- sr-users - kamailio.org

[Users] t_replicate after failure
by Douglas Garstang 30 Nov '05

30 Nov '05

In the scenario below, if the first t_replicate fails because 192.168.10.5 returns "Not Found", the second t_replicate command does not execute. Why? How can I make it execute? Call me crazy, but this doesn't seem at all intuitive to me. In a functional programming world, if the first fails, the next should still run. if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("voip.com", "subscriber")) { www_challenge("voip.com", "0"); exit; }; save("location"); t_replicate("192.168.10.5","5060"); t_replicate("192.168.10.7","5060"); exit; }; Thanks, Doug.

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[Serusers] Acc fields
by Richard C. Thompson 30 Nov '05

30 Nov '05

I hope someone can help answer this question. I am using Mysql acc table to generate cdr's and I am having problems with missing BYE records. I am using the totag field as a unique identifier to match up the INVITES, ACK and BYE's. Is this the field I should be using? I know ser changes the BYE uri to reflect the uac that issued the BYE but I don't know if it changes the totag. Thanks for your reply. Rick Thompson

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[Users] Sending Authorisation Credentials
by Douglas Garstang 30 Nov '05

30 Nov '05

I have user agents registering with OpenSER. I wanted to 'copy' the registration to an Asterisk server, so I modified a standard openser.cfg as such: if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("voip.com", "subscriber")) { www_challenge("voip.com", "0"); exit; }; save("location"); t_replicate("192.168.10.7","5060"); # Added this line... exit; }; I just stuck the t_replicate() call in there. OpenSER sends the REGISTER packet to the Asterisk box who requests credentials. How can I get OpenSER to send credentials (ie act as a UAC) for a given user, to a UAS? Thanks, Doug

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[Serusers] CDRTool
by Martin Anderberg 30 Nov '05

30 Nov '05

(I know this is a bit out of topic, please dont feel offended) Hi there! I was about to try out CDRTool to see if it does what it says - however, it doesn't run at all. There is no forum (that i know of) that discusses CDRTool so i hope someone here can help. The box is a debian unstable (tried stable aswell) and CDRTools is installed according to the instructions. When i try to execute ie something in the scripts/-directory i get the following error: Warning: main(SOAP/Client.php): failed to open stream: No such file or directory in /var/www/CDRTool/rating_lib.phtml on line 5 Fatal error: main(): Failed opening required 'SOAP/Client.php' (include_path='.:/usr/share/php:/usr/share/pear') in /var/www/CDRTool/rating_lib.phtml on line 5 Ok, SOAP/Client.php doesn't exist, SOAP/client.php does however. A symlink ought to solve that problem? Running in the scripts/-directory again gives the error: #!/usr/bin/php4 Fatal error: Cannot redeclare class db_sql in /var/www/CDRTool/phplib/db_mysql.inc on line 13 Now i'm stuck and dont know what to do. If someone has any clues, please let me know. Thanks Martin

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[Serusers] Issues with SER for TCP load testing
by Stephen McVarnock 30 Nov '05

30 Nov '05

Hi folks, I am new to the mailing lists - unsure as to which list is the most appropriate I have mailed to both - I apologise for this in advance! I am using SIPP and SER(v 0.9.4) to load test an application - SIPP is sending calls via UDP to SER at 15 cps and with a duration of 180 seconds. SER is then forwarding on the calls to the application using TCP and receiving the replies back over TCP. However I am seeing quite a few errors in SER after a while. I'd like to determine if the issues I am seeing are with SER or with the application itself so I'd appreciate some help with the following errors. Can someone tell me, for each of the error traces below, what is the root cause of it and (even better) how I can resolve it for SER - if it is indeed an issue with SER and not with the application. 1) 123(14715) ERROR: tcpconn_new: mem. allocation failure 123(14715) ERROR: tcp_main_loop: tcpconn_new failed, closing socket 2) 54(14646) ERROR: tsend_stream: failed to send: (32) Broken pipe 54(14646) ERROR: tcp_send: failed to send 3) 65(14785) ERROR: tcp_blocking_connect: SO_ERROR (146) Connection refused 65(14785) ERROR: tcpconn_connect: tcp_blocking_connect failed 65(14785) ERROR: tcp_send: connect failed 65(14785) msg_send: ERROR: tcp_send failed 4) 63(14783) ERROR: tsend_stream: send timeout (10000) 63(14783) ERROR: tcp_send: failed to send 5) There was also another error on the TCP select loop - select failed: Too many connections (or something similar). I have been playing around with the number of descriptors set using limit but without success. Like I said, my main concern is to eliminate SER where possible as the cause of these issues before focusing attention on the application. Thanks in advance, Steve.

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[Serusers] Creating CDRs from SERs accouning records
by Adrian Georgescu 30 Nov '05

30 Nov '05

Frank, you can download CDRTool from: http://cdrtool.ag-projects.com/ CDRTool it is not publicly supported, we provide support only for our customers but you may give it a try. Adrian >>>>>>>>>>> Hi all i'm looking for a free/opensource application that generates CDRs (not XDRs, must not be rated) from the accounting data/events generated by SER (i think this is usually called normalization). I think this would have to be an application that is able to match INVITEs and BYEs and form CDRs with call start timestamp, call duration and so on. I had a look at the list of accounting software on the iptel.org website, but most of there applications are highly integrated accounting/billing solutions, far more than what i need (and also most of them seem to be commercial products). In the mailinglist archive i only found some references to CDRTool which seems also to be commercial. Obv. there must be a free version of CDRTool but neither i'm sure whether it implements exactly what i need so i could "grab it" for my own purposes nor could i find a place to download the free version). Does anyone know if there is such a kind of application available? Or any other idea how to deal with my requirements? Thanks a lot for your help Frank

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[Serusers] REGISTER message headers verification
by Aisling O'Driscoll 30 Nov '05

30 Nov '05

Hi, I was wondering if someone could please cast their eyes over the register sent from one of my phones behind a nat and the 200 OK sent back from SER. There has been an issue with no audio between a natted client and a public client. I looked at the invite and noticed the private address was being used in the Contact header field and the c field in the sdp so obviously the headers are not being rewritten properly....I then checked the register messages to see if the phone was registering correctly - The message sequence is below. I notice the private header is shown but that would probably be correct seeing fix_nated_register doesnt actually change the header, it simply adds fields, is that right? i.e. thats why it replaced fix_nated-contact to be more RFC compliant Anyway its the contact header fields sent back in the 200OK that I would like verified - Should the private address be listed as one of the options here? I have gone back to the onsip feature callfwd script using mediaproxy to eliminate any errors due to changes I may have made The ONLY difference between the scripts is that I have if(uri != myself) { route(4); route(1); break; }; instead of #if (!is_uri_host_local()) { # if (is_from_local() || allow_trusted()) { # route(4); # route(1); # } else { # sl_send_reply("403", "Forbidden"); # }; # break; # }; as otherwise the phones wouldn't register. Many thanks, Aisling. SER: 157.190.74.152 Phone Private Address: 172.16.3.13 NAT Box: 84.203.148.14 U 84.203.148.14:5060 -> 157.190.74.152:5060 REGISTER sip:157.190.74.152:5060 SIP/2.0..Via: SIP/2.0/UDP 172.16.3.13;bran ch=z9hG4bK7543671c6ca1f07a..From: "Aisling 2092" <sip:2092@157.190.74.152:5 060>;tag=2ec993227a53039b..To: <sip:2092@157.190.74.152:5060>..Contact: <si p:2092@172.16.3.13>..Call-ID: 78f22336c383663f@172.16.3.13..CSeq: 100 REGIS TER..Expires: 3456000..User-Agent: Grandstream BT100 1.0.6.7..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Cont ent-Length: 0.... # U 157.190.74.152:5060 -> 84.203.148.14:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 172.16.3.13;branch=z9hG4bK7543671c6ca1 f07a;received=84.203.148.14..From: "Aisling 2092" <sip:2092@157.190.74.152: 5060>;tag=2ec993227a53039b..To: <sip:2092@157.190.74.152:5060>..Call-ID: 78 f22336c383663f@172.16.3.13..CSeq: 100 REGISTER..Server: Sip EXpress router (0.9.4 (i386/linux))..Content-Length: 0..Warning: 392 157.190.74.152:5060 " Noisy feedback tells: pid=1742 req_src_ip=84.203.148.14 req_src_port=5060 in_uri=sip:157.190.74.152:5060 out_uri=sip:157.190.74.152:5060 via_cnt==1". ... # U 157.190.74.152:5060 -> 84.203.148.14:5060 SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP 172.16.3.13;branch=z9hG4bK754367 1c6ca1f07a;rport=5060;received=84.203.148.14..From: "Aisling 2092" <sip:209 2@157.190.74.152:5060>;tag=2ec993227a53039b..To: <sip:2092@157.190.74.152:5 060>;tag=4b358c93636b59b9f78eca99791ec991.cdd2..Call-ID: 78f22336c383663f@1 72.16.3.13..CSeq: 100 REGISTER..WWW-Authenticate: Digest realm="157.190.74. 152", nonce="438dfc0821fd70b65dbd9c0db3e5aa29fa1fcd82"..Server: Sip EXpress router (0.9.4 (i386/linux))..Content-Length: 0..Warning: 392 157.190.74.15 2:5060 "Noisy feedback tells: pid=1742 req_src_ip=84.203.148.14 req_src_po rt=5060 in_uri=sip:157.190.74.152:5060 out_uri=sip:157.190.74.152:5060 via_ cnt==1".... # U 84.203.148.14:5060 -> 157.190.74.152:5060 REGISTER sip:157.190.74.152:5060 SIP/2.0..Via: SIP/2.0/UDP 172.16.3.13;bran ch=z9hG4bK9e94983538147a54..From: "Aisling 2092" <sip:2092@157.190.74.152:5 060>;tag=2ec993227a53039b..To: <sip:2092@157.190.74.152:5060>..Contact: <si p:2092@172.16.3.13>..Authorization: Digest username="2092", realm="157.190. 74.152", algorithm=MD5, uri="sip:157.190.74.152:5060", nonce="438dfc0821fd7 0b65dbd9c0db3e5aa29fa1fcd82", response="90d926c009567996b99e76fef560b9c6".. Call-ID: 78f22336c383663f@172.16.3.13..CSeq: 101 REGISTER..Expires: 3456000 ..User-Agent: Grandstream BT100 1.0.6.7..Max-Forwards: 70..Allow: INVITE,AC K,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Length: 0.... # U 157.190.74.152:5060 -> 84.203.148.14:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 172.16.3.13;branch=z9hG4bK9e9498353814 7a54;received=84.203.148.14..From: "Aisling 2092" <sip:2092@157.190.74.152: 5060>;tag=2ec993227a53039b..To: <sip:2092@157.190.74.152:5060>..Call-ID: 78 f22336c383663f@172.16.3.13..CSeq: 101 REGISTER..Server: Sip EXpress router (0.9.4 (i386/linux))..Content-Length: 0..Warning: 392 157.190.74.152:5060 " Noisy feedback tells: pid=1744 req_src_ip=84.203.148.14 req_src_port=5060 in_uri=sip:157.190.74.152:5060 out_uri=sip:157.190.74.152:5060 via_cnt==1". ... # U 157.190.74.152:5060 -> 84.203.148.14:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 172.16.3.13;branch=z9hG4bK9e94983538147a54 ;rport=5060;received=84.203.148.14..From: "Aisling 2092" <sip:2092@157.190. 74.152:5060>;tag=2ec993227a53039b..To: <sip:2092@157.190.74.152:5060>;tag=4 b358c93636b59b9f78eca99791ec991.3e0a..Call-ID: 78f22336c383663f(a)172.16.3.13 ..CSeq: 101 REGISTER..Contact: <sip:2092@157.190.74.151>;expires=3279337, < sip:2092@172.16.3.13>;expires=3447275;received="sip:84.203.148.14:50 60", <s ip:2092@84.203.148.14:5060>;expires=3456000..Server: Sip EXpress router (0. 9.4 (i386/linux))..Content-Length: 0..Warning: 392 157.190.74.152:5060 "Noi sy feedback tells: pid=1744 req_src_ip=84.203.148.14 req_src_port=5060 in_ uri=sip:157.190.74.152:5060 out_uri=sip:157.190.74.152:5060 via_cnt==1".... ##exit 91 received, 0 dropped localhost:~ # -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.

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[Users] AVPOP and openser DB key
by Matteo Piazza 30 Nov '05

30 Nov '05

I use avpops module. I use the user_preferences table (openser DB) as avp_table: modparam("avpops","avp_table","usr_preferences") But I have a question: In the description of avp_table is not required a key, in the default configuration of user_preferences table there is a key composed by user-domain-attribute. But with this key I can't use some different value for the same attribute: for example "fwdbusy". So I have delete this key, and the program is ok. Someone knows why in the user_preferences the key is composed those three values -- ======================================= Matteo Piazza, Junior Researcher CREATE-NET Via Solteri, 38 - 38100 Trento - Italy email: matteo.piazza(a)create-net.it Tel: +39-0461-408400ext:308 www.create-net.it =======================================

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Re: [Users] How to Configure a Redirect Server Using OpenSer
by Bogdan-Andrei Iancu 30 Nov '05

30 Nov '05

Hi, do you get any error? do you see the 3xx reply? are you sure the script execution gets there? regards, bogdan Ajay Srivastava wrote: >Hi bogdan, >Thanks for ur suggestion. I made changes in openser.cfg as per the example >which you have given and then restart openser, but still not getting >redirect messages, should I need to do something more for enabling this >option? Please help me and tell me.. >Regards, >ajay > >-----Original Message----- >From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] >Sent: Monday, November 28, 2005 5:02 PM >To: Ajay Srivastava >Cc: users(a)openser.org; Aviraj Saha >Subject: Re: [Users] How to Configure a Redirect Server Using OpenSer > >Hi Ajay, > >there is a script example on the cvs: > >http://cvs.sourceforge.net/viewcvs.py/openser/sip-server/examples/redirect.c >fg?rev=1.2&view=auto > >regards, >bogdan > >Ajay Srivastava wrote: > > > >>Hi >> >>We are trying to configure OpenSer as Redirect Server. Can anyone >>provide me with a configuration file that explains what options we >>have in OpenSer to configure it as redirect server. >> >> >> >>With Regards >> >>Ajay >> >>------------------------------------------------------------------------ >> >>_______________________________________________ >>Users mailing list >>Users(a)openser.org >>http://openser.org/cgi-bin/mailman/listinfo/users >> >> >> >>

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[Users] uac_replace_from errors
by Patrick Jordan-Smith 30 Nov '05

30 Nov '05

Hi there, I'm not totally sure if this is a problem with SER or not, but I have a interesting problem when using uac_replace_from together with LinkSys PAP2 ATAs (and possibly other linksys equipment, retesting tonight.) When you configure a name in the linksys ATA, you can put your name, ie, for myself I put Patrick Jordan-Smith. The linksys sends this like: From: Patrick Jordan-Smith <sip:728728@sip.qsi.net.nz> When I try to use uac_replace_from with this from address it chokes on it: ERROR: parse_to : unexpected char ["] in status 8: <<Beetlestone >> . ERROR:parse_from_header: bad from header ERROR:uac:restore_from_reply: failed to find/parse FROM hdr ERROR: parse_to : unexpected char ["] in status 8: <<Beetlestone >> . ERROR:parse_from_header: bad from header And I end up with a wrecked From address then nothing works. I have checked RFC 3261 and it only says quote marks should be used if the field contains a comma, question-mark or semi-colon. So it seems Linksys are following spec and SER is in error. Any comments on this? Thanks, Pat -- Patrick Jordan-Smith Network Operations Mercury Telecommunications Ltd / Quicksilver Internet voice +64-9-916-0300 / +64-27-5900595 fax +64-9-916-0301

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[Users] PIC/SER PA presence module port?
by Joachim Fabini 30 Nov '05

30 Nov '05

Hi, Did anyone succeed (or, at least, try) to port the new SER pa module (Jamey Hicks' additions for PIC/SER trial) to openSER? If yes: any hints on how to do this? If not a question to the developers: Are the module interfaces of SER and OpenSER still so close that a port of the PA module should be feasible without too much coding? There is much interest in presence around and having a decent presence module in OpenSER is a definite plus. tia, --Joachim

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[Users] How can I send radius authentication packet with openser
by Arda Tekin 30 Nov '05

30 Nov '05

Hi, I have installed openser, mysql, radiusclient-ng-0.5.2 successfully on REL3.0. openser works well with mysql. I need to send a radius authentication packet to a radius server(according to RFC2865). Packet contains base params: User-name (attr.1) $Username Password (attr.2) $Password NAS-Identifier (attr.4) (auto-generated) NAS-Port (attr.5) $uref State (attr.24) 0 Client-Port-DNIS (attr.30) NONE Caller-Id (attr.31) $calling I can not find a clear sample about radius. Which module is used for this purpose? Regards Arda

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[Serusers] serctl via unix socket?
by Cesc 30 Nov '05

30 Nov '05

Hi, We use SER+SEMS in my project. Recently, we moved the interface between them to unix sockets (from the "traditional" fifo). But now ... serctl won't work because it cannot find the ser fifo (for example, "serctl ul show"). Can the two (fifo and unix sock) cohexist? that is ... can ser open both at the same time and receive/send from/to both? Or, how can i make serctl use the socket interface? Regards, Cesc

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[Users] not going to failure_route
by Jayesh Nambiar 30 Nov '05

30 Nov '05

hi all, I was trying the following scenario: If a call comes from trusted source_ip, I rewritehost and forward them. But what I want is if the reply status is greater than 400, it should go to the failure route and try that destination. But the call gets replied as "500 service unavailable". I think, the call should go to the failure route if the status is greater than 300, is it right? Or i m making some logical error here. snippet of my cfg file: if (uri=~"^sip:00[0-9]*@") { #forward the trusted IPs without authentication strip(2); rewritehost("216.XX.XX.XX"); t_on_failure("1"); t_on_reply("2"); t_relay(); }; failure_route[1] { log(1, "trying next ip address"); prefix("00"); rewritehost("63.XX.XX.XX"); t_relay(); } onreply_route[2] { if(status=~"18[0-9]") { t_on_failure("0"); }; } pls help me out with this. thanx a lot in advance!! Jayesh --------------------------------- Enjoy this Diwali with Y! India Click here

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Fwd: [Users] sample configuration file for redirect server
by jyotisankar nayak 30 Nov '05

30 Nov '05

HI Iqbal, pls help me in this matter. regards jyoti Note: forwarded message attached. --------------------------------- Yahoo! Music Unlimited - Access over 1 million songs. Try it free.

