 Nov '06

08 Nov '06

Hello , I have connected openser with pstn through cisco. when I make a call from voip network to pstn it's ok. but from pstn to voip I have a problem:openser answers 403 forbiden. in openser I do the authorisation on mysql, I have disabled authorisation on sip gateway: if (src_ip!=X.X.X.X) { if (!www_authorize("DOMAIN.COM","subscriber")) { www_challenge("DOMAIN.COM","0"); exit; } }; What is the problem? X.X.X.X is cisco U X.X.X.X:54177 -> 172.17.6.2:5060 INVITE sip:820022@172.17.6.2:5060 SIP/2.0..Via: SIP/2.0/UDP X.X.X.X:5060..From: <sip:022250699@X.X.X.X>;tag=1A0FBC30-1472..To: <sip:820022@172.1 7.6.2>..Date: Wed, 08 Nov 2006 11:03:14 GMT..Call-ID: 906DA628-6E4F11DB-9034EA4F-E981BA1F@X.X.X.X..Supported: timer,100rel..Min-SE: 1800..Cisco-Guid : 2422905184-1850675675-2419190351-3917593119..User-Agent: Cisco-SIPGateway/IOS-12.x..Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBS CRIBE, NOTIFY, INFO..CSeq: 101 INVITE..Max-Forwards: 6..Remote-Party-ID: <sip:022250699@X.X.X.X>;party=calling;screen=yes;privacy=off..Timestamp: 116 2983794..Contact: <sip:022250699@X.X.X.X:5060>..Expires: 180..Allow-Events: telephone-event..Content-Type: application/sdp..Content-Length: 235....v= 0..o=CiscoSystemsSIP-GW-UserAgent 1226 5023 IN IP4 X.X.X.X..s=SIP Call..c=IN IP4 X.X.X.X..t=0 0..m=audio 16642 RTP/AVP 18 19..c=IN IP4 X.X.X.X..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:19 CN/8000..a=ptime:20.. # U 172.17.6.2:5060 -> X.X.X.X:5060 SIP/2.0 403 Use From=ID..Via: SIP/2.0/UDP X.X.X.X:5060..From: <sip:022250699@X.X.X.X>;tag=1A0FBC30-1472..To: <sip:820022@172.17.6.2>;tag=329cfeaa6ded039da25ff8cbb8668bd2.13ec..Call-ID: 906DA628-6E4F11DB-9034EA4F-E981BA1F@X.X.X.X..CSeq: 101 INVITE..Server: OpenSer (1.1.0-tls (x86_64/linux))..C ontent-Length: 0..Warning: 392 172.17.6.2:5060 "Noisy feedback tells: pid=32240 req_src_ip=X.X.X.X req_src_port=54177 in_uri=sip:820022@172.17.6.2:5 060 out_uri=sip:820022@172.17.6.2:5060 via_cnt==1".... # U X.X.X.X:54177 -> 172.17.6.2:5060 ACK sip:820022@172.17.6.2:5060 SIP/2.0..Via: SIP/2.0/UDP X.X.X.X:5060..From: <sip:022250699@X.X.X.X>;tag=1A0FBC30-1472..To: <sip:820022@172.17.6 .2>;tag=329cfeaa6ded039da25ff8cbb8668bd2.13ec..Date: Wed, 08 Nov 2006 11:03:14 GMT..Call-ID: 906DA628-6E4F11DB-9034EA4F-E981BA1F@X.X.X.X..Max-Forward s: 6..Content-Length: 0..CSeq: 101 ACK.... Best regards, Ion Minzu, Specialist Tehnologii Informationale, Administrator de sistem al Centrului de certificare, Administrator VoIP, I.S."Centrul de Telecomunicatii Speciale", tel:250-517 (office), 069501208 (mob), 382869185 (ICQ) mailto:ion.minzu@cts.md

3 2

[Users] OpenSER + SEMS + Nat?
by Brandon Armstead 08 Nov '06

08 Nov '06

Hello, I currently have an openser server running on port 5060, and a second that handles voicemail / stateful transaction stuff on port 5070. So, I am having proxy A forward to proxy B to do the handling which are both on the same server, although there seems to be a problem with NAT'ing, when I call 'voicemail' or something I can not hear anything. It seems to be because it is sending the RTP data to the wrong location? It should be sending it back to public-ip-addr-1 when it is sending to an internal ip address 192.168.1.* Here is the log from sems: 101 44811:1466724767 <sip:public-ip-addr-1:5060;nat=yes;ftag=dd9bbf561d35ee06o0;lr=on> sip:public-ip-addr-1:5060;nat=yes;ftag=dd9bbf561d35ee06o0;lr=on P-MsgFlags: 0 UA: Linksys/PAP2-3.1.9(LSa) P-Email-Address: test(a)test.com . v=0 o=- 9200357 9200357 IN IP4 192.168.1.241 s=- c=IN IP4 192.168.1.241 t=0 0 m=audio 16420 RTP/AVP 0 2 4 8 18 96 97 98 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv > (7856) DEBUG: execute (AmInterfaceHandler.cpp:99): req.method = <INVITE> (7856) DEBUG: execute (AmInterfaceHandler.cpp:100): req.user = <115> (7856) DEBUG: execute (AmInterfaceHandler.cpp:101): req.domain = <public-ip-addr-1> (7856) DEBUG: execute (AmInterfaceHandler.cpp:102): req.dstip = <public-ip-addr-1> (7856) DEBUG: execute (AmInterfaceHandler.cpp:103): req.port = <5070> (7856) DEBUG: execute (AmInterfaceHandler.cpp:104): req.r_uri = < sip:115@public-ip-addr-1:5070> (7856) DEBUG: execute (AmInterfaceHandler.cpp:105): req.from_uri = < sip:1000@192.168.1.241:5060> (7856) DEBUG: execute (AmInterfaceHandler.cpp:106): req.from = <Brandon1 < sip:1000@test.mydomain.com>> (7856) DEBUG: execute (AmInterfaceHandler.cpp:107): req.to = << sip:115@test.mydomain.com>> (7856) DEBUG: execute (AmInterfaceHandler.cpp:108): req.callid = < 81e2d596-978fccc6(a)192.168.1.241> (7856) DEBUG: execute (AmInterfaceHandler.cpp:109): req.from_tag = <dd9bbf561d35ee06o0> (7856) DEBUG: execute (AmInterfaceHandler.cpp:110): req.to_tag = <> (7856) DEBUG: execute (AmInterfaceHandler.cpp:113): cseq_str = <101> (7856) DEBUG: execute (AmInterfaceHandler.cpp:119): cseq = <101>(101) (7856) DEBUG: execute (AmInterfaceHandler.cpp:121): req.key = <44811:1466724767> (7856) DEBUG: execute (AmInterfaceHandler.cpp:122): req.route = <<sip:public-ip-addr-1:5060;nat=yes;ftag=dd9bbf561d35ee06o0;lr=on>> (7856) DEBUG: execute (AmInterfaceHandler.cpp:123): req.next_hop = <sip:public-ip-addr-1:5060;nat=yes;ftag=dd9bbf561d35ee06o0;lr=on> (7856) DEBUG: execute (AmInterfaceHandler.cpp:128): hdrs = <P-MsgFlags: 0 UA: Linksys/PAP2-3.1.9(LSa) P-Email-Address: test(a)test.com > (7856) DEBUG: execute (AmInterfaceHandler.cpp:133): body = <v=0 o=- 9200357 9200357 IN IP4 192.168.1.241 s=- c=IN IP4 192.168.1.241 t=0 0 m=audio 16420 RTP/AVP 0 2 4 8 18 96 97 98 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv > (7856) DEBUG: execute (AmInterfaceHandler.cpp:144): Request OK: dispatch it! (7856) DEBUG: postEvent (AmEventQueue.cpp:48): AmEventQueue: trying to post event (7856) DEBUG: postEvent (AmEventQueue.cpp:56): AmEventQueue: event posted (7856) DEBUG: processEvents (AmEventQueue.cpp:69): before processing event (7856) DEBUG: process (AmSession.cpp:421): AmSession::process (7856) DEBUG: process (AmSession.cpp:425): Session received SIP Event (7856) DEBUG: onSipRequest (AmSession.cpp:472): onSipRequest: method = INVITE (7856) DEBUG: parse_sdp_media (AmSdp.cpp:632): next_line=<a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv > (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:658): parse_sdp_attribute: s=rtpmap:0 PCMU/8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:693): sdp attribute: pt=0; enc=PCMU; cr=8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:658): parse_sdp_attribute: s=rtpmap:2 G726-32/8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:693): sdp attribute: pt=2; enc=G726-32; cr=8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:658): parse_sdp_attribute: s=rtpmap:4 G723/8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:693): sdp attribute: pt=4; enc=G723; cr=8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:658): parse_sdp_attribute: s=rtpmap:8 PCMA/8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:693): sdp attribute: pt=8; enc=PCMA; cr=8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:658): parse_sdp_attribute: s=rtpmap:18 G729a/8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:693): sdp attribute: pt=18; enc=G729a; cr=8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:658): parse_sdp_attribute: s=rtpmap:96 G726-40/8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:693): sdp attribute: pt=96; enc=G726-40; cr=8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:658): parse_sdp_attribute: s=rtpmap:97 G726-24/8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:693): sdp attribute: pt=97; enc=G726-24; cr=8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:658): parse_sdp_attribute: s=rtpmap:98 G726-16/8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:693): sdp attribute: pt=98; enc=G726-16; cr=8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:658): parse_sdp_attribute: s=rtpmap:100 NSE/8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:693): sdp attribute: pt=100; enc=NSE; cr=8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:658): parse_sdp_attribute: s=rtpmap:101 telephone-event/8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:693): sdp attribute: pt=101; enc=telephone-event; cr=8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:658): parse_sdp_attribute: s=fmtp:101 0-15 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:658): parse_sdp_attribute: s=ptime:30 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:740): unknown attribute definition 'ptime:30' (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:658): parse_sdp_attribute: s=sendrecv (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:745): flag definition is not yet supported (sendrecv) (7856) DEBUG: parse_sdp_media (AmSdp.cpp:651): ret=0; next_line=<> (7856) DEBUG: getCompatiblePayload (AmSdp.cpp:341): payload found: 0/PCMU/8000 (7856) DEBUG: getCompatiblePayload (AmSdp.cpp:344): using global address: 192.168.1.241 (7856) DEBUG: negotiate (AmSession.cpp:247): remote party supports telephone events (pt=101) (7856) DEBUG: setRAddr (AmRtpStream.cpp:346): RTP remote address set to 192.168.1.241:16420 (7856) DEBUG: init (AmRtpAudio.cpp:146): AmRtpAudio::init(...) (7856) DEBUG: acceptAudio (AmSession.cpp:607): Sending Rtp data to 192.168.1.241/16420 (7856) DEBUG: updateStatusReply (AmSipDialog.cpp:86): reply: transaction found! (7856) DEBUG: updateStatusReply (AmSipDialog.cpp:125): req.method = INVITE; t.method = INVITE (7856) DEBUG: init (AmCtrlInterface.cpp:231): AmUnixCtrlInterface::init @ /tmp/5D08F0A1-4552226600080041-08108600 (7856) DEBUG: write_to_socket (AmUtils.cpp:588): sending: <:t_reply:/tmp/5D08F0A1-4552226600080041-08108600 200 OK 44811:1466724766 193812C5-45522256000F0C97-08103000 Contact: <sip:115@public-ip-addr-1:5070> Content-Type: application/sdp . v=0 o=username 0 0 IN IP4 public-ip-addr-1 s=session c=IN IP4 public-ip-addr-1 t=0 0 m=audio 10022 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 . > (7856) DEBUG: write_to_socket (AmUtils.cpp:614): write to unix socket: completed (7856) DEBUG: cacheMsg (AmCtrlInterface.cpp:186): recv-ed: <200 Succeeded> (7856) DEBUG: processEvents (AmEventQueue.cpp:71): event processed (7856) DEBUG: processEvents (AmEventQueue.cpp:69): before processing event (7856) DEBUG: process (AmSession.cpp:421): AmSession::process (7856) DEBUG: process (AmSession.cpp:425): Session received SIP Event (7856) DEBUG: onSipRequest (AmSession.cpp:472): onSipRequest: method = INVITE (7856) DEBUG: parse_sdp_media (AmSdp.cpp:632): next_line=<a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv > (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:658): parse_sdp_attribute: s=rtpmap:0 PCMU/8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:693): sdp attribute: pt=0; enc=PCMU; cr=8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:658): parse_sdp_attribute: s=rtpmap:2 G726-32/8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:693): sdp attribute: pt=2; enc=G726-32; cr=8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:658): parse_sdp_attribute: s=rtpmap:4 G723/8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:693): sdp attribute: pt=4; enc=G723; cr=8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:658): parse_sdp_attribute: s=rtpmap:8 PCMA/8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:693): sdp attribute: pt=8; enc=PCMA; cr=8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:658): parse_sdp_attribute: s=rtpmap:18 G729a/8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:693): sdp attribute: pt=18; enc=G729a; cr=8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:658): parse_sdp_attribute: s=rtpmap:96 G726-40/8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:693): sdp attribute: pt=96; enc=G726-40; cr=8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:658): parse_sdp_attribute: s=rtpmap:97 G726-24/8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:693): sdp attribute: pt=97; enc=G726-24; cr=8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:658): parse_sdp_attribute: s=rtpmap:98 G726-16/8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:693): sdp attribute: pt=98; enc=G726-16; cr=8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:658): parse_sdp_attribute: s=rtpmap:100 NSE/8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:693): sdp attribute: pt=100; enc=NSE; cr=8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:658): parse_sdp_attribute: s=rtpmap:101 telephone-event/8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:693): sdp attribute: pt=101; enc=telephone-event; cr=8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:658): parse_sdp_attribute: s=fmtp:101 0-15 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:658): parse_sdp_attribute: s=ptime:30 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:740): unknown attribute definition 'ptime:30' (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:658): parse_sdp_attribute: s=sendrecv (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:745): flag definition is not yet supported (sendrecv) (7856) DEBUG: parse_sdp_media (AmSdp.cpp:651): ret=0; next_line=<> (7856) DEBUG: getCompatiblePayload (AmSdp.cpp:341): payload found: 0/PCMU/8000 (7856) DEBUG: getCompatiblePayload (AmSdp.cpp:344): using global address: 192.168.1.241 (7856) DEBUG: negotiate (AmSession.cpp:247): remote party supports telephone events (pt=101) (7856) DEBUG: setRAddr (AmRtpStream.cpp:346): RTP remote address set to 192.168.1.241:16420 (7856) DEBUG: init (AmRtpAudio.cpp:146): AmRtpAudio::init(...) (7856) DEBUG: acceptAudio (AmSession.cpp:607): Sending Rtp data to 192.168.1.241/16420 (7856) DEBUG: updateStatusReply (AmSipDialog.cpp:86): reply: transaction found! (7856) DEBUG: updateStatusReply (AmSipDialog.cpp:125): req.method = INVITE; t.method = INVITE (7856) DEBUG: init (AmCtrlInterface.cpp:231): AmUnixCtrlInterface::init @ /tmp/75EE09BE-4552226600080514-08108600 (7856) DEBUG: write_to_socket (AmUtils.cpp:588): sending: <:t_reply:/tmp/75EE09BE-4552226600080514-08108600 200 OK 44811:1466724767 193812C5-45522256000F0C97-08103000 Contact: <sip:115@public-ip-addr-1:5070> Content-Type: application/sdp . v=0 o=username 0 0 IN IP4 public-ip-addr-1 s=session c=IN IP4 public-ip-addr-1 t=0 0 m=audio 10022 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 . > (7856) DEBUG: write_to_socket (AmUtils.cpp:614): write to unix socket: completed (7856) DEBUG: cacheMsg (AmCtrlInterface.cpp:186): recv-ed: <200 Succeeded> (7856) DEBUG: processEvents (AmEventQueue.cpp:71): event processed (7856) DEBUG: run (AmSession.cpp:292): 81e2d596-978fccc6(a)192.168.1.241 dlg.getUACTransPending() = 0 Here is my config for "proxy B" the transaction handling with SEMS that "proxy A" sends to: # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # ser for sems configuration # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E) check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5070 children=4 unix_sock="/tmp/openser_sems_sock" # ------------------ module loading ---------------------------------- mpath="/usr/local/lib/openser/modules" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "avpops.so" # ----------------- setting module-specific parameters --------------- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) modparam("avpops", "avp_aliases", "email=i:67") modparam("tm", "pass_provisional_replies", 1) # appends for INVITE to voicemail modparam("tm", "tw_append", "voicemail_headers:UA=$hdr(User-Agent);P-Email-Address=$avp(s:email)") # appends for dtmf per INFO modparam("tm", "tw_append", "info_append:Content-Length=$hdr(Content-Length);Content-Type=$hdr(Content-Type);$rb") # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); return; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); return; }; if (method != "ACK" && method != "INVITE" && method != "BYE" && method != "CANCEL" && method != "INFO" ){ log("unsupported method\n"); sl_send_reply("500","unsupported method"); return; } # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing # make transaction if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); if (!t_relay()) { sl_reply_error(); } return; }; if (!t_newtran()){ log("could not create transaction\n"); sl_send_reply("500","could not create transaction"); return; } # actively absorb ACKs if (method == "ACK") { t_relay(); return; } # pass INFO to SEMS if (method=="INFO") { if(!t_write_unix("/tmp/sems_sock","sems/info_append")){ log("could not contact sems\n"); t_reply("500","could not contact media server"); } } if (uri =~ "sip:101.*@") { if (!t_write_unix("/tmp/sems_sock","myapp")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); return; } return; } if (uri =~ "sip:102.*@") { if (!t_write_unix("/tmp/sems_sock","myconfigurableapp")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); return; } return; } if (uri =~ "sip:103.*@") { if (!t_write_unix("/tmp/sems_sock","myannounceapp")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); return; } return; } if (uri =~ "sip:104.*@") { if (!t_write_unix("/tmp/sems_sock","myjukebox")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); return; } return; } if (uri =~ "sip:105.*@") { if (!t_write_unix("/tmp/sems_sock","mycc")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); return; } return; } if (uri =~ "sip:106.*@") { if (!t_write_unix("/tmp/sems_sock","ivr_announce")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); return; } return; } ############ default sems apps if (uri =~ "sip:110.*@") { if (!t_write_unix("/tmp/sems_sock","echo")){ log("could not contact echo\n"); t_reply("500","could not contact media server"); return; } return; } if (uri =~ "sip:111.*@") { if (!t_write_unix("/tmp/sems_sock","announcement")){ log("could not contact announcement\n"); t_reply("500","could not contact media server"); return; } return; } if (uri =~ "sip:112.*@") { if (!t_write_unix("/tmp/sems_sock","conference")){ log("could not contact conference\n"); t_reply("500","could not contact media server"); return; } return; } if (uri =~ "sip:113.*@") { if (!t_write_unix("/tmp/sems_sock","mailbox")){ log("could not contact mailbox\n"); t_reply("500","could not contact media server"); return; } return; } if (uri =~ "sip:114.*@") { if (!t_write_unix("/tmp/sems_sock","early_announce")){ log("could not contact early_announce\n"); t_reply("500","could not contact media server"); return; } return; } if (uri =~ "sip:115.*@") { # load email avp with some email address avp_write("test(a)test.com","$avp(s:email)"); # use voicemail_headers append to pass it to sems if (!t_write_unix("/tmp/sems_sock","voicemail/voicemail_headers")){ log("could not contact voicemail\n"); t_reply("500","could not contact media server"); return; } return; } t_reply("404","Not found"); } Thank you, any and all help is appreciated, thanks again! -- Brandon Armstead

