sr-users February 2004

sr-users@lists.kamailio.org

144 participants
382 discussions

by Jiri Kuthan

post message dumps too -- we don't know what is happening without them. -jiri At 07:56 PM 2/20/2004, Albert Benet wrote: > Hi > > I've setted up a TestBed about ToIP based on SIP. In this TestBed I've >used a SER Server running on a Linux Red Hat 9.0. > > I have had several problems with this TestBed for a long time. I cannot >call external phones across the gateway (Cisco 1760). Once the call is >established it lasts 20 second, before this time it hangs up. > > I would like to ask you, if its possible for you to review my >configuration in order to detect any kind of mistake or problem. > >Thanks for all your help > >Albert > -- Jiri Kuthan http://iptel.org/~jiri/

20 years, 1 month

[Serusers] cisco pix and ser

by laurent Schweizer

Hello, I want to know if somebody has already tested the cisco PIX (with NAT and SIP alg). because when i use it with ser and record-route then BYE messages are not correctly routed Laurent

20 years, 1 month

[Serusers] Rewriting "From" / Setting caller number for PSTN calls

by Arnd Vehling

Hi, with an usuall SER setup a typical user account looks like this: Primary sip address: sip:joe@myvoip.net Aliases: Berlin incoming number: sip:493042423@myvoip.net Los Angeles "" : sip:1310232323@myvoip.net If "joe" now calls out via a PSTN-GW the caller-id which is transmitted to the PSNT-GW would be "joe". Now "Joe" is clearly alphanumeric and no PSTN-GW (or the PSTN-Switch to which the PSTN-GW is connected) will accept "Joe" as caller-id. All i can currently do is setting a default callerid on the PSTN-GW in case the incoming caller-id isnt valid. What i would like to do though is to lookup if a user has an alias for a certain region and rewrite the the from line in the sip-packet going to the PSTN-GW. Example: "Joe" uses the Berlin/german PSTN-GW or dials a german PSTN number. Now i would look into the alias table and search for either an alias starting with "49" or "4930" and if one is found set the from field to the according alias. In this case "493042423(a)myvoip.net". This would ensure that, if possible, a matching caller-id would be set for every user. Problems: - "From" line rewriting is not RFC compliant. - I dont see any possebility to do this with the stock SER software Questions: - Which module would be a good starting point for adding this functionality? All which would be needed would be one command for looking up the alias list of a user from memory using regular expressions. - How do others cope with this problem? The easy but not very elegant solution is to just use a valid e.164 number (ex. the LA number) as primary sip address. But that still doesnt fix the problem that "Joes" Los Angeles number would be set as caller-id even if he got a valid "berlin" number too and calls via a german PSTN termination. best regards, Arnd

20 years, 1 month

[Serusers] Debian unstable: ser starts but socket not open

by Luna Kid

Just downloaded and installed the latest Debian unstable packages. When starting ser: ----------------- Listening on sip.neuropolis.org [212.52.166.10]:5060 Aliases: WARNING: no fork mode stateless - initializing Maxfwd module- initializing ------------------ Still I cannot connect (e.g. telnet) to 5060. The socket is NOT actually open. I guess my local Linux (Debian) config prevents ser from properly opening that socket (just dunno, how). # fuser -n tcp 5060 here: 5060 # uname -r 2.4.22 Tried low port numbers, too. Nothing. Strange that ser does not seem to complain.. Or does it? Here's a debug=7 log: http://sip.neuropolis.org/serlog.txt Some ser.cfg settings that may differ from the (latest) default stuff: --------------------- debug=7 fork=no log_stderror=no check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) listen=sip.neuropolis.org port=5060 children=2 fifo="/tmp/ser_fifo" Thanks, Luna Kid

20 years, 1 month

[Serusers] Porxy 2 Proxy

by Nikola Stojanoski

Hi, is there a way to make SER work like this with all accounting options: UA ---> SER ---> Cisco Proxy ---> GW ---> PSTN i've manage to do it but i can't send record BYE, so the call duration on the UA is stoped but i'm getting billed by the Cisco Proxy because BYE is not reaching SER and the Cisco Proxy. Has anyone made a scenario like this? Regards, Nikola

