- sr-users - kamailio.org

[Serusers] cancel message and to tag
by Hye-Jeong Mok 10 Oct '06

10 Oct '06

Hi, Have a question about the CANCEL message and the to tag. Let say I have a UAC sending INVITE to SER. SER responds with 183 message with a to tag = 111. Then the UAC sends CANCEL with no tag, and SER responds with 200 message with the new to tag = 222, followed by 487 message with to tag 222. SER 0.9.0 that I'm using behaves this way. I believe in 3261, the to tag of the 200 message in response to the CANCEL that has no tag, it has to be the same as the to tag that is in the previous 1xx message. So in this case, SER that I'm using is doing something wrong? Any idea how to fix this? Thanks, HJ _________________________________________________________________ FREE pop-up blocking with the new MSN Toolbar - get it now! http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/

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[Serusers] set some flag that isn't reset after some call leg has finished
by Ricardo Carvalho 10 Oct '06

10 Oct '06

How may I set some flag that isn't reset after some call leg has finished? Every flag that I rise with setflag() gets always reset after that call leg finishes!! Thanks in advance, Ricardo.

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[Serusers] store in DB some searched content present in messages
by Ricardo Carvalho 10 Oct '06

10 Oct '06

Is there a way to search and store in a regular expression basis, some content of some kind of messages in an avp and then store that content in some table of MySQL database? Thanks, Ricardo.

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[Users] openser-ser-sems
by geovani ferronetti 10 Oct '06

10 Oct '06

Hi, I have a problem. I have a working openser server, but I want and a media server, so I decide to use sems. I put everything in this way: Openser <-------> Ser<--------->sems All things are working on the same server.Openser works on 5060 port, and ser on 6060 port. In openser.cfg I redirect to ser in this way: if (uri=~"sip:100@.*") { t_relay("localhost:6060"); exit; }; and in ser.cfg: if (uri=~"sip:100.*@") { if(!t_write_unix("/tmp/am_sock","conference")) { t_reply("500","error contacting sems"); }; break; }; When I dial 100 the logs of the ser contain: Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: SIP Request: Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: method: <INVITE> Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: uri: <sip:100@domain.com> Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: version: <SIP/2.0> Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: parse_headers: flags=1 Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: Found param type 232, <branch> = <z9hG4bK2cec.f018b0b5.0>; state=16 Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: end of header reached, state=5 Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: parse_headers: Via found, flags=1 Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: parse_headers: this is the first via Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: After parse_msg... Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: preparing to run routing scripts... Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: parse_headers: flags=128 Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: Found param type 232, <branch> = <z9hG4bKOUUByO4Kp1q4sDNU>; state=16 Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: end of header reached, state=5 Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: parse_headers: Via found, flags=128 Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: parse_headers: this is the second via Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: DEBUG:maxfwd:is_maxfwd_present: value = 69 Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: DBG:maxfwd:process_maxfwd_header: value 69 decreased to 16 Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: parse_headers: flags=8 Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: DEBUG: add_param: tag=6mVbY0KQFDRYupVP Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: end of header reached, state=29 Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: parse_headers: flags=256 Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: end of header reached, state=9 Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: DEBUG: get_hdr_field: <To> [29]; uri=[sip:100@domain.com] Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: DEBUG: to body ["100" <sip:100@voip.cts.md> ] Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: get_hdr_field: cseq <CSeq>: <1> <INVITE> Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: DEBUG: get_hdr_body : content_length=232 Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: found end of header Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: find_first_route: No Route headers found Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: loose_route: There is no Route HF Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: grep_sock_info - checking if host==us: 11==9 && [domain.com] == [127.0.0.1] Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: grep_sock_info - checking if port 6060 matches port 5060 Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: grep_sock_info - checking if host==us: 11==11 && [domain.com] == [X.X.X.X] Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: grep_sock_info - checking if port 6060 matches port 5060 Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: grep_sock_info - checking if host==us: 11==10 && [domain.com] == [172.17.6.2] Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: grep_sock_info - checking if port 6060 matches port 5060 Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: grep_sock_info - checking if host==us: 11==9 && [domain.com] == [127.0.0.1] Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: grep_sock_info - checking if port 6060 matches port 5060 Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: grep_sock_info - checking if host==us: 11==11 && [domain.com] == [X.X.X.X] Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: grep_sock_info - checking if port 6060 matches port 5060 Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: grep_sock_info - checking if host==us: 11==10 && [domain.com] == [172.17.6.2] Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: grep_sock_info - checking if port 6060 matches port 5060 Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: check_self: host != me Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: DEBUG: t_newtran: msg id=3 , global msg id=2 , T onfiltered=0xffffffffffffffff Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: parse_headers: flags=-1 Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: parse_headers: flags=60 Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: t_lookup_request: start searching: hash=52930, isACK=0 Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: DEBUG: RFC3261 transaction matching failed Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: DEBUG: t_lookup_request: no transaction found Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: SER: new INVITE Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: parse_headers: flags=-1 Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: check_via_address(172.17.6.2, 172.17.6.2, 0) Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: WARNING:vqm_resize: resize(0) called Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: DEBUG: reply sent out. buf=0x5d8838: SIP/2.0 1..., shmem=0x2b9b012ee768: SIP/2.0 1 Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: DEBUG: _reply_light: finished Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: DEBUG: mk_proxy: doing DNS lookup... Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: get_record: lookup(_sip._udp.domain.com, 33) failed Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: sip_resolvehost: no SRV record found for voip.cts.md, trying 'normal' lookup... Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: check_via_address(172.17.6.2, 172.17.6.2, 0) Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: DEBUG: add_to_tail_of_timer[4]: 0x2b9b012ec368 Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: DEBUG: add_to_tail_of_timer[0]: 0x2b9b012ec388 Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: SER: new transaction fwd'ed Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: DEBUG:destroy_avp_list: destroying list (nil) Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: receive_msg: cleaning up Please show me where is the problem, in openser or ser or sems? Thanks. Ion M. __________________________________________________ Do You Yahoo!? Poco spazio e tanto spam? Yahoo! Mail ti protegge dallo spam e ti da tanto spazio gratuito per i tuoi file e i messaggi http://mail.yahoo.it

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[Serusers] anyone know about SIPP
by vijay tiwari 10 Oct '06

10 Oct '06

hi all anyone have idea to use SIPP and how to you this tool in brief thanks vijay --------------------------------- Get your own web address for just $1.99/1st yr. We'll help. Yahoo! Small Business.

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[Users] Question about handling retransmission
by Michael Ulitskiy 10 Oct '06

10 Oct '06

Hello, I have a question about a good way to handle retransmissions. I see that t_check_trans() function is able to tell me if the request is a retransmission or not. So my question is: should I apply the whole request-processing logic to retransmissions or I can simply do something like the following: if (is_method("ACK|CANCEL")) { if (t_check_trans()) { t_relay(); } else { xlog(L_NOTICE,"$rm: No matching transaction. dropping"); exit; } } if (t_check_trans()) { xlog(L_NOTICE,"$rm: is a retransmission"); t_relay(); } I.e. in the above code I assume that t_relay() would be able to match the request to the transaction and absorb it like it does for ACKs and CANCELs without applying whatever modification was done to original request. Is it true? Thanks, Michael

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[Users] [Devel] SST module changes
by Ron Winacott 10 Oct '06

10 Oct '06

Hello all, I just updated the SST module to be fully compliant with the RFC. Here is a list of the biggest changes: 1) The script function sstCheckMin() will now add the Min-SE: header and send the 422 reply if the Session-Expire: header in the INVITE is too small. (with the argument flag set to true) The old sstCheckMin() function would return true stating that the Min-SE: value was too small, but the RFC states the 422 reply can *not* be sent without stating your Min-SE: value. 2) The Min-SE: and the Session-Expires: headers will be updated to your min_se value if in the initial request if too small. 3) If the UAC does not request SST support, the SST module will request it on behave of the PROXY. 4) The PRACK issue with the dialog module has been worked around. PRACK will clobber the dialog expires lifetime value but the SST module will fix it. 5) If the UAS does not support SST, the SST module will request the UAC refresh on behave of the PROXY. 6) reINTITE and UPDATE are now supported as refreshers to an active timed session. 7) The introduction of the sst flag parameter. This is keeping with the idea that the module will not do anything to a message unless asked for by the script (openser.cfg) file. You now have to setflag(sst_flag) just like you have to setflag(dialog_flag) for the module to act on a message. 8) Code clean up and commenting of all functions and beefed up the user documentation. Hope you find this module usefull, ronw (karwin) -- Ron Winacott - SOMA Networks, Inc. --- Chaos, panic and disorder...my work here is done. _______________________________________________ Devel mailing list Devel(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/devel

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[Users] set some flag that isn't reset after some call leg has finished
by Ricardo Carvalho 10 Oct '06

10 Oct '06

How may I set some flag that isn't reset after some call leg has finished? Every flag that I rise with setflag() gets always reset after that call leg finishes!! Thanks in advance, Ricardo.

2 1

[Serusers] CDRtool Web Problem
by Davis, Chris 10 Oct '06

10 Oct '06

We just installed CDRtool over ser v.0.9.3 and appeared to install properly. However when we pull up the web portion, we get the initial login page. Login in with the default admin accounts and it takes us to the EULA. After agreeing we are taken to a mostly blank page with 2 frames, 1 pointing to callsearch.phtml and 1 pointing to blank.phtml. No errors are given in reference to database or access. Help? Thanks in advance Chris -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.407 / Virus Database: 268.13.1/466 - Release Date: 10/7/2006

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[users] TLS and OpenSER Issues
by Tracy Lofton 10 Oct '06

10 Oct '06

Hi, I recently experimented with TLS on a server that was initially enabled without openser and it worked fine. Once I remade with TLS and restarted SER, I noticed very poor reception on regular analog (w/ATA) and SIP phones, such as fading out completely on audio, one-way audio issues again, or fading in/out with regarding to sound or no sound at all and just dead, lapsing air, so I concluded that OpenSER with TLS .cfg file is restricted to TLS ONLY. Although I aware that TLS is to be strictly used with TLS enabled software, I thought I would be able to use the same .cfg file and UAs that use particular technologies would still work the same. How should it have actually worked? Is this the norm? Now, I have the TLS enabled Server on an entire different machine so that I can test more thorougly but need to route it with the main OpenSER/Asterisk system to determine how TLS functions using the PSTN. So I am now hoping to experimentally have users wishing secure lines routing thru the one server and then to another system, but sharing the same database as the first server. And I am hoping that audio this way will not be an issue. Is this the norm? Thanks for any enlightenment. Tracy _________________________________________________________________ Add fun gadgets and colorful themes to express yourself on Windows Live Spaces http://clk.atdmt.com/MSN/go/msnnkwsp0070000001msn/direct/01/?href=http://ww…

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[Users] cvs access of rel_1_1_0 failed
by Juha Heinanen 10 Oct '06

10 Oct '06

i tried to checkout rel_1_1_0 to a new host and got: cvs -z3 checkout -r rel_1_1_0 sip-server ... cvs checkout: Updating sip-server/modules/imc cvs checkout: failed to create lock directory for `/cvsroot/openser/sip-server/modules/imc' (/cvsroot/openser/sip-server/modules/imc/#cvs.lock): Permission denied cvs checkout: failed to obtain dir lock in repository `/cvsroot/openser/sip-server/modules/imc' cvs [checkout aborted]: read lock failed - giving up is imc modules supposed to be in rel_1_1_0? -- juha

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[Serusers] Invitation to OpenSER Summit @ VoN Berlin
by Daniel-Constantin Mierla 10 Oct '06

10 Oct '06

Hello everybody, we would like to invite all of you to join the OpenSER Summit to be held during the VoN Autumn Europe, Berlin, November 6-8, 2006. The access is open for anybody, free of charge. Also, all registered participants will get free entry passes to the VoN Exhibition. Courtesy of Pulver Media we can provide full conference passes (exhibition, conference, special events at VoN) during the summit, but in a limited number, so first-come, first-served. Agenda is published at openser site, and will be completed soon with the panelists and topic names: http://www.openser.org/index.php?option=com_content&task=view&id=57&Itemid=… To register, please fill and submit the form at: http://www.openser.org/index.php?option=com_content&task=view&id=57&Itemid=… or, if more convenient to you, send an email with your details at team(a)openser.org. Your request will be recorded and you will be contacted soon for confirmation. Main focus for this event is to reveal the status of OpenSER project -- achievements and roadmap (board members and main developers of the project will be there), discuss businesses cases of OpenSER, network people for business opportunities and cooperations (many executives will attend the event). Just to list a few hot topics in the agenda: - scalability and distribution - secure communication with SIP using OpenSER - 3/4G and convergence - fixed mobile convergence and peering heterogeneous networks - carrier grade and hosted infrastructures, IMS - integration with other open source (and not only!!!) VoIP application to deliver complete and robust VoIP platforms Looking forward to meet you there. Daniel Mierla -- OpenSER Project http://www.openser.org

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[Users] rel_1_1_0 CVS seems to be badly broken
by Juha Heinanen 10 Oct '06

10 Oct '06

this one failed. i didn't check if it is still relevant. -- juha --------------------------------------------------------------------- when i look rel_1_1_0 using ViewVC i see core directory mi and imc module for example. those should not be there: http://openser.cvs.sourceforge.net/openser/sip-server/?pathrev=rel_1_1_0 this in addition to the earlier problem that anon cvs checkout fails. -- juha

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[Users] Request for discussion: FIFO to MI migration
by Bogdan-Andrei Iancu 10 Oct '06

10 Oct '06

Hi everybody, also related to the new MI topic - as previously posted, there already an implementation for FIFO interface (FIFO <-> MI translation) - see http://openser.org/docs/modules/1.2.x/mi_fifo.html . There are two hot topics I would like to address: 1) backward compatibility with all fifo implementation. The new fifo syntax is a bit simplify and changed in order to allow to be mapped over an "name = value" approach - see http://openser.org/docs/modules/1.2.x/mi_fifo.html#AEN40 The big difference is when it comes to values spreading across multiple lines (as the FIFO syntax is line orientated). In old implementation the line order was known by the functions implementing the external commands- lines with a single dot were used as end-markers for sets of line that should be interpreted as a single value. The MI interface is just doing translation (is not functionality aware) so a more coherent definition was needed - for multiple line value, quoted strings are used. This will break compatibility for functions using multiple line values like t_uac_dlg(). There are just a few functions, but they are. 2) for how long should we keep in parallel the two FIFO / UNIXSOCK implementations before dropping the old ones? I am a fan of a fast transition I would drop the old impls as soon as the new one offer the same functioanlity - this will accelerate the testing and migration process. So the questions are - when to drop the old impls and how to deal with backward compatibilities (as MI will not be 100% compat with SEMS for example) One solution (based on my opinions) - drop as soon as possible and use for backward compatibility a FIFO proxy to do translation between versions (for people interested in). any other solutions are welcome for debate. Regards, Bogdan

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[Users] add_diversion working
by Helge Waastad 10 Oct '06

10 Oct '06

Hi, I was just wondering if someone could confirm that the add_redirect function works in OpenSER 1.1.0? I have some problems using this function. br hw -- Helge Waastad Senior Engineer Systemavdelingen Smartnet

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[Users] Problem with module dispatcher
by Jose Gil Navarrete 10 Oct '06

10 Oct '06

Hi In my server I have: if (!method=="REGISTER"){ setflag(1); setflag(2); record_route(); } Date: Fri, 29 Sep 2006 09:36:49 +0300 From: Daniel-Constantin Mierla <daniel(a)voice-system.ro> Subject: Re: [Users] Problem with module dispatcher To: Jose Gil Navarrete <gijoenav(a)yahoo.com.mx> Cc: users(a)openser.org Message-ID: <451CBF01.9030508(a)voice-system.ro> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hello, do you do record routing on server? The ACK should follow Route headers. A network trace (ngrep, ethereal) will help to identify the cause for the problem. Cheers, Daniel On 09/29/06 00:29, Jose Gil Navarrete wrote: > Hi all > > I have implemented the load balance in mode statefull, I obtained the > next messages: > > phone1 dispatcher > sipserver phone2 > --------INVITE--------> ----INVITE-----------> > ----INVITE--------> > <----100 TRYING---- <---100 TRYING---- <---100 TRYING---- > <----180 RINGING-- <---180 RINGING--- <---180 RINGING--- > <----200 OK---------- <----- 200 OK-------- <---200 > OK-------- > -------- ACK --------> --------- ACK --------> > > > You can see that the sipserver never sends ACK message to video_phone2. > Is this correct? > > Then I try the next, in the sipserver: > if (method =="ACK"){ > if (!t_relay()) { > sl_reply_error(); > }; > return; > } > > But the sipserver doesn't send the message ACK at the phone2? > Does some body know what it's wrong? > > --------------------------------- Do You Yahoo!? La mejor conexión a Internet y 2GB extra a tu correo por $100 al mes. http://net.yahoo.com.mx

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[Users] Using forward_tcp help
by Nhadie Ramos 10 Oct '06

10 Oct '06

Hi All, I have the REGISTER part of my ser.cfg below, what do you think is wrong because it doesn't replicate it. Do i have to do any changes on parameters of required modules? if (method=="REGISTER") { if ((src_ip=="10.10.10.20") || (src_ip=="10.10.10.30")) { if (isflagset(8)) {xlog("L_INFO", "Register: Source is either 10.10.10.20 or 10.10.10.30\n");}; save_noreply("location"); } else if (!www_authorize("mydomain.com", "subscriber")) { if (isflagset(8)) {xlog("L_INFO", "Register: asking for authentication\n");}; www_challenge("mydomain.com", "0"); break; }; if (isflagset(8)) {xlog("L_INFO", "Register: Authorized, saving location\n");}; if (!save("location")) { sl_reply_error(); } else { if (isflagset(8)) {xlog("L_INFO", "Register: Authorized, saving location\n");}; if ((!src_ip=="10.10.10.20") || (!src_ip=="10.10.10.30")) { if (isflagset(8)) {xlog("L_INFO", "Register: Replicating to 10.10.10.20\n");}; forward_tcp("10.10.10.20", 5060); forward_tcp("10.10.10.30", 5060); }; }; break; };

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[Users] Re: Minisip
by Gregoire 10 Oct '06

10 Oct '06

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi! I am using Minisip on Linux Ubuntu Dapper. For the certificates on the client side, you can generate and sign it by yourself with openssl. See ( http://www.maemo.org/platform/docs/tutorials/certman.html). On ther server side, juste use the script genrootCA.sh to generate your CA certificate and genuser.sh to get you server certificate signed by your CA created before. Don't forget to put the file "user-calist.pem" also on the client side and add it in Minisip, into the section CA Database. Sincerely Gregoire Tracy Lofton wrote: > Hi, > > One other questions, where do I put all the certificates that are > generated, such as client side certificates/keys, and server side > certificate and keys. I could not figure it out so I just stopped > trying after awhile. > > Thanks, > > Tracy > > _________________________________________________________________ > Add fun gadgets and colorful themes to express yourself on Windows > Live Spaces > http://clk.atdmt.com/MSN/go/msnnkwsp0070000001msn/direct/01/?href=http://ww… > > > -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.2.2 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFK0eeI8gmGeMTr0sRAtaDAJ9A+1ry6XUOi1otKituE4RhbTALLgCgircP Q2yforFBbAa0O78jmlRh7rs= =Ha79 -----END PGP SIGNATURE-----

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[Users] Whixh Router is Best for security...
by raviprakash sunkara 10 Oct '06

10 Oct '06

Hello Users, If I Keep the Router , My openser is Not working fine. But If place it Directly to modem, it works Kool and Fine. Actually My router is Firewall router which does have the NAT configuring , It was NetGear.. Can tell Which Router I need to Use for My Sip Server...... -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 ravi.sunkara(a)hyperion-tech.com www.hyperion-tech.com

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[Users] Openser ALG setup
by Lindsey Hans Joseph Lao 10 Oct '06

10 Oct '06

Hello All, I am setting up Openser in this configuration: Client ---------> (private interface) Router (public interface) --------> Registrar The router is configured with Openser having mediaproxy module support. What I want to achieve is for nat traversal to happen at the router side such that when the sip messages reach the registrar, it would already contain the public IP address in its contact header field. I am using Openser here more like an application level gateway. Openser is listening on the public interface only. Currently, my setup is not working with the private address of the client registering in the registrar. Is this a possible setup? Can someone show me the way? Thamk you very much. Hans

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[serusers] Connecting Another SER
by John Shellhouse 10 Oct '06

10 Oct '06

Hi Everyone, I currently have SER and Asterisk on one machine, and recently launched another SER on another machine and wanted to know how do I connect SER-1 to SER-2. Thanks in advance. John Shellhouse

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[Serusers] Need help proper way to see avpops value
by reza shah 10 Oct '06

10 Oct '06

Hi, I just try avpops to get a value from usr_preferences. I have followed the onsip.org article, and some of previous email in this list. Still have trouble to see the value inside avpops variable. May be i missed something. i`m using ser 0.9.6 with freebsd. Did i do the right way? my codes are: loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/xlog.so" modparam("avpops","avp_url","mysql://ser:heslo@localhost/ser") modparam("avpops","avp_table","usr_preferences") modparam("avpops","username_column","username") modparam("avpops","domain_column","domain") modparam("avpops","value_column","value") modparam("avpops","type_column","type") modparam("avpops","attribute_column","attribute") #inside register method avp_db_load("$from/domain","s:mapnumber"); xlog("L_NOTICE","value = $avp(s:mapnumber)\n"); __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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RE: [Serusers] DNS and IP
by Rosa De Santis 09 Oct '06

09 Oct '06

Hi again, Weiter. I already have "alias=the.domain.name" in my config, but its not working :( The complete scenario is like this: I have two SIP servers: server A and server B. I have clients registered in each server, lets call them UA1 UA2 registered on server A, and UA3 UA4 on server B. When I use ip to register the clients then I can make calls in all ways. UA1 to UA2, UA1 to UA3, and viceversa, etc... But if I use dns to register then users can call on the same server: UA1<->UA2, UA3<->UA4, but users from server A can't call to users on server B and viceversa. I have the dns registered on both servers. Please, any idea??? Thanks, Rosa. > ----------------------------------------------- > > Date: Sun, 8 Oct 2006 12:02:24 +0200 > > From: bp4mls(a)googlemail.com > > To: rosadesantis(a)hotmail.com > > Subject: Re: [Serusers] DNS and IP > > CC: serusers(a)iptel.org > > > > Make sure the box where SER is running can also resolve the name to it's own IP. > > You can also use "alias=the.domain.name" in you config. > > On 10/6/06, Rosa De Santis <rosadesantis(a)hotmail.com> wrote: > > Hello all. > > Please, I have a problem using DNS or IP on the client side. > > When I use IP all calls are fine, but if I use the DNS of the server that point to the same IP some calls can't be done. > > Is there any place to configure this on ser.cfg ??? > > Please help. > > Thanks. > > _________________________________________________________________ > > Busca desde cualquier página Web con una protección excepcional. Consigue la Barra de herramientas de Windows Live hoy mismo GRATUITAMENTE. > > http://toolbar.live.com_______________________________________________ > > Serusers mailing list > > Serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > _________________________________________________________________ > Llama a tus amigos de PC a PC: ¡Es GRATIS! > http://get.live.com/messenger/overview_____________________________________… > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers _________________________________________________________________ Consigue el nuevo Windows Live Messenger http://get.live.com/messenger/overview

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[Users] openser.org outage
by Daniel-Constantin Mierla 09 Oct '06

09 Oct '06

Hello, just to inform you, today the collocation site hosting the OpenSER site experienced lot of outages (http://www.hetzner-status.de/). We apologizes for the situation and we kindly ask you to re-summit any message which didn't get it to the mailing list. We look forward to implement a mirroring system to be able to handle smoothly such situations. Cheers, Daniel

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[Users] 404 not found retransmission
by Cseke Tamas 09 Oct '06

09 Oct '06

Hi, I have a retransmission problem, i recive an ack for an 404 reply, however opener 1.1.0 retransmit it: U 213.xxx.xxx.92:5060 -> 213.xxx.xxx.32:5060 SIP/2.0 404 Not Found..Via: SIP/2.0/UDP 192.168.1.106;rport=5060;received=2 17.xxx.xxx.22;branch=z9hG4bKf0dec68a3f9e217a..From: <sip:241@callshop.peakte lecom.me.uk>;tag=25dde6813cae8747..To: <sip:1@callshop.>;t ag=D3DB214-25D2..Date: Mon, 09 Oct 2006 11:37:18 GMT..Call-ID: 3c47a6403890 c2f2@192.168.1.106..Server: Cisco-SIPGateway/IOS-12.x..CSeq: 42170 INVITE.. Allow-Events: telephone-event..Content-Length: 0.... # U 213.xxx.xxx.32:5060 -> 213.xxx.xxx.92:5060 ACK sip:1@callshop. SIP/2.0..Via: SIP/2.0/UDP 192.168.1.10 6;branch=z9hG4bKf0dec68a3f9e217a..From: <sip:241@callshop. >;tag=25dde6813cae8747..To: <sip:1@callshop.>;tag=D3DB214- 25D2..Contact: <sip:241@192.168.1.106>..Authorization: Digest username="241 ", realm="callshop.", algorithm=MD5, uri="sip:1@callshop ", nonce="452a37e35171ad890ccd763950d954c45090f992", respon se="f68ecea0d5196e8e88abeac6a68"..Call-ID: 3c47a6403890c2f2(a)192.168.1. 106..CSeq: 42170 ACK..User-Agent: Grandstream BT110 1.0.8.23..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Cont ent-Length: 0.... # U 213.xxx.xxx.92:5060 -> 213.xxx.xxx.32:5060 SIP/2.0 404 Not Found..Via: SIP/2.0/UDP 192.168.1.106;rport=5060;received=2 17.xxx.xxx.22;branch=z9hG4bKf0dec68a3f9e217a..From: <sip:241@callshop.peakte lecom.me.uk>;tag=25dde6813cae8747..To: <sip:1@callshop.>;t ag=D3DB214-25D2..Date: Mon, 09 Oct 2006 11:37:18 GMT..Call-ID: 3c47a6403890 c2f2@192.168.1.106..Server: Cisco-SIPGateway/IOS-12.x..CSeq: 42170 INVITE.. Allow-Events: telephone-event..Content-Length: 0.... # U 213.xxx.xxx.92:5060 -> 213.xxx.xxx.32:5060 SIP/2.0 404 Not Found..Via: SIP/2.0/UDP 192.168.1.106;rport=5060;received=2 17.xxx.xxx.22;branch=z9hG4bKf0dec68a3f9e217a..From: <sip:241@callshop.peakte lecom.me.uk>;tag=25dde6813cae8747..To: <sip:1@callshop.>;t ag=D3DB214-25D2..Date: Mon, 09 Oct 2006 11:37:18 GMT..Call-ID: 3c47a6403890 c2f2@192.168.1.106..Server: Cisco-SIPGateway/IOS-12.x..CSeq: 42170 INVITE.. Allow-Events: telephone-event..Content-Length: 0.... 0(15109) DEBUG: to body [<sip:1@callshop>] 0(15109) get_hdr_field: cseq <CSeq>: <42170> <INVITE> 0(15109) parse_headers: flags=8 0(15109) DEBUG: t_reply_matching: hash 39491 label 455964109 branch 2 0(15109) DEBUG:tm:REF_UNSAFE: after is 1 0(15109) DEBUG: t_reply_matching: reply matched (T=0xb3df7618)! 0(15109) DEBUG: t_check: msg id=35 global id=35 T end=0xb3df7618 0(15109) DEBUG:tm:reply_received: org. status uas=100, uac[2]=100 local=0 is_in vite=1) 0(15109) parse_headers: flags=8 0(15109) DEBUG:tm:build_local: using FROM=<From: <sip:241@callshop me.uk>;tag=25dde6813cae8747 >, TO=<To: <sip:1@callshop>;tag=D3DB214-25D2 >, CSEQ_N=<CSeq: 42170> 0(15109) parse_headers: flags=80 0(15109) DEBUG: get_hdr_body : content_length=0 0(15109) found end of header 0(15109) DEBUG:tm:t_should_relay_response: T_code=100, new_code=404 0(15109) DEBUG:tm:t_pick_branch: picked branch 2, code 404 0(15109) DEBUG:tm:relay_reply: branch=2, save=0, relay=2 0(15109) old size: 482, new size: 427 0(15109) build_res_from_sip_res: copied size: orig:83, new: 28, rest: 399 msg= SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.106;rport=5060;received=217.xxx.xxx.22;branch=z9hG4bKf 0dec68a3f9e217a From: <sip:241@callshop>;tag=25dde6813cae8747 To: <sip:1@callshop>;tag=D3DB214-25D2 Date: Mon, 09 Oct 2006 11:37:18 GMT Call-ID: 3c47a6403890c2f2(a)192.168.1.106 Server: Cisco-SIPGateway/IOS-12.x CSeq: 42170 INVITE Allow-Events: telephone-event Content-Length: 0 0(15109) DEBUG: add_to_tail_of_timer[4]: 0xb3df76c8 0(15109) DEBUG: add_to_tail_of_timer[0]: 0xb3df76d8 0(15109) DEBUG:tm:relay_reply: sent buf=0x812e928: SIP/2.0 4..., shmem=0xb3df8f 50: SIP/2.0 4 0(15109) DEBUG: cleanup_uac_timers: RETR/FR timers reset 0(15109) DEBUG:tm:UNREF_UNSAFE: after is 0 0(15109) DEBUG:destroy_avp_list: destroying list (nil) 0(15109) receive_msg: cleaning up 0(15109) SIP Request: 0(15109) method: <ACK> 0(15109) uri: <sip:1@callshop> 0(15109) version: <SIP/2.0> 0(15109) parse_headers: flags=2 0(15109) Found param type 232, <branch> = <z9hG4bKf0dec68a3f9e217a>; state=16 0(15109) end of header reached, state=5 0(15109) parse_headers: Via found, flags=2 0(15109) parse_headers: this is the first via 0(15109) After parse_msg... 0(15109) preparing to run routing scripts... 0(15109) parse_headers: flags=8 0(15109) DEBUG: add_param: tag=D3DB214-25D2 0(15109) DEBUG:parse_to:end of header reached, state=29 0(15109) DBUG:parse_to: display={}, ruri={sip:1@callshop} 0(15109) DEBUG: get_hdr_field: <To> [53]; uri=[sip:1@callshop ] 0(15109) DEBUG: to body [<sip:1@callshop>] 0(15109) parse_headers: flags=100 0(15109) get_hdr_field: cseq <CSeq>: <42170> <ACK> 0(15109) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(15109) DEBUG: add_param: tag=25dde6813cae8747 0(15109) DEBUG:parse_to:end of header reached, state=29 0(15109) DBUG:parse_to: display={}, ruri={sip:241@callshop} 0(15109) DEBUG: t_check: msg id=36 global id=35 T start=0xffffffff 0(15109) parse_headers: flags=ffffffffffffffff 0(15109) DEBUG: get_hdr_body : content_length=0 0(15109) found end of header 0(15109) parse_headers: flags=78 0(15109) t_lookup_request: start searching: hash=39491, isACK=1 0(15109) DEBUG: RFC3261 transaction matched, tid=f0dec68a3f9e217a 0(15109) DEBUG:tm:REF_UNSAFE: after is 1 0(15109) DEBUG: t_lookup_request: transaction found (T=0xb3df7618) 0(15109) DEBUG: t_check: msg id=36 global id=36 T end=0xb3df7618 0(15109) DEBUG:t_check_status: checked status is <404> 0(15109) parse_headers: flags=58 0(15109) parse_headers: flags=200 0(15109) find_first_route: No Route headers found 0(15109) loose_route: There is no Route HF 0(15109) grep_sock_info - checking if host==us: 26==12 && [callshop. m.me.uk] == [213.166.8.92] 0(15109) grep_sock_info - checking if port 5060 matches port 5060 0(15109) grep_sock_info - checking if host==us: 26==12 && [callshop. m.me.uk] == [213.166.8.92] 0(15109) grep_sock_info - checking if port 5060 matches port 5060 0(15109) DEBUG: t_newtran: msg id=36 , global msg id=36 , T on entrance=0xb3df7 618 * 0(15109) ERROR: t_newtran: transaction already in process 0xb3df7618* 0(15109) DEBUG:tm:UNREF_UNSAFE: after is 0 0(15109) DEBUG:destroy_avp_list: destroying list (nil) 0(15109) receive_msg: cleaning up 1(15110) DEBUG: timer routine:1,tl=0xb3df78ec next=0xb3df7818 1(15110) DEBUG: timer routine:1,tl=0xb3df7818 next=0xb3df7744 1(15110) DEBUG: timer routine:1,tl=0xb3df7744 next=(nil) 1(15110) DEBUG: timer routine:4,tl=0xb3df76c8 next=0xb3df78dc 1(15110) DEBUG: retransmission_handler : reply resending (t=0xb3df7618, SIP/2.0 4 ... ) 1(15110) DEBUG:tm:t_retransmit_reply: buf=0xb7db8c60: SIP/2.0 4..., shmem=0xb3d f8f50: SIP/2.0 4 1(15110) DEBUG: add_to_tail_of_timer[5]: 0xb3df76c8 1(15110) DEBUG: retransmission_handler : done 1(15110) DEBUG: timer routine:4,tl=0xb3df78dc next=0xb3df7808 1(15110) DEBUG: timer routine:4,tl=0xb3df7808 next=0xb3df7734 1(15110) DEBUG: timer routine:4,tl=0xb3df7734 next=(nil) 1(15110) DEBUG: timer routine:5,tl=0xb3df76c8 next=(nil) 1(15110) DEBUG: retransmission_handler : reply resending (t=0xb3df7618, SIP/2.0 4 ... ) 1(15110) DEBUG:tm:t_retransmit_reply: buf=0xb7db8c60: SIP/2.0 4..., shmem=0xb3d f8f50: SIP/2.0 4 1(15110) DEBUG: add_to_tail_of_timer[6]: 0xb3df76c8 1(15110) DEBUG: retransmission_handler : done 1(15110) DEBUG: timer routine:6,tl=0xb3df76c8 next=(nil) 1(15110) DEBUG: retransmission_handler : reply resending (t=0xb3df7618, SIP/2.0 4 ... ) 1(15110) DEBUG:tm:t_retransmit_reply: buf=0xb7db8c60: SIP/2.0 4..., shmem=0xb3d f8f50: SIP/2.0 4 1(15110) DEBUG: add_to_tail_of_timer[7]: 0xb3df76c8 1(15110) DEBUG: retransmission_handler : done 1(15110) DEBUG: timer routine:7,tl=0xb3df76c8 next=(nil) 1(15110) DEBUG: retransmission_handler : reply resending (t=0xb3df7618, SIP/2.0 4 ... ) 1(15110) DEBUG:tm:t_retransmit_reply: buf=0xb7db8c60: SIP/2.0 4..., shmem=0xb3d f8f50: SIP/2.0 4 1(15110) DEBUG: add_to_tail_of_timer[7]: 0xb3df76c8 1(15110) DEBUG: retransmission_handler : done 1(15110) INFO: signal 2 received 1(15110) Memory status (pkg): 1(15110) fm_status (0x811fd80): 1(15110) heap size= 1048576 1(15110) used= 34976, used+overhead=55056, free=993520 1(15110) max used (+overhead)= 55880 1(15110) dumping free list: 1(15110) hash = 7 fragments no.: 1, unused: 0 bucket size: 56 - 56 (first 56) 1(15110) hash = 16 fragments no.: 6, unused: 0 bucket size: 128 - 128 (first 128) 1(15110) hash = 255 fragments no.: 1, unused: 0 bucket size: 2040 - 2040 (first 2040) 1(15110) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 990656) 1(15110) TOTAL: 9 free fragments = 993520 free bytes 1(15110) ----------------------------- thanks any help, Tamas

