- sr-users - kamailio.org

[Users] Avpops'exported function
by chungyu 12 Apr '07

12 Apr '07

Dear all: Avpops module' exported function avp_db_delete() can do delete one record? example: I use "test" as default table that has two fields. username | number ------------------- bob | 1 alice | 1 bob | 2 my configuration: modparam("avpops", "avp_url", "mysql://ser:heslo@localhost/ser") modparam("avpops", "avp_table", "test") modparam("avpops", "username_column", "username") modparam("avpops", "avp_aliases", "num=i:55") modparam("avpops", "db_scheme", "number_scheme:table=test;value_col=number;value_type=string") if (method=="INVITE"){ avp_db_load("$from/username", "$num/$number_scheme"); //load from table "test" ,number column as string value into AVP name '$num' avp_db_delete("$from/username", "s:test") //delete all bob' record }; if bob calling then bob'record has deleted username | number -------------------- alice | 1 But i want delete a record. ex: delete number value is 1 for bob username | number -------------------- alice | 1 --------------------- bob | 2 avp_db_delete() can do delete one record? Another question is avp_db_store() can do insert new data in table? ex: insert into test (username,number) values ("mary","3") username | number ------------------- alice | 1 ------------------- bob | 2 ------------------- mary | 3 avp_db_store() can do it? sorry,I still can't understand what it mean about avp_db_store() and avp_db_delete(). Can tell me? Thanks a lot in advance Chungyu

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[Users] un-asked-for CANCEL being generated
by Mark Price 12 Apr '07

12 Apr '07

I'm using openser 1.1.x, and seeing something rather strange. We are using openser as a proxy with openser and asterisk on box A, asterisk on box B. Openser listens on ports 5060 and 5065. Asterisk listens on port 5070 on box A and 5060 on box B. When a call arrives from PSTN, it works fine if I forward to asterisk in box B. If I forward to asterisk on box A, everything seems to be going well and then openser generates a CANCEL to the receiving phone for seemingly no good reason. You can see from the packet below that the cancel is being generated by openser, because of the user agent. What might cause this? Thanks, Mark Price U 66.129.95.24:5065 -> 71.199.89.40:16742 CANCEL sip:9043067833@71.199.89.40:16742;rinstance=f5f4de634aa6bff6;transport=udp SIP/2.0..Via: SIP/2.0/UDP 66.129.95.24:5065;branch=z9hG4bK1c31.2cb72f11.0..From: "Mark Price" <sip:9046268172@66.129.95.5>;tag=as11c24889..Call-ID: 1a095b0e1901cead54e8d68c63271aad@66.129.95.5..To: <sip:9043067833@joinuneta.com:5065>..CSeq: 102 CANCEL..User-Agen t: OpenSer (1.1.0-tls (i386/linux))..Content-Length: 0....

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[Users] OpenSER + RTPProxy as Session Border Controller (SBC)
by Martin Klisch 12 Apr '07

12 Apr '07

Hi, is it possible to use OpenSER + RTProxy as a small Session Border Controller? I want to pass ALL calls to an extra OpenSER+RTPProxy, so ALL calls use RTP Sessions via RTPProxy. Can the RTP Proxy signal to the SIP Proxy, if a RTP Session timeouts, so the SIP Proxy can signal a BYE to both legs of the call? I need something like this, to not account timeouted calls. For example broken calls, which never end. any other ideas? professional session border controllers are very very very expensive.

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[Users] TLS and asterisk ?
by Bodin Bruno 12 Apr '07

12 Apr '07

Hi, I know asterisk doesn't support tls but that something strange in SIP protocol : TLS UA -> Openser -> no tls UA : that work TLS UA -> Openser -> asterisk : that don't work any one know something about this ? to use asterisk I make rewritehostport. thank for help

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[Users] How to change the expire parameter in a reply message of REGISTER?
by 杨冬 12 Apr '07

12 Apr '07

hi, all: I want to make a client to send the REGISTER update message in a short time, which is used to prevent the NAT device to close the port for the client. So I want to rewrite the reply message for REGISTER and change the expire parameter to a value of my setting. How can I achieve this? Thanks. andy.yang 2007/04/10

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[Users] Redirecting number of ISUP-IAM to Diversion field of SIP-INVITE translation
by Kettmann, Peter 12 Apr '07

12 Apr '07

Hi, we are using Asterisk (1.2.12.1), chan_ss7 library (0.8.4) and a Digium E1 card to perform calls between IP and PSTN/GSM networks. In the next step we need to translate Redirecting number from ISUP-IAM message to Diversion field of SIP-INVITE message. Do you have an idea to make it on Asterisk ? Is it perhaps possible via the OpenSer Diversion module? Thanks for your help, Peter

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[Users] Possible to use subst() to alter first line of INVITE?
by Leif Madsen 12 Apr '07

12 Apr '07

Evening all! I'm trying to use subst() to alter the first line of an INVITE I'm sending to a remote Asterisk box, although I have been unable to get this working. Perhaps I need to use a different function? The INVITE I'm working with looks like (IPs have been changed to protect the innocent): INVITE sip:s@192.168.1.250 SIP/2.0 Record-Route: <sip:192.168.1.190;lr;ftag=as62d28c09> Via: SIP/2.0/UDP 192.168.1.190;branch=z9hG4bKcce2.b907e716.0 Via: SIP/2.0/UDP 192.168.1.147:5060;branch=z9hG4bK5a287301;rport=5060 From: "lmadsen2#lmentinc" <sip:lmadsen2#lmentinc@192.168.1.147>;tag=as62d28c09 To: <4166289921@switch1> Contact: <sip:lmadsen2#lmentinc@192.168.1.147> Call-ID: 791d9cc101c5cbdd2896c0d357cf644c(a)192.168.1.147 CSeq: 102 INVITE User-Agent: Asterisk Max-Forwards: 69 Remote-Party-ID: "lmadsen2#lmentinc" <sip:lmadsen2#lmentinc@192.168.1.147>;privacy=off;screen=no Date: Thu, 12 Apr 2007 05:00:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 240 I have two subst() I've been playing with. This one works which alters the To: line in the INVITE: subst('/^To: (.*)sip:[^@]*+(.*)$/To: \1$hdr(X-DID)\2/g'); This one does not, which I would like to alter the INVITE sip:s@192.168.1.250 SIP/2.0 of the header. This is the very first line in the INVITE. subst('/^INVITE (.*)sip:[^@]*+(.*)$/INVITE sip:$hdr(X-DID)\2/ig'); I tested the regex online as well, and it seems to work. Anything I'm missing? Is there another function I can use to alter the line I want, or am I up poop creek? Thanks! -- Leif Madsen. http://www.leifmadsen.com http://www.oreilly.com/catalog/asterisk

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re:[Users] Sip Tools
by yanlin 12 Apr '07

12 Apr '07

hi, raviprakash sunkara, http://www.voip-info.org/wiki/view/Open+Source+VOIP+Software search here, maybe you can find something useful. i use SIPp to test performance, it can generate calls in certain sequence. regards, yan lin *** Disclaimer: This message may contain privileged and/or confidential information. If you have received this e-mail in error or are not the intended recipient, you may not use, copy, disseminate or distribute it; do not open any attachments, delete it immediately from your system and notify the sender promptly by e-mail that you have done so. Thank you. ***

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[Serusers] re: Malformed CONTACT field in 302 response
by Jignesh Gandhi 11 Apr '07

11 Apr '07

Hello, I am using SER as a redirect server. I recently posted almost the same question. This time the format of the CONTACT is different. It contains call information. Recently I came across an issue where the 302 sent back by SER is not liked by a soft switch. Particularly , the format of the CONTACT field according to the softswitch.. Here is an excerpt of the 302 reply send back ... Session Initiation Protocol Status-Line: SIP/2.0 302 MovedTemporarily Status-Code: 302 Resent Packet: False Message Header Via:SIP/2.0/UDP 70.167.132.2;branch= z9hG4bK-BroadWorks.70.167.132.2-172.20.20.46V5060-0-476027561-38657848-1175625610577 - From:"7000 Arris-EMTA-SIP"<sip:6782347000@70.167.132.2 ;user=phone>;tag=38657848-1175625610577- SIP Display info: "7000 Arris-EMTA-SIP" SIP from address: sip:6782347000@70.167.132.2 SIP tag: 38657848-1175625610577- To:<sip:+16782349999;csel=noind@172.20.20.46:5060;user=phone>;tag= b27e1a1d33761e85846fc98f5f3a7e58.4baf SIP to address: sip:+16782349999 SIP tag: b27e1a1d33761e85846fc98f5f3a7e58.4baf Call-ID:BW144010577030407-1243907316@70.167.132.2 CSeq:476027561 INVITE Contact: sip:+16782349999;csel=noind@172.20.20.37 :5060;user=phone;transport=udp Server: Sip EXpress router (0.8.14 (i386/linux)) Content-Length: 0 the softswitch is wanting the contact field with <sip:xxxx@xxx.xxx.xxx.xxx:5060> like a FROM or TO URI. It is claiming that since it contains call information within the sip URI, it should be enclosed withing < >. Here is ser.cfg part that does the sl_send_reply(); # do a stateless redirect, if return code is correct if (method=="INVITE") { xlog("L_INFO", "SourceIP <%is> \n"); xlog("L_INFO", "From-uri<%fu>, r-uri <%ru> \n"); if (!gl_redirect()) << custom function that just gets a Server to redirect to { sl_send_reply("480", "TemporarilyUnavailable"); xlog("L_WARN", "Sending a 480 response with r-uri <%ru>\n"); break; } else { sl_send_reply("302", "MovedTemporarily"); xlog("L_DBG", "Sending a 302 response with r-uri <%ru>\n"); break; } }; Any help is appreciated.** thanks, -- Jignesh Gandhi jigpgandhi(a)gmail.com

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[Users] mysql database accounting problems
by Tanguy Bouzéloc 11 Apr '07

11 Apr '07

Hello, I've some issues with my mysql accounting, I can only log successful calls in the acc table and I've no informations about missed calls. In mysql accounting I've only sip 200 ok and no errors. I've not this behavior in my syslog. It's very important for us for evaluate the performance of our providers and we can only do it with a database accouting. For example, If I pass a call and I let it ring til operator timeout, I've no trace of this call in the accounting table. I've also another question, the lcr and gw tables have very small fixed limits (256 for lcr and 32 for gw), since we have much more gateways and sdas I've updated theses limits to a value which fit our needs, do you thinks it can be a problem ? Thanks in advance, Tanguy Bouzéloc Init-Sys # ----------- global configuration parameters ------------------------ #debug=9 fork=yes check_via=no dns=no rev_dns=no port=5060 children=4 sip_warning=0 # ------------------ module loading ---------------------------------- loadmodule "/usr/lib/openser/modules/mysql.so" loadmodule "/usr/lib/openser/modules/nathelper.so" loadmodule "/usr/lib/openser/modules/sl.so" loadmodule "/usr/lib/openser/modules/tm.so" loadmodule "/usr/lib/openser/modules/rr.so" loadmodule "/usr/lib/openser/modules/maxfwd.so" loadmodule "/usr/lib/openser/modules/lcr.so" loadmodule "/usr/lib/openser/modules/acc.so" loadmodule "/usr/lib/openser/modules/xlog.so" loadmodule "/usr/lib/openser/modules/textops.so" loadmodule "/usr/lib/openser/modules/permissions.so" # ----------------- setting module-specific parameters --------------- modparam("lcr", "db_url", "mysql://???") modparam("lcr", "db_mode", 0) modparam("rr","enable_full_lr",1) modparam("acc", "db_url", "mysql://???") modparam("acc", "db_flag", 1) modparam("acc", "db_localtime", 1) modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1 ) modparam("permissions", "db_url", "mysql://???") route { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); return; }; if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); return; }; if (!is_method("REGISTER")) { record_route(); }; if (is_method("BYE") || is_method("CANCEL")) { xlog("L_NOTICE", "End call($ci | $Ri): $fU -> $tU"); if (is_method("BYE")) setflag(1); unforce_rtp_proxy(); }; if (loose_route()) { route(1); return; }; if (!allow_trusted()) { xlog("L_WARN", "Host not trusted ($fu/$si/$rP)"); sl_send_reply("403", "Forbidden"); exit; }; if (!load_gws()) { sl_send_reply("500", "Internal server error, unable to load gateways"); return; } if (!next_gw()) { xlog("L_ERR", "No gw found ($ru/$fu)"); sl_send_reply("503", "Service not available, no gateways found"); return; }; t_on_failure("2"); route(1); } route[1] { setflag(1); if (is_method("INVITE") || is_method("ACK")){ if (is_method("INVITE")) xlog("L_NOTICE", "Incoming call($ci | $Ri): $fU -> $tU"); if (!search("^Content-Length:[ ]*0")) force_rtp_proxy("foc"); # if(src_ip == 217.15.85.178) { # subst('/^From:.*<sip:(.*)@(.*)$/From: "\1" <sip:\1/@\2'); # } t_on_reply("1"); }; if (!t_relay()) { sl_reply_error(); }; } onreply_route[1] { setflag(1); if (status=~"[12][0-9][0-9]"){ if (!search("^Content-Length:[ ]*0")) force_rtp_proxy("foc"); }; } failure_route[2] { if (t_check_status("400")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 400 Bad Request"); } else if (t_check_status("401")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 401 Unauthorized"); } else if (t_check_status("403")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 403 Forbidden"); } else if (t_check_status("404")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 404 Not Found"); } else if (t_check_status("405")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 405 Method Not Allowed"); } else if (t_check_status("406")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 406 Not Acceptable"); } else if (t_check_status("407")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 407 Proxy Authentication Required"); } else if (t_check_status("408")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 408 Request Timeout"); } else if (t_check_status("410")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 410 Gone"); } else if (t_check_status("413")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 413 Request Entity Too Large"); } else if (t_check_status("414")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 414 Request-URI Too Long"); } else if (t_check_status("415")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 415 Unsupported Media Type"); } else if (t_check_status("416")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 416 Unsupported URI Scheme"); } else if (t_check_status("420")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 420 Bad Extension"); } else if (t_check_status("421")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 421 Extension Required"); } else if (t_check_status("423")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 423 Interval Too Brief"); } else if (t_check_status("480")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 480 Temporarily Unavailable"); } else if (t_check_status("481")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 481 Call/Transaction Does Not Exist"); } else if (t_check_status("482")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 482 Loop Detected"); } else if (t_check_status("483")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 483 Too Many Hops"); } else if (t_check_status("484")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 484 Address Incomplete"); } else if (t_check_status("485")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 485 Ambiguous"); } else if (t_check_status("486")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 486 Busy Here"); } else if (t_check_status("487")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 487 Request Terminated"); } else if (t_check_status("488")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 488 Not Acceptable Here"); } else if (t_check_status("491")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 491 Request Pending"); } else if (t_check_status("493")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 493 Undecipherable"); } else if (t_check_status("500")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 500 Server Internal Error"); } else if (t_check_status("501")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 501 Not Implemented"); } else if (t_check_status("502")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 502 Bad Gateway"); } else if (t_check_status("503")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 503 Service Unavailable"); } else if (t_check_status("504")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 504 Server Time-out"); } else if (t_check_status("505")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 505 Version Not Supported"); } else if (t_check_status("513")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 513 Message Too Large"); } else { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU Unknown code"); } setflag(1); if (is_method("INVITE") && t_check_status("408|500|503")) { if (!next_gw()) { xlog("L_ERR", "No more gw ($ru/$fu)"); t_reply("486", "Busy"); # t_reply("503", "Service not available"); exit; } t_on_failure("2"); force_rtp_proxy("foc"); t_relay(); } }

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Re: [Users] Unable to dial with openser and UAs (Softphone)
by Tsitsi 11 Apr '07

11 Apr '07

Hi Andreas, Thank for your reply. Yes, this is a sipwise config. Openser : I use Debian Sarge apt repository : deb http://www.openser.org/debian sarge main Mysql 4.1 I have set up all tables building with openser_mysql Softphone: Windows XP and X-lite 3.0 build 34025 Regards, Tsitsi --------------------------------- Découvrez une nouvelle façon d'obtenir des réponses à toutes vos questions ! Profitez des connaissances, des opinions et des expériences des internautes sur Yahoo! Questions/Réponses.

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[Users] enabling ipv6
by shruu thii 11 Apr '07

11 Apr '07

hi all, i'm using fedora 4.i've installed openser 1.0.1.how do i enable this for ipv6? i want to use linphone clients.this is for pc-pc testing on lan. --------------------------------- Heres a new way to find what you're looking for - Yahoo! Answers

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[Users] Sip Tools
by raviprakash sunkara 11 Apr '07

11 Apr '07

Hello Urs, Anybody can tell , is there to Test load balance in OpenSER and Asterisk , Not Ethereal and ngrep or sniffers. -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 www.hyperion-tech.com Client and Parent company :- www.august-networks.com

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[Users] Openser on ucLinux
by Asterisk Expert 11 Apr '07

11 Apr '07

Hello Everyone, I would like to know if anyone has compiled and installed OpenSER on ucLinux. If yes, can you provide the hardware details and procedure if its different than normal openser installation. -- Thanks Ruchir Brahmbhatt Ecosmob Technologies

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[Users] Cisco/OpenSER called number issues after ENUM lookup
by Luca Corti 11 Apr '07

11 Apr '07

Hello, I've got an AS5350 receiving calls on ISDN E1 and forwarding calls to OpenSER. At the moment ENUM translations are performed on the AS5350 and calls received by OpenSER already contain the correct SIP URIs. Unfortunately I verified that the called party number information is lost in the process and at the called side I cannot distinguish numbers if I associate multiple PSTN numbers to the same sip user. I guess this is Cisco's fault, since it should add a Remote-Party-ID of type "called" to the INVITE. Do I have to move ENUM mapping to OpenSER or is there a quicker solution? thanks Luca

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Re:[Users] how to handle 302-moved temporarily
by Bogdan-Andrei Iancu 11 Apr '07

11 Apr '07

Hi Zahid, a cause might be a bogus received ACK (or improper handled), making openser to retransmit the final response. Please send me privately the openser debug output (use debug 9) for this scenario. regards, bogdan zm23 wrote: >> >> Hi Bogdan, >> I do see an ack sent back from the gw, however, when new >> invite is sent openser relays it according to lcr rules but then a >> 302 is sent back to the gateway. >> >> >> Would it be better to detect and process the 302 message in openser >> itself? >> >> Thanks. >> >> -- >> >> >> >> >> On Mar 30, 2007, at 9:49 AM, Bogdan-Andrei Iancu wrote: >> >>> Hi Zahid, >>> >>> is the second 302 identical to the fist? is there any ACK from CISCO >>> GW to openser as response to first 302? >>> >>> regards, >>> bogdan >>> >>> zm23 wrote: >>>> I made a mistake saying it was a REFER method. This is happening >>>> with just a 302 reply messages. >>>> Any suggestions? >>>> >>>> Thanks. >>>> >>>> >>>> On Mar 28, 2007, at 12:17 PM, zm23 wrote: >>>> >>>>> Hello All, >>>>> I am trying to find out the proper/best way to handle >>>>> REFERs coming from phone. My setup is as follows: >>>>> >>>>> >>>>> Cisco gw openser >>>>> phone gw 2 >>>>> >>>>> >>>>> DID >>>>> 1234 -----> Invite -----------------> >>>>> --------------------> >>>>> >>>>> >>>>> >>>>> <----- 302 msg <--------------------- >>>>> <--------------------302 moved.... >>>>> >>>>> new #5678 sent back >>>>> >>>>> After receiving new invite, openser sends it out to another gateway >>>>> >>>>> 5678 ----> invite ----------------------> >>>>> ---------------------------------------------------> >>>>> >>>>> yet i am still seeing the 302 messages sent back from proxy to >>>>> cisco gw. >>>>> >>>>> <------------------------------- -- 302 moved >>>>> >>>>> >>>>> Call does connect to 5678 but is dropped after few seconds with >>>>> only one way audio. >>>>> >>>>> >>>>> What needs to be done so that when second invite hits proxy, there >>>>> are no more 302s sent back? >>>>> >>>>> Should I process the 302 in proxy itself? >>>>> >>>>> What is the best way to make the 2nd call to 5678 so that it can >>>>> be identified as a separate call? >>>>> >>>>> >>>>> >>>>> >>>>> Thank you in advance for your help in this regards. >>>>> >>>>> --Zahid

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[Users] Does openser support SMS over IMS
by hu li 11 Apr '07

11 Apr '07

Hi, I want to setup an test environment for Short Message Service over IMS network, can openser simulate the IMS network, then I can send short message between two phone through openser in IMS domain? Can openser only be installed in linux OS, not in windows OS? I did not find the binary files for installing in windows. Thanks in advance!

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[Users] XCAP Lite 0.9 released
by Adrian Georgescu 11 Apr '07

11 Apr '07

Change log xcap-lite (0.9) unstable; urgency=low * Validate presrules against presence-rules.xsd * Use Xpath for node selection * Enable multiple domains (embed Domain=example.com in XCAP uri) The software can be downloaded from: http://download.dns-hosting.info/XCAP/xcap-lite-0.9.tar.gz Regards, Adrian

1 0

[Users] Debian package of Openser 1.2.0 with TLS compiled in?
by Kerker Staffan 11 Apr '07

11 Apr '07

Hi are there any debian packages released with the TLS support compiled in? i found all other packages for debian, but none with TLS support. best regards /Staffan Kerker --- Staffan Kerker, Saab Communication Ljungadalsgatan 2, 35180 Växjö, Sweden p. +46 470 42185 c. +46 705 391365 m. staffan.kerker(a)saabgroup.com w. http://www.saabgroup.com

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[Users] One way audio problem (Urgent)
by kyneoh＠dallab.com 11 Apr '07

11 Apr '07

Hello all, Ive been having a problem i cant seem to figure out, and i can see from various Google and list searches that i havent been the only one having this, but havent been able to find a solution yet for more than 3 weeks. Ive a bit frustrate to solve current problem. Ive got 4 UAs here. User agent A & C & D are located under WLAN, and user agent B located in the same LAN as SIP server and STUN server. User agent A & C connected to a restricted NAT that used STUN server to discover its own public IP. User agent A & C are mobile phone. (OS : window mobile 5.0) User agent D is softphone install in laptop. (OS : window XP) When calls gets sent from Box A to Box C (remote) i get a problem with the one-way audio(incoming caller can hear my voice, but i cant hear them). While, calls from Box A to Box D was successful. Both calls are form the same structure, how come calls work on A to D ??? (D) | (A)-----[NAT]-----------[NAT]---------------[SIP Proxy]-----(B) | | (C) [STUN server] *** SIP server and STUN server have assigned different public IP adress. SIP server : OPENSER + RTP Proxy in OPENSUSE Linux. Box A, C : Ageet softphone. Box B ,D : X-Lite softphone. Hope some of you out there are able to help me out a bit, its getting a bit annoying not being able to hear people calling me. Regards, Kum

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[Users] Proxy server only
by Lucas Diaz 11 Apr '07

11 Apr '07

This is my scenario: I got Sip devices using as Registrar and Proxy a public service from Internet. I want to OpenSER in the middle so I can track and manage the calls. How can I configure OpenSER? Do I need to do something in the external Server? Any help would be apreciated. Lucas.

