sr-users November 2006

sr-users@lists.kamailio.org

189 participants
307 discussions

[Users] loose_route() function
by Gregoire 27 Nov '06

27 Nov '06

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi! I am wondering where I can find more information about the loose_route() function. On Wiki, I can see that it is RFC 3261 compliant and understand what it's the difference between loose routing and strict routing, but in my conf. file I have a test like: if(loose_route()){...} and I 'd like to know what kind of value this function can return or when can it return something else than true? Does someone have some information about that? Thanks Greg -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.2.2 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFZv/iI8gmGeMTr0sRAgg8AKCA9CGb6hI9FEhdE8BMm5nlnXDKkACgjZBA NFrRWixi2BtEE6e2YrAfg9g= =ucXk -----END PGP SIGNATURE-----

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[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

2 1

[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.

1 0

[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

2 1

[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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sr-users November 2006