- sr-users - kamailio.org

[Serusers] Digit manipulation
by Jose Bertuzzi 16 Aug '05

16 Aug '05

Hello, i have to transform the dialed digits like this: 0[2-3][2-9][2-9]15[4-6]XXXXXX to 549[2-3][4-6]XXXXXX for example: 0385 15 6097775 to 549 385 6097775 I can do it with prefix and strip but only by adding lots of config lines. Is there any other easy method to do it? Is it possible to use a translation table in mysql to do the trick? regards, Pablo. ____________________________________________________ Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs

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Tr: RE: [Serusers] Serweb -- Installation
by harry gaillac 15 Aug '05

15 Aug '05

Remarque : message transféré en pièce jointe. ___________________________________________________________________________ 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] Core dump in custom dispatcher code
by Alistair Cunningham 15 Aug '05

15 Aug '05

I've trying to add support to the dispatcher module for reloading its configuration file on a fifo command. This allows an external daemon to monitor a set of external Asterisk machines, detect when they fail, and round robin calls between the surviving ones. I'm (I think) most of the way there. I can issue the command, and see the reload happening, but I see a core dump: [root@test dispatcher]# serctl fifo dispatcher_reload 2(31672) DISPATCHER:ds_load_list: dest [135343048/135343048/21] <sip:1.2.3.4:5060> 2(31672) DISPATCHER:ds_load_list: dest [0/1/1] <sip:1.2.3.4:5060> 2(31672) DISPATCHER:ds_load_list: found [1] dest sets 0(31670) child process 31672 exited by a signal 11 0(31670) core was generated (IP address changed to protect the guilty) I've had a look at the code, and don't really understand why it's happening. A gdb backtrace shows: (gdb) bt #0 ds_load_list (lfile=0x8112b98 "È+\021\b\025") at dispatch.c:281 281 dp = dp->next; #0 ds_load_list (lfile=0x8112b98 "È+\021\b\025") at dispatch.c:281 #1 0x0017cacf in dispatcher_reload (pipe=0xa1f5d30, response_file=0x8112378 "/tmp/ser_receiver_31623") at dispatcher.c:170 #2 0x08057456 in start_fifo_server () at fifo_server.c:540 #3 0x0805caf5 in main_loop () at main.c:988 #4 0x0805e52b in main (argc=3, argv=0xbffa3584) at main.c:1568 I've modified dispatcher.c. In mod_init, I've added: if (register_fifo_cmd(dispatcher_reload, "dispatcher_reload", 0) < 0) { LOG(L_ERR, "Cannot register dispatcher_reload\n"); return -1; } and added a new function: static int dispatcher_reload ( FILE* pipe, char* response_file ) { ds_destroy_list (); if (ds_load_list(dslistfile)==0) { fifo_reply (response_file, "200 OK\n"); return 1; } else { fifo_reply (response_file, "400 Dispatcher reload failed\n"); return -1; } } Can anyone shed light on this? -- Alistair Cunningham, Integrics Ltd, +44 (0)7870 699 479 http://integrics.com/

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[Serusers] AVPs question
by Federico Giannici 15 Aug '05

15 Aug '05

I need a way to store somewhere some computed values relative to sip messages. That is, I need some kind of "variables"! I need to access to these "variables" both from a custom SER module and from the ser.cfg script. I think that the solution are AVPs. Am I right? But I'm a little confused about the many "incarnations" of the AVP. After looking a little around I have understood that from inside the module I have to use the functions in "usr_avp.c". Am I right? Then there are the "avp" and "avpops" modules. My question is: they all refer to the SAME avps? Thas is, if I save an avp value by means of "usr_avp.c", then I can access the SAME avp by means of "avpops"? Or they are different and separate implementations? Thanks. P.S. I'm using SER 0.9.3. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

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[Serusers] call timeouts
by Iqbal 15 Aug '05

15 Aug '05

Hi Is there any standard on call timeouts. I hit a really strange problem. I have the following set PSTN ---> SER ---> asterisk All ip devices register to SER Now when a call comes in via PSTN and it destined for a coporate, it hits asterisk, which give its the auto attendant features and all, press 1, and you goto sales, via ser. Now if no one at sales answers, it goes to voicemail in asterisk again. BUT when I make a call from within that company, say to extension 111, which dials out to a IP phone registered in SER, and no one answers, I dont trip to voicemail. After hrs of testing, and a some help from some unknowns on the web, I realised it seemed to be a timeout problem. The call in SER was timing out, and hence call was being destroyed, if I set a timeout in the Dial string in asterisk to a number (which just by luck happened to be less than that in ser) it all worked. Was just wondering if there were any standard timeout parameters which people used for call, cause calls to pstn gateways would take longer than sip to sip ones. Iqbal

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[Serusers] Rating Engine ?
by Adrian Georgescu 15 Aug '05

15 Aug '05

Oznan, CDRTool has a rating and prepaid engine suitable for SER, Asterisk and Cisco. It is a commercial product. See: http://cdrtool.ag-projects.com Adrian >>>>>>>>>>>>>> Dear List, As some of you know, there would be an open-source-billing for OpenSER, i can see call duration,add remove edit users etc features but missing thing is rating. Since there may be more then one rating table it's very hard for MySql to query whole database and gives time out errors. I can manage PHP to create/delete/edit a new database with prefix,location name,initial time,initial cost,increment time,increment cost and rate id. Now there's a problem, think that there're 2 different rate table for different customers, a rate id is what i would like to write to acc table in each call. May be i should force ACC module to add rate_id together with other information like from_uri, to_sip, timestamp etc. etc. How can i manage this, is there any clue or information ? Thanks, Ozan Blotter

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[Serusers] ivr module to manage voice messages
by harry gaillac 15 Aug '05

15 Aug '05

Hello, I posted http://lists.iptel.org/pipermail/semsdev/2005-August/000437.html Nobody is interested to improve sems features ! It might not be the best solution. Some people are stranges when somebody suggest features nobody reply. therefore we can read sometimes some developpers telling "users just wait we provide a good and free project". 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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[Users] Rating Engine ?
by Ozan Blotter 15 Aug '05

15 Aug '05

Dear List, As some of you know, there would be an open-source-billing for OpenSER, i can see call duration,add remove edit users etc features but missing thing is rating. Since there may be more then one rating table it's very hard for MySql to query whole database and gives time out errors. I can manage PHP to create/delete/edit a new database with prefix,location name,initial time,initial cost,increment time,increment cost and rate id. Now there's a problem, think that there're 2 different rate table for different customers, a rate id is what i would like to write to acc table in each call. May be i should force ACC module to add rate_id together with other information like from_uri, to_sip, timestamp etc. etc. How can i manage this, is there any clue or information ? Thanks, Ozan Blotter

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[Serusers] Rating Engine ?
by Ozan Blotter 15 Aug '05

15 Aug '05

Dear List, As some of you know, there would be an open-source-billing for OpenSER, i can see call duration,add remove edit users etc features but missing thing is rating. Since there may be more then one rating table it's very hard for MySql to query whole database and gives time out errors. I can manage PHP to create/delete/edit a new database with prefix,location name,initial time,initial cost,increment time,increment cost and rate id. Now there's a problem, think that there're 2 different rate table for different customers, a rate id is what i would like to write to acc table in each call. May be i should force ACC module to add rate_id together with other information like from_uri, to_sip, timestamp etc. etc. How can i manage this, is there any clue or information ? Thanks, Ozan Blotter

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[Serusers] Blocking caller id
by Chris Tsai 15 Aug '05

15 Aug '05

Hi, I have problem about use rpid header in SIP invite message.and SER send response is 403... and RPID format is : Remote-Party-ID: Chris <sip:christsai12@iptel.org>;privacy=full;party=calling I am scratching my head on this one. Please help me...thanks for anyone... Any advice? Regards Chris My INVITE message as followsing.. Request-Line: INVITE sip:8201@iptel.org SIP/2.0 Message Header Via: SIP/2.0/UDP 61.216.39.201:5060;branch=z9hG4bK-hrs-21193 From: Anonymous <sip:anonymous@iptel.org>;tag=DBb-18789 To: <sip:8201@iptel.org> Call-ID: gnA-11414(a)61.216.39.201 CSeq: 8 INVITE Contact: <sip:christsai12@61.216.39.201> Max-Forwards: 70 Remote-Party-ID: Chris <sip:christsai12@iptel.org>;privacy=full;party=calling supported: replaces User-Agent: DrayTek UA-1.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO, REFER, NOTIFY, SUBSCRIBE Content-Type: application/sdp Content-Length: 270 Message body Session Initiation Protocol Status-Line: SIP/2.0 407 Proxy Authentication Required Message Header Via: SIP/2.0/UDP 61.216.39.201:5060;branch=z9hG4bK-hrs-21193;rport=5060;received=61.216.35.30 From: Anonymous <sip:anonymous@iptel.org>;tag=DBb-18789 To: <sip:8201@iptel.org>;tag=b51ece8fd8c195776737473dd6552d43.8aa7 Call-ID: gnA-11414(a)61.216.39.201 CSeq: 8 INVITE P-Behind-NAT: Yes Proxy-Authenticate: Digest realm="iptel.org", nonce="42bf93c8d6bc77f7ef3e6e8f24513796fda5bccc" Server: Sip EXpress router (0.9.0udpfifo (i386/linux)) Content-Length: 0 Warning: 392 195.37.77.99:5060 "Noisy feedback tells: pid=30610 req_src_ip=61.216.35.30 req_src_port=5060 in_uri=sip:8201@iptel.org out_uri=sip:8201@iptel.org via_cnt==1" Session Initiation Protocol Request-Line: ACK sip:8201@iptel.org SIP/2.0 Message Header Via: SIP/2.0/UDP 61.216.39.201:5060;branch=z9hG4bK-hrs-21193 From: Anonymous <sip:anonymous@iptel.org>;tag=DBb-18789 To: <sip:8201@iptel.org>;tag=b51ece8fd8c195776737473dd6552d43.8aa7 Call-ID: gnA-11414(a)61.216.39.201 CSeq: 8 ACK Content-Length: 0 Session Initiation Protocol Request-Line: INVITE sip:8201@iptel.org SIP/2.0 Message Header Via: SIP/2.0/UDP 61.216.39.201:5060;branch=z9hG4bK-mWf-3372 From: Anonymous <sip:anonymous@iptel.org>;tag=DBb-18789 To: <sip:8201@iptel.org> Call-ID: gnA-11414(a)61.216.39.201 CSeq: 9 INVITE Contact: <sip:christsai12@61.216.39.201> Proxy-Authorization: Digest username="christsai12", realm="iptel.org", nonce="42bf93c8d6bc77f7ef3e6e8f24513796fda5bccc", uri="sip:8201@iptel.org", response="d1cb74bf6a92df9edf18e565ee2eac79", algorithm=MD5 Max-Forwards: 70 Remote-Party-ID: Chris <sip:christsai12@iptel.org>;privacy=full;party=calling supported: replaces User-Agent: DrayTek UA-1.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO, REFER, NOTIFY, SUBSCRIBE Content-Type: application/sdp Content-Length: 270 Message body Session Initiation Protocol Status-Line: SIP/2.0 403 That is ugly -- use From=id next time (OB) Message Header Via: SIP/2.0/UDP 61.216.39.201:5060;branch=z9hG4bK-mWf-3372;rport=5060;received=61.216.35.30 From: Anonymous <sip:anonymous@iptel.org>;tag=DBb-18789 To: <sip:8201@iptel.org>;tag=b51ece8fd8c195776737473dd6552d43.b29b Call-ID: gnA-11414(a)61.216.39.201 CSeq: 9 INVITE P-Behind-NAT: Yes Server: Sip EXpress router (0.9.0udpfifo (i386/linux)) Content-Length: 0 Warning: 392 195.37.77.99:5060 "Noisy feedback tells: pid=30606 req_src_ip=61.216.35.30 req_src_port=5060 in_uri=sip:8201@iptel.org out_uri=sip:8201@iptel.org via_cnt==1"

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[Serusers] SERWEB add user
by han yibing 15 Aug '05

15 Aug '05

Hello I want to add user by SERWEB, please tell me which SERweb version support the function. or have third party software?? Thanks! ___________________________________________________________ 雅虎免费G邮箱－中国第一绝无垃圾邮件骚扰超大邮箱 http://cn.mail.yahoo.com

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[Serusers] SER hardware specification
by seehoe yee 15 Aug '05

15 Aug '05

Hello there, Any suggestion whether a P3 2Ghz ( dual P3 1Ghz processor ) running SER able to handle 1500 - 2000 users? Appreciate information sharing of the hardware spec. deployed for SER with the number of users :) seehoe

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[Serusers] About seradmin
by han yibing 14 Aug '05

14 Aug '05

Hello where can i find seradmin install guide? i don't know how to install the client in Windows XP OS. thanks ___________________________________________________________ 雅虎免费G邮箱－中国第一绝无垃圾邮件骚扰超大邮箱 http://cn.mail.yahoo.com

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[Serusers] DB documentation outdated?
by Federico Giannici 14 Aug '05

14 Aug '05

It seems that the database interface documentation in the current Developer's Guide is very outdated. For example, the bind_dbmod() function is said to have no parameters and indeed it seems that it has two parameters! Is there any uptodate documentation for developing SER modules? Thanks. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

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Re: [Serusers] Distributing "OSP enable SER" binary file w/ source code
by vmathur＠transnexus.com 13 Aug '05

13 Aug '05

Hi Joris, Yes, OSP has been slow in taking off but is definitely gaining momentum. Some of the big carriers - MCI, Primus Telecommunications, NTT etc - are already using OSP to manage their networks. The protocol is fairly mature & stable now (in version 1.4 I think) and provides a lot of flexibility; you can convey presence information, capabilities information (channles, bandwidth etc), and even pricing information for each destination. I beleive that over time it will evolve as the preferred method of inter-connection for IP based communications. Vikrant ---------------------------------------------------------------------------------------------------------------------------- I remember reading about osp a few years ago but have not heard much ever since. Has anyone deployed osp? joris pouw

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[Serusers] nathelper module/rtpproxy
by harry gaillac 13 Aug '05

