- sr-users - kamailio.org

Re: [Serusers] Re: [Serdev] rel_0_9_0 release critical bugs/missingthings
by Samuel Osorio Calvo 21 Jun '05

21 Jun '05

Undocummented code = non existing code (for almomst everyone except the developer and few guys reading the devel list, updating the working copy of the CVS, and checking the new code).....I know it is unpleasant, but two lines with an overview is not asking too much.....isn't it? my contribution, Samuel. Unclassified. >>> Jan Janak <jan(a)iptel.org> 06/21/05 11:58AM >>> On 21-06-2005 09:20, Juha Heinanen wrote: > Greger V. Teigre writes: > > > Let's try to make 0.9.x documentation as complete as possible now > > leading up to the release!! > > nathelper README doesn't describe all nat_uac_test modes. tm module > README seems to lack description of several exported functions. > > these both are example of bad iptel.org policy. no new features should > be allowed to be added/changed in the code unless also the documentation > is updated. Are you suggesting that commits should be rejected if they do not update documentation as well ? What is better, having undocumented code or having no code ? Jan. _______________________________________________ Serdev mailing list serdev(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serdev

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Re: [Serusers] Is there an AVP to alter "expires=" in REGISTERmessages?
by Samuel Osorio Calvo 21 Jun '05

21 Jun '05

I was taking a more detailled look at the code and it seems that the Contact header will not be added into the reply if you use save_noreply+sl_send_reply because the Contact buffer is added inside the registrar module....but I recall that it worked. I suggest you to give it a try and see the answer with ethereal|ngrep|* maybe it is taken after when constructing the answer from the sipmsg. Anyhow it would be easy to add a save_extended_expires command that makes what you need but, as I said, people can start using these SIP patches to build even more non-compliant proxies with SER.... Samuel. Unclassified. >>> "Greger V. Teigre" <greger(a)teigre.com> 06/21/05 07:19AM >>> You're right of course :-) Samuel's approach is more correct, but I'm not sure you will be able to generate the correct reply using save_noreply and sl_reply (with Contact) ?! Does save_noreply generate the reply? g-) ---- Original Message ---- From: Java Rockx To: Greger V. Teigre Cc: serusers Sent: Monday, June 20, 2005 02:21 PM Subject: Re: [Serusers] Is there an AVP to alter "expires=" in REGISTERmessages? > Greger, > > That would be possible, but the way SER generates the 200OK reply, it > would contain the altered "expires=" value. By doing so, the ATA > would take that new value and use it, so if the ATA sent the REGISTER > with an expires=300 and SER replied with a 200OK and a contact header > containing expires=360, the ATA would not send another REGISTER for > 360 seconds. > > I was fooling around with sip_msg.c and reply.c were I padded the > expires with 60 seconds before saving to usrloc and then removing the > pad before sending the reply and it works, but I would just feel > better by not altering the source code. > > I guess I'd just feel more comfortable if SER had some sort of AVP or > modparam for the registrar module that did this automagically. > > Regards, > Paul > > > On 6/20/05, Greger V. Teigre <greger(a)teigre.com> wrote: > It was supposed to be short, but maybe too short. I meant doing the > subst on the REGISTER message before save. > g-) > ---- Original Message ---- > From: Greger V. Teigre > To: Java Rockx ; serusers > Sent: Monday, June 20, 2005 09:01 AM > Subject: Re: [Serusers] Is there an AVP to alter "expires=" in > REGISTERmessages? > >> Hi Paul, >> Would it be possible to "do it upside down" ? ;-) I.e. do a >> subst("/^Expires:(.*)[0-9]$/ Expires: yourvalue+60/gi"); >> g-) >> >> ---- Original Message ---- >> From: Java Rockx >> To: serusers >> Sent: Sunday, June 19, 2005 04:28 PM >> Subject: [Serusers] Is there an AVP to alter "expires=" in REGISTER >> messages? >> >>> Hi All. >>> >>> We've got some ATAs that don't quite re-REGISTER in a timely fashion >>> and therefore SER expires the usrloc record a few seconds before the >>> ATA sends another REGISTER message. >>> >>> Is there an AVP or something to add, for example, 60 seconds to the >>> "expires=" value in the REGISTER message prior to calling save()? By >>> doing so, these "broken" ATAs would re-REGISTER before SER removes >>> the usrloc record from cache. >>> >>> Regards, >>> Paul >>> >>> >>> >>> >>> _______________________________________________ >>> Serusers mailing list >>> serusers(a)lists.iptel.org >>> http://lists.iptel.org/mailman/listinfo/serusers >> >> >> >> _______________________________________________ >> Serusers mailing list >> serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers

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[Serusers] openser/ser - avoiding forks
by Andrei Pelinescu-Onciul 21 Jun '05

21 Jun '05

I think that a fork is bad for everybody. Whatever way you put it, it will still involve extra overhead at least for each code merge. So this mail is an attempt to come to a compromise that would be acceptable for everybody. My first proposal would be to have another branch on ser cvs (berlios). Something like openser_0.x.y (where the version number should reflect somehow that it's newer than 0.9 but not unstable). Bogdan and Daniel would mantain this branch and they would be able to release from it as often as they want. It would be some kind of stable on steroids :-) This would have the main advantage that code merges/diffs/fixes propagation would be much easier and much faster. Administrating all the stuff will be also easier (like common bug tracking system a.s.o). It would be also easier for users, it will be much more easier to keep track of the changes and to choose an appropiate version. There must be however some rules. Something like: backporting/forwadporting from unstable/stable should be ok, the same for minor patches (e.g. xlog colors), but not major code changes, not present in unstable and without the code maintainer approval (to maintain future compatibility). Adding new stuff (new modules) should also be ok, but it would be preferable that these module come from either unstable or experimental. The mailing list for development should remain serdev. For users it could be a different one (if the changes or the mail traffic gets too big). As far as integrating new contributions faster in unstable, I agree that this is a problem, but I think we can sort it out. The experimental repository is a beginning and if there are better ideeas on how to speed things up while maintaing sanity, please speak. Maybe some kind of reviewed patches (e.g: someone sends a patch that adds a new feature in core, Bogdan reviews it decides it's ok and blesses it => I will integrate much faster something blessed by other ser developer). If you don't agree with my proposal, please say what you don't like and what you would like changed so that it would be acceptable for you. I'm sure that we can reach a compromise and hopefully improve ser development pace in the process. Andrei

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[Serusers] How to config the ser?where can I find the instruction?
by Alan bus 21 Jun '05

21 Jun '05

I am new to using SER.I installed ser-0.8.14_src.tar.gz on Linux-redhat-2.4.18-14 with mysql-standard-5.0.6-beta-linux-i686.tar.gz.Mysql can run successfully.The following is what I have done:(the main directory:/usr/local/src/ser-0.8.14) step 1:make all step 2:ser_mysql.sh create The following are the commands and the error messages: [root@redhat scripts]# ser_mysql.sh create MySql password for root: Domain (realm) for the default user 'admin': blrc.com ./ser_mysql.sh: line 1: gen_ha1: command not found HA1 calculation failed what's the matter?What should I do the next?Could you help me? Here are the ser.cfg: # # $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=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" # ------------------ 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" # 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" # ----------------- 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) # ------------------------- 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; }; # 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()) { 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==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; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; } thx! _________________________________________________________________ 享用世界上最大的电子邮件系统― MSN Hotmail。 http://www.hotmail.com

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[Serusers] avp_pushto() + append_branch() not working
by Java Rockx 21 Jun '05

21 Jun '05

Hi All. I just need a sanity check regarding avp_pushto() for serial forking. Someone please correct me if I'm wrong here. When processing an INVITE message which results in a 486 Busy status my failure_route[] gets invoked. In the failure route I use avp_pushto() to load the next leg of the call in to the $ruri and then call append_branch() and lookup(location) prior to t_relay()ing the message. Using ngrep I do see the R-URI being changed to the value used in the avp_pushto() call, but the IP address that the INVITE is then sent to is the same as the IP address of the SIP router, which I cannot understand because I do call lookup("location") before t_relay() In otherwords, if my sip router, sip.mycompany.com<http://sip.mycompany.com>has an IP of 71.72.73.74 <http://71.72.73.74> and I use avp_pushto() to set the R-URI to 1001_at_sip.mycompany.com, lookup(location) does return TRUE, but ngrep still shows the INVITE leaving the SIP router and going to 71.72.73.74<http://71.72.73.74>(ie, itself) rather than the IP address in usrloc. Is there anything obvious that I'm not doing that would cause lookup(location) to return TRUE, but not change the R-URI accordingly? Regards, Paul

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[Serusers] loading db during initialization
by Jack Wei 21 Jun '05

21 Jun '05

is there anyway to stop ser from loading users from db during startup? Jack Wei ____________________________________________________ Yahoo! Sports Rekindle the Rivalries. Sign up for Fantasy Football http://football.fantasysports.yahoo.com

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[Serusers] voice sniffing
by Iqbal 21 Jun '05

21 Jun '05

Hi I have managed to get voice calls sniffed, by using a external sniffer. Does anyone know how I could bind a username/caller/caller tag to each wav file. I am not sure this is possible because the voip stream doesnt really have this info, but since my users are making calls by sending invites etc to ser, could I use ser to get the ID's and then call the sniffer to tag the names....or has anyone come across a really good voip sniffer Iqbal

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[Serusers] Is there an AVP to alter "expires=" in REGISTER messages?
by Java Rockx 21 Jun '05

21 Jun '05

Hi All. We've got some ATAs that don't quite re-REGISTER in a timely fashion and therefore SER expires the usrloc record a few seconds before the ATA sends another REGISTER message. Is there an AVP or something to add, for example, 60 seconds to the "expires=" value in the REGISTER message prior to calling save()? By doing so, these "broken" ATAs would re-REGISTER before SER removes the usrloc record from cache. Regards, Paul

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[Serusers] UAC module: multiple PSTN gateways
by Cameron Beattie 21 Jun '05

21 Jun '05

Is it possible to specify multiple authentication credentials within the UAC module? What I want to do is access multiple PSTN gateways that each require different credentials. The basic purpose of this is to allow least cost routing. Example Provider 1 offers cheap calling to +44 (UK) Provider 2 offers cheap calling to +1 (US) When user dials a 44 destination ser should relay to Provider 1 gateway, which will require authentication using Credential 1. When user dials a 1 destination ser should relay to Provider 2 gateway, which will require authentication using Credential 2. Any advice would be greatfully received. Regards Cameron

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[Serusers] Using Sipsak to register
by Cameron Beattie 21 Jun '05

21 Jun '05

I am struggling to understand the syntax of sipsak. I want to use it to register against a proxy. I enter the command: sipsak -U -C sip:1234@192.168.0.5 -a password -s sip:1234@192.168.0.5 I get the message error: REGISTER Contact uri doesn't not contain sip:, *, or is not empty" If I enter the command sipsak -U -s sip:1234@192.168.0.5 -a password then the register message gets sent but with a contact of sip:1234@127.0.0.1 Could anyone help me with the correct syntax to use? Regards Cameron

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[Serusers] voice messages in /tmp
by Llanos Serna García-Conde 20 Jun '05

20 Jun '05

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[Serusers] SER + Prepaid = Old discussion
by Ricardo Poppi 20 Jun '05

20 Jun '05

Hi all. This list had a long discussion for the last two months about how to use SER with prepaid applications, since it does not participate into the media path. I´ll try to put things together in a easy way - at least for me... - for trying to solve some doubts about how this feature can work with SIP without creating new problems. And, of course, I´m assuming that we WILL NOT use b2bua, but try to make this work in a proxied environment. As mentioned before, one way to do so, is using the session timer (RFC-4028) definitions. The main reason those definitions were created, is for using when a BYE message don´t reach one of the UAs involved into a SIP dialog, that without those definitions would be virtualy connected "forever" since it did not receive any SIP signaling to disconect. With session timer working, even if a BYE message never reaches the other point, the maximun time that the "not-reached" UA would stay connected is the time setted into the session-timer parameter. Ok. But let´s use it to work with the prepaid environment. We implement a logic into our sip proxy (SER), that record-route all signaling messages between all UAs of our SIP network. Then we put our proxy to analize all re-invite messages that goes into our SIP dialogs and, if the customer credit (in seconds) is below that the time into the session-timer parameter, it rewrites the "seconds-credit" into the message session-timer position, decrements the last session time into the customer seconds credits table, and counts on the UAs to disconecting the call, since it - the proxy - will drop/block the next re-invite(s) to this especific dialog. In this aproach, we need to check the first invite as well, because if the customer credit is below that the value of the first/default session-timer, the SIP message needs to be rewriten too. Problems on this aproach: - If a re-invite from the caling user - the one that will be billed - never reachs the proxy, the UA will disconect the call sending a BYE, and, if the BYE never reachs the proxy either, the seconds of the last session won´t be billed, correct? - Thinking about security issues into a environment that just one UA support session-timers, It would be very easy for a malicious UA - the one that supports session-timers - when it runs out of credit, to send its last re-invite - the never answered one - withouting disconecting the media path. In this case, the proxy will "think" that the call is off, but it would be not true. Does anyone work with a true proxied prepaid environment using session timers? There is any aproach, other than using RFC-4028 - session timers, for doing this? Any clue will be welcome. Thanks in advance for the list. Regards, Ricardo Poppi

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Re: [Serusers] SER + Prepaid = Old discussion
by Samuel Osorio Calvo 20 Jun '05

20 Jun '05

I don't think that session timers have the accuracy required for a billing system. If you take a look at the specification, minimum values are 90 seconds and recommended values are 30 minutes. If you reduce these values to bill every second a user is accessing the service, you could end up in interoperability problems. Moreover, any traversed proxy may use Min-SE header to increase the minimum value to few minutes, or hours, and then your user is gonna feel really happy having minutes for free... I really think the only reliable way is to use a b2bua because it is the only entity aware of the media path and able of dropping the call "instantaneously". I don't even think a statefull proxy will work since it is not in the media path and it is not supposed to initiate BYEs... I would suggest to choose between b2bua or flat fares. I think the latter suits more the IP world: almost everywhere you pay X per month for internet access, doesn't it apply better to VoIP?? interconnecting with PSTN is another topic and I think something like skype would suit better to IP than the classic X minutes->Y money. Samuel. Unclassified. >>> Ricardo Poppi <rpoppi77(a)giro.com.br> 06/20/05 06:11PM >>> Hi all. This list had a long discussion for the last two months about how to use SER with prepaid applications, since it does not participate into the media path. I ll try to put things together in a easy way - at least for me... - for trying to solve some doubts about how this feature can work with SIP without creating new problems. And, of course, I m assuming that we WILL NOT use b2bua, but try to make this work in a proxied environment. As mentioned before, one way to do so, is using the session timer (RFC-4028) definitions. The main reason those definitions were created, is for using when a BYE message don t reach one of the UAs involved into a SIP dialog, that without those definitions would be virtualy connected "forever" since it did not receive any SIP signaling to disconect. With session timer working, even if a BYE message never reaches the other point, the maximun time that the "not-reached" UA would stay connected is the time setted into the session-timer parameter. Ok. But let s use it to work with the prepaid environment. We implement a logic into our sip proxy (SER), that record-route all signaling messages between all UAs of our SIP network. Then we put our proxy to analize all re-invite messages that goes into our SIP dialogs and, if the customer credit (in seconds) is below that the time into the session-timer parameter, it rewrites the "seconds-credit" into the message session-timer position, decrements the last session time into the customer seconds credits table, and counts on the UAs to disconecting the call, since it - the proxy - will drop/block the next re-invite(s) to this especific dialog. In this aproach, we need to check the first invite as well, because if the customer credit is below that the value of the first/default session-timer, the SIP message needs to be rewriten too. Problems on this aproach: - If a re-invite from the caling user - the one that will be billed - never reachs the proxy, the UA will disconect the call sending a BYE, and, if the BYE never reachs the proxy either, the seconds of the last session won t be billed, correct? - Thinking about security issues into a environment that just one UA support session-timers, It would be very easy for a malicious UA - the one that supports session-timers - when it runs out of credit, to send its last re-invite - the never answered one - withouting disconecting the media path. In this case, the proxy will "think" that the call is off, but it would be not true. Does anyone work with a true proxied prepaid environment using session timers? There is any aproach, other than using RFC-4028 - session timers, for doing this? Any clue will be welcome. Thanks in advance for the list. Regards, Ricardo Poppi _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[Users] RE: [Serusers] OpenSER release
by Giudice, Salvatore 20 Jun '05

20 Jun '05

I am not an advocate for either ser or openser, but I would like to comment. Is openser going to be equipped with a forum/ticket system where people can document bugs, feature requests, etc (non-configuration issues)? This is just my observation and you may not agree, but I believe this project could be much better maintained if it used a more structured ticketing style system to manage development issues instead of the current mailing lists. In my experience, mailing lists like this foster a terrible user experience where many development issues can go on without response. Ideally, if there was a mailing list to address user issues and ticketing system like the one Digium uses to manage Asterisk, I think everyone would benefit by being better informed and ser would ultimately be a better product for it. How many people out there feel that their issues have fallen through the cracks in the past couple years? -----Original Message----- From: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] Sent: Wednesday, June 15, 2005 4:28 AM To: Andrei Pelinescu-Onciul Cc: SER developer mailing list; serusers; users(a)openser.org; devel(a)openser.org Subject: Re: [Serusers] OpenSER release On 06/14/05 23:21, Andrei Pelinescu-Onciul wrote: >On Jun 14, 2005 at 22:48, Daniel-Constantin Mierla <daniel(a)voice-system.ro> wrote: > >[...] > > >>It is your opinion, but I repeat myself, that the SER code maintained by >>us will go further -- I don't think that someone can claim that we >>didn't do the job for our code (the only discrepancy is some last-minute >>adds in xlog (to print avps) - will be committed on unstable very soon >>with the new color patch). The cvs was created just to ease the >>maintainance. The patches would be a nightmare. >> >> > >Maybe I've misunderstood you: is this only a parallel "stabilized" >version + some features or is it a full fork (do you intend to fork >unstable also)? > > It is fork for the code that we changed (acc module, usrloc module ...), in the future may be other that they do not find the path in SER. We will maintain and upgrade our part of code from SER continuously. >I have no problem with another stable version, what worries me is >fragmenting the development for unstable (which is the place where major >changes are made). > > I see no fragmenting there -- the situation is the same for SER as it was before. For example, there is no fragment for acc module, it will be maintained by who did it till now, adding what he considers necessary there. But we came to meet a lot of requests of why the acc patch is not included in the CVS (it was fully backward compatible and had new features requested by many SER users) and we want to promote _more open_ approach to contributions to all parts of code. The acc patch was sent on November 1, 2004. No real response (neither negative, nor positive) from maintainer to the submission since then ... are you aware of a good reason?!?! ... should we wait just about (or more) half an year for each contribution?!? I will not do that anymore!!! Daniel > >Andrei > > > _______________________________________________ Serusers mailing list Serusers(a)iptel.org http://mail.iptel.org/mailman/listinfo/serusers

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RE: [Serusers] double INVITE in a single call
by Giudice, Salvatore 20 Jun '05

20 Jun '05

Multiple aliases will fork the call in parallel. -----Original Message----- From: Jesus Rodriguez [mailto:jesusr@voztele.com] Sent: Monday, June 20, 2005 10:05 AM To: Nerea Munguira Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] double INVITE in a single call On Mon, 20 Jun 2005, Nerea Munguira wrote: Hola Nerea, > I have several calls that have two branches, one in one way and another in > the other way, I mean, for the same call ID, user 1 calls user 2, and also, > user 2 calls user 1. > > I wonder if this is normal, or if I have a something wrong. This only occur > sometimes, so it isn?t a routing problem. > > I have this problem in different scenarios, so it has to do with ser > configuration, not with the user routing scheme. Also, I have this problem > with calls between sip users or calls between sip user and PSTN. > > For the same call, I?ve got two logs: > > User 1 -> User 2 INVITE fromtag: id1 totag: id2 CALLID1 > User 2 -> User 1 INVITE fromtag: id2 totag: id1 CALLID1 > > Where id1 and id2 are the same identifier. What you see is possibly a re-INVITE because a codec or other SDP parameter negotiation. Saludos JesusR. ------------------------------- Jesus Rodriguez VozTelecom Sistemas, S.L. jesusr(a)voztele.com http://www.voztele.com Tel. 902360305 ------------------------------- _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[Users] sems module
by harry gaillac 20 Jun '05

20 Jun '05

Hello , I've heard that nathepler could provide MOH with rttproxy. I should try to use annoucement module. I want a dedicated directory to put .wav files My problem would be to write a boucle "while <condition> do <action> . In other words play file(1), file(2), ... file(3), file(1) ... I need help to start this boucle ! Is it a good idea ? Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

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Re: [Serusers] How-to write a custom module
by Samuel Osorio Calvo 20 Jun '05

20 Jun '05

In fact there's the print module which is there just to give an example of a starting point for a new module......or is there anyone using print?? ;) Samuel. Unclassified. >>> Andreas Granig <andreas.granig(a)inode.info> 06/20/05 03:17PM >>> Steve Blair wrote: > I'd like to write a new module for SER. I see a number of other people > have asked this question but I don't see a how-to guide mentioned in > any of their messages. Does such a guide exist? If so can someone > provide a pointer to it? IMHO the "group" module is a good starting point because of it's simplicity.. Andy _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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Re: [Serusers] Is there an AVP to alter "expires=" inREGISTERmessages?
by Samuel Osorio Calvo 20 Jun '05

20 Jun '05

I think that the UA should send the re-registration message before. I would try to push UAs developers to upgrade the firmware instead of adding more stuff to SER just to correct end points which are not compliant. Said that, since we are not in a perfect world, you might try something like (just until the ATA firmware is corrected, of course ;) ): save_noreply(); //modify Expires in reply sl_send_reply; I don't know exactly how to modify the Expires.....maybe something like subst textopts. Or remove the header and add a header to the reply (there was a function to do that but I don't remember the exact syntax or how it exactly works). I can not help you more than that....and I don't know for sure if it would work, just give it a try. Samuel Unclassified. >>> Java Rockx <javarockx(a)gmail.com> 06/20/05 02:21PM >>> Greger, That would be possible, but the way SER generates the 200OK reply, it would contain the altered "expires=" value. By doing so, the ATA would take that new value and use it, so if the ATA sent the REGISTER with an expires=300 and SER replied with a 200OK and a contact header containing expires=360, the ATA would not send another REGISTER for 360 seconds. I was fooling around with sip_msg.c and reply.c were I padded the expires with 60 seconds before saving to usrloc and then removing the pad before sending the reply and it works, but I would just feel better by not altering the source code. I guess I'd just feel more comfortable if SER had some sort of AVP or modparam for the registrar module that did this automagically. Regards, Paul On 6/20/05, Greger V. Teigre <greger(a)teigre.com> wrote: > > It was supposed to be short, but maybe too short. I meant doing the subst > on the REGISTER message before save. > g-) > ---- Original Message ---- > From: Greger V. Teigre > To: Java Rockx ; serusers > Sent: Monday, June 20, 2005 09:01 AM > Subject: Re: [Serusers] Is there an AVP to alter "expires=" in > REGISTERmessages? > > > Hi Paul, > > Would it be possible to "do it upside down" ? ;-) I.e. do a > > subst("/^Expires:(.*)[0-9]$/ Expires: yourvalue+60/gi"); > > g-) > > > > ---- Original Message ---- > > From: Java Rockx > > To: serusers > > Sent: Sunday, June 19, 2005 04:28 PM > > Subject: [Serusers] Is there an AVP to alter "expires=" in REGISTER > > messages? > > > >> Hi All. > >> > >> We've got some ATAs that don't quite re-REGISTER in a timely fashion > >> and therefore SER expires the usrloc record a few seconds before the > >> ATA sends another REGISTER message. > >> > >> Is there an AVP or something to add, for example, 60 seconds to the > >> "expires=" value in the REGISTER message prior to calling save()? By > >> doing so, these "broken" ATAs would re-REGISTER before SER removes > >> the usrloc record from cache. > >> > >> Regards, > >> Paul > >> > >> > >> > >> > >> _______________________________________________ > >> Serusers mailing list > >> serusers(a)lists.iptel.org > >> http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers >

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[Serusers] double INVITE in a single call
by Nerea Munguira 20 Jun '05

20 Jun '05

I have several calls that have two branches, one in one way and another in the other way, I mean, for the same call ID, user 1 calls user 2, and also, user 2 calls user 1. I wonder if this is normal, or if I have a something wrong. This only occur sometimes, so it isn´t a routing problem. I have this problem in different scenarios, so it has to do with ser configuration, not with the user routing scheme. Also, I have this problem with calls between sip users or calls between sip user and PSTN. For the same call, I´ve got two logs: User 1 -> User 2 INVITE fromtag: id1 totag: id2 CALLID1 User 2 -> User 1 INVITE fromtag: id2 totag: id1 CALLID1 Where id1 and id2 are the same identifier. Can anybody help me? thanks a lot!!

