sr-users July 2004

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[Serusers] Asterix - Ser
by "Juan G. Castañeda" 02 Jul '04

02 Jul '04

Can Asterix be used as a PSTN gateway for Ser? If it is so, how should Ser be configured?. Thanks Juan G. Castañeda

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[Serusers] Ser and new Mediaproxy
by Tjapko Smits 02 Jul '04

02 Jul '04

Hello , Can someone provide me full working ser.cfg example for latest version of Mediaproxy (example distributed with new mediaproxy does not seem to work properly). Herunder you will find the test.cfg that we use. Regards, Tjapko. Here is the ser.cfg that we use to test mediaproxy. # Example ser.cfg for mediaproxy functionality debug=7 fork=no log_stderror=yes check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 #children=4 fifo="/tmp/ser_fifo" alias=some.site.com listen=82.100.100.100 loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" #loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" modparam("usrloc","db_mode",0) modparam("rr", "enable_full_lr", 1) modparam("mediaproxy", "natping_interval", 60) route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too many hops"); break; }; if (msg:len >= max_len) { sl_send_reply("513", "Message too big"); break; }; if (method == "BYE" || method == "CANCEL") { end_media_session(); }; if (loose_route()) { t_relay(); break; }; if (method == "INVITE") { record_route(); }; if (client_nat_test("3")) { if (method == "REGISTER" || ! search("^Record-Route:")) { fix_contact(); force_rport(); }; if (method == "REGISTER"){ save("location"); break; }; if (method=="INVITE") { use_media_proxy(); t_on_reply("1"); }; if (!t_relay()) { sl_reply_error(); }; } onreply_route[1] { if (status=~"(183)|2[0-9][0-9]") { if (client_nat_test("1")) { fix_contact(); }; fix_contact(); }; use_media_proxy(); }; if (status=~"[3-4]0[0-9]") { end_media_session(); break; }; }

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[Serusers] From-URI dependent actions
by Martin Koenig 02 Jul '04

02 Jul '04

Hi List, is it possible to rewrite the TO-URI depending on FROM-URI, RPID or other data provided by radius, mysql or other sources? So to speak to enable location based services? Usergroup A (1xxx(a)sip.com) in Building A should reach GW-A, Usergroup B (2xxx(a)sip.com) in Building B should reach GW-B? Or would it be more advised to just use Groups and assign every user in Building A to Group A? Would cause action for every user though. Thank you, Martin

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Re: [Serusers] non-unique call id problem
by GR S 02 Jul '04

02 Jul '04

Hello All, > Greg Fausak <greg(a)addabrand.com> wrote: > I am seeing non-unique call IDs from a UA. > SHould I reject a call if the Call-ID is not unique? Has > anyone else messed with this? It is a big accounting problem! > > Greg Fausak I apologize if this is an off-topic question, but this is about SIP. Hope folks dont mind... This thread is a bit old, but i am curious about a comment i saw in the Asterisk SIP channel code. It is mentioned that some vendors send multiple calls with the same Call-ID and different tags to simplify billing. It also says that it is an overhead for the other *sane* SIP implementations. I am learning SIP and dont have much idea about billing, but what makes some vendors think that it is easy for billing, where as it is a big problem for some others? What is the right way of doing it? How can such differences bring consistancy among different SIP implementations? What method does SER follow? This is from the Asterisk SIP channel code: /* In principle Call-ID's uniquely identify a call, however some vendors (i.e. Pingtel) send multiple calls with the same Call-ID and different tags in order to simplify billing. The RFC does state that we have to compare tags in addition to the call-id, but this generate substantially more overhead which is totally unnecessary for the vast majority of sane SIP implementations, and thus Asterisk does not enable this behavior by default. Short version: You'll need this option to support conferencing on the pingtel */ Regards, ===== Girish Gopinath <gr_sh2003(a)yahoo.com> __________________________________ Do you Yahoo!? New and Improved Yahoo! Mail - 100MB free storage! http://promotions.yahoo.com/new_mail