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RE: [Serusers] DNS and IP
by Rosa De Santis 09 Oct '06

09 Oct '06

Hi, Weiter. Thanks for your answer. I already have "alias=the.domain.name" in my config, but its not working :( Any ideas??? Thanks again. Rosa. ----------------------------------------------- > Date: Sun, 8 Oct 2006 12:02:24 +0200 > From: bp4mls(a)googlemail.com > To: rosadesantis(a)hotmail.com > Subject: Re: [Serusers] DNS and IP > CC: serusers(a)iptel.org > > Make sure the box where SER is running can also resolve the name to it's own IP. > You can also use "alias=the.domain.name" in you config. > On 10/6/06, Rosa De Santis <rosadesantis(a)hotmail.com> wrote: > Hello all. > Please, I have a problem using DNS or IP on the client side. > When I use IP all calls are fine, but if I use the DNS of the server that point to the same IP some calls can't be done. > Is there any place to configure this on ser.cfg ??? > Please help. > Thanks. > _________________________________________________________________ > Busca desde cualquier página Web con una protección excepcional. Consigue la Barra de herramientas de Windows Live hoy mismo GRATUITAMENTE. > http://toolbar.live.com_______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers _________________________________________________________________ Llama a tus amigos de PC a PC: ¡Es GRATIS! http://get.live.com/messenger/overview

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Re: [Users] Multi Call-Legs accounting configuration
by Ricardo Carvalho 09 Oct '06

09 Oct '06

I've done like Rafael told in the e-mail shown bellow for 1.1.x version, although I was expecting to see in acc table to be written A B and C contacts of forwarded calls (when A calls B and B forwards to C), but there is never written B contact, the one who should be billed for that call... the entries in my acc table for forwarded calls are like this: | sip_from | sip_to | i_uri | o_uri | from_uri | to_uri | src_leg | dst_leg | |---------------------------------------------------------------------------| | A | C | C | C | A | C | A | C | As seen, my acc table doesn't show B contact!! How should I edit my openser.cfg file so that B contact be shown in acc table for forwarded calls? Thanks, Ricardo. Rafael J. Risco G.V. wrote: > hi ricardo, > for 1.1.x version, try this: > modparam("acc", "multi_leg_enabled", 1) > modparam("acc", "src_leg_avp_id", 110) > modparam("acc", "dst_leg_avp_id", 111) > ... > route { > > ## populate the avps (for Multi call leg accounting): > if (!method=="REGISTER" && !method=="NOTIFY") { > avp_write("$from/username", "$avp(i:110)"); > avp_write("$ruri/username", "$avp(i:111)"); > }; > ..... > > for 1.2 or cvs : > # for mysql-based accouting, use the names of the columns > modparam("acc", "multi_leg_info", > "leg_src=$avp(i:110);leg_dst=$avp(i:111)") > > Do not forget to add src_leg and dst_leg columns in acc table > (leg_src, leg_dst for 1.2) > > rafael > > > On 10/6/06, Ricardo Carvalho <rcarvalho(a)iric.up.pt> wrote: >> I want to use Multi Call-Legs accounting to properly account forwarded >> calls made "in the phones". For example user A calls user B which >> forwards the call to user C. There is only one SIP call but with 2 legs >> ( A to B and B to C). Accounting the legs of a call is required for >> proper billing of the calls. >> >> As can be seen in acc module documentation, I've added the following 3 >> lines to my openser.cfg script: >> >> modparam("acc", "multi_leg_enabled", 1) >> modparam("acc", "src_leg_avp_id", 110) >> modparam("acc", "dst_leg_avp_id", 110) >> >> Although, after testing some forwarded calls, nothing has been written >> to src_leg or dst_leg columns of acc table!! >> What have I done wrong? How may I implement this feature correctly? >> >> Regards, >> Ricardo. >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >> >

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[Users] openser.org down?
by Brandon Price 09 Oct '06

09 Oct '06

Can't access it from NYC, or Sanfran.

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[Users] retransmission 404 not found
by Cseke Tamas 09 Oct '06

09 Oct '06

Hi, I have a retransmission problem, i recive an ack for an 404 reply, however opener 1.1.0 retransmit it: U 213.xxx.xxx.92:5060 -> 213.xxx.xxx.32:5060 SIP/2.0 404 Not Found..Via: SIP/2.0/UDP 192.168.1.106;rport=5060;received=2 17.xxx.xxx.22;branch=z9hG4bKf0dec68a3f9e217a..From: <sip:241@callshop.peakte lecom.me.uk>;tag=25dde6813cae8747..To: <sip:1@callshop.>;t ag=D3DB214-25D2..Date: Mon, 09 Oct 2006 11:37:18 GMT..Call-ID: 3c47a6403890 c2f2@192.168.1.106..Server: Cisco-SIPGateway/IOS-12.x..CSeq: 42170 INVITE.. Allow-Events: telephone-event..Content-Length: 0.... # U 213.xxx.xxx.32:5060 -> 213.xxx.xxx.92:5060 ACK sip:1@callshop. SIP/2.0..Via: SIP/2.0/UDP 192.168.1.10 6;branch=z9hG4bKf0dec68a3f9e217a..From: <sip:241@callshop. >;tag=25dde6813cae8747..To: <sip:1@callshop.>;tag=D3DB214- 25D2..Contact: <sip:241@192.168.1.106>..Authorization: Digest username="241 ", realm="callshop.", algorithm=MD5, uri="sip:1@callshop ", nonce="452a37e35171ad890ccd763950d954c45090f992", respon se="f68ecea0d5196e8e88abeac6a68"..Call-ID: 3c47a6403890c2f2(a)192.168.1. 106..CSeq: 42170 ACK..User-Agent: Grandstream BT110 1.0.8.23..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Cont ent-Length: 0.... # U 213.xxx.xxx.92:5060 -> 213.xxx.xxx.32:5060 SIP/2.0 404 Not Found..Via: SIP/2.0/UDP 192.168.1.106;rport=5060;received=2 17.xxx.xxx.22;branch=z9hG4bKf0dec68a3f9e217a..From: <sip:241@callshop.peakte lecom.me.uk>;tag=25dde6813cae8747..To: <sip:1@callshop.>;t ag=D3DB214-25D2..Date: Mon, 09 Oct 2006 11:37:18 GMT..Call-ID: 3c47a6403890 c2f2@192.168.1.106..Server: Cisco-SIPGateway/IOS-12.x..CSeq: 42170 INVITE.. Allow-Events: telephone-event..Content-Length: 0.... # U 213.xxx.xxx.92:5060 -> 213.xxx.xxx.32:5060 SIP/2.0 404 Not Found..Via: SIP/2.0/UDP 192.168.1.106;rport=5060;received=2 17.xxx.xxx.22;branch=z9hG4bKf0dec68a3f9e217a..From: <sip:241@callshop.peakte lecom.me.uk>;tag=25dde6813cae8747..To: <sip:1@callshop.>;t ag=D3DB214-25D2..Date: Mon, 09 Oct 2006 11:37:18 GMT..Call-ID: 3c47a6403890 c2f2@192.168.1.106..Server: Cisco-SIPGateway/IOS-12.x..CSeq: 42170 INVITE.. Allow-Events: telephone-event..Content-Length: 0.... 0(15109) DEBUG: to body [<sip:1@callshop>] 0(15109) get_hdr_field: cseq <CSeq>: <42170> <INVITE> 0(15109) parse_headers: flags=8 0(15109) DEBUG: t_reply_matching: hash 39491 label 455964109 branch 2 0(15109) DEBUG:tm:REF_UNSAFE: after is 1 0(15109) DEBUG: t_reply_matching: reply matched (T=0xb3df7618)! 0(15109) DEBUG: t_check: msg id=35 global id=35 T end=0xb3df7618 0(15109) DEBUG:tm:reply_received: org. status uas=100, uac[2]=100 local=0 is_in vite=1) 0(15109) parse_headers: flags=8 0(15109) DEBUG:tm:build_local: using FROM=<From: <sip:241@callshop me.uk>;tag=25dde6813cae8747 >, TO=<To: <sip:1@callshop>;tag=D3DB214-25D2 >, CSEQ_N=<CSeq: 42170> 0(15109) parse_headers: flags=80 0(15109) DEBUG: get_hdr_body : content_length=0 0(15109) found end of header 0(15109) DEBUG:tm:t_should_relay_response: T_code=100, new_code=404 0(15109) DEBUG:tm:t_pick_branch: picked branch 2, code 404 0(15109) DEBUG:tm:relay_reply: branch=2, save=0, relay=2 0(15109) old size: 482, new size: 427 0(15109) build_res_from_sip_res: copied size: orig:83, new: 28, rest: 399 msg= SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.106;rport=5060;received=217.xxx.xxx.22;branch=z9hG4bKf 0dec68a3f9e217a From: <sip:241@callshop>;tag=25dde6813cae8747 To: <sip:1@callshop>;tag=D3DB214-25D2 Date: Mon, 09 Oct 2006 11:37:18 GMT Call-ID: 3c47a6403890c2f2(a)192.168.1.106 Server: Cisco-SIPGateway/IOS-12.x CSeq: 42170 INVITE Allow-Events: telephone-event Content-Length: 0 0(15109) DEBUG: add_to_tail_of_timer[4]: 0xb3df76c8 0(15109) DEBUG: add_to_tail_of_timer[0]: 0xb3df76d8 0(15109) DEBUG:tm:relay_reply: sent buf=0x812e928: SIP/2.0 4..., shmem=0xb3df8f 50: SIP/2.0 4 0(15109) DEBUG: cleanup_uac_timers: RETR/FR timers reset 0(15109) DEBUG:tm:UNREF_UNSAFE: after is 0 0(15109) DEBUG:destroy_avp_list: destroying list (nil) 0(15109) receive_msg: cleaning up 0(15109) SIP Request: 0(15109) method: <ACK> 0(15109) uri: <sip:1@callshop> 0(15109) version: <SIP/2.0> 0(15109) parse_headers: flags=2 0(15109) Found param type 232, <branch> = <z9hG4bKf0dec68a3f9e217a>; state=16 0(15109) end of header reached, state=5 0(15109) parse_headers: Via found, flags=2 0(15109) parse_headers: this is the first via 0(15109) After parse_msg... 0(15109) preparing to run routing scripts... 0(15109) parse_headers: flags=8 0(15109) DEBUG: add_param: tag=D3DB214-25D2 0(15109) DEBUG:parse_to:end of header reached, state=29 0(15109) DBUG:parse_to: display={}, ruri={sip:1@callshop} 0(15109) DEBUG: get_hdr_field: <To> [53]; uri=[sip:1@callshop ] 0(15109) DEBUG: to body [<sip:1@callshop>] 0(15109) parse_headers: flags=100 0(15109) get_hdr_field: cseq <CSeq>: <42170> <ACK> 0(15109) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(15109) DEBUG: add_param: tag=25dde6813cae8747 0(15109) DEBUG:parse_to:end of header reached, state=29 0(15109) DBUG:parse_to: display={}, ruri={sip:241@callshop} 0(15109) DEBUG: t_check: msg id=36 global id=35 T start=0xffffffff 0(15109) parse_headers: flags=ffffffffffffffff 0(15109) DEBUG: get_hdr_body : content_length=0 0(15109) found end of header 0(15109) parse_headers: flags=78 0(15109) t_lookup_request: start searching: hash=39491, isACK=1 0(15109) DEBUG: RFC3261 transaction matched, tid=f0dec68a3f9e217a 0(15109) DEBUG:tm:REF_UNSAFE: after is 1 0(15109) DEBUG: t_lookup_request: transaction found (T=0xb3df7618) 0(15109) DEBUG: t_check: msg id=36 global id=36 T end=0xb3df7618 0(15109) DEBUG:t_check_status: checked status is <404> 0(15109) parse_headers: flags=58 0(15109) parse_headers: flags=200 0(15109) find_first_route: No Route headers found 0(15109) loose_route: There is no Route HF 0(15109) grep_sock_info - checking if host==us: 26==12 && [callshop. m.me.uk] == [213.166.8.92] 0(15109) grep_sock_info - checking if port 5060 matches port 5060 0(15109) grep_sock_info - checking if host==us: 26==12 && [callshop. m.me.uk] == [213.166.8.92] 0(15109) grep_sock_info - checking if port 5060 matches port 5060 0(15109) DEBUG: t_newtran: msg id=36 , global msg id=36 , T on entrance=0xb3df7 618 * 0(15109) ERROR: t_newtran: transaction already in process 0xb3df7618* 0(15109) DEBUG:tm:UNREF_UNSAFE: after is 0 0(15109) DEBUG:destroy_avp_list: destroying list (nil) 0(15109) receive_msg: cleaning up 1(15110) DEBUG: timer routine:1,tl=0xb3df78ec next=0xb3df7818 1(15110) DEBUG: timer routine:1,tl=0xb3df7818 next=0xb3df7744 1(15110) DEBUG: timer routine:1,tl=0xb3df7744 next=(nil) 1(15110) DEBUG: timer routine:4,tl=0xb3df76c8 next=0xb3df78dc 1(15110) DEBUG: retransmission_handler : reply resending (t=0xb3df7618, SIP/2.0 4 ... ) 1(15110) DEBUG:tm:t_retransmit_reply: buf=0xb7db8c60: SIP/2.0 4..., shmem=0xb3d f8f50: SIP/2.0 4 1(15110) DEBUG: add_to_tail_of_timer[5]: 0xb3df76c8 1(15110) DEBUG: retransmission_handler : done 1(15110) DEBUG: timer routine:4,tl=0xb3df78dc next=0xb3df7808 1(15110) DEBUG: timer routine:4,tl=0xb3df7808 next=0xb3df7734 1(15110) DEBUG: timer routine:4,tl=0xb3df7734 next=(nil) 1(15110) DEBUG: timer routine:5,tl=0xb3df76c8 next=(nil) 1(15110) DEBUG: retransmission_handler : reply resending (t=0xb3df7618, SIP/2.0 4 ... ) 1(15110) DEBUG:tm:t_retransmit_reply: buf=0xb7db8c60: SIP/2.0 4..., shmem=0xb3d f8f50: SIP/2.0 4 1(15110) DEBUG: add_to_tail_of_timer[6]: 0xb3df76c8 1(15110) DEBUG: retransmission_handler : done 1(15110) DEBUG: timer routine:6,tl=0xb3df76c8 next=(nil) 1(15110) DEBUG: retransmission_handler : reply resending (t=0xb3df7618, SIP/2.0 4 ... ) 1(15110) DEBUG:tm:t_retransmit_reply: buf=0xb7db8c60: SIP/2.0 4..., shmem=0xb3d f8f50: SIP/2.0 4 1(15110) DEBUG: add_to_tail_of_timer[7]: 0xb3df76c8 1(15110) DEBUG: retransmission_handler : done 1(15110) DEBUG: timer routine:7,tl=0xb3df76c8 next=(nil) 1(15110) DEBUG: retransmission_handler : reply resending (t=0xb3df7618, SIP/2.0 4 ... ) 1(15110) DEBUG:tm:t_retransmit_reply: buf=0xb7db8c60: SIP/2.0 4..., shmem=0xb3d f8f50: SIP/2.0 4 1(15110) DEBUG: add_to_tail_of_timer[7]: 0xb3df76c8 1(15110) DEBUG: retransmission_handler : done 1(15110) INFO: signal 2 received 1(15110) Memory status (pkg): 1(15110) fm_status (0x811fd80): 1(15110) heap size= 1048576 1(15110) used= 34976, used+overhead=55056, free=993520 1(15110) max used (+overhead)= 55880 1(15110) dumping free list: 1(15110) hash = 7 fragments no.: 1, unused: 0 bucket size: 56 - 56 (first 56) 1(15110) hash = 16 fragments no.: 6, unused: 0 bucket size: 128 - 128 (first 128) 1(15110) hash = 255 fragments no.: 1, unused: 0 bucket size: 2040 - 2040 (first 2040) 1(15110) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 990656) 1(15110) TOTAL: 9 free fragments = 993520 free bytes 1(15110) ----------------------------- thanks any help, Tamas

1 0

[Users] Multi Call-Legs accounting configuration
by Ricardo Carvalho 09 Oct '06

09 Oct '06

I want to use Multi Call-Legs accounting to properly account forwarded calls made "in the phones". For example user A calls user B which forwards the call to user C. There is only one SIP call but with 2 legs ( A to B and B to C). Accounting the legs of a call is required for proper billing of the calls. As can be seen in acc module documentation, I've added the following 3 lines to my openser.cfg script: modparam("acc", "multi_leg_enabled", 1) modparam("acc", "src_leg_avp_id", 110) modparam("acc", "dst_leg_avp_id", 110) Although, after testing some forwarded calls, nothing has been written to src_leg or dst_leg columns of acc table!! What have I done wrong? How may I implement this feature correctly? Regards, Ricardo.

3 3

[Serusers] Using forward_tcp help
by Nhadie Ramos 09 Oct '06

09 Oct '06

Hi All, I have the REGISTER part of my ser.cfg below, what do you think is wrong because it doesn't replicate it. Do i have to do any changes on parameters of required modules? if (method=="REGISTER") { if ((src_ip=="10.10.10.20") || (src_ip=="10.10.10.30")) { if (isflagset(8)) {xlog("L_INFO", "Register: Source is either 10.10.10.20 or 10.10.10.30\n");}; save_noreply("location"); } else if (!www_authorize("mydomain.com", "subscriber")) { if (isflagset(8)) {xlog("L_INFO", "Register: asking for authentication\n");}; www_challenge("mydomain.com", "0"); break; }; if (isflagset(8)) {xlog("L_INFO", "Register: Authorized, saving location\n");}; if (!save("location")) { sl_reply_error(); } else { if (isflagset(8)) {xlog("L_INFO", "Register: Authorized, saving location\n");}; if ((!src_ip=="10.10.10.20") || (!src_ip=="10.10.10.30")) { if (isflagset(8)) {xlog("L_INFO", "Register: Replicating to 10.10.10.20\n");}; forward_tcp("10.10.10.20", 5060); forward_tcp("10.10.10.30", 5060); }; }; break; };

3 4

[Serusers] RE: Serusers Digest, Vol 42, Issue 6
by Ramachandran 09 Oct '06

09 Oct '06

Deal All, Its really good news that Open IMS components are going to be available soon, Is it possible for me to download / access those currently available version of SER 0.10.99-dev41 ? Thanks a million in advance. Thanks, Ramachandran -----Original Message----- From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of serusers-request(a)lists.iptel.org Sent: Saturday, October 07, 2006 5:30 PM To: serusers(a)lists.iptel.org Subject: Serusers Digest, Vol 42, Issue 6 Send Serusers mailing list submissions to serusers(a)lists.iptel.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.iptel.org/mailman/listinfo/serusers or, via email, send a message with subject or body 'help' to serusers-request(a)lists.iptel.org You can reach the person managing the list at serusers-owner(a)lists.iptel.org When replying, please edit your Subject line so it is more specific than "Re: Contents of Serusers digest..." Today's Topics: 1. Re: SIP to AS communication (Dragos Vingarzan) 2. best way to route a huge number of calls (Sameer Ahmed) ---------------------------------------------------------------------- Message: 1 Date: Fri, 06 Oct 2006 14:14:34 +0200 From: Dragos Vingarzan <vingarzan(a)fokus.fraunhofer.de> Subject: Re: [Serusers] SIP to AS communication To: serusers(a)iptel.org Cc: Jiri Kuthan <jiri(a)iptel.org> Message-ID: <452648AA.6090106(a)fokus.fraunhofer.de> Content-Type: text/plain; charset=windows-1252; format=flowed ...This is re-post, because the previous one was sent from the wrong mail-address and now it awaits moderations... sorry for duplicates if eventually it gets accepted. Hi all, Yes, IMS is quite bloated. However, for the past years we did some work at FOKUS related to it and it will be soon released. It does not have ALL what IMS demands, but it is a start and we hope that you guys, if you think that something more is required, could help us achieve that. I am attaching the proof that it will be, on 16.11.2006 (it is also an invitation for you to participate, but I am not making publicity on this list ;-) http://www.fokus.fraunhofer.de/event/ims_ws_06/ ). If you need further info, feel free to contact us. To offer you some info about it before this date, we will release a bundle of modules: - cdp - for efficient diameter communication in SER (stands for CDiameterPeer) - pcscf - implementing some of the P-CSCF functionality - icscf - -//- ICSCF - scscf- -//- SCSCF - isc - for Initial Filter Criteria Triggering and communication between S-CSCF and AS - sip2ims - a simple MD5 to AKA authorization gateway All these modules at the moment are relatively stable on SER 0.10.99-dev41. But the problem there is that the diameter functionality is integrated into the core. We are working now on stabilizing the cdp as a module in 0.10.99-dev45. Once we succeed, I hope that the modules will be usable with any new plain-CVS SER release. Also, we will give away FHoSS - our Java HSS implementation together with the included pure Java - Java Diameter Peer - diameter functionality. So, you will have a basic IMS system to play with. Almost the same Open IMS Core is used in the ETSI/TISPAN IMS Benchmarking effort as a reference implementation for the System Under Test (look for WorkItem 06024). Don't expect too much, but at least now we have full Cx support, Initial Filter Criteria triggering, ISC support, AKA authentication, IPSec in transport mode with clients, etc... and all this plus the wonderful SER script functionality. Oh yeah :)... and don't expect carrier-grade quality though. These are research projects and we care more about experimenting with the latest features than the 99.999%. For that stability, as Jiri implies, Tekelec probably has (or will have) a good solution, but we can't do that as our budgets are much lower than those of the SPs or TEMS. As Andrey pointed out, we are not the only ones doing this, but I hope that we could converge. And Jiri is right about the complexity. There are at least 4 components that have to work together. For SigComp, because of it's complexity, I was considering TLS with compression. This could also replace the IPSec transport mode tunnel for security, which could be difficult on some embedded devices. Any thoughts on this? Anyway, SigComp is asked mainly by the mobile operators only and we should not . Looking forward for you comments, Dragos Vingarzan Jiri Kuthan wrote: > Just a warning small in size but I guess big in the impact. I think people kind > of underappreciate the effort which has to be taken. If you consider all the > 3gpp complexity (sigcomp as a prime example, one of many being real non-trivial), > size of mobile networks for which the system has to be running, number of > interfaces to speak over and achieve interoperability (management, diameter, > applications, cdr processing...), routing features (the 'scim' feature with > which this discussion began)... > > I occur to know Tekelec has gone through this exercise and generating > a reasonable result has cost quite an investment. > > > At 07:19 06/10/2006, Andrey Kuprianov wrote: > >> Hi, I suppose you need to develop modules for all the the x-CSCF + AS (where x-CSCF is P-CSCF, S-CSCF or I-CSCF). You see, the functionalities/features that x-CSCF and application servers require (for instance, P-headers or querying for subscriber information from HSS) are not available in SER. That's why you need to make your own modules *all the way*. As for HSS, i dont think you need to make some special super DIAMETER-enabled database. In my (an in many other people's btw) point of view a simple database, like MySQL is more than capable of storing subscriber info or initial filtering criterias (iFC --> im talking about SCIM/SIMF). If you are really set to develop IMS on SER, then maybe you can start with SER 0.8.14. You can still find decent development documentation for it. As for later versions, I dont think any good docs are yet available (someone, please, correct me if im wrong) >> > > The documentation is being worked on, but for the moment you are unfortunately correct. > > >> Just for your info, INT (France) joint with France Telecom are developing IMS on SER as of now (dont know if it's going to be commerial or open-source. >> Contact them if you are interested, maybe you can even help them develop it further :). Plus, Focus is also set to develop a commercial version of IMS for SER and, although, Focus said that their IMS will be available in the last quarter of this year, but nothing's been released yet. :( >> > > That's fantastic news, I'm looking forward to GPL contributions from these > very respectable institutions. This is really a big piece of work and I am > enormously positive about every single new piece of the puzzle. > > -jiri > > >> Good luck, Andrey. On 10/6/06, Kamal.Mann(a)t-systems.com <Kamal.Mann(a)t-systems.com> wrote: > Hi All > We are using SER as CSCF which communicates with HSS and Application Server. > ï§ Whenever a UA sends a request to SER then it should go to some specific table and look for services assigned to that particular UA. > ï§ If UA is authorize to use that service than SER send this to AS. > ï§ AS then send this request/response to Serving SER (S-CSCF) which further serves that UA. > I need your suggestions for the same that what kind of module we need to develop for this scenario or any valuable suggestions. > > Thanks in anticipation > Kamal Mann > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > > </x-flowed> >> _______________________________________________ >> Serusers mailing list >> Serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >> > > -- > Jiri Kuthan http://iptel.org/~jiri/ > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > -- ----------------------------------------- Dipl. Eng. Dragos Vingarzan FOKUS/NGNI Kaiserin-Augusta-Allee 31 10589 Berlin,Germany Phone +49 (0)30 - 3463 - 7385 Mobile +49 (0)163 - 159 - 5221 eMail vingarzan(a)fokus.fraunhofer.de Web www.fokus.fraunhofer.de We could change the world if God would give us the source code... ----------------------------------------------------------------- ------------------------------ Message: 2 Date: Sat, 7 Oct 2006 13:00:54 +0500 From: "Sameer Ahmed" <sameer.ahmed(a)gmail.com> Subject: [Serusers] best way to route a huge number of calls To: serusers(a)lists.iptel.org Message-ID: <413b539b0610070100l2a65474av2c99b805bb1c1db3(a)mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" Hi What is the best way to filter through a large number of subscribers registered on SER? Hundreds of IF...ELSE statements don't look efficient, nor pretty. Regards, Sameer

1 0

[Users] rel_1_1_0 CVS seems to be badly broken
by Juha Heinanen 09 Oct '06

09 Oct '06

when i look rel_1_1_0 using ViewVC i see core directory mi and imc module for example. those should not be there: http://openser.cvs.sourceforge.net/openser/sip-server/?pathrev=rel_1_1_0 this in addition to the earlier problem that anon cvs checkout fails. -- juha

1 0

[Serusers] startup warning
by Richard C. Thompson 09 Oct '06

09 Oct '06

Hi All Our SER server appears to be working just fine but I get the following message when ever I restart the service. chgrp: cannot access `/tmp/ser_fifo': No such file or directory can anyone give me a hint as to what I should look for. The directory does exist and the file too. SER is running as root. TIA Rick

1 0

[Users] What is this certificate belongs to? Please help..
by Ferianto siregar 09 Oct '06

09 Oct '06

Dear all, Thank you very much for your time to read this message. Thank you All, I have qustion about certificate that we build in openser in order the openser support TLS. They are CA root certificate and client/server certificate. Here is my question: 1. What is certificate file that we have got after running ./gen_rootca.sh belogs to? I mean, what is cacert.pem, 01.pem, cakey.pem belongs to? Is it belongs to server? and for what does it use? 2. What is certificate file that we have got after runnign ./gen_usercert.sh belongs to? I mean, what is user-cert.pem,user-privkey.pem, user-calist.pem belongs to? Is it belongs to server or client? and what does it use? I do hope anybody can help me and give me an answer. Please.. Thank you. Regards, Ferianto --------------------------------- Get your email and more, right on the new Yahoo.com

1 0

[Serusers] DNS and IP
by Rosa De Santis 08 Oct '06

08 Oct '06

Hello all. Please, I have a problem using DNS or IP on the client side. When I use IP all calls are fine, but if I use the DNS of the server that point to the same IP some calls can't be done. Is there any place to configure this on ser.cfg ??? Please help. Thanks. _________________________________________________________________ Busca desde cualquier página Web con una protección excepcional. Consigue la Barra de herramientas de Windows Live hoy mismo GRATUITAMENTE. http://toolbar.live.com