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[Serusers] How-to write a custom module
by Steve Blair 20 Jun '05

20 Jun '05

Hello: I'd like to write a new module for SER. I see a number of other people have asked this question but I don't see a how-to guide mentioned in any of their messages. Does such a guide exist? If so can someone provide a pointer to it? Thanks,Steve -- ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 voice: 215-573-8396 215-746-8001 fax: 215-898-9348 sip:blairs@upenn.edu

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[Serusers] How-to write a custom module
by Steve Blair 20 Jun '05

20 Jun '05

Hello: I'd like to write a new module for SER. I see a number of other people have asked this question but I don't see a how-to guide mentioned in any of their messages. Does such a guide exist? If so can someone provide a pointer to it? Thanks,Steve -- ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 voice: 215-573-8396 215-746-8001 fax: 215-898-9348 sip:blairs@upenn.edu

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Re: Re: [Serusers] OpenSER release
by Max Malloy 20 Jun '05

20 Jun '05

2 1

[Serusers] help on mediaproxy with ser no sound
by peiyin 20 Jun '05

20 Jun '05

Hi all, I'm having trouble configuring mediaproxy to work on my SER server. One phone is behind NAT and the other is using a public ip. Both are registered in SER. Calls from phone A (nat) to phone B(public) are successful whereby phone B will rings.However, no sound comes out from both side, also, when i hang up phone B, phone A doesn't produce any engaged tone. Calls from phone B to phone A also successful but i don't get any audio on either end. I'm storing all information in database. My config is as below (taken from example and modified): route{ if (!mf_process_maxfwd_header("10")) { if (method!="ACK") { sl_send_reply("483", "Too many hops"); }; break; }; if (msg:len >= max_len) { if (method!="ACK") { sl_send_reply("513", "Message too big"); }; break; }; if(method=="INVITE" || method=="BYE") { setflag(1); #accounting } if (method=="REGISTER") { if (is_from_local()) { # Mark as NAT'ed if (client_nat_test("3")) { setflag(6); force_rport(); fix_contact(); }; if (!www_authorize("my_ser_ip", "subscriber")) { www_challenge("my_ser_ip", "0"); break; } else if (!check_to()) { sl_send_reply("403", "Username!=To not allowed"); break; }; if (!save("location")) { sl_reply_error(); }; } else { sl_send_reply("403", "This domain is not served here"); }; break; }; if (method=="INVITE") { if (!(is_from_local() || is_uri_host_local())) { sl_send_reply("403", "Relaying is forbidden"); break; }; t_on_failure("1"); } else if (method == "BYE" || method == "CANCEL") { end_media_session(); }; if (loose_route()) { if (method=="INVITE" || method=="ACK") { use_media_proxy(); }; # end media session for BYE and CANCEL is done above # before entering the loose route. no need to call it here t_relay(); break; }; # Force subsequent messages to pass trough this proxy if (method == "INVITE") { record_route(); }; if (client_nat_test("3") && !search("^Record-Route:")) { # Mark as NAT'ed force_rport(); fix_contact(); }; if (method=="INVITE") { t_on_reply("1"); }; if (is_uri_host_local()) { if (!lookup("location")) { sl_send_reply("404", "User not found"); break; }; }; if (method=="INVITE" || method=="ACK") { use_media_proxy(); }; if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } failure_route[1] { end_media_session(); } onreply_route[1] { if (status=~"(183)|(2[0-9][0-9])") { if (client_nat_test("1")) { fix_contact(); }; use_media_proxy(); }; }

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[Serusers] (no subject)
by Chia Huey Lim 20 Jun '05

20 Jun '05

Hi. I am trying to configure the SER routing script for the NAT environment. Is there any function to replace IP in the Connection Information (c=) to its Public IP? I have tried using fix_nated_sdp("3") , but it is still giving the internal IP. Thanks a lot!

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[Serusers] Is there a more detailed radius howto?
by Ryan Pagquil 20 Jun '05

20 Jun '05

Hi, I'm currently working on making SER to authenticate with freeradius, is there any other howto that is detailed enough to guide a newbie? Thanks a lot, Ryan

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[Serusers] Asterisk+mediaproxy+loose_route
by Ah khng 20 Jun '05

20 Jun '05

Hi, I'm using the mediaproxy configuration example from onsip.org and asterisk as PSTN gateway. However, when loose routing, stream was pass through the mediaproxy which i want asterisk to handles NAT for the RTP packet itself. I have made an inelegant change to check for uri and source ip of the asterisk, but how to implement when LCR is being used? If i'm wrong please correct me. # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { #xlog("L_INFO", "loose_route(): Looking up %rm URI %ru from %is\n"); #lookup("location"); xlog("L_INFO", "loose_route(): t_relay() %rm to URI %ru\n"); if (has_totag() && (method=="INVITE" || method=="ACK")) { if (client_nat_test("3") || search("^Route:.*;nat=yes")) { setflag(6); if (!(src_ip==A.B.C.D || uri=~"^sip:.*@A.B.C.D")) { #A.B.C.D => asterisk ip use_media_proxy(); xlog("L_INFO", "loose_route(): use mediaproxy for %rm to URI %ru\n"); }; }; }; route(1); break; }; cheer seong

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[Serusers] Daniel's Commitment to Community?
by Peter L P 19 Jun '05

19 Jun '05

I am joining this discussion for the first time, so please appologize me if I provide you with novice's questions. I find the idea of SER great. As a matter of fact, I think that all software should be free and I am very thankful for people who advocate it, like Daniel orginally seemed to do. We, as a community, should fully support free software! Now what I am beginning to be concerned about is whether Daniel's team has actually the same goals! But he keeps the VNT and VPS products insufficiently transparent; I cannot download it from the Internet! How credible is this then??? The avpops patch has not commited immediatelly either! And scaringly, as Bogdan's email revealed, the commercial company voice-system has branched some other source code which is using some other forking code -- aren't they sucking serusers community to create a new branch under their control? I am concerned, because just giving away three or five modules is not much software for free and in turn for that we are being offered untransparent VNT and VPS. What assurance can we obtained we are not beeing fooled and this is not actually a red harring or even a mousetrap??? PS --- What I would also like to request is better documentation!!! __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[Serusers] getting contact info from db
by Jack Wei 19 Jun '05

19 Jun '05

how do i look up a user from the database, get the contact info from the location table, rewrite the uri with the contact info? Jack Wei __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[Serusers] rtpproxy
by Glenn Dalgliesh 19 Jun '05

19 Jun '05

I must be totally out of it. I am trying to download rtpproxy for the CVS head and I am not having at luck. Has something changed with regard to it's location? cvs -d:pserver:anonymous@cvs.ser.berlios.de:/cvsroot/ser co rtpproxy ? rtpproxy/rtpproxy cvs server: existing repository /cvsroot/ser/PortaSIP/rtpproxy does not match /cvsroot/ser/rtpproxy cvs server: ignoring module rtpproxy [root@ser software]#

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[Serusers] open letter: ser, tls, openser
by Peter Griffiths 19 Jun '05

19 Jun '05

folks -- i usually do not participate in mailing list discussions but it seems that my tls contribution became hot topic in recent ser vs. openser discussions so i think i will make an exception here. i think that openser is a bad idea. please stop using the tls code as a hostage in this dispute, i did not write it to fork the project. iptel guys made no attempt to hide the code when i approached them although they probably have their own commercial implementation. for me ser is an open and free project because: 1. i could download ser freely and use it 2. i needed tls, i was not able to buy it from iptel so i tried to implement it 3. i sent it to the mailing lists, anyone can find it there and use it if they want 4. someone else picked up the code, improved it and committed to cvs (thank you for this, by the way) what can be more open than this ? maybe sources stored in a wiki where anyone can change anything. openser claims that it will be more open, but: 1. over 20 people have access to ser cvs on berlios, but only 4 have access to openser cvs on sourceforge, not even other developers. 2. you took existing code from ser cvs, added your own improvements, but you did not give other developers who work on ser any chance to say whether such changes should be included or not, although they are still listed in AUTHORS file, in module documentation and elsewhere. from the user point of view they will be responsible for bugs introduced by you and have no chance to influence it. 3. i did not see any discussion about this move, it is a bit strange that the people on the lists were not allowed to participate in the decision, especially if you claim that it is for the interest of the community. how do you know people want the project to be split when you did not ask ? 4. the complete cvs history is missing in openser repository, effectively hiding who contributed what. I also noticed that user miconda created project named ser on sourceforge, is this a preparation for the next step - full control ? did other ser developers know about this ? i think having two projects with the same name on two sites is confusing. from my perspective the new fork is only an attempt to get more power and control over the sources. it is about who will control what, not about freedom. try to resolve it without forking the project. if ser is bad then openser is not any better: - iptel.org advertises iptelorg.com on its webpage - there is an ad of voice-system on the main page of openser.org, there are voice-system copyright statements and readme states that the project is maintained by voice-systems, proving beyond reasonable doubts that openser promotes voice-systems just like ser promotes iptel.org - there were complaints about maintainers deciding what comes into modules - so in openser only selected people have access to cvs and they will probably form a committee and decide. - there were calls for more discussions, but the fork was not discussed at all there probably is a good reason why tls was put in experimental directory first and is not in the main tree yet, i do not see this as a problem. in fact i would appreciate if someone who knows in detail how tcp code in ser works could review the code before it is committed in the main tree. there are many places in the code i don't understand. i did only limited testing because tls support in end devices is virtually non-existent. use it at your own risk. i don't know about others, but i am on alert when people talk about "the needs of the community" and "take over when someone is very busy or unwilling to cooperate", i don't remember seeing users asking for anything like this. -- peter __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[Serusers] Failure reason.
by Dmitry Isakbayev 19 Jun '05

19 Jun '05

Hello, I am using serial forking, and I need to keep track of failed calls to t_relay. While in the 'failure_route[x]' block, can the script query why the previous call to t_relay failed (user not registered, connection failure, etc)? Thank You, Dmitry

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Re: [Serusers] serweb PHP errors
by zhaomin 19 Jun '05

19 Jun '05

when I register a new account .but show me a error blow: "Sorry, there was an error when sending mail. Please try again later." Why? where can I configure mail server ? Zhaomin ----- Original Message ----- From: "Karel Kozlik" <karel(a)iptel.org> To: "PJR" <pjr_lists(a)mail.ntua.gr> Cc: <serusers(a)iptel.org> Sent: Wednesday, June 15, 2005 10:29 PM Subject: Re: [Serusers] serweb PHP errors > Hi, > try set: > > error_reporting = E_ALL & ~E_NOTICE > > in php.ini. > Regards Karel > > PJR napsal(a): > > On Fedora core 3 with all needed packages installed and config.php > > configured properly I can't login to serweb user_intrface or admin > > > > When I first load the user or admin index.php I get many errors in > > apache error logs like the following: > > > > [client 192.168.25.2] PHP Notice: Undefined index: extrahtml in /var/www/html/phplib/oohforms.inc on line 72, referer: http://192.168.25.3/iptel/user/ > > [client 192.168.25.2] PHP Notice: Undefined property: isfile in /var/www/html/phplib/oohforms.inc on line 250, referer: http://192.168.25.3/iptel/user/ > > [client 192.168.25.2] PHP Notice: Undefined property: hidden in /var/www/html/phplib/oohforms.inc on line 253, referer: http://192.168.25.3/iptel/user/ > > .....snip > > [client 192.168.25.2] PHP Notice: Undefined index: frozen in /var/www/html/phplib/oohforms.inc on line 284, referer: http://192.168.25.3/iptel/user/ > > > > When I enter username/passwd and submit I get more errors in logs > > (Notice, index frozen etc.) and also A Bad Username or Password message > > and I return to login screen. There is nothing wrong with passwords, and > > I can connect to ser db with mysql client just fine. > > > > I use serweb_2004-07-27 and ser-0.8.14. > > I have the vars in php.ini according to the instractions. > > > > Please point me to any possible solution. > > > > Thanks. > > > > _______________________________________________ > > Serusers mailing list > > Serusers(a)iptel.org > > http://mail.iptel.org/mailman/listinfo/serusers > > _______________________________________________ > Serusers mailing list > Serusers(a)iptel.org > http://mail.iptel.org/mailman/listinfo/serusers

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[Serusers] Analog Gateway & SER
by Ozan Blotter 19 Jun '05

19 Jun '05

is there anyone have good experience with analog fxo gateways, because while more basic sip proxies can work with them i still do not understand why SER does not work with... because i'm neither an operator nor have a cisco gateway... please contact me, thanks ozan -- >>> please do not use >>> l.s.d(a)dnaofdesign.com >>> above account is stand by >>> use same account as >>> cosmocid(a)ispro.net.tr

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[Serusers] ser monitoring by serweb php script
by Rosario Pingaro 18 Jun '05

18 Jun '05

I'd like to try the monitoring script by serweb but I am unable to get the tables updated. I followed the installation readme file updating my cronjob conf file, I have a working configuration of serweb so the connection to the db is fine. But I don't get any updated. Searching the var/log/messages I see the cronjob executed, I don't see any error. So I don't know where to look to get more info on my problem. Is there some one of you that uses this script? Thanks Rosario

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Re: [Serusers] rtpproxy and serctl with problem
by Klaus Darilion 17 Jun '05

17 Jun '05

It looks like you are trying to start ser, although it is already running. regards, klaus PS: Please Cc: to the list. Bruno Oliveira wrote: > hi > > I try, but no resolv. Print this in term: > uiara:~# netstat -anp|grep 5060 > > "tcp 0 0 200.236.162.3:5060 0.0.0.0:* > OUÇA 2816/ser > tcp 0 0 192.168.1.2:5060 0.0.0.0:* > OUÇA 2816/ser > udp 0 0 200.236.162.3:5060 0.0.0.0:* > 2816/ser > udp 0 0 192.168.1.2:5060 0.0.0.0:* > 2816/ser" > > it continues with exactly error. > > thanks for your time > > On 6/17/05, Klaus Darilion <klaus.mailinglists(a)pernau.at> wrote: > >>Bruno Oliveira wrote: >> >>>hi >>> >>>i have rtpproxy runing and serctl. where it is file config of rtpproxy >>>? and when I type this command "serctl start" show me in >>>/var/log/syslog this message: >>> >>>"Jun 16 14:45:33 uiara /usr/local/sbin/ser[3237]: Maxfwd module- initializing >>>Jun 16 14:45:33 uiara /usr/local/sbin/ser[3237]: INFO: udp_init: >>>SO_RCVBUF is initially 111616 >>>Jun 16 14:45:33 uiara /usr/local/sbin/ser[3237]: INFO: udp_init: >>>SO_RCVBUF is finally 223232 >>>Jun 16 14:45:33 uiara /usr/local/sbin/ser[3237]: ERROR: tcp_init: >>>bind(6, 0x813753c, 16) on 192.168.1.2: Address already in use" >> >>Looks like another process already has bind to TCP port 5060. Maybe ser >>is already running? Or maybe a SIP client? >> >>Try >>#> netstat -anp|grep 5060 >> >>regards, >>klaus >> > > >

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[Serusers] problems with ACKs and BYEs
by Jorge Crichigno 17 Jun '05

17 Jun '05

Hi all, I have a problem with in the following scenario: Lucent GW SER IPPhone IP: 1.2.3.4 IP:5.6.7.8 IP:8.9.5.5 INVITE--> <-- 100 INVITE--> <--100 <--180 <--180 <--200 <--200 ACK-->(Request-Uri = sip:5.6.7.8:5060) <-- PROBLEM HERE The problem arises only with the Lucent Gateway, because it sends the ACK with the Request-Uri as showed above (Request-Uri has the IP and port of the SER). I also found the same problem with BYEs. I attach the ser.cfg at this mail. I think this is a bug of Lucent, because I expected that Request-Uri of the ACKs should equal to the contact uri of the 200 message sent by the ipphone. The same for BYEs. Could some one clarify me that? Thanks in advance !!!

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RE: [Users] extcmd removed?
by Matt Schulte 17 Jun '05

17 Jun '05

Yes, your right. my bad. -----Original Message----- From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] Sent: Friday, June 17, 2005 1:08 PM To: Matt Schulte Cc: users(a)openser.org Subject: Re: [Users] extcmd removed? Hi Matt, there are two module which allows you to run external application : exec ext extcmd was a module developed exclusively for a FOKUS project - some interface (via udp) messaging for a JAVA application... are you sure you are not referring to "exec" as functionality?? regards, bogdan Matt Schulte wrote: >All, >I was just curious why ext/cmd were removed? > >The website says: >"extcmd - the functionality is not required anymore" > >I've ran into several cases where we require running external commands, >while I reason they're not the best idea and can cause memory >leaks/crashes, there is no alternativer other than writing a module. Of >which no one in our company can do so :-) > >ideas? Thanks. > >_______________________________________________ >Users mailing list >Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users > > >

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[Users] extcmd removed?
by Matt Schulte 17 Jun '05

17 Jun '05

All, I was just curious why ext/cmd were removed? The website says: "extcmd - the functionality is not required anymore" I've ran into several cases where we require running external commands, while I reason they're not the best idea and can cause memory leaks/crashes, there is no alternativer other than writing a module. Of which no one in our company can do so :-) ideas? Thanks.