1 0

[Serusers] Using scripts with exec module
by Prettl, Kathrin 08 Nov '06

08 Nov '06

I have written a shell script which has a return value of either 0 or 1. Depending on the outcome of this script I want SER to react. I know I have to use the exec module for this but I don't know how. Can anyone help? Thanks, Kathrin

3 3

Fw: [Users] TM : retransmission timers
by Rao Ramaratnamma 08 Nov '06

08 Nov '06

sorry for reposting -- I think this question belongs to both mailing list. I am really stuck with this. rr ----- Forwarded Message ---- From: Rao Ramaratnamma <raramarat(a)yahoo.com> To: Christian Schlatter <cs(a)unc.edu>; users(a)openser.org Sent: Tuesday, November 7, 2006 11:15:27 PM Subject: Re: [Users] TM : retransmission timers the ser ottendorf announcement does mention improved timers. Cannot openser include this feature too and cannot I merge ser with openser for good timers? I am still trying to understand the difference between ser and openser but standart compliance seems to be very important matter! Cannot people provide me with some hints? I am sure that I am not the only who is asking the difference between ser and openser. ser documentation does not appear uptodate, but the software as sannounced appears impressive. I have already asked this question but did not receive any answer. thank you in advance! rr ----- Original Message ---- From: Christian Schlatter <cs(a)unc.edu> To: users(a)openser.org Sent: Tuesday, November 7, 2006 10:52:56 PM Subject: Re: [Users] TM : retransmission timers Greg Fausak wrote: > Hello, > > I believe this is a well known bug. > Granularity of timers is 1 second. So, if you sign up for a timer to > be fired in 1 second it will happen anywhere between 0 seconds and 1 > second. > 2 seconds will happen between 1 and 2 seconds. I usually set up my > timers to be 2, 2, 4, 8. There are VOIP providers that are pretty > sticky about > the first 500ms. If you are using one of them you're out of luck. Yes, there is a timer process that wakes up every second to perform retransmissions. I was actually quite surprised that OpenSER, which is known to be very standards compliant, does not follow the RFC 3261 retransmission timeouts. On the other hand, the RFC 3261 timeout values are just suggestions and standards compliant SIP UA must accept shorter timeouts. Still it would be nice if OpenSER would support sub second timers, this would allow for shorter fail-over times. Christian > > I believe SER has made timer changes to support more exact timer > intervals. They are a completely different camp, with a different feature > set (although they share the same roots). > > -g > > > On 11/7/06, Jean-François SMIGIELSKI <jf-smig(a)ibelgique.com> wrote: >> Hello, >> >> I made strange observations about the intervals between >> retransmissions with the TM module. >> In my experiments, I used the default parameters for the TM module >> timers, and I sent an INVITE that cannot receive answers (it has a >> well known R-URI pattern that is forwarded to a place and port that >> nobody listen). >> >> When reading RFC3261, I expected to see intervals between >> retransmissions of |500ms|1s|2s|4s|8s|16s|. 7 transmissions, during 32s. >> >> But with OpenSER, (I have tested with the debian package 1.1.0-5 on a >> debian etch, and the cvs sources for 1.1.0 or 1.0.1compiled by >> myself), I can see intervals like <500ms, 2s, 4s, 4s,4s, ... until 26s >> are spent (9 sendings). The first interval is sometomes very short >> (40ms). >> >> Altough I like the sequence of 4s separated transmissions, I do not >> know why the first interval is so short, and why there is no sending >> after 1s. >> >> Did anybody observed such behaviours? Are they normal? >> >> Thanks in advance! >> >> JF Smigielski. >> >> >> ________________________________________________________________________ >> iBELGIQUE, exprimez-vous ! >> http://web.ibelgique.com/ >> >> _______________________________________________ >> 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

7 12

[Serusers] RE: XCAP server
by Jayantheesh Srimushnam - TLS, Chennai 08 Nov '06

08 Nov '06

Hi, Is it possible to run XCAP server as a standalone? I didn't see any info about this in the document. --Jayantheesh _____ From: Jayantheesh Srimushnam - TLS, Chennai Sent: Tuesday, November 07, 2006 8:44 PM To: 'serusers(a)iptel.org' Cc: Jayantheesh Srimushnam - TLS, Chennai Subject: XCAP server Hi All, I have downloaded the SER and trying to run XCAP server. I have gone through the README file for the XCAP server and understood that it is server simulation. Still I didn't get how to install and run it. Could anyone please through some light on this? Thanks, Jayantheesh DISCLAIMER The contents of this e-mail and any attachment(s) are confidential and intended for the named recipient(s) only. It shall not attach any liability on the originator or HCL or its affiliates. Any views or opinions presented in this email are solely those of the author and may not necessarily reflect the opinions of HCL or its affiliates. Any form of reproduction, dissemination, copying, disclosure, modification, distribution and / or publication of this message without the prior written consent of the author of this e-mail is strictly prohibited. If you have received this email in error please delete it and notify the sender immediately. Before opening any mail and attachments please check them for viruses and defect.

1 0

[Users] newbie wants to forward REGISTERs through DMZ to internal SIP server
by Simon Barber 08 Nov '06

08 Nov '06

Hi All, I have an internal SIP server that handles registrations, and is used for incoming and outgoing phone calls. It does not do any authentication. I'd like to set up SER in my DMZ, and have it: 1. authenticate all requests. (this should be easy - I think I can handle it, although I would like to get digest -> RADIUS auth against an Active directory going). 2. forward REGISTER requests to the internal SIP server, both for phones on the internal network and for phones on the external network. 3. forward INVITE and ACK requests performing the correct handling of NAT, using RTP helper to forward media when needed. Right now I'd love to just get the REGISTER requests doing the right thing. Any help, or even better sample config scripts would be much appreciated. Simon

2 1

[Users] Problems with DNS.
by nick 08 Nov '06

08 Nov '06

I'm trying to get my machine ready to work externally, with public Ip addresses. I believe I've got all the pieces ready for the domain. I have the PTR records published for my server's public IP address. I have the .sip_.udp.mobilia.it. 0 0 5060 sipserver.mobilia.it. record (I tried it with and without the .mobilia.it added to the udp) published, I tell the server to listen on the IP addresses which maps it's name in DNS, I have a line after listen : alias="mobilia.it" (I've tried both FQDN and this) I've exported the domain both as the command export SIP_DOMAIN="mobilia.it" and in the openserctlrc (does it need to be the FQDN or just the domain, again, neither seem to make a difference..) however, I still can't get the machine to believe that it is actually responsible for this domain. When I register I have errors along this line: (it registers, obviously, but the logic is wrong, since he thinks the user is coming from another domain). grep_sock_info - checking if host==us: 10==13 && [mobilia.it] == [89.202.252.16] Nov 3 10:13:34 sipserver /usr/local/sbin/openser[3193]: grep_sock_info - checking if port 5060 matches port 5060 Nov 3 10:13:34 sipserver /usr/local/sbin/openser[3193]: grep_sock_info - checking if host==us: 10==13 && [mobilia.it] == [89.202.252.16] Nov 3 10:13:34 sipserver /usr/local/sbin/openser[3193]: grep_sock_info - checking if port 5060 matches port 5060 Can anyone venture a guess as to what I might be missing???

