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37519 discussions

[Users] Accounting using dialog and sst module
by Asterisk Expert 03 Jan '07

03 Jan '07

Hello, I am currently trying to do accurate billing with openser. I found that dialog and sst modules can be used to determine failed transactions. i.e. If UA crashed and no BYE received at all. I tried to configure dialog and sst modules as specified in the documentation. But active dialogs keep incrementing even after hangup. It would be nice if someone can post a good example. I am using cvs head of openser. Snippet of config is as below: ###################### DIALOG ###################### modparam("dialog", "dlg_flag", 13) modparam("dialog", "timeout_avp", "$avp(i:10)") modparam("dialog", "use_tight_match", 1) modparam("dialog", "enable_stats", 1) ###################### SST ###################### modparam("sst", "enable_stats", 1) modparam("sst", "timeout_avp", "$avp(i:10)") modparam("sst", "sst_flag", 14) route { ........ ........ if(method=="INVITE") { setflag(13); # set the dialog flag setflag(14); # Set the sst flag avp_write("15","$avp(i:10)"); }; if (method=="INVITE") { record_route_preset("192.168.1.2:5060;nat=yes;did=1"); }; ......... ......... } -- Regards Asterisk Expert

3 6

[Users] Help: how can I create one SIP account
by CESAR HUAMANI H. 03 Jan '07

03 Jan '07

Hello, Today I installed the openser in my debian etch since the CVS, I was looking information about how I can creat accounts SIP for my Endpoints (softphone and Hardphone ATA), please help me I'm new with the OPENSER Best regards

2 1

[Serusers] Detecting if username is in subscriber table
by Simon 02 Jan '07

02 Jan '07

Dear All, I would like to detect if a URI is for a user that is in the subscriber table. If I just use lookup("location") then I only detect those users that have registered. If a phone is switched off it is not registered and so will not be in the location table. However it will be in the subscriber table and I need to be able to check this from within the ser.cfg file to allow correct routing etc. Any Ideas ? Many Thanks Simon

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[Users] Call forward record and Remove from callid in Openser
by Howard Tang 02 Jan '07

02 Jan '07

Hi All, I hope someone can help me with the following questions I have. 1. I have set up accounting module for record the call, simple outgoing call and incoming call are ok. However, if a sip phone activated the call forward, the call record doesn't show which sip client forward the call. e.g. A call B and B is the sip phone which forward the call to C. In the record, I can only see A call C. How can I see a record that B call C instead of A call C in the record? 2. I have connected to a PSTN gateway, and i have some internal sip users which can make a pstn call, the sip proxy will send their own private number a caller id to the pstn gateway. Since the callid is a invalid number, many Mobile phone networks provider reject the call. I want to remove the from callid when the call being send to the pstn gateway, how can i do this? If someone can send me the code that will be great. -- Howard Tang

2 1

[Users] Presence and Location
by Klaus Darilion 30 Dec '06

30 Dec '06

Hi! What kind of presence does the presence module support? Only the "basic" presence format PIDF (RFC 3863) or also PIDF extensions like the GEOPRIV Location Object Format RFC 4119? Or in other words - is the presence module aware of the XML payload or is it agnostic to the presence payload? regards klaus

2 1

[Users] totag_avpid
by Klaus Darilion 30 Dec '06

30 Dec '06

Hi! I just found this parameter in sl and presence module, but the usage is not described well. What is it's purpose? I guess that the presence module exports the totag into this AVP and the sl module use this totag in its response. I think it is not a good idea to have it configureable in 2 modules. We had the same problem with the "use_domain" parameter in several modules which was fixed recently. Maybe this can be fixed the same way - having it configurable in sl module, and presence module look ups the value on loading (or generate a error if not defined). Further, the presence module's parameter documentation lack the default values. regards klaus

