September 2007 - sr-users

[OpenSER-Users] How do I obtain or log the return code in a failure_route[]?
by Juan Carlos Castro y Castro 10 Sep '07

10 Sep '07

I know I can test the SIP return code with an "if", but how do I log it, or, alternatively, pass it to a exec_dset() script? I tried something like log(1, "Failure Route, status is $status\n"); And I get a "bad logargument" when I start openser. Clues? Juan

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[OpenSER-Users] Openser Signaling and Routing
by Kaeman Chris 08 Sep '07

08 Sep '07

Hello, Is this possible? Call comes to openser for handling signaling, and is handed over to B2BUA for topology hiding and media handling. But the routing(selection of gw to terminate to the supplier) is used from openser LCR. Or after the call is given to B2BUA the routing is to be performed by the B2BUA. Thanks!! Regards, KChris --------------------------------- 5, 50, 500, 5000 - Store N number of mails in your inbox. Click here.

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[OpenSER-Users] Empty $du
by Andreas Sikkema 07 Sep '07

07 Sep '07

Hi, I'm trying to save the destination IP address of an INVITE message received and processed by OpenSER in the acc table. I've added an extra column and told the acc module about the new column. When I looked at the contents of the new column it was empty and adding an xlog() call to the config file also showed an empty $du pseudo variable. I know the OpenSER version I'm using is not the very latest, but it should be new enough to properly fill $du. I'm calling acc for received INVITE and BYE messages and showing $du with xlog() in the last line of INVITE processing for my testcase when the destination should be available. What am I doing wrong? -- Andreas Sikkema

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[OpenSER-Users] modparam("usrloc", "nat_bflag", 6) and setflag(6) doesn't update flags column
by Halomoan Chow 07 Sep '07

07 Sep '07

Dear all I followed tutorial from SER-Getting Start document for Mediaproxy to solve nat problem. I read on the Openser document http://www.openser.org/docs/modules/1.2.x/usrloc.html#AEN83 I have to use modparam("usrloc", "nat_bflag", 6) instead of modparam("registrar", "nat_flag", 6) And from the documentation, it said "When SIP clients attempt to REGISTER with our SIP proxy we need a way to tell the registrar module to store NAT information for any particular UA. We do this by using flag 6, which has been arbitrarily chosen (but defined earlier in the loadmodule parameter section). We could have specified another integer here, but flag 6 seems to be the accepted standard for nat_flag. If the nat_flag is set before calling the save() function to store contact information, SER will preserve the NAT contact information as well as set the flags column in the MySQL location table. By doing so, we can call lookup(location) when processing messages and flag 6 will be set for NATed clients." I have a NAT UA is detected as NAT-ed UA by Openser with below rule and do setflag(6); before it does save("location"); Why in mysql, location table, i see the flags column still 0 instead of 6 ? Did I miss something? Please help :( # $Id: nat-mediaproxy.cfg 51 2006-01-31 13:28:04Z /CN=Paul Hazlett/emailAddress= paul(a)onsip.org $ debug=3 fork=no log_stderror=yes #alias="halodns.powere2e.com" listen=x.x.x.x # INSERT YOUR IP ADDRESS HERE port=5060 children=4 dns=no rev_dns=no loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/mediaproxy.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/xlog.so" loadmodule "/usr/local/lib/openser/modules/mi_fifo.so" modparam("mi_fifo","fifo_name","/tmp/openser_fifo") modparam("auth_db|domain|uri_db|usrloc", "db_url", " mysql://openser:openserrw@localhost/openser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper|registrar", "received_avp", "$avp(i:42)") modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/openser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/openser/rtp-clients") modparam("usrloc", "db_mode", 2) modparam("usrloc", "nat_bflag", 6) <- set to 6 modparam("rr", "enable_full_lr", 1) . . . . route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- xlog("L_NOTICE","$Cxg Enter Route[2]: REGISTER $Cxx\n"); sl_send_reply("100", "Trying"); if (!search("^Contact:[ ]*\*") && client_nat_test("2")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("","subscriber")) { www_challenge("","0"); return; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); return; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; }

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[OpenSER-Users] uac_auth, how to sending request in failure route?
by Cseke Tamas 07 Sep '07

07 Sep '07

Hello, I 'd like to relay calls to another sip proxy, which requires authentication. But i can't forward the 407 reply to the caller, beacause they cant authenticate themselves, so i have to authenticate the call myself, without the caller knows anything about it. witch uac_auth, i can add the proxy-authorization header, but i can't send the new request to the proxy. i try this way: route[1] { t_on_failure("1"); rewritehostport("another.proxy:5060"); if (!t_relay()) { sl_reply_error(); } exit; } failure_route[1] { xlog("L_WARN", "FAILURE Route 3 AUTH required...trying..\n"); uac_auth(); if(t_newtran()) { xlog("L_WARN", "Creating new transaction successfully...\n"); } else { xlog("L_WARN", "Creating new transaction failed...\n"); }; t_relay(); } When the proxy challange authentication with 407, the call is passed to failure route properly, but t_relay failed with the following error: 0(27975) ERROR:tm:t_forward_nonack: no branch for forwarding 0(27975) ERROR:tm:w_t_relay: t_forward_nonack failed I think that's mean that the transaction doesn't exists, becacause it acknowleged alrealy, am i right? i think i have to t_newtran(), but it cannot be done in failure route, Can you suggest any method to send the request again with the credentials? Thanks any help, Tamas

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[OpenSER-Users] my small security breach REGISTER
by Marc LEURENT 07 Sep '07

07 Sep '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I have a security matter with my configuration (default one), it's possible to register using login/password and to set anything in the contact field. So if you have an account 106/password, it's possible to be 105 in the location database! How is it possible to deny that kind of matter..? Thanks Is it useful to use: method_filtering of the REGISTRAR module Or is it better to so something whith the values below and a compare function?? $ct - reference to body of contact header $ar - realm from Authorization or Proxy-Authorization header $au - username from Authorization or Proxy-Authorization header if ($ct != $au@$ar) { sl_send_reply("403", "User and login must be the same"); }; Best Regards, Marc LEURENT # U 82.127.0.79:1045 -> 88.191.45.91:5060 REGISTER sip:sd-7501.dedibox.fr;user=phone SIP/2.0. Via: SIP/2.0/UDP 82.127.0.79:1046;branch=z9hG4bK5808036470869310420. From: <sip:105@sd-7501.dedibox.fr:5060;user=phone>;tag=c0a80101-38c0e7. To: <sip:105@sd-7501.dedibox.fr:5060;user=phone>. Call-ID: 29eb6e9-c0a80101-5-17(a)192.168.95.70. CSeq: 90 REGISTER. Max-Forwards: 70. Expires: 3600. Contact: <sip:105@82.127.0.79:1046;user=phone>. Authorization: Digest username="106", realm="sd-7501.dedibox.fr", nonce="46dfceb402cad04812873b855bc50ea65aa99ed5", uri="sip:sd-7501.dedibox.fr", response="7dca83fd358a9aea3a963f4a71ea5c9e", algorithm=MD5, qop=auth, cnonce="38c102", nc=00000001. User-Agent: THOMSON ST2030 hw0 fw1.56 00-0E-50-4E-AF-C4. Allow-Events: refer,dialog,message-summary,check-sync,talk,hold. Content-Length: 0. . AOR:: 105 Contact:: sip:105@82.127.0.79:1046;user=phone Q= Expires:: 194 Callid:: 29eb6e9-c0a80101-5-17(a)192.168.95.70 Cseq:: 92 User-agent:: THOMSON ST2030 hw0 fw1.56 00-0E-50-4E-AF-C4 Received:: sip:82.127.0.79:1045 State:: CS_SYNC Flags:: 0 Cflag:: 192 Socket:: udp:88.191.45.91:5060 Methods:: 4294967295 -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFG39AIqjpLE0HiOBYRAiUKAJ9Ilv+Zpbzw89tqWgwmHyVjU/DXugCgjEh8 5XQKEAeiF/L4RWszGC2/yzQ= =SXE9 -----END PGP SIGNATURE-----

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[OpenSER-Users] is_uri_host_local() returns 1 or 4294967295 (always TRUE !!!)
by Iñaki Baz Castillo 07 Sep '07

07 Sep '07

Hi, in order to minimize the SQL queries for functions "is_uri_host_local()" and "is_from_local()" I want to exec they just one time and store the result in a variable, but the result is unexpected: $var(is_uri_host_local) = is_uri_host_local(); xlog("------------is_uri_host_local = $var(is_uri_host_local)\n"); So if the URI is local then the result is 1. But if the URI is not local then result is 4294967295. ¿?¿?¿?¿ Obviously it makes me difficult for me to do: if ($var(is_uri_host_local)) because always is true and I need to convert the values and so. Why "is_uri_host_local()" doesn't return -1 in case of being not local domain? Or maybe the issue is when I store the result in a variable? but why? Thanks. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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[OpenSER-Users] Transaction Deletion
by Brandon Armstead 07 Sep '07