5 13

Re: [Users] Errors whie starting openser with radius server integration
by Sanjeev Manoli 08 Nov '06

08 Nov '06

Hi Ravi, Thanks for your help and suggestion. I went through avp module documentation and figured out that the avp_check syntax in config.cfg file was incorrect. I changed it as follows and now it's working, if(!avp_check("$avp(i:2)", "eq/$avp($src_ip)/ig")) { sl_send_reply("403", "Forbidden IP"); exit; }; But still I get parse error for following 2 statements in openser.cfg (I commented them to make forward progress). Please advice modparam("acc", "radius_extra", "Sip-Src-IP=$si;Sip-Src-Port=$sp") modparam("avpops", "avp_aliases", "day=i:101;time=i:102") Another thing is I am currently using the file based radius authentication for the users. (I have created a file called /usr/local/etc/raddb/users with 2 users). Do you know how to enable/use radius with user authentication done from a database (i am using mysql). I couldn't find any documentation for it. Thanks for the help - Sanjeev On 11/6/06, raviprakash sunkara <sunkara.raviprakash.feb14(a)gmail.com> wrote: > Usrs usrloc is comment, > > > > > > On 11/7/06, Sanjeev Manoli <meghsan(a)gmail.com> wrote: > > Hi, > > I am getting following errors as seen in /var/log/message while > > running openser with radius integration, > > > ***************************************************************************************** > > Nov 6 20:33:45 lx-dev monit[13565]: 'openser' start: /etc/init.d/openser > > Nov 6 20:33:45 lx-dev monit[13565]: 'openser' failed to start > > Nov 6 20:33:45 lx-dev openser: init_tcp: using epoll_lt as the io > > watch method (auto detected) > > Nov 6 20:33:45 lx-dev openser: INFO: statistics manager successfully > > initialized > > Nov 6 20:33:45 lx-dev openser: StateLess module - initializing > > Nov 6 20:33:45 lx-dev openser: TM - initializing... > > Nov 6 20:33:46 lx-dev openser: Maxfwd module- initializing > > Nov 6 20:33:46 lx-dev openser: AVPops - initializing > > Nov 6 20:33:46 lx-dev openser: TextOPS - initializing > > Nov 6 20:33:46 lx-dev openser: ACC - initializing > > Nov 6 20:33:46 lx-dev openser: AUTH module - initializing > > Nov 6 20:33:46 lx-dev openser: xl_parse_item: error - bad parameters > > Nov 6 20:33:46 lx-dev openser: ERROR:avpops:fixup_check_avp: unable > > to get pseudo-variable in P1 > > Nov 6 20:33:46 lx-dev openser: ERROR: fix_actions: fixing failed > > (code=-2) at cfg line 146 > > Nov 6 20:33:46 lx-dev openser: ERROR: fix_expr : fix_actions error > > > ***************************************************************************************** > > I am using openser (Version: openser-1.1.0-tls) and radius server > > (freeradius-1.1.3) along with radiusclient-ng (radiusclient-ng-0.5.2 ). > > I exactly followed the following radius integration documentation > > from openser web site, > > http://openser.org/docs/openser-radius-1.0.x.html > > > > If I remove the radius integration related part from openser.cfg then > > my openser server starts fine, I have tested it with kphone SIP UA and > > it works fine. > > > > One other question is I get parse error for following 2 statements in > > openser.cfg (I commented them to make forward progress). Please advice > > the right syntax to use following modparam statements. > > modparam("acc", "radius_extra", > "Sip-Src-IP=$si;Sip-Src-Port=$sp") > > modparam("avpops", "avp_aliases", "day=i:101;time=i:102") > > > > This is kind of urgent for me and I am clueless at this point so > > really appreciate all your help. > > > > Thanks, > > - San > > P.S. For your reference here is the openser.cfg file that I am using, > > # > > # $Id$ > > # > > # radius config script > > # > > > > # ----------- global configuration parameters ------------------------ > > > > debug=9 # debug level (cmd line: -dddddddddd) > > fork=no > > log_stderror=no # (cmd line: -E) > > > > check_via=no # (cmd. line: -v) > > dns=no # (cmd. line: -r) > > rev_dns=no # (cmd. line: -R) > > port=5060 > > children=4 > > listen=udp: 192.168.0.5 > > alias="192.168.0.5" > > > > #fifo="/tmp/openser_fifo" > > > > # ------------------ module loading > ---------------------------------- > > mpath="/usr/local/lib/openser/modules" > > > > loadmodule "mysql.so" > > loadmodule "sl.so" > > loadmodule "tm.so" > > loadmodule "rr.so" > > loadmodule "maxfwd.so" > > loadmodule "avpops.so" > > loadmodule " usrloc.so" > > loadmodule "registrar.so" > > loadmodule "textops.so" > > loadmodule "xlog.so" > > loadmodule "uri.so" > > loadmodule "acc.so" > > loadmodule "auth.so" > > loadmodule "auth_radius.so" > > loadmodule "group_radius.so" > > loadmodule "avp_radius.so" > > > > # ----------------- setting module-specific parameters --------------- > > > > # -- usrloc params -- > > > #modparam("usrloc","db_url","mysql://openser:openserrw@localhost/openser") > > modparam("usrloc", "db_mode", 2) > > > > # -- acc params -- > > modparam("acc", "radius_flag", 1) > > modparam("acc", "radius_missed_flag", 2) > > modparam("acc", "log_flag", 1) > > modparam("acc", "log_missed_flag", 1) > > modparam("acc", "service_type", 15) > > #modparam("acc", "radius_extra", > "Sip-Src-IP=$si;Sip-Src-Port=$sp") > > modparam("acc|auth_radius|group_radius|avp_radius", > "radius_config", > > "/etc/radiusclient-ng/radiusclient.conf") > > > > # -- group_radius params -- > > modparam("group_radius", "use_domain", 1) > > > > # -- avpops params -- > > #modparam("avpops", "avp_aliases", "day=i:101;time=i:102") > > > > # -- rr params -- > > # add value to ;lr param to make some broken UAs happy > > modparam("rr", "enable_full_lr", 1) > > > > # ------------------------- request routing logic > ------------------- > > > > # main routing logic > > > > route{ > > > > # initial sanity checks -- messages with > > # max_forwards==0, or excessively long requests > > if (!mf_process_maxfwd_header("10")) { > > sl_send_reply("483","Too Many Hops"); > > exit; > > }; > > > > if (msg:len >= 2048 ) { > > sl_send_reply("513", "Message too big"); > > exit; > > }; > > > > # check if user is suspended > > > if(is_method("REGISTER|INVITE|MESSAGE|OPTIONS|SUBSCRIBE")) > > { > > if (radius_is_user_in("From", "suspended")) { > > sl_send_reply("403", "Forbidden - > suspended"); > > exit; > > }; > > }; > > > > # we record-route all messages -- to make sure that > > # subsequent messages will go through our proxy; that's > > # particularly good if upstream and downstream entities > > # use different transport protocol > > if (!method=="REGISTER") > > record_route(); > > > > # subsequent messages withing a dialog should take the > > # path determined by record-routing > > if (loose_route()) { > > # mark routing logic in request > > append_hf("P-hint: rr-enforced\r\n"); > > if(is_method("BYE")) > > { # log it all the time > > acc_rad_request("200 ok"); > > acc_log_request("200 ok"); > > } > > route(1); > > }; > > > > if(is_method("INVITE") && !has_totag()) > > { # set the acc flags > > setflag(1); > > setflag(2); > > }; > > > > if (!uri==myself) { > > # check if user is allowed to do voip calls to other domains > > if(is_method("INVITE|MESSAGE")) { > > if (!radius_is_user_in("From", "voip")) { > > sl_send_reply("403", "Forbidden VoIP"); > > exit; > > }; > > }; > > # mark routing logic in request > > append_hf("P-hint: outbound\r\n"); > > route(1); > > }; > > > > # if the request is for other domain use UsrLoc > > # (in case, it does not work, use the following command > > # with proper names and addresses in it) > > if (uri==myself) { > > # authenticate registers > > if (method=="REGISTER") { > > if (!radius_www_authorize("192.168.0.5")) { > > www_challenge("192.168.0.5", "0"); > > exit; > > }; > > > > # check the src ip address > > > Call the method is Wrong , Check the Module doc od avp_radius and Avp, > > > > > if(!avp_check("i:2", "eq/$src_ip/ig")) > > { > > sl_send_reply("403", "Forbidden IP"); > > exit; > > }; > > > > save("location"); > > exit; > > }; > > > > # calls to pstn > > if(uri=~"sip:00[1-9][0-9]+@") { > > if(is_method("INVITE") && !has_totag()) { > > if (!radius_is_user_in("From", "pstn")) { > > sl_send_reply("403", "Forbidden > PSTN"); > > exit; > > }; > > }; > > # set gateway address > > rewritehostport(" 10.10.10.10:5090"); > > route(1); > > }; > > > > # load callee's avps > > if(avp_load_radius("callee")) > > { > > # check if user has time filter enabled > > if(avp_check("i:3", "eq/i:1")) > > { > > # print time in an avp > > avp_printf("i:100", "$Tf"); > > # extract day > > avp_subst("i:100/i:101", "/(.{3}) > .+/*\1*/"); > > if(!avp_check("i:6", "fm/$day")) { > > sl_send_reply("403", "Forbidden - > day"); > > exit; > > }; > > # extract 'hours:minutes' > > avp_subst("i:100/i:102", "/(.{10}) > (.{5}):.+/\2/"); > > if((is_avp_set("i:4") && avp_check("i:4", > "gt/$time")) > > || (is_avp_set("i:5") && avp_check("i:5", "lt/$time"))) { > > sl_send_reply("403", "Forbidden - > time"); > > exit; > > }; > > }; > > }; > > > > # native SIP destinations are handled using our USRLOC DB > > if (!lookup("location")) { > > # log to acc as missed call > > acc_rad_request("404 Not Found"); > > acc_log_request("404 Not Found"); > > sl_send_reply("404", "Not Found"); > > exit; > > }; > > append_hf("P-hint: usrloc applied\r\n"); > > }; > > > > route(1); > > } > > > > # generic forward > > route[1] { > > # send it out now; use stateful forwarding as it works reliably > > # even for UDP2TCP > > if (!t_relay()) { > > sl_reply_error(); > > }; > > exit; > > } > > # > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > -- > 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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Re: [Users] TM : retransmission timers
by Rao Ramaratnamma 08 Nov '06

08 Nov '06

the ser ottendorf announcement does mention improved timers. Cannot openser include this feature too and cannot I merge ser with openser for good timers? I am still trying to understand the difference between ser and openser but standart compliance seems to be very important matter! Cannot people provide me with some hints? I am sure that I am not the only who is asking the difference between ser and openser. ser documentation does not appear uptodate, but the software as sannounced appears impressive. I have already asked this question but did not receive any answer. thank you in advance! rr ----- Original Message ---- From: Christian Schlatter <cs(a)unc.edu> To: users(a)openser.org Sent: Tuesday, November 7, 2006 10:52:56 PM Subject: Re: [Users] TM : retransmission timers Greg Fausak wrote: > Hello, > > I believe this is a well known bug. > Granularity of timers is 1 second. So, if you sign up for a timer to > be fired in 1 second it will happen anywhere between 0 seconds and 1 > second. > 2 seconds will happen between 1 and 2 seconds. I usually set up my > timers to be 2, 2, 4, 8. There are VOIP providers that are pretty > sticky about > the first 500ms. If you are using one of them you're out of luck. Yes, there is a timer process that wakes up every second to perform retransmissions. I was actually quite surprised that OpenSER, which is known to be very standards compliant, does not follow the RFC 3261 retransmission timeouts. On the other hand, the RFC 3261 timeout values are just suggestions and standards compliant SIP UA must accept shorter timeouts. Still it would be nice if OpenSER would support sub second timers, this would allow for shorter fail-over times. Christian > > I believe SER has made timer changes to support more exact timer > intervals. They are a completely different camp, with a different feature > set (although they share the same roots). > > -g > > > On 11/7/06, Jean-François SMIGIELSKI <jf-smig(a)ibelgique.com> wrote: >> Hello, >> >> I made strange observations about the intervals between >> retransmissions with the TM module. >> In my experiments, I used the default parameters for the TM module >> timers, and I sent an INVITE that cannot receive answers (it has a >> well known R-URI pattern that is forwarded to a place and port that >> nobody listen). >> >> When reading RFC3261, I expected to see intervals between >> retransmissions of |500ms|1s|2s|4s|8s|16s|. 7 transmissions, during 32s. >> >> But with OpenSER, (I have tested with the debian package 1.1.0-5 on a >> debian etch, and the cvs sources for 1.1.0 or 1.0.1compiled by >> myself), I can see intervals like <500ms, 2s, 4s, 4s,4s, ... until 26s >> are spent (9 sendings). The first interval is sometomes very short >> (40ms). >> >> Altough I like the sequence of 4s separated transmissions, I do not >> know why the first interval is so short, and why there is no sending >> after 1s. >> >> Did anybody observed such behaviours? Are they normal? >> >> Thanks in advance! >> >> JF Smigielski. >> >> >> ________________________________________________________________________ >> iBELGIQUE, exprimez-vous ! >> http://web.ibelgique.com/ >> >> _______________________________________________ >> 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

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[Users] one way audio
by rilawich ango 08 Nov '06

08 Nov '06

Hi all, Anyone experiences an one way audio problem when a call is connected to the called party? I am using openser with mediaproxy to provide services to our customer. However, some of our customer complain about the one way audio problem (calling party can hear called party but not vice versa). The system works to most of the customer without that problem. I wonder why there are some customers experience that problem. I assume there is the port problem during RTP packet transmission. Am I right? In the mediaproxy setting, the portRange is 2000:8000. Do I need to increase it? Any method to trace the problem if my assumption is wrong? Thanks.

