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Re: [Serusers] How to configure Trunk capacity
by roayu 11 Jul '07

11 Jul '07

Hi, Klaus! Thanks for your reply. In that case, is there any software that can be integrated with SER to function as SIP trunking? Thanks. Best regards, Roa Yu -----Original Message----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Sent: Tuesday, July 10, 2007 2:34 PM To: roayu Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] How to configure Trunk capacity What do you mean by "trunk capacity"? The number of calls which may be sent to a certain gateway concurrently? If yes - you can't configure this in ser. ser is is transaction statefull, thus it does not count calls at all. Usually you just send the call to the gateway, and if the gateway is busy it will reponse with a certain response code. Then you configure ser to activate a failure to another gateway. regards klaus roayu wrote: > Hi, there! > > > > Can anyone tell me where can I configure the Trunk capacity for SER? > > > > Thank you. > > > > > > Best regards, > > Roa Yu > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers

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[Users] openser 1.1.1 available for several embedded platforms
by Ovidiu Sas 10 Jul '07

10 Jul '07

Hi, openser 1.1.1 is now available in binary format for several embedded patforms: - Linksys NSLU - Synology DS-101(g+) - Iomega NAS 100d - Freecom FSG-3 - Qnap TurboStation ... and all routers supported by the oleg or dd-wrt distributions. Check out http://www.nslu2-linux.org Regards, Ovidiu Sas

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[OpenSER-Users] mediaproxy module init error
by Rizwan Hisham 10 Jul '07

10 Jul '07

Hi all, im a new user of openser. how can i use mediaproxy found at www.ag-projects.com to destribute load on my 2 asterisk servers. I have configured a simple openser server which routes every request to asterisk. now i want to install another asterisk server and openser will be used to destribute calls between the 2. openser is running fine, i need to learn how mediaproxy works along with openser. i have installed the mediaproxy but the following error shows up when starting openser: 0(0) ERROR:mediaproxy:mod_init: Bad config - nat ping enabled, but not nat bflag set in usrloc module 0(0) init_mod(): Error while initializing module mediaproxy ERROR: error while initializing modules 0(0) INFO:mi_fifo:mi_destroy: process hasn't been created -> nothing to kill here are the modules and modules parameters: loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "mi_fifo.so" loadmodule "domain.so" loadmodule "mediaproxy.so" loadmodule "xlog.so" modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") modparam("rr", "enable_full_lr", 1) modparam("domain", "db_url", "mysql://openser:openser@192.168.0.7/openser") modparam("mediaproxy", "mediaproxy_socket", "/var/run/proxydispatcher.sock") modparam("mediaproxy", "natping_interval", 60) I have no idea what above statements are doing excpet they are bothering me. i use the following link for help http://voip-info.org/wiki/view/OpenSER+And+Mediaproxy -- Best Regards Rizwan Hisham Software Engineer AXVOICE Inc. www.axvoice.com

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[Serusers] ser mem usage
by Arun Kumar 10 Jul '07

10 Jul '07

HI I've noticed strange thing about my ser(0.9.6) day by day memory usage is getting increas and even after stopping my ser its not releasing the memory. thanks Arun

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[OpenSER-Users] radius authentication and local users database...
by liran tal 10 Jul '07

10 Jul '07

Hey everyone, I've been using the radius authentication just fine but I have a general question... Is it possible to configure it so that openser will first look if the user is found on the local users/registrar database and if not found it will then use the radius mechanism? So, is it possible to integrate both radius and local user lookup or it's one or the either? Thanks, Lir.

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[Serusers] Getting rid of SIP URI in Caller ID
by Charles Ulrich 10 Jul '07

10 Jul '07

Hi all, I'm fairly new to SER (just heard about it this afternoon), but I've already run up against a problem that's vexing me. We have a Polycom phone sitting behind a Linksys WRT54G router/firewall with DD-WRT firmware. This version of DD-WRT is the "voip" edition, meaning it has SER and rtpproxy installed to help ease NAT issues with SIP. No configuration of SER is necessary, we enter the WRT54G's IP address into the phone's outbound proxy field and it just works. The only drawback is that when using this method, the caller ID on an incoming call from the PSTN contains a SIP URI (in the form of sip:5551212@207.179.x.x) instead of a simple telephone number. Our end-users are pretty voip illiterate and thus haven't the slightest clue how VoIP works. They are only going to get confused when a SIP URI shows up on their caller ID instead of the actual PSTN phone number of the person who called them. How can this be remedied? I can't tell what version of ser is on this device because the -h and -V flags return nothing. (Probably to make it slightly smaller for the embedded platform.) The 'strings' utility wasn't helpful either. Attached is the ser.cfg that comes with DD-WRT. Any help is greatly appreciated. Let me know if I need to provide more information. Thanks! -- Charles Ulrich Ideal Solution, LLC -- http://www.idealso.com

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[OpenSER-Users] next_gw() function
by Bruno Machado 10 Jul '07

10 Jul '07

Hi all. I'm using Openser 1.1.0 and I'm trying to create a failure route using LCR. All the calls must be routed to the same IP, but when the OpenSER execute the next_gw(), I receive the error "503 Service not available, no gateways found". When I change the IP column gw.ip_addr, it works fine. I think that is not possible to route the messages through the same IP. But I need it! All of my calls must to pass through this IP. openser ----> proxy B ----> the main GW (isn't answering) Using the failure route: openser ----> proxy B ----> the secundary GW (is answering) This 'proxy B' is important and I can't remove it at all. So, how can I use the next_gw() or another function to route the call throught it? There are a parameter that I can set to help? I wouldn't like to use a static route wrote in the script file. And the Openser's version 1.2? I didn't try it yet. Is it possible using this version? Any comments are welcome. Thanks a lot. Bruno Machado ____________________________________________________________________________________ Novo Yahoo! Cadê? - Experimente uma nova busca. http://yahoo.com.br/oqueeuganhocomisso

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[Serusers] How to configure Trunk capacity
by roayu 10 Jul '07

10 Jul '07

Hi, there! Can anyone tell me where can I configure the Trunk capacity for SER? Thank you. Best regards, Roa Yu

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[OpenSER-Users] Can I disable CANCEL when parallel forking?
by Tim Madorma 10 Jul '07

10 Jul '07

Hi, I was wondering if there is an option to disable the sending of CANCEL towards the branches that have not yet responded when OpenSER performs parallel forking and one endpoint has answered. The request/question that I have is sort of outlined in RFC 3841 as follows: cancel-directive: This type of directive indicates whether the caller would like each proxy server to send a CANCEL request downstream ("cancel") in response to a 200 OK from the downstream server (which is the normal mode of operation, making it redundant), or whether this function should be left to the caller ("no-cancel"). If a proxy receives a request with this parameter set to "no- cancel", it SHOULD NOT CANCEL any outstanding branches upon receipt of a 2xx. However, it would still send CANCEL on any outstanding branches upon receipt of a 6xx. But in this case, I was wondering if there was something that was configurable. thanks, Tim

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[Serusers] SEMS stops taking audio with a particular ATA
by SIP 10 Jul '07

10 Jul '07

This is kind of odd, and should really probably be directed toward the SEMS list, but I thought I'd pose it here first to see if anyone had seen it. A particular user keeps calling our SEMS server, and each and every time he does, it crashes. It ONLY happens with that one user, and only with his Cortelco ATA (an identical model to another we have being used on our network with no problems). We get this in the log: Jul 9 20:26:32 proxy Sems[17552]: Error: (AmThread.cpp)(start)(83): pthread create failed with code 11 Jul 9 20:26:35 proxy Sems[17552]: Error: (AmThread.cpp)(stop)(101): pthread_detach failed with code ESRCH: thread could not be found. And then poof, the SEMS server stops sending audio anymore and has to be killed and restarted (it still runs and accepts connections). Any ideas? Nothing much comes up on a google search for this error. N.

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[Users] statefull handling of 6xx responses
by Christian Schlatter 09 Jul '07

09 Jul '07

RFC 3261 defines that upon reception of a 6xx response, a proxy SHOULD cancel all forked dialogs and send back the 6xx as the final response to the upstream UA. --- rfc 3261, section 16.7 ------- Under the new rules, upon receiving a 6xx, a proxy will issue a CANCEL request, which will generally result in 487 responses from all outstanding client transactions, and then at that point the 6xx is forwarded upstream. ---------------------------------- Some call forking experiments have shown that openser 1.2 does not follow these rules, instead it seems to disregard any 6xx response and continue with normal processing of existing forked dialogs. E.g. if a call forks to five phones and one of them sends back a 603 Declined response, the other four phones continue to ring until fr_inv_timer fires and failure_route is called with a status of 408 Timeout. Is it possible to make openser more SIP RFC compliant through configuration or does this need a code change? thanks, Christian

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[OpenSER-Users] How do i make openser.cfg to overwrite the domain in the Request URI
by Pande, Piyush 09 Jul '07

09 Jul '07

Hi, I would like to route all incoming requests which have a "proxy IP" in the request URI to a specific domain. How can I do this through changing openser.cfg? Thanks for the help, Piyush

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[Serusers] Problems with avp_db_load and db_query
by Ravin Suri 09 Jul '07

09 Jul '07

Hello everyone, I am facing a problem trying to load data from MySQL in ser.cfg. I need to load the subsciber's email address in order to pass it to SEMS. I've tried two modules (avpops and db_ops) both with no success. I am using Ser 2.0.0-rc1 and I added an 'email' column to the uri table. Here is the DB schema (only 1 row shown without did, flags and scheme colums): +--------------+------------------------------+--------------------------+ | username | uid | email | +--------------+------------------------------+--------------------------+ | sb4157762898 | sb4157762898(a)sq2.doclinx.com | sb4157762898(a)doclinx.com | | +--------------+------------------------------+--------------------------+ A. avpops module. I have the following module params: modparam("avpops", "avp_table", "uri") modparam("avpops", "uid_column", "uid"); modparam("avpops", "avp_aliases", "email=i:47") modparam("avpops", "db_scheme", "email_scheme:table=uri;uid_col=uid;value_col=email;value_type=string") Then when I'm going to forward to SEMS: avp_db_load("$to", "$email/$email_scheme"); xlog("L_INFO", "route INBOUND[%rm]: voicemail, email=%$email, from: [%fu], To: [%tu]\n"); append_hf("P-App-Name: voicemail\r\n"); append_hf("P-App-Param: Email-Address=%$email\r\n"); rewritehostport("127.0.0.1:5070"); t_relay_to_udp("127.0.0.1","5070"); Here is the appropriate excerpt from the SER log (from the xlog statement above): Jul 9 13:12:39 sq2 /usr/local/sbin/ser[5595]: route INBOUND[INVITE]: voicemail, email=<null>, from: [sip:sb7037160791@sq2.doclinx.com], To: [sip:sb4157762898@sq2.doclinx.com] 2. db_ops module. I have the following module params: modparam("db_ops", "declare_query", "select/uri/email/uid///%$t.uid"); Then when I'm going to forward to SEMS: $email = @db_query[0]; xlog("L_INFO", "route INBOUND[%rm]: voicemail, email=%$email, from: [%fu], To: [%tu]\n"); append_hf("P-App-Name: voicemail\r\n"); append_hf("P-App-Param: Email-Address=%$email\r\n"); rewritehostport("127.0.0.1:5070"); t_relay_to_udp("127.0.0.1","5070"); This makes SER fail on startup with this message: 0(5693) Unable to resolve select 'db_query' at level 0 0(5693) BUG: route.c:492: Unable to resolve select ERROR: error -1 while trying to fix configuration I have followed the README file in db_ops to the letter. What am I missng? I'd be be happy to post ser.cfg and the log file. Thank you very much for your help. Best regards, - Ravin

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[OpenSER-Users] NOTIFY not forwarded upon PUBLISH
by KUMAR 09 Jul '07

09 Jul '07

Hi all, I'm using openser from the SVN. The configuration file that I'm using is attached. The server upon receiving PUBLISH, replies with a 200 OK. But doesnt send out NOTIFY. I want to use the presence modules, without XCAP. I just want the server to send out SUBSCRIBE on the UA's behalf, by looking up the values from the database and also to send out NOTIFY on the UA's behalf, whenever it receives the PUBLISH, all without having anything to do with XCAP. Is this possible ? If yes, what changes do I need to do with the configuration file. Thank you in advance. Kumar

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RE: [OpenSER-Users] crash when checking empty destination uri
by Papadopoulos Georgios 09 Jul '07