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RE: [Users] extcmd removed?
by Matt Schulte 17 Jun '05

17 Jun '05

We did, it f'd up in the long run. -----Original Message----- From: Juha Heinanen [mailto:jh@tutpro.com] Sent: Friday, June 17, 2005 1:10 PM To: Matt Schulte Cc: users(a)openser.org Subject: [Users] extcmd removed? Matt Schulte writes: > leaks/crashes, there is no alternativer other than writing a module. Of > which no one in our company can do so :-) hire someone from outside to write your module. -- juha

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[Serusers] OpenSER release
by Max Malloy 17 Jun '05

17 Jun '05

2 1

[Serusers] second mediaproxy on another network, not working
by Rosario Pingaro 17 Jun '05

17 Jun '05

I have successfully installed ser and mediaproxy on my network, it is a /27. Now to load balance I installed a second mediaproxy server on a datacenter, so on another network. I configure dns and the mediaproxy.ini (on the new server) to work with the new ip address. I disbaled all firewall rules to test the proxy. But it didn't work, the call is established but no rtp flow between the ua and mediaproxy. This is the debug from the dispacher: Jun 17 18:10:32 voip proxydispatcher[26275]: command request 493086ee-17262f1a(a)10.254.0.42 10.254.0.42:16440:audio 213.178.216.33 voadsl.italycomnet.it local 213.178.208.130 remote Sipura/SPA2100-3.1.3 info=from:prova@voadsl.italycomnet.it,to:0828720607@voadsl.italycomnet.it,fromtag:56eac4de28c789eao0,totag: Jun 17 18:10:32 voip proxydispatcher[26275]: forwarding to mediaproxy on voicegw-mi.italycomnet.it:25060: got: '213.178.208.130 35006' Jun 17 18:10:32 voip proxydispatcher[26275]: command execution time: 289.48 ms Jun 17 18:10:32 voip proxydispatcher[26275]: command lookup 493086ee-17262f1a(a)10.254.0.42 213.178.208.130:13308:audio 213.178.208.130 voadsl.italycomnet.it local voadsl.italycomnet.it unknown Asterisk=20Italycom=20Mix info=from:prova@voadsl.italycomnet.it,to:0828720607@voadsl.italycomnet.it,fromtag:56eac4de28c789eao0,totag:as58d31ab2 Jun 17 18:10:33 voip proxydispatcher[26275]: forwarding to mediaproxy on voicegw-mi.italycomnet.it:25060: got: '213.178.208.130 35006' Jun 17 18:10:33 voip proxydispatcher[26275]: command execution time: 183.21 ms Jun 17 18:10:40 voip proxydispatcher[26275]: command lookup 493086ee-17262f1a(a)10.254.0.42 213.178.208.130:13308:audio 213.178.208.130 voadsl.italycomnet.it local voadsl.italycomnet.it unknown Asterisk=20Italycom=20Mix info=from:prova@voadsl.italycomnet.it,to:0828720607@voadsl.italycomnet.it,fromtag:56eac4de28c789eao0,totag:as58d31ab2 Jun 17 18:10:41 voip proxydispatcher[26275]: forwarding to mediaproxy on voicegw-mi.italycomnet.it:25060: got: '213.178.208.130 35006' Jun 17 18:10:41 voip proxydispatcher[26275]: command execution time: 175.07 ms Jun 17 18:10:46 voip proxydispatcher[26275]: command delete 493086ee-17262f1a(a)10.254.0.42 info= Jun 17 18:10:46 voip proxydispatcher[26275]: forwarding to mediaproxy on voicegw-mi.italycomnet.it:25060: got: '' Jun 17 18:10:46 voip proxydispatcher[26275]: command execution time: 191.27 ms this is the debug from the mediaproxy server: Jun 17 18:10:32 voicegw-mi mediaproxy[4213]: command request 493086ee-17262f1a(a)10.254.0.42 10.254.0.42:16440:audio 213.178.216.33 voadsl.italycomnet.it local 213.178.208.130 remote Sipura/SPA2100-3.1.3 info=from:prova@voadsl.italycomnet.it,to:0828720607@voadsl.italycomnet.it,fromtag:56eac4de28c789eao0,totag:,dispatcher Jun 17 18:10:32 voicegw-mi mediaproxy[4213]: session 493086ee-17262f1a(a)10.254.0.42: started. listening on 213.178.208.130:35006 Jun 17 18:10:32 voicegw-mi mediaproxy[4213]: command execution time: 1.61 ms Jun 17 18:10:33 voicegw-mi mediaproxy[4213]: command lookup 493086ee-17262f1a(a)10.254.0.42 213.178.208.130:13308:audio 213.178.208.130 voadsl.italycomnet.it local voadsl.italycomnet.it unknown Asterisk=20Italycom=20Mix info=from:prova@voadsl.italycomnet.it,to:0828720607@voadsl.italycomnet.it,fromtag:56eac4de28c789eao0,totag:as58d31ab2,dispatcher Jun 17 18:10:33 voicegw-mi mediaproxy[4213]: command execution time: 0.56 ms Jun 17 18:10:33 voicegw-mi mediaproxy[4213]: session 493086ee-17262f1a(a)10.254.0.42: called signed in from 213.178.208.130:13308 (RTP) (will return to 213.178.208.130:13308) Jun 17 18:10:33 voicegw-mi mediaproxy[4213]: warning: Received packet from a third party: 213.178.216.2:16440 Jun 17 18:10:40 voicegw-mi mediaproxy[4213]: command lookup 493086ee-17262f1a(a)10.254.0.42 213.178.208.130:13308:audio 213.178.208.130 voadsl.italycomnet.it local voadsl.italycomnet.it unknown Asterisk=20Italycom=20Mix info=from:prova@voadsl.italycomnet.it,to:0828720607@voadsl.italycomnet.it,fromtag:56eac4de28c789eao0,totag:as58d31ab2,dispatcher Jun 17 18:10:40 voicegw-mi mediaproxy[4213]: command execution time: 0.18 ms Jun 17 18:10:46 voicegw-mi mediaproxy[4213]: command delete 493086ee-17262f1a(a)10.254.0.42 info=dispatcher Jun 17 18:10:46 voicegw-mi mediaproxy[4213]: session 493086ee-17262f1a(a)10.254.0.42: 0/646/0 packets, 0/38760/0 bytes (caller/called/relayed) Jun 17 18:10:46 voicegw-mi mediaproxy[4213]: session 493086ee-17262f1a(a)10.254.0.42: ended. Jun 17 18:10:46 voicegw-mi mediaproxy[4213]: command execution time: 0.42 ms Jun 17 18:11:52 voicegw-mi sshd(pam_unix)[4228]: session opened for user root by root(uid=0) Seems that is not able to find the right pubblic address of the ua.... Any help is appreciated. Thanks Rosario

2 1

[Users] TLS support
by Bogdan-Andrei Iancu 17 Jun '05

17 Jun '05

Hi folks, thanks to Peter and Cesc we have TLS support buit-in OpenSER. The implementation is only on CVS head (development branch). To getting do: - if you already have a local checkout of CVS head: cvs update -dP - no local checkout of CVS head: cvs -d:pserver:anonymous@cvs.sourceforge.net:/cvsroot/openser login cvs -z3 -d:pserver:anonymous@cvs.sourceforge.net:/cvsroot/openser co -P sip-server Full documentation about how to compile and use it (script example included) can be found in the documentation repository: http://openser.org/docs/tls.html The next step will be to convince as many people as possible to use it and to test it for eventual bugs/problems. I will be very helpful in moving towards a stable version. regards, Bogdan

1 0

[Serusers] 30s calls
by Jerlique Ban 17 Jun '05

17 Jun '05

Hello, I have configured SER and asterisk to allow me to make calls to the PSTN network, however on my voip phone (behind NAT) I am having issues with the voip tx audio dropping out after 30 seconds. Now I'm guessing it's a nat issue but even that doesn't really make sense!! Why, well because the only the TX of the voip phone drops out (ie the PSTN phone cannot hear what is said on the voip phone). The PSTN phone can still transmit audio to the voip phone (through the nat). Anyway in SER, I have set the natping_interval to 5 seconds, and this still doesn't resolve the issue. Strangely at the time that the audio disconnects Asterisk is sending my phone an INVITE message. Why would it do this mid call? I'm using SER0.9.0+Asterisk as my platform. Any pointers?? JB

2 1

[Serusers] avaya
by Iqbal 17 Jun '05

17 Jun '05

Hi has anyone worked with avaya, and linked it to ser, if so are there any obstacles or is it pretty straightforward Iqbal

1 0

[Serusers] free SER
by A C 17 Jun '05

17 Jun '05

Dear Bogdan, Dear Daniel, As a user of SER for number of years now I really appreciate your work and effort to liberate SER from the clutches of iptelorg with its microsoft like mentality. I am sure it was not easy for you to stand-up against those who have paid your salary for a number of years and have given you the knowledge and software with which you have built your own company. I also agree with you that the Fraunhofer as the maintainer of the iptel.org and berlios sites has in no way the right to refer on these sites to a spinn-off in which it is involved in -having invested a huge sum in this project including your salary in the last years is surely no excuse. I am also of your opinion that the benefits of the users should be the utmost goal of all who are involved in SER. Therefore my question is: will the next release of openSER also include the VNT and VPS products your company are offering? Providing these components to the community will surely help the SER users to offer better VoIP services -which as you have stressed should be our major goal. I would recommend to publish your products under the BSD license so that we could include them into our own products as well? We are building an NGN product that would benefit greatly from your components. iptelorg has actually asked us to pay for their source code. Those guys have surely not realized yet that SER does not belong to them but to the users and community. Best regards -- ___________________________________________________________ Sign-up for Ads Free at Mail.com http://promo.mail.com/adsfreejump.htm

1 0

[Serusers] ONsip.org poll announcement regarding SER/OpenSER
by Greger V. Teigre 17 Jun '05

17 Jun '05

I believe the community should be allowed to tell the core developers their opinion and I have thus created a poll at http://onsip.org/ (http://www.onsip.org/modules/xoopspoll/index.php?poll_id=5) I apologize that you will have to log in (or register if you have not yet done so) in order to vote. This is how Xoops (the content management system) avoids voting manipulation through (otherwise allowed) multiple votes from one user. Below are the statements that you can agree to or not. Please send me a private note if you have suggestions for other statements that will be informative. Regards, Greger ------------------------------------------------------------------------------------- What is your reaction to the OpenSER announcement? Overall, a separate OpenSER will benefit the users Overall, a separate OpenSER will not benefit the users I will go for the "old" SER I will go for the new OpenSER The conflict should be resolved into one project Splitting in two is a good thing

1 0

Re: [Users] new version openser how to deal with call a SIP uri with multi registar
by jimmy way 17 Jun '05

17 Jun '05

Hi bogdan, Need I write script for failure_route and onreply_route in ser.cfg to support parallel fork or I only lets openser to do that by itself. If need, how to write the script? By the way, how to support serial fork? Regards, Jimway --- Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> wrote: > Hi Jimmy. > > see inline comments. As short note, what you are > describing in parallel > forking and it behaves exactly as you thing (see RFC > 3261). > > regards, > bogdan > > jimmy way wrote: > > >Hi all, > > > > The question is: > > 1. a user A registar from multi address, like > > UA1 registrar from address1 > > UA2 registrar from address2 > > UA3 registrar from address3 > > and so on > > so in db "location" there are multi address > in > >the list about user A > > 2. if set append_branch as on in registrar > module, > >when user B invite user A, Openser server will > append > >all these address in the invite packet. So all > these > >user will get the call. normally all UAs ringing, > then > > 3. if one of these UAs accept the call, how > about > >other UAs, will they stop ringing? > > > > > yes, the other pending branches will be > automatically cancelled. > > > 4. if one of these UAs deny(40x) the call, how > >about other UAs, will the still ringing? > > > > > yes - there are two way for a parallel forked call > to end: either one of > the branches picks up (2xx reply), either all of > them send negative > replies (>=300). > > > 5. if one of these UAs not connected, how about > >caller the callee do? > > > > > if one of the branches is not connected (but still > registered), an > internal timeout (408) will be generated for it. > > > > >tks. > >Jimway > > > > > > > __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

3 2

[Serusers] test rtpproxy
by Bruno Oliveira 17 Jun '05

17 Jun '05

hi. as I make test connection beetwen kphone ( sip client ) and me rtpproxy. thanks for your time. -- []s, Bruno "Niggas" Oliveira Belo Horizonte - MG Msn: n1gg4s(a)gmail.com Icq: 176314647 "Todo o nosso descontentamento por aquilo que nos falta procede da nossa falta de gratidão por aquilo que temos."

1 0

Re: [Serusers] rtpproxy and serctl with problem
by Bruno Oliveira 17 Jun '05

17 Jun '05

hi I try, but no resolv. Print this in term: "uiara:~# netstat -anp|grep 5060 tcp 0 0 200.236.162.3:5060 0.0.0.0:* OUÇA 2816/ser tcp 0 0 192.168.1.2:5060 0.0.0.0:* OUÇA 2816/ser udp 0 0 200.236.162.3:5060 0.0.0.0:* 2816/ser udp 0 0 192.168.1.2:5060 0.0.0.0:* 2816/ser" it continues with exactly error. thanks for your time > On 6/17/05, Klaus Darilion <klaus.mailinglists(a)pernau.at> wrote: > > Bruno Oliveira wrote: > > > hi > > > > > > i have rtpproxy runing and serctl. where it is file config of rtpproxy > > > ? and when I type this command "serctl start" show me in > > > /var/log/syslog this message: > > > > > > "Jun 16 14:45:33 uiara /usr/local/sbin/ser[3237]: Maxfwd module- initializing > > > Jun 16 14:45:33 uiara /usr/local/sbin/ser[3237]: INFO: udp_init: > > > SO_RCVBUF is initially 111616 > > > Jun 16 14:45:33 uiara /usr/local/sbin/ser[3237]: INFO: udp_init: > > > SO_RCVBUF is finally 223232 > > > Jun 16 14:45:33 uiara /usr/local/sbin/ser[3237]: ERROR: tcp_init: > > > bind(6, 0x813753c, 16) on 192.168.1.2: Address already in use" > > > > Looks like another process already has bind to TCP port 5060. Maybe ser > > is already running? Or maybe a SIP client? > > > > Try > > #> netstat -anp|grep 5060 > > > > regards, > > klaus > > > > > -- > []s, > Bruno "Niggas" Oliveira > Belo Horizonte - MG > Msn: n1gg4s(a)gmail.com > Icq: 176314647 > > "Todo o nosso descontentamento por aquilo > que nos falta procede da nossa falta de > gratidão por aquilo que temos." > -- []s, Bruno "Niggas" Oliveira Belo Horizonte - MG Msn: n1gg4s(a)gmail.com Icq: 176314647 "Todo o nosso descontentamento por aquilo que nos falta procede da nossa falta de gratidão por aquilo que temos."