2 1

[Serusers] Please tell me here whats going wrong :-(
by Ravi ***** 08 Oct '06

08 Oct '06

please see my ngrep report - and tell me why my SER cannot send ACK to the pstn gw. it is sending to private ip adress with complete wrong adressing like sip: 192.168.1.106:5060 SIP/2.0. i think it has to be like sip:phonenumber@192.168.1.106

3 2

[Users] A Constant Username?
by Kelvin Williams 07 Oct '06

07 Oct '06

Is there a constant Username that I could use within my openser.cfg? I have several database lookups based upon the Username, however I have found that if my user dials out using *67 his ATA replaces his username with "anonymous" thereby making a good majority of my config useless. kw

1 0

[Users] creating user accounts
by Chris Robson 07 Oct '06

07 Oct '06

I used to be able to create user accounts by issuing the following SER command: serctl add UserName UserPassword UserName@email Openser documentation gives the example openserctl add UserName UserPassword UserName@Email But when I do this I get the following error WARINIG: no database engine found - tried '' This command requires a database engine - none was loaded So how do you use openserctl to create new users? Thanks....Chris

4 6

[Serusers] best way to route a huge number of calls
by Sameer Ahmed 07 Oct '06

07 Oct '06

Hi What is the best way to filter through a large number of subscribers registered on SER? Hundreds of IF...ELSE statements don't look efficient, nor pretty. Regards, Sameer

1 0

[Users] UN-REGISTER
by Javier Ramirez 06 Oct '06

06 Oct '06

Hi, I have a question: the user A is register, the ip of register is 192.168.0.55. the user change PC, and this PC have the ip 192.168.0.66 the user register again, in the new machine, but, how to unregister the firsth registration from 192.168.0.55 ? best regards Javier

7 16

[Users] Implementing Round robing
by Asterisk Expert 06 Oct '06

06 Oct '06

Dear all, I have openser setup for voip service. I want to send call to termination provider gateways in round robin fashion. Can anyone suggest me if its feasible or not. Will simultaneous calls and reinvites cause problem or not. Whats the best way to implement it. Thanks -- Regards Asterisk Expert

3 2

[Serusers] NAT
by Daniel Mirrha Santos 06 Oct '06

06 Oct '06

How could I implement NAT support on ser? May I have to use a special module? What kinds of NAT is supported? Thanks, Daniel

5 7

[Users] I wont send a reply for ACK!!
by Jayesh Nambiar 06 Oct '06

06 Oct '06

Hi all, I have a doubt over this warning message appearing in my logs. When my Openser sends a 4xx message to any gateway, the gateway replies with an ACK, but then openser does not send an OK for that ACK and leaves this message in the logs. Can someone please explain why this message comes again and again. I have in my script, if(method=="ACK") { route(1); exit; }; after record routing and loose route handling. route(1) does the t_relay(). Please explain. w/regards, jayesh __________________________________________________________ Yahoo! India Answers: Share what you know. Learn something new http://in.answers.yahoo.com/

2 1

Re: [Users] Out of Memory error
by Bogdan-Andrei Iancu 06 Oct '06

06 Oct '06

Hi Bill, see http://openser.org/dokuwiki/doku.php?id=memory. regards, bogdan Bill Zhang wrote: > Hi, > > > > Anyone knows what could be wrong? This happens only when the client > using TCP sending registration. > > > > First, there are a LOT of following lines: > > > > Oct 2 16:26:52 ip67-153-142-69 /usr/local/sbin/openser[2463]: ERROR: > build_res_buf_from_sip_req: out of memory ; needs 698 > > Oct 2 16:26:52 ip67-153-142-69 /usr/local/sbin/openser[2463]: > challenge(): Error while sending response > > Oct 2 16:26:52 ip67-153-142-69 /usr/local/sbin/openser[2463]: ERROR: > build_res_buf_from_sip_req: out of memory ; needs 698 > > Oct 2 16:26:52 ip67-153-142-69 /usr/local/sbin/openser[2463]: > challenge(): Error while sending response > > > > Then, following: > > > > Oct 2 16:30:09 ip67-153-142-69 /usr/local/sbin/openser[2462]: ERROR: > build_res_buf_from_sip_req: out of memory ; needs 779 > > Oct 2 16:30:09 ip67-153-142-69 /usr/local/sbin/openser[2462]: > challenge(): Error while sending response > > Oct 2 16:30:09 ip67-153-142-69 /usr/local/sbin/openser[2462]: ERROR: > build_res_buf_from_sip_req: out of memory ; needs 779 > > Oct 2 16:30:09 ip67-153-142-69 /usr/local/sbin/openser[2462]: > challenge(): Error while sending response > > > > Then, > > Oct 2 16:49:01 ip67-153-142-69 /usr/local/sbin/openser[2462]: ERROR: > build_res_buf_from_sip_req: out of memory ; needs 780 > > Oct 2 16:49:01 ip67-153-142-69 /usr/local/sbin/openser[2462]: > challenge(): Error while sending response > > Oct 2 16:49:41 ip67-153-142-69 /usr/local/sbin/openser[2462]: ERROR: > build_res_buf_from_sip_req: out of memory ; needs 780 > > Oct 2 16:49:41 ip67-153-142-69 /usr/local/sbin/openser[2462]: > challenge(): Error while sending response > > > > And on and on. > > > > Using UDP register are fine. > > > > Thanks in advanceJ. > > Best Regards, > Bill > > > >------------------------------------------------------------------------ > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > >

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[Users] Out of Memory error
by Bill Zhang 06 Oct '06

06 Oct '06

Hi, Anyone knows what could be wrong? This happens only when the client using TCP sending registration. First, there are a LOT of following lines: Oct 2 16:26:52 ip67-153-142-69 /usr/local/sbin/openser[2463]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 698 Oct 2 16:26:52 ip67-153-142-69 /usr/local/sbin/openser[2463]: challenge(): Error while sending response Oct 2 16:26:52 ip67-153-142-69 /usr/local/sbin/openser[2463]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 698 Oct 2 16:26:52 ip67-153-142-69 /usr/local/sbin/openser[2463]: challenge(): Error while sending response Then, following: Oct 2 16:30:09 ip67-153-142-69 /usr/local/sbin/openser[2462]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 779 Oct 2 16:30:09 ip67-153-142-69 /usr/local/sbin/openser[2462]: challenge(): Error while sending response Oct 2 16:30:09 ip67-153-142-69 /usr/local/sbin/openser[2462]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 779 Oct 2 16:30:09 ip67-153-142-69 /usr/local/sbin/openser[2462]: challenge(): Error while sending response Then, Oct 2 16:49:01 ip67-153-142-69 /usr/local/sbin/openser[2462]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 780 Oct 2 16:49:01 ip67-153-142-69 /usr/local/sbin/openser[2462]: challenge(): Error while sending response Oct 2 16:49:41 ip67-153-142-69 /usr/local/sbin/openser[2462]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 780 Oct 2 16:49:41 ip67-153-142-69 /usr/local/sbin/openser[2462]: challenge(): Error while sending response And on and on. Using UDP register are fine. Thanks in advance:-). Best Regards, Bill

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[Users] NEW - Management Interface
by Bogdan-Andrei Iancu 05 Oct '06

05 Oct '06

Hi everybody, as shown by the roadmap, one of the objectives is the creation of a unique interface (management interface or MI) to replace the current FIFO and UNIXSOCK interfaces. Why? The current design does not scale, neither from core point of view, nor from modules point of view. The transport layers are currently implemented in core, so each new type of layer implies changes of the core. But the worst part is on the module side - as the implementations are transport specific, for each exported function, the module needs to implement a version for each available transport layer. So, adding a new transport layer will imply: (1) changes in core (2) reimplementation of all existent functions in modules for the new transport layer. What brings the new MI? MI will define in core an abstract layer for both functions and data types to be used between the external function implementations (from modules) and the transport layers (which now will be implemented as modules). When a module wants to export a function for external usage, it will just register it to the MI and it will use for implementation the data types defined by MI. So, the exported function will be automatically visible via all transport layer (existent or future). Also adding a new transport layer will imply only writing a new module which implements MI - no impact on the core or the module exporting functions. First version of the MI (Management interface) was uploaded on the public CVS couple of days ago. Also there is available the first module implementing it - the mi_fifo module implementing the FIFO interface. Sooner will follow a module for unixsockets and later NET UDP and TCP, HTTP, RPC, XML-RPC and so on.... Also we are in process of re-implementing the external function based on the MI specs. So far all core functions are done and cpl-c and dialog modules. The rest of the modules will be finished in the next week. Any comments, opinions on the topic are welcomed. Regards, Bogdan

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[Serusers] Australia DID incoming problem
by ram 05 Oct '06

05 Oct '06

Hi I have got Aus DID from provider I have configured in my Server Aus DID combination of below 1161XXXXXX635 161XXXXXX635 1XXXXXX635 XXXXXX635 DID As a user ID when i call 01161XXXXXX635 the call is coming to My SER, but the User is online, its not rininging on the Phone But dialed person getting Ringing sound. what format i need to use to create account DID I have simple route config till now iam able to config all USA DID but iam getting problem here with AUS some one give me idea or help Ram

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[Users] Multiple OpenSER DB Question
by Curt Moore 05 Oct '06

05 Oct '06

Hello all. I have a question with regard to sharing the location table between multiple OpenSER servers when using multiple OpenSER servers for load balancing and redundancy. I am using db_mode 1 with usrloc in OpenSER 1.1.0. When the usrloc module gets/sets information from the location table, does it pull all contact information or just the contact information where the socket field is an IP assigned to the running OpenSER process, ie. a local IP address? It appears that the preload_udomain() function pulls all location information and then makes determinations on whether it is for the current OpenSER process. I see where the dbrow2info() function in udomain.c makes the determination that the socket is non local but it seems that this is just an error message and the process continues as normal. Should there be a return just after this debug message so that the usrloc module does not continue and manipulate a location entry that does not belong to this particular OpenSER process? Should the information even be pulled from the DB on startup for records which do not have a socket belonging to the current OpenSER process? Thanks, -Curt

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[Users] openser TLS error problem
by Tracy Lofton 05 Oct '06

05 Oct '06

Hi, I noticed that the SSL dump you ran dispays RSA 1024, while, openSER-1.1.0-tls uses 2048 and was wondering if one was to change openser RSA to 1024 what type of effect would this have on the question at hand (UA trying to register...)? Noted below is the gen_rootCA.sh script with a RSA of 2048 for OpenSER: gen_rootCA.sh:( cd $CA_PATH; openssl req -config ../$CA_CONF -x509 -newkey rsa:2048 -days 365 -out ./cacert.pem -outform PEM ) I am experimenting with TLS and am very new to the TLS functionality/performance/security protocol and would like to know if the descrepancy in RSA 1024 and 2048 matters? I apologize that I could not offer assistance but just another question. Tracy ----------------------------------------------------- Message: 1 Date: Thu, 05 Oct 2006 17:21:21 +0800 From: Lindsey Hans Joseph Lao <llao(a)infoweapons.com> Subject: [Users] openser TLS error problem To: Users(a)openser.org Message-ID: <4524CE91.2050500(a)infoweapons.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hello, I have an Openser server with TLS support running ok. However, when a UA client tries to register with it, it gives off this error: 7(63031) tls_accept: Error in SSL: 7(63031) tls_error: error:1408F10B:SSL routines:SSL3_GET_RECORD:wrong version number I ran SSLDUMP and got these logs: 8 1 2.4157 (2.4157) C>S SSLv2 compatible client hello Version 3.1 cipher suites TLS_RSA_WITH_RC4_128_MD5 TLS_RSA_WITH_RC4_128_SHA TLS_RSA_WITH_3DES_EDE_CBC_SHA SSL2_CK_RC4 SSL2_CK_3DES SSL2_CK_RC2 TLS_RSA_WITH_DES_CBC_SHA SSL2_CK_DES TLS_RSA_EXPORT1024_WITH_RC4_56_SHA TLS_RSA_EXPORT1024_WITH_DES_CBC_SHA TLS_RSA_EXPORT_WITH_RC4_40_MD5 TLS_RSA_EXPORT_WITH_RC2_CBC_40_MD5 SSL2_CK_RC4_EXPORT40 SSL2_CK_RC2_EXPORT40 TLS_DHE_DSS_WITH_3DES_EDE_CBC_SHA TLS_DHE_DSS_WITH_DES_CBC_SHA TLS_DHE_DSS_EXPORT1024_WITH_DES_CBC_SHA 8 2.4163 (0.0005) S>C TCP FIN 8 2.4165 (0.0002) C>S TCP FIN Any idea of what might the problem be? Help would be greatly appreciated. Thanks! ---------------------------- _________________________________________________________________ SearchYour way, your world, right now! http://imagine-windowslive.com/minisites/searchlaunch/?locale=en-us&FORM=WL…

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[Users] Inquiry on Mediaproxy
by Lindsey Hans Joseph Lao 05 Oct '06

05 Oct '06

Hello, Just a quick question on Mediaproxy NAT traversal. When using mediaproxy on a NATed. router/firewall, would the registrar show a private contact header IP address or the public ip address of the User Agent client? I'm running openser configured to be multihomed listening to a private and public interface. However, it baffles me since the contact header registering in another Openser registrar is the private IP address of the user agent. Did mediaproxy function as its supposed to be? Thanks!

1 0

[Serusers] radius authentication on Invite with rpid header
by Rosario Pingaro 05 Oct '06

05 Oct '06

I have an ata that support rpid headr. So when I block the cid serivce I receive an anonymous from header with the rpid header that has the info about user and domain. How to have my radius authentication working on this situation? At the moment I am not able to authenticate because I get Domain=localhost instead of my domain. Reagrds Rosario Pingaro D. Lgs 196/2003 Il presente messaggio contiene informazioni confidenziali, indirizzate esclusivamente alle persone sopra indicate. Se il ricevente non è tra dette persone, non dovrà intraprendere alcuna azione, tipo copia, stampa o trasmettere il suo contenuto a terzi ed i relativi allegati, ma solo informare il mittente dell'errore e cancellare il messaggio. Il mittente dovrà, altresì, accertarsi che gli allegati non contengano virus prima di aprirli.

1 0

[Serusers] Dual number with SER
by reza shah 05 Oct '06

05 Oct '06

I want to create a service like this. User have two number given by 2 provider PA and PB called NA and NB. PA and PB can connect with each other. When user start his/her phone it will register first to PB with number NB, then PB will create register message (new message i think) with NA number to PA. PB is what i will develop. Is this possible with SER, i`m new to SER. Another question is how to create a module that i can call from SER? Thanks alot Reza __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 2

[Serusers] radius records for IM??
by technos e 05 Oct '06

05 Oct '06

Hi All, Can anyone tell me If I can generate Radius record for Instant Messaging using Iptel . Thanks in Advance. Techno --------------------------------- Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min.

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[Users] openser TLS error problem
by Lindsey Hans Joseph Lao 05 Oct '06

05 Oct '06

Hello, I have an Openser server with TLS support running ok. However, when a UA client tries to register with it, it gives off this error: 7(63031) tls_accept: Error in SSL: 7(63031) tls_error: error:1408F10B:SSL routines:SSL3_GET_RECORD:wrong version number I ran SSLDUMP and got these logs: 8 1 2.4157 (2.4157) C>S SSLv2 compatible client hello Version 3.1 cipher suites TLS_RSA_WITH_RC4_128_MD5 TLS_RSA_WITH_RC4_128_SHA TLS_RSA_WITH_3DES_EDE_CBC_SHA SSL2_CK_RC4 SSL2_CK_3DES SSL2_CK_RC2 TLS_RSA_WITH_DES_CBC_SHA SSL2_CK_DES TLS_RSA_EXPORT1024_WITH_RC4_56_SHA TLS_RSA_EXPORT1024_WITH_DES_CBC_SHA TLS_RSA_EXPORT_WITH_RC4_40_MD5 TLS_RSA_EXPORT_WITH_RC2_CBC_40_MD5 SSL2_CK_RC4_EXPORT40 SSL2_CK_RC2_EXPORT40 TLS_DHE_DSS_WITH_3DES_EDE_CBC_SHA TLS_DHE_DSS_WITH_DES_CBC_SHA TLS_DHE_DSS_EXPORT1024_WITH_DES_CBC_SHA 8 2.4163 (0.0005) S>C TCP FIN 8 2.4165 (0.0002) C>S TCP FIN Any idea of what might the problem be? Help would be greatly appreciated. Thanks!

2 1

[Serusers] Problem with TCP
by P Jayasree 05 Oct '06

05 Oct '06

Hi, I'm using ser proxy server the latest version 0.9.I'm running ser on ipv6 PC ,listening on ipv6 address whihc i configured in ser.cfg. Using some other clients which are running on ipv6 PC's can able to call from UDP-UDP properly.AS ser runs on both UDP and TCP ,the TCP-TCP calls also should go properly. But the error 'tcp_blocking_connect:(22) Invalid Argument' is coming when the next hop from the proxy is TCP. Kindly support for this problem. Regards, Jayasree

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[Users] OpenSER Summit @ VoN Europe
by Daniel-Constantin Mierla 05 Oct '06

05 Oct '06

Hello Everybody, we are pleased to announce that first OpenSER Summit will take place in Berlin, on the 8th November. The summit is hosted inside the VoN Europe Exhibition. The participants registered via OpenSER will have free access to summit and exhibition. A registration form will be available very soon on the web site, as an alternative you can send email to team(a)openser.org with your details. We are looking to network people around OpenSER, to share experiences and build business relations, among invited persons are executives from major communication companies. The board members and main developers will be there as well. There is no need for other words about VoN (http://www.von.com), is the leading conference in VoIP and we are glad that we can tie up together the OpenSER Summit. To be able to cover organizing expenses, we are looking for sponsorships. The sponsors will get the opportunity to held a presentation and do advertising inside the event. Banners will be published on the OpenSER web site and event's page. We will be able to offer to sponsors free passes to full conference (all presentations at VoN, lunches, dinners and parties) and other discounts. If you are interested to be a sponsor of the event, please contact us at team(a)openser.org and we will give you more details about the conditions and offering you will get. More details will be published on http://www.openser.org site as soon as updates come. Looking forward to meet you there, Daniel

1 0

[Serusers] Problem with TCP
by P Jayasree 05 Oct '06

05 Oct '06

Hi, I'm using ser proxy server the latest version 0.9.I'm running ser on ipv6 PC ,listening on ipv6 address whihc i configured in ser.cfg. Using some other clients which are running on ipv6 PC's can able to call from UDP-UDP properly.AS ser runs on both UDP and TCP ,the TCP-TCP calls also should go properly. But the error 'tcp_blocking_connect:(22) Invalid Argument' is coming when the next hop from the proxy is TCP. Kindly support for this problem. Regards, Jayasree The information contained in this e-mail message and/or attachments to it may contain confidential or privileged information. If you are not the intended recipient, any dissemination, use, review, distribution, printing or copying of the information contained in this e-mail message and/or attachments to it are strictly prohibited. If you have received this communication in error, please notify us by reply e-mail immediately and permanently delete the message and any attachments.

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Re: Re: [Users] tls. soket queues not flushed
by serega 05 Oct '06

05 Oct '06

Hello, I'm sorry for a big lag. I test system yesterday on other linux kernel (2.6) and obtain the same results. In this case i suppose, that this problem don't bases on system socket implementation. I tested openser on other numbers of tcp workers and obtain next results: 20 children: 2 sockets don't flushed (all had data on receive and send queues), cpu usage is above 48% on each of 2 openser processes, one of them invokes poll function in tls_server.c, other trying to write into tls socket. 1 child: 1 socket don't flushed (receive queue not empty, send queue empty), cpu usage is 90% on one process. 100 children: no problems, all work correctly 2 children: no problems. >Hello, > >On 09/29/06 16:06, serega wrote: >> Hi Klaus and all other who read this. >> >> It's not a problem of tcp socket - yesterday i rewrite gateway config for fowarding directed to extgateway requests to gateway itself. >> In tcp this problems not occured, but after i switched on tls, i see overfilled queues again. In that case i haven't extgateway, communications with >> remote sockets and etc. Did anybody test openser whith big payload? I think this problem occurs because in my case i have big traffic and several process that simultaniously use tls. Did anybody have the same problem? >> > >If you increase the number of the TLS/TCP workers do you get same >results? In case of TLS, there is more CPU used because of >encryption/decryption. How is the CPU usage in teh system while you are >testing? > >As Klaus said, the management of socket queue is done by OS. > >Cheers, >Daniel > >> >>> Hi Serega! >>> >>> I'm not sure - just a guess. openser uses "worker threads". This worker >>> threads get a SIP message from the listener thread, process them, and >>> then they send them. >>> >>> If for some reason the sending fails - e.g. the gateway sends via >>> TCP/TLS to extgateway and extgateway's receiving queue is full. Then >>> gateway's sending buffer gets full and the threads wait until they can >>> send. Thus, as the threads are all busy, also the receiving queue of >>> gateway gets full. I guess after some time the threads give up sending >>> and will read again. >>> >>> The question is: why does the sending buffer gets full? Where is gateway >>> sending the messages to? >>> >>> The sending queue must not get flushed. The queue is in the OS, not in >>> openser. As TCP guarantees that there is no loss/reordering the TCP >>> stack must not flush the queue. >>> >>> regards >>> klaus >>> >>> serega wrote: >>> >>>> Hi all. >>>> >>>> I have next problem while using openser with tls. System consists of >>>> sip statefull server (in next time, simply, server) and stateless sip >>>> gateway (gateway). Server used for connect to jabber server. Server >>>> contains rewrote jabber module logic and also use presence module. >>>> Gateway used to connect to other sip gateway (extgateway) via tls >>>> protocol. Gateway use 1.1 openser version and doesn’t contain changed >>>> code. When server together with gateway restarts and server in its >>>> database contains above 260 subscriptions (in watcherinfo table) i >>>> have error. Socket used by gateway to connect with extgateway >>>> contains in receive and sent queues a lot of data (above 50kByte on >>>> each sides). This data never flushed out. This happens because after >>>> restart sip server through gateway send notifications to extgateway. >>>> When I attached using gdb to process that send data, I saw that it >>>> was in infinity loop because tls library returns SSL_ERROR_WANT_WRITE >>>> and I think it’s correct because we have overfilled send queue. In >>>> this case I don’t interest why receive queue not empty (I think it >>>> happens because sending process have got block on socket). But I >>>> don’t understand why sent queue not flushed. I test this behavior >>>> using tcp – all was correct. Socket not closed by other side because >>>> this status can be unchanged above one day. Anybody can help me? Why >>>> this happens? >>>> >>>> I use openssl-0.9.8c and redhat os (Linux xdevel1 2.4.21-4.ELsmp #1 >>>> SMP Fri Oct 3 17:52:56 EDT 2003 i686 i686 i386 GNU/Linux). Computer >>>> have too Pentium 4 processors). >>>> >>>> >>>> Gateway.cfg (sip.qa.hbex.com – gateway address): >>>> >>>> # # $Id: router_qa.cfg,v 1.1 2006/08/02 18:14:34 ilya Exp $ # # >>>> simple quick-start config script # >>>> >>>> # ----------- global configuration parameters >>>> ------------------------ >>>> >>>> debug=9 # debug level (cmd line: -dddddddddd) >>>> log_facility=LOG_LOCAL0 fork=yes log_stderror=yes # (cmd line: -E) >>>> >>>> >>>> check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) >>>> rev_dns=no # (cmd. line: -R) children=20 >>>> fifo="/tmp/openser_fifo" server_signature=no >>>> >>>> fifo_db_url="mysql://root:@localhost/openser" >>>> >>>> listen = udp:sip.qa.hbex.com:5060 #incoming sip server address listen >>>> = tcp:sip.qa.hbex.com:5060 >>>> >>>> #tls address disable_tls = 0 listen = tls:sip.qa.hbex.com:5061 listen >>>> = udp:sip.qa.hbex.com:5061 >>>> >>>> tls_certificate = "/ home/inop/ex.com-cert.pem" tls_private_key = >>>> "/home/inop/ex.com-privkey.pem" tls_ca_list = >>>> "/home/inop/ex.com-calist.pem" tls_require_client_certificate=0 >>>> tls_verify_client=0 tls_verify_server=0 >>>> >>>> # ------------------ module loading >>>> ---------------------------------- loadmodule >>>> "/home/interop/openser/lib/openser/modules/rr.so" loadmodule >>>> "/home/interop/openser/lib/openser/modules/xlog.so" loadmodule >>>> "/home/interop/openser/lib/openser/modules/textops.so" loadmodule >>>> "/home/interop/openser/lib/openser/modules/maxfwd.so" loadmodule >>>> "/home/interop/openser/lib/openser/modules/sl.so" loadmodule >>>> "/home/interop/openser/lib/openser/modules/mysql.so" loadmodule >>>> "/home/interop/openser/lib/openser/modules/tm_unchanged.so" >>>> loadmodule "/home/interop/openser/lib/openser/modules/usrloc.so" >>>> loadmodule "/home/interop/openser/lib/openser/modules/registrar.so" >>>> >>>> # ----------------- setting module-specific parameters >>>> --------------- modparam("registrar", "default_expires", 120) >>>> modparam("registrar", "use_domain", 1) >>>> >>>> modparam("usrloc", "use_domain", 1) 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) >>>> >>>> # add value to ;lr param to make some broken UAs happy modparam("rr", >>>> "enable_full_lr", 1) modparam("maxfwd", "max_limit", 10) # >>>> ------------------------- request routing logic ------------------- >>>> # main routing logic route{ if (!mf_process_maxfwd_header("10")) { >>>> sl_send_reply("483","To Many Hops"); drop(); }; if (dst_port==5061) { >>>> if (search("^To:.*@.*cmnicator.com") || search("^To:.*@.*cihb.com")) >>>> { # rewrite destination and forward to jabber (sip server) route(1); >>>> return; }; sl_reply_error(); return; } if (method=="REGISTER") { >>>> xlog("XXX: saving location msg=$mb\n"); if(!save_noreply("location")) >>>> { log("XXX: Error saving location!\n"); sl_reply_error(); } >>>> sl_send_reply("200","OK"); return; } if >>>> (search("^To:.*@.*cmnicator.com") || search("^To:.*@.*cihb.com")) { >>>> #forward to self route(3); return; } ##forward to ext gateway >>>> route(2); } >>>> >>>> route[1]{ if (method!="MESSAGE" && method!="SUBSCRIBE" && >>>> method!="NOTIFY") { log("XXX: Request not forwarded to sip >>>> server!\n"); sl_send_reply("202","Accepted"); return; } >>>> t_on_reply("1"); record_route(); >>>> >>>> if(!lookup("location")) { if (method!="SUBSCRIBE") { log("XXX: only >>>> subscribes are processed for user that are not registered\n"); >>>> sl_reply_error(); return; } >>>> subst_uri('/(.*)@(.*)/\1#\2*jabber.qa.im.hb.com@jabber01.qa.im.hb.com/ig'); >>>> if(!t_relay()) { log("XXX: error forwarding jabber01...\n"); >>>> sl_reply_error(); return; } else { sl_send_reply("200","OK"); return; >>>> } } else { log("XXX: forwarding to the address of record...\n"); >>>> if(!t_relay()) { log("XXX: error forwarding to address of record >>>> \n"); sl_reply_error(); return; } else { sl_send_reply("200", "OK"); >>>> return; } } >>>> >>>> } >>>> >>>> # forwarding to external gateway route[2]{ log("XXX: rewriting >>>> headers\n"); >>>> subst('/^(From:[^@#]*)#([^@]*)\*jabber.qa.im.hb.com@jabber...qa.im.hb.com(.*)/\1@\2\3/ig'); >>>> >>>> subst('/^(Contact:[^@#]*)#([^@]*)\*jabber.qa.im.hb.com@jabber...qa.im.hb.com(.*)/\1@\2\3/ig'); >>>> >>>> >>>> if(!forward("tls:43.123.141.166:3000")) { log("XXX: Error forwarding >>>> to external gateway!\n"); sl_reply_error(); } else { >>>> sl_send_reply("200", "Accepted"); }; } >>>> >>>> # forwarding to SELF route[3]{ log("XXX: rewriting headers\n"); >>>> subst('/^(From:[^@#]*)#([^@]*)\*jabber.qa.im.hb.com@jabber...qa.im.hb.com(.*)/\1@\2\3/ig'); >>>> >>>> subst('/^(Contact:[^@#]*)#([^@]*)\*jabber.qa.im.hb.com@jabber...qa.im.hb.com(.*)/\1@\2\3/ig'); >>>> >>>> >>>> log("XXX: about to forward to self\n"); >>>> if(!forward("tcp:sip.qa.hbex.com:5061")) { log("XXX: Error forwarding >>>> to self!\n"); sl_reply_error(); } else { sl_send_reply("200", >>>> "Accepted"); }; } >>>> >>>> # this is executed for replies onreply_route[1]{ >>>> subst('/^(To:[^@#]*)#([^@]*)\*jabber.qa.im.hb.com@jabber...qa.im.hb.com(.*)/\1@\2\3/ig'); >>>> >>>> subst('/^(Contact:[^@#]*)#([^@]*)\*jabber.qa.im.hb.com@jabber...qa.im.hb.com(.*)/\1@\2\3/ig'); >>>> route(2); } >>>> >>>> >>>> >>>> _______________________________________________ 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] OpenserAdmin: openser log
by Ronald Wiplinger 05 Oct '06

05 Oct '06

In the log file (OpenserAdmin) I see my registered phones with different addresses. I would like to know what it means: a. public IP with a port number sip:7111@59.124.230.151:65129 b. private IP with a port number sip:7112@192.168.10.75:5060 c. private IP with NO port number sip:7113@192.168.10.45 d. Eyebeam with public IP with port and rinstance (???) sip:7311@203.70.36.126:65321;rinstance=059afa5458d206a7 e. Gizmo phone, secondary number should be 7312. sip:17476063396@proxy01.sipphone.com:5060 d. and e. are on my desktop connected to a ADSL router, which obviously makes the NAT to 203.70.36.126, my current public IP on that line. b and c are also on a private network and connected through the same ADSL router as a. Why a has a public IP, while b and c not? Why have all a port, but c not ??? bye Ronald

1 0

[Users] How to read log files?
by Ronald Wiplinger 04 Oct '06

04 Oct '06

I got two places to look: /var/log/openser and the output of ngrep. I would like to see only one case of one phone / phone call. Is there a tool to filter that out? Has anybody made a transfer from the ngrep to a mysql database? I expect that this would be easier to select single cases. Where are and how must I set-up openser to get CDRs ? bye Ronald Wiplinger

1 0

[Users] uac_redirect
by Andreas Granig 04 Oct '06

04 Oct '06

Hi, Two questions about the uac_redirect module: 1. If I extract contacts from a 302 with two contacts having the same q-value by using get_redirects("1:1"), which one of the contacts is selected? The first one? The last one? 2. If I apply a filter rule like: modparam("uac_redirect", "default_filter", "deny") modparam("uac_redirect", "accept_filter", "sip:.+@foo.bar.com") and I get a 302 with a Contact, say, "sip:foo@some.other.tld", then the module logs this error: "ERROR:uac_redirect:get_redirects: get contact from shm_reply branch 0 failed". I'd say if zero contacts are extracted from branches due to filters, it's not really an error. Any suggestions for a better error handling? Andy

2 2

[Users] check_route_param and rr_preset
by Helge Waastad 04 Oct '06

04 Oct '06

Hi, I was just wondering if the check_route_param("") works on record_route_preset Routes? It never hits when my Route: are, f.ex like xxx.xxx.xxx.xxx:<port>;nat=yes Is this true? br hw -- Helge Waastad Senior Engineer Systemavdelingen Smartnet

3 12

[Serusers] re: How to change the To/From headers?
by lawrence k.y. lin 04 Oct '06