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RE: [Serusers] B2BUA
by Vitaly Nikolaev 02 Jul '04

02 Jul '04

Hello, Is there any information about this product, how can I buy it. How can I get trial version. My company is using SER in production and we like it very much. If you guys (iptel.org) have commercial version we probably would happy to buy it (with support) but all my mails to info(a)iptel.org were unanswered. Please respond to my mail with info I need. Probably you are ignoring big customer :) Thank you -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Andy Blen Sent: Wednesday, June 30, 2004 9:46 AM To: Gala, Dipen; serusers(a)lists.iptel.org Subject: Re: [Serusers] B2BUA Application Agent, iptel's highly programmable B2BUA for use in service provider networks, is a commercial item and is not downloadable on a free basis. Andy At 03:30 PM 6/30/2004, Gala, Dipen wrote: >Has anyone installed or used B2BUA with ser. Pls share with me how to start and from where can I download the AA, and do I have to code in python in order to get it working the way I want??? > >Dipen K Gala >Fidelity Investments System Company - Telecom >Phone: (617) 563-2729 >Email: Dipen.Gala(a)FMR.com >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Andy Blen iptel.org Services _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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Re: [Serusers] SER 0.8.13 Segmentation Fault
by Bogdan-Andrei IANCU 02 Jul '04

02 Jul '04

Hi Ezequiel, thanks for info. I found the problem: you are loading acc module before tm nad acc iss based on tm (it tries to register some callbacks, but tm is not init yet). So,the fast solution is to load acc after tm. I will try to fix it properly in the mean while. thanks for report, bogdan Ezequiel Colombo wrote: > Hi Bogdan, thanks for your quickly response. > I have packed my sip_router (sources) directory with the core.22059.gz file > (the core dump [w/ulimit unlimited]) > > You can try download this tarball at > http://200.80.35.17/sip_router.seg_fault.0.8.13.tar.gz {8.3Mb long} > > Thanks again, and reply back to me if you have any solution or request. > Ezequiel Colombo > > > ----- Original Message ----- > From: "Bogdan-Andrei IANCU" <iancu(a)fokus.fraunhofer.de> > To: "Ezequiel Colombo" <ecolombo(a)arcotel.net> > Cc: <serusers(a)lists.iptel.org> > Sent: Thursday, July 01, 2004 4:02 PM > Subject: Re: [Serusers] SER 0.8.13 Segmentation Fault > > > >>Hi, >> >>thanks for the report. Please be sure to get a core (do ulimit -c >>unlimited before runing ser) and send us the backtrace. if possible put >>on some public ftp/http site the core and the sources. >> >>thanks again, >>bogdan >> >>Ezequiel Colombo wrote: >> >>>Hi, i am trying to upgrade to 0.8.13 and get Segmentation Fault when run >>>SER. >>>All the CVS source compile fine and the configuration test with "-c" >>>pass but SER dont start. >>>I have the core dump file, if any of the developers can help me please >>>request it off the list. >>> >>>[root@billing sip_router]# ./ser -f ./ser.cfg -E -D -d >>>Listening on >>> udp: 172.16.0.1 [172.16.0.1]:5060 >>> tcp: 172.16.0.1 [172.16.0.1]:5060 >>>Aliases: >>> tcp: labo:5060 >>> tcp: labo.arcotel.net:5060 >>> udp: labodomain:5060 >>> udp: labo.labodomain.net:5060 >>> *: labodomain.net:* >>> >>>WARNING: no fork mode >>>acc - initializing >>>Segmentation fault >>> >>>Thanks >>>Ezequiel Colombo >>> >>> >>>------------------------------------------------------------------------ >>> >>>_______________________________________________ >>>Serusers mailing list >>>serusers(a)lists.iptel.org >>>http://lists.iptel.org/mailman/listinfo/serusers >> >> >

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[Serusers] General ser_fifo question
by Fastnet Technologies 02 Jul '04

02 Jul '04

My ser is running fine as is mysql database. I have been monitoring the ser_fifo file and noticed it never changes in size. Is this normal? Since I am having trouble getting the serweb working properly (only admin can login), I was wondering if this is a related problem?