13 Aug '05

hello, I post the same question. Anybody would be able to explain me How nathelper module/rtpproxy could provide music on hold ? 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] RE: pa module query
by Steve Blair 13 Aug '05

13 Aug '05

Steve Blair wrote: > > I realize that. Excellent use of the exclamation points. You posted a > question > and I replied with what I thought would be helpful...sorry if it wasn't. > > -Steve > > harry gaillac wrote: > >> Hello, >> >> The problem is to get documentation for IM/presence >> !!! >> >> Harry >> --- Steve Blair <blairs(a)isc.upenn.edu> a écrit : >> >> >> >>> The following should be enough to get you started >>> with the presence (pa) module. >>> The xcap stuff I do not yet have documented or >>> working properly so I cannot help >>> with that. >>> >>> -Steve >>> >>> loadmodule "/usr/local/lib/ser/modules/pa.so" # enables >>> presence module >>> >>> # >>> # ------------- pa settings >>> # >>> modparam("pa", "default_expires", 1800) >>> modparam("pa", "timer_interval", 10) >>> modparam("pa", "use_db", 0) # >>> def=0 disable db support >>> modparam("pa", "use_place_table", 0) # >>> def=0 disable use of place table >>> modparam("pa", "use_bsearch", 0) # >>> def=0 disable use of binary search for location >>> modparam("pa", "use_location_package", 0) # >>> def=0 disable use of location package >>> modparam("pa", "pa_domain", "bigU.edu") # >>> def name of xcap server >>> modparam("pa", "db_url", >>> "mysql://ser:tespaserv@localhost/ser") # def db access string >>> modparam("pa", "presentity_table", "presentity") >>> modparam("pa", "presentity_contact_table", >>> "presentity_contact") >>> modparam("pa", "watcherinfo_table", "watcherinfo") >>> modparam("pa", "place_table", "place") >>> modparam("pa", "new_watcher_pending", 0) # >>> def=0 default state of watchers 0=active >>> modparam("pa", "pidf_priority", 1) # >>> def=1 default priority of contacts in pidf documents >>> >>> if (uri==myself) { >>> ... >>> # >>> # Handle publish and subscription requests >>> # >>> >>> if (method=="PUBLISH") { >>> if (!t_newtran()) { >>> xlog("L_INFO", "\n[SER]: PUBLISH newtran >>> problem: Time: [%Tf] From: <%fu> Source IP: <%is> \n"); >>> sl_reply_error(); >>> }; >>> handle_publish("registrar"); >>> xlog("L_INFO", "\n[SER]: Handle PUBLISH: >>> Time: [%Tf] From: <%fu> To <%tu> Source IP: <%is> \n"); >>> break; >>> }; >>> >>> if (method=="SUBSCRIBE") { >>> if (!t_newtran()) { >>> xlog("L_INFO", "\n[SER]: SUBSCRIBE newtran >>> problem: Time: [%Tf] From: <%fu> Source IP: <%is> \n"); >>> sl_reply_error(); >>> }; >>> handle_subscription("registrar"); >>> xlog("L_INFO", "\n[SER]: Handle SUBSCRIBE: >>> Time: [%Tf] From: <%fu> To <%tu> Source IP: <%is> \n"); >>> break; >>> }; >>> >>> .... >>> >>> harry gaillac wrote: >>> >>> >>> >>>> Hello, >>>> >>>> I would be happy to help you however I can't help >>>> >>> >>> you >>> >>> >>>> for setting pa module IM/presence because of I >>>> >>> >>> tried >>> >>> >>>> to do it myself. >>>> >>>> Try to join Jamey Hicks for help. >>>> I can't find doc to setup pa module and xcap >>>> >>> >>> server. >>> >>> >>>> I'm sorry ! >>>> >>>> Harry >>>> >>>> >>>> --- meera.balu(a)wipro.com a écrit : >>>> >>>> >>>> >>>> >>>> >>>>> Hi Harry >>>>> >>>>> I found your email id from SER development >>>>> >>>> >>> mailing >>> >>> >>>>> list >>>>> >>>>> You seem to be testing/enhancing "pa" module of >>>>> >>>> >>> SER >>> >>> >>>>> Iam actually testing the pa module and would need >>>>> your help in setting >>>>> up & doing basic testing. >>>>> >>>>> I have loaded the '"pa" module in ser.cfg >>>>> >>>>> I would like to know >>>>> >>>>> * Would kphone4.2 (linux based soft phone) be >>>>> >>>> >>> enough >>> >>> >>>>> to test the >>>>> "presence" aspects (Updating states, subscribing >>>>> >>>> >>> to >>> >>> >>>>> users, getting >>>>> notified of change in user states, etc.) of 'pa' ? >>>>> * Is their any SQL table available for presence ? >>>>> (from where did >>>>> you get pa.sql - >>>>> >>>>> >>>>> >>>> >>> http://lists.iptel.org/pipermail/serdev/2005-March/004129.html) >>> >>> >>>> >>>> >>>> >>>> >>>>> * Other than loading the pa module in ser.cfg >>>>> >>>> >>> file, >>> >>> >>>>> what are the >>>>> other configuration changes or settings that I >>>>> >>>> >>> need >>> >>> >>>>> to be doing for >>>>> verifying presence agent module >>>>> >>>>> Any pointers would be of much help. >>>>> >>>>> Thanks >>>>> Meera >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> Confidentiality Notice >>>>> >>>>> The information contained in this electronic >>>>> >>>> >>> message >>> >>> >>>>> and any attachments to this message are intended >>>>> for the exclusive use of the addressee(s) and may >>>>> contain confidential or privileged information. If >>>>> you are not the intended recipient, please notify >>>>> the sender at Wipro or Mailadmin(a)wipro.com >>>>> immediately >>>>> and destroy all copies of this message and any >>>>> >>>>> >>>> >>>> attachments. >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>> >>> ___________________________________________________________________________ >>> >>> >>> >>> >>>> Appel audio GRATUIT partout dans le monde avec le >>>> >>> >>> nouveau Yahoo! Messenger >>> >>>> Téléchargez cette version sur >>>> >>> >>> http://fr.messenger.yahoo.com >>> >>> >>>> _______________________________________________ >>>> Serusers mailing list >>>> serusers(a)lists.iptel.org >>>> http://lists.iptel.org/mailman/listinfo/serusers >>>> >>>> >>>> >>> >> >> >> >> >> >> >> >> ___________________________________________________________________________ >> 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] lcr module
by Sebastian Kühner 13 Aug '05

13 Aug '05

Hi! I found a patch for the lcr module on this website: http://www.linguin.org/ser.php#lcr-cap Which format has it? I tried to unpack it, put it didn't work with tar -xzvf. I have ser-0.9.3 running on my system and I need prefix support for my routing. Has anyone of you an idea how to fix that??? The problem is that before the next_gw() command I need to delete this prefix... Thanks! Sebastian

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[Serusers] invite using sipsak
by Jack Wei 13 Aug '05

13 Aug '05

does anyone know how to implement the following: ua1 is connected to ua2. insert an INVITE to ua3. now all 3 uas are connected so they can speak to each other. jack Jack Wei ____________________________________________________ Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs

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[Serusers] What is the meaning of "Noisy Feedback"?
by JamesGemini 13 Aug '05

13 Aug '05

Does the message "Noisy Feedback tells" mean that the server is configured to include debugging information in its communications? Kind of like "verbose" mode?

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[Serusers] Stateful vs Stateless
by Hernando Maisano 12 Aug '05

12 Aug '05

Hello, How know the true difference between STATEFUL and STATELESS? Somebody that can help me? Nando

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Re: [Serusers] Distributing "OSP enable SER" binary file w/ source
by vmathur＠transnexus.com 12 Aug '05

12 Aug '05

Hi Jan, This sounds good. I have another question: How do you deal with dependencies on external libraries. Do you store the libraries in CVS? For the OSP module, the toolkit library is required, and I am wondering what is the best way of providing it to the users. Also, does IPTell or anyone else distribute pre-compiled versions of SER? I am trying to explore the possiblities of distributing a OSP enabled version of SER for those who dont want to go through the trouble of compiling the OSP library. Vikrant

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Re: Tr: Re: [Serusers] hardphones MOH
by harry gaillac 12 Aug '05

12 Aug '05

Maxim Sobolev, I posted you many mails you did not reply. i won't waste time to send mails for MOH support every weeks. keep this functionality in your private repo. Have good days Harry --- Maxim Sobolev <sobomax(a)portaone.com> a écrit : > Yes, it will. I have this functionality in my > private repo, bit > unfortunately due to lack of free time I have not > merged support for > this feature into the public version of nathelper > yet. We have some > heavy changes that aren't applicable for the public > version, which has > to be filtered out. > > I am planning to do it sooner or later, maybe if you > are really > interested and can allocate some small amount of > money to this project I > can subcontract some of my friends/colleagues to do > it for me. > > -Maxim > > harry gaillac wrote: > > Hello, > > > > Does nathelper module + rtpproxy will be able to > ac as > > a 3cpp to provide MOH > > > > caller--------ser---------callee > > nathelper > > rtpproxy > > Regards > > Harry > > Remarque : message transféré en pièce jointe. > > > > > > > > > > > > > > > ___________________________________________________________________________ > > Appel audio GRATUIT partout dans le monde avec le > nouveau Yahoo! Messenger > > Téléchargez cette version sur > http://fr.messenger.yahoo.com > > > > > > > ------------------------------------------------------------------------ > > > > Subject: > > Re: [Serusers] hardphones MOH > > From: > > harry gaillac <gaillacharry(a)yahoo.fr> > > Date: > > Thu, 30 Jun 2005 23:01:32 +0200 (CEST) > > To: > > Nils Ohlmeier <lists(a)ohlmeier.org> > > > > To: > > Nils Ohlmeier <lists(a)ohlmeier.org> > > CC: > > serusers(a)lists.iptel.org > > > > > > What do you advise me ? > > > > I wish to use SER+Sems(ivr) to provide MOH. > > I've got two polycom ip300 however we just can > place > > on hold caller without music ! > > > > What can I do? what's the way? > > Does Ser or sems must act as 3cpp ? > > > > Regards > > Harry > > --- Nils Ohlmeier <lists(a)ohlmeier.org> a écrit : > > > > > >>snom UA's do, except that they spare the empty > offer > >>to the MOH server, but > >>send directly the SDP of the holdee to the MOH > >>server and expecting that the > >>server supports the chosen codec. > >>Beware that this 3pcc MOH alternative completely > >>breaks if there is any NAT > >>involved. > >> > >> Nils > >> > >>On Thursday 30 June 2005 22:29, harry gaillac > wrote: > >> > >>>hello, > >>> > >>>Anybody would be informed about ip phones able to > >>>provide moh according to > >>>http://www.tech-invite.com/Ti-sip-service-3.html > >>> > >>>regards > >>>harry > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >> > > > ___________________________________________________________________________ > > > >>>Appel audio GRATUIT partout dans le monde avec le > >> > >>nouveau Yahoo! Messenger > >> > >>>Téléchargez cette version sur > >> > >>http://fr.messenger.yahoo.com > >> > >>>_______________________________________________ > >>>Serusers mailing list > >>>serusers(a)lists.iptel.org > >>>http://lists.iptel.org/mailman/listinfo/serusers > >> > > > > > > > > > > > > > > > > > ___________________________________________________________________________ > > > Appel audio GRATUIT partout dans le monde avec le > nouveau Yahoo! Messenger > > Téléchargez cette version sur > http://fr.messenger.yahoo.com > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > ___________________________________________________________________________ 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: pa module query
by harry gaillac 12 Aug '05

12 Aug '05

Hello, I would be happy to help you however I can't help you for setting pa module IM/presence because of I tried to do it myself. Try to join Jamey Hicks for help. I can't find doc to setup pa module and xcap server. I'm sorry ! Harry --- meera.balu(a)wipro.com a écrit : > > Hi Harry > > I found your email id from SER development mailing > list > > You seem to be testing/enhancing "pa" module of SER > > Iam actually testing the pa module and would need > your help in setting > up & doing basic testing. > > I have loaded the '"pa" module in ser.cfg > > I would like to know > > * Would kphone4.2 (linux based soft phone) be enough > to test the > "presence" aspects (Updating states, subscribing to > users, getting > notified of change in user states, etc.) of 'pa' ? > * Is their any SQL table available for presence ? > (from where did > you get pa.sql - > http://lists.iptel.org/pipermail/serdev/2005-March/004129.html) > * Other than loading the pa module in ser.cfg file, > what are the > other configuration changes or settings that I need > to be doing for > verifying presence agent module > > Any pointers would be of much help. > > Thanks > Meera > > > > > > > Confidentiality Notice > > The information contained in this electronic message > and any attachments to this message are intended > for the exclusive use of the addressee(s) and may > contain confidential or privileged information. If > you are not the intended recipient, please notify > the sender at Wipro or Mailadmin(a)wipro.com > immediately > and destroy all copies of this message and any attachments. ___________________________________________________________________________ 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: FW: [Serusers] ivr module to manage voice messages
by harry gaillac 12 Aug '05

12 Aug '05

In fact I expect to use asterisk 1.2 for all telephony features IM/presence, ... I would just setup up ser + rtp proxy for far end nat and proxying . I agree you now when you say "i dropped sems along time ago cause asterisk is better." which damage sems+ser+serweb don't provide a good open source telephony project. My I call you via sip if possible send me you address out of list. regards harry --- Iqbal <iqbal(a)gigo.co.uk> a écrit : > if u want some help with asterisk let me know, it > failry simple to get > work with ser, i dropped sems along time ago cause > asterisk is better. > Have voicemail and pbx, what u want to watch out for > is how you are > grouping callers etc, what I do is all users > (corporates ) who need > features of asterisk get an entry in grp table in > ser, which says > corporate, only these calls goto asterisk, the rest > stay in ser, which > saves overhead, cause asterisk does slow things > down. > > Iqbal > > harry gaillac wrote: > > >Hello, > > > >Thanks I think I 'll do it myself when asterisk 1.2 > >will be available i will set up it like most of > people > >instead of sems. > > > >A few people are interested in sems . > > > >I' ll use asterisk as sip media server ser as proxy > >and rtpproxy (far end nat) . > > > >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] Distributing "OSP enable SER" binary file w/ source code
by vmathur＠transnexus.com 12 Aug '05

12 Aug '05

Hi Jan, The code has been written as a SER module. The details of the code are available at: http://developer.berlios.de/projects/osp-module Right now the process of compilation requires that you download the OSP toolkit source code and complie it. This would build the OSP client library that the SER would use. You should then download the SER and the OSP modules and build them. The process is rather lengthy and we were wondering if we come up with an easier way of doing this. What exactly is the experimental branch in CVS ? Thanks, Vikrant --------------------------------------------------------------------------------------------------- In what form does the come come ? Did you implement it as a SER module ? If so then we can put it in the experimental branch in CVS. If you made some changes to other modules or the core then it would be great if you could generate patches against the latest development version. Jan.