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RE: [Users] Forwarding from dispatcher
by Douglas Garstang 30 Nov '05

30 Nov '05

Yes, I think the documentation needs an update. :) -----Original Message----- From: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] Sent: Tuesday, November 29, 2005 9:52 PM To: rick(a)eltopia.com Cc: users(a)openser.org Subject: Re: [Users] Forwarding from dispatcher On 11/29/05 23:15, Rick Richardson wrote: > I might be missing something obvious. But I have set up a dispatching > openSER box using the recommended .cfg file from the dispatcher module > docs and a custom list. > > On init, The parser complains of the forward(); function because it has > no params. I looked at the core docs and they say they forward() > requires a host and port, which makes sense, however, I don't know what > to put in as a param as it is decided by the ds_select_dst() function. > > In short, what do I pass to forward to make it use the destination > decided by ds_select_dst? > use "forward(uri:host, uri:port)" Cheers, Daniel > thanks, > > Rick > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

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[Users] RE: Users Digest, Vol 6, Issue 98
by Papadopoulos Georgios 30 Nov '05

30 Nov '05

Lokesh, Comment lines start with # not ; George > Date: Wed, 30 Nov 2005 11:18:58 -0000 > From: "Lokesh Kumar" <lokesh(a)interacesso.pt> > Subject: Re: [Users] Can anyone share his openser.cfg..in which > openser is routing calls to pstn gateway > To: "Norman Brandinger" <norm(a)goes.com> > Cc: serusers(a)iptel.org, users(a)openser.org > Message-ID: <004901c5f59f$dbb87030$2f00a8c0@lokesh> > Content-Type: text/plain; charset="iso-8859-1" > > Hello Norman, > When I type openser start it gives me " ERROR: Bad config > file (4 errors). > here is the messages i got from cat /var/log/messages > --------- > Nov 30 11:05:01 ser crond(pam_unix)[26819]: session closed > for user root Nov 30 11:07:16 ser openser: ERROR: > load_module: could not open module > <modules/sl/sl.so>: modules/sl/sl.so: cannot open shared > object file: No such file or directory Nov 30 11:07:16 ser > openser: parse error (10,13-14): failed to load module Nov 30 > 11:07:16 ser openser: ERROR: load_module: could not open module > <modules/tm/tm.so>: modules/tm/tm.so: cannot open shared > object file: No such file or directory Nov 30 11:07:16 ser > openser: parse error (11,13-14): failed to load module Nov 30 > 11:07:16 ser openser: parse error (12,1-2): syntax error Nov > 30 11:07:16 ser openser: parse error (12,1-2): > Nov 30 11:09:11 ser openser: ERROR: load_module: could not open module > <modules/sl/sl.so>: modules/sl/sl.so: cannot open shared > object file: No such file or directory Nov 30 11:09:11 ser > openser: parse error (10,13-14): failed to load module Nov 30 > 11:09:11 ser openser: ERROR: load_module: could not open module > <modules/tm/tm.so>: modules/tm/tm.so: cannot open shared > object file: No such file or directory Nov 30 11:09:11 ser > openser: parse error (11,13-14): failed to load module Nov 30 > 11:09:11 ser openser: parse error (12,1-2): syntax error Nov > 30 11:09:11 ser openser: parse error (12,1-2): > [root@ser openser]# > ------------------------------ > > Here is my openser.cfg , In which i had commented mysql, and > auth modules.I want the things works without any AAA, initially. > > > > # > # $Id: pstn.cfg,v 1.2 2005/07/25 16:27:33 miconda Exp $ # # > example: ser configured as PSTN gateway guard; PSTN gateway > is located # at 192.168.0.10 # > > # ------------------ module loading ---------------------------------- > > loadmodule "modules/sl/sl.so" > loadmodule "modules/tm/tm.so" > ;loadmodule "modules/acc/acc.so" > loadmodule "modules/rr/rr.so" > loadmodule "modules/maxfwd/maxfwd.so" > ;loadmodule "modules/mysql/mysql.so" > ;loadmodule "modules/auth/auth.so" > ;loadmodule "modules/auth_db/auth_db.so" > ;loadmodule "modules/group/group.so" > loadmodule "modules/uri/uri.so" > > # ----------------- setting module-specific parameters --------------- > > ;modparam("auth_db", "db_url","mysql://ser:heslo@localhost/ser") > ;modparam("auth_db", "calculate_ha1", yes) > ;modparam("auth_db", "password_column", "password") > > # -- acc params -- > ;modparam("acc", "log_level", 1) > # that is the flag for which we will account -- don't forget > to # set the same one :-) ;modparam("acc", "log_flag", 1 ) > > # ------------------------- request routing logic ------------------- > > # main routing logic > > route{ > > /* ********* ROUTINE CHECKS > ********************************** */ > > # filter too old messages > if (!mf_process_maxfwd_header("10")) { > log("LOG: Too many hops\n"); > sl_send_reply("483","Too Many Hops"); > return; > }; > if (len_gt( max_len )) { > sl_send_reply("513", "Wow -- Message too large"); > return; > }; > > /* ********* RR ********************************** */ > > /* grant Route routing if route headers present */ > if (loose_route()) { t_relay(); return; }; > > /* record-route INVITEs -- all subsequent requests > must visit us */ > if (method=="INVITE") { > record_route(); > > > }; > > # now check if it really is a PSTN destination which > should be handled > # by our gateway; if not, and the request is an > invitation, drop it -- > # we cannot terminate it in PSTN; relay non-INVITE > requests -- it may > # be for example BYEs sent by gateway to call originator > if (!uri=~"sip:\+?[0-9]+@.*") { > if (method=="INVITE") { > sl_send_reply("403", "Call cannot be > served here"); > } else { > forward(uri:host, uri:port); > }; > return; > }; > > # account completed transactions via syslog > setflag(1); > > # free call destinations ... no authentication needed > if ( is_user_in("Request-URI", "free-pstn") /* free > destinations */ > | uri=~"sip:[79][0-9][0-9][0-9]@.*" > /* local PBX */ > | uri=~"sip:98[0-9][0-9][0-9][0-9]") { > log("free call"); > } else if (src_ip==212.xx.35.xx) { > # our gateway doesn't support digest authentication; > # verify that a request is coming from it by source > # address > log("gateway-originated request"); > } else { > # in all other cases, we need to check the > request against > # access control lists; first of all, verify request > # originator's identity > > if (!proxy_authorize( "gateway" /* realm */, > "subscriber" /* table name */)) { > proxy_challenge( "gateway" /* realm > */, "0" /* no qop */ ); > return; > }; > > # authorize only for INVITEs -- RR/Contact > may result in weird > # things showing up in d-uri that would break > our logic; our > # major concern is INVITE which causes PSTN costs > > if (method=="INVITE") { > > # does the authenticated user have a > permission for local > # calls (destinations beginning with > a single zero)? > # (i.e., is he in the "local" group?) > if (uri=~"sip:0[1-9][0-9]+@.*") { > if > (!is_user_in("credentials", "local")) { > sl_send_reply("403", > "No permission for local calls"); > return; > }; > # the same for long-distance > (destinations begin with two zeros") > } else if (uri=~"sip:00[1-9][0-9]+@.*") { > if > (!is_user_in("credentials", "ld")) { > sl_send_reply("403", > " no permission for LD "); > return; > }; > # the same for > international calls (three zeros) > } else if (uri=~"sip:000[1-9][0-9]+@.*") { > if > (!is_user_in("credentials", "int")) { > sl_send_reply("403", > "International permissions needed"); > return; > }; > # everything else (e.g., > interplanetary calls) is denied > } else { > sl_send_reply("403", "Forbidden"); > return; > }; > > }; # INVITE to authorized PSTN > > }; # authorized PSTN > > # if you have passed through all the checks, let your > call go to GW! > > rewritehostport("212.xx.35.xx:5060"); > > # forward the request now > if (!t_relay()) { > sl_reply_error(); > return; > }; > > } > Disclaimer The information in this e-mail and any attachments is confidential. It is intended solely for the attention and use of the named addressee(s). If you are not the intended recipient, or person responsible for delivering this information to the intended recipient, please notify the sender immediately. Unless you are the intended recipient or his/her representative you are not authorized to, and must not, read, copy, distribute, use or retain this message or any part of it. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.

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[Serusers] Error opening ser's FIFO /tmp/ser_fifo
by subbanarasa 30 Nov '05

30 Nov '05

/Hi, // //I had installed already the version 0.8.12 running at a MySql SERVER. The service ////at this moment, STARTS normally with no interruption but when I do use the command line : ////"serctl + [any option] - I got the error : "Error opening ser's FIFO /tmp/ser_fifo".Make sure you have line fifo=/tmp/ser_fifo in your config but ser.config file like this # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) #fork=yes log_stderror=no # (cmd line: -E) /*Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "mysql:/ser:reddy@10.232.19.169/ser") # -- 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) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; # 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); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # 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("10.232.19.169", "subscriber")) { www_challenge("10.232.19.169", "0"); break; }; save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; 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(); }; } /

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Re: [Serusers] SER and Asterisk authentication
by Voipers Portugal 30 Nov '05

30 Nov '05

Will it work with SER or only for OpenSER? I'm sorry but I don't know the difference beteween those 2. Jose Simoes On 11/28/05, Jim Dalton <jim.dalton(a)transnexus.com> wrote: > > One alternative to consider is a certificate authority/peering server that > issues a digitally signed token authorizing each call between SER and > Asterisk. This eliminates the threat of IP spoofing and the need top > maintain IP access lists on each SER and Asterisk box. See > http://osp-module.berlios.de/. > > Jim Dalton > > > ------------------------------ > *From:* serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] *On > Behalf Of *Voipers Portugal > *Sent:* Friday, November 25, 2005 2:21 PM > *To:* sip(a)perceval.net > *Cc:* serusers(a)lists.iptel.org > *Subject:* Re: [Serusers] SER and Asterisk authentication > > > Thanks, i'll read it now. But this only solves one part of the problem, > that is, Asterisk can authenticate in SER. And how SER can authenticate in > Asterisk? Is it possible only by IP? > > Jose Simoes > > > On 11/25/05, Arek Bekiersz <sip(a)perceval.net> wrote: > > > > OK, I understand your problem. > > Please read <carefully> Asterisk Wiki for sip.conf at: > > > > http://www.voip-info.org/tiki-index.php?page=Asterisk+config+sip.conf > > > > > > To register, you just need to insert this into general section of > > Asterisk's sip.conf: > > > > [general] > > ; Register with a SIP provider - register username at SER as > > local_asterisk_extension at Asterisk > > > > register => > > username:digest_password@your.ser.server/local_asterisk_extension > > > > [your.ser.server] > > type=friend > > host=your_ser.server > > username=username > > > > > > -- > > Arek > > > > > > > > Voipers Portugal wrote: > > > But what do I have to add? I can add a user, with username Asterisk, > > but > > > how can I associate that user to the INVITE message from Asterisk to > > the > > > SER? Don't understand how SER can authenticate if i cannot associate > > the > > > RADIUS/LDAP user to the Asterisk box. > > > > > > Jose Simoes > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > >

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[Serusers] JAIN SIMPLE stack?
by Andrey Kouprianov 30 Nov '05

30 Nov '05

Hi! Does anyone know, when will JAIN IM and Presence (JSP164 and JSR165) implementaions become available? Is Panasonic the one to make it?

1 0

[Serusers] SER Bug? 503 messsage is not handled properly
by Pat wang 30 Nov '05

30 Nov '05

I have a test setup like: UAC1-----SER------Third Party Proxy----UAC2 UAC1 registers to SER directly while UAC2 registers to SER via a third party proxy. When UAC1 calls UAC2, the third party proxy replies a 100 trying then 503 Service Unavailable to SER. There is no Retry-After in the 503 message. SER keeps re-sending INVITE. By RFC, If no Retry-After is given, the client MUST act as if it had received a 500 (Server Internal Error) response. Is this a bug on SER? Thanks, Pat

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RE: [Serusers] Local ACKs for SER-SEMS
by Bjorn Asmul 30 Nov '05

30 Nov '05

I'm using latest stable SER (0.9.5.pre2), and we have problems with a Grandstream GXP-2000 phone with firmware 1.0.1.9. We are trying to initiate a call from a Cisco gateway via SER to the GXP. SER tries to set up the call, but there seem to be an ACK missing somewhere, so the Cisco keeps retrying the call. >From the trace I could see this message: sl_filter_ACK: to late to be a local ACK! Would this patch fix this problem? Also I did a re-compile from CVS (rev. rel_0_9_0) today. Would this patch have been applied? Thanks, Bjorn -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of SER.Jan Janak Sent: Tuesday, November 29, 2005 7:42 AM To: Cesc Cc: OpenSER-dev; SER-Users Subject: Re: [Serusers] Local ACKs for SER-SEMS Great, thanks a lot. I just commited your fix. Jan. On 28-11-2005 18:51, Cesc wrote: > Hi, > > I found it ... now it works. There is a pseudo-patch at the end of the > message (pseudo because it cannot directly be applied ... it is just > to mark the three lines i needed to change ... ). The patch is for > ser-rel-0.9.4 (i send it to openser-devel ... i did not look, but > probably the problem is there too). > > It is just a small bug ... from 0.9.0 to 0.9.4 a part of the loop was > moved outside the loop, to the end of the function ... and in a place > where there used to be a return now the loop would just continue ... > Then, the matching_3261 function would, if a transaction matching the > ACK was found, would always return e2e ack ... even though the > ack_matching function returned it as a local transaction ... > > For me, this patch fixes it ... but as it is a delicate part, i am not > sure if it affects other parts ... i don't think so, but ... > > Tks for all who took a look at the problem (greger, jan, samuel, ... )! > > Regards, > > Cesc > PS - This should affect anyone using SEMS ... so, if no one saw it ... > does it mean there is really a few people using it? what a pitty ... > > > Index: modules/tm/t_lookup.c > =================================================================== > --- modules/tm/t_lookup.c (revision 8) > +++ modules/tm/t_lookup.c (working copy) > > @@ -292,9 > struct via_body *via1; > int is_ack; > int dlg_parsed; > - int ret; > + int ret = 0; > struct cell *e2e_ack_trans; > > e2e_ack_trans=0; > via1=p_msg->via1; > is_ack=p_msg->REQ_METHOD==METHOD_ACK; > > @@ -327,16 > if (is_ack && p_cell->uas.status<300 && e2e_ack_trans==0) { > /* make sure we have parsed all things we need for dialog > * matching */ > if (!dlg_parsed) { > dlg_parsed=1; > if (!parse_dlg(p_msg)) { > LOG(L_ERR, "ERROR: matching_3261: dlg parsing failed\n"); > return 0; > } > } > ret=ack_matching(p_cell /* t w/invite */, p_msg /* ack */); > if (ret>0) { > e2e_ack_trans=p_cell; > + break; > } > /* this ACK is neither local "negative" one, nor a proxied > * end-2-end one, nor an end-2-end one for a UAS transaction > > @@ -358,9 > > /* just check if it we found an e2e ACK previously */ > if (e2e_ack_trans) { > *trans=e2e_ack_trans; > - return 2; > + return ret; > } > DBG("DEBUG: RFC3261 transaction matching failed\n"); > return 0; > } _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[Serusers] How to Implement 3 way call confrence in SER
by sagar 30 Nov '05

30 Nov '05

Hi All, Is it possible to Implement 3 way call confrence using SER. I am using ser 0.8.14 with asterisk b2bua. If it is possible from where can i get the help or config files. Thanks in advance Vidhya sagar Dixit --------------------------------- Yahoo! Music Unlimited - Access over 1 million songs. Try it free.