1 0

[Serusers] SER does not forward ACK
by jero 17 Jun '05

17 Jun '05

I've a problem with SER forwarding ACK's when a call has stablished. I only need to make outbound call from sip-phone, every call has to be redirected to AS5300. These are the packets sended: SIP-PHONE -----------> SER -------------> AS5300 ------>PSTN _________________________________________________ |-----invite------->| | |<----trying--------| | | |-----invite------->| | |<----trying--------| | |<----ringing-------| |<----ringing-------| | | |<------ok----------| |<------ok----------| | |-------*ack*------>| | <--here is the problem | | | | |<------ok----------| |<------ok----------| | | |<------ok----------| |<------ok----------| | | | timeout As seen, the ack looses in SER and is not sended to AS5300, the call is stablished, but after 20 seconds is dropped by as5300 because of the timeout. I've read the messages on this forum about this, and it seems to be a misconfiguration problem, but i didn't find the solution. Here is mi ser.cfg and a more detailed trace of the packets. SIP-PHONE - 81.37.111.201 SER - 196.220.57.58 AS5300 - 196.220.57.51 # # $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=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" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "/usr/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/acc.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/lib/ser/modules/auth.so" #loadmodule "/usr/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # -- acc params -- # set the reporting log level modparam("acc", "log_level", 1) # number of flag, which will be used for accounting; if a message is # labeled with this flag, its completion status will be reported modparam("acc", "log_flag", 1 ) modparam("acc", "radius_config", "/usr/local/etc/radiusclient/radiusclient.conf") modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 2) # ------------------------- request routing logic ------------------- # main routing logic alias="196.220.57.58" 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; }; # Lots deleted, such as processing registrations # we record-route all messages -- to make sure that # subsequent messages will go through our proxy. if(method=="INVITE"){ record_route(); } # loose-route processing if (loose_route()) { t_relay(); break; }; if(uri==myself){ if(method=="REGISTER"){ save("location"); break; } } # IF HOST IS JUST NUMBERS SEND TO route(1) if (uri =~ "^sip:[0-9]*@"){ log(1, "SER: matched ^sip:[0-9]*@ - just numbers - process in route(3)\n"); route(1); break; }; } route[1] { rewritehostport("196.220.57.51:5060"); setflag(1); t_relay(); break; } ########################################################################## Packet trace ########################################################################## No. Time Source Destination Protocol Info 5053 17.486988 81.37.111.201 196.220.57.58 SIP/SDP Request: INVITE sip:607628000@196.220.57.58, with session description Internet Protocol, Src Addr: 81.37.111.201 (81.37.111.201), Dst Addr: 196.220.57.58 (196.220.57.58) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: INVITE sip:607628000@196.220.57.58 SIP/2.0 Message Header From: "777910020"<sip:777910020@196.220.57.58>;tag=c0a80129-13c4-3e951-f479ae6-6c8e To: <sip:607628000@196.220.57.58> Call-ID: 569ecc-c0a80129-13c4-3e951-f479ae1-4390(a)196.220.57.58 CSeq: 1 INVITE Via: SIP/2.0/UDP 192.168.1.41:5060;branch=z9hG4bK-3e951-f479ae6-2caf Max-Forwards: 70 Supported: replaces User-Agent: FXS_GW (4asipfxs.108a) Contact: <sip:777910020@81.37.111.201:5060> Content-Type: application/SDP Content-Length: 286 Message body No. Time Source Destination Protocol Info 5054 17.491678 196.220.57.58 81.37.111.201 SIP Status: 100 trying -- your call is important to us Internet Protocol, Src Addr: 196.220.57.58 (196.220.57.58), Dst Addr: 81.37.111.201 (81.37.111.201) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 100 trying -- your call is important to us Message Header From: "777910020"<sip:777910020@196.220.57.58>;tag=c0a80129-13c4-3e951-f479ae6-6c8e To: <sip:607628000@196.220.57.58> Call-ID: 569ecc-c0a80129-13c4-3e951-f479ae1-4390(a)196.220.57.58 CSeq: 1 INVITE Via: SIP/2.0/UDP 192.168.1.41:5060;branch=z9hG4bK-3e951-f479ae6-2caf;received=81.37.111.201 Server: Sip EXpress router (0.9.0 (i386/linux)) Content-Length: 0 Warning: 392 196.220.57.58:5060 "Noisy feedback tells: pid=1024 req_src_ip=81.37.111.201 req_src_port=5060 in_uri=sip:607628000@196.220.57.58 out_uri=sip:607628000@196.220.57.51:5060 via_cnt==1" No. Time Source Destination Protocol Info 5058 17.494454 196.220.57.58 196.220.57.51 SIP/SDP Request: INVITE sip:607628000@196.220.57.51:5060, with session description Internet Protocol, Src Addr: 196.220.57.58 (196.220.57.58), Dst Addr: 196.220.57.51 (196.220.57.51) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: INVITE sip:607628000@196.220.57.51:5060 SIP/2.0 Message Header Record-Route: <sip:196.220.57.58;ftag=c0a80129-13c4-3e951-f479ae6-6c8e;lr=on> From: "777910020"<sip:777910020@196.220.57.58>;tag=c0a80129-13c4-3e951-f479ae6-6c8e To: <sip:607628000@196.220.57.58> Call-ID: 569ecc-c0a80129-13c4-3e951-f479ae1-4390(a)196.220.57.58 CSeq: 1 INVITE Via: SIP/2.0/UDP 196.220.57.58;branch=z9hG4bK3e84.b6c26904.0 Via: SIP/2.0/UDP 192.168.1.41:5060;received=81.37.111.201;branch=z9hG4bK-3e951-f479ae6-2caf Max-Forwards: 16 Supported: replaces User-Agent: FXS_GW (4asipfxs.108a) Contact: <sip:777910020@81.37.111.201:5060> Content-Type: application/SDP Content-Length: 286 Message body No. Time Source Destination Protocol Info 5063 17.508105 196.220.57.51 196.220.57.58 SIP Status: 100 Trying Internet Protocol, Src Addr: 196.220.57.51 (196.220.57.51), Dst Addr: 196.220.57.58 (196.220.57.58) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 100 Trying Message Header Via: SIP/2.0/UDP 196.220.57.58;branch=z9hG4bK3e84.b6c26904.0,SIP/2.0/UDP 192.168.1.41:5060;received=81.37.111.201;branch=z9hG4bK-3e951-f479ae6-2caf From: "777910020"<sip:777910020@196.220.57.58>;tag=c0a80129-13c4-3e951-f479ae6-6c8e To: <sip:607628000@196.220.57.58>;tag=2A3EE1E8-F7B Date: Thu, 16 Jun 2005 08:44:22 GMT Call-ID: 569ecc-c0a80129-13c4-3e951-f479ae1-4390(a)196.220.57.58 Server: Cisco-SIPGateway/IOS-12.x CSeq: 1 INVITE Allow-Events: telephone-event Content-Length: 0 No. Time Source Destination Protocol Info 8037 28.278422 196.220.57.51 196.220.57.58 SIP Status: 180 Ringing Internet Protocol, Src Addr: 196.220.57.51 (196.220.57.51), Dst Addr: 196.220.57.58 (196.220.57.58) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 180 Ringing Message Header Via: SIP/2.0/UDP 196.220.57.58;branch=z9hG4bK3e84.b6c26904.0,SIP/2.0/UDP 192.168.1.41:5060;received=81.37.111.201;branch=z9hG4bK-3e951-f479ae6-2caf From: "777910020"<sip:777910020@196.220.57.58>;tag=c0a80129-13c4-3e951-f479ae6-6c8e To: <sip:607628000@196.220.57.58>;tag=2A3EE1E8-F7B Date: Thu, 16 Jun 2005 08:44:22 GMT Call-ID: 569ecc-c0a80129-13c4-3e951-f479ae1-4390(a)196.220.57.58 Server: Cisco-SIPGateway/IOS-12.x CSeq: 1 INVITE Allow-Events: telephone-event Contact: <sip:607628000@196.220.57.51:5060> Record-Route: <sip:196.220.57.58;ftag=c0a80129-13c4-3e951-f479ae6-6c8e;lr=on> Content-Length: 0 No. Time Source Destination Protocol Info 8038 28.279923 196.220.57.58 81.37.111.201 SIP Status: 180 Ringing Internet Protocol, Src Addr: 196.220.57.58 (196.220.57.58), Dst Addr: 81.37.111.201 (81.37.111.201) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 180 Ringing Message Header Via: SIP/2.0/UDP 192.168.1.41:5060;received=81.37.111.201;branch=z9hG4bK-3e951-f479ae6-2caf From: "777910020"<sip:777910020@196.220.57.58>;tag=c0a80129-13c4-3e951-f479ae6-6c8e To: <sip:607628000@196.220.57.58>;tag=2A3EE1E8-F7B Date: Thu, 16 Jun 2005 08:44:22 GMT Call-ID: 569ecc-c0a80129-13c4-3e951-f479ae1-4390(a)196.220.57.58 Server: Cisco-SIPGateway/IOS-12.x CSeq: 1 INVITE Allow-Events: telephone-event Contact: <sip:607628000@196.220.57.51:5060> Record-Route: <sip:196.220.57.58;ftag=c0a80129-13c4-3e951-f479ae6-6c8e;lr=on> Content-Length: 0 No. Time Source Destination Protocol Info 9791 33.535713 196.220.57.51 196.220.57.58 SIP/SDP Status: 200 OK, with session description Internet Protocol, Src Addr: 196.220.57.51 (196.220.57.51), Dst Addr: 196.220.57.58 (196.220.57.58) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 196.220.57.58;branch=z9hG4bK3e84.b6c26904.0,SIP/2.0/UDP 192.168.1.41:5060;received=81.37.111.201;branch=z9hG4bK-3e951-f479ae6-2caf From: "777910020"<sip:777910020@196.220.57.58>;tag=c0a80129-13c4-3e951-f479ae6-6c8e To: <sip:607628000@196.220.57.58>;tag=2A3EE1E8-F7B Date: Thu, 16 Jun 2005 08:44:22 GMT Call-ID: 569ecc-c0a80129-13c4-3e951-f479ae1-4390(a)196.220.57.58 Server: Cisco-SIPGateway/IOS-12.x CSeq: 1 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO Allow-Events: telephone-event Contact: <sip:607628000@196.220.57.51:5060> Record-Route: <sip:196.220.57.58;ftag=c0a80129-13c4-3e951-f479ae6-6c8e;lr=on> Content-Type: application/sdp Content-Length: 229 Message body No. Time Source Destination Protocol Info 9794 33.537703 196.220.57.58 81.37.111.201 SIP/SDP Status: 200 OK, with session description Internet Protocol, Src Addr: 196.220.57.58 (196.220.57.58), Dst Addr: 81.37.111.201 (81.37.111.201) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 192.168.1.41:5060;received=81.37.111.201;branch=z9hG4bK-3e951-f479ae6-2caf From: "777910020"<sip:777910020@196.220.57.58>;tag=c0a80129-13c4-3e951-f479ae6-6c8e To: <sip:607628000@196.220.57.58>;tag=2A3EE1E8-F7B Date: Thu, 16 Jun 2005 08:44:22 GMT Call-ID: 569ecc-c0a80129-13c4-3e951-f479ae1-4390(a)196.220.57.58 Server: Cisco-SIPGateway/IOS-12.x CSeq: 1 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO Allow-Events: telephone-event Contact: <sip:607628000@196.220.57.51:5060> Record-Route: <sip:196.220.57.58;ftag=c0a80129-13c4-3e951-f479ae6-6c8e;lr=on> Content-Type: application/sdp Content-Length: 229 Message body No. Time Source Destination Protocol Info 9956 34.033762 196.220.57.51 196.220.57.58 SIP/SDP Status: 200 OK, with session description Internet Protocol, Src Addr: 196.220.57.51 (196.220.57.51), Dst Addr: 196.220.57.58 (196.220.57.58) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 196.220.57.58;branch=z9hG4bK3e84.b6c26904.0,SIP/2.0/UDP 192.168.1.41:5060;received=81.37.111.201;branch=z9hG4bK-3e951-f479ae6-2caf From: "777910020"<sip:777910020@196.220.57.58>;tag=c0a80129-13c4-3e951-f479ae6-6c8e To: <sip:607628000@196.220.57.58>;tag=2A3EE1E8-F7B Date: Thu, 16 Jun 2005 08:44:22 GMT Call-ID: 569ecc-c0a80129-13c4-3e951-f479ae1-4390(a)196.220.57.58 Server: Cisco-SIPGateway/IOS-12.x CSeq: 1 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO Allow-Events: telephone-event Contact: <sip:607628000@196.220.57.51:5060> Record-Route: <sip:196.220.57.58;ftag=c0a80129-13c4-3e951-f479ae6-6c8e;lr=on> Content-Type: application/sdp Content-Length: 229 Message body No. Time Source Destination Protocol Info 9957 34.035211 196.220.57.58 81.37.111.201 SIP/SDP Status: 200 OK, with session description Internet Protocol, Src Addr: 196.220.57.58 (196.220.57.58), Dst Addr: 81.37.111.201 (81.37.111.201) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 192.168.1.41:5060;received=81.37.111.201;branch=z9hG4bK-3e951-f479ae6-2caf From: "777910020"<sip:777910020@196.220.57.58>;tag=c0a80129-13c4-3e951-f479ae6-6c8e To: <sip:607628000@196.220.57.58>;tag=2A3EE1E8-F7B Date: Thu, 16 Jun 2005 08:44:22 GMT Call-ID: 569ecc-c0a80129-13c4-3e951-f479ae1-4390(a)196.220.57.58 Server: Cisco-SIPGateway/IOS-12.x CSeq: 1 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO Allow-Events: telephone-event Contact: <sip:607628000@196.220.57.51:5060> Record-Route: <sip:196.220.57.58;ftag=c0a80129-13c4-3e951-f479ae6-6c8e;lr=on> Content-Type: application/sdp Content-Length: 229 Message body No. Time Source Destination Protocol Info 9958 34.045266 81.37.111.201 196.220.57.58 SIP Request: ACK sip:607628000@196.220.57.51:5060 Internet Protocol, Src Addr: 81.37.111.201 (81.37.111.201), Dst Addr: 196.220.57.58 (196.220.57.58) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: ACK sip:607628000@196.220.57.51:5060 SIP/2.0 Message Header From: "777910020"<sip:777910020@196.220.57.58>;tag=c0a80129-13c4-3e951-f479ae6-6c8e To: <sip:607628000@196.220.57.58>;tag=2A3EE1E8-F7B Call-ID: 569ecc-c0a80129-13c4-3e951-f479ae1-4390(a)196.220.57.58 CSeq: 1 ACK Via: SIP/2.0/UDP 192.168.1.41:5060;branch=z9hG4bK-3e962-f47db5a-57d8 Max-Forwards: 70 User-Agent: FXS_GW (4asipfxs.108a) Contact: <sip:777910020@81.37.111.201:5060> Route: <sip:196.220.57.58;lr;ftag=c0a80129-13c4-3e951-f479ae6-6c8e> Content-Length: 0 No. Time Source Destination Protocol Info 10269 35.033720 196.220.57.51 196.220.57.58 SIP/SDP Status: 200 OK, with session description Internet Protocol, Src Addr: 196.220.57.51 (196.220.57.51), Dst Addr: 196.220.57.58 (196.220.57.58) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 196.220.57.58;branch=z9hG4bK3e84.b6c26904.0,SIP/2.0/UDP 192.168.1.41:5060;received=81.37.111.201;branch=z9hG4bK-3e951-f479ae6-2caf From: "777910020"<sip:777910020@196.220.57.58>;tag=c0a80129-13c4-3e951-f479ae6-6c8e To: <sip:607628000@196.220.57.58>;tag=2A3EE1E8-F7B Date: Thu, 16 Jun 2005 08:44:22 GMT Call-ID: 569ecc-c0a80129-13c4-3e951-f479ae1-4390(a)196.220.57.58 Server: Cisco-SIPGateway/IOS-12.x CSeq: 1 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO Allow-Events: telephone-event Contact: <sip:607628000@196.220.57.51:5060> Record-Route: <sip:196.220.57.58;ftag=c0a80129-13c4-3e951-f479ae6-6c8e;lr=on> Content-Type: application/sdp Content-Length: 229 Message body No. Time Source Destination Protocol Info 10271 35.035346 196.220.57.58 81.37.111.201 SIP/SDP Status: 200 OK, with session description Internet Protocol, Src Addr: 196.220.57.58 (196.220.57.58), Dst Addr: 81.37.111.201 (81.37.111.201) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 192.168.1.41:5060;received=81.37.111.201;branch=z9hG4bK-3e951-f479ae6-2caf From: "777910020"<sip:777910020@196.220.57.58>;tag=c0a80129-13c4-3e951-f479ae6-6c8e To: <sip:607628000@196.220.57.58>;tag=2A3EE1E8-F7B Date: Thu, 16 Jun 2005 08:44:22 GMT Call-ID: 569ecc-c0a80129-13c4-3e951-f479ae1-4390(a)196.220.57.58 Server: Cisco-SIPGateway/IOS-12.x CSeq: 1 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO Allow-Events: telephone-event Contact: <sip:607628000@196.220.57.51:5060> Record-Route: <sip:196.220.57.58;ftag=c0a80129-13c4-3e951-f479ae6-6c8e;lr=on> Content-Type: application/sdp Content-Length: 229 Message body No. Time Source Destination Protocol Info 10791 36.690602 196.220.57.51 81.37.111.201 RTCP Sender Report Internet Protocol, Src Addr: 196.220.57.51 (196.220.57.51), Dst Addr: 81.37.111.201 (81.37.111.201) User Datagram Protocol, Src Port: 19343 (19343), Dst Port: 16385 (16385) Real-time Transport Control Protocol Stream setup by SDP (frame 5058) 10.. .... = Version: RFC 1889 Version (2) ..0. .... = Padding: False ...0 0001 = Reception report count: 1 Packet type: Sender Report (200) Length: 12 Sender SSRC: 459946035 Timestamp, MSW: 3327900282 Timestamp, LSW: 356227917 RTP timestamp: 0 Sender's packet count: 52 Sender's octet count: 3011 Source 1 Real-time Transport Control Protocol Stream setup by SDP (frame 5058) 10.. .... = Version: RFC 1889 Version (2) ..0. .... = Padding: False ...0 0001 = Source count: 1 Packet type: Source description (202) Length: 19 Chunk 1, SSRC/CSRC 459946035 No. Time Source Destination Protocol Info 10894 37.033690 196.220.57.51 196.220.57.58 SIP/SDP Status: 200 OK, with session description Internet Protocol, Src Addr: 196.220.57.51 (196.220.57.51), Dst Addr: 196.220.57.58 (196.220.57.58) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 196.220.57.58;branch=z9hG4bK3e84.b6c26904.0,SIP/2.0/UDP 192.168.1.41:5060;received=81.37.111.201;branch=z9hG4bK-3e951-f479ae6-2caf From: "777910020"<sip:777910020@196.220.57.58>;tag=c0a80129-13c4-3e951-f479ae6-6c8e To: <sip:607628000@196.220.57.58>;tag=2A3EE1E8-F7B Date: Thu, 16 Jun 2005 08:44:22 GMT Call-ID: 569ecc-c0a80129-13c4-3e951-f479ae1-4390(a)196.220.57.58 Server: Cisco-SIPGateway/IOS-12.x CSeq: 1 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO Allow-Events: telephone-event Contact: <sip:607628000@196.220.57.51:5060> Record-Route: <sip:196.220.57.58;ftag=c0a80129-13c4-3e951-f479ae6-6c8e;lr=on> Content-Type: application/sdp Content-Length: 229 Message body No. Time Source Destination Protocol Info 10895 37.035934 196.220.57.58 81.37.111.201 SIP/SDP Status: 200 OK, with session description Internet Protocol, Src Addr: 196.220.57.58 (196.220.57.58), Dst Addr: 81.37.111.201 (81.37.111.201) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 192.168.1.41:5060;received=81.37.111.201;branch=z9hG4bK-3e951-f479ae6-2caf From: "777910020"<sip:777910020@196.220.57.58>;tag=c0a80129-13c4-3e951-f479ae6-6c8e To: <sip:607628000@196.220.57.58>;tag=2A3EE1E8-F7B Date: Thu, 16 Jun 2005 08:44:22 GMT Call-ID: 569ecc-c0a80129-13c4-3e951-f479ae1-4390(a)196.220.57.58 Server: Cisco-SIPGateway/IOS-12.x CSeq: 1 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO Allow-Events: telephone-event Contact: <sip:607628000@196.220.57.51:5060> Record-Route: <sip:196.220.57.58;ftag=c0a80129-13c4-3e951-f479ae6-6c8e;lr=on> Content-Type: application/sdp Content-Length: 229 Message body No. Time Source Destination Protocol Info 11726 40.222188 196.220.57.51 81.37.111.201 RTCP Sender Report Internet Protocol, Src Addr: 196.220.57.51 (196.220.57.51), Dst Addr: 81.37.111.201 (81.37.111.201) User Datagram Protocol, Src Port: 19343 (19343), Dst Port: 16385 (16385) Real-time Transport Control Protocol Stream setup by SDP (frame 5058) 10.. .... = Version: RFC 1889 Version (2) ..0. .... = Padding: False ...0 0001 = Reception report count: 1 Packet type: Sender Report (200) Length: 12 Sender SSRC: 459946035 Timestamp, MSW: 3327900285 Timestamp, LSW: 2639679338 RTP timestamp: 0 Sender's packet count: 102 Sender's octet count: 5715 Source 1 Real-time Transport Control Protocol Stream setup by SDP (frame 5058) 10.. .... = Version: RFC 1889 Version (2) ..0. .... = Padding: False ...0 0001 = Source count: 1 Packet type: Source description (202) Length: 19 Chunk 1, SSRC/CSRC 459946035 No. Time Source Destination Protocol Info 12971 44.392482 196.220.57.51 81.37.111.201 RTCP Sender Report Internet Protocol, Src Addr: 196.220.57.51 (196.220.57.51), Dst Addr: 81.37.111.201 (81.37.111.201) User Datagram Protocol, Src Port: 19343 (19343), Dst Port: 16385 (16385) Real-time Transport Control Protocol Stream setup by SDP (frame 5058) 10.. .... = Version: RFC 1889 Version (2) ..0. .... = Padding: False ...0 0001 = Reception report count: 1 Packet type: Sender Report (200) Length: 12 Sender SSRC: 459946035 Timestamp, MSW: 3327900289 Timestamp, LSW: 3371471883 RTP timestamp: 0 Sender's packet count: 108 Sender's octet count: 6075 Source 1 Real-time Transport Control Protocol Stream setup by SDP (frame 5058) 10.. .... = Version: RFC 1889 Version (2) ..0. .... = Padding: False ...0 0001 = Source count: 1 Packet type: Source description (202) Length: 19 Chunk 1, SSRC/CSRC 459946035 No. Time Source Destination Protocol Info 13187 45.033530 196.220.57.51 196.220.57.58 SIP/SDP Status: 200 OK, with session description Internet Protocol, Src Addr: 196.220.57.51 (196.220.57.51), Dst Addr: 196.220.57.58 (196.220.57.58) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 196.220.57.58;branch=z9hG4bK3e84.b6c26904.0,SIP/2.0/UDP 192.168.1.41:5060;received=81.37.111.201;branch=z9hG4bK-3e951-f479ae6-2caf From: "777910020"<sip:777910020@196.220.57.58>;tag=c0a80129-13c4-3e951-f479ae6-6c8e To: <sip:607628000@196.220.57.58>;tag=2A3EE1E8-F7B Date: Thu, 16 Jun 2005 08:44:22 GMT Call-ID: 569ecc-c0a80129-13c4-3e951-f479ae1-4390(a)196.220.57.58 Server: Cisco-SIPGateway/IOS-12.x CSeq: 1 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO Allow-Events: telephone-event Contact: <sip:607628000@196.220.57.51:5060> Record-Route: <sip:196.220.57.58;ftag=c0a80129-13c4-3e951-f479ae6-6c8e;lr=on> Content-Type: application/sdp Content-Length: 229 Message body No. Time Source Destination Protocol Info 13188 45.034943 196.220.57.58 81.37.111.201 SIP/SDP Status: 200 OK, with session description Internet Protocol, Src Addr: 196.220.57.58 (196.220.57.58), Dst Addr: 81.37.111.201 (81.37.111.201) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 192.168.1.41:5060;received=81.37.111.201;branch=z9hG4bK-3e951-f479ae6-2caf From: "777910020"<sip:777910020@196.220.57.58>;tag=c0a80129-13c4-3e951-f479ae6-6c8e To: <sip:607628000@196.220.57.58>;tag=2A3EE1E8-F7B Date: Thu, 16 Jun 2005 08:44:22 GMT Call-ID: 569ecc-c0a80129-13c4-3e951-f479ae1-4390(a)196.220.57.58 Server: Cisco-SIPGateway/IOS-12.x CSeq: 1 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO Allow-Events: telephone-event Contact: <sip:607628000@196.220.57.51:5060> Record-Route: <sip:196.220.57.58;ftag=c0a80129-13c4-3e951-f479ae6-6c8e;lr=on> Content-Type: application/sdp Content-Length: 229 Message body No. Time Source Destination Protocol Info 14829 50.802294 196.220.57.51 81.37.111.201 RTCP Sender Report Internet Protocol, Src Addr: 196.220.57.51 (196.220.57.51), Dst Addr: 81.37.111.201 (81.37.111.201) User Datagram Protocol, Src Port: 19343 (19343), Dst Port: 16385 (16385) Real-time Transport Control Protocol Stream setup by SDP (frame 5058) 10.. .... = Version: RFC 1889 Version (2) ..0. .... = Padding: False ...0 0001 = Reception report count: 1 Packet type: Sender Report (200) Length: 12 Sender SSRC: 459946035 Timestamp, MSW: 3327900296 Timestamp, LSW: 837215452 RTP timestamp: 0 Sender's packet count: 203 Sender's octet count: 11330 Source 1 Real-time Transport Control Protocol Stream setup by SDP (frame 5058) 10.. .... = Version: RFC 1889 Version (2) ..0. .... = Padding: False ...0 0001 = Source count: 1 Packet type: Source description (202) Length: 19 Chunk 1, SSRC/CSRC 459946035 No. Time Source Destination Protocol Info 15481 53.037113 196.220.57.51 81.37.111.201 RTCP Receiver Report Internet Protocol, Src Addr: 196.220.57.51 (196.220.57.51), Dst Addr: 81.37.111.201 (81.37.111.201) User Datagram Protocol, Src Port: 19343 (19343), Dst Port: 16385 (16385) Real-time Transport Control Protocol Stream setup by SDP (frame 5058) 10.. .... = Version: RFC 1889 Version (2) ..0. .... = Padding: False ...0 0001 = Reception report count: 1 Packet type: Receiver Report (201) Length: 7 Sender SSRC: 459946035 Source 1 Real-time Transport Control Protocol Stream setup by SDP (frame 5058) 10.. .... = Version: RFC 1889 Version (2) ..0. .... = Padding: False ...0 0001 = Source count: 1 Packet type: Source description (202) Length: 7 Chunk 1, SSRC/CSRC 459946035 Real-time Transport Control Protocol Stream setup by SDP (frame 5058) 10.. .... = Version: RFC 1889 Version (2) ..0. .... = Padding: False ...0 0001 = Source count: 1 Packet type: Goodbye (203) Length: 1 Identifier: 459946035 No. Time Source Destination Protocol Info 17285 59.087291 81.37.111.201 196.220.57.58 SIP Request: BYE sip:607628000@196.220.57.51:5060 Internet Protocol, Src Addr: 81.37.111.201 (81.37.111.201), Dst Addr: 196.220.57.58 (196.220.57.58) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: BYE sip:607628000@196.220.57.51:5060 SIP/2.0 Message Header From: "777910020"<sip:777910020@196.220.57.58>;tag=c0a80129-13c4-3e951-f479ae6-6c8e To: <sip:607628000@196.220.57.58>;tag=2A3EE1E8-F7B Call-ID: 569ecc-c0a80129-13c4-3e951-f479ae1-4390(a)196.220.57.58 CSeq: 2 BYE Via: SIP/2.0/UDP 192.168.1.41:5060;branch=z9hG4bK-3e97b-f483cd0-e16 Max-Forwards: 70 Supported: replaces User-Agent: FXS_GW (4asipfxs.108a) Route: <sip:196.220.57.58;lr;ftag=c0a80129-13c4-3e951-f479ae6-6c8e> Content-Length: 0 No. Time Source Destination Protocol Info 17286 59.090474 196.220.57.58 196.220.57.51 SIP Request: BYE sip:607628000@196.220.57.51:5060 Internet Protocol, Src Addr: 196.220.57.58 (196.220.57.58), Dst Addr: 196.220.57.51 (196.220.57.51) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: BYE sip:607628000@196.220.57.51:5060 SIP/2.0 Message Header From: "777910020"<sip:777910020@196.220.57.58>;tag=c0a80129-13c4-3e951-f479ae6-6c8e To: <sip:607628000@196.220.57.58>;tag=2A3EE1E8-F7B Call-ID: 569ecc-c0a80129-13c4-3e951-f479ae1-4390(a)196.220.57.58 CSeq: 2 BYE Via: SIP/2.0/UDP 196.220.57.58;branch=z9hG4bK0e84.aed52364.0 Via: SIP/2.0/UDP 192.168.1.41:5060;received=81.37.111.201;branch=z9hG4bK-3e97b-f483cd0-e16 Max-Forwards: 16 Supported: replaces User-Agent: FXS_GW (4asipfxs.108a) Content-Length: 0 No. Time Source Destination Protocol Info 17289 59.094579 196.220.57.51 196.220.57.58 SIP Status: 481 Call Leg/Transaction Does Not Exist Internet Protocol, Src Addr: 196.220.57.51 (196.220.57.51), Dst Addr: 196.220.57.58 (196.220.57.58) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 481 Call Leg/Transaction Does Not Exist Message Header Via: SIP/2.0/UDP 196.220.57.58;branch=z9hG4bK0e84.aed52364.0,SIP/2.0/UDP 192.168.1.41:5060;received=81.37.111.201;branch=z9hG4bK-3e97b-f483cd0-e16 From: "777910020"<sip:777910020@196.220.57.58>;tag=c0a80129-13c4-3e951-f479ae6-6c8e To: <sip:607628000@196.220.57.58>;tag=2A3EE1E8-F7B Call-ID: 569ecc-c0a80129-13c4-3e951-f479ae1-4390(a)196.220.57.58 CSeq: 2 BYE Content-Length: 0 No. Time Source Destination Protocol Info 17290 59.096162 196.220.57.58 81.37.111.201 SIP Status: 481 Call Leg/Transaction Does Not Exist Internet Protocol, Src Addr: 196.220.57.58 (196.220.57.58), Dst Addr: 81.37.111.201 (81.37.111.201) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 481 Call Leg/Transaction Does Not Exist Message Header Via: SIP/2.0/UDP 192.168.1.41:5060;received=81.37.111.201;branch=z9hG4bK-3e97b-f483cd0-e16 From: "777910020"<sip:777910020@196.220.57.58>;tag=c0a80129-13c4-3e951-f479ae6-6c8e To: <sip:607628000@196.220.57.58>;tag=2A3EE1E8-F7B Call-ID: 569ecc-c0a80129-13c4-3e951-f479ae1-4390(a)196.220.57.58 CSeq: 2 BYE Content-Length: 0 ############################################################################## One more thing, when looking at ser debug, after receiving the ACK from SIP-PHONE a line shows the following message: DEBUG : sl_filter_ACK: to late to be a local ACK! Please any help will be appreciated. Thanks.