09 Jul '07

(gdb) frame 1 #1 0x080847ef in comp_scriptvar (msg=0x81787b0, op=21, left=0x8170a60, right=0x8170a6c) at route.c:948 948 return comp_s2s(op, &lstr, &rstr); (gdb) print lstr $1 = {s = 0x80dc726 "", len = 0} > -----Original Message----- > From: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] > Sent: Monday, July 09, 2007 5:43 PM > To: Papadopoulos Georgios > Cc: users(a)openser.org > Subject: Re: [OpenSER-Users] crash when checking empty destination uri > > Hello, > > in gdb, please do: > > frame 1 > print lstr > > and send me the output. Strange that in frame 0, the stack > seems to be corrupted. > > Thanks, > Daniel > > > On 07/09/07 15:08, Papadopoulos Georgios wrote: > > #0 0x080844a7 in comp_s2s (op=Variable "op" is not available. > > ) at route.c:748 > > #1 0x080847ef in comp_scriptvar (msg=0x81787b0, op=21, > > left=0x8170a60, > > right=0x8170a6c) at route.c:948 > > #2 0x080850a5 in eval_expr (e=0x8170a58, msg=0x81787b0, val=0x0) at > > route.c:1323 > > #3 0x0808517e in eval_expr (e=0x8170c38, msg=0x81787b0, val=0x0) at > > route.c:1365 > > #4 0x08084dcf in eval_expr (e=0x8170c60, msg=0x81787b0, val=0x0) at > > route.c:1374 > > #5 0x08050b5c in do_action (a=0x8170ed0, msg=0x81787b0) at > > action.c:783 > > #6 0x0805329c in run_action_list (a=0x81708c0, msg=0x81787b0) at > > action.c:131 > > #7 0x08051c20 in do_action (a=0x8166648, msg=0x81787b0) at > > action.c:111 > > #8 0x0805329c in run_action_list (a=0x8155010, msg=0x81787b0) at > > action.c:131 > > #9 0x080534cf in run_top_route (a=0x8155010, msg=0x81787b0) at > > action.c:111 > > #10 0x0807b4a6 in receive_msg ( > > buf=0x811b160 "ACK sip:DEMO1@213.5.43.4 SIP/2.0\r\nVia: > > SIP/2.0/UDP > > 213.5.17.78:5060;branch=z9hG4bK7a9dbc5b;rport\r\nFrom: \"GEOP > > Papadopoul\" <sip:2116872933@altecnet.gr>;tag=as4aecac7c\r\nTo: > > <sip:DEMO1@213.5.43.4>;tag=b3"..., len=411, > > rcv_info=0xbffb2d40) at receive.c:156 > > #11 0x080a4475 in udp_rcv_loop () at udp_server.c:451 > > #12 0x0806a116 in main_loop () at main.c:834 > > #13 0x0806c007 in main (argc=6, argv=0xbffb2f24) at main.c:1393 > > > > > > > >> -----Original Message----- > >> From: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] > >> Sent: Monday, July 09, 2007 3:03 PM > >> To: Papadopoulos Georgios > >> Cc: users(a)openser.org > >> Subject: Re: [OpenSER-Users] crash when checking empty destination > >> uri > >> > >> Hello, > >> > >> On 07/09/07 13:50, Papadopoulos Georgios wrote: > >> > >>> Hi Daniel, > >>> > >>> Here is the ACK packet > >>> > >>> U 2007/07/09 13:13:31.577697 213.5.17.78:5060 -> > >>> > >> 213.5.43.4:5060 ACK > >> > >>> sip:demo1@213.5.168.214:54656 SIP/2.0. > >>> Via: SIP/2.0/UDP 213.5.17.78:5060;branch=z9hG4bK504461f5;rport. > >>> Route: <sip:213.5.43.4;lr=on;ftag=as2f3574dc>. > >>> From: "GEOP Papadopoul" > <sip:2116872933@altecnet.gr>;tag=as2f3574dc. > >>> To: <sip:DEMO1@213.5.43.4>;tag=80299598395. > >>> Contact: <sip:2116872933@213.5.17.78>. > >>> Call-ID: 088e7cdc7176016349ad45ea69468b7b(a)altecnet.gr. > >>> CSeq: 102 ACK. > >>> User-Agent: Asterisk PBX. > >>> Max-Forwards: 70. > >>> Content-Length: 0. > >>> > >>> How do I get the backtrace that you need? All I get from > >>> > >> gdb is this: > >> > >>> #0 0x080844a7 in ?? () > >>> #1 0xbffb26e0 in ?? () > >>> #2 0x00b56cee in ?? () > >>> #3 0x00000000 in ?? () > >>> > >>> > >> Check if you got a core file, and then: > >> gdb /path/to/openser /path/to/core_file > >> > >> # bt > >> > >> Cheers, > >> Daniel > >> > >> > >>> Best regards > >>> > >>> George > >>> > >>> > >>> > >>> > >>> > >>>> -----Original Message----- > >>>> From: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] > >>>> Sent: Monday, July 09, 2007 12:10 PM > >>>> To: Papadopoulos Georgios > >>>> Cc: users(a)openser.org > >>>> Subject: Re: [OpenSER-Users] crash when checking empty > destination > >>>> uri > >>>> > >>>> Hello George, > >>>> > >>>> if loose_route returns true, the $du should be set. Can > >>>> > >> you send me a > >> > >>>> sample of such ACK along with the backtrace? > >>>> > >>>> Cheers, > >>>> Daniel > >>>> > >>>> > >>>> On 07/09/07 10:52, Papadopoulos Georgios wrote: > >>>> > >>>> > >>>>> Hi Daniel, > >>>>> > >>>>> Thank you for looking into this. I got from SVN the patch > >>>>> > >> for rev. > >> > >>>>> 2420 and I applied it to my 1.2 branch. OpenSER still > >>>>> > >>>>> > >>>> crashes at the > >>>> > >>>> > >>>>> same point. Did I get the right patch? The one I have > >>>>> > >>>>> > >>>> changes one line > >>>> > >>>> > >>>>> on route.c and transformations.c. I will add it at the end > >>>>> > >>>>> > >>>> of the email. > >>>> > >>>> > >>>>> Best regards > >>>>> > >>>>> George > >>>>> > >>>>> > >>>>> Index: route.c^M > >>>>> > >>>>> > >>>>> > >> > ===================================================================^M > >> > >>>> > >>>> > >>>>> --- route.c (revision 2419)^M > >>>>> +++ route.c (revision 2420)^M > >>>>> @@ -446,6 +446,9 @@^M > >>>>> char backup2; > >>>>> str res; > >>>>> xl_value_t value; > >>>>> + > >>>>> + if(ival==NULL || ival->s==NULL) > >>>>> + goto error; > >>>>> > >>>>> res.s = 0; res.len = 0; > >>>>> if(opd->type == SCRIPTVAR_ST) > >>>>> Index: transformations.c^M > >>>>> > >>>>> > >>>>> > >> > ===================================================================^M > >> > >>>> > >>>> > >>>>> --- transformations.c (revision 2419)^M > >>>>> +++ transformations.c (revision 2420)^M > >>>>> @@ -71,7 +71,8 @@^M > >>>>> char *p, *s; > >>>>> str st; > >>>>> xl_value_t v; > >>>>> - if(val==NULL) > >>>>> + > >>>>> + if(val==NULL || val->flags&XL_VAL_NULL) > >>>>> return -1; > >>>>> > >>>>> switch(subtype) > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>>> -----Original Message----- > >>>>>> From: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] > >>>>>> Sent: Friday, July 06, 2007 9:19 PM > >>>>>> To: Papadopoulos Georgios > >>>>>> Cc: users(a)openser.org > >>>>>> Subject: Re: [OpenSER-Users] crash when checking empty > >>>>>> > >> destination > >> > >>>>>> uri > >>>>>> > >>>>>> Hello, > >>>>>> > >>>>>> On 07/06/07 17:09, Papadopoulos Georgios wrote: > >>>>>> > >>>>>> > >>>>>> > >>>>>>> Hello, > >>>>>>> > >>>>>>> I have the following line in my script which causes OpenSER > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>> (1.2) to > >>>>>> > >>>>>> > >>>>>> > >>>>>>> crash when the execution gets there for an ACK. > >>>>>>> if > ($du=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)") > >>>>>>> It gets at this line when an ACK is received and after > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>> loose_route() > >>>>>> > >>>>>> > >>>>>> > >>>>>>> returns true. So I am guessing the destination uri is > >>>>>>> > >>>>>>> > >>>> null at that > >>>> > >>>> > >>>>>>> point and the check causes OpenSER to crash. > >>>>>>> > >>>>>>> If I add the following check then everything works fine. > >>>>>>> if (isdsturiset() && > >>>>>>> $du=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)") > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>> can you test with devel version? I applied a patch, and if > >>>>>> > >>>>>> > >>>> ok, I will > >>>> > >>>> > >>>>>> backport. Thanks for reporting. > >>>>>> > >>>>>> Cheers, > >>>>>> Daniel > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>>> > >>>>>>> thank you > >>>>>>> > >>>>>>> George > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> Disclaimer > >>>>>>> > >>>>>>> The information in this e-mail and any attachments is > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>> confidential. It > >>>>>> > >>>>>> > >>>>>> > >>>>>>> is intended solely for the attention and use of the named > >>>>>>> addressee(s). If you are not the intended recipient, > or person > >>>>>>> responsible for delivering this information to the intended > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>> recipient, > >>>>>> > >>>>>> > >>>>>> > >>>>>>> please notify the sender immediately. Unless you are > >>>>>>> > >> the intended > >> > >>>>>>> recipient or his/her representative you are not > >>>>>>> > >>>>>>> > >>>> authorized to, and > >>>> > >>>> > >>>>>>> must not, read, copy, distribute, use or retain this > >>>>>>> > >>>>>>> > >>>> message or any > >>>> > >>>> > >>>>>>> part of it. E-mail transmission cannot be guaranteed to be > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>> secure or > >>>>>> > >>>>>> > >>>>>> > >>>>>>> error-free as information could be intercepted, > >>>>>>> > >> corrupted, lost, > >> > >>>>>>> destroyed, arrive late or incomplete, or contain viruses. > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >> > --------------------------------------------------------------------- > >> > >>>> > >>>> > >>>>>> - > >>>>>> > >>>>>> > >>>>>> > >>>>>>> -- > >>>>>>> > >>>>>>> _______________________________________________ > >>>>>>> Users mailing list > >>>>>>> Users(a)openser.org > >>>>>>> http://openser.org/cgi-bin/mailman/listinfo/users > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>> > >>>>> > >>>>> > >>> > >>> > > > > >

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RE: [OpenSER-Users] crash when checking empty destination uri
by Papadopoulos Georgios 09 Jul '07

09 Jul '07

#0 0x080844a7 in comp_s2s (op=Variable "op" is not available. ) at route.c:748 #1 0x080847ef in comp_scriptvar (msg=0x81787b0, op=21, left=0x8170a60, right=0x8170a6c) at route.c:948 #2 0x080850a5 in eval_expr (e=0x8170a58, msg=0x81787b0, val=0x0) at route.c:1323 #3 0x0808517e in eval_expr (e=0x8170c38, msg=0x81787b0, val=0x0) at route.c:1365 #4 0x08084dcf in eval_expr (e=0x8170c60, msg=0x81787b0, val=0x0) at route.c:1374 #5 0x08050b5c in do_action (a=0x8170ed0, msg=0x81787b0) at action.c:783 #6 0x0805329c in run_action_list (a=0x81708c0, msg=0x81787b0) at action.c:131 #7 0x08051c20 in do_action (a=0x8166648, msg=0x81787b0) at action.c:111 #8 0x0805329c in run_action_list (a=0x8155010, msg=0x81787b0) at action.c:131 #9 0x080534cf in run_top_route (a=0x8155010, msg=0x81787b0) at action.c:111 #10 0x0807b4a6 in receive_msg ( buf=0x811b160 "ACK sip:DEMO1@213.5.43.4 SIP/2.0\r\nVia: SIP/2.0/UDP 213.5.17.78:5060;branch=z9hG4bK7a9dbc5b;rport\r\nFrom: \"GEOP Papadopoul\" <sip:2116872933@altecnet.gr>;tag=as4aecac7c\r\nTo: <sip:DEMO1@213.5.43.4>;tag=b3"..., len=411, rcv_info=0xbffb2d40) at receive.c:156 #11 0x080a4475 in udp_rcv_loop () at udp_server.c:451 #12 0x0806a116 in main_loop () at main.c:834 #13 0x0806c007 in main (argc=6, argv=0xbffb2f24) at main.c:1393 > -----Original Message----- > From: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] > Sent: Monday, July 09, 2007 3:03 PM > To: Papadopoulos Georgios > Cc: users(a)openser.org > Subject: Re: [OpenSER-Users] crash when checking empty destination uri > > Hello, > > On 07/09/07 13:50, Papadopoulos Georgios wrote: > > Hi Daniel, > > > > Here is the ACK packet > > > > U 2007/07/09 13:13:31.577697 213.5.17.78:5060 -> > 213.5.43.4:5060 ACK > > sip:demo1@213.5.168.214:54656 SIP/2.0. > > Via: SIP/2.0/UDP 213.5.17.78:5060;branch=z9hG4bK504461f5;rport. > > Route: <sip:213.5.43.4;lr=on;ftag=as2f3574dc>. > > From: "GEOP Papadopoul" <sip:2116872933@altecnet.gr>;tag=as2f3574dc. > > To: <sip:DEMO1@213.5.43.4>;tag=80299598395. > > Contact: <sip:2116872933@213.5.17.78>. > > Call-ID: 088e7cdc7176016349ad45ea69468b7b(a)altecnet.gr. > > CSeq: 102 ACK. > > User-Agent: Asterisk PBX. > > Max-Forwards: 70. > > Content-Length: 0. > > > > How do I get the backtrace that you need? All I get from > gdb is this: > > #0 0x080844a7 in ?? () > > #1 0xbffb26e0 in ?? () > > #2 0x00b56cee in ?? () > > #3 0x00000000 in ?? () > > > > Check if you got a core file, and then: > gdb /path/to/openser /path/to/core_file > > # bt > > Cheers, > Daniel > > > Best regards > > > > George > > > > > > > > > >> -----Original Message----- > >> From: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] > >> Sent: Monday, July 09, 2007 12:10 PM > >> To: Papadopoulos Georgios > >> Cc: users(a)openser.org > >> Subject: Re: [OpenSER-Users] crash when checking empty destination > >> uri > >> > >> Hello George, > >> > >> if loose_route returns true, the $du should be set. Can > you send me a > >> sample of such ACK along with the backtrace? > >> > >> Cheers, > >> Daniel > >> > >> > >> On 07/09/07 10:52, Papadopoulos Georgios wrote: > >> > >>> Hi Daniel, > >>> > >>> Thank you for looking into this. I got from SVN the patch > for rev. > >>> 2420 and I applied it to my 1.2 branch. OpenSER still > >>> > >> crashes at the > >> > >>> same point. Did I get the right patch? The one I have > >>> > >> changes one line > >> > >>> on route.c and transformations.c. I will add it at the end > >>> > >> of the email. > >> > >>> Best regards > >>> > >>> George > >>> > >>> > >>> Index: route.c^M > >>> > >>> > >> > ===================================================================^M > >> > >>> --- route.c (revision 2419)^M > >>> +++ route.c (revision 2420)^M > >>> @@ -446,6 +446,9 @@^M > >>> char backup2; > >>> str res; > >>> xl_value_t value; > >>> + > >>> + if(ival==NULL || ival->s==NULL) > >>> + goto error; > >>> > >>> res.s = 0; res.len = 0; > >>> if(opd->type == SCRIPTVAR_ST) > >>> Index: transformations.c^M > >>> > >>> > >> > ===================================================================^M > >> > >>> --- transformations.c (revision 2419)^M > >>> +++ transformations.c (revision 2420)^M > >>> @@ -71,7 +71,8 @@^M > >>> char *p, *s; > >>> str st; > >>> xl_value_t v; > >>> - if(val==NULL) > >>> + > >>> + if(val==NULL || val->flags&XL_VAL_NULL) > >>> return -1; > >>> > >>> switch(subtype) > >>> > >>> > >>> > >>> > >>> > >>>> -----Original Message----- > >>>> From: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] > >>>> Sent: Friday, July 06, 2007 9:19 PM > >>>> To: Papadopoulos Georgios > >>>> Cc: users(a)openser.org > >>>> Subject: Re: [OpenSER-Users] crash when checking empty > destination > >>>> uri > >>>> > >>>> Hello, > >>>> > >>>> On 07/06/07 17:09, Papadopoulos Georgios wrote: > >>>> > >>>> > >>>>> Hello, > >>>>> > >>>>> I have the following line in my script which causes OpenSER > >>>>> > >>>>> > >>>> (1.2) to > >>>> > >>>> > >>>>> crash when the execution gets there for an ACK. > >>>>> if ($du=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)") > >>>>> It gets at this line when an ACK is received and after > >>>>> > >>>>> > >>>> loose_route() > >>>> > >>>> > >>>>> returns true. So I am guessing the destination uri is > >>>>> > >> null at that > >> > >>>>> point and the check causes OpenSER to crash. > >>>>> > >>>>> If I add the following check then everything works fine. > >>>>> if (isdsturiset() && > >>>>> $du=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)") > >>>>> > >>>>> > >>>> can you test with devel version? I applied a patch, and if > >>>> > >> ok, I will > >> > >>>> backport. Thanks for reporting. > >>>> > >>>> Cheers, > >>>> Daniel > >>>> > >>>> > >>>> > >>>>> > >>>>> thank you > >>>>> > >>>>> George > >>>>> > >>>>> > >>>>> > >>>>> Disclaimer > >>>>> > >>>>> The information in this e-mail and any attachments is > >>>>> > >>>>> > >>>> confidential. It > >>>> > >>>> > >>>>> is intended solely for the attention and use of the named > >>>>> addressee(s). If you are not the intended recipient, or person > >>>>> responsible for delivering this information to the intended > >>>>> > >>>>> > >>>> recipient, > >>>> > >>>> > >>>>> please notify the sender immediately. Unless you are > the intended > >>>>> recipient or his/her representative you are not > >>>>> > >> authorized to, and > >> > >>>>> must not, read, copy, distribute, use or retain this > >>>>> > >> message or any > >> > >>>>> part of it. E-mail transmission cannot be guaranteed to be > >>>>> > >>>>> > >>>> secure or > >>>> > >>>> > >>>>> error-free as information could be intercepted, > corrupted, lost, > >>>>> destroyed, arrive late or incomplete, or contain viruses. > >>>>> > >>>>> > >>>>> > >>>>> > >> > --------------------------------------------------------------------- > >> > >>>> - > >>>> > >>>> > >>>>> -- > >>>>> > >>>>> _______________________________________________ > >>>>> Users mailing list > >>>>> Users(a)openser.org > >>>>> http://openser.org/cgi-bin/mailman/listinfo/users > >>>>> > >>>>> > >>>>> > >>> > >>> > > > > >

2 1

RE: [OpenSER-Users] crash when checking empty destination uri
by Papadopoulos Georgios 09 Jul '07

09 Jul '07

Hi Daniel, Here is the ACK packet U 2007/07/09 13:13:31.577697 213.5.17.78:5060 -> 213.5.43.4:5060 ACK sip:demo1@213.5.168.214:54656 SIP/2.0. Via: SIP/2.0/UDP 213.5.17.78:5060;branch=z9hG4bK504461f5;rport. Route: <sip:213.5.43.4;lr=on;ftag=as2f3574dc>. From: "GEOP Papadopoul" <sip:2116872933@altecnet.gr>;tag=as2f3574dc. To: <sip:DEMO1@213.5.43.4>;tag=80299598395. Contact: <sip:2116872933@213.5.17.78>. Call-ID: 088e7cdc7176016349ad45ea69468b7b(a)altecnet.gr. CSeq: 102 ACK. User-Agent: Asterisk PBX. Max-Forwards: 70. Content-Length: 0. How do I get the backtrace that you need? All I get from gdb is this: #0 0x080844a7 in ?? () #1 0xbffb26e0 in ?? () #2 0x00b56cee in ?? () #3 0x00000000 in ?? () Best regards George > -----Original Message----- > From: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] > Sent: Monday, July 09, 2007 12:10 PM > To: Papadopoulos Georgios > Cc: users(a)openser.org > Subject: Re: [OpenSER-Users] crash when checking empty destination uri > > Hello George, > > if loose_route returns true, the $du should be set. Can you > send me a sample of such ACK along with the backtrace? > > Cheers, > Daniel > > > On 07/09/07 10:52, Papadopoulos Georgios wrote: > > Hi Daniel, > > > > Thank you for looking into this. I got from SVN the patch for rev. > > 2420 and I applied it to my 1.2 branch. OpenSER still > crashes at the > > same point. Did I get the right patch? The one I have > changes one line > > on route.c and transformations.c. I will add it at the end > of the email. > > > > Best regards > > > > George > > > > > > Index: route.c^M > > > ===================================================================^M > > --- route.c (revision 2419)^M > > +++ route.c (revision 2420)^M > > @@ -446,6 +446,9 @@^M > > char backup2; > > str res; > > xl_value_t value; > > + > > + if(ival==NULL || ival->s==NULL) > > + goto error; > > > > res.s = 0; res.len = 0; > > if(opd->type == SCRIPTVAR_ST) > > Index: transformations.c^M > > > ===================================================================^M > > --- transformations.c (revision 2419)^M > > +++ transformations.c (revision 2420)^M > > @@ -71,7 +71,8 @@^M > > char *p, *s; > > str st; > > xl_value_t v; > > - if(val==NULL) > > + > > + if(val==NULL || val->flags&XL_VAL_NULL) > > return -1; > > > > switch(subtype) > > > > > > > > > >> -----Original Message----- > >> From: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] > >> Sent: Friday, July 06, 2007 9:19 PM > >> To: Papadopoulos Georgios > >> Cc: users(a)openser.org > >> Subject: Re: [OpenSER-Users] crash when checking empty destination > >> uri > >> > >> Hello, > >> > >> On 07/06/07 17:09, Papadopoulos Georgios wrote: > >> > >>> Hello, > >>> > >>> I have the following line in my script which causes OpenSER > >>> > >> (1.2) to > >> > >>> crash when the execution gets there for an ACK. > >>> if ($du=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)") > >>> It gets at this line when an ACK is received and after > >>> > >> loose_route() > >> > >>> returns true. So I am guessing the destination uri is > null at that > >>> point and the check causes OpenSER to crash. > >>> > >>> If I add the following check then everything works fine. > >>> if (isdsturiset() && > >>> $du=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)") > >>> > >> can you test with devel version? I applied a patch, and if > ok, I will > >> backport. Thanks for reporting. > >> > >> Cheers, > >> Daniel > >> > >> > >>> > >>> thank you > >>> > >>> George > >>> > >>> > >>> > >>> Disclaimer > >>> > >>> The information in this e-mail and any attachments is > >>> > >> confidential. It > >> > >>> is intended solely for the attention and use of the named > >>> addressee(s). If you are not the intended recipient, or person > >>> responsible for delivering this information to the intended > >>> > >> recipient, > >> > >>> please notify the sender immediately. Unless you are the intended > >>> recipient or his/her representative you are not > authorized to, and > >>> must not, read, copy, distribute, use or retain this > message or any > >>> part of it. E-mail transmission cannot be guaranteed to be > >>> > >> secure or > >> > >>> error-free as information could be intercepted, corrupted, lost, > >>> destroyed, arrive late or incomplete, or contain viruses. > >>> > >>> > >>> > >> > --------------------------------------------------------------------- > >> - > >> > >>> -- > >>> > >>> _______________________________________________ > >>> Users mailing list > >>> Users(a)openser.org > >>> http://openser.org/cgi-bin/mailman/listinfo/users > >>> > >>> > > > > >