20 years, 1 month

[Serusers] Fw: rtpproxy error

by Joao Carlos Moura

Hi I am trying estabilish a communication between two IPphones. One side an IPphone behind a NAT with an IP 192.168.0.2 and on the other side an IPphone on a public and valid IP. But this did not run correctly and I got an error on the /var/log/message file and the specified error is shown below: I already searched around the lists over the net, so if you do have any clue to share with me, please do it because I don t have much experience with this but I can make it run with your help. If you need more informations from me, let me know it. Thank you very much for your attention. Kind Regards, Joao Moura ++++++++++++++++++++ ERROR +++++++++++++++++++++++++++++++++++++++++++++++ > > > >/var/log/message > > > >Feb 27 18:09:47 blacksip ser[3588]: ERROR: send_rtpp_command: can't read > >reply from a RTP proxy > >Feb 27 18:09:58 blacksip ser[3586]: ERROR: send_rtpp_command: can't read > >reply from a RTP proxy > >Feb 27 18:09:58 blacksip ser[3586]: ERROR: on_reply processing failed > > > >++++++++++++++++++++ ERROR +++++++++++++++++++++++++++++++++++++++++++++++ > > > >Linux Configuration: > >RedHat 9.0 / Athlon 1900+ / 256MB > >CVS SER_0_8_12 e RTPPROXY > > > >ser.cfg > ># > ># $Id: nathelper.cfg,v 1.1.2.1 2003/11/24 14:47:18 janakj Exp $ > ># > ># simple quick-start config script including nathelper support > > > ># This default script includes nathelper support. To make it work > ># you will also have to install Maxim's RTP proxy. The proxy is enforced > ># if one of the parties is behind a NAT. > ># > ># If you have an endpoing in the public internet which is known to > ># support symmetric RTP (Cisco PSTN gateway or voicemail, for example), > ># then you don't have to force RTP proxy. If you don't want to enforce > ># RTP proxy for some destinations than simply use t_relay() instead of > ># route(1) > ># > ># Sections marked with !! Nathelper contain modifications for nathelper > ># > ># NOTE !! This config is EXPERIMENTAL ! > ># > ># ----------- global configuration parameters ------------------------ > > > >debug=3 # debug level (cmd line: -dddddddddd) > >fork=yes > >log_stderror=no # (cmd line: -E) > > > >/* Uncomment these lines to enter debugging mode > >fork=no > >log_stderror=yes > >*/ > > > >check_via=no # (cmd. line: -v) > >dns=no # (cmd. line: -r) > >rev_dns=no # (cmd. line: -R) > >port=5060 > >children=4 > >fifo="/tmp/ser_fifo" > > > ># ------------------ module loading ---------------------------------- > > > ># Uncomment this if you want to use SQL database > >#loadmodule "/usr/local/lib/ser/modules/mysql.so" > > > >loadmodule "/usr/local/lib/ser/modules/sl.so" > >loadmodule "/usr/local/lib/ser/modules/tm.so" > >loadmodule "/usr/local/lib/ser/modules/rr.so" > >loadmodule "/usr/local/lib/ser/modules/maxfwd.so" > >loadmodule "/usr/local/lib/ser/modules/usrloc.so" > >loadmodule "/usr/local/lib/ser/modules/registrar.so" > >loadmodule "/usr/local/lib/ser/modules/textops.so" > > > ># Uncomment this if you want digest authentication > ># mysql.so must be loaded ! > >#loadmodule "/usr/local/lib/ser/modules/auth.so" > >#loadmodule "/usr/local/lib/ser/modules/auth_db.so" > > > ># !! Nathelper > >loadmodule "/usr/local/lib/ser/modules/nathelper.so" > > > ># ----------------- setting module-specific parameters --------------- > > > ># -- usrloc params -- > > > >modparam("usrloc", "db_mode", 0) > > > ># Uncomment this if you want to use SQL database > ># for persistent storage and comment the previous line > >#modparam("usrloc", "db_mode", 2) > > > ># -- auth params -- > ># Uncomment if you are using auth module > ># > >#modparam("auth_db", "calculate_ha1", yes) > ># > ># If you set "calculate_ha1" parameter to yes (which true in this config), > ># uncomment also the following parameter) > ># > >#modparam("auth_db", "password_column", "password") > > > ># -- rr params -- > ># add value to ;lr param to make some broken UAs happy > >modparam("rr", "enable_full_lr", 1) > > > ># !! Nathelper > >modparam("registrar", "nat_flag", 6) > >modparam("nathelper", "natping_interval", 30) # Ping interval 30 s > >modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT > > > ># ------------------------- request routing logic ------------------- > > > ># main routing logic > > > >route{ > > > > # initial sanity checks -- messages with > > # max_forwards==0, or excessively long requests > > if (!mf_process_maxfwd_header("10")) { > > sl_send_reply("483","Too Many Hops"); > > break; > > }; > > if (msg:len >= max_len ) { > > sl_send_reply("513", "Message too big"); > > break; > > }; > > > > # !! Nathelper > > # Special handling for NATed clients; first, NAT test is > > # executed: it looks for via!=received and RFC1918 addresses > > # in Contact (may fail if line-folding is used); also, > > # the received test should, if completed, should check all > > # vias for rpesence of received > > if (nat_uac_test("3")) { > > # Allow RR-ed requests, as these may indicate that > > # a NAT-enabled proxy takes care of it; unless it is > > # a REGISTER > > > > if (method == "REGISTER" || ! search("^Record-Route:")) { > > log("LOG: Someone trying to register from private IP, > >rewriting\n"); > > > > # This will work only for user agents that support > >symmetric > > # communication. We tested quite many of them and > >majority is > > # smart enough to be symmetric. In some phones it takes > >a configuration > > # option. With Cisco 7960, it is called NAT_Enable=Yes, > >with kphone it is > > # called "symmetric media" and "symmetric signalling". > > > > fix_nated_contact(); # Rewrite contact with source IP of > >signalling > > if (method == "INVITE") { > > fix_nated_sdp("1"); # Add direction=active to SDP > > }; > > force_rport(); # Add rport parameter to topmost Via > > setflag(6); # Mark as NATed > > }; > > }; > > > > # we record-route all messages -- to make sure that > > # subsequent messages will go through our proxy; that's > > # particularly good if upstream and downstream entities > > # use different transport protocol > > if (!method=="REGISTER") record_route(); > > > > # subsequent messages withing a dialog should take the > > # path determined by record-routing > > if (loose_route()) { > > # mark routing logic in request > > append_hf("P-hint: rr-enforced\r\n"); > > route(1); > > break; > > }; > > > > if (!uri==myself) { > > # mark routing logic in request > > append_hf("P-hint: outbound\r\n"); > > route(1); > > break; > > }; > > > > # if the request is for other domain use UsrLoc > > # (in case, it does not work, use the following command > > # with proper names and addresses in it) > > if (uri==myself) { > > > > if (method=="REGISTER") { > > > ># Uncomment this if you want to use digest authentication > ># if (!www_authorize("iptel.org", "subscriber")) { > ># www_challenge("iptel.org", "0"); > ># break; > ># }; > > > > save("location"); > > break; > > }; > > > > lookup("aliases"); > > if (!uri==myself) { > > append_hf("P-hint: outbound alias\r\n"); > > route(1); > > break; > > }; > > > > # native SIP destinations are handled using our USRLOC DB > > if (!lookup("location")) { > > sl_send_reply("404", "Not Found"); > > break; > > }; > > }; > > append_hf("P-hint: usrloc applied\r\n"); > > route(1); > >} > > > >route[1] > >{ > > # !! Nathelper > > if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && > >!search("^Route:")){ > > sl_send_reply("479", "We don't forward to private IP > >addresses"); > > break; > > }; > > > > # if client or server know to be behind a NAT, enable relay > > if (isflagset(6)) { > > force_rtp_proxy(); > > }; > > > > # NAT processing of replies; apply to all transactions (for example, > > # re-INVITEs from public to private UA are hard to identify as > > # NATed at the moment of request processing); look at replies > > t_on_reply("1"); > > > > # send it out now; use stateful forwarding as it works reliably > > # even for UDP2TCP > > if (!t_relay()) { > > sl_reply_error(); > > }; > >} > > > ># !! Nathelper > >onreply_route[1] { > > # NATed transaction ? > > if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { > > fix_nated_contact(); > > force_rtp_proxy(); > > # otherwise, is it a transaction behind a NAT and we did not > > # know at time of request processing ? (RFC1918 contacts) > > } else if (nat_uac_test("1")) { > > fix_nated_contact(); > > }; > >} > > -- > Jiri Kuthan http://iptel.org/~jiri/ >