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[Users] Forwarding requests
by Harry Harcrow 07 Nov '06

07 Nov '06

_____ From: Harry Harcrow [mailto:harry.harcrow@verizonbusiness.com] Sent: Tuesday, November 07, 2006 3:36 PM To: 'users(a)openser.org' Subject: Forwarding requests I have my SER server running on a Linux machine and I am attempting to forward some requests to a Cisco SIP server. I have been running X-lite clients on windows and the same Linux box that the SER server is running on. I have no problem completing calls between the X-lite clients OR with either X-lite client calling the Cisco SIP server directly. However, when I try to make a call from either X-lite client through the SER server I eventually get timeout methods. I am trapping those calls and wrote the following route block: route [5] { record_route(); forward(uri:host, uri:port); } I have tried replacing the forward with send() and t_relay() method calls, but nothing seems to work. The header from the X-lite client directly to the Cisco SIP server, which works is: INVITE sip:2158200@65.221.7.xxx SIP/2.0 Via: SIP/2.0/UDP 166.34.149.xxx:5061;rport;branch=z9hG4bK5963AD09C90F6F3A16F55A271C98F573 From: Harry Harcrow <sip:1000@166.34.149.xxx:5061>;tag=1279481120 To: <sip:2158200@65.221.7.xxx> Contact: <sip:1000@166.34.149.xxx:5061> Call-ID: 28E5B723-9B02-BF38-1A81-1FECC0A85D5E(a)166.34.149.xxx CSeq: 6254 INVITE Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1105d Content-Length: 313 The header from SER to the Cisco server, which does not work is: INVITE sip:2158200@65.221.7.xxx SIP/2.0 Record-Route: <sip:166.34.149.xxx;ftag=2061592395;lr=on> Via: SIP/2.0/UDP 166.34.149.xxx;branch=0 Via: SIP/2.0/UDP 166.34.149.xxx:5061;rport=5061;branch=z9hG4bK4EA9A90B8CE7CE52BFB69DA431560E5 7 From: Harry Harcrow <sip:1000@166.34.149.xxx:5061>;tag=2061592395 To: <sip:2158200@65.221.7.xxx> Contact: <sip:1000@166.34.149.xxx:5061> Call-ID: 38CA166F-59A6-E44B-F34E-9E1D6307698B(a)166.34.149.xxx CSeq: 3009 INVITE Max-Forwards: 16 Content-Type: application/sdp User-Agent: X-Lite release 1105d Content-Length: 313 The only thing that I can see different is the record_route record and an additional Via record. Does anyone have any idea what I am doing wrong? Thanks Harry Harcrow

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[Serusers] ser accounting Issue
by ram 07 Nov '06

07 Nov '06

Hi i have SER running with mysql i have a question related to accounting when the calls are live, when i reboot the server i wont see the Bye messages, this kind of situation, how can i terminate the call i mean send bye message to all pending calls and account them when i server come back any help on this issue iam not sure if this issue already addressed Ram

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[Users] photos from OpenSER BoF
by Daniel-Constantin Mierla 07 Nov '06

07 Nov '06

Hello, just to get a feeling about how the event is going, we uploaded few pictures, since the wireless is very crowded, bandwidth is quite limited -- more will come in the next days, along with the slides. We are very please to have more that 75% attendees from the first days, the room for BoF session proved very small, hopefully tomorrow the places in summit room will be enough -- we got a lot of new people interested in the summit from VoN event. http://www.openser.org/pictures/openser-bof/ Cheers, Daniel

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[Serusers] error: mediaproxy/sendMediaproxyCommand(): can't connect to MediaProxy
by vijay tiwari 07 Nov '06

07 Nov '06

hi all we i call to one UA to another UA i found this error. error: mediaproxy/sendMediaproxyCommand(): can't connect to MediaProxy thanks vijay --------------------------------- Sponsored Link Free Uniden 5.8GHz Phone System with Packet8 Internet Phone Service

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[Users] OpenSER Summit @ VoN Berlin - ready to go
by Daniel-Constantin Mierla 07 Nov '06

07 Nov '06

Hello everybody, OpenSER summit is ready to start in one day. Everything is in place, although some fine tunings will be there all the time. Next days the activity on mailing lists may decrease a bit, as many of OpenSER users are joining the event. To check the event's schedule and details: http://www.voneurope.com/expo.html http://www.openser.org The event will not be broadcasted on the web, the presentations will be collected and published after the event. Photos and briefing will be available as well. Cheers, Daniel

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Re: [Serusers] new major SER pre-release, Ottendorf, is out for testing
by sip 07 Nov '06

07 Nov '06

Well it's about time!!! ;) Seriously though, thanks for all the hard work you've all put in on this release. I'm looking forward to getting the chance to install and play with it, and (most importantly for me) documenting as much as I can about how to migrate from a production 0.9.X solution to the new code -- something I imagine is going to be tricky as much of our actual web/management code will need to be rewritten. I can't express really how much SER and the Iptel team have changed my life for the better. Thanks again for everything. N. On Mon, 06 Nov 2006 12:02:24 +0100, Jiri Kuthan wrote > Ladies and gentlemen, girls and boys, > > the new testing pre-release of the best SIP proxy server ever, SER, > is out! > > This major pre-release, codenamed Ottendorf, includes fundamental > improvements to the software. The improvements allow SER users to > achieve significantly higher managability and reliability by better > data mode, interop features, security, etc. They recollect > voluminous feedback we have received from iptel customers and open > source users. In particular, the new features include but are not > limited to: - improved data model which eliminates any possible ambiguities. > - key under-the-hood changes with impact on performance: timer subsystem > overhaul (now fast, with fine resolution, high accuracy and > variability), TCP processing overhaul (higher scalability) - key > reliability changes: DNS cache and IP blacklisting - key interop > changes: support for STUN-bis, presence server aligned to latest > IETF development, sanity module for early detection of invalid messages > - more management: management features are now available via XML-RPC > remotely, as well as via an alternative binary protocol - more > security: ratelimit module, more check types in permissions module - > scripting features: select operator in combination with revised AVPs, > named flags and routes, more types of routes (onsend, timer, > branch) - applications: presence has been greatly updated to keep > pace with development in the IETF/SIMPLE (includes asynchronous subscription > authorization, extended PIDF); web interface includes many new managibility > features (multidomain, templates and multilanguage to name some). New > scripting flexibility allows safely for features like individual ringing > timers and direct access to SIP message elements. > - new tools: Nils's script execution tree tool > http://lists.iptel.org/pipermail/serdev/2006-October/008124.html > and refurbished ser_ctl > see the following links for some additional details: > - http://www.iptel.org/ser/doc/010whatsnew > - http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/NEWS?rev=HEAD > > Accompanying software, SEMS and SERWEB have been greatly overhauled > as well. SEMS now features improved performance (single-threaded > model now), service composition via python-IVR-scripting and B2BUA, > adaptive playout buffer for better voice quality, and more. SERWEB > features new multilevel administrative access system, multidomain > support, per-domain customization, remote xml-rpc access, > multilanguage support, and more. The new command-line utility, > SER_CTL, provides also many great improvements over the previous tool. > > Interop > ------- > Shortly, we have been keeping the bar high through our participation > in the IETF and SIPITs. SER implements latest interop changes from > the IETF (STUN-bis, presence + XCAP, to name the most important ones) > and both new features and legacy features were extensively and > successfully tested middle of October in SIPit. > > Next Steps > ----------- > Keep in mind, this is a pre-release for testing purposes. It is now > call on users to test the software hard and provide feedback. In the > meantime, the codebase remains frozen and we only commit patches > fixing various bugs. Once we get good confidence that Ottendorf has > stabilized and is being adequately documented, we will declare the > final release. No date has been determined yet. Also, we consider > changing CVS to SVN. > > Again, and this is very important: the final release will be as good > as the feedback we obtain for this pre-release. Please do spend your > time sending solid feedback back to [serusers]. > > Roadmap > ------- > We keep rolling. We are still debating with our generous > contributors what is the next possible good things to work on, the > topics include but are not limited to as of now: - documentation. - > databases are important. LDAP and BDB, DB-API revision. - interop: > identity, TLS revision - advanced scripting (scripting engine, vars > returns, AVPs) - uncountable improvements (futex, dev/rtc, > tm/refcnts, tcp fd cache, .....) > > Trivia > -------- > The release name Ottendorf is coming from a place in Saxonian > Switzerland, where many iptel fellows and their friends spent a > weekend during this Indian summer. http://www.iptel.org/jan/gallery/saxonian_switzerland_2006/ > http://www.iptel.org/~andi/2006-09-22-Ottendorf/ > http://www.iptel.org/~vlada/saxon_switzerland/index.html > http://www.ohlmeier.de/albums/ottendorf_2006/index.html > > Acknowledgments > --------------- > This grand achievement is result of extremely many folks who did lot > of hard work. Out of folks affiliated with iptel, Andrei and Jan > spent immense effort on moving SER forward, with great support from Hallik > (command-line tool), Michal, Miklos, Nils, Tomas, Vaclav (presence), > Vlada (new STUN). In this context, I would like to thank to our > company for the great support. Recently, Bogdan Pintea greatly > helped to make the upcoming release better and more stable under > sponsorship of his company, iptego. transnexus provided osp support. > > SEMS has been greatly improved thanks to Stefan and Raphael. > Credits for serweb go to Karel. > > Same importantly, SER has been immensely benefitting of contributors, > to name in alphabetic order at least the 'top-of-the-iceberg' > fellows: Bogdan Pintea, Cesc Santa, Greg Fausak, Greger Teigre, > Hendrik Scholz, <http://tutpro.com/>Juha Heinänen, Klaus Darilion, > Maxim Sobolev, Peter Nixon, Uli Abend. > > Great support on the mailing list has been provided by Greger V. > Teigre, Java Rockx, Klaus Darilion, Zeus Ng, and more. > > Simply, this list is too short and incomplete to capture all the > generous contributions that came in -- my appologies for everyone > else I have missed. > > Warnings and Disclaimers > ------------------------ > This is a TESTING version. We are encouraging people to give it intense > testing and provide us with feedback, so that we can properly assess > what it takes to declare it final. The volume and number of changes > is big, caution is advised. > > All GPL warnings and disclaimers apply. > > This is SER, which is not the derivate software known as openser. We > have no ideas of in what extent and timeline derivate projects follow > (or not) SER. > > ------------------------------------------------------------------------------- > > To Give it a Try > ----------------- > warning: this is just a 'jump-start' for black-belt users. It is not > a replacement for documentation or a detailed step-by-step installation > guide. It assumes knowledge of operating system, SIP, and SER. > > PREPARATION > - make sure mysql is running. In the default configuration, SER > assumes it to be running on localhost, if this does not hold, you > need to change ser.cfg. - make sure python is installed (needed for > new command-line utilities) SER INSTALLATION FROM CVS - Download > from CVS/HEAD using instructions at http://www.iptel.org/downloads - > Compile it: make include_modules="mysql xmlrpc" all - Install it > the default way of your administrative privileges allow to: make > include_modules="mysql xmlrpc" install - Download command-line tool > (see more documentation under http://cvs.berlios.de/cgi- > bin/viewcvs.cgi/*checkout*/ser/serctl/README?rev=HEAD) - cvs co serctl > - Initiate database: > ./scripts/mysql/ser_mysql.sh create > START > - Start it. Call: ser > - Begin provisioning users with the new serctl (the capital word are > just place holders) # ser_ctl domain add DOMAINNAME # ser_ctl > user add USERNAME@DOMAINNAME -p PASSWORD # If you want to have > Asserted Identity header for your user # - ser_attr add uid=UID asserted_id="PID" > # If you want to have gateway support > # - ser_db add attr_types name=gw_ip rich_type=string raw_type=2 \ > description="The gateway IP for the default ser.cfg" default_flags=33 > # - ser_attr add global gw_ip=GATEWAY-IP > - more things you may wish to do: > - familiarize yourself with current ser.cfg > - send us feedback. > > -- > Jiri Kuthan http://iptel.org/~jiri/ > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers

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[Users] Re: [Semsdev] SEMS users meeting at VON Europe 2006 (Berlin)
by Stefan Sayer 07 Nov '06

07 Nov '06

Hello, we would like to meet tomorrow, Wednesday, at 13:00 in the exhibition area. I hope to see those of you who are at VON Europe and interested in SEMS to discuss its development. Regards Stefan Raphael Coeffic wrote: > > Hello everybody, > > Stefan Sayer and I are proud to invite you to a SEMS users meeting at the > VON > Europe 2006 in Berlin on wednesday 8th november. It could a great chance > to > discuss all your favorite features and gives us the possibility to know > what > is important to you. > > At the moment, there is still no meeting point that have been agreed upon. > So > if you have any good ideas, including plans for the evening, please make > your > suggestions on the list. > > Regards > Raphael. > > -- > Dipl. Inf. Raphael Coeffic > > iptelorg GmbH > Am Borsigturm 11 > 13507 Berlin > Germany > > rco(a)iptel.org > www.iptelorg.com > > T +49-30-3251-3218 > F +49-30-6908-8248 > _______________________________________________ > Semsdev mailing list > Semsdev(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/semsdev > -- Stefan Sayer Media Services Development iptego GmbH Am Borsigturm 40 13507 Berlin Germany stefan.sayer(a)iptego.de www.iptego.de _______________________________________________ Semsdev mailing list Semsdev(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/semsdev

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[Serusers] Using different domains
by Nick Hoffman 07 Nov '06

07 Nov '06

Hi guys. I just noticed that when user agents register, the domain in their SIP URI does not have to equal their domain in the subscribers database table. For instance, I have 2 domains in the domain table: sip.altcall.com sip.voxpak.com The username "foobar" in my subscribers table has the domain "sip.altcall.com", but can register and make calls as foobar(a)sip.voxpak.com . Why is this? Cheers, -- Nick E: nick.hoffman(a)voxpak.com P: +61 7 5591 3588 F: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make any use of the email. We do not waive any privilege, confidentiality or copyright associated with it.