3 4

[Users] cvs tags and branches
by Klaus Darilion 17 Jun '05

17 Jun '05

Hi! Can someone please describe the CVS tags/branches and their relation with ser? What is the difference between MAIN and HEAD? thanks, klaus

2 3

[Serusers] freeware MCU
by Rodríguez García, Marcos 17 Jun '05

17 Jun '05

Hi, Any one now any free software that makes the functions o a MCU for voice/video multiconferences with SIP? Thanks, Marcos

1 0

[Users] new version openser how to deal with call a SIP uri with multi registar
by jimmy way 17 Jun '05

17 Jun '05

Hi all, The question is: 1. a user A registar from multi address, like UA1 registrar from address1 UA2 registrar from address2 UA3 registrar from address3 and so on so in db "location" there are multi address in the list about user A 2. if set append_branch as on in registrar module, when user B invite user A, Openser server will append all these address in the invite packet. So all these user will get the call. normally all UAs ringing, then 3. if one of these UAs accept the call, how about other UAs, will they stop ringing? 4. if one of these UAs deny(40x) the call, how about other UAs, will the still ringing? 5. if one of these UAs not connected, how about caller the callee do? tks. Jimway ____________________________________________________ Yahoo! Sports Rekindle the Rivalries. Sign up for Fantasy Football http://football.fantasysports.yahoo.com

2 1

[Users] mailing list archive does not work
by Klaus Darilion 17 Jun '05

17 Jun '05

regards, klaus

3 2

RE: [Serusers] pa cannot work as a B2BUA?
by Simith Nambiar 17 Jun '05

17 Jun '05

Hi mma, I use windows mesenger 4.6 , it has the Expires header ! ================================================================ SUBSCRIBE sip:pramukh@10.10.100.44 SIP/2.0 Via: SIP/2.0/UDP 10.10.100.242:8385 Max-Forwards: 70 From: "amith_nambiar" <sip:amith_nambiar@10.10.100.44>;tag=bdc8933dec424822be3b3d671f3660a6;epid=d 9d835a5ca To: <sip:pramukh@10.10.100.44> Call-ID: c454b0c6f13f46cc90d45be8f968d1d3(a)10.10.100.242 CSeq: 2 SUBSCRIBE Contact: <sip:10.10.100.242:8385> User-Agent: RTC/1.2 Expires: 0 Event: presence Content-Length: 0 ================================================================ I get a parse error in the function handle_subscription ! I'am trying to debug why it is giving me that error, using a debugger to step thru code, have found a problem, but i'am not 100 % sure! cheers. Simith -----Original Message----- From: mma [mailto:mmouhiha@softiphone.net] Sent: Thursday, June 16, 2005 3:46 PM To: Simith Nambiar; 'Jamey Hicks' Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] pa cannot work as a B2BUA? Hi I think the pb is that the SUBSCRIBE requests that sends Windows Messenger does not contain an "Expires" header as defined in RFC3265 ftp://ftp.rfc-editor.org/in-notes/rfc3265.txt Regards ----- Original Message ----- From: "Simith Nambiar" <simith(a)indtelesoft.com> To: "'Jamey Hicks'" <jamey.hicks(a)hp.com>; "mma" <mmouhiha(a)softiphone.net> Cc: <serusers(a)lists.iptel.org> Sent: Tuesday, June 14, 2005 6:14 AM Subject: RE: [Serusers] pa cannot work as a B2BUA? > *********************** > Pas de virus detecté dans l'attachement no filename > > Votre Mail a été vérifié par InterScan. > ***********-*********** > > > > -----Original Message----- > From: Jamey Hicks [mailto:jamey.hicks@hp.com] > Sent: Monday, June 13, 2005 11:23 PM > To: mma > Cc: serusers(a)lists.iptel.org > Subject: Re: [Serusers] pa cannot work as a B2BUA? > > > mma wrote: > > > hi, > > > > i read in pa module that it cannot subsribe to presence of users > > using SIP, > > so how can i do to manage the presence of my contact list > > > PA still does not have support to SUBSCRIBE to contacts. What you can > do is set up PA so that it only handles presence for users who register > against and publish presence information to your SER server, and to > forward subscribe requests belonging to other registrars or UAs. > > > i added pa module to SER and a had this errors in response to a > > SUBSCRIBE request > > > > parse_hfs(): Error while parsing headers > > > I would have to see the message to be able to tell what is wrong. Which > UA was generating the subscription? Which version of SER? > > -Jamey > > Hi Jamey, > I have a similar Error , when using MSN Messenger Ver 4.6, and > Ser-0.9.0 , > Please let me know if you have any hints to look for problems ! > > Cheers. > Simith > > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > ______________________________________________________________________ > This email has been scanned via a message filtering facility. > If you have any issues with your email, please contact the IT Dept. > ______________________________________________________________________ This email has been scanned via a message filtering facility. If you have any issues with your email, please contact the IT Dept.

1 0

[Serusers] Problem with cancel a invite
by Anders Hansson 17 Jun '05

17 Jun '05

Hi all, I´ve got a problem when trying to cancel an invite. The situation: A makes a call to B (INVITE). B press the "hang up" button. An here comes the problem. From As side, everything is OK, but from Bs side, the phone is still ringing... How can a solve this problem? Anyone? Thanks, Anders

5 8

[Serusers] MediaProxy No Sound
by TCCHAN 17 Jun '05

17 Jun '05

Dear All, I found out that the following: (1) if I comment out the use_media_proxy in the on_reply_route, I got half of the stream working, ie. the called party could hear the caller party. Sessions will dump a seesion as before. (2) if I comment out the use_media_proxy in the if (method=="INVITE") in main route, I got the whole sessions working, ie called party can hear caller and vice versa. But sessions didn't get a session dumped, even mediaproxy don't show any activities, what happened? ( my device are behind NAT, how could they hear if mediaproxy is not working, and if mediaproxy is working, how come there is not print out on mediaproxy and sessions show No RTP sessions) (3) If I use use_media_proxy on both the main route and the on_reply_route (which is as shown on the example ser.cfg in the mediaproxy module) I 've got nothing, both the caller and called party can't hear anything. But seesions shown a session. If seems that I can't call use_media_proxy twice or something like that, but I really not quite sure. Any suggestion? Thank for any help available. Regards, TC Chan On Thu, 2005-06-16 at 12:22 +0200, harry gaillac wrote: > Hello, > > Look at your onreply_route in ser.cfg . > > Harry > > > #debug=3 #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" #-------------------------------------------------------------------------------- #TCN: Need this line for ver 0.9.0 fifo_db_url="mysql://ser:heslo@localhost/ser" #-------------------------------------------------------------------------------- # ------------------ module loading ---------------------------------- #loadmodule "/usr/local/lib/ser/modules/nathelper.so" 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/xlog.so" loadmodule "/usr/local/lib/ser/modules/exec.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/group.so" loadmodule "/usr/local/lib/ser/modules/textops.so" #loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" #---------------------------------------------------------------------- # mediaproxy modparam("mediaproxy", "natping_interval", 60) #modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy", "mediaproxy_socket", "/var/run/proxydispatcher.sock") modparam("registrar", "nat_flag", 2) #---------------------------------------------------------------------- #---------------------------------------------------------------------- # uri #modparam("uri", "db_url", "sql://root:heslo@localhost/ser") #modparam("uri", "uri_table", "subscriber") #---------------------------------------------------------------------- #---------------------------------------------------------------------- # TCN: no forking #modparam("registrar","desc_time_order",1) modparam("usrloc","desc_time_order",1) modparam("registrar","append_branches",0) #----------------------------------------------------------------------- # ----------------- setting module-specific parameters --------------- modparam("usrloc", "db_mode", 1) #modparam("usrloc", "db_mode", 2) #------------------------------------------------------------------------------------- #TCN: To chnage the database loc to remote PC change the following lines #grep -r 'db_url' sersrc review that the following modules need to be changed: # group,vm,usrloc,uri,domain,jabber,auth_db,pdt,acc,msilo #modparam("auth_db", "db_url", "sql://root:heslo@219.94.42.174/ser") #modparam("usrloc", "db_url", "sql://root:heslo@219.94.42.174/ser") #modparam("group", "db_url", "sql://root:heslo@219.94.42.174/ser") modparam("acc", "db_url", "mysql://root:heslo@localhost/ser") modparam("auth_db", "db_url", "mysql://ser:heslo@localhost/ser") modparam("usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("group", "db_url", "mysql://ser:heslo@localhost/ser") #------------------------------------------------------------------------------------- modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 30 ) ################### NAT ##################### # We will you flag 6 to mark NATed contacts #modparam("registrar", "nat_flag", 6) # Enable NAT pinging #modparam("nathelper", "natping_interval", 3) # Ping only contacts that are known to be # behind NAT #modparam("nathelper", "ping_nated_only", 1) ############################################## #---------------------------------------------------------------------- # accounting #modparam("acc", "log_flag", 2) #modparam("acc", "db_flag", 2) #---------------------------------------------------------------------- alias="redtone" alias="redtone.sip" alias="redtone.sip.com" alias="redtone.sip.com.my" # ------------------------- request 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"); xlog ("L_INFO","-----------------------------------------------------------------\r\n\r\n\r\n\r\n"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too Large"); xlog ("L_INFO","-----------------------------------------------------------------\r\n\r\n\r\n\r\n"); break; }; if (method=="NOTIFY") { break; } if (method=="REGISTER") { xlog("L_INFO","xlog %Tf: REGISTER route uri:%ru method:%rm\r\nfrom: %fu To:%tu Contact:%ct\r\n\r\n\r\n"); #if (is_from_local()) #{ xlog("L_INFO","xlog %Tf: FROM LOCAL REGISTER route uri:%ru method:%rm\r\nfrom:%fu To:%tu Contact:%ct\r\n\r\n\r\n"); # Mark as NAT'ed if (client_nat_test("3")) { #setflag(2); force_rport(); fix_contact(); }; xlog("L_INFO","xlog %Tf: start REGISTER route uri:%ru method:%rm\r \nfrom:%fu To:%tu Contact:%ct\r\n\r\n\r\n"); if (!www_authorize("redtone", "subscriber")) { xlog("L_INFO","%fu failed authentification!\r\n"); www_challenge("redtone", "0"); xlog("L_INFO","%fu failed authentification and challenge!\r\n"); xlog ("L_INFO","-----------------------------------------------------------------\r\n\r\n\r\n\r\n"); break; }; save("location"); xlog("L_INFO","username %fu location REGISTERed!\r\n"); xlog ("L_INFO","-----------------------------------------------------------------\r\n\r\n\r\n\r\n"); break; /* } else { xlog("L_INFO","This domain is not served here!\r\n"); sl_send_reply("403", "This domain is not served here"); }; */ break; }; /* if (method=="INVITE") { if (!(is_from_local() || is_uri_host_local())) { sl_send_reply("403", "Relaying is forbidden"); break; }; t_on_failure("1"); } else */ if (method == "BYE" || method == "CANCEL") { xlog("L_INFO","end_media_session uri:%ru method:%rm \r\nfrom:%fu To:%tu Contact:%ct\r\n"); end_media_session(); }; /* if (loose_route()) { if (method=="INVITE" || method=="ACK") { use_media_proxy(); }; # end media session for BYE and CANCEL is done above # before entering the loose route. no need to call it here t_relay(); break; }; */ # Force subsequent messages to pass trough this proxy if (method == "INVITE") { xlog("L_INFO","invite record route uri:%ru method:%rm \r\nfrom:%fu To:%tu Contact:%ct\r\n"); record_route(); }; if (client_nat_test("3") && !search("^Record-Route:")) { # Mark as NAT'ed xlog("L_INFO","force port and fixcontact uri:%ru method:%rm \r \nfrom:%fu To:%tu Contact:%ct\r\n"); force_rport(); fix_contact(); }; if (method=="INVITE") { xlog("L_INFO","set t_onreply uri:%ru method:%rm \r\nfrom:%fu To:%tu Contact:%ct\r\n"); t_on_reply("1"); setflag(6); }; # if (is_uri_host_local()) { lookup("aliases"); if (!lookup("location")) { sl_send_reply("404", "User not found"); break; }; # }; #if (method=="INVITE" || method=="ACK") { if (method=="INVITE") { #if (method=="ACK") { xlog("L_INFO","use mediapproxy uri:%ru method:%rm \r\nfrom:%fu To:% tu Contact:%ct\r\n"); use_media_proxy(); }; xlog("L_INFO","relay out\r\n"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } failure_route[1] { end_media_session(); } onreply_route[1] { xlog("L_INFO","on reply uri:%ru method:%rm \r\nfrom:%fu To:%tu Contact:%ct\r\n"); #if (status=~"(183)|(2[0-9][0-9])") { if (status=="200") { xlog("L_INFO","REPLY OK uri:%ru method:%rm \r\nfrom:%fu To:%tu Contact:%ct\r\n"); #if (client_nat_test("1")) { fix_contact(); #}; if (!isflagset(7)) { use_media_proxy(); setflag(7); } }; }

1 0

[Serusers] rtpproxy and serctl with problem
by Bruno Oliveira 17 Jun '05

17 Jun '05

hi i have rtpproxy runing and serctl. where it is file config of rtpproxy ? and when I type this command "serctl start" show me in /var/log/syslog this message: "Jun 16 14:45:33 uiara /usr/local/sbin/ser[3237]: Maxfwd module- initializing Jun 16 14:45:33 uiara /usr/local/sbin/ser[3237]: INFO: udp_init: SO_RCVBUF is initially 111616 Jun 16 14:45:33 uiara /usr/local/sbin/ser[3237]: INFO: udp_init: SO_RCVBUF is finally 223232 Jun 16 14:45:33 uiara /usr/local/sbin/ser[3237]: ERROR: tcp_init: bind(6, 0x813753c, 16) on 192.168.1.2: Address already in use" and in pront command show this message: "uiara:~# serctl start Starting SER : PID file /var/run/ser.pid does not exist -- SER start failed" when I type "pstree" show me ser and rtpproxy in the list. anyone can a help me! thanks for your time! -- []s, Bruno "Niggas" Oliveira Belo Horizonte - MG Msn: n1gg4s(a)gmail.com Icq: 176314647 "Todo o nosso descontentamento por aquilo que nos falta procede da nossa falta de gratidão por aquilo que temos."

2 1

[Serusers] error: 400; check if you use aliases in SER
by TCCHAN 17 Jun '05

17 Jun '05

Dear All, After switching from 0.8.14 to 0.9.0 I have this problem of getting error: 400; check if you use aliases in SER when I try to add a user with serctl, eg.serctl add 01548900 999 01548900(a)redtone.com I searched the web and knew someone else has posted this question before and I apologize for asking again ( didn't manage to get the answer). Thanks for any help available. Regards, TC Chan

2 1

Re: Fw: [Serusers] voice mail problem on alias
by Rosario Pingaro 17 Jun '05

17 Jun '05

I fixed the problem commenting the revert_uri(), so it call each time the modified uri that i have after lookup("aliases"). very easy after all. What I am not able to preview is the future implication of commenting revert_uri() giving the fact that in all samples it is present. Thanks Rosario ----- Original Message ----- From: "Stefan Sayer" <sayer(a)fokus.fraunhofer.de> To: "Rosario Pingaro" <rpingar(a)italycom.it> Sent: Thursday, June 16, 2005 6:49 PM Subject: Re: Fw: [Serusers] voice mail problem on alias > Hello, > > that is interesting - if you find the answer can you mail me as well? > (I'd like to add that to the documentation.) > > I think it depends on how you read the email adress from the db - probably > you are loading it with something like > avp_db_load( "$ruri", "$email/$email_scheme"); > and if it is not found you'd have to check if it is an alias and lookup > the real name. > > Another option is to use user_prefix_separator, only a single ser > instance, I think with this scheme there is not this problem with the > alias (but i am not sure). > > Stefan > Rosario Pingaro wrote: >> Debugging sems seems that the probelm is related on how ser pass the user >> to sems; >> in fact when I call the alias and the use is ofline ser pass the real >> user to sems. >> when ser get a timeout pass to sems the alias, sems is not able to find >> an email for the alias and so i get the error. >> >> How to workaround this? >> >> Thanks >> Rosario >> >> >> ----- Original Message ----- >> From: Rosario Pingaro To: serusers(a)lists.iptel.org Sent: Thursday, June 16, >> 2005 5:20 PM >> Subject: [Serusers] voice mail problem on alias >> >> >> Hi I am implementing voice mail using ser and sems, ser 0.9.2 and latesr >> sems from cvs. >> >> I have the failure_route[x] to activate the voicemail also when i get an >> invite timeoute for my users. >> failure_route[1] { >> revert_uri(); >> rewritehostport("xxx.xxx.xxx.xxx:5090"); >> append_branch(); >> t_relay_to_udp("xxx.xxx.xxx.xxx", "5090"); >> } >> >> I don't have any problem when i try to leave a voice mail for an "user". >> On the "alias" the voicemail works fine when the user is offline, but >> when it is online and get a timeout sems respond with a 404 no email >> address for user >> >> the fact that the voicemail works on alias when it is off line push me to >> think the problem is into the failure_route. But I don't know what to add >> to get it working. >> >> Any help is welcome. >> >> Thanks >> Rosario >> >> >> >> -------------------------------------------------------------------------------- >> >> >> _______________________________________________ >> Serusers mailing list >> serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >> >> >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Serusers mailing list >> serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers > >

1 0

[Users] free-TLS released in experimental tree
by Cesc 16 Jun '05

16 Jun '05

Hi, This is a notification that the first release of free-TLS for SER has been commited to the experimental tree in the CVS repository for SER. For openSER it will be (i think) committed to the core CVS repository right away. User testing is required to fast-track it into the main core, so I encourage you to give it a try. More will be added soon to ease testing (for example, test certificates ready to use ... but you can also create your own). It contains a README file, which should be enough to get things started. It will be improved with a fully working tls_only.ser.cfg file in no time. Also more developer help will be added in the README, for people who want to collaborate. This is all for now. Regards, Cesc

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Re: [Serusers] RE: Domain module
by harry gaillac 16 Jun '05

16 Jun '05

yes i read it -:) I would have spent less time if i had read it but it's not wasted time . I use SER I have to read OPENSER docs. Something is wrong ! Thanks all for help. Harry --- Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> a écrit : > Harry, > > you may find this useful > > http://www.openser.org/docs/modules/domain.html > also you can find full up2date docs for all modules > there > > bogdan > > harry gaillac wrote: > > >According to domain.c file "is_uri_host_local()" > >function > >is needed to check R-URI. > > > >Harry > > > >--- Juha Heinanen <jh(a)tutpro.com> a écrit : > > > > > > > >>put xlog debug into the script. print from header > >>before the tests, > >>etc. also, wheck with fifo command domain_dump > (or > >>something like that) > >>that domain table is really loaded. > >> > >>-- juha > >> > >> > >> > ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

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[Serusers] Why does added headers show up in the CANCEL?
by Henrik Gustafsson 16 Jun '05

16 Jun '05

Hello! I'm playing around with my own module which is adding or replacing a header in proxied INVITE requests. I'm using the "parse_headers ... anchor_lump ... insert_new_lump_before" method on the proxied request and the outgoing INVITE looks great with the new header. But the request is forwarded by using the tm module and when the transaction receives a CANCEL the outgoing CANCEL gets the header added to the initial INVITE. That would not be a problem for a normal SIP UA, since unknown headers should be ignored, but this UA from a big telecom vendor throws a 400 Bad request. Why is the added headers added to the CANCEL? Shouldn't just the required and transaction identifying headers be enough in a CANCEL? Could I do anything different to prevent this from happening? /Henrik Gustafsson

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[Serusers] Handling CANCELS when a UAC doesn't answer and t_relay has maintained state
by Evan Borgström 16 Jun '05

16 Jun '05

Hello All, Looking for some input if anyone else has run into this situation and/or figured a way around it. - Call comes in from PSTN gateway to a local user - SER finds the user in the location table and t_relay pushes the request off to them - The user is behind an ACL and is not sending keep-alives (yes I know the client is broken) so the INVITE never makes it to them and hence no 180/183 Ringing message comes back - PSTN user gives up and hangs up the phone after 3 seconds of no ringing - PSTN gateway forwards the CANCEL message in and SER cancels the branch - However since t_relay is stateful and re-tries transmissions the INVITE is sent another time or two - Finally the connection times out and the failure_route set for the t_relay kicks in and the INVITE gets pushed off to the voicemail server. - Voicemail server OK's the INVITE and starts retransmitting the OK over and over and over since the PSTN GW doesn't ACK the OK since it knows that call-id as being cancelled. So, should I handle cancel's a little more explicitly by doing something similar to: if ((method=="CANCEL") && (t_lookup_request())) { t_release(); }; What side effects, if any, would this cause? Is there a better way to handle it? Thanks, -Evan

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[Users] Conclusions
by Bogdan-Andrei Iancu 16 Jun '05

16 Jun '05

Folks, There were a lot of ideas, discussions, fears vehiculated in the last days. I try to bring some response as quick and as short as possible just to close the subject. 1) there is and it will be no war - I think the idea to associate war with new thing is in human nature and we cannot fight it. Media Proxy was also welcomed with war and it proved to be a good thing (a complementary solution to nathelper); avpops got the same treatment and now I believe to be a very used module. So let's try t get rid of the war idea - OpenSER is just an alternative to SER and everybody is free to choose what to use. 2) very common idea (no idea who came with it?!) was that OpenSER is less stable and it will be mainly an place for new thing to get mature - wrong ! OpenSER's stability will be one of its strong attribute - if you look to the NEW FEATURE section we will be surprise how many bugs were hunted and fixed. Shortly - we will try to keep OpenSER as stable as SER and maybe more - of course, nothing is bug free :) 3) getting back into SER team - maybe the GCC story will happen, but for this the SER must prove (with facts and not words) that it became a real open project - politics and development are not dictated by any hidden commercial reasons; SER becomes an independent and neutral project (not faked "propriety" of anybody), anybody being free to contribute; the maintainer dictatorship must end - a module maintainer must not have "live or death" right on its modules, it must obey the management group of the project!. 4) documentation management - that a hard problem which frustrated the users a lot - OpenSER tries a different cycle of documentation which will make it easier to use and to update for user: main storage point will be sgml files on CVS -> HTML files on web for reading -> on web wiki page for user to contrib/comment -> filtered time to time by maintainers and moved into sgml; and here ends; easy and straight for both users and maintainers; also Makefiles has built-in commands for generating READMEs and HTMLS from sgml. now, some short answers to some interesting ideas: Greger ------ what was the attitude when "commercial components start popping up with alternative contributions" ? IGNORE! "Do they(voice system) contribute their commercial code to OpenSER? Of course not, they have exactly the same problems as iptelorg.com" Maybe you should look twice to avpops, speeddial, alias_db, uac modules, maintenance for xlog and pdt and all patches to usrloc, rr , register and nathlper - many enterprise solution fixes are there (nat traversal and replication) Kristian Galway, reticent, Maxim, Klaus, Giudace ------------------------------------------------ there is no war and let's forget the word! it's simple matter of choosing and liberty (even if history showed us that all this were gain through war) Everybody is free to use whatever he like. If you don't like OpenSER, fine by me, forget about it, but don't misplace words! Dave, Ingo, Maxim, Atle ------------------------------------------------ by splitting the project nothing is lost, but better (efficient) used; for developers - no more struggle to put your stuff on cvs; more time to work on real things; anyhow you worked was blocked on SER :(. For users - if they know what they want, they can have the liberty to choose based on the preferences, which is a step forward. Let's not hide behind the mentality - simple thing and less options make our life easier... Harry, Samuel ------------- interesting idea of being looser....wonder way?? If like SER you can stick to it and you don't loose or win; If you find OpenSER more suitable for you, you win.....so....where is the looser may I ask?? Cesc ----- |And this process (n. devel) must be well-defined and NOT CONTROLLED exclusively by a company (be iptel or voice-systems, same-o, same-o). If the contribution is voted as desirable by a community of users, it is well tested and so on, it should be accepted." you made a strong point here !! with this email I want to close the subject. Please *do not reply* to it, but only in privately or use openser mailing list if you have related questions. End of subject Bogdam

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[Serusers] sample cfg for ser + asterisk
by Roger Lewau 16 Jun '05

16 Jun '05

Does any one have a sample cfg for a ser + asterisk setup they would like to share? We are planning to use asterisk to provide vm, conferencing and pstn functionality. So any input on this issue would be nice.