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[Users] memlog question
by Papadopoulos Georgios 11 Apr '07

11 Apr '07

Hello all, I have a question about what is the best way to enable logging in openser 1.1.x. I want also to be able to get the memory dump so I have compiled with the appropriate flags and in my config I have debug=2 memlog=1 log_stderror=no This outputs everything into /var/log/messages, but the problem is that the memory dump section is messed up (lines do not seem to be in the right order). Also it takes a lot of time to stop openser with this configuration and I end up killing it. Is this because dumping all the memory to syslog is very slow? Is there a better way to do this? 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] users in the same OpenSer cant call each other
by Joao Pereira 11 Apr '07

11 Apr '07

Hello My Openser users can call PSTN numbers through a PSTN gateway. But they cant call each other dialing the usernames. Where is the bug in my config? here is my openser.cfg route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; if (!method=="REGISTER") record_route(); if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; # -------------------------------------------------- # Registration # -------------------------------------------------- if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("myserver.com", "subscriber")) { www_challenge("myserver.com", "0"); exit; }; save("location"); exit; }; } if (is_method("INVITE")) { # Route E.164 numbers to PSTN Gateway if (uri=~"sip:\+[1-9][0-9]*@myserver.com" ) { if (isflagset(29)){ route(2); return; } } # Route E.164 numbers to PSTN Gateway if (uri=~"sip:[1-9][0-9]*@myserver.com" ) { if (isflagset(29)){ route(2); return; } } } # method = invite 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; } # Gateway PSTN route[2] { rewritehost("gateways_IP"); t_on_reply("1"); if (!t_relay()) { sl_reply_error(); }; } Thanks for the help Joao Pereira

2 1

[Users] help plz
by shruu thii 11 Apr '07

11 Apr '07

hi all, i'm using fc4.i've installed openser 1.0.1.how should i enable this for ipv6. i'm using linphone client.how do i register? i wanna do pc-pc calling on lan. thankyou. shruthi --------------------------------- Heres a new way to find what you're looking for - Yahoo! Answers

2 1

[Users] force_rtp_proxy() and t_write_unix()
by Victor Gamov 11 Apr '07

11 Apr '07

hi! Can anybody explain me: when I do force_rtp_proxy("c") and then t_write_unix() which request body write to socket -- original or rewrited by force_rtp_proxy() ? Now I trying to configure OpenSER-1.2.0 to work with SEMS and I found that SEMS get original SDP with internal IPs. Is it my OpenSER misconfiguration? Thanks! -- CU, Victor Gamov

2 3

Re: [Serusers] re: Malformed CONTACT field in 302 response
by Olaf Bergmann 11 Apr '07

11 Apr '07

Jignesh Gandhi wrote: > Thank you for your reply. > > Options #1 and 2 are not possible. > I am looking at how to do Option 3. Can you tell me as to > how to go about setting the q value ? Sorry, I am not sure how to do this in SER 0.8.x. Maybe someone else out there? > thanks, > Jignesh gandhi > > > On 4/10/07, *Olaf Bergmann* <Olaf.Bergmann(a)freenet-ag.de > <mailto:Olaf.Bergmann@freenet-ag.de>> wrote: > > Jignesh Gandhi wrote: > > > I recently posted almost the same question. > > And you got a number of good suggestions how to get around your > problem. Largely, the options were: > > - Make your softswitch vendor fix the bug. > - Use another softswitch. > - Adapt your custom function gl_redirect() to create a Contact field > with angle brackets. > - Set the q value. > > Which one did you check out? Did it work? Why not? What about the > other options? > > Regards, > Olaf > > > > > -- > Jignesh Gandhi > jigpgandhi(a)gmail.com <mailto:jigpgandhi@gmail.com> Regards, Olaf

1 0

[Users] One way audio problem (Urgent)
by kyneoh＠dallab.com 11 Apr '07

11 Apr '07

1 0

[Users] Terminate Call
by Javier 11 Apr '07

11 Apr '07

How to I can teminate call when prepayed debit run out? __________________________________________________ Preguntá. Respondé. Descubrí. Todo lo que querías saber, y lo que ni imaginabas, está en Yahoo! Respuestas (Beta). ¡Probalo ya! http://www.yahoo.com.ar/respuestas

3 2

[Users] "detached" timer
by T.R. Missner 10 Apr '07

10 Apr '07

Does anyone know what causes this? */set_timer for 1 list called on a "detached" timer -- ignoring /* I also see */set_timer for 3 list called on a "detached" timer -- ignoring /* When this happens Openser seems to lock up for 10 seconds or so. From searching it appears this is caused by a race but I am not sure what the race is or why this results in an unresponsive openser instance for multiple seconds. Transaction expiration racing reply? Desperately need to understand how this could be triggered so I can get customer to adjust system. Any way to adjust? tried tweaking fr_inv_timer but no joy. TR

6 32

[Serusers] problem with compiling the new CVS release
by İlker Aktuna (Koç.net) 10 Apr '07

10 Apr '07

Hi, I have a working SER installation which includes presence modules etc. Today I tried to compile the latest CVS version because of a requirement to the reigstrar module. (NAT support with mediaproxy) unfortunately, the "make" process did not comlete. I got these errors: make[1]: Entering directory `/root/yeni/sip_router/modules/acc_db' gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -minline-all-stringops -falign-loops -ftree-vectorize -mtune=athlon64 -Wall -DNAME='"ser"' -DVERSION='"2.1.0-dev2"' -DARCH='"i386"' -DOS='linux_' -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 4.1.0"' -D__CPU_i386 -D__OS_linux -DSER_VER=2001000 -DCFG_DIR='"/root/ser2/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST -DDBG_QM_MALLOC -DUSE_TLS -DTLS_HOOKS -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_EPOLL -DHAVE_SIGIO_RT -DSIGINFO64_WORKARROUND -DHAVE_SELECT -c acc_db.c -o acc_db.o acc_db.c:164: error: expected â€˜=â€™, â€˜,â€™, â€˜;â€™, â€˜asmâ€™ or â€˜__attribute__â€™ before â€˜acc_dbfâ€™ acc_db.c:172: error: expected â€˜=â€™, â€˜,â€™, â€˜;â€™, â€˜asmâ€™ or â€˜__attribute__â€™ before â€˜keysâ€™ acc_db.c:173: error: expected â€˜=â€™, â€˜,â€™, â€˜;â€™, â€˜asmâ€™ or â€˜__attribute__â€™ before â€˜valsâ€™ acc_db.c: In function â€˜init_dataâ€™: acc_db.c:406: error: â€˜keysâ€™ undeclared (first use in this function) acc_db.c:406: error: (Each undeclared identifier is reported only once acc_db.c:406: error: for each function it appears in.) acc_db.c:407: error: â€˜valsâ€™ undeclared (first use in this function) acc_db.c: In function â€˜fmt2strarâ€™: acc_db.c:594: error: â€˜valsâ€™ undeclared (first use in this function) acc_db.c: In function â€˜log_requestâ€™: acc_db.c:755: error: â€˜acc_dbfâ€™ undeclared (first use in this function) acc_db.c:760: error: â€˜keysâ€™ undeclared (first use in this function) acc_db.c:760: error: â€˜valsâ€™ undeclared (first use in this function) acc_db.c: In function â€˜child_initâ€™: acc_db.c:974: error: â€˜acc_dbfâ€™ undeclared (first use in this function) acc_db.c: In function â€˜mod_destroyâ€™: acc_db.c:989: error: â€˜acc_dbfâ€™ undeclared (first use in this function) acc_db.c: In function â€˜mod_initâ€™: acc_db.c:1014: warning: implicit declaration of function â€˜bind_dbmodâ€™ acc_db.c:1014: error: â€˜acc_dbfâ€™ undeclared (first use in this function) acc_db.c:1020: warning: implicit declaration of function â€˜DB_CAPABILITYâ€™ acc_db.c:1020: error: â€˜DB_CAP_INSERTâ€™ undeclared (first use in this function) make[1]: *** [acc_db.o] Error 1 make[1]: Leaving directory `/root/yeni/sip_router/modules/acc_db' make: *** [modules] Error 1 # Does anyone have any idea what the problem is ? Thanks, ilker

8 25

[Users] help
by shruu thii 10 Apr '07

10 Apr '07

hi list, i'm a newbie to openser..i've installed openser,please do help me in proceeding further... --------------------------------- Check out what you're missing if you're not on Yahoo! Messenger

1 0

[Users] Segmentation fault when sip_trace is set to on-
by Sergio Gutierrez 09 Apr '07

09 Apr '07

Hi list. I am using openser 1.2.0, compiled to 64 bits on Solaris 10 on Sparc Platform. The service starts ok, but, when I send the sip_trace comand via fifo, the thread which writes on database crashes with segmentation fault. This is the backtrace of the core file: #0 0xffffffff7e990dc8 in strftime () from /lib/64/libc.so.1 #1 0xffffffff7be06a34 in time2mysql (_time=5051504780619285336, _result=0xffffffff7bf09f4c "", _res_len=-1) at /home/operador/openser-1.2.0-notls/modules/mysql//utils.c:54 #2 0xffffffff7be07724 in val2str (_c=0x900, _v=0, _s=0xffffffff7bf09f4c "", _len=0x2ff) at /home/operador/openser-1.2.0-notls/modules/mysql//val.c:147 #3 0xffffffff7be02480 in db_mysql_print_values (_c=0xffffff7fffebce00, _b=0xff <Address 0xff out of bounds>, _l=637645056, _v=0x32393038, _n=-1360931840) at /home/operador/openser-1.2.0-notls/modules/mysql//dbase.c:145 #4 0xffffffff7be041e4 in db_mysql_insert (_h=0xff, _k=0x10032393130, _v=0xffffff7ffff74000, _n=744703) at /home/operador/openser-1.2.0-notls/modules/mysql//dbase.c:584 #5 0xffffffff7910a5a8 in sip_trace (msg=0x101c0000, s1=0x2000 <Address 0x2000 out of bounds>, s2=0x2340 <Address 0x2340 out of bounds>) at /home/operador/openser-1.2.0-notls/modules/siptrace//siptrace.c:472 #6 0x000000010000d2a4 in do_action (a=0x0, msg=0x1000eb00046) at /home/operador/openser-1.2.0-notls//action.c:883 #7 0x000000010000e394 in run_top_route (a=0x10026fb78, msg=0x10028a0a0) at /home/operador/openser-1.2.0-notls//action.c:131 #8 0x000000010006503c in receive_msg (buf=0x100260 <Address 0x100260 out of bounds>, len=586177280, rcv_info=0xffffffff7ffff7a8) at /home/operador/openser-1.2.0-notls//receive.c:156 #9 0x00000001000a1900 in udp_rcv_loop () at /home/operador/openser- 1.2.0-notls//udp_server.c:465 #10 0x000000010003465c in main (argc=117, argv=0xffffff7910acf0ff) at /home/operador/openser-1.2.0-notls//main.c:1399 Thanks in advance for your help and attention. Kind regards, Sergio Gutiérrez EPM Telecomunicaciones.

1 0

[Users] XCAP Lite 0.8 released
by Adrian Georgescu 09 Apr '07

09 Apr '07

Change log xcap-lite (0.8) unstable; urgency=low * Synchronize watchers table with presrules (requires domxml in php4) * Add database locking for PUT * Add more debugging information * Fix realm detection in hostname * Validate documents against standardized schemas (php 5 only) * Update documentation The software can be downloaded from: http://download.dns-hosting.info/XCAP/xcap-lite-0.8.tar.gz Regards, Adrian

1 0

[Serusers] install SER
by Tuan Nguyen 09 Apr '07

09 Apr '07

Hello, I want to install SER in ubuntu. I've downloaded from cvs the version -D 06042007. I 've compiled and haven't no error, now i use "make install" to install but it didn't work without error or inform (I've installed build tools). Anyone can tell me what happen and how can i fix this problem? Thanks in advance T.

1 0

[Users] avp_db_store can't write into database
by chungyu 09 Apr '07

09 Apr '07

Hi all, I am trying to use avp_db_store() stores my table "contact" with caller uri ex: table name: contact ----------------------------- | username | domain | | ----------------------------- | | | | ----------------------------- | | | | ----------------------------- If (method=="INVITE"){ avp_write("$ruri/username", "s:name"); avp_write("$ruri/domain", "s:domain"); avo_db_store("s:name", "s:username/contact") avp_db_store("s:domain", "s:domain/contact"); }; if the call bob then save username and domain of callee in contact callee: bob at iptel.org => ----------------------------- | username | domain | | ----------------------------- | bob | iptel.org | | ----------------------------- | | | | ----------------------------- but not success, system reply ERROR:avpops:db_store: insert faild Am I setting wrong param for avo_db_store()? Thanks a lot in advance chungyu

2 4

[Users] o_uri in 1.2
by Tim 06 Apr '07

06 Apr '07

Does anyone know how I can get hold of the o_uri ( outgoing uri ) for accounting in openser 1.2? I thought it might be destination uri but that didn't work. Thanks Tim

2 3

[Users] cancels not going through
by Vadim Berezniker 06 Apr '07

06 Apr '07

A number is dialed on a phone and the INVITE goes: PHONE => ASTERISK => SER => PHONE If the other phone doesn't pickup and the calling phone is hung up, it sends a BYE to Asterisk. Asterisk sends a CANCEL to SER. SER cancels its transaction but it doesn't relay the CANCEL to the phone so it keeps ringing. Am I missing something here?

2 3

Re: [Users] proxy-authenticate causes phone to drop authenticate creds
by Daniel-Constantin Mierla 06 Apr '07

06 Apr '07

Try to run asterisk in ddebug mode, and watch debug messages, maybe you get some hints why it considers as coming from phone and not from openser. Cheers, Daniel On 04/05/07 18:00, Mark Price wrote: > Daniel, > I tried that. > I think that asterisk still sees the request as coming from the phone > instead of from openser. Is there possibly something I'm doing wrong? > Is there something I probably need to do extra? > > Here is my openser.cfg: >

1 0

Re: [Users] Presence: nothing to retr ansmit ???
by Daniel-Constantin Mierla 06 Apr '07

06 Apr '07

Hello, can you capture and send the network trace for such case (using ngrep or wireshark)? Cheers, Daniel On 04/06/07 04:28, Eddie wrote: > Hi Daniel, > Thank you very much for your advise. > I captured packets by ethereal, but there is no reply for the first > PUBLISH message. > To implement PUBLISH module, I just followed the README in > /openser..../modules/presence > Is there anything else shoule be considerd? > Thanks so much! > Brs, > Eddie > > 在2007-04-05，"Daniel-Constantin Mierla" <daniel(a)voice-system.ro> 写道： > > Hello, > > seems that this is a retransmission of PUBLISH. Check the log above the > pasted messages. There should be the messages of processing the initial > PUBLISH. Check there to see if the reply was sent (you can use > ngrep/wireshark to watch the SIP traffic, therefore, you can see wheter > reply is sent or not). > > Cheers, > Daniel > > > On 04/03/07 10:58, Eddie wrote: > > Hi all, > > I am a new to openser. I am using openser-1.2.0-notls. I met a issue > > when I setup a presence server. > > I use a UDP testing tool to send PUBLISH message : > > PUBLISH sip:111@sip.comverse.com <mailto:111@sip.comverse.com> SIP/2.0 > > To: <sip:111@sip.comverse.com <mailto:111@sip.comverse.com>> > > From: <sip:111@sip.comverse.com <mailto:111@sip.comverse.com>>;tag=aa0e9e60 > > Via: SIP/2.0/UDP > > 172.19.1.190:5060;branch=z9hG4bK-d87543-4f7005092661b51b-1--d87543-;rport > > Call-ID: dd2288324976454c@UG9ja2V0X1BD > > <mailto:dd2288324976454c@UG9ja2V0X1BD> > > CSeq: 4444 PUBLISH > > Contact: <sip:111@172.19.1.190 <mailto:111@172.19.1.190>:5060> > > Expires: 3600 > > Max-Forwards: 70 > > Content-Type: application/pidf+xml > > Event: presence > > Content-Length: 549 > > <?xml version="1.0" encoding="UTF-8" ?> > > <presence xmlns="urn:ietf:params:xml:ns:pidf" > > xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" > > xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" > > xmlns:op="urn:oma:params:xml:ns:pidf:oma-pres" > > xmlns:xsi="http://www.w3.org/2001/XMLSchema-instan > > <tuple id="t0001"> > > <status> > > <basic>open</basic> > > </status> > > <op:overriding-willingness> > > <op:basic>open</op:basic> > > </op:overriding-willingness> > > </tuple> > > </presence> > > but no 200 OK return back from the server. and I set debug level to 7, > > the log is as followed: > > 4(5485) method: <PUBLISH> > > 4(5485) uri: <sip:111@sip.comverse.com <mailto:111@sip.comverse.com>> > > 4(5485) version: <SIP/2.0> > > 4(5485) parse_headers: flags=2 > > 4(5485) DEBUG:parse_to:end of header reached, state=10 > > 4(5485) DBUG:parse_to: display={}, ruri={sip:111@sip.comverse.com <mailto:111@sip.comverse.com>} > > 4(5485) DEBUG: get_hdr_field: <To> [28]; uri=[sip:111@sip.comverse.com <mailto:111@sip.comverse.com>] > > 4(5485) DEBUG: to body [<sip:111@sip.comverse.com <mailto:111@sip.comverse.com>> > > > > 4(5485) Found param type 232, <branch> = > > <z9hG4bK-d87543-4f7005092661b51b-1--d87543->; state=6 > > 4(5485) Found param type 235, <rport> = <n/a>; state=17 > > 4(5485) end of header reached, state=5 > > 4(5485) parse_headers: Via found, flags=2 > > 4(5485) parse_headers: this is the first via > > 4(5485) After parse_msg... > > 4(5485) preparing to run routing scripts... > > 4(5485) parse_headers: flags=100 > > 4(5485) get_hdr_field: cseq <CSeq>: <4445> <PUBLISH> > > 4(5485) DEBUG:maxfwd:is_maxfwd_present: value = 70 > > 4(5485) DEBUG: add_param: tag=aa0e9e60 > > 4(5485) DEBUG:parse_to:end of header reached, state=29 > > 4(5485) DBUG:parse_to: display={}, ruri={sip:111@sip.comverse.com <mailto:111@sip.comverse.com>} > > 4(5485) parse_headers: flags=200 > > 4(5485) DEBUG: get_hdr_body : content_length=549 > > 4(5485) found end of header > > 4(5485) find_first_route: No Route headers found > > 4(5485) loose_route: There is no Route HF > > 4(5485) grep_sock_info - checking if host==us: 16==12 && > > [sip.comverse.com] == [172.19.1.170] > > 4(5485) grep_sock_info - checking if port 5060 matches port 5060 > > 4(5485) grep_sock_info - checking if host==us: 16==12 && > > [sip.comverse.com] == [172.19.1.170] > > 4(5485) grep_sock_info - checking if port 5060 matches port 5060 > > 4(5485) DEBUG: t_newtran: T on entrance=0xffffffff > > 4(5485) parse_headers: flags=ffffffffffffffff > > 4(5485) parse_headers: flags=78 > > 4(5485) t_lookup_request: start searching: hash=29736, isACK=0 > > 4(5485) DEBUG: RFC3261 transaction matched, > > tid=-d87543-4f7005092661b51b-1--d87543- > > 4(5485) DEBUG:tm:REF_UNSAFE: after is 1 > > 4(5485) DEBUG: t_lookup_request: transaction found (T=0xb6148f48) > > 4(5485) DEBUG:tm:t_retransmit_reply: nothing to retransmit > > 4(5485) DEBUG:tm:UNREF_UNSAFE: after is 0 > > 4(5485) DEBUG:destroy_avp_list: destroying list (nil) > > 4(5485) receive_msg: cleaning up > > 5(5488) PRESENCE: msg_watchers_clean:cleaning pending subscriptions > > 5(5488) PRESENCE: msg_active_watchers_clean:cleaning expired watcher > > information > > 5(5488) PRESENCE:msg_presentity_clean:cleaning expired presentity > > information > > I am confused about the issue, and I am not sure whether it is caused > > by configuration files or mysql database. > > Please help me. Thanks so much ! > > Brs, > > Eddie > > > > > > ------------------------------------------------------------------------ > > 每天上班第一件要做的事情是？马上看看>> <http://www.126.com/> > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org <mailto:Users@openser.org> > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > > ------------------------------------------------------------------------ > 每天上班第一件要做的事情是？马上看看>> <http://www.126.com/>

1 0

[Users] Help wanted
by Flavio Goncalves 06 Apr '07

06 Apr '07

Dear coleagues, I need help to translate an OpenSER eBook (building a SIP provider) from portuguese to english. I'm willing paying US$75,00 (paypal) per chapter. If any of you have interest, please contact me at flavio(a)asteriskguide.com Regards, Flavio E. Goncalves

1 0

[Serusers] Can SER do it? Extract info
by Gustavo Passos Tourinho 05 Apr '07

05 Apr '07

Hi There. Can SER extract info from request uri? For example, If I have this uri: INVITE sip:123456#0159559432@my.sip.domain How can I get data before "#" (123456)? I dont want to "strip" this information, just want to get it for further utilization. Is that possible? If it can be done, can be done using variable number of digits? For example: INVITE sip:123#0159559432@my.sip.domain INVITE sip:1234567#0159559432@my.sip.domain Thanks in advanced. Regards, Gustavo

2 1

[Serusers] support to implemente multi leg call radius accounting
by Rosario Pingaro 05 Apr '07

05 Apr '07

We are looking to find support to implement multi leg call accounting in our 0.9.6 ser configuration. We use radius accounting. Any consultant interested supporting us may submit a quotation to this email. Reagrds Rosario

1 0

[Users] t_relay() has no effect !
by JP Ramoul 05 Apr '07

05 Apr '07

2 1

[Serusers] uninstall
by Tuan Nguyen 05 Apr '07

05 Apr '07

Hi everybody, I have installed ser-0.9.6 and now i want to uninstall it to new install in correctly way because it works not property. how can i complete uninstall old version. thanks in advance T.

2 1

Re: [Serusers] error by ser-2.1
by Ladislav Andel 05 Apr '07

05 Apr '07

Here is the answer: http://lists.iptel.org/pipermail/serusers/2007-April/032657.html or http://lists.iptel.org/pipermail/serusers/2007-April/032655.html You need to download stable or earlier version of SER 2.1 because of temporary broken database API. Lada Tuan Nguyen wrote: > Hi Andel, > > I do not see anywhere say that "2 posts before question". > > I am newbie by SER, I have many problem to try to run it, how can > solve this problem without question? and what is "Support for users of > the SIP Express Router" in http://lists.iptel.org/mailman/listinfo/ ? > > Please tell me what should i do to solve my problem when i don`t find > answer in archive? > thanks, > T. > On 4/5/07, *Ladislav Andel* <ladaan(a)iptel.org > <mailto:ladaan@iptel.org>> wrote: > > Hi, > Have you been reading the serusers mailing list ? > Just 2 posts before your question you will find the answer. > > It is for everybody. Read and search the list before asking. > > Peace :), > Lada > > >

1 0

[Serusers] error by ser-2.1
by Tuan Nguyen 05 Apr '07

05 Apr '07

hi everyone, when i compile new version i get follow error: make[1]: Entering directory `/net/u/ntm/Desktop/ser- 2.1.0+cvs20070404/modules/auth_db' make[1]: Leaving directory `/net/u/ntm/Desktop/ser- 2.1.0+cvs20070404/modules/auth_db' make[1]: Entering directory `/net/u/ntm/Desktop/ser- 2.1.0+cvs20070404/modules/auth_db' gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -minline-all-stringops -falign-loops -ftree-vectorize -mtune=athlon64 -Wall -DNAME='"ser"' -DVERSION='"2.1.0+cvs20070404"' -DARCH='"i386"' -DOS='linux_' -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 4.1.2"' -D__CPU_i386 -D__OS_linux -DSER_VER=2001000 -DCFG_DIR='"/usr/local/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST -DDBG_QM_MALLOC -DUSE_TLS -DTLS_HOOKS -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_EPOLL -DHAVE_SIGIO_RT -DSIGINFO64_WORKARROUND -DHAVE_SELECT -c authdb_mod.c -o authdb_mod.o In file included from authdb_mod.c:49: authdb_mod.h:57: error: expected '=', ',', ';', 'asm' or '__attribute__' before 'auth_dbf' authdb_mod.c:106: error: expected '=', ',', ';', 'asm' or '__attribute__' before 'auth_dbf' authdb_mod.c: In function 'child_init': authdb_mod.c:165: error: 'auth_dbf' undeclared (first use in this function) authdb_mod.c:165: error: (Each undeclared identifier is reported only once authdb_mod.c:165: error: for each function it appears in.) authdb_mod.c: In function 'mod_init': authdb_mod.c:180: warning: implicit declaration of function 'bind_dbmod' authdb_mod.c:180: error: 'auth_dbf' undeclared (first use in this function) authdb_mod.c: In function 'destroy': authdb_mod.c:206: error: 'auth_dbf' undeclared (first use in this function) authdb_mod.c: In function 'authdb_fixup': authdb_mod.c:226: error: 'auth_dbf' undeclared (first use in this function) authdb_mod.c:231: warning: implicit declaration of function 'table_version' make[1]: *** [authdb_mod.o] Error 1 make[1]: Leaving directory `/net/u/ntm/Desktop/ser- 2.1.0+cvs20070404/modules/auth_db' make: *** [modules] Error 1 anyone know why and how correct it. thanks T.