04 Oct '06

Hello, I looked at the FAQ in iptel.log and did a search on it for "how to change the To/From headers?" but did not find the answer. There is an item in FAQ for "Why not change the To/From headers?" in the following url "http://www.iptel.org/FAQ_To_From_change", but did mention how to change the To/From headers. Are there variables related to To/From headers that we can change ? Please advise, thanks, Larry _________________________________________________________________ Be seen and heard with Windows Live Messenger and Microsoft LifeCams http://clk.atdmt.com/MSN/go/msnnkwme0020000001msn/direct/01/?href=http://ww…

1 0

[Serusers] blocking CID continuing using radius authentication
by Rosario Pingaro 04 Oct '06

04 Oct '06

I would like to use the ATAs features to block the CID for anonymous calls. At the same time I wouldn't like to break my accounting done by Radius Authentication. This is the ser.cfg about radius auth for international calls: if (!radius_is_user_in("From","int")) { sl_send_reply("403","Internazionali non abilitate"); break; I use to authenticate the "From" field so if I disable the CID on the ata I will receive an anonymous from field. What do I have to authenticate to mantain a working accounting? Do I have to authenticate against "Credentials"? What is the difference between Credentials and from headers? Regards Rosario

1 2

[Users] Group creation with openserctl
by Antoine Fressancourt 04 Oct '06

04 Oct '06

Hello, I am somehow trying to add a user to a group, with the openserctl command : openserctl acl grant 5220 mitel I get a response : WARINIG: Invalid privilege: acl 'mitel' ignored I tryed to browse the web about that issue and about group creation without success, so I would be really pleased to hear from your advices. By the way, "bruteforcing" the addition of the user in the group table is working. Thank you, Antoine

2 1

[Users] Mediaproxy
by Lucio Maciel 04 Oct '06

04 Oct '06

Hi How many simultaneous calls mediaproxy.py can handle with a good server (Dell poweredge 6850, dual XEON, lots of memory and bandwitch) ? Lúcio Fernando Maciel Telecom South America R. Ramiro D'Ávilla, 20 sala 306 Porto Alegre - RS - Brasil lucio(a)tesatelecom.com +55 (51) 3061-5569 +55 (51) 8442-1079

2 1

Re: [Users] ERROR 1062 (23000) at line 1: Duplicate entry '' for key 'phplib_id'
by Bogdan-Andrei Iancu 04 Oct '06

04 Oct '06

maybe we should add for the future a detection of the md5 utility to avoid this kind of problems regards, bogdan Lindsey Hans Joseph Lao wrote: > it worked! :-) thanks a lot! > Bogdan-Andrei Iancu wrote: > >> you can run "MD5=md5 openserctl ...." or for a permanent change see >> in openserctl.base file the MD5 definition. >> >> regards, >> bogdan >> >> Lindsey Hans Joseph Lao wrote: >> >>> i have a utility called md5 installed in my freeBSD system. Can you >>> show me the way on how to integrate it with openserctl? >>> >>> Thanks! >>> >>> Bogdan-Andrei Iancu wrote: >>> >>>> from 1.0.x to 1.1.x there was a change in how the phplib_id is >>>> computed..now is done via md5sum to avoid any shell compatibility >>>> problems (as before). >>>> >>>> check if you have the md5 utility on your system (typically is >>>> md5sum) or check if it has a different name than "md5sum" >>>> >>>> regards, >>>> bogdan >>>> >>>> >>>> Lindsey Hans Joseph Lao wrote: >>>> >>>>> Indeed, it has an empty value in the table... i am unable to add >>>>> subscribers because the error shown below occurs. this happened >>>>> when i switched from openser1.0.1 no tls to openser 1.1.0-tls. >>>>> Here's what happens when I try to add a user to the database: >>>>> >>>>> md5sum: not found >>>>> md5sum: not found >>>>> md5sum: not found >>>>> >>>>> then it prompts for my mysql password. I don't think I have >>>>> installed those to binaries. However, when using openser1.0.1. I >>>>> couldn't encountering this error... >>>>> Probably a bug? >>>>> >>>>> Thanks! >>>>> >>>>> Bogdan-Andrei Iancu wrote: >>>>> >>>>>> ok- it must be the phplib_id column - can you check its value in >>>>>> the table? is must not be NULL or empty... >>>>>> >>>>>> I suspect your openserctl fails to compute the phplib_id and you >>>>>> get a constant empty string for all subscribers.. do you get any >>>>>> errors? do you have "md5sum" and "awk" installed on your machine? >>>>>> >>>>>> regards, >>>>>> bogdan >>>>>> >>>> >>> >> >

1 0

[Users] ERROR 1062 (23000) at line 1: Duplicate entry '' for key 'phplib_id'
by Lindsey Hans Joseph Lao 04 Oct '06

04 Oct '06

Hello, I happen to encounter this error in Openser 1.1.0 with TLS: ERROR 1062 (23000) at line 1: Duplicate entry '' for key 'phplib_id' \E[37;31mERROR: introducing the new user '1050' to the database failed This happens when I try to add users to the mysql database using openserctl add username password email-address. Are there any workarounds for this? or is this a bug in the recent version? Thanks!

2 7

Re: [Serusers]More than one domain
by Ricardo Carvalho 04 Oct '06

04 Oct '06

Hi Kamal, modparam("usrloc", "timer_interval", 30) #Number of seconds between two timer runs. The module uses timer to delete expired contacts, synchronize with database and other tasks, that need to be run periodically. In case of crash or restart contacts that are in memory only and haven't been flushed yet will get lost. If you want minimize the risk, use shorter timer interval. Default value is 60. That description can be seen in the README file located at ser_source_dir/modules/usrloc/. Regards, Ricardo. Kamal.Mann(a)t-systems.com wrote: > Hi Ricardo > It's working now. Thanks a lot. Will you please specify the meaning of > 'modparam("usrloc", "timer_interval", 30)'. > > Thanks in anticipation > Kamal Mann > -----Original Message----- > From: serusers-bounces(a)lists.iptel.org > [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Ricardo Carvalho > Sent: Wednesday, October 04, 2006 2:42 PM > To: serusers(a)lists.iptel.org > Subject: Re: [Serusers]More than one domain > > You need to edit ser.cfg with the following parameters: > > modparam("usrloc", "use_domain", 1) > modparam("registrar", "use_domain", 1) > modparam("usrloc", "timer_interval", 30) > > and you have to live blank the domain in the www_authorize and > www_challenge parameters so that it will work for every domain. In the > domain table you have to add every domain served too. Also add the > alias=every_domain_you_want_ser_to_listen_to in ser.cfg file. > > > Regards, > Ricardo. > > > > > > > > > Kamal.Mann(a)t-systems.com wrote: > >> Hi all >> >> I need to know the changes in configuration of ser if I want that a >> single server can serve more than one domain. >> >> Right now I am changing in following lines but it server only one >> domain at a time: >> >> >> >> /if (method=="REGISTER") {/ >> >> / / >> >> / if (!www_authorize("ims.de", "subscriber")) / >> >> / {/ >> >> / www_challenge("ims.de", "0");/ >> >> / break;/ >> >> / };/ >> >> / / >> >> / save("location");/ >> >> / break;/ >> >> / };/ >> >> >> >> Thanks in anticipation >> >> Kamal Mann >> >> >> > ------------------------------------------------------------------------ > >> _______________________________________________ >> 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 > >

1 0

Re: [Users] rtpproxy has been disabled temorarily
by Bogdan-Andrei Iancu 04 Oct '06

04 Oct '06

Ronald, you need to set also the "-l" parameter (listening address). regards, bogdan Ronald Wiplinger wrote: > taipei:~ # rtpproxy -s udp:xxx.xxx.xxx.xxx:22222 > rtpproxy: explicit binding address has to be specified in UDP command > mode > > ??? but it is anyway defined: > > taipei:~ # grep 2222 /home/ronald/downloads/rtpproxy/rtpp_defines.h > #define CPORT "22222" > > What do I still miss? >

1 0

[Serusers] How to keep track of currently active calls
by Ricardo Carvalho 04 Oct '06

04 Oct '06

Hi, As much as I know, active_sessions table exists to keep track of currently active web sessions, for use by serweb. Although, I don't have Serweb installed because I'm building my own web interface so I can't see those active sections looking at that table. Anyway, when I "select * from active_sessions", it doesn't retrieve nothing... I guess even with Serweb it wouldn't show anything because this table is always empty!! What may I be doing wrong? How may I enable active calls to be shown in that table? Regards, Ricardo.

1 0

[Users] SSL: connect failed ..Please help
by Ferianto siregar 04 Oct '06

04 Oct '06

Dear all, Thank you very much for you time . All, I am sorry. May I ask you again, please..? I got error message in minisip command prompt when I tried using TLS (Transport Method = TLS and Ntwork Port = 5061).But, without TLS , I can make a call with minisip. (User`s registration success) The error message says : A. in client command prompt init 6/9: Creating MSip SIP stack init 7/9: Connecting GUI to SIP logic init 8.2/9: Starting TCP transport worker thread init 8.3/9: Starting TLS transport worker thread init 9/9: Registering Identities to registrar server Registering user didit(a)202.95.149.251 to proxy 202.95.149.251, requesting domain 202.95.149.251 SipMessageTransport: sendMessage: creating new socket Creating new SSL_CTX SSL: connect failed SipMessageTransport: sendMessage: exception thrown! SipMessageTransport: sendMessage: creating new socket SSL: connect failed SipMessageTransport: sendMessage: exception thrown! SipMessageTransport: sendMessage: creating new socket SSL: connect failed SipMessageTransport: sendMessage: exception thrown! SipMessageTransport: sendMessage: creating new socket SSL: connect failed SipMessageTransport: sendMessage: exception thrown! B. in server terminal tls_tcpconn_init: Setting in ACCEPT mode (server) 11(5927) tcpconn_add: hashes: 835, 11 11(5927) handle_new_connect: new connection: 0x422d88f0 24 flags: 0002 11(5927) send2child: to tcp child 0 7(5919), 0x422d88f0 7(5919) received n=4 con=0x422d88f0, fd=20 7(5919) DBG: io_watch_add(0x80ed320, 20, 2, 0x422d88f0), fd_no=1 7(5919) tls_update_fd: New fd is 20 7(5919) tls_accept: Error in SSL: 7(5919) tls_error: error:1408F10B:SSL routines:SSL3_GET_RECORD:wrong version number 7(5919) DBG: io_watch_del (0x80ed320, 20, 1, 0x10) fd_no=2 called 7(5919) releasing con 0x422d88f0, state -2, fd=20, id=11 7(5919) extra_data 0x422e8a08 11(5927) handle_tcp_child: reader response= 422d88f0, -2 from 0 11(5927) tcpconn_destroy: destroying connection 0x422d88f0, flags 0002 11(5927) tls_close: Closing SSL connection 11(5927) tls_update_fd: New fd is 24 11(5927) tls_shutdown: Shutdown successful 11(5927) tls_tcpconn_clean: Entered What`s wrong? How to solve the error "SSL3_GET_RECORD:wrong version number" and "SSL: connect failed"? Please forgive me for my lack of skill and many questions Please tell me if I have shown my openser.cfg file. Thanks Thank you for your attention and Have a nice day :) Regards, Ferianto Note: 1. I use Redhat 9 [root@localhost root]# rpm -qa|grep -i ssl perl-Crypt-SSLeay-0.45-7 openssl-devel-0.9.7a-2 openssl-perl-0.9.7a-2 openssl-0.9.7a-2 pyOpenSSL-0.5.1-8 mod_ssl-2.0.40-21 openssl096b-0.9.6b-3 docbook-style-dsssl-1.76-8 openssl096-0.9.6-15 [root@localhost root]# --------------------------------- How low will we go? Check out Yahoo! Messengers low PC-to-Phone call rates. --------------------------------- Do you Yahoo!? Get on board. You're invited to try the new Yahoo! Mail.

2 1

RE: [Serusers] Presence
by senthil.nachimuthu＠bt.com 04 Oct '06

04 Oct '06

Hi Javier, Thanks for your information. I need to make SEP to send/proxy the message to my presence server. SEP and presence server are physically different entities. Would be of greater help if you could let me know how can i specify the IP address of my presence server in SEP so that the presence specific messages reaching SEP gets forwarded to my presence server. Thanks and Regards, Senthil Kumar, ________________________________ From: Javier Ramirez [mailto:jvr_78@yahoo.com.ar] Sent: 25 September 2006 15:03 To: Nachimuthu,SK,Senthil,XSD4C C Subject: Re: [Serusers] Presence if (method=="SUBSCRIBE") { if (!t_newtran()) { sl_reply_error(); }; handle_subscription("registrar"); }; if (method=="PUBLISH") { if (!t_newtran()) { log(1, "newtran error\n"); sl_reply_error(); }; handle_publish("registrar"); break; }; ----- Original Message ----- From: senthil.nachimuthu(a)bt.com To: serusers(a)lists.iptel.org Sent: Monday, September 25, 2006 8:28 AM Subject: [Serusers] Presence Hi, I want to configure SER <file://ser/about> 0.9.6 in such a way that It can proxy presence messages(SUBSCRIBE,PUBLISH) to presence server. Can you let me know how can I do that? Thanks and Regards, Senthil Kumar, Presence Capability, BT-Service Execution Platform, B62-MH,Orion Building,Adastral Park,Ipswich Work:+44 (0)1473 605 668 Mobile:+44 (0)7772287447 __________ NOD32 1.1768 (20060922) Information __________ This message was checked by NOD32 antivirus system. http://www.eset.com ________________________________ _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers __________ NOD32 1.1768 (20060922) Information __________ This message was checked by NOD32 antivirus system. http://www.eset.com

4 3

[Users] rtpproxy has been disabled temorarily
by Ronald Wiplinger 04 Oct '06

04 Oct '06

If I use # -- nathelper -- modparam("nathelper", "rtpproxy_sock", "udp:59.124.210.41:22222") # default port 22222 #modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_from", "sip:59.124.210.41") # ----------------------------------------------------------------- # NAT detection # ----------------------------------------------------------------- route(2); route[1] { if (subst_uri('/(sip:.*);nat=yes/\1/')){ setflag(6); }; if (isflagset(5)||isflagset(6)) { route(3); } # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } route[2]{ force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); }; setflag(5); }; } route[3] { if (is_method("BYE|CANCEL")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); t_on_failure("1"); }; if (isflagset(5)) search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); t_on_reply("1"); } failure_route[1] { if (isflagset(6) || isflagset(5)) { unforce_rtp_proxy(); } } I get in the log file: Oct 4 09:19:01 taipei Taipei-openser[4083]: WARNING: rtpp_test: support for RTP proxy <udp:59.124.210.41:22222>has been disabled temporarily Oct 4 09:19:01 taipei Taipei-openser[4085]: WARNING: rtpp_test: support for RTP proxy <udp:59.124.210.41:22222>has been disabled temporarily Oct 4 09:19:01 taipei Taipei-openser[4087]: WARNING: rtpp_test: support for RTP proxy <udp:59.124.210.41:22222>has been disabled temporarily Oct 4 09:19:01 taipei Taipei-openser[4089]: WARNING: rtpp_test: support for RTP proxy <udp:59.124.210.41:22222>has been disabled temporarily Oct 4 09:19:01 taipei Taipei-openser[4091]: WARNING: rtpp_test: support for RTP proxy <udp:59.124.210.41:22222>has been disabled temporarily Oct 4 09:19:01 taipei Taipei-openser[4050]: WARNING: rtpp_test: support for RTP proxy <udp:59.124.210.41:22222>has been disabled temporarily Oct 4 09:19:01 taipei Taipei-openser[4067]: WARNING: rtpp_test: support for RTP proxy <udp:59.124.210.41:22222>has been disabled temporarily Oct 4 09:19:01 taipei Taipei-openser[4069]: WARNING: rtpp_test: support for RTP proxy <udp:59.124.210.41:22222>has been disabled temporarily with: modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") this error does not appear. In neither way it work for me (ring but no audio) Can anybody help me please? bye Ronald Wiplinger

2 1

[Users] NAT: STUN, nathelper, rtpproxy, mediaproxy
by Ronald Wiplinger 04 Oct '06

04 Oct '06

We have STUN, nathelper with rtpproxy or mediaproxy, ... maybe even more. Which one is good for which situation? What are the differences? How can I cover each situation? Can I combine different methods? bye Ronald

2 1

[Users] SER with multiple asterisks
by Adi Simon 04 Oct '06

04 Oct '06

Hi, I have one SER and two asterisk boxes. I'm trying to duplicate each register message from the client so it will reach each of the asterisks. I tried the following in my openser.cfg: if ( (is_method("SUBSCRIBE")) || (is_method("REGISTER")) ) { seturi("sip:nobody@10.0.0.1:5060"); append_branch("sip:parallel@10.0.0.2:5060"); if (!t_relay()) { sl_reply_error(); }; exit; } The first asterisk server registers the UA successfully. The second one gives this error: Oct 3 12:55:48 NOTICE[32364]: chan_sip.c:6395 check_auth: stale nonce received from 'sip:1234566@foo.bar' Transmitting (no NAT) to asterisk2:5060: I googled a bit and I understand I cannot fully duplicate a sip message since the nonce cannot be identical. Is there a way to recreate the sip message with another nonce? Is there any other solution to this scenario? Thanks, Adi.

1 1

[Serusers] SRV usage with SER
by Paul Antinori (pantinor) 04 Oct '06

04 Oct '06

Does SER support the SRV mechanism, sending a request to a URL with a SRV domain name? Also, if the first priority destination fails due to a timeout, will it failover to the second priority? Thanks for help, Paul

2 1

[Users] different between "silo" and "voice_silo" table
by cv aik 04 Oct '06

04 Oct '06

Hi, Thanx for the answer last time... I have add configuration of MSILO module in openser.cfg. The problem is I don't know how to test it (how to leave offline voice message between clients). If there is any application I have to use to save voice message, please let me know. And about the silo table, I've checked OPENSER database and I 've found that silo table is already there without I've to create it. But, when I configure my serweb, I found in my /var/www/html/serweb/config.php, there is another table named "voice_silo". There is no table named "voice_silo" in my OPENSER database. I'd like to know what's the different between "silo" and "voice_silo" (maybe in storing message (voice or text)). Thanx Regards, Aldi --------------------------------- Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail.

2 1

[Serusers] authenticating forwarded calls
by Mark Price 04 Oct '06

04 Oct '06

Is it possible to set up SER with asterisk so that any INVITE that is forwarded from SER to asterisk does not need to be authenticated by asterisk? Thanks, Mark

2 1

[Users] Who want to take this challenge (20US$)
by Ronald Wiplinger 04 Oct '06

04 Oct '06

Hi, I want to get a success feeling, ... and with your help I would get my openser installation running with: a. mysql (done) b. nat traversal (not working yet) c. ENUM (not working yet) d. sipbroker (not started yet) e.g., *777xxxxx should go to Sipgate.de .... If you want to help me to set it up, please send me first an email and I am happy to reserve time for your set-up try. If you succeed I pay you US$ 20 to your PayPal account !!!! Environment: 1. Eyebeam is on my internal network 2. several video phones are available outside of my network (one on a public IP: Grandstream 3000 video phone) and others behind a nat 3. several ENUM numbers are available, which are going to my asterisk server (different machine, different domain name) 4. sipbroker can also reach our asterisk server 71xx IP range 192.168.10.x -------------- NAT 7111 (public IP) | | ---------------------------------------INTERNET -------------------------------------------- | | | extern: 203.70.36.126 | | second ADSL on 192.168.250.250 | public ip address | taipei.elmit.net (Openser server) other servers with 2 Ethernet | | IP range 192.168.250.x | | | | | --------------------------------------------------------------------------------------------- | | | | phones and desk top computers bye Ronald Wiplinger

1 0

[Users] ENUM dialing
by Ronald Wiplinger 03 Oct '06

03 Oct '06

I try to call an ENUM number. I use this settings for it: mpath="/usr/local/lib64/openser/modules/" loadmodule "enum.so" # -- enum params -- modparam("enum", "domain_suffix", "e164.arpa.") route{ # ----------------------------------------------------------------- # initial sanity checks -- messages with # max_forwards==0, or excessively long requests # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 4096 ) { sl_send_reply("513", "Message too big"); exit; }; # ----------------------------------------------------------------- # NAT detection # ----------------------------------------------------------------- log("I am checking for NAT now\n"); route(2); # ----------------------------------------------------------------- # if you want to make ENUM work with numbers starting with "00", # use the following to convert "00" it into a "+" # ----------------------------------------------------------------- log("I am checking for ENUM now\n"); if (uri=~"^sip:00[1-9][0-9]*@") { # tried before: taipei\.elmit\.net") { # strip leading "00" # (change example.net to your domainname or skip the stuff after the "@") strip(2); # (adjust, if your international prefix is something else than "00") prefix("+"); }; # check if request uri starts with an international phone # number (+X.), if yes, try to ENUM resolve in e164.arpa. # if no result, try in nrenum.net if (uri=~"sip:\+[0-9]+@") { # tried before: taipei\.elmit\.net") { # (change example.net to your domainname or skip the stuff after the "@") ####if ( !enum_query("e164.arpa.") ) { #### enum_query("nrenum.net."); #### }; if (enum_query()) { t_relay(); exit; } }; I tried to add some log statements, but I never see them in /var/log/messages either. Below is the ngrep output. It seems that it does not go anywhere in route() Any ideas how to track it down? bye Ronald Content-Type: application/sdp. User-Agent: eyeBeam release 1005g stamp 32576. Content-Length: 809. . v=0. o=- 2 2 IN IP4 192.168.250.108. s=CounterPath eyeBeam 1.5. c=IN IP4 192.168.250.108. t=0 0. m=audio 48702 RTP/AVP 107 100 106 6 0 105 18 3 5 101. a=alt:1 2 : O5/HQNzE iLXVkGqH 5.32.86.166 48702. a=alt:2 1 : ok7vH1EF jO6dCAlm 192.168.250.108 48702. a=fmtp:18 annexb=yes. a=fmtp:101 0-15. a=rtpmap:107 BV32/16000. a=rtpmap:100 SPEEX/16000. a=rtpmap:106 SPEEX-FEC/16000. a=rtpmap:105 SPEEX-FEC/8000. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. a=sendrecv. a=x-rtp-session-id:E4797B0FC1A140648811BD63523C3B70. m=video 40488 RTP/AVP 125. a=alt:1 2 : urVBRqXw O9lI2kQp 5.32.86.166 40488. a=alt:2 1 : LCXVKz4X nxDzxq4V 192.168.250.108 40488. a=fmtp:125 profile-level-id=42e015; max-br=4000; max-mbps=19800. a=rtpmap:125 H264/90000. a=sendrecv. a=x-rtp-session-id:B8E2EE0EA3944D6DA47E5D98AABCD52B. # U 2006/09/28 16:57:11.567171 192.168.250.41:5060 -> 192.168.250.108:40430 SIP/2.0 404 Not Found. Via: SIP/2.0/UDP 192.168.250.108:40430;branch=z9hG4bK-d87543-0c396723de34cf5a-1--d87543-;rport=40430. To: "00886228357765"<sip:00886228357765@192.168.250.41>;tag=329cfeaa6ded039da25ff8cbb8668bd2.e6c9. From: "Ronald-taipei @ video "<sip:7311@192.168.250.41>;tag=3f7a5072. Call-ID: YmQ0YzNiYmU4MDJjMjQ4YzAyZjJjMjZiYzA3ZGYxYTk.. CSeq: 1 INVITE. Server: OpenSer (1.1.0-notls (x86_64/linux)). Content-Length: 0. Warning: 392 192.168.250.41:5060 "Noisy feedback tells: pid=11197 req_src_ip=192.168.250.108 req_src_port=40430 in_uri=sip:00886228357765@192.168.250.41 out_uri=sip:+886228357765@192.168.250.41 via_cnt==1". . # U 2006/09/28 16:57:11.569258 192.168.250.108:40430 -> 192.168.250.41:5060 ACK sip:00886228357765@192.168.250.41 SIP/2.0. Via: SIP/2.0/UDP 192.168.250.108:40430;branch=z9hG4bK-d87543-0c396723de34cf5a-1--d87543-;rport. To: "00886228357765"<sip:00886228357765@192.168.250.41>;tag=329cfeaa6ded039da25ff8cbb8668bd2.e6c9. From: "Ronald-taipei @ video "<sip:7311@192.168.250.41>;tag=3f7a5072. Call-ID: YmQ0YzNiYmU4MDJjMjQ4YzAyZjJjMjZiYzA3ZGYxYTk.. CSeq: 1 ACK. Content-Length: 0. .

2 3

[Users] Digest authentication
by Artur Hayne 03 Oct '06

03 Oct '06

Hi, Can I change the authentication protocol for any other or the Openser authenticate only with the digest? The Digest needs that the password in the DB stay in clear text and its not good. Thanks. --------------------------------- O Yahoo! está de cara nova. Venha conferir!

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[Users] Who is the rtpproxy current maintainer ?
by Jeremie Le Hen 03 Oct '06

03 Oct '06

Hi there, sorry for bugging you with such a trifling question, but I tried to send a patch to the documented RTPProxy maintainer (Maxim Sobolev) twice (@PBXPress.com and @portaone.com) but I got a mail from MAILER-DAEMON each time, claiming the user doesn't exist. Thank you. Best regards, -- Jeremie Le Hen < jeremie at le-hen dot org >< ttz at chchile dot org >

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[Serusers] Re: sems pthread issue
by jorain 03 Oct '06

03 Oct '06

Hi everyone, I'm doing some heavy tests on sems, which involves making lots of calls to ivr & isdngw modules. I noticed that the sems threads size (seen from process table or system monitor) keep increases with every call made. It wasn't release even when the call is already hangup, until it reach like 2GB, then I start getting pthread error code 12 whenever I tried calling. I have to restart sems to release the threads. I'm no good in C language and not sure how I can handle the pthread error. Can anyone give me some advice as how to utilize the memory usage for sems, so that it can keep on running like SER? Best Regards, Jorain

3 2

[Users] problem to create DB in openser cvs (1.2)
by Rafael J. Risco G.V. 03 Oct '06

03 Oct '06

Hi I've installed openser dev 1.2 to test new acc module in my lab but cant create DB in mysql server using ist openser_mysql, please see this error: ]# openser_mysql.sh create MySql password for root: Domain (realm) for the default user 'admin': 10.0.1.231 creating database openser ... /usr/local/sbin/openser_mysql.sh: line 262: method: command not found /usr/local/sbin/openser_mysql.sh: line 262: from_uri: command not found /usr/local/sbin/openser_mysql.sh: line 262: from_tag: command not found /usr/local/sbin/openser_mysql.sh: line 262: to_uri: command not found /usr/local/sbin/openser_mysql.sh: line 262: to_tag: command not found /usr/local/sbin/openser_mysql.sh: line 262: callid: command not found /usr/local/sbin/openser_mysql.sh: line 262: cseq_no: command not found /usr/local/sbin/openser_mysql.sh: line 262: sip_code: command not found /usr/local/sbin/openser_mysql.sh: line 262: sip_reason: command not found real 0m0.000s user 0m0.000s sys 0m0.000s /usr/local/sbin/openser_mysql.sh: line 262: callid: command not found /usr/local/sbin/openser_mysql.sh: line 262: method: command not found /usr/local/sbin/openser_mysql.sh: line 262: from_uri: command not found /usr/local/sbin/openser_mysql.sh: line 262: from_tag: command not found /usr/local/sbin/openser_mysql.sh: line 262: to_uri: command not found /usr/local/sbin/openser_mysql.sh: line 262: to_tag: command not found /usr/local/sbin/openser_mysql.sh: line 262: callid: command not found /usr/local/sbin/openser_mysql.sh: line 262: cseq_no: command not found /usr/local/sbin/openser_mysql.sh: line 262: sip_code: command not found /usr/local/sbin/openser_mysql.sh: line 262: sip_reason: command not found real 0m0.000s user 0m0.000s sys 0m0.000s /usr/local/sbin/openser_mysql.sh: line 262: callid: command not found ERROR 1064 (42000) at line 80: You have an error in your SQL syntax; check the manual that corresponds to your MySQL serverversion for the right syntax to use near '=0(root) gid=0(root) groups=0(root),1(bin),2(daemon),3(sys),4(adm),6(disk),10(wh'at line 2

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[Users] rewritehost failure route
by Cseke Tamas 03 Oct '06

03 Oct '06

Hello, I 've problem with rewritehost function in failure_route with openser 1.1.0 In core functions i don't know in which route can i use them, unlike functions in modules. I 'd like to make a failover system, In route block rewritehost is successfull but on failure route there is no effect i think, i pass the host to t_relay, so it is relayed to correct host, but in r-uri, there is incorrect host, and sip proxy reply with (SIP/2.0 400 Bad Request - 'Invalid IP Address) route{ ...... t_on_failure("1"); rewritehost("xxx.xxx.xxx.90"); t_relay(); ..... } failure_route[1] { t_on_failure("2"); append_branch(); rewritehost("xxx.xxx.xxx.91"); t_relay("xxx.xxx.xxx.91"); } failure_route[2] { # failover cisco (zajcek) append_branch(); rewritehost("xxx.xxx.xxx.31"); t_relay("xxx.xxx.xxx.31"); } Thanks any help, Tamas

2 1

[Users] SER / Load Balancing
by Thomas Deillon 03 Oct '06

03 Oct '06

Hi everybody, I saw this: http://www.iptel.org/faq/how_can_i_get_load_balancing_failover_with_ser. On asterisk, you can put and share on DB the dial plan and the sip definition but not the information relative to the REGISTER message (localisation). So, you cannot make perfect load-balanced servers. We have here a lot of clients and asterisk boxes in 3 towns with a very good connections between them. I wondered if it's possible with ser to put and share in the database the "REGISTER" information and so able any phone to connect to any server ? Thanks a lot for your help, Best Regards, Thomas

1 0

[Serusers] How to test basic multicast support on LAN?
by zain 03 Oct '06

03 Oct '06

Hi, I want to use basic multicast support in SER, but I don't know how to test the multicast function on LAN to see whether it's working in SER. The following are some descriptions about my version. OS: Debian kernel version: 2.6.8-2-686 SER version: 0.9.6 I add the following lines in ser.cfg. mcast_loopback=yes mcast_ttl=32 I use "/etc/ser start" to start SER. SER works good, but I don't know how to verify multicast on LAN. If the UA(User Agent) is kphone or SJphone. Thanks in advance. Justin

2 1

[Serusers] http digest authentication
by Holger Kinkelin 03 Oct '06

03 Oct '06

Hello list! I wonder how a UAC calculates the response to a 401 "Unauthorized" message from a server. I know that it works much like http digest authentication, but it's not exactely the same, since I do NOT have the Quality of Protection (qop) value. (Or am I wrong? My assumption is based on the fact that I don't see a qop-value in the REGISTER message my kphone sends after the 401) I've written some piece of code that is (in pseudo code) like that: HA1 = md5(username + ":" + realm + ":" + password) HA2 = md5(digestURI) response = md5(HA1 + ":" + nonce + ":" + recCounter + ":" + cNonce + ":" + qualOfProt + ":" + HA2) username = "1000" realm = "192.168.0.31" password = "1000" digestURI = "sip:192.168.0.31" <--- IS THIS CORRECT?? Or is it s.th like "REGISTER sip: ..."? nonce = "4520e111333a24d8c4f3d20c6171cc37dfa2be33" recCounter = "00000001" cNonce = "abcdefghi" qualOfProt IS MISSING! I know that my code is working for "normal" http digest authentication but not for "sip digest authentication". So what am I doing wrong? Thanks for replies in advance! Regards, Holger

2 1

[Serusers] error running radius client
by technos e 03 Oct '06

03 Oct '06

Hi All, I am trying to run a radius client using radiusclient -f digest localhost auth <secretkry> its giving error: error opening configuration file, can anyone pls help me ?? its very urgent.. thanks in advance. techno __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[Users] Need help in adding certificate
by Ferianto siregar 03 Oct '06

03 Oct '06

Dear all, Thank you very much for your time. All, I read from tls.html (www.openser.org) that for making good certificate, we must add rootCA to calist file by using: cat add_cacert.pem >> calist.pem 1.When I try to run it, I got error message. It can not execute? Why? It says :"There is no command add_cacert.pem" Does that command just make new file named "calist.pem"? (There is no difference between cacert.pem) 2. I compile TLS with openssl.0.9.7a. The openssl-0.9.7.a packet, I got when installing Redhat. But, How can I know that TLS that I have build using openssl or ssl (for security) version? or How can we run openssl or ssl? I do hope anybody can help me. Please..Thank you very much. Regards, Ferianto --------------------------------- Stay in the know. Pulse on the new Yahoo.com. Check it out.