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[Users] Connect two OpenSER...
by Steffen Witt 07 Nov '06

07 Nov '06

Hello, I have two OpenSER servers that I want to "connect" with each other: Phone "a" <---> Server "A" <------------> Server "B" <---> Phone "b" Phone "a" should call phone "b" without adding the domain during call initiation and vice versa. Phone numbers from "A" starts with "5xxxx" and from "B" with "6xxxx". I have tried to edit openser.cfg e.g. at server "A": route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; ... # TRY TO USE AVP # if (uri=~"69000@") { avp_printf("$avp(s:new_uri)","sip:69000@IP_OF_SERVER_B;transport=udp"); avp_pushto("$ru","$avp(s:new_uri)"); t_relay(); }; # TRY TO USE FORWARD # # if (to_uri=~"69000@") { # forward("udp:IP_OF_SERVER_B:5060"); # exit; # }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS #if(uri=~"@tls_domain1.net") { # t_relay("tls:domain1.net"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay("tls:domain2.net"); # exit; #} route(1); }; ... route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } The call receives phone "b" but there will be no connection established. I get immediatelly an error on server "A" side... "Status: 500 I'm terribly sorry, server error occured (1/SL)". Any ideas how to configure it correctly so that I can reach my phone on the other site? What rules do I have to create? What functions to use? Are there any examples available (besides the source files)? Thanks for your help! Best regards, Steffen

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[Users] Messege is too big Error Code 513
by ShivaSankar Padhy 07 Nov '06

07 Nov '06

Dear all! I am still new to the openser. Let me ask another silly question:) I am getting error messege as 513, messege is too big, May I know in what condition is is comming??What is the rule to avoid this. Expectig a sooner response as always, Thanks in Advance With cheers, ShivaSankar

3 2

[Users] FW: XMPP module
by Herchi Silviu 07 Nov '06

07 Nov '06

> Hi list, > > I'm new to OpenSER, and I'm trying to compile the XMPP module. > > So far, I've checkd out the CVS version and compiled it successfully. > However, the XMPP module wouldn't compile, even though I have taken it > out of the exclude list in the Makefile. The only way to compile it > was to run "make" in the modules/xmpp directory (I know, I'm not > supposed to, but it worked...). > > I now have the following in my configuration file: > > [...] > loadmodule "/usr/local/openser-1.2/lib/openser/modules/xmpp.so" > > modparam("xmpp", "backend", "component") > modparam("xmpp", "domain_separator", "*") > modparam("xmpp", "gateway_domain", "xmpp.test.int") > modparam("xmpp", "xmpp_domain", "openser.test.int") > modparam("xmpp", "xmpp_host", "im.test.int") > modparam("xmpp", "xmpp_port", "10015") > modparam("xmpp", "xmpp_password", "password") > [...] > > But running OpenSER results in: > > 0(24041) set_mod_param_regex: parameter <xmpp_port> not found in > module <xmpp> > 0(24041) parse error (86,20-21): Can't set module parameter > ERROR: bad config file (1 errors) > > Anyone can help? > > Thank you for your help, > > Silviu

1 0

[Serusers] Variables in Ser.cfg
by Benito Caracuel 07 Nov '06

07 Nov '06

Hello all, I have a file txt and a variable. I need to do this in ser.cfg: $variable = cat /tmp/file.txt Replace_all("bobobo", "%$variable"); But It`s not run. How can I do this? Thank you very much. _________________________________________________________________ Llama a tus amigos de PC a PC: ¡Es GRATIS! http://get.live.com/messenger/overview

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[Serusers] Re: some simple newbie help with dialplan needed...
by Evert 07 Nov '06

07 Nov '06

Hi! I have SER running on 5060, and Asterisk on 5061, so if Asterisk passes it to @localhost there should be no problem there. The remaining problem seems to be the passing on of the call. The mentioned dialplan does send the entered extension to SER, and the phone on that SER-account rings. However, when the called phone anwers the call, the calling phone does not notice this, and keeps on ringing... No connection is established. I am not sure whether this is caused by Asterisk or SER, so I am sending this to the SER-group as well... :-) Regards, Evert William Piper wrote: > Does your variable ${SERADDRESS} have the port number defined? If you > replace it with localhost, I believe port 5060 will pick up the call by > default. > > Try making sure that your ${SERADDRESS} has the IP like the following: > 127.0.0.1:5070 <http://127.0.0.1:5070> (replace 5070 with whatever port > your SER is listening on). > > I've never tried sending the call through loopback but it seems like it > should work. > > bp > > > On 11/6/06, *Evert* <evert(a)poboxes.info <mailto:evert@poboxes.info>> wrote: > > SERADDRESS is a variable that points to a SER-server (currently running > on the same IP as Asterisk, but on a different port). > > I noticed that it helped to replace @${SERADDRESS} with @localhost > > I now have: > > exten => 998,1,Answer > exten => 998,2,Background(agent-newlocation) > exten => 998,3,Set(TIMEOUT(digit)=15) > exten => 998,4,WaitExten(10) > exten => _XXXX,1,Dial(SIP/${EXTEN}@localhost,60,o) > > > This does send the entered extension to SER, and the phone rings. > However, when the called phone anwers the call, the calling phone does > not notice this, and keeps on ringing... > > > > > Regards, > Evert > > Leo Ann Boon wrote: > > Evert wrote: > >> Hi! :) > >> > >> Thanks for the tip. I'm almost there now, the only problem that I > have > >> left is that I do NOT want Asterisk to check whether the extension > >> entered is valid. In the current setup Asterisk will refuse to > forward > >> the call since it thinks the extension is invalid... :-/ > >> > >> > > Is ${SERADDRESS} the name of a valid SIP peer or just plain ole IP > > address? It should be a SIP peer. > > > > Leo

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RE: [serusers]: trusted table(permissions module)
by Kamal.Mann＠t-systems.com 07 Nov '06

07 Nov '06

Hi All In current scenario when a sip invite message comes to SER from my SIP_AS; SER take this massage trusted (as I defined in trusted table) but after this in my INVITE route(3) it sends it to route(1) default message handler. I am enclosing my ser.cfg; please take a glance on it. Thanks in anticipation Kamal Mann -----Original Message----- From: John Shellhouse [mailto:jwshel@hotmail.com] Sent: Friday, November 03, 2006 9:17 PM To: Mann, Kamal Subject: RE: [serusers]: trusted table(permissions module) Hi, you've already tried something more filled in like this... if (!proxy_authorize("somedomain.com","subscriber")) { proxy_challenge("somedomain.com","0"); break; } else if (method == "INVITE" & !check_from()) { log("from!=digest\n"); sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); T >From: <Kamal.Mann(a)t-systems.com> >To: <andrey.kouprianov(a)gmail.com>, <serusers(a)iptel.org> >Subject: RE: [serusers]: trusted table(permissions module) >Date: Fri, 3 Nov 2006 15:40:26 +0100 > >Hi All >Now SER is trusting packates from trusted table IP and in the following >part of route(3) of Invite message handler allow_trusted results 'true' and >it directly bypasses this code. >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(); >}; >But after in route(1) its sending 407-proxy auth required to trusted ip of >sip_AS. Please help me out of this dilemma. > >Thanks in anticipation >Kamal Mann > >-----Original Message----- >From: serusers-bounces(a)lists.iptel.org >[mailto:serusers-bounces@lists.iptel.org] On Behalf Of Andrey Kuprianov >Sent: Thursday, November 02, 2006 1:13 PM >To: serusers(a)iptel.org >Subject: Re: [serusers]: trusted table(permissions module) > >The one you attached, didnt have a semicolon. Anyway, try using a >fresh copy of a ready made ser.cfg. > > Bests, > > Andrey. > >On 11/2/06, Kamal.Mann(a)t-systems.com <Kamal.Mann(a)t-systems.com> wrote: > > Yup I always do the same, Might be copy paste error. Its having >semicolon & working fine here. > > > > Regards > > Kamal Mann > > > > -----Original Message----- > > From: Andrey Kuprianov [mailto:andrey.kouprianov@gmail.com] > > Sent: Thursday, November 02, 2006 10:34 AM > > To: serusers(a)iptel.org > > Subject: Re: [serusers]: trusted table(permissions module) > > > > Hi Kamal, > > > > Whenever you modified your ser.cfg did you restart SER after? Did you > > check your ser.cfg for errors using "ser -c" command? I found some > > errors in your ser.cfg code. For instance, inside the main route > > block, your "if", which processes INVITEs, is missing a semicolon at > > the end of the block. > > > > if (method=="INVITE") > > { > > # sl_send_reply("404", "INVITE "); > > route(3); > > break; > > } <----- missing semicolon here > > > > > > If you are comfortable with your current ser.cfg, that's ok, but I'd > > suggest you start with a fresh one, make your trusted table work, and > > then add a "REFER" processing, like in your currect ser.cfg. Just > > download one fresh ser.cfg from iptel.org site and add your > > allow_trusted() "if" clause there. > > > > Regards, > > > > Andrey. > > > > > > On 11/2/06, Kamal.Mann(a)t-systems.com <Kamal.Mann(a)t-systems.com> wrote: > > > Hi > > > Please find ser.cfg enclosed. > > > > > > Trusted table entry: > > > +---------------+-------+--------------+ > > > | src_ip | proto | from_pattern | > > > +---------------+-------+--------------+ > > > | 10.25.119.156 | any | ^sip:.*$ | > > > +---------------+-------+--------------+ > > > 1 row in set (0.00 sec) > > > > > > Thanks in anticipation > > > Kamal Mann > > > -----Original Message----- > > > From: serusers-bounces(a)lists.iptel.org >[mailto:serusers-bounces@lists.iptel.org] On Behalf Of Andrey Kuprianov > > > Sent: Wednesday, November 01, 2006 7:03 PM > > > To: serusers(a)iptel.org > > > Subject: Re: [serusers]: trusted table(permissions module) > > > > > > Hi Kamal, > > > > > > I understand your setup. Nevertheless, it will be difficult to figure > > > out your problem, w/o your ser.cfg file at hand. Also, send a trusted > > > table query result (i.e. that row which contains entry for your > > > SIP-AS). > > > > > > Regards, > > > > > > Andrey. > > > > > > On 11/1/06, Kamal.Mann(a)t-systems.com <Kamal.Mann(a)t-systems.com> wrote: > > > > Hi All > > > > In my scenario I need ser to communicate with my SIP-AS. This SIP-AS >is having an application xyz running on it. This XYZ sends an invite to URI >"A(a)xcv.de" which is a subscriber in SER and this is in TO header of Invite >msg & FROM header contains another URI test(a)xcv.de but this URI >(test(a)xcv.de) isn't registered / subscribed to SER. I need SER trust all >packets from SIP-AS ip. BUT SER is sending 407 reply of INVITE to XYZ >application! IP in trusted table is of SIP-AS on top of which XYZ >application is running. > > > > > > > > Thanks in anticipation > > > > Kamal Mann > > > > > > > > -----Original Message----- > > > > From: Maciej ¯wirski [mailto:mzwirek@poczta.fm] > > > > Sent: Wednesday, November 01, 2006 3:20 PM > > > > To: serusers(a)lists.iptel.org > > > > Subject: Re: [serusers]: trusted table(permissions module) > > > > > > > > Kamal.Mann(a)t-systems.com wrote: > > > > > > > > > Hi All > > > > > > > > > > I made an entry /"10.25.119.156, any, ^sip:.*$"/ into trusted >table so > > > > > that any packet sent from 10.25.119.156 would be trusted by SER >and > > > > > don't ask for its credentials. But SER is replying *407- proxy* >auth > > > > > needed!! After this I tried with permissions.allow file: > > > > > > > > > Hi, > > > > I had the same issue while interconnecting SER and Asterisk. It >turned > > > > out to be an issue on Asterisk side (I had the same user added for >SER > > > > and Asterisk, so the poor thing couldn't authorize either :)) So you > > > > could check if the 407 you're getting is from SER or from the remote >host. > > > > > > > > Regards, > > > > Maciej Zwirski > > > > > > > > >---------------------------------------------------------------------- > > > > Jestes kierowca? To poczytaj! >>> http://link.interia.pl/f199e > > > > > > > > _______________________________________________ > > > > Serusers mailing list > > > > Serusers(a)lists.iptel.org > > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > _______________________________________________ > > > > Serusers mailing list > > > > Serusers(a)lists.iptel.org > > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > > > > > > > > >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers _________________________________________________________________ Get FREE company branded e-mail accounts and business Web site from Microsoft Office Live http://clk.atdmt.com/MRT/go/mcrssaub0050001411mrt/direct/01/

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[Users] Errors whie starting openser with radius server integration
by Sanjeev Manoli 07 Nov '06