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[Users] Advanced OpenSER Programming Course, Paris, INRIA, June 4-6, 2007
by Daniel-Constantin Mierla 05 Apr '07

05 Apr '07

Hello, following some discussions started during first OpenSER Summit, we are happy to announce that first Advanced Programming Course will happen very soon. Courtesy of INRIA (many thanks to Philippe Sultan) we have the room and a good connected location. There will be three full days of deep analysis and survey of internal architecture and structures, best practices to add new features, optimizations and OpenSER's specific components (memory manager, locking system, ...). The course is dedicated to people willing to develop OpenSER extensions. It will be a class of 14 persons. The participation fee will be just to cover the logistic required for this event, estimated to be about 150-200Euro per person, for the full course (mainly is subject of how many participants will be registered). Teachers will be Bogdan-Andrei Iancu and Daniel-Constantin Mierla, from OpenSER development team, and the primary goal is to attract more developers to core components of OpenSER and build a higher development squad. Also, there will be open discussions to improve existing code and functionalities. If the interest for this event is high, the primary selection criteria will include the level of contributions so far and willingness to contribute in the future. New similar events will be scheduled if the number of people showing interest is enough for another one or more classes. For this one, we expect to have people with pretty good knowledge of SIP and OpenSER operation, to avoid long introduction lessons. If you are interested in this course, please send an email to team(a)openser.org. <mailto:team@openser.org.> News link: http://www.openser.org/mos/view/News/NewsItem/Advanced-OpenSER-Programming-… Cheers, Daniel

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[Users] Presence: nothing to retransmit ???
by Eddie 05 Apr '07

05 Apr '07

Hi all, I am a new to openser. I am using openser-1.2.0-notls. I met a issue when I setup a presence server. I use a UDP testing tool to send PUBLISH message : PUBLISH sip:111@sip.comverse.com SIP/2.0 To: <sip:111@sip.comverse.com> From: <sip:111@sip.comverse.com>;tag=aa0e9e60 Via: SIP/2.0/UDP 172.19.1.190:5060;branch=z9hG4bK-d87543-4f7005092661b51b-1--d87543-;rport Call-ID: dd2288324976454c@UG9ja2V0X1BD CSeq: 4444 PUBLISH Contact: <sip:111@172.19.1.190:5060> Expires: 3600 Max-Forwards: 70 Content-Type: application/pidf+xml Event: presence Content-Length: 549 <?xml version="1.0" encoding="UTF-8" ?> <presence xmlns="urn:ietf:params:xml:ns:pidf" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:op="urn:oma:params:xml:ns:pidf:oma-pres" xmlns:xsi="<A href="http://www.w3.org/2001/XMLSchema-instan">http://www.w3.org/2001/XMLSchema-instan</A><BR> <tuple id="t0001"><BR> <status><BR> <basic>open</basic><BR> </status><BR> <op:overriding-willingness><BR> <op:basic>open</op:basic><BR> </op:overriding-willingness><BR> </tuple><BR></presence></DIV> <DIV> </DIV> <DIV>but no 200 OK return back from the server. and I set debug level to 7, the log is as followed: </DIV> <DIV> </DIV> <DIV>4(5485) method: <PUBLISH><BR> 4(5485) uri: <sip:111@sip.comverse.com><BR> 4(5485) version: <SIP/2.0><BR> 4(5485) parse_headers: flags=2<BR> 4(5485) DEBUG:parse_to:end of header reached, state=10<BR> 4(5485) DBUG:parse_to: display={}, ruri={sip:111@sip.comverse.com}<BR> 4(5485) DEBUG: get_hdr_field: <To> [28]; uri=[sip:111@sip.comverse.com]<BR> 4(5485) DEBUG: to body [<sip:111@sip.comverse.com><BR><BR> 4(5485) Found param type 232, <branch> = <z9hG4bK-d87543-4f7005092661b51b-1--d87543->; state=6<BR> 4(5485) Found param type 235, <rport> = <n/a>; state=17<BR> 4(5485) end of header reached, state=5<BR> 4(5485) parse_headers: Via found, flags=2<BR> 4(5485) parse_headers: this is the first via<BR> 4(5485) After parse_msg...<BR> 4(5485) preparing to run routing scripts...<BR> 4(5485) parse_headers: flags=100<BR> 4(5485) get_hdr_field: cseq <CSeq>: <4445> <PUBLISH><BR> 4(5485) DEBUG:maxfwd:is_maxfwd_present: value = 70<BR> 4(5485) DEBUG: add_param: tag=aa0e9e60<BR> 4(5485) DEBUG:parse_to:end of header reached, state=29<BR> 4(5485) DBUG:parse_to: display={}, ruri={sip:111@sip.comverse.com}<BR> 4(5485) parse_headers: flags=200<BR> 4(5485) DEBUG: get_hdr_body : content_length=549<BR> 4(5485) found end of header<BR> 4(5485) find_first_route: No Route headers found<BR> 4(5485) loose_route: There is no Route HF<BR> 4(5485) grep_sock_info - checking if host==us: 16==12 && [sip.comverse.com] == [172.19.1.170]<BR> 4(5485) grep_sock_info - checking if port 5060 matches port 5060<BR> 4(5485) grep_sock_info - checking if host==us: 16==12 && [sip.comverse.com] == [172.19.1.170]<BR> 4(5485) grep_sock_info - checking if port 5060 matches port 5060<BR> 4(5485) DEBUG: t_newtran: T on entrance=0xffffffff<BR> 4(5485) parse_headers: flags=ffffffffffffffff<BR> 4(5485) parse_headers: flags=78<BR> 4(5485) t_lookup_request: start searching: hash=29736, isACK=0<BR> 4(5485) DEBUG: RFC3261 transaction matched, tid=-d87543-4f7005092661b51b-1--d87543-<BR> 4(5485) DEBUG:tm:REF_UNSAFE: after is 1<BR> 4(5485) DEBUG: t_lookup_request: transaction found (T=0xb6148f48)<BR> 4(5485) DEBUG:tm:t_retransmit_reply: nothing to retransmit<BR> 4(5485) DEBUG:tm:UNREF_UNSAFE: after is 0<BR> 4(5485) DEBUG:destroy_avp_list: destroying list (nil)<BR> 4(5485) receive_msg: cleaning up<BR> 5(5488) PRESENCE: msg_watchers_clean:cleaning pending subscriptions<BR> 5(5488) PRESENCE: msg_active_watchers_clean:cleaning expired watcher information<BR> 5(5488) PRESENCE:msg_presentity_clean:cleaning expired presentity information</DIV> <DIV> </DIV> <DIV>I am confused about the issue, and I am not sure whether it is caused by configuration files or mysql database.</DIV> <DIV> </DIV> <DIV>Please help me. Thanks so much !</DIV> <DIV> </DIV> <DIV> </DIV> <DIV>Brs,</DIV> <DIV> </DIV> <DIV>Eddie</DIV> <DIV> </DIV>

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[Users] host identification
by Joao Pereira 05 Apr '07

05 Apr '07

Hello does openSER uses the host identificayion in any way to authenticate its costumers? Because I re-installed my server and thats the only difference from the previous instalation (its even in the same machine). Thanks regards Joao pereira

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[Users] openser 1.2 avp
by laurent schweizer 05 Apr '07

05 Apr '07

Hello, with openser 1.0 I have this if ( avp_db_load("$low_c_service/username","i:/serviceLowCoastredirect")) and for openser 1.2 I try to do this if ( avp_db_load("$avp(low_c_service)/username","i:/serviceLowCoastredirect")) or if ( avp_db_load("$avp($low_c_service)/username","i:/serviceLowCoastredirect")) low_c_service is an alias avp_aliases="low_c_service=i:500" but I have always an error. 0(0) ERROR:avops:parse_avp_db:: error - bad avp flags 0(0) ERROR:avpops:fixup_db_avp: parse failed 0(0) ERROR: fix_actions: fixing failed (code=-1) at cfg line 633 0(0) ERROR: fix_expr : fix_actions error

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[Users] SIP SUBSCRIBE AND MESSAGE msgs
by Abdurrahman Sahin 05 Apr '07

05 Apr '07

hi all; i installed openser now trying to setup a basic text commnuication between two clients on the same pc (one client is windows msngr and the other one is x-lite ) it seems to me SUBSCRIBE and MESSAGE sip msgs are not processed. i generated openser.cfg with the wizard at sipwise.com site here is my ngrep dump for SUBSCRIBE msg ==================== U ua_public_ip:29350 -> 192.168.200.2:5060 SUBSCRIBE sip:burak@testsrv200 SIP/2.0. Via: SIP/2.0/UDP ua_public_ip:29347. Max-Forwards: 70. From: "apo@testsrv200" <sip:apo@testsrv200>;tag=29774bb79657418eb518068a22304b7c;epid=13d631ff91. To: <sip:burak@testsrv200>. Call-ID: 15088cb5e87d4fb887cc073d0fd9df07. CSeq: 1 SUBSCRIBE. Contact: <sip:ua_public_ip:29347>. User-Agent: RTC/1.3. Event: presence. Accept: application/xpidf+xml, text/xml+msrtc.pidf, application/pidf+xml. Supported: com.microsoft.autoextend. Supported: ms-benotify. Proxy-Require: ms-benotify. Content-Length: 0. . # U 192.168.200.2:5060 -> ua_public_ip:29350 SIP/2.0 501 Method Not Supported Here. Via: SIP/2.0/UDP ua_public_ip:29347;rport=29350. From: "apo@testsrv200" <sip:apo@testsrv200>;tag=29774bb79657418eb518068a22304b7c;epid=13d631ff91. To: <sip:burak@testsrv200>;tag=329cfeaa6ded039da25ff8cbb8668bd2.ea73. Call-ID: 15088cb5e87d4fb887cc073d0fd9df07. CSeq: 1 SUBSCRIBE. Server: OpenSER (1.2.0-notls (x86_64/linux)). Content-Length: 0. MESSAGE msg (while trying to establish a text chat) =========================== U ua_public_ip:29355 -> 192.168.200.2:5060 MESSAGE sip:apo@testsrv200 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.33:58348;branch=z9hG4bK-d87543-2a3b75359734fe48-1--d87543-;rport. Max-Forwards: 70. To: "apo"<sip:apo@testsrv200>. From: "burak"<sip:burak@testsrv200>;tag=fb1a9f68. Call-ID: NTNhMGZlMmYwODIxNmFlNmNmYThmYjQ0NzFhOGFjODI.. CSeq: 2 MESSAGE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/im-iscomposing+xml. User-Agent: X-Lite release 1006e stamp 34025. Content-Length: 259. . <?xml version='1.0' encoding='UTF-8'?> <isComposing xmlns='urn:ietf:params:xml:ns:im-iscomposing' xmlns:xsi='http://www.w3.org/2001/XMLSchema-instance'> <state>active</state> <contenttype>goober</contenttype> <refresh>60</refresh> </isComposing> # U 192.168.200.2:5060 -> ua_public_ip:29355 SIP/2.0 501 Method Not Supported Here. Via: SIP/2.0/UDP 192.168.1.33:58348;branch=z9hG4bK-d87543-2a3b75359734fe48-1--d87543-;rport=29355;received=ua_public_ip. To: "apo"<sip:apo@testsrv200>;tag=329cfeaa6ded039da25ff8cbb8668bd2.402d. From: "burak"<sip:burak@testsrv200>;tag=fb1a9f68. Call-ID: NTNhMGZlMmYwODIxNmFlNmNmYThmYjQ0NzFhOGFjODI.. CSeq: 2 MESSAGE. Server: OpenSER (1.2.0-notls (x86_64/linux)). Content-Length: 0. . .

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[Users] Strange behaviour on siptrace module
by Sergio Gutierrez 05 Apr '07

05 Apr '07

Hi list. I am having a problem with siptrace module. When I send the command sip_trace on through fifo, openser do not send back responses to user agents. On logfile (with debug=9), I just receive this message: sip_trace: storing info... On database do not appear any trace message. Thanks in advance for your attention. Kind regards. Sergio G.

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[Users] proxy-authenticate causes phone to drop authenticate creds
by Mark Price 05 Apr '07

05 Apr '07

Hi, I am using openser 1.1.x as a front end to asterisk. Here is what happens in the dialog: 1. phone sends invite 2. openser says "Unauthorized" 3. phone sends invite + Authorization header 4. openser auths the packet and forwards to asterisk 5. asterisk says "proxy authentication required" 6. openser says "proxy authentication required" here is where it gets interesting. 7.a On some devices, the phone sends invite + Authorization header + Proxy-Authorizaiton header. 7.b On some devices, the phone sends invite + Proxy-Authorization header. How does one work around 7.b? Really, what I want to see is how to make it so that asterisk does not try to authenticate the call. However, since that doesn't seem to be possible, what is the sensible way to route these calls? Thanks, Mark Price

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[Users] Authentication problem
by charl van vuuren 05 Apr '07

05 Apr '07

I'm trying to setup Openser (1.1) as registrar with mediaproxy and mysql all on the same box. I am currently using the SIP Proxy/Registrar with Mediaproxy config from sipwise. MySQL and Mediaproxy are running. Xlite registers successfully with Openser with the users I have created. The problem comes with the authentication. Both www_authorize and proxy_authorize fail. www_authorize fails at regtistration and proxy_authorize off course then fails when I try make a call. Any suggestions? Thanx

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[Users] user agent in avp
by Mark Price 05 Apr '07

05 Apr '07

Hi, In openser 1.1.x, is it possible to get similar functionality to that suggested by the following: avp_check($hdr(ua[0]),"re/Some Phone UA/") thanks Mark Price

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[Users] radius attributes
by Terry Robert 05 Apr '07

05 Apr '07

Hi openser, Following the tutorial on FreeRadius integration (http://www.openser.org/docs/openser-radius-1.0.x.html), I'm seeing accounting packets successfully sent to Radiusd. However, some attributes are missing, notably: UserName, CallingStationId, and CalledStationId. Radius client dictionary contains: ATTRIBUTE User-Name 1 string ATTRIBUTE Password 2 string ... ATTRIBUTE Called-Station-Id 30 string ATTRIBUTE Calling-Station-Id 31 string Output from 'radiusd -X' shows: ... Module: Instantiated detail (detail) Listening on authentication *:1812 Listening on accounting *:1813 Ready to process requests. rad_recv: Accounting-Request packet from host 10.10.90.198:32812, id=10, length=120 Acct-Status-Type = Start Service-Type = IAPP-Register Attr-102 = 0x000000c8 Error-Cause = 1 Event-Timestamp = "Apr 4 2007 15:10:27 PDT" Attr-105 = 0x3937333153495070546167303031 Attr-104 = 0x3937323853495070546167303131 Acct-Session-Id = "1-9731(a)10.10.90.51" NAS-Port = 5060 Acct-Delay-Time = 0 NAS-IP-Address = 10.10.90.198 Processing the preacct section of radiusd.conf modcall: entering group preacct for request 0 modcall[preacct]: module "preprocess" returns noop for request 0 rlm_acct_unique: WARNING: Attribute User-Name was not found in request, unique ID MAY be inconsistent ... Any tips on how to fix this? Thanks.

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RE: [Serusers] Problems with ACK
by Jiri Kuthan 04 Apr '07

04 Apr '07

I think the key message for problem analysis, client's ACK, is missing. This is the setting, right:? .100/UAC -----NAT/.248----> SER outbound @ .246 -----> UAS @ .239 The ACK in your message dump is only that one sent from proxy (246) to UAS (.239). The ACK which can possibly mismatch and produce the error message you mentioned must come from upstream. To be properly formatted, it must have Via identical to that of initial INVITE ( Via: SIP/2.0/UDP 192.168.1.100:5060;rport=46127;received=100.100.100.248;branch=z9hG4bK2893084087) -jiri At 17:50 27/03/2007, Ricardo Martinez wrote: No. Time Source Destination Protocol Info > 14 6.102719 100.100.100.239 100.100.100.246 SIP Status: 487 Request Terminated >Session Initiation Protocol > Status-Line: SIP/2.0 487 Request Terminated > Status-Code: 487 > Resent Packet: False > Message Header > Via: SIP/2.0/UDP 100.100.100.246;branch=z9hG4bK895c.a98d35.0 > Via: SIP/2.0/UDP 192.168.1.100:5060;rport=46127;received=100.100.100.248;branch=z9hG4bK2893084087 >No. Time Source Destination Protocol Info > 15 6.103011 100.100.100.246 100.100.100.239 SIP Request: ACK sip:005622408196@100.100.100.239 >Session Initiation Protocol > Request-Line: ACK sip:005622408196@100.100.100.239 SIP/2.0 > Method: ACK > Resent Packet: False > Message Header > Via: SIP/2.0/UDP 100.100.100.246;branch=z9hG4bK895c.a98d35.0 > f: "cl" <sip:cl@sipvoiss.desa.redvoiss.net>;tag=1587516403 > SIP Display info: "cl" > SIP from address: sip:cl@sipvoiss.desa.redvoiss.net > SIP tag: 1587516403 > i: <mailto:e94a98aa177e8579e5242a2091c0ac5f@192.168.1.100>e94a98aa177e8579e5242a2091c0ac5f(a)192.168.1.100 > To: <sip:0299005622408196@sipvoiss.desa.redvoiss.net>;tag=9146c297a4 >Thanks > >Ricardo.- >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

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[Serusers] error with CPL script( CPL module of SER)
by dillip parida 04 Apr '07

04 Apr '07

Hi all, I am trying to implement call redirection feature using cpl script. I have complied cpl module that supports the cpl script. I have uploaded the cpl script. plz help me to solve the problem.. I have written this mail regarding the error that > we are getting in testing the cpl script of Call redirection. > > We have three X-lite phones with SIP uris 2001(a)172.16.1.24, > ,2002(a)172.16.1.28 and 2003(a)172.16.1.31 and all the three soft phones > have been registered with SER proxy running on 10.2.3.123. All the > three users have added in the SER database along with mail ids. > > user mail id > 2001 2001(a)172.16.1.24 > 2002 2002(a)172.16.1.28 > 2003 2003(a)172.16.1.31 > > The call scenario is as follows.. > 1.user 2001 will make a call to user 2002. > 2. If the user 2002 is either not responding (No answer) or busy, then > the call > will be redirected to user 2003. > 3. we able to make a call to 2002,and user 2002's X-lite phone is > ringing but the call is > not getting redirected. > > The cpl script for redirection has been deployed in SER proxy. > > I am sending the call log of sip phone:2001 as an attachment along > with snapshot sip phone 2001. > > I am sending cpl script at the bottam of this mail.plz find the cpl script at the end of this mail. > As shown in the xlite screen "500 CPL script execution failed" can u > please suggest me if there is > any error in XML/CPL syntax. > > > > > > > > > > > > > > > > > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------ > > <?xml version="1.0"?> > <!DOCTYPE html cpl PUBLIC "-//IETF//DTD RFCxxx CPL 1.0//EN" "cpl.dtd"> > <cpl> > <subaction id="pda"> > <location url="sip:2003@172.16.1.31 <sip:1003@172.16.1.31>"> > <proxy/> > </location> > </subaction> > <incoming> > <location url="sip:2002@172.16.1.28 <sip:1002@172.16.1.28>"> > <proxy timeout="8"> > <busy> > <sub ref="pda"/> > </busy> > <noanswer> > <sub ref="pda"/> > </noanswer> > </proxy> > </location> > </incoming> > </cpl>

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Re: [Users] does openser support QoS ?
by NEILL WILKINSON 04 Apr '07

04 Apr '07

The more obvious question is: How does it know there is no more bandwidth available? As a signalling proxy, surely it has no visibility of the RTP usages or requirements. I guess if it counted all the complete INVITE-> OK dialogues and the SDP codec used it could estimate usage? But the RTP/media path could be different in each case of these so again the same question as above...... Neill...;o) ----- Original Message ---- From: Henning Westerholt <henning.westerholt(a)1und1.de> To: users(a)openser.org; yanlin <yanlin(a)fortinet.com> Sent: Tuesday, 3 April, 2007 10:41:31 AM Subject: Re: [Users] does openser support QoS ? On Tuesday 03 April 2007 11:15, yanlin wrote: > Hi, > > does openser support some kind of QoS ? > like, band width management, when there has no more usable bandwidth, then > new INVITE should be rejected. Hi Yanlin, if the bandwith is exceeded, no new INVITEs will get to the server and will timeout. :-) The "pike" module does some kind of DOS protection, but this is also no real QoS. There was a discussion two weeks ago about this topic on devel (the topic was "rand function in config file"), Klaus suggested some kind of bandwith management module there. But at the moment such a module don't exist in OpenSER. Cheers, Henning _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

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[Users] New features added to OpenSer - BLA support and XMPP presence gateway
by Anca-Maria Vamanu 04 Apr '07

04 Apr '07

Two new modules were just introduced to SVN -- both are related to presence services. 1) pua_bla - implements Bridged Line Appearances according to the specifications in draft-anil-sipping-bla-03.txt. BLA, aka Shared Call/Line Appearance, comes from old telephony, where many devices share same line. Translated to SIP, means that many SIP devices share same AoR, but only one call can be established at a time, either addressed or initiated from the line. The call can be put on hold, and the others devices are notified so that they can pick up the call (the typical example is that with the secretary answering the call and putting it on hold to be picked up by the boss). The SIP devices must have client-side support for BLA as well. In out tests we used POLYCOM SoundPoint IP 430 SIP phones. The behaviour for this specific device: when picking up the handset form one phone a red light appears at the other phones. If attempting to call from a second phone using the same line, you get the message "Resources full". If a call is established, when putting it on hold, the light starts to flicker at the other phones and it is possible to pick it up. 2) pua_xmpp - is a presence gateway between SIMPLE and XMPP. It comes to complete the IM gateway, by enabling presence state information exchange between SIP/SIMPLE and Jabber/XMPP clients. For testings we used Gaim 2.0.0beta5, Psi v0.10 for XMPP and Xlite v3.0 for SIP. Some sample config file to implement above features will be published soon on dokuwiki.

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RE: [Users] Problem with dialog module
by Leonid Fainshtein 04 Apr '07