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Re: [Serusers] Distributing "OSP enable SER" binary file w/ source code
by vmathur＠transnexus.com 12 Aug '05

12 Aug '05

Sure Juan, I am currently working on a document that details how to compile OSP with SER, and also a white paper which tells you about the benefits of using OSP with SER. It should be ready soon :) and will send it across. In the meantime if you want to learn more about OSP, you can go to www.transnexus.com. By the way, we have also recently implemented OSP for use with the Asterisk PBX. So, if anyone here wants to use OSP for routing, security, or billing with Asterisk, let me know and I can provide more details. Thanks, Vikrant ---------------------------------------------------------------------- Hi Vikrant. It seems very interesting to me. I would like you to send me the info you have. Regards Juan ------------------------------------------------------------ From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of vmathur(a)transnexus.com Sent: Friday, August 12, 2005 11:00 AM To: hernan_gomez_1(a)yahoo.com Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] Distributing "OSP enable SER" binary file w/ source code Hernan, OSP uses Public key based authentication and encrytion schemes, which are stronger than Radius's shared secrets. If you use OSP for authentication and authorization, you need not use Radius. OSP, just like Radius, has a client stack, which is implemented in the SER, and has a Server, which provides centralized call routing, accounting, and security. At the time of startup, the OSP server and the client, the SER in this case, exchange their public keys. The public and private keys can then be used to encode/decode messages as desired. A typical call setup procedure looks like this 1. The source SER goes to the osp server to get the SIP URI corresopnding to the destination. The message is encoded using the source SER's private key. 2. The osp server decodes the message using source SERs public key and after successful decoding (authentication) returns the route back to the source. Along with the route, it also sends back a digitally signed (using the OSP Servers private key) token. 3. The Source uses the route returned by the OSP Srver to send an INVITE. The INVITE message contains the token issued by the OSP Server 4. The destination decodes/validates the token using the OSP Servers public key. Upon successful validation (authorization) it accepts the call. This saves you the effort of mantaining cumbersome access lists for authentication. I can provide you with more documentatin on OSP and how to use SER with OSP if you wish Thanks, Vikrant ------------------------------------------------------------------------------------------------------------- Fogive my ignorance for I have never heard of OSP before ;) You mentioned Radius in your message. How does Radius authentication work in OSP? I am having a tough time getting mine to work. hernan vmathur(a)transnexus.com wrote: Dear All, I have recently implemented OSP w/ SER. OSP is an ETSI defined protocol, which I am using for ceneralised routing, and security of my inter-domain calls. The problem, however, is that the build process is a little lengthy. I want to contribute my implementation to this group so that anyone who is struggling with SER routing configurations or Radius authentication issues may benefit from it. I was, thus, wondering if we can have a binary file of the OSP enabled SER, that can be distributed with the source code. Does anyone have an opinion on this? Also, for anyone who wants to check-out this implementation, I can provide more details. Thanks, Vikrant

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[Serusers] Users authentication scheme
by Federico Giannici 12 Aug '05

12 Aug '05

I'm relatively new to SIP and I'm learning how to configure SER for a little ISP. I'm currently asking myself when we should authenticate users. Obviously, I don't wont to have an open-relay SIP server. So I'm thinking that I have to authenticate users for every message that comes and that have a "From:" header that matches one of our domains. Is this correct? Then I have to call check_to() for REGISTER messages and check_from() for all the others. Is this correct? So here it is a scheme of the logic I'm going to implement. Do you think is correct? IF uri == myself IF method == REGISTER www_authenticate() check_to() save() ELSE IF From == myself proxy_authenticate() check_from() Normal processing ELSE IF From == myself proxy_authenticate() check_from() t_relay() ELSE Error! Thanks. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

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[Serusers] vmstat: a lot of CS while writing memory back to database
by Markus Monka 12 Aug '05

12 Aug '05

Hi, can someone explain me, why we got a lot of context switches while write back the memory to database? modparam("usrloc", "db_mode", 2) modparam("usrloc", "timer_interval", 30) Looks like we have 2000 querys in that periode. procs -----------memory---------- ---swap-- -----io---- --system-- ----cpu---- r b swpd free buff cache si so bi bo in cs us sy id wa 0 0 10716 15752 111820 793044 0 0 0 136 1292 2491 1 1 98 0 0 0 10716 15752 111820 793048 0 0 0 0 1079 1935 1 1 98 0 12 0 10716 15616 111824 793136 0 0 0 0 1896 298403 10 69 21 0 13 0 10716 15616 111824 793136 0 0 0 0 2259 400436 12 88 0 0 0 0 10716 15608 111824 793152 0 0 0 0 1691 101200 5 25 70 0 Would by write through a better solution for saving data in db? While SER is saving the Data, some register request are timing out. test by sipsak: All usrloc tests completed successful. received last message 12.012 ms after first request (test duration). All usrloc tests completed successful. received last message 12.069 ms after first request (test duration). All usrloc tests completed successful. received last message 9.171 ms after first request (test duration). All usrloc tests completed successful. received last message 8.777 ms after first request (test duration). timeout after 500 ms All usrloc tests completed successful. received last message 6.185 ms after first request (test duration). All usrloc tests completed successful. received last message 7.754 ms after first request (test duration). All usrloc tests completed successful. received last message 7.645 ms after first request (test duration). All usrloc tests completed successful. received last message 10.153 ms after first request (test duration). timeout after 500 ms timeout after 1000 ms version: ser 0.8.14-4 (i386/linux) flags: STATS:Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 @(#) $Id: main.c,v 1.168.4.3.2.1 2004/10/22 17:21:32 andrei Exp $ main.c compiled on 12:47:12 Jul 12 2005 with gcc 3.3 Greets Markus

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[Serusers] Simple SIP proxy hardware
by Federico Giannici 12 Aug '05

12 Aug '05

As we do NOT want to provide a central RTP proxy, we are looking for a simple SIP proxy (in hardware) to install in small/medium offices to simplify/bypass all NAT problems. It should: 1) Work as a SIP proxy server: should use a STUN server to bypass all NAT problems (obviously a GOOD implementation of STUN) and forward all SIP requests to a downstream proxy. 2) Work as a registrat proxy: short-circuit all INVITEs between internal UACs (to bypass all hairpin problme of the NAT) and forward all SIP requests to a downstream proxy. ...and optionally... 3) Handle all the authentication requests from the downstream proxy. In this way the configuration of the phones is very simple, only phone number and sip proxy, no STUN server, no hairpin problems, etc... If 3) is implemented, they don't even have to know the username/password used to authenticate (so they cannot make free phone calls from home!). Does something like that exist? I mean, something that costs around a couple hundred of euro??? Thanks. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

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RE: [Serusers] AVP string concatenation
by Somlai Gábor 12 Aug '05

12 Aug '05

> Hi! > > You can use avp_printf. I do not know if it is also supported in ser. > But I use it with openser. Thanks all of your answers, I've found it in openser, but I have ser. I do not want to migrate because of a simple concatenation. Is not there any solution in ser? Or can I use avpops module of openser in ser? Are they binary or source compatible? Thanks, Gabor Somlai > > regards, > klaus > > Somlai Gábor wrote: > > Hello, > > > > > > > > How should I concatenate the values of two AVPs? > > > > I have not found the answer in the docs and in archive, maybe I'm lame.. > > > > > > > > Thanks, > > > > Gabor Somlai > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers

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[Users] openser core cookbook
by Daniel-Constantin Mierla 12 Aug '05

12 Aug '05

Hello, the OpenSER Core Cookbook dokuwiki got a very good shape in meanwhile and looks as an useful reference documentation for OpenSER core. The percentage of the items with text around is quite high now, just several functions and parameters are still empty, fewer and fewer day by day... http://openser.org/dokuwiki/doku.php?id=openser_core_cookbook Thanks to all contributing there. Daniel

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RE: FW: [Serusers] ivr module to manage voice messages
by harry gaillac 12 Aug '05

12 Aug '05

Hello, Thanks I think I 'll do it myself when asterisk 1.2 will be available i will set up it like most of people instead of sems. A few people are interested in sems . I' ll use asterisk as sip media server ser as proxy and rtpproxy (far end nat) . Regards Harry --- jmenjor(a)comcast.net a écrit : > Harry, > > We have a fully functional VoiceMail system that > does what you are looking for but uses Asterisk. > > The cost of installation and configuration is $700. > > Regards, > Patrick J. Menjor > -- > President & CTO > 360freephone Corporation > Tallahassee, Florida. > Phone:1.850.591.8315 > Fax:1.850.656.6975 > > -------------- Forwarded Message: -------------- > From: harry gaillac <gaillacharry(a)yahoo.fr> > To: serusers(a)lists.iptel.org > Cc: sems(a)lists.iptel.org > Subject: [Serusers] ivr module to manage voice > messages > Date: Wed, 10 Aug 2005 15:57:51 +0000 > > Hi all, > > I need help from developpers to write an python > script. > I 'm not really able to write this script however it > > could improve sems features. > > If we store voice messages on a imap server can we > write a python script (imaplib) to manage messages > from > anywhere. > > Can we set a main menu with recorded messages in > order > to read play delete voice messages ? > > phone -------- sems--------- imap server > > Does imaplib can act as a imap client? > > what are the main problems between imap protocol and > > sems? > > For example I want to listen my voices messages so i > > must send login password to imap server via sems > (DTMF) > for authentication. > If I press a key to read messages imap server send > command to sems . > How sems could play voice messages from imap server > ? > > thanks for suggestions > > Harry > > > > > > > > > > > > > > > > ___________________________________________________________________________ > > Appel audio GRATUIT partout dans le monde avec le > nouveau Yahoo! Messenger > Téléchargez cette version sur > http://fr.messenger.yahoo.com > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers ___________________________________________________________________________ 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] Billing for origination
by Marc Khayat 12 Aug '05

12 Aug '05

Hi all, I'm using SER as originating SIP server, and putting all the logs in a MySQL table. Since the international calls are as 00x or 00xx or 00xxx (no specific length), is there a nice way to do the billing based on the country code? Thanks, Marc

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Re: [Users] rport in via header
by Daniel-Constantin Mierla 12 Aug '05

12 Aug '05

Hello, I understood the issue you have. I just added on CVS a new function ( add_local_rport() ) that appends the rport parameter to the via header generated by OpenSER. You have to call this function for any SIP request that your proxy will forward and you want to have the rport parameter. The pserver cvs may take some time to synchronize (sourceforge policy). Please let me know if it works. Daniel On 08/12/05 08:10, ranveer kunal wrote: >correction : > >On 8/12/05, ranveer kunal <ranveerkunal(a)gmail.com> wrote: > > >>hi daniel, >> no basically gateway is not appending any via header, its >>acting as NAT. >>ok i'll explain the scenario onece again. >> >>proxy is behind the gateway, so any packet going out of proxy is >>natted out using gateway's ip (ip2) and port (port2), the via header >>is appended by the proxy, it has ip1:port1. ok. >>Now when next hop proxy gets the packet, it will reply to the ip from >> >> > >***********> which it got the packet ie ip1 and the port from top via >header that >***********> is port2. >which it got the packet ie ip2 and the port from top via header that >is port1. > > > > >>Ithink now you'll get the real picture. >> >>Regards, >>Ranveer. >> >> >>On 8/12/05, Daniel-Constantin Mierla <daniel(a)voice-system.ro> wrote: >> >> >>>Hello, >>> >>>so actually the gateway is a NAT, and it is not SIP aware at all. From >>>the picture I got that the gateway adds also a via header with >>>ip2:port2. In this case, I understand your question, I will double check >>>how via header is built inside openser and come back with answer during >>>the day (too late now). >>> >>>Daniel >>> >>>On 08/11/05 21:06, ranveer kunal wrote: >>> >>> >>> >>>>hi, >>>> RFC 3581 says if the the transport layer is UDP, the reply should >>>>be sent to ip of ip packet received and the port shud be taken from >>>>the via header, untill unless rport is there. >>>> >>>>since in via header port is port1(it was written by proxy). next hop >>>>proxy sends the reply to ip2:port1. >>>> >>>>I have figured a work around, i'll force gateway to send the packet >>>> >>>> >>>>from port=port1. Thus the problem will be solved. >>> >>> >>>>Thanks. >>>> >>>>Regards, >>>>Ranveer. >>>> >>>> >>>>On 8/11/05, Daniel-Constantin Mierla <daniel(a)voice-system.ro> wrote: >>>> >>>> >>>> >>>> >>>>>Hello, >>>>> >>>>>On 08/11/05 04:33, ranveer kunal wrote: >>>>> >>>>> >>>>> >>>>> >>>>> >>>>>>hey daniel, >>>>>> The packet from the ser proxy is statelessly forwarded >>>>>>by a gateway that is inline with proxy, now i want next hop proxy to >>>>>>reply on the port of the gateway rather than port of proxy. that is y >>>>>>i need proxy to send rport. >>>>>> >>>>>>proxy--------------gateway--------------next hop proxy >>>>>> >>>>>>proxy -> ip1:port1-> gateway -> ip2:port2 -> next hop proxy >>>>>>next hop proxy -> replies to ip2:port1 (hybrid) -> screws my setup >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>the next hop proxy is broken, why does it reply to ip2:port1? It should >>>>>reply to source_ip and source_port of the received request. The proxy is >>>>>not aware at all about ip2:port2 used by gateway. >>>>> >>>>>Daniel >>>>> >>>>> >>>>> >>>>> >>>>> >>>>>>can you suggest me some way to overcome this. >>>>>> >>>>>>Regards, >>>>>>Ranveer. >>>>>> >>>>>>On 8/11/05, Daniel-Constantin Mierla <daniel(a)voice-system.ro> wrote: >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>>Hello, >>>>>>> >>>>>>>On 08/10/05 21:10, ranveer kunal wrote: >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>>>hi daniel, >>>>>>>> as far as i have explored force_rport(), places the rport >>>>>>>>parameter in the topmost via header and not in its own via header. am >>>>>>>>i right ? >>>>>>>> >>>>>>>>example : >>>>>>>> >>>>>>>>UAC -> (via1:via2:via3)proxy(force_rport)->(via1:via2:via3,rport:via4)next proxy >>>>>>>> >>>>>>>>i want this : >>>>>>>>UAC -> (via1:via2:via3)proxy(force_rport)->(via1:via2:via3:via4,rport)next proxy >>>>>>>> >>>>>>>>Please clarify it ? >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>yes, it is like that. And this is how it is supposed to be, according to >>>>>>>RFC3581. Why do you need it in proxy's via? The proxy will look at the >>>>>>>second via to get the destination address for reply. >>>>>>> >>>>>>>Daniel >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>>>Regards, >>>>>>>>Ranveer. >>>>>>>> >>>>>>>>On 8/10/05, Daniel-Constantin Mierla <daniel(a)voice-system.ro> wrote: >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>>>Hello, >>>>>>>>> >>>>>>>>>you have to use force_rport() in the configuration file for the request >>>>>>>>>you want rport parameter in VIA. >>>>>>>>> >>>>>>>>>Daniel >>>>>>>>> >>>>>>>>> >>>>>>>>>On 08/10/05 20:44, ranveer kunal wrote: >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>>>hi all, >>>>>>>>>> Is there a way to configure openser so that it sends rport in >>>>>>>>>>via header ? Actually th packet from server is reverse natted through >>>>>>>>>>another system using any random port, i want it the reply to come on >>>>>>>>>>the same port . thanks .... >>>>>>>>>> >>>>>>>>>>Regards, >>>>>>>>>>Ranveer. >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>> >>>> >>>> >>>> >>-- >>Memories : They bring Diamonds and Rust >> >> >> > > > >