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[Serusers] ser and asterisk and rtpproxy
by Aldo Armiento 02 Jul '04

02 Jul '04

Hi all! I know the argument is very famouse, I think I have read all ser.cfg on the net.... The problem is that when I call extension 9916 I don't hear asterisk music-on-old sound... The scenario: ser at port 5060 asterisk at port 5090 rtpproxy I would like that, when an UA (x-lite for example) call 9916 UA can hear "music-on-hold" in asterisk. *From ser.cfg:* if (uri=~"sip:9916@") { log("\n\nLOG: 9916 - Asterisk - Test()\n\n"); force_rtp_proxy(); t_relay_to_udp("xxx.xxx.xxx.xxx","5090"); break; } *From asterisk sip.conf:* [general] port = 5090 bindaddr = xxx.xxx.xxx.xxx externip = xxx.xxx.xxx.xxx tos=lowdelay ; disallow = all allow = ulaw allow = alaw ; context = sip-callfrom language = us musicclass = default at now when I call 9916 from x-lite: *ser says:* 0(31421) SIP Request: 0(31421) method: <INVITE> 0(31421) uri: <sip:9916@xxx.xxx.xxx.xxx> 0(31421) version: <SIP/2.0> 0(31421) parse_headers: flags=1 0(31421) Found param type 235, <rport> = <n/a>; state=6 0(31421) Found param type 232, <branch> = <z9hG4bKB3CD5F29F30B4329995057709DFC7500>; state=16 0(31421) end of header reached, state=5 0(31421) parse_headers: Via found, flags=1 0(31421) parse_headers: this is the first via 0(31421) After parse_msg... 0(31421) preparing to run routing scripts... 0(31421) DEBUG : is_maxfwd_present: searching for max_forwards header 0(31421) parse_headers: flags=128 0(31421) end of header reached, state=9 0(31421) DEBUG: get_hdr_field: <To> [26]; uri=[sip:9916@xxx.xxx.xxx.xxx] 0(31421) DEBUG: to body [<sip:9916@xxx.xxx.xxx.xxx> ] 0(31421) get_hdr_field: cseq <CSeq>: <12362> <INVITE> 0(31421) DEBUG: is_maxfwd_present: value = 70 0(31421) check_via_address(192.168.6.101, 192.168.6.101, 0) 0(31421) parse_headers: flags=64 0(31421) parse_headers: flags=-1 0(31421) DEBUG: get_hdr_body : content_length=267 0(31421) found end of header 0(31421) LOG: Someone trying to register from private IP, rewriting 0(31421) parse_headers: flags=64 0(31421) parse_headers: flags=-1 0(31421) parse_headers: flags=-1 0(31421) DEBUG:check_content_type: type <application/sdp> found valid 0(31421) DEBUG: add_param: tag=2216232801 0(31421) end of header reached, state=29 0(31421) parse_headers: flags=256 0(31421) find_first_route(): No Route headers found 0(31421) loose_route(): There is no Route HF 0(31421) LOG: 9916 - Asterisk - Test() 0(31421) parse_headers: flags=-1 0(31421) DEBUG:check_content_type: type <application/sdp> found valid 0(31421) ERROR: extract_mediaip: no `c=' in SDP 0(31421) DEBUG: t_newtran: msg id=7 , global msg id=6 , T on entrance=0xffffffff 0(31421) parse_headers: flags=-1 0(31421) parse_headers: flags=60 0(31421) t_lookup_request: start searching: hash=6960, isACK=0 0(31421) DEBUG: RFC3261 transaction matching failed 0(31421) DEBUG: t_lookup_request: no transaction found 0(31421) SER: new INVITE 0(31421) parse_headers: flags=-1 0(31421) check_via_address(192.168.6.101, 