2 1

[OpenSER-Users] Re: Presence, Database Error !!
by KUMAR 09 Jul '07

09 Jul '07

Thanks Henning, The remote_cseq was spelled wrongly kumar

1 0

RE: [OpenSER-Users] crash when checking empty destination uri
by Papadopoulos Georgios 09 Jul '07

09 Jul '07

Hi Daniel, Thank you for looking into this. I got from SVN the patch for rev. 2420 and I applied it to my 1.2 branch. OpenSER still crashes at the same point. Did I get the right patch? The one I have changes one line on route.c and transformations.c. I will add it at the end of the email. Best regards George Index: route.c^M ===================================================================^M --- route.c (revision 2419)^M +++ route.c (revision 2420)^M @@ -446,6 +446,9 @@^M char backup2; str res; xl_value_t value; + + if(ival==NULL || ival->s==NULL) + goto error; res.s = 0; res.len = 0; if(opd->type == SCRIPTVAR_ST) Index: transformations.c^M ===================================================================^M --- transformations.c (revision 2419)^M +++ transformations.c (revision 2420)^M @@ -71,7 +71,8 @@^M char *p, *s; str st; xl_value_t v; - if(val==NULL) + + if(val==NULL || val->flags&XL_VAL_NULL) return -1; switch(subtype) > -----Original Message----- > From: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] > Sent: Friday, July 06, 2007 9:19 PM > To: Papadopoulos Georgios > Cc: users(a)openser.org > Subject: Re: [OpenSER-Users] crash when checking empty destination uri > > Hello, > > On 07/06/07 17:09, Papadopoulos Georgios wrote: > > Hello, > > > > I have the following line in my script which causes OpenSER > (1.2) to > > crash when the execution gets there for an ACK. > > if ($du=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)") > > It gets at this line when an ACK is received and after > loose_route() > > returns true. So I am guessing the destination uri is null at that > > point and the check causes OpenSER to crash. > > > > If I add the following check then everything works fine. > > if (isdsturiset() && > > $du=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)") > can you test with devel version? I applied a patch, and if > ok, I will backport. Thanks for reporting. > > Cheers, > Daniel > > > > > thank you > > > > George > > > > > > > > Disclaimer > > > > The information in this e-mail and any attachments is > confidential. It > > is intended solely for the attention and use of the named > > addressee(s). If you are not the intended recipient, or person > > responsible for delivering this information to the intended > recipient, > > please notify the sender immediately. Unless you are the intended > > recipient or his/her representative you are not authorized to, and > > must not, read, copy, distribute, use or retain this message or any > > part of it. E-mail transmission cannot be guaranteed to be > secure or > > error-free as information could be intercepted, corrupted, lost, > > destroyed, arrive late or incomplete, or contain viruses. > > > > > ---------------------------------------------------------------------- > > -- > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users > > >

2 1

Re: [Serusers] How SER communicate to other SIP server
by roayu 09 Jul '07

09 Jul '07

Hi, Szasz! I found that both UAC are only sending out packets but didn't receive any packets from either party. For instance, softphone A only send packets to serverA whereas softphone B only send packets to serverB. Is there any other settings that I need configure as well? Which firewall settings that I need to check (server/client)? Thanks. Roa Yu -----Original Message----- From: Szasz Szabolcs [mailto:sszasz@digicomm.ro] Sent: Monday, July 09, 2007 4:40 PM To: roayu Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] How SER communicate to other SIP server Hi! See your firewall settings, probably you don't allow to forward the RTP packets. Szasz Szabolcs ----- Original Message ----- From: "roayu" <roayu(a)ctisys.net> To: "Olaf Bergmann" <Olaf.Bergmann(a)freenet-ag.de> Cc: <serusers(a)lists.iptel.org> Sent: Monday, July 09, 2007 11:12 AM Subject: Re: [Serusers] How SER communicate to other SIP server > Hi, Olaf! > > I managed to call to the other SIP server. Unfortunately, both sides can't > hear anything. What could be the problem? Could it be the NAT issue? If > so, > where should I allocate them to use the mediaproxy(before/after > rewritehostport())? > > > Thanks. > > Best regards, > Roa Yu > > > -----Original Message----- > From: Olaf Bergmann [mailto:Olaf.Bergmann@freenet-ag.de] > Sent: Wednesday, July 04, 2007 2:14 PM > To: roayu > Cc: serusers(a)lists.iptel.org > Subject: Re: [Serusers] How SER communicate to other SIP server > > roayu wrote: >> Hi, Olaf! >> >>>From what I understand about forward() is that it will forward the >>>request >> to the given destination statelessly. Whereas for t_relay(), it will >> relay > a >> message to the destination stated in current URI statefully. How do I get >> the URI to point to the other SIP server by using t_relay() as I couldn't >> assign any IP address for t_relay(). I managed to assign an IP address by >> using forward(). >> >> I tried to use both. Unfortunately, I still failed to forward the message > to >> the another SIP server to establish the call. Is there any functions I > need >> to use before using either those 2 functions? FYI, I will only call them >> when it starts to INVITE. > > 482 Loop Detected > > The function you are looking for is called rewritehostport (or > rewritehost, if you like). But you could also use > t_relay_to_{udp,tcp} to specify a literal transport address where > you want your request being routed to. > <http://www.iptel.org/ser/doc/search> gives more information on > these functions. > > Regards, > Olaf > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] rtpproxy bridge mode ip addresses used
by Shaun Hofer 09 Jul '07

09 Jul '07

I used examples/alg.cfg to set-up SER between two subnets (both private at this stage) and had the UA (end points) connected directly to SER machine, one to each interface. This all worked fine, but the problem I've found is that when I started putting UA onto other subnets I would loose the rtp packets. It seems rtpproxy uses one IP addresses and not both for sending rtp, when in bridge mode. This causes problems when the two subnets are non-routable except through SER and rtpproxy. UA (10.0.10.212) ----- (10.0.10.1) Router (10.0.100.1) ---- (10.0.100.2) SER (10.0.11.2) ---- UA (10.0.11.150) From packet captures I can see that UA 10.0.10.212 tries to send RTP packets to 10.0.11.2 which the router knows nothing about and doesn't route to 10.0.100.2, but the SIP packets seem to have the correct src and dest. I know I can put static routes in etc for the router. We hope to deploy this where the private IP address will be replaced by public ones, and there will be routers connected to one of SER machine's interface that we have no control over. Any ideas on how I can force rtpproxy to use 10.0.100.2 when going to 10.0.10.212 and still use 10.0.11.2 for 10.0.11.150 or is this limitation of rtpproxy ? netstat when UA 10.0.11.150 calls UA 10.0.10.212: udp 0 0 10.0.11.2:8016 0.0.0.0:* 19347/rtpproxy udp 0 0 10.0.11.2:8017 0.0.0.0:* 19347/rtpproxy udp 0 0 10.0.11.2:8018 0.0.0.0:* 19347/rtpproxy udp 0 0 10.0.11.2:8019 0.0.0.0:* 19347/rtpproxy udp 0 0 127.0.0.1:7000 0.0.0.0:* 19347/rtpproxy I have also tried swapping the -l ip address around, but still lost audio in one direction on calls. rtpproxy -l 10.0.100.2 10.0.11.2 -s udp:127.0.0.1 7000 Thanks in advance -- Shaun Hofer VoxPak Pty Ltd ********************************************************** If you receive this email by mistake, please notify us and do not make any use of the email. We do not waive any privilege, confidentiality or copyright associated with it.

1 0

[OpenSER-Users] [OT] Auto-replies from "Jean-Pierre RAMOUL" <jra@teamlog.com>
by Iñaki Baz Castillo 09 Jul '07

09 Jul '07

Hi, there is a user of this list who has configured a vacation mesage and for each mail I send to this list I receive a message: "Bonjour, Je suis actuellement en congés, je traiterai votre demande dès mon retour. Cordialement, Jean-Pierre Ramoul" Could this kind of mails be blocked? Regards. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

2 1

Re: [Serusers] How SER communicate to other SIP server
by roayu 09 Jul '07

09 Jul '07

Hi, Olaf! I managed to call to the other SIP server. Unfortunately, both sides can't hear anything. What could be the problem? Could it be the NAT issue? If so, where should I allocate them to use the mediaproxy(before/after rewritehostport())? Thanks. Best regards, Roa Yu -----Original Message----- From: Olaf Bergmann [mailto:Olaf.Bergmann@freenet-ag.de] Sent: Wednesday, July 04, 2007 2:14 PM To: roayu Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] How SER communicate to other SIP server roayu wrote: > Hi, Olaf! > >>From what I understand about forward() is that it will forward the request > to the given destination statelessly. Whereas for t_relay(), it will relay a > message to the destination stated in current URI statefully. How do I get > the URI to point to the other SIP server by using t_relay() as I couldn't > assign any IP address for t_relay(). I managed to assign an IP address by > using forward(). > > I tried to use both. Unfortunately, I still failed to forward the message to > the another SIP server to establish the call. Is there any functions I need > to use before using either those 2 functions? FYI, I will only call them > when it starts to INVITE. 482 Loop Detected The function you are looking for is called rewritehostport (or rewritehost, if you like). But you could also use t_relay_to_{udp,tcp} to specify a literal transport address where you want your request being routed to. <http://www.iptel.org/ser/doc/search> gives more information on these functions. Regards, Olaf

2 1

[OpenSER-Users] Presence, Database Error !!
by KUMAR 09 Jul '07

09 Jul '07

Hi all, I'm trying to make openser work with presence. I'm having few problems making it work. I've attached the openser.cfg that I'm using. The dump I get from the server is as follows. 1(5115) submit_query: Unknown column 'remote_cseq' in 'field list' 1(5115) db_insert: Error while submitting query 1(5115) PRESENCE:update_subscription: ERROR while storing new subscription 1(5115) PRESENCE:update_presentity: ERROR occured 1(5115) PRESENCE:handle_subscribe: ERROR while updating database 4(5125) ERROR:check_content_type: invalid type for a message 4(5125) ERROR: extract_body: content type mismatching 4(5125) ERROR: force_rtp_proxy2: can't extract body from the message Here's what the UA is doing. The UA upon registration is sending a SUBSCRIBE followed by PUBLISH. The server then forwards the PUBLISH. Now, Isn't the server supposed to send NOTIFY ? And what about the error messages resulting from the database operations ? Thank you in advance kumar

2 1

[Serusers] Loading and deleting avp
by tzieleniewski 09 Jul '07

09 Jul '07

Hi, How can one delete a particular avp attribute or avps attributes from the user class and a particular track?? Is it possible to load the particular attribute to the specific class and track? Thanks in advance. Cheers Tomasz

2 3

[Serusers] insert_avp_hf and append_avp_hf result in unknown command, missing loadmodule? error
by tzieleniewski 09 Jul '07

09 Jul '07

Hi, I am using SER rel 2_0_0 from cvs and when I try to invoke insert_avp_method("xxx","yyy") or append_avp_method("xxx","yyy") I get unknown command, missing loadmodule? I have the avp module loaded. What could be the reason for this? Cheers TOmasz

2 1

[Serusers] QUERY - SER Documentation
by Parminder.Nayyar＠relianceada.com 09 Jul '07

09 Jul '07

Hello This is in reference to the product SER available as a open source. We intend to use Solaris 10 OS as a SIP Server using SER. But I could just get the Getting Started article on the website which takes care of specific cases. However , can we get some documentation on the modules and parameters there in? Thanks Parminder Nayyar

1 0

[Serusers] QUERY - SER Documentation
by Parminder.Nayyar＠relianceada.com 09 Jul '07

09 Jul '07

Hello This is in reference to the product SER available as a open source. We intend to use Solaris 10 OS as a SIP Server using SER. But I could just get the Getting Started article on the website which takes care of specific cases. However , can we get some documentation on the modules and parameters there in? Thanks Parminder Nayyar

1 0

[Serusers] Question
by Serg Nikitin 09 Jul '07

09 Jul '07

1 0

[Serusers] How to configure Trunk capacity
by roayu 09 Jul '07

09 Jul '07

Hi, there! Can anyone tell me where can I configure the Trunk capacity for SER? Thank you. Best regards, Roa Yu

1 0

[Serusers] loose_route() function doesn't work? - ser 2_0_0
by tzieleniewski 08 Jul '07

08 Jul '07

Hi, I have the following SIP scenario: UA_A -> SER_A -> other SIP Proxy -> SER_A -> UA_B I want to force SER_A instance to forward the received request to other SIP proxy element and ensure that that request will return to SER_A afterwards. To do that I use an insert_hf() function from textops module inside ser.cfg to insert a double Route header field - first value the sip uri of the 'other SIP proxy' and second one the SER_A sip uri to assure that the forwarded request will return to SER_A. Whenever request enters the ser.cfg script logic my SER code performs the loose_route() check in general. In my situation when request initialy enters the logic it doesn't have any Route headers. When request enters a particular route block responsible for handling this request there is an lookup_user() invocation to check if the user should be serviced by this SER. If lookup_user() returns true and this particular user has a flag set which indicates routing through the extra SIP proxy I add the Route headers and pass the request to the 'FORWARD' route block where I do the t_relay(). My problem is that when I do t_relay() after inserting the Route headers into the request it is not forwarded according to the topmost Route header field value?? Is it bug in ser?? Please point me what do I missed. Thank you in advance. Tomasz

3 4

[Serusers] SER - Getting Started Guide
by Shaun Hofer 08 Jul '07

08 Jul '07

Thought I would let people know that (in case no one realised) SER - Getting Started Guide (html and pdf versions) seems not to be accessible any more. Seems all the links that point to onsip.org on iptel.org/ser/doc/gettingstarted page can't load, yet onsip.org main page can be loaded. Use google cache in the mean time but not sure how up to date it is: http://www.google.com/search?q=ser+gettingstarted&ie=utf-8&oe=utf-8&aq=t&rl… -- Shaun Hofer Email: shaun.hofer(a)voxpak.com ********************************************************** If you receive this email by mistake, please notify us and do not make any use of the email. We do not waive any privilege, confidentiality or copyright associated with it.

2 1

[Serusers] t_relay_to_tcp/udp/tls
by Zappasodi Daniele 08 Jul '07

08 Jul '07

Hello, Is it possible to choose the protocol, without specifying the destination as we do with the t_relay()? With the t_relay_to_[tcp|udp|tls] I can choose the protocol but I must specify a destination address in the config file. Thanks ********************************************************************** The information in this message is confidential and may be legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, or distribution of the message, or any action or omission taken by you in reliance on it, is prohibited and may be unlawful. Please immediately contact the sender if you have received this message inerror. **********************************************************************

2 3

[Serusers] iptel.org bug tracker down?
by Jan Andres 07 Jul '07

07 Jul '07

Hi, Since yesterday (I think) I always run into timeouts when trying to access the SER bug tracker at iptel.org. Once I also got an error response (after a long delay) which indicated that the server's JVM was running out of memory. Just wanted to report this in case nobody noticed it so far. Or am I the only one having problems accessing the tracker? Regards, Jan -- Jan Andres <jan.andres(a)freenet.ag>

2 1

[OpenSER-Users] mk_proxy: could not resolve hostname
by Pande, Piyush 06 Jul '07

06 Jul '07

Hi, I am new to SER and have been having this problem for a while which I'm not sure how to resolve. Any pointers would be greatly appreciated On the openser.cfg when I use the "forward" or the "send" calls with an IP inside I see the following upon running openser ( Note I have masked the real IP's) ./openser Listening on udp: 208.xxx.xx.xx tcp: 208.xxx.xx.xx WARNING: no fork mode stateless - initializing 0(0) Maxfwd module- initializing textops - initializing 0(0) AUTH module - initializing 0(0) AUTH_DB module - initializing 0(0) ERROR: mk_proxy: could not resolve hostname: "sip:208.1xx.44.yy:5060" ERROR: error -478 while trying to fix configuration Anybody have any ideas? I have tried putting a DNS name to the IP address as well but it comes back with the same error. Regards Piyush

1 0

[OpenSER-Users] crash when checking empty destination uri
by Papadopoulos Georgios 06 Jul '07

06 Jul '07

Hello, I have the following line in my script which causes OpenSER (1.2) to crash when the execution gets there for an ACK. if ($du=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)") It gets at this line when an ACK is received and after loose_route() returns true. So I am guessing the destination uri is null at that point and the check causes OpenSER to crash. If I add the following check then everything works fine. if (isdsturiset() && $du=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)") thank you George Disclaimer The information in this e-mail and any attachments is confidential. It is intended solely for the attention and use of the named addressee(s). If you are not the intended recipient, or person responsible for delivering this information to the intended recipient, please notify the sender immediately. Unless you are the intended recipient or his/her representative you are not authorized to, and must not, read, copy, distribute, use or retain this message or any part of it. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.