1 0

[Serusers] Trusted Table Format
by sermail 12 Aug '05

12 Aug '05

Dear All, I'm getting confused, what is the contents of the trusted table is I am using the allow_trusted function and want to accept INVITES from a particular IP address ? Many Thanks Simon

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[Serusers] Distributing "OSP enable SER" binary file w/ source code
by vmathur＠transnexus.com 12 Aug '05

12 Aug '05

Dear All, I have recently implemented OSP w/ SER. OSP is an ETSI defined protocol, which I am using for ceneralised routing, and security of my inter-domain calls. The problem, however, is that the build process is a little lengthy. I want to contribute my implementation to this group so that anyone who is struggling with SER routing configurations or Radius authentication issues may benefit from it. I was, thus, wondering if we can have a binary file of the OSP enabled SER, that can be distributed with the source code. Does anyone have an opinion on this? Also, for anyone who wants to check-out this implementation, I can provide more details. Thanks, Vikrant

2 1

[Serusers] Any comments about WellTech FXO/FXS device
by Sebastian Milioto 12 Aug '05

12 Aug '05

We are an ISP about to deploy an VoIP enterprise solution for about 50 costumers. We are looking for fxo/fxs, fxo and fxs devices. We know about Sipura, Welltech, Cisco, Patton. Which one do you recommend taking into account : 1. Easy installation and configuration 2. Performance 3. Durability 4. Simultaneous calls, etc May be you could reccomend another brand also. Thank you very much for your time Sebastian Milioto Telecommunication Engineer -------------------------------- IM: sebamilioto(a)hotmail.com Mobile: +549-3571-15540598 e-mail: smilioto(a)GMAIL.com --------------------------------

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[Users] RR module changes
by Bogdan-Andrei Iancu 12 Aug '05

12 Aug '05

Hi everybody, I made some updates to the RR module on both users and developer sides: for users: new is_direction() function; more docs about dialog support via RR module for developers: RR API added; RR callbacks added; docs describing both of these (with some examples). For more and more info, please refer to the online doc: http://www.openser.org/docs/modules/0.10.x/rr.html The RR API and callbacks are not 100% tested, it's on the way - they will be used by the UAC module (for FROM/TO mangling) and by other future module which will need dialog support. regards, Bogdan

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[Serusers] ser fails after mysql restart
by Mike Williams 12 Aug '05

12 Aug '05

Hello, OpenSER 0.9.5 fails if the connection to MySQL is broken or times out while ser is running. Is there any way to have the server reconnect intelligently? I currently have the MySQL timeout set to five days, but I would prefer a real fix. Ser and mysql are both running on the same server. Thanks, Mike Williams Here's some of my syslog output: Aug 12 07:33:43 sip /usr/local/sbin/openser[12542]: submit_query(): MySQL server has gone away Aug 12 07:33:43 sip /usr/local/sbin/openser[12542]: submit_query(): Error while submitting query Aug 12 07:33:43 sip /usr/local/sbin/openser[12542]: get_ha1(): Error while querying database Aug 12 07:33:51 sip /usr/local/sbin/openser[12558]: submit_query(): MySQL server has gone away Aug 12 07:33:51 sip /usr/local/sbin/openser[12558]: submit_query(): Error while submitting query Aug 12 07:33:51 sip /usr/local/sbin/openser[12558]: get_ha1(): Error while querying database Aug 12 07:34:11 sip /usr/local/sbin/openser[12542]: submit_query(): MySQL server has gone away Aug 12 07:34:11 sip /usr/local/sbin/openser[12542]: submit_query(): Error while submitting query Aug 12 07:34:11 sip /usr/local/sbin/openser[12542]: get_ha1(): Error while querying database Aug 12 07:34:13 sip /usr/local/sbin/openser[12558]: submit_query(): MySQL server has gone away Aug 12 07:34:13 sip /usr/local/sbin/openser[12558]: submit_query(): Error while submitting query Aug 12 07:34:13 sip /usr/local/sbin/openser[12558]: get_ha1(): Error while querying database

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[Serusers] SQL Query and Accounting
by c waddy 12 Aug '05

12 Aug '05

I have followed all the advice i could find in the mailing list and google. I have mysql.so in modules directory, uncommented the sql bit in the makefile, I used "make all" when compiling. The config has the relevant parameters set i think: # --acc params-- modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) modparam("acc", "log_fmt", "fimos") #modparam("acc", "db_url", "sql://ser:heslo@localhost/ser") #modparam("acc", "db_missed_flag", 2) #modparam("acc", "db_flag", 1) modparam("usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("usrloc", "db_mode", 2) Also in SERWEB, when i click the accounting tab i get an error -> error in SQL query (1), line: 48 No calls The rest of serwbe works fine for adding users etc. Can anyone point me in the right direction.

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[Serusers] "undefined ERR_get_error_line_data" Error when loading mysql(ser 0.9.3)
by iceboy 12 Aug '05

12 Aug '05

Dear serusers, I have downloaded the ser-0.9.13 source code and compiled it on a FC4. It can be started succesfully WITHOUT mysql support and I can make phone alls. But when I add the mysql modules in ser.cfg and start ser, it shows the following error message: 0(20691) ERROR: load_module: could not open module </usr/local/lib/ser/modules/mysql.so>: /usr/local/lib/ser/modules/mysql.so: undefined symbol: ERR_get_error_line_data 0(20691) parse error (29,13-14): failed to load module ERROR: bad config file (1 errors) I checked the mysql.so file and the database and both are right. Can anyone give me a help? Thanks. Best Regards, iceboy

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[Serusers] Fw: turning 30x blind forward to INVITE
by Antanas Masevicius 12 Aug '05

12 Aug '05

Hello, this topic was last discussed here: http://lists.iptel.org/pipermail/serusers/2005-May/019551.html last few days i was trying to apply juha's move() function with almost no success to 0.9.3 stable branch. i added such code to my ser.cfg: onreply_route[1] { xlog("L_NOTICE", "Reply status %rs: Reason %rr\n"); if (t_check_status("30[12]")) { xlog("L_NOTICE", "We got a 30[12]!!!!!\n"); t_move(); # custom function by juha #t_reply(); }; } and modified move() code like this: inline static int move(struct sip_msg* _m, char *_foo, char *_bar) { //struct sip_msg* reply; regmatch_t pmatch[2]; struct hdr_field hf; contact_t* first; regex_t contact_re; // What is it? didn't find anywere. //reply = _m->final_reply; // MOD HERE if (!_m) { LOG(L_ERR, "move(): No reply found\n"); return -1; } LOG(L_ERR, "move(): unparsed part of reply: %s\n", _m->unparsed); // FIXME: where to put this? initialisation? // MOD HERE regcomp(&contact_re, "^Contact:(.*)$", REG_EXTENDED|REG_NEWLINE); if ( ( regexec(&contact_re, _m->unparsed, 2, &(pmatch[0]), 0) != 0) || (pmatch[1].rm_so == -1)) { LOG(L_ERR, "move(): No Contact header found\n"); return -1; } hf.type = HDR_CONTACT; hf.name.len = 0; hf.body.len = pmatch[1].rm_eo - pmatch[1].rm_so - 1; hf.body.s = &(_m->unparsed[pmatch[1].rm_so]); hf.len = hf.body.len + 2; hf.parsed = NULL; hf.next = NULL; // this prints out redirected contact LOG(L_ERR, "hf.body: '%.*s'\n", hf.body.len, hf.body.s); if (parse_contact(&hf) < 0) { LOG(L_ERR, "move(): Error while parsing Contact\n"); goto failure; } if (((contact_body_t*)hf.parsed)->star == 1) { LOG(L_ERR, "move(): Contact is *\n"); goto failure; } first = ((contact_body_t*)hf.parsed)->contacts; if (first) { // MOD HERE if (append_branch(_m, first->uri.s, first->uri.len, 0, 0, Q_UNSPECIFIED) == 1) { goto success; } else { LOG(L_ERR, "move(): Appending branch failed\n"); } } else { LOG(L_ERR, "move(): No contacts in Contact header\n"); } failure: if (hf.parsed) { free_contact((contact_body_t**)(&(hf.parsed))); } return -1; success: free_contact((contact_body_t**)(&(hf.parsed))); return 1; } it seems that t_move succeeds(), contact gets parsed and append_branch() gets execuded. But then, nothing happens. 302 is replied back to call originator. It is function build_res_buf_from_sip_res is called() which builds 302 message back to call originator.. It seems that after getting contact from 302 message, INVITE sould be generatated somehow. I suspect that i am passing wrong message to append_branch, but i didn't found how to access INVITE from tm module while in onreply_route[]. Can anybody with deeper ser knowledge help? Antanas

1 0

[Users] mediaproxy problem
by jayesh nambiar 12 Aug '05

12 Aug '05

hi daniel, Thanx a lot. There was actually a problem in my configuration. I corrected it. The problem was with the reply route handling. Thanx again. Can someone pls suggest what could be the best way to handle broken calls from the configuration script. I've read earlier that one can implement it through the gateway if gateway supports it. But pls suggest the best approach to handle it through the script. And can u pls make me understand what exactly are failed transactions. Thanx again. Jayesh. ____________________________________________________ Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs

3 2

RE: [Serusers] Any comments about WellTech FXO/FXS device
by Marc Khayat 12 Aug '05

12 Aug '05

You may also wanna try Addpac Marc -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Jan Janak Sent: Friday, August 12, 2005 10:51 AM To: Sebastian Milioto Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] Any comments about WellTech FXO/FXS device I would recommend Sipura. Jan. On 11-08-2005 20:24, Sebastian Milioto wrote: > We are an ISP about to deploy an VoIP enterprise solution for about 50 > costumers. We are looking for fxo/fxs, fxo and fxs devices. We know about > Sipura, Welltech, Cisco, Patton. Which one do you recommend taking into > account : > > 1. Easy installation and configuration > 2. Performance > 3. Durability > 4. Simultaneous calls, etc > > May be you could reccomend another brand also. > Thank you very much for your time > > > Sebastian Milioto > Telecommunication Engineer > -------------------------------- > IM: sebamilioto(a)hotmail.com > Mobile: +549-3571-15540598 > e-mail: smilioto(a)GMAIL.com > -------------------------------- > > > > _______________________________________________ > 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

[Serusers] [Question] About force_rtp_proxy in nathelper module..?
by da1228 12 Aug '05

12 Aug '05

Hi all: I am confused what function will signal rtpproxy the port mapping while the nat traversal is activated? Does "force_rtp_proxy" achieve that function ? In the doc http://www.iptel.org/ser/doc/modules/txt/nathelper.txt Maxim says that "force_rtp_proxy" will "Rewrites SDP body to ensure that media is passed through an RTP proxy." Does it means the function "force_rtp_proxy" will change the port in SDP, and also signal rtpproxy the port mapping that he should transport ? Thanks in advance Brian

2 1

[Serusers] how to make webbased checksession
by Asep Saifudin 12 Aug '05

12 Aug '05

Hi all, I want to log online / talking user to database so I can put them at my website to monitor them web based. Is there anyone implement it ? what table should I use ? acctive_sessions or else ? and what should I add at my ser.cfg ? btw, is I monitor talking user shell on my server, I use checksessions but I want this checksessions at my website Rgrds, Asep

1 0

Re: [Serusers] Problem with BYE
by Samuel Osorio Calvo 12 Aug '05

12 Aug '05

You have to set first all destinations and afterwards call t_relay, which will manage all the previous destinations. To add destinations, the easyest is to use append_branch(); So: rewritehost(gateway) append_branch(B2BUI) t_relay(); Check spelling the spelling of previous commands, they are just an indication of how to use them, in the tm module README you will find the right parameters. Samuel. Unclassified. >>> Teran Jesus <jteran(a)mcmtelecom.com.mx> 08/12/05 12:09AM >>> Hello, I have a problem, i need to transfere a "BYE" to my B2BUI and to my gateway at the same time, i have used the rewritehost and the t_relay_to_udp twice but only take me the first one. ¿Can you help me? _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] test
by jeff kwong 12 Aug '05

12 Aug '05

testing. my messages doesn't seem to get posted

1 0

[Serusers] AVP string concatenation
by Somlai Gábor 12 Aug '05

12 Aug '05

Hello, How should I concatenate the values of two AVPs? I have not found the answer in the docs and in archive, maybe I'm lame.. Thanks, Gabor Somlai

2 1

[Serusers] thanx
by rupesh 12 Aug '05

12 Aug '05

hello, thank you sam lee and ashutosh kumar for ur suggestions, it worked.