4 3

[Serusers] Repeated Register Message
by tcchan 30 Nov '05

30 Nov '05

Hi All, I encountered a situation that somehow a UA keep on Registering with the server with the same IP but on different port. So SER, thought that there are multiple instances of the same username and save the various location into the database. The problem is it registers so often that the memory is full not very long. I suspect it has something to do firewall but I am not sure. Has anyone encounter this problem before? Any hint?? For the time being, is there a way to limit the number of REGISTER message that the SER will take? Regards, TC Chan

3 2

[Users] sample configuration file for redirect server
by jyotisankar nayak 30 Nov '05

30 Nov '05

hi, i want to configure the openser to behave as a redurect server i.e when i send a request to the openser server it send back the request to client with zero or more addresses in the contact field. so can anybody pls send a sample configuration file which will force openser to behave as a redirect server. regards, jyoti sankar nayak. --------------------------------- Yahoo! DSL Something to write home about. Just $16.99/mo. or less

1 0

[Users] Forwarding from dispatcher
by Rick Richardson 30 Nov '05

30 Nov '05

I might be missing something obvious. But I have set up a dispatching openSER box using the recommended .cfg file from the dispatcher module docs and a custom list. On init, The parser complains of the forward(); function because it has no params. I looked at the core docs and they say they forward() requires a host and port, which makes sense, however, I don't know what to put in as a param as it is decided by the ds_select_dst() function. In short, what do I pass to forward to make it use the destination decided by ds_select_dst? thanks, Rick

2 1

[Users] error when i make a call!
by messa＠innsof.com 30 Nov '05

30 Nov '05

Hi all I have these errors when running my openser system: 0(21323) ERROR: t_newtran: transaction already in process 0xb616ba38 0(21323) ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) What do you think about it? Best regards! Serge

2 1

[Serusers] VoIP provider for SER?
by Dejan Cc 30 Nov '05

30 Nov '05

Dear friends, My company have used SER as backbone of our IP telephony system and Asterisks at subscriber's sites . Asterisks register to SER. Now, we would like to use SER to originate our international traffic to VoIP provider that would support such solution (connections from SER). Is there any VoIP provider that could provide us such service? If anyone knows any, please, help us. Any reply on this issue will be greatly appreciated. Best regards Dejan Cc

2 1

[Serusers] SER INVITE 200OK retransmission timer
by hpvpl 29 Nov '05

29 Nov '05

Hi, I'm performing performance test using SER 0.8.12 on linux red-hat with a default ser.cfg file (the one provided with installed package). I can't find in the documentation the default values used for SIP INVITE and 200OK retransmission. What are these default value ? Thanks for the support Cheers

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[Serusers] Modules mysql
by Marcel van Kaam, Fonetica Belgie 29 Nov '05

29 Nov '05

Hi All, Just checked on the list, found already some answers but still no luck. I use the getting started files from Onsip.org. Running Debian 2.6.12 ERROR: module modules/cpl/cpl.so not compiled ERROR: module modules/cpl-c/cpl-c.so not compiled ERROR: module modules/extcmd/extcmd.so not compiled ERROR: module modules/jabber/jabber.so not compiled ERROR: module modules/mysql/mysql.so not compiled When I use make all; make install, this is the outcome of my SER installation. I also tried make module etc. Is Debian the problem or do I need to make some changes to the Makefiles? Any help is welcome. Marcel

1 1

Re: [Serusers] Cisco 7960 SIP firmware
by Ladislav Andel 29 Nov '05

29 Nov '05

Thank you for pointing me at. I will look at it. Ladislav Atle Samuelsen wrote: > offtopic, and never said by me... > > broadvoice in USA has a tftpserver.. with the images on.. just point > your phone to it.. and wWops it's upgraded to atest SIP-image :-) > (ps, I've never done it myself) > > -A > > * Ladislav Andel <ladia6(a)centrum.cz> [051129 20:27]: > >> Hello, >> I'm sorry for this off topic but would anybody know where can I get SIP firmware for Cisco 7960 IPphone? >> What is the price or can anybody share, please? I'm speaking for my professor and am trying to get the firmware >> for our lab IP phones for the lab purposes, not any commercial use. It would help even me to complete and show off my results >> of my semester project. >> >> Thank you for any hints or redirecting. >> >> Ladislav >> >> >> _______________________________________________ >> Serusers mailing list >> serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >> >> > > >

1 1

[Serusers] Cisco 7960 SIP firmware
by Ladislav Andel 29 Nov '05

29 Nov '05

Hello, I'm sorry for this off topic but would anybody know where can I get SIP firmware for Cisco 7960 IPphone? What is the price or can anybody share, please? I'm speaking for my professor and am trying to get the firmware for our lab IP phones for the lab purposes, not any commercial use. It would help even me to complete and show off my results of my semester project. Thank you for any hints or redirecting. Ladislav

1 0

[Users] Is anyone using xtunnels successfully with openser....???
by Lokesh Kumar 29 Nov '05

29 Nov '05

Hello List, I just want to know if any one has been configured xtunnels successfully with openser and freeradius. Regards

1 0

[Serusers] How to Implement 3 way call confrence in SER.
by sagar 29 Nov '05

29 Nov '05

Hi All, Is it possible to Implement 3 way call confrence using SER. I am using ser 0.8.14 with asterisk b2bua. If it is possible from where can i get the help or config files. Thanks in advance Vidhya sagar Dixit --------------------------------- Yahoo! DSL Something to write home about. Just $16.99/mo. or less

1 0

[Users] postgres and openser 1.0.0
by JF 29 Nov '05

29 Nov '05

Hi, I'm trying to use postgres module with auth_db module in openser 1.0.0 but it is faling to load usrloc module. I get the following error: 0(0) find_mod_export: <db_use_table> in module mysql not found 0(0) bind_dbmod: Module mysql does not export db_use_table function 0(0) ERROR: mod_init(): Can't bind database module 0(0) init_mod(): Error while initializing module usrloc but I'm not using mysql module! In config file I have: # ------------------ module loading ---------------------------------- #loadmodule "/usr/local/openser-1.0.0/lib/openser/modules/mysql.so" loadmodule "/usr/local/openser-1.0.0/lib/openser/modules/postgres.so" loadmodule "/usr/local/openser-1.0.0/lib/openser/modules/sl.so" loadmodule "/usr/local/openser-1.0.0/lib/openser/modules/tm.so" loadmodule "/usr/local/openser-1.0.0/lib/openser/modules/rr.so" loadmodule "/usr/local/openser-1.0.0/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/openser-1.0.0/lib/openser/modules/usrloc.so" loadmodule "/usr/local/openser-1.0.0/lib/openser/modules/registrar.so" loadmodule "/usr/local/openser-1.0.0/lib/openser/modules/textops.so" # Uncomment this if you want digest authentication loadmodule "/usr/local/openser-1.0.0/lib/openser/modules/auth.so" loadmodule "/usr/local/openser-1.0.0/lib/openser/modules/auth_db.so" loadmodule "/usr/local/openser-1.0.0/lib/openser/modules/xlog.so" loadmodule "/usr/local/openser-1.0.0/lib/openser/modules/uri.so" loadmodule "/usr/local/openser-1.0.0/lib/openser/modules/lcr.so" Why is it searching in mysql module if I'm using postgres? Any help appreciated. JF

2 2

Re: [Serusers] Re: [Users] rfc3264 and nathelper module
by sip 29 Nov '05

29 Nov '05

On Tue, 29 Nov 2005 14:46:55 +0100 (CET), harry gaillac wrote > > REGISTER: > > if (!is_from_local()) { > sl_send_reply("401", "Unauthorized"); > break; > }; > rewritehostport("nxs.yi.org:5050"); > t_relay_to_udp("nxs.yi.org","5050"); So... you want your users to register both with Asterisk AND with SER? What for? Why not use some form of SER authorisation to control access to the Asterisk server? I'm not sure what you're trying to do here... > > INVITE: > rewritehostport("nxs.yi.org:5050"); What's the context around this? You want ALL calls to go straight to Asterisk? It just seems... I don't know... weird. I assume you're doing this because SER handles presence and Asterisk doesn't... but everything else you want to be controlled by Asterisk, including user registration, etc, etc. For your invite, btw, you would need to also have a t_relay_to_udp, or alternatively, you can do it as: rewritehostport("nxs.yi.org:5050"); forward(uri:host,uri:port); break; > > sip.conf: > [general] > context=local ; Default context for > incoming calls > ; if asterisk was > compiled with OSP support. > realm=nxs.yi.org ; Realm for digest > authentication > ; defaults to > "asterisk" > ; Realms MUST be > globally unique according to RFC 3261 > ; Set this to your > host name or domain name > bindport=5050 ; UDP Port to bind to > (SIP standard port is 5060) > bindaddr=nxs.yi.org ; IP address to bind > to (0.0.0.0 binds to all) > srvlookup=yes ; Enable DNS SRV > lookups on outbound calls > tos=lowdelay ; > lowdelay,throughput,reliability,mincost,none > maxexpirey=3600 ; Max length of > incoming registration we allow > defaultexpirey=1000 ; Default length of > incoming/outoing registration > allow=all ; First disallow all > codecs > musicclass=default ; Sets the default > music on hold class for all SIP calls > language=fr ; Default language > setting for all users/peers > rtptimeout=60 ; Terminate call if 60 > seconds of no RTP activity > tpholdtimeout=300 ; Terminate call if > 300 seconds of no RTP activity > useragent=Asterisk PBX ; Allows you to change > the user agent string > dtmfmode = rfc2833 ; Set default dtmfmode > for sending DTMF. Default: rfc2833 > promiscredir = no ; If yes, allows 302 > or REDIR to non-local SIP address > > [84] > type=friend > secret=84 > username=84 > context=local > host=dynamic > mailbox=84 > allow=all > > [85] > type=friend > secret=85 > username=85 > context=local > host=dynamic > mailbox=85 > allow=all > In your blocks there, you might want to add insecure=very to allow registered hosts to call without reauthenticating -- just to speed things up a little (optional) one IMPORTANT thing you want to add to each block just to make sure it's there is: canreinvite=yes This allows the RTP traffic to bypass asterisk and just go from UA to UA without taking up bandwidth. It will NOT, however, work well for NATted users without playing around with rewriting the SDP (using something like a nathelper fix_nated_sdp("3") call) Rereading your original emails, I now think I understand that this is what you were asking about before (correct me if I'm wrong)... was the fix_nated_sdp not working correctly in some way? N.

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[Users] cpl scripts and pstn gateway authentication
by Günther Starnberger 29 Nov '05

29 Nov '05

hello, what is the usual method to use the cpl module when a pstn gateway needs to know which user is responsible a call? e.g. i have a pstn gateway pstn.example. numbers are dialed by specifying them as user part of a sip uri => 00321234(a)pstn.example. as the gateway is an asterisk box i'm rather flexible in choosing a method which specifies how the name of the remote-user is transfered to the gateway. if a user does a forward to the pstn (e.g. in the case of no-answer) he should also be accounted for the call - so i either need to set the 'From' header to the owner of the cpl script or i need to add another header like 'X-account' which specifies the owner. a solution for this seems to be to store the name of the callee in an AVP before executing the script and to set the From field in proxy_route after the script has started. are there better or more-standard solutions for this problem? bye, /gst

2 4

[Users] LCR gw matching
by JF 29 Nov '05

29 Nov '05

Hi, Can anyone help clarify the following: According to LCR module docs for version 1.0.0, "gateway selection is based on caller's RPID URI (if available in caller's RPID AVP after authentication) or From URI and user part of Request-URI (telephone number)." What is the RPID AVP and how is it set? The docs also say "a gateway matches a request if user part of Request-URI and "From" URI match the prefix and From pattern of the gateway". But in gw table, there is no From pattern column, just prefix: gw_name | grp_id | ip_addr | port | uri_scheme | transport | prefix Also, it would be nice if gw matching could be based also on request method, request headers, or even arbitrary AVPs which would be set somewhere else. What do folks think about this? Thanks, JF

2 1

[Users] rfc3264 and nathelper module
by harry gaillac 29 Nov '05

29 Nov '05

Hello, Does nathelper module would support rfc3264 to provide MOH ? Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

10 19

[Serusers] domain and "myself" question
by Kyriakos Mavromichalis 29 Nov '05

29 Nov '05

Hi, currently i have a configuration that works when UA use ip adresseses for contacting the server. Also ip is used for the domain. What i would like users to do is use a domain ie sip1.otenet-telecom.net to register. Domain is setup A record and points to the ip of the server. Users register ok but when they try to make a call, if(uri!=myself) returns true and the call does not go through. uri is number(a)sip1.otenet-telecom.net sip1.otenet-telecom.net is in the domain mysql table. What do i do to get the server recognize the calls from users using the domain? thanks -- Kyriakos Mavromichalis Otenet Telecom

6 7

[Serusers] Local ACKs for SER-SEMS
by Cesc 29 Nov '05

29 Nov '05

Hi, In my lab we have SER (0.9.4) running, collocated in the same box with SEMS, communicating via the fifo. An abstraction of the config: //identify traffic for SEMS //send to route[x] route[x] { if( !t_newtran() ) { ..... } if( method == "ACK" ) { break;} //drop local acks ... t_reply("100", "msg"); send via fifo to sems ... } I have stumbled against some problems: P1) Local ACKs from the phone are not being dropped. P2) to-tag in replies from SER from sl_send_reply and OK message change P3) the t_newtran does not stop sending OKs, even it receives ACKs to them They all seem to be related ... the ACK from the phone back to SER, which are local(right?), are not dropped for two reasons: the sl_timeout is not updated, as i don't have an sl_send_reply in the route[x] block. I can fix the above by adding an sl_send_reply("183", "sems session in progress") ... but then, the second problem appears. Adding the sl_send_reply(183, ) adds a to-tag to the 183-reply, but then the following OK has a completely different to-tag ... which is the one used by my UA to create the ACK. As the to-tag in the ACK is different to that from the 183-reply ... the sl_filter_ack does not recognize the ACK as local, and does not filter it. In any case, this would not solve the problem of the t_newtran sending OK replies to the phone, even it receives ACKs to them. What can i do? should i use t_release somewhere? are these ACKs "local"? if not, where should these ACKs be routed to, to the route[x] block? Tks in advance, Cesc

4 11

[Serusers] Does SER support multithreading
by nitin ganjsinghani 29 Nov '05

29 Nov '05

hi friends does SER support POSIX compliant threads Thanks

2 1

[Users] RE: Request to mailing list Serusers rejected
by harry gaillac 29 Nov '05

29 Nov '05

serusers list moderators look like the ministere of the propaganda of Reinhard Goebel well known in germany! I'm tired of moderators Harry Gaillac from France --- serusers-bounces(a)iptel.org a écrit : > Your request to the Serusers mailing list > > Posting of your message titled "Re: [Users] > rfc3264 and nathelper > module" > > has been rejected by the list moderator. The > moderator gave the > following reason for rejecting your request: > > "Your message was deemed inappropriate by the > moderator." > > Any questions or comments should be directed to the > list administrator > at: > > serusers-owner(a)iptel.org > ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