2 1

[Users] Problem with timer and t_forward_nonack
by Antoine Fressancourt 03 Oct '06

03 Oct '06

Hello, I have a problem with my openser.cfg file, and especially with the way I should deal with the fr_inv_timer. I can't get it to work in my openser.cfgfile, and every time the timer ends I get the following error: 6(21521) ERROR:tm:t_forward_nonack: no branch for forwarding 6(21521) ERROR:tm:w_t_relay: t_forward_nonack failed The error occurs in a route bloc : route[1] { xlog("L_INFO", "INFO: route1 with r-URI $ru\n"); xlog("L_INFO", "test 14 : if(isflagset(2)) (line 200)\n"); if(isflagset(2)){ xlog("L_INFO", "-----> TRUE ,if the flag is set to 2, it calls t_on_failure\n"); t_on_failure("1"); t_relay(); exit; } xlog("L_INFO", "EXECUTION test 15 : if (!t_relay()) (ligne 206)\n"); if (!t_relay()) { xlog("L_INFO", "---------> TRUE\n"); sl_reply_error(); }; exit; } I have a log message at the beginning of failure_route[1] that shows that it is not entered before the error occurs. Could you tell me what's wrong ? Thank you very much, Antoine

1 0

[Serusers] xlog
by Legos Engineering 03 Oct '06

03 Oct '06

HI, The log of xlog are generated in messages. So i would like to know, what i must do if i want that xlog send its information in an other file for only one request L_WARN(...) I want to use this request in order to do stats of connection of UA Thx for your help.

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[Users] doubt in Loose route section
by raviprakash sunkara 03 Oct '06

03 Oct '06

Hello Users, When I working account part and Nat with OpenSER, The Transaction of route record entered into the Loose route Section. if(loose_route()) { acc_db_request(" ",""); t_relay(); exit; }; # call type transaction taking place here......... .............. ..................... ................... ...................... ..................... .................. All the Accounting and voice part are working ... but i'm doubt in loose route taking place all the call transcation.... -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 ravi.sunkara(a)hyperion-tech.com www.hyperion-tech.com

1 0

[Serusers] Ser_acct_authentication using ser
by rahman atikur 03 Oct '06

03 Oct '06

Dear all: I am new in ser. I install ser 0.9.6. And make a call using halaphone.. Now i want to use radius server for authenticate. I already install Radius server freeradius-0.9.0 and radiusclient-ng-0.5.3 .. Now the problem is auth_radius module is not installed... >From the scratch ... how i solve the problem. Regards John __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

1 0

[Users] How to leave offline-voice message?
by cv aik 02 Oct '06

02 Oct '06

Hi all, I'd like to ask some questions about MSILO module.. I have setup server and it works well, then I add msilo configuration into openser.cfg For addition, my clients are using MS Windows. I used 2 clients to test MSILO feature. I try to call the offline client, but my x-lite display that user not found. Please tell me what's the problem. Thanx Regards, Aldi --------------------------------- Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail.

2 1

[Serusers] SER connecting to SIP VoIP providers
by Voipers Portugal 02 Oct '06

02 Oct '06

Dear all, I am am using SER to provide SIP service to my clients, but I would like to allow them to make outbound calls. This is where a SIP VoIP provider comes in. As far as I know, I have 2 possibilities: - using IP authentication - using client authentication As for the first one, I only have to forward the INVITE messages, right? What I would like to know, is how to use "user authentication". In this case, it would be transparent to the VoIP provider since it should be treated as a regular SIP user agent. Can anyone help me out? Jose Simoes

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[Users] how to handle "488 Not Acceptable Here"
by Richard Bennett 02 Oct '06

02 Oct '06

hi, When calling from Sipura to softphone (either SJphone or the sip client in my e61 nokia phone), with the Sipura set to Prefered codec: g723 Use prefered codec only: no and the softphone only supporting g711, I get this reply from the softphone: U xxxx:11474 -> xxxxx:5060 SIP/2.0 488 Not Acceptable Here. Via: SIP/2.0/UDP xxxx;branch=z9hG4bK23e.248c6cd1.1,SIP/2.0/UDP xxxx:10001;branch=z9hG4bK-3e8ce192;rport=10001. To: <sip:1@test.com>;tag=ujk6mpqhbhhc6kj68irr. From: sipura line1 <sip:2@test.com>;tag=d0404120874d710eo0. Call-ID: e5052808-575f7746(a)192.168.1.52. CSeq: 102 INVITE. Warning: 304 192.168.1.3 Media type not available. Content-Length: 0. the invite was: U xxxx:5060 -> xxxx:11474 INVITE sip:1@test.co SIP/2.0. Record-Route: <sip:xxxx;lr=on;ftag=d0404120874d710eo0>. Via: SIP/2.0/UDP xxxx:branch=z9hG4bK23e.248c6cd1.1. Via: SIP/2.0/UDP xxxx:10001;branch=z9hG4bK-3e8ce192;rport=10001. From: sipura line1 <sip:2@test.com>;tag=d0404120874d710eo0. To: <sip:1@test.com>. Call-ID: e5052808-575f7746(a)192.168.1.52. CSeq: 102 INVITE. Max-Forwards: 69. Contact: sipura line1 <sip:2@xxx:10001>. Expires: 240. User-Agent: Sipura/SPA2002-3.1.5. Content-Length: 419. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura. Content-Type: application/sdp. . v=0. o=- 36143 36143 IN IP4 xxxx. s=-. c=IN IP4 xxxxxx. t=0 0. m=audio 62238 RTP/AVP 4 0 2 8 18 96 97 98 100 101. a=rtpmap:4 G723/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729a/8000. a=rtpmap:96 G726-40/8000. a=rtpmap:97 G726-24/8000. a=rtpmap:98 G726-16/8000. a=rtpmap:100 NSE/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. If I set prefered codec to g711 on the Sipura it works normally. What is the best way to handle this on the sip proxy? Thanks, Richard

2 2

[Serusers] Accounting problem with radius server
by Caxton 02 Oct '06

02 Oct '06

Dear all: I setup SER and use radius server to accounting. However, the record after I test are not complete. A(Caller) <------> B(Callee) Suppose A is caller and B is callee. The record will be only one and correct if A hang up first. Unfortunately, if B hang up faster than A, the record will be divide into two records like table1(for example.). Table 1. |-----------------------------------------------------------------------------------------------------------------| |AccSessionID AccUniqueId ...<skip>... Start Time Stop Time ...<skip>... | |-----------------------------------------------------------------------------------------------------------------| |AHandUpFirst 001 2006-10-02 2006-10-02 | | 12:40:03 12:45:32 | |-----------------------------------------------------------------------------------------------------------------| |BHandUpFirst 002 2006-10-02 0000-00-00 | | 12:58:40 00:00:00 | |-----------------------------------------------------------------------------------------------------------------| |BHandUpFirst 003 2006-10-02 2006-10-02 | | 13:05:03 13:05:03 | |-----------------------------------------------------------------------------------------------------------------| How can I solve this problem? Please give me a hand. Test User Agent: Window messenger 5.1 and X-lite Regards, Caxton

1 0

[Users] OT: searching for a softclient with SDP feature
by Thomas Balsfulland 02 Oct '06

02 Oct '06

hello group, i am searching for a sip-softclient (windows or linux - opensource, freeware preferred), which is able to send an INVITE completely without any SDP information (no message body). does someone of you know something like this? regards thomas -- thomas balsfulland tbals(a)ctrl-c.de zwischen mut und dummheit liegt nur ein grat der sich durch das vorbereitetsein unterscheidet

2 1

[Serusers] SER for inbound calls
by Voipers Portugal 02 Oct '06

02 Oct '06

Dear all, Another question that I have to do is: Imagine that I have a connection to a VoIP provider using a username and a password (so that SER register in the VoIP provider as a normal user). Once it is registered, do you think it possible to receive calls? That is, make it ring in one or all of the SIP users configured in SER? Any help would be appreciated. Jose Simoes

3 2

[Serusers] DNS SRV support ?
by Kostas Marneris 02 Oct '06

02 Oct '06

Hello, I run ser.0.9.6 and I can use either rewritehostport("1.2.3.4:5060"); or rewritehostport("pstngw.foo.com:5060"); with t_relay() command on INVITE/PSTN handler in order to route calls to a Cisco PSTN GW. This works fine. Is "DNS SRV queries" supported on ser.0.9.6 and is there a specific conf ? thanx Kostas

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[Serusers] problem
by vijay tiwari 02 Oct '06

02 Oct '06

i have install the ser and it installed but i have some problem 1. first i am not able to run the ser command 2. i want to make the scripts for ser and mediaproxy but its show me one error for running this command i am in /etc/init.d when i am running this command chkconfig --add ser error reading information on service ser: No such file or directory please help me out with regards vijay --------------------------------- How low will we go? Check out Yahoo! Messengers low PC-to-Phone call rates.

2 1

[Users] complete openser.cfg available?
by Ronald Wiplinger 02 Oct '06

02 Oct '06

I would like to see beside the sample and the "advices" to add to the cfg file a complete cfg file. Is such a more complese sample available? I am sure most of the users will use more than just the basics, and most of them will use the same modules and set-up. As new user, I face often the problem where to put the snip set from the documentation. I think it would be better to comment out all not necessary parts for the basic, or mention the spot to put in other configuration parts (anyway only the route() part needs that) bye Ronald

3 2

[Serusers] UAC multiple Registration
by Abdul Qadir 02 Oct '06

02 Oct '06

Hi, Is it possible in SER to implement login concept like yahoo messenger, when you login from another pc your prevois login is signedoff by server. i.e unregister a previously registerd client when same client tries to register from another location. Thanks in advance, Abdul Qadir --------------------------------- Get your email and more, right on the new Yahoo.com

2 1

[Users] Duplicating SIP Messages
by Adi Simon 01 Oct '06

01 Oct '06

Hi, I'm trying to figure out if it is possible to duplicate SIP Messages. In a setup of a SER proxy and two asterisk boxes I want to send all the Register messages to both of the asterisks, I tried doing the following in the openser.cfg: if (method=="REGISTER") { route(2); } else { route(1); }; route[2] { rewritehostport("10.0.0.1:5060"); if (!t_relay()) { sl_reply_error(); }; rewritehostport("10.0.0.2:5060"); if (!t_relay()) { sl_reply_error(); }; exit; } But I keep getting : 0(22183) ERROR: t_newtran: transaction already in process 0xb6136318 0(22183) DEBUG:sl:sl_reply_error: error text is I'm terribly sorry, server error occurred (1/SL) Is it possible at all? Thanks, Adi.

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[Users] Please help for tasting TLS with minisip
by Ferianto siregar 01 Oct '06

01 Oct '06

Dear Jeevan, How are you? I do hope your are fine. I would like to say thanks to you for your help before. Thank you. I am so sorry for disturb you. Sorry Jeevan, I still have problem with configuring minisip like I asked you a week ago. I still can not make a call with minisip. There is always error message. Can I ask you again, Please? Here is my problem: 1. I run ./gen_rooCA.sh and running this script, I got rootCA directory. This directory contains: a.cacert.pem b.index c.serial d.private directory which contains cakey.pem e. cert directory which contain 01.pem 2. I run ./gen_usercert.sh user and running this script,I got user directory. This directory contains: a.user-calist.pem b.user-cert.pem c.user-privkey.pem d.user-cert_req.pem 3. When running minisip (suppported TLS), I done these tasks: a.In "Preferences settings", I use TLS port in "enable TLS Local Port" and "Transport Method" is TLS. My Questions: 1. In "Personal Settings", what file should I put? I mean, in menu "choose a certificate", what should I browse to fill this? 2. In " choosen a private key", what should I browse to fill this? 3. In "CA database", what sholud I browse to fill this? Because when I fill "chosee a certificate" with "cacert.pem", the "choosen a private key" with "cakey.pem" ,and the " CA databese" with "user-cert.pem","cacert.pem","user-calist.pem","user-privkey.pem", I got the error message: The error message said that the "SSL : connect failed" Here is : SipMessageTransport: sendMessage: creating new socket IP4Address(string): sip.pcr.ac.id (202.95.149.251) Creating new SSL_CTX Enter PEM pass phrase:openser Unknown command: o Packets will be displayed to the screen Unknown command: e Unknown command: n Unknown command: s Unknown command: e SSL: connect failed SipMessageTransport: sendMessage: exception thrown! Registering all identities to their registrar: Registration request sent (username = aldi(a)pcr.ac.id) SipMessageTransport: sendMessage: creating new socket IP4Address(string): sip.pcr.ac.id (202.95.149.251) What should I do, Jeevan? I have tried it many times, but I still can not make a call with minisip. I realize that my skill is too weak. Would you mind guide me again to solve this problem? Please... Thank you very much.. I am sorry for waste your time, Thanks. Regards :) Ferianto Note: 1. I can not run openssl x509 -in cakey.pem -out cakey.crt. It says that it can not converted. TRUSTED CERTIFICATE 2. I copy all the file ("cacert.crt",cakey.pem","user-calist.pem",etc) from my linux server to windows. Because windows can not download with the extension .pem. Is it correct? --------------------------------- How low will we go? Check out Yahoo! Messengers low PC-to-Phone call rates.

1 0

[Users] [openser] Enabling TLS Support
by John Shellhouse 30 Sep '06

30 Sep '06

Hi, I am about to enable TLS support although my current openser.cfg system works fine and I do not particularly want to damage it, but I would like to incorporate TLS. I was planning to save my current openser.cfg file and just add the necessary TLS info and wonder if this can be done? 1) I understand that I can use both TLS(5061) and TCP(5060), so therefore particular portions in my current openser.cfg should still work, is this a proper assumption? 2) How would this affect users that are using port TCP(5061) ? 3) So when I remake with TLS support how will this affect the current system? 4) Will the remake write over all files stored in /usr/local/etc/openser and other openser directories? 5) Will the remake create new database tables, basiecally would I be able to keep the current database structure I currently have without reloading openser_mysql.sh? 6) In a nutshell, how will the remake negatively affect my current openser system that I should be aware of? I apologize for the lengthy amount of questions but I am just not sure in these areas. Thanks in advance. John

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Re: Re: [Users] tls. soket queues not flushed
by serega 30 Sep '06

30 Sep '06

Hi Klaus and all other who read this. It's not a problem of tcp socket - yesterday i rewrite gateway config for fowarding directed to extgateway requests to gateway itself. In tcp this problems not occured, but after i switched on tls, i see overfilled queues again. In that case i haven't extgateway, communications with remote sockets and etc. Did anybody test openser whith big payload? I think this problem occurs because in my case i have big traffic and several process that simultaniously use tls. Did anybody have the same problem? >Hi Serega! > >I'm not sure - just a guess. openser uses "worker threads". This worker >threads get a SIP message from the listener thread, process them, and >then they send them. > >If for some reason the sending fails - e.g. the gateway sends via >TCP/TLS to extgateway and extgateway's receiving queue is full. Then >gateway's sending buffer gets full and the threads wait until they can >send. Thus, as the threads are all busy, also the receiving queue of >gateway gets full. I guess after some time the threads give up sending >and will read again. > >The question is: why does the sending buffer gets full? Where is gateway >sending the messages to? > >The sending queue must not get flushed. The queue is in the OS, not in >openser. As TCP guarantees that there is no loss/reordering the TCP >stack must not flush the queue. > >regards >klaus > >serega wrote: >> Hi all. >> >> I have next problem while using openser with tls. System consists of >> sip statefull server (in next time, simply, server) and stateless sip >> gateway (gateway). Server used for connect to jabber server. Server >> contains rewrote jabber module logic and also use presence module. >> Gateway used to connect to other sip gateway (extgateway) via tls >> protocol. Gateway use 1.1 openser version and doesn’t contain changed >> code. When server together with gateway restarts and server in its >> database contains above 260 subscriptions (in watcherinfo table) i >> have error. Socket used by gateway to connect with extgateway >> contains in receive and sent queues a lot of data (above 50kByte on >> each sides). This data never flushed out. This happens because after >> restart sip server through gateway send notifications to extgateway. >> When I attached using gdb to process that send data, I saw that it >> was in infinity loop because tls library returns SSL_ERROR_WANT_WRITE >> and I think it’s correct because we have overfilled send queue. In >> this case I don’t interest why receive queue not empty (I think it >> happens because sending process have got block on socket). But I >> don’t understand why sent queue not flushed. I test this behavior >> using tcp – all was correct. Socket not closed by other side because >> this status can be unchanged above one day. Anybody can help me? Why >> this happens? >> >> I use openssl-0.9.8c and redhat os (Linux xdevel1 2.4.21-4.ELsmp #1 >> SMP Fri Oct 3 17:52:56 EDT 2003 i686 i686 i386 GNU/Linux). Computer >> have too Pentium 4 processors). >> >> >> Gateway.cfg (sip.qa.hbex.com – gateway address): >> >> # # $Id: router_qa.cfg,v 1.1 2006/08/02 18:14:34 ilya Exp $ # # >> simple quick-start config script # >> >> # ----------- global configuration parameters >> ------------------------ >> >> debug=9 # debug level (cmd line: -dddddddddd) >> log_facility=LOG_LOCAL0 fork=yes log_stderror=yes # (cmd line: -E) >> >> >> check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) >> rev_dns=no # (cmd. line: -R) children=20 >> fifo="/tmp/openser_fifo" server_signature=no >> >> fifo_db_url="mysql://root:@localhost/openser" >> >> listen = udp:sip.qa.hbex.com:5060 #incoming sip server address listen >> = tcp:sip.qa.hbex.com:5060 >> >> #tls address disable_tls = 0 listen = tls:sip.qa.hbex.com:5061 listen >> = udp:sip.qa.hbex.com:5061 >> >> tls_certificate = "/ home/inop/ex.com-cert.pem" tls_private_key = >> "/home/inop/ex.com-privkey.pem" tls_ca_list = >> "/home/inop/ex.com-calist.pem" tls_require_client_certificate=0 >> tls_verify_client=0 tls_verify_server=0 >> >> # ------------------ module loading >> ---------------------------------- loadmodule >> "/home/interop/openser/lib/openser/modules/rr.so" loadmodule >> "/home/interop/openser/lib/openser/modules/xlog.so" loadmodule >> "/home/interop/openser/lib/openser/modules/textops.so" loadmodule >> "/home/interop/openser/lib/openser/modules/maxfwd.so" loadmodule >> "/home/interop/openser/lib/openser/modules/sl.so" loadmodule >> "/home/interop/openser/lib/openser/modules/mysql.so" loadmodule >> "/home/interop/openser/lib/openser/modules/tm_unchanged.so" >> loadmodule "/home/interop/openser/lib/openser/modules/usrloc.so" >> loadmodule "/home/interop/openser/lib/openser/modules/registrar.so" >> >> # ----------------- setting module-specific parameters >> --------------- modparam("registrar", "default_expires", 120) >> modparam("registrar", "use_domain", 1) >> >> modparam("usrloc", "use_domain", 1) 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) >> >> # add value to ;lr param to make some broken UAs happy modparam("rr", >> "enable_full_lr", 1) modparam("maxfwd", "max_limit", 10) # >> ------------------------- request routing logic ------------------- >> # main routing logic route{ if (!mf_process_maxfwd_header("10")) { >> sl_send_reply("483","To Many Hops"); drop(); }; if (dst_port==5061) { >> if (search("^To:.*@.*cmnicator.com") || search("^To:.*@.*cihb.com")) >> { # rewrite destination and forward to jabber (sip server) route(1); >> return; }; sl_reply_error(); return; } if (method=="REGISTER") { >> xlog("XXX: saving location msg=$mb\n"); if(!save_noreply("location")) >> { log("XXX: Error saving location!\n"); sl_reply_error(); } >> sl_send_reply("200","OK"); return; } if >> (search("^To:.*@.*cmnicator.com") || search("^To:.*@.*cihb.com")) { >> #forward to self route(3); return; } ##forward to ext gateway >> route(2); } >> >> route[1]{ if (method!="MESSAGE" && method!="SUBSCRIBE" && >> method!="NOTIFY") { log("XXX: Request not forwarded to sip >> server!\n"); sl_send_reply("202","Accepted"); return; } >> t_on_reply("1"); record_route(); >> >> if(!lookup("location")) { if (method!="SUBSCRIBE") { log("XXX: only >> subscribes are processed for user that are not registered\n"); >> sl_reply_error(); return; } >> subst_uri('/(.*)@(.*)/\1#\2*jabber.qa.im.hb.com@jabber01.qa.im.hb.com/ig'); >> if(!t_relay()) { log("XXX: error forwarding jabber01...\n"); >> sl_reply_error(); return; } else { sl_send_reply("200","OK"); return; >> } } else { log("XXX: forwarding to the address of record...\n"); >> if(!t_relay()) { log("XXX: error forwarding to address of record >> \n"); sl_reply_error(); return; } else { sl_send_reply("200", "OK"); >> return; } } >> >> } >> >> # forwarding to external gateway route[2]{ log("XXX: rewriting >> headers\n"); >> subst('/^(From:[^@#]*)#([^@]*)\*jabber.qa.im.hb.com@jabber...qa.im.hb.com(.*)/\1@\2\3/ig'); >> >> subst('/^(Contact:[^@#]*)#([^@]*)\*jabber.qa.im.hb.com@jabber...qa.im.hb.com(.*)/\1@\2\3/ig'); >> >> >> if(!forward("tls:43.123.141.166:3000")) { log("XXX: Error forwarding >> to external gateway!\n"); sl_reply_error(); } else { >> sl_send_reply("200", "Accepted"); }; } >> >> # forwarding to SELF route[3]{ log("XXX: rewriting headers\n"); >> subst('/^(From:[^@#]*)#([^@]*)\*jabber.qa.im.hb.com@jabber...qa.im.hb.com(.*)/\1@\2\3/ig'); >> >> subst('/^(Contact:[^@#]*)#([^@]*)\*jabber.qa.im.hb.com@jabber...qa.im.hb.com(.*)/\1@\2\3/ig'); >> >> >> log("XXX: about to forward to self\n"); >> if(!forward("tcp:sip.qa.hbex.com:5061")) { log("XXX: Error forwarding >> to self!\n"); sl_reply_error(); } else { sl_send_reply("200", >> "Accepted"); }; } >> >> # this is executed for replies onreply_route[1]{ >> subst('/^(To:[^@#]*)#([^@]*)\*jabber.qa.im.hb.com@jabber...qa.im.hb.com(.*)/\1@\2\3/ig'); >> >> subst('/^(Contact:[^@#]*)#([^@]*)\*jabber.qa.im.hb.com@jabber...qa.im.hb.com(.*)/\1@\2\3/ig'); >> route(2); } >> >> >> >> _______________________________________________ Users mailing list >> Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users > >

2 1

[Users] Is there a way to give a dial tone to the caller?
by Ronald Wiplinger 30 Sep '06

30 Sep '06

When I am placing a call I will never hear a dial tone. Is there a way to add dial tone? bye Ronald

2 2

[Users] Please help for creating CA root certificate nad client certificate
by Ferianto siregar 30 Sep '06

30 Sep '06

Dear all, Thank you very much for this chance. Thank you for your time to read this message. All, I have read document about tls from openser.org In that document, I read that to create CA root and client certifacate, we just must run script ./gen_rootCA.sh and ./gen_usercert.sh where is in tls/tools. It will produce rootCA directory and user directory. Then, I put the rootCA directory and user directory into my public webserver in order the client can download it. But, when I try to download the certificate from the server (using Redhat 9.0), to the client computer (using Windows XP), why I can not download it? Do, I have to convert the extension form .pem to .crt? How can I convert it? Please help..Please.. Regards, Ferianto __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

1 0

[Serusers] Best platform for SER --->please suggest some thing
by ravi reddy 30 Sep '06

30 Sep '06

Hello guys, I want to install SER on a SERVER so please suggest me some servers (prefererably with configuration like HP, or some thing) because basically i dont have any knowledge on server types . so, any comments and suggestions are appreciated . Thank You. Regards, Ravi.

3 3

[Serusers] SER and Routing Help Question
by Mario Roeber 30 Sep '06

30 Sep '06

Hello, maybe someone can help me with the routing. here the schema. Firewall with 3 Network interfaces 2x interfaces intern 1x interface extern the default provider is sipgate.de. ser shoult route all registration default to sipgate and listen on 2 intern interfaces. ser shoult handel all intern calls and route direct intern. user auth i'dont need or radius i'dont need. all inbount calls from extern shoult route automatic to right sip- phone. current i use the default configuration with 2 options more. listen 212.21.68.62 listen 212.21.69.110 maybe someone can help me to with the routing thanks a lot for your time, if you can help me. regards mario roeber

1 0

[Users] [openser] Displaying Caller-ID
by Tracy Lofton 29 Sep '06

29 Sep '06

Hi Everybody, I was wondering if I would be able to display the CallerID in the display of the callee's phone rather than it displaying anonymous, maybe it can be done if I used something like this from the tm module: if(search("From:[0-9]{10}@.*"){ append_hf("remote-party-id:<sip:2134445555 at mydomain.com>") ) or would i use openser auth module with append_rpid_hf() in some form or fashion? If so, how can I do this? Just need a little input, guidance and assistance and to know if it actually can be done. If not this method, then what can I use to display the callID of the calling party to the caller display. Some providers are known to block the call if a certain amount of digits are not displayed. Thanks in advance. Tracy _________________________________________________________________ Add fun gadgets and colorful themes to express yourself on Windows Live Spaces http://clk.atdmt.com/MSN/go/msnnkwsp0070000001msn/direct/01/?href=http://ww…

3 3

Re: [Serusers] how to add mysql.so
by vijay tiwari 29 Sep '06

29 Sep '06

thanks when i install mysql and i type this command ./configure --prefix=/usr/local/mysql --enable-large-files --enable-shared=yes it show me some checking and show me this error checking "LinuxThreads"... "Not found" configure: error: This is a linux system and Linuxthreads was not found. On linux Linuxthreads should be used. Please install Linuxthreads (or a new glibc) and try again. See the Installation chapter in the Reference Manual for more information. and i am not able to install mysql sip <sip(a)arcdiv.com> wrote: Why not? That's a little vague. Can't add them because you don't have it? Can't add them because of permissions problems? Can't add them because the directory doesn't exist? Can't add them because you don't know the unix copy command? A little more specific information goes a long way. N. On Fri, 29 Sep 2006 07:30:18 -0700 (PDT), vijay tiwari wrote > > hi all > > > i am not able to add the mysql in that locations /usr/local/lib/ser/modules > > > thanks > vijay > --------------------------------- All-new Yahoo! Mail - Fire up a more powerful email and get things done faster. Vijay Tiwari Network Admin Sachitel Communications Noida(U.P.) M- 09871633711 --------------------------------- Do you Yahoo!? Get on board. You're invited to try the new Yahoo! Mail.

1 0

[Users] thought exercise
by Mark Kent 29 Sep '06

29 Sep '06

I wonder if anyone on this list is open to a thought exercise? I'm trying to put together a cogent message about this, but in the meantime: I've got a situation where my openser box seems to route stuff to itself and it loops around and around until I get the classic "Too Many Hops" error. This seems to happen most often on BYE, although I've seen an ACK do it too. Here is a small ngrep snippet, after PHONE has called OPENSER-ONE who statelessly relays to OPENSER-TWO who t_relays to PSTNGW, and then the pstn user hangs-up: U 23:03:53.254382 PSTNGW:5060 -> OPENSER-TWO:5060 BYE sip:JoeUser@OPENSER-ONE.example.com:5060;user=phone SIP/2.0 Call-ID: callid@PHONE From: <sip:18005551212@OPENSER-TWO.example.com:5060>;tag=26674 To: "Joe User" <sip:JoeUser@OPENSER-ONE.example.com>;tag=thetag Content-Length: 0 CSeq: 1 BYE Via: SIP/2.0/UDP PSTNGW:5060;branch=thebranch Route: <sip:JoeUser@OPENSER-ONE:5060> Max-Forwards: 70 # U 23:03:53.255013 OPENSER-TWO:5060 -> OPENSER-ONE:5060 BYE sip:JoeUser@OPENSER-ONE:5060 SIP/2.0 Record-Route: <sip:JoeUser@OPENSER-TWO;lr=on;ftag=26674> Call-ID: callid@PHONE From: <sip:18005551212@OPENSER-TWO.example.com:5060>;tag=26674 To: "Joe User" <sip:JoeUser@OPENSER-ONE.example.com>;tag=thetag Content-Length: 0 CSeq: 1 BYE Via: SIP/2.0/UDP OPENSER-TWO;branch=otherbranch.0 Via: SIP/2.0/UDP PSTNGW:5060;branch=thebranch Max-Forwards: 69 Route: <sip:JoeUser@OPENSER-ONE.example.com:5060;user=phone> Right here OPENSER-ONE freaks out, and the delay causes OPENSER-TWO to re-transmit: # U 23:03:53.611227 OPENSER-TWO:5060 -> OPENSER-ONE:5060 BYE sip:JoeUser@OPENSER-ONE:5060 SIP/2.0 Record-Route: <sip:JoeUser@OPENSER-TWO;lr=on;ftag=26674> Call-ID: callid@PHONE From: <sip:18005551212@OPENSER-TWO.example.com:5060>;tag=26674 To: "Joe User" <sip:JoeUser@OPENSER-ONE.example.com>;tag=thetag Content-Length: 0 CSeq: 1 BYE Via: SIP/2.0/UDP OPENSER-TWO;branch=otherbranch.0 Via: SIP/2.0/UDP PSTNGW:5060;branch=thebranch Max-Forwards: 69 Route: <sip:JoeUser@OPENSER-ONE.example.com:5060;user=phone> OK, so who on this list is good enough to be able to suggest plausible theories about what my openser.cfg flaw is? If you don't want to speculate openly on the list then please send me a direct email. Thanks, -mark

3 6

[Users] e2e ACK matching problem
by JF 29 Sep '06

29 Sep '06

Hi all, I'm facing a cenario where an INVITE transaction goes through one (Open)SER more than one time, and the ACK to a 480 Temporarily Unavailable is wrongly being matched as an e2e ACK. I dug through tm code and found this comment in t_lookup.c: /* here we do an exercise which will be removed from future code * versions: we try to match end-2-end ACKs if they appear at our * server. This allows some applications bound to TM via callbacks * to correlate the e2e ACKs with transaction context, e.g., for * purpose of accounting. We think it is a bad place here, among * other things because it is not reliable. If a transaction loops * via SER the ACK can't be matched to proper INVITE transaction * (it is a separate transactino with its own branch ID) and it * matches all transaction instances in the loop dialog-wise. * Eventually, regardless to which transaction in the loop the * ACK belongs, only the first one will match. */ Does anyone have a solution to this problem? Thanks, JF

1 0

[Serusers] Registration
by senthil.nachimuthu＠bt.com 29 Sep '06

29 Sep '06

> Hi, > I want to configure SER as a proxy server. I would like SER to > proxy presence messages and I have done that with the help of > record_route and t_relay_to_udp. > I would like to have SER forwarding the register message to another > server to authenticate and after successful authentication SER store > the location to route the requests. I observed SER is not allowing URI > getting registered. > > I tried the following but did not work.. > > if(!uri==myself) { > > if( method=="REGISTER") { > record_route(); > save("location"); > t_relay_to_udp("132.146.109.243","5060"); > break; > }; > }; > > > Can anyone let me know how can I enable other domain URI passing > through SER to reach other server which authenticates them? > > > Thanks and Regards, > Senthil Kumar, >

1 0

[Serusers] how to add mysql.so
by vijay tiwari 29 Sep '06

29 Sep '06

hi all i am not able to add the mysql in that locations /usr/local/lib/ser/modules thanks vijay --------------------------------- All-new Yahoo! Mail - Fire up a more powerful email and get things done faster.