1 0

[Serusers] Large Telco SER Implementation
by INNOVE Communications - Cocoy J. Claravall 12 Aug '05

12 Aug '05

Can anyone share large SER installations in a telco environment? I read somewhere about Earthlink w/ 1.6million subs on SER. Any other that you know? We would like to get some reference accounts for this platform, thanks.

1 0

[Serusers] Paid help and support
by Sebastian Muniz 12 Aug '05

12 Aug '05

Hello. We are planning an enterprise with SER + ArtDIO UA. I have already asked info(a)iptel.org for a paid expert or so to help us set up SER remotely and they have said all of their resources are busy. So, is there any person in the list willing to help us set ip up? If so, please write a personal e-mail to sebastianm(a)smartsecurity.com.mx Thanks in advance. Sebastian.

1 0

Re: [Users] rport in via header
by Daniel-Constantin Mierla 11 Aug '05

11 Aug '05

Hello, so actually the gateway is a NAT, and it is not SIP aware at all. From the picture I got that the gateway adds also a via header with ip2:port2. In this case, I understand your question, I will double check how via header is built inside openser and come back with answer during the day (too late now). Daniel On 08/11/05 21:06, ranveer kunal wrote: >hi, > RFC 3581 says if the the transport layer is UDP, the reply should >be sent to ip of ip packet received and the port shud be taken from >the via header, untill unless rport is there. > >since in via header port is port1(it was written by proxy). next hop >proxy sends the reply to ip2:port1. > >I have figured a work around, i'll force gateway to send the packet >from port=port1. Thus the problem will be solved. > >Thanks. > >Regards, >Ranveer. > > >On 8/11/05, Daniel-Constantin Mierla <daniel(a)voice-system.ro> wrote: > > >>Hello, >> >>On 08/11/05 04:33, ranveer kunal wrote: >> >> >> >>>hey daniel, >>> The packet from the ser proxy is statelessly forwarded >>>by a gateway that is inline with proxy, now i want next hop proxy to >>>reply on the port of the gateway rather than port of proxy. that is y >>>i need proxy to send rport. >>> >>>proxy--------------gateway--------------next hop proxy >>> >>>proxy -> ip1:port1-> gateway -> ip2:port2 -> next hop proxy >>>next hop proxy -> replies to ip2:port1 (hybrid) -> screws my setup >>> >>> >>> >>> >>the next hop proxy is broken, why does it reply to ip2:port1? It should >>reply to source_ip and source_port of the received request. The proxy is >>not aware at all about ip2:port2 used by gateway. >> >>Daniel >> >> >> >>>can you suggest me some way to overcome this. >>> >>>Regards, >>>Ranveer. >>> >>>On 8/11/05, Daniel-Constantin Mierla <daniel(a)voice-system.ro> wrote: >>> >>> >>> >>> >>>>Hello, >>>> >>>>On 08/10/05 21:10, ranveer kunal wrote: >>>> >>>> >>>> >>>> >>>> >>>>>hi daniel, >>>>> as far as i have explored force_rport(), places the rport >>>>>parameter in the topmost via header and not in its own via header. am >>>>>i right ? >>>>> >>>>>example : >>>>> >>>>>UAC -> (via1:via2:via3)proxy(force_rport)->(via1:via2:via3,rport:via4)next proxy >>>>> >>>>>i want this : >>>>>UAC -> (via1:via2:via3)proxy(force_rport)->(via1:via2:via3:via4,rport)next proxy >>>>> >>>>>Please clarify it ? >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>yes, it is like that. And this is how it is supposed to be, according to >>>>RFC3581. Why do you need it in proxy's via? The proxy will look at the >>>>second via to get the destination address for reply. >>>> >>>>Daniel >>>> >>>> >>>> >>>> >>>> >>>>>Regards, >>>>>Ranveer. >>>>> >>>>>On 8/10/05, Daniel-Constantin Mierla <daniel(a)voice-system.ro> wrote: >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>>>Hello, >>>>>> >>>>>>you have to use force_rport() in the configuration file for the request >>>>>>you want rport parameter in VIA. >>>>>> >>>>>>Daniel >>>>>> >>>>>> >>>>>>On 08/10/05 20:44, ranveer kunal wrote: >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>>hi all, >>>>>>> Is there a way to configure openser so that it sends rport in >>>>>>>via header ? Actually th packet from server is reverse natted through >>>>>>>another system using any random port, i want it the reply to come on >>>>>>>the same port . thanks .... >>>>>>> >>>>>>>Regards, >>>>>>>Ranveer. >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>> >>>>> >>>>> >>>>> >>> >>> >>> >>> > > > >

1 0

[Serusers] Problem with BYE
by Teran Jesus 11 Aug '05

11 Aug '05

Hello, I have a problem, i need to transfere a "BYE" to my B2BUI and to my gateway at the same time, i have used the rewritehost and the t_relay_to_udp twice but only take me the first one. ¿Can you help me?

1 0

[Users] problems with tcp_init...
by Loide Mara 11 Aug '05

11 Aug '05

Hi! When I start openser executing (openserctl start) it says me Starting SER : PID file /var/run/openser.pid does not exist -- SER start failed So, I try to watch debug messages and i see the rows DEBUG: setting SO_RCVBUF; set=262144,verify=262142 0(0) DEBUG: setting SO_RCVBUF has no effect 0(0) INFO: udp_init: SO_RCVBUF is finally 262142 0(0) ERROR: tcp_init: bind(6, 0x810580c, 16) on 127.0.0.1: Address already in use What can be the problem? =/ Has everyone some problems like that (and solutions =). Thanks. Have a good day. -------------------------------------------------- Loide Mara Verdes de Jesus 6º período - Ciência da Computação / UFAM Laboratório VoIP - UFAM

2 1

[Serusers] accounting fifo messages
by SER Users 11 Aug '05

11 Aug '05

hello, I am trying to force a call disconnect by sending a BYE to both ends. everything is ok, the call gets disconnected but I do not manage to get the BYE message accounted by the accounting module. Is it possible to do accounting on messages sent using the fifo interface, is it something special that needs to be configured ? thanks, Razvan Radu

1 0

[Serusers] TCP and UDP listeners
by Carla Ares 11 Aug '05

11 Aug '05

HI, I'm using ser 0.8.14. It opens the port 5060 udp and tcp. But I'm using only the udp port. I want to configure ser to open only the udp port. Is this possible? Carla

2 1

[Serusers] SER in production environment
by Luba Vincent 11 Aug '05

11 Aug '05

Hello, We would like to use SER in a production environment and provide presence support to at least 5000 *online* users through a publicly available service. We are particularly worried about scalability. * How stable is the presence module? * Does it support RFC3857 to allow authorization management of subscriptions? * For such a big number of users, do we need several computers running SER? * If so, how can we ensure that all SER servers have the required information to be able to send status changes to all users? Is that stored in a MySQL DB and if so, how is it determined which SER server sends the NOTIFY on state changes to all users having the user whose state changed in their list? * Are there bandwidth estimations? I have the feeling that 5000 online users will generate many NOTIFY messages due to presence changes. * Is it possible to prevent MESSAGE requests to be sent following some piece of information found in the database? We would like to connect SER to a SMS service, and prevent users to send SMS'es when their credit is not enough. That supposes some computation in the ser.cfg file, is that possible? Greetings, Luba Vincent ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Serusers] ser_fifo
by Siphiwo Dzingwe 11 Aug '05

11 Aug '05

Hi I just installed ser v0.8.14 from the *.tar.gz distribution and when I try to run serctl moni it gives me an error saying: Error opening ser's FIFO /tmp/ser_fifo Make sure you have line fifo=/tmp/ser_fifo in your config in the ser.cfg file the line fifo="/tmp/ser_fifo" is uncommented. what am I doing wrong

2 1

[Serusers] Mysql CDR time different
by TCCHAN 11 Aug '05

11 Aug '05

Hi All, I have acc module to generate cdr for calls. I noticed that all the calls are timed 4 hour different than what is shown on the real time clock. Can anyone help? Thanks for any help available. Regards, TC Chan

3 3

[Serusers] enum lookups and billing
by Iqbal 11 Aug '05

11 Aug '05

Hi Is there a easy way of doing zero billing for all datra returned from enum lookup Iqbal

3 6

RE: [Serusers] starting and stopping ser
by Marc Khayat 11 Aug '05

11 Aug '05

Mr Jan Janak helped me out on this: Use the init.d scripts (usually /etc/rc.d/init.d/ser in RedHat). So u can do: /etc/rc.d/init.d/ser start|stop|restart Marc -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Ashutosh Kumar Sent: Tuesday, August 09, 2005 5:06 PM To: serusers(a)lists.iptel.org Subject: RE: [Serusers] starting and stopping ser Hi, In my case also, using "serctl stop", gave the error ""pid file cannot be found....blah blah". And killing all running ser forked processes one by one is cumbersome. So the method I m using for the time being till I get a nice solutions is use the command "killall /usr/local/sbin/ser" , i.e the killall command is followed by the prog name along with the full path or the prog name only, depending upon where in the filesys you are standing right now. I also found the following commands to be usful when viewing running ser processes -ps -auxmchf | grep ser -pstree -Gpn | grep ser : shows ser heirachy wise, (father and sons) Regards Ashutosh ________________________________________ From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Hemant Patel (IP) Sent: Tuesday, August 09, 2005 8:23 PM To: serusers(a)lists.iptel.org Subject: [Serusers] starting and stopping ser Hi List , This is my first install and use of ser so please bear with me. I've installed ser 0.9.3 ( Linux i386 ) binary. Started and stopped ser with the default ser.cfg - fine. Amended ser.cfg to load mysql.so auth and auth_db modules as per standard instructions. Mysql version is 4.0.18. I used 'mysql_sh' script to create default ser database and ser , serro users in mysql with default password. Checked that 'admin' user also setup on ser database with default password. Have set debug level 7 in ser.cfg to follow what goes on. Logged in as normal user and then su as root So here goes.. ser start - this seems to be ok. ser stop - seems to report ser has stopped ser start - connection to 127.0.0.1 already in use message ( ???) So am I starting and stopping ser incorrectly ? What's the best way to start and stop ser - in a development environment ? Any pointers gratefully received. thanks -- Hemant Patel ipwise Ltd http://www.ipwise.com hpatel(a)ipwise.com (+44) 0208 2409433 _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

4 3

[Serusers] Is it possible to configure SER to authenticate from a third party db table?
by Gareth Holly 11 Aug '05

11 Aug '05

Hi, I'm trying to configure SER to perform proxy authentication using username and password information from a users table in my own database. Is this possible? and if it is could you please point me in the direction of configuration examples? Thanks in advance Send instant messages to your online friends http://au.messenger.yahoo.com

2 5

Re: [Users] rport in via header
by Daniel-Constantin Mierla 11 Aug '05

11 Aug '05

Hello, On 08/11/05 04:33, ranveer kunal wrote: >hey daniel, > The packet from the ser proxy is statelessly forwarded >by a gateway that is inline with proxy, now i want next hop proxy to >reply on the port of the gateway rather than port of proxy. that is y >i need proxy to send rport. > >proxy--------------gateway--------------next hop proxy > >proxy -> ip1:port1-> gateway -> ip2:port2 -> next hop proxy >next hop proxy -> replies to ip2:port1 (hybrid) -> screws my setup > > the next hop proxy is broken, why does it reply to ip2:port1? It should reply to source_ip and source_port of the received request. The proxy is not aware at all about ip2:port2 used by gateway. Daniel >can you suggest me some way to overcome this. > >Regards, >Ranveer. > >On 8/11/05, Daniel-Constantin Mierla <daniel(a)voice-system.ro> wrote: > > >>Hello, >> >>On 08/10/05 21:10, ranveer kunal wrote: >> >> >> >>>hi daniel, >>> as far as i have explored force_rport(), places the rport >>>parameter in the topmost via header and not in its own via header. am >>>i right ? >>> >>>example : >>> >>>UAC -> (via1:via2:via3)proxy(force_rport)->(via1:via2:via3,rport:via4)next proxy >>> >>>i want this : >>>UAC -> (via1:via2:via3)proxy(force_rport)->(via1:via2:via3:via4,rport)next proxy >>> >>>Please clarify it ? >>> >>> >>> >>> >>yes, it is like that. And this is how it is supposed to be, according to >>RFC3581. Why do you need it in proxy's via? The proxy will look at the >>second via to get the destination address for reply. >> >>Daniel >> >> >> >>>Regards, >>>Ranveer. >>> >>>On 8/10/05, Daniel-Constantin Mierla <daniel(a)voice-system.ro> wrote: >>> >>> >>> >>> >>>>Hello, >>>> >>>>you have to use force_rport() in the configuration file for the request >>>>you want rport parameter in VIA. >>>> >>>>Daniel >>>> >>>> >>>>On 08/10/05 20:44, ranveer kunal wrote: >>>> >>>> >>>> >>>> >>>> >>>>>hi all, >>>>> Is there a way to configure openser so that it sends rport in >>>>>via header ? Actually th packet from server is reverse natted through >>>>>another system using any random port, i want it the reply to come on >>>>>the same port . thanks .... >>>>> >>>>>Regards, >>>>>Ranveer. >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>> >>> >>> >>> > > > >