4 4

Re: [Serusers] Please help with ser/nathelper
by Klaus Darilion 29 Nov '05

29 Nov '05

Hi Enky! Please always Cc: to the list! Setting up a dedicated outboundproxy (e.g. like Jasomi peerpoint or SNOM NAT filter) is not trivial. Typical this applications do all the NAT traversal and are transparent to SIP proxy. That means the SIP proxy does not have knowledge of the existence of such an outboundproxy. This is done by replacing the Contact: URI from the client with a URI pointing to the outboundproxy, and a random username (which will be mapped to the real client URI inside the outboundproxy). Some of them are real B2BUAs, some of them are some kind of proxy. All of them have in common, that they need to have a local "location table" to store the mapping from the real SIP clients to the SIP clients presented to the main SIP proxy. Another approach is the Path extension in RFC 3327 (http://ietf.org/rfc/rfc3327). This requires support by the main proxy, which has to store the location of the client plus the path to the client (i.e. the outbound proxy). Support for this extension is currently under development in the experimental tree (search the mailing list archive for more info). Back to the former approach: for REGISTER: ser as outboundproxy has to store the contact of the NATed client, and then replaces it with an unique identifier and send the message upstream to the main proxy. If the registration was sucessfull, the mapping will be valid. In the response, ser has to change the "virtual user" in the Contact: header back to the original Contact: header, as otherwise some clients get confused and reject the 200 Ok response. AFAIK there is no way to achieve this with ser without modifying code. Maybe AVPs can help, but i'm not sure. regards klaus asterisk(a)bgopentel.com wrote: > Dear Klaus, > thank you for your attention. > Unfortunately I do not know how to handle this issue. I use a mysql module in the main proxy to store the user data. The main proxy is running on a preconfigured FC1 Linux with a specialized software, installed on it. This software is not related to the SER and VoIP at all, but I can not change the installed software versions and that's why the SER version is rather old, but I can not upgrade it. So, I decided to run a separate proxy to only handle the NAT issues. > Please tell me what I have to configure in the main and in the outbound proxy to have it working? Thank you very much. I appreciate your help and attention. > > > > Klaus Darilion wrote .. > >>How do you make sure that the main proxy sends the INVITE to the >>outbound proxy, and the outbound proxy sends the request to the client? >> >>where do you store the contact information? in the main proxy or in the >>outbound proxy? >> >>regards >>klaus >> >>Enky wrote: >> >>>Hi, >>>I am trying to run the SER as an outbound proxy. Unfortunately I have >>>some problems. I have used the nathelper/rtpproxy with a customized >>>ser.conf and the result is that the clients behind NAT are making >>>outbound calls with no problems. The audio is two-way and all seems >>>best, but when I try to react a client behind NAT I can not. It is >>>registered in the SIP Proxy, but I can not dial it. >>>My scenario is: SER with MySQL authorization on the first PC and SER >>>with nathelper/rtpproxy for outbound proxy on second PC. >>>Could someone give me some hint as I can not solve this problem. The >>>Outbound Proxy SER version is: >>>version: ser 0.9.3 (i386/linux) >>>flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, >>>DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT >>>ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, >>>MAX_URI_SIZE 1024, BUF_SIZE 65535 >>>@(#) $Id: main.c,v 1.197 2004/12/03 19:09:31 andrei Exp $ >>>main.c compiled on 20:51:59 Jun 28 2005 with gcc 3.3 >>> >>>The ser.cfg of the Outbound Proxy SER is: >>># >>># $Id: nathelper.cfg,v 1.1 2003/11/10 14:15:36 janakj Exp $ >>># >>># simple quick-start config script including nathelper support >>> >>># This default script includes nathelper support. To make it work >>># you will also have to install Maxim's RTP proxy. The proxy is enforced >>># if one of the parties is behind a NAT. >>># >>># If you have an endpoing in the public internet which is known to >>># support symmetric RTP (Cisco PSTN gateway or voicemail, for example), >>># then you don't have to force RTP proxy. If you don't want to enforce >>># RTP proxy for some destinations than simply use t_relay() instead of >>># route(1) >>># >>># Sections marked with !! Nathelper contain modifications for nathelper >>># >>># NOTE !! This config is EXPERIMENTAL ! >>># >>># ----------- global configuration parameters ------------------------ >>> >>>debug=3 # debug level (cmd line: -dddddddddd) >>>fork=yes >>>log_stderror=no # (cmd line: -E) >>> >>>/* Uncomment these lines to enter debugging mode >>>fork=no >>>log_stderror=yes >>>*/ >>> >>>check_via=no # (cmd. line: -v) >>>dns=no # (cmd. line: -r) >>>rev_dns=no # (cmd. line: -R) >>>port=5082 >>>children=4 >>>fifo="/tmp/ser_fifo" >>> >>># ------------------ module loading ---------------------------------- >>> >>># Uncomment this if you want to use SQL database >>>#loadmodule "/usr/local/lib/ser/modules/mysql.so" >>> >>>loadmodule "/usr/local/lib/ser/modules/sl.so" >>>loadmodule "/usr/local/lib/ser/modules/tm.so" >>>loadmodule "/usr/local/lib/ser/modules/rr.so" >>>loadmodule "/usr/local/lib/ser/modules/maxfwd.so" >>>loadmodule "/usr/local/lib/ser/modules/usrloc.so" >>>loadmodule "/usr/local/lib/ser/modules/registrar.so" >>>loadmodule "/usr/local/lib/ser/modules/textops.so" >>> >>># Uncomment this if you want digest authentication >>># mysql.so must be loaded ! >>>#loadmodule "/usr/local/lib/ser/modules/auth.so" >>>#loadmodule "/usr/local/lib/ser/modules/auth_db.so" >>> >>># !! Nathelper >>>loadmodule "/usr/local/lib/ser/modules/nathelper.so" >>> >>># ----------------- setting module-specific parameters --------------- >>> >>># -- usrloc params -- >>> >>>modparam("usrloc", "db_mode", 0) >>> >>># Uncomment this if you want to use SQL database >>># for persistent storage and comment the previous line >>>#modparam("usrloc", "db_mode", 2) >>> >>># -- 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) >>> >>># !! Nathelper >>>modparam("registrar", "nat_flag", 6) >>>modparam("nathelper", "natping_interval", 60) # Ping interval 60 s >>>modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind >> >>NAT >> >>> >>># ------------------------- request routing logic ------------------- >>> >>># main routing logic >>> >>>route{ >>> >>> # initial sanity checks -- messages with >>> # max_forwards==0, or excessively long requests >>> if (!mf_process_maxfwd_header("10")) { >>> sl_send_reply("483","Too Many Hops"); >>> break; >>> }; >>> if (msg:len >= max_len ) { >>> sl_send_reply("513", "Message too big"); >>> break; >>> }; >>> >>> # !! Nathelper >>> # Special handling for NATed clients; first, NAT test is >>> # executed: it looks for via!=received and RFC1918 addresses >>> # in Contact (may fail if line-folding is used); also, >>> # the received test should, if completed, should check all >>> # vias for rpesence of received >>> if (nat_uac_test("3")) { >>> # Allow RR-ed requests, as these may indicate that >>> # a NAT-enabled proxy takes care of it; unless it is >>> # a REGISTER >>> >>> if (method == "REGISTER" || ! search("^Record-Route:")) { >>> log("LOG: Someone trying to register from private IP, rewriting\n"); >>> >>> # This will work only for user agents that support symmetric >>> # communication. We tested quite many of them and majority is >>> # smart enough to be symmetric. In some phones it takes a >>>configuration >>> # option. With Cisco 7960, it is called NAT_Enable=Yes, with >>>kphone it is >>> # called "symmetric media" and "symmetric signalling". >>> >>> fix_nated_contact(); # Rewrite contact with source IP of signalling >>> if (method == "INVITE") { >>> fix_nated_sdp("1"); # Add direction=active to SDP >>> }; >>> force_rport(); # Add rport parameter to topmost Via >>> setflag(6); # Mark as NATed >>> }; >>> }; >>> >>> # 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); >>> break; >>> }; >>> >>> if (!uri==myself) { >>> # mark routing logic in request >>> append_hf("P-hint: outbound\r\n"); >>> route(1); >>> break; >>> }; >>> >>> # 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") { >>> >>> save("location"); >>> break; >>> }; >>> >>> lookup("aliases"); >>> if (!uri==myself) { >>> append_hf("P-hint: outbound alias\r\n"); >>> route(1); >>> break; >>> }; >>> >>> # native SIP destinations are handled using our USRLOC DB >>> if (!lookup("location")) { >>> sl_send_reply("404", "Not Found"); >>> break; >>> }; >>> }; >>> append_hf("P-hint: usrloc applied\r\n"); >>> route(1); >>>} >>> >>>route[1] >>>{ >>> # !! Nathelper >>> if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && >>>!search("^Route:")){ >>> sl_send_reply("479", "We don't forward to private IP addresses"); >>> break; >>> }; >>> >>> # if client or server know to be behind a NAT, enable relay >>> if (isflagset(6)) { >>> force_rtp_proxy(); >>> }; >>> >>> # NAT processing of replies; apply to all transactions (for example, >>> # re-INVITEs from public to private UA are hard to identify as >>> # NATed at the moment of request processing); look at replies >>> t_on_reply("1"); >>> >>> # send it out now; use stateful forwarding as it works reliably >>> # even for UDP2TCP >>> if (!t_relay()) { >>> sl_reply_error(); >>> }; >>>} >>> >>># !! Nathelper >>>onreply_route[1] { >>> # NATed transaction ? >>> if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { >>> fix_nated_contact(); >>> if (!search("^Content-Length:\ 0")) { >>> force_rtp_proxy(); >>> }; >>> # otherwise, is it a transaction behind a NAT and we did not >>> # know at time of request processing ? (RFC1918 contacts) >>> } else if (nat_uac_test("1")) { >>> fix_nated_contact(); >>> }; >>>} >>> >>> >>> >>> >>>------------------------------------------------------------------------ >>> >>>_______________________________________________ >>>Serusers mailing list >>>serusers(a)lists.iptel.org >>>http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] Random Value in Custom Headers
by Andrey Kouprianov 29 Nov '05

29 Nov '05

Hi all: Actually, I already posted this kind of question, but once again: can I configure SER to leave custom headers in responses, because SER seems to strip them off. The problem is that my custom headers are not the same all the time. For instance, X-Header: blah <random value> This <random value> is always changing, that's why I cannot use append_hf() function. Can I somehow make SER leave custom headers untouched?

2 1

[Serusers] Please help with ser/nathelper
by Enky 29 Nov '05

29 Nov '05

Hi, I am trying to run the SER as an outbound proxy. Unfortunately I have some problems. I have used the nathelper/rtpproxy with a customized ser.conf and the result is that the clients behind NAT are making outbound calls with no problems. The audio is two-way and all seems best, but when I try to react a client behind NAT I can not. It is registered in the SIP Proxy, but I can not dial it. My scenario is: SER with MySQL authorization on the first PC and SER with nathelper/rtpproxy for outbound proxy on second PC. Could someone give me some hint as I can not solve this problem. The Outbound Proxy SER version is: version: ser 0.9.3 (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 @(#) $Id: main.c,v 1.197 2004/12/03 19:09:31 andrei Exp $ main.c compiled on 20:51:59 Jun 28 2005 with gcc 3.3 The ser.cfg of the Outbound Proxy SER is: # # $Id: nathelper.cfg,v 1.1 2003/11/10 14:15:36 janakj Exp $ # # simple quick-start config script including nathelper support # This default script includes nathelper support. To make it work # you will also have to install Maxim's RTP proxy. The proxy is enforced # if one of the parties is behind a NAT. # # If you have an endpoing in the public internet which is known to # support symmetric RTP (Cisco PSTN gateway or voicemail, for example), # then you don't have to force RTP proxy. If you don't want to enforce # RTP proxy for some destinations than simply use t_relay() instead of # route(1) # # Sections marked with !! Nathelper contain modifications for nathelper # # NOTE !! This config is EXPERIMENTAL ! # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5082 children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/local/lib/ser/modules/auth.so" #loadmodule "/usr/local/lib/ser/modules/auth_db.so" # !! Nathelper loadmodule "/usr/local/lib/ser/modules/nathelper.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) # -- 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) # !! Nathelper modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 60) # Ping interval 60 s modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); break; }; # !! Nathelper # Special handling for NATed clients; first, NAT test is # executed: it looks for via!=received and RFC1918 addresses # in Contact (may fail if line-folding is used); also, # the received test should, if completed, should check all # vias for rpesence of received if (nat_uac_test("3")) { # Allow RR-ed requests, as these may indicate that # a NAT-enabled proxy takes care of it; unless it is # a REGISTER if (method == "REGISTER" || ! search("^Record-Route:")) { log("LOG: Someone trying to register from private IP, rewriting\n"); # This will work only for user agents that support symmetric # communication. We tested quite many of them and majority is # smart enough to be symmetric. In some phones it takes a configuration # option. With Cisco 7960, it is called NAT_Enable=Yes, with kphone it is # called "symmetric media" and "symmetric signalling". fix_nated_contact(); # Rewrite contact with source IP of signalling if (method == "INVITE") { fix_nated_sdp("1"); # Add direction=active to SDP }; force_rport(); # Add rport parameter to topmost Via setflag(6); # Mark as NATed }; }; # 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); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # 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") { save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # !! Nathelper if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && !search("^Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); break; }; # if client or server know to be behind a NAT, enable relay if (isflagset(6)) { force_rtp_proxy(); }; # NAT processing of replies; apply to all transactions (for example, # re-INVITEs from public to private UA are hard to identify as # NATed at the moment of request processing); look at replies t_on_reply("1"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } # !! Nathelper onreply_route[1] { # NATed transaction ? if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { fix_nated_contact(); if (!search("^Content-Length:\ 0")) { force_rtp_proxy(); }; # otherwise, is it a transaction behind a NAT and we did not # know at time of request processing ? (RFC1918 contacts) } else if (nat_uac_test("1")) { fix_nated_contact(); }; }

2 1

[Serusers] domain trouble
by George Lambson 29 Nov '05

29 Nov '05

For some reason my SER isn't using the sip domain I specify (export SIP_DOMAIN="whatever.edu"). When my UA registers it will only register if I take out all of the digect authentication from my config file (I leave all of them blank instead of putting in a domain) and then use the host name of my server instead of a domain. Any ideas what I am doing wrong? Thanks, George

2 1

[Users] a sample .cfg file for proxy server
by jyotisankar nayak 29 Nov '05

29 Nov '05

hi can u pls send me a sample confuguration file so that i can configure a proxy server. my requerment is simple.. i want to send request to a server through openser regards jyoti --------------------------------- Yahoo! Music Unlimited - Access over 1 million songs. Try it free.

2 1

[Users] domainkeys module
by Klaus Darilion 29 Nov '05

29 Nov '05

Hi! I've just posted a domainkeys module on the tracker on sourceforge: http://sourceforge.net/tracker/index.php?func=detail&aid=1368417&group_id=1… This is a "proof-of-concept" and at the moment not a module for real usage. Further, this is something which not standardized or documented anywhere. It's just an experiment to use the domainkeys technology (invented as anti-SPAM technology for emails) also with SIP. Following some snippets from the README file. If you would like to discuss this idea with me, you'r welcome. regards klaus Chapter 1. User's Guide 1.1. Overview This module is a proof-of-concept. It tries to use the domainkeys technologie, for signing SIP message. Domainkeys were developed as an anti-SPAM technology for emails. But as SIP is familiar with email, it might also be useful to use domainkeys for signing SIP messages. _________________________________________________________ 1.1.1. Installation Just copy the module into openser's module directory. This module needs the libdomainkeys library and the libcrypto library (from openssl). Thus make sure to have openssl installed. If the domainkeys library is not installed in the default directories, you have to modify the library path in the Makefile to point to the location of the domainkeys module and include this directory to /etc/ld.so.conf. Further you have to adopt the inlcude path of #include "/home/darilion/libdomainkeys-0.68/domainkeys.h" in the file domainkeys.c. Then compile and use the module as usual. _________________________________________________________ 1.1.2. Known Limitations * The current version signs only the From: and To: header, and the message body. * This module is far from stable and seriously contains several bugs. * Signing and validation takes time. On a 2GHz Athlon CPU, the signing off an INVITE message (From:, To: and body) took ~8ms. The validation of this message took ~9ms. Additionally, the validation takes time as at least one DNS lookup is necessary to fetch the public key of the signing domain. This can be reduced by a fast resolving name server and long TTLs for the public key. * This module uses the original received SIP message for calculating the message signatur. Thus, if you use function which modify the used headers (e.g. uac module) or which modify the body (e.g. nathelper module), then the added signature will be invalid. 1.3. Exported Parameters 1.3.1. privkey_file (string) Path to the file containing the private key for the signing. (The public key for the validation will be fetched from DNS) Default value is "". Example 1-1. privkey_file example modparam("domainkeys", "privkey_file", "/etc/domainkeys/privkeys/2005") _________________________________________________________ 1.4. Exported Functions 1.4.1. domainkeys_validate Signs the SIP message. This function can be used from REQUEST_ROUTE. Example 1-2. domainkeys_validate usage ... if (is_method("INVITE")) { if (domainkeys_sign()) { xlog("L_INFO","domainkeys - signing: SUCCESS\n"); } else { xlog("L_INFO","domainkeys - signing: FAILURE\n"); } } ... _________________________________________________________ 1.4.2. domainkeys_validate Validates the Domainkeys-Signature in the received SIP message. This function can be used from REQUEST_ROUTE. Example 1-3. search_append usage ... if (is_present_hf("Domainkey-Signature")) { xlog("L_INFO","Domainkey-Signature found, start validating...\n"); if (domainkeys_validate()) { xlog("L_INFO","domainkeys - validate: SUCCESS\n"); } else { xlog("L_INFO","domainkeys - validate: FAILURE\n"); } } ...