07 Sep '07

Hello Users, I have a weird scenario where the PSTN is responding with 200 OK, however Openser Never forwards on the ACK, after several retransmissions of 200OK to Openser, PSTN responds with BYE, anyone know what might be going on? I've played with the timers to no avail, any help is appreciated, thank you! The log below is as follows: 375 Sep 6 12:18:10 openser /usr/local/sbin/openser[17385]: DEBUG: timer routine:1,tl=0x2b78e4e73588 next=(nil), timeout=261 376 Sep 6 12:18:11 openser /usr/local/sbin/openser[17385]: DEBUG: timer routine:2,tl=0x2b78e4e733b8 next=(nil), timeout=262 377 Sep 6 12:18:11 openser /usr/local/sbin/openser[17385]: DEBUG: wait_handler : removing 0x2b78e4e73338 from table 378 Sep 6 12:18:11 openser /usr/local/sbin/openser[17385]: DEBUG: delete transaction 0x2b78e4e73338 379 Sep 6 12:18:11 openser /usr/local/sbin/openser[17385]: DEBUG: wait_handler : done 380 Sep 6 12:18:12 openser /usr/local/sbin/openser[17345]: DEBUG:parse_to:end of header reached, state=10 381 Sep 6 12:18:12 openser /usr/local/sbin/openser[17345]: DEBUG: get_hdr_field: <To> [26]; uri=[sip:openser] 382 Sep 6 12:18:12 openser /usr/local/sbin/openser[17345]: DEBUG: to body [<sip:openser> ] 383 Sep 6 12:18:12 openser /usr/local/sbin/openser[17345]: DEBUG: get_hdr_body : content_length=0 384 Sep 6 12:18:12 openser /usr/local/sbin/openser[17345]: DEBUG:destroy_avp_list: destroying list (nil) 385 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG:forward_reply: found module tm, passing reply to it 386 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG: t_check: start=0xffffffffffffffff 387 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG: add_param: tag=gK0288a919 388 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG:parse_to:end of header reached, state=29 389 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG: get_hdr_field: <To> [53]; uri=[sip:17144828306@openser] 390 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG: to body [<sip:17144828306@openser>] 391 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG: t_reply_matching: hash 53903 label 1526431020 branch 0 392 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG: t_reply_matching: no matching transaction exists 393 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG: t_reply_matching: failure to match a transaction 394 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG: t_check: end=(nil) It seems like its not matching the transaction, because a timer is removing the transaction? Any ideas? Thanks! -- Brandon Armstead

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[OpenSER-Users] Dispatcher + NAT
by kokoska rokoska 07 Sep '07

07 Sep '07

Hi all, I'm currently succesfully running OpenSER, but user population grow up and I'm affraid I need laod balancing :-) The projected scenario is: USERS <-> (NAT) <-> LB <-> P1 <-> GWs <-> P2 <-> R1 <-> R2 Where LB is Load Balancer (OpenSER with Dispatcher & Path module), and P1-P2 is proxies (OpenSER) and R1-R2 registrars (OpenSER) And the problem I need to solve is NAT handling. I can correctly process registrations with help of Path module on LB and enabling of using it on R1,R2. But, what should I do with calls? My idea is to process NAT handling on P1-P2, but I don't know how. Both proxies are unable (I mean) detect correct user IP address. Could you tell me, please, what to do? Do NAT handling on LB? It is unnecessary added load to LB... Thanx in advance! kokoska.rokoska

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[OpenSER-Users] is_from_user_enum() errors
by Eric Hiller 07 Sep '07

07 Sep '07

Hello all, I have a rather strange issue, I cannot use "is_from_user_enum()" without first using the variable $fu somewhere; or else I get the error "0(5858) get_parsed_from(): From header is not properly parsed". Originally I had an xlog there and it all worked fine, but when I took it away I started getting the error. Now I just have " $var(bleh) = $fu;" there and it makes it work fine. I had a forum post @ http://www.voipuser.org/forum_topic_10875.html that shows my entire config. Thanks for any help, Eric _________________________________________________________________ Discover the new Windows Vista http://search.msn.com/results.aspx?q=windows+vista&mkt=en-US&form=QBRE

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[OpenSER-Users] CDRTool Price is all 0.0000
by Chahn Kim 06 Sep '07

06 Sep '07

Hi all. CDRTool shows all call records with price o.oooo. Syslog logs following: Error: Cannot find rates for callid=, domain=koreanet.us, destination=1847, profile=441, app=audio Can anyone help? Thanks in advance John Chahn Kim email: johnk(a)koreanet.us

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[OpenSER-Users] Limiting maximum simultaneous calls per user
by Sebastian Mangelkramer 06 Sep '07

06 Sep '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Dear list, actually I'm searching for a solution to limit the max. simultaneous calls per user. I`ve searched the internet and list-archives for possible solutions. But until now i haven`t found a solution. What i exactly need is to define the max. simultaneous calls per user (e.g. 2,4,6,8,...). I know, that openSER is transaction-stateful, not call-stateful but is there a way to monitor the number of RTP-Sessions per user from Mediaproxy and set the "max_calls"-variable via an AVP? My setup is openSER 1.2.1-notls + Mediaproxy Has anyone implemented such a solution or anyone know such one? Perhaps i had to place a kind of SBC/B2BUA in front of my SIP-Proxy? Many thanks in advance, kind regards Sebastian -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (Darwin) iD8DBQFG3tU87EKQHe5FugIRAi8LAJ0eOVGO10FDjZcTMc7q24TXdfElDgCdHM0g l2ZanHEt/eX9yZTw/jVQxGc= =7Mz4 -----END PGP SIGNATURE-----

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[OpenSER-Users] OpenXCAP install and config problem?
by KevinKinnan 06 Sep '07

06 Sep '07

I am not sure whether or not my procedure is right as server doesnt work at the moment, but I will keep working on it. Here is the Fedora installation [1] Fedora 6 linux install; [2] pre-dependencies package; Because I installed OpenIMSCore and Openser Presence module before, so some dependencies was already installed. In this case I just installed clearsilver from Add&Remove Package, and copy lxml folder(1.3.4 download from lxml site) to /usr/lib/python2.5/site-packages/. [3] install OpenXCAP, followed installation guide/Debain unstable installation By now, that's it. Coz no daemon can be run from /etc/init.d/openxcap. Also, I haven't modified anything in the config.ini and openxcap files Cheers, Kevin _________________________________________________________________ MSN 中文网，最新时尚生活资讯，白领聚集门户。 http://cn.msn.com

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[OpenSER-Users] OpenXCAP install and config problem?
by KevinKinnan 06 Sep '07

06 Sep '07

I am trying to install openXCAP on the Fedora 6. Here is my installation procedure: python ./setup.py install; copy config.init to /etc/openserxcap/ copy openxcap to /etc/init.d/ run /etc/init.d/openxcap start update openser config file modparam("presence_xml", "force_active", 0); modparam("presence_xml", "integrated_xcap_server", 0); I havent change anything in config.ini and openxcap it shows some errors "start-stop-daemon command not found". It seems that the openxcap.ini is writed for Debian based Linux as start-stop-daemon is for Debain. How can I run the start file on a red hat based linux? Do I need to do any other modification in some files? Thank You! Kevin _________________________________________________________________ MSN 中文网，最新时尚生活资讯，白领聚集门户。 http://cn.msn.com

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[OpenSER-Users] Please help in Openser behind NAT design
by Halomoan Chow 06 Sep '07

06 Sep '07

Hi All I'm a new member and trying to use Openser as my VOIP "router" I plan to make following configuration: UA <---> Openser <---> Firewall/NAT <--INTERNET--> Firewall/NAT <---> UA a.a.a.a/24 x.x.x.x y.y.y.y b.b.b.b/24 a.a.a.a and b.b.b.b is internal IP not internet route-able x.x.x.x and y.y.y.y is Firewall/NAT external IP number I know there is mediaproxy and RTPProxy to solve the UA b.b.b.b that behind NAT but I don't know if my decision to put Openser behind NAT is correct or not. How to archive the above diagram with Openser ? If above Openser location is wrong, where should the Openser reside in the diagram? Install in Firewall/NAT or in INTERNET (dedicated server) Thank you for your help and sorry if my question is too newbie Halomoan

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Re: [Serusers] Can't make a single call. Help me please.
by Mike Trest - Personal 06 Sep '07

06 Sep '07

As a long time VoIP and Internet engineer, we have all seen various port-blocking activities at different levels. These are almost always done to attain economic or political goals. (Remember PANAMA and SINGAPORE). Seldom is it ever for security reasons or for conservation of resources. It is the first question I ask whenever I start to assist a gateway operator who is testing via a cable operator. Therefore, it is understandable that we would be groping in the dark without that bit of bank ground information. Perhaps we should add a guideline note to our FAQ which recommends placing a sniffer/etherial between the originating sip device and the IP network as well as an ethereal/tcpdump on the SER end. ..mike.. At 01:34 AM 9/3/2007, Harmeet Singh wrote: >Dear Greg > >I tried ekiga, brujula, and couple other services on my internet >yesterday. None worked. Same problem - REGISTER is OK but can't make >calls. Then after searching google again, I found that my ISP (Rogers >Cable) is dropping/blocking the INVITE messages. That's why I was able >to register and not make any calls. Then I found on the internet about >a workaround, which is to not use 5060 port. So, I just tried that. To >my surprise, Any port other than 5060 just works! It beats me really. > >Thanks again for your help. I really appreciate it. > >Harmeet --- avast! Antivirus: Outbound message clean. Virus Database (VPS): 000771-1, 09/01/2007 Tested on: 9/6/2007 10:25:41 AM avast! - copyright (c) 1988-2007 ALWIL Software. http://www.avast.com

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[OpenSER-Users] openSER rtpproxy video call - please help
by Volodymyr Fedyk 06 Sep '07

06 Sep '07

Hi, I've tried this openser configs from http://www.voip-info.org/wiki/view/OpenSER+v1.2.x+and+RTPProxy and generated by http://sipwise.com/wizard/ and I always face with the next issue: 2 UAs are behind NAT; each UA could call another; IM works; Audio works; but looks-like video streams are handled by openSER and rtpproxy incorrectly, and I'm unable to make a video call to any ua, - I'm using x-lit sp phone, and it shows "Could not start video" on one side, and "No matching video codec" on the other side... Has anybody faced with such issues later or has any ideas ? best regards, Dorin

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[OpenSER-Users] Almost the End / End calls OpenSER + mediaproxy + freeradius
by Marc LEURENT 06 Sep '07

06 Sep '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello evrybody! I have one little problem... When a call badly ends (like hard disconnection or killall mysoftphone) mediaproxy has a timeout of 10sec, after the timeout, it sends a message to freeradius to stop accounting (so CDRTool billing too), but DON'T say to openser to stop the call, so the connection to my SIP/PSTN provider is still up (and its billing too) Is there any way to say to openser to send BYEs when mediaproxy detects the end of a badly ended call? Best Regards! PS: J-20 before VoN Europe 2007 - Rome Extract of the INSTALL file: """If the media session timeouts (there is no BYE that closes the session), MediaProxy will contact back the proxy dispatcher and pass the session media information to be logged. Using this architecture, only the proxy dispatcher needs to have direct access to the database. This addresses the issue of the MediaProxy servers being distributed and not having access to the database.""" -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFG3rpwqjpLE0HiOBYRAmbmAJ9UiDDi6UWYg4v+KLR020MwG/ks2ACggg8w aDQX87YSNxjEvy5vcSSeWto= =OCrH -----END PGP SIGNATURE-----