192.168.6.101, 0) 0(31421) WARNING:vqm_resize: resize(0) called 0(31421) DEBUG: reply sent out. buf=0x80f9b98: SIP/2.0 1..., shmem=0x4019d0f0: SIP/2.0 1 0(31421) DEBUG: _reply_light: finished 0(31421) parse_headers: flags=2048 0(31421) parse_headers: flags=-1 0(31421) clen_builder: content-length: 306 (306) 0(31421) check_via_address(192.168.6.101, 192.168.6.101, 0) 0(31421) DEBUG: add_timer_unsafe[4]: 0x4019d520 0(31421) DEBUG: add_timer_unsafe[0]: 0x4019d530 0(31421) SER: new transaction fwd'ed 0(31421) receive_msg: cleaning up 0(31421) SIP Reply (status): 0(31421) version: <SIP/2.0> 0(31421) status: <100> 0(31421) reason: <Trying> 0(31421) parse_headers: flags=1 0(31421) Found param type 232, <branch> = <z9hG4bK03b1.24325446.0>; state=16 0(31421) end of header reached, state=5 0(31421) parse_headers: Via found, flags=1 0(31421) parse_headers: this is the first via 0(31421) After parse_msg... 0(31421) forward_reply: found module tm, passing reply to it 0(31421) DEBUG: t_check: msg id=8 global id=7 T start=0xffffffff 0(31421) parse_headers: flags=17 0(31421) Found param type 235, <rport> = <5060>; state=6 0(31421) Found param type 232, <branch> = <z9hG4bKB3CD5F29F30B4329995057709DFC7500>; state=16 0(31421) end of header reached, state=5 0(31421) parse_headers: Via found, flags=17 0(31421) parse_headers: this is the second via 0(31421) DEBUG: add_param: tag=as72cacbaf 0(31421) end of header reached, state=29 0(31421) DEBUG: get_hdr_field: <To> [41]; uri=[sip:9916@xxx.xxx.xxx.xxx] 0(31421) DEBUG: to body [<sip:9916@xxx.xxx.xxx.xxx>] 0(31421) get_hdr_field: cseq <CSeq>: <12362> <INVITE> 0(31421) parse_headers: flags=4 0(31421) DEBUG: t_reply_matching: hash 6960 label 1682252610 branch 0 0(31421) DEBUG: t_reply_matching: reply matched (T=0x4019d408)! 0(31421) DEBUG: t_check: msg id=8 global id=8 T end=0x4019d408 0(31421) DEBUG: reply_received: org. status uas=100, uac[0]=0 local=0 is_invite=1) 0(31421) ->>>>>>>>> T_code=100, new_code=100 0(31421) DEBUG: relay_reply: branch=0, save=0, relay=-1 0(31421) DEBUG: add_timer_unsafe[1]: 0x4019d530 0(31421) receive_msg: cleaning up 0(31421) SIP Reply (status): 0(31421) version: <SIP/2.0> 0(31421) status: <200> 0(31421) reason: <OK> 0(31421) parse_headers: flags=1 0(31421) Found param type 232, <branch> = <z9hG4bK03b1.24325446.0>; state=16 0(31421) end of header reached, state=5 0(31421) parse_headers: Via found, flags=1 0(31421) parse_headers: this is the first via 0(31421) After parse_msg... 0(31421) forward_reply: found module tm, passing reply to it 0(31421) DEBUG: t_check: msg id=9 global id=8 T start=0xffffffff 0(31421) parse_headers: flags=17 0(31421) Found param type 235, <rport> = <5060>; state=6 0(31421) Found param type 232, <branch> = <z9hG4bKB3CD5F29F30B4329995057709DFC7500>; state=16 0(31421) end of header reached, state=5 0(31421) parse_headers: Via found, flags=17 0(31421) parse_headers: this is the second via 0(31421) DEBUG: add_param: tag=as72cacbaf 0(31421) end of header reached, state=29 0(31421) DEBUG: get_hdr_field: <To> [41]; uri=[sip:9916@xxx.xxx.xxx.xxx] 0(31421) DEBUG: to body [<sip:9916@xxx.xxx.xxx.xxx>] 0(31421) get_hdr_field: cseq <CSeq>: <12362> <INVITE> 0(31421) parse_headers: flags=4 0(31421) DEBUG: t_reply_matching: hash 6960 label 1682252610 branch 0 0(31421) DEBUG: t_reply_matching: reply matched (T=0x4019d408)! 