2 2

[Users] CANCEL problem
by Klaus Darilion 29 Nov '05

29 Nov '05

Hi! looks like there is a problem if the CANCEL is received during DNS lookup of the ongoing transaction. The CANCEL is answered with 200 Ok, but the cancel will not send upstream, although the INVITE is sent. No 487 either. My CANCEL config: if ( is_method("CANCEL") && !t_check_trans() ) { # CANCEL without matching INVITE transaction, ignore # may happen if the INVITE is slower than the CANCEL # ignore the CANCEL, as the client will retransmit it, and maybe next time # the INVITE transaction is already created xlog("L_WARN","CANCEL without matching transaction ... ignore and discard.\n"); exit; } if ( is_method("CANCEL") ) { # CANCEL with matching INVITE transaction, just # t_relay xlog("L_INFO","CANCEL with matching transaction ... t_relay.\n"); t_relay(); exit; below are the SIP call flow and the syslog output regards klaus openser[3670]: SIP Request: openser[3670]: method: <CANCEL> openser[3670]: uri: <sip:klaus3000@iptel.org> openser[3670]: version: <SIP/2.0> openser[3670]: parse_headers: flags=2 openser[3670]: Found param type 232, <branch> = <z9hG4bK-d87543-5b192d3a37574d15-1--d87543->; state=6 openser[3670]: Found param type 235, <rport> = <n/a>; state=17 openser[3670]: end of header reached, state=5 openser[3670]: parse_headers: Via found, flags=2 openser[3670]: parse_headers: this is the first via openser[3670]: After parse_msg... openser[3670]: preparing to run routing scripts... openser[3670]: parse_headers: flags=100 openser[3670]: DEBUG:parse_to:end of header reached, state=9 openser[3670]: DEBUG: get_hdr_field: <To> [27]; uri=[sip:klaus3000@iptel.org] openser[3670]: DEBUG: to body [<sip:klaus3000@iptel.org>^M ] openser[3670]: get_hdr_field: cseq <CSeq>: <1> <CANCEL> openser[3670]: DEBUG: get_hdr_body : content_length=0 openser[3670]: found end of header openser[3670]: DEBUG: is_maxfwd_present: max_forwards header not found! openser[3670]: parse_headers: flags=ffffffffffffffff openser[3670]: DEBUG: add_param: tag=a83eae01 openser[3670]: DEBUG:parse_to:end of header reached, state=29 openser[3670]: parse_headers: flags=200 openser[3670]: find_first_route: No Route headers found openser[3670]: loose_route: There is no Route HF openser[3670]: parse_headers: flags=78 openser[3670]: DEBUG: t_lookupOriginalT: searching on hash entry 62046 openser[3670]: DEBUG: RFC3261 transaction matched, tid=-d87543-5b192d3a37574d15-1--d87543- openser[3670]: DEBUG: t_lookupOriginalT: canceled transaction found (0x40484280)! openser[3670]: DEBUG: t_lookupOriginalT completed openser[3670]: xl_printf: final buffer length 46 openser[3670]: CANCEL with matching transaction ... t_relay. openser[3670]: DEBUG: t_newtran: msg id=3 , global msg id=2 , T on entrance=0xffffffff openser[3670]: parse_headers: flags=ffffffffffffffff openser[3670]: parse_headers: flags=78 openser[3670]: t_lookup_request: start searching: hash=62046, isACK=0 openser[3670]: DEBUG: RFC3261 transaction matching failed openser[3670]: DEBUG: t_lookup_request: no transaction found openser[3670]: DEBUG: t_lookupOriginalT: searching on hash entry 62046 openser[3670]: DEBUG: RFC3261 transaction matched, tid=-d87543-5b192d3a37574d15-1--d87543- openser[3670]: DEBUG: t_lookupOriginalT: canceled transaction found (0x40484280)! openser[3670]: DEBUG: t_lookupOriginalT completed openser[3670]: DEBUG: e2e_cancel: e2e cancel -- no more pending branches openser[3670]: parse_headers: flags=ffffffffffffffff openser[3670]: check_via_address(83.136.33.3, 83.136.33.3, 0) openser[3670]: WARNING:vqm_resize: resize(0) called openser[3670]: DEBUG: cleanup_uac_timers: RETR/FR timers reset openser[3670]: DEBUG: add_to_tail_of_timer[2]: 0x40486168 openser[3670]: DEBUG:tm:_reply_light: reply sent out. buf=0x8117520: SIP/2.0 2..., shmem=0x40485d00: SIP/2.0 2 openser[3670]: DEBUG:tm:_reply_light: finished openser[3670]: SER: new transaction fwd'ed openser[3670]: DEBUG:destroy_avp_list: destroying list (nil) openser[3670]: receive_msg: cleaning up U 2005/11/28 13:55:39.652279 83.136.33.3:8722 -> 213.239.207.29:5060 INVITE sip:klaus3000@iptel.org SIP/2.0. Via: SIP/2.0/UDP 83.136.33.3:8722;branch=z9hG4bK-d87543-5b192d3a37574d15-1--d87543-;rport. Max-Forwards: 70. Contact: <sip:klaus@83.136.33.3:8722>. To: <sip:klaus3000@iptel.org>. From: "klaus dk.labs.nic.at"<sip:klaus@dk.labs.nic.at>;tag=a83eae01. Call-ID: 9843d9424930591d@a2xhdXNwYw... CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: eventlist. User-Agent: eyeBeam release 3010n stamp 19039. Content-Length: 201. . v=0. o=- 15208201 15208322 IN IP4 83.136.33.3. s=eyeBeam. c=IN IP4 83.136.33.3. t=0 0. m=audio 8468 RTP/AVP 0 8 3 18 101. a=fmtp:101 0-15. a=ptime:18 40. a=rtpmap:101 telephone-event/8000. a=sendrecv. # U 2005/11/28 13:55:39.660699 213.239.207.29:5060 -> 83.136.33.3:8722 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 83.136.33.3:8722;branch=z9hG4bK-d87543-5b192d3a37574d15-1--d87543-;rport=8722. To: <sip:klaus3000@iptel.org>. From: "klaus dk.labs.nic.at"<sip:klaus@dk.labs.nic.at>;tag=a83eae01. Call-ID: 9843d9424930591d@a2xhdXNwYw... CSeq: 1 INVITE. Server: OpenSer (1.0.0 (i386/linux)). Content-Length: 0. Warning: 392 213.239.207.29:5060 "Noisy feedback tells: pid=3669 req_src_ip=83.136.33.3 req_src_port=8722 in_uri=sip:klaus3000@iptel.org out_uri=sip:klaus3000@iptel.org via_cnt==1". . # U 2005/11/28 13:55:39.661561 213.239.207.29:32792 -> 213.133.98.96:53 ............._sip._udp.iptel.org..!.. # U 2005/11/28 13:55:39.718959 213.133.98.96:53 -> 213.239.207.29:32792 ............._sip._udp.iptel.org..!.....!....N%.........sip.iptel.org.........N%...ns7 fraunhofer.de.........N%...ns1.V........N%...ns2.V........N%...ns3.V........N%...ns5.V.sip........N&...%Mc.R.......>...6"..q......"....,%..............`.. # U 2005/11/28 13:55:39.719287 213.239.207.29:32792 -> 213.133.98.96:53 .............sip.iptel.org..... # U 2005/11/28 13:55:39.913552 83.136.33.3:8722 -> 213.239.207.29:5060 CANCEL sip:klaus3000@iptel.org SIP/2.0. Via: SIP/2.0/UDP 83.136.33.3:8722;branch=z9hG4bK-d87543-5b192d3a37574d15-1--d87543-;rport. To: <sip:klaus3000@iptel.org>. From: "klaus dk.labs.nic.at"<sip:klaus@dk.labs.nic.at>;tag=a83eae01. Call-ID: 9843d9424930591d@a2xhdXNwYw... CSeq: 1 CANCEL. User-Agent: eyeBeam release 3010n stamp 19039. Content-Length: 0. . # U 2005/11/28 13:55:39.918736 213.239.207.29:5060 -> 83.136.33.3:8722 SIP/2.0 200 ok -- no more pending branches. Via: SIP/2.0/UDP 83.136.33.3:8722;branch=z9hG4bK-d87543-5b192d3a37574d15-1--d87543-;rport=8722. To: <sip:klaus3000@iptel.org>;tag=fa997f81440371de71ab448ebdb9af56-3f94. From: "klaus dk.labs.nic.at"<sip:klaus@dk.labs.nic.at>;tag=a83eae01. Call-ID: 9843d9424930591d@a2xhdXNwYw... CSeq: 1 CANCEL. Server: OpenSer (1.0.0 (i386/linux)). Content-Length: 0. Warning: 392 213.239.207.29:5060 "Noisy feedback tells: pid=3670 req_src_ip=83.136.33.3 req_src_port=8722 in_uri=sip:klaus3000@iptel.org out_uri=sip:klaus3000@iptel.org via_cnt==1". . # U 2005/11/28 13:55:40.083745 213.133.98.96:53 -> 213.239.207.29:32792 .............sip.iptel.org.............N&...%Mc........N%...ns7 fraunhofer.de.........N%...ns1.?........N%...ns2.?........N%...ns3.?........N%...ns5.?.;.......>...6"..Z......"....,%..l...........`.. # U 2005/11/28 13:55:40.084090 213.239.207.29:5060 -> 195.37.77.99:5060 INVITE sip:klaus3000@iptel.org SIP/2.0. Record-Route: <sip:213.239.207.29;ftag=a83eae01;lr=on>. DomainKey-Signature: a=rsa-sha1; q=dns; c=simple; s=2005; d=dk.labs.nic.at; h=To:From; b=gQHw1V0nzlaGJgDtZX3DY+9MCck/NlieDzMnVseDjJALaG6/vtgZG98jPtas6CZx. Via: SIP/2.0/UDP 213.239.207.29;branch=z9hG4bKe52f.c88e2d42.0. Via: SIP/2.0/UDP 83.136.33.3:8722;branch=z9hG4bK-d87543-5b192d3a37574d15-1--d87543-;rport=8722. Max-Forwards: 69. Contact: <sip:klaus@83.136.33.3:8722>. To: <sip:klaus3000@iptel.org>. From: "klaus dk.labs.nic.at"<sip:klaus@dk.labs.nic.at>;tag=a83eae01. Call-ID: 9843d9424930591d@a2xhdXNwYw... CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: eventlist. User-Agent: eyeBeam release 3010n stamp 19039. Content-Length: 201. P-hint: outbound. . v=0. o=- 15208201 15208322 IN IP4 83.136.33.3. s=eyeBeam. c=IN IP4 83.136.33.3. t=0 0. m=audio 8468 RTP/AVP 0 8 3 18 101. a=fmtp:101 0-15. a=ptime:18 40. a=rtpmap:101 telephone-event/8000. a=sendrecv. # U 2005/11/28 13:55:40.105264 195.37.77.99:5060 -> 213.239.207.29:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 213.239.207.29;branch=z9hG4bKe52f.c88e2d42.0;rport=5060. Via: SIP/2.0/UDP 83.136.33.3:8722;branch=z9hG4bK-d87543-5b192d3a37574d15-1--d87543-;rport=8722. To: <sip:klaus3000@iptel.org>. From: "klaus dk.labs.nic.at"<sip:klaus@dk.labs.nic.at>;tag=a83eae01. Call-ID: 9843d9424930591d@a2xhdXNwYw... CSeq: 1 INVITE. Server: Sip EXpress router (0.9.0udpfifo (i386/linux)). Content-Length: 0. Warning: 392 195.37.77.99:5060 "Noisy feedback tells: pid=16799 req_src_ip=213.239.207.29 req_src_port=5060 in_uri=sip:klaus3000@iptel.org out_uri=sip:klaus3000@192.168.2.3:5061 via_cnt==2". . # U 2005/11/28 13:55:40.956023 195.37.77.99:5060 -> 213.239.207.29:5060 SIP/2.0 180 Ringing. To: <sip:klaus3000@iptel.org>;tag=5afee18c3cf2167ei1. From: "klaus dk.labs.nic.at"<sip:klaus@dk.labs.nic.at>;tag=a83eae01. Call-ID: 9843d9424930591d@a2xhdXNwYw... CSeq: 1 INVITE. Via: SIP/2.0/UDP 213.239.207.29;rport=5060;branch=z9hG4bKe52f.c88e2d42.0. Via: SIP/2.0/UDP 83.136.33.3:8722;branch=z9hG4bK-d87543-5b192d3a37574d15-1--d87543-;rport=8722. Record-Route: <sip:195.37.77.99;ftag=a83eae01;lr=on>. Record-Route: <sip:213.239.207.29;ftag=a83eae01;lr=on>. Server: Sipura/SPA2000-3.1.2(NTb). Content-Length: 0. P-NAT-Check: YES. . # U 2005/11/28 13:55:40.958703 213.239.207.29:5060 -> 83.136.33.3:8722 SIP/2.0 180 Ringing. To: <sip:klaus3000@iptel.org>;tag=5afee18c3cf2167ei1. From: "klaus dk.labs.nic.at"<sip:klaus@dk.labs.nic.at>;tag=a83eae01. Call-ID: 9843d9424930591d@a2xhdXNwYw... CSeq: 1 INVITE. Via: SIP/2.0/UDP 83.136.33.3:8722;branch=z9hG4bK-d87543-5b192d3a37574d15-1--d87543-;rport=8722. Record-Route: <sip:195.37.77.99;ftag=a83eae01;lr=on>. Record-Route: <sip:213.239.207.29;ftag=a83eae01;lr=on>. Server: Sipura/SPA2000-3.1.2(NTb). Content-Length: 0. P-NAT-Check: YES. .

2 2

[Serusers] Does SER support TLS?
by Chueh, Weihsin 29 Nov '05

29 Nov '05

Hi All, I download the latest version of SER, 0.9.4, but do not see anything about TLS in ser.cfg file. Does SER support TLS? If it does, how to configure and install SER with TLS feature ON? Thanks, Weihsin Chueh HMP Function Test, Intel 1515 Route 10, Parsippany, NJ 07045 973-967-6243 mailto: <mailto:weihsin.chueh@intel.com> weihsin.chueh(a)intel.com

3 2

[Serusers] rfc3264 and nathelper module
by harry gaillac 28 Nov '05

28 Nov '05

Hello, Does nathelper module would support rfc3264 to provide MOH ? Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

2 1

[Users] OpenSER 1.0.0 compile failure?
by Matt Schulte 28 Nov '05

28 Nov '05

All, I browsed around on google and saw no mention of this problem. When compiling I get: gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"openser"' -DVERSION='"1.0.0"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.3"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local/etc/openser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_MCAST -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c msg_translator.c -o msg_translator.o msg_translator.c: In function `build_req_buf_from_sip_req': msg_translator.c:1339: error: `id_buf' undeclared (first use in this function) msg_translator.c:1339: error: (Each undeclared identifier is reported only once msg_translator.c:1339: error: for each function it appears in.) msg_translator.c:1340: error: `id_len' undeclared (first use in this function) make: *** [msg_translator.o] Error 1 Simple enough error, any suggestions? I am running: Fedora Core 2 (yum updated of course) gcc 3.3.3 Thanks!

2 2

[Serusers] RADIATOR + LDAP +SER
by Alan Baker 28 Nov '05

28 Nov '05

I configured SER for authentication through RADIATOR to LDAP. I am kind of stumped in regards to the User-Name attribute that is given to the Radius server with the realm attached. Is there a way to send only the username portion? I have included the output from debug. I would appropriate any information. Thanks, Alan -------------------------------------------------------------------------- Mon Nov 28 14:32:44 2005: DEBUG: Reading dictionary file '/etc/radiator/dictionary' Mon Nov 28 14:32:44 2005: DEBUG: Creating authentication port 0.0.0.0:1645 Mon Nov 28 14:32:44 2005: DEBUG: Creating accounting port 0.0.0.0:1646 Mon Nov 28 14:32:44 2005: NOTICE: Server started: Radiator 3.13 on sip (LOCKED) Mon Nov 28 14:32:51 2005: DEBUG: Packet dump: *** Received from 127.0.0.1 port 33032 .... Code: Access-Request Identifier: 247 Authentic: <210>E<8>@m<211>h<133><34><23><16><21><28><242><187><154> Attributes: User-Name = "alan(a)sip.domain.com" Digest-Attributes = "<10><6>alan" Digest-Attributes = "<1><19>sip.domain.com" Digest-Attributes = "<2>*438b5c8fa0ca1bassjsjskakad74ec2c2bd0" Digest-Attributes = "<4><23>sip:sip.domain.com" Digest-Attributes = "<3><10>REGISTER" Digest-Response = "c469a2464876sska898jkdsjs889" Service-Type = Sip-Session Sip-Uri-User = "alan" NAS-IP-Address = UNKNOWN NAS-Port = 5060 Mon Nov 28 14:32:51 2005: DEBUG: Handling request with Handler 'Realm=DEFAULT' Mon Nov 28 14:32:51 2005: DEBUG: Deleting session for alan(a)sip.domain.com, UNKNOWN, 5060 Mon Nov 28 14:32:51 2005: DEBUG: Handling with Radius::AuthLDAP2: Mon Nov 28 14:32:51 2005: INFO: Connecting to ldap.domain.com, port 389 Mon Nov 28 14:32:51 2005: INFO: Attempting to bind to LDAP server ldap.domain.com:389 Mon Nov 28 14:32:51 2005: ERR: ldap search failed with error LDAP_PARAM_ERROR. Mon Nov 28 14:32:51 2005: ERR: LDAP_PARAM_ERROR Mon Nov 28 14:32:51 2005: DEBUG: Radius::AuthLDAP2 looks for match with alan(a)sip.domain.com Mon Nov 28 14:32:51 2005: ERR: ldap search failed with error LDAP_PARAM_ERROR. Mon Nov 28 14:32:51 2005: ERR: LDAP_PARAM_ERROR Mon Nov 28 14:32:51 2005: DEBUG: AuthBy LDAP2 result: REJECT, No such user Mon Nov 28 14:32:51 2005: INFO: Access rejected for alan(a)sip.domain.com: No such user Mon Nov 28 14:32:51 2005: DEBUG: Packet dump: *** Sending to 127.0.0.1 port 33032 .... Code: Access-Reject Identifier: 247 Authentic: <210>E<8>@m<211>h<133><34><23><16><21><28><242><187><154> Attributes: Reply-Message = "Request Denied"

1 0

[Serusers] Problem with Mysql module
by Marcel van Kaam, Fonetica Belgie 28 Nov '05

28 Nov '05

Hi All, Just checked on the list, found already some answers but still no luck. I use the getting started files from Onsip.org. Running Debian 2.6.12 ERROR: module modules/cpl/cpl.so not compiled ERROR: module modules/cpl-c/cpl-c.so not compiled ERROR: module modules/extcmd/extcmd.so not compiled ERROR: module modules/jabber/jabber.so not compiled ERROR: module modules/mysql/mysql.so not compiled When I use make all; make install, this is the outcome of my SER installation. I also tried make module etc. Is Debian the problem or do I need to make some changes to the Makefiles? Any help is welcome. Marcel

1 0

[Serusers] password encryption
by George Lambson 28 Nov '05

28 Nov '05

Hi all. I am having troubles getting my database to encrypt passwords. I set the option in the config file and even recreated the ser db using the included script, but no matter what when I add a user it shows the password in the table. Any ideas? Thanks, George

1 0

[Serusers] Re: [Users] rfc3264 and nathelper module
by Klaus Darilion 28 Nov '05

28 Nov '05

harry gaillac wrote: > Are you sure ? I am sure nathelper is independent from MOH > does nathelper is not able to handle sdp ? nathelper will parse SDP and may replace IP addresses and ports. This feature is independent from the service behind the SIP message flow (e.g. if it is a basic call, a transfered call, MOH, ...) klaus > > Harry > --- Klaus Darilion <klaus.mailinglists(a)pernau.at> a > écrit : > > >>Why do you think that nathelper is somehow related >>to MOH? This are >>total different things - they have nothing in >>common. >> >>klaus >> >>harry gaillac wrote: >> >>>I don't understand what you mean >>> >>> >>> >>>>harry gaillac wrote: >>>> >>>> >>>>>Hello, >>>>> >>>>>Maxim Sobolev has contributed to the >> >>developpement >> >>>>of >>>> >>>> >>>>>the nathelper module >>>>>He should provide this feature I can't contact >> >>him >> >>>>>though i sent a lot of mail . >>>> >>>>Hi Harry! Your really have absolutly no glue what >>>>you are talking about. >>>> >>>>klaus >>>> >>>> >>>> >>>>>Should I waste time my week to send and resend >>>> >>>>some >>>> >>>> >>>>>questions ? >>>>> >>>>>How can we build a real sip IPBX ? >>>>> >>>>>Regards >>>>>Harry >>>>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >> > ___________________________________________________________________________ > >>>Appel audio GRATUIT partout dans le monde avec le >> >>nouveau Yahoo! Messenger >> >>>Téléchargez cette version sur >> >>http://fr.messenger.yahoo.com >> >>> >> > > > > > > > > ___________________________________________________________________________ > Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger > Téléchargez cette version sur http://fr.messenger.yahoo.com > >

2 1

[Users] voice quality!
by messa＠innsof.com 28 Nov '05

28 Nov '05

hi all When i making a call, the voice quality is not well.it's entrench. What can i do to have a better voice quality? Best regards Serge!