20 years, 1 month

Re: [Serusers] Rewriting "From" / Setting caller number for PSTN calls

by Atle Samuelsen

* Arnd Vehling <av(a)nethead.de> [040228 01:13]: > Hi Atle, > > > This is the oppesit of what Im doing... > > :) Yeah, this came to my mind when i was thinking about it. He he > >but would you be needing a thing that rewrote the realname feeld? > > not an requirement at the moment. I need the ability to search and > fetch a users alias and rewrte the "From: " header. I just hope > this wont break accouting stuff. as Jiri wrote.. and as I'v been told by some of the guys here it's not safe to change the From header.. > >if > >so.. Im thinking of writing one, that can get a name out of the > >database, and say that "493042423" is Peter Pan > > If youre patch is ready ill sure will have a look at it. But i would > prefer to look up the aliases from memory because of the better > performance. Shouldnt be that hard if one looks at how "lookup(alias)" > is implemented. As I said, Im thinking of writing a patch for it.. and I will most likely do that next week.. It would be faster.. but ~6 milion phonenumbers and names would be rather much to put into memro.. - Atle

20 years, 1 month

[Serusers] Re: FW: cpl-c module

by Bogdan-Andrei IANCU

Hi Chris, please see the comments inline. Best regards, Bogdan Chris Crawford wrote: >Bogdan, >I have some easy questions for you regarding the cpl-c module. I have >looked into some of the code and the features and am pretty psyched >about what the module currently supports. > >1- Do you see any type of performance impact when processing the CPLs? > The performance impact is very difficult ot estimate since it strongly depends of the CPL script complexity. And you can estimate this complexity by the number of signaling operations done inside the script (especially proxy operations). But excepting signaling (which either you do from CPL script or from routing script, it's the same as time/resorce consuming) the additional overhead is minimal since all CPL scripts are kept in a compile version (a not as XML file), thing which tremendous reduce the interpretation time. So, I wound say the CPL execution is not more or less expensive than other SER operations. >2- Easy question... Is a cpl configuration required for every >subscriber, or can it be for ones that actually specify a cpl? For those >that don't specify a cpl, we want to use the system wide settings in the >ser.cfg. > You can use groups (ACLs) for activateing/deactivating the CPL script for specific user (on their request, for ex.). Anyhow, if a user has no script or an empty one, CPL execution will reduce to a DB query. >3- Do you have any pending functionality? > So far, I would say no. > >Cheers, > >Chris > > > >>-----Original Message----- >>From: Daniel-Constantin Mierla [mailto:Daniel- >>Constantin.Mierla(a)fokus.fraunhofer.de] >>Sent: Friday, February 27, 2004 11:35 AM >>To: Chris Crawford >>Subject: Re: cpl module >> >>Hello Chris, >>I would say to ignore that module. I was written long time ago and was >>supposed to offer basic CPL support for ser. It is using and external >>Java interpreter for CPL scripts. The module is obsoleted and it was >>replaced by cpl-c module which does the job much better and is more >>complete. >> >>Bogdan is the developer of cpl-c (also he started the Java CPL >>interpreter). What I did in cpl module was to communicate with the >> >> >Java > > >>interpreter. >> >>cpl-c follows pretty well almost all CPL specifications and we got >> >> >good > > >>feedbacks about it. If you get stuck using it send us a mail. >> >>.Daniel >> >>On 2/27/2004 5:04 PM, Chris Crawford wrote: >> >> >> >>>Hey Daniel, >>>Question regarding the CPL component. >>> >>>The module has your name on it so I figured I would ask you directly. >>>The documentation is really thin with this module. If you have any >>> >>> >to > > >>>pass on, beyond whats publicly available, that would be great. I >>>haven't really spent any time with the code. I suppose I should once >>> >>> >I > > >>>get some of the easy questions out of the way. The question I have >>> >>> >is, > > >>>does the current module support individual subscriber configuration >>> >>> >or > > >>>is a global setting? We want to use the module such that each user >>> >>> >can > > >>>provision busy, noanswer or failure on their own, without changing >>> >>> >the > > >>>ser.cfg, as we would have potentially 10,000 users, each with their >>> >>> >own > > >>>configuration. >>> >>>Chris >>> >>> >>> >>>