0(31421) parse_headers: flags=4 0(31421) DEBUG: t_check: msg id=9 global id=9 T end=0x4019d408 0(31421) DEBUG: reply_received: org. status uas=100, uac[0]=100 local=0 is_invite=1) 0(31421) ->>>>>>>>> T_code=100, new_code=200 0(31421) DEBUG: relay_reply: branch=0, save=0, relay=0 0(31421) old size: 842, new size: 780 0(31421) build_res_from_sip_res: copied size: orig:78, new: 16, rest: 764 msg= SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.6.101:5060;rport=5060;branch=z9hG4bKB3CD5F29F30B4329995057709DFC7500 Record-Route: <sip:xxx.xxx.xxx.xxx;ftag=2216232801;lr=on> From: 2001 <sip:2001@xxx.xxx.xxx.xxx>;tag=2216232801 To: <sip:9916@xxx.xxx.xxx.xxx>;tag=as72cacbaf Call-ID: 7C0EAF05-D24D-463B-BB2A-F7746D751921(a)192.168.6.101 CSeq: 12362 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:9916@xxx.xxx.xxx.xxx:5090> Content-Type: application/sdp Content-Length: 265 v=0 o=root 31456 31456 IN IP4 xxx.xxx.xxx.xxx s=session c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=audio 10100 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - 0(31421) DEBUG: update_totag_set: new totag 0(31421) DEBUG: reply relayed. buf=0x80f9de8: SIP/2.0 2..., shmem=0x4019b240: SIP/2.0 2 0(31421) DEBUG: cleanup_uac_timers: RETR/FR timers reset 0(31421) DEBUG: add_timer_unsafe[2]: 0x4019d450 0(31421) receive_msg: cleaning up 0(31421) SIP Request: 0(31421) method: <ACK> 0(31421) uri: <sip:9916@xxx.xxx.xxx.xxx:5090> 0(31421) version: <SIP/2.0> 0(31421) parse_headers: flags=1 0(31421) Found param type 235, <rport> = <n/a>; state=6 0(31421) Found param type 232, <branch> = <z9hG4bKA83370958AA746AFA673710404627DFC>; state=16 0(31421) end of header reached, state=5 0(31421) parse_headers: Via found, flags=1 0(31421) parse_headers: this is the first via 0(31421) After parse_msg... 0(31421) preparing to run routing scripts... 0(31421) DEBUG : sl_filter_ACK: to late to be a local ACK! 0(31421) DEBUG : is_maxfwd_present: searching for max_forwards header 0(31421) parse_headers: flags=128 0(31421) DEBUG: add_param: tag=as72cacbaf 0(31421) end of header reached, state=29 0(31421) DEBUG: get_hdr_field: <To> [41]; uri=[sip:9916@xxx.xxx.xxx.xxx] 0(31421) DEBUG: to body [<sip:9916@xxx.xxx.xxx.xxx>] 0(31421) get_hdr_field: cseq <CSeq>: <12362> <ACK> 0(31421) DEBUG: is_maxfwd_present: value = 70 0(31421) check_via_address(192.168.6.101, 192.168.6.101, 0) 0(31421) parse_headers: flags=64 0(31421) parse_headers: flags=-1 0(31421) DEBUG: get_hdr_body : content_length=0 0(31421) found end of header 0(31421) LOG: Someone trying to register from private IP, rewriting 0(31421) parse_headers: flags=64 0(31421) DEBUG: add_param: tag=2216232801 0(31421) end of header reached, state=29 0(31421) parse_headers: flags=256 0(31421) check_self - checking if host==us: 13==13 && [xxx.xxx.xxx.xxx] == [xxx.xxx.xxx.xxx] 0(31421) check_self - checking if port 5060 matches port 5090 0(31421) check_self - checking if host==us: 13==13 && [xxx.xxx.xxx.xxx] == [xxx.xxx.xxx.xxx] 0(31421) check_self - checking if port 5060 matches port 5090 0(31421) check_self: host != me 0(31421) check_self - checking if host==us: 13==13 && [xxx.xxx.xxx.xxx] == [xxx.xxx.xxx.xxx] 0(31421) check_self - checking if port 5060 matches port 5060 0(31421) ral(): Topmost route URI: 'sip:xxx.xxx.xxx.xxx;ftag=2216232801;lr=on' is me 0(31421) parse_headers: flags=256 0(31421) found end of header 0(31421) fnr(): No next Route HF found 0(31421) ral(): No next URI found 0(31421) LOG: 9916 - Asterisk - Test() 0(31421) DEBUG: t_newtran: msg id=10 , global msg id=9 , T on entrance=0xffffffff 0(31421) parse_headers: flags=-1 0(31421) parse_headers: flags=60 0(31421) t_lookup_request: start searching: hash=6960, isACK=1 0(31421) parse_headers: flags=28 0(31421) DEBUG: t_lookup_request: e2e proxy ACK found 0(31421) parse_headers: flags=4 0(31421) DEBUG: totag for e2e ACK found: 0 0(31421) SER: forwarding ACK statelessly 0(31421) check_via_address(192.168.6.101, 192.168.6.101, 0) 0(31421) Sending: ACK sip:9916@xxx.xxx.xxx.xxx:5090 SIP/2.0 Record-Route: <sip:xxx.xxx.xxx.xxx;ftag=2216232801;lr=on> Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;branch=0 Via: SIP/2.0/UDP 192.168.6.101:5060;rport=5060;branch=z9hG4bKA83370958AA746AFA673710404627DFC From: 2001 <sip:2001@xxx.xxx.xxx.xxx>;tag=2216232801 To: <sip:9916@xxx.xxx.xxx.xxx>;tag=as72cacbaf Contact: <sip:2001@192.168.6.101:5060> Call-ID: 7C0EAF05-D24D-463B-BB2A-F7746D751921(a)192.168.6.101 CSeq: 12362 ACK Max-Forwards: 69 Content-Length: 0 Voco-dbg: NAThelper . 0(31421) orig. len=435, new_len=509, proto=1 0(31421) receive_msg: cleaning up 1(31422) DEBUG: timer routine:1,tl=0x4019d530 next=(nil) 1(31422) DEBUG: timer routine:4,tl=0x4019d520 next=(nil) *rtpproxy -f says:* rtpproxy: new session 7C0EAF05-D24D-463B-BB2A-F7746D751921(a)192.168.6.101, tag 2216232801 requested rtpproxy: new session on a port 10018 created, tag 2216232801 rtpproxy: pre-filling caller's address with 192.168.6.101:8000 *asterisk sip debug says:* Sip read: INVITE sip:9916@xxx.xxx.xxx.xxx:5090 SIP/2.0 Record-Route: <sip:xxx.xxx.xxx.xxx;ftag=2216232801;lr=on> Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;branch=z9hG4bK03b1.24325446.0 Via: SIP/2.0/UDP 192.168.6.101:5060;rport=5060;branch=z9hG4bKB3CD5F29F30B4329995057709DFC7500 From: 2001 <sip:2001@xxx.xxx.xxx.xxx>;tag=2216232801 To: <sip:9916@xxx.xxx.xxx.xxx> Contact: <sip:2001@192.168.6.101:5060> Call-ID: 7C0EAF05-D24D-463B-BB2A-F7746D751921(a)192.168.6.101 CSeq: 12362 INVITE Max-Forwards: 69 Content-Type: application/sdp User-Agent: X-Lite release 1103m Content-Length: 306 Voco-dbg: NAThelper Voco-dbg: SDP rewrited v=0 o=2001 8220109 8220125 IN IP4 192.168.6.101 s=X-Lite c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=audio 10018 RTP/AVP 0 8 3 97 