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[Serusers] Re: How to execute failure_route block on tcp connect failure.
by Dmitry Isakbayev 16 Jun '05

16 Jun '05

Hello, I have some more information about this issue. There are actually two problems. First - When the 1st attempt fails because connection was refused, connection timed out, or there was no route to host, an error message as logged and the failure_route block is never executed. Second - When the 2nd or 3rd, etc attempt fails due to a connection problem, an error is logged and only 30 seconds later the failure_route block is executed. I would really appreciate if some body could explain what is going on or point me in the right direction. Thank You, Dmitry ----- Original Message ----- From: Dmitry Isakbayev To: serusers(a)lists.iptel.org Sent: Wednesday, June 15, 2005 9:12 PM Subject: How to execute failure_route block on tcp connect failure. Hello, I am having a problem executing a failure_route block when t_relay fails to establish a tcp connection. Everything works as expected when t_relay connects to a proxy and receives back an error like 501. Is there a way to configure t_relay to execute a failure_route block an a tcp connection error? Thank You in advance. Dmitry Here are the logs from connection failure - Jun 15 20:52:14 localhost ser[21870]: ERROR: tcp_blocking_connect: SO_ERROR (111) Connection refused Jun 15 20:52:14 localhost ser[21870]: ERROR: tcpconn_connect: tcp_blocking_connect failed Jun 15 20:52:14 localhost ser[21870]: ERROR: tcp_send: connect failed Jun 15 20:52:14 localhost ser[21870]: msg_send: ERROR: tcp_send failed Jun 15 20:52:14 localhost ser[21870]: ERROR: t_forward_nonack: sending request failed

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Re: [Serusers] peering
by Joao Pereira 16 Jun '05

16 Jun '05

Sorry, but I didnt explained my self corectly, what I want is to redirect the calls that aren't for my domain ( for example, calls to sip:maria@fwd.pulver.com or calls to sip:manel@iptel.org). What I whant to know is what to put after this: if( ! uri=~"@fccn.pt" ) { #redirect to fwd or iptel Thanks João Klaus Darilion wrote: > The suggest way is ENUM, but only a few of them creates ENUM entries > for there users. > > regards, > klaus > > Joao Pereira wrote: > >> Hello to all >> I whant my SIP clients to be able to do calls to any other SIP >> network, and to do It, I writed this line (for the IPTEL.ORG case): >> >> if(uri=~"@iptel.org"){ forward(iptel.org,5060); >> break; >> } >> >> But this way, I need to put theese 4 lines to all VoIP providers. >> Is there any way of saying: >> if( ! uri=~"@fccn.pt" ) { # if the invite isnt for my domain (fccn.pt) >> send it to the correct domain >> >> Thanks >> >> João Pereira >> >> >> >> _______________________________________________ >> Serusers mailing list >> serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >> >> > >

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[Serusers] Problems with ser and Micronet SP5012
by gsalvucci＠ing.unrc.edu.ar 16 Jun '05

16 Jun '05

Hi, I´m using ser-0.8.14 in a LAN with the ser.cfg from the distribution. I have many ATAs and softphones(x-lite) and a PSTN gateway Micronet SP5012 connected to PBX. Problem is the following: When I call to a PSTN number everything works fine but when call finishes the port FXO of the gateway stays busy. Thanks Gustavo. ---------------------------------------------------------------- Este mensaje ha sido enviado usando el servicio de Webmail de la Universidad Nacional de Río Cuarto. Basado en Horde/IMP

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[Serusers] peering
by Joao Pereira 16 Jun '05

16 Jun '05

Hello to all I whant my SIP clients to be able to do calls to any other SIP network, and to do It, I writed this line (for the IPTEL.ORG case): if(uri=~"@iptel.org"){ forward(iptel.org,5060); break; } But this way, I need to put theese 4 lines to all VoIP providers. Is there any way of saying: if( ! uri=~"@fccn.pt" ) { # if the invite isnt for my domain (fccn.pt) send it to the correct domain Thanks João Pereira

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[Serusers] ser , its forks and knives, and docs, and everything else
by Iqbal 16 Jun '05

16 Jun '05

Hi Another day of going back and forth, feel sorry for the poor newbie who might have signed up for this mailing list, and realized he'd was standing in no mans land between too factions who had dug deep into their trenches.... I cant recall the names of who posted what, and who wants what, but I think a summation, together with some sort of action towards an outcome, is really needed, so I'll try, and please add to it. 1. There is a definant call for two modes, one stable and one more fast paced for those who like to try out new things a) A mode where these two can be linked is required, someone suggested maintaining two threads in CVS, b) do you need to mailing lists, yes/no c) Do you need two websites yes/no d) Should iptel be inherently linked with SER, as most users think, or should their be a non-biased site, I think since SER has its own name, why not use something based upon that, just as other opensource has, if companies wish to sponser, add there name etc, then they should be allowed by the maintainers discretion 2. Documentation, yes it is poor, but that means it can only get better , or at least the way I look at it, a)Who maintains it, is there a overwhelming need for help on this, if so, can we not on the website above, include a link for those who would like to help contribute, and then set some standards as to how it needs to be done etc b) Informal documentation, I think sites such as onsip etc have started something good, and many users will eventually have their own websites etc, with tips and tricks, however if this could also be added to in one place it would be good for the newbie, maybe a wiki, maybe onsip. 3. Working together, now this is a tricky one, since no matter how we look at it, SER seems to be splitting (unless I am misreading the posts), which means even in theory if we want to make it work like i did, its just wont, and we need to accept that, its growing and people want/wish to do it there way, and I guess the compromise is in 1, above. No matter how many emails go back and forth, going back to what it was is not possible, cause human nature will not allow it. 4. Justification and blame, well I think alot of that has happened, and as a user, since I dont know all the details I cannot really comment, except to say, that I am sure its all justified, and everyone is too blame :-), but its time too move on, it was a project and like all projects starts a little chaotically, and somewhere down the line a radical shake up needs to be done, this also happens in companies, trust me I have built quite a few, the casual working modes are dropped for a formal approach, suddenly hand written scraps of paper are no longer acceptable, everything needs to have 3 signatures on it...such is life. I guess this is what SER is going through, and maybe this is what it needs, who's fault it is, or who realized a change is required is irrelevant, but without change I think SER will not work, simply because some of the core team want a change, and you cannot argue with that. Soooo write down what is needed take a vote, ask a few users what they think, and lets get it done. As a side note, I hope the above doesn't seem biased one way or the other, it is not meant to, its meant to get things moving and a decision reached, before everyone goes on holiday, and starts smoking a pipe of peace....without having invited me. Iqbal

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[Serusers] forwarding to voicemail (SEMS)
by Llanos Serna García-Conde 16 Jun '05

16 Jun '05

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[Serusers] Domain module
by harry gaillac 16 Jun '05

16 Jun '05

Hello, I checked my ser.cfg my table domain in db ser and re-read the documentation of domain module . After reading domain.c file I found a exported function for R-URI which is not is_from_local() for outbound domain but is_uri_host_local(): I add this function in my conf file now i can call others domains however we don't find this function anywhere else that domain.c I use documentation from onsip.org which talk about nat however the examples may be wrong or incomplete. Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

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[Serusers] RE: Domain module
by harry gaillac 16 Jun '05

16 Jun '05

According to domain.c file "is_uri_host_local()" function is needed to check R-URI. Harry --- Juha Heinanen <jh(a)tutpro.com> a écrit : > put xlog debug into the script. print from header > before the tests, > etc. also, wheck with fifo command domain_dump (or > something like that) > that domain table is really loaded. > > -- juha > ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

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Fw: [Serusers] voice mail problem on alias
by Rosario Pingaro 16 Jun '05

16 Jun '05

Debugging sems seems that the probelm is related on how ser pass the user to sems; in fact when I call the alias and the use is ofline ser pass the real user to sems. when ser get a timeout pass to sems the alias, sems is not able to find an email for the alias and so i get the error. How to workaround this? Thanks Rosario ----- Original Message ----- From: Rosario Pingaro To: serusers(a)lists.iptel.org Sent: Thursday, June 16, 2005 5:20 PM Subject: [Serusers] voice mail problem on alias Hi I am implementing voice mail using ser and sems, ser 0.9.2 and latesr sems from cvs. I have the failure_route[x] to activate the voicemail also when i get an invite timeoute for my users. failure_route[1] { revert_uri(); rewritehostport("xxx.xxx.xxx.xxx:5090"); append_branch(); t_relay_to_udp("xxx.xxx.xxx.xxx", "5090"); } I don't have any problem when i try to leave a voice mail for an "user". On the "alias" the voicemail works fine when the user is offline, but when it is online and get a timeout sems respond with a 404 no email address for user the fact that the voicemail works on alias when it is off line push me to think the problem is into the failure_route. But I don't know what to add to get it working. Any help is welcome. Thanks Rosario -------------------------------------------------------------------------------- _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[Serusers] Re: problem in ser
by Kamran Ahmad 16 Jun '05

16 Jun '05

hello i was using ser-0.9.0 with radiusclient, freeradisus, but unable to configure on this mechine. I am already using these kind of setup on my fedora core mechine with no problem. but i have many problems on this mechine. i am using same ser.cfg as my successfull setups but no success. now i am using ser-0.8.0 with radiusclient.0.4.8 with kernel 2.6.9-1.667 after lot of compiliation i am able to compile auth_radius.so but now i am getting 0(4156) auth module - initializing 0(4156) find_export: found <sl_send_reply> in module sl_module [/usr/local/lib/ser/modules/sl.so] 0(4156) DEBUG: init_mod: auth_radius 0(4156) auth_radius - Initializing 0(4156) auth_radius: No `Cisco' vendor in Radius dictionary 0(4156) find_export: found <pre_auth> in module auth [/usr/local/lib/ser/modules/auth.so] 0(4156) find_export: found <post_auth> in module auth [/usr/local/lib/ser/modules/auth.so] 0(4156) ERROR: auth_radius: can't get code for the Digest-Response attribute 0(4156) init_mod(): Error while initializing module auth_radius ERROR: error while initializing modules /var/log/messages ----------------- Jun 9 18:48:50 voice ser: WARNING: could not rev. resolve 217.8.112.104 Jun 9 18:48:50 voice ser[4153]: ERROR: auth_radius: can't get code for the Digest-Response attribute Jun 9 18:48:50 voice ser[4153]: init_mod(): Error while initializing module auth_radius __________________________________ Discover Yahoo! Find restaurants, movies, travel and more fun for the weekend. Check it out! http://discover.yahoo.com/weekend.html

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[Users] LCR module
by Bogdan-Andrei Iancu 16 Jun '05

16 Jun '05

Hi folks, As more an more people are asking for a LCR module in 0.9.4 release, I prepared an official backport of the module for OpenSER 0.9.4 (sources): http://www.openser.org/pub/openser/0.9.4/src/lcr-opsenser-0.9.4.tgz Also there is more and more positive feedback regarding the stability of the module and we are looking for including it in a second fast release on 0.9 branch - a 0.9.5 which will additionally bring the LCR module. Any opinions on this? regards, bogdan

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[Serusers] lost in cvs tags - difference between 0.9.3 and rel_0_9_0
by Klaus Darilion 16 Jun '05

16 Jun '05

Hi! I'm lost in the name of new ser versions and the CVS tags. There is talk about version 0.9, 0.9.2 and 0.9.3, although I only found rel_0_9_0 on cvs. What is the relation between this versions and the CVS tag. thanks, klaus

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[Serusers] voice mail problem on alias
by Rosario Pingaro 16 Jun '05

16 Jun '05

Hi I am implementing voice mail using ser and sems, ser 0.9.2 and latesr sems from cvs. I have the failure_route[x] to activate the voicemail also when i get an invite timeoute for my users. failure_route[1] { revert_uri(); rewritehostport("xxx.xxx.xxx.xxx:5090"); append_branch(); t_relay_to_udp("xxx.xxx.xxx.xxx", "5090"); } I don't have any problem when i try to leave a voice mail for an "user". On the "alias" the voicemail works fine when the user is offline, but when it is online and get a timeout sems respond with a 404 no email address for user the fact that the voicemail works on alias when it is off line push me to think the problem is into the failure_route. But I don't know what to add to get it working. Any help is welcome. Thanks Rosario

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[Serusers] sipath, ser on openwrt
by Klaus Darilion 16 Jun '05

16 Jun '05

Hi all! I just found this project: http://sipath.sourceforge.net/ It uses ser+rtpproxy as outboundproxy on openwrt. Looks like the modified the code - the REGISTER will be stored locally, the contact will be re-written and the REGISTER will be forwarded to the real proxy. regards, klaus

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[Users] CVS head
by Bogdan-Andrei Iancu 16 Jun '05

16 Jun '05

Hi folks, the CVS head now is up-to-date. All new changes (from 0.9.0 to head) from SER were integrated into OpenSER cvs head. LCR and UAC modules are also there. Feel free to give it a try. now the CVS head is ready for forward developing :) bogdan

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RE: [Serusers] OpenSER release
by Giudice, Salvatore 16 Jun '05

16 Jun '05

I would have to agree with you. The ser module documentation is definitely in the 'abysmal' category, although it has improved somewhat in the last 18 months. For example, at least some documentation on modules like registrar now tell you that the save function returns a 200 ok. To the best of my recollection, this sort of detail wasn't there 18 months ago. If openser can produce decent docs, I can see it getting much broader support. -----Original Message----- From: Kristin Galway [mailto:klgalway@yahoo.com] Sent: Wednesday, June 15, 2005 7:31 PM To: info(a)beeplove.com; greger(a)teigre.com; Giudice, Salvatore; serdev(a)lists.iptel.org; serusers(a)lists.iptel.org Subject: Re: [Serusers] OpenSER release This is my first posting to this list but I have been following for about 5 months now. I began to use SER back in February and have used versions from 0.8.14 to cvs versions and have seen plenty of changes here and there. The biggest thing that really troubles me is the almost non-existent documentation for the modules. There has been more than one occasion where I have run into problems to check the documentation to find no mention of moving things from one module to another, or other things such as new features and what not. Most times I have to dig through the source code to actually find information. The sum of this all is that if the fork of OpenSER is going to present itself as a moving package, with up-to-date documention and more so, not having to wait months upon months for a request or a fix to occur. If OpenSER has that intention then I am all for supporting it and using the code. Just because the current code works for some, does not mean that the rest of us users have to wait for something to happen. Kristin Chapter 2. Developer's Guide To be done. _________________________________________________________ Chapter 3. Frequently Asked Questions 3.1. What is the meaning of life ? 3.1. What is the meaning of life ? 42 --- "m36828253-1(a)imap.1and1.com" <m36828253-1(a)imap.1and1.com> wrote: > Why the bug tracking page in a different website. > Why not under iptel.org ? > > Mohammad > > > Original Message: > ----------------- > From: Greger V. Teigre greger(a)teigre.com > Date: Wed, 15 Jun 2005 21:03:21 +0200 > To: Salvatore.Giudice(a)FMR.COM, serdev(a)lists.iptel.org, > serusers(a)lists.iptel.org > Subject: Re: [Serusers] OpenSER release > > > I completely agree with you. I have been told that > there was an attempt at > introducing a bug-tracking system earlier, but that > it has been difficult. > Anyhow, in setting up policies and procedures around > the experimental > directory, we have decided that usage of > http://bugs.sip-router.org will be > mandatory. Hopefully recent, better integration > between the bug tracking > system and the CVS will make it more convenient to > use also for other CVS > modules (however, I don't have a say there). > g-) > > Giudice, Salvatore wrote: > > I am not an advocate for either ser or openser, > but I would like to > > comment. > > > > Is openser going to be equipped with a > forum/ticket system where > > people can document bugs, feature requests, etc > (non-configuration > > issues)? > > > > This is just my observation and you may not agree, > but I believe this > > project could be much better maintained if it used > a more structured > > ticketing style system to manage development > issues instead of the > > current mailing lists. In my experience, mailing > lists like this > > foster a terrible user experience where many > development issues can > > go on without response. > > > > Ideally, if there was a mailing list to address > user issues and > > ticketing system like the one Digium uses to > manage Asterisk, I think > > everyone would benefit by being better informed > and ser would > > ultimately be a better product for it. How many > people out there feel > > that their issues have fallen through the cracks > in the past couple > > years? > > > > -----Original Message----- > > From: Daniel-Constantin Mierla > [mailto:daniel@voice-system.ro] > > Sent: Wednesday, June 15, 2005 4:28 AM > > To: Andrei Pelinescu-Onciul > > Cc: SER developer mailing list; serusers; > users(a)openser.org; > > devel(a)openser.org > > Subject: Re: [Serusers] OpenSER release > > > > On 06/14/05 23:21, Andrei Pelinescu-Onciul wrote: > > > >> On Jun 14, 2005 at 22:48, Daniel-Constantin > Mierla > > <daniel(a)voice-system.ro> wrote: > >> > >> [...] > >> > >> > >>> It is your opinion, but I repeat myself, that > the SER code > >>> maintained by us will go further -- I don't > think that someone can > >>> claim that we didn't do the job for our code > (the only discrepancy > >>> is some last-minute adds in xlog (to print avps) > - will be > >>> committed on unstable very soon > > > >>> with the new color patch). The cvs was created > just to ease the > >>> maintainance. The patches would be a nightmare. > >>> > >>> > >> > >> Maybe I've misunderstood you: is this only a > parallel "stabilized" > >> version + some features or is it a full fork (do > you intend to fork > >> unstable also)? > >> > >> > > It is fork for the code that we changed (acc > module, usrloc module > > ...), > > > > in the future may be other that they do not find > the path in SER. We > > will maintain and upgrade our part of code from > SER continuously. > > > >> I have no problem with another stable version, > what worries me is > >> fragmenting the development for unstable (which > is the place where > >> major changes are made). > >> > >> > > I see no fragmenting there -- the situation is the > same for SER as it > > was before. For example, there is no fragment for > acc module, it will > > be > > > > maintained by who did it till now, adding what he > considers necessary > > there. But we came to meet a lot of requests of > why the acc patch is > > not > > > > included in the CVS (it was fully backward > compatible and had new > > features requested by many SER users) and we want > to promote _more > > open_ > > > > approach to contributions to all parts of code. > The acc patch was sent > > on November 1, 2004. No real response (neither > negative, nor positive) > > from maintainer to the submission since then ... > are you aware of a > > good > > > > reason?!?! ... should we wait just about (or more) > half an year for > > each > > > > contribution?!? I will not do that anymore!!! > > > > Daniel > > > >> > >> Andrei > >> > >> > >> > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > -------------------------------------------------------------------- > mail2web - Check your email from the web at > http://mail2web.com/ . > > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > __________________________________ Discover Yahoo! Stay in touch with email, IM, photo sharing and more. Check it out! http://discover.yahoo.com/stayintouch.html

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RE: [Serusers] Re: [Serdev] openser/ser - avoiding forks
by Giudice, Salvatore 16 Jun '05

16 Jun '05

So 31 submitted patches/feature requests were basically ignored in the past year and a half? How can this situation be improved? Is there going to be a commitment to monitor this new ticketing system? How many patches and more feature requests will need to be reconciled 18 months from now? -----Original Message----- From: Jan Janak [mailto:jan@iptel.org] Sent: Wednesday, June 15, 2005 8:57 PM To: Andrei Pelinescu-Onciul Cc: serdev(a)lists.iptel.org; serusers(a)lists.iptel.org Subject: [Serusers] Re: [Serdev] openser/ser - avoiding forks Hello, I am trying to reconcile patches and improvement suggestions that have been left unanswered on the mailing lists. I went through serusers and serdev mailing list archives from the beginning of 2004 till now. I was mainly focusing on submitted patches and non-trivial feature requests. Below is a list of issues that I was able to find in the mailing list archives. Given the number of messages that were sent to the lists in the last year I am pretty sure that the list below will miss many suggestions. If you submitted a patch or improvement and it was neither accepted nor rejected, and it is still relevant, please resubmit your proposal to serdev(a)lists.iptel.org mailing list or (preferably) create an issue in the bug tracking system at: http://bugs.sip-router.org Please re-submit also improvements sent in individual messages to developers. Summary ------- When it comes to accepting improvements, there surely is space for improvements but the situation, in my opinion, is not as bad as it may seem from the recent discussions. Properly reported bugs get usualy fixed quickly. The more detailed description the faster the fix, so I think there are no big problems with that. The list at the end of this message contains patches and feature requests that I was able to find in the archives. From the list the following issues have not yet been closed: 1, 4, 7, 18, 19, 21, 22, 23, 24, 25, 30, 31 And from those only 1, 4, 18, 21, 22, 25, 31 contained patches that have been left unanswered. That 7 patches which still need to be processed. The following patches have in my opinion low importance: 1 - date header in REGISTER replies 18 - radius fix for acc 21 - check_to for digest credentials without database The following patches can be classified as important: 4 - AVP support in acc 22 - transactional auth replies 25 - Free TLS 31 - Xten improvements of pa In addition to that there are the following improvement suggestions which did not contain any patches: 7, 19, 23, 24 So I was able to find 4 important and 3 less important patches that were not integrated in the last 1.5 year from the 31 items below. Also I created issues in the bug tracking system for all items below that are not yet closed. The winner among them in terms of e-mail volume is free TLS, of course. Jan. PS: Code back-porting is another issue and is not covered here. ------------------------------------------------------------------------ ------ SERUSERS - 2004 --------------- 1) Date Header in REGISTER responses Feature request by TeleSIP, I have a patch for that already from Robert Sanders which will be integrated Status: OPEN Bug: SER-30 2) PA interoperability with RTC, patch submitted by Klaus Darilion, integrated by Jamey HICKS Status: CLOSED 3) branch=0 problem reported on 16 Jul 2004, closed by Jiri as "not a bug" Status: CLOSED 4) AVP patch for acc module, submitted by Ramona on 31 Oct 2004 Status: OPEN Bug: SER-31 SERDEV - 2004 ------------- 5) Video - related patch for nathelper, integrated by Maxim Status: CLOSED 6) fix_nated_contact for nathelper (replied by Maxim) Status: CLOSED 7) 16 Apr 2004 Juha proposed adding npdi and pstn URI parameters (no patches) Status: OPEN Bug: SER-32 8) Maxim submitted patch adding PIDs of all processes into the pid file, resolved by another means Status: CLOSED 9) 20 Apr. 2004 Alexander Mayhofer submitted patch for rtpproxy, commited by Maxim Status: CLOSED 10) Test for realm in auth_radius too tight, reported by Cesar Hernandez, fixed by Jan Status: CLOSED 11) 23 Apr 2004 Multicast support patch, commited by Andrei Status: CLOSED 12) 24 Apr 2004 Postgres patch by Alexander Mayhofer, in CVS Status: CLOSED 13) max_expires patch submitted by Jamey Hicks, in CVS Status: CLOSED 14) start/stop commands fro SEMS into serctl by Klaus Darilion, rejected. Status: CLOSED 15) 30 Jun 2004 User agent support in xlog module by Alexander Mayhofer, in cvs Status: CLOSED 16) 7 Jul 2004 Patch for configurable user agent string by Maxim, resolved by other means Status: CLOSED 17) 21 Jul 2004 Patch for storing user agent string in location table, commited by Maxim Status: CLOSED 18) 21 Jul 2004 Radius-related patch for acc module Status: OPEN Bug: SER-33 19) 2 Oct 2004 Escaped uri characters are not interpreted properly, no patch Status: DEFERED Bug: SER-34 20) 27 Oct 2004 Mysql_ping patch by Dan Pascu, in cvs Status: CLOSED 21) 10 Nov 2004 uri_db patch to make it possible to use check_to and check_from without database (submitted by Marian Dumitru). Status: OPEN Bug: SER-35 22) 25 Dec 2004 Maxim's patch to make it possible send transactional replies from authentication modules Status: OPEN Bug: SER-36 23) Additional header fields requested by Juha (server, refer-to) Status: OPEN Bug: SER-37 SERUSERS - 2005 --------------- 24) feature request route("string") Status: DEFERED Bug: SER-38 25) Free TLS Status: OPEN Bug: SER-39 26) 19 Apr 2005 branch=0 in ACK problem, rejected by Jiri Status: CLOSED 27) 20 Apr 2005 mysql_ping patch for 0.8.14, not important enough to be commited to 0.8.14 Status: CLOSED 28) Backport of radius auth changes from unstable to stable, rejected Status: CLOSED SERDEV - 2005 ------------- 29) Transaction replies in auth modules (by Maxim) Status: OPEN Bug: SER-36 30) NAT support in usrloc and registrar, partially done Status: OPEN Bug: SER-40 31) RFC3265,RFC3903 support in pa, I could not find the patch Status: OPEN Bug: SER-41 _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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RE: [Serusers] OpenSER release
by Giudice, Salvatore 16 Jun '05