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[OpenSER-Users] openser as proxy relay
by Asim Riaz 06 Sep '07

06 Sep '07

Hi, Could any body give me some idea to what need to be done in openser for relaying all SIP packets to another third party SIP registrar and Gateway. Scenario XYZ SIP Reg------------------openser----------------------UA Thanks in advance

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[Serusers] serweb login problem
by Harmeet Singh 06 Sep '07

06 Sep '07

Hi I am facing this strange problem, if anyone can help me on this I will appreciate. Here is my configuration - Fedora 7, Apache 2.2.4, php 5.2.2, ser/serweb 0.9.6 When I go to localhost/serweb, enter login info, and submit, I get a download window pop-up asking me to save the php file. Shouldn't Apache be catching this and render the php to html? The login page itself is a php page and that renders ok. So, Apache is configured correctly for php rendering. What else am I missing here? Any help is appreciated. Thanks Harmeet

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[OpenSER-Users] XCAP setting problem?
by Adrian Georgescu 05 Sep '07

05 Sep '07

You may obtain the XCAP server and the configurations from http:// openxcap.org The latest OpenSER svn version works with it and the sample configuration. Adrian >>>>> I don't know whether or not the xcap_server is set correct. There is no external XCAP server integrated. BTW, I know presence support XCAP function, does it means this module has already included a XCAP server? Thank You! Kevin

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[OpenSER-Users] XCAP setting problem?
by KevinKinnan 05 Sep '07

05 Sep '07

The PUBLISH request works when using the following configurations(no XCAP): loadmodule "sl.so" loadmodule "mysql.so" loadmodule "presence.so" loadmodule "presence_xml.so" modparam("presence_xml", "db_url", "mysql://openser:openserrw@localhost:3306/openser") modparam("presence_xml", "force_active", 1) modparam("presence_xml", "pidf_manipulation", 1) It doesn't if starting XCAP by updating these, the server can run properly, but cann't handle PUBLISH. loadmodule "xcap_client.so" modparam("presence_xml", "xcap_table", "xcap_xml") modparam("presence_xml", "force_active", 0) modparam("presence_xml", "xcap_server", "localhost:5065") I don't know whether or not the xcap_server is set correct. There is no external XCAP server integrated. BTW, I know presence support XCAP function, does it means this module has already included a XCAP server? Thank You! Kevin _________________________________________________________________ Windows Live Spaces 中最年轻的成员！ http://miaomiaogarden2007.spaces.live.com/

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[OpenSER-Users] mi_xmlrpc compilation issues
by Purna Chandar M 05 Sep '07

05 Sep '07

Hi, Im trying to compile mi_xmlrpc module in openser but could not succeed. The following errors are displayed mi_xmlrpc.c: In function 'xmlrpc_process': mi_xmlrpc.c:161: error: 'xmlrpc_server_abyss_rpc2_handler' undeclared (first use in this function) mi_xmlrpc.c:161: error: (Each undeclared identifier is reported only once mi_xmlrpc.c:161: error: for each function it appears in.) mi_xmlrpc.c:167: error: 'xmlrpc_server_abyss_default_handler' undeclared (first use in this function) Im using Suse 10.1; The libraries xmlrpc-c-0.9.10-36, libxml-1.8.17-385, libxml2-devel-2.6.23-13, xmlrpc-c-devel-0.9.10-36 are properly installed. Few discussions on the mailing list concluded that the module mi_xmlrpc compiles well on Debian without any dependency issues. Is it true?? Can somebody tell me whether Im missing any dependencies?? Please let me know how to successfully compile this module. Thanks and regards, Purna Chandar M

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Re: [OpenSER-Users] mi_xmlrpc compilation issues
by Kriti Wadhwa 05 Sep '07

05 Sep '07

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[OpenSER-Users] NatPing stop working
by Marcello Lupo 05 Sep '07

05 Sep '07

Hi to all, i have openser 1.2.1 active from some months.. Suddenly on yesterday the natping stop to work. I have 140 contacts in user location almost all of them natted. There is some limit to user in location table at which the ping stops or a particular server load? There is a way to have something in the logs about it?? I didn't found anything in the logs. This is my config: modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) With a restart of openser all come back to work but i'm afraid if it will happen again? Thanks for help, Bye, Marcello

2 14

[OpenSER-Users] Opnser rejecting BYE packet
by Dan 05 Sep '07

05 Sep '07

Scenario 1: U 212.34.169.220:5060 -> 78.45.146.34:5060 BYE sip:test@78.45.146.34 SIP/2.0. Route: <sip:78.45.146.34;ftag=aa313a3c;lr>. CSeq: 2 BYE. Via: SIP/2.0/UDP 212.34.169.220:5060. From: <sip:1217773456@78.45.146.34>;tag=03095807065938997270316869. Call-ID: 8246bf729f62a754. To: 58.68.96.11<sip:test@78.45.146.34>;tag=aa313a3c. Content-Length: 0. # U 78.45.146.34:5060 -> 212.34.169.220:5060 SIP/2.0 404 User Not Found. CSeq: 2 BYE. Via: SIP/2.0/UDP 212.34.169.220:5060. From: <sip:1217773456@78.45.146.34>;tag=03095807065938997270316869. Call-ID: 8246bf729f62a754. To: 58.68.96.11<sip:test@78.45.146.34>;tag=aa313a3c. Server: OpenSer (1.0.1 (i386/linux)). Content-Length: 0. Warning: 392 78.45.146.34:5060 "Noisy feedback tells: pid=24525 req_src_ip=78.45.146.34 req_src_port=5060 in_uri=sip:78.45.146.34;ftag=aa313a3c;lr out_uri=sip:78.45.146.34;ftag=aa313a3c;lr via_cnt==2". Scenario 2: U 59.176.78.110:50496 -> 210.68.65.78:5060 BYE sip:210.68.65.78 SIP/2.0. Route: <sip:@209.67.171.10;transport=udp>. Via: SIP/2.0/UDP 59.176.78.110:50496. From: sip:user1@210.68.65.78;tag=4154163258. To: <sip:12818495027@210.68.65.78;user=phone>;tag=3005360702044301069847989. Call-ID: 4202237689(a)192.168.1.156. CSeq: 3 BYE. User-Agent: Cisco ATA v2.15 ata18x (020927a). Content-Length: 0. . # U 210.68.65.78:5060 -> 59.176.78.110:50496 SIP/2.0 478 Unresolvable destination (478/TM). Via: SIP/2.0/UDP 59.176.78.110:50496. From: sip:user1@210.68.65.78;tag=4154163258. To: <sip:12818495027@210.68.65.78;user=phone>;tag=3005360702044301069847989. Call-ID: 4202237689(a)192.168.1.156. CSeq: 3 BYE. Server: OpenSer (1.0.1 (i386/linux)). Content-Length: 0. Warning: 392 210.68.65.78:5060 "Noisy feedback tells: pid=2144 req_src_ip=59.176.78.110 req_src_port=50496 in_uri=sip:210.68.65.78. In both cases BYE is not getting accounted for. Can anyone help me on this ?? Regards Dan

4 19

Re: [OpenSER-Users] Broken "BYE" returned from Asterisk on TLS implementation ?
by David Loh 05 Sep '07

05 Sep '07

Hi, Arrggghh .. that's one of my attempts to eliminate the broken "BYE" problem... that's ngrep was captured when I set "modparam("rr", "enable_double_rr", "0");", I've paste another ngrep to http://pastebin.ca/674450, this time the double RR header is enabled. And I've posted my .cfg to http://pastebin.ca/Nx0Ss4Fd (key to decrypt the post is "openser"). Even though double RR header is enabled, but for BYE it's still doesn't process properly :( For the .cfg file line #130 onward, I did tried t_relay, forward and force_send_socket, but none of this will do the trick (force_send_socket was complaining TLS error due to missing certificate (?) ) Would appreciate if anyone could enlighten me why is this happen ? Thanks, David Loh Klaus Darilion wrote: > But the INVITE you posted at http://pastebin.ca/673392 also has only > one Record-Route header. > > regards > klaus > > David Loh schrieb: >> Hi, >> >> Yea, OpenSER proxy was add 2 record-route header for the INVITE/ACK >> ...but when asterisk disconnected the call and send BYE back to OpenSER, >> the TLS RR header wasn't present, the only 2 RR header was >> "SIP/2.0/UDP <OpenSER_IP>" and "SIP/2.0/UDP <Client_WAN_IP>" .... >> I'm puzzled ... is there any command to 'fix' this? >> >> >> Regards, >> David Loh >> >> Klaus Darilion wrote: >>> The openser proxy should add 2 record-route header (TLS and UDP = >>> double record route). This is why it does not work. >>> >>> regards >>> klaus >>> >>> David Loh schrieb: >>>> Hi All, >>>> >>>> Greeting. >>>> >>>> I've been struggle with OpenSER TLS implementation for more than a >>>> week, since I've ported from UDP to TLS, everything work fine >>>> except the "BYE" request from Asterisk (loose route), my >>>> implementation was something like below: >>>> >>>> [Client] --> [Router] --> [Internet] --> [SIP] --> [Asterisk] >>>> >>>> My OpenSER.cfg already configured to listen on two port which is :- >>>> "tls:eth0:5061" and "udp:eth0:5060", client make p2p or PSTN (or >>>> even voicemail) having no problem, >>>> but when the callee disconnect the call, caller will never get hang >>>> up :( >>>> >>>> I've attached my ethereal trace/ngrep to pastebin, >>>> http://pastebin.ca/673392 >>>> >>>> Wondering if anyone can help me with the broken "BYE" that returned >>>> from Asterisk ? >>>> Line #131, supposedly this line should have contain 2 Via header, >>>> one was "SIP/2.0/UDP" and another "SIP/2.0/TLS", >>>> but somehow the TLS via header was gone !! (compare to previous ACK >>>> (Line #117) /INVITE (Line #51). >>>> Due to the missing TLS via header, OpenSER log file was complaining >>>> "protocol/port mis-match". >>>> >>>> The last BYE request (Line #256) is actually firing from Client, >>>> which contain the "TLS" via. >>>> >>>> >>>> I've even tried "force_send_socket" to port 5061 (instead of 5060) >>>> from loose route, but it complaining TLS certificate error, >>>> since Asterisk doesn't support TLS natively, I've no clue why is >>>> the ACK/INVITE/CANCEL work but not BYE. >>>> if (loose_route) { >>>> .... >>>> if(is_method("BYE")) { force_send_socket(IP:5061); } >>>> } >>>> >>>> >>>> Has any one gone through of this kinda OpenSER over TLS + Asterisk >>>> setup, >>>> I'm really appreciate if you can share your experience with me, or >>>> pin point what's the mistakes I made here. >>>> >>>> Thanks in advance. >>>> >>>> Regards, >>>> David Loh >>>> >>>> >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users(a)openser.org >>>> http://openser.org/cgi-bin/mailman/listinfo/users >>> >>> >> >> > >