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=direction:active a=nortpproxy:yes 15 headers, 14 lines Using latest request as basis request Sending to xxx.xxx.xxx.xxx : 5060 (non-NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 97 Found RTP audio format 101 Peer RTP is at port xxx.xxx.xxx.xxx:0 Found description format pcmu Found description format pcma Found description format gsm Found description format speex Found description format telephone-event Capabilities: us - 0x40e(GSM|ULAW|ALAW|ILBC), peer - audio=0x20e(GSM|ULAW|ALAW|SPEEX)/video=0x0(EMPTY), combined - 0xe(GSM|ULAW|ALAW) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) Found no matching peer or user for 'xxx.xxx.xxx.xxx:5060' Looking for 9916 in sip-callfrom list_route: hop: <sip:xxx.xxx.xxx.xxx;ftag=2216232801;lr=on> list_route: hop: <sip:2001@192.168.6.101:5060> Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;branch=z9hG4bK03b1.24325446.0 Via: SIP/2.0/UDP 192.168.6.101:5060;rport=5060;branch=z9hG4bKB3CD5F29F30B4329995057709DFC7500 From: 2001 <sip:2001@xxx.xxx.xxx.xxx>;tag=2216232801 To: <sip:9916@xxx.xxx.xxx.xxx>;tag=as72cacbaf Call-ID: 7C0EAF05-D24D-463B-BB2A-F7746D751921(a)192.168.6.101 CSeq: 12362 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:9916@xxx.xxx.xxx.xxx:5090> Content-Length: 0 to xxx.xxx.xxx.xxx:5060 -- Executing Answer("SIP/xxx.xxx.xxx.xxx-081346f0", "") in new stack We're at xxx.xxx.xxx.xxx port 10100 Answering with preferred capability 0x2(GSM) Answering with preferred capability 0x4(ULAW) Answering with preferred capability 0x8(ALAW) Answering with non-codec capability 0x1(G723) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;branch=z9hG4bK03b1.24325446.0 Via: SIP/2.0/UDP 192.168.6.101:5060;rport=5060;branch=z9hG4bKB3CD5F29F30B4329995057709DFC7500 Record-Route: <sip:xxx.xxx.xxx.xxx;ftag=2216232801;lr=on> From: 2001 <sip:2001@xxx.xxx.xxx.xxx>;tag=2216232801 To: <sip:9916@xxx.xxx.xxx.xxx>;tag=as72cacbaf Call-ID: 7C0EAF05-D24D-463B-BB2A-F7746D751921(a)192.168.6.101 CSeq: 12362 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:9916@xxx.xxx.xxx.xxx:5090> Content-Type: application/sdp Content-Length: 265 v=0 o=root 31456 31456 IN IP4 xxx.xxx.xxx.xxx s=session c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=audio 10100 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to xxx.xxx.xxx.xxx:5060 -- Executing MusicOnHold("SIP/xxx.xxx.xxx.xxx-081346f0", "default") in new stack -- Started music on hold, class 'default', on SIP/xxx.xxx.xxx.xxx-081346f0 Sip read: ACK sip:9916@xxx.xxx.xxx.xxx:5090 SIP/2.0 Record-Route: <sip:xxx.xxx.xxx.xxx;ftag=2216232801;lr=on> Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;branch=0 Via: SIP/2.0/UDP 192.168.6.101:5060;rport=5060;branch=z9hG4bKA83370958AA746AFA673710404627DFC From: 2001 <sip:2001@xxx.xxx.xxx.xxx>;tag=2216232801 To: <sip:9916@xxx.xxx.xxx.xxx>;tag=as72cacbaf Contact: <sip:2001@192.168.6.101:5060> Call-ID: 7C0EAF05-D24D-463B-BB2A-F7746D751921(a)192.168.6.101 CSeq: 12362 ACK Max-Forwards: 69 Content-Length: 0 Voco-dbg: NAThelper 12 headers, 0 lines