04 Apr '07

Hi Bogdan, Please check the ngrep output. I put my comments along with the trace: 1. ****** The 203(a)172.17.4.22 calls 3525(a)172.17.4.22 U 2007/03/18 12:54:09.758887 172.17.2.11:5060 -> 172.17.4.22:5060 INVITE sip:3525@172.17.4.22:5060 SIP/2.0..Via: SIP/2.0/UDP 172.17.2.11:5060 ;branch=z9hG4bK3796032784-846..Max-Forwards: 70..Allow: INVITE,OPTIONS,CANC EL,ACK,BYE,PRACK,INFO,REFER,NOTIFY..Supported: replaces..User-Agent: Claro BS3202 A454..From: <sip:203@172.17.4.22>;tag=hssUA_3795992784-846..To: <sip :3525@172.17.4.22:5060>..Call-ID: 3795992784-846..CSeq: 2 INVITE..Contact: <sip:203@172.17.2.11:5060>..Content-Type: application/sdp..Proxy-Authorizat ion: Digest username="613",realm="172.17.4.22",nonce="45fd6fdd9e385352f9c63 342cc574c454939f3ca",uri="sip:3525@172.17.4.22:5060",response="7156ea61b 864 a4ab937d17f42c7cd45a",algorithm=md5..Content-Length: 237....v=0..o=Claro 0 1 IN IP4 172.17.2.11..s=Claro..c=IN IP4 172.17.2.11..t=0 0..m=audio 30006 R TP/AVP 0 101 120..a=rtpmap:0 PCMU/8000..a=rtpmap:101 TELEPHONE-EVENT/8000.. a=fmtp:101 0-16..a=rtpmap:120 BSRTCP/8000..a=fmtp:120 ver2..a=sendrecv.. U 2007/03/18 12:54:09.766562 172.17.4.22:5060 -> 172.17.2.11:5060 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP 172.17 .2.11:5060;branch=z9hG4bK3796032784-846..From: <sip:203@172.17.4.22>;tag=hs sUA_3795992784-846..To: <sip:3525@172.17.4.22:5060>..Call-ID: 3795992784-84 6..CSeq: 2 INVITE..Server: ClaroSIS (Openser 1.1.1 (i386/linux))..Content-L ength: 0..Warning: 392 172.17.4.22:5060 "Noisy feedback tells: pid=4003 re q_src_ip=172.17.2.11 req_src_port=5060 in_uri=sip:3525@172.17.4.22:5060 out _uri=sip:3525@172.17.2.14;user=phone via_cnt==1".... U 2007/03/18 12:54:09.766871 172.17.4.22:5060 -> 172.17.2.14:5060 INVITE sip:3525@172.17.2.14;user=phone SIP/2.0..Record-Route: <sip:172.17.4 .22;lr=on;ftag=hssUA_3795992784-846;did=55f.4491d5b2>..Via: SIP/2.0/UDP 172 .17.4.22;branch=z9hG4bKbd8b.80bd0bc3.0..Via: SIP/2.0/UDP 172.17.2.11:5060;b ranch=z9hG4bK3796032784-846..Max-Forwards: 69..Allow: INVITE,OPTIONS,CANCEL ,ACK,BYE,PRACK,INFO,REFER,NOTIFY..Supported: replaces..User-Agent: Claro BS 3202 A454..From: <sip:203@172.17.4.22>;tag=hssUA_3795992784-846..To: <sip:3 525@172.17.4.22:5060>..Call-ID: 3795992784-846..CSeq: 2 INVITE..Contact: <s ip:203@172.17.2.11:5060>..Content-Type: application/sdp..Content-Length: 23 7....v=0..o=Claro 0 1 IN IP4 172.17.2.11..s=Claro..c=IN IP4 172.17.2.11..t= 0 0..m=audio 30006 RTP/AVP 0 101 120..a=rtpmap:0 PCMU/8000..a=rtpmap:101 TE LEPHONE-EVENT/8000..a=fmtp:101 0-16..a=rtpmap:120 BSRTCP/8000..a=fmtp:120 v er2..a=sendrecv.. U 2007/03/18 12:54:09.772591 172.17.2.14:5060 -> 172.17.4.22:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 172.17.4.22;branch=z9hG4bKbd8b.80bd0bc 3.0..Via: SIP/2.0/UDP 172.17.2.11:5060;branch=z9hG4bK3796032784-846..From: <sip:203@172.17.4.22>;tag=hssUA_3795992784-846..To: <sip:3525@172.17.4.22:5 060>..Call-ID: 3795992784-846..CSeq: 2 INVITE..User-Agent: Grandstream BT11 0 1.0.8.16..Content-Length: 0.... U 2007/03/18 12:54:09.775043 172.17.2.14:5060 -> 172.17.4.22:5060 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP 172.17.4.22;branch=z9hG4bKbd8b.80bd0b c3.0..Via: SIP/2.0/UDP 172.17.2.11:5060;branch=z9hG4bK3796032784-846..Recor d-Route: <sip:172.17.4.22;lr=on;ftag=hssUA_3795992784-846;did=55f.4491d5b2> ..From: <sip:203@172.17.4.22>;tag=hssUA_3795992784-846..To: <sip:3525@172.1 7.4.22:5060>;tag=61645575ae245b55..Call-ID: 3795992784-846..CSeq: 2 INVITE. .User-Agent: Grandstream BT110 1.0.8.16..Content-Length: 0.... U 2007/03/18 12:54:09.777250 172.17.4.22:5060 -> 172.17.2.11:5060 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP 172.17.2.11:5060;branch=z9hG4bK379603 2784-846..Record-Route: <sip:172.17.4.22;lr=on;ftag=hssUA_3795992784-846;di d=55f.4491d5b2>..From: <sip:203@172.17.4.22>;tag=hssUA_3795992784-846..To: <sip:3525@172.17.4.22:5060>;tag=61645575ae245b55..Call-ID: 3795992784-846.. CSeq: 2 INVITE..User-Agent: Grandstream BT110 1.0.8.16..Content-Length: 0.. .. U 2007/03/18 12:54:13.673700 172.17.2.14:5060 -> 172.17.4.22:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 172.17.4.22;branch=z9hG4bKbd8b.80bd0bc3.0. .Via: SIP/2.0/UDP 172.17.2.11:5060;branch=z9hG4bK3796032784-846..Record-Rou te: <sip:172.17.4.22;lr=on;ftag=hssUA_3795992784-846;did=55f.4491d5b2>..Fro m: <sip:203@172.17.4.22>;tag=hssUA_3795992784-846..To: <sip:3525@172.17.4.2 2:5060>;tag=61645575ae245b55..Call-ID: 3795992784-846..CSeq: 2 INVITE..User -Agent: Grandstream BT110 1.0.8.16..Contact: <sip:3525@172.17.2.14;user=pho ne>..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Cont ent-Type: application/sdp..Supported: replaces..Content-Length: 153....v=0. .o=3525 8000 8000 IN IP4 172.17.2.14..s=SIP Call..c=IN IP4 172.17.2.14..t=0 0..m=audio 5004 RTP/AVP 0..a=sendrecv..a=rtpmap:0 PCMU/8000..a=ptime:20.. U 2007/03/18 12:54:13.676298 172.17.4.22:5060 -> 172.17.2.11:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 172.17.2.11:5060;branch=z9hG4bK3796032784- 846..Record-Route: <sip:172.17.4.22;lr=on;ftag=hssUA_3795992784-846;did=55f .4491d5b2>..From: <sip:203@172.17.4.22>;tag=hssUA_3795992784-846..To: <sip: 3525@172.17.4.22:5060>;tag=61645575ae245b55..Call-ID: 3795992784-846..CSeq: 2 INVITE..User-Agent: Grandstream BT110 1.0.8.16..Contact: <sip:3525@172.1 7.2.14:5060;user=phone>..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS, INFO,SUBSCRIBE..Content-Type: application/sdp..Supported: replaces..Content -Length: 153....v=0..o=3525 8000 8000 IN IP4 172.17.2.14..s=SIP Call..c=IN IP4 172.17.2.14..t=0 0..m=audio 5004 RTP/AVP 0..a=sendrecv..a=rtpmap:0 PCMU /8000..a=ptime:20.. U 2007/03/18 12:54:13.721732 172.17.2.11:5060 -> 172.17.4.22:5060 ACK sip:3525@172.17.2.14:5060;user=phone SIP/2.0..Via: SIP/2.0/UDP 172.17.2 .11:5060;branch=z9hG4bK3799992784-846..Route: <sip:172.17.4.22;lr=on;ftag=h ssUA_3795992784-846;did=55f.4491d5b2>..Max-Forwards: 70..Proxy-Authorizatio n: Digest username="613",realm="172.17.4.22",nonce="45fd6fdd9e385352f9c6334 2cc574c454939f3ca",uri="sip:3525@172.17.4.22:5060",response="7156ea61b86 4a4 ab937d17f42c7cd45a",algorithm=md5..User-Agent: Claro BS3202 A454..From: <si p:203@172.17.4.22>;tag=hssUA_3795992784-846..To: <sip:3525@172.17.4.22:5060 >;tag=61645575ae245b55..Call-ID: 3795992784-846..CSeq: 2 ACK..Contact: <sip :203@172.17.2.11:5060>..Content-Length: 0.... U 2007/03/18 12:54:13.724056 172.17.4.22:5060 -> 172.17.2.14:5060 ACK sip:3525@172.17.2.14:5060;user=phone SIP/2.0..Record-Route: <sip:172.17 .4.22;lr=on;ftag=hssUA_3795992784-846>..Via: SIP/2.0/UDP 172.17.4.22;branch =z9hG4bKbd8b.80bd0bc3.2..Via: SIP/2.0/UDP 172.17.2.11:5060;branch=z9hG4bK37 99992784-846..Max-Forwards: 69..Proxy-Authorization: Digest username="613", realm="172.17.4.22",nonce="45fd6fdd9e385352f9c63342cc574c454939f3ca",uri ="s ip:3525@172.17.4.22:5060",response="7156ea61b864a4ab937d17f42c7cd45a",al gor ithm=md5..User-Agent: Claro BS3202 A454..From: <sip:203@172.17.4.22>;tag=hs sUA_3795992784-846..To: <sip:3525@172.17.4.22:5060>;tag=61645575ae245b55..C all-ID: 3795992784-846..CSeq: 2 ACK..Contact: <sip:203@172.17.2.11:5060>..C ontent-Length: 0.... 2. ****** The call is established. The did=55f.4491d5b2. Now 3525(a)172.17.4.22 initiates call transfer. It sends re-INVITE in order to mute the call partner. The header filed Route contains did=55f.4491d5b2. U 2007/03/18 12:54:16.394546 172.17.2.14:5060 -> 172.17.4.22:5060 INVITE sip:203@172.17.2.11:5060 SIP/2.0..Via: SIP/2.0/UDP 172.17.2.14;branc h=z9hG4bKb9469def476f37f2..Route: <sip:172.17.4.22;lr=on;ftag=hssUA_3795992 784-846;did=55f.4491d5b2>..From: <sip:3525@172.17.4.22:5060>;tag=61645575ae 245b55..To: <sip:203@172.17.4.22>;tag=hssUA_3795992784-846..Contact: <sip:3 525(a)172.17.2.14;user=phone>..Supported: replaces..Call-ID: 3795992784-846.. CSeq: 321 INVITE..User-Agent: Grandstream BT110 1.0.8.16..Max-Forwards: 70. .Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content- Type: application/sdp..Content-Length: 320....v=0..o=3525 8000 8001 IN IP4 172.17.2.14..s=SIP Call..c=IN IP4 0.0.0.0..t=0 0..m=audio 5004 RTP/AVP 0 8 18 4 99 2 9..a=sendonly..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpm ap:18 G729/8000..a=rtpmap:4 G723/8000..a=rtpmap:99 iLBC/8000..a=fmtp:99 mod e=20..a=rtpmap:2 G726-32/8000..a=rtpmap:9 G722/16000..a=ptime:20.. U 2007/03/18 12:54:16.396983 172.17.4.22:5060 -> 172.17.2.14:5060 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP 172.17 .2.14;branch=z9hG4bKb9469def476f37f2..From: <sip:3525@172.17.4.22:5060>;tag =61645575ae245b55..To: <sip:203@172.17.4.22>;tag=hssUA_3795992784-846..Call -ID: 3795992784-846..CSeq: 321 INVITE..Server: ClaroSIS (Openser 1.1.1 (i38 6/linux))..Content-Length: 0..Warning: 392 172.17.4.22:5060 "Noisy feedback tells: pid=3997 req_src_ip=172.17.2.14 req_src_port=5060 in_uri=sip:203@1 72.17.2.11:5060 out_uri=sip:203@172.17.2.11:5060 via_cnt==1".... 3. ****** OpenSer sends the re-INVITE to the transferee. Please note, that "did" doesn't appear in the Record-Route. As result of that, BYE for this call will no have "did" parameter and the dialog will appear as not closed. U 2007/03/18 12:54:16.397307 172.17.4.22:5060 -> 172.17.2.11:5060 INVITE sip:203@172.17.2.11:5060 SIP/2.0..Record-Route: <sip:172.17.4.22;lr= on;ftag=61645575ae245b55>..Via: SIP/2.0/UDP 172.17.4.22;branch=z9hG4bK4751. 82bc6f96.0..Via: SIP/2.0/UDP 172.17.2.14;branch=z9hG4bKb9469def476f37f2..Fr om: <sip:3525@172.17.4.22:5060>;tag=61645575ae245b55..To: <sip:203@172.17.4 .22>;tag=hssUA_3795992784-846..Contact: <sip:3525@172.17.2.14;user=phone>.. Supported: replaces..Call-ID: 3795992784-846..CSeq: 321 INVITE..User-Agent: Grandstream BT110 1.0.8.16..Max-Forwards: 69..Allow: INVITE,ACK,CANCEL,BYE ,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Type: application/sdp..Conten t-Length: 320....v=0..o=3525 8000 8001 IN IP4 172.17.2.14..s=SIP Call..c=IN IP4 0.0.0.0..t=0 0..m=audio 5004 RTP/AVP 0 8 18 4 99 2 9..a=sendonly..a=rt pmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729/8000..a=rtpmap:4 G 723/8000..a=rtpmap:99 iLBC/8000..a=fmtp:99 mode=20..a=rtpmap:2 G726-32/8000 ..a=rtpmap:9 G722/16000..a=ptime:20.. U 2007/03/18 12:54:16.435210 172.17.2.11:5060 -> 172.17.4.22:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 172.17.4.22;branch=z9hG4bK4751.82bc6f96.0, SIP/2.0/UDP 172.17.2.14;branch=z9hG4bKb9469def476f37f2..Record-Route: <sip: 172.17.4.22;lr=on;ftag=61645575ae245b55>..Server: Claro BS3202 A454..From: <sip:3525@172.17.4.22:5060>;tag=61645575ae245b55..To: <sip:203@172.17.4.22> ;tag=hssUA_3795992784-846..Call-ID: 3795992784-846..CSeq: 321 INVITE..Conta ct: <sip:203@172.17.2.11:5060>..Allow: INVITE,OPTIONS,CANCEL,ACK,BYE,PRACK, INFO,REFER,NOTIFY..Content-Type: application/sdp..Content-Length: 167....v= 0..o=Claro 1174230681570 1174230681 IN IP4 172.17.2.11..s=Claro..c=IN IP4 1 72.17.2.11..t=0 0..m=audio 30006 RTP/AVP 0..a=rtpmap:0 PCMU/8000..a=ptime:2 0..a=recvonly.. U 2007/03/18 12:54:16.436772 172.17.4.22:5060 -> 172.17.2.14:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 172.17.2.14;branch=z9hG4bKb9469def476f37f2 ..Record-Route: <sip:172.17.4.22;lr=on;ftag=61645575ae245b55>..Server: Clar o BS3202 A454..From: <sip:3525@172.17.4.22:5060>;tag=61645575ae245b55..To: <sip:203@172.17.4.22>;tag=hssUA_3795992784-846..Call-ID: 3795992784-846..CS eq: 321 INVITE..Contact: <sip:203@172.17.2.11:5060>..Allow: INVITE,OPTIONS, CANCEL,ACK,BYE,PRACK,INFO,REFER,NOTIFY..Content-Type: application/sdp..Cont ent-Length: 167....v=0..o=Claro 1174230681570 1174230681 IN IP4 172.17.2.11 ..s=Claro..c=IN IP4 172.17.2.11..t=0 0..m=audio 30006 RTP/AVP 0..a=rtpmap:0 PCMU/8000..a=ptime:20..a=recvonly.. U 2007/03/18 12:54:16.443328 172.17.2.14:5060 -> 172.17.4.22:5060 ACK sip:203@172.17.2.11:5060 SIP/2.0..Via: SIP/2.0/UDP 172.17.2.14;branch=z 9hG4bK26b04409b72e06f9..Route: <sip:172.17.4.22;lr=on;ftag=61645575ae245b55 >..From: <sip:3525@172.17.4.22:5060>;tag=61645575ae245b55..To: <sip:203@172 .17.4.22>;tag=hssUA_3795992784-846..Contact: <sip:3525@172.17.2.14;user=pho ne>..Call-ID: 3795992784-846..CSeq: 321 ACK..User-Agent: Grandstream BT110 1.0.8.16..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIO NS,INFO,SUBSCRIBE..Content-Length: 0.... U 2007/03/18 12:54:16.445463 172.17.4.22:5060 -> 172.17.2.11:5060 ACK sip:203@172.17.2.11:5060 SIP/2.0..Record-Route: <sip:172.17.4.22;lr=on; ftag=61645575ae245b55>..Via: SIP/2.0/UDP 172.17.4.22;branch=z9hG4bK4751.82b c6f96.2..Via: SIP/2.0/UDP 172.17.2.14;branch=z9hG4bK26b04409b72e06f9..From: <sip:3525@172.17.4.22:5060>;tag=61645575ae245b55..To: <sip:203@172.17.4.22 >;tag=hssUA_3795992784-846..Contact: <sip:3525@172.17.2.14;user=phone>..Cal l-ID: 3795992784-846..CSeq: 321 ACK..User-Agent: Grandstream BT110 1.0.8.16 ..Max-Forwards: 69..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO, SUBSCRIBE..Content-Length: 0.... U 2007/03/18 12:54:20.272078 172.17.2.14:5060 -> 172.17.4.22:5060 REFER sip:203@172.17.2.11:5060 SIP/2.0..Via: SIP/2.0/UDP 172.17.2.14;branch =z9hG4bK9e02f3ad995f8245..Route: <sip:172.17.4.22;lr=on;ftag=61645575ae245b 55>..From: <sip:3525@172.17.4.22:5060>;tag=61645575ae245b55..To: <sip:203@1 72.17.4.22>;tag=hssUA_3795992784-846..Contact: <sip:3525@172.17.2.14;user=p hone>..Supported: replaces..Refer-To: <sip:5524@172.17.4.22;user=phone>..Re ferred-By: <sip:3525@172.17.4.22;user=phone>..Call-ID: 3795992784-846..CSeq : 322 REFER..User-Agent: Grandstream BT110 1.0.8.16..Max-Forwards: 70..Allo w: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Lengt h: 0.... U 2007/03/18 12:54:20.275997 172.17.4.22:5060 -> 172.17.2.11:5060 REFER sip:203@172.17.2.11:5060 SIP/2.0..Record-Route: <sip:172.17.4.22;lr=o n;ftag=61645575ae245b55>..Via: SIP/2.0/UDP 172.17.4.22;branch=z9hG4bK1751.e 223fee7.0..Via: SIP/2.0/UDP 172.17.2.14;branch=z9hG4bK9e02f3ad995f8245..Fro m: <sip:3525@172.17.4.22:5060>;tag=61645575ae245b55..To: <sip:203@172.17.4. 22>;tag=hssUA_3795992784-846..Contact: <sip:3525@172.17.2.14;user=phone>..S upported: replaces..Refer-To: <sip:5524@172.17.4.22;user=phone>..Referred-B y: <sip:3525@172.17.4.22;user=phone>..Call-ID: 3795992784-846..CSeq: 322 RE FER..User-Agent: Grandstream BT110 1.0.8.16..Max-Forwards: 69..Allow: INVIT E,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Length: 0.... U 2007/03/18 12:54:20.323290 172.17.2.11:5060 -> 172.17.4.22:5060 INVITE sip:5524@172.17.4.22;user=phone SIP/2.0..Via: SIP/2.0/UDP 172.17.2.1 1:5060;branch=z9hG4bK3806597784-846..Max-Forwards: 70..Allow: INVITE,OPTION S,CANCEL,ACK,BYE,PRACK,INFO,REFER,NOTIFY..Supported: replaces..From: <sip:2 03(a)172.17.4.22>;tag=hssUA_3806597784-846..To: <sip:5524@172.17.4.22;user=ph one>..Call-ID: 3806597784-846..CSeq: 1 INVITE..Contact: <sip:203@172.17.2.1 1:5060>..Referred-By: <sip:3525@172.17.4.22;user=phone>..Content-Type: appl ication/sdp..Content-Length: 237....v=0..o=Claro 0 1 IN IP4 172.17.2.11..s= Claro..c=IN IP4 172.17.2.11..t=0 0..m=audio 30006 RTP/AVP 0 101 120..a=rtpm ap:0 PCMU/8000..a=rtpmap:101 TELEPHONE-EVENT/8000..a=fmtp:101 0-16..a=rtpma p:120 BSRTCP/8000..a=fmtp:120 ver2..a=sendrecv.. U 2007/03/18 12:54:20.324935 172.17.4.22:5060 -> 172.17.2.11:5060 SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP 172.17.2.11:506 0;branch=z9hG4bK3806597784-846..From: <sip:203@172.17.4.22>;tag=hssUA_38065 97784-846..To: <sip:5524@172.17.4.22;user=phone>;tag=329cfeaa6ded039da25ff8 cbb8668bd2.dc60..Call-ID: 3806597784-846..CSeq: 1 INVITE..Proxy-Authenticat e: Digest realm="172.17.4.22", nonce="45fd6fe80721e4ce9c9b2eef4c93a66ef88b9 f2c"..Server: ClaroSIS (Openser 1.1.1 (i386/linux))..Content-Length: 0..War ning: 392 172.17.4.22:5060 "Noisy feedback tells: pid=4001 req_src_ip=172. 17.2.11 req_src_port=5060 in_uri=sip:5524@172.17.4.22;user=phone out_uri=si p:5524@172.17.4.22;user=phone via_cnt==1".... U 2007/03/18 12:54:20.328647 172.17.2.11:5060 -> 172.17.4.22:5060 SIP/2.0 202 Accepted..Via: SIP/2.0/UDP 172.17.4.22;branch=z9hG4bK1751.e223f ee7.0,SIP/2.0/UDP 172.17.2.14;branch=z9hG4bK9e02f3ad995f8245..Record-Route: <sip:172.17.4.22;lr=on;ftag=61645575ae245b55>..From: <sip:3525@172.17.4.22 :5060>;tag=61645575ae245b55..To: <sip:203@172.17.4.22>;tag=hssUA_3795992784 -846..Call-ID: 3795992784-846..CSeq: 322 REFER..Contact: <sip:203@172.17.2. 11:5060>..Content-Length: 0.... U 2007/03/18 12:54:20.330099 172.17.4.22:5060 -> 172.17.2.14:5060 SIP/2.0 202 Accepted..Via: SIP/2.0/UDP 172.17.2.14;branch=z9hG4bK9e02f3ad99 5f8245..Record-Route: <sip:172.17.4.22;lr=on;ftag=61645575ae245b55>..From: <sip:3525@172.17.4.22:5060>;tag=61645575ae245b55..To: <sip:203@172.17.4.22> ;tag=hssUA_3795992784-846..Call-ID: 3795992784-846..CSeq: 322 REFER..Contac t: <sip:203@172.17.2.11:5060>..Content-Length: 0.... U 2007/03/18 12:54:20.351529 172.17.2.11:5060 -> 172.17.4.22:5060 ACK sip:5524@172.17.4.22;user=phone SIP/2.0..Via: SIP/2.0/UDP 172.17.2.11:5 060;branch=z9hG4bK3806597784-846..Max-Forwards: 70..User-Agent: Claro BS320 2 A454..From: <sip:203@172.17.4.22>;tag=hssUA_3806597784-846..To: <sip:5524 @172.17.4.22;user=phone>;tag=329cfeaa6ded039da25ff8cbb8668bd2.dc60..Call -ID : 3806597784-846..CSeq: 1 ACK..Contact: <sip:203@172.17.2.11:5060>..Content -Length: 0.... U 2007/03/18 12:54:20.364969 172.17.2.11:5060 -> 172.17.4.22:5060 INVITE sip:5524@172.17.4.22;user=phone SIP/2.0..Via: SIP/2.0/UDP 172.17.2.1 1:5060;branch=z9hG4bK3806642784-846..Max-Forwards: 70..Allow: INVITE,OPTION S,CANCEL,ACK,BYE,PRACK,INFO,REFER,NOTIFY..Supported: replaces..User-Agent: Claro BS3202 A454..From: <sip:203@172.17.4.22>;tag=hssUA_3806597784-846..To : <sip:5524@172.17.4.22;user=phone>..Call-ID: 3806597784-846..CSeq: 2 INVIT E..Contact: <sip:203@172.17.2.11:5060>..Content-Type: application/sdp..Prox y-Authorization: Digest username="613",realm="172.17.4.22",nonce="45fd6fe80 721e4ce9c9b2eef4c93a66ef88b9f2c",uri="sip:5524@172.17.4.22",response="3f b20 f7d6e67668e07b862b99412162d",algorithm=md5..Content-Length: 237....v=0..o=C laro 0 1 IN IP4 172.17.2.11..s=Claro..c=IN IP4 172.17.2.11..t=0 0..m=audio 30006 RTP/AVP 0 101 120..a=rtpmap:0 PCMU/8000..a=rtpmap:101 TELEPHONE-EVENT /8000..a=fmtp:101 0-16..a=rtpmap:120 BSRTCP/8000..a=fmtp:120 ver2..a=sendre cv.. U 2007/03/18 12:54:20.372610 172.17.4.22:5060 -> 172.17.2.11:5060 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP 172.17 .2.11:5060;branch=z9hG4bK3806642784-846..From: <sip:203@172.17.4.22>;tag=hs sUA_3806597784-846..To: <sip:5524@172.17.4.22;user=phone>..Call-ID: 3806597 784-846..CSeq: 2 INVITE..Server: ClaroSIS (Openser 1.1.1 (i386/linux))..Con tent-Length: 0..Warning: 392 172.17.4.22:5060 "Noisy feedback tells: pid=4 003 req_src_ip=172.17.2.11 req_src_port=5060 in_uri=sip:5524@172.17.4.22;us er=phone out_uri=sip:5524@172.17.2.168:5061 via_cnt==1".... U 2007/03/18 12:54:20.372833 172.17.4.22:5060 -> 172.17.2.168:5061 INVITE sip:5524@172.17.2.168:5061 SIP/2.0..Record-Route: <sip:172.17.4.22;l r=on;ftag=hssUA_3806597784-846;did=0e3.7c5bd92>..Via: SIP/2.0/UDP 172.17.4. 22;branch=z9hG4bK95c6.337d9f56.0..Via: SIP/2.0/UDP 172.17.2.11:5060;branch= z9hG4bK3806642784-846..Max-Forwards: 69..Allow: INVITE,OPTIONS,CANCEL,ACK,B YE,PRACK,INFO,REFER,NOTIFY..Supported: replaces..User-Agent: Claro BS3202 A 454..From: <sip:203@172.17.4.22>;tag=hssUA_3806597784-846..To: <sip:5524@17 2.17.4.22;user=phone>..Call-ID: 3806597784-846..CSeq: 2 INVITE..Contact: <s ip:203@172.17.2.11:5060>..Content-Type: application/sdp..Content-Length: 23 7....v=0..o=Claro 0 1 IN IP4 172.17.2.11..s=Claro..c=IN IP4 172.17.2.11..t= 0 0..m=audio 30006 RTP/AVP 0 101 120..a=rtpmap:0 PCMU/8000..a=rtpmap:101 TE LEPHONE-EVENT/8000..a=fmtp:101 0-16..a=rtpmap:120 BSRTCP/8000..a=fmtp:120 v er2..a=sendrecv.. U 2007/03/18 12:54:20.385872 172.17.2.168:5061 -> 172.17.4.22:5060 SIP/2.0 100 Trying..To: <sip:5524@172.17.4.22;user=phone>..From: <sip:203@1 72.17.4.22>;tag=hssUA_3806597784-846..Call-ID: 3806597784-846..CSeq: 2 INVI TE..Via: SIP/2.0/UDP 172.17.4.22;branch=z9hG4bK95c6.337d9f56.0..Via: SIP/2. 