2 1

[Serusers] Used exec module
by inge 06 Jul '07

06 Jul '07

Hi all, I find a way to execute an external script when I get a trigger. The trigger could be BYE or INVITE for example. I read the documentation and some examples but I don't find tips to help me. Is it possible to have a "return false" in the script wich is executed and use it in "if (exec())" in SER ? If it's possible I can do IF ELSE in ser.cfg according to the return of the script. The best should be that I return a value in the script and I will able to test it in SER. I run SER 0.9.6. Thanks for your support and ideas. Adrien

1 1

[OpenSER-Users] OpenSer replying to wrong NAT port
by Iñaki Baz Castillo 06 Jul '07

06 Jul '07

Hi, I'm starting with OpenSer 1.2.1-1 and have a strange problem. This occurs always with users behind NAT with a mapped NAT port different of his device SIP port. For example: * user 807: device: sip:192.186.0.204:5061 is mapped in the NAT router as: sip:212.121.235.18:1055 So in "location" table I see: --------------------------------- username contact received 807 sip:807@192.168.1.204:5061 sip:212.121.235.18:1055 800 sip:800@212.121.235.18:5080 NULL --------------------------------- 800 is a user with STUN. I have some services in Asterisk accesible via public and private numbers in OpenSer. Public numbers (500) don't need auth in INVITE. When 807 call to any private number (or internal users as 800) OpenSer ask him for auth, but OpenSer replies this messaje to a wrong port: --------------------------------- # 807 INVITE 800 U 212.121.235.18:1055 -> 82.94.0.210:5060 INVITE sip:800@openser.domain.org SIP/2.0 # OpenSer 407 U 82.94.0.210:5060 -> 212.121.235.18:5061 <--- WRONG DST PORT SIP/2.0 407 Proxy Authentication Required --------------------------------- As you see, OpenSer send the 407 message to the NAT IP but to the device SIP port instead of the NAT port. So 807 sends again and again the INVITE. But if 807 calls to a public number (500) it get the call but when 807 does BYE, the response to that BYE is again wrong: --------------------------------- # 807 INVITE 500 U 212.121.235.18:1055 -> 82.94.0.210:5060 INVITE sip:500@openser.domain.org SIP/2.0 # OpenSer OK U 82.94.0.210:5060 -> 212.121.235.18:1055 SIP/2.0 200 OK # 807 BYE 500 U 212.121.235.18:1055 -> 82.94.0.210:5060 BYE sip:1800@82.94.0.210:5070 SIP/2.0 # OpenSer OK U 82.94.0.210:5060 -> 212.121.235.18:5061 <--- WRONG DST PORT SIP/2.0 200 OK --------------------------------- I hope this is a bug in my OpenSer configuration. Please, could anyone tell me with funcions related to this issue should I verify? Thanks a lot. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

2 2

[Serusers] need help with media proxu
by Kirk McCalla 06 Jul '07

06 Jul '07

Hi I am trying st set up media proxy on my sip express router setup. this is the problem that I am running into. I have placed Media proxy on a seperate box and I have configured the INI file. I am able to start the media proxy. I then go to the sip express router and I add the proxydispatcher to the box. the documentation that is on onsip.org does not tell me what directory this should go into. this is what I decided to do. I placed the file in /usr/share. I then tried to start the proxydispatcher.py by typing python proxydispatcher.py and it failed. This is the error that I got. # python proxydispatcher.py Traceback (most recent call last): File "proxydispatcher.py", line 31, in ? import daemon ImportError: No module named daemon I am a little confused as to how to set up the whole media proxy nat traversal. Can any one let me know if I did this correctluy up to this point. Part 2. I cant get any external phones to register onto the server. The getting started config works fine. I am now using a nathelper config and I cant get any external phones to register. Help help

1 0

[Users] Query in the openser.cfg file
by sreekanth 06 Jul '07

06 Jul '07

Hi, What is the use of the config item children in the openser.cfg file. How is it related to command line options (-n process-no / -N tcp process-no). Rgds, Sreekanth

2 1

Re: [Users] Many crashes on Openser system
by Klaus Darilion 06 Jul '07

06 Jul '07

looks like (not sure) process 1546 received a reply, does the tm matching, and then forwards the reply. The last entry Jun 7 10:42:04 marge openser[1546]: DBG: trans=0xb577a938, callback type 128, id 0 entered references a callback. Maybe the tm module calls a registered callback function and this function crashes. Do you use any self-written modules? Which modules are you using which uses callbacks (e.g. acc, dialog ...)? Try to disable these modules. regards klaus Antonio Higuera wrote: > Thanks for your reply. > > You have attached a file with the logs. Be aware that the coredump > backtrace > i sent in my previous mail maybe is not the same crash you can see in the > log. > > Thanks again for your help. > > > Jun 7 10:42:04 marge openser[1546]: SIP Reply (status): > > Jun 7 10:42:04 marge openser[1546]: version: <SIP/2.0> > > Jun 7 10:42:04 marge openser[1546]: status: <407> > > Jun 7 10:42:04 marge openser[1546]: reason: <Proxy Authentication Required> > > Jun 7 10:42:04 marge openser[1546]: parse_headers: flags=2 > > Jun 7 10:42:04 marge openser[1546]: Found param type 232, <branch> = < > z9hG4bK6202.244bec76.0>; state=6 > > Jun 7 10:42:04 marge openser[1546]: Found param type 234, <received> = < > 192.168.1.129>; state=16 > > Jun 7 10:42:04 marge openser[1546]: end of header reached, state=5 > > Jun 7 10:42:04 marge openser[1546]: parse_headers: Via found, flags=2 > > Jun 7 10:42:04 marge openser[1546]: parse_headers: this is the first via > > Jun 7 10:42:04 marge openser[1546]: After parse_msg... > > Jun 7 10:42:04 marge openser[1546]: DEBUG:forward_reply: found module tm, > passing reply to it > > Jun 7 10:42:04 marge openser[1546]: DEBUG: t_check: start=0xffffffff > > Jun 7 10:42:04 marge openser[1546]: parse_headers: flags=22 > > Jun 7 10:42:04 marge openser[1546]: Found param type 232, <branch> = > <z9hG4bKcc503ea3c48d6b766>; state=16 > > Jun 7 10:42:04 marge openser[1546]: end of header reached, state=5 > > Jun 7 10:42:04 marge openser[1546]: parse_headers: Via found, flags=22 > > Jun 7 10:42:04 marge openser[1546]: parse_headers: this is the second via > > Jun 7 10:42:04 marge openser[1546]: DEBUG: add_param: tag=as4b55784e > > Jun 7 10:42:04 marge openser[1546]: DEBUG:parse_to:end of header reached, > state=29 > > Jun 7 10:42:04 marge openser[1546]: DBUG:parse_to: display={21104}, ruri={ > sip:21104@barik.com} > > Jun 7 10:42:04 marge openser[1546]: DEBUG: get_hdr_field: <To> [44]; uri=[ > sip:21104@barik.com] > > Jun 7 10:42:04 marge openser[1546]: DEBUG: to body [21104 < > sip:21104@barik.com>] > > Jun 7 10:42:04 marge openser[1546]: get_hdr_field: cseq <CSeq>: <30425> > <INVITE> > > Jun 7 10:42:04 marge openser[1546]: parse_headers: flags=8 > > Jun 7 10:42:04 marge openser[1546]: DEBUG: t_reply_matching: hash 8230 > label > 1741599810 branch 0 > > Jun 7 10:42:04 marge openser[1546]: DEBUG:tm:REF_UNSAFE: after is 1 > > Jun 7 10:42:04 marge openser[1546]: DEBUG: t_reply_matching: reply matched > (T=0xb577a938)! > > Jun 7 10:42:04 marge openser[1546]: DBG: trans=0xb577a938, callback type 2, > id 0 entered > > Jun 7 10:42:04 marge openser[1546]: parse_headers: flags=8 > > Jun 7 10:42:04 marge openser[1546]: DEBUG: t_check: end=0xb577a938 > > Jun 7 10:42:04 marge openser[1546]: DEBUG:tm:reply_received: org. status > uas=100, uac[0]=0 local=0 is_invite=1) > > Jun 7 10:42:04 marge openser[1546]: parse_headers: flags=8 > > Jun 7 10:42:04 marge openser[1546]: DEBUG:tm:build_local: using FROM=<From: > Arturo <sip:104@barik.com>;tag=73529158ed >, TO=<To: 21104 < > sip:21104@barik.com>;tag=as4b55784e >, CSEQ_N=<CSeq: 30425> > > Jun 7 10:42:04 marge openser[1546]: DEBUG:tm:t_should_relay_response: > T_code=100, new_code=407 > > Jun 7 10:42:04 marge openser[1546]: DEBUG:tm:t_pick_branch: picked > branch 0, > code 407 > > Jun 7 10:42:04 marge openser[1546]: DEBUG:tm:t_should_relay_response: > dns-failover test: branch=0, last_recv=407, flags=2 > > Jun 7 10:42:04 marge openser[1546]: DEBUG:tm:relay_reply: branch=0, save=0, > relay=0 > > Jun 7 10:42:04 marge openser[1546]: old size: 533, new size: 448 > > Jun 7 10:42:04 marge openser[1546]: build_res_from_sip_res: copied size: > orig:128, new: 43, rest: 405 msg= SIP/2.0 407 Proxy Authentication Required > Via: SIP/2.0/UDP 192.168.1.64:5060;branch=z9hG4bKcc503ea3c48d6b766 From: > Arturo <sip:104@barik.com>;tag=73529158ed To: 21104 > <sip:21104@barik.com>;tag=as4b55784e > Call-ID: f35b076f65bee9c5 CSeq: 30425 INVITE User-Agent: Asterisk PBX > Allow: > INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", > nonce="6f707e3f" > Content-Length: 0 > > Jun 7 10:42:04 marge openser[1546]: DEBUG:tm:set_timer: relative timeout is > 500000 > > Jun 7 10:42:04 marge openser[1546]: DEBUG: add_to_tail_of_timer[4]: > 0xb577aa00 (52180600000) > > Jun 7 10:42:04 marge openser[1546]: DEBUG:tm:set_timer: relative timeout is > 10 > > Jun 7 10:42:04 marge openser[1546]: DEBUG: add_to_tail_of_timer[0]: > 0xb577aa1c (52190) > > Jun 7 10:42:04 marge openser[1546]: DEBUG:tm:relay_reply: sent > buf=0x819cce8: SIP/2.0 4..., shmem=0xb5778158: SIP/2.0 4 > > Jun 7 10:42:04 marge openser[1546]: DBG: trans=0xb577a938, callback type > 128, id 0 entered > > > > 2007/6/20, Klaus Darilion <klaus.mailinglists(a)pernau.at>: >> >> > The following lines are the log generated: >> > >> > Jun 7 10:42:04 marge openser[1545]: receive_msg: cleaning up >> > >> > Jun 7 10:42:04 marge openser[1565]: ERROR: receive_fd: EOF on 8 >> > >> > Jun 7 10:42:04 marge openser[1565]: DBG: handle_ser_child: dead >> child 2, >> > pid >> > 1546 (shutting down?) >> > >> > Jun 7 10:42:04 marge openser[1565]: DBG: io_watch_del (0x812d720, 8, >> -1, >> > 0x0) fd_no=23 called >> > >> > Jun 7 10:42:04 marge openser[1544]: child process 1546 exited by a >> > signal 11 >> >> Could you also post the last log lines of process 1546 ? >> >> >> regards >> klaus >>

3 7

[OpenSER-Users] Strange ACK format gives OpenSER error?
by Tobias Lindgren 06 Jul '07

06 Jul '07

Hi all, after a patch from one of our providers ACKs started to come with R-URI looking like: ACK <sip:192.168.0.1;lr=on;ftag=507454020> SIP/2.0 instead of: ACK sip:192.168.0.1;lr=on;ftag=507454020 SIP/2.0 like it did before the patch. The new ACK format gives an error in OpenSER: Jul 6 11:35:55 ser1 /sbin/openser[9634]: ERROR: parse_uri: bad uri, state 0 parsed: <<sip> (4) / <<sip:192.168.0.1;lr=on;ftag=507454020>> (38) Jul 6 11:35:55 ser1 /sbin/openser[9634]: ERROR: parse_sip_msg_uri: bad uri <<sip:192.168.0.1;lr=on;ftag=507454020>> Jul 6 11:35:55 ser1 /sbin/openser[9634]: loose_route: Error while parsing Request URI Are the new format of the ACKs valid? With the "<>"? If they are valid, the problem lies in OpenSER? Br, /Tobias

2 1

[Serusers] Problems using avps in subst
by Roger Lewau 06 Jul '07

06 Jul '07

Hello I am trying to change the Remote-Party-ID header of a forwarded call but have little luck. I found several examples on the net where avps are used inside a subst but when I try this the resulting Remote-Party-ID is wrong My code: avp_write("$ruri/username","s:calledno"); subst('/^Remote-Party-ID:(.*)>/Remote-Party-ID: <sip:$avp(calledno) user=phone>/ig'); The result: Remote-Party-ID: <sip:$avp(calledno) user=phone>;party=calling;screen=no;privacy=off. Anyone who can point me to the error? I use SER 0.9.6 Med vänliga hälsningar/Kind regards Roger Lewau

1 0

[Serusers] the avp loaded attribute if statement check doesn't work for avp of the value 1 - ser 2_0_0
by tzieleniewski 06 Jul '07

06 Jul '07

Hi, I have the following ser.cfg code: lookup_user("$tu.uid", "@ruri"); load_attrs("$tu", "$tu.uid"); if ($tu.forward_as_blind) { the if statement gives the negative result what is I think wrong because the attribute is present and it's value is 1. here is the output from dump_attrs(): Jul 6 08:35:13 sen ser[1445]: INFO: avp.c:540: class=GLOBAL Jul 6 08:35:13 sen ser[1445]: AVP["lang"]="en" Jul 6 08:35:13 sen ser[1445]: INFO: avp.c:550: track=FROM class=DOMAIN Jul 6 08:35:13 sen ser[1445]: AVP["did"]="voip.rd.touk.pl" Jul 6 08:35:13 sen ser[1445]: AVP["digest_realm"]="voip.rd.touk.pl" Jul 6 08:35:13 sen ser[1445]: INFO: avp.c:560: track=TO class=DOMAIN Jul 6 08:35:13 sen ser[1445]: AVP["did"]="voip.rd.touk.pl" Jul 6 08:35:13 sen ser[1445]: AVP["digest_realm"]="voip.rd.touk.pl" Jul 6 08:35:13 sen ser[1445]: INFO: avp.c:570: track=FROM class=USER Jul 6 08:35:13 sen ser[1445]: INFO: No AVP present Jul 6 08:35:13 sen ser[1445]: INFO: avp.c:580: track=TO class=USER Jul 6 08:35:13 sen ser[1445]: AVP["datetime_created"]="2007-04-18 07:51:34" Jul 6 08:35:13 sen ser[1445]: AVP["email"]="tzl(a)touk.pl" Jul 6 08:35:13 sen ser[1445]: AVP["forward_as"]="192.168.0.165:6060" Jul 6 08:35:13 sen ser[1445]: AVP["forward_as_blind"]=1 Jul 6 08:35:13 sen ser[1445]: AVP["forward_as_default"]=0 Jul 6 08:35:13 sen ser[1445]: AVP["forward_as_invite"]=0 Jul 6 08:35:13 sen ser[1445]: AVP["forward_as_message"]=0 Jul 6 08:35:13 sen ser[1445]: AVP["forward_voicemail"]=0 Jul 6 08:35:13 sen ser[1445]: INFO: avp.c:590: track=FROM class=URI Jul 6 08:35:13 sen ser[1445]: INFO: No AVP present Jul 6 08:35:13 sen ser[1445]: INFO: avp.c:600: track=TO class=URI Jul 6 08:35:13 sen ser[1445]: INFO: No AVP present please point me what did I miss? Thanks in advance Tomasz

1 0

[OpenSER-Users] No NOTIFY after presence unsubscription
by Michel de Boer 05 Jul '07