1 0

[Users] video rtp
by Ronald Wiplinger 29 Sep '06

29 Sep '06

Can anybody explain me this, please. How is the rtp stream with video? Which ports are usually used for that? Following thoughts I have how it could be: 1. video and audio are two indipendent different rtp streams 2. audio starts with the lower free udp numbers, video with the higher ones. Only that could give me an explanation, why audio is ok in our test, but no video, and only if we want to call to a NAT, but worked from it. Where can I get more info? bye Ronald

1 0

[Serusers] Can I configure SER to send Radius Accounting packets?
by a.kolesnikov 29 Sep '06

29 Sep '06

Hello, everyone. Can I configure SER to send Radius Accounting packets? С уважением, Колесников Андрей Netris телефон: +7 495 950 55 25 ext. 1303 <http://www.netris.ru> http://www.netris.ru

1 0

[Serusers] Busy Call Fwd..
by Natambu Obleton 29 Sep '06

29 Sep '06

I have my SER server setup to forward calls when busy and I am using polycomm 601 registered to SER. When the Polycomm is on the phone and another phone call its, they don't get a busy signal, but show up on line2 for answering. I think this is a stateful vs. stateless problem, but how can I set it up so that when the user is on the phone it rings busy? Really just limit the number of simultaneous calls. Thanks. Natambu Obleton Network Engineer FastTrack Communications nobleton(a)fasttrackcomm.net (970) 247-3366 office (970) 247-2426 fax

2 1

[Serusers] Re: SER and SPA 942
by Alex Fler 29 Sep '06

29 Sep '06

Hello All, Sorry for stupid question, I have ser server configured, and it works fine for my needs, I never had register my UA's directly to SER. When I registered I SPA 942 phone directly to ser and put call on hold, my phone display the message "Trying to hold" for a few seconds, and after that call getting dropped. Does anybody know why? Thank you for help. Regards Alex Fler

2 1

[Users] cdrtool/mediaproxy/openser
by Richard Bennett 29 Sep '06

29 Sep '06

Hi, Does anyone have an example openser.cfg file for using openser with Mediaproxy and CDRTool? I have Mediaproxy/openser working with the onsip getting started example, but this only passes calls that are behind nat to Mediaproxy. What is the best way to setup openser so that all calls pass through mediaproxy, and the corrected CDRs* can be written with accounting? * as mediaproxy disconnects zombie-calls, and corrects the accounting to the time the last rtp packets were detected. Thanks, Richard

1 0

[Users] RTPProxy manual
by Jeremie Le Hen 29 Sep '06

29 Sep '06

Hi, sorry if this is a bit off-topic, but I am looking for a document as near as a manual page as possible. I remember to have found such a document a few weeks ago, but I can't find it any more. It basically explained what did the command-line options, what is the bridge mode, etc. Pointers are welcome. Thank you. Regards, -- Jeremie Le Hen < jeremie at le-hen dot org >< ttz at chchile dot org >

1 0

[Users] mediaproxy working, but not if asterisk is involved
by Arne Van Theemsche 29 Sep '06

29 Sep '06

hi my users subscribe with openser, en asterisk is used as connectivity to pstn i am now installing a mediaproxy, for all users, so every call goes via a mediaproxy. I'm doing this as follows (relevant statements only) in route #I installed the t_on_reply here to be sure that every reply gets parsed, but normally in the INVITE section should be enough? t_on_reply("1"); if (method==INVITE) { use_media_proxy(); } onreply_route[1] { log(-3,"reply received"); if (!search("^Content-Length:[ ]*0")) { log(-3,"using mediaproxy"); use_media_proxy(); }; } the weird is, for all local users, this works fine, but as soon as asterisk is involved, the reply doesn't get triggered (not seeing the "reply received" either, only when disconnecting the call). The call get's established fine, asterisk is sending media to the mediaproxy, but the SDP towards the calling phone is not modified (since the onreply isn't triggered) am I missing something here? thanks Arne

2 4

[Users] Cancel question/problem
by Andreas M. 29 Sep '06

29 Sep '06

Hello, i have following setup: Cisco AS5300 -- Openser --- Client Now if there is an incoming call from the CiscoGW to the client and is canceled, but the "cancel" message does not arrive at the client (because any reason), the phone doesn´t stop ringing. Cisco --- OSER --- Client Cancel -> <- Canceling Cancel -> thats it. It seems that no one is missing any OK or ACKS! It happens only sometimes, but who is responsible to handle this behavior ? The Cisco Gateway or the Proxy ? Using OSER 1.0.1 regards, Andreas M.

2 1

[Users] tls. soket queues not flushed
by serega 29 Sep '06

29 Sep '06

Hi all. I have next problem while using openser with tls. System consists of sip statefull server (in next time, simply, server) and stateless sip gateway (gateway). Server used for connect to jabber server. Server contains rewrote jabber module logic and also use presence module. Gateway used to connect to other sip gateway (extgateway) via tls protocol. Gateway use 1.1 openser version and doesn’t contain changed code. When server together with gateway restarts and server in its database contains above 260 subscriptions (in watcherinfo table) i have error. Socket used by gateway to connect with extgateway contains in receive and sent queues a lot of data (above 50kByte on each sides). This data never flushed out. This happens because after restart sip server through gateway send notifications to extgateway. When I attached using gdb to process that send data, I saw that it was in infinity loop because tls library returns SSL_ERROR_WANT_WRITE and I think it’s correct because we have overfilled send queue. In this case I don’t interest why receive queue not empty (I think it happens because sending process have got block on socket). But I don’t understand why sent queue not flushed. I test this behavior using tcp – all was correct. Socket not closed by other side because this status can be unchanged above one day. Anybody can help me? Why this happens? I use openssl-0.9.8c and redhat os (Linux xdevel1 2.4.21-4.ELsmp #1 SMP Fri Oct 3 17:52:56 EDT 2003 i686 i686 i386 GNU/Linux). Computer have too Pentium 4 processors). Gateway.cfg (sip.qa.hbex.com – gateway address): # # $Id: router_qa.cfg,v 1.1 2006/08/02 18:14:34 ilya Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=9 # debug level (cmd line: -dddddddddd) log_facility=LOG_LOCAL0 fork=yes log_stderror=yes # (cmd line: -E) check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) children=20 fifo="/tmp/openser_fifo" server_signature=no fifo_db_url="mysql://root:@localhost/openser" listen = udp:sip.qa.hbex.com:5060 #incoming sip server address listen = tcp:sip.qa.hbex.com:5060 #tls address disable_tls = 0 listen = tls:sip.qa.hbex.com:5061 listen = udp:sip.qa.hbex.com:5061 tls_certificate = "/ home/inop/ex.com-cert.pem" tls_private_key = "/home/inop/ex.com-privkey.pem" tls_ca_list = "/home/inop/ex.com-calist.pem" tls_require_client_certificate=0 tls_verify_client=0 tls_verify_server=0 # ------------------ module loading ---------------------------------- loadmodule "/home/interop/openser/lib/openser/modules/rr.so" loadmodule "/home/interop/openser/lib/openser/modules/xlog.so" loadmodule "/home/interop/openser/lib/openser/modules/textops.so" loadmodule "/home/interop/openser/lib/openser/modules/maxfwd.so" loadmodule "/home/interop/openser/lib/openser/modules/sl.so" loadmodule "/home/interop/openser/lib/openser/modules/mysql.so" loadmodule "/home/interop/openser/lib/openser/modules/tm_unchanged.so" loadmodule "/home/interop/openser/lib/openser/modules/usrloc.so" loadmodule "/home/interop/openser/lib/openser/modules/registrar.so" # ----------------- setting module-specific parameters --------------- modparam("registrar", "default_expires", 120) modparam("registrar", "use_domain", 1) modparam("usrloc", "use_domain", 1) 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) # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) modparam("maxfwd", "max_limit", 10) # ------------------------- request routing logic ------------------- # main routing logic route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","To Many Hops"); drop(); }; if (dst_port==5061) { if (search("^To:.*@.*cmnicator.com") || search("^To:.*@.*cihb.com")) { # rewrite destination and forward to jabber (sip server) route(1); return; }; sl_reply_error(); return; } if (method=="REGISTER") { xlog("XXX: saving location msg=$mb\n"); if(!save_noreply("location")) { log("XXX: Error saving location!\n"); sl_reply_error(); } sl_send_reply("200","OK"); return; } if (search("^To:.*@.*cmnicator.com") || search("^To:.*@.*cihb.com")) { #forward to self route(3); return; } ##forward to ext gateway route(2); } route[1]{ if (method!="MESSAGE" && method!="SUBSCRIBE" && method!="NOTIFY") { log("XXX: Request not forwarded to sip server!\n"); sl_send_reply("202","Accepted"); return; } t_on_reply("1"); record_route(); if(!lookup("location")) { if (method!="SUBSCRIBE") { log("XXX: only subscribes are processed for user that are not registered\n"); sl_reply_error(); return; } subst_uri('/(.*)@(.*)/\1#\2*jabber.qa.im.hb.com@jabber01.qa.im.hb.com/ig'); if(!t_relay()) { log("XXX: error forwarding jabber01...\n"); sl_reply_error(); return; } else { sl_send_reply("200","OK"); return; } } else { log("XXX: forwarding to the address of record...\n"); if(!t_relay()) { log("XXX: error forwarding to address of record \n"); sl_reply_error(); return; } else { sl_send_reply("200", "OK"); return; } } } # forwarding to external gateway route[2]{ log("XXX: rewriting headers\n"); subst('/^(From:[^@#]*)#([^@]*)\*jabber.qa.im.hb.com@jabber...qa.im.hb.com(.*)/\1@\2\3/ig'); subst('/^(Contact:[^@#]*)#([^@]*)\*jabber.qa.im.hb.com@jabber...qa.im.hb.com(.*)/\1@\2\3/ig'); if(!forward("tls:43.123.141.166:3000")) { log("XXX: Error forwarding to external gateway!\n"); sl_reply_error(); } else { sl_send_reply("200", "Accepted"); }; } # forwarding to SELF route[3]{ log("XXX: rewriting headers\n"); subst('/^(From:[^@#]*)#([^@]*)\*jabber.qa.im.hb.com@jabber...qa.im.hb.com(.*)/\1@\2\3/ig'); subst('/^(Contact:[^@#]*)#([^@]*)\*jabber.qa.im.hb.com@jabber...qa.im.hb.com(.*)/\1@\2\3/ig'); log("XXX: about to forward to self\n"); if(!forward("tcp:sip.qa.hbex.com:5061")) { log("XXX: Error forwarding to self!\n"); sl_reply_error(); } else { sl_send_reply("200", "Accepted"); }; } # this is executed for replies onreply_route[1]{ subst('/^(To:[^@#]*)#([^@]*)\*jabber.qa.im.hb.com@jabber...qa.im.hb.com(.*)/\1@\2\3/ig'); subst('/^(Contact:[^@#]*)#([^@]*)\*jabber.qa.im.hb.com@jabber...qa.im.hb.com(.*)/\1@\2\3/ig'); route(2); }

2 1

[Users] TLS : Please help for TLS concept
by Ferianto siregar 29 Sep '06

29 Sep '06

Dear all, Thank you very much for all your time to read this message. All, I now have problem with TLS security concept. I have tried to read the tutorial (tls.html) from www.openser.org. I have read it many times. But, I still confused to undestand it. Would you mind help me, Please? Here are my questions: 1. How can we evidence that the communication that using TLS is secure communication? (Real proof) 2. When using TLS, how can we know that the communication has been eavesdropped? 3. Is TLS just use port 5061? If yes? How can TLS can protect the communication from sniffer who knows TLS using port 5061? 4. What kind of type encryption that TLS use? 5. What sofware that we can use to test for making a call in opeser with TLS support? I have tried minisip, but I always get error message..Please... I realize that my undestanding in reading TLS tutorial is weak. I do hope, anybody can help me, Please... Thank you very much, Regards, Ferianto --------------------------------- Stay in the know. Pulse on the new Yahoo.com. Check it out.

2 1

[Serusers] SER MySQL Support
by Holger Kinkelin 29 Sep '06

29 Sep '06

Hello everybody! I want to use the mySQL-support for SER but I've got a problem: When I start SER I get the following error DrScheme@dell:/usr/local/etc/ser> /usr/local/sbin/ser 0(7443) ERROR: load_module: could not open module </usr/local/lib/ser/modules/mysql.so>: libmysqlclient.so.12: cannot open shared object file: No such file or directory 0(7443) parse error (18,13-14): failed to load module ERROR: bad config file (1 errors) When I get this right, the libmysqlclient.so-file is missing. I looked for the file and indeed, it's not on the box. I don't know where to get that file from and where to put it, when I have it. Could please someone tell me? Incase you need to know what I did up to now: - I installed mySQL (4.1.13-3) using YaST on my Suse 10.0 box. The mySQL server is running happily - I installed SER 0.9.6 from a tar.gz package, run the script for mySQL-Tables and I use the following config-file # $Id: hello-world.cfg 46 2006-01-25 04:21:30Z /CN=Paul Hazlett/emailAddress=paul(a)onsip.org $ debug=3 fork=no log_stderror=yes listen=192.168.0.31 # put your server IP address here port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" # mysql-support fifo_db_url="mysql://ser:heslo@localhost/ser" # --- # mysql-support 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" # mysql-support loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" # --- modparam("usrloc", "db_mode", 0) modparam("rr", "enable_full_lr", 1) # mysql-support modparam("auth_db|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") # --- route { # ------------------------------------------------------------------------ # Sanity Check Section # ------------------------------------------------------------------------ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ------------------------------------------------------------------------ # Record Route Section # ------------------------------------------------------------------------ if (method!="REGISTER") { record_route(); }; # ------------------------------------------------------------------------ # Loose Route Section # ------------------------------------------------------------------------ if (loose_route()) { route(1); break; }; # ------------------------------------------------------------------------ # Call Type Processing Section # ------------------------------------------------------------------------ if (uri!=myself) { route(1); break; }; if (method=="ACK") { route(1); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } route[1] { # ------------------------------------------------------------------------ # Default Message Handler # ------------------------------------------------------------------------ if (!t_relay()) { sl_reply_error(); }; } route[2] { # ------------------------------------------------------------------------ # REGISTER Message Handler # ------------------------------------------------------------------------ # mysql support if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; # --- if (!save("location")) { sl_reply_error(); }; } Regards, Holger

4 7

[Users] What to do when the ip address of the phone changes?
by Ronald Wiplinger 29 Sep '06

29 Sep '06

When a phone registers it has an expire time of 3600 sec. The user just turned off the phone and plugged it into another network. The server does not let register anymore, since it is already registered. Need to wait 1300 seconds for next register. Is there a way (openserctl xxxxxx) to release the phone ? I don't want to remove the phone from the database!!! bye Ronald

2 1

Re: [Users] thought exercise
by Mark Kent 29 Sep '06

29 Sep '06

>> Probably OPENSER-ONE does not recognize the BYE as targeted to itself Why would a BYE be targeted at a middle-man? I acknowledge that I don't check for this in my BYE handler :-) I did add this as a debug aid: if (is_uri_host_local()) { xlog("L_ERR", "relay to myself? $rm $ru $si $fu\n"); } in my stateful relay handler: route[1] { if (is_uri_host_local()) { xlog("L_ERR", "relay to myself? $rm $ru $si $fu\n"); } t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" && isflagset(6)) { unforce_rtp_proxy(); } sl_reply_error(); } } but, obviously without understanding, I changed that to: route[1] { if (is_uri_host_local()) { xlog("L_ERR", "relay to myself? $rm $ru $si $fu\n"); sl_send_reply("200", "OK"); } else { t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" && isflagset(6)) { unforce_rtp_proxy(); } sl_reply_error(); } } } and that seems to snuff this out, but I also have the feeling that I'm just covering up for a flaw elsewhere. Thanks, -mark

2 3

[Users] Problem with module dispatcher
by Jose Gil Navarrete 29 Sep '06

29 Sep '06

Hi all I have implemented the load balance in mode statefull, I obtained the next messages: phone1 dispatcher sipserver phone2 --------INVITE--------> ----INVITE-----------> ----INVITE--------> <----100 TRYING---- <---100 TRYING---- <---100 TRYING---- <----180 RINGING-- <---180 RINGING--- <---180 RINGING--- <----200 OK---------- <----- 200 OK-------- <---200 OK-------- -------- ACK --------> --------- ACK --------> You can see that the sipserver never sends ACK message to video_phone2. Is this correct? Then I try the next, in the sipserver: if (method =="ACK"){ if (!t_relay()) { sl_reply_error(); }; return; } But the sipserver doesn't send the message ACK at the phone2? Does some body know what it's wrong? --------------------------------- Do You Yahoo!? La mejor conexión a Internet y 2GB extra a tu correo por $100 al mes. http://net.yahoo.com.mx

2 1

[Users] How to get debug print out?
by Steve Xu 29 Sep '06

29 Sep '06

Hi folks, I setup "debug=7" in openser.cfg, however I still cannot get any detail debug print out in /var/log/messages. Here is how I launch the openser: openserctl start There is a logic errors in openser.cfg and I try to trace down the call step by step, how can I do it? Thanks for your time to read my question, I am really appreciated on it. Steve

3 3

[Serusers] just start SER
by vijay tiwari 28 Sep '06

28 Sep '06

hi all today i have install ser on my Linux machine Linux bishthost.bisht 2.6.11-1.1369_FC4 #1 Thu Jun 2 22:55:56 EDT 2005 i686 i686 i386 GNU/Linux and its install everthing is ok and i found two file in /usr/local/etc/ser dictionary.ser ser.cfg please tell me can i start change in ser.cfg or take backup of this file and start working on mail ser.cfg and how can i used hello word ser.cfg file just start copy paste what can i do please guide me, which is best for me thanks vijay --------------------------------- All-new Yahoo! Mail - Fire up a more powerful email and get things done faster.

1 0

[Users] [openser] Displaying Caller-ID
by Tracy Lofton 28 Sep '06

28 Sep '06

Hi, This is probably a simply question for some of you, but is there a way to get openser to display the Caller-ID (telephone number) to the calling party's phone display screen, rather than displaying Annonymous? If so, how would I be able to do this? Thanks, Tracy _________________________________________________________________ Try the new Live Search today! http://imagine-windowslive.com/minisites/searchlaunch/?locale=en-us&FORM=WL…

1 0

[Users] "BUG: io_watch_del" logs
by Federico Giannici 28 Sep '06

28 Sep '06

I have noticed that the following errors began to appear in the logs when OpenSER (1.1.0 under OpenBSD) is closed: Sep 27 17:53:07 eowyn OpenSER[29102]: BUG: io_watch_del: invalid fd 32, not in [0, 0) Sep 27 17:53:07 eowyn OpenSER[29102]: BUG: io_watch_del: invalid fd 33, not in [0, 0) Sep 27 17:53:07 eowyn OpenSER[29102]: BUG: io_watch_del: invalid fd 34, not in [0, 0) Sep 27 17:53:07 eowyn OpenSER[29102]: BUG: io_watch_del: invalid fd 35, not in [0, 0) Sep 27 17:53:07 eowyn OpenSER[29102]: BUG: io_watch_del: invalid fd 36, not in [0, 0) It says "BUG", so is it serious? Bye. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

2 3

[Users] Can I avoid password openser.cfg ?
by Ronald Wiplinger 28 Sep '06

28 Sep '06

How can I avoid details in the openser.cfg? modparam("usrloc|auth_db","db_url","mysql://openser:openserrw@localhost/openser") bye Ronald

3 6

[Users] New Rules: no line wrapping
by Mark Kent 28 Sep '06

28 Sep '06

I'ld like to make a plea to everyone on this list: Please make sure that your mailer doesn't fold long lines in your outbound email. There are certain things that should be sent verbatim (although possibly "anonymized") for example, ngrep output and openser.cfg snippets. I hope I don't send mangled email... let me know if I do, but please make sure it's not your mail reader that is doing the mangling. Thanks, -mark

1 0

[Users] CPL module
by Nicolas Palix 28 Sep '06

28 Sep '06

Hi, I have installed openser and its CPL module and I use CPLEd to write and upload service. I have some problem with the timeout attribut of proxy XML node. It's seem to be not taken into account and the "no answer" node is interpreted about 2 min later. I set the timeout in the CPL script at 5 sec. After re-reading the CPL module documentation and check the cpl_proxy.h code, I try to set the "timer_avp" param without success. Can I have some information on how AVP and timer works ? What I should configured to have CPL enable installation of OpenSER ? Best regards -- Nicolas Palix http://phoenix.labri.fr/people/palix/

2 4

Re [serusers] Cannot Access Serweb
by Tracy Lofton 28 Sep '06

28 Sep '06

Hi All, I followed the intall directions completely and installed both runkit and classkit. I am using php 5.1.1 and when I go to http://myserver/serweb/html/admin/index.php, I get the following message: Function aggregate_methods() doesn't exists. Try install Classkit extension. http://pecl.php.net/package/classkit I can see the module sitting in the diretory as noted below: /usr/lib/php/modules/classkit.so php.ini file ; Directory in which the loadable extensions (modules) reside. extension_dir = "/usr/lib/php/modules" extension=classkit.so extension=runkit.so [runkit] runkit.internal_override=1 Does anyone have any idea of why the classkit module fails to load? Thanks for any help Tracy _________________________________________________________________ Get today's hot entertainment gossip http://movies.msn.com/movies/hotgossip

1 0

[Users] dbtext script problem
by Lindsey Hans Joseph Lao 28 Sep '06

28 Sep '06

Hello, I have a problem with sc.dbtext script. I understand that it is used to create text file for dbtext to store information such as location, address, etc. I downloaded from Openser CVS the sc.dbtext script and replaced my current openserctl file with it. Unfortunately, it still failed to run: Here's the error: /usr/local/sbin/openserctl: 463: Syntax error: Bad substitution Does anyone have an idea on how to resolve this? Thank you. Hans

2 4

[Users] dbtext openser
by Lindsey Hans Joseph Lao 28 Sep '06

28 Sep '06

Hello, I'm planning to use DBTEXT instead of mysql. I found a forum somewhere in the Internet that a script, serctl dbtext [dbtext path] I believe. Is this supported in OpenSER 1.1.0? Because when I tried to invoke this particular command, nothing happens. Thanks!

3 3

Re: [Users] Caller ID & Time of Day
by John Peters 28 Sep '06

28 Sep '06

Try configuring this on the endpoint. For instance, on Sipura ATAs you might set the following parameters if the location were using Eastern Standard Time: Time_Zone "GMT-05:00" ; Primary_NTP_Server "time-b.nist.gov" ; We just rolled out service, and a lot of users are complaining about the > time displayed on their phones being incorrect. Exactly 2 hours behind. > > The switch which we receive all of our inbound calls from is 2 hours > behind > us. Is it possible to adjust the time in the Caller ID so that their > phones > will display properly? >

2 1

[Users] trusted IP with accounting
by ram 28 Sep '06

28 Sep '06

Hi all iam trying to deploy trusted IP and Accounting so iam bit confused can some one clarify me how can i achive this PSTN--OpenerSER(myserver)------------------Customer(Asterisk or ser or any predictive Dialers)---their clients 1. how can i give access to the customer to call using his IP, and he can creat internal Number of users 2. how can i account to his IP address billing in Acc 3. how can i forward to DID from myserver to his server blindly 4. how can i bill that DID billing also, for incoming. any suggestiong will be great Ram

2 7

Re: [Users] AVPOPS error
by Bogdan-Andrei Iancu 28 Sep '06

28 Sep '06

ok -super :) bogdan Jose Silva wrote: > Thanks!!! your solution worked out fine. > > Thank you very mutch. > Jose Silva > > 2006/9/28, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro>: > >> maybe your query is about 1024 char len, bot exceeding it all the >> time.... >> >> try to increase the buffer size - see modules/avpops/avpops_impl.c line >> 66 for >> #define AVP_PRINTBUF_SIZE 1024 >> >> note you need to recompile and re-install. >> >> regards, >> bogdan >> >> >> Jose Silva wrote: >> >> > Yes it's a very long query! >> > But the query works for the first messages, but after a while it fails >> > to work. >> > How can I correct this? >> > >> > thanks, >> > Jose Silva >> > >> > >> > 2006/9/28, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro>: >> > >> >> Hi Jose, >> >> >> >> maybe you are trying to form a very long query...by default, the used >> >> buffer to build the query has a length of 1024 chars... >> >> is this the case?? >> >> >> >> regards, >> >> bogdan >> >> >> >> Jose Silva wrote: >> >> >> >> > Hi I'm using avpops module to do some DB queries, everything >> goes fine >> >> > and all the queries are executed correctly for the first >> messages, but >> >> > after receiving allot of messages what I get on my log for one >> of the >> >> > queries (just one of the queries, the other ones are executed >> >> > correctly) is the following. >> >> > >> >> > (...) >> >> > xl_printf: no more space for text [175] >> >> > 3(4820) xl_printf: buffer overflow -- increase the buffer size... >> >> > 3(4820) avpops:ops_dbquery_avps: error - cannot print the query >> >> > (...) >> >> > >> >> > How can I solve this problem? >> >> > >> >> > Thanks in advance, >> >> > Jose Silva >> >> > >> >> > _______________________________________________ >> >> > Users mailing list >> >> > Users(a)openser.org >> >> > http://openser.org/cgi-bin/mailman/listinfo/users >> >> > >> >> >> >> >> >>

1 0

[Users] Mysql error
by Nick De Cristofaro 28 Sep '06

28 Sep '06

Hey Im just trying to add users after doing the basic install. And I get this? : "ERROR 1062 (23000) at line 1: Duplicate entry '' for key 2 -e \E[37;31mERROR: introducing the new user 'test' to the database failed" Nick

2 1

[Users] AVPOPS error
by Jose Silva 28 Sep '06

28 Sep '06

Hi I'm using avpops module to do some DB queries, everything goes fine and all the queries are executed correctly for the first messages, but after receiving allot of messages what I get on my log for one of the queries (just one of the queries, the other ones are executed correctly) is the following. (...) xl_printf: no more space for text [175] 3(4820) xl_printf: buffer overflow -- increase the buffer size... 3(4820) avpops:ops_dbquery_avps: error - cannot print the query (...) How can I solve this problem? Thanks in advance, Jose Silva

2 3

[Users] CPL module
by Nicolas Palix 28 Sep '06

28 Sep '06

Hi, I have installed openser and its CPL module and I use CPLEd to write and upload service. I have some problem with the timeout attribut of proxy XML node. It's seem to be not taken into account and the "no answer" node is interpreted about 2 min later. I set the timeout in the CPL script at 5 sec. After re-reading the CPL module documentation and check the cpl_proxy.h code, I try to set the "timer_avp" param without success. Can I have some information on how AVP and timer works ? What I should configured to have CPL enable installation of OpenSER ? Best regards -- Nicolas Palix http://phoenix.labri.fr/people/palix/

1 0

[Users] rtpproxy still not working for me
by Ronald Wiplinger 28 Sep '06

28 Sep '06

Status: openser compiled, including mysql openser working with mysql (registering) openser server has two ethernet ports with one public and one private IP rtpproxy is running without any parameter 7311 is eyebeam on local desktop 7312 is gizmo on the same local desktop 7111 is a grandstream remote, behind a nat Openser adminstrator shows all three phones as registered with following sip addresses: sip:7111@192.168.10.91 sip:7311@192.168.250.108:42800;rinstance=f17a00051f63a625 sip:17476063396@proxy01.sipphone.com:5060 <=== that should be 7312(a)192.168.250.108 Internal calls: call from 7312 to 7311 works. call from 7311 to 7312 ends up at a voicemail box (most likely gizmo problem) call from 7311 to 17476063396(a)proxy01.sipphone.com ends up at the voicemail box External calls: call from 7311 (local phone) 7111 (remote phone behind NAT) does not work, see the ngrep info below. Looking at the output I feel that the nathelper is not working. As I understand the documents, the address should be replaced with an public IP address, but it uses my servers internal ethernet port (192.168.250.41) 5.32.86.166 in alt1 is the IP address of my Hamachi network. I have no idea, why this one shows up here. # U 2006/09/28 08:56:06.482374 192.168.250.108:42800 -> 192.168.250.41:5060 INVITE sip:7111@192.168.250.41 SIP/2.0. Via: SIP/2.0/UDP 192.168.250.108:42800;branch=z9hG4bK-d87543-a177451df022e84f-1--d87543-;rport. Max-Forwards: 70. Contact: <sip:7311@192.168.250.108:42800>. To: "7111"<sip:7111@192.168.250.41>. From: "Ronald-taipei @ video "<sip:7311@192.168.250.41>;tag=13766900. Call-ID: ZGIzNjlhYmIwODBmOTAzMDJiMDU1Zjc0ZTZhNDM0MzE.. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. User-Agent: eyeBeam release 1005g stamp 32576. Content-Length: 809. . v=0. o=- 0 2 IN IP4 192.168.250.108. s=CounterPath eyeBeam 1.5. c=IN IP4 192.168.250.108. t=0 0. m=audio 59382 RTP/AVP 107 100 106 6 0 105 18 3 5 101. a=alt:1 2 : eE24SyNT p1rs0KkA 5.32.86.166 59382. a=alt:2 1 : eKP8Z+Ou +E2ejbdN 192.168.250.108 59382. a=fmtp:18 annexb=yes. a=fmtp:101 0-15. a=rtpmap:107 BV32/16000. a=rtpmap:100 SPEEX/16000. a=rtpmap:106 SPEEX-FEC/16000. a=rtpmap:105 SPEEX-FEC/8000. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. a=sendrecv. a=x-rtp-session-id:B8DA69C497FC40B2AACE87E3E86D914A. m=video 46168 RTP/AVP 125. a=alt:1 2 : Ac/XN/Nt wQExOeCB 5.32.86.166 46168. a=alt:2 1 : ZKv1sx6w JY5CEQdN 192.168.250.108 46168. a=fmtp:125 profile-level-id=42e015; max-br=4000; max-mbps=19800. a=rtpmap:125 H264/90000. a=sendrecv. a=x-rtp-session-id:AD24C300DC234CCFA115E5F0B36CDEAB. # U 2006/09/28 08:56:06.483471 192.168.250.41:5060 -> 192.168.250.108:42800 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 192.168.250.108:42800;branch=z9hG4bK-d87543-a177451df022e84f-1--d87543-;rport=42800. To: "7111"<sip:7111@192.168.250.41>. From: "Ronald-taipei @ video "<sip:7311@192.168.250.41>;tag=13766900. Call-ID: ZGIzNjlhYmIwODBmOTAzMDJiMDU1Zjc0ZTZhNDM0MzE.. CSeq: 1 INVITE. Server: OpenSer (1.1.0-notls (x86_64/linux)). Content-Length: 0. Warning: 392 192.168.250.41:5060 "Noisy feedback tells: pid=8787 req_src_ip=192.168.250.108 req_src_port=42800 in_uri=sip:7111@192.168.250.41 out_uri=sip:7111@192.168.10.91 via_cnt==1". . # U 2006/09/28 08:56:30.829647 192.168.250.108:42800 -> 192.168.250.41:5060 . . # U 2006/09/28 08:56:36.004956 192.168.250.41:5060 -> 192.168.250.108:42800 SIP/2.0 408 Request Timeout. Via: SIP/2.0/UDP 192.168.250.108:42800;branch=z9hG4bK-d87543-a177451df022e84f-1--d87543-;rport=42800. To: "7111"<sip:7111@192.168.250.41>;tag=fa997f81440371de71ab448ebdb9af56-1877. From: "Ronald-taipei @ video "<sip:7311@192.168.250.41>;tag=13766900. Call-ID: ZGIzNjlhYmIwODBmOTAzMDJiMDU1Zjc0ZTZhNDM0MzE.. CSeq: 1 INVITE. Server: OpenSer (1.1.0-notls (x86_64/linux)). Content-Length: 0. Warning: 392 192.168.250.41:5060 "Noisy feedback tells: pid=8791 req_src_ip=192.168.250.108 req_src_port=42800 in_uri=sip:7111@192.168.250.41 out_uri=sip:7111@192.168.10.91 via_cnt==0". . # /var/log/messages shows: Sep 28 09:00:27 taipei /usr/local/sbin/openser[8787]: ERROR: force_rtp_proxy2: no available proxies Sep 28 09:00:40 taipei /usr/local/sbin/openser[8787]: ERROR: unforce_rtp_proxy: no available proxies Sep 28 09:00:40 taipei /usr/local/sbin/openser[8787]: ERROR: unforce_rtp_proxy: no available proxies Where is my mistake? bye Ronald