2 5

[OpenSER-Users] OpenSER with PostgreSQL
by Cliff Flood 04 Sep '07

04 Sep '07

Hi, I have an Ubuntu 6.06 install with PostgreSQL 8.1 and OpenSER 1.2.2 which I am testing at the moment. I have tested OpenSER with MySQL and it worked fine but when when I want to switch to PostgreSQL I'm having some difficulty. I have used the postgressqldb.sh script to initialise the database and it has completed without error. When I start OpenSER I get the following lines in my PostgreSQL logs: LOG: connection received: host=127.0.0.1 port=37610 LOG: connection authorized: user=openser database=openser LOG: SSL SYSCALL error: EOF detected LOG: could not receive data from client: Connection reset by peer LOG: unexpected EOF on client connection I have looked at OpenSER's logs with debugging turned on but I don't see anything that looks like an error but I would be happy to provide them. `openserctl moni` returns: ERROR: Error opening OpenSER's FIFO /tmp/openser_fifo ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo")' in your config ERROR: and also have loaded the mi_fifo module. My openser.cfg looks like this: debug=9 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) children=4 listen=192.168.20.151 dns=no rev_dns=no port=5060 mpath="/usr/local/lib/openser/modules/" loadmodule "mysql.so" loadmodule "postgres.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "mi_fifo.so" loadmodule "auth.so" loadmodule "auth_db.so" modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "postgres://openser:openserrw@localhost/openser") modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "db_url", "postgres://openser:openserrw@localhost/openser") modparam("auth_db", "password_column", "password") modparam("rr", "enable_full_lr", 1) route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS #if(uri=~"@tls_domain1.net") { # t_relay("tls:domain1.net"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay("tls:domain2.net"); # exit; #} route(1); }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication # if (!www_authorize("host.doman.tld", "subscriber")) { # www_challenge("host.domain.tld", "0"); # exit; # }; save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; 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; } Any and all help appreciated. -- Cliff Flood Systems Administrator Jazinga Inc.

2 2

[OpenSER-Users] Strange Provider SIP - DSP fields in trying
by Marc LEURENT 04 Sep '07

04 Sep '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I have a SIP provider: telecomitalia that use a cirpack! The problem is that the cirpack sends TRYING and RINGING packets with SDP fields! (I don't know if it follows RFC...) When a natted contact place a call, I use a rtp_proxy, but only the 200ok with session description is modified. Both TRYING/RINGING with session description Connection Address/port are not replaced! OpenWengo understand the modification and change the IP where it sends RTP from the one in TRYING/RINGING SDP to the one in 200ok (proxy address because of force_rtp_proxy()) But other softphones like Twinkle don't understand it and still send RTP to the first IP! Do you know how I can delete or modify TRYING/RINGING SDP? Thanks Below an example of a RINGING with SDP # U 212.129.6.65:5060 -> 88.191.45.91:5060 SIP/2.0 180 Ringing. Allow: UPDATE,REFER. Call-ID: 866489712(a)192.168.95.47. Contact: <sip:212.129.6.65:5060>. Content-Type: application/sdp. CSeq: 21 INVITE. From: "Marc LEURENT" <sip:mleurent@sd-7501.dedibox.fr>;tag=542924903. Record-Route: <sip:88.191.45.91;lr;ftag=542924903>. Server: Cirpack/v4.39a (gw_sip). To: <sip:0614730696@sd-7501.dedibox.fr>;tag=01-07627-0003a50a-10152ba67. Via: SIP/2.0/UDP 88.191.45.91;received=88.191.45.91;branch=z9hG4bK2cf4.8d35c996.0,SIP/2.0/UDP 192.168.95.47:5060;received=81.57.0.22;rport=64726;branch=z9hG4bK20204963;xxx-nat-type=sym. Content-Length: 303. . v=0. o=cp10 118890098142 118890098144 IN IP4 212.129.46.35. s=SIP Call. c=IN IP4 212.129.47.194. t=0 0. m=audio 32896 RTP/AVP 8. b=AS:64. a=rtpmap:8 PCMA/8000/1. a=ptime:10. a=sendrecv. m=video 65534 RTP/AVP 34 31. a=rtpmap:34 H263/90000/1. a=fmtp:34 . a=rtpmap:31 H261/90000/1. a=fmtp:31 . a=inactive. -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFG3TdoqjpLE0HiOBYRAjnkAKCgzB4p89oTKhxuvZcLlklTme9w3ACeP+TW FI+LkwgbaJ1XJTIm3FUSMTg= =Cp53 -----END PGP SIGNATURE-----

2 4

[OpenSER-Users] MediaProxy 1.9.0
by Mike O'Connor 04 Sep '07

04 Sep '07

Hi Guys I upgraded my mediaproxy to 1.9.0 and found out sometime later that the data usage update at call end seems to have stopped. I'm using direct mysql (not radius) for the end of call updates. No error message in the logs and a manual test of mysql show nothing wrong. Any idea. Mike