05 Jul '07

Hi, I am currently experimenting with the presence module (openser v1.2.1). It seems to work really fine. I only encountered a small problem. When my client unsubscribes from the presence event by sending a SUBSCRIBE request with expires=0, then openser responds with a 200 OK. After the 200 OK, I expect to receive a NOTIFY message, but openser never sends this NOTIFY. It is this NOTIFY that should terminate the subscription dialog through its Subscription-State header set to "terminated". RFC 3856 states this: The subscriber can terminate the subscription by sending a SUBSCRIBE, within the dialog, with an Expires header field (which indicates duration of the subscription) value of zero. This causes an immediate termination of the subscription. A NOTIFY request is then generated by the presence agent with the most recent state. This behavior is based on the general requirements for event handling as defined in RFC 3265. Is this a bug or is there some configuration stuff that enables/disables the NOTIFY for a SUBSCRIBE with epxpires=0. A SUBSCRIBE with expires=0 can also arrive as the initial and final SUBSCRIBE for a subscription. In that case it will be fetching the presence information only once. Without the NOTIFY, a fetch does not work. Cheers, Michel -- Michel de Boer www.twinklephone.com

3 8

[OpenSER-Users] Presence Error
by KUMAR 05 Jul '07

05 Jul '07

Hi all, My UA, is sending a SUBSCRIBE to the server with Event: presence and Accept: application/pidf+xml. But the openser upon receiving this returns a 489 Bad Request. The debug at the server shows: Missing or unsupported event header field value Can anyone please tell me the cause of the problem. I've also tried changing Event: presence.winfo , but that too doesn't work. Thank you. kumar

3 4

[Serusers] textops and branch route
by Zappasodi Daniele 05 Jul '07

05 Jul '07

hello, is it possible to modify the Request URI or the header fields in branch_route? If yes, why the textop functions: subst, subst_uri, replace, search, can not be used in branch_route? ********************************************************************** The information in this message is confidential and may be legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, or distribution of the message, or any action or omission taken by you in reliance on it, is prohibited and may be unlawful. Please immediately contact the sender if you have received this message inerror. **********************************************************************

1 0

[OpenSER-Users] 204 Response from XCAP-Lite for OpenSer, Is this a Bug
by Kirti Dhingra 05 Jul '07

05 Jul '07

Hi , I have a problem setting up the XCAP-Lite server v1.0.2. After setting up Apache2 with all required modules, virtual host etc. I try to use my own Presence Client to write the presrules and resource-list.xml to the database. This failes. I have successfully installed Apache2 with pear (+ extensions), php, mysql support. I created a virtual host with the following settings: <VirtualHost 10.203.144.228:80> <Location "/"> Dav On #necessary for PUT method </Location> ServerName 10.203.144.228 DocumentRoot /var/www/ #/var/www/xcap-root didn't work RewriteEngine On RewriteLog /var/log/apache2/rewrite.log RewriteLogLevel 4 RewriteRule /xcap-root/(.*) /xcap-root/index.php ProxyVia On </VirtualHost> I putted all files from XCAP-Lite archive in /var/www/xcap-root. After using PUT method through my client I receive the 204 No content message. The interesting part is that the database content doesn't change, but putted XML file overwrites index.php instead. Does anybody know why the access to database does not work? Password etc. in config.php is ok. A similar problem was also faced by Sebastian and is available here http://www.openser.org/pipermail/users/2007-April/010537.html, however no response to the query can be seen. Any help would be greatly appreciated. Thanks & Regards, Kirti Dhingra *********************** Aricent-Unclassified *********************** "DISCLAIMER: This message is proprietary to Aricent and is intended solely for the use of the individual to whom it is addressed. It may contain privileged or confidential information and should not be circulated or used for any purpose other than for what it is intended. If you have received this message in error, please notify the originator immediately. If you are not the intended recipient, you are notified that you are strictly prohibited from using, copying, altering, or disclosing the contents of this message. Aricent accepts no responsibility for loss or damage arising from the use of the information transmitted by this email including damage from virus."

2 1

[OpenSER-Users] Re: [Users] Many crashes on Openser system
by Antonio Higuera 05 Jul '07

05 Jul '07

Ok, undestood. You have the log attached from INVITE. Regards, Antonio. 2007/7/5, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro>: > > Hi Antonio, > > yes, it still crashes as that was just a patch for getting more > information at runtime. > > The log is not complete - I need the log from the entire call (starting > with the INVITE, the 200 OK, ACK). Actually the most important is the > log from the process dealing whit the 200 OK as it is the one doing the > accounting. > > Thanks and regards, > Bogdan > > Antonio Higuera wrote: > > Hi Bogdan, > > > > Thanks for the patch. I tested it and the system crashed again. Below > > you have the backtrace and the logs generated attached. > > > >

1 0

[OpenSER-Users] Help!
by Gerardo Gonzalez 05 Jul '07

05 Jul '07

Regards, It's possible to debug errors on openser.cfg file, when I try to start my server? Actually I'm trying to configure Asterisk 1.4.6 and Openser 1.1.0-9etch1, but I got the next error: Not starting openser: invalid configuration file! Thanks!! GJGM

2 1

[Serusers] XLITEPHONE PROBLEM
by Adeyeye Michael 04 Jul '07

04 Jul '07

Kindly help me check this: root@micadeyeye:/usr/src# cd xten-xlite/ root@micadeyeye:/usr/src/xten-xlite# ls README xtensoftphone root@micadeyeye:/usr/src/xten-xlite# ./xtensoftphone I/O warning : failed to load external entity "/root/.Xscrc" Warning: /dev/dsp appears to be a valid audio device, but I cannot open it. Please ensure that no other applications are using the audio device (perhaps by trying ``lsof /dev/dsp''). -- Mike

4 3

[OpenSER-Users] Wrong CSeq in NOTIFY after presence subscription refresh
by Michel de Boer 04 Jul '07

04 Jul '07

I perform the following scenario (openser v1.2.1): 1) Send a SUBCRIBE to openser for the presence event with expires = 3600 2) The subscription is established. Openser sends a NOTIFY with PIDF body. The sequence number in the CSeq header is 516. 3) 1 hour later I send an in-dialog SUBSCRIBE to extend the subscription for another hour. 4) openser sends a NOTIFY having sequence number 516 in the CSeq header. The sequence number of the NOTIFY should have been higher (517) as the sequence numbers of in-dialog messages should increase. I think this is a bug. Regards, Michel -- Michel de Boer www.twinklephone.com

2 2

Re: [OpenSER-Users] The Via Header of CANCEL Message
by fancy 04 Jul '07

04 Jul '07

Hi Klaus: RFC 3665 page 54 line 4: A CANCEL constructed by a client MUST have only a single Via header field value matching the top Via value in the request being cancelled. and RFC 3665 at page 53: CANCEL is referred to as a "hop-by-hop" request, since it is responded to at each stateful proxy hop. and RFC 3665 at page 21: Client: A client is any network element that sends SIP requests and receives SIP responses. Clients may or may not interact directly with a human user. User agent clients and proxies are clients. So, for my opinion, the role of proxy in this case of RFC 3665 section 3.8 is a client when it sends CANCEL message to Bob. If I misunderstand any RFC meanings, please correct me. Thank you very much. Best Regards, Fangyu Ling > fancy wrote: >> According to RFC 3665 section 3.8 and RFC 3261 section 9, >> there is only one via header in CANCEL message >> (message F11, F13 at page 72 of RFC 3665). > > Hi! > > I could find it in RFC 3665, but not in 3261. Where exactly in RFC 3261 > is mentioned that the CANCEL has only 1 Via header? > > Further, the example has only one Via header - but I could not find any > definition if this is a MUST or not. > >> How can I remove the 2nd via header? > > You would have to modify tm module and change the code which generates > the CANCEL message. > > regards > klaus >

3 6

[OpenSER-Users] how to see sip messages in openser
by Rizwan Hisham 04 Jul '07

04 Jul '07

Hi all, i want to see what sip messages are sent to and from openser. just like we can see them on asterisk cli. Is there a way to do that in openser? -- Best Regards Rizwan Hisham Software Engineer AXVOICE Inc. www.axvoice.com

4 3

[OpenSER-Users] DLG_count error
by [ACTIVENDY] NOC 04 Jul '07

04 Jul '07

Hi, I want to limit the simultaneous calls on my Openser server. I use this code to limit the calls: ..... isten=xxx.xxx.xxx.xxx listen=xxx.xxx.xxx.xxx. # put your server IP address here port=5060 children=4 dns=no rev_dns=no fifo="/tmp/openser_fifo" fifo_db_url="mysql://xxxxxxxxxxxxxx loadmodule "/usr/lib/openser/modules/mysql.so" loadmodule "/usr/lib/openser/modules/sl.so" loadmodule "/usr/lib/openser/modules/tm.so" loadmodule "/usr/lib/openser/modules/rr.so" loadmodule "/usr/lib/openser/modules/maxfwd.so" loadmodule "/usr/lib/openser/modules/usrloc.so" loadmodule "/usr/lib/openser/modules/registrar.so" loadmodule "/usr/lib/openser/modules/uri_db.so" loadmodule "/usr/lib/openser/modules/auth.so" loadmodule "/usr/lib/openser/modules/auth_db.so" loadmodule "/usr/lib/openser/modules/uri.so" loadmodule "/usr/lib/openser/modules/domain.so" loadmodule "/usr/lib/openser/modules/textops.so" loadmodule "/usr/lib/openser/modules/avpops.so" loadmodule "/usr/lib/openser/modules/xlog.so" loadmodule "/usr/lib/openser/modules/dispatcher.so" loadmodule "/usr/lib/openser/modules/permissions.so" loadmodule "/usr/lib/openser/modules/acc.so" loadmodule "/usr/lib/openser/modules/dialog.so" modparam("dispatcher", "list_file", "/etc/openser/dispatcher.list") modparam("acc|auth_db|uri_db|usrloc|permissions", "db_url", "mysql://xxxxxxxxxxx") modparam("avpops","db_url","mysql://xxxxxxxxxxxxxxxx) modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("usrloc", "db_mode", 2) modparam("rr", "enable_full_lr", 1) modparam("tm", "wt_timer", 30) modparam("tm", "fr_timer", 2) modparam("tm", "fr_inv_timer", 120) #modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("permissions", "default_allow_file", "/etc/openser/permissions.allow") modparam("permissions", "db_mode", 1) modparam("dispatcher", "flags", 2 ) modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) modparam("dispatcher", "cnt_avp_id", 4) modparam("avpops","avp_table","gw_avail_port") modparam("dialog", "enable_stats", 1) modparam("dialog","dlg_flag",3) ..... if (avp_check("$DLG_count", "gt/i:10")) { sl_send_reply("403","no more calls accepted"); exit; } But I get the following error when I try to call and it doesn't work: 8(4826) xl_get_spec_value: error - null sp->itf 8(4826) ERROR:avpops:ops_check_avp: cannot get src value My Openser version is 1.1.1 Thanks

2 1

[OpenSER-Users] DLG_count error
by [ACTIVENDY] NOC 04 Jul '07

04 Jul '07

Hi, I want to limit the simultaneous calls on my Openser server. I use this code to limit the calls: ..... isten=xxx.xxx.xxx.xxx listen=xxx.xxx.xxx.xxx. # put your server IP address here port=5060 children=4 dns=no rev_dns=no fifo="/tmp/openser_fifo" fifo_db_url="mysql://xxxxxxxxxxxxxx loadmodule "/usr/lib/openser/modules/mysql.so" loadmodule "/usr/lib/openser/modules/sl.so" loadmodule "/usr/lib/openser/modules/tm.so" loadmodule "/usr/lib/openser/modules/rr.so" loadmodule "/usr/lib/openser/modules/maxfwd.so" loadmodule "/usr/lib/openser/modules/usrloc.so" loadmodule "/usr/lib/openser/modules/registrar.so" loadmodule "/usr/lib/openser/modules/uri_db.so" loadmodule "/usr/lib/openser/modules/auth.so" loadmodule "/usr/lib/openser/modules/auth_db.so" loadmodule "/usr/lib/openser/modules/uri.so" loadmodule "/usr/lib/openser/modules/domain.so" loadmodule "/usr/lib/openser/modules/textops.so" loadmodule "/usr/lib/openser/modules/avpops.so" loadmodule "/usr/lib/openser/modules/xlog.so" loadmodule "/usr/lib/openser/modules/dispatcher.so" loadmodule "/usr/lib/openser/modules/permissions.so" loadmodule "/usr/lib/openser/modules/acc.so" loadmodule "/usr/lib/openser/modules/dialog.so" modparam("dispatcher", "list_file", "/etc/openser/dispatcher.list") modparam("acc|auth_db|uri_db|usrloc|permissions", "db_url", "mysql://xxxxxxxxxxx") modparam("avpops","db_url","mysql://xxxxxxxxxxxxxxxx) modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("usrloc", "db_mode", 2) modparam("rr", "enable_full_lr", 1) modparam("tm", "wt_timer", 30) modparam("tm", "fr_timer", 2) modparam("tm", "fr_inv_timer", 120) #modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("permissions", "default_allow_file", "/etc/openser/permissions.allow") modparam("permissions", "db_mode", 1) modparam("dispatcher", "flags", 2 ) modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) modparam("dispatcher", "cnt_avp_id", 4) modparam("avpops","avp_table","gw_avail_port") modparam("dialog", "enable_stats", 1) modparam("dialog","dlg_flag",3) ..... if (avp_check("$DLG_count", "gt/i:10")) { sl_send_reply("403","no more calls accepted"); exit; } But I get the following error when I try to call and it doesn't work: 8(4826) xl_get_spec_value: error - null sp->itf 8(4826) ERROR:avpops:ops_check_avp: cannot get src value My Openser version is 1.1.1 Thanks

1 0

[OpenSER-Users] DLG_count error
by [ACTIVENDY] NOC 04 Jul '07

04 Jul '07

Hi, I want to limit the simultaneous calls on my Openser server. I use this code to limit the calls: ..... isten=xxx.xxx.xxx.xxx listen=xxx.xxx.xxx.xxx. # put your server IP address here port=5060 children=4 dns=no rev_dns=no fifo="/tmp/openser_fifo" fifo_db_url="mysql://xxxxxxxxxxxxxx loadmodule "/usr/lib/openser/modules/mysql.so" loadmodule "/usr/lib/openser/modules/sl.so" loadmodule "/usr/lib/openser/modules/tm.so" loadmodule "/usr/lib/openser/modules/rr.so" loadmodule "/usr/lib/openser/modules/maxfwd.so" loadmodule "/usr/lib/openser/modules/usrloc.so" loadmodule "/usr/lib/openser/modules/registrar.so" loadmodule "/usr/lib/openser/modules/uri_db.so" loadmodule "/usr/lib/openser/modules/auth.so" loadmodule "/usr/lib/openser/modules/auth_db.so" loadmodule "/usr/lib/openser/modules/uri.so" loadmodule "/usr/lib/openser/modules/domain.so" loadmodule "/usr/lib/openser/modules/textops.so" loadmodule "/usr/lib/openser/modules/avpops.so" loadmodule "/usr/lib/openser/modules/xlog.so" loadmodule "/usr/lib/openser/modules/dispatcher.so" loadmodule "/usr/lib/openser/modules/permissions.so" loadmodule "/usr/lib/openser/modules/acc.so" loadmodule "/usr/lib/openser/modules/dialog.so" modparam("dispatcher", "list_file", "/etc/openser/dispatcher.list") modparam("acc|auth_db|uri_db|usrloc|permissions", "db_url", "mysql://xxxxxxxxxxx") modparam("avpops","db_url","mysql://xxxxxxxxxxxxxxxx) modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("usrloc", "db_mode", 2) modparam("rr", "enable_full_lr", 1) modparam("tm", "wt_timer", 30) modparam("tm", "fr_timer", 2) modparam("tm", "fr_inv_timer", 120) #modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("permissions", "default_allow_file", "/etc/openser/permissions.allow") modparam("permissions", "db_mode", 1) modparam("dispatcher", "flags", 2 ) modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) modparam("dispatcher", "cnt_avp_id", 4) modparam("avpops","avp_table","gw_avail_port") modparam("dialog", "enable_stats", 1) modparam("dialog","dlg_flag",3) ..... if (avp_check("$DLG_count", "gt/i:10")) { sl_send_reply("403","no more calls accepted"); exit; } But I get the following error when I try to call and it doesn't work: 8(4826) xl_get_spec_value: error - null sp->itf 8(4826) ERROR:avpops:ops_check_avp: cannot get src value My Openser version is 1.1.1 Thanks

1 0

[OpenSER-Users] RE: SIP Trunking in openser
by Dan-Cristian Bogos 04 Jul '07

04 Jul '07

Hi Rizwan, there can be more ways of implementing your scenario. Here are two of most popular ones: 1. Using rewritehost("IP_OF_YOUR_PROVIDER") and then call t_relay() to send the request out. 2. Using directly t_relay("IP_OF_YOUR_PROVIDER") without rewriting the RURI host. The difference between the two approaches is that in the case of rewritehost() function u will alter the domain of RURI forcing it to be the ip address of your provider. In the second case, u will no longer alter the domain, but instruct t_relay() not to find the route by itself (no dns lookup or any other way) but send the request as it is to the IP which u specify specify. For more info, please consult the documentation of the two functions: http://openser.org/dokuwiki/doku.php/core-cookbook:1.2.x#rewritehost_sethost http://www.openser.org/docs/modules/1.2.x/tm.html#AEN345 Hope that helps, DanB On 7/4/07, Rizwan Hisham <rizwanhasham(a)gmail.com> wrote: > Hi all, > i tried to throw all my INVITE messages to an external IP address by using > the rewritehost function. but calls from my openser dont passthru. i need > to implement a sip trunk and throw my calls thru sip trunk to an external > ip. how should i do this? I am a new user, so whatever your answer is, plz > try to elaborate a little more. thanx > > -- > Best Regards > Rizwan Hisham > Software Engineer > AXVOICE Inc. > www.axvoice.com > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > >

1 0

[OpenSER-Users] RE: [Users] Interaction Presence Agent - Presence User Agent
by Schumann Sebastian 04 Jul '07

04 Jul '07

Dear Anca, Dear all After several trials I didn't come to a conclusion how to use the EXPIRES -1 with PUA_MI. All the time I enter this value to my FIFO file, the answer is pua_mi:mia_pua_publish: invalid expires parameter. This seems also logical to me, as source code of mi_func.c leads there on lines 97-102 (the expires is smaller than 0). Is there a support for PUA of the negative expires field, that is not implemented in PUA_MI? As I thought, PUA cannot be used standalone is there something I did not pay attention to? Any hints? Thanks! Regards Sebastian -----Original Message----- From: Anca-Maria Vamanu [mailto:anca@voice-system.ro] Sent: Friday, May 25, 2007 11:14 AM To: Schumann Sebastian Cc: users(a)openser.org Subject: Re: [Users] Interaction Presence Agent - Presence User Agent Hello, There is already a means to do that but in the trunk version. You should just give the command with expires -1 and the pua module will refresh Publish until you give a command with expires 0. regards, Anca Schumann Sebastian wrote: > Dear all, > > > > I have a question for understanding: I have implemented one OpenSER as > Registrar and one OpenSER as Presence Agent plus XCAP Server 1.0.0. > This is all working so far with newest SVN Release (Big Thanks to > Ovidiu and Anca for the hint) of OpenSER. > > > > Now I want to deploy a Web-Front-End for user interaction with the > Presence Core, because not all clients used are SIMPLE capable. User > of non-capable devices should be able to just login and change their > presence state. They do not necessarily have to be registered at the > registrar, why PUA+ pua_usrloc on registrar side is no choice. > > > > I was thinking about another OpenSER deployed as PUA plus the pua_mi > module for a solution. It gets commands from Web-Frontend via PHP, > forwarded to the FIFO with pua_mi module. The problem I have now is > that the messages are sent just once. Means a user changes his > presence state, PUA sends a PUBLISH to PA but after the expiry the > user appears to be offline. > > > > Is there any possiblitiy to let the PUA send messages constantly, when > the user selects online - means all the time until offline is > selected? I was thinking about many possiblities (cronjob or > shellscript running all the time during online time and sending data > to FIFO) but they are more or less just work-arounds. > > > > Best regards > > Sebastian > > > > **Sebastian Schumann** > > Diploma Student Architecture and Design Department > > > > Slovak Telekom, a. s. > > T-Com, Operations Unit > > Production and Service Division > > Innovations and Enterprise Solution Subunit > > > > Address: Namestie Slobody 6, 817 62 Bratislava, Slovakia > > Office: Room 232, Jarabinkova 1, 821 09 Bratislava, Slovakia > > > > +421 910 643010 (mobile) > > > > sebastian.schumann(a)t-com.sk <mailto:sebastian.schumann@t-com.sk> > > http://www.t-com.sk > >----------------------------------------------------------------------- - > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > >

2 1

[OpenSER-Users] SIP Trunking in openser
by Rizwan Hisham 04 Jul '07

04 Jul '07

Hi all, i tried to throw all my INVITE messages to an external IP address by using the rewritehost function. but calls from my openser dont passthru. i need to implement a sip trunk and throw my calls thru sip trunk to an external ip. how should i do this? I am a new user, so whatever your answer is, plz try to elaborate a little more. thanx -- Best Regards Rizwan Hisham Software Engineer AXVOICE Inc. www.axvoice.com

1 0

[OpenSER-Users] Why so many branches of openser?
by Rizwan Hisham 04 Jul '07

04 Jul '07

Hi all, I am a new user or openser. It is difficult for me to decide which release should i install on my computer as i have no knowledge of why there are so many branches of openser production. for startup, i have installed openser version 0.9.5. but i need to know whats the difference between 0.9.x 1.0.x 1.1.x and 1.2.x? -- Best Regards Rizwan Hisham Software Engineer AXVOICE Inc. www.axvoice.com

2 1

Re: [OpenSER-Users] help with using avp...
by Daniel-Constantin Mierla 04 Jul '07

04 Jul '07

On 07/04/07 09:26, liran tal wrote: > > Thank you very much Daniel. > I was mislead that such variables are not available on 1.1.1 and that > I have to > upgrade to 2.x to use it. It is on 1.1 as well, the link provided is for release 1.1.x. You have to use avpops functions in 1.1 to deal with them. Cheers, Daniel > > Thanks alot, it works just fine. > Liran. > > On 7/2/07, *Daniel-Constantin Mierla* <daniel(a)voice-system.ro > <mailto:daniel@voice-system.ro>> wrote: > > Hello, > > that's a standalone pseudo-variable you can use directlu: $au > http://www.openser.org/dokuwiki/doku.php/pseudovariables:1.1.x#auth_username > <http://www.openser.org/dokuwiki/doku.php/pseudovariables:1.1.x#auth_username> > > You do not need to play with such complex regexp subst. > > Daniel > > On 07/01/07 10:33, liran tal wrote: > > Hey everyone, > > > > I'm using OpenSER 1.1.1 SVN version. > > I'd like to save the Authorization Digest Username as it appears in > > header into a variable (avp) so > > that later I can use it to set an accounting attribute in the > radius. > > > > From the SIP Packet the information is given like this: > > Authorization: Digest username="dsdsds", realm=" 1.2.3.4 > <http://1.2.3.4> > > <http://1.2.3.4>", algorithm=MD5, uri="sip: 1.2.3.4 > <http://1.2.3.4> <http://1.2.3.4>", > > nonce="xxx" > > Where I would like to grab the username part "dsdsds". > > > > From the docs I could only conclude that doing it would be something > > like this: > > avp_write("$hdr[Authorization]","s:12"); > > avp_subst("$avp(s:12)", "/.*username=(.*),/\1/"); > > > > > > > > I'm not even sure about that piece of code, if someone can help... > > > > > > Thanks, > > Lir. > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org <mailto:Users@openser.org> > > http://openser.org/cgi-bin/mailman/listinfo/users > > > >

1 0

[Serusers] Number is Simultaneous calls
by Jai Rangi 04 Jul '07

04 Jul '07

Hello, Is there a module that can control number of simultaneous calls in ser. Fro example we want to allow max of 10 simultaneous calls for user A and 20 simultaneous calls for user B. How can I do that. Thank you, -Jai

2 1

Re: [OpenSER-Users] The Via Header of CANCEL Message
by fancy 04 Jul '07

04 Jul '07

Hi Bogdan: I did not use forward(), but I load sl module. And I did use sl_send_reply() to reply 483 and 513 message and sl_reply_error(). The attached file is my openser config file. I use this config file to run openser. The under RFC3665 section 3.8 condition, I got two Via headers as I mentioned before. So, according to my config file, Openser is a stateless proxy server under RFC3665 section 3.8's condition? Thank you. Best Regards, Fangyu Ling ----- Original Message ----- From: "Bogdan-Andrei Iancu" <bogdan(a)voice-system.ro> To: <fancy(a)cht.com.tw> Cc: <klaus.mailinglists(a)pernau.at>; <users(a)openser.org> Sent: Wednesday, July 04, 2007 2:53 PM Subject: Re: [OpenSER-Users] The Via Header of CANCEL Message > Hi Fangyu , > > My opinion is that the RFC3665 does not cover the stateless part. > > OpenSER can be both a stateless ans stateful proxy, it is just a matter > of configuration (what functions are you using). You can get pure > stateless or stateful cfg, or a mixed one, depending of you. > > If you use the TM function for relaying and replying you will get a pure > stateful proxy. For stateless functions, see the forward() or sl module. > > regards, > bogdan > > fancy wrote: >> Hi Bogdan: >> >> RFC3665 didn't mention any stateful or stateless >> within all the document. >> So I think it is just not covering it. >> >> By the way, is openser a stateless proxy server? >> I think it is a stateful proxy server! >> Can I config it to be a pure stateless or stateful proxy server? >> >> Thank you. >> >> Best Regards, >> Fangyu Ling >> >> >> >>> Klaus, >>> >>> CANCEL is hop-by-hop only in stateful processing and in this case, yes, >>> it has only on VIA. But if you have stateless processing, CANCEL is >>> end-2-end and it will have multiple VIAs... >>> >>> Does 3665 obsoletes stateless SIP or it is just not covering it??? >>> >>> regards, >>> bogdan >>> >>> Klaus Darilion wrote: >>> >>>> Looks like it is defined clearly and openser is not 100% standard conform. >>>> >>>> regards >>>> klaus >>>> >>>> fancy wrote: >>>> >>>> >>>>> Hi Klaus: >>>>> RFC 3665 page 54 line 4: >>>>> >> Sorry it's RFC 3261 page 54 line 4: >> >>>>> A CANCEL constructed by a >>>>> client MUST have only a single Via header field value matching the >>>>> top Via value in the request being cancelled. >>>>> >>>>> and RFC 3665 at page 53: >>>>> >> Sorry it's RFC 3261 at page 53: >> >>>>> CANCEL is referred to as a "hop-by-hop" request, since it is >>>>> responded to at each stateful proxy hop. >>>>> >>>>> and RFC 3665 at page 21: >>>>> >> Sorry it's RFC 3261 at page 21: >> >>>>> Client: A client is any network element that sends SIP requests >>>>> and receives SIP responses. Clients may or may not interact >>>>> directly with a human user. User agent clients and proxies are >>>>> clients. >>>>> >>>>> So, for my opinion, the role of proxy in this case of RFC 3665 section 3.8 is a client when it sends CANCEL message to Bob. >>>>> >>>>> If I misunderstand any RFC meanings, please correct me. >>>>> Thank you very much. >>>>> >>>>> Best Regards, >>>>> Fangyu Ling >>>>> >>>>> >>>>> >>>>> >>>>>> fancy wrote: >>>>>> >>>>>> >>>>>>> According to RFC 3665 section 3.8 and RFC 3261 section 9, >>>>>>> there is only one via header in CANCEL message >>>>>>> (message F11, F13 at page 72 of RFC 3665). >>>>>>> >>>>>>> >>>>>> Hi! >>>>>> >>>>>> I could find it in RFC 3665, but not in 3261. Where exactly in RFC 3261 >>>>>> is mentioned that the CANCEL has only 1 Via header? >>>>>> >>>>>> Further, the example has only one Via header - but I could not find any >>>>>> definition if this is a MUST or not. >>>>>> >>>>>> >>>>>> >>>>>>> How can I remove the 2nd via header?

1 0

Re: [OpenSER-Users] The Via Header of CANCEL Message
by fancy 04 Jul '07

04 Jul '07

Hi Bogdan: RFC3665 didn't mention any stateful or stateless within all the document. So I think it is just not covering it. By the way, is openser a stateless proxy server? I think it is a stateful proxy server! Can I config it to be a pure stateless or stateful proxy server? Thank you. Best Regards, Fangyu Ling > Klaus, > > CANCEL is hop-by-hop only in stateful processing and in this case, yes, > it has only on VIA. But if you have stateless processing, CANCEL is > end-2-end and it will have multiple VIAs... > > Does 3665 obsoletes stateless SIP or it is just not covering it??? > > regards, > bogdan > > Klaus Darilion wrote: >> Looks like it is defined clearly and openser is not 100% standard conform. >> >> regards >> klaus >> >> fancy wrote: >> >>> Hi Klaus: >>> RFC 3665 page 54 line 4: Sorry it's RFC 3261 page 54 line 4: >>> A CANCEL constructed by a >>> client MUST have only a single Via header field value matching the >>> top Via value in the request being cancelled. >>> >>> and RFC 3665 at page 53: Sorry it's RFC 3261 at page 53: >>> CANCEL is referred to as a "hop-by-hop" request, since it is >>> responded to at each stateful proxy hop. >>> >>> and RFC 3665 at page 21: Sorry it's RFC 3261 at page 21: >>> Client: A client is any network element that sends SIP requests >>> and receives SIP responses. Clients may or may not interact >>> directly with a human user. User agent clients and proxies are >>> clients. >>> >>> So, for my opinion, the role of proxy in this case of RFC 3665 section 3.8 is a client when it sends CANCEL message to Bob. >>> >>> If I misunderstand any RFC meanings, please correct me. >>> Thank you very much. >>> >>> Best Regards, >>> Fangyu Ling >>> >>> >>> >>>> fancy wrote: >>>> >>>>> According to RFC 3665 section 3.8 and RFC 3261 section 9, >>>>> there is only one via header in CANCEL message >>>>> (message F11, F13 at page 72 of RFC 3665). >>>>> >>>> Hi! >>>> >>>> I could find it in RFC 3665, but not in 3261. Where exactly in RFC 3261 >>>> is mentioned that the CANCEL has only 1 Via header? >>>> >>>> Further, the example has only one Via header - but I could not find any >>>> definition if this is a MUST or not. >>>> >>>> >>>>> How can I remove the 2nd via header? >>>>> >>>> You would have to modify tm module and change the code which generates >>>> the CANCEL message. >>>> >>>> regards >>>> klaus >>>>

2 1

Re: [Serusers] How SER communicate to other SIP server
by roayu 04 Jul '07

04 Jul '07

Hi, Olaf! >From what I understand about forward() is that it will forward the request to the given destination statelessly. Whereas for t_relay(), it will relay a message to the destination stated in current URI statefully. How do I get the URI to point to the other SIP server by using t_relay() as I couldn't assign any IP address for t_relay(). I managed to assign an IP address by using forward(). I tried to use both. Unfortunately, I still failed to forward the message to the another SIP server to establish the call. Is there any functions I need to use before using either those 2 functions? FYI, I will only call them when it starts to INVITE. Thanks! Best regards, :) Roa Yu -----Original Message----- From: Olaf Bergmann [mailto:Olaf.Bergmann@freenet-ag.de] Sent: Tuesday, July 03, 2007 8:14 PM To: roayu Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] How SER communicate to other SIP server roayu wrote: > In that case, since it's not recommended to change the TO header, is there > any other ways I can make the SER to route to another SIP server without > using rewritehostport()? Yes, several. First of all, rewritehostport() does no forwarding, it just changes the domain part of the request URI (not To). Use forward() or t_relay() depending on your particular needs. Regards, Olaf

2 1

[OpenSER-Users] openser 1.2.1 core dumps on 302 redirect
by Brandon Price 03 Jul '07

03 Jul '07

With debug=99 I get ... 0(31783) DEBUG:t_check_status: checked status is <302> 0(31783) +15183024234 has a forward 0(31783) DEBUG:uac_redirect:get_redirect: resume branch=0 0(31783) DEBUG:uac_redirect:get_redirect: checking branch=0 (added=0) 0(31783) DEBUG:uac_redirect:get_redirect: branch=0 is a redirect (added=0) 0(31783) parse_headers: flags=7 0(31783) Found param type 232, <branch> = <z9hG4bK68c33491>; state=16 0(31783) end of header reached, state=5 0(31783) parse_headers: Via found, flags=7 0(31783) parse_headers: this is the second via 0(31783) DEBUG: get_hdr_body : content_length=0 0(31783) found end of header 0(31783) DEBUG:uac_redirect:sort_contacts: <sip: 13143212222(a)eastern.testdomain.net> q=10 0(31783) DEBUG:uac_redirect:shmcontact2dset: adding contact <sip: 13143212222(a)eastern.testdomain.net> 0(31783) ******* setting for branch 0 flags 0 0(31783) parse_headers: flags=78 Segmentation fault (core dumped) // Where is this core dump being written? related openser.cfg snippet: if(t_check_status("302")){ xlog("$tU has a forward"); get_redirects("*", "redirected"); xlog("forwarding to $tU"); t_relay("5.6.7.8"); exit; }; Any help is as always, greatly appreciated.