20 years, 1 month

[Serusers] BYE message not recognized

by Dawid Mielnik

Hi, I have a problem with SER rocognizing BYE messages sent from Asterisk - which in turn leaves open connections in my accounting database. My setup is as follows: UA--NAT--(Internet)--SER--(Internet)--Asterisk--(PSTN)--POTS sss.sss.ss.sss aaa.aaa.aa.aaa My problem is when I place a call from the SIP UA to a PSTN phone through Asterisk and the PSTN phone releases the connection. SER fails to recognize the BYE message sent by Asterisk and does not put a 'stop time' entry into my radius database. This does not happen when I call other SIP UA (with the other UA also behind NAT). Below I've attached: 1. ser.cfg for the call 2. asterisk sip debug log 3. sip trace from the UA machine Any help appreciated here, without this I have no billing ! Thanks, Dave 1. ser.cfg for the call: ---------------------------------------------------------------------------- ------------------------------------------------- # # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode debug=9 fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" # accounting loadmodule "/usr/local/lib/ser/modules/acc.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" # Nathelper loadmodule "/usr/local/lib/ser/modules/nathelper.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # -- nathelper params -- modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) # -- acc params -- modparam("acc", "radius_config", "/usr/local/etc/radiusclient/radiusclient.conf") modparam("acc", "log_level", 1) modparam("acc", "radius_flag", 1) modparam("acc", "report_ack", 0) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; # sprawdzamy czy odzywa sie ktos z za natu.... if (nat_uac_test("3")) { if (method == "REGISTER" || !search("^Record-Route:")) { log("LOG: Kolejny NATowiec...\n"); fix_nated_contact(); if (method == "INVITE") { fix_nated_sdp("1"); }; force_rport(); # dodaj do Via - topmost setflag(6); # odznacz jako nated }; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol record_route(); # loose-route processing if (loose_route()) { route(1); #t_relay(); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) # if (uri=="aaa.aaa.aa.aaa") { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("sss.sss.ss.sss", "subscriber")) { www_challenge("sss.sss.ss.sss", "0"); break; }; save("location"); break; }; setflag(1); # native SIP destinations are handled using our USRLOC DB # going to our sip users ? if (uri=~"sip:326794*" || uri=~"sip:58279*") { if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; route(1); # going to pstn } else { # }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP # coming from fax ? if (search("(f|From): .*3267940@*")) { # fax numbers # forward to fax gw rewritehostport("192.168.0.250:5060"); } else { # forward to voice gw rewritehostport("aaa.aaa.aa.aaa:5060"); }; if (!t_relay()) { sl_reply_error(); }; }; # sprawdzamy czy wysylamy do natowanych abonentow #setflag(1); #route(1); # if (!t_relay()) { # sl_reply_error(); #}; } route[1] { # if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && !search("^Route:")){ # sl_send_reply("479", "We don't forward to private IP addresses"); # break; # }; # jezeli za natem to uzywamy rtp relay if (isflagset(6)){ force_rtp_proxy(); }; # natowe przetwarzanie odpowiedzi, ma sie do wszystkich transakcji t_on_reply("1"); # nat zalatwiony wysylamy, stateful relaying if (!t_relay()){ sl_reply_error(); }; } onreply_route[1] { # nated ? if (isflagset(6) && status =~ "(183)|2[0-9][0-9]"){ fix_nated_contact(); force_rtp_proxy(); # lub, jezeli transakcja jest natowana ale nie wiedzielismy o tym # przetwarzajac zapytanie... } else if (nat_uac_test("1")) { fix_nated_contact(); }; } 2. asterisk sip debug log: ---------------------------------------------------------------------------- ------------------------------------------------- Found audio format UNKN Found description format pcmu Found description format pcma Found description format speex Found description format telephone-event Capabilities: us - 270, them - 524/0, combined - 12 Non-codec capabilities: us - 1, them - 1, combined - 1 Looking for 0225827915 in