07 Nov '06

Hi, I am getting following errors as seen in /var/log/message while running openser with radius integration, ***************************************************************************************** Nov 6 20:33:45 lx-dev monit[13565]: 'openser' start: /etc/init.d/openser Nov 6 20:33:45 lx-dev monit[13565]: 'openser' failed to start Nov 6 20:33:45 lx-dev openser: init_tcp: using epoll_lt as the io watch method (auto detected) Nov 6 20:33:45 lx-dev openser: INFO: statistics manager successfully initialized Nov 6 20:33:45 lx-dev openser: StateLess module - initializing Nov 6 20:33:45 lx-dev openser: TM - initializing... Nov 6 20:33:46 lx-dev openser: Maxfwd module- initializing Nov 6 20:33:46 lx-dev openser: AVPops - initializing Nov 6 20:33:46 lx-dev openser: TextOPS - initializing Nov 6 20:33:46 lx-dev openser: ACC - initializing Nov 6 20:33:46 lx-dev openser: AUTH module - initializing Nov 6 20:33:46 lx-dev openser: xl_parse_item: error - bad parameters Nov 6 20:33:46 lx-dev openser: ERROR:avpops:fixup_check_avp: unable to get pseudo-variable in P1 Nov 6 20:33:46 lx-dev openser: ERROR: fix_actions: fixing failed (code=-2) at cfg line 146 Nov 6 20:33:46 lx-dev openser: ERROR: fix_expr : fix_actions error ***************************************************************************************** I am using openser (Version: openser-1.1.0-tls) and radius server (freeradius-1.1.3) along with radiusclient-ng (radiusclient-ng-0.5.2). I exactly followed the following radius integration documentation from openser web site, http://openser.org/docs/openser-radius-1.0.x.html If I remove the radius integration related part from openser.cfg then my openser server starts fine, I have tested it with kphone SIP UA and it works fine. One other question is I get parse error for following 2 statements in openser.cfg (I commented them to make forward progress). Please advice the right syntax to use following modparam statements. modparam("acc", "radius_extra", "Sip-Src-IP=$si;Sip-Src-Port=$sp") modparam("avpops", "avp_aliases", "day=i:101;time=i:102") This is kind of urgent for me and I am clueless at this point so really appreciate all your help. Thanks, - San P.S. For your reference here is the openser.cfg file that I am using, # # $Id$ # # radius config script # # ----------- global configuration parameters ------------------------ debug=9 # debug level (cmd line: -dddddddddd) fork=no log_stderror=no # (cmd line: -E) check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 listen=udp:192.168.0.5 alias="192.168.0.5" #fifo="/tmp/openser_fifo" # ------------------ module loading ---------------------------------- mpath="/usr/local/lib/openser/modules" loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "avpops.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "xlog.so" loadmodule "uri.so" loadmodule "acc.so" loadmodule "auth.so" loadmodule "auth_radius.so" loadmodule "group_radius.so" loadmodule "avp_radius.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc","db_url","mysql://openser:openserrw@localhost/openser") modparam("usrloc", "db_mode", 2) # -- acc params -- modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 2) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 1) modparam("acc", "service_type", 15) #modparam("acc", "radius_extra", "Sip-Src-IP=$si;Sip-Src-Port=$sp") modparam("acc|auth_radius|group_radius|avp_radius", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") # -- group_radius params -- modparam("group_radius", "use_domain", 1) # -- avpops params -- #modparam("avpops", "avp_aliases", "day=i:101;time=i:102") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # check if user is suspended if(is_method("REGISTER|INVITE|MESSAGE|OPTIONS|SUBSCRIBE")) { if (radius_is_user_in("From", "suspended")) { sl_send_reply("403", "Forbidden - suspended"); exit; }; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); if(is_method("BYE")) { # log it all the time acc_rad_request("200 ok"); acc_log_request("200 ok"); } route(1); }; if(is_method("INVITE") && !has_totag()) { # set the acc flags setflag(1); setflag(2); }; if (!uri==myself) { # check if user is allowed to do voip calls to other domains if(is_method("INVITE|MESSAGE")) { if (!radius_is_user_in("From", "voip")) { sl_send_reply("403", "Forbidden VoIP"); exit; }; }; # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { # authenticate registers if (method=="REGISTER") { if (!radius_www_authorize("192.168.0.5")) { www_challenge("192.168.0.5", "0"); exit; }; # check the src ip address if(!avp_check("i:2", "eq/$src_ip/ig")) { sl_send_reply("403", "Forbidden IP"); exit; }; save("location"); exit; }; # calls to pstn if(uri=~"sip:00[1-9][0-9]+@") { if(is_method("INVITE") && !has_totag()) { if (!radius_is_user_in("From", "pstn")) { sl_send_reply("403", "Forbidden PSTN"); exit; }; }; # set gateway address rewritehostport("10.10.10.10:5090"); route(1); }; # load callee's avps if(avp_load_radius("callee")) { # check if user has time filter enabled if(avp_check("i:3", "eq/i:1")) { # print time in an avp avp_printf("i:100", "$Tf"); # extract day avp_subst("i:100/i:101", "/(.{3}) .+/*\1*/"); if(!avp_check("i:6", "fm/$day")) { sl_send_reply("403", "Forbidden - day"); exit; }; # extract 'hours:minutes' avp_subst("i:100/i:102", "/(.{10}) (.{5}):.+/\2/"); if((is_avp_set("i:4") && avp_check("i:4", "gt/$time")) || (is_avp_set("i:5") && avp_check("i:5", "lt/$time"))) { sl_send_reply("403", "Forbidden - time"); exit; }; }; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { # log to acc as missed call acc_rad_request("404 Not Found"); acc_log_request("404 Not Found"); sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } # generic forward route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } #

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[Serusers] flatstore data fields
by Matthew McGuire 06 Nov '06

06 Nov '06

I am using flatstore module for capturing stateful call records. This gives much faster performance than mysql inserts into acc. However, I'd like to do batch inserts so that I can still use the mysql acc table structure for accounting and billing. Unfortunately the data fields do not match up exactly. Is there a document that explains what data fields are being generated by flatstore and how to correlate them to acc? thanks, matt

1 0

[Users] OpenSER + Sems?
by Brandon＠voipo.com 06 Nov '06

06 Nov '06

Hello, Can anyone provide me with a basic implementation of confrencing and voicemail with sems? For the life of me I can't seem to figure out a few loose ends and get it working correctly, well thankyou for any advice or help you may have. Regards, Brandon Armstead Sent via BlackBerry from T-Mobile

1 0

[Serusers] 2 clients behind the same NAT
by Ira Kadin 06 Nov '06

06 Nov '06

Dear all, we are using the NAT - mediaproxy solution. Is there a way to avoid using mediaproxy in case both sides (client and server) are located behind the same NAT (something like comparing private and receiving from (NAT) IP's - and in case they are equal - don't call use_mediaproxy. Thanks Irena

2 1

Re: [Serusers] Problem with mysql module [mysql.so] file
by samuel 06 Nov '06

06 Nov '06

You can just install the dev packages and afterwards compile SER again. Samuel 2006/11/6, tezet(a)touk.pl <tezet(a)touk.pl>: > Thank u!! > > So i have to install mysql from the beginning or it is possible just to > install this missing packages? I mean would it be enough to download the > source instead of binary distribution or this development version is > separate distribution or extra package?? > > Tomasz. > > When this kind of error appear in the compilation (missing .h file) it > > usually means you have to install the developement version of the > > packet. In your case it means you have to install mysql dev pacakges. > > > > Samuel. > > 2006/11/6, tzieleniewski <tzieleniewski(a)o2.pl>: > >> Hi all > >> > >> I've installed ser as it is written in the Getting started manuall with > >> ser source 0.9.7-pre1. I've also installed the mysql5.0 binary > >> distribution. > >> unfortunatelly i've encountered a strange problem. At think that this > >> may be caused by the fact that i used o binary distribution od mysql > >> instead of source. > >> Any way i'm just starting my adventure with ser:) so i ask u ser guru's > >> to help me;) > >> > >> When i tried to compile the source again including mysql package the > >> mysql.co file did not appeared? > >> and the log contained only such lines regarding mysql: > >> make[1]: Wejďż˝ďż˝cie do katalogu > >> `/home/tezet/apps/ser/ser-0.9.7-pre1/modules/mysql' > >> make[1]: Opuszczenie katalogu > >> `/home/tezet/apps/ser/ser-0.9.7-pre1/modules/mysql' > >> make[1]: Wejďż˝ďż˝cie do katalogu > >> `/home/tezet/apps/ser/ser-0.9.7-pre1/modules/mysql' > >> gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall > >> -minline-all-stringops -malign-double -falign-loops -ftree-vectorize > >> -mtune=athlon64 -DNAME='"ser"' -DVERSION='"0.9.7-pre1"' > >> -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 4.1.2"' -D__CPU_i386 > >> -D__OS_linux -DCFG_DIR='"/usr/local/etc/ser/"' -DPKG_MALLOC -DSHM_MEM > >> -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP > >> -DDISABLE_NAGLE -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT > >> -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN > >> -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL > >> -DHAVE_ALLOCA_H -I/usr/local/include -I/usr/local/include/mysql > >> -I/usr/local/mysql/include -I/usr/include/mysql -c dbase.c -o dbase.o > >> make[1]: Opuszczenie katalogu > >> `/home/tezet/apps/ser/ser-0.9.7-pre1/modules/mysql' > >> > >> When i tried to compile the package manually using Makefile in the > >> "source"/modules/myqsl directory i got plenty of errors:): > >> dbase.c:34:25: error: mysql/mysql.h: Nie ma takiego pliku ani katalogu > >> dbase.c:35:26: error: mysql/errmsg.h: Nie ma takiego pliku ani katalogu > >> In file included from dbase.c:39: > >> val.h:44: error: expected ďż˝ďż˝ďż˝)ďż˝ďż˝ďż˝ before ďż˝ďż˝ďż˝*ďż˝ďż˝ďż˝ > >> token > >> In file included from dbase.c:40: > >> my_con.h:39: error: expected specifier-qualifier-list before > >> ďż˝ďż˝ďż˝MYSQL_RESďż˝ďż˝ďż˝ > >> dbase.c: In function ďż˝ďż˝ďż˝submit_queryďż˝ďż˝ďż˝: > >> dbase.c:67: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝timestampďż˝ďż˝ďż˝ > >> dbase.c:68: warning: implicit declaration of function > >> ďż˝ďż˝ďż˝mysql_pingďż˝ďż˝ďż˝ > >> dbase.c:68: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝conďż˝ďż˝ďż˝ > >> dbase.c:72: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝timestampďż˝ďż˝ďż˝ > >> dbase.c:90: warning: implicit declaration of function > >> ďż˝ďż˝ďż˝mysql_queryďż˝ďż˝ďż˝ > >> dbase.c:90: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝conďż˝ďż˝ďż˝ > >> dbase.c:93: warning: implicit declaration of function > >> ďż˝ďż˝ďż˝mysql_errnoďż˝ďż˝ďż˝ > >> dbase.c:93: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝conďż˝ďż˝ďż˝ > >> dbase.c:94: error: ďż˝ďż˝ďż˝CR_SERVER_GONE_ERRORďż˝ďż˝ďż˝ undeclared > >> (first use in this function) > >> dbase.c:94: error: (Each undeclared identifier is reported only once > >> dbase.c:94: error: for each function it appears in.) > >> dbase.c:94: error: ďż˝ďż˝ďż˝CR_SERVER_LOSTďż˝ďż˝ďż˝ undeclared (first > >> use in this function) > >> dbase.c:98: warning: implicit declaration of function > >> ďż˝ďż˝ďż˝mysql_errorďż˝ďż˝ďż˝ > >> dbase.c:98: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝conďż˝ďż˝ďż˝ > >> dbase.c:98: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝conďż˝ďż˝ďż˝ > >> dbase.c:98: warning: format ďż˝ďż˝ďż˝%sďż˝ďż˝ďż˝ expects type > >> ďż˝ďż˝ďż˝char *ďż˝ďż˝ďż˝, but argument 3 has type ďż˝ďż˝ďż˝intďż˝ďż˝ďż˝ > >> dbase.c:98: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝conďż˝ďż˝ďż˝ > >> dbase.c:98: warning: format ďż˝ďż˝ďż˝%sďż˝ďż˝ďż˝ expects type > >> ďż˝ďż˝ďż˝char *ďż˝ďż˝ďż˝, but argument 3 has type ďż˝ďż˝ďż˝intďż˝ďż˝ďż˝ > >> dbase.c:98: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝conďż˝ďż˝ďż˝ > >> dbase.c:98: warning: format ďż˝ďż˝ďż˝%sďż˝ďż˝ďż˝ expects type > >> ďż˝ďż˝ďż˝char *ďż˝ďż˝ďż˝, but argument 3 has type ďż˝ďż˝ďż˝intďż˝ďż˝ďż˝ > >> dbase.c:98: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝conďż˝ďż˝ďż˝ > >> dbase.c:98: warning: format ďż˝ďż˝ďż˝%sďż˝ďż˝ďż˝ expects type > >> ďż˝ďż˝ďż˝char *ďż˝ďż˝ďż˝, but argument 3 has type ďż˝ďż˝ďż˝intďż˝ďż˝ďż˝ > >> dbase.c:98: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝conďż˝ďż˝ďż˝ > >> dbase.c:98: warning: format ďż˝ďż˝ďż˝%sďż˝ďż˝ďż˝ expects type > >> ďż˝ďż˝ďż˝char *ďż˝ďż˝ďż˝, but argument 3 has type ďż˝ďż˝ďż˝intďż˝ďż˝ďż˝ > >> dbase.c:98: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝conďż˝ďż˝ďż˝ > >> dbase.c:98: warning: format ďż˝ďż˝ďż˝%sďż˝ďż˝ďż˝ expects type > >> ďż˝ďż˝ďż˝char *ďż˝ďż˝ďż˝, but argument 3 has type ďż˝ďż˝ďż˝intďż˝ďż˝ďż˝ > >> dbase.c:98: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝conďż˝ďż˝ďż˝ > >> dbase.c:98: warning: format ďż˝ďż˝ďż˝%sďż˝ďż˝ďż˝ expects type > >> ďż˝ďż˝ďż˝char *ďż˝ďż˝ďż˝, but argument 3 has type ďż˝ďż˝ďż˝intďż˝ďż˝ďż˝ > >> dbase.c: At top level: > >> dbase.c:138: error: expected ďż˝ďż˝ďż˝)ďż˝ďż˝ďż˝ before > >> ďż˝ďż˝ďż˝*ďż˝ďż˝ďż˝ token > >> dbase.c:166: error: expected ďż˝ďż˝ďż˝)ďż˝ďż˝ďż˝ before > >> ďż˝ďż˝ďż˝*ďż˝ďż˝ďż˝ token > >> dbase.c:207: error: expected ďż˝ďż˝ďż˝)ďż˝ďż˝ďż˝ before > >> ďż˝ďż˝ďż˝*ďż˝ďż˝ďż˝ token > >> dbase.c: In function ďż˝ďż˝ďż˝store_resultďż˝ďż˝ďż˝: > >> dbase.c:304: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝resďż˝ďż˝ďż˝ > >> dbase.c:304: warning: implicit declaration of function > >> ďż˝ďż˝ďż˝mysql_store_resultďż˝ďż˝ďż˝ > >> dbase.c:304: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝conďż˝ďż˝ďż˝ > >> dbase.c:305: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝resďż˝ďż˝ďż˝ > >> dbase.c:306: warning: implicit declaration of function > >> ďż˝ďż˝ďż˝mysql_field_countďż˝ďż˝ďż˝ > >> dbase.c:306: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝conďż˝ďż˝ďż˝ > >> dbase.c:311: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝conďż˝ďż˝ďż˝ > >> dbase.c:311: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝conďż˝ďż˝ďż˝ > >> dbase.c:311: warning: format ďż˝ďż˝ďż˝%sďż˝ďż˝ďż˝ expects type > >> ďż˝ďż˝ďż˝char *ďż˝ďż˝ďż˝, but argument 3 has type ďż˝ďż˝ďż˝intďż˝ďż˝ďż˝ > >> dbase.c:311: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝conďż˝ďż˝ďż˝ > >> dbase.c:311: warning: format ďż˝ďż˝ďż˝%sďż˝ďż˝ďż˝ expects type > >> ďż˝ďż˝ďż˝char *ďż˝ďż˝ďż˝, but argument 3 has type ďż˝ďż˝ďż˝intďż˝ďż˝ďż˝ > >> dbase.c:311: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝conďż˝ďż˝ďż˝ > >> dbase.c:311: warning: format ďż˝ďż˝ďż˝%sďż˝ďż˝ďż˝ expects type > >> ďż˝ďż˝ďż˝char *ďż˝ďż˝ďż˝, but argument 3 has type ďż˝ďż˝ďż˝intďż˝ďż˝ďż˝ > >> dbase.c:311: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝conďż˝ďż˝ďż˝ > >> dbase.c:311: warning: format ďż˝ďż˝ďż˝%sďż˝ďż˝ďż˝ expects type > >> ďż˝ďż˝ďż˝char *ďż˝ďż˝ďż˝, but argument 3 has type ďż˝ďż˝ďż˝intďż˝ďż˝ďż˝ > >> dbase.c:311: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝conďż˝ďż˝ďż˝ > >> dbase.c:311: warning: format ďż˝ďż˝ďż˝%sďż˝ďż˝ďż˝ expects type > >> ďż˝ďż˝ďż˝char *ďż˝ďż˝ďż˝, but argument 3 has type ďż˝ďż˝ďż˝intďż˝ďż˝ďż˝ > >> dbase.c:311: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝conďż˝ďż˝ďż˝ > >> dbase.c:311: warning: format ďż˝ďż˝ďż˝%sďż˝ďż˝ďż˝ expects type > >> ďż˝ďż˝ďż˝char *ďż˝ďż˝ďż˝, but argument 3 has type ďż˝ďż˝ďż˝intďż˝ďż˝ďż˝ > >> dbase.c:311: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝conďż˝ďż˝ďż˝ > >> dbase.c:311: warning: format ďż˝ďż˝ďż˝%sďż˝ďż˝ďż˝ expects type > >> ďż˝ďż˝ďż˝char *ďż˝ďż˝ďż˝, but argument 3 has type ďż˝ďż˝ďż˝intďż˝ďż˝ďż˝ > >> dbase.c: In function ďż˝ďż˝ďż˝db_free_resultďż˝ďż˝ďż˝: > >> dbase.c:348: warning: implicit declaration of function > >> ďż˝ďż˝ďż˝mysql_free_resultďż˝ďż˝ďż˝ > >> dbase.c:348: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝resďż˝ďż˝ďż˝ > >> dbase.c:349: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝resďż˝ďż˝ďż˝ > >> dbase.c: In function ďż˝ďż˝ďż˝db_queryďż˝ďż˝ďż˝: > >> dbase.c:398: warning: implicit declaration of function > >> ďż˝ďż˝ďż˝print_whereďż˝ďż˝ďż˝ > >> dbase.c:398: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝conďż˝ďż˝ďż˝ > >> dbase.c: In function ďż˝ďż˝ďż˝db_insertďż˝ďż˝ďż˝: > >> dbase.c:471: warning: implicit declaration of function > >> ďż˝ďż˝ďż˝print_valuesďż˝ďż˝ďż˝ > >> dbase.c:471: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝conďż˝ďż˝ďż˝ > >> dbase.c: In function ďż˝ďż˝ďż˝db_deleteďż˝ďż˝ďż˝: > >> dbase.c:516: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝conďż˝ďż˝ďż˝ > >> dbase.c: In function ďż˝ďż˝ďż˝db_updateďż˝ďż˝ďż˝: > >> dbase.c:559: warning: implicit declaration of function > >> ďż˝ďż˝ďż˝print_setďż˝ďż˝ďż˝ > >> dbase.c:559: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝conďż˝ďż˝ďż˝ > >> dbase.c:568: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝conďż˝ďż˝ďż˝ > >> make: *** [dbase.o] Bďż˝ďż˝ďż˝ďż˝d 1 > >> > >> > >> > >> > >> _______________________________________________ > >> Serusers mailing list > >> Serusers(a)lists.iptel.org > >> http://lists.iptel.org/mailman/listinfo/serusers > >> > > > > >