2 1

[OpenSER-Users] OpenSER on Xorcom TS-1
by TCB 03 Jul '07

03 Jul '07

Anyone running openser on Xorcom TS-1 with asterisk as b2bua ? -- TC

1 0

[OpenSER-Users] opeser-SVN + Freeradius 1.1.6 problem
by ram 03 Jul '07

03 Jul '07

Hi iam following the below document http://www.openser.org/docs/openser-radius-1.0.x.html iam able to compile both openser and freeradius and radius-ng after configuration files edited. 7. Testing RADIUS server radiusd -X Starting - reading configuration files ... reread_config: reading radiusd.conf Config: including file: /usr/local/etc/raddb/proxy.conf Config: including file: /usr/local/etc/raddb/clients.conf Config: including file: /usr/local/etc/raddb/snmp.conf Config: including file: /usr/local/etc/raddb/sql.conf main: prefix = "/usr/local" main: localstatedir = "/usr/local/var" main: logdir = "/usr/local/var/log/radius" main: libdir = "/usr/local/lib" main: radacctdir = "/usr/local/var/log/radius/radacct" main: hostname_lookups = no main: max_request_time = 30 main: cleanup_delay = 5 main: max_requests = 1024 main: delete_blocked_requests = 0 main: port = 0 main: allow_core_dumps = no main: log_stripped_names = no main: log_file = "/usr/local/var/log/radius/radius.log" main: log_auth = no main: log_auth_badpass = no main: log_auth_goodpass = no main: pidfile = "/usr/local/var/run/radiusd/radiusd.pid" main: user = "(null)" main: group = "(null)" main: usercollide = no main: lower_user = "no" main: lower_pass = "no" main: nospace_user = "no" main: nospace_pass = "no" main: checkrad = "/usr/local/sbin/checkrad" main: proxy_requests = yes proxy: retry_delay = 5 proxy: retry_count = 3 proxy: synchronous = no proxy: default_fallback = yes proxy: dead_time = 120 proxy: post_proxy_authorize = no proxy: wake_all_if_all_dead = no security: max_attributes = 200 security: reject_delay = 1 security: status_server = no main: debug_level = 0 read_config_files: reading dictionary Errors reading dictionary: dict_init: No ATTRIBUTE "Service-Type" defined for VALUE "SIP-Callee-AVPs" Errors reading radiusd.conf any suggestions ram

2 2

[Serusers] User update on SER
by Felipe Louback 03 Jul '07

03 Jul '07

Hey, How does SER updates a user location? For exemple, upon the first REGISTER message, it keeps the location of the user in a database or so. But this entry has a expire time. How is this expire time updated? SER updates it upon receiving another SIP REGISTER or OPTIONS messages? I know that this update is done, but I am quite confusing about how it is done. Thanks Felipe -- Master Student - Electrical Engineering Department Computer Engineering and Telecommunications Research Group Universidade Federal de Minas Gerais - Brazil "Come to me, all you who are weary and burdened, and I will give you rest." Matthews 11:28 "Failure is the opportunity to begin again more intelligently." Henry Ford

2 3

[OpenSER-Users] openser authentication not work
by Amr Elgbaly 03 Jul '07

03 Jul '07

hello !! i installed openser with mysql module but any one can register and make calls to another users openser.cfg content cat openser.cfg # # $Id: openser.cfg 1827 2007-03-12 15:22:53Z bogdan_iancu $ # # simple quick-start config script # Please refer to the Core CookBook at http://www.openser.org/dokuwiki/doku.php <http://www.voipuser.org/ship_to.php?url=http://www.openser.org/dokuwiki/dok…> # for a explanation of possible statements, functions and parameters. # # ----------- global configuration parameters ------------------------ debug=8 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E) children=4 log_facility=LOG_LOCAL0 sip_warning=yes check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) # Uncomment these lines to enter debugging mode #fork=no #log_stderror=yes # port=5060 # uncomment the following lines for TLS support #disable_tls = 0 #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/openser/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/openser/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/openser/tls/user/user-calist.pem" # ------------------ module loading ---------------------------------- #set module path mpath="/usr/local/lib/openser/modules/" # Uncomment this if you want to use SQL database loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "mi_fifo.so" loadmodule "/usr/local/lib/openser/modules/xlog.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "auth.so" loadmodule "auth_db.so" # ----------------- setting module-specific parameters --------------- # -- mi_fifo params -- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "mysql://openser:openserrw@localhost/openser") #fifo_db_url="mysql://openser:openserrw@localhost/openser" modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("usrloc", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS #if(uri=~"@tls_domain1.net") { # t_relay("tls:domain1.net"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay("tls:domain2.net"); # exit; #} route(1); }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication #if (!www_authorize("openser.org", "subscriber")) { # www_challenge("openser.org", "0"); # exit; #}; save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } Regards , -- Amr Elgbaly Gnu/Linux sytem engineer OptiServers support team http://optiservers.com Telephone : +2 02 678 679 Cell Phone : +2 012 803 2002

2 1

[OpenSER-Users] can't find cpl-c
by Richard Morrisey 03 Jul '07

03 Jul '07

I'm having trouble finding the cpl-c module. I have the binaries for Openser 1.1.1 installed on arch x86_64 with FC6 downloaded from the opensuse repository. Where can I get a compiled version of this module? Rich

2 1

[OpenSER-Users] presence module notify missing max-forwards
by Sigrid Thijs 03 Jul '07

03 Jul '07

Hi, I noticed that the NOTIFY requests that are sent by the presence module do not contain a "Max-Forwards" header. This is a mandatory header! We are using openser version 1.2.0 on a freebsd platform. kind regards, Sigrid

2 1

[OpenSER-Users] error while running openser
by Vishwadeepak Agarwal 03 Jul '07

03 Jul '07

while running openser i am getting this.can somebody help? 0(0) StateLess module - initializing 0(0) TM - initializing... 0(0) Maxfwd module- initializing 0(0) INFO:ul_init_locks: locks array size 512 0(0) TextOPS - initializing 0(0) ERROR:mi_fifo:mod_init:no fifo configured 0(0) init_mod(): Error while initializing module mi_fifo ERROR: error while initializing modules 0(0) INFO:mi_fifo:mi_destroy:memory for the child's mi_fifo_pid was not allocated -> nothing to destroy Regards, Vishwa Deepak Agarwal ______________________________________________________________________

2 1

[OpenSER-Users] Milkfish 3.2.0 released
by Daniel-Constantin Mierla 03 Jul '07

03 Jul '07

Hello, just saw that the other day Milkfish 3.2.0 was released. It includes OpenSER, so, for those playing with embedded devices, that might be interesting. http://www.milkfish.org/ http://wiki.milkfish.org/index.php?n=Boozy.Features#boozy320features Daniel

1 0

Re: [Serusers] How SER communicate to other SIP server
by roayu 03 Jul '07

03 Jul '07

Hi, Olaf! In that case, since it's not recommended to change the TO header, is there any other ways I can make the SER to route to another SIP server without using rewritehostport()? Thanks. Best regards, Roa Yu -----Original Message----- From: Olaf Bergmann [mailto:Olaf.Bergmann@freenet-ag.de] Sent: Tuesday, July 03, 2007 4:36 PM To: roayu Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] How SER communicate to other SIP server roayu wrote: > that's the reason why I need to change the URI (of the TO header). No, it is not. At least if your user agent and other servers implement the same RFC 3261 as most of the other SIP servers in the world do. > just failed to change the uri at the TO header. Therefore, it couldn't route > to another SIP server. > > Current FROM/TO uri that I get, > From : <sip:12345@mySER.net:5060> > To : <sip:88888888@mySER.net:5060> > > The actual FROM/TO uri that I WANT will be, > From : <sip:12345@mySER.net:5060> > To : <sip:88888888@anotherSIP.net:5060> > > I've checked on the regular expression, and it's fine with that. Am I using > the correct function - rewritehostport()? Well, rewritehostport() changes the request URI, not the URI part of the To header field. This is fine for most applications, but if you really want to change the contents of the To header, you will have to use something like replace_hf(). But it is not recommended to do that. Really. SIP is designed to work without replacing the To. In fact, applications are more likely to break if it is changed. Regards, Olaf

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Re: [Serusers] How SER communicate to other SIP server
by roayu 03 Jul '07

03 Jul '07

Hi, Olaf! Pls ignore the previous mail and refer to this as there's a typo error. Oops, I accidentally deleted the semi-colon(;) at the closing of the IF statement. I know that the SIP messages are routed according to the URI and that's the reason why I need to change the URI (of the TO header). But I just failed to change the uri at the TO header. Therefore, it couldn't route to another SIP server. Current FROM/TO uri that I get, From : <sip:12345@mySER.net:5060> To : <sip:88888888@mySER.net:5060> The actual FROM/TO uri that I WANT will be, From : <sip:12345@mySER.net:5060> To : <sip:88888888@anotherSIP.net:5060> I've checked on the regular expression, and it's fine with that. Am I using the correct function - rewritehostport()? Thanks. Best regards, Roa Yu -----Original Message----- From: Olaf Bergmann [mailto:Olaf.Bergmann@freenet-ag.de] Sent: Tuesday, July 03, 2007 3:07 PM To: roayu Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] How SER communicate to other SIP server roayu wrote: > Hi, Olaf! > > I've used 'rewritehostport("sipB.net:5060")' under the INVITE message > handler. > > The following are my configuration, pls correct me if I'm wrong. > if(uri =~ "^sip:[0-9{8}@.*") { > if(!proxy_authorise("","subscriber")){ > proxy_challenge("","0"); > break; > > }else if(!check_from()) { > sl_send_reply("403","Use From ID"); > break; > }; > > consume_credentials(); > rewritehostport("sipB.net:5060"); > setflag(1); > route(1); > break; > } > > When I dialed 8-digit of the number, it will pass to another SIP > server(sipB.net). But on the TO header, I still found that the TO address > still under the sipSER.net. You may refer to the following. > From : <sip:12345@sipSER.net:5060> > To : <sip:88888888@sipSER.net:5060> > Does that means that it's still looping inside sipSER.net instead of routing > to sipB.net? If yes, how do I change the TO header or make it route to > sipB.net? No, SIP messages are routed according to their request URI. I am not sure if the regular expression in your if-clause was corrupted by the E-Mail service (if not, at least a closing bracket is missing). I suggest, you start with a simplified version of this regexp to be sure that the then-part is evaluated at all. Use xlog() to add a couple of debug messages to find out which routing blocks are evaluated. Regards, Olaf _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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Re: [Serusers] How SER communicate to other SIP server
by roayu 03 Jul '07

03 Jul '07

Hi, Olaf! Oops, I accidentally deleted the semi-colon(;) at the closing of the IF statement. I know that the SIP messages are routed according to the URI and that's the reason why I need to change the URI (of the TO header). But I just failed to change the uri at the TO header. Therefore, it couldn't route to another SIP server. Current FROM/TO uri that I get, From : <sip:12345@mySER.net:5060> To : <sip:88888888@anotherSIP.net:5060> The actual FROM/TO uri that I WANT will be, From : <sip:12345@mySER.net:5060> To : <sip:88888888@anotherSIP.net:5060> I've checked on the regular expression, and it's fine with that. Am I using the correct function - rewritehostport()? Thanks. Best regards, Roa Yu -----Original Message----- From: Olaf Bergmann [mailto:Olaf.Bergmann@freenet-ag.de] Sent: Tuesday, July 03, 2007 3:07 PM To: roayu Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] How SER communicate to other SIP server roayu wrote: > Hi, Olaf! > > I've used 'rewritehostport("sipB.net:5060")' under the INVITE message > handler. > > The following are my configuration, pls correct me if I'm wrong. > if(uri =~ "^sip:[0-9{8}@.*") { > if(!proxy_authorise("","subscriber")){ > proxy_challenge("","0"); > break; > > }else if(!check_from()) { > sl_send_reply("403","Use From ID"); > break; > }; > > consume_credentials(); > rewritehostport("sipB.net:5060"); > setflag(1); > route(1); > break; > } > > When I dialed 8-digit of the number, it will pass to another SIP > server(sipB.net). But on the TO header, I still found that the TO address > still under the sipSER.net. You may refer to the following. > From : <sip:12345@sipSER.net:5060> > To : <sip:88888888@sipSER.net:5060> > Does that means that it's still looping inside sipSER.net instead of routing > to sipB.net? If yes, how do I change the TO header or make it route to > sipB.net? No, SIP messages are routed according to their request URI. I am not sure if the regular expression in your if-clause was corrupted by the E-Mail service (if not, at least a closing bracket is missing). I suggest, you start with a simplified version of this regexp to be sure that the then-part is evaluated at all. Use xlog() to add a couple of debug messages to find out which routing blocks are evaluated. Regards, Olaf

1 0

[OpenSER-Users] Permission module
by Alexander Nazarenko 03 Jul '07

03 Jul '07

Dear All, I can see that permission module initializes DB only once with Openser startup. All the changes, made in DB, will be postponed till the next Openser restart. Can anybody advise how to make real-time permission DB fetching?

2 1

Re: [Serusers] How SER communicate to other SIP server
by roayu 03 Jul '07

03 Jul '07

Hi, Olaf! I've used 'rewritehostport("sipB.net:5060")' under the INVITE message handler. The following are my configuration, pls correct me if I'm wrong. if(uri =~ "^sip:[0-9{8}@.*") { if(!proxy_authorise("","subscriber")){ proxy_challenge("","0"); break; }else if(!check_from()) { sl_send_reply("403","Use From ID"); break; }; consume_credentials(); rewritehostport("sipB.net:5060"); setflag(1); route(1); break; } When I dialed 8-digit of the number, it will pass to another SIP server(sipB.net). But on the TO header, I still found that the TO address still under the sipSER.net. You may refer to the following. From : <sip:12345@sipSER.net:5060> To : <sip:88888888@sipSER.net:5060> Does that means that it's still looping inside sipSER.net instead of routing to sipB.net? If yes, how do I change the TO header or make it route to sipB.net? Thanks. Best regards, Roa Yu -----Original Message----- From: Olaf Bergmann [mailto:Olaf.Bergmann@freenet-ag.de] Sent: Monday, July 02, 2007 4:55 PM To: roayu Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] How SER communicate to other SIP server roayu wrote: > Hi, Olaf! > > I'm not sure what functions that I can use to include inside ser.cfg. Could > you pls let me know where I can see all those functions like > sl_send_reply(), so that I can use them. Have a look at <http://www.iptel.org/doc>. You will find several tutorials as well as reference documents, especially for the available modules. In addition, there is an excellent interactive search tool at <http://www.iptel.org/ser/doc/search> to lookup function definitions etc. Regards, Olaf

2 1

[Serusers] PA module SER 2_0_0 - no xcap query with auth parameter set to xcap
by tzieleniewski 03 Jul '07

03 Jul '07

Hi, I am trying to use the xcap authorization for PA during the SUBSCRITPION message servicing. I have the following code which work fine when auth parameter is set to 'none': if(lookup_user("$tu.uid", "@ruri")) { if (!t_newtran()) sl_reply_error(); drop; } (handle_subscription("registrar")) ... } but when I switch to auth 'xcap' I cant see any query send to XCAP server simulated on apache according to the presence handbook. Please point me what do I miss. Thanks Cheers Tomasz

2 7

[Serusers] xl_mod_init warning
by Todd Lewis 02 Jul '07

02 Jul '07

I using the example configuration file that comes with ser-2.0-rc1 No matter what I do, I cannot figure out what's causing this warning /usr/local/ser/sbin/ser[43254]: Maxfwd module- initializing /usr/local/ser/sbin/ser[43254]: WARNING: xl_mod_init: more IP 10.0.0.6 not used I have two ip's assigned to the server, using one Nic card Static IP 10.0.0.6 Virtual IP 10.0.0.4 NicCard em0 I have tried -------------------- listen=10.0.0.4 fork=yes/no --------------------- -------------------- listen-10.0.0.6 fork=yes/no -------------------- -------------------- listen=10.0.0.4 listen=10.0.0.6 fork=yes/no -------------------- -------------------- listen=em0 listen=10.0.0.4 fork=yes/no -------------------- -------------------- listen=em0 fork=yes/no -------------------- I always get the same error message. Any ideas?