1 0

[Serusers] MEdiaproxy web page
by Ashutosh Kumar 11 Aug '05

11 Aug '05

Hi, The web page media_sessions.phtml supplied with mediaproxy doesn't show the rtp seesions active on my sip server. I tried changing the value in the server array list in this php file to the physical ip, localohost , but of no avail.The output is always the following, Server Version Caller traffic Called traffic Relayed traffic Sessions Status 1 localhost Connection refused Why is the connection refused? Regards, Ashutosh Kumar Chetu, Inc. Ph : 1(305) 402 6724 - Witin US Ph : 91 120 5323340 - Outside US Fax:1 (305) 832 5987 For more information, please visit http://www.chetu.com

1 0

[Serusers] Billing software
by Marc Khayat 11 Aug '05

11 Aug '05

Hi all, What do you recommend (and are using) as billing software freeware and/or shareware that interacts with SER (and, if possible, with GnuGK)? Thanks, Marc

1 0

[Serusers] Medaiproxy not working properly
by Ashutosh Kumar 11 Aug '05

11 Aug '05

Hi, I m runnunig mediaproxy with ser-0.9.0. Most of the times, it is working fine. But sometimes there is one way or both sides audio problem when both clients are NATTED. On digging deeper, I see as one of the participating NATTED phone having an aor (address of record) as username@userIP:5060. IS this assignment of port 5060 a problem? The problem is that more than one clients/phones in that NAT have this same AOR, which might be causing this problem. Further, no natted client should be assigned 5060 by the nat for port forwarding as far as I think, correct me if I m wrong. Additionally, I m using the default mediaproxy.ini, the only change is the proxyIP, which has the value of my machin's IP on which mediaproxy is running. Can someone send me a correctly working mediaproxy.ini config. Regards, Ashutosh Kumar Chetu, Inc. Ph : 1(305) 402 6724 - Witin US Ph : 91 120 5323340 - Outside US Fax:1 (305) 832 5987 For more information, please visit http://www.chetu.com

1 0

[Serusers] compilation of ser.cfg
by rupesh 11 Aug '05

11 Aug '05

hello , i am new to SER. i have made some modification in the ser.cfg file in ser 0.9.3 version . now i want to compile this ser.cfg file so that the changes which i have made can effect the ser, but am unable to compile. can someone help me with this. can u tell me the compilation steps.i have checked in the getting started document and the install document from iptel also, but of no use. waiting for someone to help me out. thank you, Rupesh

3 2

[Serusers] Billing software
by Marc Khayat 11 Aug '05

11 Aug '05

Hi all, What do you recommend (and are using) as billing software freeware and/or shareware that interacts with SER (and, if possible, with GnuGK)? Thanks, Marc

1 0

RE: [Serusers] compilation of ser.cfg
by Sam Lee 11 Aug '05

11 Aug '05

Just restart SER will do. Eg. Serctl restart -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of rupesh Sent: Thursday, August 11, 2005 2:24 PM To: serusers(a)lists.iptel.org Subject: [Serusers] compilation of ser.cfg hello , i am new to SER. i have made some modification in the ser.cfg file in ser 0.9.3 version . now i want to compile this ser.cfg file so that the changes which i have made can effect the ser, but am unable to compile. can someone help me with this. can u tell me the compilation steps.i have checked in the getting started document and the install document from iptel also, but of no use. waiting for someone to help me out. thank you, Rupesh _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] ser.cfg compilation
by rupesh 11 Aug '05

11 Aug '05

hello , i am new to SER. i have made some modification in the ser.cfg file in ser 0.9.3 version . now i want to compile this ser.cfg file so that the changes which i have made can effect the ser, but am unable to compile. can someone help me with this. can u tell me the compilation steps.i have checked in the getting started document and the install document from iptel also, but of no use. waiting for someone to help me out. thank you, Rupesh

1 0

[Serusers] AudioCodes
by Kapil Dhawan 11 Aug '05

11 Aug '05

Hi List I am using an AudioCodes MP-104 FXS with Version ID: 4.20.354.571. Authentication details: username/password 101/101. In End Point's Phone Number field i am using 102 instead of 101. The problem is....When REGISTER comes, it comes from 102 whereas it should come from 101. Second is When SER asks for realm and other details A/C is sending details of 101 in realm. U 192.168.1.2:5060 -> 192.168.1.1:5060 REGISTER sip:192.168.1.1 SIP/2.0..Via: SIP/2.0/UDP 192.168.1.2;branch=z9hG4bKacgsnejnS..From: <sip:102@192.168.1.1>;tag=1c66110..To: <sip:102@192.168.1.1>..Call-ID : 478522552255UyxO@192.168.1.2..CSeq: 90119 REGISTER..Expires: 60..Contact: <sip:102@192.168.1.2;user=phone>;expires=60..Content-Length: 0....# U 192.168.1.1:5060 -> 192.168.1.2:5060 SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP 192.168.1.2;branch=z9hG4bKacgsnejnS;rport=5060..From: <sip:102@192.168.1.1>;tag=1c66110..To: <sip:102@192.168.1.1>;tag=b27e1a1d33761e85846fc98f5f3a7e58.7aba..Call-ID: 478522552255UyxO@192.168.1.2..CSeq: 90119 REGISTER..WWW-Authenticate: Digest realm="192.168.1.1", nonce="42fad87e8fc59c528db3b626c769c8e55a3f4454"..Server: Sip EXpress router (0.10.99-dev14-tcp (i386/linux))..Content-Length: 0..Warning: 392 192.168.1.1:5060 "Noisy feedback tells: pid=17382 req_src_ip=192.168.1.2 req_src_port=5060 in_uri=sip:192.168.1.1 out_uri=sip:192.168.1.1 via_cnt==1"....# U 192.168.1.2:5060 -> 192.168.1.1:5060 REGISTER sip:192.168.1.1 SIP/2.0..Via: SIP/2.0/UDP 192.168.1.2;branch=z9hG4bKacbcphONy..From: <sip:102@192.168.1.1>;tag=1c66110..To: <sip:102@192.168.1.1>..Call-ID : 478522552255UyxO@192.168.1.2..CSeq: 90120 REGISTER..Contact: <sip:102@192.168.1.2;user=phone>;expires=60..Authorization:Digest username="101",realm="192.168.1.1",nonce="42fad87e8fc59c528db3b626c769c8e55a3f4454",uri="sip:192.168.1.1",Algorithm="MD5",response="14e3a3d63687571fc152c7d2487d0314"..Max-Forwards: 70..Expires: 60..User-Agent: Audiocodes-Sip-Gateway-MP-104 FXS/v.4.20.354.571..Content-Length: 0 ....# U 192.168.1.1:5060 -> 192.168.1.2:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.1.2;branch=z9hG4bKacbcphONy;rport=5060..From: <sip:102@192.168.1.1>;tag=1c66110..To: <sip:102@192.168.1.1>;tag=b27e1a1d33761e85846fc98f5f3a7e58.e20d..Call-ID: 478522552255UyxO@192.168.1.2..CSeq: 90120 REGISTER..Contact: <sip:102@192.168.1.2:5060;user=phone>;expires=60..Server: Sip EXpress router (0.10.99-dev14-tcp (i386/linux))..Content-Length: 0..Warning: 392192.168.1.1:5060 "Noisy feedback tells: pid=17378 req_src_ip=192.168.1.2 req_src_port=5060 in_uri=sip:192.168.1.1 out_uri=sip:192.168.1.1 via_cnt==1".... Any idea? _________________________________________________________________ Aamir Khan is back! http://server1.msn.co.in/SP05/Mangalpandey/index.html See him in the mustached avatar in Mangal Pandey.

1 0

[Serusers] stopping at mysql modules when start SER!!!!!!
by zhu lizhong 11 Aug '05

11 Aug '05

> > *hello, Jan Janak and other guys: > > I try to connect ser and freeradius. i also follow the hints(ser_radius > > how to) from iptel.com <http://iptel.com>. > > but when i run ser, it stopping at mysql parts and not errors. my > > ser.cfg is : > > > > # ------------------ module loading ---------------------------------- > > > > # Uncomment this if you want to use SQL database > > loadmodule "/usr/local/lib/ser/modules/mysql.so" > > loadmodule "/usr/local/lib/ser/modules/sl.so" > > loadmodule "/usr/local/lib/ser/modules/tm.so" > > loadmodule "/usr/local/lib/ser/modules/rr.so" > > loadmodule "/usr/local/lib/ser/modules/maxfwd.so" > > loadmodule "/usr/local/lib/ser/modules/usrloc.so" > > loadmodule "/usr/local/lib/ser/modules/registrar.so" > > loadmodule "/usr/local/lib/ser/modules/exec.so" > > > > loadmodule "/usr/local/lib/ser/modules/textops.so" > > > > # RADIUS support > > loadmodule "/usr/local/lib/ser/modules/acc.so" > > > > # Uncomment this if you want digest authentication > > # mysql.so must be loaded ! > > loadmodule "/usr/local/lib/ser/modules/auth.so" > > loadmodule "/usr/local/lib/ser/modules/auth_radius.so" > > > > # ----------------- setting module-specific parameters --------------- > > > > # -- usrloc params -- > > > > # 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 -- > > modparam("acc", "radius_config", "/etc/radiusclient/radiusclient.conf") > > modparam("acc", "radius_missed_flag", 2) > > modparam("acc", "radius_config", "/etc/radiusclient/radiusclient.conf") > > modparam("acc", "radius_missed_flag", 2) > > modparam("acc", "radius_flag", 1) > > > > # -- auth_radius params -- > > modparam("auth_radius", "radius_config", > > "/etc/radiusclient/radiusclient.conf") > > > > # ------------------------- 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")) { > > log(1,"LOG: Too many hops"); > > sl_send_reply("483","Too Many Hops"); > > break; > > }; > > if ( msg:len > max_len ) { > > log(1,"LOG: Message too big"); > > loose-route processing > > if(loose_route()){ > > t_relay(); > > break; > > }; > > > > if(uri==myself){ > > > > # All REGISTER attempts are processed and must always be authenticated > > if (method=="REGISTER"){ > > if (!radius_www_authorize("")) { > > www_challenge("", "0"); > > break; > > }; > > > > save("location"); > > break; > > }; > > > > I post only part of ser.cfg. the errors are : > > > > 0(0) find_mod_export: found <db_query> in module mysql > > [/usr/local/lib/ser/modules/mysql.so] > > 0(0) find_mod_export: found <db_raw_query> in module mysql > > [/usr/local/lib/ser/modules/mysql.so] 0(0) find_mod_export: found > > <db_free_result> in module mysql [/usr/local/lib/ser/modules/mysql.so] > > 0(0) find_mod_export: found <db_insert> in module mysql > > [/usr/local/lib/ser/modules/mysql.so] > > 0(0) find_mod_export: found <db_delete> in module mysql > > [/usr/local/lib/ser/modules/mysql.so] > > 0(0) find_mod_export: found <db_update> in module mysql > > [/usr/local/lib/ser/modules/mysql.so] > > thanks lot!* > > > zhu > > > >

1 0

RE: [Serusers] ser : rport
by Sam Lee 11 Aug '05

11 Aug '05

Question is ... How do u find out the current value of the rport in the via header ? -----Original Message----- From: ranveer kunal [mailto:ranveerkunal@gmail.com] Sent: Wednesday, August 10, 2005 10:09 PM To: Sam Lee Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] ser : rport I am sorry , i am a bit new to ser. Please tell me more elaborately, how to make sure that ser attaches rport to the via header. thanks, ranveer On 8/10/05, Sam Lee <sam.lee(a)super.net.sg> wrote: > If it is exposed as a environmental variable, you can do a export <env > variable> = something. > This can be used in conjunction with the exec_msg in the ser.cfg > > -----Original Message----- > From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] > On Behalf Of ranveer kunal > Sent: Wednesday, August 10, 2005 6:42 PM > To: serusers(a)lists.iptel.org > Subject: [Serusers] ser : rport > > hi all, > I am using ser, i want ser to add an rport to the via header, > is this possible? > > Thanks > Ranveer. > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > -- Memories : They bring Diamonds and Rust

1 0

[Serusers] Server Capacity
by Waldo Rubinstein 10 Aug '05

10 Aug '05

Hi all, I'm new to SER and was reading the admin guide and came across this: "With a $3000 dual-CPU PC, the SIP Express Router is able to power IP telephony services in an area as large as the Bay Area during peak hours." My question about this, is what is this measuring? For example, is it only measuring SIP registrations? Maybe the same SER server is connected to a few IPTSPs and SER is acting as both a registrar and a proxy server? May be all of the above, while still allowing every UA registered in the server to talk to one another? Maybe it's also providing some more advanced features like voicemail to all those Bay Area users? Can someone please clarify this for me? Thanks, Waldo

2 4

[Serusers] Codec
by Hernando Maisano 10 Aug '05

10 Aug '05

Hello, I have a question: somebody knows how is the codec use the SER? and if it is possible to be formed? Thanks Excuse i'm not speak English.

2 2

[Serusers] Both UA nated in different networks and SER
by Sebastian Muñiz 10 Aug '05

10 Aug '05

Hi. Reading Gettingstarted04.pdf at 1.6.2 from Onsip.org, seems to me that I _always_ need a rtp relay when both UA are nated. Because NAT will not be able to know what ports must be forwarded to the internal UA. Is this true? Is there any scenario where I can have both UA nated and they can just use SIP signaling with ser, and stablish the RTP flow between them? Thanks in advance, Sebastian.

1 0

[Serusers] Auto-Attendant, REFER & Sems
by Adam Sherman 10 Aug '05

10 Aug '05

(Oringinaly sent to the Sems list, thought it was relevant here as well.) So, I'm trying to build an achitecture that acceps calls from a third-party SIP PSTN gateway and runs them through a SEMS auto-attendant, then out to either local users or back to the PSTN gateway. The gateway won't support a REFER. Also, if I issue a REFER back to the PSTN, my SER will now be out of the loop, accounting-wise, right? In addition, I've got some issues with getting ivr.redirect() to work at all. (Other thread.) So, what is the smart road ahead here? If I need a B2BUA, can anyone recommend a good solution? Thanks, A. -- Adam Sherman Technologist

1 0

[Serusers] nathelper/rtpproxy Music On Hold
by harry gaillac 10 Aug '05

10 Aug '05

Hello, I posted many many mails about nathelper/rtpproxy Music On Hold I've been waiting a reply about nathelper/rtpproxy Music On Hold features . May be somebody could tell me more how to provide MOH via nathelper/rtpproxy Music On Hold I don't want to spend time to post the same question ! Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Serusers] alias flag prob
by Aisling 10 Aug '05

10 Aug '05

Hello, Has anyone else experienced the below error when adding an alias using ser version 0.9.0? 1(28100) ERROR:ul_add: flags expected 400 ul_add; flags expected I am going according to the usage info: serctl alias add <alias> <uri> Many thanks, Aisling. -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.