0/UDP 172.17.2.11:5060;branch=z9hG4bK3806642784-846..Record-Route: <sip:172 .17.4.22;lr=on;ftag=hssUA_3806597784-846;did=0e3.7c5bd92>..Server: Linksys/ PAP2-3.1.3(LS)..Content-Length: 0.... U 2007/03/18 12:54:20.393629 172.17.2.11:5060 -> 172.17.4.22:5060 NOTIFY sip:3525@172.17.2.14;user=phone SIP/2.0..Via: SIP/2.0/UDP 172.17.2.1 1:5060;branch=z9hG4bK3806672784-846..Route: <sip:172.17.4.22;lr=on;ftag=hss UA_3795992784-846;did=55f.4491d5b2>..Max-Forwards: 70..Proxy-Authorization: Digest username="613",realm="172.17.4.22",nonce="45fd6fdd9e385352f9c63342c c574c454939f3ca",uri="sip:3525@172.17.4.22:5060",response="7156ea61b864a 4ab 937d17f42c7cd45a",algorithm=md5..Allow: INVITE,OPTIONS,CANCEL,ACK,BYE,PRACK ,INFO,REFER,NOTIFY..Supported: replaces..From: <sip:203@172.17.4.22>;tag=hs sUA_3795992784-846..To: <sip:3525@172.17.4.22:5060>;tag=61645575ae245b55..C all-ID: 3795992784-846..CSeq: 3 NOTIFY..Contact: <sip:203@172.17.2.11:5060> ..Event: refer..Subscription-State: active;expires=60..Content-Type: messag e/sipfrag..Content-Length: 530....SIP/2.0 100 trying -- your call is import ant to us..Via: SIP/2.0/UDP 172.17.2.11:5060;received=172.17.4.22;branch=z9 hG4bK3806642784-846..From: <sip:203@172.17.4.22>;tag=hssUA_3806597784-846.. To: <sip:5524@172.17.4.22;user=phone>..Call-ID: 3806597784-846..CSeq: 2 INV ITE..Server: ClaroSIS (Openser 1.1.1 (i386/linux))..Content-Length: 0..Warn ing: 392 172.17.4.22:5060 "Noisy feedback tells: pid=4003 req_src_ip=172.1 7.2.11 req_src_port=5060 in_uri=sip:5524@172.17.4.22;user=phone out_uri=sip :5524@172.17.2.168:5061 via_cnt==1".... U 2007/03/18 12:54:20.396980 172.17.4.22:5060 -> 172.17.2.14:5060 NOTIFY sip:3525@172.17.2.14;user=phone SIP/2.0..Record-Route: <sip:172.17.4 .22;lr=on;ftag=hssUA_3795992784-846>..Via: SIP/2.0/UDP 172.17.4.22;branch=z 9hG4bKcd8b.81aaa3f3.0..Via: SIP/2.0/UDP 172.17.2.11:5060;branch=z9hG4bK3806 672784-846..Max-Forwards: 69..Proxy-Authorization: Digest username="613",re alm="172.17.4.22",nonce="45fd6fdd9e385352f9c63342cc574c454939f3ca",uri=" sip :3525@172.17.4.22:5060",response="7156ea61b864a4ab937d17f42c7cd45a",algo rit hm=md5..Allow: INVITE,OPTIONS,CANCEL,ACK,BYE,PRACK,INFO,REFER,NOTIFY..Suppo rted: replaces..From: <sip:203@172.17.4.22>;tag=hssUA_3795992784-846..To: < sip:3525@172.17.4.22:5060>;tag=61645575ae245b55..Call-ID: 3795992784-846..C Seq: 3 NOTIFY..Contact: <sip:203@172.17.2.11:5060>..Event: refer..Subscript ion-State: active;expires=60..Content-Type: message/sipfrag..Content-Length : 530....SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/U DP 172.17.2.11:5060;received=172.17.4.22;branch=z9hG4bK3806642784-846..From : <sip:203@172.17.4.22>;tag=hssUA_3806597784-846..To: <sip:5524@172.17.4.22 ;user=phone>..Call-ID: 3806597784-846..CSeq: 2 INVITE..Server: ClaroSIS (Op enser 1.1.1 (i386/linux))..Content-Length: 0..Warning: 392 172.17.4.22:5060 "Noisy feedback tells: pid=4003 req_src_ip=172.17.2.11 req_src_port=5060 in_uri=sip:5524@172.17.4.22;user=phone out_uri=sip:5524@172.17.2.168:5061 v ia_cnt==1".... U 2007/03/18 12:54:20.401630 172.17.2.168:5061 -> 172.17.4.22:5060 SIP/2.0 180 Ringing..To: <sip:5524@172.17.4.22;user=phone>;tag=8cb5cce9fbff 0a22i1..From: <sip:203@172.17.4.22>;tag=hssUA_3806597784-846..Call-ID: 3806 597784-846..CSeq: 2 INVITE..Via: SIP/2.0/UDP 172.17.4.22;branch=z9hG4bK95c6 .337d9f56.0..Via: SIP/2.0/UDP 172.17.2.11:5060;branch=z9hG4bK3806642784-846 ..Record-Route: <sip:172.17.4.22;lr=on;ftag=hssUA_3806597784-846;did=0e3.7c 5bd92>..Server: Linksys/PAP2-3.1.3(LS)..Content-Length: 0.... U 2007/03/18 12:54:20.403183 172.17.4.22:5060 -> 172.17.2.11:5060 SIP/2.0 180 Ringing..To: <sip:5524@172.17.4.22;user=phone>;tag=8cb5cce9fbff 0a22i1..From: <sip:203@172.17.4.22>;tag=hssUA_3806597784-846..Call-ID: 3806 597784-846..CSeq: 2 INVITE..Via: SIP/2.0/UDP 172.17.2.11:5060;branch=z9hG4b K3806642784-846..Record-Route: <sip:172.17.4.22;lr=on;ftag=hssUA_3806597784 -846;did=0e3.7c5bd92>..Server: Linksys/PAP2-3.1.3(LS)..Content-Length: 0... . U 2007/03/18 12:54:20.403878 172.17.2.14:5060 -> 172.17.4.22:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 172.17.4.22;branch=z9hG4bKcd8b.81aaa3f3.0. .Via: SIP/2.0/UDP 172.17.2.11:5060;branch=z9hG4bK3806672784-846..Record-Rou te: <sip:172.17.4.22;lr=on;ftag=hssUA_3795992784-846>..From: <sip:203@172.1 7.4.22>;tag=hssUA_3795992784-846..To: <sip:3525@172.17.4.22:5060>;tag=61645 575ae245b55..Call-ID: 3795992784-846..CSeq: 3 NOTIFY..User-Agent: Grandstre am BT110 1.0.8.16..Contact: <sip:3525@172.17.2.14;user=phone>..Allow: INVIT E,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Supported: replaces.. Content-Length: 0.... U 2007/03/18 12:54:20.407617 172.17.4.22:5060 -> 172.17.2.11:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 172.17.2.11:5060;branch=z9hG4bK3806672784- 846..Record-Route: <sip:172.17.4.22;lr=on;ftag=hssUA_3795992784-846>..From: <sip:203@172.17.4.22>;tag=hssUA_3795992784-846..To: <sip:3525@172.17.4.22: 5060>;tag=61645575ae245b55..Call-ID: 3795992784-846..CSeq: 3 NOTIFY..User-A gent: Grandstream BT110 1.0.8.16..Contact: <sip:3525@172.17.2.14:5060;user= phone>..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..S upported: replaces..Content-Length: 0.... U 2007/03/18 12:54:20.429109 172.17.2.11:5060 -> 172.17.4.22:5060 NOTIFY sip:3525@172.17.2.14;user=phone SIP/2.0..Via: SIP/2.0/UDP 172.17.2.1 1:5060;branch=z9hG4bK3806702784-846..Route: <sip:172.17.4.22;lr=on;ftag=hss UA_3795992784-846;did=55f.4491d5b2>..Max-Forwards: 70..Proxy-Authorization: Digest username="613",realm="172.17.4.22",nonce="45fd6fdd9e385352f9c63342c c574c454939f3ca",uri="sip:3525@172.17.4.22:5060",response="7156ea61b864a 4ab 937d17f42c7cd45a",algorithm=md5..Allow: INVITE,OPTIONS,CANCEL,ACK,BYE,PRACK ,INFO,REFER,NOTIFY..Supported: replaces..From: <sip:203@172.17.4.22>;tag=hs sUA_3795992784-846..To: <sip:3525@172.17.4.22:5060>;tag=61645575ae245b55..C all-ID: 3795992784-846..CSeq: 4 NOTIFY..Contact: <sip:203@172.17.2.11:5060> ..Event: refer..Subscription-State: active;expires=60..Content-Type: messag e/sipfrag..Content-Length: 397....SIP/2.0 180 Ringing..To: <sip:5524@172.17 .4.22;user=phone>;tag=8cb5cce9fbff0a22i1..From: <sip:203@172.17.4.22>;tag=h ssUA_3806597784-846..Call-ID: 3806597784-846..CSeq: 2 INVITE..Via: SIP/2.0/ UDP 172.17.2.11:5060;received=172.17.4.22;branch=z9hG4bK3806642784-846..Rec ord-Route: <sip:172.17.4.22;lr=on;ftag=hssUA_3806597784-846;did=0e3.7c5bd92 >..Server: Linksys/PAP2-3.1.3(LS)..Content-Length: 0.... U 2007/03/18 12:54:20.431776 172.17.4.22:5060 -> 172.17.2.14:5060 NOTIFY sip:3525@172.17.2.14;user=phone SIP/2.0..Record-Route: <sip:172.17.4 .22;lr=on;ftag=hssUA_3795992784-846>..Via: SIP/2.0/UDP 172.17.4.22;branch=z 9hG4bK9d8b.47c7ce13.0..Via: SIP/2.0/UDP 172.17.2.11:5060;branch=z9hG4bK3806 702784-846..Max-Forwards: 69..Proxy-Authorization: Digest username="613",re alm="172.17.4.22",nonce="45fd6fdd9e385352f9c63342cc574c454939f3ca",uri=" sip :3525@172.17.4.22:5060",response="7156ea61b864a4ab937d17f42c7cd45a",algo rit hm=md5..Allow: INVITE,OPTIONS,CANCEL,ACK,BYE,PRACK,INFO,REFER,NOTIFY..Suppo rted: replaces..From: <sip:203@172.17.4.22>;tag=hssUA_3795992784-846..To: < sip:3525@172.17.4.22:5060>;tag=61645575ae245b55..Call-ID: 3795992784-846..C Seq: 4 NOTIFY..Contact: <sip:203@172.17.2.11:5060>..Event: refer..Subscript ion-State: active;expires=60..Content-Type: message/sipfrag..Content-Length : 397....SIP/2.0 180 Ringing..To: <sip:5524@172.17.4.22;user=phone>;tag=8cb 5cce9fbff0a22i1..From: <sip:203@172.17.4.22>;tag=hssUA_3806597784-846..Call -ID: 3806597784-846..CSeq: 2 INVITE..Via: SIP/2.0/UDP 172.17.2.11:5060;rece ived=172.17.4.22;branch=z9hG4bK3806642784-846..Record-Route: <sip:172.17.4. 22;lr=on;ftag=hssUA_3806597784-846;did=0e3.7c5bd92>..Server: Linksys/PAP2-3 .1.3(LS)..Content-Length: 0.... U 2007/03/18 12:54:20.438291 172.17.2.14:5060 -> 172.17.4.22:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 172.17.4.22;branch=z9hG4bK9d8b.47c7ce13.0. .Via: SIP/2.0/UDP 172.17.2.11:5060;branch=z9hG4bK3806702784-846..Record-Rou te: <sip:172.17.4.22;lr=on;ftag=hssUA_3795992784-846>..From: <sip:203@172.1 7.4.22>;tag=hssUA_3795992784-846..To: <sip:3525@172.17.4.22:5060>;tag=61645 575ae245b55..Call-ID: 3795992784-846..CSeq: 4 NOTIFY..User-Agent: Grandstre am BT110 1.0.8.16..Contact: <sip:3525@172.17.2.14;user=phone>..Allow: INVIT E,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Supported: replaces.. Content-Length: 0.... U 2007/03/18 12:54:20.439954 172.17.4.22:5060 -> 172.17.2.11:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 172.17.2.11:5060;branch=z9hG4bK3806702784- 846..Record-Route: <sip:172.17.4.22;lr=on;ftag=hssUA_3795992784-846>..From: <sip:203@172.17.4.22>;tag=hssUA_3795992784-846..To: <sip:3525@172.17.4.22: 5060>;tag=61645575ae245b55..Call-ID: 3795992784-846..CSeq: 4 NOTIFY..User-A gent: Grandstream BT110 1.0.8.16..Contact: <sip:3525@172.17.2.14:5060;user= phone>..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..S upported: replaces..Content-Length: 0.... U 2007/03/18 12:54:23.312128 172.17.2.168:5061 -> 172.17.4.22:5060 SIP/2.0 200 OK..To: <sip:5524@172.17.4.22;user=phone>;tag=8cb5cce9fbff0a22i 1..From: <sip:203@172.17.4.22>;tag=hssUA_3806597784-846..Call-ID: 380659778 4-846..CSeq: 2 INVITE..Via: SIP/2.0/UDP 172.17.4.22;branch=z9hG4bK95c6.337d 9f56.0..Via: SIP/2.0/UDP 172.17.2.11:5060;branch=z9hG4bK3806642784-846..Rec ord-Route: <sip:172.17.4.22;lr=on;ftag=hssUA_3806597784-846;did=0e3.7c5bd92 >..Contact: 5524 <sip:5524@172.17.2.168:5061>..Server: Linksys/PAP2-3.1.3(L S)..Content-Length: 210..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPT IONS, REFER..Supported: x-sipura..Content-Type: application/sdp....v=0..o=- 23586680 23586680 IN IP4 172.17.2.168..s=-..c=IN IP4 172.17.2.168..t=0 0.. m=audio 16388 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-e vent/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 2007/03/18 12:54:23.313878 172.17.4.22:5060 -> 172.17.2.11:5060 SIP/2.0 200 OK..To: <sip:5524@172.17.4.22;user=phone>;tag=8cb5cce9fbff0a22i 1..From: <sip:203@172.17.4.22>;tag=hssUA_3806597784-846..Call-ID: 380659778 4-846..CSeq: 2 INVITE..Via: SIP/2.0/UDP 172.17.2.11:5060;branch=z9hG4bK3806 642784-846..Record-Route: <sip:172.17.4.22;lr=on;ftag=hssUA_3806597784-846; did=0e3.7c5bd92>..Contact: 5524 <sip:5524@172.17.2.168:5061>..Server: Links ys/PAP2-3.1.3(LS)..Content-Length: 210..Allow: ACK, BYE, CANCEL, INFO, INVI TE, NOTIFY, OPTIONS, REFER..Supported: x-sipura..Content-Type: application/ sdp....v=0..o=- 23586680 23586680 IN IP4 172.17.2.168..s=-..c=IN IP4 172.17 .2.168..t=0 0..m=audio 16388 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap: 101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 2007/03/18 12:54:23.355581 172.17.2.11:5060 -> 172.17.4.22:5060 27646@0:1480 ........NOTIFY sip:3525@172.17.2.14;user=phone SIP/2.0..Via: SIP/2.0/UDP 17 2.17.2.11:5060;branch=z9hG4bK3809632784-846..Route: <sip:172.17.4.22;lr=on; ftag=hssUA_3795992784-846;did=55f.4491d5b2>..Max-Forwards: 70..Proxy-Author ization: Digest username="613",realm="172.17.4.22",nonce="45fd6fdd9e385352f 9c63342cc574c454939f3ca",uri="sip:3525@172.17.4.22:5060",response="7156e a61 b864a4ab937d17f42c7cd45a",algorithm=md5..Allow: INVITE,OPTIONS,CANCEL,ACK,B YE,PRACK,INFO,REFER,NOTIFY..Supported: replaces..From: <sip:203@172.17.4.22 >;tag=hssUA_3795992784-846..To: <sip:3525@172.17.4.22:5060>;tag=61645575ae2 45b55..Call-ID: 3795992784-846..CSeq: 5 NOTIFY..Contact: <sip:203@172.17.2. 11:5060>..Event: refer..Subscription-State: terminated;reason=noresource..C ontent-Type: message/sipfrag..Content-Length: 756....SIP/2.0 200 OK..To: <s ip:5524@172.17.4.22;user=phone>;tag=8cb5cce9fbff0a22i1..From: <sip:203@172. 17.4.22>;tag=hssUA_3806597784-846..Call-ID: 3806597784-846..CSeq: 2 INVITE. .Via: SIP/2.0/UDP 172.17.2.11:5060;received=172.17.4.22;branch=z9hG4bK38066 42784-846..Record-Route: <sip:172.17.4.22;lr=on;ftag=hssUA_3806597784-846;d id=0e3.7c5bd92>..Contact: 5524 <sip:5524@172.17.2.168:5061>..Server: Linksy s/PAP2-3.1.3(LS)..Content-Length: 210..Allow: ACK,BYE,CANCEL,INFO,INVITE,NO TIFY,OPTIONS,REFER..Supported: x-sipura..Content-Type: application/sdp....v =0..o=- 23586680 23586680 IN IP4 172.17.2.168..s=-..c=IN IP4 172.17.2.168.. t=0 0..m=audio 16388 RTP/AVP 0 101..a=rtpmap:0 PCMU/800 U 2007/03/18 12:54:23.361462 172.17.4.22:5060 -> 172.17.2.14:5060 50085@0:1480 .....I..NOTIFY sip:3525@172.17.2.14;user=phone SIP/2.0..Record-Route: <sip: 172.17.4.22;lr=on;ftag=hssUA_3795992784-846>..Via: SIP/2.0/UDP 172.17.4.22; branch=z9hG4bKad8b.016fb321.0..Via: SIP/2.0/UDP 172.17.2.11:5060;branch=z9h G4bK3809632784-846..Max-Forwards: 69..Proxy-Authorization: Digest username= "613",realm="172.17.4.22",nonce="45fd6fdd9e385352f9c63342cc574c454939f3c a", uri="sip:3525@172.17.4.22:5060",response="7156ea61b864a4ab937d17f42c7cd4 5a" ,algorithm=md5..Allow: INVITE,OPTIONS,CANCEL,ACK,BYE,PRACK,INFO,REFER,NOTIF Y..Supported: replaces..From: <sip:203@172.17.4.22>;tag=hssUA_3795992784-84 6..To: <sip:3525@172.17.4.22:5060>;tag=61645575ae245b55..Call-ID: 379599278 4-846..CSeq: 5 NOTIFY..Contact: <sip:203@172.17.2.11:5060>..Event: refer..S ubscription-State: terminated;reason=noresource..Content-Type: message/sipf rag..Content-Length: 756....SIP/2.0 200 OK..To: <sip:5524@172.17.4.22;user= phone>;tag=8cb5cce9fbff0a22i1..From: <sip:203@172.17.4.22>;tag=hssUA_380659 7784-846..Call-ID: 3806597784-846..CSeq: 2 INVITE..Via: SIP/2.0/UDP 172.17. 2.11:5060;received=172.17.4.22;branch=z9hG4bK3806642784-846..Record-Rout e: <sip:172.17.4.22;lr=on;ftag=hssUA_3806597784-846;did=0e3.7c5bd92>..Conta ct: 5524 <sip:5524@172.17.2.168:5061>..Server: Linksys/PAP2-3.1.3(LS)..Content -Length: 210..Allow: ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,REFER..Suppo rted: x-sipura..Content-Type: application/sdp....v=0..o=- 23586680 23586680 IN IP4 172.17.2.168..s=-..c=IN IP4 172.17.2.168..t=0 0 U 2007/03/18 12:54:23.371944 172.17.2.11:5060 -> 172.17.4.22:5060 ACK sip:5524@172.17.2.168:5061 SIP/2.0..Via: SIP/2.0/UDP 172.17.2.11:5060;b ranch=z9hG4bK3809647784-846..Route: <sip:172.17.4.22;lr=on;ftag=hssUA_38065 97784-846;did=0e3.7c5bd92>..Max-Forwards: 70..Proxy-Authorization: Digest u sername="613",realm="172.17.4.22",nonce="45fd6fe80721e4ce9c9b2eef4c93a66 ef8 8b9f2c",uri="sip:5524@172.17.4.22",response="3fb20f7d6e67668e07b862b9941 216 2d",algorithm=md5..User-Agent: Claro BS3202 A454..From: <sip:203@172.17.4.2 2>;tag=hssUA_3806597784-846..To: <sip:5524@172.17.4.22;user=phone>;tag=8cb5 cce9fbff0a22i1..Call-ID: 3806597784-846..CSeq: 2 ACK..Contact: <sip:203@172 .17.2.11:5060>..Content-Length: 0.... U 2007/03/18 12:54:23.372334 172.17.2.14:5060 -> 172.17.4.22:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 172.17.4.22;branch=z9hG4bKad8b.016fb321.0. .Via: SIP/2.0/UDP 172.17.2.11:5060;branch=z9hG4bK3809632784-846..Record-Rou te: <sip:172.17.4.22;lr=on;ftag=hssUA_3795992784-846>..From: <sip:203@172.1 7.4.22>;tag=hssUA_3795992784-846..To: <sip:3525@172.17.4.22:5060>;tag=61645 575ae245b55..Call-ID: 3795992784-846..CSeq: 5 NOTIFY..User-Agent: Grandstre am BT110 1.0.8.16..Contact: <sip:3525@172.17.2.14;user=phone>..Allow: INVIT E,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Supported: replaces.. Content-Length: 0.... U 2007/03/18 12:54:23.373917 172.17.4.22:5060 -> 172.17.2.168:5061 ACK sip:5524@172.17.2.168:5061 SIP/2.0..Record-Route: <sip:172.17.4.22;lr=o n;ftag=hssUA_3806597784-846>..Via: SIP/2.0/UDP 172.17.4.22;branch=z9hG4bK95 c6.337d9f56.2..Via: SIP/2.0/UDP 172.17.2.11:5060;branch=z9hG4bK3809647784-8 46..Max-Forwards: 69..Proxy-Authorization: Digest username="613",realm="172 .17.4.22",nonce="45fd6fe80721e4ce9c9b2eef4c93a66ef88b9f2c",uri="sip:5524 @17 2.17.4.22",response="3fb20f7d6e67668e07b862b99412162d",algorithm=md5..Us er- Agent: Claro BS3202 A454..From: <sip:203@172.17.4.22>;tag=hssUA_3806597784- 846..To: <sip:5524@172.17.4.22;user=phone>;tag=8cb5cce9fbff0a22i1..Call-ID: 3806597784-846..CSeq: 2 ACK..Contact: <sip:203@172.17.2.11:5060>..Content- Length: 0.... U 2007/03/18 12:54:23.375179 172.17.4.22:5060 -> 172.17.2.11:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 172.17.2.11:5060;branch=z9hG4bK3809632784- 846..Record-Route: <sip:172.17.4.22;lr=on;ftag=hssUA_3795992784-846>..From: <sip:203@172.17.4.22>;tag=hssUA_3795992784-846..To: <sip:3525@172.17.4.22: 5060>;tag=61645575ae245b55..Call-ID: 3795992784-846..CSeq: 5 NOTIFY..User-A gent: Grandstream BT110 1.0.8.16..Contact: <sip:3525@172.17.2.14:5060;user= phone>..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..S upported: replaces..Content-Length: 0.... 4. ****** The message BYE doesn't have the did parameter. U 2007/03/18 12:54:23.490685 172.17.2.14:5060 -> 172.17.4.22:5060 BYE sip:203@172.17.2.11:5060 SIP/2.0..Via: SIP/2.0/UDP 172.17.2.14;branch=z 9hG4bKa419f2aaf11a7b40..Route: <sip:172.17.4.22;lr=on;ftag=61645575ae245b55 >..From: <sip:3525@172.17.4.22:5060>;tag=61645575ae245b55..To: <sip:203@172 .17.4.22>;tag=hssUA_3795992784-846..Supported: replaces..Call-ID: 379599278 4-846..CSeq: 323 BYE..User-Agent: Grandstream BT110 1.0.8.16..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Cont ent-Length: 0.... U 2007/03/18 12:54:23.494656 172.17.4.22:5060 -> 172.17.2.11:5060 BYE sip:203@172.17.2.11:5060 SIP/2.0..Record-Route: <sip:172.17.4.22;lr=on; ftag=61645575ae245b55>..Via: SIP/2.0/UDP 172.17.4.22;branch=z9hG4bK2751.609 47d11.0..Via: SIP/2.0/UDP 172.17.2.14;branch=z9hG4bKa419f2aaf11a7b40..From: <sip:3525@172.17.4.22:5060>;tag=61645575ae245b55..To: <sip:203@172.17.4.22 >;tag=hssUA_3795992784-846..Supported: replaces..Call-ID: 3795992784-846..C Seq: 323 BYE..User-Agent: Grandstream BT110 1.0.8.16..Max-Forwards: 69..All ow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Leng th: 0.... U 2007/03/18 12:54:23.517688 172.17.2.11:5060 -> 172.17.4.22:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 172.17.4.22;branch=z9hG4bK2751.60947d11.0, SIP/2.0/UDP 172.17.2.14;branch=z9hG4bKa419f2aaf11a7b40..Record-Route: <sip: 172.17.4.22;lr=on;ftag=61645575ae245b55>..Server: Claro BS3202 A454..From: <sip:3525@172.17.4.22:5060>;tag=61645575ae245b55..To: <sip:203@172.17.4.22> ;tag=hssUA_3795992784-846..Call-ID: 3795992784-846..CSeq: 323 BYE..Contact: <sip:203@172.17.2.11:5060>..Allow: INVITE,OPTIONS,CANCEL,ACK,BYE,PRACK,INF O,REFER,NOTIFY..Content-Length: 0.... U 2007/03/18 12:54:23.519138 172.17.4.22:5060 -> 172.17.2.14:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 172.17.2.14;branch=z9hG4bKa419f2aaf11a7b40 ..Record-Route: <sip:172.17.4.22;lr=on;ftag=61645575ae245b55>..Server: Clar o BS3202 A454..From: <sip:3525@172.17.4.22:5060>;tag=61645575ae245b55..To: <sip:203@172.17.4.22>;tag=hssUA_3795992784-846..Call-ID: 3795992784-846..CS eq: 323 BYE..Contact: <sip:203@172.17.2.11:5060>..Allow: INVITE,OPTIONS,CAN CEL,ACK,BYE,PRACK,INFO,REFER,NOTIFY..Content-Length: 0.... U 2007/03/18 12:54:29.780666 172.17.2.11:5060 -> 172.17.4.22:5060 BYE sip:5524@172.17.2.168:5061 SIP/2.0..Via: SIP/2.0/UDP 172.17.2.11:5060;b ranch=z9hG4bK3816062784-846..Route: <sip:172.17.4.22;lr=on;ftag=hssUA_38065 97784-846;did=0e3.7c5bd92>..Max-Forwards: 70..Proxy-Authorization: Digest u sername="613",realm="172.17.4.22",nonce="45fd6fe80721e4ce9c9b2eef4c93a66 ef8 8b9f2c",uri="sip:5524@172.17.4.22",response="3fb20f7d6e67668e07b862b9941 216 2d",algorithm=md5..User-Agent: Claro BS3202 A454..From: <sip:203@172.17.4.2 2>;tag=hssUA_3806597784-846..To: <sip:5524@172.17.4.22;user=phone>;tag=8cb5 cce9fbff0a22i1..Call-ID: 3806597784-846..CSeq: 3 BYE..Content-Length: 0.... U 2007/03/18 12:54:29.784698 172.17.4.22:5060 -> 172.17.2.168:5061 BYE sip:5524@172.17.2.168:5061 SIP/2.0..Record-Route: <sip:172.17.4.22;lr=o n;ftag=hssUA_3806597784-846>..Via: SIP/2.0/UDP 172.17.4.22;branch=z9hG4bKa5 c6.704043c7.0..Via: SIP/2.0/UDP 172.17.2.11:5060;branch=z9hG4bK3816062784-8 46..Max-Forwards: 69..Proxy-Authorization: Digest username="613",realm="172 .17.4.22",nonce="45fd6fe80721e4ce9c9b2eef4c93a66ef88b9f2c",uri="sip:5524 @17 2.17.4.22",response="3fb20f7d6e67668e07b862b99412162d",algorithm=md5..Us er- Agent: Claro BS3202 A454..From: <sip:203@172.17.4.22>;tag=hssUA_3806597784- 846..To: <sip:5524@172.17.4.22;user=phone>;tag=8cb5cce9fbff0a22i1..Call-ID: 3806597784-846..CSeq: 3 BYE..Content-Length: 0.... U 2007/03/18 12:54:29.804689 172.17.2.168:5061 -> 172.17.4.22:5060 SIP/2.0 200 OK..To: <sip:5524@172.17.4.22;user=phone>;tag=8cb5cce9fbff0a22i 1..From: <sip:203@172.17.4.22>;tag=hssUA_3806597784-846..Call-ID: 380659778 4-846..CSeq: 3 BYE..Via: SIP/2.0/UDP 172.17.4.22;branch=z9hG4bKa5c6.704043c 7.0..Via: SIP/2.0/UDP 172.17.2.11:5060;branch=z9hG4bK3816062784-846..Record -Route: <sip:172.17.4.22;lr=on;ftag=hssUA_3806597784-846>..Server: Linksys/ PAP2-3.1.3(LS)..P-RTP-Stat: PS=146,OS=34640,PR=303,OR=48162,PL=1,JI=0,LA=10 ,DU=6,EN=G711u,DE=G711u..Content-Length: 0.... U 2007/03/18 12:54:29.806174 172.17.4.22:5060 -> 172.17.2.11:5060 SIP/2.0 200 OK..To: <sip:5524@172.17.4.22;user=phone>;tag=8cb5cce9fbff0a22i 1..From: <sip:203@172.17.4.22>;tag=hssUA_3806597784-846..Call-ID: 380659778 4-846..CSeq: 3 BYE..Via: SIP/2.0/UDP 172.17.2.11:5060;branch=z9hG4bK3816062 784-846..Record-Route: <sip:172.17.4.22;lr=on;ftag=hssUA_3806597784-846>..S erver: Linksys/PAP2-3.1.3(LS)..P-RTP-Stat: PS=146,OS=34640,PR=303,OR=48162, PL=1,JI=0,LA=10,DU=6,EN=G711u,DE=G711u..Content-Length: 0.... As you can see, the dialog (callid=3795992784-846) still exists... /usr/local/sbin/openserctl fifo dlg_list 200 OK hash=3925, label=727521604, ptr=0x405c2d78 state=3, timeout=236218 callid='3795992784-846' from uri='sip:203@172.17.4.22', tag='hssUA_3795992784-846' to uri='sip:3525@172.17.4.22:5060', tag='61645575ae245b55' Thnk you in advance, Leonid Fainshtein -----Original Message----- From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] Sent: Friday, March 16, 2007 2:51 PM To: Leonid Fainshtein Cc: users(a)openser.org Subject: Re: [Users] Problem with dialog module Hi Leonid, what do you mean by "removes"?. When sequential requests are processed, the Route header corresponding to your server will be completely removed. regards, bogdan Leonid Fainshtein wrote: > Hi, > I see that when Openser 1.1 receives re-INVITE it removes parameter > "did=" from header field Route before sending it to the second call > party. > Any idea? > > Thank you in advance, > Leonid > > > ******************************************** > This outgoing mail message was scanned by McAfee GroupShield Engine > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > > ******************************************** This outgoing mail message was scanned by McAfee GroupShield Engine