1 0

RE: [OpenSER-Users] Opnser rejecting BYE packet
by Dan 04 Sep '07

04 Sep '07

Hi Ravi, Did what you suggested, but still same error. Regards Dan _____ From: raviprakash sunkara [mailto:sunkara.raviprakash.feb14@gmail.com] Sent: Tuesday, September 04, 2007 3:51 PM To: Dan Subject: Re: [OpenSER-Users] Opnser rejecting BYE packet Hello Dan In yur openser.cfg In Below this this Section route { if(loose_route()) { # mention here BYE section also with fix_nated_contact(); } if(method=="CANCEL" ) { } else if (method=="INVITE") { } else if (method=="REGISTER") { } else if ( method=="BYE" ) # here u missing in this section route(1); exit; } On 9/4/07, Dan <fiedler.dan(a)gmail.com> wrote: Where exactly? Regards Dan _____ From: raviprakash sunkara [mailto:sunkara.raviprakash.feb14@gmail.com] Sent: Tuesday, September 04, 2007 2:27 PM To: Dan Subject: Re: [OpenSER-Users] Opnser rejecting BYE packet Hi Dan Problem in ur openser,cfg file On 9/4/07, Dan <fiedler.dan(a)gmail.com> wrote: My Config File: listen=72.65.56.136 #alias=0.0.0.0 alias=dstpl.com check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=6 fifo="/tmp/openser_fifo" fifo_db_url="postgres://openser:openserrw@localhost/openser" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/openser/modules/postgres.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/uac.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/textops.so" # Uncomment this if you want diest authentication mysql.so must be loaded ! loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_radius.so" loadmodule "/usr/local/lib/openser/modules/acc.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" loadmodule "/usr/local/lib/openser/modules/avp_radius.so" loadmodule "/usr/local/lib/openser/modules/group_radius.so" # loadmodule "/usr/local/lib/openser/modules/lcr.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" #loadmodule "/usr/local/lib/openser/modules/mediaproxy.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/permissions.so" modparam("auth_db|permissions|uri_db|usrloc","db_url", "postgres://openser:openserrw@localhost/openser") # ----------------- setting module-specific parameters --------------- # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) modparam("maxfwd", "max_limit", 32) modparam("auth_radius","radius_config","/etc/radiusclient-ng/radiusclient.co nf") modparam("auth_radius", "service_type", 15) modparam("avp_radius", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") modparam("group_radius", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") # -- Accounting param -- modparam("acc", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") modparam("acc", "service_type", 16) modparam("acc", "radius_flag", 1) modparam("acc", "log_level", 1) # modparam("acc", "radius_missed_flag", 1) modparam("acc", "report_ack", 0) # 1 reporta dos starts en acc modparam("acc", "radius_extra", "Sip-Source-IP-Address=$si") # -- auth params -- # Uncomment if you are using auth module #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) #modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy # modparam("rr", "enable_full_lr", 1) # modparam("avpops", "avp_aliases", "c_uri=i:888; s_ip=i:999" ) modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") #modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if((method=="INVITE") || (method=="ACK") || (method=="BYE") || (method=="CANCEL")) { setflag(1); } 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; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method!="REGISTER") { record_route(); }; if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); } # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); }; force_rtp_proxy("l"); }; route(1); exit; }; if (!uri==myself) { route(5); route(1); exit; }; if (uri==myself) { if (method=="CANCEL") { route(1); exit; } else if (method=="INVITE") { route(3); exit; } else if (method=="REGISTER") { route(2); exit; } else if (method=="BYE") { route(1); exit; }; lookup("aliases"); if (uri!=myself) { route(5); route(1); exit; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); exit; }; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" && isflagset(6)) { unforce_rtp_proxy(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- if (!search("^Contact:\ +\*") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; sl_send_reply("100", "Trying"); if (!radius_www_authorize(" <http://72.65.56.136> 72.65.56.136")) { www_challenge("72.65.56.136", "0"); exit; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); exit; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } onreply_route[1] { if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if (!search("^Content-Length:\ +0")) { force_rtp_proxy(); }; }; if (nat_uac_test("1")) { fix_nated_contact(); }; } route[5] { # ----------------------------------------------------------------- # RTP Proxy Enabler # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); }; } route[3] { # ------------------------------------------------------------------------ # INVITE Handler # ------------------------------------------------------------------------ if (nat_uac_test("19")) { setflag(6); } lookup("aliases"); if (uri!=myself) { route(5); route(1); exit; }; if (!allow_trusted() && uri=~"^sip:[0-9]*@.*") { if (!radius_www_authorize(" 72.65.56.136")) { www_challenge("72.65.56.136", "0"); exit; }; }; if (uri=~"^sip:[0-9]*@.*"){ route(11); exit; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); exit; }; if (method=="CANCEL") { route(1); exit; }; route(5); route(1); } route[11]{ # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- strip(0); prefix(""); rewritehostport(" 206.234.108.104:5060 <http://206.234.108.104:5060> "); route(5); route(1); } -----Original Message----- From: users-bounces(a)openser.org [mailto: <mailto:users-bounces@openser.org> users-bounces(a)openser.org] On Behalf Of Dan-Cristian Bogos Sent: Tuesday, September 04, 2007 1:30 PM To: users openser.org Subject: Re: [OpenSER-Users] Opnser rejecting BYE packet Hello Inaki, yes, it is almost sure that it does (except receiving this answer from remote party). This is the reason all the people place loose-route block before location query and this is why I have assumed that it does not match the loose-routing and ends up to location query automatically. DanB On 9/4/07, Marc LEURENT <lftsy(a)free.fr> wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Like Dan said, > send us your configuration, you may have forbidden something! > > Dan a écrit : > > *_Scenario 1:_* > > > > > > > > > > > > U 212.34.169.220:5060 -> 78.45.146.34:5060 > > > > BYE sip:test@78.45.146.34 SIP/2.0. > > > > Route: <sip:78.45.146.34;ftag=aa313a3c;lr>. > > > > CSeq: 2 BYE. > > > > Via: SIP/2.0/UDP 212.34.169.220:5060. > > > > From: <sip:1217773456@78.45.146.34>;tag=03095807065938997270316869. > > > > Call-ID: 8246bf729f62a754. > > > > To: 58.68.96.11< sip:test@78.45.146.34 <mailto:sip:test@78.45.146.34> >;tag=aa313a3c. > > > > Content-Length: 0. > > > > > > > > # > > > > U 78.45.146.34:5060 -> 212.34.169.220:5060 > > > > SIP/2.0 404 User Not Found. > > > > CSeq: 2 BYE. > > > > Via: SIP/2.0/UDP 212.34.169.220:5060. > > > > From: < <mailto:sip:1217773456@78.45.146.34> sip:1217773456@78.45.146.34>;tag=03095807065938997270316869. > > > > Call-ID: 8246bf729f62a754. > > > > To: 58.68.96.11< sip:test@78.45.146.34 <mailto:sip:test@78.45.146.34> >;tag=aa313a3c. > > > > Server: OpenSer (1.0.1 (i386/linux)). > > > > Content-Length: 0. > > > > Warning: 392 78.45.146.34:5060 "Noisy feedback tells: pid=24525 > > req_src_ip=78.45.146.34 req_src_port=5060 > > in_uri=sip:78.45.146.34;ftag=aa313a3c;lr > > out_uri=sip:78.45.146.34;ftag=aa313a3c;lr via_cnt==2". > > > > > > > > > > > > *_Scenario 2:_* > > > > > > > > U 59.176.78.110:50496 -> 210.68.65.78:5060 > > > > BYE sip:210.68.65.78 <http://210.68.65.78> SIP/2.0. > > > > Route: <sip:@209.67.171.10;transport=udp>. > > > > Via: SIP/2.0/UDP 59.176.78.110:50496 . > > > > From: sip:user1@210.68.65.78;tag=4154163258. > > > > To: > > <sip:12818495027@210.68.65.78 ;user=phone>;tag=3005360702044301069847989. > > > > Call-ID: 4202237689(a)192.168.1.156. > > > > CSeq: 3 BYE. > > > > User-Agent: Cisco ATA v2.15 ata18x (020927a). > > > > Content-Length: 0. > > > > . > > > > > > > > # > > > > U 210.68.65.78:5060 -> 59.176.78.110:50496 > > > > SIP/2.0 478 Unresolvable destination (478/TM). > > > > Via: SIP/2.0/UDP 59.176.78.110:50496. > > > > From: sip:user1@210.68.65.78;tag=4154163258. > > > > To: > > < <mailto:sip:12818495027@210.68.65.78> sip:12818495027@210.68.65.78;user=phone>;tag=3005360702044301069847989. > > > > Call-ID: 4202237689(a)192.168.1.156. > > > > CSeq: 3 BYE. > > > > Server: OpenSer (1.0.1 (i386/linux)). > > > > Content-Length: 0. > > > > Warning: 392 210.68.65.78:5060 "Noisy feedback tells: pid=2144 > > req_src_ip=59.176.78.110 req_src_port=50496 in_uri=sip:210.68.65.78. > > > > > > > > > > > > In both cases BYE is not getting accounted for. Can anyone help me on > > this ?? > > > > > > > > > > > > Regards > > > > Dan > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.7 (GNU/Linux) > Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org <http://enigmail.mozdev.org> > > iD8DBQFG3Q9/qjpLE0HiOBYRArgmAKCec0eFK2QQyBmV5eGoCWRmUOMM8ACgmFgp > tOKvu9D6afXK2f8if8FABnM= > =66HB > -----END PGP SIGNATURE----- > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users -- Thanks &Regards Ravi Prakash Sunkara -- Thanks &Regards Ravi Prakash Sunkara

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[Serusers] SER+PATTON1+PATTON2+PATTON3
by pokerino76 04 Sep '07

04 Sep '07

I do not have the possibility to use Patton 4960 with 1xPRI. I wished an alternative method for Patton1 + patton2 + patton3: 8+8+8=32. thanks This is the code for 2 Patton Hi! On the pattons you can created Error Messages when not circuit channel is available And on Ser you have to register to Failure Route to the route of the first patton and the failure route point to the second Patton. PATTON1: profile sip default map causes to-sip no-circuit-channel-available to 486 map causes to-sip resources-unavailable to 486 service hunt-group HUNT-BRI cyclic timeout 1 drop-causes normal-unspecified drop-causes no-circuit-channel-available drop-causes network-out-of-order drop-causes temporary-failure drop-causes switching-equipment-congestion drop-causes access-info-discarded drop-causes circuit-channel-not-available drop-causes resources-unavailable route call 1 dest-interface ISDN_01 route call 2 dest-interface ISDN_02 route call 3 dest-interface ISDN_03 SER: Ser.cfg: if (uri=~ ". *sip: 012345 ") # Patton 1 {t_on_failure (1); t_relay_to_udp (11.11.11.10, 5060); break; } failure_route [1] { if (method== " INVITE ") { if (t_check_status (600|486)) #### ErrorCode from Patton 1 { #####call from Patton 1 -- 486 (Busy) ret -- call switch to Patton 2 ###### if (uri=~ ". *sip: 012345 ") # Costumer xy { append_branch (); t_relay_to_udp (11.11.11.11, 5060); ##Unterroute ist hier nicht notwendig, break; ##da sich die Verrechnung hier nicht aendert } } } Which code to use for 3 patton? ------------------------------------------------------ Leggi GRATIS le tue mail con il telefonino i-mode di Wind http://i-mode.wind.it/

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[OpenSER-Users] Openser rejecting BYE packet
by Dan 04 Sep '07

04 Sep '07

Scenario 1: U 212.34.169.220:5060 -> 78.45.146.34:5060 BYE sip:test@78.45.146.34 SIP/2.0. Route: <sip:78.45.146.34;ftag=aa313a3c;lr>. CSeq: 2 BYE. Via: SIP/2.0/UDP 212.34.169.220:5060. From: <sip:1217773456@78.45.146.34>;tag=03095807065938997270316869. Call-ID: 8246bf729f62a754. To: 58.68.96.11<sip:test@78.45.146.34>;tag=aa313a3c. Content-Length: 0. # U 78.45.146.34:5060 -> 212.34.169.220:5060 SIP/2.0 404 User Not Found. CSeq: 2 BYE. Via: SIP/2.0/UDP 212.34.169.220:5060. From: <sip:1217773456@78.45.146.34>;tag=03095807065938997270316869. Call-ID: 8246bf729f62a754. To: 58.68.96.11<sip:test@78.45.146.34>;tag=aa313a3c. Server: OpenSer (1.0.1 (i386/linux)). Content-Length: 0. Warning: 392 78.45.146.34:5060 "Noisy feedback tells: pid=24525 req_src_ip=78.45.146.34 req_src_port=5060 in_uri=sip:78.45.146.34;ftag=aa313a3c;lr out_uri=sip:78.45.146.34;ftag=aa313a3c;lr via_cnt==2". Scenario 2: U 59.176.78.110:50496 -> 210.68.65.78:5060 BYE sip:210.68.65.78 SIP/2.0. Route: <sip:@209.67.171.10;transport=udp>. Via: SIP/2.0/UDP 59.176.78.110:50496. From: sip:user1@210.68.65.78;tag=4154163258. To: <sip:12818495027@210.68.65.78;user=phone>;tag=3005360702044301069847989. Call-ID: 4202237689(a)192.168.1.156. CSeq: 3 BYE. User-Agent: Cisco ATA v2.15 ata18x (020927a). Content-Length: 0. . # U 210.68.65.78:5060 -> 59.176.78.110:50496 SIP/2.0 478 Unresolvable destination (478/TM). Via: SIP/2.0/UDP 59.176.78.110:50496. From: sip:user1@210.68.65.78;tag=4154163258. To: <sip:12818495027@210.68.65.78;user=phone>;tag=3005360702044301069847989. Call-ID: 4202237689(a)192.168.1.156. CSeq: 3 BYE. Server: OpenSer (1.0.1 (i386/linux)). Content-Length: 0. Warning: 392 210.68.65.78:5060 "Noisy feedback tells: pid=2144 req_src_ip=59.176.78.110 req_src_port=50496 in_uri=sip:210.68.65.78. In both cases BYE is not getting accounted for. Can anyone help me on this ?? Regards Dan