2 1

[Users] Inspiring New Year wishes from Sunkara Ravi
by Sunkara Ravi 30 Dec '06

30 Dec '06

Dear Users, Sunkara Ravi [ sunkara.raviprakash.feb14(a)gmail.com ] has sent you an ecard from 123Greetings.com. Send free ecards from 123greetings.com with your choice of colors, words and music. Your ecard will be available with us for the next 30 days. If you wish to keep the ecard longer, you may save it on your computer or take a print. To view your ecard, choose from any of the following options: -------- OPTION 1 -------- Click on the following Internet address or copy & paste it into your browser's address box. http://www.123greetings.com/view/GX91230040549763 -------- OPTION 2 -------- Copy & paste the ecard number in the "View Your Card" box at http://www.123greetings.com Your ecard number is GX91230040549763 If you need help in viewing your card or any other assistance, please visit our Help / FAQ section located at http://www.123greetings.com/help/ If you need further help, feel free to write to us at support(a)123greetings.com Best wishes, Postmaster, 123Greetings.com *If you would like to send someone an ecard, you can do so at http://www.123greetings.com

1 0

Re: Re: [Users] how to configure failure route for b2bua or GW
by liu york 30 Dec '06

30 Dec '06

yes,the failure route works,but the client phone still receive 408 messeage. I try to use if(t_check_status(408|483)) drop; at failure route ,but faild :( ======= 2006-12-29 16:28:57 您在来信中写道：======= >Do no use forward(), just rewritehost and afterwards t_on_failure and t_relay. > >Samuel. > >2006/12/29, liu york <york(a)goldentek.biz>: >> hi all >> >> I try to build failure route to another backup b2bua or pstn gw,it seems can work after timeout ,but unfortunate,the openser send a 408 messeage to phone before failure route.,config like that,have any general config to failover or loadbalance for me.Many many thx. >> >> route[4] { >> >> log(1, "b2bua"); >> rewritehost("192.168.1.204"); >---> #forward("192.168.1.204:80"); >> t_on_failure("2"); >> t_relay(); >> } >> >> failure_route[2] { >> log(1, "b2bua"); >> rewritehost("192.168.1.226"); >---> #forward("192.168.1.226:5060"); >> t_relay(); >> } >> >> >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >> > = = = = = = = = = = = = = = = = = = = = 　　　　　　　　致礼！　　　　　　　　york 　　　　　　　　york(a)goldentek.biz 　　　　　　　　　　2006-12-30

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[Serusers] Re: [Serdev] SIP Stack development
by Greger V. Teigre 29 Dec '06