default list_route: hop: <sip:0225827915@sss.sss.ss.sss;ftag=896605854;lr=on> list_route: hop: <sip:3267915@80.55.21.254:1184> Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP sss.sss.ss.sss;branch=z9hG4bKd56d.7b60dd2.0 Via: SIP/2.0/UDP 192.168.2.32:5060;received=80.55.21.254;rport=1184;branch=z9hG4bK3AB182BF553 74A818403FAF5511D2C7B From: Dawid Mielnik <sip:3267915@sss.sss.ss.sss>;tag=896605854 To: <sip:0225827915@sss.sss.ss.sss>;tag=as37250f4f Call-ID: AB33A916-A194-4C39-AF7C-7BAF0ED22473(a)192.168.2.32 CSeq: 58806 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:0225827915@aaa.aaa.aa.aaa> Content-Length: 0 to sss.sss.ss.sss:5060 -- Executing SetCallerID("SIP/-08161e80", "223267915") in new stack -- Executing Dial("SIP/-08161e80", "Zap/g1/0225827915") in new stack -- Called g1/0225827915 We're at aaa.aaa.aa.aaa port 10548 Answering with preferred capability 4 Answering with preferred capability 8 Answering with non-codec capability 1 Transmitting (no NAT): SIP/2.0 183 Session Progress Via: SIP/2.0/UDP sss.sss.ss.sss;branch=z9hG4bKd56d.7b60dd2.0 Via: SIP/2.0/UDP 192.168.2.32:5060;received=80.55.21.254;rport=1184;branch=z9hG4bK3AB182BF553 74A818403FAF5511D2C7B From: Dawid Mielnik <sip:3267915@sss.sss.ss.sss>;tag=896605854 To: <sip:0225827915@sss.sss.ss.sss>;tag=as37250f4f Call-ID: AB33A916-A194-4C39-AF7C-7BAF0ED22473(a)192.168.2.32 CSeq: 58806 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:0225827915@aaa.aaa.aa.aaa> Content-Type: application/sdp Content-Length: 217 v=0 o=root 26423 26423 IN IP4 aaa.aaa.aa.aaa s=session c=IN IP4 aaa.aaa.aa.aaa t=0 0 m=audio 10548 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 to sss.sss.ss.sss:5060 Feb 27 18:21:48 NOTICE[1236268096]: chan_zap.c:3587 zt_read: Fax detected, but no fax extension -- Zap/1-1 is making progress passing it to SIP/-08161e80 -- Zap/1-1 is ringing 11 headers, 0 lines 10 headers, 0 lines -- Zap/1-1 answered SIP/-08161e80 We're at aaa.aaa.aa.aaa port 10548 Answering with preferred capability 4 Answering with preferred capability 8 Answering with non-codec capability 1 Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP sss.sss.ss.sss;branch=z9hG4bKd56d.7b60dd2.0 Via: SIP/2.0/UDP 192.168.2.32:5060;received=80.55.21.254;rport=1184;branch=z9hG4bK3AB182BF553 74A818403FAF5511D2C7B Record-Route: <sip:0225827915@sss.sss.ss.sss;ftag=896605854;lr=on> From: Dawid Mielnik <sip:3267915@sss.sss.ss.sss>;tag=896605854 To: <sip:0225827915@sss.sss.ss.sss>;tag=as37250f4f Call-ID: AB33A916-A194-4C39-AF7C-7BAF0ED22473(a)192.168.2.32 CSeq: 58806 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:0225827915@aaa.aaa.aa.aaa> Content-Type: application/sdp Content-Length: 217 v=0 o=root 26423 26424 IN IP4 aaa.aaa.aa.aaa s=session c=IN IP4 aaa.aaa.aa.aaa t=0 0 m=audio 10548 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 to sss.sss.ss.sss:5060 asterisk*CLI> Sip read: ACK sip:0225827915@aaa.aaa.aa.aaa:5060 SIP/2.0 Record-Route: <sip:0225827915@sss.sss.ss.sss;ftag=896605854;lr=on> Via: SIP/2.0/UDP sss.sss.ss.sss;branch=0 Via: SIP/2.0/UDP 192.168.2.32:5060;received=80.55.21.254;rport=1184;branch=z9hG4bK33D0B46E959 A49B2914EB72B18029B74 From: Dawid Mielnik <sip:3267915@sss.sss.ss.sss>;tag=896605854 To: <sip:0225827915@sss.sss.ss.sss>;tag=as37250f4f Contact: <sip:3267915@80.55.21.254:1184> Call-ID: AB33A916-A194-4C39-AF7C-7BAF0ED22473(a)192.168.2.32 CSeq: 58806 ACK Max-Forwards: 69 Content-Length: 0 11 headers, 0 lines -- Channel 1, span 1 got hangup -- Hungup 'Zap/1-1' == Spawn extension (default, 0225827915, 2) exited non-zero on 'SIP/-08161e80' set_destination: Parsing <sip:0225827915@sss.sss.ss.sss;ftag=896605854;lr=on> for address/port to send to set_destination: set destination to sss.sss.ss.sss, port 5060 Reliably Transmitting: BYE sip:3267915@80.55.21.254:1184 SIP/2.0 Via: SIP/2.0/UDP aaa.aaa.aa.aaa:5060;branch=z9hG4bK49c1001d Route: <sip:3267915@80.55.21.254:1184> From: <sip:0225827915@sss.sss.ss.sss>;tag=as37250f4f To: Dawid Mielnik <sip:3267915@sss.sss.ss.sss>;tag=896605854 Contact: <sip:0225827915@aaa.aaa.aa.aaa> Call-ID: AB33A916-A194-4C39-AF7C-7BAF0ED22473(a)192.168.2.32 CSeq: 102 BYE User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to