1 0

[Serusers] ivr module to manage voice messages
by harry gaillac 10 Aug '05

10 Aug '05

Hi all, I need help from developpers to write an python script. I 'm not really able to write this script however it could improve sems features. If we store voice messages on a imap server can we write a python script (imaplib) to manage messages from anywhere. Can we set a main menu with recorded messages in order to read play delete voice messages ? phone -------- sems--------- imap server Does imaplib can act as a imap client? what are the main problems between imap protocol and sems? For example I want to listen my voices messages so i must send login password to imap server via sems (DTMF) for authentication. If I press a key to read messages imap server send command to sems . How sems could play voice messages from imap server ? thanks for suggestions Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Serusers] Re: CDRtool
by harry gaillac 10 Aug '05

10 Aug '05

Hello, Even "I do not meet the profile of your customers" You can explain me how to setup prepaid/postpaid applications vith cdrtool and mediaproxy ! How can you tell "I dont meet the profile of your customers"! regards Harry --- Adrian Georgescu <ag(a)ag-projects.com> a écrit : > Sorry Harry, I cannot advise you on anything. You do > not meet the > profile of our customers, you have to search some > place else. > > Adrian > > On Aug 9, 2005, at 12:48 AM, harry gaillac wrote: > > > Hello, > > > > I downloaded CDRtool and mediaproxy. > > Does CDRtool and mediaproxy can provide a > > prepaid/postpaid solution ? > > > > What do you advise me ? > > > > from /INSTALL.txt > > > > For SIP Tracing, SER quota and Prepaid you need > > additional modules from AG > > Projects. > > what's SER call control module ? > > > > Can you tell me more ? > > > > Regards > > Harry > > ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Users] mediaproxy problem
by jayesh nambiar 10 Aug '05

10 Aug '05

hi, I m sorry for my earlier mail as the picture was not very clear. The scenario is something like this: Client A (Public IP) \ ________ __________________ __________ \| My | | Foreign Openser | | PSTN | |Openser |-----------------------| trusts me through |--------------| Gateway | / |_______ | | source IP | |_________| / |__________________| / Client B (Private IP) Whenever My Openser receives a call with prefix for eg. 011, it t_relay it to the foreign Ser which trusts My Openser with source IP. Now when the client A with public IP makes a call it gets established but when the client B with private IP makes a call I receive the ringing message but the RTP sessions does not start. I have configured the mediaproxy in My Openser but still it does'nt seem to work. Pls help me figure out the problem. The RTP Proxy session is not happening and that might be the reason for the problem. Can u pls help me understand why RTP proxy is not getting established for Nated clients. Below is my configuration script: debug=3 fork=yes log_stderror=no listen=203.XXX.64.XX # INSERT YOUR IP ADDRESS HERE port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://openser:openserrw@localhost/openser" loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/mediaproxy.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/permissions.so" loadmodule "/usr/local/lib/openser/modules/acc.so" modparam("auth_db|permissions|uri_db|usrloc|acc", "db_url", "mysql://openser:openserrw@localhost/openser") 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-asymmetric-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-asymmetric-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", "log_level", 1) modparam("acc", "log_flag", 1) modparam("acc", "db_flag", 1) 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")) { # INSERT YOUR IP ADDRESS HERE record_route_preset("203.XXX.64.XX:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { 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 # ----------------------------------------------------------------- t_on_reply("1"); 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("my_domain","subscriber")) { www_challenge("my_domain","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 # ----------------------------------------------------------------- setflag(1); if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; if (method=="INVITE" && !allow_trusted()) { if (!proxy_authorize("my_domain","subscriber")) { proxy_challenge("my_domain","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:011[0-9]*@") { # International PSTN route(4); break; }; if (!lookup("location")) { if (uri=~"^sip:[0-9]{10}@") { # Domestic PSTN route(4); break; }; sl_send_reply("404", "User Not Found"); break; }; if (method=="CANCEL") { route(1); break; }; route(5); route(1); } route[4] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- rewritehost("66.XXX.11.XX"); avp_write("i:45", "inv_timeout"); route(5); 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(5); 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(); }; } --------------------------------- Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. Learn more.

3 2

[Serusers] Dispatcher and a dynamic list file
by Alistair Cunningham 10 Aug '05

10 Aug '05

I'm looking for ideas on how to use the dispatcher module with a list file that is frequently updated by another process (specifically, it checks the availability of the remote machines, and also interacts with the heartbeat program). In the past, I've always used an exec_dst() call to run an external program which does this, but would now like to migrate to using dispatcher for performance. If that's not possible, I'll go back to the old method. I've considered adding a call to ds_select_dst() to call ds_destroy_list(); ds_load_list() on each request with file locking. It looks like it should work. Has anyone implemented this? Does it work? Does it have a significant impact on performance? I'd also be open to having the other process write to a database table, which then SER reads. I'm not sure what's the best mechanism within SER to look up the table - perhaps avpops? I'd even be willing write a new module to do it - 'dispatcher_db' perhaps, but this wouldn't be in the short future. -- Alistair Cunningham, Integrics Ltd, +44 (0)7870 699 479 http://integrics.com/

2 2

[Users] MediaProxy Related (argh my email client)
by Ozan Blotter 10 Aug '05

10 Aug '05

Dear List, This is a big explanation with enough information except ngrep/ethereal captures, in both cases : ADSL Connection Speed : 512/128, ATA's : Zyxel P2002,Codec : G729,RTP Port Range : 19000-20000, and OpenSER on Public IP without NAT etc. I think i should change the ser.cfg to use MediaProxy just for SIP2SIP calls not for PSTN calls and i do not know how to do it, any clues ? Thanks,Ozan * A Side client use Zoom 4X, port forwarding to ATA in modem setting i register as port:5060 (openserctl ul show),* B Side client use Zyxel 660R, port forwarding to ATA in modem setting and register as port:10XXX (openserctl ul show),* Make a call to other sip client it's okay voice quality is acceptable most of time, but caller or callee losts ADSL connection after 30-60 seconds,* From A Side (Zoom 4X), make a call to PSTN, ADSL crashes, everything locks up,* From B Side (Zyxel 660R), make a call to PSTN, everything is normal, connection never down. ps: If i change A Side client modem to Zyxel 660R too as B Side client, the problems are totaly same. here is my basic ser.cfg >>>debug=3fork=yeslog_stderror=nocheck_via=nolisten=212.XXX.104.XXX # OpenSER Server on Public IP port=5060children=4dns=norev_dns=nofifo="/tmp/openser_fifo"loadmodule "/usr/local/lib/openser/modules/sl.so"loadmodule "/usr/local/lib/openser/modules/tm.so"loadmodule "/usr/local/lib/openser/modules/rr.so"loadmodule "/usr/local/lib/openser/modules/maxfwd.so"loadmodule "/usr/local/lib/openser/modules/usrloc.so"loadmodule "/usr/local/lib/openser/modules/registrar.so"modparam("usrloc", "db_mode", 0)modparam("rr", "enable_full_lr", 1)route { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; if (method!="REGISTER") { record_route(); }; if (loose_route()) { route(1); break; }; if (uri!=myself) { route(1); break; }; if (uri==myself) { if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (uri=~"^sip:0[0-9]*@*") { rewritehost("195.XXX.122.XXX"); # PSTN Gateway on Public IP route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); };}route[1] { if (!t_relay()) { sl_reply_error(); };}route[2] { if (!save("location")) { sl_reply_error(); };}In my advanced SER.CFG ;* A Side client use Zoom 4X, port forwarding to ATA in modem setting i register as port:1XXXX (openserctl ul show),* B Side client use Zyxel 660R, port forwarding to ATA in modem setting and register as port:1XXXX (openserctl ul show),* Make a call to other sip client it's okay voice quality is worse then first case most of time, connection never down,* From A Side (Zoom 4X), make a call to PSTN, everything is normal, connection never down,* From B Side (Zyxel 660R), make a call to PSTN, everything is normal, connection never down. ps : If i change both A Side client and B Side client modems not to do port forwarding, problems are totaly same. here is my advanced ser.cfg >>> debug=3 fork=yes log_stderror=no listen=212.XXX.104.XXX # OpenSER Server on Public IP port=5060 children=4 alias=212.XXX.104.XXX # OpenSER Server on Public IP dns=no rev_dns=no fifo="/tmp/openser_fifo" fifo_db_url="mysql://openser:openserrw@localhost/openser" loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/mediaproxy.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/acc.so" modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("auth_db", "use_domain", 1) modparam("domain", "db_mode", 1) modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 180) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/openser/sip-asymmetric-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/openser/rtp-asymmetric-clients") modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", 1) modparam("registrar", "default_expires", 60) modparam("registrar", "min_expires", 30) modparam("registrar", "nat_flag", 6) modparam("registrar", "use_domain", 1) modparam("rr", "enable_full_lr", 1) modparam("acc", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("acc", "failed_transactions", 1) modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) modparam("acc", "db_flag", 1) 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 and Acc section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { record_route_preset("212.XXX.104.XXX:5060;nat=yes"); } else if (method!="REGISTER") { if!(uri=~"^sip:833[0-9]*@") { record_route(); setflag(1); } }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { 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(1); break; }; if (uri==myself) { 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(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- if (!search("^Contact:\ +\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; sl_send_reply("100", "Trying"); 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(); }; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (uri=~"^sip:0[0-9]*@*") { rewritehost("195.XXX.122.XXX"); # PSTN Gateway on Public IP route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; if (method=="CANCEL") { route(1); break; }; if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (isflagset(6) || isflagset(7)) { use_media_proxy(); }; 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(); }; }

1 0

[Serusers] OT: test
by Simone Cittadini 10 Aug '05

10 Aug '05

it seems my mails aren't arriving ....

1 0

RE: [Serusers] ser : rport
by Sam Lee 10 Aug '05

10 Aug '05

If it is exposed as a environmental variable, you can do a export <env variable> = something. This can be used in conjunction with the exec_msg in the ser.cfg -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of ranveer kunal Sent: Wednesday, August 10, 2005 6:42 PM To: serusers(a)lists.iptel.org Subject: [Serusers] ser : rport hi all, I am using ser, i want ser to add an rport to the via header, is this possible? Thanks Ranveer. _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

2 1

[Users] MediaProxy Related
by Ozan Blotter 10 Aug '05

10 Aug '05

Dear List,This is a big explanation with enough information except ngrep/ethereal captures, in both cases :ADSL Connection Speed : 512/128,ATA's : Zyxel P2002,Codec : G729,RTP Port Range : 19000-20000,and OpenSER on Public IP without NAT etc.I think i should change the ser.cfg to use MediaProxy just for SIP2SIP calls not for PSTN calls and i do not know how to do it, any clues ?Thanks,Ozan* A Side client use Zoom 4X, port forwarding to ATA in modem setting i register as port:5060 (openserctl ul show),* B Side client use Zyxel 660R, port forwarding to ATA in modem setting and register as port:10XXX (openserctl ul show),* Make a call to other sip client it's okay voice quality is acceptable most of time, but caller or callee losts ADSL connection after 30-60 seconds,* From A Side (Zoom 4X), make a call to PSTN, ADSL crashes, everything locks up,* From B Side (Zyxel 660R), make a call to PSTN, everything is normal, connection never down.ps: If i change A Side client modem to Zyxel 660R too as B Side client, the problems are totaly same.here is my basic ser.cfg >>>debug=3fork=yeslog_stderror=nocheck_via=nolisten=212.XXX.104.XXX # OpenSER Server on Public IP port=5060children=4dns=norev_dns=nofifo="/tmp/openser_fifo"loadmodule "/usr/local/lib/openser/modules/sl.so"loadmodule "/usr/local/lib/openser/modules/tm.so"loadmodule "/usr/local/lib/openser/modules/rr.so"loadmodule "/usr/local/lib/openser/modules/maxfwd.so"loadmodule "/usr/local/lib/openser/modules/usrloc.so"loadmodule "/usr/local/lib/openser/modules/registrar.so"modparam("usrloc", "db_mode", 0)modparam("rr", "enable_full_lr", 1)route { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; if (method!="REGISTER") { record_route(); }; if (loose_route()) { route(1); break; }; if (uri!=myself) { route(1); break; }; if (uri==myself) { if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (uri=~"^sip:0[0-9]*@*") { rewritehost("195.XXX.122.XXX"); # PSTN Gateway on Public IP route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); };}route[1] { if (!t_relay()) { sl_reply_error(); };}route[2] { if (!save("location")) { sl_reply_error(); };}In my advanced SER.CFG ;* A Side client use Zoom 4X, port forwarding to ATA in modem setting i register as port:1XXXX (openserctl ul show),* B Side client use Zyxel 660R, port forwarding to ATA in modem setting and register as port:1XXXX (openserctl ul show),* Make a call to other sip client it's okay voice quality is worse then first case most of time, connection never down,* From A Side (Zoom 4X), make a call to PSTN, everything is normal, connection never down,* From B Side (Zyxel 660R), make a call to PSTN, everything is normal, connection never down.ps : If i change both A Side client and B Side client modems not to do port forwarding, problems are totaly same.here is my advanced ser.cfg >>>debug=3 fork=yes log_stderror=no listen=212.XXX.104.XXX # OpenSER Server on Public IP port=5060 children=4 alias=212.XXX.104.XXX # OpenSER Server on Public IP dns=no rev_dns=no fifo="/tmp/openser_fifo" fifo_db_url="mysql://openser:openserrw@localhost/openser" loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/mediaproxy.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/acc.so" modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("auth_db", "use_domain", 1) modparam("domain", "db_mode", 1) modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 180) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/openser/sip-asymmetric-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/openser/rtp-asymmetric-clients") modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", 1) modparam("registrar", "default_expires", 60) modparam("registrar", "min_expires", 30) modparam("registrar", "nat_flag", 6) modparam("registrar", "use_domain", 1) modparam("rr", "enable_full_lr", 1) modparam("acc", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("acc", "failed_transactions", 1) modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) modparam("acc", "db_flag", 1) 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 and Acc section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { record_route_preset("212.XXX.104.XXX:5060;nat=yes"); } else if (method!="REGISTER") { if!(uri=~"^sip:833[0-9]*@") { record_route(); setflag(1); } }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { 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(1); break; }; if (uri==myself) { 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(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- if (!search("^Contact:\ +\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; sl_send_reply("100", "Trying"); 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(); }; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (uri=~"^sip:0[0-9]*@*") { rewritehost("195.XXX.122.XXX"); # PSTN Gateway on Public IP route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; if (method=="CANCEL") { route(1); break; }; if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (isflagset(6) || isflagset(7)) { use_media_proxy(); }; 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(); }; }