1 0

[Users] Re: [Serusers] new major SER pre-release, Ottendorf, is out for testing
by Rao Ramaratnamma 06 Nov '06

06 Nov '06

Thank you very much. I am now in learning phase, can someone possibly bring me to the speed and explain how SER compares to OPENSER. I only found this: http://openser.org/index.php?option=com_content&task=view&id=44 and it seems to be outdated now. rr P.S.: excuse my crossposting but I think this is relevant with this question. ----- Original Message ---- From: Jiri Kuthan <jiri(a)iptel.org> To: serusers(a)iptel.org Sent: Monday, November 6, 2006 12:02:24 PM Subject: [Serusers] new major SER pre-release, Ottendorf, is out for testing Ladies and gentlemen, girls and boys, the new testing pre-release of the best SIP proxy server ever, SER, is out! This major pre-release, codenamed Ottendorf, includes fundamental improvements to the software. The improvements allow SER users to achieve significantly higher managability and reliability by better data mode, interop features, security, etc. They recollect voluminous feedback we have received from iptel customers and open source users. In particular, the new features include but are not limited to: - improved data model which eliminates any possible ambiguities. - key under-the-hood changes with impact on performance: timer subsystem overhaul (now fast, with fine resolution, high accuracy and variability), TCP processing overhaul (higher scalability) - key reliability changes: DNS cache and IP blacklisting - key interop changes: support for STUN-bis, presence server aligned to latest IETF development, sanity module for early detection of invalid messages - more management: management features are now available via XML-RPC remotely, as well as via an alternative binary protocol - more security: ratelimit module, more check types in permissions module - scripting features: select operator in combination with revised AVPs, named flags and routes, more types of routes (onsend, timer, branch) - applications: presence has been greatly updated to keep pace with development in the IETF/SIMPLE (includes asynchronous subscription authorization, extended PIDF); web interface includes many new managibility features (multidomain, templates and multilanguage to name some). New scripting flexibility allows safely for features like individual ringing timers and direct access to SIP message elements. - new tools: Nils's script execution tree tool http://lists.iptel.org/pipermail/serdev/2006-October/008124.html and refurbished ser_ctl see the following links for some additional details: - http://www.iptel.org/ser/doc/010whatsnew - http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/NEWS?rev=HEAD Accompanying software, SEMS and SERWEB have been greatly overhauled as well. SEMS now features improved performance (single-threaded model now), service composition via python-IVR-scripting and B2BUA, adaptive playout buffer for better voice quality, and more. SERWEB features new multilevel administrative access system, multidomain support, per-domain customization, remote xml-rpc access, multilanguage support, and more. The new command-line utility, SER_CTL, provides also many great improvements over the previous tool. Interop ------- Shortly, we have been keeping the bar high through our participation in the IETF and SIPITs. SER implements latest interop changes from the IETF (STUN-bis, presence + XCAP, to name the most important ones) and both new features and legacy features were extensively and successfully tested middle of October in SIPit. Next Steps ----------- Keep in mind, this is a pre-release for testing purposes. It is now call on users to test the software hard and provide feedback. In the meantime, the codebase remains frozen and we only commit patches fixing various bugs. Once we get good confidence that Ottendorf has stabilized and is being adequately documented, we will declare the final release. No date has been determined yet. Also, we consider changing CVS to SVN. Again, and this is very important: the final release will be as good as the feedback we obtain for this pre-release. Please do spend your time sending solid feedback back to [serusers]. Roadmap ------- We keep rolling. We are still debating with our generous contributors what is the next possible good things to work on, the topics include but are not limited to as of now: - documentation. - databases are important. LDAP and BDB, DB-API revision. - interop: identity, TLS revision - advanced scripting (scripting engine, vars returns, AVPs) - uncountable improvements (futex, dev/rtc, tm/refcnts, tcp fd cache, .....) Trivia -------- The release name Ottendorf is coming from a place in Saxonian Switzerland, where many iptel fellows and their friends spent a weekend during this Indian summer. http://www.iptel.org/jan/gallery/saxonian_switzerland_2006/ http://www.iptel.org/~andi/2006-09-22-Ottendorf/ http://www.iptel.org/~vlada/saxon_switzerland/index.html http://www.ohlmeier.de/albums/ottendorf_2006/index.html Acknowledgments --------------- This grand achievement is result of extremely many folks who did lot of hard work. Out of folks affiliated with iptel, Andrei and Jan spent immense effort on moving SER forward, with great support from Hallik (command-line tool), Michal, Miklos, Nils, Tomas, Vaclav (presence), Vlada (new STUN). In this context, I would like to thank to our company for the great support. Recently, Bogdan Pintea greatly helped to make the upcoming release better and more stable under sponsorship of his company, iptego. transnexus provided osp support. SEMS has been greatly improved thanks to Stefan and Raphael. Credits for serweb go to Karel. Same importantly, SER has been immensely benefitting of contributors, to name in alphabetic order at least the 'top-of-the-iceberg' fellows: Bogdan Pintea, Cesc Santa, Greg Fausak, Greger Teigre, Hendrik Scholz, <http://tutpro.com/>Juha Heinänen, Klaus Darilion, Maxim Sobolev, Peter Nixon, Uli Abend. Great support on the mailing list has been provided by Greger V. Teigre, Java Rockx, Klaus Darilion, Zeus Ng, and more. Simply, this list is too short and incomplete to capture all the generous contributions that came in -- my appologies for everyone else I have missed. Warnings and Disclaimers ------------------------ This is a TESTING version. We are encouraging people to give it intense testing and provide us with feedback, so that we can properly assess what it takes to declare it final. The volume and number of changes is big, caution is advised. All GPL warnings and disclaimers apply. This is SER, which is not the derivate software known as openser. We have no ideas of in what extent and timeline derivate projects follow (or not) SER. ------------------------------------------------------------------------------- To Give it a Try ----------------- warning: this is just a 'jump-start' for black-belt users. It is not a replacement for documentation or a detailed step-by-step installation guide. It assumes knowledge of operating system, SIP, and SER. PREPARATION - make sure mysql is running. In the default configuration, SER assumes it to be running on localhost, if this does not hold, you need to change ser.cfg. - make sure python is installed (needed for new command-line utilities) SER INSTALLATION FROM CVS - Download from CVS/HEAD using instructions at http://www.iptel.org/downloads - Compile it: make include_modules="mysql xmlrpc" all - Install it the default way of your administrative privileges allow to: make include_modules="mysql xmlrpc" install - Download command-line tool (see more documentation under http://cvs.berlios.de/cgi-bin/viewcvs.cgi/*checkout*/ser/serctl/README?rev=…) - cvs co serctl - Initiate database: ./scripts/mysql/ser_mysql.sh create START - Start it. Call: ser - Begin provisioning users with the new serctl (the capital word are just place holders) # ser_ctl domain add DOMAINNAME # ser_ctl user add USERNAME@DOMAINNAME -p PASSWORD # If you want to have Asserted Identity header for your user # - ser_attr add uid=UID asserted_id="PID" # If you want to have gateway support # - ser_db add attr_types name=gw_ip rich_type=string raw_type=2 \ description="The gateway IP for the default ser.cfg" default_flags=33 # - ser_attr add global gw_ip=GATEWAY-IP - more things you may wish to do: - familiarize yourself with current ser.cfg - send us feedback. -- Jiri Kuthan http://iptel.org/~jiri/ _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[Serusers] Problem with mysql module [mysql.so] file
by tzieleniewski 06 Nov '06