sss.sss.ss.sss:5060 asterisk*CLI> Sip read: SIP/2.0 200 Ok Via: SIP/2.0/UDP aaa.aaa.aa.aaa:5060;branch=z9hG4bK49c1001d From: <sip:0225827915@sss.sss.ss.sss>;tag=as37250f4f To: Dawid Mielnik <sip:3267915@sss.sss.ss.sss>;tag=896605854 Contact: <sip:3267915@192.168.2.32:5060> Record-Route: <sip:0225827915@sss.sss.ss.sss;ftag=896605854;lr=on> Call-ID: AB33A916-A194-4C39-AF7C-7BAF0ED22473(a)192.168.2.32 CSeq: 102 BYE Server: X-Lite build 1088 Content-Length: 0 10 headers, 0 lines Message is BYE 3. sip trace from the UA machine: ---------------------------------------------------------------------------- ------------------------------------------------- Sip Scenario Trace File: radius_calledstat_ends4 Generated: Fri Feb 27 18:32:43 2004 Traced on: Fri Feb 27 18:23:34 2004 Created by:\\Mielonka\Techniczny\sip_project\test_tools\sip_scenario\sip_scenario.ex e version=1.2.0 192.168.2.32:5060 sss.sss.ss.sss:5060 | | <Call><PFrame><Time> | | |>F1 INVITE (sdp)-------------------------------->| 1 PF:224 18:23:51.4621 | | |<- trying -- your call is important to us 100 F2<| 1 PF:227 18:23:51.5577 | | |<------------------(sdp) Session Progress 183 F3<| 1 PF:233 18:23:51.5942 | | |>F4 (sip incomplete) >>>------------------------>| 1 PF:1097 18:24:1.2413 | | |<--------------------------------(sdp) OK 200 F5<| 1 PF:1407 18:24:4.5217 | | |>F6 ACK ---------------------------------------->| 1 PF:1410 18:24:4.5381 | | |<---------------------------------------- BYE F7<| 1 PF:1542 18:24:5.9842 | | |>F8 200 Ok ------------------------------------->| 1 PF:1543 18:24:5.9935 | | |<--------------------------<<< (sip fragment) F9<| 2 PF:2500 18:26:9.9414 ============================================================================ ==== SIP MESSAGE 1 192.168.2.32:5060(1) -> sss.sss.ss.sss:5060(2) UDP Frame 224 27/Feb/04 18:23:51.4621 TimeFromPreviousSipFrame=17.4501 TimeFromStart=17.4501 INVITE sip:0225827915@sss.sss.ss.sss SIP/2.0 Via: SIP/2.0/UDP 192.168.2.32:5060;rport;branch=z9hG4bK3AB182BF55374A818403FAF5511D2C7B From: Dawid Mielnik <sip:3267915@sss.sss.ss.sss>;tag=896605854 To: <sip:0225827915@sss.sss.ss.sss> Contact: <sip:3267915@192.168.2.32:5060> Call-ID: AB33A916-A194-4C39-AF7C-7BAF0ED22473(a)192.168.2.32 CSeq: 58806 INVITE Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite build 1088 Content-Length: 247 v=0 o=3267915 27731078 27731078 IN IP4 192.168.2.32 s=X-Lite c=IN IP4 192.168.2.32 t=0 0 m=audio 8000 RTP/AVP 0 8 97 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 ============================================================================ ==== SIP MESSAGE 2 sss.sss.ss.sss:5060(2) -> 192.168.2.32:5060(1) UDP Frame 227 27/Feb/04 18:23:51.5577 TimeFromPreviousSipFrame=0.0956 TimeFromStart=17.5457 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 192.168.2.32:5060;rport=1184;branch=z9hG4bK3AB182BF55374A818403FAF5511D2C7B; received=80.55.21.254 From: Dawid Mielnik <sip:3267915@sss.sss.ss.sss>;tag=896605854 To: <sip:0225827915@sss.sss.ss.sss> Call-ID: AB33A916-A194-4C39-AF7C-7BAF0ED22473(a)192.168.2.32 CSeq: 58806 INVITE Server: Sip EXpress router (0.8.12-tcp_nonb (i386/linux)) Content-Length: 0 Warning: 392 sss.sss.ss.sss:5060 "Noisy feedback tells: pid=7250 req_src_ip=80.55.21.254 req_src_port=1184 in_uri=sip:0225827915@sss.sss.ss.sss out_uri=sip:0225827915@aaa.aaa.aa.aaa:5060 via_cnt==1" ============================================================================ ==== SIP MESSAGE 3 sss.sss.ss.sss:5060(2) -> 192.168.2.32:5060(1) UDP Frame 233 27/Feb/04 18:23:51.5942 TimeFromPreviousSipFrame=0.0365 TimeFromStart=17.5822 SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.2.32:5060;received=80.55.21.254;rport=1184;branch=z9hG4bK3AB182BF553 74A818403FAF5511D2C7B From: Dawid Mielnik <sip:3267915@sss.sss.ss.sss>;tag=896605854 To: <sip:0225827915@sss.sss.ss.sss>;tag=as37250f4f Call-ID: AB33A916-A194-4C39-AF7C-7BAF0ED22473(a)192.168.2.32 CSeq: 58806 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:0225827915@aaa.aaa.aa.aaa> Content-Type: application/sdp Content-Length: 217 v=0 o=root 26423 26423 IN IP4 aaa.aaa.aa.aaa s=session c=IN IP4 aaa.aaa.aa.aaa t=0 0 m=audio 10548 