2 2

[Serusers] SER with TLS
by Katty Xiong 04 Apr '07

04 Apr '07

I am using SER ottendorf with TLS protocol and have the following issues. Does anybody experience similar problems? SER cannot run with the following setup in the configuration file: (I follow this link to setup key and certificate: http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/modules/tls/README…) modparam("tls", "private_key", "cakey.pem") modparam("tls", "certificate", "cacert.pem") modparam("tls", "ca_list", "calist.pem") modparam("tls", "cipher_list", "HIGH"); With the last line commented out: #modparam("tls", "cipher_list", "HIGH"); SER can start, but the tls connection cannot be established. Network trace shows SER does not responde to ClientHello sent by client. Thanks, Joy ____________________________________________________________________________________ Food fight? Enjoy some healthy debate in the Yahoo! Answers Food & Drink Q&A. http://answers.yahoo.com/dir/?link=list&sid=396545367

3 9

[Users] response message content
by Victor Gamov 03 Apr '07

03 Apr '07

Hi all! Can somebody explain me how I can fill response with my useful headers? For example, I need to send 305 "Use proxy" response with Contact headers. Is it possible? If so, how can I populate it? Thanks! -- CU, Victor Gamov

3 5

Re: [Serusers] How to disable RTP proxy connection in
by Amanda Mattiuz 03 Apr '07

03 Apr '07

What happens if I disable rtpproxy but still load the nathelper? Can I still solve NAT problems with that?? -- ------------------------------------------------------------------- Amanda Remes Mattiuz amattiuz(a)gmail.com

3 2

[Users] RFC 3428
by Helmut Kuper 03 Apr '07

03 Apr '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I have a question about openser: Does openser support RFC 3428 (Instant Messaging (IM))? If so, How do I have to configure openser to relay SIP method MESSAGE ? regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGERzN4tZeNddg3dwRAkNgAJ9t+ffNXs8+tiISFPEZlSyf7A5m2ACfSzJM rKsinfXGvAWBBW6mMSO9a8Y= =ixv4 -----END PGP SIGNATURE-----

3 3

Re: Fwd: [Users] SNMPstats module not returning data
by Terry Robert 03 Apr '07

03 Apr '07

I experienced the same problem with the SNMPstats module. If you run openser in foreground mode (fork=no), the module doesn't initialize. Running in background mode fixes the problem.

3 3

Re: [Users] Openser fails when compiled on Solaris 64bit
by Sergio Gutierrez 03 Apr '07

03 Apr '07

Hi Daniel. This is the backtrace of the core. Thanks #0 0xffffffff7d90ee9c in preload_udomain (_c=0x10026b7d800, _d=0x0) at /home/operador/openser-1.2.0-notls/modules/usrloc//udomain.c:404 #1 0xffffffff7d9131c4 in child_init (_rank=1344858528) at /home/operador/openser-1.2.0-notls/modules/usrloc//ul_mod.c:344 #2 0x0000000100082500 in init_mod_child (m=0x1, rank=1, type=0x1000f41b0 "CHILD") at /home/operador/openser-1.2.0-notls//sr_module.c:400 #3 0x0000000100082460 in init_mod_child (m=0x1, rank=1, type=0x1000f41b0 "CHILD") at /home/operador/openser-1.2.0-notls//sr_module.c:394 #4 0x0000000100082460 in init_mod_child (m=0x1, rank=1, type=0x1000f41b0 "CHILD") at /home/operador/openser-1.2.0-notls//sr_module.c:394 #5 0x0000000100082770 in init_child (rank=1) at /home/operador/openser- 1.2.0-notls//sr_module.c:394 #6 0x0000000100032414 in main_loop () at /home/operador/openser-1.2.0-notls //main.c:724 #7 0x00000001000345a4 in main (argc=2, argv=0xffffff7e7007a000) at /home/operador/openser-1.2.0-notls//main.c:1399 On 3/26/07, Daniel-Constantin Mierla <daniel(a)voice-system.ro> wrote: > > Hello, > > can you send a gdb backtrace :-D -- I cannot read adb backtrace. > > Cheers, > Daniel > > > On 03/26/07 19:42, Sergio Gutierrez wrote: > > Hi Daniel. > > > > This is the backtrace, obtained with adb: > > > > usrloc.so`preload_udomain+0x4cc(1, ffffffff7228e9f0, 100215a00, > > ffffffff7d919e58, ffffffffffefe3f8, 0) > > 0xffffffff7d9131bc(2, ffffffff7da1c9b0, ffffffff7d91a170, > > ffffffff7228e9a0, ffffffff7da1b0a8, ffffffffffeff0c8) > > init_mod_child+0xd8(100269930, 1, 1000f41b0, 1000f40f0, 1, 100269870) > > init_mod_child+0x38(100269bb0, 1, 1000f41b0, 1000f40f0, 1, 1002699f0) > > init_mod_child+0x38(100269d30, 1, 1000f41b0, 1000f40f0, 1, 100269c70) > > init_child+0xa8(1, 100269df0, 1, 1000f4168, 10020d000, 100269eb0) > > main_loop+0xf34(0, 8c, 1000eefd8, ffffffff720083c4, 1002694b0, 0) > > main+0x1e10(100215, 9, 0, 10020d000, 1000ef5b8, 10020d000) > > _start+0x7c(0, 0, 0, 0, 0, 0) > > > > On 3/26/07, *Sergio Gutierrez* <saguti(a)gmail.com > > <mailto:saguti@gmail.com>> wrote: > > > > Hi Daniel. > > > > I used 512 for -m parameter. > > > > Would you like the backtrace of the core file? > > > > Thanks. > > > > > > On 3/26/07, *Daniel-Constantin Mierla* < daniel(a)voice-system.ro > > <mailto:daniel@voice-system.ro>> wrote: > > > > Hello, > > > > could you send a backtrace? What was the value for parameter -m? > > > > Cheers, > > Daniel > > > > > > On 03/26/07 19:29, Sergio Gutierrez wrote: > > > Hi > > > > > > I am trying to run Openser compiled on 64 bits on a SPARC > > Machine > > > running Solaris 10. > > > > > > When I create a simple configuration file for testing radius > > > integration, Openser starts to consume the whole memory > > reservation > > > (-m parameter) and fails with segmentation fault error. > > > > > > after several tests, I have found that the error is caused by > > save() > > > function (registrar module). > > > > > > This is the main route my configuration file: > > > > > > route { > > > if(method=="REGISTER") > > > { > > > if(!radius_www_authorize("")) > > > { > > > www_challenge("", "0"); > > > return; > > > }; > > > > > > if(!save("location")) > > > { > > > sl_reply_error(); > > > }; > > > return; > > > > > > } > > > else > > > { > > > } > > > } > > > > > > Thanks in advance for your help. > > > > > > Sergio G. > > > > > > > ------------------------------------------------------------------------ > > > > > > _______________________________________________ > > > Users mailing list > > > Users(a)openser.org <mailto:Users@openser.org> > > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > > > > >

3 10

[Users] Module Permission (register.allow and register.deny)
by Andrew And 03 Apr '07

03 Apr '07

In http://www.openser.org/docs/modules/1.2.x/permissions.html don´t have example with files register.allow and register.deny. Does someone has one example with rules these files?

1 0

[Users] does openser support QoS ?
by yanlin 03 Apr '07

03 Apr '07

Hi, does openser support some kind of QoS ? like, band width management, when there has no more usable bandwidth, then new INVITE should be rejected. so far, i know that dialog module has a bit dialog awareness. Any clues? thanks. yan lin

4 3

[Users] Serial forking broken wrt dst_uri/path?
by Frey, Daniel (external) 03 Apr '07

03 Apr '07

Hi, I tried to use append_branch() with dst_uri and path parameters set (and using set_dst_uri() / set_path_vector() for the first message). It works well as long as only parallel forking is involved, but when I use serialize_branches and next_branches and serial forking is required, the information given in dst_uri and path is discarded. This also seems obvious when reading the code from serialize.c, which simply doesn't store it on the contacts[n]-structs. I could add it to the sturct, but I'm at a loss how to store/retrieve it from the AVP. Are there any plans to fix this? Or any help on how I can fix it myself? Regards, Daniel

2 1

[Users] expires in Register message
by Dimo 03 Apr '07

03 Apr '07

Hi guys, Congratulations on the recent release, I will be testing it soon. I wonder if you can help me with a problem. I am trying to set UAs with a certain user-agent header to have different max_expires than all the rest. I am trying to override the default min_expires and max_expires values from within the config, after the definitions. So far i cannot do it - the modparam("registrar", "min_expires", xx) does not seem to accept an AVP for the xx part. Do you know if it is possible to change these timers per client from the code? Do they accept AVPs or is there any other way i can make it work? Thanks in advance Dimo

2 1

[Users] RTP without proxying, Linksys 3102
by Amanda Mattiuz 03 Apr '07

03 Apr '07

Hello Bill... it would be great if you could send me the cfg file you used.... -- ------------------------------------------------------------------- Amanda Remes Mattiuz amattiuz(a)gmail.com

1 0

[Serusers] mail client for SER UA
by Meidiana Ten 03 Apr '07

03 Apr '07

Hi there, I'm setting up voicemail server using SER. Do I need to setup mail client in client machine so it can access mailbox in server? If so, please tell me the software and how to configure the mail client. Thanx Regards, Meidiana ____________________________________________________________________________________ Never miss an email again! Yahoo! Toolbar alerts you the instant new Mail arrives. http://tools.search.yahoo.com/toolbar/features/mail/

1 0

RE: [Serusers] global contact list
by Barta János 03 Apr '07

03 Apr '07

Hello! Thanks for your answer. I'm using the standard config file (http://ftp.iptel.org/pub/ser/presence/cfg/full-no-failover.cfg), with a little modification, because I'm using radius authentication, accounting and media proxy. loadmodule "/opt/lib/ser/modules/acc_radius.so" loadmodule "/opt/lib/ser/modules/mediaproxy.so" loadmodule "/opt/lib/ser/modules/auth_radius.so" modparam("acc_radius","radius_config","/etc/radiusclient-ng/radiusclient.conf") modparam("acc_radius", "log_flag", 1) modparam("acc_radius", "log_missed_flag", 2) modparam("rls", "db_url", "mysql://yyyyyyy:xxxxxx@10.22.1.222/ser_pa") modparam("domain|uri_db|auth_db|usrloc|msilo", "db_url", "mysql://yyyyyyy:xxxxxx@10.22.1.222/ser_pa") if ((method=="INVITE") or (method=="BYE") or (method=="CANCEL")) { setflag(1); } if (!radius_www_authorize("test.intra")) { www_challenge("test.intra", "0"); break; }; handle_rls_subscription("1"); route[1] { if (method=="INVITE") { t_on_reply("1"); t_on_failure("1"); use_media_proxy(); } else if (method=="BYE|CANCEL") { end_media_session(); } # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } onreply_route[1] { if (status=~"(183)|2[0-9][0-9]") { use_media_proxy(); }; if (status=~"[3-6]0[0-9]") { end_media_session(); break; } } failure_route[1] { # forwarding failed -- check if the request was a MESSAGE if (!method=="MESSAGE") { break; }; log(1, "MSILO: MESSAGE forward failed - storing it\n"); # we have changed the R-URI with the contact address, ignore it now if (m_store("0", "")) { t_reply("202", "Accepted"); } else { log(1, "MSILO: offline message NOT stored\n"); t_reply("503", "Service Unavailable"); }; end_media_session(); } ----------------- I made a capture with Ethereal, and I can see that eyeBeam sent the "Supported: eventlist" field. I inserted the lists of both type: Global-list: Message Header Via: SIP/2.0/UDP 10.22.2.134:9510;branch=z9hG4bK-d87543-dd118615e6316f41-1--d87543- Max-Forwards: 70 Contact: <sip:12345@10.22.2.134:9510> To: <sip:bartaj-list@test.intra> SIP to address: sip:bartaj-list@test.intra From: "12345"<sip:12345@test.intra>;tag=0a48df34 SIP Display info: "12345" SIP from address: sip:12345@test.intra SIP tag: 0a48df34 Call-ID: ee79a62abd227204@S2FhbGlhLXhwLndlc3RlbC53ZXN0ZWw5MDAuaHU. CSeq: 1 SUBSCRIBE Expires: 60 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Supported: eventlist User-Agent: CounterPath eyeBeam Event: presence Content-Length: 0 User-list: Message Header Via: SIP/2.0/UDP 10.22.2.134:9510;branch=z9hG4bK-d87543-6300f275d049656e-1--d87543- Max-Forwards: 70 Contact: <sip:12345@10.22.2.134:9510> To: <sip:12345@test.intra> SIP to address: sip:12345@test.intra From: "12345"<sip:12345@test.intra>;tag=cb3e2c5c SIP Display info: "12345" SIP from address: sip:12345@test.intra SIP tag: cb3e2c5c Call-ID: a46e4b35e4754144@S2FhbGlhLXhwLndlc3RlbC53ZXN0ZWw5MDAuaHU. CSeq: 1 SUBSCRIBE Expires: 3600 Accept: message/external-body Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Supported: eventlist User-Agent: CounterPath eyeBeam Event: sip-profile;profile-type=application;app-id=resource-lists Content-Length: 0 The second case is working, but the first one is not. Is this maybe an EyeBeam problem? Which version of EyeBeam works fine (or which other client can I test it with)? And I have another question. What can I do with those clients, that does not support the presence information (can I do something anyway?). For example: I have a "Cisco Call Manager Express" box, and some Cisco IP Phones, that reach other PC clients through the SER. The Cisco makes the registration (and reregistration) with the phone's identifiers too into the SER (but the presence information is not handled by the Cisco), and therefore if a PC client would like to add the Cisco phone into his buddy list, the presence won't be working. ...Record(0x405cc588)... domain: 'location' aor : '101$' ~~~Contact(0x405cc5c8)~~~ domain : 'location' aor : '101$' Contact : 'sip:101$@10.22.2.233:5060' Expires : Expired q : Call-ID : '9C5717E5-D7D211DB-804EA958-71810899' CSeq : 4103 User-Agent: 'Cisco-SIPGateway/IOS-12.x' received : '' State : CS_SYNC Flags : 0 next : (nil) prev : (nil) ~~~/Contact~~~~ .../Record... ...Record(0x405cc698)... domain: 'location' aor : '100$' ~~~Contact(0x405cc6d8)~~~ domain : 'location' aor : '100$' Contact : 'sip:100$@10.22.2.233:5060' Expires : Expired q : Call-ID : '9C5717E5-D7D211DB-804DA958-71810899' CSeq : 1364 User-Agent: 'Cisco-SIPGateway/IOS-12.x' received : '' State : CS_SYNC Flags : 0 next : (nil) prev : (nil) ~~~/Contact~~~~ .../Record... Is it possible, that the SER (in these cases) simulates a presence client (if the user registration is active, then the user is online, anyway offline)? Thanks, Jani -----Original Message----- From: Vaclav Kubart [mailto:vaclav.kubart@iptel.org] Sent: Wednesday, March 28, 2007 3:46 PM To: Barta János Cc: serusers(a)iptel.org Subject: Re: [Serusers] global contact list Hi, please can you send here your config file? By the way, in some EyeBeam versions was not possible to use "Subscribe to contact list" feature because in such cases EyeBeam didn't send header "Supported: eventlist". Vaclav On Tue, Mar 27, 2007 at 08:54:15AM +0200, Barta János wrote: > Hello! > > I have a little problem with the (global) contact list subscription > of the presence, please help me. > I installed the presence server (0.10.99-dev35-pa-4.1), and it works > without any problems. > But I can not use the "Subscribe to contact list" of the eyeBeam. (The > user subscribe list works fine.) I typed the next content into this > field (of the eyeBeam): > > sip:bartaj-list@test.intra > > > and my config file contains: > > sip:/var/xcap-root/rls-services/global# cat ./index <?xml > version="1.0" encoding="UTF-8"?> <rls-services> > > <service uri="sip:bartaj-list@test.intra"> > <list name="Human resources"> > <entry uri="sip:test1@test.intra"> > > <display-name>bartaj</display-name> > </entry> > <entry uri="sip:test2@test.intra"> > > <display-name>12345</display-name> > </entry> > </list> > <packages> > <package>presence</package> > </packages> > </service> > > </rls-services> > > But I can see only the user subscribe list, and can't see the global > one. > Why? > Can anybody help me? > Thx, > Jani > _____________________________________________________________________________________________________ Jean Michel Jarre: Téo és Téa, az új, ünnepi album 1990 Forintért, kizárólag a T-Online Zeneáruházban! http://zenearuhaz.t-online.hu/index.php?m=info&albumid=30593&sp=1&sty=21

2 1

[Users] [resolv] radius problem in 1.2
by Bodin Bruno 03 Apr '07

03 Apr '07

Hello, During few days I've a problem with openser 1.2 and radius authentication but I just solve it ! This is a little note about my method. *Please, you're welcome to check and comment it for a better solution.* *Issue* : Radius authentication fail for openser 1.2 but not for 1.1.1 version with the same config. *Description* : freeradius server sucess (Sending Access-Accept in log) but it's not good in openser log. *Search method* : analyse of differences between 1.1.1 and 1.2 version >lsof | grep radius | grep -v radiusd /_openser 1.1.1_/ : openser 6226 root mem REG 8,2 50984 1333396 /usr/lib64/libradiusclient-ng.so.2.0.0 openser 6226 root mem REG 8,2 71642 30076344 /opt/openser-1.1.1/lib64/openser/modules/auth_radius.so openser 6226 root mem REG 8,2 41637 30076357 /opt/openser-1.1.1/lib64/openser/modules/group_radius.so openser 6226 root mem REG 8,2 43683 30076346 /opt/openser-1.1.1/lib64/openser/modules/avp_radius.so openser 6226 root mem REG 8,2 38171 30389824 /opt/openser-1.1.1/lib64/openser/modules/uri_radius.so _/openser1.2 :/_ openser 6299 root mem REG 8,2 50952 1333316 /usr/lib64/libfreeradius-client.so.2.0.0 openser 6299 root mem REG 8,2 76493 30356347 /opt/openser-1.2/lib64/openser/modules/auth_radius.so openser 6299 root mem REG 8,2 41769 30356361 /opt/openser-1.2/lib64/openser/modules/group_radius.so openser 6299 root mem REG 8,2 44923 30356349 /opt/openser-1.2/lib64/openser/modules/avp_radius.so openser 6299 root mem REG 8,2 39411 30076331 /opt/openser-1.2/lib64/openser/modules/uri_radius.so *openser 1.2.0 use libfreeradius-client.so.2.0.0 openser 1.1.1 use libradiusclient-ng.so.2.0.0 * * my solve method :* >cd /usr/lib64/ >mv libfreeradius-client.so.2.0.0 libfreeradius-client.so.2.0.0.old >ln -s libradiusclient-ng.so.2.0.0 libfreeradius-client.so.2.0.0

2 1

[Users] DLG: BYE not detected because of route reccord
by Michel Bensoussan 03 Apr '07

03 Apr '07

Hello The dialog module checks SIP messages only if the OpenSER in on record-route header in function dlg_onroute(). When the callee in registered on another SIP Server and the callee sends BYE, the OpenSER is not on record-route header. In this case, the dialog module doesn't detect end of call. Is there a way to callback the dlg_onroute() function not only in case of record-route? The configuration is as following: - Tel 101 registered to 192.168.13.8 (siproxd) (URI: 101(a)192.168.13.8) (IP: 192.168.13.101) - Tel 155 registered to 192.168.13.86 (OpenSER) (URI: 155(a)192.168.13.86) (IP: 192.168.13.155) - 155 is the caller and 101 is the callee. Attached, message capture (ethereal) when the callee (101) terminate the call. Thanks. Regards, Michel.