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[Fwd: Re: [Serusers] SER on port xxxx to SER on port 5060]
by Aldo Armiento 02 Jul '04

02 Jul '04

-------- Original Message -------- Subject: Re: [Serusers] SER on port xxxx to SER on port 5060 Date: Thu, 01 Jul 2004 14:56:50 +0200 From: Aldo Armiento <aldo(a)cerca.com> Organization: Cerca.com s.r.l. To: daniel(a)iptel.org References: <40E34AFC.40309(a)armiento.com> <40E3C3E1.6010300(a)iptel.org> Dear Daniel, ok, i known "port" in global config. My question is: how can I tell (routing rules) to "fake" ser at port 5090 to work as "proxy" for "real" ser at port 5060? UA(1) --> NAT --> SER (5090) --> SER (5060) --> NAT --> UA(2) | |--> this NAT currupts packets to port 5060 (!!) so I have to use another port but I want other "normal" UA (with or without a good nat) work on port 5060. Thanks you!! Aldo. Daniel-Constantin Mierla wrote: > Hello, > use port=5090 in your ser.cfg config file -- in global configuration > parameters section. > > Daniel > > On 7/1/2004 1:21 AM, Aldo Armiento wrote: > >> Hi all! >> Some my sip clients are behind a nat on an internet public network >> that "corrupt" UDP packect (wrong checksum) if is directed to port 5060. >> Is possible to have a SER on port (for example) 5090 as a "proxy" for >> the real ser on port 5060? How I have to configure ser.cfg for "fake" >> ser on port 5090? >> >> Thanks you, >> Aldo. >> >> _______________________________________________ >> Serusers mailing list >> serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >>

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[Serusers] problem with fifo server
by Emanuel Schmid 02 Jul '04

02 Jul '04

Hello I'm trying to send a SIP-Info via the ser fifo server. This works fine for one time, but if I want to send a second message, I have to wait a certain time or I get the following errors: ------------------------- 9(12790) ERROR: fifo_server: command must begin with :: INFO 9(12790) ERROR: fifo_server: command must have at least 3 chars 9(12790) ERROR: fifo_server: command must begin with :: To: <sip:dummy@pc33.faerun.ch> 9(12790) ERROR: fifo_server: command must begin with :: Content-Type: text/plain; charset=iso8859-1 9(12790) ERROR: fifo_server: command must begin with :: And this is a text again. ------------------------ i used the following skript to get the message to the fifo: ------------------------ cat > /tmp/ser_fifo <<EOF :t_uac_dlg:hh INFO sip:dummy@pc33.faerun.ch . From: <sip:ser@faerun.ch> To: <sip:dummy@pc33.faerun.ch> Contact: <sip:devnull@pc38.faerun.ch> Content-Type: text/plain; charset=iso8859-1 . And this is a text again. . EOF -------------------------- If the message is acceptet, everything works as expected. Counting the lines in the script and comparing them to the errors, it looks as if the second time I tried to send the message the first line was consumed without being processed (by the last call?). What am I doing wrong? thanks a lot emanuel

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sr-users July 2004