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[Users] proxy authorization required sent to the wrong port
by Benko 28 Sep '06

28 Sep '06

Hello! I've tried to modify my setup to use rtpproxy, following the guide posted at http://siprouter.onsip.org/doc/gettingstarted/, most of it works fine, but i have a small but annoying problem: When i receive a invite from a UA, i verify the identity with proxy_authorize. However, the proxy-authenticate-request is not sent to the originating sip-port of the UA but to 5060 for some reason. In my old setup this worked flawless, the proxy authorize request is sent to the correct external port of the phone. To me it seems there is no difference in this part of the route-plan, so how does this happen? I have attached both config-files and a ngrep of the requests(*.old is the working setup, *.rtpproxy is the new setup that doesn't work), maybe someone can help me with this? Thanks in advance! Regards Christian

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Re: [Users] OPENSER 1.1.0 equivalent version for SER
by Bogdan-Andrei Iancu 28 Sep '06

28 Sep '06

Hi Bill, Bill Zhang wrote: > Hi, > > I just sew the new face of www.iptel.org <http://www.iptel.org/> and > see they start to move things forward now. Got some questions about > SER versus OPENSER: > > 1. What is the OPENSER 1.1.0 equivalent version for SER? 0.10.x or > 0.9.x? Is there a lot of difference? > OpenSER 1.1.0 is the latest stable release. AFAIK, last stable release for SER is 0.9.x for more than one year ago.... ..yes there are a lot of differences... > 1. SER has a SEMS server, does OPENSER has something equivalent? > SEMS can be also used with OpenSER > > 1. When will OPENSER have the new Presence module? I know it’s > soon, but how soon? > in a matter of weeks - are you interested in testing it? Regards, Bogdan

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[Users] Openser Administrator misses /var/log/openser
by Ronald Wiplinger 28 Sep '06

28 Sep '06

Openser Administrator says: No such file or directory - /var/log/openser - I agree with that! Where do I set it? tail /var/log/messages Sep 27 19:59:41 taipei /usr/local/sbin/openser[13042]: ERROR:registrar:update_contacts: invalid cseq for aor <7111> Sep 27 19:59:41 taipei /usr/local/sbin/openser[13044]: ERROR:registrar:update_contacts: invalid cseq for aor <7211> Sep 27 19:59:44 taipei /usr/local/sbin/openser[13044]: ERROR:registrar:update_contacts: invalid cseq for aor <7111> Sep 27 19:59:47 taipei /usr/local/sbin/openser[13044]: ERROR:registrar:update_contacts: invalid cseq for aor <7211> Sep 27 19:59:50 taipei /usr/local/sbin/openser[13042]: ERROR:registrar:update_contacts: invalid cseq for aor <7111> Sep 27 19:59:59 taipei /usr/local/sbin/openser[13044]: ERROR:registrar:update_contacts: invalid cseq for aor <7113> Sep 27 19:59:59 taipei /usr/local/sbin/openser[13042]: ERROR:registrar:update_contacts: invalid cseq for aor <7211> Sep 27 20:00:02 taipei /usr/local/sbin/openser[13044]: ERROR:registrar:update_contacts: invalid cseq for aor <7111> Sep 27 20:00:03 taipei /usr/local/sbin/openser[13042]: ERROR:registrar:update_contacts: invalid cseq for aor <7113> Sep 27 20:00:07 taipei /usr/local/sbin/openser[13044]: ERROR:registrar:update_contacts: invalid cseq for aor <7113> BTW, what do these error messages mean? bye Ronald

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[Users] t_on_failure does not work for me
by Jeremie Le Hen 28 Sep '06

28 Sep '06

Hi list, my configuration file contains the following block: % if (is_method("INVITE")){ % force_rtp_proxy(""); % t_on_failure("1"); % append_hf("P-hint: RTP session ports reserved by rtpproxy\r\n"); % }; If I dump the SIP dialog on the network, I can indeed see the "P-hint" header. Then, the configuration file contains: % failure_route[1] { % if (!(status == "407")) { % unforce_rtp_proxy(); % append_hf("P-hint: RTP session dropped\r\n"); % } else { % append_hf("P-hint: Proxy authentication required, don't drop RTP session\r\n"); % } % } Unfortunately, when a 407 reply code comes back from the registrar, the outgoing SIP packet doesn't have any "P-hint" header. I didn't manage to test the other cases, unfortunately. Did I miss something in the configuration file ? Thank you. Regards, -- Jeremie Le Hen < jeremie at le-hen dot org >< ttz at chchile dot org >

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RE: [Serusers] Implementing call fwd
by Kamal.Mann＠t-systems.com 28 Sep '06

28 Sep '06

Hi all Please help out ________________________________ From: Mann, Kamal Sent: Tuesday, September 26, 2006 5:46 PM To: 'serusers(a)lists.iptel.org' Subject: RE: [Serusers] Implementing call fwd Hi all I am implementing Blind call forwarding with 0.9.6 SER version. But with current configuration it's not working. Please help on same. I am enclosing small part of usr_prefrences table & my ser.cfg.

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Re: [Serusers] Implementing call fwd
by Greger V. Teigre 28 Sep '06

28 Sep '06

Please copy the list. It depends on what you want to achieve. Blind forwarding is to me the following: - Check if this is a local user (myself) - Check if this is an alias (lookup("aliases")) - Check if the AOR is registered (lookup("location")) - Check if the user has blind call forwarding - If no, relay to looked up location - If yes, rewrite ruri. (Then what you do depends on your setup complexity, either you forward to PSTN, so just rewrite to GW and relay, or you allow recursive forwarding. Then you need to either relay to yourself or call a route that runs the above described process. Or simpler, you can do as you do by duplicating the lookups, now with the new ruri. Then you allow only one step of forwarding, not recursive.) g-) Kamal.Mann(a)t-systems.com wrote: > > I think so, because we need to check is r-uri (to which call get > forwarded) is registered or not. > > I am newbie if anything wrong please suggests me. > > > > Thanks in anticipation > > Kamal Mann > > ** ** > > *From:* Greger V. Teigre [mailto:greger@teigre.com] > *Sent:* Wednesday, September 27, 2006 8:19 PM > *To:* Mann, Kamal > *Cc:* serusers(a)lists.iptel.org > *Subject:* Re: [Serusers] Implementing call fwd > > > > You do lookup("aliases") and lookup("location") AFTER you have changed > the uri through avp_pushto from usr_preferences. > Is this really what you want? > g-) > > Kamal.Mann(a)t-systems.com <mailto:Kamal.Mann@t-systems.com> wrote: > > > Hi All > I want to setup Blind call forwarding. Like if a uri - mann(a)iptel.org <mailto:mann@iptel.org> > is not registered with SER but this user had already configured in ser > database table usr_prefrences that his call should be forwarde to > another uri kamal(a)iptel.org <mailto:kamal@iptel.org>. For that my usr_prefrence table is like > > > >> USR_PREFRENCES table >> >> +------+----------+-----------+-----------+---------------------+------ >> > + > >> ---------------+ >> >> | uuid | username | domain | attribute | value | type >> > | > >> modified | >> >> +------+----------+-----------+-----------+---------------------+------ >> > + > >> ----------------+ >> >> | | bhaskar | debian | callfwd | kamal(a)iptel.org <mailto:kamal@iptel.org> >> | 0 | 20060926165852 | >> >> | | mann | iptel.org | callfwd | sip:kamal@iptel.org <mailto:sip:kamal@iptel.org> | 0 >> > | > >> 20060926150443 | >> >> | | prateek | iptel.org | callfwd | kamal(a)iptel.org <mailto:kamal@iptel.org> | 0 >> > | > >> 20060926170701 | >> >> +------+----------+-----------+-----------+---------------------+------ >> > + > >> ----------------+ >> > > > Is this entry is correct?? Is there any flaw in my ser.cfg?? > Any help would be appreciated. > Regards > Kamal Mann > -----Original Message----- > From: Steve Blair [mailto:blairs@isc.upenn.edu] > Sent: Wednesday, September 27, 2006 2:26 PM > To: Mann, Kamal > Cc: serusers(a)lists.iptel.org <mailto:serusers@lists.iptel.org> > Subject: Re: [Serusers] Implementing call fwd > > > > Does your config file relay REFER messages? It needs to in order to > perform the transfer. > > Kamal.Mann(a)t-systems.com <mailto:Kamal.Mann@t-systems.com> wrote: > > >> Hi all >> >> Please help out >> >> >> > ------------------------------------------------------------------------ > >> *From:* Mann, Kamal >> *Sent:* Tuesday, September 26, 2006 5:46 PM >> *To:* 'serusers(a)lists.iptel.org <mailto:serusers@lists.iptel.org>' >> *Subject:* RE: [Serusers] Implementing call fwd >> >> Hi all >> >> I am implementing Blind call forwarding with 0.9.6 SER version. But >> with current configuration it's not working. Please help on same. >> >> I am enclosing small part of usr_prefrences table & my ser.cfg. >> >> ----------------------------------------------------------------------- >> > - > >> _______________________________________________ >> Serusers mailing list >> Serusers(a)lists.iptel.org <mailto:Serusers@lists.iptel.org> >> http://lists.iptel.org/mailman/listinfo/serusers >> >> >> > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org <mailto:Serusers@lists.iptel.org> > http://lists.iptel.org/mailman/listinfo/serusers > > >

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[Users] Mediaproxy / proxydispatcher: couldn't retrieve DNS SRV record
by kjcsb 28 Sep '06

28 Sep '06

I am getting the following error in the log file: Sep 28 06:24:25 beta proxydispatcher[24129]: error: couldn't retrieve DNS SRV record '_mediaproxy._tcp.domain.co.nz': (111, 'Connection refused') However I think the DNS records are OK. dig SRV _mediaproxy._tcp.domain.co.nz ; <<>> DiG 9.2.3 <<>> SRV _mediaproxy._tcp.domain.co.nz ;; global options: printcmd ;; Got answer: ;; ->>HEADER<<- opcode: QUERY, status: NOERROR, id: 63894 ;; flags: qr rd ra; QUERY: 1, ANSWER: 3, AUTHORITY: 2, ADDITIONAL: 3 ;; QUESTION SECTION: ;_mediaproxy._tcp.domain.co.nz. IN SRV ;; ANSWER SECTION: _mediaproxy._tcp.domain.co.nz. 86100 IN SRV 0 100 25060 c-mp-03.domain.co.nz. _mediaproxy._tcp.domain.co.nz. 86100 IN SRV 10 50 25060 c-mp-01.domain.co.nz. _mediaproxy._tcp.domain.co.nz. 86100 IN SRV 10 50 25060 c-mp-02.domain.co.nz. ;; AUTHORITY SECTION: domain.co.nz. 258929 IN NS ns2.widge.net. domain.co.nz. 258929 IN NS ns1.widge.net. ;; ADDITIONAL SECTION: c-mp-03.domain.co.nz. 86129 IN A 60.234.nnn.nnn ns1.widge.net. 170576 IN A 82.165.160.211 ns2.widge.net. 170576 IN A 82.165.163.115 ;; Query time: 83 msec ;; SERVER: 64.62.196.100#53(64.62.196.100) ;; WHEN: Thu Sep 28 06:28:07 2006 ;; MSG SIZE rcvd: 276 On the target server (c-mp-03.domain.co.nz) there's something getting through to to mediaproxy as shown in the log file: Sep 28 06:26:21 domain01 mediaproxy[16128]: command status Sep 28 06:26:21 domain01 mediaproxy[16128]: command execution time: 0.54 ms Sep 28 06:26:41 domain01 mediaproxy[16128]: command status Sep 28 06:26:41 domain01 mediaproxy[16128]: command execution time: 0.55 ms Could anyone advise how to resolve this issue? Regards Cameron

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[Serusers] unhandled SIP state transition
by jorain 28 Sep '06

28 Sep '06

Hi all, When i call from pstn to sip phone, and hang up it before someone pick up the sip phone, then my sems will show this error and immediately end the sems. Error: changeSIPState(IsdnCall.cpp:339):unhandled SIP state transition(inbound) from (DISCONNECTED|DISCONNECTED) to (CONNECTED|DISCONNECTED) What could be the problem? Any help is appreciated. Thanks Regards, jorain

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[Users] OPENSER 1.1.0 equivalent version for SER
by Bill Zhang 27 Sep '06

27 Sep '06

Hi, I just sew the new face of www.iptel.org <http://www.iptel.org/> and see they start to move things forward now. Got some questions about SER versus OPENSER: 1. What is the OPENSER 1.1.0 equivalent version for SER? 0.10.x or 0.9.x? Is there a lot of difference? 2. SER has a SEMS server, does OPENSER has something equivalent? 3. When will OPENSER have the new Presence module? I know it's soon, but how soon? Thanks in advance. Best Regards, Bill

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[Users] Re: Problem handling multiple calls in FreeBSD 6.1-RELEASE
by Mark Kent 27 Sep '06

27 Sep '06

> >The problem is easy to reproduce: just reducing the children parameter > >to 1, and openser will not handle a second call (obviously that doesn't > >happen in Debian, using the same configuration for openser). I just tested this on FreeBSD 6.1-RELEASE, using openser 1.1.0, and I could place two simultaneous calls without any issues or delays. As you may know, freebsd has a few different thread libraries. I made sure all my compiles, for everything in and around openser, was compiled with -pthread on the command line. -mark

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Re: [Users] OpenserAdmin Can't Login
by John Shellhouse 27 Sep '06

27 Sep '06

Hi, First, let me thank you for your assistance and time with my problem. But I am not sure what UR configuration of openser you are speaking of. in the config/database.yml I have indicated this info: development: adapter: mysql database: openser username: openser password: openserrw host: localhost and did the same for production in database.yml. There was not much to do. what other config could it be. How did you do your UR configuration of openser? Thanks, jwshel >From: "raviprakash sunkara" <sunkara.raviprakash.feb14(a)gmail.com> >To: "John Shellhouse" <jwshel(a)hotmail.com> >Subject: Re: [Users] OpenserAdmin Can't Login >Date: Wed, 27 Sep 2006 18:27:54 +0530 > >Username is openser >passwors is openserrw > >My Guess is that Ur configuaration of openser admin is not Clear > > > >On 9/27/06, John Shellhouse <jwshel(a)hotmail.com> wrote: >> >>Hi, >> >>I do not understand what you are describing, can you be more specific? I >>did try to login at: >>http://localhost:3000/login/login, but password and username not accepted. >> >>jwshel >> >> >> >From: "raviprakash sunkara" <sunkara.raviprakash.feb14(a)gmail.com> >> >To: "John Shellhouse" <jwshel(a)hotmail.com> >> >Subject: Re: [Users] OpenserAdmin Can't Login >> >Date: Wed, 27 Sep 2006 14:51:12 +0530 >> > >> >Hello , >> >U can't configured the openser admin 0.3 the Database.dl in that :P >> > >> >Try this ............. >> >http://localhost:3000/login/login >> > >> > >> >On 9/27/06, John Shellhouse <jwshel(a)hotmail.com> wrote: >> >> >> >>Hi, >> >> >> >>Second request, I cannot login to openser Admin. Followed directions, >> >>added >> >>ruby rails, gems, etc. started ./script/server entered default >> >>username:password of openser:openserrw. >> >> >> >>This is the only result: >> >>All login in attempts are logged. Unauthorized access is forbidden. >> >> >> >>Need help by someone, anyone even will be appreciated. >> >> >> >>thanks >> >>jswhel >> >> >> >> >> >> >From: "John Shellhouse" <jwshel(a)hotmail.com> >> >> >To: users(a)openser.org >> >> >Subject: [Users] [openser] OpenserAdmin Login >> >> >Date: Sun, 24 Sep 2006 05:57:01 -0700 >> >> > >> >> >Hi, >> >> > >> >> >I am cannot login to the initial Webpage for OpenserAdmin after >>starting >> >> >./script/server. I put in the default username:password of >> >> >openser:openserrw but I am not allowed access. Any suggestions >>anyone? >> >> > >> >> >Thanks, >> >> > >> >> >jswhel >> >> > >> >> > >> >> > >> >> >_______________________________________________ >> >> >Users mailing list >> >> >Users(a)openser.org >> >> >http://openser.org/cgi-bin/mailman/listinfo/users >> >> >> >> >> >> >> >>_______________________________________________ >> >>Users mailing list >> >>Users(a)openser.org >> >>http://openser.org/cgi-bin/mailman/listinfo/users >> >> >> > >> > >> > >> >-- >> >Thanks and Regards >> >Ravi Prakash Sunkara >> >ravi.sunkara(a)hyperion-tech.com >> >M:+91 9985077535 >> >O:+91 40 23114549 >> >F:+91 40 40208727 >> >ravi.sunkara(a)hyperion-tech.com >> >www.hyperion-tech.com >> >> >> >>_______________________________________________ >>Users mailing list >>Users(a)openser.org >>http://openser.org/cgi-bin/mailman/listinfo/users >> > > > >-- >Thanks and Regards >Ravi Prakash Sunkara >ravi.sunkara(a)hyperion-tech.com >M:+91 9985077535 >O:+91 40 23114549 >F:+91 40 40208727 >ravi.sunkara(a)hyperion-tech.com >www.hyperion-tech.com

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[Users] Caller ID & Time of Day
by Kelvin Williams 27 Sep '06

27 Sep '06

We just rolled out service, and a lot of users are complaining about the time displayed on their phones being incorrect. Exactly 2 hours behind. The switch which we receive all of our inbound calls from is 2 hours behind us. Is it possible to adjust the time in the Caller ID so that their phones will display properly? Kw

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Re: [Users] mysql.so missing
by Ronald Wiplinger 27 Sep '06

27 Sep '06

Largo wrote: > Hi Ronald > > That means U didint compile the > mysql module ! > > Fix it like this : > > - compiling all the packets : > > *make all* > > - make with a prefix : > > *make prefix=/usr/local/ install* > > > - now install the excluded packets, as example here mysql for mysql > database or pgsql for postgresql database support: > > *make modules include_modules=”mysql”* > > *make install include_modules=”mysql”* > Was the same error at the time I compiled it, ... I had to do first: "unset CPU" This made and installed the module mysql.so > ** > > **** > > **** > > - copying the init script to the “/etc/init.d” does not lead to a > result. The init script already in /usr/local/sbin to the init.d > directory as shown in the following line: > > *ln -s /usr/local/sbin/openser /etc/init.d/openser* > Are you sure? shouldn't it be openserctl? I tried ln -s /usr/local/sbin/openserctl /etc/rc.d/openser (SuSE 10.1 has all init files in /etc/rc.d) # /etc/rc.d/openser start Starting OpenSER : ERROR: PID file /var/run/openser.pid does not exist -- OpenSER start failed Where have I lost the pid file? BTW, Largo, you have a nice email address !!! bye Ronald

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[Serusers] Customize SERWeb
by Legos Engineering 27 Sep '06

27 Sep '06

Dear all, I find a way to add a new column in accounting page. So in this way, I think that I can create a new table with my price/prefix and then make a calcul base with the duration of call. But maybe someone knows if this already exist. If not, what's the best way to do something like that ? Help will appreciated. Thanks. Best regards

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[Users] Problem handling multiple calls in FreeBSD 6.1-RELEASE
by Hernán A. Pérez Masci 27 Sep '06

27 Sep '06

Hi there, We've been running openser for a long time on Debian Sarge boxes without problems, but recently we installed it (version 1.0.1) on a FreeBSD 6.1-RELEASE box, and started having problems when testing it even with very low load conditions. The problem is easy to reproduce: just reducing the children parameter to 1, and openser will not handle a second call (obviously that doesn't happen in Debian, using the same configuration for openser). Is this a known issue? I haven't found any message related to it. Regards, Hernan.

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[Users] mysql.so missing
by Ronald Wiplinger 27 Sep '06

27 Sep '06

taipei:/usr/local/etc/openser # openser ERROR: bad config file (1 errors) /var/log/messages shows: Sep 27 19:23:16 taipei openser: ERROR: load_module: could not open module </usr/local/lib64/openser/modules/mysql.so>: /usr/local/lib64/openser/modules/mysql.so: cannot open shared object file: No such file or directory Sep 27 19:23:16 taipei openser: parse error (39,13-14): failed to load module mysql.so exists at /usr/lib64/php5/extensions/mysql.so However, a symlink does not work for that either. I use SuSE 10.1 on a Semperon (Athlon) CPU. I have installed mysql-devel. Besides this, I am confused about the openser.cfg file, which does not match the SER - Getting Started. Is there a better document available? What do I need to install to get the system running with phones behind NAT ? The document suggests many different things (Nathelper, STUN, rtpproxy and mediaproxy). Which one do I need? Is it either rtpproxy or mediaproxy? Is it in addition to STUN or Nathelper???? I also do not really know what I need. My server is on a public IP, with a second Ethernet card for the internal LAN. Where the user has his phone I do not know yet. Maybe on a symetric NAT or asymetric NAT, maybe on a public IP. For all it should work. bye Ronald (obviously confused)

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[Users] openser working with pre-paid and post-paid?
by moacir FERRARI NETO 27 Sep '06

27 Sep '06

Hello guys... It would like that to openser it worked with service of pre-paid end post-paid... I need a "how to" install this service. I saw many diagrams working with B2BUA and I still with many doubts about it. thanks, Moacir. _______________________________________________________ Você quer respostas para suas perguntas? Ou você sabe muito e quer compartilhar seu conhecimento? Experimente o Yahoo! Respostas ! http://br.answers.yahoo.com/

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[Serusers] t_on_retransmission route
by Hendrik Scholz 27 Sep '06

27 Sep '06

Hi! We were poking the idea around of having an optional route that would be invoked upon a retransmission. The fr timers all fire way too late if you'd have to implement some sort of fallback if your peering partner or PSTN gateway died. I magine a routing mechanism much like t_on_failure() / failure_route() to allow something like this: - set transmission timer via AVP / t_set_fr() like - t_on_retransmission(1); - t_relay_to_udp() and then you'd have something like retransmission_route[1] { # select new gateway t_relay_to_udp() } The retransmission route should be invoked if no provisional (i.e. 100 Trying) nor final response was received within the given interval. I'm not sure about how to handle TCP connections, though. A simple TCP ACK might be enough to assume the other side is up and happy to answer our request. This would allow for redirecting traffic without having to wait for a final response (408 timeout). The timer value would be fr_timer - fr_inv_timer starting at the same time as fr_timer itself. Last but not least I'm not sure about the impact of having quite a lot of timers firing after a couple of msecs. If anybody thinks this would make sense I'd give it a try ;) Cheers, Hendrik -- Hendrik Scholz - <hscholz(a)raisdorf.net> - http://www.wormulon.net/ drag me, drop me - treat me like an object

3 3

Re: [Users] OpenserAdmin Can't Login
by John Shellhouse 27 Sep '06

27 Sep '06

Hi, I do not understand what you are describing, can you be more specific? I did try to login at: http://localhost:3000/login/login, but password and username not accepted. jwshel >From: "raviprakash sunkara" <sunkara.raviprakash.feb14(a)gmail.com> >To: "John Shellhouse" <jwshel(a)hotmail.com> >Subject: Re: [Users] OpenserAdmin Can't Login >Date: Wed, 27 Sep 2006 14:51:12 +0530 > >Hello , >U can't configured the openser admin 0.3 the Database.dl in that :P > >Try this ............. >http://localhost:3000/login/login > > >On 9/27/06, John Shellhouse <jwshel(a)hotmail.com> wrote: >> >>Hi, >> >>Second request, I cannot login to openser Admin. Followed directions, >>added >>ruby rails, gems, etc. started ./script/server entered default >>username:password of openser:openserrw. >> >>This is the only result: >>All login in attempts are logged. Unauthorized access is forbidden. >> >>Need help by someone, anyone even will be appreciated. >> >>thanks >>jswhel >> >> >> >From: "John Shellhouse" <jwshel(a)hotmail.com> >> >To: users(a)openser.org >> >Subject: [Users] [openser] OpenserAdmin Login >> >Date: Sun, 24 Sep 2006 05:57:01 -0700 >> > >> >Hi, >> > >> >I am cannot login to the initial Webpage for OpenserAdmin after starting >> >./script/server. I put in the default username:password of >> >openser:openserrw but I am not allowed access. Any suggestions anyone? >> > >> >Thanks, >> > >> >jswhel >> > >> > >> > >> >_______________________________________________ >> >Users mailing list >> >Users(a)openser.org >> >http://openser.org/cgi-bin/mailman/listinfo/users >> >> >> >>_______________________________________________ >>Users mailing list >>Users(a)openser.org >>http://openser.org/cgi-bin/mailman/listinfo/users >> > > > >-- >Thanks and Regards >Ravi Prakash Sunkara >ravi.sunkara(a)hyperion-tech.com >M:+91 9985077535 >O:+91 40 23114549 >F:+91 40 40208727 >ravi.sunkara(a)hyperion-tech.com >www.hyperion-tech.com

1 0

[Serusers] Ser vovida B2BUA
by prabhakar g 27 Sep '06

27 Sep '06

I'm looking for Ser vovida B2BUA & Radius howto, related documentation. Can anybody help me. -- Prabhakar.g Network Engineer Hyderabad-34,India

3 2

[Serusers] Voice recorder?
by Tobias Lindgren 27 Sep '06

27 Sep '06

Hi all, I'm looking for a good voice recorder software (OpenSource pref.) that can record a conversation in a phonecall. Please let me know if you have tested one. Br, Tobias

1 0

[Users] mediaproxy signIn IP address
by Klaus Darilion 27 Sep '06

27 Sep '06

Hi! It looks like there is some restriction of the IP address allowed to signIn to mediaproxy. Can this be configured or is this hardcoded? thanks klaus

1 0

Re: [Serusers] Nathelper Ping OPTIONS
by Greger V. Teigre 27 Sep '06

27 Sep '06

:-D If she really is beautiful, I would take whatever she could give... However, this one doesn't want to talk to you. You need to find some tricks. g-) olivier wrote: > The beautifullest girl of the world can't give you what she doesn't > have ;) > > Thanks anyway, > > Olivier > > Greger V. Teigre a écrit : >> Yes, I saw that. Last time I tried, it worked. I couldn't see what >> possibly is wrong from the output and still can't. Sorry. >> g-) >> >> olivier.taylor wrote: >>> Hi Greger, >>> >>> I have tried to apply the patch to 0.9.6, see the result below :( >>> >>> I have downloaded the last 0.9.6. >>> >>> Olivier >>> >>> Greger V. Teigre a écrit : >>>> The patch should work for 0.9.6. AFAIK, 0.10.x already has the >>>> OPTIONS support. >>>> g-) >>>> >>>> olivier.taylor wrote: >>>>> Hello, >>>>> >>>>> just trying to enable the ping functionality in nathelper to send >>>>> SIP message (i.e. OPTIONS) instead of 4-byte UDP packets >>>>> on SER 0.9.6 and 0.10.0 >>>>> >>>>> as shown below, patch doesn't apply >>>>> Is there another possibility to have this feature? >>>>> >>>>> Olivier >>>>> >>>>> >>>>> patch -p1 < nathelper_ping.patch >>>>> Hmm... Looks like a unified diff to me... >>>>> The text leading up to this was: >>>>> -------------------------- >>>>> |diff -Naur -x CVS nathelper/nathelper.c nathelper_ping/nathelper.c >>>>> |--- nathelper/nathelper.c 2006-08-16 08:45:48.000000000 +0200 >>>>> |+++ nathelper_ping/nathelper.c 2006-08-10 14:54:14.000000000 +0200 >>>>> -------------------------- >>>>> Patching file nathelper.c using Plan A... >>>>> Hunk #1 failed at 118. >>>>> Hunk #2 failed at 172. >>>>> Hunk #3 failed at 193. >>>>> Hunk #4 failed at 212. >>>>> Hunk #5 failed at 239. >>>>> Hunk #6 failed at 257. >>>>> Hunk #7 failed at 267. >>>>> Hunk #8 failed at 276. >>>>> Hunk #9 failed at 320. >>>>> Hunk #10 failed at 367. >>>>> Hunk #11 failed at 1543. >>>>> 11 out of 11 hunks failed--saving rejects to nathelper.c.rej >>>>> Hmm... The next patch looks like a unified diff to me... >>>>> T >>>>> _______________________________________________ >>>>> Serusers mailing list >>>>> Serusers(a)lists.iptel.org >>>>> http://lists.iptel.org/mailman/listinfo/serusers >>>>> >>>>> >>>> >>>> >>> >>> >> >> > >

1 0

[Serusers] Carrier-grade framework for SER
by Andreas Granig 27 Sep '06

27 Sep '06

Hi guys, I'd like to propose another possibility for a highly-available and scalable system design based on SER: The problems I've encountered for scalable systems are: - Distribution of the user location and alias location among the nodes (user location is based on registrations, alias location comes from web interfaces and is used for call forwarding). - Reloading up to date location tables after breakdown and recovery of a node So I'm just thinking loud about the following provisioning system: - Write a client which fulfills the this demands: - Receive one or more locations from SER via a SER module or from a web application and distribute them to other known clients. Take care of retransmissions if a client isn't reachable or reports a temporary failure. - Receive one or more locations from other clients and write them into the SER FIFO. If writing into the FIFO fails, try to write directly into the database (location-table, alias-table etc.). Report a temporary failure if this also fails. Maybe a centralized server should be used which receives the locations from the clients and distributes them to other clients, so that the nodes just know about the server and nothing about other nodes. This would make integration of new nodes easy. On the other hand, it's another single point of failure, so a decentralized solution should be considered. But that would mean that you've to inform every node about the existence of a new node. The protocol used between the nodes should be simple and fast. So I think SOAP drops out here. Maybe XMLRPC or ICE (http://www.zeroc.com/ice.html) could be used. One might think now why not just use replication on SIP layer, but t_replicate only supports one peer and you've no possibility to get locations on a node while it's down. Replication of other location tables like the alias-table is also not possible. I'd be willing to release these parts as GPL for creating an open framework for carrier-grade SER integration, so any feedback, improvements or flames are highly welcome. Cheers, Andy

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[Users] auth ACK and BYE?
by Mark Kent 27 Sep '06