2 5

[Users] BUG in TM
by Sam Lee 03 Apr '07

03 Apr '07

Hey guys, I'm getting BUG:tm:run_failure_handlers: stack level is 1!! in my syslog. It is generating when i place a call out to PSTN. Any ideas ? Sam

2 5

Re: [Users] Module UAC to authentication and redirect calls
by Bogdan-Andrei Iancu 03 Apr '07

03 Apr '07

See the "credential" module parameter: http://www.openser.org/docs/modules/1.2.x/uac.html#AEN106 regards, bogdan Thiago Maluf wrote: > what`s parameters I will use to do this authentication? > > 2007/3/2, Bogdan-Andrei Iancu < bogdan(a)voice-system.ro > <mailto:bogdan@voice-system.ro>>: > > Hi, > > try something like: > > failure_route[1] { > xlog("-----------$T_reply_code ---------\n"); > if (t_check_status("401|407")) { > log(1,"doing auth\n"); > if (uac_auth()) { > append_branch(); > t_relay(); > } > } > } > > regards, > bogdan > > Thiago Maluf wrote: > > Hi List, > > I need config one module UAC to do authentication and redirect my > > calls to other Voip Provider. > > Somebody already did it? If can tell me what functions you used. > > Thanks in advanced! > > Thiago Maluf. > > > > ---------------------------------------------------------------- > > THIAGO MALUF RESENDE > > Consultor Voip e Programador WEB (Voip Developer and Web Developer) > > e-mail: malufrj(a)gmail.com <mailto:malufrj@gmail.com> <mailto: > malufrj(a)gmail.com <mailto:malufrj@gmail.com>> > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org <mailto:Users@openser.org> > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > > -- > ---------------------------------------------------------------- > THIAGO MALUF RESENDE > Consultor Voip e Programador WEB (Voip Developer and Web Developer) > Tel: +55 21 86042100 > e-mail: malufrj(a)gmail.com <mailto:malufrj@gmail.com>

1 0

Re: [Users] 3xx requests and t_relay
by Bogdan-Andrei Iancu 03 Apr '07

03 Apr '07

Hi Shane, please post the network trace from the the server and the full debug log of openser (use debug=9). regards, bogdan Shane Burrell wrote: > I used that as an example. On the sip side I see the 302 come back and the > openser ack it. Then I never see the new call go out. The user gets a busy > or not avaible message from the switch. In openser I have some logging and > it seems to start a new route after the ACK but bombs somewhere. > > -----Original Message----- > From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > Sent: Wednesday, February 28, 2007 12:54 PM > To: Shane Burrell; users openser.org > Subject: Re: [Users] 3xx requests and t_relay > > Hi Shane, > > have you followed this ex: > http://www.openser.org/docs/modules/1.2.x/uac_redirect.html#AEN251 > > can you be more specific about the errors you see? > > regards, > bogdan > > Shane Burrell wrote: > >> I am calling the t_relay after getting the contacts. However internally I >> see in openser it makes it back to the route and then dies in loose route. >> I assumed there was something I need to check to allow the new call to be >> processed. >> >> -----Original Message----- >> From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] >> Sent: Wednesday, February 28, 2007 12:26 PM >> To: Shane Burrell >> Cc: users(a)openser.org >> Subject: Re: [Users] 3xx requests and t_relay >> >> Hi, >> >> let me see if I get it right...you get a 3xx reply and try to generate >> INVITE to the new contacts? if so, take a look at the uac_redirect module. >> >> regards, >> bogdan >> >> Shane Burrell wrote: >> >> >>> I have this working to the point it tries to relay the call but I see >>> that internally openser is not making the new call. I assume t_relay >>> calls a new route but how to I make sure that all relays get >>> forwarded. Some of my logic else where seems to be killing the >>> rerouted call and I need any 302 routed calls to be accepted and routed. >>> >>> >>> >>> TIA, >>> >>> >>> >>> Shane >>> >>> >>> *************************************************************** >>> >>> >>> >>> ------------------------------------------------------------------------ >>> >>> _______________________________________________ >>> Users mailing list >>> Users(a)openser.org >>> http://openser.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >> >> > > >

1 0

[Users] TCP lifetime
by Klaus Darilion 03 Apr '07

03 Apr '07

Hi! I thought setting the tcp_connection_lifetime to 0 should keep them open endlessly, but in my case the proxy closes the TCP connection ~10 seconds after the last SIP message. What am I doing wrong? regards klaus -- Klaus Darilion nic.at

2 7

[Users] NEW FEATURE: IP blacklists
by Bogdan-Andrei Iancu 03 Apr '07

03 Apr '07

Hi everybody, OpenSER 1.2.0 has new feature - IP Blacklist support. This is a low level filtering engine for the outgoing requests; low level, because the filtering is done based on IP, protocol, port, etc. Its primary purposes will be to prevent sending requests to critical IPs (like GWs) due DNS or to avoid sending to destinations that are known to be unavailable (temporary or permanent). Because of flexibility concerns, the filtering rules can be groups inside multiple lists. A rule: - matches based on IP/mask, proto, port and text pattern criteria - can be reversed applied A list: - can be read-only - it does not change during execution - have timeout per elements - elements expires after a configured timeout. How to use: =========== currently there are 2 ways of using the blacklists: 1) statically defining list in the configuration file and selecting which ones should be used for each request. You can define blacklists as follow: # filter out requests going to ips of my gws dst_blacklist = gw:{( tcp , 192.168.2.100 , 5060 , "" ),( any , 192.168.2.101 , 0 , "" )} # block requests going to "evil" networks dst_blacklist = net_filter:{ ( any , 192.168.1.100/255.255.255.0 , 0 , "" )} # block message requests with nasty words dst_blacklist = msg_filter:{ ( any , 192.168.20.0/255.255.255.0 , 0 , "MESSAGE*ugly_word" )} # block requests not going to a specific subnet dst_blacklist = net_filter2:{ !( any , 192.168.30.0/255.255.255.0 , 0 , "" )} a rule is defined by: protocol : TCP, UDP, TLS or "any" for anything port : number or 0 for any ip/mask test patter - is a filename like matching (see "man 3 fnmatch") applied on the outgoing request buffer (first_line+hdrs+body) From routing script, you can use the use_blacklist("name") function to select what blacklist to be applied for the current request. More than one list can be selected. If the destination address matches on of the selected rules, the send will fail. 2) via DNS The DNS resolver, when configured with failover, can automatically store in a temporary blacklist the failed destinations. This will prevent (for a limited period of time) openser to send requests to destination known as failed. So, the blacklist can be used as a memory for the DNS resolver. To use it, you have to enabled it - the rest is done automatically. disable_dns_blacklist = no By default is enabled. The temporary blacklist created by DNS resolver is named "dns" and it is by default selected for usage (no need use the use_blacklist() function. The rules from this list have a life time of 4 minutes - you can change it at compile time, from blacklists.h . To give you an internal snapshot, a new MI function - "list_blacklists" - was added to print all existent blacklists and their rules. Any suggestions/reports are welcome! regards, bogdan

2 5

re: [Users] SIPp and registration
by yanlin 03 Apr '07

03 Apr '07

Hi, Scott Yagel, you could find some clues in the file scenario.cpp of sipp project. there were some embeded scenario scripts in sipp, stored in char* default_scenario[], you can sent a register sip message by doing this: " <send retrans=\"500\">\n" " <![CDATA[\n" "\n" " REGISTER sip:CA.cym.com SIP/2.0\n" " Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]\n" " From: ua1 <sip:ua1@nnl.cym:[local_port]>;tag=[pid]SIPpTag07[call_number]\n" " To: ua1 <sip:ua1@nnl.cym:[local_port]>\n" " Call-ID: [call_id]\n" " CSeq: 1 REGISTER\n" " Contact: sip:ua1@[local_ip]:[local_port]\n" " Content-Length: 0\n" " Expires: 300\n" "\n" " ]]>\n" " </send>\n" regards. Yan Lin

2 1

[Users] What does this mean?
by nick 03 Apr '07

03 Apr '07

pre_auth(): Credentials with given realm not found After looking around google it seems to be based on the fact that the first sip: value doesn't contain the userid (if I understand correctly...) I can't figure out exactly why the RURI is coming in this fashion.. Should I be changing something in the clients, or my config?? These worked on an older config, now with this section, they just don't get accepted. I've tried it with and without sipserver.mobilia.it in the realm. I've also tried it with the server generating ha1 and static ha1 (from the ha1 and ha1b fields in the db), no difference. I have use domain 1 in registrar and auth_db, and strip "sipserver." sl_send_reply("100", "Trying"); if(!www_authorize("", "subscriber")) { xlog("L_INFO", "Register authentication failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); www_challenge("", "1"); exit; } Mar 29 12:07:40 sipserver /usr/local/sbin/openser[22732]: New request - M=REGISTER RURI=sip:mobilia.it F=sip:rice@mobilia.it T=sip:rice@mobilia.it IP=x.x.x.105 ID=fmq-17240(a)x.x.x.105 Mar 29 12:07:40 sipserver /usr/local/sbin/openser[22732]: parse_headers: flags=100 Mar 29 12:07:40 sipserver /usr/local/sbin/openser[22732]: get_hdr_field: cseq <CSeq>: <100> <REGISTER> Mar 29 12:07:40 sipserver /usr/local/sbin/openser[22732]: DEBUG:maxfwd:is_maxfwd_present: value = 70 Mar 29 12:07:40 sipserver /usr/local/sbin/openser[22732]: parse_headers: flags=200 Mar 29 12:07:40 sipserver /usr/local/sbin/openser[22732]: DEBUG: get_hdr_body : content_length=0 Mar 29 12:07:40 sipserver /usr/local/sbin/openser[22732]: found end of header Mar 29 12:07:40 sipserver /usr/local/sbin/openser[22732]: find_first_route: No Route headers found Mar 29 12:07:40 sipserver /usr/local/sbin/openser[22732]: loose_route: There is no Route HF Mar 29 12:07:40 sipserver /usr/local/sbin/openser[22732]: parse_headers: flags=ffffffffffffffff Mar 29 12:07:40 sipserver /usr/local/sbin/openser[22732]: check_via_address(x.x.x.105, x.x.x.105, 0) Mar 29 12:07:40 sipserver /usr/local/sbin/openser[22732]: parse_headers: flags=4000 Mar 29 12:07:40 sipserver /usr/local/sbin/openser[22732]: pre_auth(): Credentials with given realm not found Mar 29 12:07:40 sipserver /usr/local/sbin/openser[22732]: Register authentication failed - M=REGISTER RURI=sip:mobilia.it F=sip:rice@mobilia.it T=sip:rice@mobilia.it IP=x.x.x.105 ID=fmq-17240(a)x.x.x.105 Mar 29 12:07:40 sipserver /usr/local/sbin/openser[22732]: build_auth_hf(): 'WWW-Authenticate: Digest realm="mobilia.it", nonce="460b9118a6a044d8837a01f924be2a9a3b7e0271", qop="auth"^M '

4 5

[Users] Simple xcap-emulator for OpenSER
by Adrian Georgescu 02 Apr '07

02 Apr '07

Hello, We wrote a simple xcap emulator to be able to manage the XCAP documents used by OpenSER presence agent, by using PHP scripts in combination with Apache web server. I have tested it against OpenSER with MySQL and Eyebeam for Apple. Klaus Darilion has tested it against Postgress and Eyebeam for Windows. The software can be downloaded from: http://download.dns-hosting.info/XCAP/xcap-0.5.tar.gz Regards, Adrian

2 3

[Serusers] How to disable RTP proxy connection in nathelper module
by tzieleniewski 02 Apr '07

02 Apr '07

Hi, I am using nathelper and because I have an CISCO router in front of the SER I don't need to use RTP Proxy. Whenever I start SER I get the following errors: Apr 2 09:13:33 sen ser[3442]: ERROR: send_rtpp_command: can't connect to RTP proxy Apr 2 09:13:33 sen ser[3442]: send_rtpp_command(): proxy <unix:/var/run/rtpproxy.sock> does not responding, disable it Apr 2 09:13:33 sen ser[3442]: WARNING: rtpp_test: can't get version of the RTP proxy Apr 2 09:13:33 sen ser[3442]: WARNING: rtpp_test: support for RTP proxy <unix:/var/run/rtpproxy.sock>has been disabled temporarily Apr 2 09:13:33 sen ser[3444]: ERROR: send_rtpp_command: can't connect to RTP proxy Is there a way to disable this trials through ser.cfg by nathelper mod parameter in order not to get those errors? Cheers Tomasz

2 1

[Users] WeSIP C2C and freeswitch
by Helmut Kuper 02 Apr '07

02 Apr '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I try to use WeSIP C2C with UACs registered to freeswitch. freeswitch never answeres the INVITEs for caller (that INVITE with no SDP and from=to=caller's sip address). So my question is, is it because freeswitch maybe doesn't support rfc 3725? If so, is there any C2C proven SIP-PBX I can use instead? Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFF7s8L4tZeNddg3dwRArGjAKCRJ+Kqi4FqdE2J9S/Q6t8L3VEVzwCgncCc G2rxVs9s3fqs+XgUXnsefRs= =RqW7 -----END PGP SIGNATURE-----

4 9

[Users] How URI is parsed...
by Stuart Sheldon 02 Apr '07

02 Apr '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Newbe question: Can someone explain what causes a match when lookup() and does_uri_exist() is used? Does it look at the least significant characters is the URI? Does it drop certain leading numbers like '1'? Stu -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.3 (GNU/Linux) iD8DBQFGDs0BK69Y+xPZrWYRAg5XAJ9kL3FNuImfu6wasPZ9z32uDYmoGwCfWKhW 5al96b0dhkmFNYnwBzyHmoY= =mNnn -----END PGP SIGNATURE-----

2 2

[Users] use of AVPs in core functions
by Jens Carl 02 Apr '07

02 Apr '07

Hello List, is it possible to use AVPs in core function like rewritehost(), etc. Regards Jens

2 1

Re: [Users] Openser on Sun Ultrasparc T1 Processors - Mismatch architecture detection
by Henning Westerholt 02 Apr '07

02 Apr '07

On Tuesday 27 March 2007 17:41, you wrote: > Hi Henning. > > I tested the compilation and it does not detect the architecture. > > I think that maybe is because of uname -p outputs sparc, and uname -m > outputs sun4v. > [..] Hi Sergio, thank you for the testing. I add another check to the patch according your suggestions, could please try it another time? Cheers, Henning

3 2

re: [Users] postgres
by Mark Price 02 Apr '07

02 Apr '07

On 12/20/06, Klaus Darilion <klaus.mailinglists(a)pernau.at > wrote: > > Hi Mark! > > Postgres should work well - I use it since ser 0.8. Just make sure that > the hard disk does not get full, because this breaks the index inside > postgres and postgres is getting real slow (re-create the index if it > happens). > > Last time I tested openser_postgres.sh it worked fine. If you find a bug > please let us know. > > Please check to use the latest versions from CVS (for 1.1 use CVS > rel_1_1_0) The latest version from cvs doesn't work out of the box with openser build from the release tarball, because the release tarball doesn't included /usr/sbin/openser_gen_ha1 (although I could get past this by replacing it with the md5sum equivelant). However, the release tarball doesn't work because of this bug: https://sourceforge.net/tracker/?func=detail&atid=743020&aid=1518732&group_… So, 1. is the cvs version of rel_1_1_0 considered safe for production use? 2. if so and there are important fixes such as this in CVS, would openser.org consider releasing a point release containing such changes? Thanks, Mark Price

6 8

RE: [Users] out of memory
by Papadopoulos Georgios 02 Apr '07

02 Apr '07

Hi Bogdan, I increased PKG_MEM_POOL_SIZE to 2MB and of course it took much longer to run out of memory. Usually it happens once every day, this time it ran for 3 days before running out of memory. If you don't mind, I will send the memory dump by personal email since I don't have easy access to an ftp server. Can you explain what you mean by 1.1.x? The version I am using is openser-1.1.1-notls_src.tar.gz, downloaded from your site under 1.1.1/. I am guessing that to get the fix I should use whatever is under latest-1.1.x/ Is this right? thank you George > -----Original Message----- > From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > Sent: Friday, March 30, 2007 3:33 PM > To: Papadopoulos Georgios > Cc: users(a)openser.org > Subject: Re: [Users] out of memory > > Papadopoulos Georgios wrote: > > > > Hi Bogdan, > > > > I found the following thread in the list > > http://www.openser.org/pipermail/users/2006-June/005339.html > > I suspect that I have the same issue. If I assume that our DNS is > > strange, then my 1.1.1-notls version will be affected, > right? Is there > > a patch to fix this? Eventually I want to go to 1.2 but I > have to do a > > lot of testing before I can put it in production. So for > the moment I > > need a fix for 1.1.1. > > > that fix is present in 1.1.x also. > > Also back in December I sent this to the list > > http://www.openser.org/pipermail/users/2006-December/008205.html > > >From the start, we had the issue of openser stopping to process any > > incoming requests. On December I thought it was DNS related. I am > > still in the dark about this. Lately I added logging and I > discovered > > the "out of memory" error. I am guessing that the error was > there from > > the start, I just did not know about it. Do all of these make any > > sense to you? And is there a way to test our DNS to > determine whether it is buggy? > > > I;m afraid not :) > > In the meantime I will follow the instructions on the wiki > to get the > > memory dump. > > > yes - that will be the most relevant info. > > regards, > bogdan > > thank you > > > > George > > > > > > > > > >> -----Original Message----- > >> From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > >> Sent: Wednesday, March 28, 2007 11:54 AM > >> To: Papadopoulos Georgios > >> Cc: users(a)openser.org > >> Subject: Re: [Users] out of memory > >> > >> Hi Papadopoulos, > >> > >> the error you get complains about the lack of private memory (per > >> process) - the statistics show info about the shared memory. See: > >> http://www.openser.org/dokuwiki/doku.php/troubleshooting:memory > >> > >> but look for pkg instead of shm..... > >> > >> regards, > >> bogdan > >> > >> Papadopoulos Georgios wrote: > >> > >>> Hello, > >>> > >>> Lately our OpenSER 1.1.1 is running out of memory and I get the > >>> following message > >>> 4(25190) get_hdr_field: out of memory The result is that > it stops > >>> processing any incoming request. This is a production > system and of > >>> course it causes a great deal of frustration. > >>> > >>> The last time this happened I got the following statistics before > >>> restarting > >>> > >>>> ./openserctl fifo get_statistics > >>>> > >>> Total statistics = 37 > >>> Total modules = 5 > >>> Module name = core; statistics=8 > >>> core:rcv_requests = 3259334 > >>> core:rcv_replies = 8816 > >>> core:fwd_requests = 736 > >>> core:fwd_replies = 137 > >>> core:drop_requests = 743 > >>> core:drop_replies = 0 > >>> core:err_requests = 0 > >>> core:err_replies = 43 > >>> Module name = shmem; statistics=6 > >>> shmem:total_size = 268435456 > >>> shmem:used_size = 1675864 > >>> shmem:real_used_size = 1835104 > >>> shmem:max_used_size = 4399048 > >>> shmem:free_size = 266600352 > >>> shmem:fragments = 17710 > >>> Module name = sl; statistics=9 > >>> sl:1xx_replies = 0 > >>> sl:2xx_replies = 1690410 > >>> sl:3xx_replies = 43 > >>> sl:4xx_replies = 334827 > >>> sl:5xx_replies = 1185629 > >>> sl:6xx_replies = 0 > >>> sl:sent_replies = 3210909 > >>> sl:sent_err_replies = 0 > >>> sl:received_ACKs = 743 > >>> Module name = tm; statistics=11 > >>> tm:received_replies = 8636 > >>> tm:relayed_replies = 7096 > >>> tm:local_replies = 1718 > >>> tm:UAS_transactions = 7003 > >>> tm:UAC_transactions = 94 > >>> tm:2xx_transactions = 1528 > >>> tm:3xx_transactions = 3 > >>> tm:4xx_transactions = 5129 > >>> tm:5xx_transactions = 468 > >>> tm:6xx_transactions = 0 > >>> tm:inuse_transactions = 0 > >>> Module name = usrloc; statistics=3 > >>> usrloc:location-users = 0 > >>> usrloc:location-contacts = 0 > >>> usrloc:location-expires = 7282 > >>> > >>> I am not sure how to read these numbers but I see a big > "free_size" > >>> and still OpenSER is out of memory. Any suggestions? > >>> > >>> thank you > >>> > >>> George > >>> > >>> > >>> > >>> > >>> > >>> > >>> Disclaimer > >>> > >>> The information in this e-mail and any attachments is > >>> > >> confidential. It > >> > >>> is intended solely for the attention and use of the named > >>> addressee(s). If you are not the intended recipient, or person > >>> responsible for delivering this information to the intended > >>> > >> recipient, > >> > >>> please notify the sender immediately. Unless you are the intended > >>> recipient or his/her representative you are not > authorized to, and > >>> must not, read, copy, distribute, use or retain this > message or any > >>> part of it. E-mail transmission cannot be guaranteed to be > >>> > >> secure or > >> > >>> error-free as information could be intercepted, corrupted, lost, > >>> destroyed, arrive late or incomplete, or contain viruses. > >>> > >>> > >>> > >> > --------------------------------------------------------------------- > >> - > >> > >>> -- > >>> > >>> _______________________________________________ > >>> Users mailing list > >>> Users(a)openser.org > >>> http://openser.org/cgi-bin/mailman/listinfo/users > >>> > >>> > >> > > > > > >

3 2

[Users] OpenSER v1.2.0 released
by Daniel-Constantin Mierla 02 Apr '07

02 Apr '07

Hello, version 1.2.0 of OpenSER has been released. It is major release, which means that you get along a lot of new features and improvements to existing ones in old versions. The configuration file and database structure suffered some changers, please read careful release notes to learn how to update. Just to list few of the new features: - SIMPLE Presence support (http://openser.org/dokuwiki/doku.php/presence:presence-module) - SNMP (http://openser.org/docs/modules/devel/snmpstats.html) - PERL API (http://openser.org/docs/modules/devel/perl.html) - JAVA SIP Servlet Application Server (http://wesip.eu) - script variables and transformations (http://openser.org/dokuwiki/doku.php/transformations:devel) - DNS failover - IP blacklists - XMPP IM gateway - XMLRPC - SIP session timers - secure federation peering - retransmission timer accuracy - usrloc refurbished for better performances - automatic error handling - AVPs in reply routes Read full release notes at: http://www.openser.org/mos/view/OpenSER-v1.2.x-Release-Notes/ Source tarball can be downloaded from: http://www.openser.org/pub/openser/latest/src/ Or SourceForge: http://sourceforge.net/project/showfiles.php?group_id=139143&package_id=224… This is the first release coming out o SVN. Old CVS repository was migrated to SVN, and with these, a few changes apply to get sources from repository. Please read instructions from download page: http://www.openser.org/mos/view/Download/ Migration instructions (although not complete, I'm sure) are available at: http://openser.org/dokuwiki/doku.php/install:1.1.x-to-1.2.x The documentation for modules is posted at: http://openser.org/docs/modules/1.2.x/ We would like to thank to all developers and contributors for their work with coding and testing for this new release. Cheers, Daniel

9 20

[Serusers] Header Selects
by Andrey Kuprianov 02 Apr '07

02 Apr '07

Hi, @msg.event or @msg.supported, or even @msg.["header-name"] is not implemented yet? im using latest ser from CVS.. btw, is it ok to use latest SER 2.0 for presence instead of the suggested snapshot v.4? I've noticed xcap-root and auth_xcap_root params were removed from rls and pa modules, so i've set xcap_root param in xcap module instead.