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 ============================================================================ ==== SIP MESSAGE 4 192.168.2.32:5060(1) -> sss.sss.ss.sss:5060(2) UDP Frame 1097 27/Feb/04 18:24:1.2413 TimeFromPreviousSipFrame=9.6471 TimeFromStart=27.2293 Extra Information: Packet is not a complete SIP message ============================================================================ ==== SIP MESSAGE 5 sss.sss.ss.sss:5060(2) -> 192.168.2.32:5060(1) UDP Frame 1407 27/Feb/04 18:24:4.5217 TimeFromPreviousSipFrame=3.2805 TimeFromStart=30.5098 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.32:5060;received=80.55.21.254;rport=1184;branch=z9hG4bK3AB182BF553 74A818403FAF5511D2C7B Record-Route: <sip:0225827915@sss.sss.ss.sss;ftag=896605854;lr=on> From: Dawid Mielnik <sip:3267915@sss.sss.ss.sss>;tag=896605854 To: <sip:0225827915@sss.sss.ss.sss>;tag=as37250f4f Call-ID: AB33A916-A194-4C39-AF7C-7BAF0ED22473(a)192.168.2.32 CSeq: 58806 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:0225827915@aaa.aaa.aa.aaa> Content-Type: application/sdp Content-Length: 217 v=0 o=root 26423 26424 IN IP4 aaa.aaa.aa.aaa s=session c=IN IP4 aaa.aaa.aa.aaa t=0 0 m=audio 10548 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 ============================================================================ ==== SIP MESSAGE 6 192.168.2.32:5060(1) -> sss.sss.ss.sss:5060(2) UDP Frame 1410 27/Feb/04 18:24:4.5381 TimeFromPreviousSipFrame=0.0164 TimeFromStart=30.5261 ACK sip:0225827915@aaa.aaa.aa.aaa SIP/2.0 Via: SIP/2.0/UDP 192.168.2.32:5060;rport;branch=z9hG4bK33D0B46E959A49B2914EB72B18029B74 From: Dawid Mielnik <sip:3267915@sss.sss.ss.sss>;tag=896605854 To: <sip:0225827915@sss.sss.ss.sss>;tag=as37250f4f Contact: <sip:3267915@192.168.2.32:5060> Route: <sip:0225827915@sss.sss.ss.sss;ftag=896605854;lr=on> Call-ID: AB33A916-A194-4C39-AF7C-7BAF0ED22473(a)192.168.2.32 CSeq: 58806 ACK Max-Forwards: 70 Content-Length: 0 ============================================================================ ==== SIP MESSAGE 7 sss.sss.ss.sss:5060(2) -> 192.168.2.32:5060(1) UDP Frame 1542 27/Feb/04 18:24:5.9842 TimeFromPreviousSipFrame=1.4461 TimeFromStart=31.9723 BYE sip:3267915@80.55.21.254:1184 SIP/2.0 Max-Forwards: 10 Record-Route: <sip:3267915@sss.sss.ss.sss;ftag=as37250f4f;lr=on> Via: SIP/2.0/UDP sss.sss.ss.sss;branch=z9hG4bKe74a.b59c9824.0 Via: SIP/2.0/UDP aaa.aaa.aa.aaa:5060;branch=z9hG4bK49c1001d From: <sip:0225827915@sss.sss.ss.sss>;tag=as37250f4f To: Dawid Mielnik <sip:3267915@sss.sss.ss.sss>;tag=896605854 Contact: <sip:0225827915@aaa.aaa.aa.aaa> Call-ID: AB33A916-A194-4C39-AF7C-7BAF0ED22473(a)192.168.2.32 CSeq: 102 BYE User-Agent: Asterisk PBX Content-Length: 0 Route: <sip:3267915@80.55.21.254:1184> ============================================================================ ==== SIP MESSAGE 8 192.168.2.32:5060(1) -> sss.sss.ss.sss:5060(2) UDP Frame 1543 27/Feb/04 18:24:5.9935 TimeFromPreviousSipFrame=0.0092 TimeFromStart=31.9815 SIP/2.0 200 Ok Via: SIP/2.0/UDP sss.sss.ss.sss;branch=z9hG4bKe74a.b59c9824.0 Via: SIP/2.0/UDP aaa.aaa.aa.aaa:5060;branch=z9hG4bK49c1001d From: <sip:0225827915@sss.sss.ss.sss>;tag=as37250f4f To: Dawid Mielnik <sip:3267915@sss.sss.ss.sss>;tag=896605854 Contact: <sip:3267915@192.168.2.32:5060> Record-Route: <sip:0225827915@sss.sss.ss.sss;ftag=896605854;lr=on> Call-ID: AB33A916-A194-4C39-AF7C-7BAF0ED22473(a)192.168.2.32 CSeq: 102 BYE Server: X-Lite build 1088 Content-Length: 0 ============================================================================ ==== SIP MESSAGE 9 sss.sss.ss.sss:5060(2) -> 192.168.2.32:5060(1) UDP Frame 2500 27/Feb/04 18:26:9.9414 TimeFromPreviousSipFrame=123.9479 TimeFromStart=155.9294 Extra Information: Packet was continued from Frame=1643 Extra Information: Packet was continued from Frame=2179 Extra Information: Packet was continued from Frame=2279 Extra Information: Packet was continued from Frame=2369 Extra Information: Packet is not a complete SIP message Extra Information: Packet does NOT contain a SIP Header but was in the same connection as Frame=1542 ============================================================================ ==== 2 incomplete sip message(s) encountered