2 1

Re: [Serusers] Re: [Users] rfc3264 and nathelper module
by harry gaillac 28 Nov '05

28 Nov '05

Here is an ethereal file. You can see there is a problem with contact field in sip HF Regards Harry --- Iqbal <iqbal(a)gigo.co.uk> a écrit : > send the sip trace, it will help my understanding of > it > > harry gaillac wrote: > > >I've ever said I can't configure ser as outbound > proxy > >so asterisk record > >sip agents behind nat with a private address in > AOR. > > > >I need to be adviced how can I send to asterisk an > >AOR with a public ip . > > > >Harry > > > >--- Iqbal <iqbal(a)gigo.co.uk> a écrit : > > > > > > > >>you have ser working, and you have ser and > asterisk > >>working together, if > >>you show the ngrep traces of all elements in the > >>call path, then we can > >>see where the problem is > >> > >>Iqbal > >> > >>harry gaillac wrote: > >> > >> > >> > >>>Give me you simple way please ! > >>> > >>>Harry > >>>--- Iqbal <iqbal(a)gigo.co.uk> a écrit : > >>> > >>> > >>> > >>> > >>> > >>>>also sip ipbx, simple add ser to asterisk, thats > >>>>all, > >>>> > >>>>Atle Samuelsen wrote: > >>>> > >>>> > >>>> > >>>> > >>>> > >>>>>Hi Harry, > >>>>> > >>>>>Does Maxim get paied to do Nathelper module for > >>>>> > >>>>> > >>>>> > >>>>> > >>>>you? if not, then he > >>>> > >>>> > >>>> > >>>> > >>>>>should'nt even need to think of it. > >>>>> > >>>>>if I dont remember wrong, Maxim said some time > >>>>> > >>>>> > >>ago > >> > >> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>that he had limited > >>>> > >>>> > >>>> > >>>> > >>>>>with time now, and that if you would give him a > >>>>> > >>>>> > >>>>> > >>>>> > >>>>small amount of money, > >>>> > >>>> > >>>> > >>>> > >>>>>he could get a friend to do some porting of > some > >>>>> > >>>>> > >>>>> > >>>>> > >>>>other features into the > >>>> > >>>> > >>>> > >>>> > >>>>>public version. > >>>>> > >>>>> > >>>>>Anyways, I think you should calm down, and > second > >>>>> > >>>>> > >>>>> > >>>>> > >>>>of all, start > >>>> > >>>> > >>>> > >>>> > >>>>>understanding what you talk about. Read the > F... > >>>>> > >>>>> > >>>>> > >>>>> > >>>>Rfc3261 a few more > >>>> > >>>> > >>>> > >>>> > >>>>>times. > >>>>> > >>>>>And, sorry for being an asswhole, It's just the > >>>>> > >>>>> > >>way > >> > >> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>I am. > >>>> > >>>> > >>>> > >>>> > >>>>>- Atle > >>>>> > >>>>> > >>>>> > >>>>>* harry gaillac <gaillacharry(a)yahoo.fr> [051128 > >>>>> > >>>>> > >>>>> > >>>>> > >>>>14:16]: > >>>> > >>>> > >>>> > >>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>>>Hello, > >>>>>> > >>>>>>Maxim Sobolev has contributed to the > >>>>>> > >>>>>> > >>developpement > >> > >> > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>of > >>>> > >>>> > >>>> > >>>> > >>>>>>the nathelper module > >>>>>>He should provide this feature I can't contact > >>>>>> > >>>>>> > >>him > >> > >> > >>>>>>though i sent a lot of mail . > >>>>>> > >>>>>>Should I waste time my week to send and resend > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>some > >>>> > >>>> > >>>> > >>>> > >>>>>>questions ? > >>>>>> > >>>>>>How can we build a real sip IPBX ? > >>>>>> > === message truncated === ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

Re: [Serusers] Re: [Users] rfc3264 and nathelper module
by harry gaillac 28 Nov '05

28 Nov '05

I've ever said I can't configure ser as outbound proxy so asterisk record sip agents behind nat with a private address in AOR. I need to be adviced how can I send to asterisk an AOR with a public ip . Harry --- Iqbal <iqbal(a)gigo.co.uk> a écrit : > you have ser working, and you have ser and asterisk > working together, if > you show the ngrep traces of all elements in the > call path, then we can > see where the problem is > > Iqbal > > harry gaillac wrote: > > >Give me you simple way please ! > > > >Harry > >--- Iqbal <iqbal(a)gigo.co.uk> a écrit : > > > > > > > >>also sip ipbx, simple add ser to asterisk, thats > >>all, > >> > >>Atle Samuelsen wrote: > >> > >> > >> > >>>Hi Harry, > >>> > >>>Does Maxim get paied to do Nathelper module for > >>> > >>> > >>you? if not, then he > >> > >> > >>>should'nt even need to think of it. > >>> > >>>if I dont remember wrong, Maxim said some time > ago > >>> > >>> > >>that he had limited > >> > >> > >>>with time now, and that if you would give him a > >>> > >>> > >>small amount of money, > >> > >> > >>>he could get a friend to do some porting of some > >>> > >>> > >>other features into the > >> > >> > >>>public version. > >>> > >>> > >>>Anyways, I think you should calm down, and second > >>> > >>> > >>of all, start > >> > >> > >>>understanding what you talk about. Read the F... > >>> > >>> > >>Rfc3261 a few more > >> > >> > >>>times. > >>> > >>>And, sorry for being an asswhole, It's just the > way > >>> > >>> > >>I am. > >> > >> > >>>- Atle > >>> > >>> > >>> > >>>* harry gaillac <gaillacharry(a)yahoo.fr> [051128 > >>> > >>> > >>14:16]: > >> > >> > >>> > >>> > >>> > >>> > >>>>Hello, > >>>> > >>>>Maxim Sobolev has contributed to the > developpement > >>>> > >>>> > >>of > >> > >> > >>>>the nathelper module > >>>>He should provide this feature I can't contact > him > >>>>though i sent a lot of mail . > >>>> > >>>>Should I waste time my week to send and resend > >>>> > >>>> > >>some > >> > >> > >>>>questions ? > >>>> > >>>>How can we build a real sip IPBX ? > >>>> > >>>>Regards > >>>>Harry > >>>> > >>>>--- Klaus Darilion > <klaus.mailinglists(a)pernau.at> > >>>> > >>>> > >>a > >> > >> > >>>>?crit : > >>>> > >>>> > >>>> > >>>> > >>>> > >>>>>no > >>>>> > >>>>>harry gaillac wrote: > >>>>> > >>>>> > >>>>> > >>>>> > >>>>>>Hello, > >>>>>> > >>>>>>Does nathelper module would support rfc3264 to > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>provide > >>>>> > >>>>> > >>>>> > >>>>> > >>>>>>MOH ? > >>>>>> > >>>>>>Harry > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>___________________________________________________________________________ > >>> > >>> > >>>> > >>>> > >>>> > >>>> > >>>>>>Appel audio GRATUIT partout dans le monde avec > >>>>>> > >>>>>> > >>le > >> > >> > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>nouveau Yahoo! Messenger > >>>>> > >>>>> > >>>>> > >>>>> > >>>>>>T?l?chargez cette version sur > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>http://fr.messenger.yahoo.com > >>>>> > >>>>> > >>>>> > >>>>> > >>>>>>_______________________________________________ > >>>>>>Users mailing list > === message truncated === ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Users] Broken Link
by Douglas Garstang 28 Nov '05

28 Nov '05

I don't know how many official people read the messages on this list, but there's a broken link on the openser.org xlog module page where it points to the pseudo variables at http://openser.org/docs/pseudo-variables.html. It used to work, and now it doesn't.... right when I need it! :P Doug.

2 2

[Serusers] SER and Asterisk authentication
by Voipers Portugal 28 Nov '05

28 Nov '05

Hello, I am trying to authenticate the call redirection between my SER and my Asterisk server. My idea is to create a User for each Server, but still don´t know how to do it. Any ideas? Regards, Jose Simoes

5 11

[Serusers] Adding Dial Plans without restarting SER.
by Leo Papadopoulos 28 Nov '05

28 Nov '05

Dear SerUsers, I have been able to do great things with dial plan inside of the ser.cfg file. HOWEVER, I would like to be able to do these same things without having to restart SER in order for the changes to be applied. Has anyone ever done anything like this? I don't want to re-invent this if it has already been done. If I do have to do my own.... I think I will design it such that certain SIP prefixes will force a lookup in MYSQL. Of course you will not want to slow down SER with MYSQL lookups for every SIP URI, so there will be an "IF" statement in the ser.cfg that tells whether or not to do a MYSQL lookup or route according to the SER.CFG statements. Leo P.

1 0

[Serusers] Re: [Users] rfc3264 and nathelper module
by Klaus Darilion 28 Nov '05

28 Nov '05

Why do you think that nathelper is somehow related to MOH? This are total different things - they have nothing in common. klaus harry gaillac wrote: > I don't understand what you mean > > >>harry gaillac wrote: >> >>>Hello, >>> >>>Maxim Sobolev has contributed to the developpement >> >>of >> >>>the nathelper module >>>He should provide this feature I can't contact him >>>though i sent a lot of mail . >> >>Hi Harry! Your really have absolutly no glue what >>you are talking about. >> >>klaus >> >> >>>Should I waste time my week to send and resend >> >>some >> >>>questions ? >>> >>>How can we build a real sip IPBX ? >>> >>>Regards >>>Harry >>> > > > > > > > > > ___________________________________________________________________________ > Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger > Téléchargez cette version sur http://fr.messenger.yahoo.com > >

1 0

[Users] Problem with pa module
by Jon Hing Yip 28 Nov '05

28 Nov '05

Hello, I've installed the presence server module (pa) and get following messages on the console after I've sent a SUBSCRIBE request to the server. The server crashs after these message occured. If you need more informations to this, just let me know. Regards, Jon H. Yip 6(2727) handle_subscription() entered 6(2727) parsing accept header 6(2727) check_message -0- _m=0x812e0e0 6(2727) check_message -1- 6(2727) pa check_message: accept=application/pidf+xml parsed=:0x30005 6(2727) check_message -2- accepts_mimes=0x812eb38 6(2727) check_message -3- 6(2727) check_message -4- parsed_event=0x812eaf8 6(2727) check_message -4- eventtype=0x0001 6(2727) check_message -4a- eventtype=0x1 epm[i].event_type=0x1 6(2727) check_message -4c- eventtype=0x1 mimetype=0x30005 accepts_mimes[k]=0x30005 6(2727) check_message -4b- eventtype=0x1 accepts_mime=0x30005 6(2727) get_pres_uri: _puri=sip:wsl.informatik.uni-ulm.de 6(2727) handle_subscription(): -1- 0(2708) child process 2727 exited by a signal 11 0(2708) core was not generated 0(2708) INFO: terminating due to SIGCHLD 10(2739) INFO: signal 15 received 11(2742) INFO: signal 15 received 12(2745) INFO: signal 15 received 13(2748) INFO: signal 15 received 14(2751) INFO: signal 15 received 1(2712) INFO: signal 15 received 2(2715) INFO: signal 15 received 3(2718) INFO: signal 15 received 4(2721) INFO: signal 15 received 5(2724) INFO: signal 15 received 7(2730) INFO: signal 15 received 8(2733) INFO: signal 15 received 9(2736) INFO: signal 15 received 15(2754) INFO: signal 15 received 0(2708) BUG: shutdown timeout triggered, dying...

2 2

Re: [Users] Remote PSTN gateway
by Klaus Darilion 28 Nov '05

28 Nov '05

hi Christoffer! Please always cc to the list. Christoffer Soop wrote: > Thank you very much! This is more than enough to get me started. > > One small remaining question though: > > Klaus Darilion wrote: > >>>- after the session with the remote proxy has been set up OpenSER either >>>works as proxy between the client and the remote proxy, or better >>>transfers the session to the client >> >>If you use record_route your proxy stays inbetween the client and the >>PSTN GW. If not, your proxy stays out of the folowing conversations. But >>you will loose the ability to account BYE messages, and will have >>problem when doing NAT traversal. > > In my scenario our proxy is the only one which should be able to > authenticate to the PSTN gateway. If the client talks directly to the > PSTN gateway will it have to re-supply the credentials? There are different credentials: 1. The credentials between your SIP proxy and your clients. 2. The credentials needed to authenticate to the remote proxy/GW. Digest authentication is normally used between clients and proxies. Thus, if you want to be standard conform, the clients would need additional credentials (a second set of realm/username/password) to authenticate also against the GW. But this is not supported by most clients, and you also do not want to give the GW credentials to you clients. The problem arises when using digest authentication between proxies. This is usually not possible. You would need a B2BUA, which receives the calls from your clients, and forward them to the GW with proper credentials. openser includes the uac module which can be used for this, but you might experience problems using this module as it is not standard conform. Another way to authentication against the GW would be with TLS or based on IP addresses (do not use UDP here as it can be spoofed easily). regards klaus

1 0

[Serusers] Error:mysql command not found
by subbanarasa 28 Nov '05

28 Nov '05

hi , Iam new to SER Iam tryied to srart SER but i got errors like this if anybody knows solution please help me. 1.[root@localhost sbin]# serctl start Starting SER : PID file /var/run/ser.pid does not exist -- SER start failed 2. [root@localhost sbin]# serctl domain add homwlabs.mirl.net MySql password: /usr/local/sbin/serctl: line 407: mysql: command not found SQL Error 3.[root@localhost sbin]# ser_mysql.sh create MySql password for root: creating database ser ... /usr/local/sbin/ser_mysql.sh: line 89: mysql: command not found Thanks and Regards Subbanarasa Reddy.

2 1

Re: [Users] fix_nat_contact() function
by Bogdan-Andrei Iancu 28 Nov '05

28 Nov '05

so, the problem is for INVITEs? bogdan harry gaillac wrote: >Hello, > >In fact I have this configuration > > Asterisk > || >------ppp0[ SER+nathelper+rtpproxy ] eth0---------sip >agents > >I told to the lists the last week I wish to use SER as >outbound proxy . >I can't rewrite contact with ppp0's address ! > >Regards >Harry > >--- Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> a >écrit : > > > >>Hi Harry, >> >>maybe you are mixing fix_nated_contact() with >>fix_nated_sdp() ?? >> >>regards, >>bogdan >> >>harry gaillac wrote: >> >> >> >>>Hello, >>> >>>I read nathelper documentation for >>> >>> >>fix_nated_contact() >> >> >>>function. >>>We can read:" Rewrites Contact HF to contain >>> >>> >>request's >> >> >>>source address:port." >>> >>>However when I use this function for nated agents >>> >>> >>it's >> >> >>>not the contact field in sip hf but the connection >>>information in sdp which is rewritten. >>> >>>Could you check it ? >>> >>>Harry >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>___________________________________________________________________________ >> >> >> >>>Appel audio GRATUIT partout dans le monde avec le >>> >>> >>nouveau Yahoo! Messenger >> >> >>>Téléchargez cette version sur >>> >>> >>http://fr.messenger.yahoo.com >> >> >>>_______________________________________________ >>>Users mailing list >>>Users(a)openser.org >>>http://openser.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >>> >>> >> >> > > > > > > >___________________________________________________________________________ >Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger >Téléchargez cette version sur http://fr.messenger.yahoo.com >

1 0

Re: [Serusers] outbound proxy tables
by Greger V. Teigre 28 Nov '05

28 Nov '05

Please copy the list. if(!lookup("location_internal")) { if(!lookup("location_external")) { # Not found } } g-) ----- Original Message ----- From: "Joao Pereira" <joao.pereira(a)fccn.pt> To: "Greger V. Teigre" <greger(a)teigre.com> Sent: Saturday, November 26, 2005 3:36 PM Subject: Re: [Serusers] outbound proxy tables > But if they are two instances of the location table... when SER is > checking if a user is online and searches the location table, it will fail > to find the users in the location_internal and location_external tables. > Maybe SER needs a special tunning to search the 3 tables... > > Joao > > > Greger V. Teigre wrote: >> I have never tried that example, but I assume they are just two other >> instances of the location table. >> g-) >> ----- Original Message ----- From: "Joao Pereira" <joao.pereira(a)fccn.pt> >> To: <serusers(a)lists.iptel.org> >> Sent: Friday, November 25, 2005 7:40 PM >> Subject: [Serusers] outbound proxy tables >> >> >>> Hello to all >>> I have SER configured as a multihomed outbound proxy, like in this page: >>> >>> http://www.voip-info.org/wiki/index.php?page=SER+example+outboundproxy >>> >>> In this configuration, SER puts users in diferent tables: >>> >>> if (dst_ip == 192.168.0.1) { if (!save("location_internal")) { >>> (...) >>> >>> Does someone have these tables? >>> The page says >>> "For this example to work, you must also follow the nathelper module >>> instructions for creating new location_internal and location_external >>> tables. Be careful to use underscores ("_") and not hyphens ("-") when >>> creating the new tables." >>> >>> But I cant find these tables .... Does someone have them??? >>> >>> Thanks >>> Joao >>> >>> _______________________________________________ >>> Serusers mailing list >>> serusers(a)lists.iptel.org >>> http://lists.iptel.org/mailman/listinfo/serusers >>> >> >