16 Jun '05

Maybe extending the documentation with a Wiki would prove effective without putting too much responsibility on any one maintainer's shoulders. This would also cut down on the number of repeat questions on the SER user's list. -----Original Message----- From: Maxim Sobolev [mailto:sobomax@portaone.com] Sent: Thursday, June 16, 2005 4:30 AM To: Alberto Cruz Cc: Greger V. Teigre; SER developer mailing list; Giudice, Salvatore; serusers Subject: Re: [Serusers] OpenSER release On Wed, Jun 15, 2005 at 10:57:06PM -0500, Alberto Cruz wrote: > And what about documentation? When was the last time that Administration > Guide was updated? And what about the Modules documentation? Are you volunteering? -Maxim

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[Serusers] storing voice messages
by Llanos Serna García-Conde 16 Jun '05

16 Jun '05

Hi all, I'm using SEMS and it works fine. I know it stores the voices messages in /tmp directory for a very short time before sending them to the e-mail addresses. I would like to store them permanently, how can I do that? Thanks for your help, Llanos

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[Users] RE: [Serusers] OpenSER release
by Giudice, Salvatore 16 Jun '05

16 Jun '05

Policy is only as good as its communication to the people it applies to. Besides the initial announcement, how many times was this site publicized in the mailing list or enforced as standard practice to have patches added to the project? -----Original Message----- From: Jan Janak [mailto:jan@iptel.org] Sent: Wednesday, June 15, 2005 9:08 PM To: info(a)beeplove.com Cc: Giudice, Salvatore; daniel(a)voice-system.ro; andrei(a)iptel.org; serdev(a)iptel.org; serusers(a)iptel.org; devel(a)openser.org; users(a)openser.org Subject: Re: [Serusers] OpenSER release There is such a bug tracking system running at http://bugs.sip-router.org It is powered by Jira and is quite easy to use. I announced it to serdev mailing list some time ago but there seemed to be no interest, from core developers only myself and Andrei created accounts. I was using it for a while and I find it easy to use and convenient. Jan. On 15-06-2005 14:24, info(a)beeplove.com wrote: > I agree with you. > We should have proper bug tracking system, including feature request. > > MOhammad > > > > Original Message: > ----------------- > From: Giudice, Salvatore Salvatore.Giudice(a)FMR.COM > Date: Wed, 15 Jun 2005 14:12:15 -0400 > To: daniel(a)voice-system.ro, andrei(a)iptel.org, serdev(a)iptel.org, > serusers(a)iptel.org, devel(a)openser.org, users(a)openser.org > Subject: RE: [Serusers] OpenSER release > > > I am not an advocate for either ser or openser, but I would like to > comment. > > Is openser going to be equipped with a forum/ticket system where people > can document bugs, feature requests, etc (non-configuration issues)? > > This is just my observation and you may not agree, but I believe this > project could be much better maintained if it used a more structured > ticketing style system to manage development issues instead of the > current mailing lists. In my experience, mailing lists like this foster > a terrible user experience where many development issues can go on > without response. > > Ideally, if there was a mailing list to address user issues and > ticketing system like the one Digium uses to manage Asterisk, I think > everyone would benefit by being better informed and ser would ultimately > be a better product for it. How many people out there feel that their > issues have fallen through the cracks in the past couple years? > > -----Original Message----- > From: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] > Sent: Wednesday, June 15, 2005 4:28 AM > To: Andrei Pelinescu-Onciul > Cc: SER developer mailing list; serusers; users(a)openser.org; > devel(a)openser.org > Subject: Re: [Serusers] OpenSER release > > On 06/14/05 23:21, Andrei Pelinescu-Onciul wrote: > > >On Jun 14, 2005 at 22:48, Daniel-Constantin Mierla > <daniel(a)voice-system.ro> wrote: > > > >[...] > > > > > >>It is your opinion, but I repeat myself, that the SER code maintained > by > >>us will go further -- I don't think that someone can claim that we > >>didn't do the job for our code (the only discrepancy is some > last-minute > >>adds in xlog (to print avps) - will be committed on unstable very soon > > >>with the new color patch). The cvs was created just to ease the > >>maintainance. The patches would be a nightmare. > >> > >> > > > >Maybe I've misunderstood you: is this only a parallel "stabilized" > >version + some features or is it a full fork (do you intend to fork > >unstable also)? > > > > > It is fork for the code that we changed (acc module, usrloc module ...), > > in the future may be other that they do not find the path in SER. We > will maintain and upgrade our part of code from SER continuously. > > >I have no problem with another stable version, what worries me is > >fragmenting the development for unstable (which is the place where > major > >changes are made). > > > > > I see no fragmenting there -- the situation is the same for SER as it > was before. For example, there is no fragment for acc module, it will be > > maintained by who did it till now, adding what he considers necessary > there. But we came to meet a lot of requests of why the acc patch is not > > included in the CVS (it was fully backward compatible and had new > features requested by many SER users) and we want to promote _more open_ > > approach to contributions to all parts of code. The acc patch was sent > on November 1, 2004. No real response (neither negative, nor positive) > from maintainer to the submission since then ... are you aware of a good > > reason?!?! ... should we wait just about (or more) half an year for each > > contribution?!? I will not do that anymore!!! > > Daniel > > > > >Andrei > > > > > > > > _______________________________________________ > Serusers mailing list > Serusers(a)iptel.org > http://mail.iptel.org/mailman/listinfo/serusers > > _______________________________________________ > Serusers mailing list > Serusers(a)iptel.org > http://mail.iptel.org/mailman/listinfo/serusers > > -------------------------------------------------------------------- > mail2web - Check your email from the web at > http://mail2web.com/ . > > > _______________________________________________ > Serusers mailing list > Serusers(a)iptel.org > http://mail.iptel.org/mailman/listinfo/serusers

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[Serusers] Problem on pthread and dtmf
by Nicky 16 Jun '05

16 Jun '05

Dear all, 1. Please help, when i run sems -E i experienced this error, what can be the caused of it:~ ~~11352) ERROR: start (AmThread.cpp:73): pthread create failed with code 12 2. There is signal of DTMF which is not pressed but is being send to the server, please see below for the log. Please advice what can be the caused and problem. Thank you in advance. ~~(4395) DEBUG: isdn_audio_eval_dtmf_relative (IvrDtmfDetector.cpp:349): Posting D TMF 0: 5 (5) (4395) DEBUG: postEvent (AmEventQueue.cpp:47): AmEventQueue: trying to post even t (4395) DEBUG: postEvent (AmEventQueue.cpp:55): AmEventQueue: event posted (4395) DEBUG: processEvents (AmEventQueue.cpp:68): before processing event (4395) DEBUG: process (IvrPython.cpp:988): IvrPython processing event... (4395) DEBUG: process (IvrPython.cpp:991): IvrDialog processing event... (4395) DEBUG: onDTMFEvent (IvrPython.cpp:1130): IvrPython::onDTMFEvent(): callin g onDTMFCallback key is 5... (4395) DEBUG: ivrEmptyMediaQueue (IvrPython.cpp:102): IVR: emptying media queue . (4395) DEBUG: IvrMediaEvent (IvrEvents.cpp:31): New Media Event: 1, (4395) DEBUG: postEvent (AmEventQueue.cpp:47): AmEventQueue: trying to post even t (4395) DEBUG: postEvent (AmEventQueue.cpp:55): AmEventQueue: event posted (4395) DEBUG: ivrStopRecording (IvrPython.cpp:132): IVR: stop recording. best regards, nicky

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[Serusers] MediaProxy NO sound
by TCCHAN 16 Jun '05

16 Jun '05

Dear All, I apologize for asking a repeated question. I tried ser 0.8.14 and 0.9 to work with mediaproxy. I could setup call but there are silence. I did try to bypass the proxydispatcher modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy.sock") #modparam("mediaproxy", "mediaproxy_socket","/var/run/proxydispatcher.sock") and the result is the same. I have been stuck for days. Thanks in advenced for any help available. Regards, TC Chan The proxydispatcher dump the following: [root@ser mediaproxy]# ./proxydispatcher.py --no-fork Listening for commands on local socket `/var/run/proxydispatcher.sock' Default mediaproxy server is on `/var/run/mediaproxy.sock' command request fe5a6f36-c6092310(a)192.228.118.240 192.228.118.240:18278:audio 210.187.111.86 219.94.42.174 remote 210.187.111.86 remote Sipura/SPA2000-2.0.13(g) info=from:01548900@219.94.42.174,to:01548901@219.94.42.174,fromtag:d9170fe31e4cbe8o0,totag: forwarding to mediaproxy on /var/run/mediaproxy.sock: got: '127.0.0.1 35000' command execution time: 16.89 ms command lookup fe5a6f36-c6092310(a)192.228.118.240 192.228.118.240:18280:audio 210.187.111.86 219.94.42.174 remote 219.94.42.174 unknown Sipura/SPA2000-2.0.13(g) info=from:01548900@219.94.42.174,to:01548901@219.94.42.174,fromtag:d9170fe31e4cbe8o0,totag:8f2251e89d625018i1 forwarding to mediaproxy on /var/run/mediaproxy.sock: got: '127.0.0.1 35000' command execution time: 3.92 ms After a long while .......... command delete fe5a6f36-c6092310(a)192.228.118.240 info= forwarding to mediaproxy on /var/run/mediaproxy.sock: got: '' command execution time: 2.53 ms While Mediaproxy dump the following: [root@ser mediaproxy]# ./mediaproxy.py --no-fork Listening for commands on local socket `/var/run/mediaproxy.sock' Listening for remote commands is disabled Using IP address `127.0.0.1' for the RTP/RTCP proxy command request fe5a6f36-c6092310(a)192.228.118.240 192.228.118.240:18278:audio 210.187.111.86 219.94.42.174 remote 210.187.111.86 remote Sipura/SPA2000-2.0.13(g) info=from:01548900@219.94.42.174,to:01548901@219.94.42.174,fromtag:d9170fe31e4cbe8o0,totag:,dispatcher session fe5a6f36-c6092310(a)192.228.118.240: started. listening on 127.0.0.1:35000 command execution time: 11.72 ms command lookup fe5a6f36-c6092310(a)192.228.118.240 192.228.118.240:18280:audio 210.187.111.86 219.94.42.174 remote 219.94.42.174 unknown Sipura/SPA2000-2.0.13(g) info=from:01548900@219.94.42.174,to:01548901@219.94.42.174,fromtag:d9170fe31e4cbe8o0,totag:8f2251e89d625018i1,dispatcher command execution time: 1.32 ms command status command execution time: 0.36 ms session fe5a6f36-c6092310(a)192.228.118.240: 0/0/0 packets, 0/0/0 bytes (caller/called/relayed) session fe5a6f36-c6092310(a)192.228.118.240: ended (did timeout). command status command execution time: 0.18 ms command delete fe5a6f36-c6092310(a)192.228.118.240 info=dispatcher command execution time: 0.24 ms Session dump the following: ./[root@ser mediaproxy]# ./sessions.py Caller Via Called Status Duration Codec Type Traffic ------------------------------------------------------------------------------------------------------------ 192.228.118.240:18278 - 127.0.0.1:35000 - 192.228.118.240:18280 inactive 0'04" Unknown Audio 0/0/0 Total traffic: 0bps/0bps/0bps (in1/in2/out) Session count: 1 [root@ser mediaproxy]# ./sessions.py No RTP sessions My ser script is as follow: #debug=3 #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 ---------------------------------- #loadmodule "/usr/local/lib/ser/modules/nathelper.so" 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/xlog.so" loadmodule "/usr/local/lib/ser/modules/exec.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/group.so" loadmodule "/usr/local/lib/ser/modules/textops.so" #loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" #---------------------------------------------------------------------- # mediaproxy modparam("mediaproxy", "natping_interval", 60) #modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy", "mediaproxy_socket", "/var/run/proxydispatcher.sock") modparam("registrar", "nat_flag", 2) #---------------------------------------------------------------------- #---------------------------------------------------------------------- # uri #modparam("uri", "db_url", "sql://root:heslo@localhost/ser") #modparam("uri", "uri_table", "subscriber") #---------------------------------------------------------------------- #---------------------------------------------------------------------- # TCN: no forking #modparam("registrar","desc_time_order",1) #modparam("usrloc","desc_time_order",1) #modparam("registrar","append_branches",0) #----------------------------------------------------------------------- # ----------------- setting module-specific parameters --------------- modparam("usrloc", "db_mode", 1) #modparam("usrloc", "db_mode", 2) #------------------------------------------------------------------------------------- #TCN: To chnage the database loc to remote PC change the following lines #grep -r 'db_url' sersrc review that the following modules need to be changed: # group,vm,usrloc,uri,domain,jabber,auth_db,pdt,acc,msilo #modparam("auth_db", "db_url", "sql://root:heslo@219.94.42.174/ser") #modparam("usrloc", "db_url", "sql://root:heslo@219.94.42.174/ser") #modparam("group", "db_url", "sql://root:heslo@219.94.42.174/ser") modparam("acc", "db_url", "mysql://root:heslo@localhost/ser") modparam("auth_db", "db_url", "mysql://root:heslo@localhost/ser") modparam("usrloc", "db_url", "mysql://root:heslo@localhost/ser") modparam("group", "db_url", "mysql://root:heslo@localhost/ser") #------------------------------------------------------------------------------------- modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 30 ) ################### NAT ##################### # We will you flag 6 to mark NATed contacts #modparam("registrar", "nat_flag", 6) # Enable NAT pinging #modparam("nathelper", "natping_interval", 3) # Ping only contacts that are known to be # behind NAT #modparam("nathelper", "ping_nated_only", 1) ############################################## #---------------------------------------------------------------------- # accounting #modparam("acc", "log_flag", 2) #modparam("acc", "db_flag", 2) #---------------------------------------------------------------------- alias="redtone" alias="redtone.sip" alias="redtone.sip.com" alias="redtone.sip.com.my" # ------------------------- request 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"); xlog ("L_ERR","-----------------------------------------------------------------\r\n\r\n\r\n\r\n"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too Large"); xlog ("L_ERR","-----------------------------------------------------------------\r\n\r\n\r\n\r\n"); break; }; if (method=="REGISTER") { xlog("L_ERR","xlog %Tf: REGISTER route uri:%ru method:%rm\r\nfrom:% fu To:%tu Contact:%ct\r\n\r\n\r\n"); #if (is_from_local()) #{ xlog("L_ERR","xlog %Tf: FROM LOCAL REGISTER route uri:%ru method:%rm\r\nfrom:%fu To:%tu Contact:%ct\r\n\r\n\r\n"); # Mark as NAT'ed if (client_nat_test("3")) { #setflag(2); force_rport(); fix_contact(); }; xlog("L_ERR","xlog %Tf: start REGISTER route uri:%ru method:%rm\r \nfrom:%fu To:%tu Contact:%ct\r\n\r\n\r\n"); if (!www_authorize("redtone", "subscriber")) { xlog("L_ERR","%fu failed authentification!\r\n"); www_challenge("redtone", "0"); xlog("L_ERR","%fu failed authentification and challenge!\r\n"); xlog ("L_ERR","-----------------------------------------------------------------\r\n\r\n\r\n\r\n"); break; }; save("location"); xlog("L_ERR","username %fu location REGISTERed!\r\n"); xlog ("L_ERR","-----------------------------------------------------------------\r\n\r\n\r\n\r\n"); break; /* } else { xlog("L_ERR","This domain is not served here!\r\n"); sl_send_reply("403", "This domain is not served here"); }; */ break; }; /* if (method=="INVITE") { if (!(is_from_local() || is_uri_host_local())) { sl_send_reply("403", "Relaying is forbidden"); break; }; t_on_failure("1"); } else */ if (method == "BYE" || method == "CANCEL") { end_media_session(); }; /* if (loose_route()) { if (method=="INVITE" || method=="ACK") { use_media_proxy(); }; # end media session for BYE and CANCEL is done above # before entering the loose route. no need to call it here t_relay(); break; }; */ # Force subsequent messages to pass trough this proxy if (method == "INVITE") { record_route(); }; if (client_nat_test("3") && !search("^Record-Route:")) { # Mark as NAT'ed force_rport(); fix_contact(); }; if (method=="INVITE") { t_on_reply("1"); }; # if (is_uri_host_local()) { if (!lookup("location")) { sl_send_reply("404", "User not found"); break; }; # }; if (method=="INVITE" || method=="ACK") { use_media_proxy(); }; if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } failure_route[1] { end_media_session(); } onreply_route[1] { if (status=~"(183)|(2[0-9][0-9])") { #if (status=~"200") { if (client_nat_test("1")) { fix_contact(); }; use_media_proxy(); }; }

3 2

[Users] documentation site
by Daniel-Constantin Mierla 16 Jun '05

16 Jun '05

Hello, there is a Wiki page where users can contribute to OpenSER documentation. It is open for everybody and we hope that everybody will have a fair attitude. http://openser.org/dokuwiki/ I have started a new document "OpenSER Core CookBook", we hope there will be many contributions. Best regards, Daniel

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[Serusers] uri matching for dial plans
by Iqbal 16 Jun '05

16 Jun '05

Hi I have a nice table of prefixes and gateways, which works great using lcr, however I am now working on dial plans, i.e user can call 001, not 0044 etc etc, based on what I have in my grp table. Problem here is with codes like 001 since this would allow 001234 also, which for instance could be puerto rico (which it aint) but if it was I wouldnt want the call to go there. so I am doing something like if (uri=001 && !uri=001234) make the call...BUT this is a pain, especially for 001, I have had to block out 30 odd routes manually, which is making my ser config look really ugly. add into that other routes and suddenly we have a painful config file. So what I wanted to know is it possible to pull info from a DB, using exec, and store the value in a variable which can be used later. the grp table only really allows u to bind a "name" and then see if the from user is in that group, but it dont let you to the break that down into finer options, especially dial plans Iqbal

2 2

[serusers] PA module
by michael p 16 Jun '05

16 Jun '05

hi i tried to load the module pa but it doesn't work. how can i check for the presentity? i tried with the softphone eyebeam to add a contact to my contact list but the little man is like off line. i added in ser.cfg : loadmodule "/usr/lib/ser/modules/pa.so" and modparam("pa", "default_expires", 3600) and " handle_subscription("registrar"); " in : if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("dev.annatel.net", "subscriber")) { www_challenge("dev.annatel.net", "0"); break; }; save("location"); handle_subscription("registrar"); break; }; i don't use jabber so i want when the user registrar that do the suscribe and transmit to the others... is that correct what i did? why there are so little doc about all the modules of SER? thanks for the help... M.

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[Serusers] ask about using SER 0.9 with Cisco PSTN Gateway
by gunawan 16 Jun '05

16 Jun '05

i hav problem using ser 0.9 with Cisco PSTN gateway... i use Cisco Router 2620XM with voice module VIC2FXO as the RJ-11 connection and IOS Cisco 12.2(13)T4... does anyone in forum ever use cisco 2620XM as a PSTN gateway???? I need help in Cisco Configuration and Ser.cfg scripts.... it's work when i use Cisco Call Manager which use H323 Protocol as my server , and right now i want to try use SIP as my protocol and SER as my server.... thanks in advance..... __________________________________ Discover Yahoo! Stay in touch with email, IM, photo sharing and more. Check it out! http://discover.yahoo.com/stayintouch.html

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[serusers] PA module
by michael p 16 Jun '05

16 Jun '05

hi where can i got informations about pa module? Is that true it doesn't work with sip messages and only work with jabber? where can i find a step by step for jabber and pa? thanks M.