2 2

[OpenSER-Users] call duration limit
by Nick 'TARANTUL' Novikov 04 Sep '07

04 Sep '07

Hi, ppl! I have installed OpenSER and radius server. OpenSER authorized all call through radius. How I can limit call duration by "Session-Timeout" attribute from radius ? -- tarantul

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[Serusers] Route SIP Calls To Asterisk
by Kevin Kibuku 03 Sep '07

03 Sep '07

Hi Team, Please someone guide me in how I could forward for instance a call to 1111111 made on the SER to an * server. I have tried quite a lot of things but nothing seems to work. Thanks -- Kevin Kibuku, Intersat Africa Limited Internet Via Satellite Cell: +254720789158

2 2

[OpenSER-Users] t_relay after failure_route -- problem
by Zahid Mehmood 03 Sep '07

03 Sep '07

Hi All, I am trying to implement serial forking and am using the example on voice-systems website as a guide. The problem I am experiencing is as follows. Failure route rewrite ruri with the next uri to try and calls route[17] route 17 performs a lookup("location"), finds the usrloc information and rewrites ruri, however, when t_relay is called it uses the original ruri (one before the lookup) to do t_relay. Another issue I'm experiencing is that when processing call after failure route, ruri_user_avp. Once i rewrite the uri in failure route, I have to use the following to manually write a value for ruri_user_avp. Can someone explain why this is happening? Thanks in advance for your help!! -- Zahid ] Some relevant config and debug info: modparam("lcr", "gw_uri_avp", "1400") modparam("lcr", "ruri_user_avp", "1402") route[18] { # used to process pstn/ off system calls ...... if (!is_avp_set("$avp(i:1402)")) { xlog("L_INFO","$ci: route[22] ** setting ruri_user_avp **\n"); $avp(i:1402) = $rU; xlog("L_INFO","$ci: route[22] ** setting ruri_user_avp = $avp (i:1402) **\n"); } ........ } #end route 18 route[11] { # load all numbers asscociated with the pilot number in $avp (serial_fork) array xlog("L_INFO","$ci: route[11] - M=$rm RURI=$ru F=$fu T=$tu IP=$si \n"); $avp(serial_fork) = "sip:*6" + $rU + "@sipdev.oursiphost.com"; if (avp_db_query("select username from fork where pilot = $rU order by pri" , "$avp(serial_fork)")) { xlog("L_INFO", "route[11] $ci: method $rm r-uri <$ru> \n"); xlog("L_INFO", "route[11] serial_fork = $avp (serial_fork) remote-user = $rU \n"); t_on_failure("5"); route(17); exit; } else { xlog("L_ERR", "route[11] failed to load serial_fork info.\n"); } avp_print(); # debugging only setflag(8); } # end of route[11] route[17] { xlog("L_INFO","$ci: route[17] - M=$rm RURI=$ru F=$fu T=$tu IP=$si \n"); $avp(s:inv_timeout) = "10"; if (lookup("location")) { # currently online registered UA. xlog("L_INFO","$ci: route[17] - M=$rm RURI=$ru F=$fu T=$tu IP=$si -- inside location lookup \n"); if (!t_relay()) { sl_reply_error(); }; return; } else { # not online if (does_uri_exist()) { # but exists xlog("L_INFO", "$ci: Subscriber is offline [$ru]\n"); sl_send_reply("480", "Temporarily Unavailable."); return; } xlog("L_INFO","$ci: route[17] - M=$rm RURI=$ru F=$fu T=$tu IP=$si -- outside location lookup \n"); }; if (uri=~"^sip:(([0-9()+-])|(%20))+@") { subst_user('/[()+-]//g'); # strip out the punctuation. subst_user('/%20//g'); # strip out the white space. route(18); # all numerics just go to route(18) -- off system calls return; }; } # end route[17] failure_route[5] { xlog("L_INFO","$ci: failure_route[5] - M=$rm RURI=$ru F=$fu T=$tu IP=$si\n"); avp_print(); if (avp_pushto("$ruri", "$avp(serial_fork)")) { append_branch(); xlog("L_INFO","$ci: failure_route[5] $rm r-uri <$ru> from <$fu> to <$tu>\n"); avp_delete("$avp(serial_fork)"); t_on_failure("5"); route(17); } else { xlog("L_INFO","$ci: failure_route[5] $rm r-uri <$ru> from <$fu> to <$tu>\n"); } } Debug info Aug 31 10:54:56 mousse openser[6815]: a410a3c1- e4befc20-961fd447(a)10.1.12.174: failure_route[5] - M=INVITE RURI=sip: 10512@10.1.12.172:8891 F=sip:10510@sipdev.oursiphost.com T=sip: 10512(a)sipdev.oursiphost.com;user=phone IP=10.1.12.174 Aug 31 10:54:56 mousse openser[6815]: INFO:avpops:print_avp: p=0xb6039c28, flags=0x0003 Aug 31 10:54:56 mousse openser[6815]: INFO: name=<inv_timeout> Aug 31 10:54:56 mousse openser[6815]: INFO: val_str=<10 / 2> Aug 31 10:54:56 mousse openser[6815]: INFO:avpops:print_avp: p=0xb6039bf0, flags=0x0002 Aug 31 10:54:56 mousse openser[6815]: INFO: id=<655> Aug 31 10:54:56 mousse openser[6815]: INFO: val_str=<sip:10511@sipdev.oursiphost.com / 29> Aug 31 10:54:56 mousse openser[6815]: INFO:avpops:print_avp: p=0xb6039bb8, flags=0x0002 Aug 31 10:54:56 mousse openser[6815]: INFO: id=<655> Aug 31 10:54:56 mousse openser[6815]: INFO: val_str=<sip:10210@sipdev.oursiphost.com / 29> Aug 31 10:54:56 mousse openser[6815]: INFO:avpops:print_avp: p=0xb6039b80, flags=0x0002 Aug 31 10:54:56 mousse openser[6815]: INFO: id=<655> Aug 31 10:54:56 mousse openser[6815]: INFO: val_str=<sip:10510@sipdev.oursiphost.com / 29> Aug 31 10:54:56 mousse openser[6815]: INFO:avpops:print_avp: p=0xb6039b48, flags=0x0002 Aug 31 10:54:56 mousse openser[6815]: INFO: id=<655> Aug 31 10:54:56 mousse openser[6815]: INFO: val_str=<sip:*610512@sipdev.oursiphost.com / 31> Aug 31 10:54:56 mousse openser[6815]: INFO:avpops:print_avp: p=0xb6039b18, flags=0x0083 Aug 31 10:54:56 mousse openser[6815]: INFO: name=<is_pilot> Aug 31 10:54:56 mousse openser[6815]: INFO: val_str=<y / 1> Aug 31 10:54:56 mousse openser[6815]: DEBUG:avpops:pushto_avps: 1 avps were processed Aug 31 10:54:56 mousse openser[6815]: ******* setting for branch 0 flags 0 Aug 31 10:54:56 mousse openser[6815]: a410a3c1- e4befc20-961fd447(a)10.1.12.174: failure_route[5] INVITE r-uri <sip: 10511(a)sipdev.oursiphost.com> from <sip:10510@sipdev.oursiphost.com> to <sip:10512@sipdev.oursiphost.com;user=phone> Aug 31 10:54:56 mousse openser[6815]: DEBUG:avpops:delete_avps: 1 avps were removed Aug 31 10:54:56 mousse openser[6815]: a410a3c1- e4befc20-961fd447(a)10.1.12.174: route[17] - M=INVITE RURI=sip: 10511(a)sipdev.oursiphost.com F=sip:10510@sipdev.oursiphost.com T=sip: 10512(a)sipdev.oursiphost.com;user=phone IP=10.1.12.174 Aug 31 10:54:56 mousse openser[6815]: rewrite_uri: Rewriting Request- URI with 'sip:10511@10.1.12.172:8891' Aug 31 10:54:56 mousse openser[6815]: a410a3c1- e4befc20-961fd447(a)10.1.12.174: route[17] - M=INVITE RURI=sip: 10511@10.1.12.172:8891 F=sip:10510@sipdev.oursiphost.com T=sip: 10512(a)sipdev.oursiphost.com;user=phone IP=10.1.12.174 -- inside location lookup Aug 31 10:54:56 mousse openser[6815]: DEBUG: mk_proxy: doing DNS lookup... Aug 31 10:54:56 mousse openser[6815]: DEBUG:sip_resolvehost2: no port, no proto -> do NAPTR lookup! Aug 31 10:54:56 mousse openser[6815]: DEBUG:filter_and_sort_naptr: found valid SIP+D2U -> _sip._udp.sipdev.oursiphost.com Aug 31 10:54:56 mousse openser[6815]: DEBUG:do_srv_lookup: SRV (_sip._udp.sipdev.oursiphost.com) = mousse.cc.oursiphost.com:5060 Aug 31 10:54:56 mousse openser[6815]: DEBUG:sip_resolvehost2: found! Aug 31 10:54:56 mousse openser[6815]: check_via_address(10.1.12.174, 10.1.12.174, 0) Aug 31 10:54:56 mousse openser[6815]: DEBUG:tm:set_timer: relative timeout is 500000 Aug 31 10:54:56 mousse openser[6815]: DEBUG: add_to_tail_of_timer[4]: 0xb6039eb4 (61500000) Aug 31 10:54:56 mousse openser[6815]: DEBUG:tm:set_timer: relative timeout is 30 Aug 31 10:54:56 mousse openser[6815]: DEBUG: add_to_tail_of_timer[0]: 0xb6039ed0 (91) Aug 31 10:54:56 mousse openser[6815]: DEBUG:tm:relay_reply: branch=0, save=1, relay=-1 >, CSEQ_N=<CSeq: 1> 2(a)sipdev.oursiphost.com;user=phone>ocal: using FROM=<From: "Zahid 10510" <sip: 10510(a)sipdev.oursiphost.com>;tag=BF787F6-F6DD44ED Aug 31 10:54:56 mousse openser[6815]: DEBUG: cancel_branch: sending cancel... Aug 31 10:54:56 mousse openser[6815]: DEBUG:tm:set_timer: relative timeout is 500000 Aug 31 10:54:56 mousse openser[6815]: DEBUG: add_to_tail_of_timer[4]: 0xb6039e20 (61500000) Aug 31 10:54:56 mousse openser[6815]: DEBUG:tm:set_timer: relative timeout is 30 Aug 31 10:54:56 mousse openser[6815]: DEBUG: add_to_tail_of_timer[0]: 0xb6039e3c (91)