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[Users] fix_nat_contact() function
by harry gaillac 28 Nov '05

28 Nov '05

Hello, I read nathelper documentation for fix_nated_contact() function. We can read:" Rewrites Contact HF to contain request's source address:port." However when I use this function for nated agents it's not the contact field in sip hf but the connection information in sdp which is rewritten. Could you check it ? Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

2 1

[Users] How to Configure a Redirect Server Using OpenSer
by Ajay Srivastava 28 Nov '05

28 Nov '05

Hi We are trying to configure OpenSer as Redirect Server. Can anyone provide me with a configuration file that explains what options we have in OpenSer to configure it as redirect server. With Regards Ajay

3 2

[Serusers] Remote Access for SIP trace
by Rodrigo P. Telles 28 Nov '05

28 Nov '05

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Folks, I'm using SER in a carrier grade mode and I need to create an interface (GUI) to our support team run SIP traces in our SER box. I think I have an idea to solve that problem but I don't know if it's the best one, follow the idea: SERVER (SER) 1 - Run an application in daemon mode using libpcap to capture traffic on port 5060 - listening on a TCP port - capture traffic all the time - push all captured traffic to that TCP port (any one who connect/telnet on that port can see the traffic - without authentication by now) CLIENT (GUI) 2 - Developed using JAVA || PHP-GTK || C++ || .... - Connect to remote port to listen the traffic - Can filter what do you want to see (show only filtered traffic or all) - Colorized matches - Can save the result of your dump/filter to a file - etc So I did a concept proof... 1 - Wrote a simple server program using Perl who run ngrep in SER box and push the captured traffic through it's listening TCP port; 2 - Wrote a simple client program using Perl who connect to a remote port and filter what you want to see or all the traffic; ..and works like a charm :-) I'd like to hear opnions from SER members about the idea. Best regards, - -- ============================================ Rodrigo P. Telles <telles(a)devel.it> IT Manager Devel-IT - http://www.devel.it IVOZ # 1029 +55 14 3324-1200 Bestcom Group ============================================ -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.4 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFDg3BWiLK8unYgEMQRAiqlAJ97fGI6OMAJvXzki77J9a5WS+KXpACeMX98 TpmB5w1kvF7xkTc1XC3o+7Y= =fkKs -----END PGP SIGNATURE-----

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RE: [Users] Avpops Module
by Douglas Garstang 28 Nov '05

28 Nov '05

Elena, My parameters to avp_db_load, as requested, are: modparam("avpops","avp_url","mysql://openser:openserrw@localhost/openser") ... modparam("avpops","db_scheme", "scheme1:uuid_col=uuid;value_col=phone;value_type=string;table=subscriber") The columns uuid, and phone are valid columns from the subscriber table. This causes the following to be logged to syslog on openser startup: Nov 27 23:53:08 ser1 openser[2085]: Maxfwd module- initializing Nov 27 23:53:08 ser1 openser[2085]: AVPops - initializing Nov 27 23:53:08 ser1 openser[2085]: ERROR:avpops_init: "AVP_DB" present but "AVP_TABLE" found empty Nov 27 23:53:08 ser1 openser[2085]: init_mod(): Error while initializing module avpops I did base it on the example from the avpops tutorial. I don't understand why this error is generated. I thought the whole point of using db_scheme was that you didn't have to call modparam on the table name (as well as the uuid, attribute columns etc). I could really do with a little help! Thanks. Douglas. -----Original Message----- From: Elena Ramona Modroiu [mailto:ramona@voice-system.ro] Sent: Sun 11/27/2005 4:26 PM To: Douglas Garstang Cc: users(a)openser.org Subject: Re: [Users] Avpops Module Hi, Douglas Garstang wrote: >I swear, getting OpenSER to do anything is worse than pulling teeth. > > :) > >The avpops module documentation at: >http://www.openser.org/docs/modules/1.0.x/avpops.html#AEN165 > >has the following example under the db_scheme command: >... >modparam("avpops","db_scheme", >"scheme1:table=subscriber;uuid_column=uuid;value_column=first_name") >... > >which causes the following error to be reported in syslog.conf: >Nov 27 14:24:52 ser1 openser: ERROR:avpops:parse_avp_sb_scheme: unknown attribute <uuid_column> >Nov 27 14:24:52 ser1 openser: ERROR:avpops:avp_add_db_scheme: falied to parse scheme >Nov 27 14:24:52 ser1 openser: parse error (102,3-4): Can't set module parameter > >Why? > > Thanks for reporting this, docs will be updated. > >The example at the 'alleged' tutorial at: >http://www.voice-system.ro/docs/avpops/ > >which is just the docs at openser.org cut and pasted mostly has this example: >... >modparam("avpops","db_scheme", >"scheme1:uuid_col=uid;value_col=job;value_type=string;table=emp") >... > >which does not generate any errors in syslog. >HOWEVER, when avp_db_load() is called, the following is logged to syslog.conf: > >Nov 27 14:21:42 ser1 openser[2424]: ERROR:avpops_init: "AVP_DB" present but "AVP_TABLE" found empty >Nov 27 14:21:42 ser1 openser[2424]: init_mod(): Error while initializing module avpops > >Why??? The docs say I can use dbscheme to use a non standard AVP table, which is what I am trying to do, namely the subscriber table. If I can use dbscheme to define the columns from another table, why is OpenSER complaining that I haven't called modparam("avpops","avp_table","avptable")????? > > What are the parameters of avp_db_load() in your case? Maybe this example (also from avpops tutorial) helps you: modparam("avpops","db_scheme", "scheme1:uuid_col=uid;value_col=job;value_type=string;table=emp") avp_db_load("$uuid","s:jobs/*$scheme1*"); - load from table 'emp', using 'uid' column as uuid and 'job' column as string value into AVP name 'jobs' Bye, Ramona >Doug. > > > > > > > > > >------------------------------------------------------------------------ > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > >

3 2

[Serusers] Problem to install SER with support for presence
by victoria beltran 28 Nov '05

28 Nov '05

Hi, I am trying to install the last unstable SER version with support for SIP-based presence. When I compile and install SER with presence modules (make install include_modules="pa rls mysql), it does several errors: receive.o: In the function 'receive_msg': /root/sip_router/receive.c:231:undefined reference to 'reset_user_avps' receive.o:/root/sip_router/receive.c:232:undefined reference to 'reset_user_avps' receive.o:/root/sip_router/receive.c:251:undefined reference to 'reset_user_avps' receive.o:/root/sip_router/receive.c:252:undefined reference to 'reset_domain_avps' I have downloaded current ser sources by "cvs -d :pserver:anonymous@cvs.berlios.de:/cvsroot/ser checkout sip_router" and I have seen the usr_avp.h and user_avp.c files. In this files, the reset_user_avps and reset_domain_avps functions hasnt been defined. Could anybody help me? Thanks!! ______________________________________________ Renovamos el Correo Yahoo! Nuevos servicios, más seguridad http://correo.yahoo.es

1 0

[Users] Questions
by Douglas Garstang 28 Nov '05

28 Nov '05

I would REALLY appreciate if someone could answer the following questions for me. 1. What is the definition of a 'negative' result in relation to t_reply()? I am seeing situations where t_relay() seems to think that 407 Proxy Auth is a negative result, because a previously set t_on_failure is executing when it connects to the gateway sucessfully. 2. Under what circumstances does t_reply() return 0, and under what circumstaces does it return -1? Seems like it always returns 0, no matter what happens. 3. When t_relay() fails to connect to a gateway, if t_on_failure(N) had previously been set, where does execution jump to t_on_failure(N)? It isn't right after t_relay() returns, and it isn't _within_ t_relay, so where is it? Is it at some arbitrary point? Is it after the current route() block ends? Thanks, Doug.

2 1

[Users] Remote PSTN gateway
by Christoffer Soop 28 Nov '05

28 Nov '05

Hi there! I would like to implement the following scenario using OpenSER: - when a SIP client sends a request consisting of only a simple phone number (on the form +CC-AAA-NNNNNN, with or without plus sign, hyphens, possible spaces etc where C, A and N are simple numbers) OpenSER should route the request to another, remote SIP-proxy acting as a PSTN gateway. - either the same credentials should be used or preferebly credentials dependent on the domain name of the callee for all requests to the SIP-proxy - after the session with the remote proxy has been set up OpenSER either works as proxy between the client and the remote proxy, or better transfers the session to the client - all requests for a client not on the form of a phone number (i.e. with a fully qualified domain name should be routed to the correct location as decided by SRV-records etc. Is this possible? Any hints on how to accomplish this? Sincerely, Kris PS/Please note that I am new to OpenSER, SIP and SIP terminology...

2 1

[Users] OT Question
by Tavis P 28 Nov '05

28 Nov '05

Does anyone know if it is possible (and then how) to specify the RTP port range on cisco TDM gateways (3640, 5300, 5400, 5800, etc) It seems inconceivable that it can't be done on cisco hardware, but i havn't been able to find any information on how to do it, i've talked to several cisco engineers that don't know how to do it either. Again, sorry for the OT question tavis

2 1

[Users] Prevent bandwidth lost
by Christophe Irles 28 Nov '05

28 Nov '05

Hi, I'm wondering how to prevent bandwitdh lost with (open)SER+MediaProxy in this type of network configuration: UA0 --- [(open)SER+MediaProxy] --- [ Internet ] --- [ FW/NAT ] --- [UA1, UA2] UA1, UA2 are one the same network and they are registered in the (open)SER server. If UA1 call UA2 with default media proxy configuration, the RTP flow will not be direct between this two UA. So I lost twice bandwith (from SER to UA1 and SER to UA2). How can i prevent this ? On the REGISTRAR process, I think I could store the public IP used by UA1 and UA2 and the two private IP of this UA. Since they have the same public IP but a different private IP, on the INVITE procces I suppose it's could be possible to indicate that RTP flow can be direct between this two devices, no ? Now which tools (modules) could be used to resolve this ? Thanks, Christophe

3 2

[Serusers] Is there a way to get ACC to NOT log ACK's?
by Leo Papadopoulos 28 Nov '05

28 Nov '05

Dear serusers, I am running 0.9.4 and am successfully logging calls to MySQL using the ACC module. I have added a statement to my ser.cfg to get the ACC module to NOT log OPTIONS as follows: if (method!="OPTIONS") { setflags.. } The above works and I am not logging OPTIONS. I tried the modification below, but it didn't work to get rid of ACK's. They are still being logged. if ((method!="OPTIONS") & (method!="ACK")) { setflags.. } How can I get ACC NOT to log ACK's records into the database? Leo P.

4 3

[Serusers] Problem about SEMS implement voicemail...
by 昂達 28 Nov '05

28 Nov '05

Hello All : I have installed SER + SEMS to implement voicemail, but only the first call can connect to voicemail, and can mail message.wav to callee's e-mail address . but after the first call , i can not connect to voicemail until i restart SEMS then i execute "sems -E " it shows that: (29104) ERROR: run (AmServer.cpp:161): odd trailer (29104) ERROR: run (AmServer.cpp:161): odd trailer (29104) ERROR: run (AmServer.cpp:161): odd trailer (29104) ERROR: run (AmServer.cpp:161): odd trailer (29104) ERROR: run (AmServer.cpp:161): odd trailer (29104) ERROR: run (AmServer.cpp:161): odd trailer (29104) ERROR: run (AmServer.cpp:161): odd trailer (29104) ERROR: run (AmServer.cpp:161): odd trailer (29104) ERROR: run (AmServer.cpp:161): odd trailer (29104) ERROR: run (AmServer.cpp:161): odd trailer (29104) ERROR: run (AmServer.cpp:161): odd trailer (29104) ERROR: run (AmServer.cpp:161): odd trailer (29104) WARNING: sig_usr_fifo (sems.cpp:84): signal 2 received (29103) WARNING: sig_usr_fifo (sems.cpp:84): signal 2 received (29103) WARNING: sig_usr_fifo (sems.cpp:84): signal 17 received How to solve this problem ?? *ser.cfg and sems.conf was attached. 本郵件附件清單如下： (1). sems.conf.txt (1.4 K) (2). ser.cfg.txt (7.3 K)

1 0

[Serusers] SEMS---how to configure MYSQL DB use unix socket ??
by 昂達 28 Nov '05

28 Nov '05

Hello All In "ser.cfg" file how to configure MYSQL DB to use unix socket?? In my "sems.conf" have configure as fllow : ...... socket_name=/tmp/am_sock ser_socket_name=/tmp/ser_socket send_method=unix_socket ......

1 0

[Serusers] Problem regarding init_hfname_parser & init_digest_parser specified in serdev.txt
by nitin ganjsinghani 28 Nov '05

28 Nov '05

hi friends there are no functions with name *init_hfname_parser* & *init_digest_parser *in any file of the TAR package. can anybody plz tell me the function name corresponding to the above functions in SER-0.9.4 Thanks Nitin

1 0

[Users] Avpops Module
by Douglas Garstang 27 Nov '05

27 Nov '05

I swear, getting OpenSER to do anything is worse than pulling teeth. The avpops module documentation at: http://www.openser.org/docs/modules/1.0.x/avpops.html#AEN165 has the following example under the db_scheme command: ... modparam("avpops","db_scheme", "scheme1:table=subscriber;uuid_column=uuid;value_column=first_name") ... which causes the following error to be reported in syslog.conf: Nov 27 14:24:52 ser1 openser: ERROR:avpops:parse_avp_sb_scheme: unknown attribute <uuid_column> Nov 27 14:24:52 ser1 openser: ERROR:avpops:avp_add_db_scheme: falied to parse scheme Nov 27 14:24:52 ser1 openser: parse error (102,3-4): Can't set module parameter Why? The example at the 'alleged' tutorial at: http://www.voice-system.ro/docs/avpops/ which is just the docs at openser.org cut and pasted mostly has this example: ... modparam("avpops","db_scheme", "scheme1:uuid_col=uid;value_col=job;value_type=string;table=emp") ... which does not generate any errors in syslog. HOWEVER, when avp_db_load() is called, the following is logged to syslog.conf: Nov 27 14:21:42 ser1 openser[2424]: ERROR:avpops_init: "AVP_DB" present but "AVP_TABLE" found empty Nov 27 14:21:42 ser1 openser[2424]: init_mod(): Error while initializing module avpops Why??? The docs say I can use dbscheme to use a non standard AVP table, which is what I am trying to do, namely the subscriber table. If I can use dbscheme to define the columns from another table, why is OpenSER complaining that I haven't called modparam("avpops","avp_table","avptable")????? Doug.

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RE: [Users] t_check_status
by Douglas Garstang 27 Nov '05

27 Nov '05

That doesn't look like an RFC to me. It's not cataloged at rfc.net, and doesn't have a number. I didn't even know it existed. Thanks though. Like most things related to Openser, the knowledge is just assumed. -----Original Message----- From: Norman Brandinger [mailto:norm@goes.com] Sent: Sun 11/27/2005 9:02 AM To: Douglas Garstang Cc: users(a)openser.org Subject: Re: [Users] t_check_status Before you can take advantage of what OpenSER has to offer, you should become familiar with the the various RFC's associated with SIP communication. AVP stands for Attribute - Value Pair. Basically, it maps an attribute for example, "NPA" to a value such as "212". The AVP's do have the ability to use databases to store and retrieve values, however, this is not a requirement. The configuration file can set, test and/or access an AVP without any database access. In real life situations, you will most probably want to use a database to store and retrieve AVP's. RPID is defined here: http://www.iptel.org/ietf/security/identity/draft-ietf-sip-privacy-04.txt Regards, Norm Douglas Garstang wrote: > I thought AVP stuff was database related.... this confuses me even more. I am reading the documentation you know.... constantly. Considering it looks like it was written by the people who developed OpenSER, there's a lot of implied/assumed knowledge there, and the scripting language is very high level, it's bloody difficult to work out what the documentation means. > > For example... where's the definitiaion of AVP? I still can't find what an 'rpid' is. > > -----Original Message----- > From: Juha Heinanen [mailto:jh@tutpro.com] > Sent: Sat 11/26/2005 9:29 PM > To: Douglas Garstang > Cc: users(a)openser.org > Subject: RE: [Users] t_check_status > > > > Douglas Garstang said: > > > I really don't even get the point of what an AVP is. Something to do with > > databases. I don't see why database interaction is needed to set a timer. > > That's crazy. > > > > i don't know what is cracy, since no database interaction is needed to set > a timer. timers can be set in many different ways. read usrav > documentation. > > --juha > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users >

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[Serusers] Zyxel 2602HW bug?
by Federico Giannici 27 Nov '05

27 Nov '05

I have found a strange bug in the ZyXEL 2602HW (firmware "V3.40(MV.3)") that make it impossible to use it to make calls: the Request URI of the ACK and BYE messages is addressed to the SIP server ("sip:195.120.250.50") instead of the other UA (i.e. something like "sip:123@1.2.3.4"). I cannot understand how such an huge bug is not experienced by somebody else. Anybody had the same problem with the 2602HW? Thanks. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

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RE: [Users] t_check_status
by Douglas Garstang 27 Nov '05