3 6

[Users] presence logging....
by Helge Waastad 02 Apr '07

02 Apr '07

Hi, I'm just wondering if it's possible to turn off logging of some presence status messages? I have debug=0 But still I see a lot of: pr 2 08:06:11 proxy-02 /usr/sbin/openser[28662]: PRESENCE: generate_ETag: etag= a.1175493921.28662.1 / 20 Apr 2 08:06:11 proxy-02 /usr/sbin/openser[28662]: PRESENCE: get_subs_dialog:The query for subscribtion for [user]= xxx,[domain]= yyy for [event]= presence returned no result Apr 2 08:06:11 proxy-02 /usr/sbin/openser[28662]: PRESENCE:query_db_notify: Could not get subs_dialog from database /32] 08:06:11 proxy-02 /usr/sbin/openser[28662]: PRESENCE: handle_publish: sip-etag: [SIP-ETag: a.1175493921.28662.1 Is the only way do only have L_ERR filtered in syslog daemon? br hw

1 0

[Serusers] About avp_db_store()
by chungyu 02 Apr '07

02 Apr '07

Hi all, I am trying to use avp_db_store() stores my table "contact" with caller uri ex: table name: contact ----------------------------- | username | domain | ----------------------------- | | | ----------------------------- | | | ----------------------------- If (method=="INVITE"){ avp_write("$ruri/username", "s:name"); avp_db_store("$ruri", "s:name/contact"); }; if the incoming call then save username and domain of caller in contact caller: bob(a)iptel.org => ----------------------------- | username | domain | ----------------------------- | bob | iptel.org | ----------------------------- | | | ----------------------------- but not success, system reply ERROR:avpops:db_store: insert faild Please help me in the problem Thanks a lot in advance chungyu

1 0

[Users] Call dropped after 6 ACKs. Branch=0 problem?
by kjcsb 02 Apr '07

02 Apr '07

I have the following scenario: PSTN gateway (202.180.nnn.nnn) -> OpenSER 1.0.1 (147.202.nnn.nnn) -> Asterisk 1.2.16 (203.89.nnn.nnn) When an incoming call is received, the call is established but often (but not always) eventually dropped. In the following dialogue, it can be seen that when the ACK is passed from the gateway on to Asterisk, OpenSER appends a Via with branch=0. After a number of these ACKs are passed on, the call is eventually dropped. It has been suggested by someone on the Asterisk list that "While "branch=0" is valid in RFC 2543, I don't think an INVITE can start-off as RFC 3261 and then the ACK can switch over to RFC 2543 in the middle of the transaction." i.e. the implication is that it's an OpenSER problem. Does that seem right? If so how can it be corrected? If not, could anyone advise what the problem might be or suggest any resolutions? U 202.180.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 INVITE sip:6499777777@domain.co.nz SIP/2.0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz> Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 02 Apr 2007 02:53:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 336 v=0 o=root 8802 8802 IN IP4 202.180.nnn.nnn s=session c=IN IP4 202.180.nnn.nnn t=0 0 m=audio 35268 RTP/AVP 18 97 3 0 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:97 iLBC/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - U 147.202.nnn.nnn:5060 -> 202.180.nnn.nnn:5060 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE Server: OpenSer (1.0.1 (i386/linux)) Content-Length: 0 Warning: 392 147.202.nnn.nnn:5060 "Noisy feedback tells: pid=29716 req_src_ip=202.180.nnn.nnn req_src_port=5060 in_uri=sip:6499777777@domain.co.nz out_uri=sip:6499777777@203.89.nnn.nnn via_cnt==1" U 147.202.nnn.nnn:5060 -> 203.89.nnn.nnn:5060 INVITE sip:6499777777@203.89.nnn.nnn SIP/2.0 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKd5ab.736d6085.0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz> Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 69 Date: Mon, 02 Apr 2007 02:53:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 336 v=0 o=root 8802 8802 IN IP4 202.180.nnn.nnn s=session c=IN IP4 202.180.nnn.nnn t=0 0 m=audio 35268 RTP/AVP 18 97 3 0 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:97 iLBC/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - U 147.202.nnn.nnn:5060 -> 203.89.nnn.nnn:5060 INVITE sip:6499777777@203.89.nnn.nnn SIP/2.0 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKd5ab.736d6085.0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz> Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 69 Date: Mon, 02 Apr 2007 02:53:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 336 v=0 o=root 8802 8802 IN IP4 202.180.nnn.nnn s=session c=IN IP4 202.180.nnn.nnn t=0 0 m=audio 35268 RTP/AVP 18 97 3 0 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:97 iLBC/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - U 203.89.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKd5ab.736d6085.0;received=147.202.nnn.nnn Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:6499777777@203.89.nnn.nnn> Content-Length: 0 U 203.89.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKd5ab.736d6085.0;received=147.202.nnn.nnn Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:6499777777@203.89.nnn.nnn> Content-Type: application/sdp Content-Length: 244 v=0 o=root 16300 16300 IN IP4 203.89.nnn.nnn s=session c=IN IP4 203.89.nnn.nnn t=0 0 m=audio 19952 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - U 147.202.nnn.nnn:5060 -> 202.180.nnn.nnn:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:6499777777@203.89.nnn.nnn> Content-Type: application/sdp Content-Length: 244 v=0 o=root 16300 16300 IN IP4 203.89.nnn.nnn s=session c=IN IP4 203.89.nnn.nnn t=0 0 m=audio 19952 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - U 203.89.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKd5ab.736d6085.0;received=147.202.nnn.nnn Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:6499777777@203.89.nnn.nnn> Content-Type: application/sdp Content-Length: 244 v=0 o=root 16300 16301 IN IP4 203.89.nnn.nnn s=session c=IN IP4 203.89.nnn.nnn t=0 0 m=audio 19952 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - U 147.202.nnn.nnn:5060 -> 202.180.nnn.nnn:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:6499777777@203.89.nnn.nnn> Content-Type: application/sdp Content-Length: 244 v=0 o=root 16300 16301 IN IP4 203.89.nnn.nnn s=session c=IN IP4 203.89.nnn.nnn t=0 0 m=audio 19952 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - U 202.180.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 ACK sip:6499777777@203.89.nnn.nnn SIP/2.0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK013cd54d;rport Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 U 147.202.nnn.nnn:5060 -> 203.89.nnn.nnn:5060 ACK sip:6499777777@203.89.nnn.nnn SIP/2.0 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK013cd54d;rport=5060 From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 69 Content-Length: 0 U 202.180.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 ACK sip:6499777777@203.89.nnn.nnn SIP/2.0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK54b95a19;rport Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 U 147.202.nnn.nnn:5060 -> 203.89.nnn.nnn:5060 ACK sip:6499777777@203.89.nnn.nnn SIP/2.0 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK54b95a19;rport=5060 From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 69 Content-Length: 0 U 203.89.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKd5ab.736d6085.0;received=147.202.nnn.nnn Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:6499777777@203.89.nnn.nnn> Content-Type: application/sdp Content-Length: 244 v=0 o=root 16300 16300 IN IP4 203.89.nnn.nnn s=session c=IN IP4 203.89.nnn.nnn t=0 0 m=audio 19952 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - U 147.202.nnn.nnn:5060 -> 202.180.nnn.nnn:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:6499777777@203.89.nnn.nnn> Content-Type: application/sdp Content-Length: 244 v=0 o=root 16300 16300 IN IP4 203.89.nnn.nnn s=session c=IN IP4 203.89.nnn.nnn t=0 0 m=audio 19952 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - U 202.180.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 ACK sip:6499777777@203.89.nnn.nnn SIP/2.0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK46a352b9;rport Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 U 147.202.nnn.nnn:5060 -> 203.89.nnn.nnn:5060 ACK sip:6499777777@203.89.nnn.nnn SIP/2.0 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK46a352b9;rport=5060 From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 69 Content-Length: 0 U 203.89.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKd5ab.736d6085.0;received=147.202.nnn.nnn Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:6499777777@203.89.nnn.nnn> Content-Type: application/sdp Content-Length: 244 v=0 o=root 16300 16300 IN IP4 203.89.nnn.nnn s=session c=IN IP4 203.89.nnn.nnn t=0 0 m=audio 19952 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - U 147.202.nnn.nnn:5060 -> 202.180.nnn.nnn:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:6499777777@203.89.nnn.nnn> Content-Type: application/sdp Content-Length: 244 v=0 o=root 16300 16300 IN IP4 203.89.nnn.nnn s=session c=IN IP4 203.89.nnn.nnn t=0 0 m=audio 19952 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - U 202.180.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 ACK sip:6499777777@203.89.nnn.nnn SIP/2.0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK2da46c51;rport Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 U 147.202.nnn.nnn:5060 -> 203.89.nnn.nnn:5060 ACK sip:6499777777@203.89.nnn.nnn SIP/2.0 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK2da46c51;rport=5060 From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 69 Content-Length: 0 U 203.89.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKd5ab.736d6085.0;received=147.202.nnn.nnn Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:6499777777@203.89.nnn.nnn> Content-Type: application/sdp Content-Length: 244 v=0 o=root 16300 16300 IN IP4 203.89.nnn.nnn s=session c=IN IP4 203.89.nnn.nnn t=0 0 m=audio 19952 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - U 147.202.nnn.nnn:5060 -> 202.180.nnn.nnn:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:6499777777@203.89.nnn.nnn> Content-Type: application/sdp Content-Length: 244 v=0 o=root 16300 16300 IN IP4 203.89.nnn.nnn s=session c=IN IP4 203.89.nnn.nnn t=0 0 m=audio 19952 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - U 202.180.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 ACK sip:6499777777@203.89.nnn.nnn SIP/2.0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK23a5accc;rport Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 U 147.202.nnn.nnn:5060 -> 203.89.nnn.nnn:5060 ACK sip:6499777777@203.89.nnn.nnn SIP/2.0 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK23a5accc;rport=5060 From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 69 Content-Length: 0 U 203.89.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKd5ab.736d6085.0;received=147.202.nnn.nnn Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:6499777777@203.89.nnn.nnn> Content-Type: application/sdp Content-Length: 244 v=0 o=root 16300 16300 IN IP4 203.89.nnn.nnn s=session c=IN IP4 203.89.nnn.nnn t=0 0 m=audio 19952 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - U 147.202.nnn.nnn:5060 -> 202.180.nnn.nnn:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:6499777777@203.89.nnn.nnn> Content-Type: application/sdp Content-Length: 244 v=0 o=root 16300 16300 IN IP4 203.89.nnn.nnn s=session c=IN IP4 203.89.nnn.nnn t=0 0 m=audio 19952 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - U 202.180.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 ACK sip:6499777777@203.89.nnn.nnn SIP/2.0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK153c7273;rport Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 U 147.202.nnn.nnn:5060 -> 203.89.nnn.nnn:5060 ACK sip:6499777777@203.89.nnn.nnn SIP/2.0 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK153c7273;rport=5060 From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 69 Content-Length: 0 U 203.89.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKd5ab.736d6085.0;received=147.202.nnn.nnn Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:6499777777@203.89.nnn.nnn> Content-Type: application/sdp Content-Length: 244 v=0 o=root 16300 16300 IN IP4 203.89.nnn.nnn s=session c=IN IP4 203.89.nnn.nnn t=0 0 m=audio 19952 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - U 147.202.nnn.nnn:5060 -> 202.180.nnn.nnn:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:6499777777@203.89.nnn.nnn> Content-Type: application/sdp Content-Length: 244 v=0 o=root 16300 16300 IN IP4 203.89.nnn.nnn s=session c=IN IP4 203.89.nnn.nnn t=0 0 m=audio 19952 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - U 202.180.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 ACK sip:6499777777@203.89.nnn.nnn SIP/2.0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK01e676a5;rport Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 U 147.202.nnn.nnn:5060 -> 203.89.nnn.nnn:5060 ACK sip:6499777777@203.89.nnn.nnn SIP/2.0 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK01e676a5;rport=5060 From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 69 Content-Length: 0 U 203.89.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKd5ab.736d6085.0;received=147.202.nnn.nnn Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:6499777777@203.89.nnn.nnn> Content-Type: application/sdp Content-Length: 244 v=0 o=root 16300 16300 IN IP4 203.89.nnn.nnn s=session c=IN IP4 203.89.nnn.nnn t=0 0 m=audio 19952 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - U 147.202.nnn.nnn:5060 -> 202.180.nnn.nnn:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:6499777777@203.89.nnn.nnn> Content-Type: application/sdp Content-Length: 244 v=0 o=root 16300 16300 IN IP4 203.89.nnn.nnn s=session c=IN IP4 203.89.nnn.nnn t=0 0 m=audio 19952 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - U 202.180.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 ACK sip:6499777777@203.89.nnn.nnn SIP/2.0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0a0ddae8;rport Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 U 147.202.nnn.nnn:5060 -> 203.89.nnn.nnn:5060 ACK sip:6499777777@203.89.nnn.nnn SIP/2.0 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0a0ddae8;rport=5060 From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 69 Content-Length: 0 U 202.180.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 BYE sip:6499777777@203.89.nnn.nnn SIP/2.0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK12646817;rport Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 U 147.202.nnn.nnn:5060 -> 203.89.nnn.nnn:5060 BYE sip:6499777777@203.89.nnn.nnn SIP/2.0 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKe5ab.311dd295.0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK12646817;rport=5060 From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 69 Content-Length: 0 U 203.89.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKe5ab.311dd295.0;received=147.202.nnn.nnn Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK12646817;rport=5060 From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 103 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 U 147.202.nnn.nnn:5060 -> 202.180.nnn.nnn:5060 SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK12646817;rport=5060 From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 103 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 Thanks Cameron

1 0

[Serusers] Unreacheable destination problems
by Amanda Mattiuz 01 Apr '07

01 Apr '07

Hello! I'm using SER + rtpproxy, but altough the NAT problem seems to be solved, I get no audio ... analysing the traffic with ethereal, I realized that the traffic going from the proxy to the UAs never arrives (ICMP unreacheable destination)... I assume it is because the users have their ports blocked, is that correct? I changed the rtpproxy port range (PORT_MIN and PORT_MAX ) to lower ports, but still I got the same messages... Does anybody knows anything that can be done by me in terms of configurations of the rtpproxy or ser.cfg or whatever, given the fact that I don't have power to open the ports??..... Thanks, Amanda -- ------------------------------------------------------------------- Amanda Remes Mattiuz amattiuz(a)gmail.com

1 0

[Users] Re: Digest Authentication Problem with header parsing problem
by Abdurrahman Sahin 01 Apr '07

01 Apr '07

hi all; i solved my problem; i was setting my server's ip address as the realm, and i guess providing an ip address as the realm causes problem. so i changed the realm name to a domain name , then set a sip proxy address for the client i also changed SIP_DOMAIN to a domain name. ----- Original Message ----- From: Abdurrahman Sahin To: users(a)openser.org Sent: Sunday, April 01, 2007 3:00 PM Subject: Digest Authentication Problem with header parsing problem hi all; i'm trying to activate digest authentication with mysql, but i see a strange parsing error in the logs. here is my logser log ================== Apr 1 13:39:04 testsrv200 /usr/local/sbin/openser[577]: New request - M=REGISTER RURI=sip:server_public_ip_address F=<null> T=sip:apo@server_public_ip_address IP=ua_public_ip_address ID=NWYyZGNjMWRjYjFlNTk5YjlhMjE5ODA4MWQ0ZGFiNTY. Apr 1 13:39:04 testsrv200 /usr/local/sbin/openser[577]: DEBUG:maxfwd:is_maxfwd_present: value = 70 Apr 1 13:39:04 testsrv200 /usr/local/sbin/openser[577]: parse_headers: flags=200 Apr 1 13:39:04 testsrv200 /usr/local/sbin/openser[577]: get_hdr_field: cseq <CSeq>: <1> <REGISTER> Apr 1 13:39:04 testsrv200 /usr/local/sbin/openser[577]: DEBUG: get_hdr_body : content_length=0 Apr 1 13:39:04 testsrv200 /usr/local/sbin/openser[577]: found end of header Apr 1 13:39:04 testsrv200 /usr/local/sbin/openser[577]: find_first_route: No Route headers found Apr 1 13:39:04 testsrv200 /usr/local/sbin/openser[577]: loose_route: There is no Route HF Apr 1 13:39:04 testsrv200 /usr/local/sbin/openser[577]: parse_headers: flags=ffffffffffffffff Apr 1 13:39:04 testsrv200 /usr/local/sbin/openser[577]: check_via_address(ua_public_ip_address, ua_public_ip_address, 0) Apr 1 13:39:04 testsrv200 /usr/local/sbin/openser[577]: parse_headers: flags=4000 Apr 1 13:39:04 testsrv200 /usr/local/sbin/openser[577]: pre_auth(): Credentials with given realm not found ] in status 6: <<"apo"<sip:apo@ua_public_ip_address;tag=5d692f7d>> . R: parse_to : unexpected char [ Apr 1 13:39:05 testsrv200 /usr/local/sbin/openser[577]: ERROR:parse_from_header: bad from header Apr 1 13:39:05 testsrv200 /usr/local/sbin/openser[577]: set_err_info: ec: 1, el: 3, ei: 'error parsing From' Apr 1 13:39:05 testsrv200 /usr/local/sbin/openser[577]: xl_get_from_attr: ERROR cannot parse FROM header Apr 1 13:39:05 testsrv200 /usr/local/sbin/openser[577]: Register authentication failed - M=REGISTER RURI=sip:server_public_ip_address F=<null> T=sip:apo@server_public_ip_address IP=ua_public_ip_address ID=NWYyZGNjMWRjYjFlNTk5YjlhMjE5ODA4MWQ0ZGFiNTY. Apr 1 13:39:05 testsrv200 /usr/local/sbin/openser[577]: build_auth_hf(): 'WWW-Authenticate: Digest realm="server_public_ip_address", nonce ' 60f8cf5a7a5eb40c26f914280272b51f58caaa7" Apr 1 13:39:05 testsrv200 /usr/local/sbin/openser[577]: parse_headers: flags=ffffffffffffffff Apr 1 13:39:05 testsrv200 /usr/local/sbin/openser[577]: check_via_address(ua_public_ip_address, ua_public_ip_address, 0) Apr 1 13:39:05 testsrv200 /usr/local/sbin/openser[577]: DEBUG:destroy_avp_list: destroying list (nil) Apr 1 13:39:05 testsrv200 /usr/local/sbin/openser[577]: receive_msg: cleaning up and sip msg received at the server side ============================== U ua_public_ip_address:28443 -> 192.168.200.2:5060 REGISTER sip:server_public_ip_address SIP/2.0. Via: SIP/2.0/UDP ua_public_ip_address:28443;branch=z9hG4bK-d87543-08523511e5151c18-1--d87543-;rport. Max-Forwards: 70. Contact: <sip:apo@ua_public_ip_address:28443;rinstance=1ff3024a97431241>. To: "apo"<sip:apo@server_public_ip_address>. From: "apo"<sip:apo@ua_public_ip_address;tag=5d692f7d. Call-ID: NWYyZGNjMWRjYjFlNTk5YjlhMjE5ODA4MWQ0ZGFiNTY.. CSeq: 1 REGISTER. Expires: 3600. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. User-Agent: X-Lite release 1006e stamp 34025. Content-Length: 0. . # U 192.168.200.2:5060 -> ua_public_ip_address:28443 SIP/2.0 100 Trying. Via: SIP/2.0/UDP ua_public_ip_address:28443;branch=z9hG4bK-d87543-08523511e5151c18-1--d87543-;rport=28443. To: "apo"<sip:apo@server_public_ip_address>. From: "apo"<sip:apo@ua_public_ip_address;tag=5d692f7d. Call-ID: NWYyZGNjMWRjYjFlNTk5YjlhMjE5ODA4MWQ0ZGFiNTY.. CSeq: 1 REGISTER. Server: OpenSER (1.2.0-notls (x86_64/linux)). Content-Length: 0. . # U 192.168.200.2:5060 -> ua_public_ip_address:28443 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP ua_public_ip_address:28443;branch=z9hG4bK-d87543-08523511e5151c18-1--d87543-;rport=28443. To: "apo"<sip:apo@server_public_ip_address>;tag=329cfeaa6ded039da25ff8cbb8668bd2.4f01. From: "apo"<sip:apo@ua_public_ip_address;tag=5d692f7d. Call-ID: NWYyZGNjMWRjYjFlNTk5YjlhMjE5ODA4MWQ0ZGFiNTY.. CSeq: 1 REGISTER. WWW-Authenticate: Digest realm="server_public_ip_address", nonce="460f8cf5a7a5eb40c26f914280272b51f58caaa7". Server: OpenSER (1.2.0-notls (x86_64/linux)). Content-Length: 0. openser.cfg file register section ======================================= # Request route 'base-route-register' ######################################################################## route[2] { sl_send_reply("100", "Trying"); if(!www_authorize("server_public_ip_address", "subscriber")) { xlog("L_INFO", "Register authentication failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); www_challenge("server_public_ip_address", "0"); exit; } if(!check_to()) { xlog("L_INFO", "Spoofed To-URI detected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Spoofed To-URI Detected"); exit; } consume_credentials(); if(!search("^Contact:[ ]*\*") && nat_uac_test("19")) { fix_nated_register(); setbflag(6); } if(!save("location")) { xlog("L_ERR", "Saving contact failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_reply_error(); exit; } xlog("L_INFO", "Registration successful - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); exit; }

1 0

[Users] Digest Authentication Problem with header parsing problem
by Abdurrahman Sahin 01 Apr '07

01 Apr '07

hi all; i'm trying to activate digest authentication with mysql, but i see a strange parsing error in the logs. here is my logser log ================== Apr 1 13:39:04 testsrv200 /usr/local/sbin/openser[577]: New request - M=REGISTER RURI=sip:server_public_ip_address F=<null> T=sip:apo@server_public_ip_address IP=ua_public_ip_address ID=NWYyZGNjMWRjYjFlNTk5YjlhMjE5ODA4MWQ0ZGFiNTY. Apr 1 13:39:04 testsrv200 /usr/local/sbin/openser[577]: DEBUG:maxfwd:is_maxfwd_present: value = 70 Apr 1 13:39:04 testsrv200 /usr/local/sbin/openser[577]: parse_headers: flags=200 Apr 1 13:39:04 testsrv200 /usr/local/sbin/openser[577]: get_hdr_field: cseq <CSeq>: <1> <REGISTER> Apr 1 13:39:04 testsrv200 /usr/local/sbin/openser[577]: DEBUG: get_hdr_body : content_length=0 Apr 1 13:39:04 testsrv200 /usr/local/sbin/openser[577]: found end of header Apr 1 13:39:04 testsrv200 /usr/local/sbin/openser[577]: find_first_route: No Route headers found Apr 1 13:39:04 testsrv200 /usr/local/sbin/openser[577]: loose_route: There is no Route HF Apr 1 13:39:04 testsrv200 /usr/local/sbin/openser[577]: parse_headers: flags=ffffffffffffffff Apr 1 13:39:04 testsrv200 /usr/local/sbin/openser[577]: check_via_address(ua_public_ip_address, ua_public_ip_address, 0) Apr 1 13:39:04 testsrv200 /usr/local/sbin/openser[577]: parse_headers: flags=4000 Apr 1 13:39:04 testsrv200 /usr/local/sbin/openser[577]: pre_auth(): Credentials with given realm not found ] in status 6: <<"apo"<sip:apo@ua_public_ip_address;tag=5d692f7d>> . R: parse_to : unexpected char [ Apr 1 13:39:05 testsrv200 /usr/local/sbin/openser[577]: ERROR:parse_from_header: bad from header Apr 1 13:39:05 testsrv200 /usr/local/sbin/openser[577]: set_err_info: ec: 1, el: 3, ei: 'error parsing From' Apr 1 13:39:05 testsrv200 /usr/local/sbin/openser[577]: xl_get_from_attr: ERROR cannot parse FROM header Apr 1 13:39:05 testsrv200 /usr/local/sbin/openser[577]: Register authentication failed - M=REGISTER RURI=sip:server_public_ip_address F=<null> T=sip:apo@server_public_ip_address IP=ua_public_ip_address ID=NWYyZGNjMWRjYjFlNTk5YjlhMjE5ODA4MWQ0ZGFiNTY. Apr 1 13:39:05 testsrv200 /usr/local/sbin/openser[577]: build_auth_hf(): 'WWW-Authenticate: Digest realm="server_public_ip_address", nonce ' 60f8cf5a7a5eb40c26f914280272b51f58caaa7" Apr 1 13:39:05 testsrv200 /usr/local/sbin/openser[577]: parse_headers: flags=ffffffffffffffff Apr 1 13:39:05 testsrv200 /usr/local/sbin/openser[577]: check_via_address(ua_public_ip_address, ua_public_ip_address, 0) Apr 1 13:39:05 testsrv200 /usr/local/sbin/openser[577]: DEBUG:destroy_avp_list: destroying list (nil) Apr 1 13:39:05 testsrv200 /usr/local/sbin/openser[577]: receive_msg: cleaning up and sip msg received at the server side ============================== U ua_public_ip_address:28443 -> 192.168.200.2:5060 REGISTER sip:server_public_ip_address SIP/2.0. Via: SIP/2.0/UDP ua_public_ip_address:28443;branch=z9hG4bK-d87543-08523511e5151c18-1--d87543-;rport. Max-Forwards: 70. Contact: <sip:apo@ua_public_ip_address:28443;rinstance=1ff3024a97431241>. To: "apo"<sip:apo@server_public_ip_address>. From: "apo"<sip:apo@ua_public_ip_address;tag=5d692f7d. Call-ID: NWYyZGNjMWRjYjFlNTk5YjlhMjE5ODA4MWQ0ZGFiNTY.. CSeq: 1 REGISTER. Expires: 3600. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. User-Agent: X-Lite release 1006e stamp 34025. Content-Length: 0. . # U 192.168.200.2:5060 -> ua_public_ip_address:28443 SIP/2.0 100 Trying. Via: SIP/2.0/UDP ua_public_ip_address:28443;branch=z9hG4bK-d87543-08523511e5151c18-1--d87543-;rport=28443. To: "apo"<sip:apo@server_public_ip_address>. From: "apo"<sip:apo@ua_public_ip_address;tag=5d692f7d. Call-ID: NWYyZGNjMWRjYjFlNTk5YjlhMjE5ODA4MWQ0ZGFiNTY.. CSeq: 1 REGISTER. Server: OpenSER (1.2.0-notls (x86_64/linux)). Content-Length: 0. . # U 192.168.200.2:5060 -> ua_public_ip_address:28443 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP ua_public_ip_address:28443;branch=z9hG4bK-d87543-08523511e5151c18-1--d87543-;rport=28443. To: "apo"<sip:apo@server_public_ip_address>;tag=329cfeaa6ded039da25ff8cbb8668bd2.4f01. From: "apo"<sip:apo@ua_public_ip_address;tag=5d692f7d. Call-ID: NWYyZGNjMWRjYjFlNTk5YjlhMjE5ODA4MWQ0ZGFiNTY.. CSeq: 1 REGISTER. WWW-Authenticate: Digest realm="server_public_ip_address", nonce="460f8cf5a7a5eb40c26f914280272b51f58caaa7". Server: OpenSER (1.2.0-notls (x86_64/linux)). Content-Length: 0. openser.cfg file register section ======================================= # Request route 'base-route-register' ######################################################################## route[2] { sl_send_reply("100", "Trying"); if(!www_authorize("server_public_ip_address", "subscriber")) { xlog("L_INFO", "Register authentication failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); www_challenge("server_public_ip_address", "0"); exit; } if(!check_to()) { xlog("L_INFO", "Spoofed To-URI detected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Spoofed To-URI Detected"); exit; } consume_credentials(); if(!search("^Contact:[ ]*\*") && nat_uac_test("19")) { fix_nated_register(); setbflag(6); } if(!save("location")) { xlog("L_ERR", "Saving contact failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_reply_error(); exit; } xlog("L_INFO", "Registration successful - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); exit; }

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[Users] UA simulator
by Michel Bensoussan 01 Apr '07

01 Apr '07

Hello This question is for openser developers. How do you test your code? Is there a UA simulator to generate many SIP messages and tests different scenarios? Thanks and regards, Michel.

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