1 2

[OpenSER-Users] Bernard Le est absent(e).
by bernard.le＠orange-ftgroup.com 03 Sep '07

03 Sep '07

Je serai absent(e) du 03/09/2007 au 17/09/2007. Je répondrai à votre message dès mon retour. Contacter le service Ingénierie : Patrice Galleau au +33 1 44 55 69 36, patrice.galleau(a)orange-ftgroup.com ou Tony Capochichi au +33 1 44 55 66 34, tony capochichi(a)orange-ftgroup.com I will reply to your message when I'll be back. Please contact the Engineer department: Patrice Galleau at +33 1 44 55 69 36, patrice.galleau(a)orange-ftgroup.com or Tony Capochichi at +33 1 44 55 66 34, tony capochichi(a)orange-ftgroup.com ********************************* This message and any attachments (the "message") are confidential and intended solely for the addressees. Any unauthorised use or dissemination is prohibited. Messages are susceptible to alteration. France Telecom Group shall not be liable for the message if altered, changed or falsified. If you are not the intended addressee of this message, please cancel it immediately and inform the sender. ********************************

1 0

[OpenSER-Users] Does Presence Module support 'policy document'?
by KevinKinnan 03 Sep '07

03 Sep '07

Does Presence Module support 'policy document'? That means the user can do some settings to forbid specified watcher or something like this. Thank you! Kevin _________________________________________________________________ Windows Live Custom Domain，您的免费电子邮局。 https://domains.live.com/default.aspx

2 1

[OpenSER-Users] OpenXCAP launch
by Adrian Georgescu 03 Sep '07

03 Sep '07

Hello, I would like to announce the launch of OpenXCAP.org OpenXCAP is an open source, easy extensible, fully featured XCAP server with TLS security and support for multiple realms. OpenXCAP server can be used in combination with a SIP server that supports PUBLISH/SUBSCRIBE/NOTIFY methods to provide a complete SIP SIMPLE server solution The server is written in the Python programming language and is based on the following RFCs: - The Extensible Markup Language Configuration Access Protocol RFC 4825 - Formats for Representing Resource Lists RFC 4826 - Usage for Manipulating Presence Document Contents RFC 4827 - Presence Authorization Rules draft-ietf-simple-presence-rules-10 The server supports TLS security, multiple back-end storage systems and works out of the box with OpenSER Proxy/Registrar/Presence server. The software is provided under the BSD license. I wish to thank here to my IETF friends who helped me with advice, testing and guidance during the development phase of this project. I would also like to thank to Mircea Amarascu, the author of the software, without his passion and dedication this project would not have been possible. The project is hosted at: http://openxcap.org The project is supported via the wiki collaboration system setup by AG Projects. To open ticket please Register first. We are looking forward your feedback from real-life field deployments. Kind regards, Adrian Georgescu AG Projects

2 1

[OpenSER-Users] Media control with openser
by srinivas Antarvedi 03 Sep '07

03 Sep '07

Hello all, Actually i have openser box which i used to make calls among my offices which spread different geographical places. i am using in the configuration file just to make changes to my rtp packets so without RTPPROXY and MEDIAPROXY modules i was able to make calls even my end UAC are behind NAT. but if i want to control my media between different UAs like a prepaid system as we used to see in case of mobile prepaid system which stops you from continuing your call when there was no balance left on ur mobile . can anybody give me a hint how can i do that ? if so what architecture it needs like any mediaproxy? will session border controller help me out? just give me a hint so that i can go ahead thanks once again -- Srinivas Antarvedi

3 2

[OpenSER-Users] Wanted: VoIP Engineer for Warsaw!
by laurent schweizer 03 Sep '07

03 Sep '07

Peoplefone AG jest firmą oferującą usługi telekomunikacyjne oparte na technologii Voice over IP (VoIP) w wyjątkowo niskich stawkach. Peoplefone jest certyfikowanym partnerem firm Siemens<http://www.siemens.ch/index.jsp?sdc_p=c175fi1012637lmno1012637psuz1&sdc_sid…>i AVM/FRITZ!Box <http://www.fritz-shop.ch/>. W związku z naszym dynamicznym rozwojem, do nowo tworzonej spółki w Polsce poszukujemy osoby na stanowisko: SPECJALISTA D/S VOIP Miejsce pracy: Warszawa *Wymagania:* - wykształcenie wyższe techniczne (lub w trakcie studiów), najlepiej informatyczne - praktyczna umiejętność programowania w środowisku w PHP i językach C i C++ - znajomość Linux - znajomość mySQL i PostgreSQL - znajomość zagadnień związanych z sieciami IP, UDP, TCP - znajomość narzędzi do analizy sieci np.. Ethereal - wiedza z zakresu VoIP, serwerów i konfiguracji oprzyrządowania - umiejętność rozwiązywania problemów informatycznych - znajomość języka angielskiego w mowie i piśmie w stopniu zaawansowanym - umiejętność szybkiego uczenia się - chęć podnoszenia swoich kwalifikacji zawodowych - profesjonalne i rzetelne podejście do obowiązków służbowych, umiejętność pracy w zespole, samodzielność, komunikatywność, systematyczność, terminowość - doświadczenie zawodowe mile widziane *Dodatkowe wymagania (mile widziane):* - umiejętność programowania w PERL i JAVA - znajomość Asterisk / SER / OPENSER - umiejętność konfigurowania bramek i ruterów Cisco, Patton, lub innych - znajomość protokołów SIP lub STUN - znajomość problematyki NAT I funkcjonowania outband Proxy - znajomość narzędzi monitorujących jak Cactus, Nagios, MRTG - znajomość narzędzi jak DRBD, Hearthbeat - dyspozycyjność Wszystkich zainteresowanych prosimy o przesłanie życiorysu oraz listu motywacyjnego (zawierającego zgodę na przetwarzanie danych osobowych) *do 10 września br. ** *na adres: *ewa.ciesla(a)peoplefone.com* W temacie wiadomości wysyłanej pocztą elektroniczną prosimy umieścić nazwę stanowiska. Skontaktujemy się tylko z wybranymi kandydatami. Prosimy o dopisanie następującej klauzuli "Wyrażam zgodę na przetwarzanie moich danych osobowych zawartych w mojej ofercie pracy dla potrzeb niezbędnych do realizacji procesu rekrutacji (zgodnie z ustawą z dn. 29.08.97 roku o Ochronie Danych Osobowych Dz. Ust Nr 133 poz. 883)".

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Re: [Serusers] Can't make a single call. Help me please.
by Harmeet Singh 03 Sep '07

03 Sep '07

It's strange, but I do see the SIP INVITE messages being sent in the ethereal log. Why they don't reach to the server? SER is connected directly to the internet, without any NAT. On the other hand, REGISTER messages do get to SER and the client recieves the 200 OK and all intermediate replies as well. I have attached the clientside ethereal log for you to see. I am lost. Harmeet

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[Serusers] exec_dset
by Jesús Terán 02 Sep '07

02 Sep '07

Hello, I am using the exec_dset for access to my database mysql and change the uri in the calls, but I think this use more CPU processing, Are there other module to do the same? thanks Jesus Teran B.

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Re: [OpenSER-Users] OpenSER to PSTN
by Ricardo.ferreira 02 Sep '07

02 Sep '07

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[OpenSER-Users] OpenSER to PSTN
by Arya 01 Sep '07

01 Sep '07

Hi Everyone Is there any way to send all the calls that come to OpenSER to PSTN lines (PRI, BRI) ? -- Thank You

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Re: [OpenSER-Users] checking radius return attributes
by danish.samad＠vocalseeds.com 01 Sep '07