29 Dec '06

Hi Alan, See inline. Alan Basinger wrote: > Thanks Greger, > I know ho hard it is to keep up with the documentation of a rapidly > changing project and also know what a challenge it is to manage the > code. I hope we can use the base of SER to create some new and > exciting applications uing SIP and add some usefull features and > functionality to the base code for everyone to benifit from. > > I appreciate the quick reply and answering a newbs questions as I may > have more as we move along. :-) Sure, I realize that lack of documentation can be replaced (partially and only temporarily) by responsive support on mailing lists. It is also a way to establish the actual scope of documentation work. > > Hi Alan, > I'm heading the new documentation effort. An updated list of the > supported rfcs and standards is on my list of needed docs... I'm on > mobile right now, so I > cannot search for you, but on iptel.org/listsearch you can search for > Janak and supported rfcs. I think he posted an update recently. > > As for the specific draft, I'm not familiar with it. In general > though, media is not handled by ser, only messaging. > The draft is for allowing messaging as well as media to be sent to a > loopback to mesure actual network conditions to and from the device > without actually setting up a call. Very usefull and we have many > vendors implimeting it in there CPE. (Linksys, Polycom, SNOM, > Panasonic, etc.) I woul like to see about developing this and many > other features / functionaility into SER. I worked with one of the > authors of this draft and it realy makes sense. Here is a link to it. > > http://www.ietf.org/internet-drafts/draft-ietf-mmusic-media-loopback-05.txt > Aha, I recognize the name now :-) Without knowing the details of the draft, I have the following comments: * In general you want your servers out of the rtp path (i.e. direct media between UAs), in fact, ser is never in the rtp path. However ser is very good at doing message processing, forwarding and all sorts of "tricks" * RTP may be handled by three separate software components: rtpproxy, mediaproxy, and sems (mediaproxy is not an iptel.org project, the others are). SER modules (nathelper and mediaproxy) are used to communicate with these external components (locally or over udp) and for mangling the SDP payload * Your measurements will probably need to follow the media path of a normal session. If not, you will not be able to measure for example real latency on your ex. rtpproxy server(s). SEMS is a very good general-purpose media server, while rtpproxy/mediaproxy are more dedicated for proxying of rtp for NAT-purposes etc * Hence, you probably need to implement sdp loopback intp the rtp handling component. SEMS may be the easiest way due to its very modular and powerful plugin interface (it's also a B2BUA). However, as sems don't stay in the media path except in conferences etc, you don't really measure "real world..." * I suspect you can create a ser module (or extend nathelper) that implements the basic loopback stuff using rtpproxy without touching rtpproxy (too much) > As for development, ser has a very efficient core exposing a module > interfaces. Modules can implement functions, parameters, avpairs (i.e. > set variables), and > selects (give access to certain info related to a message). > I found a copy of the 2002 developers guide and have been starting > there but an concerned that it may be too outdated any idea when a new > draft will be available for review? The basic concepts are still relevant. It has not been updated to cover to the attribute value pairs and the selects. We are currently focused on the user documentation for the next release, but we are debating what the next steps are. A how-to for module development is already on the list, possible with a retouch of the old developers guide. You can start out with one of the simpler modules (like textops or dispatcher) and modify to your needs. Understand the plugin interface, lumps and shared memory, and you are pretty much there ;-) > > Contributions are always welcome. We have just introduced a new > classification of modules, thus allowing experimental modules into the > cvs without too much > fuss. > Good to hear as me and my partner in crime and I hope to be able to > add some value with our ideas. :-) g-) > > BTW, start with ser ottendorf. It is getting close to release and has > many new important improvements. > > Feel free to ask questions on this list. > g-) > > Thanks again > > Alan Basinger > > ------- Original message ------- > From: Alan Basinger <droidgeneral(a)yahoo.com> > Sent: 27.12.'06, 12:43 > > > Hello all, > > I definitely am a newby with SER but not with SIP and I am trying to > find out where the supported SIP methods are? > > Specifically how would I go about finding out if SER supports the > Media Loopback draft or other ratified or non ratified components. > > Also if not supported where would I being the coder that I am not be > able to modify the code myself to add these functions and then > distribute them back to > the community? > > > > Thanks in advance > > > > Alan > > > > __________________________________________________ > > Do You Yahoo!? > > Tired of spam? Yahoo! Mail has the best spam protection around > > http://mail.yahoo.com <http://mail.yahoo.com/> > > > __________________________________________________ > Do You Yahoo!? > Tired of spam? Yahoo! Mail has the best spam protection around > http://mail.yahoo.com

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[Users] New Year's Eve invitation from Sunkara Ravi
by Sunkara Ravi 29 Dec '06

29 Dec '06

Dear Users, Sunkara Ravi [ sunkara.raviprakash.feb14(a)gmail.com ] has sent you an ecard from 123Greetings.com. Send free ecards from 123Greetings.com with your choice of colors, words and music. Your ecard will be available with us for the next 30 days. If you wish to keep the ecard longer, you may save it on your computer or take a print. To view your ecard, choose from any of the following options: -------- OPTION 1 -------- Click on the following Internet address or copy & paste it into your browser's address box. http://www.123greetings.com/view/HW31229040825964 -------- OPTION 2 -------- Copy & paste the ecard number in the "View Your Card" box at http://www.123greetings.com Your ecard number is HW31229040825964 If you need help in viewing your ecard or any other assistance, please visit our Help/ FAQ section located at http://www.123greetings.com/help/ If you need further help, feel free to write to us at support(a)123greetings.com Best wishes, Postmaster, 123Greetings.com *If you would like to send someone an ecard, you can do so at http://www.123greetings.com

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