06 Nov '06

Hi all I've installed ser as it is written in the Getting started manuall with ser source 0.9.7-pre1. I've also installed the mysql5.0 binary distribution. unfortunatelly i've encountered a strange problem. At think that this may be caused by the fact that i used o binary distribution od mysql instead of source. Any way i'm just starting my adventure with ser:) so i ask u ser guru's to help me;) When i tried to compile the source again including mysql package the mysql.co file did not appeared? and the log contained only such lines regarding mysql: make[1]: Wej��cie do katalogu `/home/tezet/apps/ser/ser-0.9.7-pre1/modules/mysql' make[1]: Opuszczenie katalogu `/home/tezet/apps/ser/ser-0.9.7-pre1/modules/mysql' make[1]: Wej��cie do katalogu `/home/tezet/apps/ser/ser-0.9.7-pre1/modules/mysql' gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -ftree-vectorize -mtune=athlon64 -DNAME='"ser"' -DVERSION='"0.9.7-pre1"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 4.1.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -I/usr/local/include -I/usr/local/include/mysql -I/usr/local/mysql/include -I/usr/include/mysql -c dbase.c -o dbase.o make[1]: Opuszczenie katalogu `/home/tezet/apps/ser/ser-0.9.7-pre1/modules/mysql' When i tried to compile the package manually using Makefile in the "source"/modules/myqsl directory i got plenty of errors:): dbase.c:34:25: error: mysql/mysql.h: Nie ma takiego pliku ani katalogu dbase.c:35:26: error: mysql/errmsg.h: Nie ma takiego pliku ani katalogu In file included from dbase.c:39: val.h:44: error: expected ��)�� before ��*�� token In file included from dbase.c:40: my_con.h:39: error: expected specifier-qualifier-list before ��MYSQL_RES�� dbase.c: In function ��submit_query��: dbase.c:67: error: ��struct my_con�� has no member named ��timestamp�� dbase.c:68: warning: implicit declaration of function ��mysql_ping�� dbase.c:68: error: ��struct my_con�� has no member named ��con�� dbase.c:72: error: ��struct my_con�� has no member named ��timestamp�� dbase.c:90: warning: implicit declaration of function ��mysql_query�� dbase.c:90: error: ��struct my_con�� has no member named ��con�� dbase.c:93: warning: implicit declaration of function ��mysql_errno�� dbase.c:93: error: ��struct my_con�� has no member named ��con�� dbase.c:94: error: ��CR_SERVER_GONE_ERROR�� undeclared (first use in this function) dbase.c:94: error: (Each undeclared identifier is reported only once dbase.c:94: error: for each function it appears in.) dbase.c:94: error: ��CR_SERVER_LOST�� undeclared (first use in this function) dbase.c:98: warning: implicit declaration of function ��mysql_error�� dbase.c:98: error: ��struct my_con�� has no member named ��con�� dbase.c:98: error: ��struct my_con�� has no member named ��con�� dbase.c:98: warning: format ��%s�� expects type ��char *��, but argument 3 has type ��int�� dbase.c:98: error: ��struct my_con�� has no member named ��con�� dbase.c:98: warning: format ��%s�� expects type ��char *��, but argument 3 has type ��int�� dbase.c:98: error: ��struct my_con�� has no member named ��con�� dbase.c:98: warning: format ��%s�� expects type ��char *��, but argument 3 has type ��int�� dbase.c:98: error: ��struct my_con�� has no member named ��con�� dbase.c:98: warning: format ��%s�� expects type ��char *��, but argument 3 has type ��int�� dbase.c:98: error: ��struct my_con�� has no member named ��con�� dbase.c:98: warning: format ��%s�� expects type ��char *��, but argument 3 has type ��int�� dbase.c:98: error: ��struct my_con�� has no member named ��con�� dbase.c:98: warning: format ��%s�� expects type ��char *��, but argument 3 has type ��int�� dbase.c:98: error: ��struct my_con�� has no member named ��con�� dbase.c:98: warning: format ��%s�� expects type ��char *��, but argument 3 has type ��int�� dbase.c: At top level: dbase.c:138: error: expected ��)�� before ��*�� token dbase.c:166: error: expected ��)�� before ��*�� token dbase.c:207: error: expected ��)�� before ��*�� token dbase.c: In function ��store_result��: dbase.c:304: error: ��struct my_con�� has no member named ��res�� dbase.c:304: warning: implicit declaration of function ��mysql_store_result�� dbase.c:304: error: ��struct my_con�� has no member named ��con�� dbase.c:305: error: ��struct my_con�� has no member named ��res�� dbase.c:306: warning: implicit declaration of function ��mysql_field_count�� dbase.c:306: error: ��struct my_con�� has no member named ��con�� dbase.c:311: error: ��struct my_con�� has no member named ��con�� dbase.c:311: error: ��struct my_con�� has no member named ��con�� dbase.c:311: warning: format ��%s�� expects type ��char *��, but argument 3 has type ��int�� dbase.c:311: error: ��struct my_con�� has no member named ��con�� dbase.c:311: warning: format ��%s�� expects type ��char *��, but argument 3 has type ��int�� dbase.c:311: error: ��struct my_con�� has no member named ��con�� dbase.c:311: warning: format ��%s�� expects type ��char *��, but argument 3 has type ��int�� dbase.c:311: error: ��struct my_con�� has no member named ��con�� dbase.c:311: warning: format ��%s�� expects type ��char *��, but argument 3 has type ��int�� dbase.c:311: error: ��struct my_con�� has no member named ��con�� dbase.c:311: warning: format ��%s�� expects type ��char *��, but argument 3 has type ��int�� dbase.c:311: error: ��struct my_con�� has no member named ��con�� dbase.c:311: warning: format ��%s�� expects type ��char *��, but argument 3 has type ��int�� dbase.c:311: error: ��struct my_con�� has no member named ��con�� dbase.c:311: warning: format ��%s�� expects type ��char *��, but argument 3 has type ��int�� dbase.c: In function ��db_free_result��: dbase.c:348: warning: implicit declaration of function ��mysql_free_result�� dbase.c:348: error: ��struct my_con�� has no member named ��res�� dbase.c:349: error: ��struct my_con�� has no member named ��res�� dbase.c: In function ��db_query��: dbase.c:398: warning: implicit declaration of function ��print_where�� dbase.c:398: error: ��struct my_con�� has no member named ��con�� dbase.c: In function ��db_insert��: dbase.c:471: warning: implicit declaration of function ��print_values�� dbase.c:471: error: ��struct my_con�� has no member named ��con�� dbase.c: In function ��db_delete��: dbase.c:516: error: ��struct my_con�� has no member named ��con�� dbase.c: In function ��db_update��: dbase.c:559: warning: implicit declaration of function ��print_set�� dbase.c:559: error: ��struct my_con�� has no member named ��con�� dbase.c:568: error: ��struct my_con�� has no member named ��con�� make: *** [dbase.o] B��d 1

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Re: [Users] Messege is too big Error Code 513
by Rao Ramaratnamma 06 Nov '06

06 Nov '06

I just began to study the differences between SER and OPENSER. So I am not sure it is the same and it is not some OPENSER related error but the SER doc says this: http://old.iptel.org/ser/doc/seruser/seruser.html#AEN1039 -rr P.S.: viva google ----- Original Message ---- From: ShivaSankar Padhy <shivapd1983(a)gmail.com> To: users(a)openser.org Sent: Monday, November 6, 2006 10:35:42 AM Subject: [Users] Messege is too big Error Code 513 Dear all! I am still new to the openser. Let me ask another silly question:) I am getting error messege as 513, messege is too big, May I know in what condition is is comming??What is the rule to avoid this. Expectig a sooner response as always, Thanks in Advance With cheers, ShivaSankar _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

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

06 Nov '06

Hello User, A small Question , When I My Sip Server without any firewall_ Router .. Hung up is working Fine,,,,,, But I Keep Sip Server with in a Firewall_router, it not Hung upping , When i saw logs from X-lite in Bye method it Changing the uri into Private IP, But should it public IP( THis is not happening ) Please help me............. -- 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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[Users] OT: sipit19 conclusions
by Klaus Darilion 06 Nov '06

06 Nov '06

Hi! I found this link on the sip mailing list. http://www.sipit.net/report19.txt regards klaus -- Klaus Darilion nic.at

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[Serusers] t_replicate for REGISTERs
by SER LIST 06 Nov '06

06 Nov '06

I am replicating a REGISTER from the SER PROXY (0.9.6) to a presence server (0.10.99) using t_replicate. On the presence server I am doing a save_noreply("location"). Everything works fine but the only "issue" I have is that SER PROXY sends the REGISTER to the presence server THREE TIMES. It seems that since SER PROXY is not receiving a reply to the REGISTER sent using t_replicate, it is re-transmitting the REGISTER. I just do not think that this is "clean". I could have the presence send back a "200 OK" to circumvent this but would like to know whether this behaviour is normal? Should t_replicate not simply replicate the request without awaiting a response. Any insight will be appreciated. _________________________________________________________________ Say hello to the next generation of Search. Live Search try it now. http://www.live.com/?mkt=en-ca

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RE: [Serusers] Ser with remote mysql
by John Shellhouse 05 Nov '06

05 Nov '06

Hi, You might want to make sure that db_url parameter is set correctly, because i believe by default the param is set to point to localhost. j >From: ram <talk2ram(a)gmail.com> >To: serusers(a)iptel.org >Subject: [Serusers] Ser with remote mysql >Date: Sun, 5 Nov 2006 22:33:06 +0530 > >Hi all > >I have Ser running one server ( centos) >and another server Database ( FC5) , Mysql mysql-5.0.X > > >iam able to connect from ser server to mysql server from prompt >iam able to connect. > >wheni configure in ser.cfg and run ,iam getting this following error. > >why this still conencting to local server ? > >Nov 5 22:26:11 ser2 ser[22959]: Maxfwd module- initializing >Nov 5 22:26:11 ser2 ser[22959]: new_connection(): Can't connect to local >MySQL server through socket '/var/lib/mysql/mysql.sock' (2) >Nov 5 22:26:11 ser2 ser[22959]: db_init(): Could not create a connection >Nov 5 22:26:11 ser2 ser[22959]: ERROR:domain_db_init: cannot initialize >database connection >Nov 5 22:26:11 ser2 ser[22959]: init_mod(): Error while initializing >module >domain > > >ser.cfg in short >fifo_db_url="mysql://ser:heslo@databaseserverip/ser" >modparam("usrloc", "db_url", "mysql://ser:heslo@databaseserverip/ser") >modparam("acc", "db_url", "mysql://ser:heslo@databaseserverip/ser") > >how can i resolve this problem > >Ram >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers _________________________________________________________________ Find a local pizza place, music store, museum and morethen map the best route! http://local.live.com?FORM=MGA001

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[Serusers] Ser with remote mysql
by ram 05 Nov '06

05 Nov '06

Hi all I have Ser running one server ( centos) and another server Database ( FC5) , Mysql mysql-5.0.X iam able to connect from ser server to mysql server from prompt iam able to connect. wheni configure in ser.cfg and run ,iam getting this following error. why this still conencting to local server ? Nov 5 22:26:11 ser2 ser[22959]: Maxfwd module- initializing Nov 5 22:26:11 ser2 ser[22959]: new_connection(): Can't connect to local MySQL server through socket '/var/lib/mysql/mysql.sock' (2) Nov 5 22:26:11 ser2 ser[22959]: db_init(): Could not create a connection Nov 5 22:26:11 ser2 ser[22959]: ERROR:domain_db_init: cannot initialize database connection Nov 5 22:26:11 ser2 ser[22959]: init_mod(): Error while initializing module domain ser.cfg in short fifo_db_url="mysql://ser:heslo@databaseserverip/ser" modparam("usrloc", "db_url", "mysql://ser:heslo@databaseserverip/ser") modparam("acc", "db_url", "mysql://ser:heslo@databaseserverip/ser") how can i resolve this problem Ram

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Re: [Serusers] Ser with remote mysql
by Atle Samuelsen 05 Nov '06

05 Nov '06

ALWAYS REPLY TO THE LIST. The problem Im guessing your runnin ginto is that the serro user is still trying to use localhost as DB host. Check in config.h. -A * ram <talk2ram(a)gmail.com> [061105 19:44]: > hi > > what problems ? > > Ram > > > On 11/5/06, Atle Samuelsen <clona(a)cyberhouse.no> wrote: > > > >Hi, > > > >Without knowing, I'm betting your getting into tourble related to serro > >user. > > > >- Atle > > > > > >* ram <talk2ram(a)gmail.com> [061105 18:22]: > >> Hi all > >> > >> I have Ser running one server ( centos) > >> and another server Database ( FC5) , Mysql mysql-5.0.X > >> > >> > >> iam able to connect from ser server to mysql server from prompt > >> iam able to connect. > >> > >> wheni configure in ser.cfg and run ,iam getting this following error. > >> > >> why this still conencting to local server ? > >> > >> Nov 5 22:26:11 ser2 ser[22959]: Maxfwd module- initializing > >> Nov 5 22:26:11 ser2 ser[22959]: new_connection(): Can't connect to > >local > >> MySQL server through socket '/var/lib/mysql/mysql.sock' (2) > >> Nov 5 22:26:11 ser2 ser[22959]: db_init(): Could not create a > >connection > >> Nov 5 22:26:11 ser2 ser[22959]: ERROR:domain_db_init: cannot initialize > >> database connection > >> Nov 5 22:26:11 ser2 ser[22959]: init_mod(): Error while initializing > >module > >> domain > >> > >> > >> ser.cfg in short > >> fifo_db_url="mysql://ser:heslo@databaseserverip/ser" > >> modparam("usrloc", "db_url", "mysql://ser:heslo@databaseserverip/ser") > >> modparam("acc", "db_url", "mysql://ser:heslo@databaseserverip/ser") > >> > >> how can i resolve this problem > >> > >> Ram > > > >> _______________________________________________ > >> Serusers mailing list > >> Serusers(a)lists.iptel.org > >> http://lists.iptel.org/mailman/listinfo/serusers > > > >

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[Users] Fw: [Serusers] rewriting From
by Rao Ramaratnamma 05 Nov '06

05 Nov '06

Can possibly the openser folks give me some answer? ----- Forwarded Message ---- From: Rao Ramaratnamma <raramarat(a)yahoo.com> To: serusers(a)iptel.org Sent: Saturday, November 4, 2006 4:34:55 PM Subject: [Serusers] rewriting From Hello, I am new to the community so please forgive me if I am missing some obvious answer but till now I have found only some suggestions what one could or could not do, but is there any known-to-work proven configuration file which rewrites From? rr _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[Serusers] Service Provider Solution
by John Shellhouse 05 Nov '06

05 Nov '06

Hello Everyone, I have finally up ser up and its running well, tested it to be true -thus far - with asterisk backend w/PSTN. Now, i want to start offering services, but only have one pstn channel. I was hoping that someone could advice me on a service provider that is willing to allow me to route calls to their sip router for a monthly fee, and if this is possible so I can offer services to users without being limited to a single channel and see how how sip really works in action with simulatenous calls? Advice? How much is monthly for this type of situation to me (range)? Help? jwshel _________________________________________________________________ Try the next generation of search with Windows Live Search today! http://imagine-windowslive.com/minisites/searchlaunch/?locale=en-us&source=…

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