27 Nov '05

I thought AVP stuff was database related.... this confuses me even more. I am reading the documentation you know.... constantly. Considering it looks like it was written by the people who developed OpenSER, there's a lot of implied/assumed knowledge there, and the scripting language is very high level, it's bloody difficult to work out what the documentation means. For example... where's the definitiaion of AVP? I still can't find what an 'rpid' is. -----Original Message----- From: Juha Heinanen [mailto:jh@tutpro.com] Sent: Sat 11/26/2005 9:29 PM To: Douglas Garstang Cc: users(a)openser.org Subject: RE: [Users] t_check_status Douglas Garstang said: > I really don't even get the point of what an AVP is. Something to do with > databases. I don't see why database interaction is needed to set a timer. > That's crazy. > i don't know what is cracy, since no database interaction is needed to set a timer. timers can be set in many different ways. read usrav documentation. --juha

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[Serusers] Gateway registration and call forwarding
by Voipers Portugal 27 Nov '05

27 Nov '05

Hello again Imagine the following scenario: I have a SER server working with my SIP users. Imagine that i want to offer them the possibility of making PSTN calls. I have just made an agreement with a termination provider and they gave me a user id, a password and the gateway address. They told me i needed to register before making calls.. How can i make SER register in that gateway? And then to allow the PSTN calls, what should I do? Jose simoes

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RE: [Users] t_check_status
by Douglas Garstang 27 Nov '05

27 Nov '05

I have read the tm documentation, many many times. I guess I don't follow, probably because I didn't write the code. The docs for tm say that setting fr_timer_avp overrides fr_timer. So? Why does it need to be overridden? Why not just use fr_timer? If the lcr module doesn't have any timers of it's own, then what these for...? modparam("lcr","fr_inv_timer_avp","fr_inv_timer_avp") modparam("lcr","fr_inv_timer",90) modparam("lcr","fr_inv_timer_next",30) I really don't even get the point of what an AVP is. Something to do with databases. I don't see why database interaction is needed to set a timer. That's crazy. Doug. -----Original Message----- From: Juha Heinanen [mailto:jh@tutpro.com] Sent: Sat 11/26/2005 5:38 PM To: Douglas Garstang Cc: Subject: RE: [Users] t_check_status Douglas Garstang writes: > Why do I need to set this timer using an avp? Is this mandatory? Why? read tm documentation. > I find it hard to believe I need to integrate with a database to set > a timer. Why can't I just set the other timers instead? And, as > before, what's the relationship between them? Why don't the lcr > timers have an effect? In fact, why are they even there? They don't > seem to have any use. lcr module doesn't have any timers of its own. it just uses tm timers. > And you didn't answer my question about t_check_status. That has > nothing to do with timers. t_relay() times out, t_check_status is > used to check the code of the last SIP message, which was null > because there _was no_ reply, and somehow t_check_status magically > matches 408|500|503....? i only responded to your lcr related question. t_check_status has nothing to do with lcr. -- juha

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RE: [Users] openser.cfg Samples
by Douglas Garstang 26 Nov '05

26 Nov '05

Thanks, but unfortunately there's nothing there relating to lcr, dispatcher, redundancy or failover, which I've been torturing over for a week now. -----Original Message----- From: Elena Ramona Modroiu [mailto:ramona@voice-system.ro] Sent: Sat 11/26/2005 3:18 AM To: Douglas Garstang Cc: users(a)openser.org Subject: Re: [Users] openser.cfg Samples Hi, in sources there is a directory 'examples' under the main directory. http://cvs.sourceforge.net/viewcvs.py/openser/sip-server/examples/ Ramona Douglas Garstang wrote: >Does anyone know where I could find some openser.cfg sample files? Been looking and there's absolutely squat available. (hint hint). This would be a very very useful resource... > >Thanks, >Doug. > > > >------------------------------------------------------------------------ > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > >

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RE: [Users] openser.cfg Samples
by Douglas Garstang 26 Nov '05

26 Nov '05

Thanks for that. I'll remember for future use. Unfortunately nothing to do with the dispatcher or lcr modules or anything to do with failover or redundancy. That would have been extremely helpful. Doug. -----Original Message----- From: JF [mailto:jfkavaka@gmail.com] Sent: Sat 11/26/2005 3:03 AM To: Douglas Garstang Cc: users(a)openser.org Subject: Re: [Users] openser.cfg Samples Hi, Check http://www.onsip.org There's a resource called "All Config FIles for Getting Started" which I found useful. JF On 11/26/05, Douglas Garstang <dgarstang(a)oneeighty.com> wrote: > Does anyone know where I could find some openser.cfg sample files? Been looking and there's absolutely squat available. (hint hint). This would be a very very useful resource... > > Thanks, > Doug. > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > > >

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[Serusers] LCR question
by Stefan Prelle 26 Nov '05

26 Nov '05

Hi all, I'm testing the LCR module and I am having problems with its performance. In my setup I use a call generator (sipp) to place 50 calls, 10 calls each second - which works like a charm without any retransmissions when NOT using LCR functionality (no load_gws()). As soon as I add this code in my route_block if (!load_gws()) { sl_send_reply("500", "Server Internal Error - Cannot load gateways"); return; }; I am having huge performance problems. The OpenSER drops to effectivly 1 CPS. My lcr table contains only 20 entries and I have just 2 gateways configured, so this shouldn't be a problem. I guess I made a big mistake somewhere, since this can't be the intended behaviour of the LCR module. Can somebody help me? Regards, Stefan

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Re: [Serusers] Re: NAT again
by Atle Samuelsen 26 Nov '05

26 Nov '05

To be honist, I have'nt looked at your problem that much. but, if your setup is ser -> asterisk and you want to do all nat triversal at ser, it's possible. just make a simple ser.config with only something like : if(src_ip=="asterisk_ip"){ rewritehost("asterisk"); }else{ t_relay(); }; * harry gaillac <gaillacharry(a)yahoo.fr> [051126 13:47]: > However I need help to fix my problems ! > A few people have sent me some help. > > Do you understand the problems in my posts ? > > Harry > > --- Atle Samuelsen <clona(a)cyberhouse.no> a ?crit : > > > goood, > > > > Then you have learnt, never send more then one copy > > of a mail, and never > > send email "Place answer me for fuck's sake" or > > something simular. > > > > - Atle > > > > * harry gaillac <gaillacharry(a)yahoo.fr> [051126 > > 11:04]: > > > I agree you > > > excuse me for sent and resent the same question I > > just > > > expected more some help . > > > I say help not someone who would do what i wish. > > > > > > Harry > > > --- Atle Samuelsen <clona(a)cyberhouse.no> a ?crit : > > > > > > > Okey, the reason why I asked, is that I see you > > have > > > > started sending > > > > more then one copy of the same mail to the > > mailig > > > > list.. again.. > > > > > > > > i tought you had understood that it's not smart, > > and > > > > that people get > > > > pissed when you do that. > > > > > > > > - Atle > > > > > > > > * harry gaillac <gaillacharry(a)yahoo.fr> [051126 > > > > 00:33]: > > > > > Dear Atle, > > > > > I'm not too old I'm just so tired of being > > > > unemployed. > > > > > Harry > > > > > > > > > > --- Atle Samuelsen <clona(a)cyberhouse.no> a > > ?crit : > > > > > > > > > > > Hi Harry, > > > > > > > > > > > > Sorry to ask, How old are you ? > > > > > > > > > > > > -Atle > > > > > > > > > > > > * Doug Meredith <doug.meredith(a)skyridge.com> > > > > [051124 > > > > > > 19:25]: > > > > > > > bosinclar(a)laposte.net wrote: > > > > > > > > > > > > > > >I need to add some functions in my > > ser.cfg > > > > for > > > > > > nat > > > > > > > >support. > > > > > > > >I use my open(ser) as outbound proxy with > > two > > > > > > > >interfaces ppp0 and eth0. > > > > > > > >sip agents send requests to eth0. > > > > > > > >I want my packets passing through the > > > > register > > > > > > and > > > > > > > >invite blocks by fixing contact with a > > public > > > > ip. > > > > > > > > > > > > > > > >All packets from private network must be > > sent > > > > to > > > > > > eth0. > > > > > > > > > > > > > > > >Any suggestions > > > > > > > > > > > > > > > >Bob > > > > > > > > > > > > > > Looks like Harry got a new e-mail account > > to > > > > avoid > > > > > > the moderation. > > > > > > > > > > > > > > Doug > > > > > > > -- > > > > > > > Doug Meredith > > (doug.meredith(a)systemguard.com) > > > > > > > SystemGuard - Oracle remote support > > > > > > > 877-974-8273 (87-SYSGUARD) > > > > > > > 506-854-7997 > > > > > > > www.systemguard.com > > > > > > > > > > > > > > > > > > _______________________________________________ > > > > > > > Serusers mailing list > > > > > > > serusers(a)lists.iptel.org > > > > > > > > > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > > > > > > > > > > > > > > _______________________________________________ > > > > > > Serusers mailing list > > > > > > serusers(a)lists.iptel.org > > > > > > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > ___________________________________________________________________________ > > > > > > > > > Appel audio GRATUIT partout dans le monde avec > > le > > > > nouveau Yahoo! Messenger > > > > > T?l?chargez cette version sur > > > > http://fr.messenger.yahoo.com > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > ___________________________________________________________________________ > > > > > Appel audio GRATUIT partout dans le monde avec le > > nouveau Yahoo! Messenger > > > T?l?chargez cette version sur > > http://fr.messenger.yahoo.com > > > > > > > > > > > > > ___________________________________________________________________________ > Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger > T?l?chargez cette version sur http://fr.messenger.yahoo.com >

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[Users] error in nathelper module
by harry gaillac 26 Nov '05

26 Nov '05

Hello, I read nathelper documentation for fix_nated_contact() function. We can read:" Rewrites Contact HF to contain request's source address:port." However when I use this function for nated agents it's not the contact field in sip hf but the connection information in sdp which is rewritten. Could check what i say ? Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

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[Users] detecting re-INVITEs and corresponding replies
by JF 26 Nov '05

26 Nov '05

Hi all, For detection of a re-INVITE request from the script is it sufficient to detect the presence of a to tag using has_to_tag() function? Eg: if (method=="INVITE" && has_to_tag()) { ... }; What about detecting a reply to a re-INVITE in an onreply_route ? Is it doable from the script? How? Thanks in advance for your help. JF

2 1

[Users] Persistently storing headers from onreply_route()?
by João Plácido 26 Nov '05

26 Nov '05

Hi, I tried the patch for synchronization of on_reply routes with the 1.0.0 release, changed the avpops module interface adding the ONREPLY_ROUTE flag and it seems to work. (http://openser.org/pipermail/users/2005-October/001281.html) Any plans for global AVP support to avoid having to do this "trick"? Thank you very much, JF

3 2

Re: [Serusers] Re: NAT again
by Atle Samuelsen 26 Nov '05

26 Nov '05

Okey, the reason why I asked, is that I see you have started sending more then one copy of the same mail to the mailig list.. again.. i tought you had understood that it's not smart, and that people get pissed when you do that. - Atle * harry gaillac <gaillacharry(a)yahoo.fr> [051126 00:33]: > Dear Atle, > I'm not too old I'm just so tired of being unemployed. > Harry > > --- Atle Samuelsen <clona(a)cyberhouse.no> a ?crit : > > > Hi Harry, > > > > Sorry to ask, How old are you ? > > > > -Atle > > > > * Doug Meredith <doug.meredith(a)skyridge.com> [051124 > > 19:25]: > > > bosinclar(a)laposte.net wrote: > > > > > > >I need to add some functions in my ser.cfg for > > nat > > > >support. > > > >I use my open(ser) as outbound proxy with two > > > >interfaces ppp0 and eth0. > > > >sip agents send requests to eth0. > > > >I want my packets passing through the register > > and > > > >invite blocks by fixing contact with a public ip. > > > > > > > >All packets from private network must be sent to > > eth0. > > > > > > > >Any suggestions > > > > > > > >Bob > > > > > > Looks like Harry got a new e-mail account to avoid > > the moderation. > > > > > > Doug > > > -- > > > Doug Meredith (doug.meredith(a)systemguard.com) > > > SystemGuard - Oracle remote support > > > 877-974-8273 (87-SYSGUARD) > > > 506-854-7997 > > > www.systemguard.com > > > > > > _______________________________________________ > > > Serusers mailing list > > > serusers(a)lists.iptel.org > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > > > ___________________________________________________________________________ > Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger > T?l?chargez cette version sur http://fr.messenger.yahoo.com >

2 2

[Users] openser.cfg Samples
by Douglas Garstang 26 Nov '05

26 Nov '05

Does anyone know where I could find some openser.cfg sample files? Been looking and there's absolutely squat available. (hint hint). This would be a very very useful resource... Thanks, Doug.

3 2

[Users] t_check_status
by Douglas Garstang 26 Nov '05

26 Nov '05

This really has me confused. My openser.cfg has the following code: modparam("lcr","fr_inv_timer",2) modparam("lcr","fr_inv_timer_next",2) modparam("tm", "fr_timer", 1) # Heck, might as well set all three! I don't know the difference between them! # fr_timer seems to actually cause INVITE's to time out though, unlike fr_inv_timer and fr_inv_timer_next which do nothing. ... t_on_failure("2"); if (!next_gw()) { sl_send_reply("503", "Service not available, no gateways found"); return; } if ( !t_relay() ) { xlog ("L_INFO","Call to t_relay() returned -1"); # Never happens even if gateway connection times out } else { xlog ("L_INFO","Call to t_relay() returned 0"); # Always happens, even when gateway connection times out } failure_route[2] { if (method == "INVITE" ) && t_check_status("408|500|503")) { } } Failure route failure_route[2] is being called when t_relay fails to contact a gateway. I'm still trying to understand how that can actually happen because t_relay() always returns 0, but anyway.... it seems that calling t_on_failure(2) somehow magically tells failure_route[2] to execute eventhough the previous t_relay() returns 0. If t_relay() always returns 0, what triggers execution of failure_route[2]??? Now for the tricky part. Function t_relay() never gets a connection to a gateway, and no final response code is received. It just so happens I had the t_check_status in there because I copied and pasted it from someones example. If no final response code is received, why does this if statement return true? It is returning true eventhough no response code was received. How can that happen? Yes, I read the docs on using t_check_status in a failure route block, and it says "the code of the selected negative final reply". Sounds about what I am talking about here, although the difference between "the code of the selected negative final reply" and a reply blocks "the code of the current received reply." isn't really spelled out anywhere. And yes, I read the docs. The docs don't explain to me why this weird stuff is occurring. Doug.

2 1

FW: [Users] Setting t_relay() timeout in lcr
by Douglas Garstang 26 Nov '05

26 Nov '05

And just what is 'sequential forwarding' anyway? I just searched through every single module one-by-one at openser.org (which is painful when you don't know what module a specific piece of documentation is in) and could not find any reference to 'sequential forwarding' except when the lcr module says it uses it. So, what is it exactly? Where's the documentation that points to how it works? I really don't think my questions are unreasonable, and I'm not trying to be difficult. I wouldn't be asking these questions if I knew or could find the answers. As I said, I'm just trying to find a solution to a problem which is ultimately _SOME_ degree of round-robin and redundancy in what is otherwise a great product. At this point it looks like round-robin is just not feesible with OpenSER. I'll settle for redundancy... ie trying another server if one is down. Doug. -----Original Message----- From: Douglas Garstang Sent: Fri 11/25/2005 7:09 PM To: Cc: users(a)openser.org Subject: RE: [Users] Setting t_relay() timeout in lcr Yes. I have read sections 1.3.17 and 1.3.18 of the module documentation for lcr. That's why I made the post. Section 1.3.17 says: 'Sets the value of the fist INVITE's Final Response timeout to be used during sequential forwarding:" Section 1.3.18 says: "Sets the value of the next INVITE's Final Response timeouts to be used during sequential fowarding:" So, I went and set: modparam("lcr","fr_inv_timer",2) modparam("lcr","fr_inv_timer_next",2) and it had no effect on the timeout that t_relay() used. However, when I set: modparam("tm", "fr_timer", 1) it _DID_ have an effect in the timeout that t_relay() used. I _have_ experimed with the variables and functions WAY more than I would have if the documentation was complete. Experimenting with the functions showed me that t_relay() is always returning 0, even when there is a failure. My question therefore stands. Why does setting fr_inv_timer and fr_inv_timer_next have no effect? What is the relationship between the settings in the tm module and the lcr module? Why are there two settings? Which should I use? Why does t_relay() always return 0 even on failure? I could be doing things completely wrong, as there is a lack of good examples accompanying the documentation, and it's easy to see the documentation was written by the same people who wrote the code - there's an assumed level of knowledge when reading it that most people don't have. Douglas. -----Original Message----- From: Juha Heinanen [mailto:jh@tutpro.com] Sent: Fri 11/25/2005 6:32 PM To: Douglas Garstang Cc: users(a)openser.org Subject: RE: [Users] Setting t_relay() timeout in lcr Douglas Garstang writes: > I just managed to set the timeout with fr_inv_timer from the tm > module, and it seems to have the right effect. So, what are > fr_inv_timer and fr_inv_timer_next in lcr module for then??? before you sent this message, have you checked sections 1.3.17 and 1.3.18 of lcr README file? you should read all existing documentation, experiment with the variables and the functions and if you after that still have problems, you post your message to the list. -- juha

2 1

[Serusers] ser as outbound proxy only
by jeff kwong 26 Nov '05

26 Nov '05

Hi to everybody! Im currently trying to setup an outbound proxy using ser. Im still a newbie with this ser and im not yet that good with the scripting. Could anyone tell me how to configure ser for outbound proxy only using rtpproxy? There are samples files in the Getting started manual but im having troubles eliminating the authentication/mysql part in script since im not yet familiar which modules depends on which. A striped down version of the nat-rtpproxy.cfg file would be really appreciated. Thanks in advance. _jeff

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