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[Serusers] Onsip.org ser.cfg + pstn + uac - can't authenticate on gateway
by Cameron Beattie 16 Jun '05

16 Jun '05

I am trying the get the uac module working with the onsip.org ser.cfg (the pstn + mediaproxy version). I think the problem is related to the authentication on the gateway device. It doesn't seem to be challenging SER. Below is the SIP traffic and then the ser.cfg file. Any suggestions would be appreciated. For information, the gateway (198.22.67.70) is provided by Nufone.net and is an Asterisk server, I believe. Thanks in advance for any advice. Regards Cameron U 147.202.xx.xxx:5060 -> 60.234.xxx.xxx:5060 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bK-fff17e3f;rport=5060;received=60.234 .xxx.xxx..From: <sip:user1@mydomain.com>;tag=f9493da2f8a6d10ao0..To: <sip:00442070800000@mydomain.com>..Call-ID: ce674b 82-e2fd1caa@192.168.0.11..CSeq: 102 INVITE..Server: Sip EXpress router (0.9.2 (i386/linux))..Content-Length: 0..Warning: 392 147.202.xx.xxx:5060 "Noisy feedback tells: pid=6752 req_src_ip=60.234.xxx.xxx req_src_port=5060 in_uri=sip:00442070800000@mydomain.com out_uri =sip:00442070800000@198.22.67.70 via_cnt==1".... ## U 147.202.xx.xxx:5060 -> 198.22.67.70:5060 INVITE sip:00442070800000@198.22.67.70 SIP/2.0..Record-Route: <sip:00442070800000@147.202.xx.xxx:5060;nat=yes;ftag=f9493da2f8a6d10ao0;lr=on>. .Via: SIP/2.0/UDP 147.202.xx.xxx;branch=z9hG4bK8b0a.2871d5c2.0..Via: SIP/2.0/UDP 192.168.0.11:5060;rport=5060;received=60.234.xxx.xxx;bran ch=z9hG4bK-fff17e3f..From: <sip:user1@mydomain.com>;tag=f9493da2f8a6d10ao0..To: <sip:00442070800000@mydomain.com>..Cal l-ID: ce674b82-e2fd1caa@192.168.0.11..CSeq: 102 INVITE..Max-Forwards: 16..Contact: <sip:user1@60.234.xxx.xxx:5060>..Expires: 240..User- Agent: Sipura/SPA3000-3.1.3(GWa)..Content-Length: 424..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: x-sipura ..Content-Type: application/sdp....v=0..o=- 757701 757701 IN IP4 192.168.0.11..s=-..c=IN IP4 147.202.xx.xxx..t=0 0..m=audio 35026 RTP/AVP 0 2 4 8 18 96 97 98 100 101..a=rtpmap:0 PCMU/8000..a=rtpmap:2 G726-32/8000..a=rtpmap:4 G723/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729 a/8000..a=rtpmap:96 G726-40/8000..a=rtpmap:97 G726-24/8000..a=rtpmap:98 G726-16/8000..a=rtpmap:100 NSE/8000..a=rtpmap:101 telephone-event /8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. # U 147.202.xx.xxx:5060 -> 60.234.xxx.xxx:5060 SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL)..Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bK-fff17e3f;rport=5060;recei ved=60.234.xxx.xxx..From: <sip:user1@mydomain.com>;tag=f9493da2f8a6d10ao0..To: <sip:00442070800000@mydomain.com>;tag=66 9cac32ae43cfbf664b867e0fb4dd5a.572f..Call-ID: ce674b82-e2fd1caa@192.168.0.11..CSeq: 102 INVITE..Server: Sip EXpress router (0.9.2 (i386/l inux))..Content-Length: 0..Warning: 392 147.202.xx.xxx:5060 "Noisy feedback tells: pid=6752 req_src_ip=60.234.xxx.xxx req_src_port=5060 i n_uri=sip:00442070800000@mydomain.com out_uri=sip:00442070800000@198.22.67.70 via_cnt==1".... ## U 198.22.67.70:5060 -> 147.202.xx.xxx:5060 SIP/2.0 404 Not Found..Via: SIP/2.0/UDP 147.202.xx.xxx;branch=z9hG4bK8b0a.2871d5c2.0..Via: SIP/2.0/UDP 192.168.0.11:5060;received=60.234. xxx.xxx;branch=z9hG4bK-fff17e3f..From: <sip:user1@mydomain.com>;tag=f9493da2f8a6d10ao0..To: <sip:00442070800000@mydomain.com>;tag=as40c6caec..Call-ID: ce674b82-e2fd1caa@192.168.0.11..CSeq: 102 INVITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, O PTIONS, BYE, REFER, NOTIFY..Contact: <sip:00442070800000@198.22.67.70>..Content-Length: 0.... # U 147.202.xx.xxx:5060 -> 198.22.67.70:5060 ACK sip:00442070800000@198.22.67.70 SIP/2.0..Via: SIP/2.0/UDP 147.202.xx.xxx;branch=z9hG4bK8b0a.2871d5c2.0..From: <sip:user1@mydomain.com>;tag=f9493da2f8a6d10ao0..Call-ID: ce674b82-e2fd1caa@192.168.0.11..To: <sip:00442070800000@mydomain.com>;tag=as40c6caec. .CSeq: 102 ACK..User-Agent: Sip EXpress router(0.9.2 (i386/linux))..Content-Length: 0.... My ser.cfg is below: debug=3 fork=yes #fork=no log_stderror=yes check_via=no dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://user:password@localhost/ser" sock_mode=0666 #fifo_mode=0666 #fifo_user=root listen=147.202.xx.xxx port=5060 children=4 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/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/uac.so" modparam("auth_db|permissions|uri_db|usrloc|acc", "db_url", "mysql:// user:password @localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 3) modparam("uac","credential","user:asterisk:password") route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { record_route_preset("147.202.xx.xxx:5060;nat=yes"); # insert IP address } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { setflag(1); end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (client_nat_test("3") || search("^Route:.*;nat=yes")) { setflag(6); use_media_proxy(); }; }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(5); route(1); break; }; if (uri==myself) { if (method=="ACK") { route(6); break; } else if (method=="CANCEL") { route(3); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(5); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact: \*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # CANCEL and INVITE Message Handler # ----------------------------------------------------------------- if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; if (method=="INVITE" && !allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; lookup("aliases"); if (uri!=myself) { route(5); route(1); break; }; if (uri=~"^sip:[+|00][0-9]*@") { # International PSTN route(4); break; }; if (!lookup("location")) { if (uri=~"^sip:[0-9]{8}@") { # Domestic PSTN route(4); break; }; sl_send_reply("404", "User Not Found"); break; }; if (method=="CANCEL") { route(1); break; }; setflag(1); setflag(3); route(5); route(1); } route[4] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- avp_write("i:45", "inv_timeout"); route(5); t_on_failure("4"); resetflag(8); t_relay_to_udp("198.22.67.70","5060"); route(1); } route[5] { # ----------------------------------------------------------------- # RTP Proxy Enabler # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { use_media_proxy(); }; } route[6] { # ------------------------------------------------------------------------ # ACK Handler # ------------------------------------------------------------------------ # ------------------------------------------------------------------------ # Aliases Section # ------------------------------------------------------------------------ lookup("aliases"); if (uri!=myself) { route(1); break; }; lookup("location"); route(1); } onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:\ 0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; } failure_route[5] { if (t_check_status("401|407")) { if (isflagset(8)) { t_reply("503","Auth failed"); break; } if (uac_auth()) { setflag(8); t_on_failure("5"); append_branch(); t_relay(); } } }

2 1

[Serusers] ultra monkey & ser
by Jack Wei 16 Jun '05

16 Jun '05

hi, I'm trying to set up a cluster of ser boxes using ultramonkey. i want ldirectord to detect if any of the ser boxes is down, but am not able to set it up properly. Here's my ldirector.cf: checktimeout=10 checkinterval=2 autoreload=no logfile="local0" quiescent=yes virtual=10.1.201.189:5060 real=10.1.201.191:5060 gate real=10.1.201.193:5060 gate scheduler=wlc protocol=udp # checktype=connect persistent=600 Using the above configuration, ipvsadm reports the real servers are up even if they aren't. If i enable "checktype=connect", ipvsadm reports the real servers are always down. if anyone know of solutions, please let me know. thanks. Jack Wei __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 1

Re: [Serusers] OpenSER release
by info＠beeplove.com 16 Jun '05

16 Jun '05

Are you guys planning to have a feature request page as well? Mohammad Original Message: ----------------- From: Greger V. Teigre greger(a)teigre.com Date: Wed, 15 Jun 2005 21:03:21 +0200 To: Salvatore.Giudice(a)FMR.COM, serdev(a)iptel.org, serusers(a)iptel.org Subject: Re: [Serusers] OpenSER release I completely agree with you. I have been told that there was an attempt at introducing a bug-tracking system earlier, but that it has been difficult. Anyhow, in setting up policies and procedures around the experimental directory, we have decided that usage of http://bugs.sip-router.org will be mandatory. Hopefully recent, better integration between the bug tracking system and the CVS will make it more convenient to use also for other CVS modules (however, I don't have a say there). g-) Giudice, Salvatore wrote: > I am not an advocate for either ser or openser, but I would like to > comment. > > Is openser going to be equipped with a forum/ticket system where > people can document bugs, feature requests, etc (non-configuration > issues)? > > This is just my observation and you may not agree, but I believe this > project could be much better maintained if it used a more structured > ticketing style system to manage development issues instead of the > current mailing lists. In my experience, mailing lists like this > foster a terrible user experience where many development issues can > go on without response. > > Ideally, if there was a mailing list to address user issues and > ticketing system like the one Digium uses to manage Asterisk, I think > everyone would benefit by being better informed and ser would > ultimately be a better product for it. How many people out there feel > that their issues have fallen through the cracks in the past couple > years? > > -----Original Message----- > From: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] > Sent: Wednesday, June 15, 2005 4:28 AM > To: Andrei Pelinescu-Onciul > Cc: SER developer mailing list; serusers; users(a)openser.org; > devel(a)openser.org > Subject: Re: [Serusers] OpenSER release > > On 06/14/05 23:21, Andrei Pelinescu-Onciul wrote: > >> On Jun 14, 2005 at 22:48, Daniel-Constantin Mierla > <daniel(a)voice-system.ro> wrote: >> >> [...] >> >> >>> It is your opinion, but I repeat myself, that the SER code >>> maintained by us will go further -- I don't think that someone can >>> claim that we didn't do the job for our code (the only discrepancy >>> is some last-minute adds in xlog (to print avps) - will be >>> committed on unstable very soon > >>> with the new color patch). The cvs was created just to ease the >>> maintainance. The patches would be a nightmare. >>> >>> >> >> Maybe I've misunderstood you: is this only a parallel "stabilized" >> version + some features or is it a full fork (do you intend to fork >> unstable also)? >> >> > It is fork for the code that we changed (acc module, usrloc module > ...), > > in the future may be other that they do not find the path in SER. We > will maintain and upgrade our part of code from SER continuously. > >> I have no problem with another stable version, what worries me is >> fragmenting the development for unstable (which is the place where >> major changes are made). >> >> > I see no fragmenting there -- the situation is the same for SER as it > was before. For example, there is no fragment for acc module, it will > be > > maintained by who did it till now, adding what he considers necessary > there. But we came to meet a lot of requests of why the acc patch is > not > > included in the CVS (it was fully backward compatible and had new > features requested by many SER users) and we want to promote _more > open_ > > approach to contributions to all parts of code. The acc patch was sent > on November 1, 2004. No real response (neither negative, nor positive) > from maintainer to the submission since then ... are you aware of a > good > > reason?!?! ... should we wait just about (or more) half an year for > each > > contribution?!? I will not do that anymore!!! > > Daniel > >> >> Andrei >> >> >> > > _______________________________________________ > Serusers mailing list > Serusers(a)iptel.org > http://mail.iptel.org/mailman/listinfo/serusers > > _______________________________________________ > Serusers mailing list > Serusers(a)iptel.org > http://mail.iptel.org/mailman/listinfo/serusers _______________________________________________ Serusers mailing list Serusers(a)iptel.org http://mail.iptel.org/mailman/listinfo/serusers -------------------------------------------------------------------- mail2web - Check your email from the web at http://mail2web.com/ .

2 1

RE: [Serusers] OpenSER release
by Giudice, Salvatore 16 Jun '05

16 Jun '05

Is anyone using this bug tracking system? Is this system sponsored by iptel.org? -----Original Message----- From: m36828253-1(a)imap.1and1.com [mailto:m36828253-1@pop.1and1.com] Sent: Wednesday, June 15, 2005 3:16 PM To: greger(a)teigre.com; Giudice, Salvatore; serdev(a)lists.iptel.org; serusers(a)lists.iptel.org Subject: Re: [Serusers] OpenSER release Why the bug tracking page in a different website. Why not under iptel.org ? Mohammad Original Message: ----------------- From: Greger V. Teigre greger(a)teigre.com Date: Wed, 15 Jun 2005 21:03:21 +0200 To: Salvatore.Giudice(a)FMR.COM, serdev(a)lists.iptel.org, serusers(a)lists.iptel.org Subject: Re: [Serusers] OpenSER release I completely agree with you. I have been told that there was an attempt at introducing a bug-tracking system earlier, but that it has been difficult. Anyhow, in setting up policies and procedures around the experimental directory, we have decided that usage of http://bugs.sip-router.org will be mandatory. Hopefully recent, better integration between the bug tracking system and the CVS will make it more convenient to use also for other CVS modules (however, I don't have a say there). g-) Giudice, Salvatore wrote: > I am not an advocate for either ser or openser, but I would like to > comment. > > Is openser going to be equipped with a forum/ticket system where > people can document bugs, feature requests, etc (non-configuration > issues)? > > This is just my observation and you may not agree, but I believe this > project could be much better maintained if it used a more structured > ticketing style system to manage development issues instead of the > current mailing lists. In my experience, mailing lists like this > foster a terrible user experience where many development issues can > go on without response. > > Ideally, if there was a mailing list to address user issues and > ticketing system like the one Digium uses to manage Asterisk, I think > everyone would benefit by being better informed and ser would > ultimately be a better product for it. How many people out there feel > that their issues have fallen through the cracks in the past couple > years? > > -----Original Message----- > From: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] > Sent: Wednesday, June 15, 2005 4:28 AM > To: Andrei Pelinescu-Onciul > Cc: SER developer mailing list; serusers; users(a)openser.org; > devel(a)openser.org > Subject: Re: [Serusers] OpenSER release > > On 06/14/05 23:21, Andrei Pelinescu-Onciul wrote: > >> On Jun 14, 2005 at 22:48, Daniel-Constantin Mierla > <daniel(a)voice-system.ro> wrote: >> >> [...] >> >> >>> It is your opinion, but I repeat myself, that the SER code >>> maintained by us will go further -- I don't think that someone can >>> claim that we didn't do the job for our code (the only discrepancy >>> is some last-minute adds in xlog (to print avps) - will be >>> committed on unstable very soon > >>> with the new color patch). The cvs was created just to ease the >>> maintainance. The patches would be a nightmare. >>> >>> >> >> Maybe I've misunderstood you: is this only a parallel "stabilized" >> version + some features or is it a full fork (do you intend to fork >> unstable also)? >> >> > It is fork for the code that we changed (acc module, usrloc module > ...), > > in the future may be other that they do not find the path in SER. We > will maintain and upgrade our part of code from SER continuously. > >> I have no problem with another stable version, what worries me is >> fragmenting the development for unstable (which is the place where >> major changes are made). >> >> > I see no fragmenting there -- the situation is the same for SER as it > was before. For example, there is no fragment for acc module, it will > be > > maintained by who did it till now, adding what he considers necessary > there. But we came to meet a lot of requests of why the acc patch is > not > > included in the CVS (it was fully backward compatible and had new > features requested by many SER users) and we want to promote _more > open_ > > approach to contributions to all parts of code. The acc patch was sent > on November 1, 2004. No real response (neither negative, nor positive) > from maintainer to the submission since then ... are you aware of a > good > > reason?!?! ... should we wait just about (or more) half an year for > each > > contribution?!? I will not do that anymore!!! > > Daniel > >> >> Andrei >> >> >> > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers -------------------------------------------------------------------- mail2web - Check your email from the web at http://mail2web.com/ .

2 1

Re: [Serusers] OpenSER release
by m36828253-1＠imap.1and1.com 16 Jun '05

16 Jun '05

Why the bug tracking page in a different website. Why not under iptel.org ? Mohammad Original Message: ----------------- From: Greger V. Teigre greger(a)teigre.com Date: Wed, 15 Jun 2005 21:03:21 +0200 To: Salvatore.Giudice(a)FMR.COM, serdev(a)iptel.org, serusers(a)iptel.org Subject: Re: [Serusers] OpenSER release I completely agree with you. I have been told that there was an attempt at introducing a bug-tracking system earlier, but that it has been difficult. Anyhow, in setting up policies and procedures around the experimental directory, we have decided that usage of http://bugs.sip-router.org will be mandatory. Hopefully recent, better integration between the bug tracking system and the CVS will make it more convenient to use also for other CVS modules (however, I don't have a say there). g-) Giudice, Salvatore wrote: > I am not an advocate for either ser or openser, but I would like to > comment. > > Is openser going to be equipped with a forum/ticket system where > people can document bugs, feature requests, etc (non-configuration > issues)? > > This is just my observation and you may not agree, but I believe this > project could be much better maintained if it used a more structured > ticketing style system to manage development issues instead of the > current mailing lists. In my experience, mailing lists like this > foster a terrible user experience where many development issues can > go on without response. > > Ideally, if there was a mailing list to address user issues and > ticketing system like the one Digium uses to manage Asterisk, I think > everyone would benefit by being better informed and ser would > ultimately be a better product for it. How many people out there feel > that their issues have fallen through the cracks in the past couple > years? > > -----Original Message----- > From: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] > Sent: Wednesday, June 15, 2005 4:28 AM > To: Andrei Pelinescu-Onciul > Cc: SER developer mailing list; serusers; users(a)openser.org; > devel(a)openser.org > Subject: Re: [Serusers] OpenSER release > > On 06/14/05 23:21, Andrei Pelinescu-Onciul wrote: > >> On Jun 14, 2005 at 22:48, Daniel-Constantin Mierla > <daniel(a)voice-system.ro> wrote: >> >> [...] >> >> >>> It is your opinion, but I repeat myself, that the SER code >>> maintained by us will go further -- I don't think that someone can >>> claim that we didn't do the job for our code (the only discrepancy >>> is some last-minute adds in xlog (to print avps) - will be >>> committed on unstable very soon > >>> with the new color patch). The cvs was created just to ease the >>> maintainance. The patches would be a nightmare. >>> >>> >> >> Maybe I've misunderstood you: is this only a parallel "stabilized" >> version + some features or is it a full fork (do you intend to fork >> unstable also)? >> >> > It is fork for the code that we changed (acc module, usrloc module > ...), > > in the future may be other that they do not find the path in SER. We > will maintain and upgrade our part of code from SER continuously. > >> I have no problem with another stable version, what worries me is >> fragmenting the development for unstable (which is the place where >> major changes are made). >> >> > I see no fragmenting there -- the situation is the same for SER as it > was before. For example, there is no fragment for acc module, it will > be > > maintained by who did it till now, adding what he considers necessary > there. But we came to meet a lot of requests of why the acc patch is > not > > included in the CVS (it was fully backward compatible and had new > features requested by many SER users) and we want to promote _more > open_ > > approach to contributions to all parts of code. The acc patch was sent > on November 1, 2004. No real response (neither negative, nor positive) > from maintainer to the submission since then ... are you aware of a > good > > reason?!?! ... should we wait just about (or more) half an year for > each > > contribution?!? I will not do that anymore!!! > > Daniel > >> >> Andrei >> >> >> > > _______________________________________________ > Serusers mailing list > Serusers(a)iptel.org > http://mail.iptel.org/mailman/listinfo/serusers > > _______________________________________________ > Serusers mailing list > Serusers(a)iptel.org > http://mail.iptel.org/mailman/listinfo/serusers _______________________________________________ Serusers mailing list Serusers(a)iptel.org http://mail.iptel.org/mailman/listinfo/serusers -------------------------------------------------------------------- mail2web - Check your email from the web at http://mail2web.com/ .

4 3

[Serusers] SER not responding
by Ryan Pagquil 16 Jun '05

16 Jun '05

Hi guys, I have a problem with SER... whenever I register using X-lite i never get logged in.... X-lite always prompts me that the connection timed out.... in my SER machine I run tcpdump -i gem0 | grep 5060 and i only see the request comming from my firewall and the SER machine is not responding on the request. here is the diagnostic of x-lite: SEND TIME: 3530347 SEND >> 202.84.24.107:5060 REGISTER sip:philonline.com SIP/2.0 Via: SIP/2.0/UDP 202.84.24.126:5060;rport;branch=z9hG4bK1165436EDDDD11D9BFB500E04CAB4AB4 From: Ryan Pagquil <sip:rpagquil@philonline.com>;tag=2916895101 To: Ryan Pagquil <sip:rpagquil@philonline.com> Contact: "Ryan Pagquil" <sip:rpagquil@202.84.24.126:5060> Call-ID: 11654266DDDD11D9BFB500E04CAB4AB4(a)philonline.com CSeq: 53597 REGISTER Expires: 1800 Max-Forwards: 70 User-Agent: X-Lite release 1103m Content-Length: 0 here is my ser.cfg: #debug=4 # debug level (cmd line: -dddddddddd) #fork=no #log_stderror=yes # (cmd line: -E) # Uncomment these lines to enter debugging mode #debug=3 #fork=yes #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 ---------------------------------- 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/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" modparam("usrloc", "db_mode", 2) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (len_gt( max_len )) { sl_send_reply("513", "Message too big"); break; }; record_route(); if (loose_route()) { t_relay(); break; }; if (uri==mysql) { if (method=="REGISTER") { if (!www_authorize("philonline.com", "subscriber")) { www_challenge("philonline.com", "0"); break; }; save("location"); break; }; if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; if (!t_relay()) { sl_reply_error(); }; } my ser.cfg contains the default configuration and i just edited and uncommented some parts of it. Thanks, -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com

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[Serusers] How to execute failure_route block on tcp connect failure.
by Dmitry Isakbayev 16 Jun '05

16 Jun '05

Hello, I am having a problem executing a failure_route block when t_relay fails to establish a tcp connection. Everything works as expected when t_relay connects to a proxy and receives back an error like 501. Is there a way to configure t_relay to execute a failure_route block an a tcp connection error? Thank You in advance. Dmitry Here are the logs from connection failure - Jun 15 20:52:14 localhost ser[21870]: ERROR: tcp_blocking_connect: SO_ERROR (111) Connection refused Jun 15 20:52:14 localhost ser[21870]: ERROR: tcpconn_connect: tcp_blocking_connect failed Jun 15 20:52:14 localhost ser[21870]: ERROR: tcp_send: connect failed Jun 15 20:52:14 localhost ser[21870]: msg_send: ERROR: tcp_send failed Jun 15 20:52:14 localhost ser[21870]: ERROR: t_forward_nonack: sending request failed

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