1 0

[Serusers] radius accounting, duplicate rows and missing columns
by Simone Cittadini 10 Aug '05

10 Aug '05

I've this ser installation whose purpose in life is accounting all the calls and redirecting them to asterisk, which routes them to ISDN or a h323 GSM gateway. SIP phones -> Ser -> Asterisk -> [isdn | gsm ] all the calls are "from left to right", asterisk will never see incoming stuff from the isdn/gsm channels radius has mysql as backend, the calls works well but : if the sip phone hangsup radius writes two identical lines in the radacct table, missing the AcctSessionTime value (writes '0') if the remote phone hangsup there are the same two lines and a third line with caller and called inverted, and start and stop time with the same value Can someone tell me what's wrong with my route ? or I'm missing something in radius configuration ? some columns and lines from the radacct table "RadAcctId","AcctSessionId","AcctUniqueId","UserName","AcctStartTime","AcctStopTime" #these lines inserted if the caller hangsup first 49,"e7e71b1b-70eec1ee(a)172.18.1.13","d9428a25afbfdf52","[sip number]","2005-08-10 08:54:53","2005-08-10 08:54:58" 50,"e7e71b1b-70eec1ee(a)172.18.1.13","d9428a25afbfdf52","[sip number]","2005-08-10 08:54:53","2005-08-10 08:54:58" #these if the called number hangsup 51,"72660925-e480d1a5(a)172.18.1.13","43e7c757d53c2970","[sip number]","2005-08-10 08:57:10","2005-08-10 08:57:22" 52,"72660925-e480d1a5(a)172.18.1.13","43e7c757d53c2970","[sip number]","2005-08-10 08:57:10","2005-08-10 08:57:22" 53,"72660925-e480d1a5(a)172.18.1.13","5b732c77e7e3f37d","[external number]","2005-08-10 08:57:13","2005-08-10 08:57:13" acc_radius paramethers : modparam("acc", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") modparam("acc", "log_level", 2) modparam("acc", "log_fmt", "cdfimorstup") modparam("acc", "radius_flag", 1) routing configuration : # initial sanity checks 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; }; # registration (and authentication) if (uri==myself) { if (method=="REGISTER") { if (!radius_www_authorize("")) { www_challenge("", "0"); break; }; save("location"); break; }; lookup("aliases"); }; setflag(1); if (method=="INVITE") record_route(); rewritehost("172.18.1.11"); if (!t_relay()) { sl_reply_error(); };

1 0

[Serusers] forking with SER
by davor jovanovic 10 Aug '05

10 Aug '05

Hi all, I have a question about routing a call to GW when user isn't in location table. If I have a user with SIP address: sip:mister@no.com, and his alias in form of his business phone on office PBX: alias = 1234 can SER use number in aliases and rewrite sip:mister@no.com to <mailto:1234@no.com> 1234(a)no.com? sip:mister@no.com -> "404 Not Found" -> use aliases -> sip:1234@no.com -> t_relay -> GW -> PBX or something like that. So if someone could explain me how can I do this. Do I need some external application to do that and return PSTN number to SER? Or how can I do forking with SER? thanks

3 5

[Serusers] looping through SER-Asterisk
by Victor Alvarez 10 Aug '05

10 Aug '05

Hi, Trying to do the following: pstn->SER->asterisk (call forward)->SER->Pstn The far end rings but when you answer the call, I get an error in Asterisk: "Got SIP response 481 "Invalid CSeq Number" back from XXX.XX.XX.XX (SER). I have tried the same using two SER machines with the same result, but if I change the scenery to pstn->SER->asterisk (call forward) -> asterisk-> Pstn, it works fine! What could be wrong here? Does anybody implemented the scenery explained above? Could anybody give me a clue? Thanks a lot. Regards, Victor.

2 2

[Serusers] CallId: 0094f942e99f-zmnf5ytp1ngc@snomSoft
by fedora＠voice.dyndns.tv 10 Aug '05

10 Aug '05

Date and time Username Source Destination Method Status 1 2005-08-10 05:45:42 CallId: 0094f942e99f-zmnf5ytp1ngc@snomSoft 1000 <sip:1000@voice.dyndns.tv:5062> sip:8000@221.186.150.68:3852 "8000" <sip:8000@voice.dyndns.tv:5062> sip:8000@221.186.150.68:3852 BYE Ok (200) 2 2005-08-10 05:43:52 CallId: 0094f942e99f-zmnf5ytp1ngc@snomSoft 8000 "8000" <sip:8000@voice.dyndns.tv:5062> sip:1000@voice.dyndns.tv:5062 <sip:1000@voice.dyndns.tv:5062> sip:1000@voice.dyndns.tv:5062 INVITE Ok (200) 3 2005-08-10 05:43:52 CallId: 0094f942e99f-zmnf5ytp1ngc@snomSoft 8000 "8000" <sip:8000@voice.dyndns.tv:5062> sip:1000@221.186.150.67:5062 <sip:1000@voice.dyndns.tv:5062> sip:1000@221.186.150.67:5062 ACK 4 2005-08-10 05:43:52 CallId: 0094f942e99f-xzxrtdd5xb03@snomSoft 8000 <sip:8000@voice.dyndns.tv:5062> sip:1000@voice.dyndns.tv:5062 <sip:1000@voice.dyndns.tv:5062> sip:1000@voice.dyndns.tv:5062 SUBSCRIBE Ok (200) 5 2005-08-10 05:43:51 CallId: 0094f942e99f-xzxrtdd5xb03@snomSoft 8000 <sip:8000@voice.dyndns.tv:5062> sip:1000@voice.dyndns.tv:5062 <sip:1000@voice.dyndns.tv:5062> sip:1000@voice.dyndns.tv:5062 SUBSCRIBE Ok (200) 6 2005-08-10 05:42:28 CallId: 3493f9426ce8-n0nujrl64fac@snomSoft 1234 <sip:1234@voice.dyndns.tv:5062> sip:8000@221.186.150.68:3852 "8000" <sip:8000@voice.dyndns.tv:5062> sip:8000@221.186.150.68:3852 BYE Ok (200)

2 1

[Users] mediaproxy problem
by jayesh nambiar 10 Aug '05

10 Aug '05

hi, I did admin the other SER and that works. Thanx. Now if the request is for foreign calls I send the request from my OPENSER to a foreign SER . That SER does trust me through the source IP. We buy pstn minutes from them. Now the calls are possible when the users are with public IP and also behind NAT. But when the call is for the Cisco phone that is to be handled by the Foriegn SER it does not get established neither the rtp sesions gets established between my SER and the end called party. Can u pls suggest a particular way to handle such calls. Can u pls suggest wat might b the problem. Thanx a lot My configuration is: debug=3 fork=yes log_stderror=no listen=203.XXX.64.XX # INSERT YOUR IP ADDRESS HERE port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://openser:openserrw@localhost/openser" loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/mediaproxy.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/permissions.so" loadmodule "/usr/local/lib/openser/modules/acc.so" modparam("auth_db|permissions|uri_db|usrloc|acc", "db_url", "mysql://openser:openserrw@localhost/openser") 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-asymmetric-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-asymmetric-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", "log_level", 1) modparam("acc", "log_flag", 1) modparam("acc", "db_flag", 1) 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")) { # INSERT YOUR IP ADDRESS HERE record_route_preset("203.XXX.64.XX:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { 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 # ----------------------------------------------------------------- t_on_reply("1"); 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("my_domain","subscriber")) { www_challenge("my_domain","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 # ----------------------------------------------------------------- setflag(1); if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; if (method=="INVITE" && !allow_trusted()) { if (!proxy_authorize("my_domain","subscriber")) { proxy_challenge("my_domain","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:011[0-9]*@") { # International PSTN route(4); break; }; if (!lookup("location")) { if (uri=~"^sip:[0-9]{10}@") { # Domestic PSTN route(4); break; }; sl_send_reply("404", "User Not Found"); break; }; if (method=="CANCEL") { route(1); break; }; route(5); route(1); } route[4] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- #rewritehost("66.XXX.11.XX"); # INSERT YOUR PSTN GATEWAY IP ADDRESS avp_write("i:45", "inv_timeout"); route(5); 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(5); 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(); }; } --------------------------------- Start your day with Yahoo! - make it your home page

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[Serusers] register pass through in ser
by jeff kwong 10 Aug '05

10 Aug '05

Hi! Is there a way for ser to recieve register message and then pass it to another provider but the ser machine will do the rtp proxying?That is all media packets will be relayed to ser to the user agents. UA---NAT---SER----ANOTHER SS Thanks! _Jeff

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[Serusers] question about multiple SER server register.
by ZHANG, YI2 -HCHBJ 10 Aug '05

10 Aug '05

Can SER server cascaded connected? That's to say. There a centeralized SER registrar server. below top layer, there are serval subordinate SER registrar server which must be registered on that centeralized registrar server. In this case, how to configure those subordinate SER registrar servers to register onto that server?how to send "REGISTER" message onto center server?

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[Serusers] DB error in SER
by Salony Agrawal 09 Aug '05

09 Aug '05

All, I am trying to connect to the database and write the enteries of a field in a file but when trying to bind to the database I am getting the following error. /tmp/ccgppbh8.o(.text+0x28): In function `main': : undefined reference to `bind_dbmod' The code is written within the PA module #include <stdio.h> #include <stdlib.h> #include "../../db/db.h" #define DB_URL "mysql://user:pass@127.0.0.1:3306/dbname" int main() { FILE *fp; db_con_t* h; db_res_t* res; db_func_t* dbf; if (bind_dbmod(DB_URL, dbf)) { printf("Error while binding database module, did you forget to load a database module ?\n"); return -1; } Please let me know. Any help will be highly appreciated. Thanks, Salony "...Live a good, honorable life. Then, when you get older and think back, you'll be able to enjoy it a second time."

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[Serusers] URGENT:No Audio with mediaproxy
by Ashutosh Kumar 09 Aug '05

09 Aug '05

Hi, I m using mediaproxy for nat raversal and rtp handling. Nat is working fine. Clients behind the nat are connected after ringing, but no audo transmission takes place in either side. On running the sessions.py of mediaproxy, the output is as following, [root@sip ser]# python /usr/local/mediaproxy/sessions.py Caller Via Called Status Duration Codec Type Traffic ---------------------------------------------------------------------------- --------------------------- 61.16.236.34:24742 - 66.114.236.182:35012 - ?.?.?.?:? inactive 0'12" G711u Audio 19.42k/704/480 Total traffic: 0bps/0bps/0bps (in1/in2/out) Session count: 1 Proxy version: 1.4.2 --------------- Why is the Called filed showing a ?.?.?.? even though the client is fully resolved. Please help urgent!!! Regards, Ashutosh Kumar Chetu, Inc. Ph : 1(305) 402 6724 - Witin US Ph : 91 120 5323340 - Outside US Fax:1 (305) 832 5987 For more information, please visit http://www.chetu.com

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Re: [Serusers] serctl start|stop|restart
by seehoe yee 09 Aug '05

09 Aug '05

----- Original Message ----- From: "Marc Khayat" <marc(a)globalcarrier.net> Date: Tuesday, August 9, 2005 9:00 pm Subject: [Serusers] serctl start|stop|restart > Hi all, > The first few times I do START STOP or RESTART, everything works fine. > But after that, it gives the following: > [root@linux ~]# serctl stop > Stopping SER : /usr/local/sbin/serctl: line 614: kill: (8771) - No > suchprocess > stopped > [root@linux ~]# > [root@linux ~]# serctl start > > Starting SER : PID file exists! (/var/run/ser.pid) already running? > [root@linux ~]# > [root@linux ~]# serctl restart > Stopping SER : /usr/local/sbin/serctl: line 627: kill: (8771) - No > suchprocess > stopped > Starting SER : PID file exists! (/var/run/ser.pid) already running? > > How can I solve it? > I've encounter this issue before, my previous encounter was caused by misconfiguration in 'ser.cfg'. I can't remember what was my configuration. If you have made any changes recently to ser.cfg. It could be the cause. To stop ser, kill /var/run/ser.pid then start again. seehoe

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[Serusers] MEdiaproxy unable to resolve private IP to public IP
by Ashutosh Kumar 09 Aug '05

09 Aug '05

Hi, I m using ser-0.9.0 with mediaproxy, non natted clients can talk fine, but natted clients addresses are not resolved to public addresses.Also, the culprit I suppose is a function fix_nated_register() used in the ser.cfg which was not recognized by ser. The relevant code from ser.cfg is as follows: ... ... # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact:\ +\*") && client_nat_test("7")) { setflag(6); # I commented the following line since it was not recognized by ser # fix_nated_register(); force_rport(); }; ... ... Regards, Ashutosh Kumar Chetu, Inc. Ph : 1(305) 402 6724 - Witin US Ph : 91 120 5323340 - Outside US Fax:1 (305) 832 5987 For more information, please visit http://www.chetu.com

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