27 Sep '06

I just noticed that my SIP phone sends authentication credentials for ACK and BYE in a session where the INVITE required authentication credentials. I recognize this as a good thing... but it also surprised me. Currently, I don't handle it in my openser.cfg... I neither ask for the credentials nor do I strip them off before passing the message down the line... other than for INVITE. I'm going to correct this in my config, and I thought it worth mentioning for others just in case others also hadn't considered it. -mark

2 1

[Serusers] digit matching
by Richard C. Thompson 27 Sep '06

27 Sep '06

Hi All I hope someone can point me in the right direction or just give me a hint. I have the following route in ser.cfg and it works great. All I would like to do is change it so it matches on any number that starts with 002 regardless of length not just 12 more digits or a fixed length. if(uri=~"sip:(002............)@") I tried if(uri=~"sip:(002*)@") but it didn't work TIA Rick

4 3

[Users] Beer in oslo tomorrow
by Atle Samuelsen 27 Sep '06

27 Sep '06

This is just a beer reminder.. Remember beer in Oslo tomorrow. :) Topic: ser 0.10 _A

2 1

[Serusers] Callfwd example not routing
by Murray Hooper 27 Sep '06

27 Sep '06

I am trying to get a call forwarding to a PSTN gateway. I see the SIP call come in but appears I am missing something to get the routing/call forward to be recognized. Below is SIP log. Sep 26 13:04:52 localhost ./ser[7102]: SIP Request: Sep 26 13:04:52 localhost ./ser[7102]: method: <INVITE> Sep 26 13:04:52 localhost ./ser[7102]: uri: <sip:5199571788@216.183.86.133> Sep 26 13:04:52 localhost ./ser[7102]: version: <SIP/2.0> Sep 26 13:04:52 localhost ./ser[7102]: parse_headers: flags=1 Sep 26 13:04:52 localhost ./ser[7102]: Found param type 232, <branch> = <z9hG4bK0a45e9d0>; state=6 Sep 26 13:04:52 localhost ./ser[7102]: Found param type 235, <rport> = <n/a>; state=17 Sep 26 13:04:52 localhost ./ser[7102]: end of header reached, state=5 Sep 26 13:04:52 localhost ./ser[7102]: parse_headers: Via found, flags=1 Sep 26 13:04:52 localhost ./ser[7102]: parse_headers: this is the first via Sep 26 13:04:52 localhost ./ser[7102]: After parse_msg... Sep 26 13:04:52 localhost ./ser[7102]: preparing to run routing scripts... Sep 26 13:04:52 localhost ./ser[7102]: parse_headers: flags=128 Sep 26 13:04:52 localhost ./ser[7102]: end of header reached, state=9 Sep 26 13:04:52 localhost ./ser[7102]: DEBUG: get_hdr_field: <To> [33]; uri=[sip:5199571788@216.183.86.133] Sep 26 13:04:52 localhost ./ser[7102]: DEBUG: to body [<sip:5199571788@216.183.86.133> ] Sep 26 13:04:52 localhost ./ser[7102]: get_hdr_field: cseq <CSeq>: <102> <INVITE> Sep 26 13:04:52 localhost ./ser[7102]: DEBUG: get_hdr_body : content_length=218 Sep 26 13:04:52 localhost ./ser[7102]: found end of header Sep 26 13:04:52 localhost ./ser[7102]: DEBUG: is_maxfwd_present: max_forwards header not found! Sep 26 13:04:52 localhost ./ser[7102]: parse_headers: flags=64 Sep 26 13:04:52 localhost ./ser[7102]: check_via_address(216.143.130.61, 216.143.130.61, 0) Sep 26 13:04:52 localhost ./ser[7102]: parse_headers: flags=33554432 Sep 26 13:04:52 localhost ./ser[7102]: DEBUG: add_param: tag=as35382f6d Sep 26 13:04:52 localhost ./ser[7102]: end of header reached, state=29 Sep 26 13:04:52 localhost ./ser[7102]: parse_headers: flags=256 Sep 26 13:04:52 localhost ./ser[7102]: find_first_route: No Route headers found Sep 26 13:04:52 localhost ./ser[7102]: loose_route: There is no Route HF Sep 26 13:04:52 localhost ./ser[7102]: is_local(): Realm '216.183.86.133' is not local Sep 26 13:04:52 localhost ./ser[7102]: is_local(): Realm '216.143.130.61' is not local Sep 26 13:04:52 localhost ./ser[7102]: parse_headers: flags=-1 Sep 26 13:04:52 localhost ./ser[7102]: check_via_address(216.143.130.61, 216.143.130.61, 0) Sep 26 13:04:52 localhost ./ser[7102]: DEBUG:destroy_avp_list: destroying list (nil) Sep 26 13:04:52 localhost ./ser[7102]: receive_msg: cleaning up Sep 26 13:04:52 localhost ./ser[7098]: SIP Request: Sep 26 13:04:52 localhost ./ser[7098]: method: <ACK> Sep 26 13:04:52 localhost ./ser[7098]: uri: <sip:5199571788@216.183.86.133> Sep 26 13:04:52 localhost ./ser[7098]: version: <SIP/2.0> Sep 26 13:04:52 localhost ./ser[7098]: parse_headers: flags=1 Sep 26 13:04:52 localhost ./ser[7098]: Found param type 232, <branch> = <z9hG4bK0a45e9d0>; state=6 Sep 26 13:04:52 localhost ./ser[7098]: Found param type 235, <rport> = <n/a>; state=17 Sep 26 13:04:52 localhost ./ser[7098]: end of header reached, state=5 Sep 26 13:04:52 localhost ./ser[7098]: parse_headers: Via found, flags=1 Sep 26 13:04:52 localhost ./ser[7098]: parse_headers: this is the first via Sep 26 13:04:52 localhost ./ser[7098]: After parse_msg... Sep 26 13:04:52 localhost ./ser[7098]: preparing to run routing scripts... Sep 26 13:04:52 localhost ./ser[7098]: parse_headers: flags=4 Sep 26 13:04:52 localhost ./ser[7098]: DEBUG: add_param: tag=7770e7624e56f146f4f19d50775e8c7e.230c Sep 26 13:04:52 localhost ./ser[7098]: end of header reached, state=29 Sep 26 13:04:52 localhost ./ser[7098]: DEBUG: get_hdr_field: <To> [75]; uri=[sip:5199571788@216.183.86.133] Sep 26 13:04:52 localhost ./ser[7098]: DEBUG: to body [<sip:5199571788@216.183.86.133>] Sep 26 13:04:52 localhost ./ser[7098]: DEBUG: sl_filter_ACK : local ACK found -> dropping it! Sep 26 13:04:52 localhost ./ser[7098]: DEBUG:destroy_avp_list: destroying list (nil) Sep 26 13:04:52 localhost ./ser[7098]: receive_msg: cleaning up

2 1

[Serusers] Nathelper Ping OPTIONS
by olivier.taylor 27 Sep '06

27 Sep '06

Hello, just trying to enable the ping functionality in nathelper to send SIP message (i.e. OPTIONS) instead of 4-byte UDP packets on SER 0.9.6 and 0.10.0 as shown below, patch doesn't apply Is there another possibility to have this feature? Olivier patch -p1 < nathelper_ping.patch Hmm... Looks like a unified diff to me... The text leading up to this was: -------------------------- |diff -Naur -x CVS nathelper/nathelper.c nathelper_ping/nathelper.c |--- nathelper/nathelper.c 2006-08-16 08:45:48.000000000 +0200 |+++ nathelper_ping/nathelper.c 2006-08-10 14:54:14.000000000 +0200 -------------------------- Patching file nathelper.c using Plan A... Hunk #1 failed at 118. Hunk #2 failed at 172. Hunk #3 failed at 193. Hunk #4 failed at 212. Hunk #5 failed at 239. Hunk #6 failed at 257. Hunk #7 failed at 267. Hunk #8 failed at 276. Hunk #9 failed at 320. Hunk #10 failed at 367. Hunk #11 failed at 1543. 11 out of 11 hunks failed--saving rejects to nathelper.c.rej Hmm... The next patch looks like a unified diff to me... T

2 3

[Serusers] How to restrict only one call per user at the same time
by Ricardo Carvalho 27 Sep '06

27 Sep '06

Hi all, In an environment where the same user can be registered at the same time in multiple phones, how may I restrict users to only be able to make one call simultaneously? I mean... only make other call after hanging up the last one, never simultaneously made from different phones. How may I implement that? Regards, Ricardo.

2 1

[Serusers] Re: Ser Radius Howto
by Alex Fler 27 Sep '06

27 Sep '06

Hello All, I'm looking for Ser Radius howto, or any other related documentation. For some reason docs were removed from iptel website. Regards Alex Fler

2 1

[Serusers] Call Forwarding Using PSTN Gateway
by Murray Hooper 27 Sep '06

27 Sep '06

Newbie to SER. I am a little confused about how much I have to set up in databases to get an incoming SIP call to call forward to a PSTN number through SER. I have setup the username, attribute and value for my number, "callfwd", sip:1234567890@gateway.com in usr_preferences and the username in the subscriber file , added gateway ip to trusted, but call is immediately dropped when it arrives at SER. I have been through the getting started and have had success in modifying the values in the scripts, but don't see what data must be added into MySQL. Getting pointed in the right direction would be greatly appreciated Thanx murray

2 1

[Serusers] Difference between subscription handed by rls and pa
by Quan Huo Sheng 27 Sep '06

27 Sep '06

Hi; I am using ser-0.10.99-dev35-pa-4.1 to talk with SIP client(UA) and XCAP server for handling subscription and publish. Anyone can give me some comments for: 1. Difference between subscription handled by rls (handle_rls_subscription) and pa (handle_subscription). 2. Difference between table ser.subscriber and ser.rls_subscription. Thanks Smile [This e-mail is confidential and may be priviledged. If you are not the intended recipient, please kindly notify us immediately and delete the message from your system; please do not copy or use it for any purpose, nor disclose its contents to any other person. Thank you.] ---ST Electronics Group---

2 1

[Serusers] rls, xcap and remote storage
by Quan Huo Sheng 27 Sep '06

27 Sep '06

Hi; I have a softphone (ie. Eyebeam) that talks with SER. The question is whether Eyebeam can upload or download the contact list/buddy list file in xml format from XCAP server via RLS modules of SER. Please advise. Regards Smile [This e-mail is confidential and may be priviledged. If you are not the intended recipient, please kindly notify us immediately and delete the message from your system; please do not copy or use it for any purpose, nor disclose its contents to any other person. Thank you.] ---ST Electronics Group---

2 1

[Users] want a sample openser.cfg can forward calls to Asterisk (PSTN)
by Steve Xu 27 Sep '06

27 Sep '06

Hi folks, I have Asterisk as a PSTN termination GW, Openser will handle the user registration. Openser users has name format 7xxxxxxxx ( all digistes), which can call each other peer to peer ( it is a basic openser function). I want to forward the call ( 1xxxxxxxxxx or 011xxxxxxxxxxxxx ) to Asterisk when openser user try to terminate US or International calls. I tried to use the setting like this: if(is_method("INVITE") && !has_totag() && uri=~"sip:011[0-9]+@.*") { route(4); exit; }; route[4] { strip(3); # route to Asterisk Media Server rewritehostport("222.222.222.79:5060"); if (!t_relay()) { sl_reply_error(); }; exit; } OpenSER keep looking the 011xxxxxxxxx from location and got 408 always , it does not forward the call to Asterisk. I read a doc about how to share acc and voice meail with Asterisk, will it have to use failure_route to accomplish it ? I am new to Openser and dont know much about the routing logic. Anybody could point me to the right way? A sample cfg will really appreciated. Steve

1 0

[Users] [openser] OpenserAdmin Login
by John Shellhouse 27 Sep '06

27 Sep '06

Hi, I am cannot login to the initial Webpage for OpenserAdmin after starting ./script/server. I put in the default username:password of openser:openserrw but I am not allowed access. Any suggestions anyone? Thanks, jswhel

1 1

FW: [Serusers] digit matching
by Richard C. Thompson 26 Sep '06

26 Sep '06

To All I answered it myself if(uri=~"sip:002[0-9]*@") Thanks Rick _____ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Richard C. Thompson Sent: Tuesday, September 26, 2006 4:31 PM To: serusers(a)iptel.org Subject: [Serusers] digit matching Hi All I hope someone can point me in the right direction or just give me a hint. I have the following route in ser.cfg and it works great. All I would like to do is change it so it matches on any number that starts with 002 regardless of length not just 12 more digits or a fixed length. if(uri=~"sip:(002............)@") I tried if(uri=~"sip:(002*)@") but it didn't work TIA Rick

1 0

[Users] avp_write with $ruri and /username /domain
by Papadopoulos Georgios 26 Sep '06

26 Sep '06

hello, Trying to migrate from 1.0.0 to 1.1.0 I came across a problem that has already been discussed in the list (of course if I had first searched the archives I would have saved myself from a couple of days of debbuging). Here are the two discussions: http://openser.org/pipermail/users/2006-August/006246.html http://openser.org/pipermail/users/2006-August/006284.html I am just sending this again because I think it looks like a bug to me. In the avpops documentation for 1.0.0 the example says avp_write("$ruri/username","$email"); In 1.1.0 it has changed to avp_write("$ru/username","$avp(email)"); Neither of those two will produce the username in avp email with openser 1.1.0. The same holds for avp_write("$ru/domain","$avp(whatever)"); avp_write("$ru/domain","$avp(whatever)"); The solution is to use: avp_write("$rU","$avp(whatever)"); avp_write("$rd","$avp(whatever)"); If the /username, /domain are not supposed to work, then shouldn't they be removed from the document? And shouldn't they cause some syntax error? thanks a lot George Disclaimer The information in this e-mail and any attachments is confidential. It is intended solely for the attention and use of the named addressee(s). If you are not the intended recipient, or person responsible for delivering this information to the intended recipient, please notify the sender immediately. Unless you are the intended recipient or his/her representative you are not authorized to, and must not, read, copy, distribute, use or retain this message or any part of it. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.

2 1

[Serusers] how to make SER send Digest-Username(115) attribute
by a.kolesnikov 26 Sep '06

26 Sep '06

Is it possible to make SER to send Digest-Username(115) attribute defined in RFC 4590 instead or in addition to Sip-Uri-Users(208) attribute? С уважением, Колесников Андрей Netris <http://www.netris.ru> http://www.netris.ru

2 1

[Serusers] Help needed to connect to a PSTN Gateway
by Ralf Eisele 26 Sep '06

26 Sep '06

Hi, I want connect my SER-Server to a public PSTN Gateway. The sample config found in 'SER - Getting Started' doesn't work. Someone out there who can help me? Mit besten Gruessen Ralf Eisele --------- eXtension World Wide Connections GmbH Partner der abbex-group.ag Finninger Strasse 56 Fon: +49 731 9216333 D-89231 Neu-Ulm Fax: +49 731 9216335 Germany Mobile: +49 179 2991333 http://www.extension.net - info(a)extension.net PGP Fingerprint: C29E 9EE9 6A36 8EDA 7146 1613 0BA8 9080 5B4A 36A3

1 0

[Users] gen_CA.sh
by Gregoire 26 Sep '06

26 Sep '06

Hi everyone! I am running openser 1.1.0-tls installed with debian package. I am trying to configure correctly TLS, and generate certificate, but in tutorials I found, people talk about a script located in /etc/openser/tls/tools named "gen_rootCA.sh", but I don't have the directory "tools" in /etc/openser/tls. Does someone have any idea about this script? Another thing is when I tried to send a mail to a user with the command: openserctl mail <username> Openser makes errors like: Write email to mail: now ... mail: You must specify direct recipients with -s, -c, or -b ERROR: sending message failed I had to change the line 1777 in /usr/sbin/openserctl in send_email $2 instead of send_email $1 to make it work. Greg

2 1

[Users] NEW SIP-to-XMPP (Google Talk/Jabber) gateway
by Daniel-Constantin Mierla 26 Sep '06

26 Sep '06

Hello, thanks to Andrea Spirea, a new SIP-to-XMPP gateway was included in OpenSER. There is a new module, XMPP, which enables SIP users to chat with any XMPP user (Google Talk, Jabber) user in the world. The design permits flexibility in configuration and deployments, the new module can at alone as a XMPP server or can work in tandem with a standalone XMPP server (for more complex communication scenarios like TLS ...). Also, the module can be plugged in your main SIP server or can be deployed alone, enabling in a easy manner geographic distribution and redundancy. There is no need of mapping SIP addresses to XMPP addresses, no need to store translation profiles, everything is done in the fly. The conversation will survive to restarts, the session being recovered form the messages. As a result, there is no need of a persistent storage, the footprint is very small, embedding the gateway in small devices should be straightforward. The news is posted here: http://www.openser.org/index.php?option=com_content&task=view&id=56&Itemid=9 Module's documentation: http://www.openser.org/docs/modules/1.2.x/xmpp.html Feedback will be very appreciated, the next steps are to enhance the standalone-xmpp-server version of the module with other features of a xmpp server and to add support for presence and voice. Testing and reports are always welcome. Cheers, Daniel

2 2

[Serusers] store in AVP something searched with a regular expression
by Ricardo Carvalho 26 Sep '06

26 Sep '06

How may I store in AVP something searched with a regular expression from SIP messages? Regards, Ricardo

1 0

[Users] Loose route...again
by Helge Waastad 26 Sep '06

26 Sep '06

Hi, I've just a quick question regarding the loose_route function We had a discussion earlier this year when the answer for loose_route was: "by default, loose_route() function detects the pre-loaded route: if there is only one Route hdr pointing to the server, it will be consumed and loose_route() will return false. If more than one Route hdr is present, the routing will be done as usual - function will consume the Route and set the destination to the next Route - true is returned." I just wonder about the "returning false" since I always see loose_route return true...is this due to the lr tag? eks: My yyy.yyy.yyy.yyy receives a BYE message BYE sip:67512391@10.46.11.132 SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;branch=z9hG4bK84fb.7836a4d5.0 Via: SIP/2.0/UDP 10.46.11.101:5060;rport=5060;branch=z9hG4bK7cae83600f656a72 Route: <sip:yyy.yyy.yyy.yyy:5060;lr> >From the explaiantion I would guess if (loose_route()){ do-something} should not do-something??? I hope Im not totally out of my head right now br hw -- Helge Waastad Senior Engineer Systemavdelingen Smartnet

2 1

[Users] pdt problem
by Francesco Castellano 26 Sep '06

26 Sep '06

Dear list members, I've a problem trying to get pdt module working. These are the log I get with debug=9. pdt_print_tree: [sip.mydomain.com] pdt_print_node: [2] [proxy01.sipphone.com] pdt_print_node: [2393] [other.host.at.domain.com] pdt_print_node: [51] [my.host.at.domain.com] PDT:pdt_sync_cache: ... PDT:pdt_sync_cache: sync op[1]=1... prefix alredy allocated pdt_add_to_tree internal error! PDT:pdt_sync_cache: Error to insert into tree PDT:prefix2domain: cannot update the cache I'm working with openser 1.1.0-notls, and pdt.c version "v 1.8 2006/07/03 15:26:17 miconda Exp". MySQL description of pdt table is: +---------+--------------+------+-----+---------+-------+ | Field | Type | Null | Key | Default | Extra | +---------+--------------+------+-----+---------+-------+ | sdomain | varchar(255) | NO | PRI | NULL | | | prefix | varchar(32) | NO | PRI | NULL | | | domain | varchar(255) | NO | | NULL | | +---------+--------------+------+-----+---------+-------+ but the association sdomain,prefix,domain seems to be correct, as the pdt_print log lines report. Strange behaviour also occurs when I try to list from fifo the table every two times once I get: host:~# openserctl fifo pdt_list database engine 'MYSQL' loaded Control engine 'FIFO' loaded entering fifo_cmd pdt_list 400 pdt_fifo_list - domain not found FIFO command was: :pdt_list:openser_receiver_25901 while the second time openserctl hangs until I kill it with ctrl+c. Any suggestion will be helpful, thanks Francesco Castellano

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[Users] db-script not working
by Gerd Feiner 26 Sep '06

26 Sep '06

hi list, i have strange problems with the openser_mysql.sh script: # openser_mysql.sh create MySql password for root: Domain (realm) for the default user 'admin': sip.de creating database openser ... ERROR 1133 (42000) at line 7: Can't find any matching row in the user table Install SERWEB tables ?(y/n): for some reason it does not do anything besides creating a database (with no tables in it). how am i supposed to create the database instead? -- mit freundlichen Grüßen / with best regards Gerd Feiner Leitung IP-Dienste Kabelfernsehen München ServiCenter GmbH & Co. KG Abt. Mediaport Medienallee 24 85774 Unterföhring T: +49 89 95083 243 F: +49 89 95083 132 E: g.feiner(a)cablesurf.de W: www.cablesurf.de

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[Users] fix_nated_contact not rewriting contact?
by kjcsb 26 Sep '06

26 Sep '06

I have the following code in openser.cfg if (client_nat_test("3")) { #IP address in contact is private IP #Via is different to address client has contacted us from #If so then setflag7 to indicate that sender is behind NAT xlog("client nat test 3 returned true for r-uri <$ru> in route 3 so will setflag7"); setflag(7); xlog("contact before update <$ct>"); xlog("via before update <$hdr(via)>"); force_rport(); #Rewrite Contact: header to contain Sip client's public IP:port fix_nated_contact(); xlog("contact is now <$ct>"); xlog("via is now <$hdr(via)>"); }; However the value returned by $ct is the same before and after fix_nated_contact(). Is that because I'm doing something wrong, or because I've misunderstood fix_nated_contact or because the $ct variable doesn't get updated by fix_nated_contact? If the latter, how can I find out the result of fix_nated_contact? The same applies to force_rport and the via details. Any advice appreciated. Regards Cameron

2 1

[Users] undefined symbol: mem_lock
by Ovidiu Sas 26 Sep '06

26 Sep '06

Hi, I'm trying to build openser 1.1.0 (notls) for arm for Linksys NSLU - OpenSLUG OS. The cross compilation went ok, but when I try to run the program I encounter the following error: root@LKG7BE78E # openser -c 0(4517) ERROR: load_module: could not open module </usr/lib/openser/modules/sl.so>: /usr/lib/openser/modules/sl.so: undefined symbol: mem_lock 0(4517) parse error (41,13-14): failed to load module Here are the compile flags: NOTE: make -e MAKEFLAGS= bin-target=/usr/sbin doc-target=/usr/share/doc/openser lib-target=/usr/lib/openser/ modules-target=/usr/lib/openser/modules/ cfg-target=/etc/openser/ prefix=/usr cfg-prefix=/ ARCH=armeb LD=ccache armeb-linux-gcc -march=armv5te -mtune=xscale -mthumb-interwork -mno-thumb all bison -d -b cfg cfg.y cfg.y: conflicts: 1 shift/reduce flex cfg.lex Compiling action.c ccache armeb-linux-gcc -march=armv5te -mtune=xscale -mthumb-interwork -mno-thumb -isystem/home/admin/slug-3.10/releases/slugos-3.10-beta/openslug-nslu2.tmp/staging/armeb-linux/include -Os -fomit-frame-pointer -frename-registers -fno-unroll-loops -fno-inline-functions -fweb -funit-at-a-time -DNAME='"openser"' -DVERSION='"1.1.0-notls"' -DARCH='"armeb"' -DOS='"linux"' -DCOMPILER='"ccache 3.4.4"' -D__CPU_armeb -D__OS_linux -D__SMP_no -DCFG_DIR='"/etc/openser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DF_MALLOC -DSTATISTICS -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DUSE_SYSV_SEM -DHAVE_SIGIO_RT -DHAVE_SELECT -c action.c -o action.o ... Linking openser ccache armeb-linux-gcc -march=armv5te -mtune=xscale -mthumb-interwork -mno-thumb -L/home/admin/slug-3.10/releases/slugos-3.10-beta/openslug-nslu2.tmp/staging/armeb-linux/lib -Wl,-rpath-link,/home/admin/slug-3.10/releases/slugos-3.10-beta/openslug-nslu2.tmp/staging/armeb-linux/lib -Wl,-O1 action.o core_stats.o crc.o daemonize.o data_lump.o data_lump_rpl.o dprint.o dset.o error.o fifo_server.o flags.o forward.o io_wait.o ip_addr.o items.o items_extra.o main.o md5.o md5utils.o modparam.o msg_translator.o pass_fd.o proxy.o qvalue.o re.o receive.o resolve.o route.o route_struct.o script_cb.o serialize.o socket_info.o sr_module.o statistics.o tcp_main.o tcp_read.o timer.o tsend.o udp_server.o unixsock_server.o usr_avp.o mem/f_malloc.o mem/mem.o mem/memtest.o mem/q_malloc.o mem/shm_mem.o mem/vq_malloc.o parser/hf.o parser/msg_parser.o parser/parse_allow.o parser/parse_content.o parser/parse_cseq.o parser/parse_disposition.o parser/parse_diversion.o parser/parse_event.o parser/parse_expires.o parser/parse_fline.o parser/parse_from.o parser/parse_hname2.o parser/parse_hostport.o parser/parse_methods.o parser/parse_nameaddr.o parser/parse_param.o parser/parse_refer_to.o parser/parse_rpid.o parser/parse_rr.o parser/parse_sipifmatch.o parser/parse_sst.o parser/parse_supported.o parser/parse_to.o parser/parse_uri.o parser/parse_via.o parser/parser_f.o parser/digest/digest.o parser/digest/digest_parser.o parser/digest/param_parser.o parser/contact/contact.o parser/contact/parse_contact.o db/db.o db/db_fifo.o db/db_id.o db/db_pool.o lex.yy.o cfg.tab.o -ldl -lresolv -o openser ... make[1]: Entering directory `/home/admin/slug-3.10/releases/slugos-3.10-beta/openslug-nslu2.tmp/work/openser-1.1.0-notls-r0/openser-1.1.0-notls/mod ules/sl' Compiling sl.c ccache armeb-linux-gcc -march=armv5te -mtune=xscale -mthumb-interwork -mno-thumb -fPIC -DPIC -isystem/home/admin/slug-3.10/releases/slugos-3.10-beta/openslug-nslu2.tmp/staging/armeb-linux/include -Os -fomit-frame-pointer -frename-registers -fno-unroll-loops -fno-inline-functions -fweb -funit-at-a-time -DNAME='"openser"' -DVERSION='"1.1.0-notls"' -DARCH='"armeb"' -DOS='"linux"' -DCOMPILER='"ccache 3.4.4"' -D__CPU_armeb -D__OS_linux -D__SMP_no -DCFG_DIR='"/etc/openser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DF_MALLOC -DSTATISTICS -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DUSE_SYSV_SEM -DHAVE_SIGIO_RT -DHAVE_SELECT -c sl.c -o sl.o ... Linking sl.so ccache armeb-linux-gcc -march=armv5te -mtune=xscale -mthumb-interwork -mno-thumb -shared -L/home/admin/slug-3.10/releases/slugos-3.10-beta/openslug-nslu2.tmp/staging/armeb-linux/lib -Wl,-rpath-link,/home/admin/slug-3.10/releases/slugos-3.10-beta/openslug-nslu2.tmp/staging/armeb-linux/lib -Wl,-O1 sl.o sl_cb.o sl_funcs.o -o sl.so Here's the output of the nm -D for openser: root@LKG7BE78E:/usr/sbin# nm -D /usr/sbin/openser U _IO_getc 00066f38 R _IO_stdin_used w _Jv_RegisterClasses U __assert_fail U __ctype_b_loc w __deregister_frame_info U __dn_expand U __errno_location U __fxstat w __gmon_start__ 00089000 A __init_array_end 00089000 A __init_array_start U __libc_current_sigrtmax U __libc_current_sigrtmin U __libc_start_main U __lxstat w __register_frame_info U __res_init U __res_search U __res_state U __xstat U abort U accept U alarm U atoi U bind U chdir U chmod U chown U chroot U clearerr U close U closelog U connect U ctime U difftime U dlclose U dlerror U dlopen U dlsym U exit U fclose U fcntl U fdopen U ferror U fflush U fgets U fileno U fopen U fork U fprintf U fputc U fputs U fread U free U freopen U fscanf U fwrite U getcwd U geteuid U getgrnam U gethostbyaddr U gethostbyname U gethostbyname2 U getopt U getpgid U getpid U getprotobyname U getpwnam U getrlimit U getsockname U getsockopt U inet_pton U ioctl U isatty U kill U listen U malloc U memchr U memcmp U memcpy U memmove U memset U mkfifo U mktime U mmap U munmap U open U openlog 0008b444 B optarg 0008b434 B opterr 0008b448 B optopt U pathconf U pause U perror U poll U printf U putchar U puts U rand U read U realloc U recv U recvfrom U recvmsg U regcomp U regexec U regfree U select U semctl U semget U semop U send U sendmsg U sendto U setbuf U seteuid U setgid U setrlimit U setsid U setsockopt U setuid U sigaddset U sigdelset U sigemptyset U sigismember U signal U sigprocmask U sigtimedwait U sleep U snprintf U socket U socketpair U srand 0008b43c B stderr 0008b440 B stdin 0008b438 B stdout U strcasecmp U strchr U strcmp U strcpy U strdup U strerror U strlen U strncasecmp U strncmp U strncpy U strtol U syslog U system U time U uname U unlink U vfprintf U vsnprintf U wait U waitpid U writev root@LKG7BE78E:/usr/sbin# nm -D /usr/lib/openser/modules/sl.so U MDStringArray 0000b260 A _DYNAMIC w _Jv_RegisterClasses 0000b60c A __bss_end__ 0000b570 A __bss_start 0000b570 A __bss_start__ w __cxa_finalize 0000b454 D __data_start w __deregister_frame_info 0000b60c A __end__ U __errno_location w __gmon_start__ w __register_frame_info 0000b60c A _bss_end__ 0000b570 A _edata 0000b60c A _end 00002b68 T _fini 000011e4 T _init U add_lump_rpl U build_res_buf_from_sip_req U crcitt_string_array U debug 000018b0 T destroy_slcb_lists U dprint U err2reason_phrase 0000b460 D exports U fm_free U fm_malloc U get_send_socket U get_ticks U log_facility U log_stderr U mem_block U mem_lock U memcmp U memcpy U memset U mhomed 0000b4e8 D mod_stats 0000b560 D module_flags 0000b564 D module_version 0000b5e4 B modules U parse_headers 0000b5e0 B pgid_file 0000b5f0 B pid_file U prev_ser_error U print_dset 0000b600 B rcv_acks U register_script_cb 00001900 T register_slcb U reply_to_via 000019e0 T run_sl_callbacks U semop 0000b5f4 B sent_err_rpls 0000b5ec B sent_rpls U shm_block 0000b45c D sl_enable_stats 00002868 T sl_filter_ACK 000026a8 T sl_reply_error 00001f58 T sl_send_reply 00001ef0 T sl_shutdown 00001d3c T sl_startup 0000b58c B slcb_hl U stat_lock U strerror U strlen U syslog U tcp_disable 0000b608 B tcp_listen U tcp_send 0000b5d8 B tx_1xx_rpls 0000b5d4 B tx_2xx_rpls 0000b5dc B tx_3xx_rpls 0000b5f8 B tx_4xx_rpls 0000b5e8 B tx_5xx_rpls 0000b5fc B tx_6xx_rpls 0000b604 B udp_listen U udp_send U update_sock_struct_from_via root@LKG7BE78E:/usr/sbin# A similar problem was encounter and fixed for OpenBSD .... Any help would be greatly appreciated. -ovi

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[Serusers] Prepaid and SERWeb
by Legos Engineering 26 Sep '06

26 Sep '06

Hi, I find something about prepaid system for my SERWeb. In fact, I want to print the price of communication in the accounting page for my final customers. I'm thinking about develop myself a db with price by minute and do the calcul on demand in accounting page. But if something already exist, can you tell me more about it ? Thanks

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