1 0

Re: [OpenSER-Users] FreeRADIUS-CDRTool Rel. 1.0
by Dan-Cristian Bogos 02 Jul '07

02 Jul '07

Hi, my code has no connection to any BBUA. It is just providing the maximum call duration information back to OpenSER which can later do what u want it to do with this info. I personally use yate (yate.null.ro) to provide call cut when the calls are going out on PSTN, by adding an extra SIP Header in OpenSER and then later process it in Yate to timeout the calls. I guess there are unlimited possibilities later, if u have the information available in OpenSER AVPs, including creating your own timer module which sends a session cancelation order to RTPProxy (heard that it is possible to kill the calls with RTPProxy, but never done it). Let me know if I was clear enough. Cheers, DanB On 7/2/07, A.M. <agentm(a)themactionfaction.com> wrote: > > On Jul 2, 2007, at 11:10 , Dan-Cristian Bogos wrote: > > > All, > > > > as a personal gratitude for all the efforts involved in opensource > > projects, especially in OpenSER community, I would like to contribute > > myself with a bit of code, investing in something I considered useful > > for this community. > > > > Hereby, I inform u that the first release of FreeRADIUS-CDRTool > > connector module is available for download and use at sourceforge.net. > > Project link: http://sourceforge.net/projects/frad-cdrtool/ > > > > Basically, FreeRADIUS-CDRTool is a FreeRADIUS module written in python > > (therefore one might need to compile rlm_python in FreeRADIUS to make > > it work), able to connect on CDRTool Engine via the Telnet socket (the > > only way I found recommended by CDRTool folks) and perform different > > actions via Prepaid API, like return maximum value for the call, > > instruct CDRTool to lock accounts already in a call, instruct > > unlocking at end of the call and balance debiting, etc. This module > > should be used together with OpenSER radius authentication and > > accounting. > > If proper configured, FreeRADIUS-CDRTool can authorize calls for > > CDRTool Prepaid users. At the starting of each call the module will > > return the maximum call duration allowed, information which can be > > processed later in a bbua scenario to timeout the calls reaching 0 > > balance, and total credit available at starting of the call, all > > through SIP-AVP Radius Attributes. > > Hi- thanks for the information. How does your code interact with the > b2bua? Which b2buas are supported? > > Cheers, > M >

1 0

[OpenSER-Users] FreeRADIUS-CDRTool Rel. 1.0
by Dan-Cristian Bogos 02 Jul '07

02 Jul '07

All, as a personal gratitude for all the efforts involved in opensource projects, especially in OpenSER community, I would like to contribute myself with a bit of code, investing in something I considered useful for this community. Hereby, I inform u that the first release of FreeRADIUS-CDRTool connector module is available for download and use at sourceforge.net. Project link: http://sourceforge.net/projects/frad-cdrtool/ Basically, FreeRADIUS-CDRTool is a FreeRADIUS module written in python (therefore one might need to compile rlm_python in FreeRADIUS to make it work), able to connect on CDRTool Engine via the Telnet socket (the only way I found recommended by CDRTool folks) and perform different actions via Prepaid API, like return maximum value for the call, instruct CDRTool to lock accounts already in a call, instruct unlocking at end of the call and balance debiting, etc. This module should be used together with OpenSER radius authentication and accounting. If proper configured, FreeRADIUS-CDRTool can authorize calls for CDRTool Prepaid users. At the starting of each call the module will return the maximum call duration allowed, information which can be processed later in a bbua scenario to timeout the calls reaching 0 balance, and total credit available at starting of the call, all through SIP-AVP Radius Attributes. I have tested the module in a Debian Linux environment, python2.4 and FreeRADIUS 1.6, CDRTool version 5.1.4, on low call load. I would not recommend yet using it in a production environment till we get some stability feedback. Have fun! Kind regards, DanB

1 0

[OpenSER-Users] mediaproxy StopRecordSerializer when closing channel
by Dzejms 02 Jul '07

02 Jul '07

Hi, I have this common problem Jul 2 16:03:25 localhost mediaproxy[5566]: error: uncaptured python exception, closing channel <rtphandler.CommandHandler connected xxx.xxx.xxx.xxx:46948 at -0x5f92e794> (exceptions.NameError:global name 'StopRecordSerializer' is not defined [asyncore.py|readwrite|94] [asyncore.py|handle_read_event|391] [/usr/local/mediaproxy/modules/rtphandler.py|handle_read|441] [/usr/local/mediaproxy/modules/rtphandler.py|process|569]) Mediaproxy version is 1.8.2 it's running on debian 4.0 etch, witch python 2.4 (also tried 2.5), i have python-mysqldb module installed, also python-dev-all . Problem still exist. When i'm using 1.7.2 version od mediaproxy, it seems to be ok. I'm also using 1.8.2 mediaproxy version on debian sarge with python 2.3 where everything seems to be ok. Any ideas? Broken debian packages? Dzejms.

1 0

[OpenSER-Users] manipulate INVITES
by Patrick Miccio 02 Jul '07

02 Jul '07

Hello everybody, since this is my first post and I am kinda new to OpenSER, bare with me ;) I have been trying to manipulate invites for the last 2 days and cant find the proper way to reach what is expected. my goal: send invites to specific PBX (Asterisk for example) with only the desired extension and not the full username, this is based on a avp_check if that UAC needs only extensions, some do, some don't ;) Here is what I do: First off all I extract the real username (in subscribers) and extension from the R-URI and place it in "$avp(callee_user)" and "$avp(callee_exten)" ############# my solution No1 is "working" avp_pushto("$ru/username", "$avp(callee_user)"); avp_copy("$avp(callee_exten)","$avp(callee_tmp)/gd"); if( subst('/^To:(.*)sip:[^@]*(@[a-zA-Z0-9.]+.*)$/To:\1sip:$avp(callee_tmp)\2/ig') ){}; -> callee_user is 012345678 -> callee_exten is 200 -> $ru is sip:012345678@xxx.xxx.xxx.xxx;user=phone -> subst [ To: <sip:012345678200@xxx.xxx.xxx.xxx;user=phone> ] with [To: <sip:200@xxx.xxx.xxx.xxx;user=phone> ] Those 3 lines work "fine", the testing Asterisk PBX gets the call correctly. The problem is that I should not change the "To:" part, it will make problems with call forwards etc... ############# my solution No.2 NOT working. if (subst_uri('/^sip:([0-9]+)@(.*)$/sip:$avp(callee_tmp)@\2;/i')){$}; doesn't work because OpenSER of course cant find 200 in the userloc.... ############# Generally I would think that changing the INVITE uri itself like in solution No.2 is the best way, couldn't find any proper way to make that work though. Does anyone have an idea how to realize this properly, or even if this makes sense ;) ? thx in advance, Patrick.

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[OpenSER-Users] help with using avp...
by liran tal 02 Jul '07

02 Jul '07

Hey everyone, I'm using OpenSER 1.1.1 SVN version. I'd like to save the Authorization Digest Username as it appears in header into a variable (avp) so that later I can use it to set an accounting attribute in the radius. >From the SIP Packet the information is given like this: Authorization: Digest username="dsdsds", realm="1.2.3.4", algorithm=MD5, uri="sip:1.2.3.4", nonce="xxx" Where I would like to grab the username part "dsdsds". >From the docs I could only conclude that doing it would be something like this: avp_write("$hdr[Authorization]","s:12"); avp_subst("$avp(s:12)", "/.*username=(.*),/\1/"); I'm not even sure about that piece of code, if someone can help... Thanks, Lir.

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[Serusers] RLS module in ser 2_0_0 - subscriptions handling
by tzieleniewski 02 Jul '07

02 Jul '07

Hi, I am using the RLS module for RLS service and resource list subscriptions. Ive done everything according to the presence handbook. When SER sends NOTIFY messages to the subscribing UAs every instances of the subscribed resources is in the pending state. How are those subscriptions handled by ser?? Do I have to include presence_b2b molude so that those subscriptions in the pending state would be handled?? Thanks in advance Cheers Tomasz

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[Serusers] routing rtpstream through nat within vpn
by Martin Drasar 02 Jul '07

02 Jul '07

Hello everyone, I have following scenario: +----------+ | VPN | |+--------+| || MRCP || |+--------+| | || | |+--------+| +---------+ || Proxy ====| Solaris | |+--------+| +---------+ +----------+ I want both sip messages and rtp stream to go through proxy between mrcp servers and solaris box. Currently I am transferring sip messages using call forwarding (avpops module) and trying to use rtpproxy for rtp stream forwarding. For rtpproxy to work I have set up NAT on proxy, but still have issues. This is the invite packet sent from proxy to MRCP server: INVITE sip:UAS@10.1.1.100:5060 SIP/2.0 Record-Route: <sip:10.0.0.1;ftag=3d423348;lr=on> <=## Via: SIP/2.0/UDP 10.0.0.1;branch=z9hG4bKe359.5e010b95.0 <=## Via: SIP/2.0/UDP 10.0.0.6:5070;branch=z9hG4bK-d8754z-5212be016a2d3c2a-1---d8754z-;rport=5070 Max-Forwards: 16 Contact: <sip:sender@10.0.0.6:5070> To: <sip:synthesizer@10.0.0.1:5060;transport=UDP> <=## From: <sip:sender@10.0.0.6:5070>;tag=3d423348 Call-ID: MTZkYWE2MGYxNTgyNmY0YTRmNzkwNjAyNmU5OGQ0ZGU. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY Content-Type: application/sdp Content-Length: 282 v=0 o=- 1183369050 1183369050 IN IP4 10.0.0.6 s=- t=0 0 m=application 35050 TCP/MRCPv2 <=## c=IN IP4 10.0.0.1 <=## a=connection:new a=setup:active a=cmid:1 a=resource:speechsynth m=audio 53384 RTP/AVP 0 c=IN IP4 10.0.0.1 <=## a=mid:1 a=recvonly a=direction:active a=nortpproxy:yes Lines with "<=##" are the lines with which I think I have problem. Because mrcp servers receive 'To' and 'Via' with address that is outside vpn, they cannot reply. As a result I send x times invite and end with 408 timeout. Also there is problem with line "m=application 35050 TCP/MRCPv2". rtpproxy changes that port, but according to mrcp specification I need it to be exactly 9. Even when changing PORT_MIN and PORT_MAX both to 9, I sometimes get port 10. Could you please advice me what to do? Am I trying in right direction? Will I have to alter the sources to make it work like I need to? Thanks for your answers Martin

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Re: [Serusers] How SER communicate to other SIP server
by roayu 02 Jul '07

02 Jul '07

Hi, Olaf! I'm not sure what functions that I can use to include inside ser.cfg. Could you pls let me know where I can see all those functions like sl_send_reply(), so that I can use them. Thanks. Best regards, Roa Yu -----Original Message----- From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of roayu Sent: Friday, June 29, 2007 8:47 AM To: Olaf Bergmann Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] How SER communicate to other SIP server Hi, Olaf! I'll have a look and try on it. Thanks. Best regards, Roa Yu -----Original Message----- From: Olaf Bergmann [mailto:Olaf.Bergmann@freenet-ag.de] Sent: Thursday, June 28, 2007 5:34 PM To: roayu Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] How SER communicate to other SIP server roayu wrote: > Hi, Olaf! > > Ok, but which section that I need to concentrate on the most? I've search > through the whole documentation and I can't find the configuration/setting > on the ser.cfg file. Really appreciate on your help... Look at the "Hello World ser.cfg" <http://siprouter.onsip.org/doc/gettingstarted/ch06.html>. For unknown SIP domains, route[1] does the DNS-based forwarding mentioned before. You should give it a try and see how it works. HTH, Olaf _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[OpenSER-Users] not able to run openser
by Vishwadeepak Agarwal 02 Jul '07

02 Jul '07

hi all, I have installed openser. when i gave ./openser.init start i got this Command Monit to start OpenSER... ./openser.init: line 64: /usr/local/bin/monit: No such file or directory when i tried to run ./openser i got this ERROR: bad config file (16 errors) can anyone help? thanks in advance regards, Vishwadeepak ______________________________________________________________________

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[Serusers] Efficient config and routing of 100's of prefix's
by Shaun Hofer 29 Jun '07

29 Jun '07

Hi, To terminate a call to the PSTN, my SER box has to route between multiple PSTN gateways depending on the destination phone number/prefix. For instance, -Use gateway1 if the destination begins with any of these prefixes: 07550[0-4] 075509 07551[0-2] 075514 075515[7-8] 0755187 0755337 07553[7-9] 075551[0-1] 07555[2-8] 07556[0-4] etc.. -Use gateway2 if the destination begins with any of these prefixes: 0730[0-2] 07303[0-6] 0731[0-2] 073131 07313[5-9] 073140 07314[4-6] 0731[6-7] 0732[1-7] 07329[1-2] 07329[5-6] etc.. -Use gateway3 if etc... -Use gatewayX if etc... Obviously, one or a small number of regular expressions can't efficiently cover all of these prefixes. However, doing it like this isn't ideal because ser.cfg would be huge and cumbersome. Also, the prefixes for each gateway may change over time: if ( (uri =~ "^sip:07550[0-4]") || (uri =~ "^sip:075509") || ... ) { route to gateway1 } if ( (uri =~ "^sip:0730[0-2]") || (uri =~ "^sip:07303[0-6]") || ... ) { route to gateway2 } if ( ... ) { route to gatewayX } What's the most efficient way of doing this with SER? Thanks in advanced -Shaun

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[Serusers] Difficult NAT issues :external UAs behind NAT and internal UAs on private IPs but _not_ NATted
by Michael Grigoni 29 Jun '07

29 Jun '07

Greetings: I really need some fresh eyes on my ser.cfg and help with this difficult NAT issue: our internal UAs are on private IPs, registered with ser running on the border router (it has aliases for the private IP internal LANs and the public IP on which it listens). Calls between UAs on public IPs and our internal UAs (on their private IPs) work well. Calls between UAs behind NAT (from the Internet) and our internal UAs (on their private IPs) have no audio from us to them. Only private IP addresses in routes from the Internet must be rewritten for NAT; our internal UAs on their private IPs must be routed without change (which works if the foreign UA isn't behind NAT). We are running '0.8.99-dev19 openbsd/sparc' and 'rtpproxy 1.21 2004/08/13' Our 'ser.cfg', a packet dump of a call from an Internet UA behind NAT ('adams.dmp') and a segment of the logfile ('serlog') are at: http://www.cybertheque.org/sw/ser/configs All help is truly appreciated. Michael Grigoni Cybertheque Museum

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[OpenSER-Users] Integrating XCAP-lite with openser
by ashok.akumar＠wipro.com 29 Jun '07

29 Jun '07

Hi all, I have downloaded XCAP-lite to support XCAP. I already have openser running. But the problem is how to integrate XCAP-lite with openser? I guess for that I need to configure config.php file in xcap-root directory of XCAP-lite. Can anyone tell me how to configure config.php or what i need to do for integrating XCAP-lite with openser? Any help will be appreciated. Rgds, Ashok The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

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[OpenSER-Users] mysql.so not included in 0.9.5
by Rizwan Hisham 29 Jun '07

29 Jun '07

Hi all, I installed the openser 0.9.5. when i tried to include mysql module in openser.cfg, it gave me this error. 0(8531) ERROR: load_module: could not open module </usr/local/lib/openser/modules/mysql.so>: /usr/local/lib/openser/modules/mysql.so: cannot open shared object file: No such file or directory 0(8531) parse error (28,13-14): failed to load module I checked the source directory. It has the mysql module files included in its source then why it did not compile them and place mysql.so in the required directory. This is how i installed: -->make -->make modules -->make all -->make install Anybody with answeres ??? -- Best Regards Rizwan Hisham Software Engineer AXVOICE Inc. www.axvoice.com

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Subject=[OpenSER-Users] openser.cfg debug howto
by Patrick Miccio 29 Jun '07

29 Jun '07

just use: /path/to/openser -c -f /path/to/openser.cfg cheers, Patrick.

1 0

[OpenSER-Users] Re: A problem with CDRTool
by Dan-Cristian Bogos 29 Jun '07

29 Jun '07

Hi Jacky, looks like your rating engine is not running. Did u install and start it? I am using debian packages for cdrtool and it is automatically installed. If u used same, all u must do is edit the /etc/defaults/cdrtool to enable both ratingengine as well as memcached and then start cdrtool with /etc/init.d/cdrtool start. Hope that helps. Cheers, DanB

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[OpenSER-Users] accounting parallel invite
by yoshiki 29 Jun '07

29 Jun '07

hello ! I am weak in English. Please teach it about a accounting request-uri on parallel invite. UAC ----> OpenSER ----> UAS1 | ----> UAS2 1. UAC call to UAS. e.g. from: UAC(a)test.com -> to: UAS(a)test.com 2. avpop module in OpenSER search UAS. 3. hit UAS1 and UAS2. 4. I enumerate contact address that was found from a location table to request-URI. 5. OpenSER relay to UAS1 and UAS2. 6. UAS1 pick up call. 7. acc module account to database src_user UAC dst_user UAS1 But I want to account parallel number (UAS). How may you write in UAS to database ? -------------------------------------- Start Yahoo! Auction now! Check out the cool campaign http://pr.mail.yahoo.co.jp/auction/

2 1

[OpenSER-Users] Openser as reverse proxy
by hdpml 29 Jun '07

29 Jun '07

Dear Mailinglist, I am a rookie in SER and joined this mailinglist a few minutes ago. Hope you can help me with my question. I found ser at my search for a sip / rtp proxy. I want to install the following scenario: client -> Internet or WLAN -> SER proxy + rtpproxy(DMZ) -> Asterisk (Internal LAN) Is it possible with openser that the authentication is passed through to asterisk and openser accept the call if the user is valid? My idea is, not to have any userdata on the SER proxy. I just want that the proxy forwards the request to the asteriskserver and checks if the REGISTER is OK or not. If the user is invalid, the proxy should deny requests from this user to the asteriskserver. If this will be possible, is it possible to integrate encryption of RTP and SIP too? Like a http reverseproxy? Thank you very much. Dominic HDPnet GmbH Erwin-Rohde-Str. 18 69120 Heidelberg Geschaeftsfuehrer: Marc Hermann Registergericht: Mannheim HRB 337012 Sitz: Heidelberg Umsatzsteuer ID Nr.: DE 211 257 470 www.hdpnet.de Diese E-Mail enthaelt vertrauliche und/oder rechtlich geschuetzte Informationen. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrtuemlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser Mail ist nicht gestattet.

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[OpenSER-Users] The Via Header of CANCEL Message
by fancy 29 Jun '07

29 Jun '07

According to RFC 3665 section 3.8 and RFC 3261 section 9, there is only one via header in CANCEL message (message F11, F13 at page 72 of RFC 3665). But when I ran the similar scenario(through one proxy), I got two via header in CANCEL message such as following: CANCEL sip:UA12@node11.under.test.com;transport=udp SIP/2.0 Record-Route: <sip:192.0.50.50;lr=on;ftag=11380> Via: SIP/2.0/UDP ss.under.test.com;branch=z9hG4bKd068.25909417.0 Via: SIP/2.0/UDP node.under.test.com:5060;received=192.0.10.10;branch=z9hG4bKPUA4295483 Max-Forwards: 69 From: UA11 <sip:UA11@under.test.com>;tag=11380 To: <sip:UA12@under.test.com> Call-ID: 11380(a)tom.example.com CSeq: 4 CANCEL Content-Length: 0 P-hint: usrloc applied How can I remove the 2nd via header? Thank you Best Regards, Fangyu Ling

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