01 Sep '07

Hi Dan, Thanks for all your help and support, I really appreciate it. I was able to get it to work. Although I had to replace all VSA'a with the SIP-AVP attributes. For some reason the SIP-AVP's were not loaded due to the presence of other VSA's. Maybe it was because I was using a = sign in the value prefixes instead of : or #. Is this a known issue. Regards, Danish ps: Someone asked earlier about the radius server I was using. The radius server along with the billing solution is from advancedvoip.com, so far our experience with them has been quite good. > Hi Danish, > > sorry for replying to you so late, but I was out from the office today. > > Here you go: > > First of all, my bad, the returned value should be in the form: > return-code#0 if you want your value to be treated as an integer or > return-code:0 if 0 the AVP should be considered as string - has been a > while since I worked last time with those. By using a higher layer of > debug you should be able to see your avps converted inside when > received from radius. > The rest of your configuration looks good. Note that you can return > more AVP values with the same reply, so you can include the credit > amount as well: > SIP-AVP="return-code#0" > SIP-AVP="h323-credit-amount#4206" > > Hope you will rock now. > > Cheers, > DanB > > On 8/30/07, danish.samad(a)vocalseeds.com <danish.samad(a)vocalseeds.com> > wrote: >> Hi Dan, >> To test the functionality of SIP-AVP, I am trying to get theses avp's in >> the registration block, shown below. Basically on sucessful registration >> the radius server returns the following attributes in acces-accpet >> >> Attributes: >> SIP-AVP = "return-code=0" >> cisco-h323-credit-amount = "h323-credit-amount=4206" >> >> so after registration I am trying to get the return code value. The >> registration block is shown below >> >> route[2] { >> # ----------------------------------------------------------------- >> # REGISTER Message Handler >> # ---------------------------------------------------------------- >> xlog("L_NOTICE","Danish: Entering route[2]\n"); >> xlog("L_NOTICE","Danish: Register message IPsrcaddr [$si], >> RecvdIPaddr[$Ri]\n"); >> if (!search("^Contact:[ ]*\*") && nat_uac_test("2")) { >> xlog("L_NOTICE","Danish: In route[2] Found Nated >> contact\n"); >> setflag(6); >> setbflag(6); >> # if you want OPTIONS natpings uncomment next >> setbflag(7); >> fix_nated_register(); >> force_rport(); >> }; >> >> sl_send_reply("100", "Trying"); >> if (!radius_www_authorize("192.168.0.99")) { >> www_challenge("192.168.0.99", "1"); >> }; >> consume_credentials(); >> >> #avp_print(); >> xlog("L_NOTICE","Danish: Before is set check\n"); >> if(is_avp_set("$avp(s:return-code)")){ >> xlog("L_NOTICE","Danish: return code is set \n"); >> }; >> xlog("L_NOTICE","Danish: After is set check\n"); >> if(avp_check("$avp(s:return-code)", "eq/i:0/g")) >> { >> xlog("L_NOTICE","Danish: return code 0 received \n"); >> }; >> >> if (!save("location")) { >> sl_reply_error(); >> }; >> >> } >> >> In my logs I see >> >> 058) DEBUG:auth_radius:generate_avps: getting SIP AVPs from avpair 225 >> 4(11058) Danish: Before is set check >> 4(11058) Danish: After is set check >> 4(11058) DEBUG:avpops:ops_check_avp: no src avp found >> >> so both conditional avp checks fail and the last debug line is a bit >> worrying >> I traced the first debug line above to sterman.c->generate_avps() >> so just under this debug line I added this piece of code >> >> vp=rc_avpair_get(vp,attrs[A_SIP_AVP].v,0); >> if (vp != NULL) >> { >> DBG("DEBUG:auth_radius:generate_avps: strvalue %s\n", >> vp->strvalue); >> } >> else >> DBG("DEBUG:auth_radius:generate_avps: vp is null\n"); >> >> and this is what I got >> >> 4(11058) DEBUG:auth_radius:generate_avps: getting SIP AVPs from avpair >> 225 >> 4(11058) DEBUG:auth_radius:generate_avps: vp is null >> >> hence openser is not loading the SIP-avp in $avp(s:return-code). I think >> either I am missing something in the openser script or the way I am >> returning the attribute. >> >> all relevant definitions in the cfg are given below: >> >> loadmodule "/usr/local/lib/openser/modules/auth_radius.so" >> loadmodule "/usr/local/lib/openser/modules/uri_radius.so" >> loadmodule "/usr/local/lib/openser/modules/avp_radius.so" >> loadmodule "/usr/local/lib/openser/modules/avpops.so" >> >> modparam("auth_radius|uri_radius|avp_radius", "radius_config", >> "/usr/local/etc/radiusclient-ng/radiusclient.conf") >> modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") >> >> modparam("uri_db|usrloc", "db_url", >> "mysql://openser:openserrw@localhost/openser") >> >> modparam("nathelper", "natping_interval", 20) >> modparam("nathelper", "ping_nated_only", 1) >> modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") >> modparam("nathelper", "received_avp", "$avp(i:42)") >> modparam("nathelper", "sipping_bflag", 7) >> modparam("nathelper", "sipping_from", "sip:pinger@abc.org") >> >> modparam("registrar", "received_avp", "$avp(i:42)") >> modparam("usrloc", "db_mode", 2) >> modparam("usrloc", "nat_bflag", 6) >> >> Hope this explains everything and you would be in a better position to >> help. >> >> Regards, >> Danish >> ps: isnt there any sample script I can refer to for sip-avp radius >> configurations. >> >> >> > Hi Danish, >> > >> > can u post the full block where you are doing these checks? >> > >> > DanB >> > >> > On 8/29/07, danish.samad(a)vocalseeds.com <danish.samad(a)vocalseeds.com> >> > wrote: >> >> Hi Dan, >> >> Thanks for the explanation, things are more clear now. Now I am >> >> returning >> >> a SIP-AVP from radius (for authentication), this is what I see in the >> >> logs >> >> >> >> Attributes: >> >> SIP-AVP = "return-code=0" >> >> ....... >> >> >> >> In my script I do >> >> avp_print(); >> >> if(is_avp_set("$avp(s:return-code)")){ >> >> xlog("L_NOTICE","return code is set \n"); >> >> }; >> >> if(avp_check("$avp(s:return-code)", "eq/0/g")) >> >> { >> >> xlog("L_NOTICE","return code 0 received \n"); >> >> }; >> >> >> >> none of the checks pass and nothing is printed from avp_print. In >> >> openser >> >> logs I see this line >> >> >> >> 1(32550) DEBUG:auth_radius:generate_avps: getting SIP AVPs from >> avpair >> >> 225 >> >> >> >> am I missing some pre-requisites or what am I doing wrong. >> >> >> >> Regards, >> >> Danish >> >> >> >> >> >> > Hi Danish, >> >> > >> >> > for the moment you cannot process attributes other than SIP-AVP >> inside >> >> > openser script. This one is automatically converted into avp. >> >> > EG. A SIP-AVP attribute in the form of: "reason='No Credit'" should >> >> > automatically create an AVP named reason with value "No Credit". >> >> > >> >> > Hope that helps, >> >> > DanB >> >> > >> >> > >> >> > On 8/29/07, danish.samad(a)vocalseeds.com >> <danish.samad(a)vocalseeds.com> >> >> > wrote: >> >> >> Hi Dan, >> >> >> >> >> >> Thanks for your reply. >> >> >> >> >> >> Actually I am currently returning an integer status value in >> >> >> cisco-h323-return-code (a vendor specific attribute). So basically >> >> how >> >> >> can >> >> >> I extract the value returned in this attribute, compare it and >> then >> >> send >> >> >> and sl reply. >> >> >> >> >> >> Probably there is a way of loading this attr value in an avp and >> >> >> comparing >> >> >> it? >> >> >> >> >> >> Regards, >> >> >> Danish >> >> >> >> >> >> > Hello Danish, >> >> >> > >> >> >> > a solution would be returning a SIP-AVP, containing the reason >> for >> >> >> > failure, and then forward this to the client using sl. >> >> >> > Another one would be completely rewriting R-URI again from your >> >> >> > SIP-AVP and then send the request to an announcement server (eg: >> >> >> > ann:paymentrequired@your.server - which should play Payment >> >> Required >> >> >> > annoucement). >> >> >> > >> >> >> > Cheers, >> >> >> > DanB >> >> >> > >> >> >> > On 8/29/07, danish.samad(a)vocalseeds.com >> >> <danish.samad(a)vocalseeds.com> >> >> >> > wrote: >> >> >> >> Hi, >> >> >> >> >> >> >> >> I have configured openser-1.2 with a (commercial) third party >> >> radius >> >> >> >> server and it is working great. Now I have a requirement where >> I >> >> need >> >> >> to >> >> >> >> check a radius return attribute in case an access reject is >> >> received >> >> >> in >> >> >> >> response to an authorization request. Is there some way I can >> do >> >> >> this >> >> >> >> in >> >> >> >> the openser script. >> >> >> >> >> >> >> >> Basically I want to send a 402 payment required message to a >> sip >> >> >> client >> >> >> >> in >> >> >> >> case the call is rejected due low balance. In such cases radius >> >> >> returns >> >> >> >> a >> >> >> >> status response code with access reject. >> >> >> >> >> >> >> >> >> >> >> >> Thanks, >> >> >> >> Danish >> >> >> >> >> >> >> >> _______________________________________________ >> >> >> >> Users mailing list >> >> >> >> Users(a)openser.org >> >> >> >> http://openser.org/cgi-bin/mailman/listinfo/users >> >> >> >> >> >> >> > >> >> >> >> >> >> >> >> > >> >> >> >> >> > >> >> >

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[OpenSER-Users] checking radius return attributes
by danish.samad＠vocalseeds.com 01 Sep '07

01 Sep '07

Hi, I have configured openser-1.2 with a (commercial) third party radius server and it is working great. Now I have a requirement where I need to check a radius return attribute in case an access reject is received in response to an authorization request. Is there some way I can do this in the openser script. Basically I want to send a 402 payment required message to a sip client in case the call is rejected due low balance. In such cases radius returns a status response code with access reject. Thanks, Danish

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Re: [Serusers] INFO Method
by Greger Viken Teigre 01 Sep '07

01 Sep '07

As I wrote: you don't, you need another component to do that. SER has a management interface that can be used to inject a message, but you still need an external component to do that. g-) ------- Original message ------- From: kapilg <kapilg(a)amdale.com> Sent: 31.8.'07, 13:34 > Oh Sorry! Actually I am using SER. > > > Well, if one has to send INFO using ser then how one can send? > > > > > Regards, > > > Kapil > > > > --------------- > > From: Greger V. Teigre [mailto:greger@teigre.com] > Sent: Friday, August 31, 2007 4:19 PM > To: kapilg > Cc: serusers(a)iptel.org > Subject: Re: [Serusers] INFO Method > > > > > Wrong list, this is the ser, not openser list. > A proxy just proxies, it doesn't create messages. But maybe your media server? > g-) > > kapilg wrote: > > > Dear Users, > > > > > Could any one tell or give me one sample of sending INFO Method with MSCML as Message Body part? > > > I am working on one project in which XLite makes a call to a Media Server using openser proxy and after some seconds delay proxy forms one packet of INFO Method filling MSCML script in this and send to Media Server. This will be an ivr call. > > > > > Any help will be appreciated. > > > > > Thanks and Regards, > > > Kapil Goel > > > > > > --------------- > > > > > > _______________________________________________ > > Serusers mailing list > > Serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > >

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sr-users September 2007