20 years, 1 month

RE: [Serusers] replication and rtpproxy

by Darren Nay

Very True. We use the same UAC for all of our clients and it has an option in the configuration to force a DNS lookup every time that it sends a packet to the proxy. This does add some overhead, especially to the DNS servers, but it adds this functionality as well. Also, Its worth mentioning that your method of testing SER via SipSak and modifying the DNS entries when SER fails to respond is more efficient in that the failover is based on SER availability. This DNS load balancing technique that I posted will only work for fault tolerance if the server itself dies. If SER crashes, but bind keeps running then it won't do the job. Regards, Darren Nay -----Original Message----- From: Arnd Vehling [mailto:av@nethead.de] Sent: Thursday, February 26, 2004 3:25 PM To: Darren Nay Cc: 'Klaus Darilion'; Andres; Nils Ohlmeier; serusers(a)lists.iptel.org Subject: Re: [Serusers] replication and rtpproxy Darren Nay wrote: > Check out this method. > http://www.presttun.org/kare/DNS/DNS-LB-FT.pdf Sounds nice but it makes you dependant on the dns resolver implementation in every UAC on the market. In addition it may happen that caching DNS servers will not honor your expire setting of 10 Seconds and the UAC will therefore continue to contact a dead proxy. best regards, Arnd

20 years, 1 month

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sr-users February 2004